diff options
Diffstat (limited to 'sound')
253 files changed, 13343 insertions, 2418 deletions
diff --git a/sound/aoa/soundbus/core.c b/sound/aoa/soundbus/core.c index 99ca7120e269..7487eb76e034 100644 --- a/sound/aoa/soundbus/core.c +++ b/sound/aoa/soundbus/core.c @@ -59,7 +59,7 @@ static int soundbus_probe(struct device *dev) static int soundbus_uevent(struct device *dev, struct kobj_uevent_env *env) { struct soundbus_dev * soundbus_dev; - struct of_device * of; + struct platform_device * of; const char *compat; int retval = 0; int cplen, seen = 0; diff --git a/sound/aoa/soundbus/soundbus.h b/sound/aoa/soundbus/soundbus.h index a0f223c13f66..adecbf36f4f6 100644 --- a/sound/aoa/soundbus/soundbus.h +++ b/sound/aoa/soundbus/soundbus.h @@ -141,7 +141,7 @@ struct soundbus_dev { struct list_head onbuslist; /* the of device it represents */ - struct of_device ofdev; + struct platform_device ofdev; /* what modules go by */ char modalias[32]; diff --git a/sound/aoa/soundbus/sysfs.c b/sound/aoa/soundbus/sysfs.c index 6496e754f00a..e0980b5c2cd8 100644 --- a/sound/aoa/soundbus/sysfs.c +++ b/sound/aoa/soundbus/sysfs.c @@ -16,7 +16,7 @@ static ssize_t modalias_show(struct device *dev, struct device_attribute *attr, char *buf) { struct soundbus_dev *sdev = to_soundbus_device(dev); - struct of_device *of = &sdev->ofdev; + struct platform_device *of = &sdev->ofdev; int length; if (*sdev->modalias) { diff --git a/sound/core/control.c b/sound/core/control.c index 070aab490191..45a818002d99 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -31,6 +31,7 @@ /* max number of user-defined controls */ #define MAX_USER_CONTROLS 32 +#define MAX_CONTROL_COUNT 1028 struct snd_kctl_ioctl { struct list_head list; /* list of all ioctls */ @@ -195,6 +196,10 @@ static struct snd_kcontrol *snd_ctl_new(struct snd_kcontrol *control, if (snd_BUG_ON(!control || !control->count)) return NULL; + + if (control->count > MAX_CONTROL_COUNT) + return NULL; + kctl = kzalloc(sizeof(*kctl) + sizeof(struct snd_kcontrol_volatile) * control->count, GFP_KERNEL); if (kctl == NULL) { snd_printk(KERN_ERR "Cannot allocate control instance\n"); diff --git a/sound/core/init.c b/sound/core/init.c index ec4a50ce5656..2de45fbd70fb 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -607,11 +607,16 @@ card_id_store_attr(struct device *dev, struct device_attribute *attr, return -EEXIST; } for (idx = 0; idx < snd_ecards_limit; idx++) { - if (snd_cards[idx] && !strcmp(snd_cards[idx]->id, buf1)) - goto __exist; + if (snd_cards[idx] && !strcmp(snd_cards[idx]->id, buf1)) { + if (card == snd_cards[idx]) + goto __ok; + else + goto __exist; + } } strcpy(card->id, buf1); snd_info_card_id_change(card); +__ok: mutex_unlock(&snd_card_mutex); return count; diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index 8442a088677d..822dd56993ca 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -77,7 +77,7 @@ static int snd_mixer_oss_release(struct inode *inode, struct file *file) struct snd_mixer_oss_file *fmixer; if (file->private_data) { - fmixer = (struct snd_mixer_oss_file *) file->private_data; + fmixer = file->private_data; module_put(fmixer->card->module); snd_card_file_remove(fmixer->card, file); kfree(fmixer); @@ -368,7 +368,7 @@ static int snd_mixer_oss_ioctl1(struct snd_mixer_oss_file *fmixer, unsigned int static long snd_mixer_oss_ioctl(struct file *file, unsigned int cmd, unsigned long arg) { - return snd_mixer_oss_ioctl1((struct snd_mixer_oss_file *) file->private_data, cmd, arg); + return snd_mixer_oss_ioctl1(file->private_data, cmd, arg); } int snd_mixer_oss_ioctl_card(struct snd_card *card, unsigned int cmd, unsigned long arg) @@ -582,7 +582,7 @@ static int snd_mixer_oss_get_volume1(struct snd_mixer_oss_file *fmixer, struct snd_mixer_oss_slot *pslot, int *left, int *right) { - struct slot *slot = (struct slot *)pslot->private_data; + struct slot *slot = pslot->private_data; *left = *right = 100; if (slot->present & SNDRV_MIXER_OSS_PRESENT_PVOLUME) { @@ -693,7 +693,7 @@ static int snd_mixer_oss_put_volume1(struct snd_mixer_oss_file *fmixer, struct snd_mixer_oss_slot *pslot, int left, int right) { - struct slot *slot = (struct slot *)pslot->private_data; + struct slot *slot = pslot->private_data; if (slot->present & SNDRV_MIXER_OSS_PRESENT_PVOLUME) { snd_mixer_oss_put_volume1_vol(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_PVOLUME], left, right); @@ -742,7 +742,7 @@ static int snd_mixer_oss_get_recsrc1_sw(struct snd_mixer_oss_file *fmixer, struct snd_mixer_oss_slot *pslot, int *active) { - struct slot *slot = (struct slot *)pslot->private_data; + struct slot *slot = pslot->private_data; int left, right; left = right = 1; @@ -755,7 +755,7 @@ static int snd_mixer_oss_get_recsrc1_route(struct snd_mixer_oss_file *fmixer, struct snd_mixer_oss_slot *pslot, int *active) { - struct slot *slot = (struct slot *)pslot->private_data; + struct slot *slot = pslot->private_data; int left, right; left = right = 1; @@ -768,7 +768,7 @@ static int snd_mixer_oss_put_recsrc1_sw(struct snd_mixer_oss_file *fmixer, struct snd_mixer_oss_slot *pslot, int active) { - struct slot *slot = (struct slot *)pslot->private_data; + struct slot *slot = pslot->private_data; snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_CSWITCH], active, active, 0); return 0; @@ -778,7 +778,7 @@ static int snd_mixer_oss_put_recsrc1_route(struct snd_mixer_oss_file *fmixer, struct snd_mixer_oss_slot *pslot, int active) { - struct slot *slot = (struct slot *)pslot->private_data; + struct slot *slot = pslot->private_data; snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_CROUTE], active, active, 1); return 0; @@ -815,7 +815,7 @@ static int snd_mixer_oss_get_recsrc2(struct snd_mixer_oss_file *fmixer, unsigned if (!(mixer->mask_recsrc & (1 << idx))) continue; pslot = &mixer->slots[idx]; - slot = (struct slot *)pslot->private_data; + slot = pslot->private_data; if (slot->signature != SNDRV_MIXER_OSS_SIGNATURE) continue; if (!(slot->present & SNDRV_MIXER_OSS_PRESENT_CAPTURE)) @@ -864,7 +864,7 @@ static int snd_mixer_oss_put_recsrc2(struct snd_mixer_oss_file *fmixer, unsigned if (!(mixer->mask_recsrc & (1 << idx))) continue; pslot = &mixer->slots[idx]; - slot = (struct slot *)pslot->private_data; + slot = pslot->private_data; if (slot->signature != SNDRV_MIXER_OSS_SIGNATURE) continue; if (!(slot->present & SNDRV_MIXER_OSS_PRESENT_CAPTURE)) @@ -929,7 +929,7 @@ static int snd_mixer_oss_build_test(struct snd_mixer_oss *mixer, struct slot *sl static void snd_mixer_oss_slot_free(struct snd_mixer_oss_slot *chn) { - struct slot *p = (struct slot *)chn->private_data; + struct slot *p = chn->private_data; if (p) { if (p->allocated && p->assigned) { kfree(p->assigned->name); diff --git a/sound/core/pcm.c b/sound/core/pcm.c index cbe815dfbdc8..6b4b1287b314 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -203,10 +203,16 @@ static char *snd_pcm_format_names[] = { FORMAT(S18_3BE), FORMAT(U18_3LE), FORMAT(U18_3BE), + FORMAT(G723_24), + FORMAT(G723_24_1B), + FORMAT(G723_40), + FORMAT(G723_40_1B), }; const char *snd_pcm_format_name(snd_pcm_format_t format) { + if (format >= ARRAY_SIZE(snd_pcm_format_names)) + return "Unknown"; return snd_pcm_format_names[format]; } EXPORT_SYMBOL_GPL(snd_pcm_format_name); @@ -358,22 +364,24 @@ static void snd_pcm_stream_proc_info_read(struct snd_info_entry *entry, static void snd_pcm_substream_proc_info_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { - snd_pcm_proc_info_read((struct snd_pcm_substream *)entry->private_data, - buffer); + snd_pcm_proc_info_read(entry->private_data, buffer); } static void snd_pcm_substream_proc_hw_params_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_pcm_substream *substream = entry->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_pcm_runtime *runtime; + + mutex_lock(&substream->pcm->open_mutex); + runtime = substream->runtime; if (!runtime) { snd_iprintf(buffer, "closed\n"); - return; + goto unlock; } if (runtime->status->state == SNDRV_PCM_STATE_OPEN) { snd_iprintf(buffer, "no setup\n"); - return; + goto unlock; } snd_iprintf(buffer, "access: %s\n", snd_pcm_access_name(runtime->access)); snd_iprintf(buffer, "format: %s\n", snd_pcm_format_name(runtime->format)); @@ -392,20 +400,25 @@ static void snd_pcm_substream_proc_hw_params_read(struct snd_info_entry *entry, snd_iprintf(buffer, "OSS period frames: %lu\n", (unsigned long)runtime->oss.period_frames); } #endif + unlock: + mutex_unlock(&substream->pcm->open_mutex); } static void snd_pcm_substream_proc_sw_params_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_pcm_substream *substream = entry->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_pcm_runtime *runtime; + + mutex_lock(&substream->pcm->open_mutex); + runtime = substream->runtime; if (!runtime) { snd_iprintf(buffer, "closed\n"); - return; + goto unlock; } if (runtime->status->state == SNDRV_PCM_STATE_OPEN) { snd_iprintf(buffer, "no setup\n"); - return; + goto unlock; } snd_iprintf(buffer, "tstamp_mode: %s\n", snd_pcm_tstamp_mode_name(runtime->tstamp_mode)); snd_iprintf(buffer, "period_step: %u\n", runtime->period_step); @@ -415,24 +428,29 @@ static void snd_pcm_substream_proc_sw_params_read(struct snd_info_entry *entry, snd_iprintf(buffer, "silence_threshold: %lu\n", runtime->silence_threshold); snd_iprintf(buffer, "silence_size: %lu\n", runtime->silence_size); snd_iprintf(buffer, "boundary: %lu\n", runtime->boundary); + unlock: + mutex_unlock(&substream->pcm->open_mutex); } static void snd_pcm_substream_proc_status_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_pcm_substream *substream = entry->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_pcm_runtime *runtime; struct snd_pcm_status status; int err; + + mutex_lock(&substream->pcm->open_mutex); + runtime = substream->runtime; if (!runtime) { snd_iprintf(buffer, "closed\n"); - return; + goto unlock; } memset(&status, 0, sizeof(status)); err = snd_pcm_status(substream, &status); if (err < 0) { snd_iprintf(buffer, "error %d\n", err); - return; + goto unlock; } snd_iprintf(buffer, "state: %s\n", snd_pcm_state_name(status.state)); snd_iprintf(buffer, "owner_pid : %d\n", pid_vnr(substream->pid)); @@ -446,6 +464,8 @@ static void snd_pcm_substream_proc_status_read(struct snd_info_entry *entry, snd_iprintf(buffer, "-----\n"); snd_iprintf(buffer, "hw_ptr : %ld\n", runtime->status->hw_ptr); snd_iprintf(buffer, "appl_ptr : %ld\n", runtime->control->appl_ptr); + unlock: + mutex_unlock(&substream->pcm->open_mutex); } #ifdef CONFIG_SND_PCM_XRUN_DEBUG diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index d6ecca27bb68..a1707cca9c66 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -67,6 +67,8 @@ void snd_pcm_playback_silence(struct snd_pcm_substream *substream, snd_pcm_ufram } else { if (new_hw_ptr == ULONG_MAX) { /* initialization */ snd_pcm_sframes_t avail = snd_pcm_playback_hw_avail(runtime); + if (avail > runtime->buffer_size) + avail = runtime->buffer_size; runtime->silence_filled = avail > 0 ? avail : 0; runtime->silence_start = (runtime->status->hw_ptr + runtime->silence_filled) % @@ -287,8 +289,11 @@ int snd_pcm_update_state(struct snd_pcm_substream *substream, return -EPIPE; } } - if (avail >= runtime->control->avail_min) - wake_up(runtime->twake ? &runtime->tsleep : &runtime->sleep); + if (runtime->twake) { + if (avail >= runtime->twake) + wake_up(&runtime->tsleep); + } else if (avail >= runtime->control->avail_min) + wake_up(&runtime->sleep); return 0; } @@ -1711,7 +1716,7 @@ EXPORT_SYMBOL(snd_pcm_period_elapsed); * The available space is stored on availp. When err = 0 and avail = 0 * on the capture stream, it indicates the stream is in DRAINING state. */ -static int wait_for_avail_min(struct snd_pcm_substream *substream, +static int wait_for_avail(struct snd_pcm_substream *substream, snd_pcm_uframes_t *availp) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -1761,7 +1766,7 @@ static int wait_for_avail_min(struct snd_pcm_substream *substream, avail = snd_pcm_playback_avail(runtime); else avail = snd_pcm_capture_avail(runtime); - if (avail >= runtime->control->avail_min) + if (avail >= runtime->twake) break; } _endloop: @@ -1824,7 +1829,7 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, goto _end_unlock; } - runtime->twake = 1; + runtime->twake = runtime->control->avail_min ? : 1; while (size > 0) { snd_pcm_uframes_t frames, appl_ptr, appl_ofs; snd_pcm_uframes_t avail; @@ -1837,7 +1842,9 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, err = -EAGAIN; goto _end_unlock; } - err = wait_for_avail_min(substream, &avail); + runtime->twake = min_t(snd_pcm_uframes_t, size, + runtime->control->avail_min ? : 1); + err = wait_for_avail(substream, &avail); if (err < 0) goto _end_unlock; } @@ -2046,7 +2053,7 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, goto _end_unlock; } - runtime->twake = 1; + runtime->twake = runtime->control->avail_min ? : 1; while (size > 0) { snd_pcm_uframes_t frames, appl_ptr, appl_ofs; snd_pcm_uframes_t avail; @@ -2064,7 +2071,9 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, err = -EAGAIN; goto _end_unlock; } - err = wait_for_avail_min(substream, &avail); + runtime->twake = min_t(snd_pcm_uframes_t, size, + runtime->control->avail_min ? : 1); + err = wait_for_avail(substream, &avail); if (err < 0) goto _end_unlock; if (!avail) diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c index ea2bf82c9373..434af3c56d52 100644 --- a/sound/core/pcm_misc.c +++ b/sound/core/pcm_misc.c @@ -128,6 +128,14 @@ static struct pcm_format_data pcm_formats[SNDRV_PCM_FORMAT_LAST+1] = { .width = 4, .phys = 4, .le = -1, .signd = -1, .silence = {}, }, + [SNDRV_PCM_FORMAT_G723_24] = { + .width = 3, .phys = 3, .le = -1, .signd = -1, + .silence = {}, + }, + [SNDRV_PCM_FORMAT_G723_40] = { + .width = 5, .phys = 5, .le = -1, .signd = -1, + .silence = {}, + }, /* FIXME: the following three formats are not defined properly yet */ [SNDRV_PCM_FORMAT_MPEG] = { .le = -1, .signd = -1, @@ -186,6 +194,14 @@ static struct pcm_format_data pcm_formats[SNDRV_PCM_FORMAT_LAST+1] = { .width = 18, .phys = 24, .le = 0, .signd = 0, .silence = { 0x02, 0x00, 0x00 }, }, + [SNDRV_PCM_FORMAT_G723_24_1B] = { + .width = 3, .phys = 8, .le = -1, .signd = -1, + .silence = {}, + }, + [SNDRV_PCM_FORMAT_G723_40_1B] = { + .width = 5, .phys = 8, .le = -1, .signd = -1, + .silence = {}, + }, }; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 2d2e1b65ee9a..8bc7cb3db330 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -142,7 +142,7 @@ int snd_pcm_info_user(struct snd_pcm_substream *substream, #ifdef RULES_DEBUG #define HW_PARAM(v) [SNDRV_PCM_HW_PARAM_##v] = #v -char *snd_pcm_hw_param_names[] = { +static const char * const snd_pcm_hw_param_names[] = { HW_PARAM(ACCESS), HW_PARAM(FORMAT), HW_PARAM(SUBFORMAT), @@ -451,13 +451,11 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, snd_pcm_timer_resolution_change(substream); runtime->status->state = SNDRV_PCM_STATE_SETUP; - if (substream->latency_pm_qos_req) { - pm_qos_remove_request(substream->latency_pm_qos_req); - substream->latency_pm_qos_req = NULL; - } + if (pm_qos_request_active(&substream->latency_pm_qos_req)) + pm_qos_remove_request(&substream->latency_pm_qos_req); if ((usecs = period_to_usecs(runtime)) >= 0) - substream->latency_pm_qos_req = pm_qos_add_request( - PM_QOS_CPU_DMA_LATENCY, usecs); + pm_qos_add_request(&substream->latency_pm_qos_req, + PM_QOS_CPU_DMA_LATENCY, usecs); return 0; _error: /* hardware might be unuseable from this time, @@ -512,8 +510,7 @@ static int snd_pcm_hw_free(struct snd_pcm_substream *substream) if (substream->ops->hw_free) result = substream->ops->hw_free(substream); runtime->status->state = SNDRV_PCM_STATE_OPEN; - pm_qos_remove_request(substream->latency_pm_qos_req); - substream->latency_pm_qos_req = NULL; + pm_qos_remove_request(&substream->latency_pm_qos_req); return result; } @@ -983,6 +980,10 @@ static int snd_pcm_do_pause(struct snd_pcm_substream *substream, int push) { if (substream->runtime->trigger_master != substream) return 0; + /* some drivers might use hw_ptr to recover from the pause - + update the hw_ptr now */ + if (push) + snd_pcm_update_hw_ptr(substream); /* The jiffies check in snd_pcm_update_hw_ptr*() is done by * a delta betwen the current jiffies, this gives a large enough * delta, effectively to skip the check once. @@ -1993,6 +1994,8 @@ void snd_pcm_release_substream(struct snd_pcm_substream *substream) substream->ops->close(substream); substream->hw_opened = 0; } + if (pm_qos_request_active(&substream->latency_pm_qos_req)) + pm_qos_remove_request(&substream->latency_pm_qos_req); if (substream->pcm_release) { substream->pcm_release(substream); substream->pcm_release = NULL; diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index eb68326c37d4..a7868ad4d530 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -829,6 +829,8 @@ static int snd_rawmidi_control_ioctl(struct snd_card *card, if (get_user(device, (int __user *)argp)) return -EFAULT; + if (device >= SNDRV_RAWMIDI_DEVICES) /* next device is -1 */ + device = SNDRV_RAWMIDI_DEVICES - 1; mutex_lock(®ister_mutex); device = device < 0 ? 0 : device + 1; while (device < SNDRV_RAWMIDI_DEVICES) { diff --git a/sound/core/seq/oss/seq_oss_init.c b/sound/core/seq/oss/seq_oss_init.c index 685712276ac9..69cd7b3c362d 100644 --- a/sound/core/seq/oss/seq_oss_init.c +++ b/sound/core/seq/oss/seq_oss_init.c @@ -281,13 +281,10 @@ snd_seq_oss_open(struct file *file, int level) return 0; _error: - snd_seq_oss_writeq_delete(dp->writeq); - snd_seq_oss_readq_delete(dp->readq); snd_seq_oss_synth_cleanup(dp); snd_seq_oss_midi_cleanup(dp); - delete_port(dp); delete_seq_queue(dp->queue); - kfree(dp); + delete_port(dp); return rc; } @@ -350,8 +347,10 @@ create_port(struct seq_oss_devinfo *dp) static int delete_port(struct seq_oss_devinfo *dp) { - if (dp->port < 0) + if (dp->port < 0) { + kfree(dp); return 0; + } debug_printk(("delete_port %i\n", dp->port)); return snd_seq_event_port_detach(dp->cseq, dp->port); diff --git a/sound/drivers/Kconfig b/sound/drivers/Kconfig index b6ae76285255..c8961165277c 100644 --- a/sound/drivers/Kconfig +++ b/sound/drivers/Kconfig @@ -174,7 +174,7 @@ config SND_ML403_AC97CR select SND_AC97_CODEC help Say Y here to include support for the - opb_ac97_controller_ref_v1_00_a ip core found in Xilinx' ML403 + opb_ac97_controller_ref_v1_00_a ip core found in Xilinx's ML403 reference design. To compile this driver as a module, choose M here: the module @@ -189,9 +189,25 @@ config SND_AC97_POWER_SAVE AC97 codecs. In this mode, the power-mode is dynamically controlled at each open/close. - The mode is activated by passing power_save=1 option to - snd-ac97-codec driver. You can toggle it dynamically over - sysfs, too. + The mode is activated by passing 'power_save=X' to the + snd-ac97-codec driver module, where 'X' is the time-out + value, a nonnegative integer that specifies how many + seconds of idle time the driver must count before it may + put the AC97 into power-save mode; a value of 0 (zero) + disables the use of this power-save mode. + + After the snd-ac97-codec driver module has been loaded, + the 'power_save' parameter can be set via sysfs as follows: + + echo 10 > /sys/module/snd_ac97_codec/parameters/power_save + + In this case, the time-out is set to 10 seconds; setting + the time-out to 1 second (the minimum activation value) + isn't recommended because many applications try to reopen + the device frequently. A value of 10 seconds would be a + good choice for normal operations. + + See Documentation/sound/alsa/powersave.txt for more details. config SND_AC97_POWER_SAVE_DEFAULT int "Default time-out for AC97 power-save mode" @@ -201,4 +217,6 @@ config SND_AC97_POWER_SAVE_DEFAULT The default time-out value in seconds for AC97 automatic power-save mode. 0 means to disable the power-save mode. + See SND_AC97_POWER_SAVE for more details. + endif # SND_DRIVERS diff --git a/sound/drivers/virmidi.c b/sound/drivers/virmidi.c index 0e631c3221e3..f4cd49336f33 100644 --- a/sound/drivers/virmidi.c +++ b/sound/drivers/virmidi.c @@ -94,7 +94,7 @@ static int __devinit snd_virmidi_probe(struct platform_device *devptr) sizeof(struct snd_card_virmidi), &card); if (err < 0) return err; - vmidi = (struct snd_card_virmidi *)card->private_data; + vmidi = card->private_data; vmidi->card = card; if (midi_devs[dev] > MAX_MIDI_DEVICES) { diff --git a/sound/i2c/other/ak4xxx-adda.c b/sound/i2c/other/ak4xxx-adda.c index 1adb8a3c2b62..57ccba88700d 100644 --- a/sound/i2c/other/ak4xxx-adda.c +++ b/sound/i2c/other/ak4xxx-adda.c @@ -878,7 +878,7 @@ static int build_deemphasis(struct snd_akm4xxx *ak, int num_emphs) static void proc_regs_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { - struct snd_akm4xxx *ak = (struct snd_akm4xxx *)entry->private_data; + struct snd_akm4xxx *ak = entry->private_data; int reg, val, chip; for (chip = 0; chip < ak->num_chips; chip++) { for (reg = 0; reg < ak->total_regs; reg++) { @@ -900,7 +900,7 @@ static int proc_init(struct snd_akm4xxx *ak) return 0; } #else /* !CONFIG_PROC_FS */ -static int proc_init(struct snd_akm4xxx *ak) {} +static int proc_init(struct snd_akm4xxx *ak) { return 0; } #endif int snd_akm4xxx_build_controls(struct snd_akm4xxx *ak) diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index c6990c680796..52064cfa91f3 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -77,6 +77,32 @@ config SND_ALS100 To compile this driver as a module, choose M here: the module will be called snd-als100. +config SND_AZT1605 + tristate "Aztech AZT1605 Driver" + depends on SND + select SND_WSS_LIB + select SND_MPU401_UART + select SND_OPL3_LIB + help + Say Y here to include support for Aztech Sound Galaxy cards + based on the AZT1605 chipset. + + To compile this driver as a module, choose M here: the module + will be called snd-azt1605. + +config SND_AZT2316 + tristate "Aztech AZT2316 Driver" + depends on SND + select SND_WSS_LIB + select SND_MPU401_UART + select SND_OPL3_LIB + help + Say Y here to include support for Aztech Sound Galaxy cards + based on the AZT2316 chipset. + + To compile this driver as a module, choose M here: the module + will be called snd-azt2316. + config SND_AZT2320 tristate "Aztech Systems AZT2320" depends on PNP @@ -351,16 +377,6 @@ config SND_SB16_CSP coprocessor can do variable tasks like various compression and decompression algorithms. -config SND_SGALAXY - tristate "Aztech Sound Galaxy" - select SND_WSS_LIB - help - Say Y here to include support for Aztech Sound Galaxy - soundcards. - - To compile this driver as a module, choose M here: the module - will be called snd-sgalaxy. - config SND_SSCAPE tristate "Ensoniq SoundScape driver" select SND_MPU401_UART diff --git a/sound/isa/Makefile b/sound/isa/Makefile index c73d30c4f462..8d781e419e2e 100644 --- a/sound/isa/Makefile +++ b/sound/isa/Makefile @@ -10,7 +10,6 @@ snd-cmi8330-objs := cmi8330.o snd-es18xx-objs := es18xx.o snd-opl3sa2-objs := opl3sa2.o snd-sc6000-objs := sc6000.o -snd-sgalaxy-objs := sgalaxy.o snd-sscape-objs := sscape.o # Toplevel Module Dependency @@ -21,8 +20,7 @@ obj-$(CONFIG_SND_CMI8330) += snd-cmi8330.o obj-$(CONFIG_SND_ES18XX) += snd-es18xx.o obj-$(CONFIG_SND_OPL3SA2) += snd-opl3sa2.o obj-$(CONFIG_SND_SC6000) += snd-sc6000.o -obj-$(CONFIG_SND_SGALAXY) += snd-sgalaxy.o obj-$(CONFIG_SND_SSCAPE) += snd-sscape.o -obj-$(CONFIG_SND) += ad1816a/ ad1848/ cs423x/ es1688/ gus/ msnd/ opti9xx/ \ +obj-$(CONFIG_SND) += ad1816a/ ad1848/ cs423x/ es1688/ galaxy/ gus/ msnd/ opti9xx/ \ sb/ wavefront/ wss/ diff --git a/sound/isa/ad1816a/ad1816a.c b/sound/isa/ad1816a/ad1816a.c index bbcbf92a8ebe..3cb75bc97699 100644 --- a/sound/isa/ad1816a/ad1816a.c +++ b/sound/isa/ad1816a/ad1816a.c @@ -162,7 +162,7 @@ static int __devinit snd_card_ad1816a_probe(int dev, struct pnp_card_link *pcard sizeof(struct snd_card_ad1816a), &card); if (error < 0) return error; - acard = (struct snd_card_ad1816a *)card->private_data; + acard = card->private_data; if ((error = snd_card_ad1816a_pnp(dev, acard, pcard, pid))) { snd_card_free(card); diff --git a/sound/isa/azt2320.c b/sound/isa/azt2320.c index f7aa637b0d18..aac8dc15c2fe 100644 --- a/sound/isa/azt2320.c +++ b/sound/isa/azt2320.c @@ -188,7 +188,7 @@ static int __devinit snd_card_azt2320_probe(int dev, sizeof(struct snd_card_azt2320), &card); if (error < 0) return error; - acard = (struct snd_card_azt2320 *)card->private_data; + acard = card->private_data; if ((error = snd_card_azt2320_pnp(dev, acard, pcard, pid))) { snd_card_free(card); diff --git a/sound/isa/galaxy/Makefile b/sound/isa/galaxy/Makefile new file mode 100644 index 000000000000..e307066d4315 --- /dev/null +++ b/sound/isa/galaxy/Makefile @@ -0,0 +1,10 @@ +# +# Makefile for ALSA +# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz> +# + +snd-azt1605-objs := azt1605.o +snd-azt2316-objs := azt2316.o + +obj-$(CONFIG_SND_AZT1605) += snd-azt1605.o +obj-$(CONFIG_SND_AZT2316) += snd-azt2316.o diff --git a/sound/isa/galaxy/azt1605.c b/sound/isa/galaxy/azt1605.c new file mode 100644 index 000000000000..9a97643cb713 --- /dev/null +++ b/sound/isa/galaxy/azt1605.c @@ -0,0 +1,91 @@ +/* + * Aztech AZT1605 Driver + * Copyright (C) 2007,2010 Rene Herman + * + * This program is free software: you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation, either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>. + * + */ + +#define AZT1605 + +#define CRD_NAME "Aztech AZT1605" +#define DRV_NAME "AZT1605" +#define DEV_NAME "azt1605" + +#define GALAXY_DSP_MAJOR 2 +#define GALAXY_DSP_MINOR 1 + +#define GALAXY_CONFIG_SIZE 3 + +/* + * 24-bit config register + */ + +#define GALAXY_CONFIG_SBA_220 (0 << 0) +#define GALAXY_CONFIG_SBA_240 (1 << 0) +#define GALAXY_CONFIG_SBA_260 (2 << 0) +#define GALAXY_CONFIG_SBA_280 (3 << 0) +#define GALAXY_CONFIG_SBA_MASK GALAXY_CONFIG_SBA_280 + +#define GALAXY_CONFIG_MPUA_300 (0 << 2) +#define GALAXY_CONFIG_MPUA_330 (1 << 2) + +#define GALAXY_CONFIG_MPU_ENABLE (1 << 3) + +#define GALAXY_CONFIG_GAME_ENABLE (1 << 4) + +#define GALAXY_CONFIG_CD_PANASONIC (1 << 5) +#define GALAXY_CONFIG_CD_MITSUMI (1 << 6) +#define GALAXY_CONFIG_CD_MASK (\ + GALAXY_CONFIG_CD_PANASONIC | GALAXY_CONFIG_CD_MITSUMI) + +#define GALAXY_CONFIG_UNUSED (1 << 7) +#define GALAXY_CONFIG_UNUSED_MASK GALAXY_CONFIG_UNUSED + +#define GALAXY_CONFIG_SBIRQ_2 (1 << 8) +#define GALAXY_CONFIG_SBIRQ_3 (1 << 9) +#define GALAXY_CONFIG_SBIRQ_5 (1 << 10) +#define GALAXY_CONFIG_SBIRQ_7 (1 << 11) + +#define GALAXY_CONFIG_MPUIRQ_2 (1 << 12) +#define GALAXY_CONFIG_MPUIRQ_3 (1 << 13) +#define GALAXY_CONFIG_MPUIRQ_5 (1 << 14) +#define GALAXY_CONFIG_MPUIRQ_7 (1 << 15) + +#define GALAXY_CONFIG_WSSA_530 (0 << 16) +#define GALAXY_CONFIG_WSSA_604 (1 << 16) +#define GALAXY_CONFIG_WSSA_E80 (2 << 16) +#define GALAXY_CONFIG_WSSA_F40 (3 << 16) + +#define GALAXY_CONFIG_WSS_ENABLE (1 << 18) + +#define GALAXY_CONFIG_CDIRQ_11 (1 << 19) +#define GALAXY_CONFIG_CDIRQ_12 (1 << 20) +#define GALAXY_CONFIG_CDIRQ_15 (1 << 21) +#define GALAXY_CONFIG_CDIRQ_MASK (\ + GALAXY_CONFIG_CDIRQ_11 | GALAXY_CONFIG_CDIRQ_12 |\ + GALAXY_CONFIG_CDIRQ_15) + +#define GALAXY_CONFIG_CDDMA_DISABLE (0 << 22) +#define GALAXY_CONFIG_CDDMA_0 (1 << 22) +#define GALAXY_CONFIG_CDDMA_1 (2 << 22) +#define GALAXY_CONFIG_CDDMA_3 (3 << 22) +#define GALAXY_CONFIG_CDDMA_MASK GALAXY_CONFIG_CDDMA_3 + +#define GALAXY_CONFIG_MASK (\ + GALAXY_CONFIG_SBA_MASK | GALAXY_CONFIG_CD_MASK |\ + GALAXY_CONFIG_UNUSED_MASK | GALAXY_CONFIG_CDIRQ_MASK |\ + GALAXY_CONFIG_CDDMA_MASK) + +#include "galaxy.c" diff --git a/sound/isa/galaxy/azt2316.c b/sound/isa/galaxy/azt2316.c new file mode 100644 index 000000000000..189441141df6 --- /dev/null +++ b/sound/isa/galaxy/azt2316.c @@ -0,0 +1,111 @@ +/* + * Aztech AZT2316 Driver + * Copyright (C) 2007,2010 Rene Herman + * + * This program is free software: you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation, either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>. + * + */ + +#define AZT2316 + +#define CRD_NAME "Aztech AZT2316" +#define DRV_NAME "AZT2316" +#define DEV_NAME "azt2316" + +#define GALAXY_DSP_MAJOR 3 +#define GALAXY_DSP_MINOR 1 + +#define GALAXY_CONFIG_SIZE 4 + +/* + * 32-bit config register + */ + +#define GALAXY_CONFIG_SBA_220 (0 << 0) +#define GALAXY_CONFIG_SBA_240 (1 << 0) +#define GALAXY_CONFIG_SBA_260 (2 << 0) +#define GALAXY_CONFIG_SBA_280 (3 << 0) +#define GALAXY_CONFIG_SBA_MASK GALAXY_CONFIG_SBA_280 + +#define GALAXY_CONFIG_SBIRQ_2 (1 << 2) +#define GALAXY_CONFIG_SBIRQ_5 (1 << 3) +#define GALAXY_CONFIG_SBIRQ_7 (1 << 4) +#define GALAXY_CONFIG_SBIRQ_10 (1 << 5) + +#define GALAXY_CONFIG_SBDMA_DISABLE (0 << 6) +#define GALAXY_CONFIG_SBDMA_0 (1 << 6) +#define GALAXY_CONFIG_SBDMA_1 (2 << 6) +#define GALAXY_CONFIG_SBDMA_3 (3 << 6) + +#define GALAXY_CONFIG_WSSA_530 (0 << 8) +#define GALAXY_CONFIG_WSSA_604 (1 << 8) +#define GALAXY_CONFIG_WSSA_E80 (2 << 8) +#define GALAXY_CONFIG_WSSA_F40 (3 << 8) + +#define GALAXY_CONFIG_WSS_ENABLE (1 << 10) + +#define GALAXY_CONFIG_GAME_ENABLE (1 << 11) + +#define GALAXY_CONFIG_MPUA_300 (0 << 12) +#define GALAXY_CONFIG_MPUA_330 (1 << 12) + +#define GALAXY_CONFIG_MPU_ENABLE (1 << 13) + +#define GALAXY_CONFIG_CDA_310 (0 << 14) +#define GALAXY_CONFIG_CDA_320 (1 << 14) +#define GALAXY_CONFIG_CDA_340 (2 << 14) +#define GALAXY_CONFIG_CDA_350 (3 << 14) +#define GALAXY_CONFIG_CDA_MASK GALAXY_CONFIG_CDA_350 + +#define GALAXY_CONFIG_CD_DISABLE (0 << 16) +#define GALAXY_CONFIG_CD_PANASONIC (1 << 16) +#define GALAXY_CONFIG_CD_SONY (2 << 16) +#define GALAXY_CONFIG_CD_MITSUMI (3 << 16) +#define GALAXY_CONFIG_CD_AZTECH (4 << 16) +#define GALAXY_CONFIG_CD_UNUSED_5 (5 << 16) +#define GALAXY_CONFIG_CD_UNUSED_6 (6 << 16) +#define GALAXY_CONFIG_CD_UNUSED_7 (7 << 16) +#define GALAXY_CONFIG_CD_MASK GALAXY_CONFIG_CD_UNUSED_7 + +#define GALAXY_CONFIG_CDDMA8_DISABLE (0 << 20) +#define GALAXY_CONFIG_CDDMA8_0 (1 << 20) +#define GALAXY_CONFIG_CDDMA8_1 (2 << 20) +#define GALAXY_CONFIG_CDDMA8_3 (3 << 20) +#define GALAXY_CONFIG_CDDMA8_MASK GALAXY_CONFIG_CDDMA8_3 + +#define GALAXY_CONFIG_CDDMA16_DISABLE (0 << 22) +#define GALAXY_CONFIG_CDDMA16_5 (1 << 22) +#define GALAXY_CONFIG_CDDMA16_6 (2 << 22) +#define GALAXY_CONFIG_CDDMA16_7 (3 << 22) +#define GALAXY_CONFIG_CDDMA16_MASK GALAXY_CONFIG_CDDMA16_7 + +#define GALAXY_CONFIG_MPUIRQ_2 (1 << 24) +#define GALAXY_CONFIG_MPUIRQ_5 (1 << 25) +#define GALAXY_CONFIG_MPUIRQ_7 (1 << 26) +#define GALAXY_CONFIG_MPUIRQ_10 (1 << 27) + +#define GALAXY_CONFIG_CDIRQ_5 (1 << 28) +#define GALAXY_CONFIG_CDIRQ_11 (1 << 29) +#define GALAXY_CONFIG_CDIRQ_12 (1 << 30) +#define GALAXY_CONFIG_CDIRQ_15 (1 << 31) +#define GALAXY_CONFIG_CDIRQ_MASK (\ + GALAXY_CONFIG_CDIRQ_5 | GALAXY_CONFIG_CDIRQ_11 |\ + GALAXY_CONFIG_CDIRQ_12 | GALAXY_CONFIG_CDIRQ_15) + +#define GALAXY_CONFIG_MASK (\ + GALAXY_CONFIG_SBA_MASK | GALAXY_CONFIG_CDA_MASK |\ + GALAXY_CONFIG_CD_MASK | GALAXY_CONFIG_CDDMA16_MASK |\ + GALAXY_CONFIG_CDDMA8_MASK | GALAXY_CONFIG_CDIRQ_MASK) + +#include "galaxy.c" diff --git a/sound/isa/galaxy/galaxy.c b/sound/isa/galaxy/galaxy.c new file mode 100644 index 000000000000..ee54df082b9c --- /dev/null +++ b/sound/isa/galaxy/galaxy.c @@ -0,0 +1,652 @@ +/* + * Aztech AZT1605/AZT2316 Driver + * Copyright (C) 2007,2010 Rene Herman + * + * This program is free software: you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation, either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>. + * + */ + +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/isa.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <asm/processor.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/wss.h> +#include <sound/mpu401.h> +#include <sound/opl3.h> + +MODULE_DESCRIPTION(CRD_NAME); +MODULE_AUTHOR("Rene Herman"); +MODULE_LICENSE("GPL"); + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for " CRD_NAME " soundcard."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for " CRD_NAME " soundcard."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable " CRD_NAME " soundcard."); + +static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +static long wss_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +static long mpu_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +static long fm_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; +static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; +static int dma1[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; +static int dma2[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; + +module_param_array(port, long, NULL, 0444); +MODULE_PARM_DESC(port, "Port # for " CRD_NAME " driver."); +module_param_array(wss_port, long, NULL, 0444); +MODULE_PARM_DESC(wss_port, "WSS port # for " CRD_NAME " driver."); +module_param_array(mpu_port, long, NULL, 0444); +MODULE_PARM_DESC(mpu_port, "MPU-401 port # for " CRD_NAME " driver."); +module_param_array(fm_port, long, NULL, 0444); +MODULE_PARM_DESC(fm_port, "FM port # for " CRD_NAME " driver."); +module_param_array(irq, int, NULL, 0444); +MODULE_PARM_DESC(irq, "IRQ # for " CRD_NAME " driver."); +module_param_array(mpu_irq, int, NULL, 0444); +MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for " CRD_NAME " driver."); +module_param_array(dma1, int, NULL, 0444); +MODULE_PARM_DESC(dma1, "Playback DMA # for " CRD_NAME " driver."); +module_param_array(dma2, int, NULL, 0444); +MODULE_PARM_DESC(dma2, "Capture DMA # for " CRD_NAME " driver."); + +/* + * Generic SB DSP support routines + */ + +#define DSP_PORT_RESET 0x6 +#define DSP_PORT_READ 0xa +#define DSP_PORT_COMMAND 0xc +#define DSP_PORT_STATUS 0xc +#define DSP_PORT_DATA_AVAIL 0xe + +#define DSP_SIGNATURE 0xaa + +#define DSP_COMMAND_GET_VERSION 0xe1 + +static int __devinit dsp_get_byte(void __iomem *port, u8 *val) +{ + int loops = 1000; + + while (!(ioread8(port + DSP_PORT_DATA_AVAIL) & 0x80)) { + if (!loops--) + return -EIO; + cpu_relax(); + } + *val = ioread8(port + DSP_PORT_READ); + return 0; +} + +static int __devinit dsp_reset(void __iomem *port) +{ + u8 val; + + iowrite8(1, port + DSP_PORT_RESET); + udelay(10); + iowrite8(0, port + DSP_PORT_RESET); + + if (dsp_get_byte(port, &val) < 0 || val != DSP_SIGNATURE) + return -ENODEV; + + return 0; +} + +static int __devinit dsp_command(void __iomem *port, u8 cmd) +{ + int loops = 1000; + + while (ioread8(port + DSP_PORT_STATUS) & 0x80) { + if (!loops--) + return -EIO; + cpu_relax(); + } + iowrite8(cmd, port + DSP_PORT_COMMAND); + return 0; +} + +static int __devinit dsp_get_version(void __iomem *port, u8 *major, u8 *minor) +{ + int err; + + err = dsp_command(port, DSP_COMMAND_GET_VERSION); + if (err < 0) + return err; + + err = dsp_get_byte(port, major); + if (err < 0) + return err; + + err = dsp_get_byte(port, minor); + if (err < 0) + return err; + + return 0; +} + +/* + * Generic WSS support routines + */ + +#define WSS_CONFIG_DMA_0 (1 << 0) +#define WSS_CONFIG_DMA_1 (2 << 0) +#define WSS_CONFIG_DMA_3 (3 << 0) +#define WSS_CONFIG_DUPLEX (1 << 2) +#define WSS_CONFIG_IRQ_7 (1 << 3) +#define WSS_CONFIG_IRQ_9 (2 << 3) +#define WSS_CONFIG_IRQ_10 (3 << 3) +#define WSS_CONFIG_IRQ_11 (4 << 3) + +#define WSS_PORT_CONFIG 0 +#define WSS_PORT_SIGNATURE 3 + +#define WSS_SIGNATURE 4 + +static int __devinit wss_detect(void __iomem *wss_port) +{ + if ((ioread8(wss_port + WSS_PORT_SIGNATURE) & 0x3f) != WSS_SIGNATURE) + return -ENODEV; + + return 0; +} + +static void wss_set_config(void __iomem *wss_port, u8 wss_config) +{ + iowrite8(wss_config, wss_port + WSS_PORT_CONFIG); +} + +/* + * Aztech Sound Galaxy specifics + */ + +#define GALAXY_PORT_CONFIG 1024 +#define CONFIG_PORT_SET 4 + +#define DSP_COMMAND_GALAXY_8 8 +#define GALAXY_COMMAND_GET_TYPE 5 + +#define DSP_COMMAND_GALAXY_9 9 +#define GALAXY_COMMAND_WSSMODE 0 +#define GALAXY_COMMAND_SB8MODE 1 + +#define GALAXY_MODE_WSS GALAXY_COMMAND_WSSMODE +#define GALAXY_MODE_SB8 GALAXY_COMMAND_SB8MODE + +struct snd_galaxy { + void __iomem *port; + void __iomem *config_port; + void __iomem *wss_port; + u32 config; + struct resource *res_port; + struct resource *res_config_port; + struct resource *res_wss_port; +}; + +static u32 config[SNDRV_CARDS]; +static u8 wss_config[SNDRV_CARDS]; + +static int __devinit snd_galaxy_match(struct device *dev, unsigned int n) +{ + if (!enable[n]) + return 0; + + switch (port[n]) { + case SNDRV_AUTO_PORT: + dev_err(dev, "please specify port\n"); + return 0; + case 0x220: + config[n] |= GALAXY_CONFIG_SBA_220; + break; + case 0x240: + config[n] |= GALAXY_CONFIG_SBA_240; + break; + case 0x260: + config[n] |= GALAXY_CONFIG_SBA_260; + break; + case 0x280: + config[n] |= GALAXY_CONFIG_SBA_280; + break; + default: + dev_err(dev, "invalid port %#lx\n", port[n]); + return 0; + } + + switch (wss_port[n]) { + case SNDRV_AUTO_PORT: + dev_err(dev, "please specify wss_port\n"); + return 0; + case 0x530: + config[n] |= GALAXY_CONFIG_WSS_ENABLE | GALAXY_CONFIG_WSSA_530; + break; + case 0x604: + config[n] |= GALAXY_CONFIG_WSS_ENABLE | GALAXY_CONFIG_WSSA_604; + break; + case 0xe80: + config[n] |= GALAXY_CONFIG_WSS_ENABLE | GALAXY_CONFIG_WSSA_E80; + break; + case 0xf40: + config[n] |= GALAXY_CONFIG_WSS_ENABLE | GALAXY_CONFIG_WSSA_F40; + break; + default: + dev_err(dev, "invalid WSS port %#lx\n", wss_port[n]); + return 0; + } + + switch (irq[n]) { + case SNDRV_AUTO_IRQ: + dev_err(dev, "please specify irq\n"); + return 0; + case 7: + wss_config[n] |= WSS_CONFIG_IRQ_7; + break; + case 2: + irq[n] = 9; + case 9: + wss_config[n] |= WSS_CONFIG_IRQ_9; + break; + case 10: + wss_config[n] |= WSS_CONFIG_IRQ_10; + break; + case 11: + wss_config[n] |= WSS_CONFIG_IRQ_11; + break; + default: + dev_err(dev, "invalid IRQ %d\n", irq[n]); + return 0; + } + + switch (dma1[n]) { + case SNDRV_AUTO_DMA: + dev_err(dev, "please specify dma1\n"); + return 0; + case 0: + wss_config[n] |= WSS_CONFIG_DMA_0; + break; + case 1: + wss_config[n] |= WSS_CONFIG_DMA_1; + break; + case 3: + wss_config[n] |= WSS_CONFIG_DMA_3; + break; + default: + dev_err(dev, "invalid playback DMA %d\n", dma1[n]); + return 0; + } + + if (dma2[n] == SNDRV_AUTO_DMA || dma2[n] == dma1[n]) { + dma2[n] = -1; + goto mpu; + } + + wss_config[n] |= WSS_CONFIG_DUPLEX; + switch (dma2[n]) { + case 0: + break; + case 1: + if (dma1[n] == 0) + break; + default: + dev_err(dev, "invalid capture DMA %d\n", dma2[n]); + return 0; + } + +mpu: + switch (mpu_port[n]) { + case SNDRV_AUTO_PORT: + dev_warn(dev, "mpu_port not specified; not using MPU-401\n"); + mpu_port[n] = -1; + goto fm; + case 0x300: + config[n] |= GALAXY_CONFIG_MPU_ENABLE | GALAXY_CONFIG_MPUA_300; + break; + case 0x330: + config[n] |= GALAXY_CONFIG_MPU_ENABLE | GALAXY_CONFIG_MPUA_330; + break; + default: + dev_err(dev, "invalid MPU port %#lx\n", mpu_port[n]); + return 0; + } + + switch (mpu_irq[n]) { + case SNDRV_AUTO_IRQ: + dev_warn(dev, "mpu_irq not specified: using polling mode\n"); + mpu_irq[n] = -1; + break; + case 2: + mpu_irq[n] = 9; + case 9: + config[n] |= GALAXY_CONFIG_MPUIRQ_2; + break; +#ifdef AZT1605 + case 3: + config[n] |= GALAXY_CONFIG_MPUIRQ_3; + break; +#endif + case 5: + config[n] |= GALAXY_CONFIG_MPUIRQ_5; + break; + case 7: + config[n] |= GALAXY_CONFIG_MPUIRQ_7; + break; +#ifdef AZT2316 + case 10: + config[n] |= GALAXY_CONFIG_MPUIRQ_10; + break; +#endif + default: + dev_err(dev, "invalid MPU IRQ %d\n", mpu_irq[n]); + return 0; + } + + if (mpu_irq[n] == irq[n]) { + dev_err(dev, "cannot share IRQ between WSS and MPU-401\n"); + return 0; + } + +fm: + switch (fm_port[n]) { + case SNDRV_AUTO_PORT: + dev_warn(dev, "fm_port not specified: not using OPL3\n"); + fm_port[n] = -1; + break; + case 0x388: + break; + default: + dev_err(dev, "illegal FM port %#lx\n", fm_port[n]); + return 0; + } + + config[n] |= GALAXY_CONFIG_GAME_ENABLE; + return 1; +} + +static int __devinit galaxy_init(struct snd_galaxy *galaxy, u8 *type) +{ + u8 major; + u8 minor; + int err; + + err = dsp_reset(galaxy->port); + if (err < 0) + return err; + + err = dsp_get_version(galaxy->port, &major, &minor); + if (err < 0) + return err; + + if (major != GALAXY_DSP_MAJOR || minor != GALAXY_DSP_MINOR) + return -ENODEV; + + err = dsp_command(galaxy->port, DSP_COMMAND_GALAXY_8); + if (err < 0) + return err; + + err = dsp_command(galaxy->port, GALAXY_COMMAND_GET_TYPE); + if (err < 0) + return err; + + err = dsp_get_byte(galaxy->port, type); + if (err < 0) + return err; + + return 0; +} + +static int __devinit galaxy_set_mode(struct snd_galaxy *galaxy, u8 mode) +{ + int err; + + err = dsp_command(galaxy->port, DSP_COMMAND_GALAXY_9); + if (err < 0) + return err; + + err = dsp_command(galaxy->port, mode); + if (err < 0) + return err; + +#ifdef AZT1605 + /* + * Needed for MPU IRQ on AZT1605, but AZT2316 loses WSS again + */ + err = dsp_reset(galaxy->port); + if (err < 0) + return err; +#endif + + return 0; +} + +static void galaxy_set_config(struct snd_galaxy *galaxy, u32 config) +{ + u8 tmp = ioread8(galaxy->config_port + CONFIG_PORT_SET); + int i; + + iowrite8(tmp | 0x80, galaxy->config_port + CONFIG_PORT_SET); + for (i = 0; i < GALAXY_CONFIG_SIZE; i++) { + iowrite8(config, galaxy->config_port + i); + config >>= 8; + } + iowrite8(tmp & 0x7f, galaxy->config_port + CONFIG_PORT_SET); + msleep(10); +} + +static void __devinit galaxy_config(struct snd_galaxy *galaxy, u32 config) +{ + int i; + + for (i = GALAXY_CONFIG_SIZE; i; i--) { + u8 tmp = ioread8(galaxy->config_port + i - 1); + galaxy->config = (galaxy->config << 8) | tmp; + } + config |= galaxy->config & GALAXY_CONFIG_MASK; + galaxy_set_config(galaxy, config); +} + +static int __devinit galaxy_wss_config(struct snd_galaxy *galaxy, u8 wss_config) +{ + int err; + + err = wss_detect(galaxy->wss_port); + if (err < 0) + return err; + + wss_set_config(galaxy->wss_port, wss_config); + + err = galaxy_set_mode(galaxy, GALAXY_MODE_WSS); + if (err < 0) + return err; + + return 0; +} + +static void snd_galaxy_free(struct snd_card *card) +{ + struct snd_galaxy *galaxy = card->private_data; + + if (galaxy->wss_port) { + wss_set_config(galaxy->wss_port, 0); + ioport_unmap(galaxy->wss_port); + release_and_free_resource(galaxy->res_wss_port); + } + if (galaxy->config_port) { + galaxy_set_config(galaxy, galaxy->config); + ioport_unmap(galaxy->config_port); + release_and_free_resource(galaxy->res_config_port); + } + if (galaxy->port) { + ioport_unmap(galaxy->port); + release_and_free_resource(galaxy->res_port); + } +} + +static int __devinit snd_galaxy_probe(struct device *dev, unsigned int n) +{ + struct snd_galaxy *galaxy; + struct snd_wss *chip; + struct snd_card *card; + u8 type; + int err; + + err = snd_card_create(index[n], id[n], THIS_MODULE, sizeof *galaxy, + &card); + if (err < 0) + return err; + + snd_card_set_dev(card, dev); + + card->private_free = snd_galaxy_free; + galaxy = card->private_data; + + galaxy->res_port = request_region(port[n], 16, DRV_NAME); + if (!galaxy->res_port) { + dev_err(dev, "could not grab ports %#lx-%#lx\n", port[n], + port[n] + 15); + err = -EBUSY; + goto error; + } + galaxy->port = ioport_map(port[n], 16); + + err = galaxy_init(galaxy, &type); + if (err < 0) { + dev_err(dev, "did not find a Sound Galaxy at %#lx\n", port[n]); + goto error; + } + dev_info(dev, "Sound Galaxy (type %d) found at %#lx\n", type, port[n]); + + galaxy->res_config_port = request_region(port[n] + GALAXY_PORT_CONFIG, + 16, DRV_NAME); + if (!galaxy->res_config_port) { + dev_err(dev, "could not grab ports %#lx-%#lx\n", + port[n] + GALAXY_PORT_CONFIG, + port[n] + GALAXY_PORT_CONFIG + 15); + err = -EBUSY; + goto error; + } + galaxy->config_port = ioport_map(port[n] + GALAXY_PORT_CONFIG, 16); + + galaxy_config(galaxy, config[n]); + + galaxy->res_wss_port = request_region(wss_port[n], 4, DRV_NAME); + if (!galaxy->res_wss_port) { + dev_err(dev, "could not grab ports %#lx-%#lx\n", wss_port[n], + wss_port[n] + 3); + err = -EBUSY; + goto error; + } + galaxy->wss_port = ioport_map(wss_port[n], 4); + + err = galaxy_wss_config(galaxy, wss_config[n]); + if (err < 0) { + dev_err(dev, "could not configure WSS\n"); + goto error; + } + + strcpy(card->driver, DRV_NAME); + strcpy(card->shortname, DRV_NAME); + sprintf(card->longname, "%s at %#lx/%#lx, irq %d, dma %d/%d", + card->shortname, port[n], wss_port[n], irq[n], dma1[n], + dma2[n]); + + err = snd_wss_create(card, wss_port[n] + 4, -1, irq[n], dma1[n], + dma2[n], WSS_HW_DETECT, 0, &chip); + if (err < 0) + goto error; + + err = snd_wss_pcm(chip, 0, NULL); + if (err < 0) + goto error; + + err = snd_wss_mixer(chip); + if (err < 0) + goto error; + + err = snd_wss_timer(chip, 0, NULL); + if (err < 0) + goto error; + + if (mpu_port[n] >= 0) { + err = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, + mpu_port[n], 0, mpu_irq[n], + IRQF_DISABLED, NULL); + if (err < 0) + goto error; + } + + if (fm_port[n] >= 0) { + struct snd_opl3 *opl3; + + err = snd_opl3_create(card, fm_port[n], fm_port[n] + 2, + OPL3_HW_AUTO, 0, &opl3); + if (err < 0) { + dev_err(dev, "no OPL device at %#lx\n", fm_port[n]); + goto error; + } + err = snd_opl3_timer_new(opl3, 1, 2); + if (err < 0) + goto error; + + err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); + if (err < 0) + goto error; + } + + err = snd_card_register(card); + if (err < 0) + goto error; + + dev_set_drvdata(dev, card); + return 0; + +error: + snd_card_free(card); + return err; +} + +static int __devexit snd_galaxy_remove(struct device *dev, unsigned int n) +{ + snd_card_free(dev_get_drvdata(dev)); + dev_set_drvdata(dev, NULL); + return 0; +} + +static struct isa_driver snd_galaxy_driver = { + .match = snd_galaxy_match, + .probe = snd_galaxy_probe, + .remove = __devexit_p(snd_galaxy_remove), + + .driver = { + .name = DEV_NAME + } +}; + +static int __init alsa_card_galaxy_init(void) +{ + return isa_register_driver(&snd_galaxy_driver, SNDRV_CARDS); +} + +static void __exit alsa_card_galaxy_exit(void) +{ + isa_unregister_driver(&snd_galaxy_driver); +} + +module_init(alsa_card_galaxy_init); +module_exit(alsa_card_galaxy_exit); diff --git a/sound/isa/gus/gusmax.c b/sound/isa/gus/gusmax.c index f26eac8d8110..3e4a58b72913 100644 --- a/sound/isa/gus/gusmax.c +++ b/sound/isa/gus/gusmax.c @@ -191,7 +191,7 @@ static int __devinit snd_gusmax_mixer(struct snd_wss *chip) static void snd_gusmax_free(struct snd_card *card) { - struct snd_gusmax *maxcard = (struct snd_gusmax *)card->private_data; + struct snd_gusmax *maxcard = card->private_data; if (maxcard == NULL) return; @@ -219,7 +219,7 @@ static int __devinit snd_gusmax_probe(struct device *pdev, unsigned int dev) if (err < 0) return err; card->private_free = snd_gusmax_free; - maxcard = (struct snd_gusmax *)card->private_data; + maxcard = card->private_data; maxcard->card = card; maxcard->irq = -1; diff --git a/sound/isa/msnd/msnd_pinnacle.c b/sound/isa/msnd/msnd_pinnacle.c index 60b6abd71612..91d6023a63e5 100644 --- a/sound/isa/msnd/msnd_pinnacle.c +++ b/sound/isa/msnd/msnd_pinnacle.c @@ -549,7 +549,10 @@ static int __devinit snd_msnd_attach(struct snd_card *card) printk(KERN_ERR LOGNAME ": Couldn't grab IRQ %d\n", chip->irq); return err; } - request_region(chip->io, DSP_NUMIO, card->shortname); + if (request_region(chip->io, DSP_NUMIO, card->shortname) == NULL) { + free_irq(chip->irq, chip); + return -EBUSY; + } if (!request_mem_region(chip->base, BUFFSIZE, card->shortname)) { printk(KERN_ERR LOGNAME @@ -761,9 +764,9 @@ static long io[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; static long mem[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +#ifndef MSND_CLASSIC static long cfg[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; -#ifndef MSND_CLASSIC /* Extra Peripheral Configuration (Default: Disable) */ static long ide_io0[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; static long ide_io1[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; @@ -891,7 +894,11 @@ static int __devinit snd_msnd_isa_probe(struct device *pdev, unsigned int idx) struct snd_card *card; struct snd_msnd *chip; - if (has_isapnp(idx) || cfg[idx] == SNDRV_AUTO_PORT) { + if (has_isapnp(idx) +#ifndef MSND_CLASSIC + || cfg[idx] == SNDRV_AUTO_PORT +#endif + ) { printk(KERN_INFO LOGNAME ": Assuming PnP mode\n"); return -ENODEV; } diff --git a/sound/isa/sb/emu8000_pcm.c b/sound/isa/sb/emu8000_pcm.c index ccedbfed061a..2f85c66f8e38 100644 --- a/sound/isa/sb/emu8000_pcm.c +++ b/sound/isa/sb/emu8000_pcm.c @@ -433,7 +433,8 @@ static int emu8k_transfer_block(struct snd_emu8000 *emu, int offset, unsigned sh while (count > 0) { unsigned short sval; CHECK_SCHEDULER(); - get_user(sval, buf); + if (get_user(sval, buf)) + return -EFAULT; EMU8000_SMLD_WRITE(emu, sval); buf++; count--; @@ -525,12 +526,14 @@ static int emu8k_pcm_copy(struct snd_pcm_substream *subs, while (count-- > 0) { unsigned short sval; CHECK_SCHEDULER(); - get_user(sval, buf); + if (get_user(sval, buf)) + return -EFAULT; EMU8000_SMLD_WRITE(emu, sval); buf++; if (rec->voices > 1) { CHECK_SCHEDULER(); - get_user(sval, buf); + if (get_user(sval, buf)) + return -EFAULT; EMU8000_SMRD_WRITE(emu, sval); buf++; } diff --git a/sound/isa/sb/sb8.c b/sound/isa/sb/sb8.c index 81284a8fa0ce..2259e3f726a7 100644 --- a/sound/isa/sb/sb8.c +++ b/sound/isa/sb/sb8.c @@ -72,7 +72,7 @@ static irqreturn_t snd_sb8_interrupt(int irq, void *dev_id) static void snd_sb8_free(struct snd_card *card) { - struct snd_sb8 *acard = (struct snd_sb8 *)card->private_data; + struct snd_sb8 *acard = card->private_data; if (acard == NULL) return; diff --git a/sound/isa/sgalaxy.c b/sound/isa/sgalaxy.c deleted file mode 100644 index 6fe27b9d9440..000000000000 --- a/sound/isa/sgalaxy.c +++ /dev/null @@ -1,369 +0,0 @@ -/* - * Driver for Aztech Sound Galaxy cards - * Copyright (c) by Christopher Butler <chrisb@sandy.force9.co.uk. - * - * I don't have documentation for this card, I based this driver on the - * driver for OSS/Free included in the kernel source (drivers/sound/sgalaxy.c) - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * - */ - -#include <linux/init.h> -#include <linux/err.h> -#include <linux/isa.h> -#include <linux/delay.h> -#include <linux/time.h> -#include <linux/interrupt.h> -#include <linux/moduleparam.h> -#include <asm/dma.h> -#include <sound/core.h> -#include <sound/sb.h> -#include <sound/wss.h> -#include <sound/control.h> -#define SNDRV_LEGACY_FIND_FREE_IRQ -#define SNDRV_LEGACY_FIND_FREE_DMA -#include <sound/initval.h> - -MODULE_AUTHOR("Christopher Butler <chrisb@sandy.force9.co.uk>"); -MODULE_DESCRIPTION("Aztech Sound Galaxy"); -MODULE_LICENSE("GPL"); -MODULE_SUPPORTED_DEVICE("{{Aztech Systems,Sound Galaxy}}"); - -static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ -static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ -static long sbport[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x220,0x240 */ -static long wssport[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x530,0xe80,0xf40,0x604 */ -static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 7,9,10,11 */ -static int dma1[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 0,1,3 */ - -module_param_array(index, int, NULL, 0444); -MODULE_PARM_DESC(index, "Index value for Sound Galaxy soundcard."); -module_param_array(id, charp, NULL, 0444); -MODULE_PARM_DESC(id, "ID string for Sound Galaxy soundcard."); -module_param_array(sbport, long, NULL, 0444); -MODULE_PARM_DESC(sbport, "Port # for Sound Galaxy SB driver."); -module_param_array(wssport, long, NULL, 0444); -MODULE_PARM_DESC(wssport, "Port # for Sound Galaxy WSS driver."); -module_param_array(irq, int, NULL, 0444); -MODULE_PARM_DESC(irq, "IRQ # for Sound Galaxy driver."); -module_param_array(dma1, int, NULL, 0444); -MODULE_PARM_DESC(dma1, "DMA1 # for Sound Galaxy driver."); - -#define SGALAXY_AUXC_LEFT 18 -#define SGALAXY_AUXC_RIGHT 19 - -#define PFX "sgalaxy: " - -/* - - */ - -#define AD1848P1( port, x ) ( port + c_d_c_AD1848##x ) - -/* from lowlevel/sb/sb.c - to avoid having to allocate a struct snd_sb for the */ -/* short time we actually need it.. */ - -static int snd_sgalaxy_sbdsp_reset(unsigned long port) -{ - int i; - - outb(1, SBP1(port, RESET)); - udelay(10); - outb(0, SBP1(port, RESET)); - udelay(30); - for (i = 0; i < 1000 && !(inb(SBP1(port, DATA_AVAIL)) & 0x80); i++); - if (inb(SBP1(port, READ)) != 0xaa) { - snd_printd("sb_reset: failed at 0x%lx!!!\n", port); - return -ENODEV; - } - return 0; -} - -static int __devinit snd_sgalaxy_sbdsp_command(unsigned long port, - unsigned char val) -{ - int i; - - for (i = 10000; i; i--) - if ((inb(SBP1(port, STATUS)) & 0x80) == 0) { - outb(val, SBP1(port, COMMAND)); - return 1; - } - - return 0; -} - -static irqreturn_t snd_sgalaxy_dummy_interrupt(int irq, void *dev_id) -{ - return IRQ_NONE; -} - -static int __devinit snd_sgalaxy_setup_wss(unsigned long port, int irq, int dma) -{ - static int interrupt_bits[] = {-1, -1, -1, -1, -1, -1, -1, 0x08, -1, - 0x10, 0x18, 0x20, -1, -1, -1, -1}; - static int dma_bits[] = {1, 2, 0, 3}; - int tmp, tmp1; - - if ((tmp = inb(port + 3)) == 0xff) - { - snd_printdd("I/O address dead (0x%lx)\n", port); - return 0; - } -#if 0 - snd_printdd("WSS signature = 0x%x\n", tmp); -#endif - - if ((tmp & 0x3f) != 0x04 && - (tmp & 0x3f) != 0x0f && - (tmp & 0x3f) != 0x00) { - snd_printdd("No WSS signature detected on port 0x%lx\n", - port + 3); - return 0; - } - -#if 0 - snd_printdd(PFX "setting up IRQ/DMA for WSS\n"); -#endif - - /* initialize IRQ for WSS codec */ - tmp = interrupt_bits[irq % 16]; - if (tmp < 0) - return -EINVAL; - - if (request_irq(irq, snd_sgalaxy_dummy_interrupt, IRQF_DISABLED, "sgalaxy", NULL)) { - snd_printk(KERN_ERR "sgalaxy: can't grab irq %d\n", irq); - return -EIO; - } - - outb(tmp | 0x40, port); - tmp1 = dma_bits[dma % 4]; - outb(tmp | tmp1, port); - - free_irq(irq, NULL); - - return 0; -} - -static int __devinit snd_sgalaxy_detect(int dev, int irq, int dma) -{ -#if 0 - snd_printdd(PFX "switching to WSS mode\n"); -#endif - - /* switch to WSS mode */ - snd_sgalaxy_sbdsp_reset(sbport[dev]); - - snd_sgalaxy_sbdsp_command(sbport[dev], 9); - snd_sgalaxy_sbdsp_command(sbport[dev], 0); - - udelay(400); - return snd_sgalaxy_setup_wss(wssport[dev], irq, dma); -} - -static struct snd_kcontrol_new snd_sgalaxy_controls[] = { -WSS_DOUBLE("Aux Playback Switch", 0, - SGALAXY_AUXC_LEFT, SGALAXY_AUXC_RIGHT, 7, 7, 1, 1), -WSS_DOUBLE("Aux Playback Volume", 0, - SGALAXY_AUXC_LEFT, SGALAXY_AUXC_RIGHT, 0, 0, 31, 0) -}; - -static int __devinit snd_sgalaxy_mixer(struct snd_wss *chip) -{ - struct snd_card *card = chip->card; - struct snd_ctl_elem_id id1, id2; - unsigned int idx; - int err; - - memset(&id1, 0, sizeof(id1)); - memset(&id2, 0, sizeof(id2)); - id1.iface = id2.iface = SNDRV_CTL_ELEM_IFACE_MIXER; - /* reassign AUX0 to LINE */ - strcpy(id1.name, "Aux Playback Switch"); - strcpy(id2.name, "Line Playback Switch"); - if ((err = snd_ctl_rename_id(card, &id1, &id2)) < 0) - return err; - strcpy(id1.name, "Aux Playback Volume"); - strcpy(id2.name, "Line Playback Volume"); - if ((err = snd_ctl_rename_id(card, &id1, &id2)) < 0) - return err; - /* reassign AUX1 to FM */ - strcpy(id1.name, "Aux Playback Switch"); id1.index = 1; - strcpy(id2.name, "FM Playback Switch"); - if ((err = snd_ctl_rename_id(card, &id1, &id2)) < 0) - return err; - strcpy(id1.name, "Aux Playback Volume"); - strcpy(id2.name, "FM Playback Volume"); - if ((err = snd_ctl_rename_id(card, &id1, &id2)) < 0) - return err; - /* build AUX2 input */ - for (idx = 0; idx < ARRAY_SIZE(snd_sgalaxy_controls); idx++) { - err = snd_ctl_add(card, - snd_ctl_new1(&snd_sgalaxy_controls[idx], chip)); - if (err < 0) - return err; - } - return 0; -} - -static int __devinit snd_sgalaxy_match(struct device *devptr, unsigned int dev) -{ - if (!enable[dev]) - return 0; - if (sbport[dev] == SNDRV_AUTO_PORT) { - snd_printk(KERN_ERR PFX "specify SB port\n"); - return 0; - } - if (wssport[dev] == SNDRV_AUTO_PORT) { - snd_printk(KERN_ERR PFX "specify WSS port\n"); - return 0; - } - return 1; -} - -static int __devinit snd_sgalaxy_probe(struct device *devptr, unsigned int dev) -{ - static int possible_irqs[] = {7, 9, 10, 11, -1}; - static int possible_dmas[] = {1, 3, 0, -1}; - int err, xirq, xdma1; - struct snd_card *card; - struct snd_wss *chip; - - err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); - if (err < 0) - return err; - - xirq = irq[dev]; - if (xirq == SNDRV_AUTO_IRQ) { - if ((xirq = snd_legacy_find_free_irq(possible_irqs)) < 0) { - snd_printk(KERN_ERR PFX "unable to find a free IRQ\n"); - err = -EBUSY; - goto _err; - } - } - xdma1 = dma1[dev]; - if (xdma1 == SNDRV_AUTO_DMA) { - if ((xdma1 = snd_legacy_find_free_dma(possible_dmas)) < 0) { - snd_printk(KERN_ERR PFX "unable to find a free DMA\n"); - err = -EBUSY; - goto _err; - } - } - - if ((err = snd_sgalaxy_detect(dev, xirq, xdma1)) < 0) - goto _err; - - err = snd_wss_create(card, wssport[dev] + 4, -1, - xirq, xdma1, -1, - WSS_HW_DETECT, 0, &chip); - if (err < 0) - goto _err; - card->private_data = chip; - - err = snd_wss_pcm(chip, 0, NULL); - if (err < 0) { - snd_printdd(PFX "error creating new WSS PCM device\n"); - goto _err; - } - err = snd_wss_mixer(chip); - if (err < 0) { - snd_printdd(PFX "error creating new WSS mixer\n"); - goto _err; - } - if ((err = snd_sgalaxy_mixer(chip)) < 0) { - snd_printdd(PFX "the mixer rewrite failed\n"); - goto _err; - } - - strcpy(card->driver, "Sound Galaxy"); - strcpy(card->shortname, "Sound Galaxy"); - sprintf(card->longname, "Sound Galaxy at 0x%lx, irq %d, dma %d", - wssport[dev], xirq, xdma1); - - snd_card_set_dev(card, devptr); - - if ((err = snd_card_register(card)) < 0) - goto _err; - - dev_set_drvdata(devptr, card); - return 0; - - _err: - snd_card_free(card); - return err; -} - -static int __devexit snd_sgalaxy_remove(struct device *devptr, unsigned int dev) -{ - snd_card_free(dev_get_drvdata(devptr)); - dev_set_drvdata(devptr, NULL); - return 0; -} - -#ifdef CONFIG_PM -static int snd_sgalaxy_suspend(struct device *pdev, unsigned int n, - pm_message_t state) -{ - struct snd_card *card = dev_get_drvdata(pdev); - struct snd_wss *chip = card->private_data; - - snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - chip->suspend(chip); - return 0; -} - -static int snd_sgalaxy_resume(struct device *pdev, unsigned int n) -{ - struct snd_card *card = dev_get_drvdata(pdev); - struct snd_wss *chip = card->private_data; - - chip->resume(chip); - snd_wss_out(chip, SGALAXY_AUXC_LEFT, chip->image[SGALAXY_AUXC_LEFT]); - snd_wss_out(chip, SGALAXY_AUXC_RIGHT, chip->image[SGALAXY_AUXC_RIGHT]); - - snd_power_change_state(card, SNDRV_CTL_POWER_D0); - return 0; -} -#endif - -#define DEV_NAME "sgalaxy" - -static struct isa_driver snd_sgalaxy_driver = { - .match = snd_sgalaxy_match, - .probe = snd_sgalaxy_probe, - .remove = __devexit_p(snd_sgalaxy_remove), -#ifdef CONFIG_PM - .suspend = snd_sgalaxy_suspend, - .resume = snd_sgalaxy_resume, -#endif - .driver = { - .name = DEV_NAME - }, -}; - -static int __init alsa_card_sgalaxy_init(void) -{ - return isa_register_driver(&snd_sgalaxy_driver, SNDRV_CARDS); -} - -static void __exit alsa_card_sgalaxy_exit(void) -{ - isa_unregister_driver(&snd_sgalaxy_driver); -} - -module_init(alsa_card_sgalaxy_init) -module_exit(alsa_card_sgalaxy_exit) diff --git a/sound/oss/Kconfig b/sound/oss/Kconfig index a513651fa149..76c090218073 100644 --- a/sound/oss/Kconfig +++ b/sound/oss/Kconfig @@ -545,11 +545,3 @@ config SOUND_KAHLUA endif # SOUND_OSS -config SOUND_SH_DAC_AUDIO - tristate "SuperH DAC audio support" - depends on CPU_SH3 && HIGH_RES_TIMERS - -config SOUND_SH_DAC_AUDIO_CHANNEL - int "DAC channel" - default "1" - depends on SOUND_SH_DAC_AUDIO diff --git a/sound/oss/Makefile b/sound/oss/Makefile index 567b8a74178a..96f14dcd0cd1 100644 --- a/sound/oss/Makefile +++ b/sound/oss/Makefile @@ -9,7 +9,6 @@ obj-$(CONFIG_SOUND_OSS) += sound.o # Please leave it as is, cause the link order is significant ! -obj-$(CONFIG_SOUND_SH_DAC_AUDIO) += sh_dac_audio.o obj-$(CONFIG_SOUND_AEDSP16) += aedsp16.o obj-$(CONFIG_SOUND_PSS) += pss.o ad1848.o mpu401.o obj-$(CONFIG_SOUND_TRIX) += trix.o ad1848.o sb_lib.o uart401.o diff --git a/sound/oss/ad1848.c b/sound/oss/ad1848.c index 24793c5b65ac..4d2a6ae978f7 100644 --- a/sound/oss/ad1848.c +++ b/sound/oss/ad1848.c @@ -716,7 +716,7 @@ static int ad1848_mixer_ioctl(int dev, unsigned int cmd, void __user *arg) default: if (get_user(val, (int __user *)arg)) - return -EFAULT; + return -EFAULT; val = ad1848_mixer_set(devc, cmd & 0xff, val); break; } diff --git a/sound/oss/au1550_ac97.c b/sound/oss/au1550_ac97.c index c1070e33b32f..a8f626d99c5b 100644 --- a/sound/oss/au1550_ac97.c +++ b/sound/oss/au1550_ac97.c @@ -49,7 +49,6 @@ #include <linux/poll.h> #include <linux/bitops.h> #include <linux/spinlock.h> -#include <linux/smp_lock.h> #include <linux/ac97_codec.h> #include <linux/mutex.h> @@ -77,6 +76,7 @@ /* Boot options * 0 = no VRA, 1 = use VRA if codec supports it */ +static DEFINE_MUTEX(au1550_ac97_mutex); static int vra = 1; module_param(vra, bool, 0); MODULE_PARM_DESC(vra, "if 1 use VRA if codec supports it"); @@ -162,25 +162,16 @@ ld2(unsigned int x) static void au1550_delay(int msec) { - unsigned long tmo; - signed long tmo2; - if (in_interrupt()) return; - tmo = jiffies + (msec * HZ) / 1000; - for (;;) { - tmo2 = tmo - jiffies; - if (tmo2 <= 0) - break; - schedule_timeout(tmo2); - } + schedule_timeout_uninterruptible(msecs_to_jiffies(msec)); } static u16 rdcodec(struct ac97_codec *codec, u8 addr) { - struct au1550_state *s = (struct au1550_state *)codec->private_data; + struct au1550_state *s = codec->private_data; unsigned long flags; u32 cmd, val; u16 data; @@ -248,7 +239,7 @@ rdcodec(struct ac97_codec *codec, u8 addr) static void wrcodec(struct ac97_codec *codec, u8 addr, u16 data) { - struct au1550_state *s = (struct au1550_state *)codec->private_data; + struct au1550_state *s = codec->private_data; unsigned long flags; u32 cmd, val; int i; @@ -807,7 +798,9 @@ au1550_llseek(struct file *file, loff_t offset, int origin) static int au1550_open_mixdev(struct inode *inode, struct file *file) { + mutex_lock(&au1550_ac97_mutex); file->private_data = &au1550_state; + mutex_unlock(&au1550_ac97_mutex); return 0; } @@ -824,22 +817,26 @@ mixdev_ioctl(struct ac97_codec *codec, unsigned int cmd, return codec->mixer_ioctl(codec, cmd, arg); } -static int -au1550_ioctl_mixdev(struct inode *inode, struct file *file, - unsigned int cmd, unsigned long arg) +static long +au1550_ioctl_mixdev(struct file *file, unsigned int cmd, unsigned long arg) { - struct au1550_state *s = (struct au1550_state *)file->private_data; + struct au1550_state *s = file->private_data; struct ac97_codec *codec = s->codec; + int ret; + + mutex_lock(&au1550_ac97_mutex); + ret = mixdev_ioctl(codec, cmd, arg); + mutex_unlock(&au1550_ac97_mutex); - return mixdev_ioctl(codec, cmd, arg); + return ret; } static /*const */ struct file_operations au1550_mixer_fops = { - owner:THIS_MODULE, - llseek:au1550_llseek, - ioctl:au1550_ioctl_mixdev, - open:au1550_open_mixdev, - release:au1550_release_mixdev, + .owner = THIS_MODULE, + .llseek = au1550_llseek, + .unlocked_ioctl = au1550_ioctl_mixdev, + .open = au1550_open_mixdev, + .release = au1550_release_mixdev, }; static int @@ -1034,7 +1031,7 @@ copy_dmabuf_user(struct dmabuf *db, char* userbuf, int count, int to_user) static ssize_t au1550_read(struct file *file, char *buffer, size_t count, loff_t *ppos) { - struct au1550_state *s = (struct au1550_state *)file->private_data; + struct au1550_state *s = file->private_data; struct dmabuf *db = &s->dma_adc; DECLARE_WAITQUEUE(wait, current); ssize_t ret; @@ -1114,7 +1111,7 @@ out2: static ssize_t au1550_write(struct file *file, const char *buffer, size_t count, loff_t * ppos) { - struct au1550_state *s = (struct au1550_state *)file->private_data; + struct au1550_state *s = file->private_data; struct dmabuf *db = &s->dma_dac; DECLARE_WAITQUEUE(wait, current); ssize_t ret = 0; @@ -1214,7 +1211,7 @@ out2: static unsigned int au1550_poll(struct file *file, struct poll_table_struct *wait) { - struct au1550_state *s = (struct au1550_state *)file->private_data; + struct au1550_state *s = file->private_data; unsigned long flags; unsigned int mask = 0; @@ -1253,12 +1250,12 @@ au1550_poll(struct file *file, struct poll_table_struct *wait) static int au1550_mmap(struct file *file, struct vm_area_struct *vma) { - struct au1550_state *s = (struct au1550_state *)file->private_data; + struct au1550_state *s = file->private_data; struct dmabuf *db; unsigned long size; int ret = 0; - lock_kernel(); + mutex_lock(&au1550_ac97_mutex); mutex_lock(&s->sem); if (vma->vm_flags & VM_WRITE) db = &s->dma_dac; @@ -1286,7 +1283,7 @@ au1550_mmap(struct file *file, struct vm_area_struct *vma) db->mapped = 1; out: mutex_unlock(&s->sem); - unlock_kernel(); + mutex_unlock(&au1550_ac97_mutex); return ret; } @@ -1343,10 +1340,9 @@ dma_count_done(struct dmabuf *db) static int -au1550_ioctl(struct inode *inode, struct file *file, unsigned int cmd, - unsigned long arg) +au1550_ioctl(struct file *file, unsigned int cmd, unsigned long arg) { - struct au1550_state *s = (struct au1550_state *)file->private_data; + struct au1550_state *s = file->private_data; unsigned long flags; audio_buf_info abinfo; count_info cinfo; @@ -1780,6 +1776,17 @@ au1550_ioctl(struct inode *inode, struct file *file, unsigned int cmd, return mixdev_ioctl(s->codec, cmd, arg); } +static long +au1550_unlocked_ioctl(struct file *file, unsigned int cmd, unsigned long arg) +{ + int ret; + + mutex_lock(&au1550_ac97_mutex); + ret = au1550_ioctl(file, cmd, arg); + mutex_unlock(&au1550_ac97_mutex); + + return ret; +} static int au1550_open(struct inode *inode, struct file *file) @@ -1797,21 +1804,22 @@ au1550_open(struct inode *inode, struct file *file) #endif file->private_data = s; + mutex_lock(&au1550_ac97_mutex); /* wait for device to become free */ mutex_lock(&s->open_mutex); while (s->open_mode & file->f_mode) { - if (file->f_flags & O_NONBLOCK) { - mutex_unlock(&s->open_mutex); - return -EBUSY; - } + ret = -EBUSY; + if (file->f_flags & O_NONBLOCK) + goto out; add_wait_queue(&s->open_wait, &wait); __set_current_state(TASK_INTERRUPTIBLE); mutex_unlock(&s->open_mutex); schedule(); remove_wait_queue(&s->open_wait, &wait); set_current_state(TASK_RUNNING); + ret = -ERESTARTSYS; if (signal_pending(current)) - return -ERESTARTSYS; + goto out2; mutex_lock(&s->open_mutex); } @@ -1840,30 +1848,34 @@ au1550_open(struct inode *inode, struct file *file) if (file->f_mode & FMODE_READ) { if ((ret = prog_dmabuf_adc(s))) - return ret; + goto out; } if (file->f_mode & FMODE_WRITE) { if ((ret = prog_dmabuf_dac(s))) - return ret; + goto out; } s->open_mode |= file->f_mode & (FMODE_READ | FMODE_WRITE); - mutex_unlock(&s->open_mutex); mutex_init(&s->sem); - return 0; + ret = 0; +out: + mutex_unlock(&s->open_mutex); +out2: + mutex_unlock(&au1550_ac97_mutex); + return ret; } static int au1550_release(struct inode *inode, struct file *file) { - struct au1550_state *s = (struct au1550_state *)file->private_data; + struct au1550_state *s = file->private_data; - lock_kernel(); + mutex_lock(&au1550_ac97_mutex); if (file->f_mode & FMODE_WRITE) { - unlock_kernel(); + mutex_unlock(&au1550_ac97_mutex); drain_dac(s, file->f_flags & O_NONBLOCK); - lock_kernel(); + mutex_lock(&au1550_ac97_mutex); } mutex_lock(&s->open_mutex); @@ -1880,20 +1892,20 @@ au1550_release(struct inode *inode, struct file *file) s->open_mode &= ((~file->f_mode) & (FMODE_READ|FMODE_WRITE)); mutex_unlock(&s->open_mutex); wake_up(&s->open_wait); - unlock_kernel(); + mutex_unlock(&au1550_ac97_mutex); return 0; } static /*const */ struct file_operations au1550_audio_fops = { - owner: THIS_MODULE, - llseek: au1550_llseek, - read: au1550_read, - write: au1550_write, - poll: au1550_poll, - ioctl: au1550_ioctl, - mmap: au1550_mmap, - open: au1550_open, - release: au1550_release, + .owner = THIS_MODULE, + .llseek = au1550_llseek, + .read = au1550_read, + .write = au1550_write, + .poll = au1550_poll, + .unlocked_ioctl = au1550_unlocked_ioctl, + .mmap = au1550_mmap, + .open = au1550_open, + .release = au1550_release, }; MODULE_AUTHOR("Advanced Micro Devices (AMD), dan@embeddededge.com"); diff --git a/sound/oss/dmasound/dmasound_core.c b/sound/oss/dmasound/dmasound_core.c index 3f3c3f71db4b..87e2c72651f5 100644 --- a/sound/oss/dmasound/dmasound_core.c +++ b/sound/oss/dmasound/dmasound_core.c @@ -181,7 +181,7 @@ #include <linux/init.h> #include <linux/soundcard.h> #include <linux/poll.h> -#include <linux/smp_lock.h> +#include <linux/mutex.h> #include <asm/uaccess.h> @@ -194,6 +194,7 @@ * Declarations */ +static DEFINE_MUTEX(dmasound_core_mutex); int dmasound_catchRadius = 0; module_param(dmasound_catchRadius, int, 0); @@ -323,22 +324,26 @@ static struct { static int mixer_open(struct inode *inode, struct file *file) { - if (!try_module_get(dmasound.mach.owner)) + mutex_lock(&dmasound_core_mutex); + if (!try_module_get(dmasound.mach.owner)) { + mutex_unlock(&dmasound_core_mutex); return -ENODEV; + } mixer.busy = 1; + mutex_unlock(&dmasound_core_mutex); return 0; } static int mixer_release(struct inode *inode, struct file *file) { - lock_kernel(); + mutex_lock(&dmasound_core_mutex); mixer.busy = 0; module_put(dmasound.mach.owner); - unlock_kernel(); + mutex_unlock(&dmasound_core_mutex); return 0; } -static int mixer_ioctl(struct inode *inode, struct file *file, u_int cmd, - u_long arg) + +static int mixer_ioctl(struct file *file, u_int cmd, u_long arg) { if (_SIOC_DIR(cmd) & _SIOC_WRITE) mixer.modify_counter++; @@ -362,11 +367,22 @@ static int mixer_ioctl(struct inode *inode, struct file *file, u_int cmd, return -EINVAL; } +static long mixer_unlocked_ioctl(struct file *file, u_int cmd, u_long arg) +{ + int ret; + + mutex_lock(&dmasound_core_mutex); + ret = mixer_ioctl(file, cmd, arg); + mutex_unlock(&dmasound_core_mutex); + + return ret; +} + static const struct file_operations mixer_fops = { .owner = THIS_MODULE, .llseek = no_llseek, - .ioctl = mixer_ioctl, + .unlocked_ioctl = mixer_unlocked_ioctl, .open = mixer_open, .release = mixer_release, }; @@ -737,8 +753,11 @@ static int sq_open(struct inode *inode, struct file *file) { int rc; - if (!try_module_get(dmasound.mach.owner)) + mutex_lock(&dmasound_core_mutex); + if (!try_module_get(dmasound.mach.owner)) { + mutex_unlock(&dmasound_core_mutex); return -ENODEV; + } rc = write_sq_open(file); /* checks the f_mode */ if (rc) @@ -781,10 +800,11 @@ static int sq_open(struct inode *inode, struct file *file) sound_set_format(AFMT_MU_LAW); } #endif - + mutex_unlock(&dmasound_core_mutex); return 0; out: module_put(dmasound.mach.owner); + mutex_unlock(&dmasound_core_mutex); return rc; } @@ -850,7 +870,7 @@ static int sq_release(struct inode *inode, struct file *file) { int rc = 0; - lock_kernel(); + mutex_lock(&dmasound_core_mutex); if (file->f_mode & FMODE_WRITE) { if (write_sq.busy) @@ -881,7 +901,7 @@ static int sq_release(struct inode *inode, struct file *file) write_sq_wake_up(file); /* checks f_mode */ #endif /* blocking open() */ - unlock_kernel(); + mutex_unlock(&dmasound_core_mutex); return rc; } @@ -955,8 +975,7 @@ printk("dmasound_core: tried to set_queue_frags on a locked queue\n") ; return 0 ; } -static int sq_ioctl(struct inode *inode, struct file *file, u_int cmd, - u_long arg) +static int sq_ioctl(struct file *file, u_int cmd, u_long arg) { int val, result; u_long fmt; @@ -1114,18 +1133,29 @@ static int sq_ioctl(struct inode *inode, struct file *file, u_int cmd, return IOCTL_OUT(arg,val); default: - return mixer_ioctl(inode, file, cmd, arg); + return mixer_ioctl(file, cmd, arg); } return -EINVAL; } +static long sq_unlocked_ioctl(struct file *file, u_int cmd, u_long arg) +{ + int ret; + + mutex_lock(&dmasound_core_mutex); + ret = sq_ioctl(file, cmd, arg); + mutex_unlock(&dmasound_core_mutex); + + return ret; +} + static const struct file_operations sq_fops = { .owner = THIS_MODULE, .llseek = no_llseek, .write = sq_write, .poll = sq_poll, - .ioctl = sq_ioctl, + .unlocked_ioctl = sq_unlocked_ioctl, .open = sq_open, .release = sq_release, }; @@ -1226,12 +1256,17 @@ static int state_open(struct inode *inode, struct file *file) { char *buffer = state.buf; int len = 0; + int ret; + mutex_lock(&dmasound_core_mutex); + ret = -EBUSY; if (state.busy) - return -EBUSY; + goto out; + ret = -ENODEV; if (!try_module_get(dmasound.mach.owner)) - return -ENODEV; + goto out; + state.ptr = 0; state.busy = 1; @@ -1293,15 +1328,18 @@ printk("dmasound: stat buffer used %d bytes\n", len) ; printk(KERN_ERR "dmasound_core: stat buffer overflowed!\n"); state.len = len; - return 0; + ret = 0; +out: + mutex_unlock(&dmasound_core_mutex); + return ret; } static int state_release(struct inode *inode, struct file *file) { - lock_kernel(); + mutex_lock(&dmasound_core_mutex); state.busy = 0; module_put(dmasound.mach.owner); - unlock_kernel(); + mutex_unlock(&dmasound_core_mutex); return 0; } diff --git a/sound/oss/midi_synth.c b/sound/oss/midi_synth.c index 3bc7104c5379..3c09374ea5bf 100644 --- a/sound/oss/midi_synth.c +++ b/sound/oss/midi_synth.c @@ -523,7 +523,9 @@ midi_synth_load_patch(int dev, int format, const char __user *addr, { unsigned char data; - get_user(*(unsigned char *) &data, (unsigned char __user *) &((addr)[hdr_size + i])); + if (get_user(data, + (unsigned char __user *)(addr + hdr_size + i))) + return -EFAULT; eox_seen = (i > 0 && data & 0x80); /* End of sysex */ diff --git a/sound/oss/msnd_pinnacle.c b/sound/oss/msnd_pinnacle.c index a1e3f9671bea..b4c1eb504c22 100644 --- a/sound/oss/msnd_pinnacle.c +++ b/sound/oss/msnd_pinnacle.c @@ -39,7 +39,7 @@ #include <linux/delay.h> #include <linux/init.h> #include <linux/interrupt.h> -#include <linux/smp_lock.h> +#include <linux/mutex.h> #include <linux/gfp.h> #include <asm/irq.h> #include <asm/io.h> @@ -79,6 +79,7 @@ dev.rec_sample_rate / \ dev.rec_channels) +static DEFINE_MUTEX(msnd_pinnacle_mutex); static multisound_dev_t dev; #ifndef HAVE_DSPCODEH @@ -639,21 +640,26 @@ static int mixer_ioctl(unsigned int cmd, unsigned long arg) return -EINVAL; } -static int dev_ioctl(struct inode *inode, struct file *file, unsigned int cmd, unsigned long arg) +static long dev_ioctl(struct file *file, unsigned int cmd, unsigned long arg) { - int minor = iminor(inode); + int minor = iminor(file->f_path.dentry->d_inode); + int ret; if (cmd == OSS_GETVERSION) { int sound_version = SOUND_VERSION; return put_user(sound_version, (int __user *)arg); } + ret = -EINVAL; + + mutex_lock(&msnd_pinnacle_mutex); if (minor == dev.dsp_minor) - return dsp_ioctl(file, cmd, arg); + ret = dsp_ioctl(file, cmd, arg); else if (minor == dev.mixer_minor) - return mixer_ioctl(cmd, arg); + ret = mixer_ioctl(cmd, arg); + mutex_unlock(&msnd_pinnacle_mutex); - return -EINVAL; + return ret; } static void dsp_write_flush(void) @@ -756,12 +762,15 @@ static int dev_open(struct inode *inode, struct file *file) int minor = iminor(inode); int err = 0; + mutex_lock(&msnd_pinnacle_mutex); if (minor == dev.dsp_minor) { if ((file->f_mode & FMODE_WRITE && test_bit(F_AUDIO_WRITE_INUSE, &dev.flags)) || (file->f_mode & FMODE_READ && - test_bit(F_AUDIO_READ_INUSE, &dev.flags))) - return -EBUSY; + test_bit(F_AUDIO_READ_INUSE, &dev.flags))) { + err = -EBUSY; + goto out; + } if ((err = dsp_open(file)) >= 0) { dev.nresets = 0; @@ -782,7 +791,8 @@ static int dev_open(struct inode *inode, struct file *file) /* nothing */ } else err = -EINVAL; - +out: + mutex_unlock(&msnd_pinnacle_mutex); return err; } @@ -791,14 +801,14 @@ static int dev_release(struct inode *inode, struct file *file) int minor = iminor(inode); int err = 0; - lock_kernel(); + mutex_lock(&msnd_pinnacle_mutex); if (minor == dev.dsp_minor) err = dsp_release(file); else if (minor == dev.mixer_minor) { /* nothing */ } else err = -EINVAL; - unlock_kernel(); + mutex_unlock(&msnd_pinnacle_mutex); return err; } @@ -1105,7 +1115,7 @@ static const struct file_operations dev_fileops = { .owner = THIS_MODULE, .read = dev_read, .write = dev_write, - .ioctl = dev_ioctl, + .unlocked_ioctl = dev_ioctl, .open = dev_open, .release = dev_release, }; @@ -1391,9 +1401,13 @@ static int __init attach_multisound(void) printk(KERN_ERR LOGNAME ": Couldn't grab IRQ %d\n", dev.irq); return err; } - request_region(dev.io, dev.numio, dev.name); + if (request_region(dev.io, dev.numio, dev.name) == NULL) { + free_irq(dev.irq, &dev); + return -EBUSY; + } - if ((err = dsp_full_reset()) < 0) { + err = dsp_full_reset(); + if (err < 0) { release_region(dev.io, dev.numio); free_irq(dev.irq, &dev); return err; diff --git a/sound/oss/sh_dac_audio.c b/sound/oss/sh_dac_audio.c deleted file mode 100644 index 4153752507e3..000000000000 --- a/sound/oss/sh_dac_audio.c +++ /dev/null @@ -1,309 +0,0 @@ -/* - * sound/oss/sh_dac_audio.c - * - * SH DAC based sound :( - * - * Copyright (C) 2004,2005 Andriy Skulysh - * - * This file is subject to the terms and conditions of the GNU General Public - * License. See the file "COPYING" in the main directory of this archive - * for more details. - */ -#include <linux/module.h> -#include <linux/init.h> -#include <linux/sched.h> -#include <linux/linkage.h> -#include <linux/slab.h> -#include <linux/fs.h> -#include <linux/sound.h> -#include <linux/soundcard.h> -#include <linux/interrupt.h> -#include <linux/hrtimer.h> -#include <asm/io.h> -#include <asm/uaccess.h> -#include <asm/irq.h> -#include <asm/delay.h> -#include <asm/clock.h> -#include <cpu/dac.h> -#include <asm/machvec.h> -#include <mach/hp6xx.h> -#include <asm/hd64461.h> - -#define MODNAME "sh_dac_audio" - -#define BUFFER_SIZE 48000 - -static int rate; -static int empty; -static char *data_buffer, *buffer_begin, *buffer_end; -static int in_use, device_major; -static struct hrtimer hrtimer; -static ktime_t wakeups_per_second; - -static void dac_audio_start_timer(void) -{ - hrtimer_start(&hrtimer, wakeups_per_second, HRTIMER_MODE_REL); -} - -static void dac_audio_stop_timer(void) -{ - hrtimer_cancel(&hrtimer); -} - -static void dac_audio_reset(void) -{ - dac_audio_stop_timer(); - buffer_begin = buffer_end = data_buffer; - empty = 1; -} - -static void dac_audio_sync(void) -{ - while (!empty) - schedule(); -} - -static void dac_audio_start(void) -{ - if (mach_is_hp6xx()) { - u16 v = __raw_readw(HD64461_GPADR); - v &= ~HD64461_GPADR_SPEAKER; - __raw_writew(v, HD64461_GPADR); - } - - sh_dac_enable(CONFIG_SOUND_SH_DAC_AUDIO_CHANNEL); -} -static void dac_audio_stop(void) -{ - dac_audio_stop_timer(); - - if (mach_is_hp6xx()) { - u16 v = __raw_readw(HD64461_GPADR); - v |= HD64461_GPADR_SPEAKER; - __raw_writew(v, HD64461_GPADR); - } - - sh_dac_output(0, CONFIG_SOUND_SH_DAC_AUDIO_CHANNEL); - sh_dac_disable(CONFIG_SOUND_SH_DAC_AUDIO_CHANNEL); -} - -static void dac_audio_set_rate(void) -{ - wakeups_per_second = ktime_set(0, 1000000000 / rate); -} - -static int dac_audio_ioctl(struct inode *inode, struct file *file, - unsigned int cmd, unsigned long arg) -{ - int val; - - switch (cmd) { - case OSS_GETVERSION: - return put_user(SOUND_VERSION, (int *)arg); - - case SNDCTL_DSP_SYNC: - dac_audio_sync(); - return 0; - - case SNDCTL_DSP_RESET: - dac_audio_reset(); - return 0; - - case SNDCTL_DSP_GETFMTS: - return put_user(AFMT_U8, (int *)arg); - - case SNDCTL_DSP_SETFMT: - return put_user(AFMT_U8, (int *)arg); - - case SNDCTL_DSP_NONBLOCK: - spin_lock(&file->f_lock); - file->f_flags |= O_NONBLOCK; - spin_unlock(&file->f_lock); - return 0; - - case SNDCTL_DSP_GETCAPS: - return 0; - - case SOUND_PCM_WRITE_RATE: - val = *(int *)arg; - if (val > 0) { - rate = val; - dac_audio_set_rate(); - } - return put_user(rate, (int *)arg); - - case SNDCTL_DSP_STEREO: - return put_user(0, (int *)arg); - - case SOUND_PCM_WRITE_CHANNELS: - return put_user(1, (int *)arg); - - case SNDCTL_DSP_SETDUPLEX: - return -EINVAL; - - case SNDCTL_DSP_PROFILE: - return -EINVAL; - - case SNDCTL_DSP_GETBLKSIZE: - return put_user(BUFFER_SIZE, (int *)arg); - - case SNDCTL_DSP_SETFRAGMENT: - return 0; - - default: - printk(KERN_ERR "sh_dac_audio: unimplemented ioctl=0x%x\n", - cmd); - return -EINVAL; - } - return -EINVAL; -} - -static ssize_t dac_audio_write(struct file *file, const char *buf, size_t count, - loff_t * ppos) -{ - int free; - int nbytes; - - if (!count) { - dac_audio_sync(); - return 0; - } - - free = buffer_begin - buffer_end; - - if (free < 0) - free += BUFFER_SIZE; - if ((free == 0) && (empty)) - free = BUFFER_SIZE; - if (count > free) - count = free; - if (buffer_begin > buffer_end) { - if (copy_from_user((void *)buffer_end, buf, count)) - return -EFAULT; - - buffer_end += count; - } else { - nbytes = data_buffer + BUFFER_SIZE - buffer_end; - if (nbytes > count) { - if (copy_from_user((void *)buffer_end, buf, count)) - return -EFAULT; - buffer_end += count; - } else { - if (copy_from_user((void *)buffer_end, buf, nbytes)) - return -EFAULT; - if (copy_from_user - ((void *)data_buffer, buf + nbytes, count - nbytes)) - return -EFAULT; - buffer_end = data_buffer + count - nbytes; - } - } - - if (empty) { - empty = 0; - dac_audio_start_timer(); - } - - return count; -} - -static ssize_t dac_audio_read(struct file *file, char *buf, size_t count, - loff_t * ppos) -{ - return -EINVAL; -} - -static int dac_audio_open(struct inode *inode, struct file *file) -{ - if (file->f_mode & FMODE_READ) - return -ENODEV; - if (in_use) - return -EBUSY; - - in_use = 1; - - dac_audio_start(); - - return 0; -} - -static int dac_audio_release(struct inode *inode, struct file *file) -{ - dac_audio_sync(); - dac_audio_stop(); - in_use = 0; - - return 0; -} - -const struct file_operations dac_audio_fops = { - .read = dac_audio_read, - .write = dac_audio_write, - .ioctl = dac_audio_ioctl, - .open = dac_audio_open, - .release = dac_audio_release, -}; - -static enum hrtimer_restart sh_dac_audio_timer(struct hrtimer *handle) -{ - if (!empty) { - sh_dac_output(*buffer_begin, CONFIG_SOUND_SH_DAC_AUDIO_CHANNEL); - buffer_begin++; - - if (buffer_begin == data_buffer + BUFFER_SIZE) - buffer_begin = data_buffer; - if (buffer_begin == buffer_end) - empty = 1; - } - - if (!empty) - hrtimer_start(&hrtimer, wakeups_per_second, HRTIMER_MODE_REL); - - return HRTIMER_NORESTART; -} - -static int __init dac_audio_init(void) -{ - if ((device_major = register_sound_dsp(&dac_audio_fops, -1)) < 0) { - printk(KERN_ERR "Cannot register dsp device"); - return device_major; - } - - in_use = 0; - - data_buffer = kmalloc(BUFFER_SIZE, GFP_KERNEL); - if (data_buffer == NULL) - return -ENOMEM; - - dac_audio_reset(); - rate = 8000; - dac_audio_set_rate(); - - /* Today: High Resolution Timer driven DAC playback. - * The timer callback gets called once per sample. Ouch. - * - * Future: A much better approach would be to use the - * SH7720 CMT+DMAC+DAC hardware combination like this: - * - Program sample rate using CMT0 or CMT1 - * - Program DMAC to use CMT for timing and output to DAC - * - Play sound using DMAC, let CPU sleep. - * - While at it, rewrite this driver to use ALSA. - */ - - hrtimer_init(&hrtimer, CLOCK_MONOTONIC, HRTIMER_MODE_REL); - hrtimer.function = sh_dac_audio_timer; - - return 0; -} - -static void __exit dac_audio_exit(void) -{ - unregister_sound_dsp(device_major); - kfree((void *)data_buffer); -} - -module_init(dac_audio_init); -module_exit(dac_audio_exit); - -MODULE_AUTHOR("Andriy Skulysh, askulysh@image.kiev.ua"); -MODULE_DESCRIPTION("SH DAC sound driver"); -MODULE_LICENSE("GPL"); diff --git a/sound/oss/sound_timer.c b/sound/oss/sound_timer.c index f0f0c19fbff7..48cda6c4c257 100644 --- a/sound/oss/sound_timer.c +++ b/sound/oss/sound_timer.c @@ -26,7 +26,7 @@ static unsigned long prev_event_time; static volatile unsigned long usecs_per_tmr; /* Length of the current interval */ static struct sound_lowlev_timer *tmr; -static spinlock_t lock; +static DEFINE_SPINLOCK(lock); static unsigned long tmr2ticks(int tmr_value) { diff --git a/sound/oss/soundcard.c b/sound/oss/soundcard.c index 2d9c51312622..46c0d03dbecc 100644 --- a/sound/oss/soundcard.c +++ b/sound/oss/soundcard.c @@ -40,7 +40,7 @@ #include <linux/major.h> #include <linux/delay.h> #include <linux/proc_fs.h> -#include <linux/smp_lock.h> +#include <linux/mutex.h> #include <linux/module.h> #include <linux/mm.h> #include <linux/device.h> @@ -56,6 +56,7 @@ * Table for permanently allocated memory (used when unloading the module) */ void * sound_mem_blocks[MAX_MEM_BLOCKS]; +static DEFINE_MUTEX(soundcard_mutex); int sound_nblocks = 0; /* Persistent DMA buffers */ @@ -151,7 +152,7 @@ static ssize_t sound_read(struct file *file, char __user *buf, size_t count, lof * big one anyway, we might as well bandage here.. */ - lock_kernel(); + mutex_lock(&soundcard_mutex); DEB(printk("sound_read(dev=%d, count=%d)\n", dev, count)); switch (dev & 0x0f) { @@ -169,7 +170,7 @@ static ssize_t sound_read(struct file *file, char __user *buf, size_t count, lof case SND_DEV_MIDIN: ret = MIDIbuf_read(dev, file, buf, count); } - unlock_kernel(); + mutex_unlock(&soundcard_mutex); return ret; } @@ -178,7 +179,7 @@ static ssize_t sound_write(struct file *file, const char __user *buf, size_t cou int dev = iminor(file->f_path.dentry->d_inode); int ret = -EINVAL; - lock_kernel(); + mutex_lock(&soundcard_mutex); DEB(printk("sound_write(dev=%d, count=%d)\n", dev, count)); switch (dev & 0x0f) { case SND_DEV_SEQ: @@ -196,7 +197,7 @@ static ssize_t sound_write(struct file *file, const char __user *buf, size_t cou ret = MIDIbuf_write(dev, file, buf, count); break; } - unlock_kernel(); + mutex_unlock(&soundcard_mutex); return ret; } @@ -210,50 +211,52 @@ static int sound_open(struct inode *inode, struct file *file) printk(KERN_ERR "Invalid minor device %d\n", dev); return -ENXIO; } + mutex_lock(&soundcard_mutex); switch (dev & 0x0f) { case SND_DEV_CTL: dev >>= 4; if (dev >= 0 && dev < MAX_MIXER_DEV && mixer_devs[dev] == NULL) { request_module("mixer%d", dev); } + retval = -ENXIO; if (dev && (dev >= num_mixers || mixer_devs[dev] == NULL)) - return -ENXIO; + break; if (!try_module_get(mixer_devs[dev]->owner)) - return -ENXIO; + break; + + retval = 0; break; case SND_DEV_SEQ: case SND_DEV_SEQ2: - if ((retval = sequencer_open(dev, file)) < 0) - return retval; + retval = sequencer_open(dev, file); break; case SND_DEV_MIDIN: - if ((retval = MIDIbuf_open(dev, file)) < 0) - return retval; + retval = MIDIbuf_open(dev, file); break; case SND_DEV_DSP: case SND_DEV_DSP16: case SND_DEV_AUDIO: - if ((retval = audio_open(dev, file)) < 0) - return retval; + retval = audio_open(dev, file); break; default: printk(KERN_ERR "Invalid minor device %d\n", dev); - return -ENXIO; + retval = -ENXIO; } - return 0; + mutex_unlock(&soundcard_mutex); + return retval; } static int sound_release(struct inode *inode, struct file *file) { int dev = iminor(inode); - lock_kernel(); + mutex_lock(&soundcard_mutex); DEB(printk("sound_release(dev=%d)\n", dev)); switch (dev & 0x0f) { case SND_DEV_CTL: @@ -278,7 +281,7 @@ static int sound_release(struct inode *inode, struct file *file) default: printk(KERN_ERR "Sound error: Releasing unknown device 0x%02x\n", dev); } - unlock_kernel(); + mutex_unlock(&soundcard_mutex); return 0; } @@ -352,7 +355,7 @@ static long sound_ioctl(struct file *file, unsigned int cmd, unsigned long arg) if (cmd == OSS_GETVERSION) return __put_user(SOUND_VERSION, (int __user *)p); - lock_kernel(); + mutex_lock(&soundcard_mutex); if (_IOC_TYPE(cmd) == 'M' && num_mixers > 0 && /* Mixer ioctl */ (dev & 0x0f) != SND_DEV_CTL) { dtype = dev & 0x0f; @@ -367,7 +370,7 @@ static long sound_ioctl(struct file *file, unsigned int cmd, unsigned long arg) ret = sound_mixer_ioctl(dev >> 4, cmd, p); break; } - unlock_kernel(); + mutex_unlock(&soundcard_mutex); return ret; } @@ -389,15 +392,15 @@ static long sound_ioctl(struct file *file, unsigned int cmd, unsigned long arg) case SND_DEV_DSP: case SND_DEV_DSP16: case SND_DEV_AUDIO: - return audio_ioctl(dev, file, cmd, p); + ret = audio_ioctl(dev, file, cmd, p); break; case SND_DEV_MIDIN: - return MIDIbuf_ioctl(dev, file, cmd, p); + ret = MIDIbuf_ioctl(dev, file, cmd, p); break; } - unlock_kernel(); + mutex_unlock(&soundcard_mutex); return ret; } @@ -437,35 +440,35 @@ static int sound_mmap(struct file *file, struct vm_area_struct *vma) printk(KERN_ERR "Sound: mmap() not supported for other than audio devices\n"); return -EINVAL; } - lock_kernel(); + mutex_lock(&soundcard_mutex); if (vma->vm_flags & VM_WRITE) /* Map write and read/write to the output buf */ dmap = audio_devs[dev]->dmap_out; else if (vma->vm_flags & VM_READ) dmap = audio_devs[dev]->dmap_in; else { printk(KERN_ERR "Sound: Undefined mmap() access\n"); - unlock_kernel(); + mutex_unlock(&soundcard_mutex); return -EINVAL; } if (dmap == NULL) { printk(KERN_ERR "Sound: mmap() error. dmap == NULL\n"); - unlock_kernel(); + mutex_unlock(&soundcard_mutex); return -EIO; } if (dmap->raw_buf == NULL) { printk(KERN_ERR "Sound: mmap() called when raw_buf == NULL\n"); - unlock_kernel(); + mutex_unlock(&soundcard_mutex); return -EIO; } if (dmap->mapping_flags) { printk(KERN_ERR "Sound: mmap() called twice for the same DMA buffer\n"); - unlock_kernel(); + mutex_unlock(&soundcard_mutex); return -EIO; } if (vma->vm_pgoff != 0) { printk(KERN_ERR "Sound: mmap() offset must be 0.\n"); - unlock_kernel(); + mutex_unlock(&soundcard_mutex); return -EINVAL; } size = vma->vm_end - vma->vm_start; @@ -476,7 +479,7 @@ static int sound_mmap(struct file *file, struct vm_area_struct *vma) if (remap_pfn_range(vma, vma->vm_start, virt_to_phys(dmap->raw_buf) >> PAGE_SHIFT, vma->vm_end - vma->vm_start, vma->vm_page_prot)) { - unlock_kernel(); + mutex_unlock(&soundcard_mutex); return -EAGAIN; } @@ -488,7 +491,7 @@ static int sound_mmap(struct file *file, struct vm_area_struct *vma) memset(dmap->raw_buf, dmap->neutral_byte, dmap->bytes_in_use); - unlock_kernel(); + mutex_unlock(&soundcard_mutex); return 0; } diff --git a/sound/oss/swarm_cs4297a.c b/sound/oss/swarm_cs4297a.c index 3136c88eacdf..44357d877a27 100644 --- a/sound/oss/swarm_cs4297a.c +++ b/sound/oss/swarm_cs4297a.c @@ -93,6 +93,7 @@ struct cs4297a_state; +static DEFINE_MUTEX(swarm_cs4297a_mutex); static void stop_dac(struct cs4297a_state *s); static void stop_adc(struct cs4297a_state *s); static void start_dac(struct cs4297a_state *s); @@ -1534,6 +1535,7 @@ static int cs4297a_open_mixdev(struct inode *inode, struct file *file) CS_DBGOUT(CS_FUNCTION | CS_OPEN, 4, printk(KERN_INFO "cs4297a: cs4297a_open_mixdev()+\n")); + mutex_lock(&swarm_cs4297a_mutex); list_for_each(entry, &cs4297a_devs) { s = list_entry(entry, struct cs4297a_state, list); @@ -1544,6 +1546,8 @@ static int cs4297a_open_mixdev(struct inode *inode, struct file *file) { CS_DBGOUT(CS_FUNCTION | CS_OPEN | CS_ERROR, 2, printk(KERN_INFO "cs4297a: cs4297a_open_mixdev()- -ENODEV\n")); + + mutex_unlock(&swarm_cs4297a_mutex); return -ENODEV; } VALIDATE_STATE(s); @@ -1551,6 +1555,7 @@ static int cs4297a_open_mixdev(struct inode *inode, struct file *file) CS_DBGOUT(CS_FUNCTION | CS_OPEN, 4, printk(KERN_INFO "cs4297a: cs4297a_open_mixdev()- 0\n")); + mutex_unlock(&swarm_cs4297a_mutex); return nonseekable_open(inode, file); } @@ -1566,11 +1571,15 @@ static int cs4297a_release_mixdev(struct inode *inode, struct file *file) } -static int cs4297a_ioctl_mixdev(struct inode *inode, struct file *file, +static int cs4297a_ioctl_mixdev(struct file *file, unsigned int cmd, unsigned long arg) { - return mixer_ioctl((struct cs4297a_state *) file->private_data, cmd, + int ret; + mutex_lock(&swarm_cs4297a_mutex); + ret = mixer_ioctl((struct cs4297a_state *) file->private_data, cmd, arg); + mutex_unlock(&swarm_cs4297a_mutex); + return ret; } @@ -1580,7 +1589,7 @@ static int cs4297a_ioctl_mixdev(struct inode *inode, struct file *file, static const struct file_operations cs4297a_mixer_fops = { .owner = THIS_MODULE, .llseek = no_llseek, - .ioctl = cs4297a_ioctl_mixdev, + .unlocked_ioctl = cs4297a_ioctl_mixdev, .open = cs4297a_open_mixdev, .release = cs4297a_release_mixdev, }; @@ -1944,7 +1953,7 @@ static int cs4297a_mmap(struct file *file, struct vm_area_struct *vma) } -static int cs4297a_ioctl(struct inode *inode, struct file *file, +static int cs4297a_ioctl(struct file *file, unsigned int cmd, unsigned long arg) { struct cs4297a_state *s = @@ -2337,6 +2346,16 @@ static int cs4297a_ioctl(struct inode *inode, struct file *file, return mixer_ioctl(s, cmd, arg); } +static long cs4297a_unlocked_ioctl(struct file *file, u_int cmd, u_long arg) +{ + int ret; + + mutex_lock(&swarm_cs4297a_mutex); + ret = cs4297a_ioctl(file, cmd, arg); + mutex_unlock(&swarm_cs4297a_mutex); + + return ret; +} static int cs4297a_release(struct inode *inode, struct file *file) { @@ -2369,7 +2388,7 @@ static int cs4297a_release(struct inode *inode, struct file *file) return 0; } -static int cs4297a_open(struct inode *inode, struct file *file) +static int cs4297a_locked_open(struct inode *inode, struct file *file) { int minor = iminor(inode); struct cs4297a_state *s=NULL; @@ -2486,6 +2505,16 @@ static int cs4297a_open(struct inode *inode, struct file *file) return nonseekable_open(inode, file); } +static int cs4297a_open(struct inode *inode, struct file *file) +{ + int ret; + + mutex_lock(&swarm_cs4297a_mutex); + ret = cs4297a_open(inode, file); + mutex_unlock(&swarm_cs4297a_mutex); + + return ret; +} // ****************************************************************************************** // Wave (audio) file operations struct. @@ -2496,7 +2525,7 @@ static const struct file_operations cs4297a_audio_fops = { .read = cs4297a_read, .write = cs4297a_write, .poll = cs4297a_poll, - .ioctl = cs4297a_ioctl, + .unlocked_ioctl = cs4297a_unlocked_ioctl, .mmap = cs4297a_mmap, .open = cs4297a_open, .release = cs4297a_release, diff --git a/sound/oss/vidc.c b/sound/oss/vidc.c index ac39a531df19..f0e0caa53200 100644 --- a/sound/oss/vidc.c +++ b/sound/oss/vidc.c @@ -491,9 +491,6 @@ static void __init attach_vidc(struct address_info *hw_config) vidc_adev = adev; vidc_mixer_set(SOUND_MIXER_VOLUME, (85 | 85 << 8)); -#if defined(CONFIG_SOUND_SOFTOSS) || defined(CONFIG_SOUND_SOFTOSS_MODULE) - softoss_dev = adev; -#endif return; irq_failed: diff --git a/sound/oss/vwsnd.c b/sound/oss/vwsnd.c index 20b3b325aa80..643f1113b1d8 100644 --- a/sound/oss/vwsnd.c +++ b/sound/oss/vwsnd.c @@ -145,7 +145,6 @@ #include <linux/init.h> #include <linux/spinlock.h> -#include <linux/smp_lock.h> #include <linux/wait.h> #include <linux/interrupt.h> #include <linux/mutex.h> @@ -160,6 +159,7 @@ #ifdef VWSND_DEBUG +static DEFINE_MUTEX(vwsnd_mutex); static int shut_up = 1; /* @@ -2429,8 +2429,7 @@ static unsigned int vwsnd_audio_poll(struct file *file, return mask; } -static int vwsnd_audio_do_ioctl(struct inode *inode, - struct file *file, +static int vwsnd_audio_do_ioctl(struct file *file, unsigned int cmd, unsigned long arg) { @@ -2446,8 +2445,8 @@ static int vwsnd_audio_do_ioctl(struct inode *inode, int ival; - DBGEV("(inode=0x%p, file=0x%p, cmd=0x%x, arg=0x%lx)\n", - inode, file, cmd, arg); + DBGEV("(file=0x%p, cmd=0x%x, arg=0x%lx)\n", + file, cmd, arg); switch (cmd) { case OSS_GETVERSION: /* _SIOR ('M', 118, int) */ DBGX("OSS_GETVERSION\n"); @@ -2885,17 +2884,19 @@ static int vwsnd_audio_do_ioctl(struct inode *inode, return -EINVAL; } -static int vwsnd_audio_ioctl(struct inode *inode, - struct file *file, +static long vwsnd_audio_ioctl(struct file *file, unsigned int cmd, unsigned long arg) { vwsnd_dev_t *devc = (vwsnd_dev_t *) file->private_data; int ret; + mutex_lock(&vwsnd_mutex); mutex_lock(&devc->io_mutex); - ret = vwsnd_audio_do_ioctl(inode, file, cmd, arg); + ret = vwsnd_audio_do_ioctl(file, cmd, arg); mutex_unlock(&devc->io_mutex); + mutex_unlock(&vwsnd_mutex); + return ret; } @@ -2921,6 +2922,7 @@ static int vwsnd_audio_open(struct inode *inode, struct file *file) DBGE("(inode=0x%p, file=0x%p)\n", inode, file); + mutex_lock(&vwsnd_mutex); INC_USE_COUNT; for (devc = vwsnd_dev_list; devc; devc = devc->next_dev) if ((devc->audio_minor & ~0x0F) == (minor & ~0x0F)) @@ -2928,6 +2930,7 @@ static int vwsnd_audio_open(struct inode *inode, struct file *file) if (devc == NULL) { DEC_USE_COUNT; + mutex_unlock(&vwsnd_mutex); return -ENODEV; } @@ -2936,11 +2939,13 @@ static int vwsnd_audio_open(struct inode *inode, struct file *file) mutex_unlock(&devc->open_mutex); if (file->f_flags & O_NONBLOCK) { DEC_USE_COUNT; + mutex_unlock(&vwsnd_mutex); return -EBUSY; } interruptible_sleep_on(&devc->open_wait); if (signal_pending(current)) { DEC_USE_COUNT; + mutex_unlock(&vwsnd_mutex); return -ERESTARTSYS; } mutex_lock(&devc->open_mutex); @@ -2993,6 +2998,7 @@ static int vwsnd_audio_open(struct inode *inode, struct file *file) file->private_data = devc; DBGRV(); + mutex_unlock(&vwsnd_mutex); return 0; } @@ -3006,7 +3012,7 @@ static int vwsnd_audio_release(struct inode *inode, struct file *file) vwsnd_port_t *wport = NULL, *rport = NULL; int err = 0; - lock_kernel(); + mutex_lock(&vwsnd_mutex); mutex_lock(&devc->io_mutex); { DBGEV("(inode=0x%p, file=0x%p)\n", inode, file); @@ -3034,7 +3040,7 @@ static int vwsnd_audio_release(struct inode *inode, struct file *file) wake_up(&devc->open_wait); DEC_USE_COUNT; DBGR(); - unlock_kernel(); + mutex_unlock(&vwsnd_mutex); return err; } @@ -3044,7 +3050,7 @@ static const struct file_operations vwsnd_audio_fops = { .read = vwsnd_audio_read, .write = vwsnd_audio_write, .poll = vwsnd_audio_poll, - .ioctl = vwsnd_audio_ioctl, + .unlocked_ioctl = vwsnd_audio_ioctl, .mmap = vwsnd_audio_mmap, .open = vwsnd_audio_open, .release = vwsnd_audio_release, @@ -3062,15 +3068,18 @@ static int vwsnd_mixer_open(struct inode *inode, struct file *file) DBGEV("(inode=0x%p, file=0x%p)\n", inode, file); INC_USE_COUNT; + mutex_lock(&vwsnd_mutex); for (devc = vwsnd_dev_list; devc; devc = devc->next_dev) if (devc->mixer_minor == iminor(inode)) break; if (devc == NULL) { DEC_USE_COUNT; + mutex_unlock(&vwsnd_mutex); return -ENODEV; } file->private_data = devc; + mutex_unlock(&vwsnd_mutex); return 0; } @@ -3203,8 +3212,7 @@ static int mixer_write_ioctl(vwsnd_dev_t *devc, unsigned int nr, void __user *ar /* This is the ioctl entry to the mixer driver. */ -static int vwsnd_mixer_ioctl(struct inode *ioctl, - struct file *file, +static long vwsnd_mixer_ioctl(struct file *file, unsigned int cmd, unsigned long arg) { @@ -3215,6 +3223,7 @@ static int vwsnd_mixer_ioctl(struct inode *ioctl, DBGEV("(devc=0x%p, cmd=0x%x, arg=0x%lx)\n", devc, cmd, arg); + mutex_lock(&vwsnd_mutex); mutex_lock(&devc->mix_mutex); { if ((cmd & ~nrmask) == MIXER_READ(0)) @@ -3225,13 +3234,14 @@ static int vwsnd_mixer_ioctl(struct inode *ioctl, retval = -EINVAL; } mutex_unlock(&devc->mix_mutex); + mutex_unlock(&vwsnd_mutex); return retval; } static const struct file_operations vwsnd_mixer_fops = { .owner = THIS_MODULE, .llseek = no_llseek, - .ioctl = vwsnd_mixer_ioctl, + .unlocked_ioctl = vwsnd_mixer_ioctl, .open = vwsnd_mixer_open, .release = vwsnd_mixer_release, }; diff --git a/sound/oss/waveartist.c b/sound/oss/waveartist.c index e688dde6bbde..52468742d9f2 100644 --- a/sound/oss/waveartist.c +++ b/sound/oss/waveartist.c @@ -184,14 +184,8 @@ waveartist_iack(wavnc_info *devc) static inline int waveartist_sleep(int timeout_ms) { - unsigned int timeout = timeout_ms * 10 * HZ / 100; - - do { - set_current_state(TASK_INTERRUPTIBLE); - timeout = schedule_timeout(timeout); - } while (timeout); - - return 0; + unsigned int timeout = msecs_to_jiffies(timeout_ms*100); + return schedule_timeout_interruptible(timeout); } static int diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index e7a8cd058efb..12e34653b8a8 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -207,12 +207,12 @@ config SND_CMIPCI config SND_OXYGEN_LIB tristate - select SND_PCM - select SND_MPU401_UART config SND_OXYGEN tristate "C-Media 8788 (Oxygen)" select SND_OXYGEN_LIB + select SND_PCM + select SND_MPU401_UART help Say Y here to include support for sound cards based on the C-Media CMI8788 (Oxygen HD Audio) chip: @@ -581,6 +581,8 @@ config SND_HDSPM config SND_HIFIER tristate "TempoTec HiFier Fantasia" select SND_OXYGEN_LIB + select SND_PCM + select SND_MPU401_UART help Say Y here to include support for the MediaTek/TempoTec HiFier Fantasia sound card. @@ -815,14 +817,17 @@ config SND_VIA82XX_MODEM will be called snd-via82xx-modem. config SND_VIRTUOSO - tristate "Asus Virtuoso 100/200 (Xonar)" + tristate "Asus Virtuoso 66/100/200 (Xonar)" select SND_OXYGEN_LIB + select SND_PCM + select SND_MPU401_UART + select SND_JACK if INPUT=y || INPUT=SND help Say Y here to include support for sound cards based on the - Asus AV100/AV200 chips, i.e., Xonar D1, DX, D2, D2X, + Asus AV66/AV100/AV200 chips, i.e., Xonar D1, DX, D2, D2X, DS, Essence ST (Deluxe), and Essence STX. - Support for the DS is experimental. - Support for the HDAV1.3 (Deluxe) is very experimental. + Support for the HDAV1.3 (Deluxe) is incomplete; for the + HDAV1.3 Slim and Xense, missing. To compile this driver as a module, choose M here: the module will be called snd-virtuoso. diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c index 6cf1de8042e8..0e247cb90ecc 100644 --- a/sound/pci/als4000.c +++ b/sound/pci/als4000.c @@ -763,9 +763,9 @@ static void snd_als4000_configure(struct snd_sb *chip) /* SPECS_PAGE: 39 */ for (i = ALS4K_GCR91_DMA0_ADDR; i <= ALS4K_GCR96_DMA3_MODE_COUNT; ++i) snd_als4k_gcr_write(chip, i, 0); - + /* enable burst mode to prevent dropouts during high PCI bus usage */ snd_als4k_gcr_write(chip, ALS4K_GCR99_DMA_EMULATION_CTRL, - snd_als4k_gcr_read(chip, ALS4K_GCR99_DMA_EMULATION_CTRL)); + (snd_als4k_gcr_read(chip, ALS4K_GCR99_DMA_EMULATION_CTRL) & ~0x07) | 0x04); spin_unlock_irq(&chip->reg_lock); } diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 1db586af4f9c..c80b0b863c54 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -460,6 +460,7 @@ static int snd_card_asihpi_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_card_asihpi *card = snd_pcm_substream_chip(substream); int err; u16 format; + int width; unsigned int bytes_per_sec; print_hwparams(params); @@ -512,9 +513,10 @@ static int snd_card_asihpi_pcm_hw_params(struct snd_pcm_substream *substream, dpcm->hpi_buffer_attached); } bytes_per_sec = params_rate(params) * params_channels(params); - bytes_per_sec *= snd_pcm_format_width(params_format(params)); + width = snd_pcm_format_width(params_format(params)); + bytes_per_sec *= width; bytes_per_sec /= 8; - if (bytes_per_sec <= 0) + if (width < 0 || bytes_per_sec == 0) return -EINVAL; dpcm->bytes_per_sec = bytes_per_sec; @@ -1383,7 +1385,7 @@ static char *asihpi_src_names[] = compile_time_assert( (ARRAY_SIZE(asihpi_src_names) == - (HPI_SOURCENODE_LAST_INDEX-HPI_SOURCENODE_BASE+1)), + (HPI_SOURCENODE_LAST_INDEX-HPI_SOURCENODE_NONE+1)), assert_src_names_size); #if ASI_STYLE_NAMES @@ -1414,7 +1416,7 @@ static char *asihpi_dst_names[] = compile_time_assert( (ARRAY_SIZE(asihpi_dst_names) == - (HPI_DESTNODE_LAST_INDEX-HPI_DESTNODE_BASE+1)), + (HPI_DESTNODE_LAST_INDEX-HPI_DESTNODE_NONE+1)), assert_dst_names_size); static inline int ctl_add(struct snd_card *card, struct snd_kcontrol_new *ctl, @@ -2171,7 +2173,7 @@ static int snd_asihpi_mux_info(struct snd_kcontrol *kcontrol, &src_node_type, &src_node_index); sprintf(uinfo->value.enumerated.name, "%s %d", - asihpi_src_names[src_node_type - HPI_SOURCENODE_BASE], + asihpi_src_names[src_node_type - HPI_SOURCENODE_NONE], src_node_index); return 0; } @@ -2603,8 +2605,8 @@ static int __devinit snd_card_asihpi_mixer_new(struct snd_card_asihpi *asihpi) } - hpi_ctl.src_node_type -= HPI_SOURCENODE_BASE; - hpi_ctl.dst_node_type -= HPI_DESTNODE_BASE; + hpi_ctl.src_node_type -= HPI_SOURCENODE_NONE; + hpi_ctl.dst_node_type -= HPI_DESTNODE_NONE; /* ASI50xx in SSX mode has multiple meters on the same node. Use subindex to create distinct ALSA controls diff --git a/sound/pci/asihpi/hpi.h b/sound/pci/asihpi/hpi.h index 0173bbe62b67..23399d02f666 100644 --- a/sound/pci/asihpi/hpi.h +++ b/sound/pci/asihpi/hpi.h @@ -50,7 +50,8 @@ i.e 3.05.02 is a development version #define HPI_VER_RELEASE(v) ((int)(v & 0xFF)) /* Use single digits for versions less that 10 to avoid octal. */ -#define HPI_VER HPI_VERSION_CONSTRUCTOR(4L, 3, 25) +#define HPI_VER HPI_VERSION_CONSTRUCTOR(4L, 4, 1) +#define HPI_VER_STRING "4.04.01" /* Library version as documented in hpi-api-versions.txt */ #define HPI_LIB_VER HPI_VERSION_CONSTRUCTOR(9, 0, 0) @@ -203,8 +204,6 @@ enum HPI_SOURCENODES { exists on a destination node can be searched for using a source node value of either 0, or HPI_SOURCENODE_NONE */ HPI_SOURCENODE_NONE = 100, - /** \deprecated Use HPI_SOURCENODE_NONE instead. */ - HPI_SOURCENODE_BASE = 100, /** Out Stream (Play) node. */ HPI_SOURCENODE_OSTREAM = 101, /** Line in node - could be analog, AES/EBU or network. */ @@ -235,8 +234,6 @@ enum HPI_DESTNODES { exists on a source node can be searched for using a destination node value of either 0, or HPI_DESTNODE_NONE */ HPI_DESTNODE_NONE = 200, - /** \deprecated Use HPI_DESTNODE_NONE instead. */ - HPI_DESTNODE_BASE = 200, /** In Stream (Record) node. */ HPI_DESTNODE_ISTREAM = 201, HPI_DESTNODE_LINEOUT = 202, /**< line out node. */ @@ -432,7 +429,18 @@ Property 2 - adapter can do stream grouping (supports SSX2) Property 1 - adapter can do samplerate conversion (MRX) Property 2 - adapter can do timestretch (TSX) */ - HPI_ADAPTER_PROPERTY_CAPS2 = 269 + HPI_ADAPTER_PROPERTY_CAPS2 = 269, + +/** Readonly adapter sync header connection count. +*/ + HPI_ADAPTER_PROPERTY_SYNC_HEADER_CONNECTIONS = 270, +/** Readonly supports SSX2 property. +Indicates the adapter supports SSX2 in some mode setting. The +return value is true (1) or false (0). If the current adapter +mode is MONO SSX2 is disabled, even though this property will +return true. +*/ + HPI_ADAPTER_PROPERTY_SUPPORTS_SSX2 = 271 }; /** Adapter mode commands @@ -813,8 +821,6 @@ enum HPI_SAMPLECLOCK_SOURCES { /** The sampleclock output is derived from its local samplerate generator. The local samplerate may be set using HPI_SampleClock_SetLocalRate(). */ HPI_SAMPLECLOCK_SOURCE_LOCAL = 1, -/** \deprecated Use HPI_SAMPLECLOCK_SOURCE_LOCAL instead */ - HPI_SAMPLECLOCK_SOURCE_ADAPTER = 1, /** The adapter is clocked from a dedicated AES/EBU SampleClock input.*/ HPI_SAMPLECLOCK_SOURCE_AESEBU_SYNC = 2, /** From external wordclock connector */ @@ -825,10 +831,6 @@ enum HPI_SAMPLECLOCK_SOURCES { HPI_SAMPLECLOCK_SOURCE_SMPTE = 5, /** One of the aesebu inputs */ HPI_SAMPLECLOCK_SOURCE_AESEBU_INPUT = 6, -/** \deprecated The first aesebu input with a valid signal -Superseded by separate Auto enable flag -*/ - HPI_SAMPLECLOCK_SOURCE_AESEBU_AUTO = 7, /** From a network interface e.g. Cobranet or Livewire at either 48 or 96kHz */ HPI_SAMPLECLOCK_SOURCE_NETWORK = 8, /** From previous adjacent module (ASI2416 only)*/ @@ -1015,8 +1017,6 @@ enum HPI_ERROR_CODES { HPI_ERROR_CONTROL_DISABLED = 404, /** I2C transaction failed due to a missing ACK. */ HPI_ERROR_CONTROL_I2C_MISSING_ACK = 405, - /** Control attribute is valid, but not supported by this hardware. */ - HPI_ERROR_UNSUPPORTED_CONTROL_ATTRIBUTE = 406, /** Control is busy, or coming out of reset and cannot be accessed at this time. */ HPI_ERROR_CONTROL_NOT_READY = 407, @@ -1827,13 +1827,41 @@ u16 hpi_parametricEQ__get_coeffs(const struct hpi_hsubsys *ph_subsys, Compressor Expander control *******************************/ -u16 hpi_compander_set(const struct hpi_hsubsys *ph_subsys, u32 h_control, - u16 attack, u16 decay, short ratio100, short threshold0_01dB, - short makeup_gain0_01dB); +u16 hpi_compander_set_enable(const struct hpi_hsubsys *ph_subsys, + u32 h_control, u32 on); + +u16 hpi_compander_get_enable(const struct hpi_hsubsys *ph_subsys, + u32 h_control, u32 *pon); + +u16 hpi_compander_set_makeup_gain(const struct hpi_hsubsys *ph_subsys, + u32 h_control, short makeup_gain0_01dB); + +u16 hpi_compander_get_makeup_gain(const struct hpi_hsubsys *ph_subsys, + u32 h_control, short *pn_makeup_gain0_01dB); + +u16 hpi_compander_set_attack_time_constant(const struct hpi_hsubsys + *ph_subsys, u32 h_control, u32 index, u32 attack); + +u16 hpi_compander_get_attack_time_constant(const struct hpi_hsubsys + *ph_subsys, u32 h_control, u32 index, u32 *pw_attack); + +u16 hpi_compander_set_decay_time_constant(const struct hpi_hsubsys *ph_subsys, + u32 h_control, u32 index, u32 decay); + +u16 hpi_compander_get_decay_time_constant(const struct hpi_hsubsys *ph_subsys, + u32 h_control, u32 index, u32 *pw_decay); + +u16 hpi_compander_set_threshold(const struct hpi_hsubsys *ph_subsys, + u32 h_control, u32 index, short threshold0_01dB); + +u16 hpi_compander_get_threshold(const struct hpi_hsubsys *ph_subsys, + u32 h_control, u32 index, short *pn_threshold0_01dB); + +u16 hpi_compander_set_ratio(const struct hpi_hsubsys *ph_subsys, + u32 h_control, u32 index, u32 ratio100); -u16 hpi_compander_get(const struct hpi_hsubsys *ph_subsys, u32 h_control, - u16 *pw_attack, u16 *pw_decay, short *pw_ratio100, - short *pn_threshold0_01dB, short *pn_makeup_gain0_01dB); +u16 hpi_compander_get_ratio(const struct hpi_hsubsys *ph_subsys, + u32 h_control, u32 index, u32 *pw_ratio100); /******************************* Cobranet HMI control diff --git a/sound/pci/asihpi/hpi6000.c b/sound/pci/asihpi/hpi6000.c index 12dab5e4892c..f7e374ec4414 100644 --- a/sound/pci/asihpi/hpi6000.c +++ b/sound/pci/asihpi/hpi6000.c @@ -687,6 +687,7 @@ static short hpi6000_adapter_boot_load_dsp(struct hpi_adapter_obj *pao, switch (pao->pci.subsys_device_id) { case 0x5100: case 0x5110: /* ASI5100 revB or higher with C6711D */ + case 0x5200: /* ASI5200 PC_ie version of ASI5100 */ case 0x6100: case 0x6200: boot_load_family = HPI_ADAPTER_FAMILY_ASI(0x6200); @@ -1133,6 +1134,12 @@ static short hpi6000_adapter_boot_load_dsp(struct hpi_adapter_obj *pao, subsys_device_id) == HPI_ADAPTER_FAMILY_ASI(0x5100)) mask = 0x00000000L; + /* ASI5200 uses AX6 code, */ + /* but has no PLD r/w register to test */ + if (HPI_ADAPTER_FAMILY_ASI(pao->pci. + subsys_device_id) == + HPI_ADAPTER_FAMILY_ASI(0x5200)) + mask = 0x00000000L; break; case HPI_ADAPTER_FAMILY_ASI(0x8800): /* ASI8800 has 16bit path to FPGA */ diff --git a/sound/pci/asihpi/hpi6205.c b/sound/pci/asihpi/hpi6205.c index 3b4413448226..22c5fc625533 100644 --- a/sound/pci/asihpi/hpi6205.c +++ b/sound/pci/asihpi/hpi6205.c @@ -941,8 +941,7 @@ static void outstream_host_buffer_free(struct hpi_adapter_obj *pao, } -static u32 outstream_get_space_available(struct hpi_hostbuffer_status - *status) +static u32 outstream_get_space_available(struct hpi_hostbuffer_status *status) { return status->size_in_bytes - (status->host_index - status->dSP_index); @@ -987,6 +986,10 @@ static void outstream_write(struct hpi_adapter_obj *pao, /* write it */ phm->function = HPI_OSTREAM_WRITE; hw_message(pao, phm, phr); + + if (phr->error) + return; + /* update status information that the DSP would typically * update (and will update next time the DSP * buffer update task reads data from the host BBM buffer) diff --git a/sound/pci/asihpi/hpi_internal.h b/sound/pci/asihpi/hpi_internal.h index fdd0ce02aa68..16f502d459de 100644 --- a/sound/pci/asihpi/hpi_internal.h +++ b/sound/pci/asihpi/hpi_internal.h @@ -104,9 +104,9 @@ typedef void hpi_handler_func(struct hpi_message *, struct hpi_response *); #define STR_ROLE_FIELD_MAX 255U struct hpi_entity_str { - uint16_t size; - uint8_t type; - uint8_t role; + u16 size; + u8 type; + u8 role; }; #if defined(_MSC_VER) @@ -119,11 +119,11 @@ struct hpi_entity { #if ! defined(HPI_OS_DSP_C6000) || (defined(HPI_OS_DSP_C6000) && (__TI_COMPILER_VERSION__ > 6000008)) /* DSP C6000 compiler v6.0.8 and lower do not support flexible array member */ - uint8_t value[]; + u8 value[]; #else /* NOTE! Using sizeof(struct hpi_entity) will give erroneous results */ #define HPI_INTERNAL_WARN_ABOUT_ENTITY_VALUE - uint8_t value[1]; + u8 value[1]; #endif }; @@ -142,12 +142,15 @@ enum HPI_BUSES { /******************************************* CONTROL ATTRIBUTES ****/ /* (in order of control type ID */ - /* This allows for 255 control types, 256 unique attributes each */ +/* This allows for 255 control types, 256 unique attributes each */ #define HPI_CTL_ATTR(ctl, ai) (HPI_CONTROL_##ctl * 0x100 + ai) /* Get the sub-index of the attribute for a control type */ #define HPI_CTL_ATTR_INDEX(i) (i&0xff) +/* Extract the control from the control attribute */ +#define HPI_CTL_ATTR_CONTROL(i) (i>>8) + /* Generic control attributes. */ /** Enable a control. @@ -311,8 +314,7 @@ Used for HPI_ChannelModeSet/Get() /* Microphone control attributes */ #define HPI_MICROPHONE_PHANTOM_POWER HPI_CTL_ATTR(MICROPHONE, 1) -/** Equalizer control attributes -*/ +/** Equalizer control attributes */ /** Used to get number of filters in an EQ. (Can't set) */ #define HPI_EQUALIZER_NUM_FILTERS HPI_CTL_ATTR(EQUALIZER, 1) /** Set/get the filter by type, freq, Q, gain */ @@ -320,13 +322,15 @@ Used for HPI_ChannelModeSet/Get() /** Get the biquad coefficients */ #define HPI_EQUALIZER_COEFFICIENTS HPI_CTL_ATTR(EQUALIZER, 3) -#define HPI_COMPANDER_PARAMS HPI_CTL_ATTR(COMPANDER, 1) +/* Note compander also uses HPI_GENERIC_ENABLE */ +#define HPI_COMPANDER_PARAMS HPI_CTL_ATTR(COMPANDER, 1) +#define HPI_COMPANDER_MAKEUPGAIN HPI_CTL_ATTR(COMPANDER, 2) +#define HPI_COMPANDER_THRESHOLD HPI_CTL_ATTR(COMPANDER, 3) +#define HPI_COMPANDER_RATIO HPI_CTL_ATTR(COMPANDER, 4) +#define HPI_COMPANDER_ATTACK HPI_CTL_ATTR(COMPANDER, 5) +#define HPI_COMPANDER_DECAY HPI_CTL_ATTR(COMPANDER, 6) -/* Cobranet control attributes. - MUST be distinct from all other control attributes. - This is so that host side processing can easily identify a Cobranet control - and apply additional host side operations (like copying data) as required. -*/ +/* Cobranet control attributes. */ #define HPI_COBRANET_SET HPI_CTL_ATTR(COBRANET, 1) #define HPI_COBRANET_GET HPI_CTL_ATTR(COBRANET, 2) #define HPI_COBRANET_SET_DATA HPI_CTL_ATTR(COBRANET, 3) @@ -1512,11 +1516,11 @@ struct hpi_control_cache_single { struct hpi_control_cache_info i; union { struct { /* volume */ - u16 an_log[2]; + short an_log[2]; } v; struct { /* peak meter */ - u16 an_log_peak[2]; - u16 an_logRMS[2]; + short an_log_peak[2]; + short an_logRMS[2]; } p; struct { /* channel mode */ u16 mode; @@ -1526,7 +1530,7 @@ struct hpi_control_cache_single { u16 source_node_index; } x; struct { /* level/trim */ - u16 an_log[2]; + short an_log[2]; } l; struct { /* tuner - partial caching. some attributes go to the DSP. */ diff --git a/sound/pci/asihpi/hpicmn.c b/sound/pci/asihpi/hpicmn.c index fcd64539d9ef..dda4f1c6f658 100644 --- a/sound/pci/asihpi/hpicmn.c +++ b/sound/pci/asihpi/hpicmn.c @@ -353,7 +353,12 @@ short hpi_check_control_cache(struct hpi_control_cache *p_cache, phr->u.c.param1 = pC->u.t.band; else if ((phm->u.c.attribute == HPI_TUNER_LEVEL) && (phm->u.c.param1 == HPI_TUNER_LEVEL_AVERAGE)) - phr->u.c.param1 = pC->u.t.level; + if (pC->u.t.level == HPI_ERROR_ILLEGAL_CACHE_VALUE) { + phr->u.c.param1 = 0; + phr->error = + HPI_ERROR_INVALID_CONTROL_ATTRIBUTE; + } else + phr->u.c.param1 = pC->u.t.level; else found = 0; break; @@ -397,7 +402,8 @@ short hpi_check_control_cache(struct hpi_control_cache *p_cache, if (pC->u.clk.source_index == HPI_ERROR_ILLEGAL_CACHE_VALUE) { phr->u.c.param1 = 0; - phr->error = HPI_ERROR_INVALID_OPERATION; + phr->error = + HPI_ERROR_INVALID_CONTROL_ATTRIBUTE; } else phr->u.c.param1 = pC->u.clk.source_index; } else if (phm->u.c.attribute == HPI_SAMPLECLOCK_SAMPLERATE) diff --git a/sound/pci/asihpi/hpidebug.c b/sound/pci/asihpi/hpidebug.c index 4cd85a401b34..949836ec913a 100644 --- a/sound/pci/asihpi/hpidebug.c +++ b/sound/pci/asihpi/hpidebug.c @@ -111,7 +111,7 @@ make_treenode_from_array(hpi_control_type_strings, HPI_CONTROL_TYPE_STRINGS) &hpi_profile_strings,\ &hpi_control_strings, \ &hpi_asyncevent_strings \ -}; +} make_treenode_from_array(hpi_function_strings, HPI_FUNCTION_STRINGS) compile_time_assert(HPI_OBJ_MAXINDEX == 14, obj_list_doesnt_match); diff --git a/sound/pci/asihpi/hpidebug.h b/sound/pci/asihpi/hpidebug.h index 44dccadcc25b..a2f0952a99f0 100644 --- a/sound/pci/asihpi/hpidebug.h +++ b/sound/pci/asihpi/hpidebug.h @@ -356,7 +356,7 @@ compile_time_assert((HPI_CONTROL_LAST_INDEX + 1 == 27), "HPI_SOURCENODE_ADAPTER" \ } -compile_time_assert((HPI_SOURCENODE_LAST_INDEX - HPI_SOURCENODE_BASE + 1) == +compile_time_assert((HPI_SOURCENODE_LAST_INDEX - HPI_SOURCENODE_NONE + 1) == (12), sourcenode_strings_match_defs); #define HPI_DESTNODE_STRINGS \ @@ -370,7 +370,7 @@ compile_time_assert((HPI_SOURCENODE_LAST_INDEX - HPI_SOURCENODE_BASE + 1) == "HPI_DESTNODE_COBRANET", \ "HPI_DESTNODE_ANALOG" \ } -compile_time_assert((HPI_DESTNODE_LAST_INDEX - HPI_DESTNODE_BASE + 1) == (8), +compile_time_assert((HPI_DESTNODE_LAST_INDEX - HPI_DESTNODE_NONE + 1) == (8), destnode_strings_match_defs); #define HPI_CONTROL_CHANNEL_MODE_STRINGS \ diff --git a/sound/pci/asihpi/hpifunc.c b/sound/pci/asihpi/hpifunc.c index 298eef3e20e9..1e92eb6dd509 100644 --- a/sound/pci/asihpi/hpifunc.c +++ b/sound/pci/asihpi/hpifunc.c @@ -96,8 +96,7 @@ void hpi_stream_response_to_legacy(struct hpi_stream_res *pSR) static struct hpi_hsubsys gh_subsys; -struct hpi_hsubsys *hpi_subsys_create(void - ) +struct hpi_hsubsys *hpi_subsys_create(void) { struct hpi_message hm; struct hpi_response hr; @@ -302,6 +301,7 @@ u16 hpi_adapter_set_mode_ex(const struct hpi_hsubsys *ph_subsys, { struct hpi_message hm; struct hpi_response hr; + hpi_init_message_response(&hm, &hr, HPI_OBJ_ADAPTER, HPI_ADAPTER_SET_MODE); hm.adapter_index = adapter_index; @@ -510,7 +510,7 @@ u16 hpi_adapter_debug_read(const struct hpi_hsubsys *ph_subsys, hm.adapter_index = adapter_index; hm.u.ax.debug_read.dsp_address = dsp_address; - if (*count_bytes > sizeof(hr.u.bytes)) + if (*count_bytes > (int)sizeof(hr.u.bytes)) *count_bytes = sizeof(hr.u.bytes); hm.u.ax.debug_read.count_bytes = *count_bytes; @@ -976,6 +976,7 @@ u16 hpi_outstream_ancillary_read(const struct hpi_hsubsys *ph_subsys, { struct hpi_message hm; struct hpi_response hr; + hpi_init_message_response(&hm, &hr, HPI_OBJ_OSTREAM, HPI_OSTREAM_ANC_READ); u32TOINDEXES(h_outstream, &hm.adapter_index, &hm.obj_index); @@ -1581,6 +1582,7 @@ u16 hpi_control_param_set(const struct hpi_hsubsys *ph_subsys, { struct hpi_message hm; struct hpi_response hr; + hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, HPI_CONTROL_SET_STATE); u32TOINDEXES(h_control, &hm.adapter_index, &hm.obj_index); @@ -1591,6 +1593,22 @@ u16 hpi_control_param_set(const struct hpi_hsubsys *ph_subsys, return hr.error; } +static u16 hpi_control_log_set2(u32 h_control, u16 attrib, short sv0, + short sv1) +{ + struct hpi_message hm; + struct hpi_response hr; + + hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, + HPI_CONTROL_SET_STATE); + u32TOINDEXES(h_control, &hm.adapter_index, &hm.obj_index); + hm.u.c.attribute = attrib; + hm.u.c.an_log_value[0] = sv0; + hm.u.c.an_log_value[1] = sv1; + hpi_send_recv(&hm, &hr); + return hr.error; +} + static u16 hpi_control_param_get(const struct hpi_hsubsys *ph_subsys, const u32 h_control, const u16 attrib, u32 param1, u32 param2, @@ -1598,6 +1616,7 @@ u16 hpi_control_param_get(const struct hpi_hsubsys *ph_subsys, { struct hpi_message hm; struct hpi_response hr; + hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, HPI_CONTROL_GET_STATE); u32TOINDEXES(h_control, &hm.adapter_index, &hm.obj_index); @@ -1605,8 +1624,8 @@ u16 hpi_control_param_get(const struct hpi_hsubsys *ph_subsys, hm.u.c.param1 = param1; hm.u.c.param2 = param2; hpi_send_recv(&hm, &hr); - if (pparam1) - *pparam1 = hr.u.c.param1; + + *pparam1 = hr.u.c.param1; if (pparam2) *pparam2 = hr.u.c.param2; @@ -1617,10 +1636,23 @@ u16 hpi_control_param_get(const struct hpi_hsubsys *ph_subsys, hpi_control_param_get(s, h, a, 0, 0, p1, NULL) #define hpi_control_param2_get(s, h, a, p1, p2) \ hpi_control_param_get(s, h, a, 0, 0, p1, p2) -#define hpi_control_ex_param1_get(s, h, a, p1) \ - hpi_control_ex_param_get(s, h, a, 0, 0, p1, NULL) -#define hpi_control_ex_param2_get(s, h, a, p1, p2) \ - hpi_control_ex_param_get(s, h, a, 0, 0, p1, p2) + +static u16 hpi_control_log_get2(const struct hpi_hsubsys *ph_subsys, + u32 h_control, u16 attrib, short *sv0, short *sv1) +{ + struct hpi_message hm; + struct hpi_response hr; + hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, + HPI_CONTROL_GET_STATE); + u32TOINDEXES(h_control, &hm.adapter_index, &hm.obj_index); + hm.u.c.attribute = attrib; + + hpi_send_recv(&hm, &hr); + *sv0 = hr.u.c.an_log_value[0]; + if (sv1) + *sv1 = hr.u.c.an_log_value[1]; + return hr.error; +} static u16 hpi_control_query(const struct hpi_hsubsys *ph_subsys, @@ -1629,6 +1661,7 @@ u16 hpi_control_query(const struct hpi_hsubsys *ph_subsys, { struct hpi_message hm; struct hpi_response hr; + hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, HPI_CONTROL_GET_INFO); u32TOINDEXES(h_control, &hm.adapter_index, &hm.obj_index); @@ -1643,9 +1676,8 @@ u16 hpi_control_query(const struct hpi_hsubsys *ph_subsys, return hr.error; } -static u16 hpi_control_get_string(const struct hpi_hsubsys *ph_subsys, - const u32 h_control, const u16 attribute, char *psz_string, - const u32 string_length) +static u16 hpi_control_get_string(const u32 h_control, const u16 attribute, + char *psz_string, const u32 string_length) { unsigned int sub_string_index = 0, j = 0; char c = 0; @@ -1916,6 +1948,7 @@ u16 hpi_cobranet_hmi_write(const struct hpi_hsubsys *ph_subsys, u32 h_control, { struct hpi_message hm; struct hpi_response hr; + hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROLEX, HPI_CONTROL_SET_STATE); u32TOINDEXES(h_control, &hm.adapter_index, &hm.obj_index); @@ -1941,6 +1974,7 @@ u16 hpi_cobranet_hmi_read(const struct hpi_hsubsys *ph_subsys, u32 h_control, { struct hpi_message hm; struct hpi_response hr; + hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROLEX, HPI_CONTROL_GET_STATE); u32TOINDEXES(h_control, &hm.adapter_index, &hm.obj_index); @@ -1980,6 +2014,7 @@ u16 hpi_cobranet_hmi_get_status(const struct hpi_hsubsys *ph_subsys, { struct hpi_message hm; struct hpi_response hr; + hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROLEX, HPI_CONTROL_GET_STATE); u32TOINDEXES(h_control, &hm.adapter_index, &hm.obj_index); @@ -2006,6 +2041,7 @@ u16 hpi_cobranet_getI_paddress(const struct hpi_hsubsys *ph_subsys, u32 byte_count; u32 iP; u16 error; + error = hpi_cobranet_hmi_read(ph_subsys, h_control, HPI_COBRANET_HMI_cobra_ip_mon_currentIP, 4, &byte_count, (u8 *)&iP); @@ -2082,6 +2118,7 @@ u16 hpi_cobranet_getMA_caddress(const struct hpi_hsubsys *ph_subsys, u32 byte_count; u16 error; u32 mAC; + error = hpi_cobranet_hmi_read(ph_subsys, h_control, HPI_COBRANET_HMI_cobra_if_phy_address, 4, &byte_count, (u8 *)&mAC); @@ -2103,53 +2140,111 @@ u16 hpi_cobranet_getMA_caddress(const struct hpi_hsubsys *ph_subsys, return error; } -u16 hpi_compander_set(const struct hpi_hsubsys *ph_subsys, u32 h_control, - u16 attack, u16 decay, short ratio100, short threshold0_01dB, - short makeup_gain0_01dB) +u16 hpi_compander_set_enable(const struct hpi_hsubsys *ph_subsys, + u32 h_control, u32 enable) +{ + return hpi_control_param_set(ph_subsys, h_control, HPI_GENERIC_ENABLE, + enable, 0); +} + +u16 hpi_compander_get_enable(const struct hpi_hsubsys *ph_subsys, + u32 h_control, u32 *enable) +{ + return hpi_control_param1_get(ph_subsys, h_control, + HPI_GENERIC_ENABLE, enable); +} + +u16 hpi_compander_set_makeup_gain(const struct hpi_hsubsys *ph_subsys, + u32 h_control, short makeup_gain0_01dB) +{ + return hpi_control_log_set2(h_control, HPI_COMPANDER_MAKEUPGAIN, + makeup_gain0_01dB, 0); +} + +u16 hpi_compander_get_makeup_gain(const struct hpi_hsubsys *ph_subsys, + u32 h_control, short *makeup_gain0_01dB) +{ + return hpi_control_log_get2(ph_subsys, h_control, + HPI_COMPANDER_MAKEUPGAIN, makeup_gain0_01dB, NULL); +} + +u16 hpi_compander_set_attack_time_constant(const struct hpi_hsubsys + *ph_subsys, u32 h_control, unsigned int index, u32 attack) +{ + return hpi_control_param_set(ph_subsys, h_control, + HPI_COMPANDER_ATTACK, attack, index); +} + +u16 hpi_compander_get_attack_time_constant(const struct hpi_hsubsys + *ph_subsys, u32 h_control, unsigned int index, u32 *attack) +{ + return hpi_control_param_get(ph_subsys, h_control, + HPI_COMPANDER_ATTACK, 0, index, attack, NULL); +} + +u16 hpi_compander_set_decay_time_constant(const struct hpi_hsubsys *ph_subsys, + u32 h_control, unsigned int index, u32 decay) +{ + return hpi_control_param_set(ph_subsys, h_control, + HPI_COMPANDER_DECAY, decay, index); +} + +u16 hpi_compander_get_decay_time_constant(const struct hpi_hsubsys *ph_subsys, + u32 h_control, unsigned int index, u32 *decay) +{ + return hpi_control_param_get(ph_subsys, h_control, + HPI_COMPANDER_DECAY, 0, index, decay, NULL); + +} + +u16 hpi_compander_set_threshold(const struct hpi_hsubsys *ph_subsys, + u32 h_control, unsigned int index, short threshold0_01dB) { struct hpi_message hm; struct hpi_response hr; + hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, HPI_CONTROL_SET_STATE); u32TOINDEXES(h_control, &hm.adapter_index, &hm.obj_index); - - hm.u.c.param1 = attack + ((u32)ratio100 << 16); - hm.u.c.param2 = (decay & 0xFFFFL); + hm.u.c.attribute = HPI_COMPANDER_THRESHOLD; + hm.u.c.param2 = index; hm.u.c.an_log_value[0] = threshold0_01dB; - hm.u.c.an_log_value[1] = makeup_gain0_01dB; - hm.u.c.attribute = HPI_COMPANDER_PARAMS; hpi_send_recv(&hm, &hr); return hr.error; } -u16 hpi_compander_get(const struct hpi_hsubsys *ph_subsys, u32 h_control, - u16 *pw_attack, u16 *pw_decay, short *pw_ratio100, - short *pn_threshold0_01dB, short *pn_makeup_gain0_01dB) +u16 hpi_compander_get_threshold(const struct hpi_hsubsys *ph_subsys, + u32 h_control, unsigned int index, short *threshold0_01dB) { struct hpi_message hm; struct hpi_response hr; + hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, HPI_CONTROL_GET_STATE); u32TOINDEXES(h_control, &hm.adapter_index, &hm.obj_index); - hm.u.c.attribute = HPI_COMPANDER_PARAMS; + hm.u.c.attribute = HPI_COMPANDER_THRESHOLD; + hm.u.c.param2 = index; hpi_send_recv(&hm, &hr); + *threshold0_01dB = hr.u.c.an_log_value[0]; - if (pw_attack) - *pw_attack = (short)(hr.u.c.param1 & 0xFFFF); - if (pw_decay) - *pw_decay = (short)(hr.u.c.param2 & 0xFFFF); - if (pw_ratio100) - *pw_ratio100 = (short)(hr.u.c.param1 >> 16); + return hr.error; +} - if (pn_threshold0_01dB) - *pn_threshold0_01dB = hr.u.c.an_log_value[0]; - if (pn_makeup_gain0_01dB) - *pn_makeup_gain0_01dB = hr.u.c.an_log_value[1]; +u16 hpi_compander_set_ratio(const struct hpi_hsubsys *ph_subsys, + u32 h_control, u32 index, u32 ratio100) +{ + return hpi_control_param_set(ph_subsys, h_control, + HPI_COMPANDER_RATIO, ratio100, index); +} - return hr.error; +u16 hpi_compander_get_ratio(const struct hpi_hsubsys *ph_subsys, + u32 h_control, u32 index, u32 *ratio100) +{ + return hpi_control_param_get(ph_subsys, h_control, + HPI_COMPANDER_RATIO, 0, index, ratio100, NULL); } u16 hpi_level_query_range(const struct hpi_hsubsys *ph_subsys, u32 h_control, @@ -2157,6 +2252,7 @@ u16 hpi_level_query_range(const struct hpi_hsubsys *ph_subsys, u32 h_control, { struct hpi_message hm; struct hpi_response hr; + hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, HPI_CONTROL_GET_STATE); u32TOINDEXES(h_control, &hm.adapter_index, &hm.obj_index); @@ -2181,37 +2277,16 @@ u16 hpi_level_set_gain(const struct hpi_hsubsys *ph_subsys, u32 h_control, short an_gain0_01dB[HPI_MAX_CHANNELS] ) { - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, - HPI_CONTROL_SET_STATE); - u32TOINDEXES(h_control, &hm.adapter_index, &hm.obj_index); - memcpy(hm.u.c.an_log_value, an_gain0_01dB, - sizeof(short) * HPI_MAX_CHANNELS); - hm.u.c.attribute = HPI_LEVEL_GAIN; - - hpi_send_recv(&hm, &hr); - - return hr.error; + return hpi_control_log_set2(h_control, HPI_LEVEL_GAIN, + an_gain0_01dB[0], an_gain0_01dB[1]); } u16 hpi_level_get_gain(const struct hpi_hsubsys *ph_subsys, u32 h_control, short an_gain0_01dB[HPI_MAX_CHANNELS] ) { - struct hpi_message hm; - struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, - HPI_CONTROL_GET_STATE); - u32TOINDEXES(h_control, &hm.adapter_index, &hm.obj_index); - hm.u.c.attribute = HPI_LEVEL_GAIN; - - hpi_send_recv(&hm, &hr); - - memcpy(an_gain0_01dB, hr.u.c.an_log_value, - sizeof(short) * HPI_MAX_CHANNELS); - return hr.error; + return hpi_control_log_get2(ph_subsys, h_control, HPI_LEVEL_GAIN, + &an_gain0_01dB[0], &an_gain0_01dB[1]); } u16 hpi_meter_query_channels(const struct hpi_hsubsys *ph_subsys, @@ -2413,6 +2488,7 @@ u16 hpi_parametricEQ__get_band(const struct hpi_hsubsys *ph_subsys, { struct hpi_message hm; struct hpi_response hr; + hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, HPI_CONTROL_GET_STATE); u32TOINDEXES(h_control, &hm.adapter_index, &hm.obj_index); @@ -2439,6 +2515,7 @@ u16 hpi_parametricEQ__set_band(const struct hpi_hsubsys *ph_subsys, { struct hpi_message hm; struct hpi_response hr; + hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, HPI_CONTROL_SET_STATE); u32TOINDEXES(h_control, &hm.adapter_index, &hm.obj_index); @@ -2460,6 +2537,7 @@ u16 hpi_parametricEQ__get_coeffs(const struct hpi_hsubsys *ph_subsys, { struct hpi_message hm; struct hpi_response hr; + hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, HPI_CONTROL_GET_STATE); u32TOINDEXES(h_control, &hm.adapter_index, &hm.obj_index); @@ -2623,8 +2701,8 @@ u16 hpi_tone_detector_get_frequency(const struct hpi_hsubsys *ph_subsys, u16 hpi_tone_detector_get_state(const struct hpi_hsubsys *ph_subsys, u32 h_control, u32 *state) { - return hpi_control_param_get(ph_subsys, h_control, - HPI_TONEDETECTOR_STATE, 0, 0, (u32 *)state, NULL); + return hpi_control_param1_get(ph_subsys, h_control, + HPI_TONEDETECTOR_STATE, state); } u16 hpi_tone_detector_set_enable(const struct hpi_hsubsys *ph_subsys, @@ -2637,8 +2715,8 @@ u16 hpi_tone_detector_set_enable(const struct hpi_hsubsys *ph_subsys, u16 hpi_tone_detector_get_enable(const struct hpi_hsubsys *ph_subsys, u32 h_control, u32 *enable) { - return hpi_control_param_get(ph_subsys, h_control, HPI_GENERIC_ENABLE, - 0, 0, (u32 *)enable, NULL); + return hpi_control_param1_get(ph_subsys, h_control, + HPI_GENERIC_ENABLE, enable); } u16 hpi_tone_detector_set_event_enable(const struct hpi_hsubsys *ph_subsys, @@ -2651,8 +2729,8 @@ u16 hpi_tone_detector_set_event_enable(const struct hpi_hsubsys *ph_subsys, u16 hpi_tone_detector_get_event_enable(const struct hpi_hsubsys *ph_subsys, u32 h_control, u32 *event_enable) { - return hpi_control_param_get(ph_subsys, h_control, - HPI_GENERIC_EVENT_ENABLE, 0, 0, (u32 *)event_enable, NULL); + return hpi_control_param1_get(ph_subsys, h_control, + HPI_GENERIC_EVENT_ENABLE, event_enable); } u16 hpi_tone_detector_set_threshold(const struct hpi_hsubsys *ph_subsys, @@ -2665,15 +2743,15 @@ u16 hpi_tone_detector_set_threshold(const struct hpi_hsubsys *ph_subsys, u16 hpi_tone_detector_get_threshold(const struct hpi_hsubsys *ph_subsys, u32 h_control, int *threshold) { - return hpi_control_param_get(ph_subsys, h_control, - HPI_TONEDETECTOR_THRESHOLD, 0, 0, (u32 *)threshold, NULL); + return hpi_control_param1_get(ph_subsys, h_control, + HPI_TONEDETECTOR_THRESHOLD, (u32 *)threshold); } u16 hpi_silence_detector_get_state(const struct hpi_hsubsys *ph_subsys, u32 h_control, u32 *state) { - return hpi_control_param_get(ph_subsys, h_control, - HPI_SILENCEDETECTOR_STATE, 0, 0, (u32 *)state, NULL); + return hpi_control_param1_get(ph_subsys, h_control, + HPI_SILENCEDETECTOR_STATE, state); } u16 hpi_silence_detector_set_enable(const struct hpi_hsubsys *ph_subsys, @@ -2686,50 +2764,50 @@ u16 hpi_silence_detector_set_enable(const struct hpi_hsubsys *ph_subsys, u16 hpi_silence_detector_get_enable(const struct hpi_hsubsys *ph_subsys, u32 h_control, u32 *enable) { - return hpi_control_param_get(ph_subsys, h_control, HPI_GENERIC_ENABLE, - 0, 0, (u32 *)enable, NULL); + return hpi_control_param1_get(ph_subsys, h_control, + HPI_GENERIC_ENABLE, enable); } u16 hpi_silence_detector_set_event_enable(const struct hpi_hsubsys *ph_subsys, u32 h_control, u32 event_enable) { return hpi_control_param_set(ph_subsys, h_control, - HPI_GENERIC_EVENT_ENABLE, (u32)event_enable, 0); + HPI_GENERIC_EVENT_ENABLE, event_enable, 0); } u16 hpi_silence_detector_get_event_enable(const struct hpi_hsubsys *ph_subsys, u32 h_control, u32 *event_enable) { - return hpi_control_param_get(ph_subsys, h_control, - HPI_GENERIC_EVENT_ENABLE, 0, 0, (u32 *)event_enable, NULL); + return hpi_control_param1_get(ph_subsys, h_control, + HPI_GENERIC_EVENT_ENABLE, event_enable); } u16 hpi_silence_detector_set_delay(const struct hpi_hsubsys *ph_subsys, u32 h_control, u32 delay) { return hpi_control_param_set(ph_subsys, h_control, - HPI_SILENCEDETECTOR_DELAY, (u32)delay, 0); + HPI_SILENCEDETECTOR_DELAY, delay, 0); } u16 hpi_silence_detector_get_delay(const struct hpi_hsubsys *ph_subsys, u32 h_control, u32 *delay) { - return hpi_control_param_get(ph_subsys, h_control, - HPI_SILENCEDETECTOR_DELAY, 0, 0, (u32 *)delay, NULL); + return hpi_control_param1_get(ph_subsys, h_control, + HPI_SILENCEDETECTOR_DELAY, delay); } u16 hpi_silence_detector_set_threshold(const struct hpi_hsubsys *ph_subsys, u32 h_control, int threshold) { return hpi_control_param_set(ph_subsys, h_control, - HPI_SILENCEDETECTOR_THRESHOLD, (u32)threshold, 0); + HPI_SILENCEDETECTOR_THRESHOLD, threshold, 0); } u16 hpi_silence_detector_get_threshold(const struct hpi_hsubsys *ph_subsys, u32 h_control, int *threshold) { - return hpi_control_param_get(ph_subsys, h_control, - HPI_SILENCEDETECTOR_THRESHOLD, 0, 0, (u32 *)threshold, NULL); + return hpi_control_param1_get(ph_subsys, h_control, + HPI_SILENCEDETECTOR_THRESHOLD, (u32 *)threshold); } u16 hpi_tuner_query_band(const struct hpi_hsubsys *ph_subsys, @@ -2822,6 +2900,7 @@ u16 hpi_tuner_getRF_level(const struct hpi_hsubsys *ph_subsys, u32 h_control, { struct hpi_message hm; struct hpi_response hr; + hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, HPI_CONTROL_GET_STATE); u32TOINDEXES(h_control, &hm.adapter_index, &hm.obj_index); @@ -2838,6 +2917,7 @@ u16 hpi_tuner_get_rawRF_level(const struct hpi_hsubsys *ph_subsys, { struct hpi_message hm; struct hpi_response hr; + hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, HPI_CONTROL_GET_STATE); u32TOINDEXES(h_control, &hm.adapter_index, &hm.obj_index); @@ -2894,14 +2974,14 @@ u16 hpi_tuner_get_program(const struct hpi_hsubsys *ph_subsys, u32 h_control, u16 hpi_tuner_get_hd_radio_dsp_version(const struct hpi_hsubsys *ph_subsys, u32 h_control, char *psz_dsp_version, const u32 string_size) { - return hpi_control_get_string(ph_subsys, h_control, + return hpi_control_get_string(h_control, HPI_TUNER_HDRADIO_DSP_VERSION, psz_dsp_version, string_size); } u16 hpi_tuner_get_hd_radio_sdk_version(const struct hpi_hsubsys *ph_subsys, u32 h_control, char *psz_sdk_version, const u32 string_size) { - return hpi_control_get_string(ph_subsys, h_control, + return hpi_control_get_string(h_control, HPI_TUNER_HDRADIO_SDK_VERSION, psz_sdk_version, string_size); } @@ -2942,15 +3022,15 @@ u16 hpi_tuner_get_mode(const struct hpi_hsubsys *ph_subsys, u32 h_control, u16 hpi_tuner_get_hd_radio_signal_quality(const struct hpi_hsubsys *ph_subsys, u32 h_control, u32 *pquality) { - return hpi_control_param_get(ph_subsys, h_control, - HPI_TUNER_HDRADIO_SIGNAL_QUALITY, 0, 0, pquality, NULL); + return hpi_control_param1_get(ph_subsys, h_control, + HPI_TUNER_HDRADIO_SIGNAL_QUALITY, pquality); } u16 hpi_tuner_get_hd_radio_signal_blend(const struct hpi_hsubsys *ph_subsys, u32 h_control, u32 *pblend) { - return hpi_control_param_get(ph_subsys, h_control, - HPI_TUNER_HDRADIO_BLEND, 0, 0, pblend, NULL); + return hpi_control_param1_get(ph_subsys, h_control, + HPI_TUNER_HDRADIO_BLEND, pblend); } u16 hpi_tuner_set_hd_radio_signal_blend(const struct hpi_hsubsys *ph_subsys, @@ -2965,6 +3045,7 @@ u16 hpi_tuner_getRDS(const struct hpi_hsubsys *ph_subsys, u32 h_control, { struct hpi_message hm; struct hpi_response hr; + hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, HPI_CONTROL_GET_STATE); u32TOINDEXES(h_control, &hm.adapter_index, &hm.obj_index); @@ -2981,43 +3062,43 @@ u16 hpi_tuner_getRDS(const struct hpi_hsubsys *ph_subsys, u32 h_control, u16 HPI_PAD__get_channel_name(const struct hpi_hsubsys *ph_subsys, u32 h_control, char *psz_string, const u32 data_length) { - return hpi_control_get_string(ph_subsys, h_control, - HPI_PAD_CHANNEL_NAME, psz_string, data_length); + return hpi_control_get_string(h_control, HPI_PAD_CHANNEL_NAME, + psz_string, data_length); } u16 HPI_PAD__get_artist(const struct hpi_hsubsys *ph_subsys, u32 h_control, char *psz_string, const u32 data_length) { - return hpi_control_get_string(ph_subsys, h_control, HPI_PAD_ARTIST, - psz_string, data_length); + return hpi_control_get_string(h_control, HPI_PAD_ARTIST, psz_string, + data_length); } u16 HPI_PAD__get_title(const struct hpi_hsubsys *ph_subsys, u32 h_control, char *psz_string, const u32 data_length) { - return hpi_control_get_string(ph_subsys, h_control, HPI_PAD_TITLE, - psz_string, data_length); + return hpi_control_get_string(h_control, HPI_PAD_TITLE, psz_string, + data_length); } u16 HPI_PAD__get_comment(const struct hpi_hsubsys *ph_subsys, u32 h_control, char *psz_string, const u32 data_length) { - return hpi_control_get_string(ph_subsys, h_control, HPI_PAD_COMMENT, - psz_string, data_length); + return hpi_control_get_string(h_control, HPI_PAD_COMMENT, psz_string, + data_length); } u16 HPI_PAD__get_program_type(const struct hpi_hsubsys *ph_subsys, u32 h_control, u32 *ppTY) { - return hpi_control_param_get(ph_subsys, h_control, - HPI_PAD_PROGRAM_TYPE, 0, 0, ppTY, NULL); + return hpi_control_param1_get(ph_subsys, h_control, + HPI_PAD_PROGRAM_TYPE, ppTY); } u16 HPI_PAD__get_rdsPI(const struct hpi_hsubsys *ph_subsys, u32 h_control, u32 *ppI) { - return hpi_control_param_get(ph_subsys, h_control, HPI_PAD_PROGRAM_ID, - 0, 0, ppI, NULL); + return hpi_control_param1_get(ph_subsys, h_control, + HPI_PAD_PROGRAM_ID, ppI); } u16 hpi_volume_query_channels(const struct hpi_hsubsys *ph_subsys, @@ -3031,36 +3112,16 @@ u16 hpi_volume_set_gain(const struct hpi_hsubsys *ph_subsys, u32 h_control, short an_log_gain[HPI_MAX_CHANNELS] ) { - struct hpi_message hm; - struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, - HPI_CONTROL_SET_STATE); - u32TOINDEXES(h_control, &hm.adapter_index, &hm.obj_index); - memcpy(hm.u.c.an_log_value, an_log_gain, - sizeof(short) * HPI_MAX_CHANNELS); - hm.u.c.attribute = HPI_VOLUME_GAIN; - - hpi_send_recv(&hm, &hr); - - return hr.error; + return hpi_control_log_set2(h_control, HPI_VOLUME_GAIN, + an_log_gain[0], an_log_gain[1]); } u16 hpi_volume_get_gain(const struct hpi_hsubsys *ph_subsys, u32 h_control, short an_log_gain[HPI_MAX_CHANNELS] ) { - struct hpi_message hm; - struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, - HPI_CONTROL_GET_STATE); - u32TOINDEXES(h_control, &hm.adapter_index, &hm.obj_index); - hm.u.c.attribute = HPI_VOLUME_GAIN; - - hpi_send_recv(&hm, &hr); - - memcpy(an_log_gain, hr.u.c.an_log_value, - sizeof(short) * HPI_MAX_CHANNELS); - return hr.error; + return hpi_control_log_get2(ph_subsys, h_control, HPI_VOLUME_GAIN, + &an_log_gain[0], &an_log_gain[1]); } u16 hpi_volume_query_range(const struct hpi_hsubsys *ph_subsys, u32 h_control, @@ -3068,6 +3129,7 @@ u16 hpi_volume_query_range(const struct hpi_hsubsys *ph_subsys, u32 h_control, { struct hpi_message hm; struct hpi_response hr; + hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, HPI_CONTROL_GET_STATE); u32TOINDEXES(h_control, &hm.adapter_index, &hm.obj_index); @@ -3094,6 +3156,7 @@ u16 hpi_volume_auto_fade_profile(const struct hpi_hsubsys *ph_subsys, { struct hpi_message hm; struct hpi_response hr; + hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, HPI_CONTROL_SET_STATE); u32TOINDEXES(h_control, &hm.adapter_index, &hm.obj_index); @@ -3170,43 +3233,42 @@ static size_t entity_type_to_size[LAST_ENTITY_TYPE] = { 6 * sizeof(char), }; -inline size_t hpi_entity_size(struct hpi_entity *entity_ptr) +static inline size_t hpi_entity_size(struct hpi_entity *entity_ptr) { return entity_ptr->header.size; } -inline size_t hpi_entity_header_size(struct hpi_entity *entity_ptr) +static inline size_t hpi_entity_header_size(struct hpi_entity *entity_ptr) { return sizeof(entity_ptr->header); } -inline size_t hpi_entity_value_size(struct hpi_entity *entity_ptr) +static inline size_t hpi_entity_value_size(struct hpi_entity *entity_ptr) { return hpi_entity_size(entity_ptr) - hpi_entity_header_size(entity_ptr); } -inline size_t hpi_entity_item_count(struct hpi_entity *entity_ptr) +static inline size_t hpi_entity_item_count(struct hpi_entity *entity_ptr) { return hpi_entity_value_size(entity_ptr) / entity_type_to_size[entity_ptr->header.type]; } -inline struct hpi_entity *hpi_entity_ptr_to_next(struct hpi_entity +static inline struct hpi_entity *hpi_entity_ptr_to_next(struct hpi_entity *entity_ptr) { - return (void *)(((uint8_t *) entity_ptr) + - hpi_entity_size(entity_ptr)); + return (void *)(((u8 *)entity_ptr) + hpi_entity_size(entity_ptr)); } -inline u16 hpi_entity_check_type(const enum e_entity_type t) +static inline u16 hpi_entity_check_type(const enum e_entity_type t) { if (t >= 0 && t < STR_TYPE_FIELD_MAX) return 0; return HPI_ERROR_ENTITY_TYPE_INVALID; } -inline u16 hpi_entity_check_role(const enum e_entity_role r) +static inline u16 hpi_entity_check_role(const enum e_entity_role r) { if (r >= 0 && r < STR_ROLE_FIELD_MAX) return 0; @@ -3624,6 +3686,7 @@ u16 hpi_async_event_wait(const struct hpi_hsubsys *ph_subsys, u32 h_async, u16 maximum_events, struct hpi_async_event *p_events, u16 *pw_number_returned) { + return 0; } diff --git a/sound/pci/asihpi/hpimsgx.c b/sound/pci/asihpi/hpimsgx.c index 2ee90dc3d897..f01ab964f602 100644 --- a/sound/pci/asihpi/hpimsgx.c +++ b/sound/pci/asihpi/hpimsgx.c @@ -741,7 +741,7 @@ static void HPIMSGX__reset(u16 adapter_index) hpi_init_response(&hr, HPI_OBJ_SUBSYSTEM, HPI_SUBSYS_FIND_ADAPTERS, 0); memcpy(&gRESP_HPI_SUBSYS_FIND_ADAPTERS, &hr, - sizeof(&gRESP_HPI_SUBSYS_FIND_ADAPTERS)); + sizeof(gRESP_HPI_SUBSYS_FIND_ADAPTERS)); for (adapter = 0; adapter < HPI_MAX_ADAPTERS; adapter++) { diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c index 7396ac54e99f..62895a719fcb 100644 --- a/sound/pci/asihpi/hpioctl.c +++ b/sound/pci/asihpi/hpioctl.c @@ -121,11 +121,17 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg) phpi_ioctl_data = (struct hpi_ioctl_linux __user *)arg; /* Read the message and response pointers from user space. */ - get_user(puhm, &phpi_ioctl_data->phm); - get_user(puhr, &phpi_ioctl_data->phr); + if (get_user(puhm, &phpi_ioctl_data->phm) || + get_user(puhr, &phpi_ioctl_data->phr)) { + err = -EFAULT; + goto out; + } /* Now read the message size and data from user space. */ - get_user(hm->h.size, (u16 __user *)puhm); + if (get_user(hm->h.size, (u16 __user *)puhm)) { + err = -EFAULT; + goto out; + } if (hm->h.size > sizeof(*hm)) hm->h.size = sizeof(*hm); @@ -138,7 +144,10 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg) goto out; } - get_user(res_max_size, (u16 __user *)puhr); + if (get_user(res_max_size, (u16 __user *)puhr)) { + err = -EFAULT; + goto out; + } /* printk(KERN_INFO "user response size %d\n", res_max_size); */ if (res_max_size < sizeof(struct hpi_response_header)) { HPI_DEBUG_LOG(WARNING, "small res size %d\n", res_max_size); @@ -464,9 +473,7 @@ void __init asihpi_init(void) memset(adapters, 0, sizeof(adapters)); - printk(KERN_INFO "ASIHPI driver %d.%02d.%02d\n", - HPI_VER_MAJOR(HPI_VER), HPI_VER_MINOR(HPI_VER), - HPI_VER_RELEASE(HPI_VER)); + printk(KERN_INFO "ASIHPI driver " HPI_VER_STRING "\n"); hpi_init_message_response(&hm, &hr, HPI_OBJ_SUBSYSTEM, HPI_SUBSYS_DRIVER_LOAD); diff --git a/sound/pci/au88x0/au88x0_mixer.c b/sound/pci/au88x0/au88x0_mixer.c index c92f493d341e..557c782ae4fc 100644 --- a/sound/pci/au88x0/au88x0_mixer.c +++ b/sound/pci/au88x0/au88x0_mixer.c @@ -23,7 +23,7 @@ static int __devinit snd_vortex_mixer(vortex_t * vortex) if ((err = snd_ac97_bus(vortex->card, 0, &ops, NULL, &pbus)) < 0) return err; memset(&ac97, 0, sizeof(ac97)); - // Intialize AC97 codec stuff. + // Initialize AC97 codec stuff. ac97.private_data = vortex; ac97.scaps = AC97_SCAP_NO_SPDIF; err = snd_ac97_mixer(pbus, &ac97, &vortex->codec); diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index 0a3d3d6e77b4..8e69620da20b 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -1002,29 +1002,27 @@ snd_ca0106_pcm_pointer_playback(struct snd_pcm_substream *substream) struct snd_ca0106 *emu = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; struct snd_ca0106_pcm *epcm = runtime->private_data; - snd_pcm_uframes_t ptr, ptr1, ptr2,ptr3,ptr4 = 0; + unsigned int ptr, prev_ptr; int channel = epcm->channel_id; + int timeout = 10; if (!epcm->running) return 0; - ptr3 = snd_ca0106_ptr_read(emu, PLAYBACK_LIST_PTR, channel); - ptr1 = snd_ca0106_ptr_read(emu, PLAYBACK_POINTER, channel); - ptr4 = snd_ca0106_ptr_read(emu, PLAYBACK_LIST_PTR, channel); - if (ptr3 != ptr4) ptr1 = snd_ca0106_ptr_read(emu, PLAYBACK_POINTER, channel); - ptr2 = bytes_to_frames(runtime, ptr1); - ptr2+= (ptr4 >> 3) * runtime->period_size; - ptr=ptr2; - if (ptr >= runtime->buffer_size) - ptr -= runtime->buffer_size; - /* - printk(KERN_DEBUG "ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, " - "buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n", - ptr1, ptr2, ptr, (int)runtime->buffer_size, - (int)runtime->period_size, (int)runtime->frame_bits, - (int)runtime->rate); - */ - return ptr; + prev_ptr = -1; + do { + ptr = snd_ca0106_ptr_read(emu, PLAYBACK_LIST_PTR, channel); + ptr = (ptr >> 3) * runtime->period_size; + ptr += bytes_to_frames(runtime, + snd_ca0106_ptr_read(emu, PLAYBACK_POINTER, channel)); + if (ptr >= runtime->buffer_size) + ptr -= runtime->buffer_size; + if (prev_ptr == ptr) + return ptr; + prev_ptr = ptr; + } while (--timeout); + snd_printk(KERN_WARNING "ca0106: unstable DMA pointer!\n"); + return 0; } /* pointer_capture callback */ diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 668a5ec04499..20763dd03fa0 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -2250,6 +2250,8 @@ static int snd_echo_resume(struct pci_dev *pci) DE_INIT(("resume start\n")); pci_restore_state(pci); commpage_bak = kmalloc(sizeof(struct echoaudio), GFP_KERNEL); + if (commpage_bak == NULL) + return -ENOMEM; commpage = chip->comm_page; memcpy(commpage_bak, commpage, sizeof(struct comm_page)); diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c index 4203782d7cb7..aff8387c45cf 100644 --- a/sound/pci/emu10k1/emu10k1.c +++ b/sound/pci/emu10k1/emu10k1.c @@ -52,6 +52,7 @@ static int max_synth_voices[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 64}; static int max_buffer_size[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 128}; static int enable_ir[SNDRV_CARDS]; static uint subsystem[SNDRV_CARDS]; /* Force card subsystem model */ +static uint delay_pcm_irq[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 2}; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for the EMU10K1 soundcard."); @@ -73,6 +74,8 @@ module_param_array(enable_ir, bool, NULL, 0444); MODULE_PARM_DESC(enable_ir, "Enable IR."); module_param_array(subsystem, uint, NULL, 0444); MODULE_PARM_DESC(subsystem, "Force card subsystem model."); +module_param_array(delay_pcm_irq, uint, NULL, 0444); +MODULE_PARM_DESC(delay_pcm_irq, "Delay PCM interrupt by specified number of samples (default 0)."); /* * Class 0401: 1102:0008 (rev 00) Subsystem: 1102:1001 -> Audigy2 Value Model:SB0400 */ @@ -127,6 +130,7 @@ static int __devinit snd_card_emu10k1_probe(struct pci_dev *pci, &emu)) < 0) goto error; card->private_data = emu; + emu->delay_pcm_irq = delay_pcm_irq[dev] & 0x1f; if ((err = snd_emu10k1_pcm(emu, 0, NULL)) < 0) goto error; if ((err = snd_emu10k1_pcm_mic(emu, 1, NULL)) < 0) diff --git a/sound/pci/emu10k1/emumpu401.c b/sound/pci/emu10k1/emumpu401.c index 8578c70c61f2..bab564824efe 100644 --- a/sound/pci/emu10k1/emumpu401.c +++ b/sound/pci/emu10k1/emumpu401.c @@ -321,7 +321,7 @@ static struct snd_rawmidi_ops snd_emu10k1_midi_input = static void snd_emu10k1_midi_free(struct snd_rawmidi *rmidi) { - struct snd_emu10k1_midi *midi = (struct snd_emu10k1_midi *)rmidi->private_data; + struct snd_emu10k1_midi *midi = rmidi->private_data; midi->interrupt = NULL; midi->rmidi = NULL; } diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index 55b83ef73c63..622bace148e3 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -332,7 +332,7 @@ static void snd_emu10k1_pcm_init_voice(struct snd_emu10k1 *emu, evoice->epcm->ccca_start_addr = start_addr + ccis; if (extra) { start_addr += ccis; - end_addr += ccis; + end_addr += ccis + emu->delay_pcm_irq; } if (stereo && !extra) { snd_emu10k1_ptr_write(emu, CPF, voice, CPF_STEREO_MASK); @@ -360,7 +360,9 @@ static void snd_emu10k1_pcm_init_voice(struct snd_emu10k1 *emu, /* Assumption that PT is already 0 so no harm overwriting */ snd_emu10k1_ptr_write(emu, PTRX, voice, (send_amount[0] << 8) | send_amount[1]); snd_emu10k1_ptr_write(emu, DSL, voice, end_addr | (send_amount[3] << 24)); - snd_emu10k1_ptr_write(emu, PSST, voice, start_addr | (send_amount[2] << 24)); + snd_emu10k1_ptr_write(emu, PSST, voice, + (start_addr + (extra ? emu->delay_pcm_irq : 0)) | + (send_amount[2] << 24)); if (emu->card_capabilities->emu_model) pitch_target = PITCH_48000; /* Disable interpolators on emu1010 card */ else @@ -732,6 +734,23 @@ static void snd_emu10k1_playback_stop_voice(struct snd_emu10k1 *emu, struct snd_ snd_emu10k1_ptr_write(emu, IP, voice, 0); } +static inline void snd_emu10k1_playback_mangle_extra(struct snd_emu10k1 *emu, + struct snd_emu10k1_pcm *epcm, + struct snd_pcm_substream *substream, + struct snd_pcm_runtime *runtime) +{ + unsigned int ptr, period_pos; + + /* try to sychronize the current position for the interrupt + source voice */ + period_pos = runtime->status->hw_ptr - runtime->hw_ptr_interrupt; + period_pos %= runtime->period_size; + ptr = snd_emu10k1_ptr_read(emu, CCCA, epcm->extra->number); + ptr &= ~0x00ffffff; + ptr |= epcm->ccca_start_addr + period_pos; + snd_emu10k1_ptr_write(emu, CCCA, epcm->extra->number, ptr); +} + static int snd_emu10k1_playback_trigger(struct snd_pcm_substream *substream, int cmd) { @@ -753,6 +772,8 @@ static int snd_emu10k1_playback_trigger(struct snd_pcm_substream *substream, /* follow thru */ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: + if (cmd == SNDRV_PCM_TRIGGER_PAUSE_RELEASE) + snd_emu10k1_playback_mangle_extra(emu, epcm, substream, runtime); mix = &emu->pcm_mixer[substream->number]; snd_emu10k1_playback_prepare_voice(emu, epcm->voices[0], 1, 0, mix); snd_emu10k1_playback_prepare_voice(emu, epcm->voices[1], 0, 0, mix); @@ -869,8 +890,9 @@ static snd_pcm_uframes_t snd_emu10k1_playback_pointer(struct snd_pcm_substream * #endif /* printk(KERN_DEBUG - "ptr = 0x%x, buffer_size = 0x%x, period_size = 0x%x\n", - ptr, runtime->buffer_size, runtime->period_size); + "ptr = 0x%lx, buffer_size = 0x%lx, period_size = 0x%lx\n", + (long)ptr, (long)runtime->buffer_size, + (long)runtime->period_size); */ return ptr; } diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index ffb1ddb8dc28..957a311514c8 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -310,8 +310,10 @@ snd_emu10k1_alloc_pages(struct snd_emu10k1 *emu, struct snd_pcm_substream *subst if (snd_BUG_ON(!hdr)) return NULL; + idx = runtime->period_size >= runtime->buffer_size ? + (emu->delay_pcm_irq * 2) : 0; mutex_lock(&hdr->block_mutex); - blk = search_empty(emu, runtime->dma_bytes); + blk = search_empty(emu, runtime->dma_bytes + idx); if (blk == NULL) { mutex_unlock(&hdr->block_mutex); return NULL; diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index ba2098d20ccc..14829210ef0b 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -396,15 +396,18 @@ int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid, } for (n = prev_nid + 1; n <= val; n++) { if (conns >= max_conns) { - snd_printk(KERN_ERR - "Too many connections\n"); + snd_printk(KERN_ERR "hda_codec: " + "Too many connections %d for NID 0x%x\n", + conns, nid); return -EINVAL; } conn_list[conns++] = n; } } else { if (conns >= max_conns) { - snd_printk(KERN_ERR "Too many connections\n"); + snd_printk(KERN_ERR "hda_codec: " + "Too many connections %d for NID 0x%x\n", + conns, nid); return -EINVAL; } conn_list[conns++] = val; @@ -586,6 +589,7 @@ int /*__devinit*/ snd_hda_bus_new(struct snd_card *card, bus->ops = temp->ops; mutex_init(&bus->cmd_mutex); + mutex_init(&bus->prepare_mutex); INIT_LIST_HEAD(&bus->codec_list); snprintf(bus->workq_name, sizeof(bus->workq_name), @@ -730,15 +734,17 @@ static void /*__devinit*/ setup_fg_nodes(struct hda_codec *codec) total_nodes = snd_hda_get_sub_nodes(codec, AC_NODE_ROOT, &nid); for (i = 0; i < total_nodes; i++, nid++) { function_id = snd_hda_param_read(codec, nid, - AC_PAR_FUNCTION_TYPE) & 0xff; - switch (function_id) { + AC_PAR_FUNCTION_TYPE); + switch (function_id & 0xff) { case AC_GRP_AUDIO_FUNCTION: codec->afg = nid; - codec->function_id = function_id; + codec->afg_function_id = function_id & 0xff; + codec->afg_unsol = (function_id >> 8) & 1; break; case AC_GRP_MODEM_FUNCTION: codec->mfg = nid; - codec->function_id = function_id; + codec->mfg_function_id = function_id & 0xff; + codec->mfg_unsol = (function_id >> 8) & 1; break; default: break; @@ -966,6 +972,36 @@ static void restore_init_pincfgs(struct hda_codec *codec) } /* + * audio-converter setup caches + */ +struct hda_cvt_setup { + hda_nid_t nid; + u8 stream_tag; + u8 channel_id; + u16 format_id; + unsigned char active; /* cvt is currently used */ + unsigned char dirty; /* setups should be cleared */ +}; + +/* get or create a cache entry for the given audio converter NID */ +static struct hda_cvt_setup * +get_hda_cvt_setup(struct hda_codec *codec, hda_nid_t nid) +{ + struct hda_cvt_setup *p; + int i; + + for (i = 0; i < codec->cvt_setups.used; i++) { + p = snd_array_elem(&codec->cvt_setups, i); + if (p->nid == nid) + return p; + } + p = snd_array_new(&codec->cvt_setups); + if (p) + p->nid = nid; + return p; +} + +/* * codec destructor */ static void snd_hda_codec_free(struct hda_codec *codec) @@ -1039,6 +1075,7 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, snd_array_init(&codec->nids, sizeof(struct hda_nid_item), 32); snd_array_init(&codec->init_pins, sizeof(struct hda_pincfg), 16); snd_array_init(&codec->driver_pins, sizeof(struct hda_pincfg), 16); + snd_array_init(&codec->cvt_setups, sizeof(struct hda_cvt_setup), 8); if (codec->bus->modelname) { codec->modelname = kstrdup(codec->bus->modelname, GFP_KERNEL); if (!codec->modelname) { @@ -1176,37 +1213,126 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, u32 stream_tag, int channel_id, int format) { + struct hda_codec *c; + struct hda_cvt_setup *p; + unsigned int oldval, newval; + int i; + if (!nid) return; snd_printdd("hda_codec_setup_stream: " "NID=0x%x, stream=0x%x, channel=%d, format=0x%x\n", nid, stream_tag, channel_id, format); - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CHANNEL_STREAMID, - (stream_tag << 4) | channel_id); - msleep(1); - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, format); + p = get_hda_cvt_setup(codec, nid); + if (!p) + return; + /* update the stream-id if changed */ + if (p->stream_tag != stream_tag || p->channel_id != channel_id) { + oldval = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0); + newval = (stream_tag << 4) | channel_id; + if (oldval != newval) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_CHANNEL_STREAMID, + newval); + p->stream_tag = stream_tag; + p->channel_id = channel_id; + } + /* update the format-id if changed */ + if (p->format_id != format) { + oldval = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_STREAM_FORMAT, 0); + if (oldval != format) { + msleep(1); + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_STREAM_FORMAT, + format); + } + p->format_id = format; + } + p->active = 1; + p->dirty = 0; + + /* make other inactive cvts with the same stream-tag dirty */ + list_for_each_entry(c, &codec->bus->codec_list, list) { + for (i = 0; i < c->cvt_setups.used; i++) { + p = snd_array_elem(&c->cvt_setups, i); + if (!p->active && p->stream_tag == stream_tag) + p->dirty = 1; + } + } } EXPORT_SYMBOL_HDA(snd_hda_codec_setup_stream); +static void really_cleanup_stream(struct hda_codec *codec, + struct hda_cvt_setup *q); + /** - * snd_hda_codec_cleanup_stream - clean up the codec for closing + * __snd_hda_codec_cleanup_stream - clean up the codec for closing * @codec: the CODEC to clean up * @nid: the NID to clean up + * @do_now: really clean up the stream instead of clearing the active flag */ -void snd_hda_codec_cleanup_stream(struct hda_codec *codec, hda_nid_t nid) +void __snd_hda_codec_cleanup_stream(struct hda_codec *codec, hda_nid_t nid, + int do_now) { + struct hda_cvt_setup *p; + if (!nid) return; snd_printdd("hda_codec_cleanup_stream: NID=0x%x\n", nid); + p = get_hda_cvt_setup(codec, nid); + if (p) { + /* here we just clear the active flag when do_now isn't set; + * actual clean-ups will be done later in + * purify_inactive_streams() called from snd_hda_codec_prpapre() + */ + if (do_now) + really_cleanup_stream(codec, p); + else + p->active = 0; + } +} +EXPORT_SYMBOL_HDA(__snd_hda_codec_cleanup_stream); + +static void really_cleanup_stream(struct hda_codec *codec, + struct hda_cvt_setup *q) +{ + hda_nid_t nid = q->nid; snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CHANNEL_STREAMID, 0); -#if 0 /* keep the format */ - msleep(1); snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, 0); -#endif + memset(q, 0, sizeof(*q)); + q->nid = nid; +} + +/* clean up the all conflicting obsolete streams */ +static void purify_inactive_streams(struct hda_codec *codec) +{ + struct hda_codec *c; + int i; + + list_for_each_entry(c, &codec->bus->codec_list, list) { + for (i = 0; i < c->cvt_setups.used; i++) { + struct hda_cvt_setup *p; + p = snd_array_elem(&c->cvt_setups, i); + if (p->dirty) + really_cleanup_stream(c, p); + } + } +} + +/* clean up all streams; called from suspend */ +static void hda_cleanup_all_streams(struct hda_codec *codec) +{ + int i; + + for (i = 0; i < codec->cvt_setups.used; i++) { + struct hda_cvt_setup *p = snd_array_elem(&codec->cvt_setups, i); + if (p->stream_tag) + really_cleanup_stream(codec, p); + } } -EXPORT_SYMBOL_HDA(snd_hda_codec_cleanup_stream); /* * amp access functions @@ -1565,6 +1691,17 @@ void snd_hda_codec_resume_amp(struct hda_codec *codec) EXPORT_SYMBOL_HDA(snd_hda_codec_resume_amp); #endif /* SND_HDA_NEEDS_RESUME */ +static u32 get_amp_max_value(struct hda_codec *codec, hda_nid_t nid, int dir, + unsigned int ofs) +{ + u32 caps = query_amp_caps(codec, nid, dir); + /* get num steps */ + caps = (caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT; + if (ofs < caps) + caps -= ofs; + return caps; +} + /** * snd_hda_mixer_amp_volume_info - Info callback for a standard AMP mixer * @@ -1579,23 +1716,17 @@ int snd_hda_mixer_amp_volume_info(struct snd_kcontrol *kcontrol, u8 chs = get_amp_channels(kcontrol); int dir = get_amp_direction(kcontrol); unsigned int ofs = get_amp_offset(kcontrol); - u32 caps; - caps = query_amp_caps(codec, nid, dir); - /* num steps */ - caps = (caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT; - if (!caps) { + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = chs == 3 ? 2 : 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = get_amp_max_value(codec, nid, dir, ofs); + if (!uinfo->value.integer.max) { printk(KERN_WARNING "hda_codec: " "num_steps = 0 for NID=0x%x (ctl = %s)\n", nid, kcontrol->id.name); return -EINVAL; } - if (ofs < caps) - caps -= ofs; - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = chs == 3 ? 2 : 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = caps; return 0; } EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_volume_info); @@ -1620,8 +1751,14 @@ update_amp_value(struct hda_codec *codec, hda_nid_t nid, int ch, int dir, int idx, unsigned int ofs, unsigned int val) { + unsigned int maxval; + if (val > 0) val += ofs; + /* ofs = 0: raw max value */ + maxval = get_amp_max_value(codec, nid, dir, 0); + if (val > maxval) + val = maxval; return snd_hda_codec_amp_update(codec, nid, ch, dir, idx, HDA_AMP_VOLMASK, val); } @@ -2912,6 +3049,7 @@ static void hda_call_codec_suspend(struct hda_codec *codec) { if (codec->patch_ops.suspend) codec->patch_ops.suspend(codec, PMSG_SUSPEND); + hda_cleanup_all_streams(codec); hda_set_power_state(codec, codec->afg ? codec->afg : codec->mfg, AC_PWRST_D3); @@ -2999,26 +3137,31 @@ struct hda_rate_tbl { unsigned int hda_fmt; }; +/* rate = base * mult / div */ +#define HDA_RATE(base, mult, div) \ + (AC_FMT_BASE_##base##K | (((mult) - 1) << AC_FMT_MULT_SHIFT) | \ + (((div) - 1) << AC_FMT_DIV_SHIFT)) + static struct hda_rate_tbl rate_bits[] = { /* rate in Hz, ALSA rate bitmask, HDA format value */ /* autodetected value used in snd_hda_query_supported_pcm */ - { 8000, SNDRV_PCM_RATE_8000, 0x0500 }, /* 1/6 x 48 */ - { 11025, SNDRV_PCM_RATE_11025, 0x4300 }, /* 1/4 x 44 */ - { 16000, SNDRV_PCM_RATE_16000, 0x0200 }, /* 1/3 x 48 */ - { 22050, SNDRV_PCM_RATE_22050, 0x4100 }, /* 1/2 x 44 */ - { 32000, SNDRV_PCM_RATE_32000, 0x0a00 }, /* 2/3 x 48 */ - { 44100, SNDRV_PCM_RATE_44100, 0x4000 }, /* 44 */ - { 48000, SNDRV_PCM_RATE_48000, 0x0000 }, /* 48 */ - { 88200, SNDRV_PCM_RATE_88200, 0x4800 }, /* 2 x 44 */ - { 96000, SNDRV_PCM_RATE_96000, 0x0800 }, /* 2 x 48 */ - { 176400, SNDRV_PCM_RATE_176400, 0x5800 },/* 4 x 44 */ - { 192000, SNDRV_PCM_RATE_192000, 0x1800 }, /* 4 x 48 */ + { 8000, SNDRV_PCM_RATE_8000, HDA_RATE(48, 1, 6) }, + { 11025, SNDRV_PCM_RATE_11025, HDA_RATE(44, 1, 4) }, + { 16000, SNDRV_PCM_RATE_16000, HDA_RATE(48, 1, 3) }, + { 22050, SNDRV_PCM_RATE_22050, HDA_RATE(44, 1, 2) }, + { 32000, SNDRV_PCM_RATE_32000, HDA_RATE(48, 2, 3) }, + { 44100, SNDRV_PCM_RATE_44100, HDA_RATE(44, 1, 1) }, + { 48000, SNDRV_PCM_RATE_48000, HDA_RATE(48, 1, 1) }, + { 88200, SNDRV_PCM_RATE_88200, HDA_RATE(44, 2, 1) }, + { 96000, SNDRV_PCM_RATE_96000, HDA_RATE(48, 2, 1) }, + { 176400, SNDRV_PCM_RATE_176400, HDA_RATE(44, 4, 1) }, + { 192000, SNDRV_PCM_RATE_192000, HDA_RATE(48, 4, 1) }, #define AC_PAR_PCM_RATE_BITS 11 /* up to bits 10, 384kHZ isn't supported properly */ /* not autodetected value */ - { 9600, SNDRV_PCM_RATE_KNOT, 0x0400 }, /* 1/5 x 48 */ + { 9600, SNDRV_PCM_RATE_KNOT, HDA_RATE(48, 1, 5) }, { 0 } /* terminator */ }; @@ -3037,7 +3180,8 @@ static struct hda_rate_tbl rate_bits[] = { unsigned int snd_hda_calc_stream_format(unsigned int rate, unsigned int channels, unsigned int format, - unsigned int maxbps) + unsigned int maxbps, + unsigned short spdif_ctls) { int i; unsigned int val = 0; @@ -3060,20 +3204,20 @@ unsigned int snd_hda_calc_stream_format(unsigned int rate, switch (snd_pcm_format_width(format)) { case 8: - val |= 0x00; + val |= AC_FMT_BITS_8; break; case 16: - val |= 0x10; + val |= AC_FMT_BITS_16; break; case 20: case 24: case 32: if (maxbps >= 32 || format == SNDRV_PCM_FORMAT_FLOAT_LE) - val |= 0x40; + val |= AC_FMT_BITS_32; else if (maxbps >= 24) - val |= 0x30; + val |= AC_FMT_BITS_24; else - val |= 0x20; + val |= AC_FMT_BITS_20; break; default: snd_printdd("invalid format width %d\n", @@ -3081,6 +3225,9 @@ unsigned int snd_hda_calc_stream_format(unsigned int rate, return 0; } + if (spdif_ctls & AC_DIG1_NONAUDIO) + val |= AC_FMT_TYPE_NON_PCM; + return val; } EXPORT_SYMBOL_HDA(snd_hda_calc_stream_format); @@ -3352,6 +3499,35 @@ static int set_pcm_default_values(struct hda_codec *codec, return 0; } +/* + * codec prepare/cleanup entries + */ +int snd_hda_codec_prepare(struct hda_codec *codec, + struct hda_pcm_stream *hinfo, + unsigned int stream, + unsigned int format, + struct snd_pcm_substream *substream) +{ + int ret; + mutex_lock(&codec->bus->prepare_mutex); + ret = hinfo->ops.prepare(hinfo, codec, stream, format, substream); + if (ret >= 0) + purify_inactive_streams(codec); + mutex_unlock(&codec->bus->prepare_mutex); + return ret; +} +EXPORT_SYMBOL_HDA(snd_hda_codec_prepare); + +void snd_hda_codec_cleanup(struct hda_codec *codec, + struct hda_pcm_stream *hinfo, + struct snd_pcm_substream *substream) +{ + mutex_lock(&codec->bus->prepare_mutex); + hinfo->ops.cleanup(hinfo, codec, substream); + mutex_unlock(&codec->bus->prepare_mutex); +} +EXPORT_SYMBOL_HDA(snd_hda_codec_cleanup); + /* global */ const char *snd_hda_pcm_type_name[HDA_PCM_NTYPES] = { "Audio", "SPDIF", "HDMI", "Modem" @@ -4360,7 +4536,7 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, cfg->hp_outs--; memmove(cfg->hp_pins + i, cfg->hp_pins + i + 1, sizeof(cfg->hp_pins[0]) * (cfg->hp_outs - i)); - memmove(sequences_hp + i - 1, sequences_hp + i, + memmove(sequences_hp + i, sequences_hp + i + 1, sizeof(sequences_hp[0]) * (cfg->hp_outs - i)); } } diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 5991d14e1ec0..62c702240108 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -224,6 +224,27 @@ enum { /* Input converter SDI select */ #define AC_SDI_SELECT (0xf<<0) +/* stream format id */ +#define AC_FMT_CHAN_SHIFT 0 +#define AC_FMT_CHAN_MASK (0x0f << 0) +#define AC_FMT_BITS_SHIFT 4 +#define AC_FMT_BITS_MASK (7 << 4) +#define AC_FMT_BITS_8 (0 << 4) +#define AC_FMT_BITS_16 (1 << 4) +#define AC_FMT_BITS_20 (2 << 4) +#define AC_FMT_BITS_24 (3 << 4) +#define AC_FMT_BITS_32 (4 << 4) +#define AC_FMT_DIV_SHIFT 8 +#define AC_FMT_DIV_MASK (7 << 8) +#define AC_FMT_MULT_SHIFT 11 +#define AC_FMT_MULT_MASK (7 << 11) +#define AC_FMT_BASE_SHIFT 14 +#define AC_FMT_BASE_48K (0 << 14) +#define AC_FMT_BASE_44K (1 << 14) +#define AC_FMT_TYPE_SHIFT 15 +#define AC_FMT_TYPE_PCM (0 << 15) +#define AC_FMT_TYPE_NON_PCM (1 << 15) + /* Unsolicited response control */ #define AC_UNSOL_TAG (0x3f<<0) #define AC_UNSOL_ENABLED (1<<7) @@ -364,6 +385,9 @@ enum { #define AC_DIG2_CC (0x7f<<0) /* Pin widget control - 8bit */ +#define AC_PINCTL_EPT (0x3<<0) +#define AC_PINCTL_EPT_NATIVE 0 +#define AC_PINCTL_EPT_HBR 3 #define AC_PINCTL_VREFEN (0x7<<0) #define AC_PINCTL_VREF_HIZ 0 /* Hi-Z */ #define AC_PINCTL_VREF_50 1 /* 50% */ @@ -624,6 +648,7 @@ struct hda_bus { struct hda_codec *caddr_tbl[HDA_MAX_CODEC_ADDRESS + 1]; struct mutex cmd_mutex; + struct mutex prepare_mutex; /* unsolicited event queue */ struct hda_bus_unsolicited *unsol; @@ -760,7 +785,10 @@ struct hda_codec { hda_nid_t mfg; /* MFG node id */ /* ids */ - u32 function_id; + u8 afg_function_id; + u8 mfg_function_id; + u8 afg_unsol; + u8 mfg_unsol; u32 vendor_id; u32 subsystem_id; u32 revision_id; @@ -805,6 +833,7 @@ struct hda_codec { hda_nid_t *slave_dig_outs; /* optional digital out slave widgets */ struct snd_array init_pins; /* initial (BIOS) pin configurations */ struct snd_array driver_pins; /* pin configs set by codec parser */ + struct snd_array cvt_setups; /* audio convert setups */ #ifdef CONFIG_SND_HDA_HWDEP struct snd_hwdep *hwdep; /* assigned hwdep device */ @@ -921,14 +950,28 @@ int snd_hda_codec_build_controls(struct hda_codec *codec); */ int snd_hda_build_pcms(struct hda_bus *bus); int snd_hda_codec_build_pcms(struct hda_codec *codec); + +int snd_hda_codec_prepare(struct hda_codec *codec, + struct hda_pcm_stream *hinfo, + unsigned int stream, + unsigned int format, + struct snd_pcm_substream *substream); +void snd_hda_codec_cleanup(struct hda_codec *codec, + struct hda_pcm_stream *hinfo, + struct snd_pcm_substream *substream); + void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, u32 stream_tag, int channel_id, int format); -void snd_hda_codec_cleanup_stream(struct hda_codec *codec, hda_nid_t nid); +void __snd_hda_codec_cleanup_stream(struct hda_codec *codec, hda_nid_t nid, + int do_now); +#define snd_hda_codec_cleanup_stream(codec, nid) \ + __snd_hda_codec_cleanup_stream(codec, nid, 0) unsigned int snd_hda_calc_stream_format(unsigned int rate, unsigned int channels, unsigned int format, - unsigned int maxbps); + unsigned int maxbps, + unsigned short spdif_ctls); int snd_hda_is_supported_format(struct hda_codec *codec, hda_nid_t nid, unsigned int format); diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index d8da18a9e98b..26c3ade73583 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -597,3 +597,52 @@ void snd_hda_eld_proc_free(struct hda_codec *codec, struct hdmi_eld *eld) EXPORT_SYMBOL_HDA(snd_hda_eld_proc_free); #endif /* CONFIG_PROC_FS */ + +/* update PCM info based on ELD */ +void hdmi_eld_update_pcm_info(struct hdmi_eld *eld, struct hda_pcm_stream *pcm, + struct hda_pcm_stream *codec_pars) +{ + int i; + + pcm->rates = 0; + pcm->formats = 0; + pcm->maxbps = 0; + pcm->channels_min = -1; + pcm->channels_max = 0; + for (i = 0; i < eld->sad_count; i++) { + struct cea_sad *a = &eld->sad[i]; + pcm->rates |= a->rates; + if (a->channels < pcm->channels_min) + pcm->channels_min = a->channels; + if (a->channels > pcm->channels_max) + pcm->channels_max = a->channels; + if (a->format == AUDIO_CODING_TYPE_LPCM) { + if (a->sample_bits & AC_SUPPCM_BITS_16) { + pcm->formats |= SNDRV_PCM_FMTBIT_S16_LE; + if (pcm->maxbps < 16) + pcm->maxbps = 16; + } + if (a->sample_bits & AC_SUPPCM_BITS_20) { + pcm->formats |= SNDRV_PCM_FMTBIT_S32_LE; + if (pcm->maxbps < 20) + pcm->maxbps = 20; + } + if (a->sample_bits & AC_SUPPCM_BITS_24) { + pcm->formats |= SNDRV_PCM_FMTBIT_S32_LE; + if (pcm->maxbps < 24) + pcm->maxbps = 24; + } + } + } + + if (!codec_pars) + return; + + /* restrict the parameters by the values the codec provides */ + pcm->rates &= codec_pars->rates; + pcm->formats &= codec_pars->formats; + pcm->channels_min = max(pcm->channels_min, codec_pars->channels_min); + pcm->channels_max = min(pcm->channels_max, codec_pars->channels_max); + pcm->maxbps = min(pcm->maxbps, codec_pars->maxbps); +} +EXPORT_SYMBOL_HDA(hdmi_eld_update_pcm_info); diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index a1fc83753cc6..bf3ced51e0f8 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -649,7 +649,9 @@ static void parse_codec_mode(char *buf, struct hda_bus *bus, *codecp = NULL; if (sscanf(buf, "%i %i %i", &vendorid, &subid, &caddr) == 3) { list_for_each_entry(codec, &bus->codec_list, list) { - if (codec->addr == caddr) { + if (codec->vendor_id == vendorid && + codec->subsystem_id == subid && + codec->addr == caddr) { *codecp = codec; break; } diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1df25cf5ce38..34940a079051 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -126,6 +126,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6}," "{Intel, ICH10}," "{Intel, PCH}," "{Intel, CPT}," + "{Intel, PBG}," "{Intel, SCH}," "{ATI, SB450}," "{ATI, SB600}," @@ -1634,7 +1635,7 @@ static int azx_pcm_hw_free(struct snd_pcm_substream *substream) azx_dev->period_bytes = 0; azx_dev->format_val = 0; - hinfo->ops.cleanup(hinfo, apcm->codec, substream); + snd_hda_codec_cleanup(apcm->codec, hinfo, substream); return snd_pcm_lib_free_pages(substream); } @@ -1653,7 +1654,8 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) format_val = snd_hda_calc_stream_format(runtime->rate, runtime->channels, runtime->format, - hinfo->maxbps); + hinfo->maxbps, + apcm->codec->spdif_ctls); if (!format_val) { snd_printk(KERN_ERR SFX "invalid format_val, rate=%d, ch=%d, format=%d\n", @@ -1687,8 +1689,8 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) else azx_dev->fifo_size = 0; - return hinfo->ops.prepare(hinfo, apcm->codec, azx_dev->stream_tag, - azx_dev->format_val, substream); + return snd_hda_codec_prepare(apcm->codec, hinfo, azx_dev->stream_tag, + azx_dev->format_val, substream); } static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) @@ -1960,7 +1962,7 @@ static void azx_irq_pending_work(struct work_struct *work) spin_unlock_irq(&chip->reg_lock); if (!pending) return; - cond_resched(); + msleep(1); } } @@ -2748,6 +2750,8 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { { PCI_DEVICE(0x8086, 0x3b57), .driver_data = AZX_DRIVER_ICH }, /* CPT */ { PCI_DEVICE(0x8086, 0x1c20), .driver_data = AZX_DRIVER_PCH }, + /* PBG */ + { PCI_DEVICE(0x8086, 0x1d20), .driver_data = AZX_DRIVER_PCH }, /* SCH */ { PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH }, /* ATI SB 450/600 */ diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 7a97f126f6f7..28ab4aead48f 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -604,6 +604,8 @@ struct hdmi_eld { int snd_hdmi_get_eld_size(struct hda_codec *codec, hda_nid_t nid); int snd_hdmi_get_eld(struct hdmi_eld *, struct hda_codec *, hda_nid_t); void snd_hdmi_show_eld(struct hdmi_eld *eld); +void hdmi_eld_update_pcm_info(struct hdmi_eld *eld, struct hda_pcm_stream *pcm, + struct hda_pcm_stream *codec_pars); #ifdef CONFIG_PROC_FS int snd_hda_eld_proc_new(struct hda_codec *codec, struct hdmi_eld *eld, diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index f97d35de66c4..f025200f2a62 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -557,7 +557,12 @@ static void print_codec_info(struct snd_info_entry *entry, else snd_iprintf(buffer, "Not Set\n"); snd_iprintf(buffer, "Address: %d\n", codec->addr); - snd_iprintf(buffer, "Function Id: 0x%x\n", codec->function_id); + if (codec->afg) + snd_iprintf(buffer, "AFG Function Id: 0x%x (unsol %u)\n", + codec->afg_function_id, codec->afg_unsol); + if (codec->mfg) + snd_iprintf(buffer, "MFG Function Id: 0x%x (unsol %u)\n", + codec->mfg_function_id, codec->mfg_unsol); snd_iprintf(buffer, "Vendor Id: 0x%08x\n", codec->vendor_id); snd_iprintf(buffer, "Subsystem Id: 0x%08x\n", codec->subsystem_id); snd_iprintf(buffer, "Revision Id: 0x%x\n", codec->revision_id); diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index afbe314a5bf3..10bbbaf6ebc3 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -3641,6 +3641,7 @@ static struct snd_pci_quirk ad1984_cfg_tbl[] = { /* Lenovo Thinkpad T61/X61 */ SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo Thinkpad", AD1984_THINKPAD), SND_PCI_QUIRK(0x1028, 0x0214, "Dell T3400", AD1984_DELL_DESKTOP), + SND_PCI_QUIRK(0x1028, 0x0233, "Dell Latitude E6400", AD1984_DELL_DESKTOP), {} }; @@ -3662,7 +3663,12 @@ static int patch_ad1984(struct hda_codec *codec) codec->patch_ops.build_pcms = ad1984_build_pcms; break; case AD1984_THINKPAD: - spec->multiout.dig_out_nid = AD1884_SPDIF_OUT; + if (codec->subsystem_id == 0x17aa20fb) { + /* Thinpad X300 does not have the ability to do SPDIF, + or attach to docking station to use SPDIF */ + spec->multiout.dig_out_nid = 0; + } else + spec->multiout.dig_out_nid = AD1884_SPDIF_OUT; spec->input_mux = &ad1984_thinkpad_capture_source; spec->mixers[0] = ad1984_thinkpad_mixers; spec->init_verbs[spec->num_init_verbs++] = ad1984_thinkpad_init_verbs; diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 350ee8ac4153..488fd9ade1ba 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -656,7 +656,7 @@ static int change_cur_input(struct hda_codec *codec, unsigned int idx, return 0; if (spec->cur_adc && spec->cur_adc != spec->adc_nid[idx]) { /* stream is running, let's swap the current ADC */ - snd_hda_codec_cleanup_stream(codec, spec->cur_adc); + __snd_hda_codec_cleanup_stream(codec, spec->cur_adc, 1); spec->cur_adc = spec->adc_nid[idx]; snd_hda_codec_setup_stream(codec, spec->cur_adc, spec->cur_adc_stream_tag, 0, @@ -972,6 +972,53 @@ static struct hda_verb cs_coef_init_verbs[] = { {} /* terminator */ }; +/* Errata: CS4207 rev C0/C1/C2 Silicon + * + * http://www.cirrus.com/en/pubs/errata/ER880C3.pdf + * + * 6. At high temperature (TA > +85°C), the digital supply current (IVD) + * may be excessive (up to an additional 200 μA), which is most easily + * observed while the part is being held in reset (RESET# active low). + * + * Root Cause: At initial powerup of the device, the logic that drives + * the clock and write enable to the S/PDIF SRC RAMs is not properly + * initialized. + * Certain random patterns will cause a steady leakage current in those + * RAM cells. The issue will resolve once the SRCs are used (turned on). + * + * Workaround: The following verb sequence briefly turns on the S/PDIF SRC + * blocks, which will alleviate the issue. + */ + +static struct hda_verb cs_errata_init_verbs[] = { + {0x01, AC_VERB_SET_POWER_STATE, 0x00}, /* AFG: D0 */ + {0x11, AC_VERB_SET_PROC_STATE, 0x01}, /* VPW: processing on */ + + {0x11, AC_VERB_SET_COEF_INDEX, 0x0008}, + {0x11, AC_VERB_SET_PROC_COEF, 0x9999}, + {0x11, AC_VERB_SET_COEF_INDEX, 0x0017}, + {0x11, AC_VERB_SET_PROC_COEF, 0xa412}, + {0x11, AC_VERB_SET_COEF_INDEX, 0x0001}, + {0x11, AC_VERB_SET_PROC_COEF, 0x0009}, + + {0x07, AC_VERB_SET_POWER_STATE, 0x00}, /* S/PDIF Rx: D0 */ + {0x08, AC_VERB_SET_POWER_STATE, 0x00}, /* S/PDIF Tx: D0 */ + + {0x11, AC_VERB_SET_COEF_INDEX, 0x0017}, + {0x11, AC_VERB_SET_PROC_COEF, 0x2412}, + {0x11, AC_VERB_SET_COEF_INDEX, 0x0008}, + {0x11, AC_VERB_SET_PROC_COEF, 0x0000}, + {0x11, AC_VERB_SET_COEF_INDEX, 0x0001}, + {0x11, AC_VERB_SET_PROC_COEF, 0x0008}, + {0x11, AC_VERB_SET_PROC_STATE, 0x00}, + + {0x07, AC_VERB_SET_POWER_STATE, 0x03}, /* S/PDIF Rx: D3 */ + {0x08, AC_VERB_SET_POWER_STATE, 0x03}, /* S/PDIF Tx: D3 */ + /*{0x01, AC_VERB_SET_POWER_STATE, 0x03},*/ /* AFG: D3 This is already handled */ + + {} /* terminator */ +}; + /* SPDIF setup */ static void init_digital(struct hda_codec *codec) { @@ -991,6 +1038,9 @@ static int cs_init(struct hda_codec *codec) { struct cs_spec *spec = codec->spec; + /* init_verb sequence for C0/C1/C2 errata*/ + snd_hda_sequence_write(codec, cs_errata_init_verbs); + snd_hda_sequence_write(codec, cs_coef_init_verbs); if (spec->gpio_mask) { diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 2bf2cb5da956..972e7c453b3d 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -116,6 +116,7 @@ struct conexant_spec { unsigned int dell_vostro:1; unsigned int ideapad:1; unsigned int thinkpad:1; + unsigned int hp_laptop:1; unsigned int ext_mic_present; unsigned int recording; @@ -131,6 +132,8 @@ struct conexant_spec { unsigned int dc_enable; unsigned int dc_input_bias; /* offset into cxt5066_olpc_dc_bias */ unsigned int mic_boost; /* offset into cxt5066_analog_mic_boost */ + + unsigned int beep_amp; }; static int conexant_playback_pcm_open(struct hda_pcm_stream *hinfo, @@ -515,6 +518,15 @@ static struct snd_kcontrol_new cxt_capture_mixers[] = { {} }; +#ifdef CONFIG_SND_HDA_INPUT_BEEP +/* additional beep mixers; the actual parameters are overwritten at build */ +static struct snd_kcontrol_new cxt_beep_mixer[] = { + HDA_CODEC_VOLUME_MONO("Beep Playback Volume", 0, 1, 0, HDA_OUTPUT), + HDA_CODEC_MUTE_BEEP_MONO("Beep Playback Switch", 0, 1, 0, HDA_OUTPUT), + { } /* end */ +}; +#endif + static const char *slave_vols[] = { "Headphone Playback Volume", "Speaker Playback Volume", @@ -580,16 +592,52 @@ static int conexant_build_controls(struct hda_codec *codec) return err; } +#ifdef CONFIG_SND_HDA_INPUT_BEEP + /* create beep controls if needed */ + if (spec->beep_amp) { + struct snd_kcontrol_new *knew; + for (knew = cxt_beep_mixer; knew->name; knew++) { + struct snd_kcontrol *kctl; + kctl = snd_ctl_new1(knew, codec); + if (!kctl) + return -ENOMEM; + kctl->private_value = spec->beep_amp; + err = snd_hda_ctl_add(codec, 0, kctl); + if (err < 0) + return err; + } + } +#endif + return 0; } +#ifdef CONFIG_SND_HDA_POWER_SAVE +static int conexant_suspend(struct hda_codec *codec, pm_message_t state) +{ + snd_hda_shutup_pins(codec); + return 0; +} +#endif + static struct hda_codec_ops conexant_patch_ops = { .build_controls = conexant_build_controls, .build_pcms = conexant_build_pcms, .init = conexant_init, .free = conexant_free, +#ifdef CONFIG_SND_HDA_POWER_SAVE + .suspend = conexant_suspend, +#endif + .reboot_notify = snd_hda_shutup_pins, }; +#ifdef CONFIG_SND_HDA_INPUT_BEEP +#define set_beep_amp(spec, nid, idx, dir) \ + ((spec)->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 1, idx, dir)) +#else +#define set_beep_amp(spec, nid, idx, dir) /* NOP */ +#endif + /* * EAPD control * the private value = nid | (invert << 8) @@ -1130,9 +1178,10 @@ static int patch_cxt5045(struct hda_codec *codec) spec->num_init_verbs = 1; spec->init_verbs[0] = cxt5045_init_verbs; spec->spdif_route = 0; - spec->num_channel_mode = ARRAY_SIZE(cxt5045_modes), - spec->channel_mode = cxt5045_modes, + spec->num_channel_mode = ARRAY_SIZE(cxt5045_modes); + spec->channel_mode = cxt5045_modes; + set_beep_amp(spec, 0x16, 0, 1); codec->patch_ops = conexant_patch_ops; @@ -1211,6 +1260,9 @@ static int patch_cxt5045(struct hda_codec *codec) break; } + if (spec->beep_amp) + snd_hda_attach_beep_device(codec, spec->beep_amp); + return 0; } @@ -1632,6 +1684,11 @@ static void cxt5051_update_speaker(struct hda_codec *codec) pinctl = (!spec->hp_present && spec->cur_eapd) ? PIN_OUT : 0; snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl); + /* on ideapad there is an aditional speaker (subwoofer) to mute */ + if (spec->ideapad) + snd_hda_codec_write(codec, 0x1b, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + pinctl); } /* turn on/off EAPD (+ mute HP) as a master switch */ @@ -1677,7 +1734,7 @@ static void cxt5051_portc_automic(struct hda_codec *codec) new_adc = spec->adc_nids[spec->cur_adc_idx]; if (spec->cur_adc && spec->cur_adc != new_adc) { /* stream is running, let's swap the current ADC */ - snd_hda_codec_cleanup_stream(codec, spec->cur_adc); + __snd_hda_codec_cleanup_stream(codec, spec->cur_adc, 1); spec->cur_adc = new_adc; snd_hda_codec_setup_stream(codec, new_adc, spec->cur_adc_stream_tag, 0, @@ -1888,6 +1945,13 @@ static void cxt5051_init_mic_port(struct hda_codec *codec, hda_nid_t nid, #endif } +static struct hda_verb cxt5051_ideapad_init_verbs[] = { + /* Subwoofer */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, + { } /* end */ +}; + /* initialize jack-sensing, too */ static int cxt5051_init(struct hda_codec *codec) { @@ -1917,6 +1981,7 @@ enum { CXT5051_LENOVO_X200, /* Lenovo X200 laptop, also used for Advanced Mini Dock 250410 */ CXT5051_F700, /* HP Compaq Presario F700 */ CXT5051_TOSHIBA, /* Toshiba M300 & co */ + CXT5051_IDEAPAD, /* Lenovo IdeaPad Y430 */ CXT5051_MODELS }; @@ -1927,6 +1992,7 @@ static const char *cxt5051_models[CXT5051_MODELS] = { [CXT5051_LENOVO_X200] = "lenovo-x200", [CXT5051_F700] = "hp-700", [CXT5051_TOSHIBA] = "toshiba", + [CXT5051_IDEAPAD] = "ideapad", }; static struct snd_pci_quirk cxt5051_cfg_tbl[] = { @@ -1938,6 +2004,7 @@ static struct snd_pci_quirk cxt5051_cfg_tbl[] = { CXT5051_LAPTOP), SND_PCI_QUIRK(0x14f1, 0x5051, "HP Spartan 1.1", CXT5051_HP), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT5051_LENOVO_X200), + SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo IdeaPad", CXT5051_IDEAPAD), {} }; @@ -1972,6 +2039,8 @@ static int patch_cxt5051(struct hda_codec *codec) spec->cur_adc = 0; spec->cur_adc_idx = 0; + set_beep_amp(spec, 0x13, 0, HDA_OUTPUT); + codec->patch_ops.unsol_event = cxt5051_hp_unsol_event; board_config = snd_hda_check_board_config(codec, CXT5051_MODELS, @@ -1989,6 +2058,10 @@ static int patch_cxt5051(struct hda_codec *codec) break; case CXT5051_LENOVO_X200: spec->init_verbs[0] = cxt5051_lenovo_x200_init_verbs; + /* Thinkpad X301 does not have S/PDIF wired and no ability + to use a docking station. */ + if (codec->subsystem_id == 0x17aa211f) + spec->multiout.dig_out_nid = 0; break; case CXT5051_F700: spec->init_verbs[0] = cxt5051_f700_init_verbs; @@ -1999,8 +2072,16 @@ static int patch_cxt5051(struct hda_codec *codec) spec->mixers[0] = cxt5051_toshiba_mixers; spec->auto_mic = AUTO_MIC_PORTB; break; + case CXT5051_IDEAPAD: + spec->init_verbs[spec->num_init_verbs++] = + cxt5051_ideapad_init_verbs; + spec->ideapad = 1; + break; } + if (spec->beep_amp) + snd_hda_attach_beep_device(codec, spec->beep_amp); + return 0; } @@ -2219,6 +2300,18 @@ static void cxt5066_ideapad_automic(struct hda_codec *codec) } } +/* toggle input of built-in digital mic and mic jack appropriately */ +static void cxt5066_hp_laptop_automic(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_jack_detect(codec, 0x1b); + snd_printdd("CXT5066: external microphone present=%d\n", present); + snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_CONNECT_SEL, + present ? 1 : 3); +} + + /* toggle input of built-in digital mic and mic jack appropriately order is: external mic -> dock mic -> interal mic */ static void cxt5066_thinkpad_automic(struct hda_codec *codec) @@ -2328,6 +2421,20 @@ static void cxt5066_ideapad_event(struct hda_codec *codec, unsigned int res) } /* unsolicited event for jack sensing */ +static void cxt5066_hp_laptop_event(struct hda_codec *codec, unsigned int res) +{ + snd_printdd("CXT5066_hp_laptop: unsol event %x (%x)\n", res, res >> 26); + switch (res >> 26) { + case CONEXANT_HP_EVENT: + cxt5066_hp_automute(codec); + break; + case CONEXANT_MIC_EVENT: + cxt5066_hp_laptop_automic(codec); + break; + } +} + +/* unsolicited event for jack sensing */ static void cxt5066_thinkpad_event(struct hda_codec *codec, unsigned int res) { snd_printdd("CXT5066_thinkpad: unsol event %x (%x)\n", res, res >> 26); @@ -2616,7 +2723,6 @@ static struct snd_kcontrol_new cxt5066_vostro_mixers[] = { .put = cxt5066_mic_boost_mux_enum_put, .private_value = 0x23 | 0x100, }, - HDA_CODEC_VOLUME_MONO("Beep Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT), {} }; @@ -2910,6 +3016,14 @@ static struct hda_verb cxt5066_init_verbs_portd_lo[] = { { } /* end */ }; + +static struct hda_verb cxt5066_init_verbs_hp_laptop[] = { + {0x14, AC_VERB_SET_CONNECT_SEL, 0x0}, + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, + { } /* end */ +}; + /* initialize jack-sensing, too */ static int cxt5066_init(struct hda_codec *codec) { @@ -2925,6 +3039,8 @@ static int cxt5066_init(struct hda_codec *codec) cxt5066_ideapad_automic(codec); else if (spec->thinkpad) cxt5066_thinkpad_automic(codec); + else if (spec->hp_laptop) + cxt5066_hp_laptop_automic(codec); } cxt5066_set_mic_boost(codec); return 0; @@ -2952,6 +3068,7 @@ enum { CXT5066_DELL_VOSTO, /* Dell Vostro 1015i */ CXT5066_IDEAPAD, /* Lenovo IdeaPad U150 */ CXT5066_THINKPAD, /* Lenovo ThinkPad T410s, others? */ + CXT5066_HP_LAPTOP, /* HP Laptop */ CXT5066_MODELS }; @@ -2962,6 +3079,7 @@ static const char *cxt5066_models[CXT5066_MODELS] = { [CXT5066_DELL_VOSTO] = "dell-vostro", [CXT5066_IDEAPAD] = "ideapad", [CXT5066_THINKPAD] = "thinkpad", + [CXT5066_HP_LAPTOP] = "hp-laptop", }; static struct snd_pci_quirk cxt5066_cfg_tbl[] = { @@ -2970,15 +3088,22 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x1028, 0x02f5, "Dell", CXT5066_DELL_LAPTOP), SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5), + SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTO), SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTO), SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), + SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP), + SND_PCI_QUIRK(0x1179, 0xff1e, "Toshiba Satellite C650D", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5), + SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21b2, "Thinkpad X100e", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x21b3, "Thinkpad Edge 13 (197)", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x21b4, "Thinkpad Edge", CXT5066_IDEAPAD), + SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD), + SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G series", CXT5066_IDEAPAD), + SND_PCI_QUIRK(0x17aa, 0x390a, "Lenovo S10-3t", CXT5066_IDEAPAD), + SND_PCI_QUIRK(0x17aa, 0x3938, "Lenovo G series (AMD)", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "ideapad", CXT5066_IDEAPAD), - SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD), {} }; @@ -3014,6 +3139,8 @@ static int patch_cxt5066(struct hda_codec *codec) spec->cur_adc = 0; spec->cur_adc_idx = 0; + set_beep_amp(spec, 0x13, 0, HDA_OUTPUT); + board_config = snd_hda_check_board_config(codec, CXT5066_MODELS, cxt5066_models, cxt5066_cfg_tbl); switch (board_config) { @@ -3031,6 +3158,23 @@ static int patch_cxt5066(struct hda_codec *codec) spec->num_init_verbs++; spec->dell_automute = 1; break; + case CXT5066_HP_LAPTOP: + codec->patch_ops.init = cxt5066_init; + codec->patch_ops.unsol_event = cxt5066_hp_laptop_event; + spec->init_verbs[spec->num_init_verbs] = + cxt5066_init_verbs_hp_laptop; + spec->num_init_verbs++; + spec->hp_laptop = 1; + spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; + spec->mixers[spec->num_mixers++] = cxt5066_mixers; + /* no S/PDIF out */ + spec->multiout.dig_out_nid = 0; + /* input source automatically selected */ + spec->input_mux = NULL; + spec->port_d_mode = 0; + spec->mic_boost = 3; /* default 30dB gain */ + break; + case CXT5066_OLPC_XO_1_5: codec->patch_ops.init = cxt5066_olpc_init; codec->patch_ops.unsol_event = cxt5066_olpc_unsol_event; @@ -3062,7 +3206,6 @@ static int patch_cxt5066(struct hda_codec *codec) spec->port_d_mode = 0; spec->dell_vostro = 1; spec->mic_boost = 3; /* default 30dB gain */ - snd_hda_attach_beep_device(codec, 0x13); /* no S/PDIF out */ spec->multiout.dig_out_nid = 0; @@ -3104,6 +3247,9 @@ static int patch_cxt5066(struct hda_codec *codec) break; } + if (spec->beep_amp) + snd_hda_attach_beep_device(codec, spec->beep_amp); + return 0; } @@ -3121,6 +3267,8 @@ static struct hda_codec_preset snd_hda_preset_conexant[] = { .patch = patch_cxt5066 }, { .id = 0x14f15067, .name = "CX20583 (Pebble HSF)", .patch = patch_cxt5066 }, + { .id = 0x14f15068, .name = "CX20584", + .patch = patch_cxt5066 }, { .id = 0x14f15069, .name = "CX20585", .patch = patch_cxt5066 }, {} /* terminator */ @@ -3131,6 +3279,7 @@ MODULE_ALIAS("snd-hda-codec-id:14f15047"); MODULE_ALIAS("snd-hda-codec-id:14f15051"); MODULE_ALIAS("snd-hda-codec-id:14f15066"); MODULE_ALIAS("snd-hda-codec-id:14f15067"); +MODULE_ALIAS("snd-hda-codec-id:14f15068"); MODULE_ALIAS("snd-hda-codec-id:14f15069"); MODULE_LICENSE("GPL"); diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 2fc53961054e..afd6022a96a7 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -46,6 +46,7 @@ struct hdmi_spec { * export one pcm per pipe */ struct hda_pcm pcm_rec[MAX_HDMI_CVTS]; + struct hda_pcm_stream codec_pcm_pars[MAX_HDMI_CVTS]; /* * nvhdmi specific @@ -698,30 +699,93 @@ static void hdmi_unsol_event(struct hda_codec *codec, unsigned int res) * Callbacks */ -static void hdmi_setup_stream(struct hda_codec *codec, hda_nid_t nid, +/* HBR should be Non-PCM, 8 channels */ +#define is_hbr_format(format) \ + ((format & AC_FMT_TYPE_NON_PCM) && (format & AC_FMT_CHAN_MASK) == 7) + +static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t nid, u32 stream_tag, int format) { - int tag; - int fmt; + struct hdmi_spec *spec = codec->spec; + int pinctl; + int new_pinctl = 0; + int i; + + for (i = 0; i < spec->num_pins; i++) { + if (spec->pin_cvt[i] != nid) + continue; + if (!(snd_hda_query_pin_caps(codec, spec->pin[i]) & AC_PINCAP_HBR)) + continue; - tag = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0) >> 4; - fmt = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_STREAM_FORMAT, 0); + pinctl = snd_hda_codec_read(codec, spec->pin[i], 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + + new_pinctl = pinctl & ~AC_PINCTL_EPT; + if (is_hbr_format(format)) + new_pinctl |= AC_PINCTL_EPT_HBR; + else + new_pinctl |= AC_PINCTL_EPT_NATIVE; + + snd_printdd("hdmi_setup_stream: " + "NID=0x%x, %spinctl=0x%x\n", + spec->pin[i], + pinctl == new_pinctl ? "" : "new-", + new_pinctl); + + if (pinctl != new_pinctl) + snd_hda_codec_write(codec, spec->pin[i], 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + new_pinctl); + } - snd_printdd("hdmi_setup_stream: " - "NID=0x%x, %sstream=0x%x, %sformat=0x%x\n", - nid, - tag == stream_tag ? "" : "new-", - stream_tag, - fmt == format ? "" : "new-", - format); + if (is_hbr_format(format) && !new_pinctl) { + snd_printdd("hdmi_setup_stream: HBR is not supported\n"); + return -EINVAL; + } - if (tag != stream_tag) - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_CHANNEL_STREAMID, - stream_tag << 4); - if (fmt != format) - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_STREAM_FORMAT, format); + snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format); + return 0; +} + +/* + * HDA PCM callbacks + */ +static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct hdmi_spec *spec = codec->spec; + struct hdmi_eld *eld; + struct hda_pcm_stream *codec_pars; + unsigned int idx; + + for (idx = 0; idx < spec->num_cvts; idx++) + if (hinfo->nid == spec->cvt[idx]) + break; + if (snd_BUG_ON(idx >= spec->num_cvts) || + snd_BUG_ON(idx >= spec->num_pins)) + return -EINVAL; + + /* save the PCM info the codec provides */ + codec_pars = &spec->codec_pcm_pars[idx]; + if (!codec_pars->rates) + *codec_pars = *hinfo; + + eld = &spec->sink_eld[idx]; + if (eld->sad_count > 0) { + hdmi_eld_update_pcm_info(eld, hinfo, codec_pars); + if (hinfo->channels_min > hinfo->channels_max || + !hinfo->rates || !hinfo->formats) + return -ENODEV; + } else { + /* fallback to the codec default */ + hinfo->channels_min = codec_pars->channels_min; + hinfo->channels_max = codec_pars->channels_max; + hinfo->rates = codec_pars->rates; + hinfo->formats = codec_pars->formats; + hinfo->maxbps = codec_pars->maxbps; + } + return 0; } /* diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c index b81d23e42ace..36a9b83a6174 100644 --- a/sound/pci/hda/patch_intelhdmi.c +++ b/sound/pci/hda/patch_intelhdmi.c @@ -66,23 +66,15 @@ static int intel_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, hdmi_setup_audio_infoframe(codec, hinfo->nid, substream); - hdmi_setup_stream(codec, hinfo->nid, stream_tag, format); - return 0; -} - -static int intel_hdmi_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - return 0; + return hdmi_setup_stream(codec, hinfo->nid, stream_tag, format); } static struct hda_pcm_stream intel_hdmi_pcm_playback = { .substreams = 1, .channels_min = 2, .ops = { + .open = hdmi_pcm_open, .prepare = intel_hdmi_playback_pcm_prepare, - .cleanup = intel_hdmi_playback_pcm_cleanup, }, }; diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c index b0652acee9b2..baa108b9d6aa 100644 --- a/sound/pci/hda/patch_nvhdmi.c +++ b/sound/pci/hda/patch_nvhdmi.c @@ -84,7 +84,7 @@ static struct hda_verb nvhdmi_basic_init_7x[] = { #else /* support all rates and formats */ #define SUPPORTED_RATES \ - (SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ + (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 |\ SNDRV_PCM_RATE_192000) #define SUPPORTED_MAXBPS 24 @@ -202,8 +202,7 @@ static int nvhdmi_dig_playback_pcm_prepare_8ch_89(struct hda_pcm_stream *hinfo, hdmi_setup_audio_infoframe(codec, hinfo->nid, substream); - hdmi_setup_stream(codec, hinfo->nid, stream_tag, format); - return 0; + return hdmi_setup_stream(codec, hinfo->nid, stream_tag, format); } static int nvhdmi_dig_playback_pcm_prepare_8ch(struct hda_pcm_stream *hinfo, @@ -327,13 +326,6 @@ static int nvhdmi_dig_playback_pcm_prepare_8ch(struct hda_pcm_stream *hinfo, return 0; } -static int nvhdmi_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - return 0; -} - static int nvhdmi_dig_playback_pcm_prepare_2ch(struct hda_pcm_stream *hinfo, struct hda_codec *codec, unsigned int stream_tag, @@ -348,12 +340,9 @@ static int nvhdmi_dig_playback_pcm_prepare_2ch(struct hda_pcm_stream *hinfo, static struct hda_pcm_stream nvhdmi_pcm_digital_playback_8ch_89 = { .substreams = 1, .channels_min = 2, - .rates = SUPPORTED_RATES, - .maxbps = SUPPORTED_MAXBPS, - .formats = SUPPORTED_FORMATS, .ops = { + .open = hdmi_pcm_open, .prepare = nvhdmi_dig_playback_pcm_prepare_8ch_89, - .cleanup = nvhdmi_playback_pcm_cleanup, }, }; @@ -541,26 +530,32 @@ static int patch_nvhdmi_2ch(struct hda_codec *codec) * patch entries */ static struct hda_codec_preset snd_hda_preset_nvhdmi[] = { - { .id = 0x10de0002, .name = "MCP77/78 HDMI", - .patch = patch_nvhdmi_8ch_7x }, - { .id = 0x10de0003, .name = "MCP77/78 HDMI", - .patch = patch_nvhdmi_8ch_7x }, - { .id = 0x10de0005, .name = "MCP77/78 HDMI", - .patch = patch_nvhdmi_8ch_7x }, - { .id = 0x10de0006, .name = "MCP77/78 HDMI", - .patch = patch_nvhdmi_8ch_7x }, - { .id = 0x10de0007, .name = "MCP79/7A HDMI", - .patch = patch_nvhdmi_8ch_7x }, - { .id = 0x10de000a, .name = "GT220 HDMI", - .patch = patch_nvhdmi_8ch_89 }, - { .id = 0x10de000b, .name = "GT21x HDMI", - .patch = patch_nvhdmi_8ch_89 }, - { .id = 0x10de000c, .name = "MCP89 HDMI", - .patch = patch_nvhdmi_8ch_89 }, - { .id = 0x10de000d, .name = "GT240 HDMI", - .patch = patch_nvhdmi_8ch_89 }, - { .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch }, - { .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi_2ch }, + { .id = 0x10de0002, .name = "MCP77/78 HDMI", .patch = patch_nvhdmi_8ch_7x }, + { .id = 0x10de0003, .name = "MCP77/78 HDMI", .patch = patch_nvhdmi_8ch_7x }, + { .id = 0x10de0005, .name = "MCP77/78 HDMI", .patch = patch_nvhdmi_8ch_7x }, + { .id = 0x10de0006, .name = "MCP77/78 HDMI", .patch = patch_nvhdmi_8ch_7x }, + { .id = 0x10de0007, .name = "MCP79/7A HDMI", .patch = patch_nvhdmi_8ch_7x }, + { .id = 0x10de000a, .name = "GPU 0a HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de000b, .name = "GPU 0b HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de000c, .name = "MCP89 HDMI", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de000d, .name = "GPU 0d HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0010, .name = "GPU 10 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0011, .name = "GPU 11 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0012, .name = "GPU 12 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0013, .name = "GPU 13 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0014, .name = "GPU 14 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0018, .name = "GPU 18 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0019, .name = "GPU 19 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de001a, .name = "GPU 1a HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de001b, .name = "GPU 1b HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de001c, .name = "GPU 1c HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0040, .name = "GPU 40 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0041, .name = "GPU 41 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0042, .name = "GPU 42 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0043, .name = "GPU 43 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0044, .name = "GPU 44 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch }, + { .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi_2ch }, {} /* terminator */ }; @@ -573,6 +568,21 @@ MODULE_ALIAS("snd-hda-codec-id:10de000a"); MODULE_ALIAS("snd-hda-codec-id:10de000b"); MODULE_ALIAS("snd-hda-codec-id:10de000c"); MODULE_ALIAS("snd-hda-codec-id:10de000d"); +MODULE_ALIAS("snd-hda-codec-id:10de0010"); +MODULE_ALIAS("snd-hda-codec-id:10de0011"); +MODULE_ALIAS("snd-hda-codec-id:10de0012"); +MODULE_ALIAS("snd-hda-codec-id:10de0013"); +MODULE_ALIAS("snd-hda-codec-id:10de0014"); +MODULE_ALIAS("snd-hda-codec-id:10de0018"); +MODULE_ALIAS("snd-hda-codec-id:10de0019"); +MODULE_ALIAS("snd-hda-codec-id:10de001a"); +MODULE_ALIAS("snd-hda-codec-id:10de001b"); +MODULE_ALIAS("snd-hda-codec-id:10de001c"); +MODULE_ALIAS("snd-hda-codec-id:10de0040"); +MODULE_ALIAS("snd-hda-codec-id:10de0041"); +MODULE_ALIAS("snd-hda-codec-id:10de0042"); +MODULE_ALIAS("snd-hda-codec-id:10de0043"); +MODULE_ALIAS("snd-hda-codec-id:10de0044"); MODULE_ALIAS("snd-hda-codec-id:10de0067"); MODULE_ALIAS("snd-hda-codec-id:10de8001"); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 596ea2f12cf6..a432e6efd19b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -137,6 +137,7 @@ enum { ALC269VB_DMIC, ALC269_FUJITSU, ALC269_LIFEBOOK, + ALC271_ACER, ALC269_AUTO, ALC269_MODEL_LAST /* last tag */ }; @@ -256,6 +257,13 @@ enum { ALC882_MODEL_LAST, }; +/* ALC680 models */ +enum { + ALC680_BASE, + ALC680_AUTO, + ALC680_MODEL_LAST, +}; + /* for GPIO Poll */ #define GPIO_MASK 0x03 @@ -326,6 +334,12 @@ struct alc_spec { hda_nid_t *capsrc_nids; hda_nid_t dig_in_nid; /* digital-in NID; optional */ + /* capture setup for dynamic dual-adc switch */ + unsigned int cur_adc_idx; + hda_nid_t cur_adc; + unsigned int cur_adc_stream_tag; + unsigned int cur_adc_format; + /* capture source */ unsigned int num_mux_defs; const struct hda_input_mux *input_mux; @@ -367,6 +381,7 @@ struct alc_spec { /* other flags */ unsigned int no_analog :1; /* digital I/O only */ + unsigned int dual_adc_switch:1; /* switch ADCs (for ALC275) */ int init_amp; /* for virtual master */ @@ -833,9 +848,13 @@ static void alc_set_input_pin(struct hda_codec *codec, hda_nid_t nid, if (auto_pin_type <= AUTO_PIN_FRONT_MIC) { unsigned int pincap; + unsigned int oldval; + oldval = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); pincap = snd_hda_query_pin_caps(codec, nid); pincap = (pincap & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT; - if (pincap & AC_PINCAP_VREF_80) + /* if the default pin setup is vref50, we give it priority */ + if ((pincap & AC_PINCAP_VREF_80) && oldval != PIN_VREF50) val = PIN_VREF80; else if (pincap & AC_PINCAP_VREF_50) val = PIN_VREF50; @@ -1003,6 +1022,29 @@ static int get_connection_index(struct hda_codec *codec, hda_nid_t mux, return -1; } +/* switch the current ADC according to the jack state */ +static void alc_dual_mic_adc_auto_switch(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + unsigned int present; + hda_nid_t new_adc; + + present = snd_hda_jack_detect(codec, spec->ext_mic.pin); + if (present) + spec->cur_adc_idx = 1; + else + spec->cur_adc_idx = 0; + new_adc = spec->adc_nids[spec->cur_adc_idx]; + if (spec->cur_adc && spec->cur_adc != new_adc) { + /* stream is running, let's swap the current ADC */ + __snd_hda_codec_cleanup_stream(codec, spec->cur_adc, 1); + spec->cur_adc = new_adc; + snd_hda_codec_setup_stream(codec, new_adc, + spec->cur_adc_stream_tag, 0, + spec->cur_adc_format); + } +} + static void alc_mic_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -1017,6 +1059,11 @@ static void alc_mic_automute(struct hda_codec *codec) if (snd_BUG_ON(!spec->adc_nids)) return; + if (spec->dual_adc_switch) { + alc_dual_mic_adc_auto_switch(codec); + return; + } + cap_nid = spec->capsrc_nids ? spec->capsrc_nids[0] : spec->adc_nids[0]; present = snd_hda_jack_detect(codec, spec->ext_mic.pin); @@ -1499,6 +1546,73 @@ static int alc_read_coef_idx(struct hda_codec *codec, return val; } +/* set right pin controls for digital I/O */ +static void alc_auto_init_digital(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + int i; + hda_nid_t pin; + + for (i = 0; i < spec->autocfg.dig_outs; i++) { + pin = spec->autocfg.dig_out_pins[i]; + if (pin) { + snd_hda_codec_write(codec, pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + PIN_OUT); + } + } + pin = spec->autocfg.dig_in_pin; + if (pin) + snd_hda_codec_write(codec, pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + PIN_IN); +} + +/* parse digital I/Os and set up NIDs in BIOS auto-parse mode */ +static void alc_auto_parse_digital(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + int i, err; + hda_nid_t dig_nid; + + /* support multiple SPDIFs; the secondary is set up as a slave */ + for (i = 0; i < spec->autocfg.dig_outs; i++) { + err = snd_hda_get_connections(codec, + spec->autocfg.dig_out_pins[i], + &dig_nid, 1); + if (err < 0) + continue; + if (!i) { + spec->multiout.dig_out_nid = dig_nid; + spec->dig_out_type = spec->autocfg.dig_out_type[0]; + } else { + spec->multiout.slave_dig_outs = spec->slave_dig_outs; + if (i >= ARRAY_SIZE(spec->slave_dig_outs) - 1) + break; + spec->slave_dig_outs[i - 1] = dig_nid; + } + } + + if (spec->autocfg.dig_in_pin) { + dig_nid = codec->start_nid; + for (i = 0; i < codec->num_nodes; i++, dig_nid++) { + unsigned int wcaps = get_wcaps(codec, dig_nid); + if (get_wcaps_type(wcaps) != AC_WID_AUD_IN) + continue; + if (!(wcaps & AC_WCAP_DIGITAL)) + continue; + if (!(wcaps & AC_WCAP_CONN_LIST)) + continue; + err = get_connection_index(codec, dig_nid, + spec->autocfg.dig_in_pin); + if (err >= 0) { + spec->dig_in_nid = dig_nid; + break; + } + } + } +} + /* * ALC888 */ @@ -3607,6 +3721,41 @@ static int alc880_alt_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, return 0; } +/* analog capture with dynamic dual-adc changes */ +static int dualmic_capture_pcm_prepare(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) +{ + struct alc_spec *spec = codec->spec; + spec->cur_adc = spec->adc_nids[spec->cur_adc_idx]; + spec->cur_adc_stream_tag = stream_tag; + spec->cur_adc_format = format; + snd_hda_codec_setup_stream(codec, spec->cur_adc, stream_tag, 0, format); + return 0; +} + +static int dualmic_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct alc_spec *spec = codec->spec; + snd_hda_codec_cleanup_stream(codec, spec->cur_adc); + spec->cur_adc = 0; + return 0; +} + +static struct hda_pcm_stream dualmic_pcm_analog_capture = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + .nid = 0, /* fill later */ + .ops = { + .prepare = dualmic_capture_pcm_prepare, + .cleanup = dualmic_capture_pcm_cleanup + }, +}; /* */ @@ -4936,7 +5085,7 @@ static void alc880_auto_init_input_src(struct hda_codec *codec) static int alc880_parse_auto_config(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - int i, err; + int err; static hda_nid_t alc880_ignore[] = { 0x1d, 0 }; err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, @@ -4967,25 +5116,7 @@ static int alc880_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - /* check multiple SPDIF-out (for recent codecs) */ - for (i = 0; i < spec->autocfg.dig_outs; i++) { - hda_nid_t dig_nid; - err = snd_hda_get_connections(codec, - spec->autocfg.dig_out_pins[i], - &dig_nid, 1); - if (err < 0) - continue; - if (!i) - spec->multiout.dig_out_nid = dig_nid; - else { - spec->multiout.slave_dig_outs = spec->slave_dig_outs; - if (i >= ARRAY_SIZE(spec->slave_dig_outs) - 1) - break; - spec->slave_dig_outs[i - 1] = dig_nid; - } - } - if (spec->autocfg.dig_in_pin) - spec->dig_in_nid = ALC880_DIGIN_NID; + alc_auto_parse_digital(codec); if (spec->kctls.list) add_mixer(spec, spec->kctls.list); @@ -5008,6 +5139,7 @@ static void alc880_auto_init(struct hda_codec *codec) alc880_auto_init_extra_out(codec); alc880_auto_init_analog_input(codec); alc880_auto_init_input_src(codec); + alc_auto_init_digital(codec); if (spec->unsol_event) alc_inithook(codec); } @@ -5045,6 +5177,39 @@ static void fixup_automic_adc(struct hda_codec *codec) spec->auto_mic = 0; /* disable auto-mic to be sure */ } +/* select or unmute the given capsrc route */ +static void select_or_unmute_capsrc(struct hda_codec *codec, hda_nid_t cap, + int idx) +{ + if (get_wcaps_type(get_wcaps(codec, cap)) == AC_WID_AUD_MIX) { + snd_hda_codec_amp_stereo(codec, cap, HDA_INPUT, idx, + HDA_AMP_MUTE, 0); + } else { + snd_hda_codec_write_cache(codec, cap, 0, + AC_VERB_SET_CONNECT_SEL, idx); + } +} + +/* set the default connection to that pin */ +static int init_capsrc_for_pin(struct hda_codec *codec, hda_nid_t pin) +{ + struct alc_spec *spec = codec->spec; + int i; + + for (i = 0; i < spec->num_adc_nids; i++) { + hda_nid_t cap = spec->capsrc_nids ? + spec->capsrc_nids[i] : spec->adc_nids[i]; + int idx; + + idx = get_connection_index(codec, cap, pin); + if (idx < 0) + continue; + select_or_unmute_capsrc(codec, cap, idx); + return i; /* return the found index */ + } + return -1; /* not found */ +} + /* choose the ADC/MUX containing the input pin and initialize the setup */ static void fixup_single_adc(struct hda_codec *codec) { @@ -5061,33 +5226,24 @@ static void fixup_single_adc(struct hda_codec *codec) } if (!pin) return; - - /* set the default connection to that pin */ - for (i = 0; i < spec->num_adc_nids; i++) { - hda_nid_t cap = spec->capsrc_nids ? - spec->capsrc_nids[i] : spec->adc_nids[i]; - int idx; - - idx = get_connection_index(codec, cap, pin); - if (idx < 0) - continue; + i = init_capsrc_for_pin(codec, pin); + if (i >= 0) { /* use only this ADC */ if (spec->capsrc_nids) spec->capsrc_nids += i; spec->adc_nids += i; spec->num_adc_nids = 1; - /* select or unmute this route */ - if (get_wcaps_type(get_wcaps(codec, cap)) == AC_WID_AUD_MIX) { - snd_hda_codec_amp_stereo(codec, cap, HDA_INPUT, idx, - HDA_AMP_MUTE, 0); - } else { - snd_hda_codec_write_cache(codec, cap, 0, - AC_VERB_SET_CONNECT_SEL, idx); - } - return; } } +/* initialize dual adcs */ +static void fixup_dual_adc_switch(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + init_capsrc_for_pin(codec, spec->ext_mic.pin); + init_capsrc_for_pin(codec, spec->int_mic.pin); +} + static void set_capture_mixer(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -5101,7 +5257,10 @@ static void set_capture_mixer(struct hda_codec *codec) }; if (spec->num_adc_nids > 0 && spec->num_adc_nids <= 3) { int mux = 0; - if (spec->auto_mic) + int num_adcs = spec->num_adc_nids; + if (spec->dual_adc_switch) + fixup_dual_adc_switch(codec); + else if (spec->auto_mic) fixup_automic_adc(codec); else if (spec->input_mux) { if (spec->input_mux->num_items > 1) @@ -5109,7 +5268,9 @@ static void set_capture_mixer(struct hda_codec *codec) else if (spec->input_mux->num_items == 1) fixup_single_adc(codec); } - spec->cap_mixer = caps[mux][spec->num_adc_nids - 1]; + if (spec->dual_adc_switch) + num_adcs = 1; + spec->cap_mixer = caps[mux][num_adcs - 1]; } } @@ -5183,6 +5344,8 @@ static void fillup_priv_adc_nids(struct hda_codec *codec, hda_nid_t *nids, static struct snd_pci_quirk beep_white_list[] = { SND_PCI_QUIRK(0x1043, 0x829f, "ASUS", 1), + SND_PCI_QUIRK(0x1043, 0x83ce, "EeePC", 1), + SND_PCI_QUIRK(0x8086, 0xd613, "Intel", 1), {} }; @@ -6624,6 +6787,7 @@ static void alc260_auto_init(struct hda_codec *codec) alc260_auto_init_multi_out(codec); alc260_auto_init_analog_input(codec); alc260_auto_init_input_src(codec); + alc_auto_init_digital(codec); if (spec->unsol_event) alc_inithook(codec); } @@ -6640,6 +6804,29 @@ static struct hda_amp_list alc260_loopbacks[] = { #endif /* + * Pin config fixes + */ +enum { + PINFIX_HP_DC5750, +}; + +static struct alc_pincfg alc260_hp_dc5750_pinfix[] = { + { 0x11, 0x90130110 }, /* speaker */ + { } +}; + +static const struct alc_fixup alc260_fixups[] = { + [PINFIX_HP_DC5750] = { + .pins = alc260_hp_dc5750_pinfix + }, +}; + +static struct snd_pci_quirk alc260_fixup_tbl[] = { + SND_PCI_QUIRK(0x103c, 0x280a, "HP dc5750", PINFIX_HP_DC5750), + {} +}; + +/* * ALC260 configurations */ static const char *alc260_models[ALC260_MODEL_LAST] = { @@ -6838,6 +7025,9 @@ static int patch_alc260(struct hda_codec *codec) board_config = ALC260_AUTO; } + if (board_config == ALC260_AUTO) + alc_pick_fixup(codec, alc260_fixup_tbl, alc260_fixups, 1); + if (board_config == ALC260_AUTO) { /* automatic parse from the BIOS config */ err = alc260_parse_auto_config(codec); @@ -6863,6 +7053,7 @@ static int patch_alc260(struct hda_codec *codec) spec->stream_analog_playback = &alc260_pcm_analog_playback; spec->stream_analog_capture = &alc260_pcm_analog_capture; + spec->stream_analog_alt_capture = &alc260_pcm_analog_capture; spec->stream_digital_playback = &alc260_pcm_digital_playback; spec->stream_digital_capture = &alc260_pcm_digital_capture; @@ -6883,6 +7074,9 @@ static int patch_alc260(struct hda_codec *codec) set_capture_mixer(codec); set_beep_amp(spec, 0x07, 0x05, HDA_INPUT); + if (board_config == ALC260_AUTO) + alc_pick_fixup(codec, alc260_fixup_tbl, alc260_fixups, 0); + spec->vmaster_nid = 0x08; codec->patch_ops = alc_patch_ops; @@ -7003,7 +7197,7 @@ static struct hda_input_mux alc883_lenovo_nb0763_capture_source = { .num_items = 4, .items = { { "Mic", 0x0 }, - { "iMic", 0x1 }, + { "Int Mic", 0x1 }, { "Line", 0x2 }, { "CD", 0x4 }, }, @@ -8573,8 +8767,8 @@ static struct snd_kcontrol_new alc883_lenovo_nb0763_mixer[] = { HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("iMic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("iMic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), { } /* end */ }; @@ -10265,7 +10459,8 @@ static struct alc_config_preset alc882_presets[] = { * Pin config fixes */ enum { - PINFIX_ABIT_AW9D_MAX + PINFIX_ABIT_AW9D_MAX, + PINFIX_PB_M5210, }; static struct alc_pincfg alc882_abit_aw9d_pinfix[] = { @@ -10275,13 +10470,22 @@ static struct alc_pincfg alc882_abit_aw9d_pinfix[] = { { } }; +static const struct hda_verb pb_m5210_verbs[] = { + { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50 }, + {} +}; + static const struct alc_fixup alc882_fixups[] = { [PINFIX_ABIT_AW9D_MAX] = { .pins = alc882_abit_aw9d_pinfix }, + [PINFIX_PB_M5210] = { + .verbs = pb_m5210_verbs + }, }; static struct snd_pci_quirk alc882_fixup_tbl[] = { + SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", PINFIX_PB_M5210), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX), {} }; @@ -10446,7 +10650,7 @@ static int alc882_parse_auto_config(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; static hda_nid_t alc882_ignore[] = { 0x1d, 0 }; - int i, err; + int err; err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, alc882_ignore); @@ -10476,25 +10680,7 @@ static int alc882_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - /* check multiple SPDIF-out (for recent codecs) */ - for (i = 0; i < spec->autocfg.dig_outs; i++) { - hda_nid_t dig_nid; - err = snd_hda_get_connections(codec, - spec->autocfg.dig_out_pins[i], - &dig_nid, 1); - if (err < 0) - continue; - if (!i) - spec->multiout.dig_out_nid = dig_nid; - else { - spec->multiout.slave_dig_outs = spec->slave_dig_outs; - if (i >= ARRAY_SIZE(spec->slave_dig_outs) - 1) - break; - spec->slave_dig_outs[i - 1] = dig_nid; - } - } - if (spec->autocfg.dig_in_pin) - spec->dig_in_nid = ALC880_DIGIN_NID; + alc_auto_parse_digital(codec); if (spec->kctls.list) add_mixer(spec, spec->kctls.list); @@ -10524,6 +10710,7 @@ static void alc882_auto_init(struct hda_codec *codec) alc882_auto_init_hp_out(codec); alc882_auto_init_analog_input(codec); alc882_auto_init_input_src(codec); + alc_auto_init_digital(codec); if (spec->unsol_event) alc_inithook(codec); } @@ -12054,12 +12241,7 @@ static int alc262_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; dig_only: - if (spec->autocfg.dig_outs) { - spec->multiout.dig_out_nid = ALC262_DIGOUT_NID; - spec->dig_out_type = spec->autocfg.dig_out_type[0]; - } - if (spec->autocfg.dig_in_pin) - spec->dig_in_nid = ALC262_DIGIN_NID; + alc_auto_parse_digital(codec); if (spec->kctls.list) add_mixer(spec, spec->kctls.list); @@ -12091,6 +12273,7 @@ static void alc262_auto_init(struct hda_codec *codec) alc262_auto_init_hp_out(codec); alc262_auto_init_analog_input(codec); alc262_auto_init_input_src(codec); + alc_auto_init_digital(codec); if (spec->unsol_event) alc_inithook(codec); } @@ -13024,10 +13207,14 @@ static int alc268_new_analog_output(struct alc_spec *spec, hda_nid_t nid, dac = 0x02; break; case 0x15: + case 0x1a: /* ALC259/269 only */ + case 0x1b: /* ALC259/269 only */ case 0x21: /* ALC269vb has this pin, too */ dac = 0x03; break; default: + snd_printd(KERN_WARNING "hda_codec: " + "ignoring pin 0x%x as unknown\n", nid); return 0; } if (spec->multiout.dac_nids[0] != dac && @@ -13078,7 +13265,7 @@ static int alc268_auto_create_multi_out_ctls(struct alc_spec *spec, HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT)); if (err < 0) return err; - } else { + } else if (nid) { err = alc268_new_analog_output(spec, nid, "Speaker", 0); if (err < 0) return err; @@ -13227,10 +13414,7 @@ static int alc268_parse_auto_config(struct hda_codec *codec) dig_only: /* digital only support output */ - if (spec->autocfg.dig_outs) { - spec->multiout.dig_out_nid = ALC268_DIGOUT_NID; - spec->dig_out_type = spec->autocfg.dig_out_type[0]; - } + alc_auto_parse_digital(codec); if (spec->kctls.list) add_mixer(spec, spec->kctls.list); @@ -13260,6 +13444,7 @@ static void alc268_auto_init(struct hda_codec *codec) alc268_auto_init_hp_out(codec); alc268_auto_init_mono_speaker_out(codec); alc268_auto_init_analog_input(codec); + alc_auto_init_digital(codec); if (spec->unsol_event) alc_inithook(codec); } @@ -13303,7 +13488,6 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = { SND_PCI_QUIRK(0x14c0, 0x0025, "COMPAL IFL90/JFL-92", ALC268_TOSHIBA), SND_PCI_QUIRK(0x152d, 0x0763, "Diverse (CPR2000)", ALC268_ACER), SND_PCI_QUIRK(0x152d, 0x0771, "Quanta IL1", ALC267_QUANTA_IL1), - SND_PCI_QUIRK(0x1854, 0x1775, "LG R510", ALC268_DELL), {} }; @@ -13694,6 +13878,12 @@ static struct snd_kcontrol_new alc269vb_laptop_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc269_asus_mixer[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x02, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Master Playback Switch", 0x0c, 0x0, HDA_INPUT), + { } /* end */ +}; + /* capture mixer elements */ static struct snd_kcontrol_new alc269_laptop_analog_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), @@ -13914,6 +14104,20 @@ static struct hda_verb alc269vb_laptop_amic_init_verbs[] = { {} }; +static struct hda_verb alc271_acer_dmic_verbs[] = { + {0x20, AC_VERB_SET_COEF_INDEX, 0x0d}, + {0x20, AC_VERB_SET_PROC_COEF, 0x4000}, + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x21, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT}, + {0x22, AC_VERB_SET_CONNECT_SEL, 6}, + { } +}; + /* toggle speaker-output according to the hp-jack state */ static void alc269_speaker_automute(struct hda_codec *codec) { @@ -14152,6 +14356,36 @@ static int alc269_mic2_mute_check_ps(struct hda_codec *codec, hda_nid_t nid) } #endif /* CONFIG_SND_HDA_POWER_SAVE */ +static int alc275_setup_dual_adc(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + if (codec->vendor_id != 0x10ec0275 || !spec->auto_mic) + return 0; + if ((spec->ext_mic.pin >= 0x18 && spec->int_mic.pin <= 0x13) || + (spec->ext_mic.pin <= 0x12 && spec->int_mic.pin >= 0x18)) { + if (spec->ext_mic.pin <= 0x12) { + spec->private_adc_nids[0] = 0x08; + spec->private_adc_nids[1] = 0x11; + spec->private_capsrc_nids[0] = 0x23; + spec->private_capsrc_nids[1] = 0x22; + } else { + spec->private_adc_nids[0] = 0x11; + spec->private_adc_nids[1] = 0x08; + spec->private_capsrc_nids[0] = 0x22; + spec->private_capsrc_nids[1] = 0x23; + } + spec->adc_nids = spec->private_adc_nids; + spec->capsrc_nids = spec->private_capsrc_nids; + spec->num_adc_nids = 2; + spec->dual_adc_switch = 1; + snd_printdd("realtek: enabling dual ADC switchg (%02x:%02x)\n", + spec->adc_nids[0], spec->adc_nids[1]); + return 1; + } + return 0; +} + /* * BIOS auto configuration */ @@ -14175,8 +14409,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_outs) - spec->multiout.dig_out_nid = ALC269_DIGOUT_NID; + alc_auto_parse_digital(codec); if (spec->kctls.list) add_mixer(spec, spec->kctls.list); @@ -14191,13 +14424,15 @@ static int alc269_parse_auto_config(struct hda_codec *codec) spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux[0]; - fillup_priv_adc_nids(codec, alc269_adc_candidates, - sizeof(alc269_adc_candidates)); + + if (!alc275_setup_dual_adc(codec)) + fillup_priv_adc_nids(codec, alc269_adc_candidates, + sizeof(alc269_adc_candidates)); /* set default input source */ - snd_hda_codec_write_cache(codec, spec->capsrc_nids[0], - 0, AC_VERB_SET_CONNECT_SEL, - spec->input_mux->items[0].index); + if (!spec->dual_adc_switch) + select_or_unmute_capsrc(codec, spec->capsrc_nids[0], + spec->input_mux->items[0].index); err = alc_auto_add_mic_boost(codec); if (err < 0) @@ -14221,12 +14456,14 @@ static void alc269_auto_init(struct hda_codec *codec) alc269_auto_init_multi_out(codec); alc269_auto_init_hp_out(codec); alc269_auto_init_analog_input(codec); + alc_auto_init_digital(codec); if (spec->unsol_event) alc_inithook(codec); } enum { ALC269_FIXUP_SONY_VAIO, + ALC269_FIXUP_DELL_M101Z, }; static const struct hda_verb alc269_sony_vaio_fixup_verbs[] = { @@ -14238,10 +14475,20 @@ static const struct alc_fixup alc269_fixups[] = { [ALC269_FIXUP_SONY_VAIO] = { .verbs = alc269_sony_vaio_fixup_verbs }, + [ALC269_FIXUP_DELL_M101Z] = { + .verbs = (const struct hda_verb[]) { + /* Enables internal speaker */ + {0x20, AC_VERB_SET_COEF_INDEX, 13}, + {0x20, AC_VERB_SET_PROC_COEF, 0x4040}, + {} + } + }, }; static struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x104d, 0x9071, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), + SND_PCI_QUIRK(0x104d, 0x9077, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), + SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), {} }; @@ -14261,6 +14508,7 @@ static const char *alc269_models[ALC269_MODEL_LAST] = { static struct snd_pci_quirk alc269_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_QUANTA_FL1), + SND_PCI_QUIRK(0x1025, 0x047c, "ACER ZGA", ALC271_ACER), SND_PCI_QUIRK(0x1043, 0x8330, "ASUS Eeepc P703 P900A", ALC269_AMIC), SND_PCI_QUIRK(0x1043, 0x1013, "ASUS N61Da", ALC269VB_AMIC), @@ -14422,6 +14670,23 @@ static struct alc_config_preset alc269_presets[] = { .unsol_event = alc269_lifebook_unsol_event, .init_hook = alc269_lifebook_init_hook, }, + [ALC271_ACER] = { + .mixers = { alc269_asus_mixer }, + .cap_mixer = alc269vb_laptop_digital_capture_mixer, + .init_verbs = { alc269_init_verbs, alc271_acer_dmic_verbs }, + .num_dacs = ARRAY_SIZE(alc269_dac_nids), + .dac_nids = alc269_dac_nids, + .adc_nids = alc262_dmic_adc_nids, + .num_adc_nids = ARRAY_SIZE(alc262_dmic_adc_nids), + .capsrc_nids = alc262_dmic_capsrc_nids, + .num_channel_mode = ARRAY_SIZE(alc269_modes), + .channel_mode = alc269_modes, + .input_mux = &alc269_capture_source, + .dig_out_nid = ALC880_DIGOUT_NID, + .unsol_event = alc_sku_unsol_event, + .setup = alc269vb_laptop_dmic_setup, + .init_hook = alc_inithook, + }, }; static int patch_alc269(struct hda_codec *codec) @@ -14493,6 +14758,10 @@ static int patch_alc269(struct hda_codec *codec) */ spec->stream_analog_playback = &alc269_44k_pcm_analog_playback; spec->stream_analog_capture = &alc269_44k_pcm_analog_capture; + } else if (spec->dual_adc_switch) { + spec->stream_analog_playback = &alc269_pcm_analog_playback; + /* switch ADC dynamically */ + spec->stream_analog_capture = &dualmic_pcm_analog_capture; } else { spec->stream_analog_playback = &alc269_pcm_analog_playback; spec->stream_analog_capture = &alc269_pcm_analog_capture; @@ -15378,8 +15647,7 @@ static int alc861_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_outs) - spec->multiout.dig_out_nid = ALC861_DIGOUT_NID; + alc_auto_parse_digital(codec); if (spec->kctls.list) add_mixer(spec, spec->kctls.list); @@ -15405,6 +15673,7 @@ static void alc861_auto_init(struct hda_codec *codec) alc861_auto_init_multi_out(codec); alc861_auto_init_hp_out(codec); alc861_auto_init_analog_input(codec); + alc_auto_init_digital(codec); if (spec->unsol_event) alc_inithook(codec); } @@ -16509,8 +16778,7 @@ static int alc861vd_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_outs) - spec->multiout.dig_out_nid = ALC861VD_DIGOUT_NID; + alc_auto_parse_digital(codec); if (spec->kctls.list) add_mixer(spec, spec->kctls.list); @@ -16537,6 +16805,7 @@ static void alc861vd_auto_init(struct hda_codec *codec) alc861vd_auto_init_hp_out(codec); alc861vd_auto_init_analog_input(codec); alc861vd_auto_init_input_src(codec); + alc_auto_init_digital(codec); if (spec->unsol_event) alc_inithook(codec); } @@ -18520,7 +18789,7 @@ static void alc662_auto_set_output_and_unmute(struct hda_codec *codec, hda_nid_t dac) { int i, num; - hda_nid_t srcs[4]; + hda_nid_t srcs[HDA_MAX_CONNECTIONS]; alc_set_pin_output(codec, nid, pin_type); /* need the manual connection? */ @@ -18624,8 +18893,7 @@ static int alc662_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_outs) - spec->multiout.dig_out_nid = ALC880_DIGOUT_NID; + alc_auto_parse_digital(codec); if (spec->kctls.list) add_mixer(spec, spec->kctls.list); @@ -18635,7 +18903,7 @@ static int alc662_parse_auto_config(struct hda_codec *codec) add_verb(spec, alc662_init_verbs); if (codec->vendor_id == 0x10ec0272 || codec->vendor_id == 0x10ec0663 || - codec->vendor_id == 0x10ec0665) + codec->vendor_id == 0x10ec0665 || codec->vendor_id == 0x10ec0670) add_verb(spec, alc663_init_verbs); if (codec->vendor_id == 0x10ec0272) @@ -18662,6 +18930,7 @@ static void alc662_auto_init(struct hda_codec *codec) alc662_auto_init_hp_out(codec); alc662_auto_init_analog_input(codec); alc662_auto_init_input_src(codec); + alc_auto_init_digital(codec); if (spec->unsol_event) alc_inithook(codec); } @@ -18781,6 +19050,445 @@ static int patch_alc888(struct hda_codec *codec) } /* + * ALC680 support + */ +#define ALC680_DIGIN_NID ALC880_DIGIN_NID +#define ALC680_DIGOUT_NID ALC880_DIGOUT_NID +#define alc680_modes alc260_modes + +static hda_nid_t alc680_dac_nids[3] = { + /* Lout1, Lout2, hp */ + 0x02, 0x03, 0x04 +}; + +static hda_nid_t alc680_adc_nids[3] = { + /* ADC0-2 */ + /* DMIC, MIC, Line-in*/ + 0x07, 0x08, 0x09 +}; + +/* + * Analog capture ADC cgange + */ +static int alc680_capture_pcm_prepare(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) +{ + struct alc_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + unsigned int pre_mic, pre_line; + + pre_mic = snd_hda_jack_detect(codec, cfg->input_pins[AUTO_PIN_MIC]); + pre_line = snd_hda_jack_detect(codec, cfg->input_pins[AUTO_PIN_LINE]); + + spec->cur_adc_stream_tag = stream_tag; + spec->cur_adc_format = format; + + if (pre_mic || pre_line) { + if (pre_mic) + snd_hda_codec_setup_stream(codec, 0x08, stream_tag, 0, + format); + else + snd_hda_codec_setup_stream(codec, 0x09, stream_tag, 0, + format); + } else + snd_hda_codec_setup_stream(codec, 0x07, stream_tag, 0, format); + return 0; +} + +static int alc680_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + snd_hda_codec_cleanup_stream(codec, 0x07); + snd_hda_codec_cleanup_stream(codec, 0x08); + snd_hda_codec_cleanup_stream(codec, 0x09); + return 0; +} + +static struct hda_pcm_stream alc680_pcm_analog_auto_capture = { + .substreams = 1, /* can be overridden */ + .channels_min = 2, + .channels_max = 2, + /* NID is set in alc_build_pcms */ + .ops = { + .prepare = alc680_capture_pcm_prepare, + .cleanup = alc680_capture_pcm_cleanup + }, +}; + +static struct snd_kcontrol_new alc680_base_mixer[] = { + /* output mixer control */ + HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x4, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x16, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Int Mic Boost", 0x12, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Line In Boost", 0x19, 0, HDA_INPUT), + { } +}; + +static struct hda_bind_ctls alc680_bind_cap_vol = { + .ops = &snd_hda_bind_vol, + .values = { + HDA_COMPOSE_AMP_VAL(0x07, 3, 0, HDA_INPUT), + HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_INPUT), + HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT), + 0 + }, +}; + +static struct hda_bind_ctls alc680_bind_cap_switch = { + .ops = &snd_hda_bind_sw, + .values = { + HDA_COMPOSE_AMP_VAL(0x07, 3, 0, HDA_INPUT), + HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_INPUT), + HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT), + 0 + }, +}; + +static struct snd_kcontrol_new alc680_master_capture_mixer[] = { + HDA_BIND_VOL("Capture Volume", &alc680_bind_cap_vol), + HDA_BIND_SW("Capture Switch", &alc680_bind_cap_switch), + { } /* end */ +}; + +/* + * generic initialization of ADC, input mixers and output mixers + */ +static struct hda_verb alc680_init_verbs[] = { + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + + {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT | AC_USRSP_EN}, + + { } +}; + +/* toggle speaker-output according to the hp-jack state */ +static void alc680_base_setup(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x16; + spec->autocfg.speaker_pins[0] = 0x14; + spec->autocfg.speaker_pins[1] = 0x15; + spec->autocfg.input_pins[AUTO_PIN_MIC] = 0x18; + spec->autocfg.input_pins[AUTO_PIN_LINE] = 0x19; +} + +static void alc680_rec_autoswitch(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + unsigned int present; + hda_nid_t new_adc; + + present = snd_hda_jack_detect(codec, cfg->input_pins[AUTO_PIN_MIC]); + + new_adc = present ? 0x8 : 0x7; + __snd_hda_codec_cleanup_stream(codec, !present ? 0x8 : 0x7, 1); + snd_hda_codec_setup_stream(codec, new_adc, + spec->cur_adc_stream_tag, 0, + spec->cur_adc_format); + +} + +static void alc680_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if ((res >> 26) == ALC880_HP_EVENT) + alc_automute_amp(codec); + if ((res >> 26) == ALC880_MIC_EVENT) + alc680_rec_autoswitch(codec); +} + +static void alc680_inithook(struct hda_codec *codec) +{ + alc_automute_amp(codec); + alc680_rec_autoswitch(codec); +} + +/* create input playback/capture controls for the given pin */ +static int alc680_new_analog_output(struct alc_spec *spec, hda_nid_t nid, + const char *ctlname, int idx) +{ + hda_nid_t dac; + int err; + + switch (nid) { + case 0x14: + dac = 0x02; + break; + case 0x15: + dac = 0x03; + break; + case 0x16: + dac = 0x04; + break; + default: + return 0; + } + if (spec->multiout.dac_nids[0] != dac && + spec->multiout.dac_nids[1] != dac) { + err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, ctlname, + HDA_COMPOSE_AMP_VAL(dac, 3, idx, + HDA_OUTPUT)); + if (err < 0) + return err; + + err = add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, ctlname, + HDA_COMPOSE_AMP_VAL(nid, 3, idx, HDA_OUTPUT)); + + if (err < 0) + return err; + spec->multiout.dac_nids[spec->multiout.num_dacs++] = dac; + } + + return 0; +} + +/* add playback controls from the parsed DAC table */ +static int alc680_auto_create_multi_out_ctls(struct alc_spec *spec, + const struct auto_pin_cfg *cfg) +{ + hda_nid_t nid; + int err; + + spec->multiout.dac_nids = spec->private_dac_nids; + + nid = cfg->line_out_pins[0]; + if (nid) { + const char *name; + if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) + name = "Speaker"; + else + name = "Front"; + err = alc680_new_analog_output(spec, nid, name, 0); + if (err < 0) + return err; + } + + nid = cfg->speaker_pins[0]; + if (nid) { + err = alc680_new_analog_output(spec, nid, "Speaker", 0); + if (err < 0) + return err; + } + nid = cfg->hp_pins[0]; + if (nid) { + err = alc680_new_analog_output(spec, nid, "Headphone", 0); + if (err < 0) + return err; + } + + return 0; +} + +static void alc680_auto_set_output_and_unmute(struct hda_codec *codec, + hda_nid_t nid, int pin_type) +{ + alc_set_pin_output(codec, nid, pin_type); +} + +static void alc680_auto_init_multi_out(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + hda_nid_t nid = spec->autocfg.line_out_pins[0]; + if (nid) { + int pin_type = get_pin_type(spec->autocfg.line_out_type); + alc680_auto_set_output_and_unmute(codec, nid, pin_type); + } +} + +static void alc680_auto_init_hp_out(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + hda_nid_t pin; + + pin = spec->autocfg.hp_pins[0]; + if (pin) + alc680_auto_set_output_and_unmute(codec, pin, PIN_HP); + pin = spec->autocfg.speaker_pins[0]; + if (pin) + alc680_auto_set_output_and_unmute(codec, pin, PIN_OUT); +} + +/* pcm configuration: identical with ALC880 */ +#define alc680_pcm_analog_playback alc880_pcm_analog_playback +#define alc680_pcm_analog_capture alc880_pcm_analog_capture +#define alc680_pcm_analog_alt_capture alc880_pcm_analog_alt_capture +#define alc680_pcm_digital_playback alc880_pcm_digital_playback +#define alc680_pcm_digital_capture alc880_pcm_digital_capture + +/* + * BIOS auto configuration + */ +static int alc680_parse_auto_config(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + int err; + static hda_nid_t alc680_ignore[] = { 0 }; + + err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, + alc680_ignore); + if (err < 0) + return err; + + if (!spec->autocfg.line_outs) { + if (spec->autocfg.dig_outs || spec->autocfg.dig_in_pin) { + spec->multiout.max_channels = 2; + spec->no_analog = 1; + goto dig_only; + } + return 0; /* can't find valid BIOS pin config */ + } + err = alc680_auto_create_multi_out_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + + spec->multiout.max_channels = 2; + + dig_only: + /* digital only support output */ + alc_auto_parse_digital(codec); + if (spec->kctls.list) + add_mixer(spec, spec->kctls.list); + + add_verb(spec, alc680_init_verbs); + + err = alc_auto_add_mic_boost(codec); + if (err < 0) + return err; + + return 1; +} + +#define alc680_auto_init_analog_input alc882_auto_init_analog_input + +/* init callback for auto-configuration model -- overriding the default init */ +static void alc680_auto_init(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + alc680_auto_init_multi_out(codec); + alc680_auto_init_hp_out(codec); + alc680_auto_init_analog_input(codec); + alc_auto_init_digital(codec); + if (spec->unsol_event) + alc_inithook(codec); +} + +/* + * configuration and preset + */ +static const char *alc680_models[ALC680_MODEL_LAST] = { + [ALC680_BASE] = "base", + [ALC680_AUTO] = "auto", +}; + +static struct snd_pci_quirk alc680_cfg_tbl[] = { + SND_PCI_QUIRK(0x1043, 0x12f3, "ASUS NX90", ALC680_BASE), + {} +}; + +static struct alc_config_preset alc680_presets[] = { + [ALC680_BASE] = { + .mixers = { alc680_base_mixer }, + .cap_mixer = alc680_master_capture_mixer, + .init_verbs = { alc680_init_verbs }, + .num_dacs = ARRAY_SIZE(alc680_dac_nids), + .dac_nids = alc680_dac_nids, + .dig_out_nid = ALC680_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc680_modes), + .channel_mode = alc680_modes, + .unsol_event = alc680_unsol_event, + .setup = alc680_base_setup, + .init_hook = alc680_inithook, + + }, +}; + +static int patch_alc680(struct hda_codec *codec) +{ + struct alc_spec *spec; + int board_config; + int err; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + + board_config = snd_hda_check_board_config(codec, ALC680_MODEL_LAST, + alc680_models, + alc680_cfg_tbl); + + if (board_config < 0 || board_config >= ALC680_MODEL_LAST) { + printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", + codec->chip_name); + board_config = ALC680_AUTO; + } + + if (board_config == ALC680_AUTO) { + /* automatic parse from the BIOS config */ + err = alc680_parse_auto_config(codec); + if (err < 0) { + alc_free(codec); + return err; + } else if (!err) { + printk(KERN_INFO + "hda_codec: Cannot set up configuration " + "from BIOS. Using base mode...\n"); + board_config = ALC680_BASE; + } + } + + if (board_config != ALC680_AUTO) + setup_preset(codec, &alc680_presets[board_config]); + + spec->stream_analog_playback = &alc680_pcm_analog_playback; + spec->stream_analog_capture = &alc680_pcm_analog_auto_capture; + spec->stream_digital_playback = &alc680_pcm_digital_playback; + spec->stream_digital_capture = &alc680_pcm_digital_capture; + + if (!spec->adc_nids) { + spec->adc_nids = alc680_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(alc680_adc_nids); + } + + if (!spec->cap_mixer) + set_capture_mixer(codec); + + spec->vmaster_nid = 0x02; + + codec->patch_ops = alc_patch_ops; + if (board_config == ALC680_AUTO) + spec->init_hook = alc680_auto_init; + + return 0; +} + +/* * patch entries */ static struct hda_codec_preset snd_hda_preset_realtek[] = { @@ -18804,6 +19512,7 @@ static struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0663, .name = "ALC663", .patch = patch_alc662 }, { .id = 0x10ec0665, .name = "ALC665", .patch = patch_alc662 }, { .id = 0x10ec0670, .name = "ALC670", .patch = patch_alc662 }, + { .id = 0x10ec0680, .name = "ALC680", .patch = patch_alc680 }, { .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 }, { .id = 0x10ec0882, .name = "ALC882", .patch = patch_alc882 }, { .id = 0x10ec0883, .name = "ALC883", .patch = patch_alc882 }, diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index f1e7babd6920..95148e58026c 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -94,6 +94,7 @@ enum { STAC_92HD83XXX_PWR_REF, STAC_DELL_S14, STAC_92HD83XXX_HP, + STAC_HP_DV7_4000, STAC_92HD83XXX_MODELS }; @@ -202,6 +203,7 @@ struct sigmatel_spec { unsigned int spdif_mute: 1; unsigned int check_volume_offset:1; unsigned int auto_mic:1; + unsigned int linear_tone_beep:1; /* gpio lines */ unsigned int eapd_mask; @@ -1631,10 +1633,17 @@ static unsigned int dell_s14_pin_configs[10] = { 0x40f000f0, 0x40f000f0, }; +static unsigned int hp_dv7_4000_pin_configs[10] = { + 0x03a12050, 0x0321201f, 0x40f000f0, 0x90170110, + 0x40f000f0, 0x40f000f0, 0x90170110, 0xd5a30140, + 0x40f000f0, 0x40f000f0, +}; + static unsigned int *stac92hd83xxx_brd_tbl[STAC_92HD83XXX_MODELS] = { [STAC_92HD83XXX_REF] = ref92hd83xxx_pin_configs, [STAC_92HD83XXX_PWR_REF] = ref92hd83xxx_pin_configs, [STAC_DELL_S14] = dell_s14_pin_configs, + [STAC_HP_DV7_4000] = hp_dv7_4000_pin_configs, }; static const char *stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = { @@ -1643,6 +1652,7 @@ static const char *stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = { [STAC_92HD83XXX_PWR_REF] = "mic-ref", [STAC_DELL_S14] = "dell-s14", [STAC_92HD83XXX_HP] = "hp", + [STAC_HP_DV7_4000] = "hp-dv7-4000", }; static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { @@ -3802,7 +3812,7 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out return err; if (codec->beep) { /* IDT/STAC codecs have linear beep tone parameter */ - codec->beep->linear_tone = 1; + codec->beep->linear_tone = spec->linear_tone_beep; /* if no beep switch is available, make its own one */ caps = query_amp_caps(codec, nid, HDA_OUTPUT); if (!(caps & AC_AMPCAP_MUTE)) { @@ -5005,6 +5015,7 @@ static int patch_stac9200(struct hda_codec *codec) codec->no_trigger_sense = 1; codec->spec = spec; + spec->linear_tone_beep = 1; spec->num_pins = ARRAY_SIZE(stac9200_pin_nids); spec->pin_nids = stac9200_pin_nids; spec->board_config = snd_hda_check_board_config(codec, STAC_9200_MODELS, @@ -5068,6 +5079,7 @@ static int patch_stac925x(struct hda_codec *codec) codec->no_trigger_sense = 1; codec->spec = spec; + spec->linear_tone_beep = 1; spec->num_pins = ARRAY_SIZE(stac925x_pin_nids); spec->pin_nids = stac925x_pin_nids; @@ -5153,6 +5165,7 @@ static int patch_stac92hd73xx(struct hda_codec *codec) codec->no_trigger_sense = 1; codec->spec = spec; + spec->linear_tone_beep = 0; codec->slave_dig_outs = stac92hd73xx_slave_dig_outs; spec->num_pins = ARRAY_SIZE(stac92hd73xx_pin_nids); spec->pin_nids = stac92hd73xx_pin_nids; @@ -5300,6 +5313,7 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) codec->no_trigger_sense = 1; codec->spec = spec; + spec->linear_tone_beep = 1; codec->slave_dig_outs = stac92hd83xxx_slave_dig_outs; spec->digbeep_nid = 0x21; spec->mux_nids = stac92hd83xxx_mux_nids; @@ -5335,6 +5349,8 @@ again: case 0x111d7667: case 0x111d7668: case 0x111d7669: + case 0x111d76d1: + case 0x111d76d9: spec->num_pins = ARRAY_SIZE(stac92hd88xxx_pin_nids); spec->pin_nids = stac92hd88xxx_pin_nids; spec->mono_nid = 0; @@ -5522,6 +5538,7 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) codec->no_trigger_sense = 1; codec->spec = spec; + spec->linear_tone_beep = 0; codec->patch_ops = stac92xx_patch_ops; spec->num_pins = STAC92HD71BXX_NUM_PINS; switch (codec->vendor_id) { @@ -5779,6 +5796,7 @@ static int patch_stac922x(struct hda_codec *codec) codec->no_trigger_sense = 1; codec->spec = spec; + spec->linear_tone_beep = 1; spec->num_pins = ARRAY_SIZE(stac922x_pin_nids); spec->pin_nids = stac922x_pin_nids; spec->board_config = snd_hda_check_board_config(codec, STAC_922X_MODELS, @@ -5883,6 +5901,7 @@ static int patch_stac927x(struct hda_codec *codec) codec->no_trigger_sense = 1; codec->spec = spec; + spec->linear_tone_beep = 1; codec->slave_dig_outs = stac927x_slave_dig_outs; spec->num_pins = ARRAY_SIZE(stac927x_pin_nids); spec->pin_nids = stac927x_pin_nids; @@ -6018,6 +6037,7 @@ static int patch_stac9205(struct hda_codec *codec) codec->no_trigger_sense = 1; codec->spec = spec; + spec->linear_tone_beep = 1; spec->num_pins = ARRAY_SIZE(stac9205_pin_nids); spec->pin_nids = stac9205_pin_nids; spec->board_config = snd_hda_check_board_config(codec, STAC_9205_MODELS, @@ -6174,6 +6194,7 @@ static int patch_stac9872(struct hda_codec *codec) return -ENOMEM; codec->no_trigger_sense = 1; codec->spec = spec; + spec->linear_tone_beep = 1; spec->num_pins = ARRAY_SIZE(stac9872_pin_nids); spec->pin_nids = stac9872_pin_nids; @@ -6264,6 +6285,8 @@ static struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x111d76d4, .name = "92HD83C1C5", .patch = patch_stac92hd83xxx}, { .id = 0x111d7605, .name = "92HD81B1X5", .patch = patch_stac92hd83xxx}, { .id = 0x111d76d5, .name = "92HD81B1C5", .patch = patch_stac92hd83xxx}, + { .id = 0x111d76d1, .name = "92HD87B1/3", .patch = patch_stac92hd83xxx}, + { .id = 0x111d76d9, .name = "92HD87B2/4", .patch = patch_stac92hd83xxx}, { .id = 0x111d7666, .name = "92HD88B3", .patch = patch_stac92hd83xxx}, { .id = 0x111d7667, .name = "92HD88B1", .patch = patch_stac92hd83xxx}, { .id = 0x111d7668, .name = "92HD88B2", .patch = patch_stac92hd83xxx}, @@ -6280,6 +6303,21 @@ static struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x111d76b5, .name = "92HD71B6X", .patch = patch_stac92hd71bxx }, { .id = 0x111d76b6, .name = "92HD71B5X", .patch = patch_stac92hd71bxx }, { .id = 0x111d76b7, .name = "92HD71B5X", .patch = patch_stac92hd71bxx }, + { .id = 0x111d76c0, .name = "92HD89C3", .patch = patch_stac92hd73xx }, + { .id = 0x111d76c1, .name = "92HD89C2", .patch = patch_stac92hd73xx }, + { .id = 0x111d76c2, .name = "92HD89C1", .patch = patch_stac92hd73xx }, + { .id = 0x111d76c3, .name = "92HD89B3", .patch = patch_stac92hd73xx }, + { .id = 0x111d76c4, .name = "92HD89B2", .patch = patch_stac92hd73xx }, + { .id = 0x111d76c5, .name = "92HD89B1", .patch = patch_stac92hd73xx }, + { .id = 0x111d76c6, .name = "92HD89E3", .patch = patch_stac92hd73xx }, + { .id = 0x111d76c7, .name = "92HD89E2", .patch = patch_stac92hd73xx }, + { .id = 0x111d76c8, .name = "92HD89E1", .patch = patch_stac92hd73xx }, + { .id = 0x111d76c9, .name = "92HD89D3", .patch = patch_stac92hd73xx }, + { .id = 0x111d76ca, .name = "92HD89D2", .patch = patch_stac92hd73xx }, + { .id = 0x111d76cb, .name = "92HD89D1", .patch = patch_stac92hd73xx }, + { .id = 0x111d76cc, .name = "92HD89F3", .patch = patch_stac92hd73xx }, + { .id = 0x111d76cd, .name = "92HD89F2", .patch = patch_stac92hd73xx }, + { .id = 0x111d76ce, .name = "92HD89F1", .patch = patch_stac92hd73xx }, {} /* terminator */ }; diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 73453814e098..ae3acb2b42d1 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -552,24 +552,30 @@ static void via_auto_init_hp_out(struct hda_codec *codec) } } +static int is_smart51_pins(struct via_spec *spec, hda_nid_t pin); + static void via_auto_init_analog_input(struct hda_codec *codec) { struct via_spec *spec = codec->spec; + unsigned int ctl; int i; for (i = 0; i < AUTO_PIN_LAST; i++) { hda_nid_t nid = spec->autocfg.input_pins[i]; + if (!nid) + continue; + if (spec->smart51_enabled && is_smart51_pins(spec, nid)) + ctl = PIN_OUT; + else if (i <= AUTO_PIN_FRONT_MIC) + ctl = PIN_VREF50; + else + ctl = PIN_IN; snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - (i <= AUTO_PIN_FRONT_MIC ? - PIN_VREF50 : PIN_IN)); - + AC_VERB_SET_PIN_WIDGET_CONTROL, ctl); } } -static int is_smart51_pins(struct via_spec *spec, hda_nid_t pin); - static void set_pin_power_state(struct hda_codec *codec, hda_nid_t nid, unsigned int *affected_parm) { @@ -658,6 +664,8 @@ static void set_jack_power_state(struct hda_codec *codec) /* PW0 (19h), SW1 (18h), AOW1 (11h) */ parm = AC_PWRST_D3; set_pin_power_state(codec, 0x19, &parm); + if (spec->smart51_enabled) + parm = AC_PWRST_D0; snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, parm); snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, @@ -667,6 +675,8 @@ static void set_jack_power_state(struct hda_codec *codec) if (is_8ch) { parm = AC_PWRST_D3; set_pin_power_state(codec, 0x22, &parm); + if (spec->smart51_enabled) + parm = AC_PWRST_D0; snd_hda_codec_write(codec, 0x26, 0, AC_VERB_SET_POWER_STATE, parm); snd_hda_codec_write(codec, 0x24, 0, @@ -3915,6 +3925,13 @@ static int vt1708S_auto_fill_dac_nids(struct via_spec *spec, } } + /* for Smart 5.1, line/mic inputs double as output pins */ + if (cfg->line_outs == 1) { + spec->multiout.num_dacs = 3; + spec->multiout.dac_nids[AUTO_SEQ_SURROUND] = 0x11; + spec->multiout.dac_nids[AUTO_SEQ_CENLFE] = 0x24; + } + return 0; } @@ -3932,7 +3949,8 @@ static int vt1708S_auto_create_multi_out_ctls(struct via_spec *spec, for (i = 0; i <= AUTO_SEQ_SIDE; i++) { nid = cfg->line_out_pins[i]; - if (!nid) + /* for Smart 5.1, there are always at least six channels */ + if (!nid && i > AUTO_SEQ_CENLFE) continue; nid_vol = nid_vols[i]; diff --git a/sound/pci/ice1712/delta.c b/sound/pci/ice1712/delta.c index d216362626d0..712c1710f9a2 100644 --- a/sound/pci/ice1712/delta.c +++ b/sound/pci/ice1712/delta.c @@ -563,6 +563,7 @@ static int __devinit snd_ice1712_delta_init(struct snd_ice1712 *ice) case ICE1712_SUBDEVICE_DELTA1010E: case ICE1712_SUBDEVICE_DELTA1010LT: case ICE1712_SUBDEVICE_MEDIASTATION: + case ICE1712_SUBDEVICE_EDIROLDA2496: ice->num_total_dacs = 8; ice->num_total_adcs = 8; break; @@ -635,6 +636,7 @@ static int __devinit snd_ice1712_delta_init(struct snd_ice1712 *ice) err = snd_ice1712_akm4xxx_init(ak, &akm_delta410, &akm_delta410_priv, ice); break; case ICE1712_SUBDEVICE_DELTA1010LT: + case ICE1712_SUBDEVICE_EDIROLDA2496: err = snd_ice1712_akm4xxx_init(ak, &akm_delta1010lt, &akm_delta1010lt_priv, ice); break; case ICE1712_SUBDEVICE_DELTA66: @@ -734,6 +736,7 @@ static int __devinit snd_ice1712_delta_add_controls(struct snd_ice1712 *ice) case ICE1712_SUBDEVICE_DELTA66: case ICE1712_SUBDEVICE_VX442: case ICE1712_SUBDEVICE_DELTA66E: + case ICE1712_SUBDEVICE_EDIROLDA2496: err = snd_ice1712_akm4xxx_build_controls(ice); if (err < 0) return err; @@ -813,5 +816,12 @@ struct snd_ice1712_card_info snd_ice1712_delta_cards[] __devinitdata = { .chip_init = snd_ice1712_delta_init, .build_controls = snd_ice1712_delta_add_controls, }, + { + .subvendor = ICE1712_SUBDEVICE_EDIROLDA2496, + .name = "Edirol DA2496", + .model = "da2496", + .chip_init = snd_ice1712_delta_init, + .build_controls = snd_ice1712_delta_add_controls, + }, { } /* terminator */ }; diff --git a/sound/pci/ice1712/delta.h b/sound/pci/ice1712/delta.h index f7f14df81f26..1a0ac6cd6501 100644 --- a/sound/pci/ice1712/delta.h +++ b/sound/pci/ice1712/delta.h @@ -34,7 +34,8 @@ "{MidiMan M Audio,Delta 410},"\ "{MidiMan M Audio,Audiophile 24/96},"\ "{Digigram,VX442},"\ - "{Lionstracs,Mediastation}," + "{Lionstracs,Mediastation},"\ + "{Edirol,DA2496}," #define ICE1712_SUBDEVICE_DELTA1010 0x121430d6 #define ICE1712_SUBDEVICE_DELTA1010E 0xff1430d6 @@ -47,6 +48,7 @@ #define ICE1712_SUBDEVICE_DELTA1010LT 0x12143bd6 #define ICE1712_SUBDEVICE_VX442 0x12143cd6 #define ICE1712_SUBDEVICE_MEDIASTATION 0x694c0100 +#define ICE1712_SUBDEVICE_EDIROLDA2496 0xce164010 /* entry point */ extern struct snd_ice1712_card_info snd_ice1712_delta_cards[]; diff --git a/sound/pci/ice1712/pontis.c b/sound/pci/ice1712/pontis.c index 6bc3f91b7281..cdb873f5da50 100644 --- a/sound/pci/ice1712/pontis.c +++ b/sound/pci/ice1712/pontis.c @@ -638,7 +638,7 @@ static struct snd_kcontrol_new pontis_controls[] __devinitdata = { */ static void wm_proc_regs_write(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { - struct snd_ice1712 *ice = (struct snd_ice1712 *)entry->private_data; + struct snd_ice1712 *ice = entry->private_data; char line[64]; unsigned int reg, val; mutex_lock(&ice->gpio_mutex); @@ -653,7 +653,7 @@ static void wm_proc_regs_write(struct snd_info_entry *entry, struct snd_info_buf static void wm_proc_regs_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { - struct snd_ice1712 *ice = (struct snd_ice1712 *)entry->private_data; + struct snd_ice1712 *ice = entry->private_data; int reg, val; mutex_lock(&ice->gpio_mutex); @@ -676,7 +676,7 @@ static void wm_proc_init(struct snd_ice1712 *ice) static void cs_proc_regs_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { - struct snd_ice1712 *ice = (struct snd_ice1712 *)entry->private_data; + struct snd_ice1712 *ice = entry->private_data; int reg, val; mutex_lock(&ice->gpio_mutex); diff --git a/sound/pci/ice1712/prodigy192.c b/sound/pci/ice1712/prodigy192.c index 2a8e5cd8f2d8..e36ddb94c382 100644 --- a/sound/pci/ice1712/prodigy192.c +++ b/sound/pci/ice1712/prodigy192.c @@ -654,7 +654,7 @@ static int prodigy192_ak4114_init(struct snd_ice1712 *ice) static void stac9460_proc_regs_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { - struct snd_ice1712 *ice = (struct snd_ice1712 *)entry->private_data; + struct snd_ice1712 *ice = entry->private_data; int reg, val; /* registers 0x0 - 0x14 */ for (reg = 0; reg <= 0x15; reg++) { diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 6433e65c9507..467749249576 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -1776,6 +1776,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = { }, { .subvendor = 0x1014, + .subdevice = 0x0534, + .name = "ThinkPad X31", + .type = AC97_TUNE_INV_EAPD + }, + { + .subvendor = 0x1014, .subdevice = 0x1f00, .name = "MS-9128", .type = AC97_TUNE_ALC_JACK diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index 289cb4dacfc7..98a8eb3c92f7 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -79,6 +79,7 @@ static DEFINE_PCI_DEVICE_TABLE(oxygen_ids) = { { OXYGEN_PCI_SUBID(0x13f6, 0x0001), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x13f6, 0x0010), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x13f6, 0x8788), .driver_data = MODEL_CMEDIA_REF }, + { OXYGEN_PCI_SUBID(0x13f6, 0xffff), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x147a, 0xa017), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x1a58, 0x0910), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x415a, 0x5431), .driver_data = MODEL_MERIDIAN }, @@ -505,7 +506,8 @@ static const struct oxygen_model model_generic = { PLAYBACK_2_TO_AC97_1 | CAPTURE_0_FROM_I2S_1 | CAPTURE_1_FROM_SPDIF | - CAPTURE_2_FROM_AC97_1, + CAPTURE_2_FROM_AC97_1 | + AC97_CD_INPUT, .dac_channels = 8, .dac_volume_min = 0, .dac_volume_max = 255, @@ -543,6 +545,10 @@ static int __devinit get_oxygen_model(struct oxygen *chip, chip->model.suspend = claro_suspend; chip->model.resume = claro_resume; chip->model.set_adc_params = set_ak5385_params; + chip->model.device_config = PLAYBACK_0_TO_I2S | + PLAYBACK_1_TO_SPDIF | + CAPTURE_0_FROM_I2S_2 | + CAPTURE_1_FROM_SPDIF; break; } if (id->driver_data == MODEL_MERIDIAN || diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index 6147216af744..7d5222caa0a9 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -34,6 +34,7 @@ /* CAPTURE_3_FROM_I2S_3 not implemented */ #define MIDI_OUTPUT 0x0800 #define MIDI_INPUT 0x1000 +#define AC97_CD_INPUT 0x2000 enum { CONTROL_SPDIF_PCM, @@ -155,6 +156,7 @@ void oxygen_pci_remove(struct pci_dev *pci); int oxygen_pci_suspend(struct pci_dev *pci, pm_message_t state); int oxygen_pci_resume(struct pci_dev *pci); #endif +void oxygen_pci_shutdown(struct pci_dev *pci); /* oxygen_mixer.c */ diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index fad03d64e3ad..e5ebe56fb0c5 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -308,25 +308,46 @@ static void oxygen_restore_eeprom(struct oxygen *chip, } } -static void pci_bridge_magic(void) +static void configure_pcie_bridge(struct pci_dev *pci) { - struct pci_dev *pci = NULL; + enum { PEX811X, PI7C9X110 }; + static const struct pci_device_id bridge_ids[] = { + { PCI_VDEVICE(PLX, 0x8111), .driver_data = PEX811X }, + { PCI_VDEVICE(PLX, 0x8112), .driver_data = PEX811X }, + { PCI_DEVICE(0x12d8, 0xe110), .driver_data = PI7C9X110 }, + { } + }; + struct pci_dev *bridge; + const struct pci_device_id *id; u32 tmp; - for (;;) { - /* If there is any Pericom PI7C9X110 PCI-E/PCI bridge ... */ - pci = pci_get_device(0x12d8, 0xe110, pci); - if (!pci) - break; - /* - * ... configure its secondary internal arbiter to park to - * the secondary port, instead of to the last master. - */ - if (!pci_read_config_dword(pci, 0x40, &tmp)) { - tmp |= 1; - pci_write_config_dword(pci, 0x40, tmp); - } - /* Why? Try asking C-Media. */ + if (!pci->bus || !pci->bus->self) + return; + bridge = pci->bus->self; + + id = pci_match_id(bridge_ids, bridge); + if (!id) + return; + + switch (id->driver_data) { + case PEX811X: /* PLX PEX8111/PEX8112 PCIe/PCI bridge */ + pci_read_config_dword(bridge, 0x48, &tmp); + tmp |= 1; /* enable blind prefetching */ + tmp |= 1 << 11; /* enable beacon generation */ + pci_write_config_dword(bridge, 0x48, tmp); + + pci_write_config_dword(bridge, 0x84, 0x0c); + pci_read_config_dword(bridge, 0x88, &tmp); + tmp &= ~(7 << 27); + tmp |= 2 << 27; /* set prefetch size to 128 bytes */ + pci_write_config_dword(bridge, 0x88, tmp); + break; + + case PI7C9X110: /* Pericom PI7C9X110 PCIe/PCI bridge */ + pci_read_config_dword(bridge, 0x40, &tmp); + tmp |= 1; /* park the PCI arbiter to the sound chip */ + pci_write_config_dword(bridge, 0x40, tmp); + break; } } @@ -519,16 +540,21 @@ static void oxygen_init(struct oxygen *chip) } } -static void oxygen_card_free(struct snd_card *card) +static void oxygen_shutdown(struct oxygen *chip) { - struct oxygen *chip = card->private_data; - spin_lock_irq(&chip->reg_lock); chip->interrupt_mask = 0; chip->pcm_running = 0; oxygen_write16(chip, OXYGEN_DMA_STATUS, 0); oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, 0); spin_unlock_irq(&chip->reg_lock); +} + +static void oxygen_card_free(struct snd_card *card) +{ + struct oxygen *chip = card->private_data; + + oxygen_shutdown(chip); if (chip->irq >= 0) free_irq(chip->irq, chip); flush_scheduled_work(); @@ -608,7 +634,7 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id, snd_card_set_dev(card, &pci->dev); card->private_free = oxygen_card_free; - pci_bridge_magic(); + configure_pcie_bridge(pci); oxygen_init(chip); chip->model.init(chip); @@ -778,3 +804,13 @@ int oxygen_pci_resume(struct pci_dev *pci) } EXPORT_SYMBOL(oxygen_pci_resume); #endif /* CONFIG_PM */ + +void oxygen_pci_shutdown(struct pci_dev *pci) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct oxygen *chip = card->private_data; + + oxygen_shutdown(chip); + chip->model.cleanup(chip); +} +EXPORT_SYMBOL(oxygen_pci_shutdown); diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index f375b8a27862..2849b36f5f7e 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -708,7 +708,7 @@ static int ac97_fp_rec_volume_put(struct snd_kcontrol *ctl, .private_value = ((codec) << 24) | ((stereo) << 16) | (index), \ } -static DECLARE_TLV_DB_SCALE(monitor_db_scale, -1000, 1000, 0); +static DECLARE_TLV_DB_SCALE(monitor_db_scale, -600, 600, 0); static DECLARE_TLV_DB_SCALE(ac97_db_scale, -3450, 150, 0); static DECLARE_TLV_DB_SCALE(ac97_rec_db_scale, 0, 150, 0); @@ -972,6 +972,9 @@ static int add_controls(struct oxygen *chip, if (!strcmp(template.name, "Stereo Upmixing") && chip->model.dac_channels == 2) continue; + if (!strncmp(template.name, "CD Capture ", 11) && + !(chip->model.device_config & AC97_CD_INPUT)) + continue; if (!strcmp(template.name, "Master Playback Volume") && chip->model.dac_tlv) { template.tlv.p = chip->model.dac_tlv; diff --git a/sound/pci/oxygen/oxygen_pcm.c b/sound/pci/oxygen/oxygen_pcm.c index 9dff6954c397..814667442eb0 100644 --- a/sound/pci/oxygen/oxygen_pcm.c +++ b/sound/pci/oxygen/oxygen_pcm.c @@ -56,8 +56,8 @@ static const struct snd_pcm_hardware oxygen_stereo_hardware = { .channels_max = 2, .buffer_bytes_max = BUFFER_BYTES_MAX, .period_bytes_min = PERIOD_BYTES_MIN, - .period_bytes_max = BUFFER_BYTES_MAX / 2, - .periods_min = 2, + .period_bytes_max = BUFFER_BYTES_MAX, + .periods_min = 1, .periods_max = BUFFER_BYTES_MAX / PERIOD_BYTES_MIN, }; static const struct snd_pcm_hardware oxygen_multichannel_hardware = { @@ -82,8 +82,8 @@ static const struct snd_pcm_hardware oxygen_multichannel_hardware = { .channels_max = 8, .buffer_bytes_max = BUFFER_BYTES_MAX_MULTICH, .period_bytes_min = PERIOD_BYTES_MIN, - .period_bytes_max = BUFFER_BYTES_MAX_MULTICH / 2, - .periods_min = 2, + .period_bytes_max = BUFFER_BYTES_MAX_MULTICH, + .periods_min = 1, .periods_max = BUFFER_BYTES_MAX_MULTICH / PERIOD_BYTES_MIN, }; static const struct snd_pcm_hardware oxygen_ac97_hardware = { @@ -100,8 +100,8 @@ static const struct snd_pcm_hardware oxygen_ac97_hardware = { .channels_max = 2, .buffer_bytes_max = BUFFER_BYTES_MAX, .period_bytes_min = PERIOD_BYTES_MIN, - .period_bytes_max = BUFFER_BYTES_MAX / 2, - .periods_min = 2, + .period_bytes_max = BUFFER_BYTES_MAX, + .periods_min = 1, .periods_max = BUFFER_BYTES_MAX / PERIOD_BYTES_MIN, }; diff --git a/sound/pci/oxygen/oxygen_regs.h b/sound/pci/oxygen/oxygen_regs.h index 72de159d4567..4dcd41b78258 100644 --- a/sound/pci/oxygen/oxygen_regs.h +++ b/sound/pci/oxygen/oxygen_regs.h @@ -436,13 +436,15 @@ /* OXYGEN_CHANNEL_* */ #define OXYGEN_CODEC_VERSION 0xe4 -#define OXYGEN_XCID_MASK 0x07 +#define OXYGEN_CODEC_ID_MASK 0x07 #define OXYGEN_REVISION 0xe6 -#define OXYGEN_REVISION_XPKGID_MASK 0x0007 +#define OXYGEN_PACKAGE_ID_MASK 0x0007 +#define OXYGEN_PACKAGE_ID_8786 0x0004 +#define OXYGEN_PACKAGE_ID_8787 0x0006 +#define OXYGEN_PACKAGE_ID_8788 0x0007 #define OXYGEN_REVISION_MASK 0xfff8 -#define OXYGEN_REVISION_2 0x0008 /* bit flag */ -#define OXYGEN_REVISION_8787 0x0014 /* 8 bits */ +#define OXYGEN_REVISION_2 0x0008 #define OXYGEN_OFFSIN_48K 0xe8 #define OXYGEN_OFFSBASE_48K 0xe9 diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index f03a2f2cffee..469010a8b849 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -25,9 +25,9 @@ #include "xonar.h" MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>"); -MODULE_DESCRIPTION("Asus AVx00 driver"); +MODULE_DESCRIPTION("Asus Virtuoso driver"); MODULE_LICENSE("GPL v2"); -MODULE_SUPPORTED_DEVICE("{{Asus,AV100},{Asus,AV200}}"); +MODULE_SUPPORTED_DEVICE("{{Asus,AV66},{Asus,AV100},{Asus,AV200}}"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; @@ -49,6 +49,7 @@ static DEFINE_PCI_DEVICE_TABLE(xonar_ids) = { { OXYGEN_PCI_SUBID(0x1043, 0x834f) }, { OXYGEN_PCI_SUBID(0x1043, 0x835c) }, { OXYGEN_PCI_SUBID(0x1043, 0x835d) }, + { OXYGEN_PCI_SUBID(0x1043, 0x835e) }, { OXYGEN_PCI_SUBID(0x1043, 0x838e) }, { OXYGEN_PCI_SUBID_BROKEN_EEPROM }, { } @@ -95,6 +96,7 @@ static struct pci_driver xonar_driver = { .suspend = oxygen_pci_suspend, .resume = oxygen_pci_resume, #endif + .shutdown = oxygen_pci_shutdown, }; static int __init alsa_card_xonar_init(void) diff --git a/sound/pci/oxygen/xonar_cs43xx.c b/sound/pci/oxygen/xonar_cs43xx.c index 7c4986b27f2b..aa27c31049af 100644 --- a/sound/pci/oxygen/xonar_cs43xx.c +++ b/sound/pci/oxygen/xonar_cs43xx.c @@ -367,13 +367,6 @@ static void xonar_d1_line_mic_ac97_switch(struct oxygen *chip, static const DECLARE_TLV_DB_SCALE(cs4362a_db_scale, -6000, 100, 0); -static int xonar_d1_control_filter(struct snd_kcontrol_new *template) -{ - if (!strncmp(template->name, "CD Capture ", 11)) - return 1; /* no CD input */ - return 0; -} - static int xonar_d1_mixer_init(struct oxygen *chip) { int err; @@ -391,7 +384,6 @@ static const struct oxygen_model model_xonar_d1 = { .longname = "Asus Virtuoso 100", .chip = "AV200", .init = xonar_d1_init, - .control_filter = xonar_d1_control_filter, .mixer_init = xonar_d1_mixer_init, .cleanup = xonar_d1_cleanup, .suspend = xonar_d1_suspend, diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c index ba18fb546b4f..d491fd6c0be2 100644 --- a/sound/pci/oxygen/xonar_pcm179x.c +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -132,6 +132,18 @@ * GPIO 5 <- 0 */ +/* + * Xonar HDAV1.3 Slim + * ------------------ + * + * CMI8788: + * + * GPIO 1 -> enable output + * + * TXD -> HDMI controller + * RXD <- HDMI controller + */ + #include <linux/pci.h> #include <linux/delay.h> #include <linux/mutex.h> @@ -362,7 +374,6 @@ static void xonar_st_init_common(struct oxygen *chip) { struct xonar_pcm179x *data = chip->model_data; - data->generic.anti_pop_delay = 100; data->generic.output_enable_bit = GPIO_ST_OUTPUT_ENABLE; data->dacs = chip->model.private_data ? 4 : 1; data->hp_gain_offset = 2*-18; @@ -408,6 +419,7 @@ static void xonar_st_init(struct oxygen *chip) { struct xonar_pcm179x *data = chip->model_data; + data->generic.anti_pop_delay = 100; data->has_cs2000 = 1; data->cs2000_fun_cfg_1 = CS2000_REF_CLK_DIV_1; @@ -428,6 +440,7 @@ static void xonar_stx_init(struct oxygen *chip) struct xonar_pcm179x *data = chip->model_data; xonar_st_init_i2c(chip); + data->generic.anti_pop_delay = 800; data->generic.ext_power_reg = OXYGEN_GPI_DATA; data->generic.ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK; data->generic.ext_power_bit = GPI_EXT_POWER; @@ -915,13 +928,6 @@ static int xonar_d2_control_filter(struct snd_kcontrol_new *template) return 0; } -static int xonar_st_control_filter(struct snd_kcontrol_new *template) -{ - if (!strncmp(template->name, "CD Capture ", 11)) - return 1; /* no CD input */ - return 0; -} - static int add_pcm1796_controls(struct oxygen *chip) { int err; @@ -991,7 +997,8 @@ static const struct oxygen_model model_xonar_d2 = { CAPTURE_0_FROM_I2S_2 | CAPTURE_1_FROM_SPDIF | MIDI_OUTPUT | - MIDI_INPUT, + MIDI_INPUT | + AC97_CD_INPUT, .dac_channels = 8, .dac_volume_min = 255 - 2*60, .dac_volume_max = 255, @@ -1037,7 +1044,6 @@ static const struct oxygen_model model_xonar_st = { .longname = "Asus Virtuoso 100", .chip = "AV200", .init = xonar_st_init, - .control_filter = xonar_st_control_filter, .mixer_init = xonar_st_mixer_init, .cleanup = xonar_st_cleanup, .suspend = xonar_st_suspend, @@ -1108,6 +1114,9 @@ int __devinit get_xonar_pcm179x_model(struct oxygen *chip, chip->model.resume = xonar_stx_resume; chip->model.set_dac_params = set_pcm1796_params; break; + case 0x835e: + snd_printk(KERN_ERR "the HDAV1.3 Slim is not supported\n"); + return -ENODEV; default: return -EINVAL; } diff --git a/sound/pci/oxygen/xonar_wm87x6.c b/sound/pci/oxygen/xonar_wm87x6.c index dbc4b89d74e4..200f7601276f 100644 --- a/sound/pci/oxygen/xonar_wm87x6.c +++ b/sound/pci/oxygen/xonar_wm87x6.c @@ -25,16 +25,24 @@ * SPI 0 -> WM8766 (surround, center/LFE, back) * SPI 1 -> WM8776 (front, input) * - * GPIO 4 <- headphone detect - * GPIO 6 -> route input jack to input 1/2 (1/0) - * GPIO 7 -> enable output to speakers - * GPIO 8 -> enable output to speakers + * GPIO 4 <- headphone detect, 0 = plugged + * GPIO 6 -> route input jack to mic-in (0) or line-in (1) + * GPIO 7 -> enable output to front L/R speaker channels + * GPIO 8 -> enable output to other speaker channels and front panel headphone + * + * WM8766: + * + * input 1 <- line + * input 2 <- mic + * input 3 <- front mic + * input 4 <- aux */ #include <linux/pci.h> #include <linux/delay.h> #include <sound/control.h> #include <sound/core.h> +#include <sound/jack.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/tlv.h> @@ -44,7 +52,8 @@ #define GPIO_DS_HP_DETECT 0x0010 #define GPIO_DS_INPUT_ROUTE 0x0040 -#define GPIO_DS_OUTPUT_ENABLE 0x0180 +#define GPIO_DS_OUTPUT_FRONTLR 0x0080 +#define GPIO_DS_OUTPUT_ENABLE 0x0100 #define LC_CONTROL_LIMITER 0x40000000 #define LC_CONTROL_ALC 0x20000000 @@ -53,7 +62,10 @@ struct xonar_wm87x6 { struct xonar_generic generic; u16 wm8776_regs[0x17]; u16 wm8766_regs[0x10]; + struct snd_kcontrol *line_adcmux_control; + struct snd_kcontrol *mic_adcmux_control; struct snd_kcontrol *lc_controls[13]; + struct snd_jack *hp_jack; }; static void wm8776_write(struct oxygen *chip, @@ -95,8 +107,12 @@ static void wm8766_write(struct oxygen *chip, (0 << OXYGEN_SPI_CODEC_SHIFT) | OXYGEN_SPI_CEN_LATCH_CLOCK_LO, (reg << 9) | value); - if (reg < ARRAY_SIZE(data->wm8766_regs)) + if (reg < ARRAY_SIZE(data->wm8766_regs)) { + if ((reg >= WM8766_LDA1 && reg <= WM8766_RDA1) || + (reg >= WM8766_LDA2 && reg <= WM8766_MASTDA)) + value &= ~WM8766_UPDATE; data->wm8766_regs[reg] = value; + } } static void wm8766_write_cached(struct oxygen *chip, @@ -105,12 +121,8 @@ static void wm8766_write_cached(struct oxygen *chip, struct xonar_wm87x6 *data = chip->model_data; if (reg >= ARRAY_SIZE(data->wm8766_regs) || - value != data->wm8766_regs[reg]) { - if ((reg >= WM8766_LDA1 && reg <= WM8766_RDA1) || - (reg >= WM8766_LDA2 && reg <= WM8766_MASTDA)) - value &= ~WM8766_UPDATE; + value != data->wm8766_regs[reg]) wm8766_write(chip, reg, value); - } } static void wm8776_registers_init(struct oxygen *chip) @@ -139,7 +151,10 @@ static void wm8776_registers_init(struct oxygen *chip) static void wm8766_registers_init(struct oxygen *chip) { + struct xonar_wm87x6 *data = chip->model_data; + wm8766_write(chip, WM8766_RESET, 0); + wm8766_write(chip, WM8766_DAC_CTRL, data->wm8766_regs[WM8766_DAC_CTRL]); wm8766_write(chip, WM8766_INT_CTRL, WM8766_FMT_LJUST | WM8766_IWL_24); wm8766_write(chip, WM8766_DAC_CTRL2, WM8766_ZCD | (chip->dac_mute ? WM8766_DMUTE_MASK : 0)); @@ -168,6 +183,40 @@ static void wm8776_init(struct oxygen *chip) wm8776_registers_init(chip); } +static void wm8766_init(struct oxygen *chip) +{ + struct xonar_wm87x6 *data = chip->model_data; + + data->wm8766_regs[WM8766_DAC_CTRL] = + WM8766_PL_LEFT_LEFT | WM8766_PL_RIGHT_RIGHT; + wm8766_registers_init(chip); +} + +static void xonar_ds_handle_hp_jack(struct oxygen *chip) +{ + struct xonar_wm87x6 *data = chip->model_data; + bool hp_plugged; + unsigned int reg; + + mutex_lock(&chip->mutex); + + hp_plugged = !(oxygen_read16(chip, OXYGEN_GPIO_DATA) & + GPIO_DS_HP_DETECT); + + oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, + hp_plugged ? 0 : GPIO_DS_OUTPUT_FRONTLR, + GPIO_DS_OUTPUT_FRONTLR); + + reg = data->wm8766_regs[WM8766_DAC_CTRL] & ~WM8766_MUTEALL; + if (hp_plugged) + reg |= WM8766_MUTEALL; + wm8766_write_cached(chip, WM8766_DAC_CTRL, reg); + + snd_jack_report(data->hp_jack, hp_plugged ? SND_JACK_HEADPHONE : 0); + + mutex_unlock(&chip->mutex); +} + static void xonar_ds_init(struct oxygen *chip) { struct xonar_wm87x6 *data = chip->model_data; @@ -176,16 +225,22 @@ static void xonar_ds_init(struct oxygen *chip) data->generic.output_enable_bit = GPIO_DS_OUTPUT_ENABLE; wm8776_init(chip); - wm8766_registers_init(chip); + wm8766_init(chip); - oxygen_write16_masked(chip, OXYGEN_GPIO_CONTROL, GPIO_DS_INPUT_ROUTE, - GPIO_DS_HP_DETECT | GPIO_DS_INPUT_ROUTE); + oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, + GPIO_DS_INPUT_ROUTE | GPIO_DS_OUTPUT_FRONTLR); + oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL, + GPIO_DS_HP_DETECT); oxygen_set_bits16(chip, OXYGEN_GPIO_DATA, GPIO_DS_INPUT_ROUTE); oxygen_set_bits16(chip, OXYGEN_GPIO_INTERRUPT_MASK, GPIO_DS_HP_DETECT); chip->interrupt_mask |= OXYGEN_INT_GPIO; xonar_enable_output(chip); + snd_jack_new(chip->card, "Headphone", + SND_JACK_HEADPHONE, &data->hp_jack); + xonar_ds_handle_hp_jack(chip); + snd_component_add(chip->card, "WM8776"); snd_component_add(chip->card, "WM8766"); } @@ -193,6 +248,7 @@ static void xonar_ds_init(struct oxygen *chip) static void xonar_ds_cleanup(struct oxygen *chip) { xonar_disable_output(chip); + wm8776_write(chip, WM8776_RESET, 0); } static void xonar_ds_suspend(struct oxygen *chip) @@ -205,6 +261,7 @@ static void xonar_ds_resume(struct oxygen *chip) wm8776_registers_init(chip); wm8766_registers_init(chip); xonar_enable_output(chip); + xonar_ds_handle_hp_jack(chip); } static void wm8776_adc_hardware_filter(unsigned int channel, @@ -320,12 +377,27 @@ static void update_wm87x6_mute(struct oxygen *chip) (chip->dac_mute ? WM8766_DMUTE_MASK : 0)); } -static void xonar_ds_gpio_changed(struct oxygen *chip) +static void update_wm8766_center_lfe_mix(struct oxygen *chip, bool mixed) { - u16 bits; + struct xonar_wm87x6 *data = chip->model_data; + unsigned int reg; - bits = oxygen_read16(chip, OXYGEN_GPIO_DATA); - snd_printk(KERN_INFO "HP detect: %d\n", !!(bits & GPIO_DS_HP_DETECT)); + /* + * The WM8766 can mix left and right channels, but this setting + * applies to all three stereo pairs. + */ + reg = data->wm8766_regs[WM8766_DAC_CTRL] & + ~(WM8766_PL_LEFT_MASK | WM8766_PL_RIGHT_MASK); + if (mixed) + reg |= WM8766_PL_LEFT_LRMIX | WM8766_PL_RIGHT_LRMIX; + else + reg |= WM8766_PL_LEFT_LEFT | WM8766_PL_RIGHT_RIGHT; + wm8766_write_cached(chip, WM8766_DAC_CTRL, reg); +} + +static void xonar_ds_gpio_changed(struct oxygen *chip) +{ + xonar_ds_handle_hp_jack(chip); } static int wm8776_bit_switch_get(struct snd_kcontrol *ctl, @@ -603,6 +675,7 @@ static int wm8776_input_mux_put(struct snd_kcontrol *ctl, { struct oxygen *chip = ctl->private_data; struct xonar_wm87x6 *data = chip->model_data; + struct snd_kcontrol *other_ctl; unsigned int mux_bit = ctl->private_value; u16 reg; int changed; @@ -610,8 +683,18 @@ static int wm8776_input_mux_put(struct snd_kcontrol *ctl, mutex_lock(&chip->mutex); reg = data->wm8776_regs[WM8776_ADCMUX]; if (value->value.integer.value[0]) { - reg &= ~0x003; reg |= mux_bit; + /* line-in and mic-in are exclusive */ + mux_bit ^= 3; + if (reg & mux_bit) { + reg &= ~mux_bit; + if (mux_bit == 1) + other_ctl = data->line_adcmux_control; + else + other_ctl = data->mic_adcmux_control; + snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, + &other_ctl->id); + } } else reg &= ~mux_bit; changed = reg != data->wm8776_regs[WM8776_ADCMUX]; @@ -882,7 +965,10 @@ static const struct snd_kcontrol_new ds_controls[] = { .put = wm8776_input_mux_put, .private_value = 1 << 1, }, - WM8776_BIT_SWITCH("Aux", WM8776_ADCMUX, 1 << 2, 0, 0), + WM8776_BIT_SWITCH("Front Mic Capture Switch", + WM8776_ADCMUX, 1 << 2, 0, 0), + WM8776_BIT_SWITCH("Aux Capture Switch", + WM8776_ADCMUX, 1 << 3, 0, 0), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "ADC Filter Capture Enum", @@ -942,13 +1028,6 @@ static const struct snd_kcontrol_new lc_controls[] = { LC_CONTROL_ALC, wm8776_ngth_db_scale), }; -static int xonar_ds_control_filter(struct snd_kcontrol_new *template) -{ - if (!strncmp(template->name, "CD Capture ", 11)) - return 1; /* no CD input */ - return 0; -} - static int xonar_ds_mixer_init(struct oxygen *chip) { struct xonar_wm87x6 *data = chip->model_data; @@ -963,7 +1042,13 @@ static int xonar_ds_mixer_init(struct oxygen *chip) err = snd_ctl_add(chip->card, ctl); if (err < 0) return err; + if (!strcmp(ctl->id.name, "Line Capture Switch")) + data->line_adcmux_control = ctl; + else if (!strcmp(ctl->id.name, "Mic Capture Switch")) + data->mic_adcmux_control = ctl; } + if (!data->line_adcmux_control || !data->mic_adcmux_control) + return -ENXIO; BUILD_BUG_ON(ARRAY_SIZE(lc_controls) != ARRAY_SIZE(data->lc_controls)); for (i = 0; i < ARRAY_SIZE(lc_controls); ++i) { ctl = snd_ctl_new1(&lc_controls[i], chip); @@ -979,10 +1064,9 @@ static int xonar_ds_mixer_init(struct oxygen *chip) static const struct oxygen_model model_xonar_ds = { .shortname = "Xonar DS", - .longname = "Asus Virtuoso 200", + .longname = "Asus Virtuoso 66", .chip = "AV200", .init = xonar_ds_init, - .control_filter = xonar_ds_control_filter, .mixer_init = xonar_ds_mixer_init, .cleanup = xonar_ds_cleanup, .suspend = xonar_ds_suspend, @@ -993,6 +1077,7 @@ static const struct oxygen_model model_xonar_ds = { .set_adc_params = set_wm8776_adc_params, .update_dac_volume = update_wm87x6_volume, .update_dac_mute = update_wm87x6_mute, + .update_center_lfe_mix = update_wm8766_center_lfe_mix, .gpio_changed = xonar_ds_gpio_changed, .dac_tlv = wm87x6_dac_db_scale, .model_data_size = sizeof(struct xonar_wm87x6), diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index ad4462677615..ad5202efd7a9 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -97,6 +97,7 @@ #include <linux/gameport.h> #include <linux/device.h> #include <linux/firmware.h> +#include <linux/kernel.h> #include <asm/io.h> #include <sound/core.h> #include <sound/info.h> @@ -667,13 +668,12 @@ static u32 atoh(const unsigned char *in, unsigned int len) unsigned char c; while (len) { + int value; + c = in[len - 1]; - if ((c >= '0') && (c <= '9')) - sum += mult * (c - '0'); - else if ((c >= 'A') && (c <= 'F')) - sum += mult * (c - ('A' - 10)); - else if ((c >= 'a') && (c <= 'f')) - sum += mult * (c - ('a' - 10)); + value = hex_to_bin(c); + if (value >= 0) + sum += mult * value; mult *= 16; --len; } @@ -1224,15 +1224,14 @@ static int try_to_load_firmware(struct cmdif *cif, struct snd_riptide *chip) firmware.firmware.ASIC, firmware.firmware.CODEC, firmware.firmware.AUXDSP, firmware.firmware.PROG); + if (!chip) + return 1; + for (i = 0; i < FIRMWARE_VERSIONS; i++) { if (!memcmp(&firmware_versions[i], &firmware, sizeof(firmware))) - break; - } - if (i >= FIRMWARE_VERSIONS) - return 0; /* no match */ + return 1; /* OK */ - if (!chip) - return 1; /* OK */ + } snd_printdd("Writing Firmware\n"); if (!chip->fw_entry) { @@ -1615,7 +1614,10 @@ static int snd_riptide_playback_open(struct snd_pcm_substream *substream) chip->playback_substream[sub_num] = substream; runtime->hw = snd_riptide_playback; + data = kzalloc(sizeof(struct pcmhw), GFP_KERNEL); + if (data == NULL) + return -ENOMEM; data->paths = lbus_play_paths[sub_num]; data->id = play_ids[sub_num]; data->source = play_sources[sub_num]; @@ -1635,7 +1637,10 @@ static int snd_riptide_capture_open(struct snd_pcm_substream *substream) chip->capture_substream = substream; runtime->hw = snd_riptide_capture; + data = kzalloc(sizeof(struct pcmhw), GFP_KERNEL); + if (data == NULL) + return -ENOMEM; data->paths = lbus_rec_path; data->id = PADC; data->source = ACLNK2PADC; diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index d19dc052c391..d5f5b440fc40 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -1527,14 +1527,14 @@ snd_rme96_free(void *private_data) static void snd_rme96_free_spdif_pcm(struct snd_pcm *pcm) { - struct rme96 *rme96 = (struct rme96 *) pcm->private_data; + struct rme96 *rme96 = pcm->private_data; rme96->spdif_pcm = NULL; } static void snd_rme96_free_adat_pcm(struct snd_pcm *pcm) { - struct rme96 *rme96 = (struct rme96 *) pcm->private_data; + struct rme96 *rme96 = pcm->private_data; rme96->adat_pcm = NULL; } @@ -1661,7 +1661,7 @@ static void snd_rme96_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { int n; - struct rme96 *rme96 = (struct rme96 *)entry->private_data; + struct rme96 *rme96 = entry->private_data; rme96->rcreg = readl(rme96->iobase + RME96_IO_CONTROL_REGISTER); @@ -2348,7 +2348,7 @@ snd_rme96_probe(struct pci_dev *pci, if (err < 0) return err; card->private_free = snd_rme96_card_free; - rme96 = (struct rme96 *)card->private_data; + rme96 = card->private_data; rme96->card = card; rme96->pci = pci; snd_card_set_dev(card, &pci->dev); diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index b92adef8e81e..0b720cf7783e 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -3284,7 +3284,7 @@ static int snd_hdsp_create_controls(struct snd_card *card, struct hdsp *hdsp) static void snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { - struct hdsp *hdsp = (struct hdsp *) entry->private_data; + struct hdsp *hdsp = entry->private_data; unsigned int status; unsigned int status2; char *pref_sync_ref; @@ -4566,7 +4566,7 @@ static int hdsp_get_peak(struct hdsp *hdsp, struct hdsp_peak_rms __user *peak_rm static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigned int cmd, unsigned long arg) { - struct hdsp *hdsp = (struct hdsp *)hw->private_data; + struct hdsp *hdsp = hw->private_data; void __user *argp = (void __user *)arg; int err; @@ -4609,6 +4609,7 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne if (err < 0) return err; + memset(&info, 0, sizeof(info)); spin_lock_irqsave(&hdsp->lock, flags); info.pref_sync_ref = (unsigned char)hdsp_pref_sync_ref(hdsp); info.wordclock_sync_check = (unsigned char)hdsp_wc_sync_check(hdsp); @@ -5155,7 +5156,7 @@ static int snd_hdsp_free(struct hdsp *hdsp) static void snd_hdsp_card_free(struct snd_card *card) { - struct hdsp *hdsp = (struct hdsp *) card->private_data; + struct hdsp *hdsp = card->private_data; if (hdsp) snd_hdsp_free(hdsp); @@ -5181,7 +5182,7 @@ static int __devinit snd_hdsp_probe(struct pci_dev *pci, if (err < 0) return err; - hdsp = (struct hdsp *) card->private_data; + hdsp = card->private_data; card->private_free = snd_hdsp_card_free; hdsp->dev = dev; hdsp->pci = pci; diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 547b713d7204..0c98ef9156d8 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -4127,6 +4127,7 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep * hw, struct file *file, case SNDRV_HDSPM_IOCTL_GET_CONFIG_INFO: + memset(&info, 0, sizeof(info)); spin_lock_irq(&hdspm->lock); info.pref_sync_ref = hdspm_pref_sync_ref(hdspm); info.wordclock_sync_check = hdspm_wc_sync_check(hdspm); diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c index 9cc1b5aa0148..1b8f6742b5fa 100644 --- a/sound/pci/sis7019.c +++ b/sound/pci/sis7019.c @@ -264,11 +264,13 @@ static void sis_update_voice(struct voice *voice) * if using small periods. * * If we're less than 9 samples behind, we're on target. + * Otherwise, shorten the next vperiod by the amount we've + * been delayed. */ if (sync > -9) voice->vperiod = voice->sync_period_size + 1; else - voice->vperiod = voice->sync_period_size - 4; + voice->vperiod = voice->sync_period_size + sync + 10; if (voice->vperiod < voice->buffer_size) { sis_update_sso(voice, voice->vperiod); @@ -736,7 +738,7 @@ static void sis_prepare_timing_voice(struct voice *voice, period_size = buffer_size; /* Initially, we want to interrupt just a bit behind the end of - * the period we're clocking out. 10 samples seems to give a good + * the period we're clocking out. 12 samples seems to give a good * delay. * * We want to spread our interrupts throughout the virtual period, @@ -747,7 +749,7 @@ static void sis_prepare_timing_voice(struct voice *voice, * * This is all moot if we don't need to use virtual periods. */ - vperiod = runtime->period_size + 10; + vperiod = runtime->period_size + 12; if (vperiod > period_size) { u16 tail = vperiod % period_size; u16 quarter_period = period_size / 4; @@ -776,7 +778,7 @@ static void sis_prepare_timing_voice(struct voice *voice, */ timing->flags |= VOICE_SYNC_TIMING; timing->sync_base = voice->ctrl_base; - timing->sync_cso = runtime->period_size - 1; + timing->sync_cso = runtime->period_size; timing->sync_period_size = runtime->period_size; timing->sync_buffer_size = runtime->buffer_size; timing->period_size = period_size; @@ -1047,7 +1049,7 @@ static int sis_chip_free(struct sis7019 *sis) /* Reset the chip, and disable all interrputs. */ outl(SIS_GCR_SOFTWARE_RESET, sis->ioport + SIS_GCR); - udelay(10); + udelay(25); outl(0, sis->ioport + SIS_GCR); outl(0, sis->ioport + SIS_GIER); @@ -1083,7 +1085,7 @@ static int sis_chip_init(struct sis7019 *sis) /* Reset the audio controller */ outl(SIS_GCR_SOFTWARE_RESET, io + SIS_GCR); - udelay(10); + udelay(25); outl(0, io + SIS_GCR); /* Get the AC-link semaphore, and reset the codecs @@ -1096,7 +1098,7 @@ static int sis_chip_init(struct sis7019 *sis) return -EIO; outl(SIS_AC97_CMD_CODEC_COLD_RESET, io + SIS_AC97_CMD); - udelay(10); + udelay(250); count = 0xffff; while ((inw(io + SIS_AC97_STATUS) & SIS_AC97_STATUS_BUSY) && --count) diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c index 6d943f6f6b70..2870a4fdc130 100644 --- a/sound/pci/trident/trident_main.c +++ b/sound/pci/trident/trident_main.c @@ -1055,7 +1055,7 @@ static int snd_trident_capture_prepare(struct snd_pcm_substream *substream) spin_lock_irq(&trident->reg_lock); - // Initilize the channel and set channel Mode + // Initialize the channel and set channel Mode outb(0, TRID_REG(trident, LEGACY_DMAR15)); // Set DMA channel operation mode register diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 7e494b6a1d0e..8c5f8b5a59f0 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -85,6 +85,7 @@ static int joystick; static int ac97_clock = 48000; static char *ac97_quirk; static int dxs_support; +static int dxs_init_volume = 31; static int nodelay; module_param(index, int, 0444); @@ -103,6 +104,8 @@ module_param(ac97_quirk, charp, 0444); MODULE_PARM_DESC(ac97_quirk, "AC'97 workaround for strange hardware."); module_param(dxs_support, int, 0444); MODULE_PARM_DESC(dxs_support, "Support for DXS channels (0 = auto, 1 = enable, 2 = disable, 3 = 48k only, 4 = no VRA, 5 = enable any sample rate)"); +module_param(dxs_init_volume, int, 0644); +MODULE_PARM_DESC(dxs_init_volume, "initial DXS volume (0-31)"); module_param(nodelay, int, 0444); MODULE_PARM_DESC(nodelay, "Disable 500ms init delay"); @@ -1245,8 +1248,10 @@ static int snd_via8233_playback_open(struct snd_pcm_substream *substream) return err; stream = viadev->reg_offset / 0x10; if (chip->dxs_controls[stream]) { - chip->playback_volume[stream][0] = 0; - chip->playback_volume[stream][1] = 0; + chip->playback_volume[stream][0] = + VIA_DXS_MAX_VOLUME - (dxs_init_volume & 31); + chip->playback_volume[stream][1] = + VIA_DXS_MAX_VOLUME - (dxs_init_volume & 31); chip->dxs_controls[stream]->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE; snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE | diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.c b/sound/pcmcia/pdaudiocf/pdaudiocf.c index df110df52a8b..7ab9174a8a84 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf.c @@ -139,8 +139,8 @@ static int snd_pdacf_probe(struct pcmcia_device *link) pdacf->p_dev = link; link->priv = pdacf; - link->io.Attributes1 = IO_DATA_PATH_WIDTH_AUTO; - link->io.NumPorts1 = 16; + link->resource[0]->flags |= IO_DATA_PATH_WIDTH_AUTO; + link->resource[0]->end = 16; link->conf.Attributes = CONF_ENABLE_IRQ | CONF_ENABLE_PULSE_IRQ; link->conf.IntType = INT_MEMORY_AND_IO; @@ -219,7 +219,7 @@ static int pdacf_config(struct pcmcia_device *link) snd_printdd(KERN_DEBUG "pdacf_config called\n"); link->conf.ConfigIndex = 0x5; - ret = pcmcia_request_io(link, &link->io); + ret = pcmcia_request_io(link); if (ret) goto failed; @@ -231,7 +231,8 @@ static int pdacf_config(struct pcmcia_device *link) if (ret) goto failed; - if (snd_pdacf_assign_resources(pdacf, link->io.BasePort1, link->irq) < 0) + if (snd_pdacf_assign_resources(pdacf, link->resource[0]->start, + link->irq) < 0) goto failed; return 0; diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.h b/sound/pcmcia/pdaudiocf/pdaudiocf.h index a0a7ec64222a..5cc3e4573074 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf.h +++ b/sound/pcmcia/pdaudiocf/pdaudiocf.h @@ -24,7 +24,6 @@ #include <sound/pcm.h> #include <asm/io.h> #include <linux/interrupt.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/sound/pcmcia/vx/vxpocket.c b/sound/pcmcia/vx/vxpocket.c index 624b47a85f0a..a6edfc3be29a 100644 --- a/sound/pcmcia/vx/vxpocket.c +++ b/sound/pcmcia/vx/vxpocket.c @@ -159,8 +159,8 @@ static int snd_vxpocket_new(struct snd_card *card, int ibl, vxp->p_dev = link; link->priv = chip; - link->io.Attributes1 = IO_DATA_PATH_WIDTH_AUTO; - link->io.NumPorts1 = 16; + link->resource[0]->flags |= IO_DATA_PATH_WIDTH_AUTO; + link->resource[0]->end = 16; link->conf.Attributes = CONF_ENABLE_IRQ; link->conf.IntType = INT_MEMORY_AND_IO; @@ -226,7 +226,7 @@ static int vxpocket_config(struct pcmcia_device *link) strcpy(chip->card->driver, vxp440_hw.name); } - ret = pcmcia_request_io(link, &link->io); + ret = pcmcia_request_io(link); if (ret) goto failed; @@ -241,7 +241,8 @@ static int vxpocket_config(struct pcmcia_device *link) chip->dev = &link->dev; snd_card_set_dev(chip->card, chip->dev); - if (snd_vxpocket_assign_resources(chip, link->io.BasePort1, link->irq) < 0) + if (snd_vxpocket_assign_resources(chip, link->resource[0]->start, + link->irq) < 0) goto failed; return 0; diff --git a/sound/pcmcia/vx/vxpocket.h b/sound/pcmcia/vx/vxpocket.h index ea4df16a28ef..d9110669d042 100644 --- a/sound/pcmcia/vx/vxpocket.h +++ b/sound/pcmcia/vx/vxpocket.h @@ -23,7 +23,6 @@ #include <sound/vx_core.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c index 2f12da4da561..581a670e8261 100644 --- a/sound/ppc/snd_ps3.c +++ b/sound/ppc/snd_ps3.c @@ -579,7 +579,7 @@ static int snd_ps3_delay_to_bytes(struct snd_pcm_substream *substream, rate * delay_ms / 1000) * substream->runtime->channels; - pr_debug(KERN_ERR "%s: time=%d rate=%d bytes=%ld, frames=%d, ret=%d\n", + pr_debug("%s: time=%d rate=%d bytes=%ld, frames=%d, ret=%d\n", __func__, delay_ms, rate, diff --git a/sound/ppc/tumbler.c b/sound/ppc/tumbler.c index 20afdf9772ee..961d98297695 100644 --- a/sound/ppc/tumbler.c +++ b/sound/ppc/tumbler.c @@ -785,7 +785,7 @@ static int snapper_set_capture_source(struct pmac_tumbler *mix) if (! mix->i2c.client) return -ENODEV; if (mix->capture_source) - mix->acs = mix->acs |= 2; + mix->acs |= 2; else mix->acs &= ~2; return i2c_smbus_write_byte_data(mix->i2c.client, TAS_REG_ACS, mix->acs); diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index b1749bc67979..3e598e756e54 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -28,9 +28,13 @@ source "sound/soc/atmel/Kconfig" source "sound/soc/au1x/Kconfig" source "sound/soc/blackfin/Kconfig" source "sound/soc/davinci/Kconfig" +source "sound/soc/ep93xx/Kconfig" source "sound/soc/fsl/Kconfig" source "sound/soc/imx/Kconfig" +source "sound/soc/jz4740/Kconfig" +source "sound/soc/nuc900/Kconfig" source "sound/soc/omap/Kconfig" +source "sound/soc/kirkwood/Kconfig" source "sound/soc/pxa/Kconfig" source "sound/soc/s3c24xx/Kconfig" source "sound/soc/s6000/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 1470141d4167..eb183443eee4 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -6,9 +6,13 @@ obj-$(CONFIG_SND_SOC) += atmel/ obj-$(CONFIG_SND_SOC) += au1x/ obj-$(CONFIG_SND_SOC) += blackfin/ obj-$(CONFIG_SND_SOC) += davinci/ +obj-$(CONFIG_SND_SOC) += ep93xx/ obj-$(CONFIG_SND_SOC) += fsl/ obj-$(CONFIG_SND_SOC) += imx/ +obj-$(CONFIG_SND_SOC) += jz4740/ +obj-$(CONFIG_SND_SOC) += nuc900/ obj-$(CONFIG_SND_SOC) += omap/ +obj-$(CONFIG_SND_SOC) += kirkwood/ obj-$(CONFIG_SND_SOC) += pxa/ obj-$(CONFIG_SND_SOC) += s3c24xx/ obj-$(CONFIG_SND_SOC) += s6000/ diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c index f6b3cc04b34b..dc5249fba85c 100644 --- a/sound/soc/atmel/atmel-pcm.c +++ b/sound/soc/atmel/atmel-pcm.c @@ -77,7 +77,6 @@ struct atmel_runtime_data { size_t period_size; dma_addr_t period_ptr; /* physical address of next period */ - int periods; /* period index of period_ptr */ /* PDC register save */ u32 pdc_xpr_save; diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 0b59806905d1..c85844d4845b 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -549,7 +549,6 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, printk(KERN_WARNING "atmel_ssc_dai: unsupported DAI format 0x%x\n", ssc_p->daifmt); return -EINVAL; - break; } pr_debug("atmel_ssc_hw_params: " "RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n", diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index a61ccd2d505f..d14a5a91a465 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -375,12 +375,10 @@ static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev) } ret = -EBUSY; - wd->ioarea = request_mem_region(r->start, r->end - r->start + 1, - "au1xpsc_ac97"); - if (!wd->ioarea) + if (!request_mem_region(r->start, resource_size(r), pdev->name)) goto out0; - wd->mmio = ioremap(r->start, 0xffff); + wd->mmio = ioremap(r->start, resource_size(r)); if (!wd->mmio) goto out1; @@ -410,8 +408,7 @@ static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev) snd_soc_unregister_dai(&au1xpsc_ac97_dai); out1: - release_resource(wd->ioarea); - kfree(wd->ioarea); + release_mem_region(r->start, resource_size(r)); out0: kfree(wd); return ret; @@ -420,6 +417,7 @@ out0: static int __devexit au1xpsc_ac97_drvremove(struct platform_device *pdev) { struct au1xpsc_audio_data *wd = platform_get_drvdata(pdev); + struct resource *r = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (wd->dmapd) au1xpsc_pcm_destroy(wd->dmapd); @@ -433,8 +431,7 @@ static int __devexit au1xpsc_ac97_drvremove(struct platform_device *pdev) au_sync(); iounmap(wd->mmio); - release_resource(wd->ioarea); - kfree(wd->ioarea); + release_mem_region(r->start, resource_size(r)); kfree(wd); au1xpsc_ac97_workdata = NULL; /* MDEV */ diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index 24454c98d0ee..6083fe7799fa 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -321,12 +321,10 @@ static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev) } ret = -EBUSY; - wd->ioarea = request_mem_region(r->start, r->end - r->start + 1, - "au1xpsc_i2s"); - if (!wd->ioarea) + if (!request_mem_region(r->start, resource_size(r), pdev->name)) goto out0; - wd->mmio = ioremap(r->start, 0xffff); + wd->mmio = ioremap(r->start, resource_size(r)); if (!wd->mmio) goto out1; @@ -362,8 +360,7 @@ static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev) snd_soc_unregister_dai(&au1xpsc_i2s_dai); out1: - release_resource(wd->ioarea); - kfree(wd->ioarea); + release_mem_region(r->start, resource_size(r)); out0: kfree(wd); return ret; @@ -372,6 +369,7 @@ out0: static int __devexit au1xpsc_i2s_drvremove(struct platform_device *pdev) { struct au1xpsc_audio_data *wd = platform_get_drvdata(pdev); + struct resource *r = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (wd->dmapd) au1xpsc_pcm_destroy(wd->dmapd); @@ -384,8 +382,7 @@ static int __devexit au1xpsc_i2s_drvremove(struct platform_device *pdev) au_sync(); iounmap(wd->mmio); - release_resource(wd->ioarea); - kfree(wd->ioarea); + release_mem_region(r->start, resource_size(r)); kfree(wd); au1xpsc_i2s_workdata = NULL; /* MDEV */ diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h index 32d3807d3f5a..093775d4dc3e 100644 --- a/sound/soc/au1x/psc.h +++ b/sound/soc/au1x/psc.h @@ -32,7 +32,6 @@ struct au1xpsc_audio_data { unsigned long rate; unsigned long pm[2]; - struct resource *ioarea; struct mutex lock; struct platform_device *dmapd; }; diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index 8ef25025f3dc..3abeeddc67d3 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -105,13 +105,18 @@ config SND_BF5XX_RESET_GPIO_NUM Set the correct GPIO for RESET the sound chip. config SND_BF5XX_SOC_AD1980 - tristate "SoC AD1980/1 Audio support for BF5xx" + tristate "SoC AD1980/1 Audio support for BF5xx (Obsolete)" depends on SND_BF5XX_AC97 select SND_BF5XX_SOC_AC97 select SND_SOC_AD1980 help Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT. + Warning: + Because Analog Devices Inc. discontinued the ad1980 sound chip since + Sep. 2009, this ad1980 driver is not maintained, tested and supported + by ADI now. + config SND_BF5XX_SOC_SPORT tristate diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index 523b7fc33f4e..c0eba5109980 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -255,8 +255,7 @@ EXPORT_SYMBOL_GPL(soc_ac97_ops); #ifdef CONFIG_PM static int bf5xx_ac97_suspend(struct snd_soc_dai *dai) { - struct sport_device *sport = - (struct sport_device *)dai->private_data; + struct sport_device *sport = dai->private_data; pr_debug("%s : sport %d\n", __func__, dai->id); if (!dai->active) @@ -271,8 +270,7 @@ static int bf5xx_ac97_suspend(struct snd_soc_dai *dai) static int bf5xx_ac97_resume(struct snd_soc_dai *dai) { int ret; - struct sport_device *sport = - (struct sport_device *)dai->private_data; + struct sport_device *sport = dai->private_data; pr_debug("%s : sport %d\n", __func__, dai->id); if (!dai->active) diff --git a/sound/soc/blackfin/bf5xx-ad1980.c b/sound/soc/blackfin/bf5xx-ad1980.c index d8f591273778..92f7c327bb7a 100644 --- a/sound/soc/blackfin/bf5xx-ad1980.c +++ b/sound/soc/blackfin/bf5xx-ad1980.c @@ -26,6 +26,14 @@ * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA */ +/* + * WARNING: + * + * Because Analog Devices Inc. discontinued the ad1980 sound chip since + * Sep. 2009, this ad1980 driver is not maintained, tested and supported + * by ADI now. + */ + #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/device.h> @@ -109,5 +117,5 @@ module_exit(bf5xx_board_exit); /* Module information */ MODULE_AUTHOR("Cliff Cai"); -MODULE_DESCRIPTION("ALSA SoC AD1980/1 BF5xx board"); +MODULE_DESCRIPTION("ALSA SoC AD1980/1 BF5xx board (Obsolete)"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c index 4b360124083e..24c14269f4bc 100644 --- a/sound/soc/blackfin/bf5xx-tdm.c +++ b/sound/soc/blackfin/bf5xx-tdm.c @@ -210,8 +210,7 @@ static int bf5xx_tdm_set_channel_map(struct snd_soc_dai *dai, #ifdef CONFIG_PM static int bf5xx_tdm_suspend(struct snd_soc_dai *dai) { - struct sport_device *sport = - (struct sport_device *)dai->private_data; + struct sport_device *sport = dai->private_data; if (!dai->active) return 0; @@ -225,8 +224,7 @@ static int bf5xx_tdm_suspend(struct snd_soc_dai *dai) static int bf5xx_tdm_resume(struct snd_soc_dai *dai) { int ret; - struct sport_device *sport = - (struct sport_device *)dai->private_data; + struct sport_device *sport = dai->private_data; if (!dai->active) return 0; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 5da30eb6ad00..83f5c67d3c41 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -22,9 +22,11 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4642 if I2C select SND_SOC_AK4671 if I2C select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC + select SND_SOC_CS42L51 if I2C select SND_SOC_CS4270 if I2C - select SND_SOC_MAX9877 if I2C select SND_SOC_DA7210 if I2C + select SND_SOC_JZ4740 if SOC_JZ4740 + select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 select SND_SOC_SPDIF select SND_SOC_SSM2602 if I2C @@ -48,6 +50,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8727 select SND_SOC_WM8728 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8731 if SND_SOC_I2C_AND_SPI + select SND_SOC_WM8741 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8750 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8753 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8776 if SND_SOC_I2C_AND_SPI @@ -120,13 +123,13 @@ config SND_SOC_AK4671 config SND_SOC_CQ0093VC tristate +config SND_SOC_CS42L51 + tristate + # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate -config SND_SOC_DA7210 - tristate - # Cirrus Logic CS4270 Codec VD = 3.3V Errata # Select if you are affected by the errata where the part will not function # if MCLK divide-by-1.5 is selected and VD is set to 3.3V. The driver will @@ -138,9 +141,15 @@ config SND_SOC_CS4270_VD33_ERRATA config SND_SOC_CX20442 tristate +config SND_SOC_JZ4740_CODEC + tristate + config SND_SOC_L3 tristate +config SND_SOC_DA7210 + tristate + config SND_SOC_PCM3008 tristate @@ -206,6 +215,9 @@ config SND_SOC_WM8728 config SND_SOC_WM8731 tristate +config SND_SOC_WM8741 + tristate + config SND_SOC_WM8750 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 91429eab0707..53524095759c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -9,6 +9,7 @@ snd-soc-ak4535-objs := ak4535.o snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o snd-soc-cq93vc-objs := cq93vc.o +snd-soc-cs42l51-objs := cs42l51.o snd-soc-cs4270-objs := cs4270.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o @@ -34,6 +35,7 @@ snd-soc-wm8711-objs := wm8711.o snd-soc-wm8727-objs := wm8727.o snd-soc-wm8728-objs := wm8728.o snd-soc-wm8731-objs := wm8731.o +snd-soc-wm8741-objs := wm8741.o snd-soc-wm8750-objs := wm8750.o snd-soc-wm8753-objs := wm8753.o snd-soc-wm8776-objs := wm8776.o @@ -56,6 +58,7 @@ snd-soc-wm9705-objs := wm9705.o snd-soc-wm9712-objs := wm9712.o snd-soc-wm9713-objs := wm9713.o snd-soc-wm-hubs-objs := wm_hubs.o +snd-soc-jz4740-codec-objs := jz4740.o # Amp snd-soc-max9877-objs := max9877.o @@ -74,10 +77,12 @@ obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o +obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o +obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o @@ -99,6 +104,7 @@ obj-$(CONFIG_SND_SOC_WM8711) += snd-soc-wm8711.o obj-$(CONFIG_SND_SOC_WM8727) += snd-soc-wm8727.o obj-$(CONFIG_SND_SOC_WM8728) += snd-soc-wm8728.o obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o +obj-$(CONFIG_SND_SOC_WM8741) += snd-soc-wm8741.o obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o obj-$(CONFIG_SND_SOC_WM8776) += snd-soc-wm8776.o diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 217538423225..a01006c8c606 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -272,6 +272,7 @@ static int ad1836_register(struct ad1836_priv *ad1836) if (ad1836_codec) { dev_err(codec->dev, "Another ad1836 is registered\n"); + kfree(ad1836); return -EINVAL; } diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index c8ca1142b2f4..1def75e4862f 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -24,6 +24,7 @@ /* codec private data */ struct ad193x_priv { + unsigned int sysclk; struct snd_soc_codec codec; u8 reg_cache[AD193X_NUM_REGS]; }; @@ -251,15 +252,32 @@ static int ad193x_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } +static int ad193x_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec); + switch (freq) { + case 12288000: + case 18432000: + case 24576000: + case 36864000: + ad193x->sysclk = freq; + return 0; + } + return -EINVAL; +} + static int ad193x_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - int word_len = 0, reg = 0; + int word_len = 0, reg = 0, master_rate = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; + struct ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec); /* bit size */ switch (params_format(params)) { @@ -275,6 +293,25 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream, break; } + switch (ad193x->sysclk) { + case 12288000: + master_rate = AD193X_PLL_INPUT_256; + break; + case 18432000: + master_rate = AD193X_PLL_INPUT_384; + break; + case 24576000: + master_rate = AD193X_PLL_INPUT_512; + break; + case 36864000: + master_rate = AD193X_PLL_INPUT_768; + break; + } + + reg = snd_soc_read(codec, AD193X_PLL_CLK_CTRL0); + reg = (reg & AD193X_PLL_INPUT_MASK) | master_rate; + snd_soc_write(codec, AD193X_PLL_CLK_CTRL0, reg); + reg = snd_soc_read(codec, AD193X_DAC_CTRL2); reg = (reg & (~AD193X_DAC_WORD_LEN_MASK)) | word_len; snd_soc_write(codec, AD193X_DAC_CTRL2, reg); @@ -348,6 +385,7 @@ static int ad193x_bus_probe(struct device *dev, void *ctrl_data, int bus_type) /* pll input: mclki/xi */ snd_soc_write(codec, AD193X_PLL_CLK_CTRL0, 0x99); /* mclk=24.576Mhz: 0x9D; mclk=12.288Mhz: 0x99 */ snd_soc_write(codec, AD193X_PLL_CLK_CTRL1, 0x04); + ad193x->sysclk = 12288000; ret = snd_soc_register_codec(codec); if (ret != 0) { @@ -383,6 +421,7 @@ static struct snd_soc_dai_ops ad193x_dai_ops = { .hw_params = ad193x_hw_params, .digital_mute = ad193x_mute, .set_tdm_slot = ad193x_set_tdm_slot, + .set_sysclk = ad193x_set_dai_sysclk, .set_fmt = ad193x_set_dai_fmt, }; diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h index a03c880d52f9..654ba64ae04c 100644 --- a/sound/soc/codecs/ad193x.h +++ b/sound/soc/codecs/ad193x.h @@ -11,6 +11,11 @@ #define AD193X_PLL_CLK_CTRL0 0x800 #define AD193X_PLL_POWERDOWN 0x01 +#define AD193X_PLL_INPUT_MASK (~0x6) +#define AD193X_PLL_INPUT_256 (0 << 1) +#define AD193X_PLL_INPUT_384 (1 << 1) +#define AD193X_PLL_INPUT_512 (2 << 1) +#define AD193X_PLL_INPUT_768 (3 << 1) #define AD193X_PLL_CLK_CTRL1 0x801 #define AD193X_DAC_CTRL0 0x802 #define AD193X_DAC_POWERDOWN 0x01 diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 042072738cdc..70cfaec3be2c 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -11,6 +11,14 @@ * option) any later version. */ +/* + * WARNING: + * + * Because Analog Devices Inc. discontinued the ad1980 sound chip since + * Sep. 2009, this ad1980 driver is not maintained, tested and supported + * by ADI now. + */ + #include <linux/init.h> #include <linux/slab.h> #include <linux/module.h> @@ -298,6 +306,6 @@ struct snd_soc_codec_device soc_codec_dev_ad1980 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_ad1980); -MODULE_DESCRIPTION("ASoC ad1980 driver"); +MODULE_DESCRIPTION("ASoC ad1980 driver (Obsolete)"); MODULE_AUTHOR("Roy Huang, Cliff Cai"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ad1980.h b/sound/soc/codecs/ad1980.h index db6c8500d66b..538f37c90806 100644 --- a/sound/soc/codecs/ad1980.h +++ b/sound/soc/codecs/ad1980.h @@ -1,5 +1,11 @@ /* * ad1980.h -- ad1980 Soc Audio driver + * + * WARNING: + * + * Because Analog Devices Inc. discontinued the ad1980 sound chip since + * Sep. 2009, this ad1980 driver is not maintained, tested and supported + * by ADI now. */ #ifndef _AD1980_H diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 7528a54102b5..3d7dc55305ec 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -22,20 +22,13 @@ * AK4643 is tested. */ -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/init.h> #include <linux/delay.h> -#include <linux/pm.h> #include <linux/i2c.h> #include <linux/platform_device.h> #include <linux/slab.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> #include <sound/soc-dapm.h> #include <sound/initval.h> +#include <sound/tlv.h> #include "ak4642.h" @@ -111,6 +104,23 @@ struct snd_soc_codec_device soc_codec_dev_ak4642; +/* + * Playback Volume (table 39) + * + * max : 0x00 : +12.0 dB + * ( 0.5 dB step ) + * min : 0xFE : -115.0 dB + * mute: 0xFF + */ +static const DECLARE_TLV_DB_SCALE(out_tlv, -11500, 50, 1); + +static const struct snd_kcontrol_new ak4642_snd_controls[] = { + + SOC_DOUBLE_R_TLV("Digital Playback Volume", L_DVC, R_DVC, + 0, 0xFF, 1, out_tlv), +}; + + /* codec private data */ struct ak4642_priv { struct snd_soc_codec codec; @@ -204,7 +214,6 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, * * PLL, Master Mode * Audio I/F Format :MSB justified (ADC & DAC) - * Digital Volume: -8dB * Bass Boost Level : Middle * * This operation came from example code of @@ -214,8 +223,6 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, ak4642_write(codec, 0x0e, 0x19); ak4642_write(codec, 0x09, 0x91); ak4642_write(codec, 0x0c, 0x91); - ak4642_write(codec, 0x0a, 0x28); - ak4642_write(codec, 0x0d, 0x28); ak4642_write(codec, 0x00, 0x64); snd_soc_update_bits(codec, PW_MGMT2, PMHP_MASK, PMHP); snd_soc_update_bits(codec, PW_MGMT2, HPMTN, HPMTN); @@ -491,8 +498,10 @@ static int ak4642_i2c_probe(struct i2c_client *i2c, codec->control_data = i2c; ret = ak4642_init(ak4642); - if (ret < 0) + if (ret < 0) { printk(KERN_ERR "failed to initialise AK4642\n"); + kfree(ak4642); + } return ret; } @@ -548,6 +557,9 @@ static int ak4642_probe(struct platform_device *pdev) goto pcm_err; } + snd_soc_add_controls(ak4642_codec, ak4642_snd_controls, + ARRAY_SIZE(ak4642_snd_controls)); + dev_info(&pdev->dev, "AK4642 Audio Codec %s", AK4642_VERSION); return ret; diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c new file mode 100644 index 000000000000..dd9b8550c402 --- /dev/null +++ b/sound/soc/codecs/cs42l51.c @@ -0,0 +1,763 @@ +/* + * cs42l51.c + * + * ASoC Driver for Cirrus Logic CS42L51 codecs + * + * Copyright (c) 2010 Arnaud Patard <apatard@mandriva.com> + * + * Based on cs4270.c - Copyright (c) Freescale Semiconductor + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * For now: + * - Only I2C is support. Not SPI + * - master mode *NOT* supported + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include <sound/initval.h> +#include <sound/pcm_params.h> +#include <sound/pcm.h> +#include <linux/i2c.h> + +#include "cs42l51.h" + +enum master_slave_mode { + MODE_SLAVE, + MODE_SLAVE_AUTO, + MODE_MASTER, +}; + +struct cs42l51_private { + unsigned int mclk; + unsigned int audio_mode; /* The mode (I2S or left-justified) */ + enum master_slave_mode func; + struct snd_soc_codec codec; + u8 reg_cache[CS42L51_NUMREGS]; +}; + +static struct snd_soc_codec *cs42l51_codec; + +#define CS42L51_FORMATS ( \ + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE) + +static int cs42l51_fill_cache(struct snd_soc_codec *codec) +{ + u8 *cache = codec->reg_cache + 1; + struct i2c_client *i2c_client = codec->control_data; + s32 length; + + length = i2c_smbus_read_i2c_block_data(i2c_client, + CS42L51_FIRSTREG | 0x80, CS42L51_NUMREGS, cache); + if (length != CS42L51_NUMREGS) { + dev_err(&i2c_client->dev, + "I2C read failure, addr=0x%x (ret=%d vs %d)\n", + i2c_client->addr, length, CS42L51_NUMREGS); + return -EIO; + } + + return 0; +} + +static int cs42l51_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct snd_soc_codec *codec; + struct cs42l51_private *cs42l51; + int ret = 0; + int reg; + + if (cs42l51_codec) + return -EBUSY; + + /* Verify that we have a CS42L51 */ + ret = i2c_smbus_read_byte_data(i2c_client, CS42L51_CHIP_REV_ID); + if (ret < 0) { + dev_err(&i2c_client->dev, "failed to read I2C\n"); + goto error; + } + + if ((ret != CS42L51_MK_CHIP_REV(CS42L51_CHIP_ID, CS42L51_CHIP_REV_A)) && + (ret != CS42L51_MK_CHIP_REV(CS42L51_CHIP_ID, CS42L51_CHIP_REV_B))) { + dev_err(&i2c_client->dev, "Invalid chip id\n"); + ret = -ENODEV; + goto error; + } + + dev_info(&i2c_client->dev, "found device cs42l51 rev %d\n", + ret & 7); + + cs42l51 = kzalloc(sizeof(struct cs42l51_private), GFP_KERNEL); + if (!cs42l51) { + dev_err(&i2c_client->dev, "could not allocate codec\n"); + return -ENOMEM; + } + codec = &cs42l51->codec; + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->dev = &i2c_client->dev; + codec->name = "CS42L51"; + codec->owner = THIS_MODULE; + codec->dai = &cs42l51_dai; + codec->num_dai = 1; + snd_soc_codec_set_drvdata(codec, cs42l51); + + codec->control_data = i2c_client; + codec->reg_cache = cs42l51->reg_cache; + codec->reg_cache_size = CS42L51_NUMREGS; + i2c_set_clientdata(i2c_client, codec); + + ret = cs42l51_fill_cache(codec); + if (ret < 0) { + dev_err(&i2c_client->dev, "failed to fill register cache\n"); + goto error_alloc; + } + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + if (ret < 0) { + dev_err(&i2c_client->dev, "Failed to set cache I/O: %d\n", ret); + goto error_alloc; + } + + /* + * DAC configuration + * - Use signal processor + * - auto mute + * - vol changes immediate + * - no de-emphasize + */ + reg = CS42L51_DAC_CTL_DATA_SEL(1) + | CS42L51_DAC_CTL_AMUTE | CS42L51_DAC_CTL_DACSZ(0); + ret = snd_soc_write(codec, CS42L51_DAC_CTL, reg); + if (ret < 0) + goto error_alloc; + + cs42l51_dai.dev = codec->dev; + cs42l51_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto error_alloc; + } + + ret = snd_soc_register_dai(&cs42l51_dai); + if (ret < 0) { + dev_err(&i2c_client->dev, "failed to register DAIe\n"); + goto error_reg; + } + + return 0; + +error_reg: + snd_soc_unregister_codec(codec); +error_alloc: + kfree(cs42l51); +error: + return ret; +} + +static int cs42l51_i2c_remove(struct i2c_client *client) +{ + struct cs42l51_private *cs42l51 = i2c_get_clientdata(client); + snd_soc_unregister_dai(&cs42l51_dai); + snd_soc_unregister_codec(cs42l51_codec); + cs42l51_codec = NULL; + kfree(cs42l51); + return 0; +} + + +static const struct i2c_device_id cs42l51_id[] = { + {"cs42l51", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, cs42l51_id); + +static struct i2c_driver cs42l51_i2c_driver = { + .driver = { + .name = "CS42L51 I2C", + .owner = THIS_MODULE, + }, + .id_table = cs42l51_id, + .probe = cs42l51_i2c_probe, + .remove = cs42l51_i2c_remove, +}; + +static int cs42l51_get_chan_mix(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned long value = snd_soc_read(codec, CS42L51_PCM_MIXER)&3; + + switch (value) { + default: + case 0: + ucontrol->value.integer.value[0] = 0; + break; + /* same value : (L+R)/2 and (R+L)/2 */ + case 1: + case 2: + ucontrol->value.integer.value[0] = 1; + break; + case 3: + ucontrol->value.integer.value[0] = 2; + break; + } + + return 0; +} + +#define CHAN_MIX_NORMAL 0x00 +#define CHAN_MIX_BOTH 0x55 +#define CHAN_MIX_SWAP 0xFF + +static int cs42l51_set_chan_mix(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned char val; + + switch (ucontrol->value.integer.value[0]) { + default: + case 0: + val = CHAN_MIX_NORMAL; + break; + case 1: + val = CHAN_MIX_BOTH; + break; + case 2: + val = CHAN_MIX_SWAP; + break; + } + + snd_soc_write(codec, CS42L51_PCM_MIXER, val); + + return 1; +} + +static const DECLARE_TLV_DB_SCALE(adc_pcm_tlv, -5150, 50, 0); +static const DECLARE_TLV_DB_SCALE(tone_tlv, -1050, 150, 0); +/* This is a lie. after -102 db, it stays at -102 */ +/* maybe a range would be better */ +static const DECLARE_TLV_DB_SCALE(aout_tlv, -11550, 50, 0); + +static const DECLARE_TLV_DB_SCALE(boost_tlv, 1600, 1600, 0); +static const char *chan_mix[] = { + "L R", + "L+R", + "R L", +}; + +static const struct soc_enum cs42l51_chan_mix = + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(chan_mix), chan_mix); + +static const struct snd_kcontrol_new cs42l51_snd_controls[] = { + SOC_DOUBLE_R_SX_TLV("PCM Playback Volume", + CS42L51_PCMA_VOL, CS42L51_PCMB_VOL, + 7, 0xffffff99, 0x18, adc_pcm_tlv), + SOC_DOUBLE_R("PCM Playback Switch", + CS42L51_PCMA_VOL, CS42L51_PCMB_VOL, 7, 1, 1), + SOC_DOUBLE_R_SX_TLV("Analog Playback Volume", + CS42L51_AOUTA_VOL, CS42L51_AOUTB_VOL, + 8, 0xffffff19, 0x18, aout_tlv), + SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", + CS42L51_ADCA_VOL, CS42L51_ADCB_VOL, + 7, 0xffffff99, 0x18, adc_pcm_tlv), + SOC_DOUBLE_R("ADC Mixer Switch", + CS42L51_ADCA_VOL, CS42L51_ADCB_VOL, 7, 1, 1), + SOC_SINGLE("Playback Deemphasis Switch", CS42L51_DAC_CTL, 3, 1, 0), + SOC_SINGLE("Auto-Mute Switch", CS42L51_DAC_CTL, 2, 1, 0), + SOC_SINGLE("Soft Ramp Switch", CS42L51_DAC_CTL, 1, 1, 0), + SOC_SINGLE("Zero Cross Switch", CS42L51_DAC_CTL, 0, 0, 0), + SOC_DOUBLE_TLV("Mic Boost Volume", + CS42L51_MIC_CTL, 0, 1, 1, 0, boost_tlv), + SOC_SINGLE_TLV("Bass Volume", CS42L51_TONE_CTL, 0, 0xf, 1, tone_tlv), + SOC_SINGLE_TLV("Treble Volume", CS42L51_TONE_CTL, 4, 0xf, 1, tone_tlv), + SOC_ENUM_EXT("PCM channel mixer", + cs42l51_chan_mix, + cs42l51_get_chan_mix, cs42l51_set_chan_mix), +}; + +/* + * to power down, one must: + * 1.) Enable the PDN bit + * 2.) enable power-down for the select channels + * 3.) disable the PDN bit. + */ +static int cs42l51_pdn_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + unsigned long value; + + value = snd_soc_read(w->codec, CS42L51_POWER_CTL1); + value &= ~CS42L51_POWER_CTL1_PDN; + + switch (event) { + case SND_SOC_DAPM_PRE_PMD: + value |= CS42L51_POWER_CTL1_PDN; + break; + default: + case SND_SOC_DAPM_POST_PMD: + break; + } + snd_soc_update_bits(w->codec, CS42L51_POWER_CTL1, + CS42L51_POWER_CTL1_PDN, value); + + return 0; +} + +static const char *cs42l51_dac_names[] = {"Direct PCM", + "DSP PCM", "ADC"}; +static const struct soc_enum cs42l51_dac_mux_enum = + SOC_ENUM_SINGLE(CS42L51_DAC_CTL, 6, 3, cs42l51_dac_names); +static const struct snd_kcontrol_new cs42l51_dac_mux_controls = + SOC_DAPM_ENUM("Route", cs42l51_dac_mux_enum); + +static const char *cs42l51_adcl_names[] = {"AIN1 Left", "AIN2 Left", + "MIC Left", "MIC+preamp Left"}; +static const struct soc_enum cs42l51_adcl_mux_enum = + SOC_ENUM_SINGLE(CS42L51_ADC_INPUT, 4, 4, cs42l51_adcl_names); +static const struct snd_kcontrol_new cs42l51_adcl_mux_controls = + SOC_DAPM_ENUM("Route", cs42l51_adcl_mux_enum); + +static const char *cs42l51_adcr_names[] = {"AIN1 Right", "AIN2 Right", + "MIC Right", "MIC+preamp Right"}; +static const struct soc_enum cs42l51_adcr_mux_enum = + SOC_ENUM_SINGLE(CS42L51_ADC_INPUT, 6, 4, cs42l51_adcr_names); +static const struct snd_kcontrol_new cs42l51_adcr_mux_controls = + SOC_DAPM_ENUM("Route", cs42l51_adcr_mux_enum); + +static const struct snd_soc_dapm_widget cs42l51_dapm_widgets[] = { + SND_SOC_DAPM_MICBIAS("Mic Bias", CS42L51_MIC_POWER_CTL, 1, 1), + SND_SOC_DAPM_PGA_E("Left PGA", CS42L51_POWER_CTL1, 3, 1, NULL, 0, + cs42l51_pdn_event, SND_SOC_DAPM_PRE_POST_PMD), + SND_SOC_DAPM_PGA_E("Right PGA", CS42L51_POWER_CTL1, 4, 1, NULL, 0, + cs42l51_pdn_event, SND_SOC_DAPM_PRE_POST_PMD), + SND_SOC_DAPM_ADC_E("Left ADC", "Left HiFi Capture", + CS42L51_POWER_CTL1, 1, 1, + cs42l51_pdn_event, SND_SOC_DAPM_PRE_POST_PMD), + SND_SOC_DAPM_ADC_E("Right ADC", "Right HiFi Capture", + CS42L51_POWER_CTL1, 2, 1, + cs42l51_pdn_event, SND_SOC_DAPM_PRE_POST_PMD), + SND_SOC_DAPM_DAC_E("Left DAC", "Left HiFi Playback", + CS42L51_POWER_CTL1, 5, 1, + cs42l51_pdn_event, SND_SOC_DAPM_PRE_POST_PMD), + SND_SOC_DAPM_DAC_E("Right DAC", "Right HiFi Playback", + CS42L51_POWER_CTL1, 6, 1, + cs42l51_pdn_event, SND_SOC_DAPM_PRE_POST_PMD), + + /* analog/mic */ + SND_SOC_DAPM_INPUT("AIN1L"), + SND_SOC_DAPM_INPUT("AIN1R"), + SND_SOC_DAPM_INPUT("AIN2L"), + SND_SOC_DAPM_INPUT("AIN2R"), + SND_SOC_DAPM_INPUT("MICL"), + SND_SOC_DAPM_INPUT("MICR"), + + SND_SOC_DAPM_MIXER("Mic Preamp Left", + CS42L51_MIC_POWER_CTL, 2, 1, NULL, 0), + SND_SOC_DAPM_MIXER("Mic Preamp Right", + CS42L51_MIC_POWER_CTL, 3, 1, NULL, 0), + + /* HP */ + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), + + /* mux */ + SND_SOC_DAPM_MUX("DAC Mux", SND_SOC_NOPM, 0, 0, + &cs42l51_dac_mux_controls), + SND_SOC_DAPM_MUX("PGA-ADC Mux Left", SND_SOC_NOPM, 0, 0, + &cs42l51_adcl_mux_controls), + SND_SOC_DAPM_MUX("PGA-ADC Mux Right", SND_SOC_NOPM, 0, 0, + &cs42l51_adcr_mux_controls), +}; + +static const struct snd_soc_dapm_route cs42l51_routes[] = { + {"HPL", NULL, "Left DAC"}, + {"HPR", NULL, "Right DAC"}, + + {"Left ADC", NULL, "Left PGA"}, + {"Right ADC", NULL, "Right PGA"}, + + {"Mic Preamp Left", NULL, "MICL"}, + {"Mic Preamp Right", NULL, "MICR"}, + + {"PGA-ADC Mux Left", "AIN1 Left", "AIN1L" }, + {"PGA-ADC Mux Left", "AIN2 Left", "AIN2L" }, + {"PGA-ADC Mux Left", "MIC Left", "MICL" }, + {"PGA-ADC Mux Left", "MIC+preamp Left", "Mic Preamp Left" }, + {"PGA-ADC Mux Right", "AIN1 Right", "AIN1R" }, + {"PGA-ADC Mux Right", "AIN2 Right", "AIN2R" }, + {"PGA-ADC Mux Right", "MIC Right", "MICR" }, + {"PGA-ADC Mux Right", "MIC+preamp Right", "Mic Preamp Right" }, + + {"Left PGA", NULL, "PGA-ADC Mux Left"}, + {"Right PGA", NULL, "PGA-ADC Mux Right"}, +}; + +static int cs42l51_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42l51_private *cs42l51 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_RIGHT_J: + cs42l51->audio_mode = format & SND_SOC_DAIFMT_FORMAT_MASK; + break; + default: + dev_err(codec->dev, "invalid DAI format\n"); + ret = -EINVAL; + } + + switch (format & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + cs42l51->func = MODE_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + cs42l51->func = MODE_SLAVE_AUTO; + break; + default: + ret = -EINVAL; + break; + } + + return ret; +} + +struct cs42l51_ratios { + unsigned int ratio; + unsigned char speed_mode; + unsigned char mclk; +}; + +static struct cs42l51_ratios slave_ratios[] = { + { 512, CS42L51_QSM_MODE, 0 }, { 768, CS42L51_QSM_MODE, 0 }, + { 1024, CS42L51_QSM_MODE, 0 }, { 1536, CS42L51_QSM_MODE, 0 }, + { 2048, CS42L51_QSM_MODE, 0 }, { 3072, CS42L51_QSM_MODE, 0 }, + { 256, CS42L51_HSM_MODE, 0 }, { 384, CS42L51_HSM_MODE, 0 }, + { 512, CS42L51_HSM_MODE, 0 }, { 768, CS42L51_HSM_MODE, 0 }, + { 1024, CS42L51_HSM_MODE, 0 }, { 1536, CS42L51_HSM_MODE, 0 }, + { 128, CS42L51_SSM_MODE, 0 }, { 192, CS42L51_SSM_MODE, 0 }, + { 256, CS42L51_SSM_MODE, 0 }, { 384, CS42L51_SSM_MODE, 0 }, + { 512, CS42L51_SSM_MODE, 0 }, { 768, CS42L51_SSM_MODE, 0 }, + { 128, CS42L51_DSM_MODE, 0 }, { 192, CS42L51_DSM_MODE, 0 }, + { 256, CS42L51_DSM_MODE, 0 }, { 384, CS42L51_DSM_MODE, 0 }, +}; + +static struct cs42l51_ratios slave_auto_ratios[] = { + { 1024, CS42L51_QSM_MODE, 0 }, { 1536, CS42L51_QSM_MODE, 0 }, + { 2048, CS42L51_QSM_MODE, 1 }, { 3072, CS42L51_QSM_MODE, 1 }, + { 512, CS42L51_HSM_MODE, 0 }, { 768, CS42L51_HSM_MODE, 0 }, + { 1024, CS42L51_HSM_MODE, 1 }, { 1536, CS42L51_HSM_MODE, 1 }, + { 256, CS42L51_SSM_MODE, 0 }, { 384, CS42L51_SSM_MODE, 0 }, + { 512, CS42L51_SSM_MODE, 1 }, { 768, CS42L51_SSM_MODE, 1 }, + { 128, CS42L51_DSM_MODE, 0 }, { 192, CS42L51_DSM_MODE, 0 }, + { 256, CS42L51_DSM_MODE, 1 }, { 384, CS42L51_DSM_MODE, 1 }, +}; + +static int cs42l51_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42l51_private *cs42l51 = snd_soc_codec_get_drvdata(codec); + struct cs42l51_ratios *ratios = NULL; + int nr_ratios = 0; + unsigned int rates = 0; + unsigned int rate_min = -1; + unsigned int rate_max = 0; + int i; + + cs42l51->mclk = freq; + + switch (cs42l51->func) { + case MODE_MASTER: + return -EINVAL; + case MODE_SLAVE: + ratios = slave_ratios; + nr_ratios = ARRAY_SIZE(slave_ratios); + break; + case MODE_SLAVE_AUTO: + ratios = slave_auto_ratios; + nr_ratios = ARRAY_SIZE(slave_auto_ratios); + break; + } + + for (i = 0; i < nr_ratios; i++) { + unsigned int rate = freq / ratios[i].ratio; + rates |= snd_pcm_rate_to_rate_bit(rate); + if (rate < rate_min) + rate_min = rate; + if (rate > rate_max) + rate_max = rate; + } + rates &= ~SNDRV_PCM_RATE_KNOT; + + if (!rates) { + dev_err(codec->dev, "could not find a valid sample rate\n"); + return -EINVAL; + } + + codec_dai->playback.rates = rates; + codec_dai->playback.rate_min = rate_min; + codec_dai->playback.rate_max = rate_max; + + codec_dai->capture.rates = rates; + codec_dai->capture.rate_min = rate_min; + codec_dai->capture.rate_max = rate_max; + + return 0; +} + +static int cs42l51_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct cs42l51_private *cs42l51 = snd_soc_codec_get_drvdata(codec); + int ret; + unsigned int i; + unsigned int rate; + unsigned int ratio; + struct cs42l51_ratios *ratios = NULL; + int nr_ratios = 0; + int intf_ctl, power_ctl, fmt; + + switch (cs42l51->func) { + case MODE_MASTER: + return -EINVAL; + case MODE_SLAVE: + ratios = slave_ratios; + nr_ratios = ARRAY_SIZE(slave_ratios); + break; + case MODE_SLAVE_AUTO: + ratios = slave_auto_ratios; + nr_ratios = ARRAY_SIZE(slave_auto_ratios); + break; + } + + /* Figure out which MCLK/LRCK ratio to use */ + rate = params_rate(params); /* Sampling rate, in Hz */ + ratio = cs42l51->mclk / rate; /* MCLK/LRCK ratio */ + for (i = 0; i < nr_ratios; i++) { + if (ratios[i].ratio == ratio) + break; + } + + if (i == nr_ratios) { + /* We did not find a matching ratio */ + dev_err(codec->dev, "could not find matching ratio\n"); + return -EINVAL; + } + + intf_ctl = snd_soc_read(codec, CS42L51_INTF_CTL); + power_ctl = snd_soc_read(codec, CS42L51_MIC_POWER_CTL); + + intf_ctl &= ~(CS42L51_INTF_CTL_MASTER | CS42L51_INTF_CTL_ADC_I2S + | CS42L51_INTF_CTL_DAC_FORMAT(7)); + power_ctl &= ~(CS42L51_MIC_POWER_CTL_SPEED(3) + | CS42L51_MIC_POWER_CTL_MCLK_DIV2); + + switch (cs42l51->func) { + case MODE_MASTER: + intf_ctl |= CS42L51_INTF_CTL_MASTER; + power_ctl |= CS42L51_MIC_POWER_CTL_SPEED(ratios[i].speed_mode); + break; + case MODE_SLAVE: + power_ctl |= CS42L51_MIC_POWER_CTL_SPEED(ratios[i].speed_mode); + break; + case MODE_SLAVE_AUTO: + power_ctl |= CS42L51_MIC_POWER_CTL_AUTO; + break; + } + + switch (cs42l51->audio_mode) { + case SND_SOC_DAIFMT_I2S: + intf_ctl |= CS42L51_INTF_CTL_ADC_I2S; + intf_ctl |= CS42L51_INTF_CTL_DAC_FORMAT(CS42L51_DAC_DIF_I2S); + break; + case SND_SOC_DAIFMT_LEFT_J: + intf_ctl |= CS42L51_INTF_CTL_DAC_FORMAT(CS42L51_DAC_DIF_LJ24); + break; + case SND_SOC_DAIFMT_RIGHT_J: + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + case SNDRV_PCM_FORMAT_S16_BE: + fmt = CS42L51_DAC_DIF_RJ16; + break; + case SNDRV_PCM_FORMAT_S18_3LE: + case SNDRV_PCM_FORMAT_S18_3BE: + fmt = CS42L51_DAC_DIF_RJ18; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + case SNDRV_PCM_FORMAT_S20_3BE: + fmt = CS42L51_DAC_DIF_RJ20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S24_BE: + fmt = CS42L51_DAC_DIF_RJ24; + break; + default: + dev_err(codec->dev, "unknown format\n"); + return -EINVAL; + } + intf_ctl |= CS42L51_INTF_CTL_DAC_FORMAT(fmt); + break; + default: + dev_err(codec->dev, "unknown format\n"); + return -EINVAL; + } + + if (ratios[i].mclk) + power_ctl |= CS42L51_MIC_POWER_CTL_MCLK_DIV2; + + ret = snd_soc_write(codec, CS42L51_INTF_CTL, intf_ctl); + if (ret < 0) + return ret; + + ret = snd_soc_write(codec, CS42L51_MIC_POWER_CTL, power_ctl); + if (ret < 0) + return ret; + + return 0; +} + +static int cs42l51_dai_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + int reg; + int mask = CS42L51_DAC_OUT_CTL_DACA_MUTE|CS42L51_DAC_OUT_CTL_DACB_MUTE; + + reg = snd_soc_read(codec, CS42L51_DAC_OUT_CTL); + + if (mute) + reg |= mask; + else + reg &= ~mask; + + return snd_soc_write(codec, CS42L51_DAC_OUT_CTL, reg); +} + +static struct snd_soc_dai_ops cs42l51_dai_ops = { + .hw_params = cs42l51_hw_params, + .set_sysclk = cs42l51_set_dai_sysclk, + .set_fmt = cs42l51_set_dai_fmt, + .digital_mute = cs42l51_dai_mute, +}; + +struct snd_soc_dai cs42l51_dai = { + .name = "CS42L51 HiFi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = CS42L51_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = CS42L51_FORMATS, + }, + .ops = &cs42l51_dai_ops, +}; +EXPORT_SYMBOL_GPL(cs42l51_dai); + + +static int cs42l51_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (!cs42l51_codec) { + dev_err(&pdev->dev, "CS42L51 codec not yet registered\n"); + return -EINVAL; + } + + socdev->card->codec = cs42l51_codec; + codec = socdev->card->codec; + + /* Register PCMs */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(&pdev->dev, "failed to create PCMs\n"); + return ret; + } + + snd_soc_add_controls(codec, cs42l51_snd_controls, + ARRAY_SIZE(cs42l51_snd_controls)); + snd_soc_dapm_new_controls(codec, cs42l51_dapm_widgets, + ARRAY_SIZE(cs42l51_dapm_widgets)); + snd_soc_dapm_add_routes(codec, cs42l51_routes, + ARRAY_SIZE(cs42l51_routes)); + + return 0; +} + + +static int cs42l51_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_device_cs42l51 = { + .probe = cs42l51_probe, + .remove = cs42l51_remove +}; +EXPORT_SYMBOL_GPL(soc_codec_device_cs42l51); + +static int __init cs42l51_init(void) +{ + int ret; + + ret = i2c_add_driver(&cs42l51_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "%s: can't add i2c driver\n", __func__); + return ret; + } + return 0; +} +module_init(cs42l51_init); + +static void __exit cs42l51_exit(void) +{ + i2c_del_driver(&cs42l51_i2c_driver); +} +module_exit(cs42l51_exit); + +MODULE_AUTHOR("Arnaud Patard <apatard@mandriva.com>"); +MODULE_DESCRIPTION("Cirrus Logic CS42L51 ALSA SoC Codec Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42l51.h b/sound/soc/codecs/cs42l51.h new file mode 100644 index 000000000000..8f0bd9786ad2 --- /dev/null +++ b/sound/soc/codecs/cs42l51.h @@ -0,0 +1,163 @@ +/* + * cs42l51.h + * + * ASoC Driver for Cirrus Logic CS42L51 codecs + * + * Copyright (c) 2010 Arnaud Patard <apatard@mandriva.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ +#ifndef _CS42L51_H +#define _CS42L51_H + +#define CS42L51_CHIP_ID 0x1B +#define CS42L51_CHIP_REV_A 0x00 +#define CS42L51_CHIP_REV_B 0x01 + +#define CS42L51_CHIP_REV_ID 0x01 +#define CS42L51_MK_CHIP_REV(a, b) ((a)<<3|(b)) + +#define CS42L51_POWER_CTL1 0x02 +#define CS42L51_POWER_CTL1_PDN_DACB (1<<6) +#define CS42L51_POWER_CTL1_PDN_DACA (1<<5) +#define CS42L51_POWER_CTL1_PDN_PGAB (1<<4) +#define CS42L51_POWER_CTL1_PDN_PGAA (1<<3) +#define CS42L51_POWER_CTL1_PDN_ADCB (1<<2) +#define CS42L51_POWER_CTL1_PDN_ADCA (1<<1) +#define CS42L51_POWER_CTL1_PDN (1<<0) + +#define CS42L51_MIC_POWER_CTL 0x03 +#define CS42L51_MIC_POWER_CTL_AUTO (1<<7) +#define CS42L51_MIC_POWER_CTL_SPEED(x) (((x)&3)<<5) +#define CS42L51_QSM_MODE 3 +#define CS42L51_HSM_MODE 2 +#define CS42L51_SSM_MODE 1 +#define CS42L51_DSM_MODE 0 +#define CS42L51_MIC_POWER_CTL_3ST_SP (1<<4) +#define CS42L51_MIC_POWER_CTL_PDN_MICB (1<<3) +#define CS42L51_MIC_POWER_CTL_PDN_MICA (1<<2) +#define CS42L51_MIC_POWER_CTL_PDN_BIAS (1<<1) +#define CS42L51_MIC_POWER_CTL_MCLK_DIV2 (1<<0) + +#define CS42L51_INTF_CTL 0x04 +#define CS42L51_INTF_CTL_LOOPBACK (1<<7) +#define CS42L51_INTF_CTL_MASTER (1<<6) +#define CS42L51_INTF_CTL_DAC_FORMAT(x) (((x)&7)<<3) +#define CS42L51_DAC_DIF_LJ24 0x00 +#define CS42L51_DAC_DIF_I2S 0x01 +#define CS42L51_DAC_DIF_RJ24 0x02 +#define CS42L51_DAC_DIF_RJ20 0x03 +#define CS42L51_DAC_DIF_RJ18 0x04 +#define CS42L51_DAC_DIF_RJ16 0x05 +#define CS42L51_INTF_CTL_ADC_I2S (1<<2) +#define CS42L51_INTF_CTL_DIGMIX (1<<1) +#define CS42L51_INTF_CTL_MICMIX (1<<0) + +#define CS42L51_MIC_CTL 0x05 +#define CS42L51_MIC_CTL_ADC_SNGVOL (1<<7) +#define CS42L51_MIC_CTL_ADCD_DBOOST (1<<6) +#define CS42L51_MIC_CTL_ADCA_DBOOST (1<<5) +#define CS42L51_MIC_CTL_MICBIAS_SEL (1<<4) +#define CS42L51_MIC_CTL_MICBIAS_LVL(x) (((x)&3)<<2) +#define CS42L51_MIC_CTL_MICB_BOOST (1<<1) +#define CS42L51_MIC_CTL_MICA_BOOST (1<<0) + +#define CS42L51_ADC_CTL 0x06 +#define CS42L51_ADC_CTL_ADCB_HPFEN (1<<7) +#define CS42L51_ADC_CTL_ADCB_HPFRZ (1<<6) +#define CS42L51_ADC_CTL_ADCA_HPFEN (1<<5) +#define CS42L51_ADC_CTL_ADCA_HPFRZ (1<<4) +#define CS42L51_ADC_CTL_SOFTB (1<<3) +#define CS42L51_ADC_CTL_ZCROSSB (1<<2) +#define CS42L51_ADC_CTL_SOFTA (1<<1) +#define CS42L51_ADC_CTL_ZCROSSA (1<<0) + +#define CS42L51_ADC_INPUT 0x07 +#define CS42L51_ADC_INPUT_AINB_MUX(x) (((x)&3)<<6) +#define CS42L51_ADC_INPUT_AINA_MUX(x) (((x)&3)<<4) +#define CS42L51_ADC_INPUT_INV_ADCB (1<<3) +#define CS42L51_ADC_INPUT_INV_ADCA (1<<2) +#define CS42L51_ADC_INPUT_ADCB_MUTE (1<<1) +#define CS42L51_ADC_INPUT_ADCA_MUTE (1<<0) + +#define CS42L51_DAC_OUT_CTL 0x08 +#define CS42L51_DAC_OUT_CTL_HP_GAIN(x) (((x)&7)<<5) +#define CS42L51_DAC_OUT_CTL_DAC_SNGVOL (1<<4) +#define CS42L51_DAC_OUT_CTL_INV_PCMB (1<<3) +#define CS42L51_DAC_OUT_CTL_INV_PCMA (1<<2) +#define CS42L51_DAC_OUT_CTL_DACB_MUTE (1<<1) +#define CS42L51_DAC_OUT_CTL_DACA_MUTE (1<<0) + +#define CS42L51_DAC_CTL 0x09 +#define CS42L51_DAC_CTL_DATA_SEL(x) (((x)&3)<<6) +#define CS42L51_DAC_CTL_FREEZE (1<<5) +#define CS42L51_DAC_CTL_DEEMPH (1<<3) +#define CS42L51_DAC_CTL_AMUTE (1<<2) +#define CS42L51_DAC_CTL_DACSZ(x) (((x)&3)<<0) + +#define CS42L51_ALC_PGA_CTL 0x0A +#define CS42L51_ALC_PGB_CTL 0x0B +#define CS42L51_ALC_PGX_ALCX_SRDIS (1<<7) +#define CS42L51_ALC_PGX_ALCX_ZCDIS (1<<6) +#define CS42L51_ALC_PGX_PGX_VOL(x) (((x)&0x1f)<<0) + +#define CS42L51_ADCA_ATT 0x0C +#define CS42L51_ADCB_ATT 0x0D + +#define CS42L51_ADCA_VOL 0x0E +#define CS42L51_ADCB_VOL 0x0F +#define CS42L51_PCMA_VOL 0x10 +#define CS42L51_PCMB_VOL 0x11 +#define CS42L51_MIX_MUTE_ADCMIX (1<<7) +#define CS42L51_MIX_VOLUME(x) (((x)&0x7f)<<0) + +#define CS42L51_BEEP_FREQ 0x12 +#define CS42L51_BEEP_VOL 0x13 +#define CS42L51_BEEP_CONF 0x14 + +#define CS42L51_TONE_CTL 0x15 +#define CS42L51_TONE_CTL_TREB(x) (((x)&0xf)<<4) +#define CS42L51_TONE_CTL_BASS(x) (((x)&0xf)<<0) + +#define CS42L51_AOUTA_VOL 0x16 +#define CS42L51_AOUTB_VOL 0x17 +#define CS42L51_PCM_MIXER 0x18 +#define CS42L51_LIMIT_THRES_DIS 0x19 +#define CS42L51_LIMIT_REL 0x1A +#define CS42L51_LIMIT_ATT 0x1B +#define CS42L51_ALC_EN 0x1C +#define CS42L51_ALC_REL 0x1D +#define CS42L51_ALC_THRES 0x1E +#define CS42L51_NOISE_CONF 0x1F + +#define CS42L51_STATUS 0x20 +#define CS42L51_STATUS_SP_CLKERR (1<<6) +#define CS42L51_STATUS_SPEA_OVFL (1<<5) +#define CS42L51_STATUS_SPEB_OVFL (1<<4) +#define CS42L51_STATUS_PCMA_OVFL (1<<3) +#define CS42L51_STATUS_PCMB_OVFL (1<<2) +#define CS42L51_STATUS_ADCA_OVFL (1<<1) +#define CS42L51_STATUS_ADCB_OVFL (1<<0) + +#define CS42L51_CHARGE_FREQ 0x21 + +#define CS42L51_FIRSTREG 0x01 +/* + * Hack: with register 0x21, it makes 33 registers. Looks like someone in the + * i2c layer doesn't like i2c smbus block read of 33 regs. Workaround by using + * 32 regs + */ +#define CS42L51_LASTREG 0x20 +#define CS42L51_NUMREGS (CS42L51_LASTREG - CS42L51_FIRSTREG + 1) + +extern struct snd_soc_dai cs42l51_dai; +extern struct snd_soc_codec_device soc_codec_device_cs42l51; +#endif diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 75af2d6e0e78..3c51d6a57523 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -15,23 +15,15 @@ * option) any later version. */ -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/kernel.h> -#include <linux/init.h> #include <linux/delay.h> -#include <linux/pm.h> #include <linux/i2c.h> #include <linux/platform_device.h> #include <linux/slab.h> -#include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> -#include <sound/soc.h> #include <sound/soc-dapm.h> -#include <sound/tlv.h> #include <sound/initval.h> -#include <asm/div64.h> +#include <sound/tlv.h> #include "da7210.h" @@ -145,6 +137,29 @@ #define DA7210_VERSION "0.0.1" +/* + * Playback Volume + * + * max : 0x3F (+15.0 dB) + * (1.5 dB step) + * min : 0x11 (-54.0 dB) + * mute : 0x10 + * reserved : 0x00 - 0x0F + * + * ** FIXME ** + * + * Reserved area are considered as "mute". + * -> min = -79.5 dB + */ +static const DECLARE_TLV_DB_SCALE(hp_out_tlv, -7950, 150, 1); + +static const struct snd_kcontrol_new da7210_snd_controls[] = { + + SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", + DA7210_HP_L_VOL, DA7210_HP_R_VOL, + 0, 0x3F, 0, hp_out_tlv), +}; + /* Codec private data */ struct da7210_priv { struct snd_soc_codec codec; @@ -227,10 +242,6 @@ static int da7210_startup(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; if (is_play) { - /* PlayBack Volume 40 */ - snd_soc_update_bits(codec, DA7210_HP_L_VOL, 0x3F, 40); - snd_soc_update_bits(codec, DA7210_HP_R_VOL, 0x3F, 40); - /* Enable Out */ snd_soc_update_bits(codec, DA7210_OUTMIX_L, 0x1F, 0x10); snd_soc_update_bits(codec, DA7210_OUTMIX_R, 0x1F, 0x10); @@ -488,7 +499,7 @@ static int da7210_init(struct da7210_priv *da7210) ret = snd_soc_register_dai(&da7210_dai); if (ret) { dev_err(codec->dev, "Failed to register DAI: %d\n", ret); - goto init_err; + goto codec_err; } /* FIXME @@ -574,6 +585,8 @@ static int da7210_init(struct da7210_priv *da7210) return ret; +codec_err: + snd_soc_unregister_codec(codec); init_err: kfree(codec->reg_cache); codec->reg_cache = NULL; @@ -601,8 +614,10 @@ static int __devinit da7210_i2c_probe(struct i2c_client *i2c, codec->control_data = i2c; ret = da7210_init(da7210); - if (ret < 0) + if (ret < 0) { pr_err("Failed to initialise da7210 audio codec\n"); + kfree(da7210); + } return ret; } @@ -656,6 +671,9 @@ static int da7210_probe(struct platform_device *pdev) if (ret < 0) goto pcm_err; + snd_soc_add_controls(da7210_codec, da7210_snd_controls, + ARRAY_SIZE(da7210_snd_controls)); + dev_info(&pdev->dev, "DA7210 Audio Codec %s\n", DA7210_VERSION); pcm_err: diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c new file mode 100644 index 000000000000..66557de1e4fe --- /dev/null +++ b/sound/soc/codecs/jz4740.c @@ -0,0 +1,511 @@ +/* + * Copyright (C) 2009-2010, Lars-Peter Clausen <lars@metafoo.de> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 675 Mass Ave, Cambridge, MA 02139, USA. + * + */ + +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> + +#include <linux/delay.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc-dapm.h> +#include <sound/soc.h> + +#define JZ4740_REG_CODEC_1 0x0 +#define JZ4740_REG_CODEC_2 0x1 + +#define JZ4740_CODEC_1_LINE_ENABLE BIT(29) +#define JZ4740_CODEC_1_MIC_ENABLE BIT(28) +#define JZ4740_CODEC_1_SW1_ENABLE BIT(27) +#define JZ4740_CODEC_1_ADC_ENABLE BIT(26) +#define JZ4740_CODEC_1_SW2_ENABLE BIT(25) +#define JZ4740_CODEC_1_DAC_ENABLE BIT(24) +#define JZ4740_CODEC_1_VREF_DISABLE BIT(20) +#define JZ4740_CODEC_1_VREF_AMP_DISABLE BIT(19) +#define JZ4740_CODEC_1_VREF_PULLDOWN BIT(18) +#define JZ4740_CODEC_1_VREF_LOW_CURRENT BIT(17) +#define JZ4740_CODEC_1_VREF_HIGH_CURRENT BIT(16) +#define JZ4740_CODEC_1_HEADPHONE_DISABLE BIT(14) +#define JZ4740_CODEC_1_HEADPHONE_AMP_CHANGE_ANY BIT(13) +#define JZ4740_CODEC_1_HEADPHONE_CHARGE BIT(12) +#define JZ4740_CODEC_1_HEADPHONE_PULLDOWN (BIT(11) | BIT(10)) +#define JZ4740_CODEC_1_HEADPHONE_POWERDOWN_M BIT(9) +#define JZ4740_CODEC_1_HEADPHONE_POWERDOWN BIT(8) +#define JZ4740_CODEC_1_SUSPEND BIT(1) +#define JZ4740_CODEC_1_RESET BIT(0) + +#define JZ4740_CODEC_1_LINE_ENABLE_OFFSET 29 +#define JZ4740_CODEC_1_MIC_ENABLE_OFFSET 28 +#define JZ4740_CODEC_1_SW1_ENABLE_OFFSET 27 +#define JZ4740_CODEC_1_ADC_ENABLE_OFFSET 26 +#define JZ4740_CODEC_1_SW2_ENABLE_OFFSET 25 +#define JZ4740_CODEC_1_DAC_ENABLE_OFFSET 24 +#define JZ4740_CODEC_1_HEADPHONE_DISABLE_OFFSET 14 +#define JZ4740_CODEC_1_HEADPHONE_POWERDOWN_OFFSET 8 + +#define JZ4740_CODEC_2_INPUT_VOLUME_MASK 0x1f0000 +#define JZ4740_CODEC_2_SAMPLE_RATE_MASK 0x000f00 +#define JZ4740_CODEC_2_MIC_BOOST_GAIN_MASK 0x000030 +#define JZ4740_CODEC_2_HEADPHONE_VOLUME_MASK 0x000003 + +#define JZ4740_CODEC_2_INPUT_VOLUME_OFFSET 16 +#define JZ4740_CODEC_2_SAMPLE_RATE_OFFSET 8 +#define JZ4740_CODEC_2_MIC_BOOST_GAIN_OFFSET 4 +#define JZ4740_CODEC_2_HEADPHONE_VOLUME_OFFSET 0 + +static const uint32_t jz4740_codec_regs[] = { + 0x021b2302, 0x00170803, +}; + +struct jz4740_codec { + void __iomem *base; + struct resource *mem; + + uint32_t reg_cache[2]; + struct snd_soc_codec codec; +}; + +static inline struct jz4740_codec *codec_to_jz4740(struct snd_soc_codec *codec) +{ + return container_of(codec, struct jz4740_codec, codec); +} + +static unsigned int jz4740_codec_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + struct jz4740_codec *jz4740_codec = codec_to_jz4740(codec); + return readl(jz4740_codec->base + (reg << 2)); +} + +static int jz4740_codec_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int val) +{ + struct jz4740_codec *jz4740_codec = codec_to_jz4740(codec); + + jz4740_codec->reg_cache[reg] = val; + writel(val, jz4740_codec->base + (reg << 2)); + + return 0; +} + +static const struct snd_kcontrol_new jz4740_codec_controls[] = { + SOC_SINGLE("Master Playback Volume", JZ4740_REG_CODEC_2, + JZ4740_CODEC_2_HEADPHONE_VOLUME_OFFSET, 3, 0), + SOC_SINGLE("Master Capture Volume", JZ4740_REG_CODEC_2, + JZ4740_CODEC_2_INPUT_VOLUME_OFFSET, 31, 0), + SOC_SINGLE("Master Playback Switch", JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_HEADPHONE_DISABLE_OFFSET, 1, 1), + SOC_SINGLE("Mic Capture Volume", JZ4740_REG_CODEC_2, + JZ4740_CODEC_2_MIC_BOOST_GAIN_OFFSET, 3, 0), +}; + +static const struct snd_kcontrol_new jz4740_codec_output_controls[] = { + SOC_DAPM_SINGLE("Bypass Switch", JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_SW1_ENABLE_OFFSET, 1, 0), + SOC_DAPM_SINGLE("DAC Switch", JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_SW2_ENABLE_OFFSET, 1, 0), +}; + +static const struct snd_kcontrol_new jz4740_codec_input_controls[] = { + SOC_DAPM_SINGLE("Line Capture Switch", JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_LINE_ENABLE_OFFSET, 1, 0), + SOC_DAPM_SINGLE("Mic Capture Switch", JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_MIC_ENABLE_OFFSET, 1, 0), +}; + +static const struct snd_soc_dapm_widget jz4740_codec_dapm_widgets[] = { + SND_SOC_DAPM_ADC("ADC", "Capture", JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_ADC_ENABLE_OFFSET, 0), + SND_SOC_DAPM_DAC("DAC", "Playback", JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_DAC_ENABLE_OFFSET, 0), + + SND_SOC_DAPM_MIXER("Output Mixer", JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_HEADPHONE_POWERDOWN_OFFSET, 1, + jz4740_codec_output_controls, + ARRAY_SIZE(jz4740_codec_output_controls)), + + SND_SOC_DAPM_MIXER_NAMED_CTL("Input Mixer", SND_SOC_NOPM, 0, 0, + jz4740_codec_input_controls, + ARRAY_SIZE(jz4740_codec_input_controls)), + SND_SOC_DAPM_MIXER("Line Input", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_OUTPUT("LOUT"), + SND_SOC_DAPM_OUTPUT("ROUT"), + + SND_SOC_DAPM_INPUT("MIC"), + SND_SOC_DAPM_INPUT("LIN"), + SND_SOC_DAPM_INPUT("RIN"), +}; + +static const struct snd_soc_dapm_route jz4740_codec_dapm_routes[] = { + {"Line Input", NULL, "LIN"}, + {"Line Input", NULL, "RIN"}, + + {"Input Mixer", "Line Capture Switch", "Line Input"}, + {"Input Mixer", "Mic Capture Switch", "MIC"}, + + {"ADC", NULL, "Input Mixer"}, + + {"Output Mixer", "Bypass Switch", "Input Mixer"}, + {"Output Mixer", "DAC Switch", "DAC"}, + + {"LOUT", NULL, "Output Mixer"}, + {"ROUT", NULL, "Output Mixer"}, +}; + +static int jz4740_codec_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + uint32_t val; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + + switch (params_rate(params)) { + case 8000: + val = 0; + break; + case 11025: + val = 1; + break; + case 12000: + val = 2; + break; + case 16000: + val = 3; + break; + case 22050: + val = 4; + break; + case 24000: + val = 5; + break; + case 32000: + val = 6; + break; + case 44100: + val = 7; + break; + case 48000: + val = 8; + break; + default: + return -EINVAL; + } + + val <<= JZ4740_CODEC_2_SAMPLE_RATE_OFFSET; + + snd_soc_update_bits(codec, JZ4740_REG_CODEC_2, + JZ4740_CODEC_2_SAMPLE_RATE_MASK, val); + + return 0; +} + +static struct snd_soc_dai_ops jz4740_codec_dai_ops = { + .hw_params = jz4740_codec_hw_params, +}; + +struct snd_soc_dai jz4740_codec_dai = { + .name = "jz4740", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8, + }, + .ops = &jz4740_codec_dai_ops, + .symmetric_rates = 1, +}; +EXPORT_SYMBOL_GPL(jz4740_codec_dai); + +static void jz4740_codec_wakeup(struct snd_soc_codec *codec) +{ + int i; + uint32_t *cache = codec->reg_cache; + + snd_soc_update_bits(codec, JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_RESET, JZ4740_CODEC_1_RESET); + udelay(2); + + snd_soc_update_bits(codec, JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_SUSPEND | JZ4740_CODEC_1_RESET, 0); + + for (i = 0; i < ARRAY_SIZE(jz4740_codec_regs); ++i) + jz4740_codec_write(codec, i, cache[i]); +} + +static int jz4740_codec_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + unsigned int mask; + unsigned int value; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + mask = JZ4740_CODEC_1_VREF_DISABLE | + JZ4740_CODEC_1_VREF_AMP_DISABLE | + JZ4740_CODEC_1_HEADPHONE_POWERDOWN_M; + value = 0; + + snd_soc_update_bits(codec, JZ4740_REG_CODEC_1, mask, value); + break; + case SND_SOC_BIAS_STANDBY: + /* The only way to clear the suspend flag is to reset the codec */ + if (codec->bias_level == SND_SOC_BIAS_OFF) + jz4740_codec_wakeup(codec); + + mask = JZ4740_CODEC_1_VREF_DISABLE | + JZ4740_CODEC_1_VREF_AMP_DISABLE | + JZ4740_CODEC_1_HEADPHONE_POWERDOWN_M; + value = JZ4740_CODEC_1_VREF_DISABLE | + JZ4740_CODEC_1_VREF_AMP_DISABLE | + JZ4740_CODEC_1_HEADPHONE_POWERDOWN_M; + + snd_soc_update_bits(codec, JZ4740_REG_CODEC_1, mask, value); + break; + case SND_SOC_BIAS_OFF: + mask = JZ4740_CODEC_1_SUSPEND; + value = JZ4740_CODEC_1_SUSPEND; + + snd_soc_update_bits(codec, JZ4740_REG_CODEC_1, mask, value); + break; + default: + break; + } + + codec->bias_level = level; + + return 0; +} + +static struct snd_soc_codec *jz4740_codec_codec; + +static int jz4740_codec_dev_probe(struct platform_device *pdev) +{ + int ret; + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = jz4740_codec_codec; + + BUG_ON(!codec); + + socdev->card->codec = codec; + + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret) { + dev_err(&pdev->dev, "Failed to create pcms: %d\n", ret); + return ret; + } + + snd_soc_add_controls(codec, jz4740_codec_controls, + ARRAY_SIZE(jz4740_codec_controls)); + + snd_soc_dapm_new_controls(codec, jz4740_codec_dapm_widgets, + ARRAY_SIZE(jz4740_codec_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, jz4740_codec_dapm_routes, + ARRAY_SIZE(jz4740_codec_dapm_routes)); + + snd_soc_dapm_new_widgets(codec); + + return 0; +} + +static int jz4740_codec_dev_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +#ifdef CONFIG_PM_SLEEP + +static int jz4740_codec_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + return jz4740_codec_set_bias_level(codec, SND_SOC_BIAS_OFF); +} + +static int jz4740_codec_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + return jz4740_codec_set_bias_level(codec, SND_SOC_BIAS_STANDBY); +} + +#else +#define jz4740_codec_suspend NULL +#define jz4740_codec_resume NULL +#endif + +struct snd_soc_codec_device soc_codec_dev_jz4740_codec = { + .probe = jz4740_codec_dev_probe, + .remove = jz4740_codec_dev_remove, + .suspend = jz4740_codec_suspend, + .resume = jz4740_codec_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_jz4740_codec); + +static int __devinit jz4740_codec_probe(struct platform_device *pdev) +{ + int ret; + struct jz4740_codec *jz4740_codec; + struct snd_soc_codec *codec; + struct resource *mem; + + jz4740_codec = kzalloc(sizeof(*jz4740_codec), GFP_KERNEL); + if (!jz4740_codec) + return -ENOMEM; + + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!mem) { + dev_err(&pdev->dev, "Failed to get mmio memory resource\n"); + ret = -ENOENT; + goto err_free_codec; + } + + mem = request_mem_region(mem->start, resource_size(mem), pdev->name); + if (!mem) { + dev_err(&pdev->dev, "Failed to request mmio memory region\n"); + ret = -EBUSY; + goto err_free_codec; + } + + jz4740_codec->base = ioremap(mem->start, resource_size(mem)); + if (!jz4740_codec->base) { + dev_err(&pdev->dev, "Failed to ioremap mmio memory\n"); + ret = -EBUSY; + goto err_release_mem_region; + } + jz4740_codec->mem = mem; + + jz4740_codec_dai.dev = &pdev->dev; + + codec = &jz4740_codec->codec; + + codec->dev = &pdev->dev; + codec->name = "jz4740"; + codec->owner = THIS_MODULE; + + codec->read = jz4740_codec_read; + codec->write = jz4740_codec_write; + codec->set_bias_level = jz4740_codec_set_bias_level; + codec->bias_level = SND_SOC_BIAS_OFF; + + codec->dai = &jz4740_codec_dai; + codec->num_dai = 1; + + codec->reg_cache = jz4740_codec->reg_cache; + codec->reg_cache_size = 2; + memcpy(codec->reg_cache, jz4740_codec_regs, sizeof(jz4740_codec_regs)); + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + jz4740_codec_codec = codec; + + snd_soc_update_bits(codec, JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_SW2_ENABLE, JZ4740_CODEC_1_SW2_ENABLE); + + platform_set_drvdata(pdev, jz4740_codec); + + ret = snd_soc_register_codec(codec); + if (ret) { + dev_err(&pdev->dev, "Failed to register codec\n"); + goto err_iounmap; + } + + ret = snd_soc_register_dai(&jz4740_codec_dai); + if (ret) { + dev_err(&pdev->dev, "Failed to register codec dai\n"); + goto err_unregister_codec; + } + + jz4740_codec_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; + +err_unregister_codec: + snd_soc_unregister_codec(codec); +err_iounmap: + iounmap(jz4740_codec->base); +err_release_mem_region: + release_mem_region(mem->start, resource_size(mem)); +err_free_codec: + kfree(jz4740_codec); + + return ret; +} + +static int __devexit jz4740_codec_remove(struct platform_device *pdev) +{ + struct jz4740_codec *jz4740_codec = platform_get_drvdata(pdev); + struct resource *mem = jz4740_codec->mem; + + snd_soc_unregister_dai(&jz4740_codec_dai); + snd_soc_unregister_codec(&jz4740_codec->codec); + + iounmap(jz4740_codec->base); + release_mem_region(mem->start, resource_size(mem)); + + platform_set_drvdata(pdev, NULL); + kfree(jz4740_codec); + + return 0; +} + +static struct platform_driver jz4740_codec_driver = { + .probe = jz4740_codec_probe, + .remove = __devexit_p(jz4740_codec_remove), + .driver = { + .name = "jz4740-codec", + .owner = THIS_MODULE, + }, +}; + +static int __init jz4740_codec_init(void) +{ + return platform_driver_register(&jz4740_codec_driver); +} +module_init(jz4740_codec_init); + +static void __exit jz4740_codec_exit(void) +{ + platform_driver_unregister(&jz4740_codec_driver); +} +module_exit(jz4740_codec_exit); + +MODULE_DESCRIPTION("JZ4740 SoC internal codec driver"); +MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:jz4740-codec"); diff --git a/sound/soc/codecs/jz4740.h b/sound/soc/codecs/jz4740.h new file mode 100644 index 000000000000..b5a0691be763 --- /dev/null +++ b/sound/soc/codecs/jz4740.h @@ -0,0 +1,20 @@ +/* + * Copyright (C) 2009, Lars-Peter Clausen <lars@metafoo.de> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 675 Mass Ave, Cambridge, MA 02139, USA. + * + */ + +#ifndef __SND_SOC_CODECS_JZ4740_CODEC_H__ +#define __SND_SOC_CODECS_JZ4740_CODEC_H__ + +extern struct snd_soc_dai jz4740_codec_dai; +extern struct snd_soc_codec_device soc_codec_dev_jz4740_codec; + +#endif diff --git a/sound/soc/codecs/spdif_transciever.c b/sound/soc/codecs/spdif_transciever.c index a63191141052..9119836051a4 100644 --- a/sound/soc/codecs/spdif_transciever.c +++ b/sound/soc/codecs/spdif_transciever.c @@ -16,8 +16,10 @@ #include <linux/module.h> #include <linux/moduleparam.h> +#include <linux/slab.h> #include <sound/soc.h> #include <sound/pcm.h> +#include <sound/initval.h> #include "spdif_transciever.h" @@ -26,6 +28,48 @@ MODULE_LICENSE("GPL"); #define STUB_RATES SNDRV_PCM_RATE_8000_96000 #define STUB_FORMATS SNDRV_PCM_FMTBIT_S16_LE +static struct snd_soc_codec *spdif_dit_codec; + +static int spdif_dit_codec_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret; + + if (spdif_dit_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = spdif_dit_codec; + codec = spdif_dit_codec; + + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms: %d\n", ret); + goto err_create_pcms; + } + + return 0; + +err_create_pcms: + return ret; +} + +static int spdif_dit_codec_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_spdif_dit = { + .probe = spdif_dit_codec_probe, + .remove = spdif_dit_codec_remove, +}; EXPORT_SYMBOL_GPL(soc_codec_dev_spdif_dit); + struct snd_soc_dai dit_stub_dai = { .name = "DIT", .playback = { @@ -40,13 +84,61 @@ EXPORT_SYMBOL_GPL(dit_stub_dai); static int spdif_dit_probe(struct platform_device *pdev) { + struct snd_soc_codec *codec; + int ret; + + if (spdif_dit_codec) { + dev_err(&pdev->dev, "Another Codec is registered\n"); + ret = -EINVAL; + goto err_reg_codec; + } + + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + codec->dev = &pdev->dev; + + mutex_init(&codec->mutex); + + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->name = "spdif-dit"; + codec->owner = THIS_MODULE; + codec->dai = &dit_stub_dai; + codec->num_dai = 1; + + spdif_dit_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto err_reg_codec; + } + dit_stub_dai.dev = &pdev->dev; - return snd_soc_register_dai(&dit_stub_dai); + ret = snd_soc_register_dai(&dit_stub_dai); + if (ret < 0) { + dev_err(codec->dev, "Failed to register dai: %d\n", ret); + goto err_reg_dai; + } + + return 0; + +err_reg_dai: + snd_soc_unregister_codec(codec); +err_reg_codec: + kfree(spdif_dit_codec); + return ret; } static int spdif_dit_remove(struct platform_device *pdev) { snd_soc_unregister_dai(&dit_stub_dai); + snd_soc_unregister_codec(spdif_dit_codec); + kfree(spdif_dit_codec); + spdif_dit_codec = NULL; return 0; } diff --git a/sound/soc/codecs/spdif_transciever.h b/sound/soc/codecs/spdif_transciever.h index 296f2eb6c4ef..1e102124f546 100644 --- a/sound/soc/codecs/spdif_transciever.h +++ b/sound/soc/codecs/spdif_transciever.h @@ -12,6 +12,7 @@ #ifndef CODEC_STUBS_H #define CODEC_STUBS_H +extern struct snd_soc_codec_device soc_codec_dev_spdif_dit; extern struct snd_soc_dai dit_stub_dai; #endif /* CODEC_STUBS_H */ diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index b0bae3508b29..0a4b0fef3355 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -560,13 +560,16 @@ static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: /* vref/mid, osc on, dac unmute */ + reg &= ~(TLV320AIC23_DEVICE_PWR_OFF | TLV320AIC23_OSC_OFF | \ + TLV320AIC23_DAC_OFF); tlv320aic23_write(codec, TLV320AIC23_PWR, reg); break; case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: /* everything off except vref/vmid, */ - tlv320aic23_write(codec, TLV320AIC23_PWR, reg | 0x0040); + tlv320aic23_write(codec, TLV320AIC23_PWR, reg | \ + TLV320AIC23_CLK_OFF); break; case SND_SOC_BIAS_OFF: /* everything off, dac mute, inactive */ @@ -615,7 +618,6 @@ static int tlv320aic23_suspend(struct platform_device *pdev, struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; - tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0); tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; @@ -632,7 +634,6 @@ static int tlv320aic23_resume(struct platform_device *pdev) u16 val = tlv320aic23_read_reg_cache(codec, reg); tlv320aic23_write(codec, reg, val); } - tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 65adc77eada1..8651b01ed223 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -49,8 +49,6 @@ #define NSAMPLE_MAX 5700 -#define LATENCY_TIME_MS 20 - #define MODE7_LTHR 10 #define MODE7_UTHR (DAC33_BUFFER_SIZE_SAMPLES - 10) @@ -62,6 +60,9 @@ #define US_TO_SAMPLES(rate, us) \ (rate / (1000000 / us)) +#define UTHR_FROM_PERIOD_SIZE(samples, playrate, burstrate) \ + ((samples * 5000) / ((burstrate * 5000) / (burstrate - playrate))) + static void dac33_calculate_times(struct snd_pcm_substream *substream); static int dac33_prepare_chip(struct snd_pcm_substream *substream); @@ -107,6 +108,10 @@ struct tlv320dac33_priv { * this */ enum dac33_fifo_modes fifo_mode;/* FIFO mode selection */ unsigned int nsample; /* burst read amount from host */ + int mode1_latency; /* latency caused by the i2c writes in + * us */ + int auto_fifo_config; /* Configure the FIFO based on the + * period size */ u8 burst_bclkdiv; /* BCLK divider value in burst mode */ unsigned int burst_rate; /* Interface speed in Burst modes */ @@ -120,6 +125,8 @@ struct tlv320dac33_priv { * samples */ unsigned int mode7_us_to_lthr; /* Time to reach lthr from uthr */ + unsigned int uthr; + enum dac33_state state; }; @@ -442,6 +449,39 @@ static int dac33_set_nsample(struct snd_kcontrol *kcontrol, return ret; } +static int dac33_get_uthr(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = dac33->uthr; + + return 0; +} + +static int dac33_set_uthr(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + if (dac33->substream) + return -EBUSY; + + if (dac33->uthr == ucontrol->value.integer.value[0]) + return 0; + + if (ucontrol->value.integer.value[0] < (MODE7_LTHR + 10) || + ucontrol->value.integer.value[0] > MODE7_UTHR) + ret = -EINVAL; + else + dac33->uthr = ucontrol->value.integer.value[0]; + + return ret; +} + static int dac33_get_fifo_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -503,13 +543,18 @@ static const struct snd_kcontrol_new dac33_snd_controls[] = { DAC33_LINEL_TO_LLO_VOL, DAC33_LINER_TO_RLO_VOL, 0, 127, 1), }; -static const struct snd_kcontrol_new dac33_nsample_snd_controls[] = { - SOC_SINGLE_EXT("nSample", 0, 0, 5900, 0, - dac33_get_nsample, dac33_set_nsample), +static const struct snd_kcontrol_new dac33_mode_snd_controls[] = { SOC_ENUM_EXT("FIFO Mode", dac33_fifo_mode_enum, dac33_get_fifo_mode, dac33_set_fifo_mode), }; +static const struct snd_kcontrol_new dac33_fifo_snd_controls[] = { + SOC_SINGLE_EXT("nSample", 0, 0, 5900, 0, + dac33_get_nsample, dac33_set_nsample), + SOC_SINGLE_EXT("UTHR", 0, 0, MODE7_UTHR, 0, + dac33_get_uthr, dac33_set_uthr), +}; + /* Analog bypass */ static const struct snd_kcontrol_new dac33_dapm_abypassl_control = SOC_DAPM_SINGLE("Switch", DAC33_LINEL_TO_LLO_VOL, 7, 1, 1); @@ -612,7 +657,7 @@ static inline void dac33_prefill_handler(struct tlv320dac33_priv *dac33) switch (dac33->fifo_mode) { case DAC33_FIFO_MODE1: dac33_write16(codec, DAC33_NSAMPLE_MSB, - DAC33_THRREG(dac33->nsample + dac33->alarm_threshold)); + DAC33_THRREG(dac33->nsample)); /* Take the timestamps */ spin_lock_irq(&dac33->lock); @@ -761,6 +806,10 @@ static void dac33_shutdown(struct snd_pcm_substream *substream, struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); dac33->substream = NULL; + + /* Reset the nSample restrictions */ + dac33->nsample_min = 0; + dac33->nsample_max = NSAMPLE_MAX; } static int dac33_hw_params(struct snd_pcm_substream *substream, @@ -985,7 +1034,7 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream) * Configure the threshold levels, and leave 10 sample space * at the bottom, and also at the top of the FIFO */ - dac33_write16(codec, DAC33_UTHR_MSB, DAC33_THRREG(MODE7_UTHR)); + dac33_write16(codec, DAC33_UTHR_MSB, DAC33_THRREG(dac33->uthr)); dac33_write16(codec, DAC33_LTHR_MSB, DAC33_THRREG(MODE7_LTHR)); break; default: @@ -1003,57 +1052,71 @@ static void dac33_calculate_times(struct snd_pcm_substream *substream) struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); + unsigned int period_size = substream->runtime->period_size; + unsigned int rate = substream->runtime->rate; unsigned int nsample_limit; /* In bypass mode we don't need to calculate */ if (!dac33->fifo_mode) return; - /* Number of samples (16bit, stereo) in one period */ - dac33->nsample_min = snd_pcm_lib_period_bytes(substream) / 4; - - /* Number of samples (16bit, stereo) in ALSA buffer */ - dac33->nsample_max = snd_pcm_lib_buffer_bytes(substream) / 4; - /* Subtract one period from the total */ - dac33->nsample_max -= dac33->nsample_min; - - /* Number of samples for LATENCY_TIME_MS / 2 */ - dac33->alarm_threshold = substream->runtime->rate / - (1000 / (LATENCY_TIME_MS / 2)); - - /* Find and fix up the lowest nsmaple limit */ - nsample_limit = substream->runtime->rate / (1000 / LATENCY_TIME_MS); - - if (dac33->nsample_min < nsample_limit) - dac33->nsample_min = nsample_limit; - - if (dac33->nsample < dac33->nsample_min) - dac33->nsample = dac33->nsample_min; - - /* - * Find and fix up the highest nsmaple limit - * In order to not overflow the DAC33 buffer substract the - * alarm_threshold value from the size of the DAC33 buffer - */ - nsample_limit = DAC33_BUFFER_SIZE_SAMPLES - dac33->alarm_threshold; - - if (dac33->nsample_max > nsample_limit) - dac33->nsample_max = nsample_limit; - - if (dac33->nsample > dac33->nsample_max) - dac33->nsample = dac33->nsample_max; - switch (dac33->fifo_mode) { case DAC33_FIFO_MODE1: + /* Number of samples under i2c latency */ + dac33->alarm_threshold = US_TO_SAMPLES(rate, + dac33->mode1_latency); + if (dac33->auto_fifo_config) { + if (period_size <= dac33->alarm_threshold) + /* + * Configure nSamaple to number of periods, + * which covers the latency requironment. + */ + dac33->nsample = period_size * + ((dac33->alarm_threshold / period_size) + + (dac33->alarm_threshold % period_size ? + 1 : 0)); + else + dac33->nsample = period_size; + } else { + /* nSample time shall not be shorter than i2c latency */ + dac33->nsample_min = dac33->alarm_threshold; + /* + * nSample should not be bigger than alsa buffer minus + * size of one period to avoid overruns + */ + dac33->nsample_max = substream->runtime->buffer_size - + period_size; + nsample_limit = DAC33_BUFFER_SIZE_SAMPLES - + dac33->alarm_threshold; + if (dac33->nsample_max > nsample_limit) + dac33->nsample_max = nsample_limit; + + /* Correct the nSample if it is outside of the ranges */ + if (dac33->nsample < dac33->nsample_min) + dac33->nsample = dac33->nsample_min; + if (dac33->nsample > dac33->nsample_max) + dac33->nsample = dac33->nsample_max; + } + dac33->mode1_us_burst = SAMPLES_TO_US(dac33->burst_rate, dac33->nsample); dac33->t_stamp1 = 0; dac33->t_stamp2 = 0; break; case DAC33_FIFO_MODE7: + if (dac33->auto_fifo_config) { + dac33->uthr = UTHR_FROM_PERIOD_SIZE( + period_size, + rate, + dac33->burst_rate) + 9; + if (dac33->uthr > MODE7_UTHR) + dac33->uthr = MODE7_UTHR; + if (dac33->uthr < (MODE7_LTHR + 10)) + dac33->uthr = (MODE7_LTHR + 10); + } dac33->mode7_us_to_lthr = - SAMPLES_TO_US(substream->runtime->rate, - MODE7_UTHR - MODE7_LTHR + 1); + SAMPLES_TO_US(substream->runtime->rate, + dac33->uthr - MODE7_LTHR + 1); dac33->t_stamp1 = 0; break; default: @@ -1104,7 +1167,7 @@ static snd_pcm_sframes_t dac33_dai_delay( struct snd_soc_codec *codec = socdev->card->codec; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); unsigned long long t0, t1, t_now; - unsigned int time_delta; + unsigned int time_delta, uthr; int samples_out, samples_in, samples; snd_pcm_sframes_t delay = 0; @@ -1182,6 +1245,7 @@ static snd_pcm_sframes_t dac33_dai_delay( case DAC33_FIFO_MODE7: spin_lock(&dac33->lock); t0 = dac33->t_stamp1; + uthr = dac33->uthr; spin_unlock(&dac33->lock); t_now = ktime_to_us(ktime_get()); @@ -1194,7 +1258,7 @@ static snd_pcm_sframes_t dac33_dai_delay( * Either the timestamps are messed or equal. Report * maximum delay */ - delay = MODE7_UTHR; + delay = uthr; goto out; } @@ -1208,8 +1272,8 @@ static snd_pcm_sframes_t dac33_dai_delay( substream->runtime->rate, time_delta); - if (likely(MODE7_UTHR > samples_out)) - delay = MODE7_UTHR - samples_out; + if (likely(uthr > samples_out)) + delay = uthr - samples_out; else delay = 0; } else { @@ -1227,8 +1291,8 @@ static snd_pcm_sframes_t dac33_dai_delay( time_delta); delay = MODE7_LTHR + samples_in - samples_out; - if (unlikely(delay > MODE7_UTHR)) - delay = MODE7_UTHR; + if (unlikely(delay > uthr)) + delay = uthr; } break; default: @@ -1347,10 +1411,15 @@ static int dac33_soc_probe(struct platform_device *pdev) snd_soc_add_controls(codec, dac33_snd_controls, ARRAY_SIZE(dac33_snd_controls)); - /* Only add the nSample controls, if we have valid IRQ number */ - if (dac33->irq >= 0) - snd_soc_add_controls(codec, dac33_nsample_snd_controls, - ARRAY_SIZE(dac33_nsample_snd_controls)); + /* Only add the FIFO controls, if we have valid IRQ number */ + if (dac33->irq >= 0) { + snd_soc_add_controls(codec, dac33_mode_snd_controls, + ARRAY_SIZE(dac33_mode_snd_controls)); + /* FIFO usage controls only, if autoio config is not selected */ + if (!dac33->auto_fifo_config) + snd_soc_add_controls(codec, dac33_fifo_snd_controls, + ARRAY_SIZE(dac33_fifo_snd_controls)); + } dac33_add_widgets(codec); @@ -1481,9 +1550,14 @@ static int __devinit dac33_i2c_probe(struct i2c_client *client, /* Pre calculate the burst rate */ dac33->burst_rate = BURST_BASEFREQ_HZ / dac33->burst_bclkdiv / 32; dac33->keep_bclk = pdata->keep_bclk; + dac33->auto_fifo_config = pdata->auto_fifo_config; + dac33->mode1_latency = pdata->mode1_latency; + if (!dac33->mode1_latency) + dac33->mode1_latency = 10000; /* 10ms */ dac33->irq = client->irq; dac33->nsample = NSAMPLE_MAX; dac33->nsample_max = NSAMPLE_MAX; + dac33->uthr = MODE7_UTHR; /* Disable FIFO use by default */ dac33->fifo_mode = DAC33_FIFO_BYPASS; diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index b4fcdb01fc49..7b618bbff884 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -43,37 +43,37 @@ */ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0x00, /* this register not used */ - 0x91, /* REG_CODEC_MODE (0x1) */ - 0xc3, /* REG_OPTION (0x2) */ + 0x00, /* REG_CODEC_MODE (0x1) */ + 0x00, /* REG_OPTION (0x2) */ 0x00, /* REG_UNKNOWN (0x3) */ 0x00, /* REG_MICBIAS_CTL (0x4) */ - 0x20, /* REG_ANAMICL (0x5) */ + 0x00, /* REG_ANAMICL (0x5) */ 0x00, /* REG_ANAMICR (0x6) */ 0x00, /* REG_AVADC_CTL (0x7) */ 0x00, /* REG_ADCMICSEL (0x8) */ 0x00, /* REG_DIGMIXING (0x9) */ - 0x0c, /* REG_ATXL1PGA (0xA) */ - 0x0c, /* REG_ATXR1PGA (0xB) */ - 0x00, /* REG_AVTXL2PGA (0xC) */ - 0x00, /* REG_AVTXR2PGA (0xD) */ + 0x0f, /* REG_ATXL1PGA (0xA) */ + 0x0f, /* REG_ATXR1PGA (0xB) */ + 0x0f, /* REG_AVTXL2PGA (0xC) */ + 0x0f, /* REG_AVTXR2PGA (0xD) */ 0x00, /* REG_AUDIO_IF (0xE) */ 0x00, /* REG_VOICE_IF (0xF) */ - 0x00, /* REG_ARXR1PGA (0x10) */ - 0x00, /* REG_ARXL1PGA (0x11) */ - 0x6c, /* REG_ARXR2PGA (0x12) */ - 0x6c, /* REG_ARXL2PGA (0x13) */ - 0x00, /* REG_VRXPGA (0x14) */ + 0x3f, /* REG_ARXR1PGA (0x10) */ + 0x3f, /* REG_ARXL1PGA (0x11) */ + 0x3f, /* REG_ARXR2PGA (0x12) */ + 0x3f, /* REG_ARXL2PGA (0x13) */ + 0x25, /* REG_VRXPGA (0x14) */ 0x00, /* REG_VSTPGA (0x15) */ 0x00, /* REG_VRX2ARXPGA (0x16) */ 0x00, /* REG_AVDAC_CTL (0x17) */ 0x00, /* REG_ARX2VTXPGA (0x18) */ - 0x00, /* REG_ARXL1_APGA_CTL (0x19) */ - 0x00, /* REG_ARXR1_APGA_CTL (0x1A) */ - 0x4a, /* REG_ARXL2_APGA_CTL (0x1B) */ - 0x4a, /* REG_ARXR2_APGA_CTL (0x1C) */ + 0x32, /* REG_ARXL1_APGA_CTL (0x19) */ + 0x32, /* REG_ARXR1_APGA_CTL (0x1A) */ + 0x32, /* REG_ARXL2_APGA_CTL (0x1B) */ + 0x32, /* REG_ARXR2_APGA_CTL (0x1C) */ 0x00, /* REG_ATX2ARXPGA (0x1D) */ 0x00, /* REG_BT_IF (0x1E) */ - 0x00, /* REG_BTPGA (0x1F) */ + 0x55, /* REG_BTPGA (0x1F) */ 0x00, /* REG_BTSTPGA (0x20) */ 0x00, /* REG_EAR_CTL (0x21) */ 0x00, /* REG_HS_SEL (0x22) */ @@ -85,32 +85,32 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0x00, /* REG_PRECKR_CTL (0x28) */ 0x00, /* REG_HFL_CTL (0x29) */ 0x00, /* REG_HFR_CTL (0x2A) */ - 0x00, /* REG_ALC_CTL (0x2B) */ + 0x05, /* REG_ALC_CTL (0x2B) */ 0x00, /* REG_ALC_SET1 (0x2C) */ 0x00, /* REG_ALC_SET2 (0x2D) */ 0x00, /* REG_BOOST_CTL (0x2E) */ 0x00, /* REG_SOFTVOL_CTL (0x2F) */ - 0x00, /* REG_DTMF_FREQSEL (0x30) */ + 0x13, /* REG_DTMF_FREQSEL (0x30) */ 0x00, /* REG_DTMF_TONEXT1H (0x31) */ 0x00, /* REG_DTMF_TONEXT1L (0x32) */ 0x00, /* REG_DTMF_TONEXT2H (0x33) */ 0x00, /* REG_DTMF_TONEXT2L (0x34) */ - 0x00, /* REG_DTMF_TONOFF (0x35) */ - 0x00, /* REG_DTMF_WANONOFF (0x36) */ + 0x79, /* REG_DTMF_TONOFF (0x35) */ + 0x11, /* REG_DTMF_WANONOFF (0x36) */ 0x00, /* REG_I2S_RX_SCRAMBLE_H (0x37) */ 0x00, /* REG_I2S_RX_SCRAMBLE_M (0x38) */ 0x00, /* REG_I2S_RX_SCRAMBLE_L (0x39) */ 0x06, /* REG_APLL_CTL (0x3A) */ 0x00, /* REG_DTMF_CTL (0x3B) */ - 0x00, /* REG_DTMF_PGA_CTL2 (0x3C) */ - 0x00, /* REG_DTMF_PGA_CTL1 (0x3D) */ + 0x44, /* REG_DTMF_PGA_CTL2 (0x3C) */ + 0x69, /* REG_DTMF_PGA_CTL1 (0x3D) */ 0x00, /* REG_MISC_SET_1 (0x3E) */ 0x00, /* REG_PCMBTMUX (0x3F) */ 0x00, /* not used (0x40) */ 0x00, /* not used (0x41) */ 0x00, /* not used (0x42) */ 0x00, /* REG_RX_PATH_SEL (0x43) */ - 0x00, /* REG_VDL_APGA_CTL (0x44) */ + 0x32, /* REG_VDL_APGA_CTL (0x44) */ 0x00, /* REG_VIBRA_CTL (0x45) */ 0x00, /* REG_VIBRA_SET (0x46) */ 0x00, /* REG_VIBRA_PWM_SET (0x47) */ @@ -143,6 +143,9 @@ struct twl4030_priv { u8 earpiece_enabled; u8 predrivel_enabled, predriver_enabled; u8 carkitl_enabled, carkitr_enabled; + + /* Delay needed after enabling the digimic interface */ + unsigned int digimic_delay; }; /* @@ -244,58 +247,95 @@ static void twl4030_codec_enable(struct snd_soc_codec *codec, int enable) udelay(10); } -static void twl4030_init_chip(struct snd_soc_codec *codec) +static inline void twl4030_check_defaults(struct snd_soc_codec *codec) { - u8 *cache = codec->reg_cache; - int i; + int i, difference = 0; + u8 val; + + dev_dbg(codec->dev, "Checking TWL audio default configuration\n"); + for (i = 1; i <= TWL4030_REG_MISC_SET_2; i++) { + twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &val, i); + if (val != twl4030_reg[i]) { + difference++; + dev_dbg(codec->dev, + "Reg 0x%02x: chip: 0x%02x driver: 0x%02x\n", + i, val, twl4030_reg[i]); + } + } + dev_dbg(codec->dev, "Found %d non maching registers. %s\n", + difference, difference ? "Not OK" : "OK"); +} - /* clear CODECPDZ prior to setting register defaults */ - twl4030_codec_enable(codec, 0); +static inline void twl4030_reset_registers(struct snd_soc_codec *codec) +{ + int i; /* set all audio section registers to reasonable defaults */ for (i = TWL4030_REG_OPTION; i <= TWL4030_REG_MISC_SET_2; i++) if (i != TWL4030_REG_APLL_CTL) - twl4030_write(codec, i, cache[i]); + twl4030_write(codec, i, twl4030_reg[i]); } -static void twl4030_apll_enable(struct snd_soc_codec *codec, int enable) +static void twl4030_init_chip(struct platform_device *pdev) { + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct twl4030_setup_data *setup = socdev->codec_data; + struct snd_soc_codec *codec = socdev->card->codec; struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); - int status = -1; + u8 reg, byte; + int i = 0; - if (enable) { - twl4030->apll_enabled++; - if (twl4030->apll_enabled == 1) - status = twl4030_codec_enable_resource( - TWL4030_CODEC_RES_APLL); - } else { - twl4030->apll_enabled--; - if (!twl4030->apll_enabled) - status = twl4030_codec_disable_resource( - TWL4030_CODEC_RES_APLL); - } + /* Check defaults, if instructed before anything else */ + if (setup && setup->check_defaults) + twl4030_check_defaults(codec); - if (status >= 0) - twl4030_write_reg_cache(codec, TWL4030_REG_APLL_CTL, status); -} + /* Reset registers, if no setup data or if instructed to do so */ + if (!setup || (setup && setup->reset_registers)) + twl4030_reset_registers(codec); -static void twl4030_power_up(struct snd_soc_codec *codec) -{ - struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); - u8 anamicl, regmisc1, byte; - int i = 0; + /* Refresh APLL_CTL register from HW */ + twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte, + TWL4030_REG_APLL_CTL); + twl4030_write_reg_cache(codec, TWL4030_REG_APLL_CTL, byte); + + /* anti-pop when changing analog gain */ + reg = twl4030_read_reg_cache(codec, TWL4030_REG_MISC_SET_1); + twl4030_write(codec, TWL4030_REG_MISC_SET_1, + reg | TWL4030_SMOOTH_ANAVOL_EN); - if (twl4030->codec_powered) + twl4030_write(codec, TWL4030_REG_OPTION, + TWL4030_ATXL1_EN | TWL4030_ATXR1_EN | + TWL4030_ARXL2_EN | TWL4030_ARXR2_EN); + + /* REG_ARXR2_APGA_CTL reset according to the TRM: 0dB, DA_EN */ + twl4030_write(codec, TWL4030_REG_ARXR2_APGA_CTL, 0x32); + + /* Machine dependent setup */ + if (!setup) return; - /* set CODECPDZ to turn on codec */ - twl4030_codec_enable(codec, 1); + twl4030->digimic_delay = setup->digimic_delay; + + /* Configuration for headset ramp delay from setup data */ + if (setup->sysclk != twl4030->sysclk) + dev_warn(codec->dev, + "Mismatch in APLL mclk: %u (configured: %u)\n", + setup->sysclk, twl4030->sysclk); + + reg = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET); + reg &= ~TWL4030_RAMP_DELAY; + reg |= (setup->ramp_delay_value << 2); + twl4030_write_reg_cache(codec, TWL4030_REG_HS_POPN_SET, reg); /* initiate offset cancellation */ - anamicl = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL); + twl4030_codec_enable(codec, 1); + + reg = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL); + reg &= ~TWL4030_OFFSET_CNCL_SEL; + reg |= setup->offset_cncl_path; twl4030_write(codec, TWL4030_REG_ANAMICL, - anamicl | TWL4030_CNCL_OFFSET_START); + reg | TWL4030_CNCL_OFFSET_START); /* wait for offset cancellation to complete */ do { @@ -310,23 +350,28 @@ static void twl4030_power_up(struct snd_soc_codec *codec) /* Make sure that the reg_cache has the same value as the HW */ twl4030_write_reg_cache(codec, TWL4030_REG_ANAMICL, byte); - /* anti-pop when changing analog gain */ - regmisc1 = twl4030_read_reg_cache(codec, TWL4030_REG_MISC_SET_1); - twl4030_write(codec, TWL4030_REG_MISC_SET_1, - regmisc1 | TWL4030_SMOOTH_ANAVOL_EN); - - /* toggle CODECPDZ as per TRM */ twl4030_codec_enable(codec, 0); - twl4030_codec_enable(codec, 1); } -/* - * Unconditional power down - */ -static void twl4030_power_down(struct snd_soc_codec *codec) +static void twl4030_apll_enable(struct snd_soc_codec *codec, int enable) { - /* power down */ - twl4030_codec_enable(codec, 0); + struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); + int status = -1; + + if (enable) { + twl4030->apll_enabled++; + if (twl4030->apll_enabled == 1) + status = twl4030_codec_enable_resource( + TWL4030_CODEC_RES_APLL); + } else { + twl4030->apll_enabled--; + if (!twl4030->apll_enabled) + status = twl4030_codec_disable_resource( + TWL4030_CODEC_RES_APLL); + } + + if (status >= 0) + twl4030_write_reg_cache(codec, TWL4030_REG_APLL_CTL, status); } /* Earpiece */ @@ -500,10 +545,11 @@ static const struct snd_kcontrol_new twl4030_dapm_abypassl2_control = static const struct snd_kcontrol_new twl4030_dapm_abypassv_control = SOC_DAPM_SINGLE("Switch", TWL4030_REG_VDL_APGA_CTL, 2, 1, 0); -/* Digital bypass gain, 0 mutes the bypass */ +/* Digital bypass gain, mute instead of -30dB */ static const unsigned int twl4030_dapm_dbypass_tlv[] = { - TLV_DB_RANGE_HEAD(2), - 0, 3, TLV_DB_SCALE_ITEM(-2400, 0, 1), + TLV_DB_RANGE_HEAD(3), + 0, 1, TLV_DB_SCALE_ITEM(-3000, 600, 1), + 2, 3, TLV_DB_SCALE_ITEM(-2400, 0, 0), 4, 7, TLV_DB_SCALE_ITEM(-1800, 600, 0), }; @@ -531,36 +577,6 @@ static const struct snd_kcontrol_new twl4030_dapm_dbypassv_control = TWL4030_REG_VSTPGA, 0, 0x29, 0, twl4030_dapm_dbypassv_tlv); -static int micpath_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - struct soc_enum *e = (struct soc_enum *)w->kcontrols->private_value; - unsigned char adcmicsel, micbias_ctl; - - adcmicsel = twl4030_read_reg_cache(w->codec, TWL4030_REG_ADCMICSEL); - micbias_ctl = twl4030_read_reg_cache(w->codec, TWL4030_REG_MICBIAS_CTL); - /* Prepare the bits for the given TX path: - * shift_l == 0: TX1 microphone path - * shift_l == 2: TX2 microphone path */ - if (e->shift_l) { - /* TX2 microphone path */ - if (adcmicsel & TWL4030_TX2IN_SEL) - micbias_ctl |= TWL4030_MICBIAS2_CTL; /* digimic */ - else - micbias_ctl &= ~TWL4030_MICBIAS2_CTL; - } else { - /* TX1 microphone path */ - if (adcmicsel & TWL4030_TX1IN_SEL) - micbias_ctl |= TWL4030_MICBIAS1_CTL; /* digimic */ - else - micbias_ctl &= ~TWL4030_MICBIAS1_CTL; - } - - twl4030_write(w->codec, TWL4030_REG_MICBIAS_CTL, micbias_ctl); - - return 0; -} - /* * Output PGA builder: * Handle the muting and unmuting of the given output (turning off the @@ -814,6 +830,16 @@ static int headsetrpga_event(struct snd_soc_dapm_widget *w, return 0; } +static int digimic_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(w->codec); + + if (twl4030->digimic_delay) + mdelay(twl4030->digimic_delay); + return 0; +} + /* * Some of the gain controls in TWL (mostly those which are associated with * the outputs) are implemented in an interesting way: @@ -1374,14 +1400,10 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { /* Analog/Digital mic path selection. TX1 Left/Right: either analog Left/Right or Digimic0 TX2 Left/Right: either analog Left/Right or Digimic1 */ - SND_SOC_DAPM_MUX_E("TX1 Capture Route", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_micpathtx1_control, micpath_event, - SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD| - SND_SOC_DAPM_POST_REG), - SND_SOC_DAPM_MUX_E("TX2 Capture Route", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_micpathtx2_control, micpath_event, - SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD| - SND_SOC_DAPM_POST_REG), + SND_SOC_DAPM_MUX("TX1 Capture Route", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_micpathtx1_control), + SND_SOC_DAPM_MUX("TX2 Capture Route", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_micpathtx2_control), /* Analog input mixers for the capture amplifiers */ SND_SOC_DAPM_MIXER("Analog Left", @@ -1398,10 +1420,17 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_PGA("ADC Physical Right", TWL4030_REG_AVADC_CTL, 1, 0, NULL, 0), - SND_SOC_DAPM_PGA("Digimic0 Enable", - TWL4030_REG_ADCMICSEL, 1, 0, NULL, 0), - SND_SOC_DAPM_PGA("Digimic1 Enable", - TWL4030_REG_ADCMICSEL, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA_E("Digimic0 Enable", + TWL4030_REG_ADCMICSEL, 1, 0, NULL, 0, + digimic_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_E("Digimic1 Enable", + TWL4030_REG_ADCMICSEL, 3, 0, NULL, 0, + digimic_event, SND_SOC_DAPM_POST_PMU), + + SND_SOC_DAPM_SUPPLY("micbias1 select", TWL4030_REG_MICBIAS_CTL, 5, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("micbias2 select", TWL4030_REG_MICBIAS_CTL, 6, 0, + NULL, 0), SND_SOC_DAPM_MICBIAS("Mic Bias 1", TWL4030_REG_MICBIAS_CTL, 0, 0), SND_SOC_DAPM_MICBIAS("Mic Bias 2", TWL4030_REG_MICBIAS_CTL, 1, 0), @@ -1419,8 +1448,11 @@ static const struct snd_soc_dapm_route intercon[] = { /* Supply for the digital part (APLL) */ {"Digital Voice Playback Mixer", NULL, "APLL Enable"}, - {"Digital R1 Playback Mixer", NULL, "AIF Enable"}, - {"Digital L1 Playback Mixer", NULL, "AIF Enable"}, + {"DAC Left1", NULL, "AIF Enable"}, + {"DAC Right1", NULL, "AIF Enable"}, + {"DAC Left2", NULL, "AIF Enable"}, + {"DAC Right1", NULL, "AIF Enable"}, + {"Digital R2 Playback Mixer", NULL, "AIF Enable"}, {"Digital L2 Playback Mixer", NULL, "AIF Enable"}, @@ -1491,10 +1523,10 @@ static const struct snd_soc_dapm_route intercon[] = { /* outputs */ /* Must be always connected (for AIF and APLL) */ - {"Virtual HiFi OUT", NULL, "Digital L1 Playback Mixer"}, - {"Virtual HiFi OUT", NULL, "Digital R1 Playback Mixer"}, - {"Virtual HiFi OUT", NULL, "Digital L2 Playback Mixer"}, - {"Virtual HiFi OUT", NULL, "Digital R2 Playback Mixer"}, + {"Virtual HiFi OUT", NULL, "DAC Left1"}, + {"Virtual HiFi OUT", NULL, "DAC Right1"}, + {"Virtual HiFi OUT", NULL, "DAC Left2"}, + {"Virtual HiFi OUT", NULL, "DAC Right2"}, /* Must be always connected (for APLL) */ {"Virtual Voice OUT", NULL, "Digital Voice Playback Mixer"}, /* Physical outputs */ @@ -1531,6 +1563,9 @@ static const struct snd_soc_dapm_route intercon[] = { {"Digimic0 Enable", NULL, "DIGIMIC0"}, {"Digimic1 Enable", NULL, "DIGIMIC1"}, + {"DIGIMIC0", NULL, "micbias1 select"}, + {"DIGIMIC1", NULL, "micbias2 select"}, + /* TX1 Left capture path */ {"TX1 Capture Route", "Analog", "ADC Physical Left"}, {"TX1 Capture Route", "Digimic0", "Digimic0 Enable"}, @@ -1605,10 +1640,10 @@ static int twl4030_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: if (codec->bias_level == SND_SOC_BIAS_OFF) - twl4030_power_up(codec); + twl4030_codec_enable(codec, 1); break; case SND_SOC_BIAS_OFF: - twl4030_power_down(codec); + twl4030_codec_enable(codec, 0); break; } codec->bias_level = level; @@ -1794,13 +1829,6 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - if (mode != old_mode) { - /* change rate and set CODECPDZ */ - twl4030_codec_enable(codec, 0); - twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode); - twl4030_codec_enable(codec, 1); - } - /* sample size */ old_format = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF); format = old_format; @@ -1818,16 +1846,20 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - if (format != old_format) { - - /* clear CODECPDZ before changing format (codec requirement) */ - twl4030_codec_enable(codec, 0); - - /* change format */ - twl4030_write(codec, TWL4030_REG_AUDIO_IF, format); - - /* set CODECPDZ afterwards */ - twl4030_codec_enable(codec, 1); + if (format != old_format || mode != old_mode) { + if (twl4030->codec_powered) { + /* + * If the codec is powered, than we need to toggle the + * codec power. + */ + twl4030_codec_enable(codec, 0); + twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode); + twl4030_write(codec, TWL4030_REG_AUDIO_IF, format); + twl4030_codec_enable(codec, 1); + } else { + twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode); + twl4030_write(codec, TWL4030_REG_AUDIO_IF, format); + } } /* Store the important parameters for the DAI configuration and set @@ -1877,6 +1909,7 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; + struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); u8 old_format, format; /* get format */ @@ -1911,15 +1944,17 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, } if (format != old_format) { - - /* clear CODECPDZ before changing format (codec requirement) */ - twl4030_codec_enable(codec, 0); - - /* change format */ - twl4030_write(codec, TWL4030_REG_AUDIO_IF, format); - - /* set CODECPDZ afterwards */ - twl4030_codec_enable(codec, 1); + if (twl4030->codec_powered) { + /* + * If the codec is powered, than we need to toggle the + * codec power. + */ + twl4030_codec_enable(codec, 0); + twl4030_write(codec, TWL4030_REG_AUDIO_IF, format); + twl4030_codec_enable(codec, 1); + } else { + twl4030_write(codec, TWL4030_REG_AUDIO_IF, format); + } } return 0; @@ -2011,6 +2046,7 @@ static int twl4030_voice_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; + struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); u8 old_mode, mode; /* Enable voice digital filters */ @@ -2035,10 +2071,17 @@ static int twl4030_voice_hw_params(struct snd_pcm_substream *substream, } if (mode != old_mode) { - /* change rate and set CODECPDZ */ - twl4030_codec_enable(codec, 0); - twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode); - twl4030_codec_enable(codec, 1); + if (twl4030->codec_powered) { + /* + * If the codec is powered, than we need to toggle the + * codec power. + */ + twl4030_codec_enable(codec, 0); + twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode); + twl4030_codec_enable(codec, 1); + } else { + twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode); + } } return 0; @@ -2068,6 +2111,7 @@ static int twl4030_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; + struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); u8 old_format, format; /* get format */ @@ -2099,10 +2143,17 @@ static int twl4030_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, } if (format != old_format) { - /* change format and set CODECPDZ */ - twl4030_codec_enable(codec, 0); - twl4030_write(codec, TWL4030_REG_VOICE_IF, format); - twl4030_codec_enable(codec, 1); + if (twl4030->codec_powered) { + /* + * If the codec is powered, than we need to toggle the + * codec power. + */ + twl4030_codec_enable(codec, 0); + twl4030_write(codec, TWL4030_REG_VOICE_IF, format); + twl4030_codec_enable(codec, 1); + } else { + twl4030_write(codec, TWL4030_REG_VOICE_IF, format); + } } return 0; @@ -2202,31 +2253,15 @@ static struct snd_soc_codec *twl4030_codec; static int twl4030_soc_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct twl4030_setup_data *setup = socdev->codec_data; struct snd_soc_codec *codec; - struct twl4030_priv *twl4030; int ret; BUG_ON(!twl4030_codec); codec = twl4030_codec; - twl4030 = snd_soc_codec_get_drvdata(codec); socdev->card->codec = codec; - /* Configuration for headset ramp delay from setup data */ - if (setup) { - unsigned char hs_pop; - - if (setup->sysclk != twl4030->sysclk) - dev_warn(&pdev->dev, - "Mismatch in APLL mclk: %u (configured: %u)\n", - setup->sysclk, twl4030->sysclk); - - hs_pop = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET); - hs_pop &= ~TWL4030_RAMP_DELAY; - hs_pop |= (setup->ramp_delay_value << 2); - twl4030_write_reg_cache(codec, TWL4030_REG_HS_POPN_SET, hs_pop); - } + twl4030_init_chip(pdev); /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); @@ -2247,6 +2282,8 @@ static int twl4030_soc_remove(struct platform_device *pdev) struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; + /* Reset registers to their chip default before leaving */ + twl4030_reset_registers(codec); twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF); snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); @@ -2287,6 +2324,7 @@ static int __devinit twl4030_codec_probe(struct platform_device *pdev) codec->read = twl4030_read_reg_cache; codec->write = twl4030_write; codec->set_bias_level = twl4030_set_bias_level; + codec->idle_bias_off = 1; codec->dai = twl4030_dai; codec->num_dai = ARRAY_SIZE(twl4030_dai); codec->reg_cache_size = sizeof(twl4030_reg); @@ -2302,9 +2340,7 @@ static int __devinit twl4030_codec_probe(struct platform_device *pdev) /* Set the defaults, and power up the codec */ twl4030->sysclk = twl4030_codec_get_mclk() / 1000; - twl4030_init_chip(codec); codec->bias_level = SND_SOC_BIAS_OFF; - twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY); ret = snd_soc_register_codec(codec); if (ret != 0) { @@ -2322,7 +2358,7 @@ static int __devinit twl4030_codec_probe(struct platform_device *pdev) return 0; error_codec: - twl4030_power_down(codec); + twl4030_codec_enable(codec, 0); kfree(codec->reg_cache); error_cache: kfree(twl4030); diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index f206d242ca31..6c57430f6e24 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -41,7 +41,11 @@ extern struct snd_soc_codec_device soc_codec_dev_twl4030; struct twl4030_setup_data { unsigned int ramp_delay_value; + unsigned int digimic_delay; /* in ms */ unsigned int sysclk; + unsigned int offset_cncl_path; + unsigned int check_defaults:1; + unsigned int reset_registers:1; unsigned int hs_extmute:1; void (*set_hs_extmute)(int mute); }; diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index af36346ff336..64a807f1a8a1 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -360,6 +360,13 @@ static int headset_power_mode(struct snd_soc_codec *codec, int high_perf) return 0; } +static int twl6040_hs_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + msleep(1); + return 0; +} + static int twl6040_power_mode_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -371,6 +378,8 @@ static int twl6040_power_mode_event(struct snd_soc_dapm_widget *w, else priv->non_lp--; + msleep(1); + return 0; } @@ -471,20 +480,6 @@ static const struct snd_kcontrol_new hfdacl_switch_controls = static const struct snd_kcontrol_new hfdacr_switch_controls = SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFRCTL, 2, 1, 0); -/* Headset driver switches */ -static const struct snd_kcontrol_new hsl_driver_switch_controls = - SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSLCTL, 2, 1, 0); - -static const struct snd_kcontrol_new hsr_driver_switch_controls = - SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSRCTL, 2, 1, 0); - -/* Handsfree driver switches */ -static const struct snd_kcontrol_new hfl_driver_switch_controls = - SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFLCTL, 4, 1, 0); - -static const struct snd_kcontrol_new hfr_driver_switch_controls = - SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFRCTL, 4, 1, 0); - static const struct snd_kcontrol_new ep_driver_switch_controls = SOC_DAPM_SINGLE("Switch", TWL6040_REG_EARCTL, 0, 1, 0); @@ -548,10 +543,14 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { TWL6040_REG_DMICBCTL, 4, 0), /* DACs */ - SND_SOC_DAPM_DAC("HSDAC Left", "Headset Playback", - TWL6040_REG_HSLCTL, 0, 0), - SND_SOC_DAPM_DAC("HSDAC Right", "Headset Playback", - TWL6040_REG_HSRCTL, 0, 0), + SND_SOC_DAPM_DAC_E("HSDAC Left", "Headset Playback", + TWL6040_REG_HSLCTL, 0, 0, + twl6040_hs_dac_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_DAC_E("HSDAC Right", "Headset Playback", + TWL6040_REG_HSRCTL, 0, 0, + twl6040_hs_dac_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_DAC_E("HFDAC Left", "Handsfree Playback", TWL6040_REG_HFLCTL, 0, 0, twl6040_power_mode_event, @@ -571,18 +570,19 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { SND_SOC_DAPM_SWITCH("HFDAC Right Playback", SND_SOC_NOPM, 0, 0, &hfdacr_switch_controls), - SND_SOC_DAPM_SWITCH("Headset Left Driver", - SND_SOC_NOPM, 0, 0, &hsl_driver_switch_controls), - SND_SOC_DAPM_SWITCH("Headset Right Driver", - SND_SOC_NOPM, 0, 0, &hsr_driver_switch_controls), - SND_SOC_DAPM_SWITCH_E("Handsfree Left Driver", - SND_SOC_NOPM, 0, 0, &hfl_driver_switch_controls, + /* Analog playback drivers */ + SND_SOC_DAPM_PGA_E("Handsfree Left Driver", + TWL6040_REG_HFLCTL, 4, 0, NULL, 0, twl6040_power_mode_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_SWITCH_E("Handsfree Right Driver", - SND_SOC_NOPM, 0, 0, &hfr_driver_switch_controls, + SND_SOC_DAPM_PGA_E("Handsfree Right Driver", + TWL6040_REG_HFRCTL, 4, 0, NULL, 0, twl6040_power_mode_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PGA("Headset Left Driver", + TWL6040_REG_HSLCTL, 2, 0, NULL, 0), + SND_SOC_DAPM_PGA("Headset Right Driver", + TWL6040_REG_HSRCTL, 2, 0, NULL, 0), SND_SOC_DAPM_SWITCH_E("Earphone Driver", SND_SOC_NOPM, 0, 0, &ep_driver_switch_controls, twl6040_power_mode_event, @@ -616,8 +616,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"HSDAC Left Playback", "Switch", "HSDAC Left"}, {"HSDAC Right Playback", "Switch", "HSDAC Right"}, - {"Headset Left Driver", "Switch", "HSDAC Left Playback"}, - {"Headset Right Driver", "Switch", "HSDAC Right Playback"}, + {"Headset Left Driver", NULL, "HSDAC Left Playback"}, + {"Headset Right Driver", NULL, "HSDAC Right Playback"}, {"HSOL", NULL, "Headset Left Driver"}, {"HSOR", NULL, "Headset Right Driver"}, @@ -928,7 +928,7 @@ static int twl6040_set_dai_sysclk(struct snd_soc_dai *codec_dai, case 19200000: /* mclk input, pll disabled */ hppllctl |= TWL6040_MCLK_19200KHZ | - TWL6040_HPLLSQRBP | + TWL6040_HPLLSQRENA | TWL6040_HPLLBP; break; case 26000000: diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index 28aac53c97bb..f3b4c1d6a82d 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -28,19 +28,6 @@ #include "uda134x.h" -#define POWER_OFF_ON_STANDBY 1 -/* - ALSA SOC usually puts the device in standby mode when it's not used - for sometime. If you define POWER_OFF_ON_STANDBY the driver will - turn off the ADC/DAC when this callback is invoked and turn it back - on when needed. Unfortunately this will result in a very light bump - (it can be audible only with good earphones). If this bothers you - just comment this line, you will have slightly higher power - consumption . Please note that sending the L3 command for ADC is - enough to make the bump, so it doesn't make difference if you - completely take off power from the codec. - */ - #define UDA134X_RATES SNDRV_PCM_RATE_8000_48000 #define UDA134X_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S20_3LE) @@ -58,7 +45,7 @@ static const char uda134x_reg[UDA134X_REGS_NUM] = { /* Extended address registers */ 0x04, 0x04, 0x04, 0x00, 0x00, 0x00, 0x00, 0x00, /* Status, data regs */ - 0x00, 0x83, 0x00, 0x40, 0x80, 0x00, + 0x00, 0x83, 0x00, 0x40, 0x80, 0xC0, 0x00, }; /* @@ -117,6 +104,7 @@ static int uda134x_write(struct snd_soc_codec *codec, unsigned int reg, case UDA134X_DATA000: case UDA134X_DATA001: case UDA134X_DATA010: + case UDA134X_DATA011: addr = UDA134X_DATA0_ADDR; break; case UDA134X_DATA1: @@ -353,8 +341,22 @@ static int uda134x_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: /* ADC, DAC on */ - reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1); - uda134x_write(codec, UDA134X_STATUS1, reg | 0x03); + switch (pd->model) { + case UDA134X_UDA1340: + case UDA134X_UDA1344: + case UDA134X_UDA1345: + reg = uda134x_read_reg_cache(codec, UDA134X_DATA011); + uda134x_write(codec, UDA134X_DATA011, reg | 0x03); + break; + case UDA134X_UDA1341: + reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1); + uda134x_write(codec, UDA134X_STATUS1, reg | 0x03); + break; + default: + printk(KERN_ERR "UDA134X SoC codec: " + "unsupported model %d\n", pd->model); + return -EINVAL; + } break; case SND_SOC_BIAS_PREPARE: /* power on */ @@ -367,8 +369,22 @@ static int uda134x_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: /* ADC, DAC power off */ - reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1); - uda134x_write(codec, UDA134X_STATUS1, reg & ~(0x03)); + switch (pd->model) { + case UDA134X_UDA1340: + case UDA134X_UDA1344: + case UDA134X_UDA1345: + reg = uda134x_read_reg_cache(codec, UDA134X_DATA011); + uda134x_write(codec, UDA134X_DATA011, reg & ~(0x03)); + break; + case UDA134X_UDA1341: + reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1); + uda134x_write(codec, UDA134X_STATUS1, reg & ~(0x03)); + break; + default: + printk(KERN_ERR "UDA134X SoC codec: " + "unsupported model %d\n", pd->model); + return -EINVAL; + } break; case SND_SOC_BIAS_OFF: /* power off */ @@ -531,9 +547,7 @@ static int uda134x_soc_probe(struct platform_device *pdev) codec->num_dai = 1; codec->read = uda134x_read_reg_cache; codec->write = uda134x_write; -#ifdef POWER_OFF_ON_STANDBY - codec->set_bias_level = uda134x_set_bias_level; -#endif + INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); @@ -544,6 +558,14 @@ static int uda134x_soc_probe(struct platform_device *pdev) uda134x_reset(codec); + if (pd->is_powered_on_standby) { + codec->set_bias_level = NULL; + uda134x_set_bias_level(codec, SND_SOC_BIAS_ON); + } else { + codec->set_bias_level = uda134x_set_bias_level; + uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + } + /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) { diff --git a/sound/soc/codecs/uda134x.h b/sound/soc/codecs/uda134x.h index 94f440490b31..205f03b3eaf8 100644 --- a/sound/soc/codecs/uda134x.h +++ b/sound/soc/codecs/uda134x.h @@ -23,9 +23,10 @@ #define UDA134X_DATA000 10 #define UDA134X_DATA001 11 #define UDA134X_DATA010 12 -#define UDA134X_DATA1 13 +#define UDA134X_DATA011 13 +#define UDA134X_DATA1 14 -#define UDA134X_REGS_NUM 14 +#define UDA134X_REGS_NUM 15 #define STATUS0_DAIFMT_MASK (~(7<<1)) #define STATUS0_SYSCLK_MASK (~(3<<4)) diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 002e289d1255..4bcd168794e1 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -795,6 +795,8 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c, dev_set_drvdata(&i2c->dev, wm2000); wm2000->anc_eng_ena = 1; + wm2000->anc_active = 1; + wm2000->spk_ena = 1; wm2000->i2c = i2c; wm2000_reset(wm2000); diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 37242a7d3077..0ad039b4adf5 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -482,7 +482,8 @@ static int wm8523_register(struct wm8523_priv *wm8523, if (wm8523_codec) { dev_err(codec->dev, "Another WM8523 is registered\n"); - return -EINVAL; + ret = -EINVAL; + goto err; } mutex_init(&codec->mutex); @@ -570,18 +571,19 @@ static int wm8523_register(struct wm8523_priv *wm8523, ret = snd_soc_register_codec(codec); if (ret != 0) { dev_err(codec->dev, "Failed to register codec: %d\n", ret); - return ret; + goto err_enable; } ret = snd_soc_register_dai(&wm8523_dai); if (ret != 0) { dev_err(codec->dev, "Failed to register DAI: %d\n", ret); - snd_soc_unregister_codec(codec); - return ret; + goto err_codec; } return 0; +err_codec: + snd_soc_unregister_codec(codec); err_enable: regulator_bulk_disable(ARRAY_SIZE(wm8523->supplies), wm8523->supplies); err_get: diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index c3571ee5c11b..72deeabef4fe 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -269,9 +269,9 @@ SOC_DOUBLE("DAC2 Invert Switch", WM8580_DAC_CONTROL4, 2, 3, 1, 0), SOC_DOUBLE("DAC3 Invert Switch", WM8580_DAC_CONTROL4, 4, 5, 1, 0), SOC_SINGLE("DAC ZC Switch", WM8580_DAC_CONTROL5, 5, 1, 0), -SOC_SINGLE("DAC1 Switch", WM8580_DAC_CONTROL5, 0, 1, 0), -SOC_SINGLE("DAC2 Switch", WM8580_DAC_CONTROL5, 1, 1, 0), -SOC_SINGLE("DAC3 Switch", WM8580_DAC_CONTROL5, 2, 1, 0), +SOC_SINGLE("DAC1 Switch", WM8580_DAC_CONTROL5, 0, 1, 1), +SOC_SINGLE("DAC2 Switch", WM8580_DAC_CONTROL5, 1, 1, 1), +SOC_SINGLE("DAC3 Switch", WM8580_DAC_CONTROL5, 2, 1, 1), SOC_DOUBLE("ADC Mute Switch", WM8580_ADC_CONTROL1, 0, 1, 1, 0), SOC_SINGLE("ADC High-Pass Filter Switch", WM8580_ADC_CONTROL1, 4, 1, 0), diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index effb14eee7d4..e2dba07f0260 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -439,7 +439,8 @@ static int wm8711_register(struct wm8711_priv *wm8711, if (wm8711_codec) { dev_err(codec->dev, "Another WM8711 is registered\n"); - return -EINVAL; + ret = -EINVAL; + goto err; } mutex_init(&codec->mutex); diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c new file mode 100644 index 000000000000..b9ea8904ad4b --- /dev/null +++ b/sound/soc/codecs/wm8741.c @@ -0,0 +1,579 @@ +/* + * wm8741.c -- WM8741 ALSA SoC Audio driver + * + * Copyright 2010 Wolfson Microelectronics plc + * + * Author: Ian Lartey <ian@opensource.wolfsonmicro.com> + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/regulator/consumer.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "wm8741.h" + +static struct snd_soc_codec *wm8741_codec; +struct snd_soc_codec_device soc_codec_dev_wm8741; + +#define WM8741_NUM_SUPPLIES 2 +static const char *wm8741_supply_names[WM8741_NUM_SUPPLIES] = { + "AVDD", + "DVDD", +}; + +#define WM8741_NUM_RATES 4 + +/* codec private data */ +struct wm8741_priv { + struct snd_soc_codec codec; + u16 reg_cache[WM8741_REGISTER_COUNT]; + struct regulator_bulk_data supplies[WM8741_NUM_SUPPLIES]; + unsigned int sysclk; + unsigned int rate_constraint_list[WM8741_NUM_RATES]; + struct snd_pcm_hw_constraint_list rate_constraint; +}; + +static const u16 wm8741_reg_defaults[WM8741_REGISTER_COUNT] = { + 0x0000, /* R0 - DACLLSB Attenuation */ + 0x0000, /* R1 - DACLMSB Attenuation */ + 0x0000, /* R2 - DACRLSB Attenuation */ + 0x0000, /* R3 - DACRMSB Attenuation */ + 0x0000, /* R4 - Volume Control */ + 0x000A, /* R5 - Format Control */ + 0x0000, /* R6 - Filter Control */ + 0x0000, /* R7 - Mode Control 1 */ + 0x0002, /* R8 - Mode Control 2 */ + 0x0000, /* R9 - Reset */ + 0x0002, /* R32 - ADDITONAL_CONTROL_1 */ +}; + + +static int wm8741_reset(struct snd_soc_codec *codec) +{ + return snd_soc_write(codec, WM8741_RESET, 0); +} + +static const DECLARE_TLV_DB_SCALE(dac_tlv_fine, -12700, 13, 0); +static const DECLARE_TLV_DB_SCALE(dac_tlv, -12700, 400, 0); + +static const struct snd_kcontrol_new wm8741_snd_controls[] = { +SOC_DOUBLE_R_TLV("Fine Playback Volume", WM8741_DACLLSB_ATTENUATION, + WM8741_DACRLSB_ATTENUATION, 1, 255, 1, dac_tlv_fine), +SOC_DOUBLE_R_TLV("Playback Volume", WM8741_DACLMSB_ATTENUATION, + WM8741_DACRMSB_ATTENUATION, 0, 511, 1, dac_tlv), +}; + +static const struct snd_soc_dapm_widget wm8741_dapm_widgets[] = { +SND_SOC_DAPM_DAC("DACL", "Playback", SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_DAC("DACR", "Playback", SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_OUTPUT("VOUTLP"), +SND_SOC_DAPM_OUTPUT("VOUTLN"), +SND_SOC_DAPM_OUTPUT("VOUTRP"), +SND_SOC_DAPM_OUTPUT("VOUTRN"), +}; + +static const struct snd_soc_dapm_route intercon[] = { + { "VOUTLP", NULL, "DACL" }, + { "VOUTLN", NULL, "DACL" }, + { "VOUTRP", NULL, "DACR" }, + { "VOUTRN", NULL, "DACR" }, +}; + +static int wm8741_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, wm8741_dapm_widgets, + ARRAY_SIZE(wm8741_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + return 0; +} + +static struct { + int value; + int ratio; +} lrclk_ratios[WM8741_NUM_RATES] = { + { 1, 256 }, + { 2, 384 }, + { 3, 512 }, + { 4, 768 }, +}; + + +static int wm8741_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); + + /* The set of sample rates that can be supported depends on the + * MCLK supplied to the CODEC - enforce this. + */ + if (!wm8741->sysclk) { + dev_err(codec->dev, + "No MCLK configured, call set_sysclk() on init\n"); + return -EINVAL; + } + + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &wm8741->rate_constraint); + + return 0; +} + +static int wm8741_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); + u16 iface = snd_soc_read(codec, WM8741_FORMAT_CONTROL) & 0x1FC; + int i; + + /* Find a supported LRCLK ratio */ + for (i = 0; i < ARRAY_SIZE(lrclk_ratios); i++) { + if (wm8741->sysclk / params_rate(params) == + lrclk_ratios[i].ratio) + break; + } + + /* Should never happen, should be handled by constraints */ + if (i == ARRAY_SIZE(lrclk_ratios)) { + dev_err(codec->dev, "MCLK/fs ratio %d unsupported\n", + wm8741->sysclk / params_rate(params)); + return -EINVAL; + } + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface |= 0x0001; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface |= 0x0002; + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface |= 0x0003; + break; + default: + dev_dbg(codec->dev, "wm8741_hw_params: Unsupported bit size param = %d", + params_format(params)); + return -EINVAL; + } + + dev_dbg(codec->dev, "wm8741_hw_params: bit size param = %d", + params_format(params)); + + snd_soc_write(codec, WM8741_FORMAT_CONTROL, iface); + return 0; +} + +static int wm8741_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); + unsigned int val; + int i; + + dev_dbg(codec->dev, "wm8741_set_dai_sysclk info: freq=%dHz\n", freq); + + wm8741->sysclk = freq; + + wm8741->rate_constraint.count = 0; + + for (i = 0; i < ARRAY_SIZE(lrclk_ratios); i++) { + dev_dbg(codec->dev, "index = %d, ratio = %d, freq = %d", + i, lrclk_ratios[i].ratio, freq); + + val = freq / lrclk_ratios[i].ratio; + /* Check that it's a standard rate since core can't + * cope with others and having the odd rates confuses + * constraint matching. + */ + switch (val) { + case 32000: + case 44100: + case 48000: + case 64000: + case 88200: + case 96000: + dev_dbg(codec->dev, "Supported sample rate: %dHz\n", + val); + wm8741->rate_constraint_list[i] = val; + wm8741->rate_constraint.count++; + break; + default: + dev_dbg(codec->dev, "Skipping sample rate: %dHz\n", + val); + } + } + + /* Need at least one supported rate... */ + if (wm8741->rate_constraint.count == 0) + return -EINVAL; + + return 0; +} + +static int wm8741_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 iface = snd_soc_read(codec, WM8741_FORMAT_CONTROL) & 0x1C3; + + /* check master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= 0x0008; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= 0x0004; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= 0x0003; + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= 0x0013; + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= 0x0010; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= 0x0020; + break; + case SND_SOC_DAIFMT_NB_IF: + iface |= 0x0030; + break; + default: + return -EINVAL; + } + + + dev_dbg(codec->dev, "wm8741_set_dai_fmt: Format=%x, Clock Inv=%x\n", + fmt & SND_SOC_DAIFMT_FORMAT_MASK, + ((fmt & SND_SOC_DAIFMT_INV_MASK))); + + snd_soc_write(codec, WM8741_FORMAT_CONTROL, iface); + return 0; +} + +#define WM8741_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | \ + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | \ + SNDRV_PCM_RATE_192000) + +#define WM8741_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops wm8741_dai_ops = { + .startup = wm8741_startup, + .hw_params = wm8741_hw_params, + .set_sysclk = wm8741_set_dai_sysclk, + .set_fmt = wm8741_set_dai_fmt, +}; + +struct snd_soc_dai wm8741_dai = { + .name = "WM8741", + .playback = { + .stream_name = "Playback", + .channels_min = 2, /* Mono modes not yet supported */ + .channels_max = 2, + .rates = WM8741_RATES, + .formats = WM8741_FORMATS, + }, + .ops = &wm8741_dai_ops, +}; +EXPORT_SYMBOL_GPL(wm8741_dai); + +#ifdef CONFIG_PM +static int wm8741_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + u16 *cache = codec->reg_cache; + int i; + + /* RESTORE REG Cache */ + for (i = 0; i < WM8741_REGISTER_COUNT; i++) { + if (cache[i] == wm8741_reg_defaults[i] || WM8741_RESET == i) + continue; + snd_soc_write(codec, i, cache[i]); + } + return 0; +} +#else +#define wm8741_suspend NULL +#define wm8741_resume NULL +#endif + +static int wm8741_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (wm8741_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = wm8741_codec; + codec = wm8741_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms: %d\n", ret); + goto pcm_err; + } + + snd_soc_add_controls(codec, wm8741_snd_controls, + ARRAY_SIZE(wm8741_snd_controls)); + wm8741_add_widgets(codec); + + return ret; + +pcm_err: + return ret; +} + +static int wm8741_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8741 = { + .probe = wm8741_probe, + .remove = wm8741_remove, + .resume = wm8741_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8741); + +static int wm8741_register(struct wm8741_priv *wm8741, + enum snd_soc_control_type control) +{ + int ret; + struct snd_soc_codec *codec = &wm8741->codec; + int i; + + if (wm8741_codec) { + dev_err(codec->dev, "Another WM8741 is registered\n"); + return -EINVAL; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + snd_soc_codec_set_drvdata(codec, wm8741); + codec->name = "WM8741"; + codec->owner = THIS_MODULE; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = NULL; + codec->dai = &wm8741_dai; + codec->num_dai = 1; + codec->reg_cache_size = WM8741_REGISTER_COUNT; + codec->reg_cache = &wm8741->reg_cache; + + wm8741->rate_constraint.list = &wm8741->rate_constraint_list[0]; + wm8741->rate_constraint.count = + ARRAY_SIZE(wm8741->rate_constraint_list); + + memcpy(codec->reg_cache, wm8741_reg_defaults, + sizeof(wm8741->reg_cache)); + + ret = snd_soc_codec_set_cache_io(codec, 7, 9, control); + if (ret != 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + goto err; + } + + for (i = 0; i < ARRAY_SIZE(wm8741->supplies); i++) + wm8741->supplies[i].supply = wm8741_supply_names[i]; + + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8741->supplies), + wm8741->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to request supplies: %d\n", ret); + goto err; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(wm8741->supplies), + wm8741->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + goto err_get; + } + + ret = wm8741_reset(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to issue reset\n"); + goto err_enable; + } + + wm8741_dai.dev = codec->dev; + + /* Change some default settings - latch VU */ + wm8741->reg_cache[WM8741_DACLLSB_ATTENUATION] |= WM8741_UPDATELL; + wm8741->reg_cache[WM8741_DACLMSB_ATTENUATION] |= WM8741_UPDATELM; + wm8741->reg_cache[WM8741_DACRLSB_ATTENUATION] |= WM8741_UPDATERL; + wm8741->reg_cache[WM8741_DACRLSB_ATTENUATION] |= WM8741_UPDATERM; + + wm8741_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + return ret; + } + + ret = snd_soc_register_dai(&wm8741_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + snd_soc_unregister_codec(codec); + return ret; + } + + dev_dbg(codec->dev, "Successful registration\n"); + return 0; + +err_enable: + regulator_bulk_disable(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); + +err_get: + regulator_bulk_free(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); + +err: + kfree(wm8741); + return ret; +} + +static void wm8741_unregister(struct wm8741_priv *wm8741) +{ + regulator_bulk_free(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); + + snd_soc_unregister_dai(&wm8741_dai); + snd_soc_unregister_codec(&wm8741->codec); + kfree(wm8741); + wm8741_codec = NULL; +} + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static __devinit int wm8741_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct wm8741_priv *wm8741; + struct snd_soc_codec *codec; + + wm8741 = kzalloc(sizeof(struct wm8741_priv), GFP_KERNEL); + if (wm8741 == NULL) + return -ENOMEM; + + codec = &wm8741->codec; + codec->hw_write = (hw_write_t)i2c_master_send; + + i2c_set_clientdata(i2c, wm8741); + codec->control_data = i2c; + + codec->dev = &i2c->dev; + + return wm8741_register(wm8741, SND_SOC_I2C); +} + +static __devexit int wm8741_i2c_remove(struct i2c_client *client) +{ + struct wm8741_priv *wm8741 = i2c_get_clientdata(client); + wm8741_unregister(wm8741); + return 0; +} + +static const struct i2c_device_id wm8741_i2c_id[] = { + { "wm8741", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8741_i2c_id); + + +static struct i2c_driver wm8741_i2c_driver = { + .driver = { + .name = "WM8741", + .owner = THIS_MODULE, + }, + .probe = wm8741_i2c_probe, + .remove = __devexit_p(wm8741_i2c_remove), + .id_table = wm8741_i2c_id, +}; +#endif + +static int __init wm8741_modinit(void) +{ + int ret; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&wm8741_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register WM8741 I2C driver: %d\n", + ret); + } +#endif + return 0; +} +module_init(wm8741_modinit); + +static void __exit wm8741_exit(void) +{ +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&wm8741_i2c_driver); +#endif +} +module_exit(wm8741_exit); + +MODULE_DESCRIPTION("ASoC WM8741 driver"); +MODULE_AUTHOR("Ian Lartey <ian@opensource.wolfsonmicro.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8741.h b/sound/soc/codecs/wm8741.h new file mode 100644 index 000000000000..fdef6ecd1f6f --- /dev/null +++ b/sound/soc/codecs/wm8741.h @@ -0,0 +1,214 @@ +/* + * wm8741.h -- WM8423 ASoC driver + * + * Copyright 2010 Wolfson Microelectronics, plc + * + * Author: Ian Lartey <ian@opensource.wolfsonmicro.com> + * + * Based on wm8753.h + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _WM8741_H +#define _WM8741_H + +/* + * Register values. + */ +#define WM8741_DACLLSB_ATTENUATION 0x00 +#define WM8741_DACLMSB_ATTENUATION 0x01 +#define WM8741_DACRLSB_ATTENUATION 0x02 +#define WM8741_DACRMSB_ATTENUATION 0x03 +#define WM8741_VOLUME_CONTROL 0x04 +#define WM8741_FORMAT_CONTROL 0x05 +#define WM8741_FILTER_CONTROL 0x06 +#define WM8741_MODE_CONTROL_1 0x07 +#define WM8741_MODE_CONTROL_2 0x08 +#define WM8741_RESET 0x09 +#define WM8741_ADDITIONAL_CONTROL_1 0x20 + +#define WM8741_REGISTER_COUNT 11 +#define WM8741_MAX_REGISTER 0x20 + +/* + * Field Definitions. + */ + +/* + * R0 (0x00) - DACLLSB_ATTENUATION + */ +#define WM8741_UPDATELL 0x0020 /* UPDATELL */ +#define WM8741_UPDATELL_MASK 0x0020 /* UPDATELL */ +#define WM8741_UPDATELL_SHIFT 5 /* UPDATELL */ +#define WM8741_UPDATELL_WIDTH 1 /* UPDATELL */ +#define WM8741_LAT_4_0_MASK 0x001F /* LAT[4:0] - [4:0] */ +#define WM8741_LAT_4_0_SHIFT 0 /* LAT[4:0] - [4:0] */ +#define WM8741_LAT_4_0_WIDTH 5 /* LAT[4:0] - [4:0] */ + +/* + * R1 (0x01) - DACLMSB_ATTENUATION + */ +#define WM8741_UPDATELM 0x0020 /* UPDATELM */ +#define WM8741_UPDATELM_MASK 0x0020 /* UPDATELM */ +#define WM8741_UPDATELM_SHIFT 5 /* UPDATELM */ +#define WM8741_UPDATELM_WIDTH 1 /* UPDATELM */ +#define WM8741_LAT_9_5_0_MASK 0x001F /* LAT[9:5] - [4:0] */ +#define WM8741_LAT_9_5_0_SHIFT 0 /* LAT[9:5] - [4:0] */ +#define WM8741_LAT_9_5_0_WIDTH 5 /* LAT[9:5] - [4:0] */ + +/* + * R2 (0x02) - DACRLSB_ATTENUATION + */ +#define WM8741_UPDATERL 0x0020 /* UPDATERL */ +#define WM8741_UPDATERL_MASK 0x0020 /* UPDATERL */ +#define WM8741_UPDATERL_SHIFT 5 /* UPDATERL */ +#define WM8741_UPDATERL_WIDTH 1 /* UPDATERL */ +#define WM8741_RAT_4_0_MASK 0x001F /* RAT[4:0] - [4:0] */ +#define WM8741_RAT_4_0_SHIFT 0 /* RAT[4:0] - [4:0] */ +#define WM8741_RAT_4_0_WIDTH 5 /* RAT[4:0] - [4:0] */ + +/* + * R3 (0x03) - DACRMSB_ATTENUATION + */ +#define WM8741_UPDATERM 0x0020 /* UPDATERM */ +#define WM8741_UPDATERM_MASK 0x0020 /* UPDATERM */ +#define WM8741_UPDATERM_SHIFT 5 /* UPDATERM */ +#define WM8741_UPDATERM_WIDTH 1 /* UPDATERM */ +#define WM8741_RAT_9_5_0_MASK 0x001F /* RAT[9:5] - [4:0] */ +#define WM8741_RAT_9_5_0_SHIFT 0 /* RAT[9:5] - [4:0] */ +#define WM8741_RAT_9_5_0_WIDTH 5 /* RAT[9:5] - [4:0] */ + +/* + * R4 (0x04) - VOLUME_CONTROL + */ +#define WM8741_AMUTE 0x0080 /* AMUTE */ +#define WM8741_AMUTE_MASK 0x0080 /* AMUTE */ +#define WM8741_AMUTE_SHIFT 7 /* AMUTE */ +#define WM8741_AMUTE_WIDTH 1 /* AMUTE */ +#define WM8741_ZFLAG_MASK 0x0060 /* ZFLAG - [6:5] */ +#define WM8741_ZFLAG_SHIFT 5 /* ZFLAG - [6:5] */ +#define WM8741_ZFLAG_WIDTH 2 /* ZFLAG - [6:5] */ +#define WM8741_IZD 0x0010 /* IZD */ +#define WM8741_IZD_MASK 0x0010 /* IZD */ +#define WM8741_IZD_SHIFT 4 /* IZD */ +#define WM8741_IZD_WIDTH 1 /* IZD */ +#define WM8741_SOFT 0x0008 /* SOFT MUTE */ +#define WM8741_SOFT_MASK 0x0008 /* SOFT MUTE */ +#define WM8741_SOFT_SHIFT 3 /* SOFT MUTE */ +#define WM8741_SOFT_WIDTH 1 /* SOFT MUTE */ +#define WM8741_ATC 0x0004 /* ATC */ +#define WM8741_ATC_MASK 0x0004 /* ATC */ +#define WM8741_ATC_SHIFT 2 /* ATC */ +#define WM8741_ATC_WIDTH 1 /* ATC */ +#define WM8741_ATT2DB 0x0002 /* ATT2DB */ +#define WM8741_ATT2DB_MASK 0x0002 /* ATT2DB */ +#define WM8741_ATT2DB_SHIFT 1 /* ATT2DB */ +#define WM8741_ATT2DB_WIDTH 1 /* ATT2DB */ +#define WM8741_VOL_RAMP 0x0001 /* VOL_RAMP */ +#define WM8741_VOL_RAMP_MASK 0x0001 /* VOL_RAMP */ +#define WM8741_VOL_RAMP_SHIFT 0 /* VOL_RAMP */ +#define WM8741_VOL_RAMP_WIDTH 1 /* VOL_RAMP */ + +/* + * R5 (0x05) - FORMAT_CONTROL + */ +#define WM8741_PWDN 0x0080 /* PWDN */ +#define WM8741_PWDN_MASK 0x0080 /* PWDN */ +#define WM8741_PWDN_SHIFT 7 /* PWDN */ +#define WM8741_PWDN_WIDTH 1 /* PWDN */ +#define WM8741_REV 0x0040 /* REV */ +#define WM8741_REV_MASK 0x0040 /* REV */ +#define WM8741_REV_SHIFT 6 /* REV */ +#define WM8741_REV_WIDTH 1 /* REV */ +#define WM8741_BCP 0x0020 /* BCP */ +#define WM8741_BCP_MASK 0x0020 /* BCP */ +#define WM8741_BCP_SHIFT 5 /* BCP */ +#define WM8741_BCP_WIDTH 1 /* BCP */ +#define WM8741_LRP 0x0010 /* LRP */ +#define WM8741_LRP_MASK 0x0010 /* LRP */ +#define WM8741_LRP_SHIFT 4 /* LRP */ +#define WM8741_LRP_WIDTH 1 /* LRP */ +#define WM8741_FMT_MASK 0x000C /* FMT - [3:2] */ +#define WM8741_FMT_SHIFT 2 /* FMT - [3:2] */ +#define WM8741_FMT_WIDTH 2 /* FMT - [3:2] */ +#define WM8741_IWL_MASK 0x0003 /* IWL - [1:0] */ +#define WM8741_IWL_SHIFT 0 /* IWL - [1:0] */ +#define WM8741_IWL_WIDTH 2 /* IWL - [1:0] */ + +/* + * R6 (0x06) - FILTER_CONTROL + */ +#define WM8741_ZFLAG_HI 0x0080 /* ZFLAG_HI */ +#define WM8741_ZFLAG_HI_MASK 0x0080 /* ZFLAG_HI */ +#define WM8741_ZFLAG_HI_SHIFT 7 /* ZFLAG_HI */ +#define WM8741_ZFLAG_HI_WIDTH 1 /* ZFLAG_HI */ +#define WM8741_DEEMPH_MASK 0x0060 /* DEEMPH - [6:5] */ +#define WM8741_DEEMPH_SHIFT 5 /* DEEMPH - [6:5] */ +#define WM8741_DEEMPH_WIDTH 2 /* DEEMPH - [6:5] */ +#define WM8741_DSDFILT_MASK 0x0018 /* DSDFILT - [4:3] */ +#define WM8741_DSDFILT_SHIFT 3 /* DSDFILT - [4:3] */ +#define WM8741_DSDFILT_WIDTH 2 /* DSDFILT - [4:3] */ +#define WM8741_FIRSEL_MASK 0x0007 /* FIRSEL - [2:0] */ +#define WM8741_FIRSEL_SHIFT 0 /* FIRSEL - [2:0] */ +#define WM8741_FIRSEL_WIDTH 3 /* FIRSEL - [2:0] */ + +/* + * R7 (0x07) - MODE_CONTROL_1 + */ +#define WM8741_MODE8X 0x0080 /* MODE8X */ +#define WM8741_MODE8X_MASK 0x0080 /* MODE8X */ +#define WM8741_MODE8X_SHIFT 7 /* MODE8X */ +#define WM8741_MODE8X_WIDTH 1 /* MODE8X */ +#define WM8741_OSR_MASK 0x0060 /* OSR - [6:5] */ +#define WM8741_OSR_SHIFT 5 /* OSR - [6:5] */ +#define WM8741_OSR_WIDTH 2 /* OSR - [6:5] */ +#define WM8741_SR_MASK 0x001C /* SR - [4:2] */ +#define WM8741_SR_SHIFT 2 /* SR - [4:2] */ +#define WM8741_SR_WIDTH 3 /* SR - [4:2] */ +#define WM8741_MODESEL_MASK 0x0003 /* MODESEL - [1:0] */ +#define WM8741_MODESEL_SHIFT 0 /* MODESEL - [1:0] */ +#define WM8741_MODESEL_WIDTH 2 /* MODESEL - [1:0] */ + +/* + * R8 (0x08) - MODE_CONTROL_2 + */ +#define WM8741_DSD_GAIN 0x0040 /* DSD_GAIN */ +#define WM8741_DSD_GAIN_MASK 0x0040 /* DSD_GAIN */ +#define WM8741_DSD_GAIN_SHIFT 6 /* DSD_GAIN */ +#define WM8741_DSD_GAIN_WIDTH 1 /* DSD_GAIN */ +#define WM8741_SDOUT 0x0020 /* SDOUT */ +#define WM8741_SDOUT_MASK 0x0020 /* SDOUT */ +#define WM8741_SDOUT_SHIFT 5 /* SDOUT */ +#define WM8741_SDOUT_WIDTH 1 /* SDOUT */ +#define WM8741_DOUT 0x0010 /* DOUT */ +#define WM8741_DOUT_MASK 0x0010 /* DOUT */ +#define WM8741_DOUT_SHIFT 4 /* DOUT */ +#define WM8741_DOUT_WIDTH 1 /* DOUT */ +#define WM8741_DIFF_MASK 0x000C /* DIFF - [3:2] */ +#define WM8741_DIFF_SHIFT 2 /* DIFF - [3:2] */ +#define WM8741_DIFF_WIDTH 2 /* DIFF - [3:2] */ +#define WM8741_DITHER_MASK 0x0003 /* DITHER - [1:0] */ +#define WM8741_DITHER_SHIFT 0 /* DITHER - [1:0] */ +#define WM8741_DITHER_WIDTH 2 /* DITHER - [1:0] */ + +/* + * R32 (0x20) - ADDITONAL_CONTROL_1 + */ +#define WM8741_DSD_LEVEL 0x0002 /* DSD_LEVEL */ +#define WM8741_DSD_LEVEL_MASK 0x0002 /* DSD_LEVEL */ +#define WM8741_DSD_LEVEL_SHIFT 1 /* DSD_LEVEL */ +#define WM8741_DSD_LEVEL_WIDTH 1 /* DSD_LEVEL */ +#define WM8741_DSD_NO_NOTCH 0x0001 /* DSD_NO_NOTCH */ +#define WM8741_DSD_NO_NOTCH_MASK 0x0001 /* DSD_NO_NOTCH */ +#define WM8741_DSD_NO_NOTCH_SHIFT 0 /* DSD_NO_NOTCH */ +#define WM8741_DSD_NO_NOTCH_WIDTH 1 /* DSD_NO_NOTCH */ + +#define WM8741_SYSCLK 0 + +extern struct snd_soc_dai wm8741_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm8741; + +#endif diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 9407e193fcc3..e2c05e3e323a 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -884,6 +884,7 @@ static int wm8750_i2c_remove(struct i2c_client *client) static const struct i2c_device_id wm8750_i2c_id[] = { { "wm8750", 0 }, + { "wm8987", 0 }, /* WM8987 is register compatible with WM8750 */ { } }; MODULE_DEVICE_TABLE(i2c, wm8750_i2c_id); @@ -925,14 +926,22 @@ static int __devexit wm8750_spi_remove(struct spi_device *spi) return 0; } +static const struct spi_device_id wm8750_spi_id[] = { + { "wm8750", 0 }, + { "wm8987", 0 }, + { } +}; +MODULE_DEVICE_TABLE(spi, wm8750_spi_id); + static struct spi_driver wm8750_spi_driver = { .driver = { - .name = "wm8750", + .name = "WM8750 SPI Codec", .bus = &spi_bus_type, .owner = THIS_MODULE, }, .probe = wm8750_spi_probe, .remove = __devexit_p(wm8750_spi_remove), + .id_table = wm8750_spi_id, }; #endif diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index 4e212ed62ea6..f8154e661524 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -178,13 +178,6 @@ static int wm8776_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) case SND_SOC_DAIFMT_LEFT_J: iface |= 0x0001; break; - /* FIXME: CHECK A/B */ - case SND_SOC_DAIFMT_DSP_A: - iface |= 0x0003; - break; - case SND_SOC_DAIFMT_DSP_B: - iface |= 0x0007; - break; default: return -EINVAL; } diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 87f14f8675fa..f7dcabf6283c 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2433,7 +2433,8 @@ static int wm8904_register(struct wm8904_priv *wm8904, if (wm8904_codec) { dev_err(codec->dev, "Another WM8904 is registered\n"); - return -EINVAL; + ret = -EINVAL; + goto err; } mutex_init(&codec->mutex); @@ -2462,7 +2463,8 @@ static int wm8904_register(struct wm8904_priv *wm8904, default: dev_err(codec->dev, "Unknown device type %d\n", wm8904->devtype); - return -EINVAL; + ret = -EINVAL; + goto err; } memcpy(codec->reg_cache, wm8904_reg, sizeof(wm8904_reg)); @@ -2566,18 +2568,19 @@ static int wm8904_register(struct wm8904_priv *wm8904, ret = snd_soc_register_codec(codec); if (ret != 0) { dev_err(codec->dev, "Failed to register codec: %d\n", ret); - return ret; + goto err_enable; } ret = snd_soc_register_dai(&wm8904_dai); if (ret != 0) { dev_err(codec->dev, "Failed to register DAI: %d\n", ret); - snd_soc_unregister_codec(codec); - return ret; + goto err_codec; } return 0; +err_codec: + snd_soc_unregister_codec(codec); err_enable: regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); err_get: diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index e3c4bbfaae27..f0c11138e610 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -845,6 +845,7 @@ static void wm8940_unregister(struct wm8940_priv *wm8940) static int wm8940_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { + int ret; struct wm8940_priv *wm8940; struct snd_soc_codec *codec; @@ -858,7 +859,11 @@ static int wm8940_i2c_probe(struct i2c_client *i2c, codec->control_data = i2c; codec->dev = &i2c->dev; - return wm8940_register(wm8940, SND_SOC_I2C); + ret = wm8940_register(wm8940, SND_SOC_I2C); + if (ret < 0) + kfree(wm8940); + + return ret; } static int __devexit wm8940_i2c_remove(struct i2c_client *client) diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index fedb76452f1b..5f025593d84d 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -964,7 +964,8 @@ static int wm8955_register(struct wm8955_priv *wm8955, if (wm8955_codec) { dev_err(codec->dev, "Another WM8955 is registered\n"); - return -EINVAL; + ret = -EINVAL; + goto err; } mutex_init(&codec->mutex); @@ -1047,18 +1048,19 @@ static int wm8955_register(struct wm8955_priv *wm8955, ret = snd_soc_register_codec(codec); if (ret != 0) { dev_err(codec->dev, "Failed to register codec: %d\n", ret); - return ret; + goto err_enable; } ret = snd_soc_register_dai(&wm8955_dai); if (ret != 0) { dev_err(codec->dev, "Failed to register DAI: %d\n", ret); - snd_soc_unregister_codec(codec); - return ret; + goto err_codec; } return 0; +err_codec: + snd_soc_unregister_codec(codec); err_enable: regulator_bulk_disable(ARRAY_SIZE(wm8955->supplies), wm8955->supplies); err_get: diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 7233cc68435a..3c6ee61f6c95 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -79,12 +79,13 @@ struct wm8960_priv { struct snd_soc_dapm_widget *lout1; struct snd_soc_dapm_widget *rout1; struct snd_soc_dapm_widget *out3; + bool deemph; + int playback_fs; }; #define wm8960_reset(c) snd_soc_write(c, WM8960_RESET, 0) /* enumerated controls */ -static const char *wm8960_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"}; static const char *wm8960_polarity[] = {"No Inversion", "Left Inverted", "Right Inverted", "Stereo Inversion"}; static const char *wm8960_3d_upper_cutoff[] = {"High", "Low"}; @@ -93,7 +94,6 @@ static const char *wm8960_alcfunc[] = {"Off", "Right", "Left", "Stereo"}; static const char *wm8960_alcmode[] = {"ALC", "Limiter"}; static const struct soc_enum wm8960_enum[] = { - SOC_ENUM_SINGLE(WM8960_DACCTL1, 1, 4, wm8960_deemph), SOC_ENUM_SINGLE(WM8960_DACCTL1, 5, 4, wm8960_polarity), SOC_ENUM_SINGLE(WM8960_DACCTL2, 5, 4, wm8960_polarity), SOC_ENUM_SINGLE(WM8960_3D, 6, 2, wm8960_3d_upper_cutoff), @@ -102,6 +102,59 @@ static const struct soc_enum wm8960_enum[] = { SOC_ENUM_SINGLE(WM8960_ALC3, 8, 2, wm8960_alcmode), }; +static const int deemph_settings[] = { 0, 32000, 44100, 48000 }; + +static int wm8960_set_deemph(struct snd_soc_codec *codec) +{ + struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + int val, i, best; + + /* If we're using deemphasis select the nearest available sample + * rate. + */ + if (wm8960->deemph) { + best = 1; + for (i = 2; i < ARRAY_SIZE(deemph_settings); i++) { + if (abs(deemph_settings[i] - wm8960->playback_fs) < + abs(deemph_settings[best] - wm8960->playback_fs)) + best = i; + } + + val = best << 1; + } else { + val = 0; + } + + dev_dbg(codec->dev, "Set deemphasis %d\n", val); + + return snd_soc_update_bits(codec, WM8960_DACCTL1, + 0x6, val); +} + +static int wm8960_get_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + + return wm8960->deemph; +} + +static int wm8960_put_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + int deemph = ucontrol->value.enumerated.item[0]; + + if (deemph > 1) + return -EINVAL; + + wm8960->deemph = deemph; + + return wm8960_set_deemph(codec); +} + static const DECLARE_TLV_DB_SCALE(adc_tlv, -9700, 50, 0); static const DECLARE_TLV_DB_SCALE(dac_tlv, -12700, 50, 1); static const DECLARE_TLV_DB_SCALE(bypass_tlv, -2100, 300, 0); @@ -131,23 +184,24 @@ SOC_SINGLE("Speaker DC Volume", WM8960_CLASSD3, 3, 5, 0), SOC_SINGLE("Speaker AC Volume", WM8960_CLASSD3, 0, 5, 0), SOC_SINGLE("PCM Playback -6dB Switch", WM8960_DACCTL1, 7, 1, 0), -SOC_ENUM("ADC Polarity", wm8960_enum[1]), -SOC_ENUM("Playback De-emphasis", wm8960_enum[0]), +SOC_ENUM("ADC Polarity", wm8960_enum[0]), SOC_SINGLE("ADC High Pass Filter Switch", WM8960_DACCTL1, 0, 1, 0), SOC_ENUM("DAC Polarity", wm8960_enum[2]), +SOC_SINGLE_BOOL_EXT("DAC Deemphasis Switch", 0, + wm8960_get_deemph, wm8960_put_deemph), -SOC_ENUM("3D Filter Upper Cut-Off", wm8960_enum[3]), -SOC_ENUM("3D Filter Lower Cut-Off", wm8960_enum[4]), +SOC_ENUM("3D Filter Upper Cut-Off", wm8960_enum[2]), +SOC_ENUM("3D Filter Lower Cut-Off", wm8960_enum[3]), SOC_SINGLE("3D Volume", WM8960_3D, 1, 15, 0), SOC_SINGLE("3D Switch", WM8960_3D, 0, 1, 0), -SOC_ENUM("ALC Function", wm8960_enum[5]), +SOC_ENUM("ALC Function", wm8960_enum[4]), SOC_SINGLE("ALC Max Gain", WM8960_ALC1, 4, 7, 0), SOC_SINGLE("ALC Target", WM8960_ALC1, 0, 15, 1), SOC_SINGLE("ALC Min Gain", WM8960_ALC2, 4, 7, 0), SOC_SINGLE("ALC Hold Time", WM8960_ALC2, 0, 15, 0), -SOC_ENUM("ALC Mode", wm8960_enum[6]), +SOC_ENUM("ALC Mode", wm8960_enum[5]), SOC_SINGLE("ALC Decay", WM8960_ALC3, 4, 15, 0), SOC_SINGLE("ALC Attack", WM8960_ALC3, 0, 15, 0), @@ -433,6 +487,21 @@ static int wm8960_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } +static struct { + int rate; + unsigned int val; +} alc_rates[] = { + { 48000, 0 }, + { 44100, 0 }, + { 32000, 1 }, + { 22050, 2 }, + { 24000, 2 }, + { 16000, 3 }, + { 11250, 4 }, + { 12000, 4 }, + { 8000, 5 }, +}; + static int wm8960_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -440,7 +509,9 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; + struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); u16 iface = snd_soc_read(codec, WM8960_IFACE1) & 0xfff3; + int i; /* bit size */ switch (params_format(params)) { @@ -454,6 +525,18 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, break; } + /* Update filters for the new rate */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + wm8960->playback_fs = params_rate(params); + wm8960_set_deemph(codec); + } else { + for (i = 0; i < ARRAY_SIZE(alc_rates); i++) + if (alc_rates[i].rate == params_rate(params)) + snd_soc_update_bits(codec, + WM8960_ADDCTL3, 0x7, + alc_rates[i].val); + } + /* set iface */ snd_soc_write(codec, WM8960_IFACE1, iface); return 0; diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index 5b9a756242f1..2549d3a297ab 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -1102,7 +1102,7 @@ static int wm8961_register(struct wm8961_priv *wm8961) ret = wm8961_reset(codec); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset\n"); - return ret; + goto err; } /* Enable class W */ @@ -1147,18 +1147,19 @@ static int wm8961_register(struct wm8961_priv *wm8961) ret = snd_soc_register_codec(codec); if (ret != 0) { dev_err(codec->dev, "Failed to register codec: %d\n", ret); - return ret; + goto err; } ret = snd_soc_register_dai(&wm8961_dai); if (ret != 0) { dev_err(codec->dev, "Failed to register DAI: %d\n", ret); - snd_soc_unregister_codec(codec); - return ret; + goto err_codec; } return 0; +err_codec: + snd_soc_unregister_codec(codec); err: kfree(wm8961); return ret; diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index a2c4b2f37cca..1468fe10cbbe 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -670,7 +670,8 @@ static __devinit int wm8974_register(struct wm8974_priv *wm8974) if (wm8974_codec) { dev_err(codec->dev, "Another WM8974 is registered\n"); - return -EINVAL; + ret = -EINVAL; + goto err; } mutex_init(&codec->mutex); diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 51d5f433215c..8a1ad778e7e3 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -1076,7 +1076,6 @@ static __devinit int wm8978_register(struct wm8978_priv *wm8978) err_codec: snd_soc_unregister_codec(codec); err: - kfree(wm8978); return ret; } @@ -1085,13 +1084,13 @@ static __devexit void wm8978_unregister(struct wm8978_priv *wm8978) wm8978_set_bias_level(&wm8978->codec, SND_SOC_BIAS_OFF); snd_soc_unregister_dai(&wm8978_dai); snd_soc_unregister_codec(&wm8978->codec); - kfree(wm8978); wm8978_codec = NULL; } static __devinit int wm8978_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { + int ret; struct wm8978_priv *wm8978; struct snd_soc_codec *codec; @@ -1107,13 +1106,18 @@ static __devinit int wm8978_i2c_probe(struct i2c_client *i2c, codec->dev = &i2c->dev; - return wm8978_register(wm8978); + ret = wm8978_register(wm8978); + if (ret < 0) + kfree(wm8978); + + return ret; } static __devexit int wm8978_i2c_remove(struct i2c_client *client) { struct wm8978_priv *wm8978 = i2c_get_clientdata(client); wm8978_unregister(wm8978); + kfree(wm8978); return 0; } diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index c018772cc430..dd8d909788c1 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -30,8 +30,6 @@ #include "wm8990.h" -#define WM8990_VERSION "0.2" - /* codec private data */ struct wm8990_priv { unsigned int sysclk; @@ -1511,8 +1509,6 @@ static int wm8990_probe(struct platform_device *pdev) struct wm8990_priv *wm8990; int ret; - pr_info("WM8990 Audio Codec %s\n", WM8990_VERSION); - setup = socdev->codec_data; codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); if (codec == NULL) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index e84a1177f350..522249d5c2b4 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -95,6 +95,7 @@ struct wm8994_priv { struct wm8994_micdet micdet[2]; + int revision; struct wm8994_pdata *pdata; }; @@ -1677,6 +1678,26 @@ static struct { static int wm8994_readable(unsigned int reg) { + switch (reg) { + case WM8994_GPIO_1: + case WM8994_GPIO_2: + case WM8994_GPIO_3: + case WM8994_GPIO_4: + case WM8994_GPIO_5: + case WM8994_GPIO_6: + case WM8994_GPIO_7: + case WM8994_GPIO_8: + case WM8994_GPIO_9: + case WM8994_GPIO_10: + case WM8994_GPIO_11: + case WM8994_INTERRUPT_STATUS_1: + case WM8994_INTERRUPT_STATUS_2: + case WM8994_INTERRUPT_RAW_STATUS_2: + return 1; + default: + break; + } + if (reg >= ARRAY_SIZE(access_masks)) return 0; return access_masks[reg].readable != 0; @@ -2341,6 +2362,20 @@ SOC_DAPM_SINGLE("AIF2 Switch", WM8994_AIF1_ADC1_RIGHT_MIXER_ROUTING, 0, 1, 0), }; +static const struct snd_kcontrol_new aif1adc2l_mix[] = { +SOC_DAPM_SINGLE("DMIC Switch", WM8994_AIF1_ADC2_LEFT_MIXER_ROUTING, + 1, 1, 0), +SOC_DAPM_SINGLE("AIF2 Switch", WM8994_AIF1_ADC2_LEFT_MIXER_ROUTING, + 0, 1, 0), +}; + +static const struct snd_kcontrol_new aif1adc2r_mix[] = { +SOC_DAPM_SINGLE("DMIC Switch", WM8994_AIF1_ADC2_RIGHT_MIXER_ROUTING, + 1, 1, 0), +SOC_DAPM_SINGLE("AIF2 Switch", WM8994_AIF1_ADC2_RIGHT_MIXER_ROUTING, + 0, 1, 0), +}; + static const struct snd_kcontrol_new aif2dac2l_mix[] = { SOC_DAPM_SINGLE("Right Sidetone Switch", WM8994_DAC2_LEFT_MIXER_ROUTING, 5, 1, 0), @@ -2472,6 +2507,7 @@ static const struct snd_kcontrol_new aif3adc_mux = static const struct snd_soc_dapm_widget wm8994_dapm_widgets[] = { SND_SOC_DAPM_INPUT("DMIC1DAT"), SND_SOC_DAPM_INPUT("DMIC2DAT"), +SND_SOC_DAPM_INPUT("Clock"), SND_SOC_DAPM_SUPPLY("CLK_SYS", SND_SOC_NOPM, 0, 0, clk_sys_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), @@ -2506,6 +2542,11 @@ SND_SOC_DAPM_MIXER("AIF1ADC1L Mixer", SND_SOC_NOPM, 0, 0, SND_SOC_DAPM_MIXER("AIF1ADC1R Mixer", SND_SOC_NOPM, 0, 0, aif1adc1r_mix, ARRAY_SIZE(aif1adc1r_mix)), +SND_SOC_DAPM_MIXER("AIF1ADC2L Mixer", SND_SOC_NOPM, 0, 0, + aif1adc2l_mix, ARRAY_SIZE(aif1adc2l_mix)), +SND_SOC_DAPM_MIXER("AIF1ADC2R Mixer", SND_SOC_NOPM, 0, 0, + aif1adc2r_mix, ARRAY_SIZE(aif1adc2r_mix)), + SND_SOC_DAPM_MIXER("AIF2DAC2L Mixer", SND_SOC_NOPM, 0, 0, aif2dac2l_mix, ARRAY_SIZE(aif2dac2l_mix)), SND_SOC_DAPM_MIXER("AIF2DAC2R Mixer", SND_SOC_NOPM, 0, 0, @@ -2668,6 +2709,14 @@ static const struct snd_soc_dapm_route intercon[] = { { "AIF1ADC1R Mixer", "ADC/DMIC Switch", "ADCR Mux" }, { "AIF1ADC1R Mixer", "AIF2 Switch", "AIF2DACR" }, + { "AIF1ADC2L", NULL, "AIF1ADC2L Mixer" }, + { "AIF1ADC2L Mixer", "DMIC Switch", "DMIC2L" }, + { "AIF1ADC2L Mixer", "AIF2 Switch", "AIF2DACL" }, + + { "AIF1ADC2R", NULL, "AIF1ADC2R Mixer" }, + { "AIF1ADC2R Mixer", "DMIC Switch", "DMIC2R" }, + { "AIF1ADC2R Mixer", "AIF2 Switch", "AIF2DACR" }, + /* Pin level routing for AIF3 */ { "AIF1DAC1L", NULL, "AIF1DAC Mux" }, { "AIF1DAC1R", NULL, "AIF1DAC Mux" }, @@ -2946,11 +2995,14 @@ static int wm8994_set_fll(struct snd_soc_dai *dai, int id, int src, return 0; } +static int opclk_divs[] = { 10, 20, 30, 40, 55, 60, 80, 120, 160 }; + static int wm8994_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = dai->codec; struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + int i; switch (dai->id) { case 1: @@ -2988,6 +3040,25 @@ static int wm8994_set_dai_sysclk(struct snd_soc_dai *dai, dev_dbg(dai->dev, "AIF%d using FLL2\n", dai->id); break; + case WM8994_SYSCLK_OPCLK: + /* Special case - a division (times 10) is given and + * no effect on main clocking. + */ + if (freq) { + for (i = 0; i < ARRAY_SIZE(opclk_divs); i++) + if (opclk_divs[i] == freq) + break; + if (i == ARRAY_SIZE(opclk_divs)) + return -EINVAL; + snd_soc_update_bits(codec, WM8994_CLOCKING_2, + WM8994_OPCLK_DIV_MASK, i); + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_2, + WM8994_OPCLK_ENA, WM8994_OPCLK_ENA); + } else { + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_2, + WM8994_OPCLK_ENA, 0); + } + default: return -EINVAL; } @@ -3000,6 +3071,8 @@ static int wm8994_set_dai_sysclk(struct snd_soc_dai *dai, static int wm8994_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + switch (level) { case SND_SOC_BIAS_ON: break; @@ -3012,11 +3085,16 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->bias_level == SND_SOC_BIAS_OFF) { - /* Tweak DC servo configuration for improved - * performance. */ - snd_soc_write(codec, 0x102, 0x3); - snd_soc_write(codec, 0x56, 0x3); - snd_soc_write(codec, 0x102, 0); + /* Tweak DC servo and DSP configuration for + * improved performance. */ + if (wm8994->revision < 4) { + /* Tweak DC servo and DSP configuration for + * improved performance. */ + snd_soc_write(codec, 0x102, 0x3); + snd_soc_write(codec, 0x56, 0x3); + snd_soc_write(codec, 0x817, 0); + snd_soc_write(codec, 0x102, 0); + } /* Discharge LINEOUT1 & 2 */ snd_soc_update_bits(codec, WM8994_ANTIPOP_1, @@ -3849,7 +3927,6 @@ static int wm8994_codec_probe(struct platform_device *pdev) struct wm8994_priv *wm8994; struct snd_soc_codec *codec; int i; - u16 rev; if (wm8994_codec) { dev_err(&pdev->dev, "Another WM8994 is registered\n"); @@ -3903,8 +3980,8 @@ static int wm8994_codec_probe(struct platform_device *pdev) wm8994->reg_cache[i] = 0; /* Set revision-specific configuration */ - rev = snd_soc_read(codec, WM8994_CHIP_REVISION); - switch (rev) { + wm8994->revision = snd_soc_read(codec, WM8994_CHIP_REVISION); + switch (wm8994->revision) { case 2: case 3: wm8994->hubs.dcs_codes = -5; @@ -4004,6 +4081,11 @@ static int wm8994_codec_probe(struct platform_device *pdev) 1 << WM8994_AIF2DAC_3D_GAIN_SHIFT, 1 << WM8994_AIF2DAC_3D_GAIN_SHIFT); + /* Unconditionally enable AIF1 ADC TDM mode; it only affects + * behaviour on idle TDM clock cycles. */ + snd_soc_update_bits(codec, WM8994_AIF1_CONTROL_1, + WM8994_AIF1ADC_TDM, WM8994_AIF1ADC_TDM); + wm8994_update_class_w(codec); ret = snd_soc_register_codec(codec); diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index 7072dc539354..2e0ca67a8df7 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -20,6 +20,9 @@ extern struct snd_soc_dai wm8994_dai[]; #define WM8994_SYSCLK_FLL1 3 #define WM8994_SYSCLK_FLL2 4 +/* OPCLK is also configured with set_dai_sysclk, specify division*10 as rate. */ +#define WM8994_SYSCLK_OPCLK 5 + #define WM8994_FLL1 1 #define WM8994_FLL2 2 diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 13186fb4dcb4..76b37ff6c264 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1356,7 +1356,7 @@ static int wm9081_register(struct wm9081_priv *wm9081, ret = snd_soc_codec_set_cache_io(codec, 8, 16, control); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; + goto err; } reg = snd_soc_read(codec, WM9081_SOFTWARE_RESET); @@ -1369,7 +1369,7 @@ static int wm9081_register(struct wm9081_priv *wm9081, ret = wm9081_reset(codec); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset\n"); - return ret; + goto err; } wm9081_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -1388,18 +1388,19 @@ static int wm9081_register(struct wm9081_priv *wm9081, ret = snd_soc_register_codec(codec); if (ret != 0) { dev_err(codec->dev, "Failed to register codec: %d\n", ret); - return ret; + goto err; } ret = snd_soc_register_dai(&wm9081_dai); if (ret != 0) { dev_err(codec->dev, "Failed to register DAI: %d\n", ret); - snd_soc_unregister_codec(codec); - return ret; + goto err_codec; } return 0; +err_codec: + snd_soc_unregister_codec(codec); err: kfree(wm9081); return ret; diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 16f1a57da08a..2cb81538cd91 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -410,6 +410,8 @@ static int hp_event(struct snd_soc_dapm_widget *w, WM8993_HPOUT1L_DLY | WM8993_HPOUT1R_DLY, 0); + snd_soc_write(codec, WM8993_DC_SERVO_0, 0); + snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_1, WM8993_HPOUT1L_ENA | WM8993_HPOUT1R_ENA, 0); diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index adadcd3aa1b1..9e8932abf158 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -26,6 +26,7 @@ #include <mach/asp.h> #include "davinci-pcm.h" +#include "davinci-i2s.h" /* @@ -68,16 +69,21 @@ #define DAVINCI_MCBSP_RCR_RDATDLY(v) ((v) << 16) #define DAVINCI_MCBSP_RCR_RFIG (1 << 18) #define DAVINCI_MCBSP_RCR_RWDLEN2(v) ((v) << 21) +#define DAVINCI_MCBSP_RCR_RFRLEN2(v) ((v) << 24) +#define DAVINCI_MCBSP_RCR_RPHASE BIT(31) #define DAVINCI_MCBSP_XCR_XWDLEN1(v) ((v) << 5) #define DAVINCI_MCBSP_XCR_XFRLEN1(v) ((v) << 8) #define DAVINCI_MCBSP_XCR_XDATDLY(v) ((v) << 16) #define DAVINCI_MCBSP_XCR_XFIG (1 << 18) #define DAVINCI_MCBSP_XCR_XWDLEN2(v) ((v) << 21) +#define DAVINCI_MCBSP_XCR_XFRLEN2(v) ((v) << 24) +#define DAVINCI_MCBSP_XCR_XPHASE BIT(31) #define DAVINCI_MCBSP_SRGR_FWID(v) ((v) << 8) #define DAVINCI_MCBSP_SRGR_FPER(v) ((v) << 16) #define DAVINCI_MCBSP_SRGR_FSGM (1 << 28) +#define DAVINCI_MCBSP_SRGR_CLKSM BIT(29) #define DAVINCI_MCBSP_PCR_CLKRP (1 << 0) #define DAVINCI_MCBSP_PCR_CLKXP (1 << 1) @@ -116,6 +122,7 @@ static const unsigned char double_fmt[SNDRV_PCM_FORMAT_S32_LE + 1] = { }; struct davinci_mcbsp_dev { + struct device *dev; struct davinci_pcm_dma_params dma_params[2]; void __iomem *base; #define MOD_DSP_A 0 @@ -144,6 +151,11 @@ struct davinci_mcbsp_dev { * won't end up being swapped because of the underrun. */ unsigned enable_channel_combine:1; + + unsigned int fmt; + int clk_div; + int clk_input_pin; + bool i2s_accurate_sck; }; static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev, @@ -254,10 +266,12 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, struct davinci_mcbsp_dev *dev = cpu_dai->private_data; unsigned int pcr; unsigned int srgr; + /* Attention srgr is updated by hw_params! */ srgr = DAVINCI_MCBSP_SRGR_FSGM | DAVINCI_MCBSP_SRGR_FPER(DEFAULT_BITPERSAMPLE * 2 - 1) | DAVINCI_MCBSP_SRGR_FWID(DEFAULT_BITPERSAMPLE - 1); + dev->fmt = fmt; /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: @@ -268,11 +282,26 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, DAVINCI_MCBSP_PCR_CLKRM; break; case SND_SOC_DAIFMT_CBM_CFS: - /* McBSP CLKR pin is the input for the Sample Rate Generator. - * McBSP FSR and FSX are driven by the Sample Rate Generator. */ - pcr = DAVINCI_MCBSP_PCR_SCLKME | - DAVINCI_MCBSP_PCR_FSXM | - DAVINCI_MCBSP_PCR_FSRM; + pcr = DAVINCI_MCBSP_PCR_FSRM | DAVINCI_MCBSP_PCR_FSXM; + /* + * Selection of the clock input pin that is the + * input for the Sample Rate Generator. + * McBSP FSR and FSX are driven by the Sample Rate + * Generator. + */ + switch (dev->clk_input_pin) { + case MCBSP_CLKS: + pcr |= DAVINCI_MCBSP_PCR_CLKXM | + DAVINCI_MCBSP_PCR_CLKRM; + break; + case MCBSP_CLKR: + pcr |= DAVINCI_MCBSP_PCR_SCLKME; + break; + default: + dev_err(dev->dev, "bad clk_input_pin\n"); + return -EINVAL; + } + break; case SND_SOC_DAIFMT_CBM_CFM: /* codec is master */ @@ -372,6 +401,18 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, return 0; } +static int davinci_i2s_dai_set_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct davinci_mcbsp_dev *dev = cpu_dai->private_data; + + if (div_id != DAVINCI_MCBSP_CLKGDV) + return -ENODEV; + + dev->clk_div = div; + return 0; +} + static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -380,8 +421,8 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, struct davinci_pcm_dma_params *dma_params = &dev->dma_params[substream->stream]; struct snd_interval *i = NULL; - int mcbsp_word_length; - unsigned int rcr, xcr, srgr; + int mcbsp_word_length, master; + unsigned int rcr, xcr, srgr, clk_div, freq, framesize; u32 spcr; snd_pcm_format_t fmt; unsigned element_cnt = 1; @@ -396,12 +437,59 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); } - i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS); - srgr = DAVINCI_MCBSP_SRGR_FSGM; - srgr |= DAVINCI_MCBSP_SRGR_FWID(snd_interval_value(i) - 1); + master = dev->fmt & SND_SOC_DAIFMT_MASTER_MASK; + fmt = params_format(params); + mcbsp_word_length = asp_word_length[fmt]; - i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_FRAME_BITS); - srgr |= DAVINCI_MCBSP_SRGR_FPER(snd_interval_value(i) - 1); + switch (master) { + case SND_SOC_DAIFMT_CBS_CFS: + freq = clk_get_rate(dev->clk); + srgr = DAVINCI_MCBSP_SRGR_FSGM | + DAVINCI_MCBSP_SRGR_CLKSM; + srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length * + 8 - 1); + if (dev->i2s_accurate_sck) { + clk_div = 256; + do { + framesize = (freq / (--clk_div)) / + params->rate_num * + params->rate_den; + } while (((framesize < 33) || (framesize > 4095)) && + (clk_div)); + clk_div--; + srgr |= DAVINCI_MCBSP_SRGR_FPER(framesize - 1); + } else { + /* symmetric waveforms */ + clk_div = freq / (mcbsp_word_length * 16) / + params->rate_num * params->rate_den; + srgr |= DAVINCI_MCBSP_SRGR_FPER(mcbsp_word_length * + 16 - 1); + } + clk_div &= 0xFF; + srgr |= clk_div; + break; + case SND_SOC_DAIFMT_CBM_CFS: + srgr = DAVINCI_MCBSP_SRGR_FSGM; + clk_div = dev->clk_div - 1; + srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length * 8 - 1); + srgr |= DAVINCI_MCBSP_SRGR_FPER(mcbsp_word_length * 16 - 1); + clk_div &= 0xFF; + srgr |= clk_div; + break; + case SND_SOC_DAIFMT_CBM_CFM: + /* Clock and frame sync given from external sources */ + i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS); + srgr = DAVINCI_MCBSP_SRGR_FSGM; + srgr |= DAVINCI_MCBSP_SRGR_FWID(snd_interval_value(i) - 1); + pr_debug("%s - %d FWID set: re-read srgr = %X\n", + __func__, __LINE__, snd_interval_value(i) - 1); + + i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_FRAME_BITS); + srgr |= DAVINCI_MCBSP_SRGR_FPER(snd_interval_value(i) - 1); + break; + default: + return -EINVAL; + } davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, srgr); rcr = DAVINCI_MCBSP_RCR_RFIG; @@ -426,12 +514,41 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, element_cnt = 1; fmt = double_fmt[fmt]; } + switch (master) { + case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBS_CFM: + rcr |= DAVINCI_MCBSP_RCR_RFRLEN2(0); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN2(0); + rcr |= DAVINCI_MCBSP_RCR_RPHASE; + xcr |= DAVINCI_MCBSP_XCR_XPHASE; + break; + case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBM_CFS: + rcr |= DAVINCI_MCBSP_RCR_RFRLEN2(element_cnt - 1); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN2(element_cnt - 1); + break; + default: + return -EINVAL; + } } dma_params->acnt = dma_params->data_type = data_type[fmt]; dma_params->fifo_level = 0; mcbsp_word_length = asp_word_length[fmt]; - rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(element_cnt - 1); - xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(element_cnt - 1); + + switch (master) { + case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBS_CFM: + rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(0); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(0); + break; + case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBM_CFS: + rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(element_cnt - 1); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(element_cnt - 1); + break; + default: + return -EINVAL; + } rcr |= DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) | DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length); @@ -442,6 +559,10 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, xcr); else davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, rcr); + + pr_debug("%s - %d srgr=%X\n", __func__, __LINE__, srgr); + pr_debug("%s - %d xcr=%X\n", __func__, __LINE__, xcr); + pr_debug("%s - %d rcr=%X\n", __func__, __LINE__, rcr); return 0; } @@ -500,6 +621,7 @@ static struct snd_soc_dai_ops davinci_i2s_dai_ops = { .trigger = davinci_i2s_trigger, .hw_params = davinci_i2s_hw_params, .set_fmt = davinci_i2s_set_dai_fmt, + .set_clkdiv = davinci_i2s_dai_set_clkdiv, }; @@ -526,6 +648,8 @@ static int davinci_i2s_probe(struct platform_device *pdev) struct snd_platform_data *pdata = pdev->dev.platform_data; struct davinci_mcbsp_dev *dev; struct resource *mem, *ioarea, *res; + enum dma_event_q asp_chan_q = EVENTQ_0; + enum dma_event_q ram_chan_q = EVENTQ_1; int ret; mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); @@ -552,7 +676,17 @@ static int davinci_i2s_probe(struct platform_device *pdev) pdata->sram_size_playback; dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].sram_size = pdata->sram_size_capture; + dev->clk_input_pin = pdata->clk_input_pin; + dev->i2s_accurate_sck = pdata->i2s_accurate_sck; + asp_chan_q = pdata->asp_chan_q; + ram_chan_q = pdata->ram_chan_q; } + + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].asp_chan_q = asp_chan_q; + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].ram_chan_q = ram_chan_q; + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].asp_chan_q = asp_chan_q; + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].ram_chan_q = ram_chan_q; + dev->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(dev->clk)) { ret = -ENODEV; @@ -584,6 +718,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) goto err_free_mem; } dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = res->start; + dev->dev = &pdev->dev; davinci_i2s_dai.private_data = dev; davinci_i2s_dai.capture.dma_data = dev->dma_params; diff --git a/sound/soc/davinci/davinci-i2s.h b/sound/soc/davinci/davinci-i2s.h index 241648ce8873..0b1e77b8c279 100644 --- a/sound/soc/davinci/davinci-i2s.h +++ b/sound/soc/davinci/davinci-i2s.h @@ -12,6 +12,11 @@ #ifndef _DAVINCI_I2S_H #define _DAVINCI_I2S_H +/* McBSP dividers */ +enum davinci_mcbsp_div { + DAVINCI_MCBSP_CLKGDV, /* Sample rate generator divider */ +}; + extern struct snd_soc_dai davinci_i2s_dai; #endif diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index d3955096d872..b24720894af6 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -890,7 +890,8 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dev->rxnumevt = pdata->rxnumevt; dma_data = &dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]; - dma_data->eventq_no = pdata->eventq_no; + dma_data->asp_chan_q = pdata->asp_chan_q; + dma_data->ram_chan_q = pdata->ram_chan_q; dma_data->dma_addr = (dma_addr_t) (pdata->tx_dma_offset + io_v2p(dev->base)); @@ -904,7 +905,8 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dma_data->channel = res->start; dma_data = &dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]; - dma_data->eventq_no = pdata->eventq_no; + dma_data->asp_chan_q = pdata->asp_chan_q; + dma_data->ram_chan_q = pdata->ram_chan_q; dma_data->dma_addr = (dma_addr_t)(pdata->rx_dma_offset + io_v2p(dev->base)); diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 2dc406f42fe7..a7124116d2e0 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -381,7 +381,7 @@ static int request_ping_pong(struct snd_pcm_substream *substream, /* Request ram master channel */ link = prtd->ram_channel = edma_alloc_channel(EDMA_CHANNEL_ANY, davinci_pcm_dma_irq, substream, - EVENTQ_1); + prtd->params->ram_chan_q); if (link < 0) goto exit1; @@ -477,7 +477,8 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) /* Request asp master DMA channel */ link = prtd->asp_channel = edma_alloc_channel(params->channel, - davinci_pcm_dma_irq, substream, EVENTQ_0); + davinci_pcm_dma_irq, substream, + prtd->params->asp_chan_q); if (link < 0) goto exit1; @@ -800,7 +801,7 @@ static void davinci_pcm_free(struct snd_pcm *pcm) dma_free_writecombine(pcm->card->dev, buf->bytes, buf->area, buf->addr); buf->area = NULL; - iram_dma = (struct snd_dma_buffer *)buf->private_data; + iram_dma = buf->private_data; if (iram_dma) { sram_free(iram_dma->area, iram_dma->bytes); kfree(iram_dma); diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index 0764944cf10f..b799a02333d8 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -21,7 +21,8 @@ struct davinci_pcm_dma_params { unsigned short acnt; dma_addr_t dma_addr; /* device physical address for DMA */ unsigned sram_size; - enum dma_event_q eventq_no; /* event queue number */ + enum dma_event_q asp_chan_q; /* event queue number for ASP channel */ + enum dma_event_q ram_chan_q; /* event queue number for RAM channel */ unsigned char data_type; /* xfer data type */ unsigned char convert_mono_stereo; unsigned int fifo_level; diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c index 40eccfe9e358..4948a79f86a0 100644 --- a/sound/soc/davinci/davinci-sffsdr.c +++ b/sound/soc/davinci/davinci-sffsdr.c @@ -150,7 +150,7 @@ static int __init sffsdr_init(void) sffsdr_snd_resources, ARRAY_SIZE(sffsdr_snd_resources)); if (ret) { - printk(KERN_ERR "platform device add ressources failed\n"); + printk(KERN_ERR "platform device add resources failed\n"); goto error; } diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c index 9aa980d38231..48678533da7a 100644 --- a/sound/soc/davinci/davinci-vcif.c +++ b/sound/soc/davinci/davinci-vcif.c @@ -203,7 +203,7 @@ static int davinci_vcif_probe(struct platform_device *pdev) int ret; davinci_vcif_dev = kzalloc(sizeof(struct davinci_vcif_dev), GFP_KERNEL); - if (!davinci_vc) { + if (!davinci_vcif_dev) { dev_dbg(&pdev->dev, "could not allocate memory for private data\n"); return -ENOMEM; diff --git a/sound/soc/ep93xx/Kconfig b/sound/soc/ep93xx/Kconfig new file mode 100644 index 000000000000..f617f560f46b --- /dev/null +++ b/sound/soc/ep93xx/Kconfig @@ -0,0 +1,18 @@ +config SND_EP93XX_SOC + tristate "SoC Audio support for the Cirrus Logic EP93xx series" + depends on ARCH_EP93XX && SND_SOC + help + Say Y or M if you want to add support for codecs attached to + the EP93xx I2S interface. + +config SND_EP93XX_SOC_I2S + tristate + +config SND_EP93XX_SOC_SNAPPERCL15 + tristate "SoC Audio support for Bluewater Systems Snapper CL15 module" + depends on SND_EP93XX_SOC && MACH_SNAPPER_CL15 + select SND_EP93XX_SOC_I2S + select SND_SOC_TLV320AIC23 + help + Say Y or M here if you want to add support for I2S audio on the + Bluewater Systems Snapper CL15 module. diff --git a/sound/soc/ep93xx/Makefile b/sound/soc/ep93xx/Makefile new file mode 100644 index 000000000000..272e60f57b9a --- /dev/null +++ b/sound/soc/ep93xx/Makefile @@ -0,0 +1,11 @@ +# EP93xx Platform Support +snd-soc-ep93xx-objs := ep93xx-pcm.o +snd-soc-ep93xx-i2s-objs := ep93xx-i2s.o + +obj-$(CONFIG_SND_EP93XX_SOC) += snd-soc-ep93xx.o +obj-$(CONFIG_SND_EP93XX_SOC_I2S) += snd-soc-ep93xx-i2s.o + +# EP93XX Machine Support +snd-soc-snappercl15-objs := snappercl15.o + +obj-$(CONFIG_SND_EP93XX_SOC_SNAPPERCL15) += snd-soc-snappercl15.o diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c new file mode 100644 index 000000000000..00b946632184 --- /dev/null +++ b/sound/soc/ep93xx/ep93xx-i2s.c @@ -0,0 +1,487 @@ +/* + * linux/sound/soc/ep93xx-i2s.c + * EP93xx I2S driver + * + * Copyright (C) 2010 Ryan Mallon <ryan@bluewatersys.com> + * + * Based on the original driver by: + * Copyright (C) 2007 Chase Douglas <chasedouglas@gmail> + * Copyright (C) 2006 Lennert Buytenhek <buytenh@wantstofly.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/slab.h> +#include <linux/clk.h> +#include <linux/io.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include <mach/hardware.h> +#include <mach/ep93xx-regs.h> +#include <mach/dma.h> + +#include "ep93xx-pcm.h" +#include "ep93xx-i2s.h" + +#define EP93XX_I2S_TXCLKCFG 0x00 +#define EP93XX_I2S_RXCLKCFG 0x04 +#define EP93XX_I2S_GLCTRL 0x0C + +#define EP93XX_I2S_TXLINCTRLDATA 0x28 +#define EP93XX_I2S_TXCTRL 0x2C +#define EP93XX_I2S_TXWRDLEN 0x30 +#define EP93XX_I2S_TX0EN 0x34 + +#define EP93XX_I2S_RXLINCTRLDATA 0x58 +#define EP93XX_I2S_RXCTRL 0x5C +#define EP93XX_I2S_RXWRDLEN 0x60 +#define EP93XX_I2S_RX0EN 0x64 + +#define EP93XX_I2S_WRDLEN_16 (0 << 0) +#define EP93XX_I2S_WRDLEN_24 (1 << 0) +#define EP93XX_I2S_WRDLEN_32 (2 << 0) + +#define EP93XX_I2S_LINCTRLDATA_R_JUST (1 << 2) /* Right justify */ + +#define EP93XX_I2S_CLKCFG_LRS (1 << 0) /* lrclk polarity */ +#define EP93XX_I2S_CLKCFG_CKP (1 << 1) /* Bit clock polarity */ +#define EP93XX_I2S_CLKCFG_REL (1 << 2) /* First bit transition */ +#define EP93XX_I2S_CLKCFG_MASTER (1 << 3) /* Master mode */ +#define EP93XX_I2S_CLKCFG_NBCG (1 << 4) /* Not bit clock gating */ + +struct ep93xx_i2s_info { + struct clk *mclk; + struct clk *sclk; + struct clk *lrclk; + struct ep93xx_pcm_dma_params *dma_params; + struct resource *mem; + void __iomem *regs; +}; + +struct ep93xx_pcm_dma_params ep93xx_i2s_dma_params[] = { + [SNDRV_PCM_STREAM_PLAYBACK] = { + .name = "i2s-pcm-out", + .dma_port = EP93XX_DMA_M2P_PORT_I2S1, + }, + [SNDRV_PCM_STREAM_CAPTURE] = { + .name = "i2s-pcm-in", + .dma_port = EP93XX_DMA_M2P_PORT_I2S1, + }, +}; + +static inline void ep93xx_i2s_write_reg(struct ep93xx_i2s_info *info, + unsigned reg, unsigned val) +{ + __raw_writel(val, info->regs + reg); +} + +static inline unsigned ep93xx_i2s_read_reg(struct ep93xx_i2s_info *info, + unsigned reg) +{ + return __raw_readl(info->regs + reg); +} + +static void ep93xx_i2s_enable(struct ep93xx_i2s_info *info, int stream) +{ + unsigned base_reg; + int i; + + if ((ep93xx_i2s_read_reg(info, EP93XX_I2S_TX0EN) & 0x1) == 0 && + (ep93xx_i2s_read_reg(info, EP93XX_I2S_RX0EN) & 0x1) == 0) { + /* Enable clocks */ + clk_enable(info->mclk); + clk_enable(info->sclk); + clk_enable(info->lrclk); + + /* Enable i2s */ + ep93xx_i2s_write_reg(info, EP93XX_I2S_GLCTRL, 1); + } + + /* Enable fifos */ + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + base_reg = EP93XX_I2S_TX0EN; + else + base_reg = EP93XX_I2S_RX0EN; + for (i = 0; i < 3; i++) + ep93xx_i2s_write_reg(info, base_reg + (i * 4), 1); +} + +static void ep93xx_i2s_disable(struct ep93xx_i2s_info *info, int stream) +{ + unsigned base_reg; + int i; + + /* Disable fifos */ + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + base_reg = EP93XX_I2S_TX0EN; + else + base_reg = EP93XX_I2S_RX0EN; + for (i = 0; i < 3; i++) + ep93xx_i2s_write_reg(info, base_reg + (i * 4), 0); + + if ((ep93xx_i2s_read_reg(info, EP93XX_I2S_TX0EN) & 0x1) == 0 && + (ep93xx_i2s_read_reg(info, EP93XX_I2S_RX0EN) & 0x1) == 0) { + /* Disable i2s */ + ep93xx_i2s_write_reg(info, EP93XX_I2S_GLCTRL, 0); + + /* Disable clocks */ + clk_disable(info->lrclk); + clk_disable(info->sclk); + clk_disable(info->mclk); + } +} + +static int ep93xx_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct ep93xx_i2s_info *info = rtd->dai->cpu_dai->private_data; + + snd_soc_dai_set_dma_data(cpu_dai, substream, + &info->dma_params[substream->stream]); + return 0; +} + +static void ep93xx_i2s_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct ep93xx_i2s_info *info = rtd->dai->cpu_dai->private_data; + + ep93xx_i2s_disable(info, substream->stream); +} + +static int ep93xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct ep93xx_i2s_info *info = cpu_dai->private_data; + unsigned int clk_cfg, lin_ctrl; + + clk_cfg = ep93xx_i2s_read_reg(info, EP93XX_I2S_RXCLKCFG); + lin_ctrl = ep93xx_i2s_read_reg(info, EP93XX_I2S_RXLINCTRLDATA); + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + clk_cfg |= EP93XX_I2S_CLKCFG_REL; + lin_ctrl &= ~EP93XX_I2S_LINCTRLDATA_R_JUST; + break; + + case SND_SOC_DAIFMT_LEFT_J: + clk_cfg &= ~EP93XX_I2S_CLKCFG_REL; + lin_ctrl &= ~EP93XX_I2S_LINCTRLDATA_R_JUST; + break; + + case SND_SOC_DAIFMT_RIGHT_J: + clk_cfg &= ~EP93XX_I2S_CLKCFG_REL; + lin_ctrl |= EP93XX_I2S_LINCTRLDATA_R_JUST; + break; + + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + /* CPU is master */ + clk_cfg |= EP93XX_I2S_CLKCFG_MASTER; + break; + + case SND_SOC_DAIFMT_CBM_CFM: + /* Codec is master */ + clk_cfg &= ~EP93XX_I2S_CLKCFG_MASTER; + break; + + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + /* Negative bit clock, lrclk low on left word */ + clk_cfg &= ~(EP93XX_I2S_CLKCFG_CKP | EP93XX_I2S_CLKCFG_REL); + break; + + case SND_SOC_DAIFMT_NB_IF: + /* Negative bit clock, lrclk low on right word */ + clk_cfg &= ~EP93XX_I2S_CLKCFG_CKP; + clk_cfg |= EP93XX_I2S_CLKCFG_REL; + break; + + case SND_SOC_DAIFMT_IB_NF: + /* Positive bit clock, lrclk low on left word */ + clk_cfg |= EP93XX_I2S_CLKCFG_CKP; + clk_cfg &= ~EP93XX_I2S_CLKCFG_REL; + break; + + case SND_SOC_DAIFMT_IB_IF: + /* Positive bit clock, lrclk low on right word */ + clk_cfg |= EP93XX_I2S_CLKCFG_CKP | EP93XX_I2S_CLKCFG_REL; + break; + } + + /* Write new register values */ + ep93xx_i2s_write_reg(info, EP93XX_I2S_RXCLKCFG, clk_cfg); + ep93xx_i2s_write_reg(info, EP93XX_I2S_TXCLKCFG, clk_cfg); + ep93xx_i2s_write_reg(info, EP93XX_I2S_RXLINCTRLDATA, lin_ctrl); + ep93xx_i2s_write_reg(info, EP93XX_I2S_TXLINCTRLDATA, lin_ctrl); + return 0; +} + +static int ep93xx_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct ep93xx_i2s_info *info = cpu_dai->private_data; + unsigned word_len, div, sdiv, lrdiv; + int found = 0, err; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + word_len = EP93XX_I2S_WRDLEN_16; + break; + + case SNDRV_PCM_FORMAT_S24_LE: + word_len = EP93XX_I2S_WRDLEN_24; + break; + + case SNDRV_PCM_FORMAT_S32_LE: + word_len = EP93XX_I2S_WRDLEN_32; + break; + + default: + return -EINVAL; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + ep93xx_i2s_write_reg(info, EP93XX_I2S_TXWRDLEN, word_len); + else + ep93xx_i2s_write_reg(info, EP93XX_I2S_RXWRDLEN, word_len); + + /* + * Calculate the sdiv (bit clock) and lrdiv (left/right clock) values. + * If the lrclk is pulse length is larger than the word size, then the + * bit clock will be gated for the unused bits. + */ + div = (clk_get_rate(info->mclk) / params_rate(params)) * + params_channels(params); + for (sdiv = 2; sdiv <= 4; sdiv += 2) + for (lrdiv = 32; lrdiv <= 128; lrdiv <<= 1) + if (sdiv * lrdiv == div) { + found = 1; + goto out; + } +out: + if (!found) + return -EINVAL; + + err = clk_set_rate(info->sclk, clk_get_rate(info->mclk) / sdiv); + if (err) + return err; + + err = clk_set_rate(info->lrclk, clk_get_rate(info->sclk) / lrdiv); + if (err) + return err; + + ep93xx_i2s_enable(info, substream->stream); + return 0; +} + +static int ep93xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, + unsigned int freq, int dir) +{ + struct ep93xx_i2s_info *info = cpu_dai->private_data; + + if (dir == SND_SOC_CLOCK_IN || clk_id != 0) + return -EINVAL; + + return clk_set_rate(info->mclk, freq); +} + +#ifdef CONFIG_PM +static int ep93xx_i2s_suspend(struct snd_soc_dai *dai) +{ + struct ep93xx_i2s_info *info = dai->private_data; + + if (!dai->active) + return; + + ep93xx_i2s_disable(info, SNDRV_PCM_STREAM_PLAYBACK); + ep93xx_i2s_disable(info, SNDRV_PCM_STREAM_CAPTURE); +} + +static int ep93xx_i2s_resume(struct snd_soc_dai *dai) +{ + struct ep93xx_i2s_info *info = dai->private_data; + + if (!dai->active) + return; + + ep93xx_i2s_enable(info, SNDRV_PCM_STREAM_PLAYBACK); + ep93xx_i2s_enable(info, SNDRV_PCM_STREAM_CAPTURE); +} +#else +#define ep93xx_i2s_suspend NULL +#define ep93xx_i2s_resume NULL +#endif + +static struct snd_soc_dai_ops ep93xx_i2s_dai_ops = { + .startup = ep93xx_i2s_startup, + .shutdown = ep93xx_i2s_shutdown, + .hw_params = ep93xx_i2s_hw_params, + .set_sysclk = ep93xx_i2s_set_sysclk, + .set_fmt = ep93xx_i2s_set_dai_fmt, +}; + +#define EP93XX_I2S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_dai ep93xx_i2s_dai = { + .name = "ep93xx-i2s", + .id = 0, + .symmetric_rates= 1, + .suspend = ep93xx_i2s_suspend, + .resume = ep93xx_i2s_resume, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = EP93XX_I2S_FORMATS, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = EP93XX_I2S_FORMATS, + }, + .ops = &ep93xx_i2s_dai_ops, +}; +EXPORT_SYMBOL_GPL(ep93xx_i2s_dai); + +static int ep93xx_i2s_probe(struct platform_device *pdev) +{ + struct ep93xx_i2s_info *info; + struct resource *res; + int err; + + info = kzalloc(sizeof(struct ep93xx_i2s_info), GFP_KERNEL); + if (!info) { + err = -ENOMEM; + goto fail; + } + + ep93xx_i2s_dai.dev = &pdev->dev; + ep93xx_i2s_dai.private_data = info; + info->dma_params = ep93xx_i2s_dma_params; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + err = -ENODEV; + goto fail; + } + + info->mem = request_mem_region(res->start, resource_size(res), + pdev->name); + if (!info->mem) { + err = -EBUSY; + goto fail; + } + + info->regs = ioremap(info->mem->start, resource_size(info->mem)); + if (!info->regs) { + err = -ENXIO; + goto fail_release_mem; + } + + info->mclk = clk_get(&pdev->dev, "mclk"); + if (IS_ERR(info->mclk)) { + err = PTR_ERR(info->mclk); + goto fail_unmap_mem; + } + + info->sclk = clk_get(&pdev->dev, "sclk"); + if (IS_ERR(info->sclk)) { + err = PTR_ERR(info->sclk); + goto fail_put_mclk; + } + + info->lrclk = clk_get(&pdev->dev, "lrclk"); + if (IS_ERR(info->lrclk)) { + err = PTR_ERR(info->lrclk); + goto fail_put_sclk; + } + + err = snd_soc_register_dai(&ep93xx_i2s_dai); + if (err) + goto fail_put_lrclk; + + return 0; + +fail_put_lrclk: + clk_put(info->lrclk); +fail_put_sclk: + clk_put(info->sclk); +fail_put_mclk: + clk_put(info->mclk); +fail_unmap_mem: + iounmap(info->regs); +fail_release_mem: + release_mem_region(info->mem->start, resource_size(info->mem)); + kfree(info); +fail: + return err; +} + +static int __devexit ep93xx_i2s_remove(struct platform_device *pdev) +{ + struct ep93xx_i2s_info *info = ep93xx_i2s_dai.private_data; + + snd_soc_unregister_dai(&ep93xx_i2s_dai); + clk_put(info->lrclk); + clk_put(info->sclk); + clk_put(info->mclk); + iounmap(info->regs); + release_mem_region(info->mem->start, resource_size(info->mem)); + kfree(info); + return 0; +} + +static struct platform_driver ep93xx_i2s_driver = { + .probe = ep93xx_i2s_probe, + .remove = __devexit_p(ep93xx_i2s_remove), + .driver = { + .name = "ep93xx-i2s", + .owner = THIS_MODULE, + }, +}; + +static int __init ep93xx_i2s_init(void) +{ + return platform_driver_register(&ep93xx_i2s_driver); +} + +static void __exit ep93xx_i2s_exit(void) +{ + platform_driver_unregister(&ep93xx_i2s_driver); +} + +module_init(ep93xx_i2s_init); +module_exit(ep93xx_i2s_exit); + +MODULE_ALIAS("platform:ep93xx-i2s"); +MODULE_AUTHOR("Ryan Mallon <ryan@bluewatersys.com>"); +MODULE_DESCRIPTION("EP93XX I2S driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/ep93xx/ep93xx-i2s.h b/sound/soc/ep93xx/ep93xx-i2s.h new file mode 100644 index 000000000000..3bd4ebfaa1de --- /dev/null +++ b/sound/soc/ep93xx/ep93xx-i2s.h @@ -0,0 +1,18 @@ +/* + * linux/sound/soc/ep93xx-i2s.h + * EP93xx I2S driver + * + * Copyright (C) 2010 Ryan Mallon <ryan@bluewatersys.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#ifndef _EP93XX_SND_SOC_I2S_H +#define _EP93XX_SND_SOC_I2S_H + +extern struct snd_soc_dai ep93xx_i2s_dai; + +#endif /* _EP93XX_SND_SOC_I2S_H */ diff --git a/sound/soc/ep93xx/ep93xx-pcm.c b/sound/soc/ep93xx/ep93xx-pcm.c new file mode 100644 index 000000000000..4ba938400791 --- /dev/null +++ b/sound/soc/ep93xx/ep93xx-pcm.c @@ -0,0 +1,319 @@ +/* + * linux/sound/arm/ep93xx-pcm.c - EP93xx ALSA PCM interface + * + * Copyright (C) 2006 Lennert Buytenhek <buytenh@wantstofly.org> + * Copyright (C) 2006 Applied Data Systems + * + * Rewritten for the SoC audio subsystem (Based on PXA2xx code): + * Copyright (c) 2008 Ryan Mallon <ryan@bluewatersys.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/device.h> +#include <linux/slab.h> +#include <linux/dma-mapping.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <mach/dma.h> +#include <mach/hardware.h> +#include <mach/ep93xx-regs.h> + +#include "ep93xx-pcm.h" + +static const struct snd_pcm_hardware ep93xx_pcm_hardware = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER), + + .rates = SNDRV_PCM_RATE_8000_48000, + .rate_min = SNDRV_PCM_RATE_8000, + .rate_max = SNDRV_PCM_RATE_48000, + + .formats = (SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE), + + .buffer_bytes_max = 131072, + .period_bytes_min = 32, + .period_bytes_max = 32768, + .periods_min = 1, + .periods_max = 32, + .fifo_size = 32, +}; + +struct ep93xx_runtime_data +{ + struct ep93xx_dma_m2p_client cl; + struct ep93xx_pcm_dma_params *params; + int pointer_bytes; + struct tasklet_struct period_tasklet; + int periods; + struct ep93xx_dma_buffer buf[32]; +}; + +static void ep93xx_pcm_period_elapsed(unsigned long data) +{ + struct snd_pcm_substream *substream = (struct snd_pcm_substream *)data; + snd_pcm_period_elapsed(substream); +} + +static void ep93xx_pcm_buffer_started(void *cookie, + struct ep93xx_dma_buffer *buf) +{ +} + +static void ep93xx_pcm_buffer_finished(void *cookie, + struct ep93xx_dma_buffer *buf, + int bytes, int error) +{ + struct snd_pcm_substream *substream = cookie; + struct ep93xx_runtime_data *rtd = substream->runtime->private_data; + + if (buf == rtd->buf + rtd->periods - 1) + rtd->pointer_bytes = 0; + else + rtd->pointer_bytes += buf->size; + + if (!error) { + ep93xx_dma_m2p_submit_recursive(&rtd->cl, buf); + tasklet_schedule(&rtd->period_tasklet); + } else { + snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + } +} + +static int ep93xx_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *soc_rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = soc_rtd->dai->cpu_dai; + struct ep93xx_pcm_dma_params *dma_params; + struct ep93xx_runtime_data *rtd; + int ret; + + dma_params = snd_soc_dai_get_dma_data(cpu_dai, substream); + snd_soc_set_runtime_hwparams(substream, &ep93xx_pcm_hardware); + + rtd = kmalloc(sizeof(*rtd), GFP_KERNEL); + if (!rtd) + return -ENOMEM; + + memset(&rtd->period_tasklet, 0, sizeof(rtd->period_tasklet)); + rtd->period_tasklet.func = ep93xx_pcm_period_elapsed; + rtd->period_tasklet.data = (unsigned long)substream; + + rtd->cl.name = dma_params->name; + rtd->cl.flags = dma_params->dma_port | EP93XX_DMA_M2P_IGNORE_ERROR | + ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + EP93XX_DMA_M2P_TX : EP93XX_DMA_M2P_RX); + rtd->cl.cookie = substream; + rtd->cl.buffer_started = ep93xx_pcm_buffer_started; + rtd->cl.buffer_finished = ep93xx_pcm_buffer_finished; + ret = ep93xx_dma_m2p_client_register(&rtd->cl); + if (ret < 0) { + kfree(rtd); + return ret; + } + + substream->runtime->private_data = rtd; + return 0; +} + +static int ep93xx_pcm_close(struct snd_pcm_substream *substream) +{ + struct ep93xx_runtime_data *rtd = substream->runtime->private_data; + + ep93xx_dma_m2p_client_unregister(&rtd->cl); + kfree(rtd); + return 0; +} + +static int ep93xx_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct ep93xx_runtime_data *rtd = runtime->private_data; + size_t totsize = params_buffer_bytes(params); + size_t period = params_period_bytes(params); + int i; + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = totsize; + + rtd->periods = (totsize + period - 1) / period; + for (i = 0; i < rtd->periods; i++) { + rtd->buf[i].bus_addr = runtime->dma_addr + (i * period); + rtd->buf[i].size = period; + if ((i + 1) * period > totsize) + rtd->buf[i].size = totsize - (i * period); + } + + return 0; +} + +static int ep93xx_pcm_hw_free(struct snd_pcm_substream *substream) +{ + snd_pcm_set_runtime_buffer(substream, NULL); + return 0; +} + +static int ep93xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct ep93xx_runtime_data *rtd = substream->runtime->private_data; + int ret; + int i; + + ret = 0; + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + rtd->pointer_bytes = 0; + for (i = 0; i < rtd->periods; i++) + ep93xx_dma_m2p_submit(&rtd->cl, rtd->buf + i); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ep93xx_dma_m2p_flush(&rtd->cl); + break; + + default: + ret = -EINVAL; + break; + } + + return ret; +} + +static snd_pcm_uframes_t ep93xx_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct ep93xx_runtime_data *rtd = substream->runtime->private_data; + + /* FIXME: implement this with sub-period granularity */ + return bytes_to_frames(runtime, rtd->pointer_bytes); +} + +static int ep93xx_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + return dma_mmap_writecombine(substream->pcm->card->dev, vma, + runtime->dma_area, + runtime->dma_addr, + runtime->dma_bytes); +} + +static struct snd_pcm_ops ep93xx_pcm_ops = { + .open = ep93xx_pcm_open, + .close = ep93xx_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = ep93xx_pcm_hw_params, + .hw_free = ep93xx_pcm_hw_free, + .trigger = ep93xx_pcm_trigger, + .pointer = ep93xx_pcm_pointer, + .mmap = ep93xx_pcm_mmap, +}; + +static int ep93xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = ep93xx_pcm_hardware.buffer_bytes_max; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->area = dma_alloc_writecombine(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + buf->bytes = size; + + return (buf->area == NULL) ? -ENOMEM : 0; +} + +static void ep93xx_pcm_free_dma_buffers(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + + dma_free_writecombine(pcm->card->dev, buf->bytes, buf->area, + buf->addr); + buf->area = NULL; + } +} + +static u64 ep93xx_pcm_dmamask = 0xffffffff; + +static int ep93xx_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, + struct snd_pcm *pcm) +{ + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &ep93xx_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = 0xffffffff; + + if (dai->playback.channels_min) { + ret = ep93xx_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + return ret; + } + + if (dai->capture.channels_min) { + ret = ep93xx_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + return ret; + } + + return 0; +} + +struct snd_soc_platform ep93xx_soc_platform = { + .name = "ep93xx-audio", + .pcm_ops = &ep93xx_pcm_ops, + .pcm_new = &ep93xx_pcm_new, + .pcm_free = &ep93xx_pcm_free_dma_buffers, +}; +EXPORT_SYMBOL_GPL(ep93xx_soc_platform); + +static int __init ep93xx_soc_platform_init(void) +{ + return snd_soc_register_platform(&ep93xx_soc_platform); +} + +static void __exit ep93xx_soc_platform_exit(void) +{ + snd_soc_unregister_platform(&ep93xx_soc_platform); +} + +module_init(ep93xx_soc_platform_init); +module_exit(ep93xx_soc_platform_exit); + +MODULE_AUTHOR("Ryan Mallon <ryan@bluewatersys.com>"); +MODULE_DESCRIPTION("EP93xx ALSA PCM interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/ep93xx/ep93xx-pcm.h b/sound/soc/ep93xx/ep93xx-pcm.h new file mode 100644 index 000000000000..4ffdd3f62fe9 --- /dev/null +++ b/sound/soc/ep93xx/ep93xx-pcm.h @@ -0,0 +1,22 @@ +/* + * sound/soc/ep93xx/ep93xx-pcm.h - EP93xx ALSA PCM interface + * + * Copyright (C) 2006 Lennert Buytenhek <buytenh@wantstofly.org> + * Copyright (C) 2006 Applied Data Systems + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _EP93XX_SND_SOC_PCM_H +#define _EP93XX_SND_SOC_PCM_H + +struct ep93xx_pcm_dma_params { + char *name; + int dma_port; +}; + +extern struct snd_soc_platform ep93xx_soc_platform; + +#endif /* _EP93XX_SND_SOC_PCM_H */ diff --git a/sound/soc/ep93xx/snappercl15.c b/sound/soc/ep93xx/snappercl15.c new file mode 100644 index 000000000000..64955340ff75 --- /dev/null +++ b/sound/soc/ep93xx/snappercl15.c @@ -0,0 +1,150 @@ +/* + * snappercl15.c -- SoC audio for Bluewater Systems Snapper CL15 module + * + * Copyright (C) 2008 Bluewater Systems Ltd + * Author: Ryan Mallon <ryan@bluewatersys.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <mach/hardware.h> + +#include "../codecs/tlv320aic23.h" +#include "ep93xx-pcm.h" +#include "ep93xx-i2s.h" + +#define CODEC_CLOCK 5644800 + +static int snappercl15_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int err; + + err = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBS_CFS); + + err = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBS_CFS); + if (err) + return err; + + err = snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, + SND_SOC_CLOCK_IN); + if (err) + return err; + + err = snd_soc_dai_set_sysclk(cpu_dai, 0, CODEC_CLOCK, + SND_SOC_CLOCK_OUT); + if (err) + return err; + + return 0; +} + +static struct snd_soc_ops snappercl15_ops = { + .hw_params = snappercl15_hw_params, +}; + +static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + {"Headphone Jack", NULL, "LHPOUT"}, + {"Headphone Jack", NULL, "RHPOUT"}, + + {"LLINEIN", NULL, "Line In"}, + {"RLINEIN", NULL, "Line In"}, + + {"MICIN", NULL, "Mic Jack"}, +}; + +static int snappercl15_tlv320aic23_init(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets, + ARRAY_SIZE(tlv320aic23_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + return 0; +} + +static struct snd_soc_dai_link snappercl15_dai = { + .name = "tlv320aic23", + .stream_name = "AIC23", + .cpu_dai = &ep93xx_i2s_dai, + .codec_dai = &tlv320aic23_dai, + .init = snappercl15_tlv320aic23_init, + .ops = &snappercl15_ops, +}; + +static struct snd_soc_card snd_soc_snappercl15 = { + .name = "Snapper CL15", + .platform = &ep93xx_soc_platform, + .dai_link = &snappercl15_dai, + .num_links = 1, +}; + +static struct snd_soc_device snappercl15_snd_devdata = { + .card = &snd_soc_snappercl15, + .codec_dev = &soc_codec_dev_tlv320aic23, +}; + +static struct platform_device *snappercl15_snd_device; + +static int __init snappercl15_init(void) +{ + int ret; + + if (!machine_is_snapper_cl15()) + return -ENODEV; + + ret = ep93xx_i2s_acquire(EP93XX_SYSCON_DEVCFG_I2SONAC97, + EP93XX_SYSCON_I2SCLKDIV_ORIDE | + EP93XX_SYSCON_I2SCLKDIV_SPOL); + if (ret) + return ret; + + snappercl15_snd_device = platform_device_alloc("soc-audio", -1); + if (!snappercl15_snd_device) + return -ENOMEM; + + platform_set_drvdata(snappercl15_snd_device, &snappercl15_snd_devdata); + snappercl15_snd_devdata.dev = &snappercl15_snd_device->dev; + ret = platform_device_add(snappercl15_snd_device); + if (ret) + platform_device_put(snappercl15_snd_device); + + return ret; +} + +static void __exit snappercl15_exit(void) +{ + platform_device_unregister(snappercl15_snd_device); + ep93xx_i2s_release(); +} + +module_init(snappercl15_init); +module_exit(snappercl15_exit); + +MODULE_AUTHOR("Ryan Mallon <ryan@bluewatersys.com>"); +MODULE_DESCRIPTION("ALSA SoC Snapper CL15"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index 1d4e7164e80a..3dcd1469f283 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -369,7 +369,7 @@ struct snd_soc_platform mpc5200_audio_dma_platform = { }; EXPORT_SYMBOL_GPL(mpc5200_audio_dma_platform); -int mpc5200_audio_dma_create(struct of_device *op) +int mpc5200_audio_dma_create(struct platform_device *op) { phys_addr_t fifo; struct psc_dma *psc_dma; @@ -488,7 +488,7 @@ out_unmap: } EXPORT_SYMBOL_GPL(mpc5200_audio_dma_create); -int mpc5200_audio_dma_destroy(struct of_device *op) +int mpc5200_audio_dma_destroy(struct platform_device *op) { struct psc_dma *psc_dma = dev_get_drvdata(&op->dev); diff --git a/sound/soc/fsl/mpc5200_dma.h b/sound/soc/fsl/mpc5200_dma.h index e1ec6d91ea38..ca99586f2ad9 100644 --- a/sound/soc/fsl/mpc5200_dma.h +++ b/sound/soc/fsl/mpc5200_dma.h @@ -81,8 +81,8 @@ to_psc_dma_stream(struct snd_pcm_substream *substream, struct psc_dma *psc_dma) return &psc_dma->playback; } -int mpc5200_audio_dma_create(struct of_device *op); -int mpc5200_audio_dma_destroy(struct of_device *op); +int mpc5200_audio_dma_create(struct platform_device *op); +int mpc5200_audio_dma_destroy(struct platform_device *op); extern struct snd_soc_platform mpc5200_audio_dma_platform; diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c index e2ee220bfb7e..a9560235daee 100644 --- a/sound/soc/fsl/mpc5200_psc_ac97.c +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -20,6 +20,7 @@ #include <asm/time.h> #include <asm/delay.h> +#include <asm/mpc52xx.h> #include <asm/mpc52xx_psc.h> #include "mpc5200_dma.h" @@ -100,19 +101,32 @@ static void psc_ac97_warm_reset(struct snd_ac97 *ac97) { struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; + mutex_lock(&psc_dma->mutex); + out_be32(®s->sicr, psc_dma->sicr | MPC52xx_PSC_SICR_AWR); udelay(3); out_be32(®s->sicr, psc_dma->sicr); + + mutex_unlock(&psc_dma->mutex); } static void psc_ac97_cold_reset(struct snd_ac97 *ac97) { struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; - /* Do a cold reset */ - out_8(®s->op1, MPC52xx_PSC_OP_RES); - udelay(10); - out_8(®s->op0, MPC52xx_PSC_OP_RES); + mutex_lock(&psc_dma->mutex); + dev_dbg(psc_dma->dev, "cold reset\n"); + + mpc5200_psc_ac97_gpio_reset(psc_dma->id); + + /* Notify the PSC that a reset has occurred */ + out_be32(®s->sicr, psc_dma->sicr | MPC52xx_PSC_SICR_ACRB); + + /* Re-enable RX and TX */ + out_8(®s->command, MPC52xx_PSC_TX_ENABLE | MPC52xx_PSC_RX_ENABLE); + + mutex_unlock(&psc_dma->mutex); + msleep(1); psc_ac97_warm_reset(ac97); } @@ -263,7 +277,7 @@ EXPORT_SYMBOL_GPL(psc_ac97_dai); * - Probe/remove operations * - OF device match table */ -static int __devinit psc_ac97_of_probe(struct of_device *op, +static int __devinit psc_ac97_of_probe(struct platform_device *op, const struct of_device_id *match) { int rc, i; @@ -303,7 +317,7 @@ static int __devinit psc_ac97_of_probe(struct of_device *op, return 0; } -static int __devexit psc_ac97_of_remove(struct of_device *op) +static int __devexit psc_ac97_of_remove(struct platform_device *op) { return mpc5200_audio_dma_destroy(op); } diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c index 4f455bd6851f..534f04cb15d7 100644 --- a/sound/soc/fsl/mpc5200_psc_i2s.c +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -16,7 +16,6 @@ #include <asm/mpc52xx_psc.h> -#include "mpc5200_psc_i2s.h" #include "mpc5200_dma.h" /** @@ -153,7 +152,7 @@ EXPORT_SYMBOL_GPL(psc_i2s_dai); * - Probe/remove operations * - OF device match table */ -static int __devinit psc_i2s_of_probe(struct of_device *op, +static int __devinit psc_i2s_of_probe(struct platform_device *op, const struct of_device_id *match) { int rc; @@ -206,7 +205,7 @@ static int __devinit psc_i2s_of_probe(struct of_device *op, } -static int __devexit psc_i2s_of_remove(struct of_device *op) +static int __devexit psc_i2s_of_remove(struct platform_device *op) { return mpc5200_audio_dma_destroy(op); } diff --git a/sound/soc/fsl/mpc5200_psc_i2s.h b/sound/soc/fsl/mpc5200_psc_i2s.h deleted file mode 100644 index ce55e070fdf3..000000000000 --- a/sound/soc/fsl/mpc5200_psc_i2s.h +++ /dev/null @@ -1,12 +0,0 @@ -/* - * Freescale MPC5200 PSC in I2S mode - * ALSA SoC Digital Audio Interface (DAI) driver - * - */ - -#ifndef __SOUND_SOC_FSL_MPC52xx_PSC_I2S_H__ -#define __SOUND_SOC_FSL_MPC52xx_PSC_I2S_H__ - -extern struct snd_soc_dai psc_i2s_dai[]; - -#endif /* __SOUND_SOC_FSL_MPC52xx_PSC_I2S_H__ */ diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 6a2764ee8203..3b13b8d65262 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -46,7 +46,7 @@ struct mpc8610_hpcd_data { }; /** - * mpc8610_hpcd_machine_probe: initalize the board + * mpc8610_hpcd_machine_probe: initialize the board * * This function is called when platform_device_add() is called. It is used * to initialize the board-specific hardware. @@ -200,7 +200,7 @@ static struct snd_soc_ops mpc8610_hpcd_ops = { * SSI devices. We also probably aren't compatible with the generic Elo DMA * device driver. */ -static int mpc8610_hpcd_probe(struct of_device *ofdev, +static int mpc8610_hpcd_probe(struct platform_device *ofdev, const struct of_device_id *match) { struct device_node *np = ofdev->dev.of_node; @@ -534,7 +534,7 @@ error: * * This function is called when the OF device is removed. */ -static int mpc8610_hpcd_remove(struct of_device *ofdev) +static int mpc8610_hpcd_remove(struct platform_device *ofdev) { struct platform_device *sound_device = dev_get_drvdata(&ofdev->dev); struct mpc8610_hpcd_data *machine_data = diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index 252defea93b5..687c76fc0839 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -1,4 +1,4 @@ -config SND_IMX_SOC +menuconfig SND_IMX_SOC tristate "SoC Audio for Freescale i.MX CPUs" depends on ARCH_MXC select SND_PCM @@ -8,14 +8,12 @@ config SND_IMX_SOC Say Y or M if you want to add support for codecs attached to the i.MX SSI interface. -config SND_MXC_SOC_SSI - tristate +if SND_IMX_SOC config SND_MXC_SOC_WM1133_EV1 tristate "Audio on the the i.MX31ADS with WM1133-EV1 fitted" - depends on SND_IMX_SOC && MACH_MX31ADS_WM1133_EV1 && EXPERIMENTAL + depends on MACH_MX31ADS_WM1133_EV1 && EXPERIMENTAL select SND_SOC_WM8350 - select SND_MXC_SOC_SSI help Enable support for audio on the i.MX31ADS with the WM1133-EV1 PMIC board with WM8835x fitted. @@ -23,8 +21,19 @@ config SND_MXC_SOC_WM1133_EV1 config SND_SOC_PHYCORE_AC97 tristate "SoC Audio support for Phytec phyCORE (and phyCARD) boards" depends on MACH_PCM043 || MACH_PCA100 - select SND_MXC_SOC_SSI select SND_SOC_WM9712 help Say Y if you want to add support for SoC audio on Phytec phyCORE and phyCARD boards in AC97 mode + +config SND_SOC_EUKREA_TLV320 + tristate "Eukrea TLV320" + depends on MACH_EUKREA_MBIMX27_BASEBOARD \ + || MACH_EUKREA_MBIMXSD25_BASEBOARD \ + || MACH_EUKREA_MBIMXSD35_BASEBOARD + select SND_SOC_TLV320AIC23 + help + Enable I2S based access to the TLV320AIC23B codec attached + to the SSI interface + +endif # SND_IMX_SOC diff --git a/sound/soc/imx/Makefile b/sound/soc/imx/Makefile index 2d203635ac11..7bc57baf2b0e 100644 --- a/sound/soc/imx/Makefile +++ b/sound/soc/imx/Makefile @@ -8,8 +8,10 @@ endif obj-$(CONFIG_SND_IMX_SOC) += snd-soc-imx.o # i.MX Machine Support +snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o snd-soc-phycore-ac97-objs := phycore-ac97.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o +obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o diff --git a/sound/soc/imx/eukrea-tlv320.c b/sound/soc/imx/eukrea-tlv320.c new file mode 100644 index 000000000000..f15dfbdc47ee --- /dev/null +++ b/sound/soc/imx/eukrea-tlv320.c @@ -0,0 +1,137 @@ +/* + * eukrea-tlv320.c -- SoC audio for eukrea_cpuimxXX in I2S mode + * + * Copyright 2010 Eric Bénard, Eukréa Electromatique <eric@eukrea.com> + * + * based on sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c + * which is Copyright 2009 Simtec Electronics + * and on sound/soc/imx/phycore-ac97.c which is + * Copyright 2009 Sascha Hauer, Pengutronix <s.hauer@pengutronix.de> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/device.h> +#include <linux/i2c.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <asm/mach-types.h> + +#include "../codecs/tlv320aic23.h" +#include "imx-ssi.h" + +#define CODEC_CLOCK 12000000 + +static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret) { + pr_err("%s: failed set cpu dai format\n", __func__); + return ret; + } + + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret) { + pr_err("%s: failed set codec dai format\n", __func__); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, + CODEC_CLOCK, SND_SOC_CLOCK_OUT); + if (ret) { + pr_err("%s: failed setting codec sysclk\n", __func__); + return ret; + } + snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0); + + ret = snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0, + SND_SOC_CLOCK_IN); + if (ret) { + pr_err("can't set CPU system clock IMX_SSP_SYS_CLK\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops eukrea_tlv320_snd_ops = { + .hw_params = eukrea_tlv320_hw_params, +}; + +static struct snd_soc_dai_link eukrea_tlv320_dai = { + .name = "tlv320aic23", + .stream_name = "TLV320AIC23", + .codec_dai = &tlv320aic23_dai, + .ops = &eukrea_tlv320_snd_ops, +}; + +static struct snd_soc_card eukrea_tlv320 = { + .name = "cpuimx-audio", + .platform = &imx_soc_platform, + .dai_link = &eukrea_tlv320_dai, + .num_links = 1, +}; + +static struct snd_soc_device eukrea_tlv320_snd_devdata = { + .card = &eukrea_tlv320, + .codec_dev = &soc_codec_dev_tlv320aic23, +}; + +static struct platform_device *eukrea_tlv320_snd_device; + +static int __init eukrea_tlv320_init(void) +{ + int ret; + + if (!machine_is_eukrea_cpuimx27() && !machine_is_eukrea_cpuimx25sd() + && !machine_is_eukrea_cpuimx35sd()) + /* return happy. We might run on a totally different machine */ + return 0; + + eukrea_tlv320_snd_device = platform_device_alloc("soc-audio", -1); + if (!eukrea_tlv320_snd_device) + return -ENOMEM; + + eukrea_tlv320_dai.cpu_dai = &imx_ssi_pcm_dai[0]; + + platform_set_drvdata(eukrea_tlv320_snd_device, &eukrea_tlv320_snd_devdata); + eukrea_tlv320_snd_devdata.dev = &eukrea_tlv320_snd_device->dev; + ret = platform_device_add(eukrea_tlv320_snd_device); + + if (ret) { + printk(KERN_ERR "ASoC: Platform device allocation failed\n"); + platform_device_put(eukrea_tlv320_snd_device); + } + + return ret; +} + +static void __exit eukrea_tlv320_exit(void) +{ + platform_device_unregister(eukrea_tlv320_snd_device); +} + +module_init(eukrea_tlv320_init); +module_exit(eukrea_tlv320_exit); + +MODULE_AUTHOR("Eric Bénard <eric@eukrea.com>"); +MODULE_DESCRIPTION("CPUIMX ALSA SoC driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c index 05f19c9284f4..0a595da4811d 100644 --- a/sound/soc/imx/imx-pcm-dma-mx2.c +++ b/sound/soc/imx/imx-pcm-dma-mx2.c @@ -292,12 +292,16 @@ static int snd_imx_open(struct snd_pcm_substream *substream) int ret; iprtd = kzalloc(sizeof(*iprtd), GFP_KERNEL); + if (iprtd == NULL) + return -ENOMEM; runtime->private_data = iprtd; ret = snd_pcm_hw_constraint_integer(substream->runtime, SNDRV_PCM_HW_PARAM_PERIODS); - if (ret < 0) + if (ret < 0) { + kfree(iprtd); return ret; + } snd_soc_set_runtime_hwparams(substream, &snd_imx_hardware); return 0; diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c index 6b518e07eea9..b2bf27282cd2 100644 --- a/sound/soc/imx/imx-pcm-fiq.c +++ b/sound/soc/imx/imx-pcm-fiq.c @@ -192,6 +192,8 @@ static int snd_imx_open(struct snd_pcm_substream *substream) int ret; iprtd = kzalloc(sizeof(*iprtd), GFP_KERNEL); + if (iprtd == NULL) + return -ENOMEM; runtime->private_data = iprtd; iprtd->substream = substream; @@ -202,8 +204,10 @@ static int snd_imx_open(struct snd_pcm_substream *substream) ret = snd_pcm_hw_constraint_integer(substream->runtime, SNDRV_PCM_HW_PARAM_PERIODS); - if (ret < 0) + if (ret < 0) { + kfree(iprtd); return ret; + } snd_soc_set_runtime_hwparams(substream, &snd_imx_hardware); return 0; diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index 80b4fee2442b..c81da05a4f11 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -23,7 +23,7 @@ * between pcm data and GPIO status data changes. Our FIQ handler is not * able to handle this, hence this driver only works with 48000Hz sampling * rate. - * Reading and writing AC97 registers is another challange. The core + * Reading and writing AC97 registers is another challenge. The core * provides us status bits when the read register is updated with *another* * value. When we read the same register two times (and the register still * contains the same value) these status bits are not set. We work @@ -83,8 +83,6 @@ static int imx_ssi_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, /* * SSI DAI format configuration. * Should only be called when port is inactive (i.e. SSIEN = 0). - * Note: We don't use the I2S modes but instead manually configure the - * SSI for I2S because the I2S mode is only a register preset. */ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { @@ -99,6 +97,10 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) /* data on rising edge of bclk, frame low 1clk before data */ strcr |= SSI_STCR_TFSI | SSI_STCR_TEFS | SSI_STCR_TXBIT0; scr |= SSI_SCR_NET; + if (ssi->flags & IMX_SSI_USE_I2S_SLAVE) { + scr &= ~SSI_I2S_MODE_MASK; + scr |= SSI_SCR_I2S_MODE_SLAVE; + } break; case SND_SOC_DAIFMT_LEFT_J: /* data on rising edge of bclk, frame high with data */ @@ -143,6 +145,11 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) strcr |= SSI_STCR_TFEN0; + if (ssi->flags & IMX_SSI_NET) + scr |= SSI_SCR_NET; + if (ssi->flags & IMX_SSI_SYN) + scr |= SSI_SCR_SYN; + writel(strcr, ssi->base + SSI_STCR); writel(strcr, ssi->base + SSI_SRCR); writel(scr, ssi->base + SSI_SCR); @@ -247,6 +254,9 @@ static int imx_ssi_hw_params(struct snd_pcm_substream *substream, dma_data = &ssi->dma_params_rx; } + if (ssi->flags & IMX_SSI_SYN) + reg = SSI_STCCR; + snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); sccr = readl(ssi->base + reg) & ~SSI_STCCR_WL_MASK; diff --git a/sound/soc/jz4740/Kconfig b/sound/soc/jz4740/Kconfig new file mode 100644 index 000000000000..5351cba66c9e --- /dev/null +++ b/sound/soc/jz4740/Kconfig @@ -0,0 +1,23 @@ +config SND_JZ4740_SOC + tristate "SoC Audio for Ingenic JZ4740 SoC" + depends on MACH_JZ4740 && SND_SOC + help + Say Y or M if you want to add support for codecs attached to + the JZ4740 I2S interface. You will also need to select the audio + interfaces to support below. + +config SND_JZ4740_SOC_I2S + depends on SND_JZ4740_SOC + tristate "SoC Audio (I2S protocol) for Ingenic JZ4740 SoC" + help + Say Y if you want to use I2S protocol and I2S codec on Ingenic JZ4740 + based boards. + +config SND_JZ4740_SOC_QI_LB60 + tristate "SoC Audio support for Qi LB60" + depends on SND_JZ4740_SOC && JZ4740_QI_LB60 + select SND_JZ4740_SOC_I2S + select SND_SOC_JZ4740_CODEC + help + Say Y if you want to add support for ASoC audio on the Qi LB60 board + a.k.a Qi Ben NanoNote. diff --git a/sound/soc/jz4740/Makefile b/sound/soc/jz4740/Makefile new file mode 100644 index 000000000000..be873c1b0c20 --- /dev/null +++ b/sound/soc/jz4740/Makefile @@ -0,0 +1,13 @@ +# +# Jz4740 Platform Support +# +snd-soc-jz4740-objs := jz4740-pcm.o +snd-soc-jz4740-i2s-objs := jz4740-i2s.o + +obj-$(CONFIG_SND_JZ4740_SOC) += snd-soc-jz4740.o +obj-$(CONFIG_SND_JZ4740_SOC_I2S) += snd-soc-jz4740-i2s.o + +# Jz4740 Machine Support +snd-soc-qi-lb60-objs := qi_lb60.o + +obj-$(CONFIG_SND_JZ4740_SOC_QI_LB60) += snd-soc-qi-lb60.o diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c new file mode 100644 index 000000000000..eb518f0c5e01 --- /dev/null +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -0,0 +1,540 @@ +/* + * Copyright (C) 2010, Lars-Peter Clausen <lars@metafoo.de> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 675 Mass Ave, Cambridge, MA 02139, USA. + * + */ + +#include <linux/init.h> +#include <linux/io.h> +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> + +#include <linux/clk.h> +#include <linux/delay.h> + +#include <linux/dma-mapping.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include "jz4740-i2s.h" +#include "jz4740-pcm.h" + +#define JZ_REG_AIC_CONF 0x00 +#define JZ_REG_AIC_CTRL 0x04 +#define JZ_REG_AIC_I2S_FMT 0x10 +#define JZ_REG_AIC_FIFO_STATUS 0x14 +#define JZ_REG_AIC_I2S_STATUS 0x1c +#define JZ_REG_AIC_CLK_DIV 0x30 +#define JZ_REG_AIC_FIFO 0x34 + +#define JZ_AIC_CONF_FIFO_RX_THRESHOLD_MASK (0xf << 12) +#define JZ_AIC_CONF_FIFO_TX_THRESHOLD_MASK (0xf << 8) +#define JZ_AIC_CONF_OVERFLOW_PLAY_LAST BIT(6) +#define JZ_AIC_CONF_INTERNAL_CODEC BIT(5) +#define JZ_AIC_CONF_I2S BIT(4) +#define JZ_AIC_CONF_RESET BIT(3) +#define JZ_AIC_CONF_BIT_CLK_MASTER BIT(2) +#define JZ_AIC_CONF_SYNC_CLK_MASTER BIT(1) +#define JZ_AIC_CONF_ENABLE BIT(0) + +#define JZ_AIC_CONF_FIFO_RX_THRESHOLD_OFFSET 12 +#define JZ_AIC_CONF_FIFO_TX_THRESHOLD_OFFSET 8 + +#define JZ_AIC_CTRL_OUTPUT_SAMPLE_SIZE_MASK (0x7 << 19) +#define JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_MASK (0x7 << 16) +#define JZ_AIC_CTRL_ENABLE_RX_DMA BIT(15) +#define JZ_AIC_CTRL_ENABLE_TX_DMA BIT(14) +#define JZ_AIC_CTRL_MONO_TO_STEREO BIT(11) +#define JZ_AIC_CTRL_SWITCH_ENDIANNESS BIT(10) +#define JZ_AIC_CTRL_SIGNED_TO_UNSIGNED BIT(9) +#define JZ_AIC_CTRL_FLUSH BIT(8) +#define JZ_AIC_CTRL_ENABLE_ROR_INT BIT(6) +#define JZ_AIC_CTRL_ENABLE_TUR_INT BIT(5) +#define JZ_AIC_CTRL_ENABLE_RFS_INT BIT(4) +#define JZ_AIC_CTRL_ENABLE_TFS_INT BIT(3) +#define JZ_AIC_CTRL_ENABLE_LOOPBACK BIT(2) +#define JZ_AIC_CTRL_ENABLE_PLAYBACK BIT(1) +#define JZ_AIC_CTRL_ENABLE_CAPTURE BIT(0) + +#define JZ_AIC_CTRL_OUTPUT_SAMPLE_SIZE_OFFSET 19 +#define JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_OFFSET 16 + +#define JZ_AIC_I2S_FMT_DISABLE_BIT_CLK BIT(12) +#define JZ_AIC_I2S_FMT_ENABLE_SYS_CLK BIT(4) +#define JZ_AIC_I2S_FMT_MSB BIT(0) + +#define JZ_AIC_I2S_STATUS_BUSY BIT(2) + +#define JZ_AIC_CLK_DIV_MASK 0xf + +struct jz4740_i2s { + struct resource *mem; + void __iomem *base; + dma_addr_t phys_base; + + struct clk *clk_aic; + struct clk *clk_i2s; + + struct jz4740_pcm_config pcm_config_playback; + struct jz4740_pcm_config pcm_config_capture; +}; + +static inline uint32_t jz4740_i2s_read(const struct jz4740_i2s *i2s, + unsigned int reg) +{ + return readl(i2s->base + reg); +} + +static inline void jz4740_i2s_write(const struct jz4740_i2s *i2s, + unsigned int reg, uint32_t value) +{ + writel(value, i2s->base + reg); +} + +static inline struct jz4740_i2s *jz4740_dai_to_i2s(struct snd_soc_dai *dai) +{ + return dai->private_data; +} + +static int jz4740_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct jz4740_i2s *i2s = jz4740_dai_to_i2s(dai); + uint32_t conf, ctrl; + + if (dai->active) + return 0; + + ctrl = jz4740_i2s_read(i2s, JZ_REG_AIC_CTRL); + ctrl |= JZ_AIC_CTRL_FLUSH; + jz4740_i2s_write(i2s, JZ_REG_AIC_CTRL, ctrl); + + clk_enable(i2s->clk_i2s); + + conf = jz4740_i2s_read(i2s, JZ_REG_AIC_CONF); + conf |= JZ_AIC_CONF_ENABLE; + jz4740_i2s_write(i2s, JZ_REG_AIC_CONF, conf); + + return 0; +} + +static void jz4740_i2s_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct jz4740_i2s *i2s = jz4740_dai_to_i2s(dai); + uint32_t conf; + + if (!dai->active) + return; + + conf = jz4740_i2s_read(i2s, JZ_REG_AIC_CONF); + conf &= ~JZ_AIC_CONF_ENABLE; + jz4740_i2s_write(i2s, JZ_REG_AIC_CONF, conf); + + clk_disable(i2s->clk_i2s); +} + +static int jz4740_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct jz4740_i2s *i2s = jz4740_dai_to_i2s(dai); + + uint32_t ctrl; + uint32_t mask; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + mask = JZ_AIC_CTRL_ENABLE_PLAYBACK | JZ_AIC_CTRL_ENABLE_TX_DMA; + else + mask = JZ_AIC_CTRL_ENABLE_CAPTURE | JZ_AIC_CTRL_ENABLE_RX_DMA; + + ctrl = jz4740_i2s_read(i2s, JZ_REG_AIC_CTRL); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ctrl |= mask; + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ctrl &= ~mask; + break; + default: + return -EINVAL; + } + + jz4740_i2s_write(i2s, JZ_REG_AIC_CTRL, ctrl); + + return 0; +} + +static int jz4740_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct jz4740_i2s *i2s = jz4740_dai_to_i2s(dai); + + uint32_t format = 0; + uint32_t conf; + + conf = jz4740_i2s_read(i2s, JZ_REG_AIC_CONF); + + conf &= ~(JZ_AIC_CONF_BIT_CLK_MASTER | JZ_AIC_CONF_SYNC_CLK_MASTER); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + conf |= JZ_AIC_CONF_BIT_CLK_MASTER | JZ_AIC_CONF_SYNC_CLK_MASTER; + format |= JZ_AIC_I2S_FMT_ENABLE_SYS_CLK; + break; + case SND_SOC_DAIFMT_CBM_CFS: + conf |= JZ_AIC_CONF_SYNC_CLK_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFM: + conf |= JZ_AIC_CONF_BIT_CLK_MASTER; + break; + case SND_SOC_DAIFMT_CBM_CFM: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_MSB: + format |= JZ_AIC_I2S_FMT_MSB; + break; + case SND_SOC_DAIFMT_I2S: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + jz4740_i2s_write(i2s, JZ_REG_AIC_CONF, conf); + jz4740_i2s_write(i2s, JZ_REG_AIC_I2S_FMT, format); + + return 0; +} + +static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct jz4740_i2s *i2s = jz4740_dai_to_i2s(dai); + enum jz4740_dma_width dma_width; + struct jz4740_pcm_config *pcm_config; + unsigned int sample_size; + uint32_t ctrl; + + ctrl = jz4740_i2s_read(i2s, JZ_REG_AIC_CTRL); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + sample_size = 0; + dma_width = JZ4740_DMA_WIDTH_8BIT; + break; + case SNDRV_PCM_FORMAT_S16: + sample_size = 1; + dma_width = JZ4740_DMA_WIDTH_16BIT; + break; + default: + return -EINVAL; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + ctrl &= ~JZ_AIC_CTRL_OUTPUT_SAMPLE_SIZE_MASK; + ctrl |= sample_size << JZ_AIC_CTRL_OUTPUT_SAMPLE_SIZE_OFFSET; + if (params_channels(params) == 1) + ctrl |= JZ_AIC_CTRL_MONO_TO_STEREO; + else + ctrl &= ~JZ_AIC_CTRL_MONO_TO_STEREO; + + pcm_config = &i2s->pcm_config_playback; + pcm_config->dma_config.dst_width = dma_width; + + } else { + ctrl &= ~JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_MASK; + ctrl |= sample_size << JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_OFFSET; + + pcm_config = &i2s->pcm_config_capture; + pcm_config->dma_config.src_width = dma_width; + } + + jz4740_i2s_write(i2s, JZ_REG_AIC_CTRL, ctrl); + + snd_soc_dai_set_dma_data(dai, substream, pcm_config); + + return 0; +} + +static int jz4740_i2s_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct jz4740_i2s *i2s = jz4740_dai_to_i2s(dai); + struct clk *parent; + int ret = 0; + + switch (clk_id) { + case JZ4740_I2S_CLKSRC_EXT: + parent = clk_get(NULL, "ext"); + clk_set_parent(i2s->clk_i2s, parent); + break; + case JZ4740_I2S_CLKSRC_PLL: + parent = clk_get(NULL, "pll half"); + clk_set_parent(i2s->clk_i2s, parent); + ret = clk_set_rate(i2s->clk_i2s, freq); + break; + default: + return -EINVAL; + } + clk_put(parent); + + return ret; +} + +static int jz4740_i2s_suspend(struct snd_soc_dai *dai) +{ + struct jz4740_i2s *i2s = jz4740_dai_to_i2s(dai); + uint32_t conf; + + if (dai->active) { + conf = jz4740_i2s_read(i2s, JZ_REG_AIC_CONF); + conf &= ~JZ_AIC_CONF_ENABLE; + jz4740_i2s_write(i2s, JZ_REG_AIC_CONF, conf); + + clk_disable(i2s->clk_i2s); + } + + clk_disable(i2s->clk_aic); + + return 0; +} + +static int jz4740_i2s_resume(struct snd_soc_dai *dai) +{ + struct jz4740_i2s *i2s = jz4740_dai_to_i2s(dai); + uint32_t conf; + + clk_enable(i2s->clk_aic); + + if (dai->active) { + clk_enable(i2s->clk_i2s); + + conf = jz4740_i2s_read(i2s, JZ_REG_AIC_CONF); + conf |= JZ_AIC_CONF_ENABLE; + jz4740_i2s_write(i2s, JZ_REG_AIC_CONF, conf); + } + + return 0; +} + +static int jz4740_i2s_probe(struct platform_device *pdev, struct snd_soc_dai *dai) +{ + struct jz4740_i2s *i2s = jz4740_dai_to_i2s(dai); + uint32_t conf; + + conf = (7 << JZ_AIC_CONF_FIFO_RX_THRESHOLD_OFFSET) | + (8 << JZ_AIC_CONF_FIFO_TX_THRESHOLD_OFFSET) | + JZ_AIC_CONF_OVERFLOW_PLAY_LAST | + JZ_AIC_CONF_I2S | + JZ_AIC_CONF_INTERNAL_CODEC; + + jz4740_i2s_write(i2s, JZ_REG_AIC_CONF, JZ_AIC_CONF_RESET); + jz4740_i2s_write(i2s, JZ_REG_AIC_CONF, conf); + + return 0; +} + +static struct snd_soc_dai_ops jz4740_i2s_dai_ops = { + .startup = jz4740_i2s_startup, + .shutdown = jz4740_i2s_shutdown, + .trigger = jz4740_i2s_trigger, + .hw_params = jz4740_i2s_hw_params, + .set_fmt = jz4740_i2s_set_fmt, + .set_sysclk = jz4740_i2s_set_sysclk, +}; + +#define JZ4740_I2S_FMTS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE) + +struct snd_soc_dai jz4740_i2s_dai = { + .name = "jz4740-i2s", + .probe = jz4740_i2s_probe, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = JZ4740_I2S_FMTS, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = JZ4740_I2S_FMTS, + }, + .symmetric_rates = 1, + .ops = &jz4740_i2s_dai_ops, + .suspend = jz4740_i2s_suspend, + .resume = jz4740_i2s_resume, +}; +EXPORT_SYMBOL_GPL(jz4740_i2s_dai); + +static void __devinit jz4740_i2c_init_pcm_config(struct jz4740_i2s *i2s) +{ + struct jz4740_dma_config *dma_config; + + /* Playback */ + dma_config = &i2s->pcm_config_playback.dma_config; + dma_config->src_width = JZ4740_DMA_WIDTH_32BIT, + dma_config->transfer_size = JZ4740_DMA_TRANSFER_SIZE_16BYTE; + dma_config->request_type = JZ4740_DMA_TYPE_AIC_TRANSMIT; + dma_config->flags = JZ4740_DMA_SRC_AUTOINC; + dma_config->mode = JZ4740_DMA_MODE_SINGLE; + i2s->pcm_config_playback.fifo_addr = i2s->phys_base + JZ_REG_AIC_FIFO; + + /* Capture */ + dma_config = &i2s->pcm_config_capture.dma_config; + dma_config->dst_width = JZ4740_DMA_WIDTH_32BIT, + dma_config->transfer_size = JZ4740_DMA_TRANSFER_SIZE_16BYTE; + dma_config->request_type = JZ4740_DMA_TYPE_AIC_RECEIVE; + dma_config->flags = JZ4740_DMA_DST_AUTOINC; + dma_config->mode = JZ4740_DMA_MODE_SINGLE; + i2s->pcm_config_capture.fifo_addr = i2s->phys_base + JZ_REG_AIC_FIFO; +} + +static int __devinit jz4740_i2s_dev_probe(struct platform_device *pdev) +{ + struct jz4740_i2s *i2s; + int ret; + + i2s = kzalloc(sizeof(*i2s), GFP_KERNEL); + + if (!i2s) + return -ENOMEM; + + i2s->mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!i2s->mem) { + ret = -ENOENT; + goto err_free; + } + + i2s->mem = request_mem_region(i2s->mem->start, resource_size(i2s->mem), + pdev->name); + if (!i2s->mem) { + ret = -EBUSY; + goto err_free; + } + + i2s->base = ioremap_nocache(i2s->mem->start, resource_size(i2s->mem)); + if (!i2s->base) { + ret = -EBUSY; + goto err_release_mem_region; + } + + i2s->phys_base = i2s->mem->start; + + i2s->clk_aic = clk_get(&pdev->dev, "aic"); + if (IS_ERR(i2s->clk_aic)) { + ret = PTR_ERR(i2s->clk_aic); + goto err_iounmap; + } + + i2s->clk_i2s = clk_get(&pdev->dev, "i2s"); + if (IS_ERR(i2s->clk_i2s)) { + ret = PTR_ERR(i2s->clk_i2s); + goto err_clk_put_aic; + } + + clk_enable(i2s->clk_aic); + + jz4740_i2c_init_pcm_config(i2s); + + jz4740_i2s_dai.private_data = i2s; + ret = snd_soc_register_dai(&jz4740_i2s_dai); + + if (ret) { + dev_err(&pdev->dev, "Failed to register DAI\n"); + goto err_clk_put_i2s; + } + + platform_set_drvdata(pdev, i2s); + + return 0; + +err_clk_put_i2s: + clk_disable(i2s->clk_aic); + clk_put(i2s->clk_i2s); +err_clk_put_aic: + clk_put(i2s->clk_aic); +err_iounmap: + iounmap(i2s->base); +err_release_mem_region: + release_mem_region(i2s->mem->start, resource_size(i2s->mem)); +err_free: + kfree(i2s); + + return ret; +} + +static int __devexit jz4740_i2s_dev_remove(struct platform_device *pdev) +{ + struct jz4740_i2s *i2s = platform_get_drvdata(pdev); + + snd_soc_unregister_dai(&jz4740_i2s_dai); + + clk_disable(i2s->clk_aic); + clk_put(i2s->clk_i2s); + clk_put(i2s->clk_aic); + + iounmap(i2s->base); + release_mem_region(i2s->mem->start, resource_size(i2s->mem)); + + platform_set_drvdata(pdev, NULL); + kfree(i2s); + + return 0; +} + +static struct platform_driver jz4740_i2s_driver = { + .probe = jz4740_i2s_dev_probe, + .remove = __devexit_p(jz4740_i2s_dev_remove), + .driver = { + .name = "jz4740-i2s", + .owner = THIS_MODULE, + }, +}; + +static int __init jz4740_i2s_init(void) +{ + return platform_driver_register(&jz4740_i2s_driver); +} +module_init(jz4740_i2s_init); + +static void __exit jz4740_i2s_exit(void) +{ + platform_driver_unregister(&jz4740_i2s_driver); +} +module_exit(jz4740_i2s_exit); + +MODULE_AUTHOR("Lars-Peter Clausen, <lars@metafoo.de>"); +MODULE_DESCRIPTION("Ingenic JZ4740 SoC I2S driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:jz4740-i2s"); diff --git a/sound/soc/jz4740/jz4740-i2s.h b/sound/soc/jz4740/jz4740-i2s.h new file mode 100644 index 000000000000..da22ed88a589 --- /dev/null +++ b/sound/soc/jz4740/jz4740-i2s.h @@ -0,0 +1,18 @@ +/* + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _JZ4740_I2S_H +#define _JZ4740_I2S_H + +/* I2S clock source */ +#define JZ4740_I2S_CLKSRC_EXT 0 +#define JZ4740_I2S_CLKSRC_PLL 1 + +#define JZ4740_I2S_BIT_CLK 0 + +extern struct snd_soc_dai jz4740_i2s_dai; + +#endif diff --git a/sound/soc/jz4740/jz4740-pcm.c b/sound/soc/jz4740/jz4740-pcm.c new file mode 100644 index 000000000000..ee68d850c8dd --- /dev/null +++ b/sound/soc/jz4740/jz4740-pcm.c @@ -0,0 +1,373 @@ +/* + * Copyright (C) 2010, Lars-Peter Clausen <lars@metafoo.de> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 675 Mass Ave, Cambridge, MA 02139, USA. + * + */ + +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> + +#include <linux/dma-mapping.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <asm/mach-jz4740/dma.h> +#include "jz4740-pcm.h" + +struct jz4740_runtime_data { + unsigned long dma_period; + dma_addr_t dma_start; + dma_addr_t dma_pos; + dma_addr_t dma_end; + + struct jz4740_dma_chan *dma; + + dma_addr_t fifo_addr; +}; + +/* identify hardware playback capabilities */ +static const struct snd_pcm_hardware jz4740_pcm_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8, + + .rates = SNDRV_PCM_RATE_8000_48000, + .channels_min = 1, + .channels_max = 2, + .period_bytes_min = 16, + .period_bytes_max = 2 * PAGE_SIZE, + .periods_min = 2, + .periods_max = 128, + .buffer_bytes_max = 128 * 2 * PAGE_SIZE, + .fifo_size = 32, +}; + +static void jz4740_pcm_start_transfer(struct jz4740_runtime_data *prtd, + struct snd_pcm_substream *substream) +{ + unsigned long count; + + if (prtd->dma_pos == prtd->dma_end) + prtd->dma_pos = prtd->dma_start; + + if (prtd->dma_pos + prtd->dma_period > prtd->dma_end) + count = prtd->dma_end - prtd->dma_pos; + else + count = prtd->dma_period; + + jz4740_dma_disable(prtd->dma); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + jz4740_dma_set_src_addr(prtd->dma, prtd->dma_pos); + jz4740_dma_set_dst_addr(prtd->dma, prtd->fifo_addr); + } else { + jz4740_dma_set_src_addr(prtd->dma, prtd->fifo_addr); + jz4740_dma_set_dst_addr(prtd->dma, prtd->dma_pos); + } + + jz4740_dma_set_transfer_count(prtd->dma, count); + + prtd->dma_pos += count; + + jz4740_dma_enable(prtd->dma); +} + +static void jz4740_pcm_dma_transfer_done(struct jz4740_dma_chan *dma, int err, + void *dev_id) +{ + struct snd_pcm_substream *substream = dev_id; + struct snd_pcm_runtime *runtime = substream->runtime; + struct jz4740_runtime_data *prtd = runtime->private_data; + + snd_pcm_period_elapsed(substream); + + jz4740_pcm_start_transfer(prtd, substream); +} + +static int jz4740_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct jz4740_runtime_data *prtd = runtime->private_data; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct jz4740_pcm_config *config; + + config = snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); + + if (!config) + return 0; + + if (!prtd->dma) { + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + prtd->dma = jz4740_dma_request(substream, "PCM Capture"); + else + prtd->dma = jz4740_dma_request(substream, "PCM Playback"); + } + + if (!prtd->dma) + return -EBUSY; + + jz4740_dma_configure(prtd->dma, &config->dma_config); + prtd->fifo_addr = config->fifo_addr; + + jz4740_dma_set_complete_cb(prtd->dma, jz4740_pcm_dma_transfer_done); + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = params_buffer_bytes(params); + + prtd->dma_period = params_period_bytes(params); + prtd->dma_start = runtime->dma_addr; + prtd->dma_pos = prtd->dma_start; + prtd->dma_end = prtd->dma_start + runtime->dma_bytes; + + return 0; +} + +static int jz4740_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct jz4740_runtime_data *prtd = substream->runtime->private_data; + + snd_pcm_set_runtime_buffer(substream, NULL); + if (prtd->dma) { + jz4740_dma_free(prtd->dma); + prtd->dma = NULL; + } + + return 0; +} + +static int jz4740_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct jz4740_runtime_data *prtd = substream->runtime->private_data; + + if (!prtd->dma) + return -EBUSY; + + prtd->dma_pos = prtd->dma_start; + + return 0; +} + +static int jz4740_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct jz4740_runtime_data *prtd = runtime->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + jz4740_pcm_start_transfer(prtd, substream); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + jz4740_dma_disable(prtd->dma); + break; + default: + break; + } + + return 0; +} + +static snd_pcm_uframes_t jz4740_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct jz4740_runtime_data *prtd = runtime->private_data; + unsigned long byte_offset; + snd_pcm_uframes_t offset; + struct jz4740_dma_chan *dma = prtd->dma; + + /* prtd->dma_pos points to the end of the current transfer. So by + * subtracting prdt->dma_start we get the offset to the end of the + * current period in bytes. By subtracting the residue of the transfer + * we get the current offset in bytes. */ + byte_offset = prtd->dma_pos - prtd->dma_start; + byte_offset -= jz4740_dma_get_residue(dma); + + offset = bytes_to_frames(runtime, byte_offset); + if (offset >= runtime->buffer_size) + offset = 0; + + return offset; +} + +static int jz4740_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct jz4740_runtime_data *prtd; + + prtd = kzalloc(sizeof(*prtd), GFP_KERNEL); + if (prtd == NULL) + return -ENOMEM; + + snd_soc_set_runtime_hwparams(substream, &jz4740_pcm_hardware); + + runtime->private_data = prtd; + + return 0; +} + +static int jz4740_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct jz4740_runtime_data *prtd = runtime->private_data; + + kfree(prtd); + + return 0; +} + +static int jz4740_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + return remap_pfn_range(vma, vma->vm_start, + substream->dma_buffer.addr >> PAGE_SHIFT, + vma->vm_end - vma->vm_start, vma->vm_page_prot); +} + +static struct snd_pcm_ops jz4740_pcm_ops = { + .open = jz4740_pcm_open, + .close = jz4740_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = jz4740_pcm_hw_params, + .hw_free = jz4740_pcm_hw_free, + .prepare = jz4740_pcm_prepare, + .trigger = jz4740_pcm_trigger, + .pointer = jz4740_pcm_pointer, + .mmap = jz4740_pcm_mmap, +}; + +static int jz4740_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = jz4740_pcm_hardware.buffer_bytes_max; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + + buf->area = dma_alloc_noncoherent(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + if (!buf->area) + return -ENOMEM; + + buf->bytes = size; + + return 0; +} + +static void jz4740_pcm_free(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < SNDRV_PCM_STREAM_LAST; ++stream) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + + dma_free_noncoherent(pcm->card->dev, buf->bytes, buf->area, + buf->addr); + buf->area = NULL; + } +} + +static u64 jz4740_pcm_dmamask = DMA_BIT_MASK(32); + +int jz4740_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, + struct snd_pcm *pcm) +{ + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &jz4740_pcm_dmamask; + + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); + + if (dai->playback.channels_min) { + ret = jz4740_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto err; + } + + if (dai->capture.channels_min) { + ret = jz4740_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto err; + } + +err: + return ret; +} + +struct snd_soc_platform jz4740_soc_platform = { + .name = "jz4740-pcm", + .pcm_ops = &jz4740_pcm_ops, + .pcm_new = jz4740_pcm_new, + .pcm_free = jz4740_pcm_free, +}; +EXPORT_SYMBOL_GPL(jz4740_soc_platform); + +static int __devinit jz4740_pcm_probe(struct platform_device *pdev) +{ + return snd_soc_register_platform(&jz4740_soc_platform); +} + +static int __devexit jz4740_pcm_remove(struct platform_device *pdev) +{ + snd_soc_unregister_platform(&jz4740_soc_platform); + return 0; +} + +static struct platform_driver jz4740_pcm_driver = { + .probe = jz4740_pcm_probe, + .remove = __devexit_p(jz4740_pcm_remove), + .driver = { + .name = "jz4740-pcm", + .owner = THIS_MODULE, + }, +}; + +static int __init jz4740_soc_platform_init(void) +{ + return platform_driver_register(&jz4740_pcm_driver); +} +module_init(jz4740_soc_platform_init); + +static void __exit jz4740_soc_platform_exit(void) +{ + return platform_driver_unregister(&jz4740_pcm_driver); +} +module_exit(jz4740_soc_platform_exit); + +MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>"); +MODULE_DESCRIPTION("Ingenic SoC JZ4740 PCM driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/jz4740/jz4740-pcm.h b/sound/soc/jz4740/jz4740-pcm.h new file mode 100644 index 000000000000..e3f221e2779c --- /dev/null +++ b/sound/soc/jz4740/jz4740-pcm.h @@ -0,0 +1,22 @@ +/* + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _JZ4740_PCM_H +#define _JZ4740_PCM_H + +#include <linux/dma-mapping.h> +#include <asm/mach-jz4740/dma.h> + +/* platform data */ +extern struct snd_soc_platform jz4740_soc_platform; + +struct jz4740_pcm_config { + struct jz4740_dma_config dma_config; + phys_addr_t fifo_addr; +}; + +#endif diff --git a/sound/soc/jz4740/qi_lb60.c b/sound/soc/jz4740/qi_lb60.c new file mode 100644 index 000000000000..f15f4918f15f --- /dev/null +++ b/sound/soc/jz4740/qi_lb60.c @@ -0,0 +1,166 @@ +/* + * Copyright (C) 2009, Lars-Peter Clausen <lars@metafoo.de> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 675 Mass Ave, Cambridge, MA 02139, USA. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <linux/gpio.h> + +#include "../codecs/jz4740.h" +#include "jz4740-pcm.h" +#include "jz4740-i2s.h" + + +#define QI_LB60_SND_GPIO JZ_GPIO_PORTB(29) +#define QI_LB60_AMP_GPIO JZ_GPIO_PORTD(4) + +static int qi_lb60_spk_event(struct snd_soc_dapm_widget *widget, + struct snd_kcontrol *ctrl, int event) +{ + int on = 0; + if (event & SND_SOC_DAPM_POST_PMU) + on = 1; + else if (event & SND_SOC_DAPM_PRE_PMD) + on = 0; + + gpio_set_value(QI_LB60_SND_GPIO, on); + gpio_set_value(QI_LB60_AMP_GPIO, on); + + return 0; +} + +static const struct snd_soc_dapm_widget qi_lb60_widgets[] = { + SND_SOC_DAPM_SPK("Speaker", qi_lb60_spk_event), + SND_SOC_DAPM_MIC("Mic", NULL), +}; + +static const struct snd_soc_dapm_route qi_lb60_routes[] = { + {"Mic", NULL, "MIC"}, + {"Speaker", NULL, "LOUT"}, + {"Speaker", NULL, "ROUT"}, +}; + +#define QI_LB60_DAIFMT (SND_SOC_DAIFMT_I2S | \ + SND_SOC_DAIFMT_NB_NF | \ + SND_SOC_DAIFMT_CBM_CFM) + +static int qi_lb60_codec_init(struct snd_soc_codec *codec) +{ + int ret; + struct snd_soc_dai *cpu_dai = codec->socdev->card->dai_link->cpu_dai; + + snd_soc_dapm_nc_pin(codec, "LIN"); + snd_soc_dapm_nc_pin(codec, "RIN"); + + ret = snd_soc_dai_set_fmt(cpu_dai, QI_LB60_DAIFMT); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cpu dai format: %d\n", ret); + return ret; + } + + snd_soc_dapm_new_controls(codec, qi_lb60_widgets, ARRAY_SIZE(qi_lb60_widgets)); + snd_soc_dapm_add_routes(codec, qi_lb60_routes, ARRAY_SIZE(qi_lb60_routes)); + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link qi_lb60_dai = { + .name = "jz4740", + .stream_name = "jz4740", + .cpu_dai = &jz4740_i2s_dai, + .codec_dai = &jz4740_codec_dai, + .init = qi_lb60_codec_init, +}; + +static struct snd_soc_card qi_lb60 = { + .name = "QI LB60", + .dai_link = &qi_lb60_dai, + .num_links = 1, + .platform = &jz4740_soc_platform, +}; + +static struct snd_soc_device qi_lb60_snd_devdata = { + .card = &qi_lb60, + .codec_dev = &soc_codec_dev_jz4740_codec, +}; + +static struct platform_device *qi_lb60_snd_device; + +static int __init qi_lb60_init(void) +{ + int ret; + + qi_lb60_snd_device = platform_device_alloc("soc-audio", -1); + + if (!qi_lb60_snd_device) + return -ENOMEM; + + ret = gpio_request(QI_LB60_SND_GPIO, "SND"); + if (ret) { + pr_err("qi_lb60 snd: Failed to request SND GPIO(%d): %d\n", + QI_LB60_SND_GPIO, ret); + goto err_device_put; + } + + ret = gpio_request(QI_LB60_AMP_GPIO, "AMP"); + if (ret) { + pr_err("qi_lb60 snd: Failed to request AMP GPIO(%d): %d\n", + QI_LB60_AMP_GPIO, ret); + goto err_gpio_free_snd; + } + + gpio_direction_output(QI_LB60_SND_GPIO, 0); + gpio_direction_output(QI_LB60_AMP_GPIO, 0); + + platform_set_drvdata(qi_lb60_snd_device, &qi_lb60_snd_devdata); + qi_lb60_snd_devdata.dev = &qi_lb60_snd_device->dev; + + ret = platform_device_add(qi_lb60_snd_device); + if (ret) { + pr_err("qi_lb60 snd: Failed to add snd soc device: %d\n", ret); + goto err_unset_pdata; + } + + return 0; + +err_unset_pdata: + platform_set_drvdata(qi_lb60_snd_device, NULL); +/*err_gpio_free_amp:*/ + gpio_free(QI_LB60_AMP_GPIO); +err_gpio_free_snd: + gpio_free(QI_LB60_SND_GPIO); +err_device_put: + platform_device_put(qi_lb60_snd_device); + + return ret; +} +module_init(qi_lb60_init); + +static void __exit qi_lb60_exit(void) +{ + gpio_free(QI_LB60_AMP_GPIO); + gpio_free(QI_LB60_SND_GPIO); + platform_device_unregister(qi_lb60_snd_device); +} +module_exit(qi_lb60_exit); + +MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>"); +MODULE_DESCRIPTION("ALSA SoC QI LB60 Audio support"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig new file mode 100644 index 000000000000..16ec2a2dba4d --- /dev/null +++ b/sound/soc/kirkwood/Kconfig @@ -0,0 +1,20 @@ +config SND_KIRKWOOD_SOC + tristate "SoC Audio for the Marvell Kirkwood chip" + depends on ARCH_KIRKWOOD + help + Say Y or M if you want to add support for codecs attached to + the Kirkwood I2S interface. You will also need to select the + audio interfaces to support below. + +config SND_KIRKWOOD_SOC_I2S + tristate + +config SND_KIRKWOOD_SOC_OPENRD + tristate "SoC Audio support for Kirkwood Openrd Client" + depends on SND_KIRKWOOD_SOC && MACH_OPENRD_CLIENT + select SND_KIRKWOOD_SOC_I2S + select SND_SOC_CS42L51 + help + Say Y if you want to add support for SoC audio on + Openrd Client. + diff --git a/sound/soc/kirkwood/Makefile b/sound/soc/kirkwood/Makefile new file mode 100644 index 000000000000..33a16dcab5b5 --- /dev/null +++ b/sound/soc/kirkwood/Makefile @@ -0,0 +1,9 @@ +snd-soc-kirkwood-objs := kirkwood-dma.o +snd-soc-kirkwood-i2s-objs := kirkwood-i2s.o + +obj-$(CONFIG_SND_KIRKWOOD_SOC) += snd-soc-kirkwood.o +obj-$(CONFIG_SND_KIRKWOOD_SOC_I2S) += snd-soc-kirkwood-i2s.o + +snd-soc-openrd-objs := kirkwood-openrd.o + +obj-$(CONFIG_SND_KIRKWOOD_SOC_OPENRD) += snd-soc-openrd.o diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c new file mode 100644 index 000000000000..a30205be3e2b --- /dev/null +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -0,0 +1,383 @@ +/* + * kirkwood-dma.c + * + * (c) 2010 Arnaud Patard <apatard@mandriva.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/io.h> +#include <linux/slab.h> +#include <linux/interrupt.h> +#include <linux/dma-mapping.h> +#include <linux/mbus.h> +#include <sound/soc.h> +#include "kirkwood-dma.h" +#include "kirkwood.h" + +#define KIRKWOOD_RATES \ + (SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000) +#define KIRKWOOD_FORMATS \ + (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +struct kirkwood_dma_priv { + struct snd_pcm_substream *play_stream; + struct snd_pcm_substream *rec_stream; + struct kirkwood_dma_data *data; +}; + +static struct snd_pcm_hardware kirkwood_dma_snd_hw = { + .info = (SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_PAUSE), + .formats = KIRKWOOD_FORMATS, + .rates = KIRKWOOD_RATES, + .rate_min = 44100, + .rate_max = 96000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = KIRKWOOD_SND_MAX_PERIOD_BYTES * KIRKWOOD_SND_MAX_PERIODS, + .period_bytes_min = KIRKWOOD_SND_MIN_PERIOD_BYTES, + .period_bytes_max = KIRKWOOD_SND_MAX_PERIOD_BYTES, + .periods_min = KIRKWOOD_SND_MIN_PERIODS, + .periods_max = KIRKWOOD_SND_MAX_PERIODS, + .fifo_size = 0, +}; + +static u64 kirkwood_dma_dmamask = 0xFFFFFFFFUL; + +static irqreturn_t kirkwood_dma_irq(int irq, void *dev_id) +{ + struct kirkwood_dma_priv *prdata = dev_id; + struct kirkwood_dma_data *priv = prdata->data; + unsigned long mask, status, cause; + + mask = readl(priv->io + KIRKWOOD_INT_MASK); + status = readl(priv->io + KIRKWOOD_INT_CAUSE) & mask; + + cause = readl(priv->io + KIRKWOOD_ERR_CAUSE); + if (unlikely(cause)) { + printk(KERN_WARNING "%s: got err interrupt 0x%lx\n", + __func__, cause); + writel(cause, priv->io + KIRKWOOD_ERR_CAUSE); + return IRQ_HANDLED; + } + + /* we've enabled only bytes interrupts ... */ + if (status & ~(KIRKWOOD_INT_CAUSE_PLAY_BYTES | \ + KIRKWOOD_INT_CAUSE_REC_BYTES)) { + printk(KERN_WARNING "%s: unexpected interrupt %lx\n", + __func__, status); + return IRQ_NONE; + } + + /* ack int */ + writel(status, priv->io + KIRKWOOD_INT_CAUSE); + + if (status & KIRKWOOD_INT_CAUSE_PLAY_BYTES) + snd_pcm_period_elapsed(prdata->play_stream); + + if (status & KIRKWOOD_INT_CAUSE_REC_BYTES) + snd_pcm_period_elapsed(prdata->rec_stream); + + return IRQ_HANDLED; +} + +static void kirkwood_dma_conf_mbus_windows(void __iomem *base, int win, + unsigned long dma, + struct mbus_dram_target_info *dram) +{ + int i; + + /* First disable and clear windows */ + writel(0, base + KIRKWOOD_AUDIO_WIN_CTRL_REG(win)); + writel(0, base + KIRKWOOD_AUDIO_WIN_BASE_REG(win)); + + /* try to find matching cs for current dma address */ + for (i = 0; i < dram->num_cs; i++) { + struct mbus_dram_window *cs = dram->cs + i; + if ((cs->base & 0xffff0000) < (dma & 0xffff0000)) { + writel(cs->base & 0xffff0000, + base + KIRKWOOD_AUDIO_WIN_BASE_REG(win)); + writel(((cs->size - 1) & 0xffff0000) | + (cs->mbus_attr << 8) | + (dram->mbus_dram_target_id << 4) | 1, + base + KIRKWOOD_AUDIO_WIN_CTRL_REG(win)); + } + } +} + +static int kirkwood_dma_open(struct snd_pcm_substream *substream) +{ + int err; + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; + struct snd_soc_dai *cpu_dai = soc_runtime->dai->cpu_dai; + struct kirkwood_dma_data *priv; + struct kirkwood_dma_priv *prdata = cpu_dai->private_data; + unsigned long addr; + + priv = snd_soc_dai_get_dma_data(cpu_dai, substream); + snd_soc_set_runtime_hwparams(substream, &kirkwood_dma_snd_hw); + + /* Ensure that all constraints linked to dma burst are fullfilled */ + err = snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, + priv->burst * 2, + KIRKWOOD_AUDIO_BUF_MAX-1); + if (err < 0) + return err; + + err = snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, + priv->burst); + if (err < 0) + return err; + + err = snd_pcm_hw_constraint_step(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, + priv->burst); + if (err < 0) + return err; + + if (soc_runtime->dai->cpu_dai->private_data == NULL) { + prdata = kzalloc(sizeof(struct kirkwood_dma_priv), GFP_KERNEL); + if (prdata == NULL) + return -ENOMEM; + + prdata->data = priv; + + err = request_irq(priv->irq, kirkwood_dma_irq, IRQF_SHARED, + "kirkwood-i2s", prdata); + if (err) { + kfree(prdata); + return -EBUSY; + } + + soc_runtime->dai->cpu_dai->private_data = prdata; + + /* + * Enable Error interrupts. We're only ack'ing them but + * it's usefull for diagnostics + */ + writel((unsigned long)-1, priv->io + KIRKWOOD_ERR_MASK); + } + + addr = virt_to_phys(substream->dma_buffer.area); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + prdata->play_stream = substream; + kirkwood_dma_conf_mbus_windows(priv->io, + KIRKWOOD_PLAYBACK_WIN, addr, priv->dram); + } else { + prdata->rec_stream = substream; + kirkwood_dma_conf_mbus_windows(priv->io, + KIRKWOOD_RECORD_WIN, addr, priv->dram); + } + + return 0; +} + +static int kirkwood_dma_close(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; + struct snd_soc_dai *cpu_dai = soc_runtime->dai->cpu_dai; + struct kirkwood_dma_priv *prdata = cpu_dai->private_data; + struct kirkwood_dma_data *priv; + + priv = snd_soc_dai_get_dma_data(cpu_dai, substream); + + if (!prdata || !priv) + return 0; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + prdata->play_stream = NULL; + else + prdata->rec_stream = NULL; + + if (!prdata->play_stream && !prdata->rec_stream) { + writel(0, priv->io + KIRKWOOD_ERR_MASK); + free_irq(priv->irq, prdata); + kfree(prdata); + soc_runtime->dai->cpu_dai->private_data = NULL; + } + + return 0; +} + +static int kirkwood_dma_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = params_buffer_bytes(params); + + return 0; +} + +static int kirkwood_dma_hw_free(struct snd_pcm_substream *substream) +{ + snd_pcm_set_runtime_buffer(substream, NULL); + return 0; +} + +static int kirkwood_dma_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; + struct snd_soc_dai *cpu_dai = soc_runtime->dai->cpu_dai; + struct kirkwood_dma_data *priv; + unsigned long size, count; + + priv = snd_soc_dai_get_dma_data(cpu_dai, substream); + + /* compute buffer size in term of "words" as requested in specs */ + size = frames_to_bytes(runtime, runtime->buffer_size); + size = (size>>2)-1; + count = snd_pcm_lib_period_bytes(substream); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + writel(count, priv->io + KIRKWOOD_PLAY_BYTE_INT_COUNT); + writel(runtime->dma_addr, priv->io + KIRKWOOD_PLAY_BUF_ADDR); + writel(size, priv->io + KIRKWOOD_PLAY_BUF_SIZE); + } else { + writel(count, priv->io + KIRKWOOD_REC_BYTE_INT_COUNT); + writel(runtime->dma_addr, priv->io + KIRKWOOD_REC_BUF_ADDR); + writel(size, priv->io + KIRKWOOD_REC_BUF_SIZE); + } + + + return 0; +} + +static snd_pcm_uframes_t kirkwood_dma_pointer(struct snd_pcm_substream + *substream) +{ + struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; + struct snd_soc_dai *cpu_dai = soc_runtime->dai->cpu_dai; + struct kirkwood_dma_data *priv; + snd_pcm_uframes_t count; + + priv = snd_soc_dai_get_dma_data(cpu_dai, substream); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + count = bytes_to_frames(substream->runtime, + readl(priv->io + KIRKWOOD_PLAY_BYTE_COUNT)); + else + count = bytes_to_frames(substream->runtime, + readl(priv->io + KIRKWOOD_REC_BYTE_COUNT)); + + return count; +} + +struct snd_pcm_ops kirkwood_dma_ops = { + .open = kirkwood_dma_open, + .close = kirkwood_dma_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = kirkwood_dma_hw_params, + .hw_free = kirkwood_dma_hw_free, + .prepare = kirkwood_dma_prepare, + .pointer = kirkwood_dma_pointer, +}; + +static int kirkwood_dma_preallocate_dma_buffer(struct snd_pcm *pcm, + int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = kirkwood_dma_snd_hw.buffer_bytes_max; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->area = dma_alloc_coherent(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + if (!buf->area) + return -ENOMEM; + buf->bytes = size; + buf->private_data = NULL; + + return 0; +} + +static int kirkwood_dma_new(struct snd_card *card, + struct snd_soc_dai *dai, struct snd_pcm *pcm) +{ + int ret; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &kirkwood_dma_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = 0xffffffff; + + if (dai->playback.channels_min) { + ret = kirkwood_dma_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + return ret; + } + + if (dai->capture.channels_min) { + ret = kirkwood_dma_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + return ret; + } + + return 0; +} + +static void kirkwood_dma_free_dma_buffers(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + buf = &substream->dma_buffer; + if (!buf->area) + continue; + + dma_free_coherent(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + +struct snd_soc_platform kirkwood_soc_platform = { + .name = "kirkwood-dma", + .pcm_ops = &kirkwood_dma_ops, + .pcm_new = kirkwood_dma_new, + .pcm_free = kirkwood_dma_free_dma_buffers, +}; +EXPORT_SYMBOL_GPL(kirkwood_soc_platform); + +static int __init kirkwood_soc_platform_init(void) +{ + return snd_soc_register_platform(&kirkwood_soc_platform); +} +module_init(kirkwood_soc_platform_init); + +static void __exit kirkwood_soc_platform_exit(void) +{ + snd_soc_unregister_platform(&kirkwood_soc_platform); +} +module_exit(kirkwood_soc_platform_exit); + +MODULE_AUTHOR("Arnaud Patard <apatard@mandriva.com>"); +MODULE_DESCRIPTION("Marvell Kirkwood Audio DMA module"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/kirkwood/kirkwood-dma.h b/sound/soc/kirkwood/kirkwood-dma.h new file mode 100644 index 000000000000..ba4454cd34f1 --- /dev/null +++ b/sound/soc/kirkwood/kirkwood-dma.h @@ -0,0 +1,17 @@ +/* + * kirkwood-dma.h + * + * (c) 2010 Arnaud Patard <apatard@mandriva.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#ifndef _KIRKWOOD_DMA_H +#define _KIRKWOOD_DMA_H + +extern struct snd_soc_platform kirkwood_soc_platform; + +#endif diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c new file mode 100644 index 000000000000..981ffc2a13c8 --- /dev/null +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -0,0 +1,495 @@ +/* + * kirkwood-i2s.c + * + * (c) 2010 Arnaud Patard <apatard@mandriva.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/io.h> +#include <linux/slab.h> +#include <linux/mbus.h> +#include <linux/delay.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <plat/audio.h> +#include "kirkwood-i2s.h" +#include "kirkwood.h" + +#define DRV_NAME "kirkwood-i2s" + +#define KIRKWOOD_I2S_RATES \ + (SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000) +#define KIRKWOOD_I2S_FORMATS \ + (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + + +struct snd_soc_dai kirkwood_i2s_dai; +static struct kirkwood_dma_data *priv; + +static int kirkwood_i2s_set_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + unsigned long mask; + unsigned long value; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + mask = KIRKWOOD_I2S_CTL_RJ; + break; + case SND_SOC_DAIFMT_LEFT_J: + mask = KIRKWOOD_I2S_CTL_LJ; + break; + case SND_SOC_DAIFMT_I2S: + mask = KIRKWOOD_I2S_CTL_I2S; + break; + default: + return -EINVAL; + } + + /* + * Set same format for playback and record + * This avoids some troubles. + */ + value = readl(priv->io+KIRKWOOD_I2S_PLAYCTL); + value &= ~KIRKWOOD_I2S_CTL_JUST_MASK; + value |= mask; + writel(value, priv->io+KIRKWOOD_I2S_PLAYCTL); + + value = readl(priv->io+KIRKWOOD_I2S_RECCTL); + value &= ~KIRKWOOD_I2S_CTL_JUST_MASK; + value |= mask; + writel(value, priv->io+KIRKWOOD_I2S_RECCTL); + + return 0; +} + +static inline void kirkwood_set_dco(void __iomem *io, unsigned long rate) +{ + unsigned long value; + + value = KIRKWOOD_DCO_CTL_OFFSET_0; + switch (rate) { + default: + case 44100: + value |= KIRKWOOD_DCO_CTL_FREQ_11; + break; + case 48000: + value |= KIRKWOOD_DCO_CTL_FREQ_12; + break; + case 96000: + value |= KIRKWOOD_DCO_CTL_FREQ_24; + break; + } + writel(value, io + KIRKWOOD_DCO_CTL); + + /* wait for dco locked */ + do { + cpu_relax(); + value = readl(io + KIRKWOOD_DCO_SPCR_STATUS); + value &= KIRKWOOD_DCO_SPCR_STATUS; + } while (value == 0); +} + +static int kirkwood_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + unsigned int i2s_reg, reg; + unsigned long i2s_value, value; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + i2s_reg = KIRKWOOD_I2S_PLAYCTL; + reg = KIRKWOOD_PLAYCTL; + } else { + i2s_reg = KIRKWOOD_I2S_RECCTL; + reg = KIRKWOOD_RECCTL; + } + + /* set dco conf */ + kirkwood_set_dco(priv->io, params_rate(params)); + + i2s_value = readl(priv->io+i2s_reg); + i2s_value &= ~KIRKWOOD_I2S_CTL_SIZE_MASK; + + value = readl(priv->io+reg); + value &= ~KIRKWOOD_PLAYCTL_SIZE_MASK; + + /* + * Size settings in play/rec i2s control regs and play/rec control + * regs must be the same. + */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + i2s_value |= KIRKWOOD_I2S_CTL_SIZE_16; + value |= KIRKWOOD_PLAYCTL_SIZE_16_C; + break; + /* + * doesn't work... S20_3LE != kirkwood 20bit format ? + * + case SNDRV_PCM_FORMAT_S20_3LE: + i2s_value |= KIRKWOOD_I2S_CTL_SIZE_20; + value |= KIRKWOOD_PLAYCTL_SIZE_20; + break; + */ + case SNDRV_PCM_FORMAT_S24_LE: + i2s_value |= KIRKWOOD_I2S_CTL_SIZE_24; + value |= KIRKWOOD_PLAYCTL_SIZE_24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + i2s_value |= KIRKWOOD_I2S_CTL_SIZE_32; + value |= KIRKWOOD_PLAYCTL_SIZE_32; + break; + default: + return -EINVAL; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + value &= ~KIRKWOOD_PLAYCTL_MONO_MASK; + if (params_channels(params) == 1) + value |= KIRKWOOD_PLAYCTL_MONO_BOTH; + else + value |= KIRKWOOD_PLAYCTL_MONO_OFF; + } + + writel(i2s_value, priv->io+i2s_reg); + writel(value, priv->io+reg); + + return 0; +} + +static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + unsigned long value; + + /* + * specs says KIRKWOOD_PLAYCTL must be read 2 times before + * changing it. So read 1 time here and 1 later. + */ + value = readl(priv->io + KIRKWOOD_PLAYCTL); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + /* stop audio, enable interrupts */ + value = readl(priv->io + KIRKWOOD_PLAYCTL); + value |= KIRKWOOD_PLAYCTL_PAUSE; + writel(value, priv->io + KIRKWOOD_PLAYCTL); + + value = readl(priv->io + KIRKWOOD_INT_MASK); + value |= KIRKWOOD_INT_CAUSE_PLAY_BYTES; + writel(value, priv->io + KIRKWOOD_INT_MASK); + + /* configure audio & enable i2s playback */ + value = readl(priv->io + KIRKWOOD_PLAYCTL); + value &= ~KIRKWOOD_PLAYCTL_BURST_MASK; + value &= ~(KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE + | KIRKWOOD_PLAYCTL_SPDIF_EN); + + if (priv->burst == 32) + value |= KIRKWOOD_PLAYCTL_BURST_32; + else + value |= KIRKWOOD_PLAYCTL_BURST_128; + value |= KIRKWOOD_PLAYCTL_I2S_EN; + writel(value, priv->io + KIRKWOOD_PLAYCTL); + break; + + case SNDRV_PCM_TRIGGER_STOP: + /* stop audio, disable interrupts */ + value = readl(priv->io + KIRKWOOD_PLAYCTL); + value |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE; + writel(value, priv->io + KIRKWOOD_PLAYCTL); + + value = readl(priv->io + KIRKWOOD_INT_MASK); + value &= ~KIRKWOOD_INT_CAUSE_PLAY_BYTES; + writel(value, priv->io + KIRKWOOD_INT_MASK); + + /* disable all playbacks */ + value = readl(priv->io + KIRKWOOD_PLAYCTL); + value &= ~(KIRKWOOD_PLAYCTL_I2S_EN | KIRKWOOD_PLAYCTL_SPDIF_EN); + writel(value, priv->io + KIRKWOOD_PLAYCTL); + break; + + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + value = readl(priv->io + KIRKWOOD_PLAYCTL); + value |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE; + writel(value, priv->io + KIRKWOOD_PLAYCTL); + break; + + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + value = readl(priv->io + KIRKWOOD_PLAYCTL); + value &= ~(KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE); + writel(value, priv->io + KIRKWOOD_PLAYCTL); + break; + + default: + return -EINVAL; + } + + return 0; +} + +static int kirkwood_i2s_rec_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + unsigned long value; + + value = readl(priv->io + KIRKWOOD_RECCTL); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + /* stop audio, enable interrupts */ + value = readl(priv->io + KIRKWOOD_RECCTL); + value |= KIRKWOOD_RECCTL_PAUSE; + writel(value, priv->io + KIRKWOOD_RECCTL); + + value = readl(priv->io + KIRKWOOD_INT_MASK); + value |= KIRKWOOD_INT_CAUSE_REC_BYTES; + writel(value, priv->io + KIRKWOOD_INT_MASK); + + /* configure audio & enable i2s record */ + value = readl(priv->io + KIRKWOOD_RECCTL); + value &= ~KIRKWOOD_RECCTL_BURST_MASK; + value &= ~KIRKWOOD_RECCTL_MONO; + value &= ~(KIRKWOOD_RECCTL_PAUSE | KIRKWOOD_RECCTL_MUTE + | KIRKWOOD_RECCTL_SPDIF_EN); + + if (priv->burst == 32) + value |= KIRKWOOD_RECCTL_BURST_32; + else + value |= KIRKWOOD_RECCTL_BURST_128; + value |= KIRKWOOD_RECCTL_I2S_EN; + + writel(value, priv->io + KIRKWOOD_RECCTL); + break; + + case SNDRV_PCM_TRIGGER_STOP: + /* stop audio, disable interrupts */ + value = readl(priv->io + KIRKWOOD_RECCTL); + value |= KIRKWOOD_RECCTL_PAUSE | KIRKWOOD_RECCTL_MUTE; + writel(value, priv->io + KIRKWOOD_RECCTL); + + value = readl(priv->io + KIRKWOOD_INT_MASK); + value &= ~KIRKWOOD_INT_CAUSE_REC_BYTES; + writel(value, priv->io + KIRKWOOD_INT_MASK); + + /* disable all records */ + value = readl(priv->io + KIRKWOOD_RECCTL); + value &= ~(KIRKWOOD_RECCTL_I2S_EN | KIRKWOOD_RECCTL_SPDIF_EN); + writel(value, priv->io + KIRKWOOD_RECCTL); + break; + + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + value = readl(priv->io + KIRKWOOD_RECCTL); + value |= KIRKWOOD_RECCTL_PAUSE | KIRKWOOD_RECCTL_MUTE; + writel(value, priv->io + KIRKWOOD_RECCTL); + break; + + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + value = readl(priv->io + KIRKWOOD_RECCTL); + value &= ~(KIRKWOOD_RECCTL_PAUSE | KIRKWOOD_RECCTL_MUTE); + writel(value, priv->io + KIRKWOOD_RECCTL); + break; + + default: + return -EINVAL; + } + + return 0; +} + +static int kirkwood_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + return kirkwood_i2s_play_trigger(substream, cmd, dai); + else + return kirkwood_i2s_rec_trigger(substream, cmd, dai); + + return 0; +} + +static int kirkwood_i2s_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + unsigned long value; + unsigned int reg_data; + + /* put system in a "safe" state : */ + /* disable audio interrupts */ + writel(0xffffffff, priv->io + KIRKWOOD_INT_CAUSE); + writel(0, priv->io + KIRKWOOD_INT_MASK); + + reg_data = readl(priv->io + 0x1200); + reg_data &= (~(0x333FF8)); + reg_data |= 0x111D18; + writel(reg_data, priv->io + 0x1200); + + msleep(500); + + reg_data = readl(priv->io + 0x1200); + reg_data &= (~(0x333FF8)); + reg_data |= 0x111D18; + writel(reg_data, priv->io + 0x1200); + + /* disable playback/record */ + value = readl(priv->io + KIRKWOOD_PLAYCTL); + value &= ~(KIRKWOOD_PLAYCTL_I2S_EN|KIRKWOOD_PLAYCTL_SPDIF_EN); + writel(value, priv->io + KIRKWOOD_PLAYCTL); + + value = readl(priv->io + KIRKWOOD_RECCTL); + value &= ~(KIRKWOOD_RECCTL_I2S_EN | KIRKWOOD_RECCTL_SPDIF_EN); + writel(value, priv->io + KIRKWOOD_RECCTL); + + return 0; + +} + +static void kirkwood_i2s_remove(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ +} + +static struct snd_soc_dai_ops kirkwood_i2s_dai_ops = { + .trigger = kirkwood_i2s_trigger, + .hw_params = kirkwood_i2s_hw_params, + .set_fmt = kirkwood_i2s_set_fmt, +}; + + +struct snd_soc_dai kirkwood_i2s_dai = { + .name = DRV_NAME, + .id = 0, + .probe = kirkwood_i2s_probe, + .remove = kirkwood_i2s_remove, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = KIRKWOOD_I2S_RATES, + .formats = KIRKWOOD_I2S_FORMATS,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = KIRKWOOD_I2S_RATES, + .formats = KIRKWOOD_I2S_FORMATS,}, + .ops = &kirkwood_i2s_dai_ops, +}; +EXPORT_SYMBOL_GPL(kirkwood_i2s_dai); + +static __devinit int kirkwood_i2s_dev_probe(struct platform_device *pdev) +{ + struct resource *mem; + struct kirkwood_asoc_platform_data *data = + pdev->dev.platform_data; + int err; + + priv = kzalloc(sizeof(struct kirkwood_dma_data), GFP_KERNEL); + if (!priv) { + dev_err(&pdev->dev, "allocation failed\n"); + err = -ENOMEM; + goto error; + } + + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!mem) { + dev_err(&pdev->dev, "platform_get_resource failed\n"); + err = -ENXIO; + goto err_alloc; + } + + priv->mem = request_mem_region(mem->start, SZ_16K, DRV_NAME); + if (!priv->mem) { + dev_err(&pdev->dev, "request_mem_region failed\n"); + err = -EBUSY; + goto error; + } + + priv->io = ioremap(priv->mem->start, SZ_16K); + if (!priv->io) { + dev_err(&pdev->dev, "ioremap failed\n"); + err = -ENOMEM; + goto err_iomem; + } + + priv->irq = platform_get_irq(pdev, 0); + if (priv->irq <= 0) { + dev_err(&pdev->dev, "platform_get_irq failed\n"); + err = -ENXIO; + goto err_ioremap; + } + + if (!data || !data->dram) { + dev_err(&pdev->dev, "no platform data ?!\n"); + err = -EINVAL; + goto err_ioremap; + } + + priv->dram = data->dram; + priv->burst = data->burst; + + kirkwood_i2s_dai.capture.dma_data = priv; + kirkwood_i2s_dai.playback.dma_data = priv; + + return snd_soc_register_dai(&kirkwood_i2s_dai); + +err_ioremap: + iounmap(priv->io); +err_iomem: + release_mem_region(priv->mem->start, SZ_16K); +err_alloc: + kfree(priv); +error: + return err; +} + +static __devexit int kirkwood_i2s_dev_remove(struct platform_device *pdev) +{ + if (priv) { + iounmap(priv->io); + release_mem_region(priv->mem->start, SZ_16K); + kfree(priv); + } + snd_soc_unregister_dai(&kirkwood_i2s_dai); + return 0; +} + +static struct platform_driver kirkwood_i2s_driver = { + .probe = kirkwood_i2s_dev_probe, + .remove = kirkwood_i2s_dev_remove, + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + }, +}; + +static int __init kirkwood_i2s_init(void) +{ + return platform_driver_register(&kirkwood_i2s_driver); +} +module_init(kirkwood_i2s_init); + +static void __exit kirkwood_i2s_exit(void) +{ + platform_driver_unregister(&kirkwood_i2s_driver); +} +module_exit(kirkwood_i2s_exit); + +/* Module information */ +MODULE_AUTHOR("Arnaud Patard, <apatard@mandriva.com>"); +MODULE_DESCRIPTION("Kirkwood I2S SoC Interface"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:kirkwood-i2s"); diff --git a/sound/soc/kirkwood/kirkwood-i2s.h b/sound/soc/kirkwood/kirkwood-i2s.h new file mode 100644 index 000000000000..c5595c616d7a --- /dev/null +++ b/sound/soc/kirkwood/kirkwood-i2s.h @@ -0,0 +1,17 @@ +/* + * kirkwood-i2s.h + * + * (c) 2010 Arnaud Patard <apatard@mandriva.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#ifndef _KIRKWOOD_I2S_H +#define _KIRKWOOD_I2S_H + +extern struct snd_soc_dai kirkwood_i2s_dai; + +#endif diff --git a/sound/soc/kirkwood/kirkwood-openrd.c b/sound/soc/kirkwood/kirkwood-openrd.c new file mode 100644 index 000000000000..0353d06bc41a --- /dev/null +++ b/sound/soc/kirkwood/kirkwood-openrd.c @@ -0,0 +1,126 @@ +/* + * kirkwood-openrd.c + * + * (c) 2010 Arnaud Patard <apatard@mandriva.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <sound/soc.h> +#include <mach/kirkwood.h> +#include <plat/audio.h> +#include <asm/mach-types.h> +#include "kirkwood-i2s.h" +#include "kirkwood-dma.h" +#include "../codecs/cs42l51.h" + +static int openrd_client_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + unsigned int freq, fmt; + + fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS; + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) + return ret; + + switch (params_rate(params)) { + default: + case 44100: + freq = 11289600; + break; + case 48000: + freq = 12288000; + break; + case 96000: + freq = 24576000; + break; + } + + return snd_soc_dai_set_sysclk(codec_dai, 0, freq, SND_SOC_CLOCK_IN); + +} + +static struct snd_soc_ops openrd_client_ops = { + .hw_params = openrd_client_hw_params, +}; + + +static struct snd_soc_dai_link openrd_client_dai[] = { +{ + .name = "CS42L51", + .stream_name = "CS42L51 HiFi", + .cpu_dai = &kirkwood_i2s_dai, + .codec_dai = &cs42l51_dai, + .ops = &openrd_client_ops, +}, +}; + + +static struct snd_soc_card openrd_client = { + .name = "OpenRD Client", + .platform = &kirkwood_soc_platform, + .dai_link = openrd_client_dai, + .num_links = ARRAY_SIZE(openrd_client_dai), +}; + +static struct snd_soc_device openrd_client_snd_devdata = { + .card = &openrd_client, + .codec_dev = &soc_codec_device_cs42l51, +}; + +static struct platform_device *openrd_client_snd_device; + +static int __init openrd_client_init(void) +{ + int ret; + + if (!machine_is_openrd_client()) + return 0; + + openrd_client_snd_device = platform_device_alloc("soc-audio", -1); + if (!openrd_client_snd_device) + return -ENOMEM; + + platform_set_drvdata(openrd_client_snd_device, + &openrd_client_snd_devdata); + openrd_client_snd_devdata.dev = &openrd_client_snd_device->dev; + + ret = platform_device_add(openrd_client_snd_device); + if (ret) { + printk(KERN_ERR "%s: platform_device_add failed\n", __func__); + platform_device_put(openrd_client_snd_device); + } + + return ret; +} + +static void __exit openrd_client_exit(void) +{ + platform_device_unregister(openrd_client_snd_device); +} + +module_init(openrd_client_init); +module_exit(openrd_client_exit); + +/* Module information */ +MODULE_AUTHOR("Arnaud Patard <apatard@mandriva.com>"); +MODULE_DESCRIPTION("ALSA SoC OpenRD Client"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:soc-audio"); diff --git a/sound/soc/kirkwood/kirkwood.h b/sound/soc/kirkwood/kirkwood.h new file mode 100644 index 000000000000..bb6e6a5648c9 --- /dev/null +++ b/sound/soc/kirkwood/kirkwood.h @@ -0,0 +1,129 @@ +/* + * kirkwood.h + * + * (c) 2010 Arnaud Patard <apatard@mandriva.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#ifndef _KIRKWOOD_AUDIO_H +#define _KIRKWOOD_AUDIO_H + +#define KIRKWOOD_RECORD_WIN 0 +#define KIRKWOOD_PLAYBACK_WIN 1 +#define KIRKWOOD_MAX_AUDIO_WIN 2 + +#define KIRKWOOD_AUDIO_WIN_BASE_REG(win) (0xA00 + ((win)<<3)) +#define KIRKWOOD_AUDIO_WIN_CTRL_REG(win) (0xA04 + ((win)<<3)) + + +#define KIRKWOOD_RECCTL 0x1000 +#define KIRKWOOD_RECCTL_SPDIF_EN (1<<11) +#define KIRKWOOD_RECCTL_I2S_EN (1<<10) +#define KIRKWOOD_RECCTL_PAUSE (1<<9) +#define KIRKWOOD_RECCTL_MUTE (1<<8) +#define KIRKWOOD_RECCTL_BURST_MASK (3<<5) +#define KIRKWOOD_RECCTL_BURST_128 (2<<5) +#define KIRKWOOD_RECCTL_BURST_32 (1<<5) +#define KIRKWOOD_RECCTL_MONO (1<<4) +#define KIRKWOOD_RECCTL_MONO_CHAN_RIGHT (1<<3) +#define KIRKWOOD_RECCTL_MONO_CHAN_LEFT (0<<3) +#define KIRKWOOD_RECCTL_SIZE_MASK (7<<0) +#define KIRKWOOD_RECCTL_SIZE_16 (7<<0) +#define KIRKWOOD_RECCTL_SIZE_16_C (3<<0) +#define KIRKWOOD_RECCTL_SIZE_20 (2<<0) +#define KIRKWOOD_RECCTL_SIZE_24 (1<<0) +#define KIRKWOOD_RECCTL_SIZE_32 (0<<0) + +#define KIRKWOOD_REC_BUF_ADDR 0x1004 +#define KIRKWOOD_REC_BUF_SIZE 0x1008 +#define KIRKWOOD_REC_BYTE_COUNT 0x100C + +#define KIRKWOOD_PLAYCTL 0x1100 +#define KIRKWOOD_PLAYCTL_PLAY_BUSY (1<<16) +#define KIRKWOOD_PLAYCTL_BURST_MASK (3<<11) +#define KIRKWOOD_PLAYCTL_BURST_128 (2<<11) +#define KIRKWOOD_PLAYCTL_BURST_32 (1<<11) +#define KIRKWOOD_PLAYCTL_PAUSE (1<<9) +#define KIRKWOOD_PLAYCTL_SPDIF_MUTE (1<<8) +#define KIRKWOOD_PLAYCTL_MONO_MASK (3<<5) +#define KIRKWOOD_PLAYCTL_MONO_BOTH (3<<5) +#define KIRKWOOD_PLAYCTL_MONO_OFF (0<<5) +#define KIRKWOOD_PLAYCTL_I2S_MUTE (1<<7) +#define KIRKWOOD_PLAYCTL_SPDIF_EN (1<<4) +#define KIRKWOOD_PLAYCTL_I2S_EN (1<<3) +#define KIRKWOOD_PLAYCTL_SIZE_MASK (7<<0) +#define KIRKWOOD_PLAYCTL_SIZE_16 (7<<0) +#define KIRKWOOD_PLAYCTL_SIZE_16_C (3<<0) +#define KIRKWOOD_PLAYCTL_SIZE_20 (2<<0) +#define KIRKWOOD_PLAYCTL_SIZE_24 (1<<0) +#define KIRKWOOD_PLAYCTL_SIZE_32 (0<<0) + +#define KIRKWOOD_PLAY_BUF_ADDR 0x1104 +#define KIRKWOOD_PLAY_BUF_SIZE 0x1108 +#define KIRKWOOD_PLAY_BYTE_COUNT 0x110C + +#define KIRKWOOD_DCO_CTL 0x1204 +#define KIRKWOOD_DCO_CTL_OFFSET_MASK (0xFFF<<2) +#define KIRKWOOD_DCO_CTL_OFFSET_0 (0x800<<2) +#define KIRKWOOD_DCO_CTL_FREQ_MASK (3<<0) +#define KIRKWOOD_DCO_CTL_FREQ_11 (0<<0) +#define KIRKWOOD_DCO_CTL_FREQ_12 (1<<0) +#define KIRKWOOD_DCO_CTL_FREQ_24 (2<<0) + +#define KIRKWOOD_DCO_SPCR_STATUS 0x120c +#define KIRKWOOD_DCO_SPCR_STATUS_DCO_LOCK (1<<16) + +#define KIRKWOOD_ERR_CAUSE 0x1300 +#define KIRKWOOD_ERR_MASK 0x1304 + +#define KIRKWOOD_INT_CAUSE 0x1308 +#define KIRKWOOD_INT_MASK 0x130C +#define KIRKWOOD_INT_CAUSE_PLAY_BYTES (1<<14) +#define KIRKWOOD_INT_CAUSE_REC_BYTES (1<<13) +#define KIRKWOOD_INT_CAUSE_DMA_PLAY_END (1<<7) +#define KIRKWOOD_INT_CAUSE_DMA_PLAY_3Q (1<<6) +#define KIRKWOOD_INT_CAUSE_DMA_PLAY_HALF (1<<5) +#define KIRKWOOD_INT_CAUSE_DMA_PLAY_1Q (1<<4) +#define KIRKWOOD_INT_CAUSE_DMA_REC_END (1<<3) +#define KIRKWOOD_INT_CAUSE_DMA_REC_3Q (1<<2) +#define KIRKWOOD_INT_CAUSE_DMA_REC_HALF (1<<1) +#define KIRKWOOD_INT_CAUSE_DMA_REC_1Q (1<<0) + +#define KIRKWOOD_REC_BYTE_INT_COUNT 0x1310 +#define KIRKWOOD_PLAY_BYTE_INT_COUNT 0x1314 +#define KIRKWOOD_BYTE_INT_COUNT_MASK 0xffffff + +#define KIRKWOOD_I2S_PLAYCTL 0x2508 +#define KIRKWOOD_I2S_RECCTL 0x2408 +#define KIRKWOOD_I2S_CTL_JUST_MASK (0xf<<26) +#define KIRKWOOD_I2S_CTL_LJ (0<<26) +#define KIRKWOOD_I2S_CTL_I2S (5<<26) +#define KIRKWOOD_I2S_CTL_RJ (8<<26) +#define KIRKWOOD_I2S_CTL_SIZE_MASK (3<<30) +#define KIRKWOOD_I2S_CTL_SIZE_16 (3<<30) +#define KIRKWOOD_I2S_CTL_SIZE_20 (2<<30) +#define KIRKWOOD_I2S_CTL_SIZE_24 (1<<30) +#define KIRKWOOD_I2S_CTL_SIZE_32 (0<<30) + +#define KIRKWOOD_AUDIO_BUF_MAX (16*1024*1024) + +/* Theses values come from the marvell alsa driver */ +/* need to find where they come from */ +#define KIRKWOOD_SND_MIN_PERIODS 8 +#define KIRKWOOD_SND_MAX_PERIODS 16 +#define KIRKWOOD_SND_MIN_PERIOD_BYTES 0x4000 +#define KIRKWOOD_SND_MAX_PERIOD_BYTES 0x4000 + +struct kirkwood_dma_data { + struct resource *mem; + void __iomem *io; + int irq; + int burst; + struct mbus_dram_target_info *dram; +}; + +#endif diff --git a/sound/soc/nuc900/Kconfig b/sound/soc/nuc900/Kconfig new file mode 100644 index 000000000000..a0ed1c618f60 --- /dev/null +++ b/sound/soc/nuc900/Kconfig @@ -0,0 +1,27 @@ +## +## NUC900 series AC97 API +## +config SND_SOC_NUC900 + tristate "SoC Audio for NUC900 series" + depends on ARCH_W90X900 + help + This option enables support for AC97 mode on the NUC900 SoC. + +config SND_SOC_NUC900_AC97 + tristate + select AC97_BUS + select SND_AC97_CODEC + select SND_SOC_AC97_BUS + + +## +## Boards +## +config SND_SOC_NUC900EVB + tristate "NUC900 AC97 support for demo board" + depends on SND_SOC_NUC900 + select SND_SOC_NUC900_AC97 + select SND_SOC_AC97_CODEC + help + Select this option to enable audio (AC97) on the + NUC900 demoboard. diff --git a/sound/soc/nuc900/Makefile b/sound/soc/nuc900/Makefile new file mode 100644 index 000000000000..7e46c7150316 --- /dev/null +++ b/sound/soc/nuc900/Makefile @@ -0,0 +1,11 @@ +# NUC900 series audio +snd-soc-nuc900-pcm-objs := nuc900-pcm.o +snd-soc-nuc900-ac97-objs := nuc900-ac97.o + +obj-$(CONFIG_SND_SOC_NUC900) += snd-soc-nuc900-pcm.o +obj-$(CONFIG_SND_SOC_NUC900_AC97) += snd-soc-nuc900-ac97.o + +# Boards +snd-soc-nuc900-audio-objs := nuc900-audio.o + +obj-$(CONFIG_SND_SOC_NUC900EVB) += snd-soc-nuc900-audio.o diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c new file mode 100644 index 000000000000..caa7c901bc2e --- /dev/null +++ b/sound/soc/nuc900/nuc900-ac97.c @@ -0,0 +1,430 @@ +/* + * Copyright (c) 2009-2010 Nuvoton technology corporation. + * + * Wan ZongShun <mcuos.com@gmail.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation;version 2 of the License. + * + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/mutex.h> +#include <linux/suspend.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <linux/device.h> +#include <linux/clk.h> + +#include <mach/mfp.h> + +#include "nuc900-audio.h" + +static DEFINE_MUTEX(ac97_mutex); +struct nuc900_audio *nuc900_ac97_data; + +static int nuc900_checkready(void) +{ + struct nuc900_audio *nuc900_audio = nuc900_ac97_data; + + if (!(AUDIO_READ(nuc900_audio->mmio + ACTL_ACIS0) & CODEC_READY)) + return -EPERM; + + return 0; +} + +/* AC97 controller reads codec register */ +static unsigned short nuc900_ac97_read(struct snd_ac97 *ac97, + unsigned short reg) +{ + struct nuc900_audio *nuc900_audio = nuc900_ac97_data; + unsigned long timeout = 0x10000, val; + + mutex_lock(&ac97_mutex); + + val = nuc900_checkready(); + if (!!val) { + dev_err(nuc900_audio->dev, "AC97 codec is not ready\n"); + goto out; + } + + /* set the R_WB bit and write register index */ + AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS1, R_WB | reg); + + /* set the valid frame bit and valid slots */ + val = AUDIO_READ(nuc900_audio->mmio + ACTL_ACOS0); + val |= (VALID_FRAME | SLOT1_VALID); + AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS0, val); + + udelay(100); + + /* polling the AC_R_FINISH */ + while (!(AUDIO_READ(nuc900_audio->mmio + ACTL_ACCON) & AC_R_FINISH) + && timeout--) + mdelay(1); + + if (!timeout) { + dev_err(nuc900_audio->dev, "AC97 read register time out !\n"); + val = -EPERM; + goto out; + } + + val = AUDIO_READ(nuc900_audio->mmio + ACTL_ACOS0) ; + val &= ~SLOT1_VALID; + AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS0, val); + + if (AUDIO_READ(nuc900_audio->mmio + ACTL_ACIS1) >> 2 != reg) { + dev_err(nuc900_audio->dev, + "R_INDEX of REG_ACTL_ACIS1 not match!\n"); + } + + udelay(100); + val = (AUDIO_READ(nuc900_audio->mmio + ACTL_ACIS2) & 0xFFFF); + +out: + mutex_unlock(&ac97_mutex); + return val; +} + +/* AC97 controller writes to codec register */ +static void nuc900_ac97_write(struct snd_ac97 *ac97, unsigned short reg, + unsigned short val) +{ + struct nuc900_audio *nuc900_audio = nuc900_ac97_data; + unsigned long tmp, timeout = 0x10000; + + mutex_lock(&ac97_mutex); + + tmp = nuc900_checkready(); + if (!!tmp) + dev_err(nuc900_audio->dev, "AC97 codec is not ready\n"); + + /* clear the R_WB bit and write register index */ + AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS1, reg); + + /* write register value */ + AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS2, val); + + /* set the valid frame bit and valid slots */ + tmp = AUDIO_READ(nuc900_audio->mmio + ACTL_ACOS0); + tmp |= SLOT1_VALID | SLOT2_VALID | VALID_FRAME; + AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS0, tmp); + + udelay(100); + + /* polling the AC_W_FINISH */ + while ((AUDIO_READ(nuc900_audio->mmio + ACTL_ACCON) & AC_W_FINISH) + && timeout--) + mdelay(1); + + if (!timeout) + dev_err(nuc900_audio->dev, "AC97 write register time out !\n"); + + tmp = AUDIO_READ(nuc900_audio->mmio + ACTL_ACOS0); + tmp &= ~(SLOT1_VALID | SLOT2_VALID); + AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS0, tmp); + + mutex_unlock(&ac97_mutex); + +} + +static void nuc900_ac97_warm_reset(struct snd_ac97 *ac97) +{ + struct nuc900_audio *nuc900_audio = nuc900_ac97_data; + unsigned long val; + + mutex_lock(&ac97_mutex); + + /* warm reset AC 97 */ + val = AUDIO_READ(nuc900_audio->mmio + ACTL_ACCON); + val |= AC_W_RES; + AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACCON, val); + + udelay(100); + + val = nuc900_checkready(); + if (!!val) + dev_err(nuc900_audio->dev, "AC97 codec is not ready\n"); + + mutex_unlock(&ac97_mutex); +} + +static void nuc900_ac97_cold_reset(struct snd_ac97 *ac97) +{ + struct nuc900_audio *nuc900_audio = nuc900_ac97_data; + unsigned long val; + + mutex_lock(&ac97_mutex); + + /* reset Audio Controller */ + val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET); + val |= ACTL_RESET_BIT; + AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val); + + val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET); + val &= (~ACTL_RESET_BIT); + AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val); + + /* reset AC-link interface */ + + val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET); + val |= AC_RESET; + AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val); + + val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET); + val &= ~AC_RESET; + AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val); + + /* cold reset AC 97 */ + val = AUDIO_READ(nuc900_audio->mmio + ACTL_ACCON); + val |= AC_C_RES; + AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACCON, val); + + val = AUDIO_READ(nuc900_audio->mmio + ACTL_ACCON); + val &= (~AC_C_RES); + AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACCON, val); + + udelay(100); + + mutex_unlock(&ac97_mutex); + +} + +/* AC97 controller operations */ +struct snd_ac97_bus_ops soc_ac97_ops = { + .read = nuc900_ac97_read, + .write = nuc900_ac97_write, + .reset = nuc900_ac97_cold_reset, + .warm_reset = nuc900_ac97_warm_reset, +} +EXPORT_SYMBOL_GPL(soc_ac97_ops); + +static int nuc900_ac97_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct nuc900_audio *nuc900_audio = nuc900_ac97_data; + int ret; + unsigned long val, tmp; + + ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + tmp = AUDIO_READ(nuc900_audio->mmio + ACTL_ACOS0); + tmp |= (SLOT3_VALID | SLOT4_VALID | VALID_FRAME); + AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS0, tmp); + + tmp = AUDIO_READ(nuc900_audio->mmio + ACTL_PSR); + tmp |= (P_DMA_END_IRQ | P_DMA_MIDDLE_IRQ); + AUDIO_WRITE(nuc900_audio->mmio + ACTL_PSR, tmp); + val |= AC_PLAY; + } else { + tmp = AUDIO_READ(nuc900_audio->mmio + ACTL_RSR); + tmp |= (R_DMA_END_IRQ | R_DMA_MIDDLE_IRQ); + + AUDIO_WRITE(nuc900_audio->mmio + ACTL_RSR, tmp); + val |= AC_RECORD; + } + + AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val); + + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + tmp = AUDIO_READ(nuc900_audio->mmio + ACTL_ACOS0); + tmp &= ~(SLOT3_VALID | SLOT4_VALID); + AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS0, tmp); + + AUDIO_WRITE(nuc900_audio->mmio + ACTL_PSR, RESET_PRSR); + val &= ~AC_PLAY; + } else { + AUDIO_WRITE(nuc900_audio->mmio + ACTL_RSR, RESET_PRSR); + val &= ~AC_RECORD; + } + + AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val); + + break; + default: + ret = -EINVAL; + } + + return ret; +} + +static int nuc900_ac97_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + struct nuc900_audio *nuc900_audio = nuc900_ac97_data; + unsigned long val; + + mutex_lock(&ac97_mutex); + + /* enable unit clock */ + clk_enable(nuc900_audio->clk); + + /* enable audio controller and AC-link interface */ + val = AUDIO_READ(nuc900_audio->mmio + ACTL_CON); + val |= (IIS_AC_PIN_SEL | ACLINK_EN); + AUDIO_WRITE(nuc900_audio->mmio + ACTL_CON, val); + + mutex_unlock(&ac97_mutex); + + return 0; +} + +static void nuc900_ac97_remove(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + struct nuc900_audio *nuc900_audio = nuc900_ac97_data; + + clk_disable(nuc900_audio->clk); +} + +static struct snd_soc_dai_ops nuc900_ac97_dai_ops = { + .trigger = nuc900_ac97_trigger, +}; + +struct snd_soc_dai nuc900_ac97_dai = { + .name = "nuc900-ac97", + .probe = nuc900_ac97_probe, + .remove = nuc900_ac97_remove, + .ac97_control = 1, + .playback = { + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .channels_min = 1, + .channels_max = 2, + }, + .capture = { + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .channels_min = 1, + .channels_max = 2, + }, + .ops = &nuc900_ac97_dai_ops, +} +EXPORT_SYMBOL_GPL(nuc900_ac97_dai); + +static int __devinit nuc900_ac97_drvprobe(struct platform_device *pdev) +{ + struct nuc900_audio *nuc900_audio; + int ret; + + if (nuc900_ac97_data) + return -EBUSY; + + nuc900_audio = kzalloc(sizeof(struct nuc900_audio), GFP_KERNEL); + if (!nuc900_audio) + return -ENOMEM; + + spin_lock_init(&nuc900_audio->lock); + + nuc900_audio->res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!nuc900_audio->res) { + ret = -ENODEV; + goto out0; + } + + if (!request_mem_region(nuc900_audio->res->start, + resource_size(nuc900_audio->res), pdev->name)) { + ret = -EBUSY; + goto out0; + } + + nuc900_audio->mmio = ioremap(nuc900_audio->res->start, + resource_size(nuc900_audio->res)); + if (!nuc900_audio->mmio) { + ret = -ENOMEM; + goto out1; + } + + nuc900_audio->clk = clk_get(&pdev->dev, NULL); + if (IS_ERR(nuc900_audio->clk)) { + ret = PTR_ERR(nuc900_audio->clk); + goto out2; + } + + nuc900_audio->irq_num = platform_get_irq(pdev, 0); + if (!nuc900_audio->irq_num) { + ret = -EBUSY; + goto out2; + } + + nuc900_ac97_data = nuc900_audio; + + nuc900_audio->dev = nuc900_ac97_dai.dev = &pdev->dev; + + ret = snd_soc_register_dai(&nuc900_ac97_dai); + if (ret) + goto out3; + + mfp_set_groupg(nuc900_audio->dev); /* enbale ac97 multifunction pin*/ + + return 0; + +out3: + clk_put(nuc900_audio->clk); +out2: + iounmap(nuc900_audio->mmio); +out1: + release_mem_region(nuc900_audio->res->start, + resource_size(nuc900_audio->res)); +out0: + kfree(nuc900_audio); + return ret; +} + +static int __devexit nuc900_ac97_drvremove(struct platform_device *pdev) +{ + + snd_soc_unregister_dai(&nuc900_ac97_dai); + + clk_put(nuc900_ac97_data->clk); + iounmap(nuc900_ac97_data->mmio); + release_mem_region(nuc900_ac97_data->res->start, + resource_size(nuc900_ac97_data->res)); + + nuc900_ac97_data = NULL; + + return 0; +} + +static struct platform_driver nuc900_ac97_driver = { + .driver = { + .name = "nuc900-audio", + .owner = THIS_MODULE, + }, + .probe = nuc900_ac97_drvprobe, + .remove = __devexit_p(nuc900_ac97_drvremove), +}; + +static int __init nuc900_ac97_init(void) +{ + return platform_driver_register(&nuc900_ac97_driver); +} + +static void __exit nuc900_ac97_exit(void) +{ + platform_driver_unregister(&nuc900_ac97_driver); +} + +module_init(nuc900_ac97_init); +module_exit(nuc900_ac97_exit); + +MODULE_AUTHOR("Wan ZongShun <mcuos.com@gmail.com>"); +MODULE_DESCRIPTION("NUC900 AC97 SoC driver!"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:nuc900-ac97"); diff --git a/sound/soc/nuc900/nuc900-audio.c b/sound/soc/nuc900/nuc900-audio.c new file mode 100644 index 000000000000..72e6f518f7b2 --- /dev/null +++ b/sound/soc/nuc900/nuc900-audio.c @@ -0,0 +1,81 @@ +/* + * Copyright (c) 2010 Nuvoton technology corporation. + * + * Wan ZongShun <mcuos.com@gmail.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation;version 2 of the License. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include "../codecs/ac97.h" +#include "nuc900-audio.h" + +static struct snd_soc_dai_link nuc900evb_ac97_dai = { + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &nuc900_ac97_dai, + .codec_dai = &ac97_dai, +}; + +static struct snd_soc_card nuc900evb_audio_machine = { + .name = "NUC900EVB_AC97", + .dai_link = &nuc900evb_ac97_dai, + .num_links = 1, + .platform = &nuc900_soc_platform, +}; + +static struct snd_soc_device nuc900evb_ac97_devdata = { + .card = &nuc900evb_audio_machine, + .codec_dev = &soc_codec_dev_ac97, +}; + +static struct platform_device *nuc900evb_asoc_dev; + +static int __init nuc900evb_audio_init(void) +{ + int ret; + + ret = -ENOMEM; + nuc900evb_asoc_dev = platform_device_alloc("soc-audio", -1); + if (!nuc900evb_asoc_dev) + goto out; + + /* nuc900 board audio device */ + platform_set_drvdata(nuc900evb_asoc_dev, &nuc900evb_ac97_devdata); + + nuc900evb_ac97_devdata.dev = &nuc900evb_asoc_dev->dev; + ret = platform_device_add(nuc900evb_asoc_dev); + + if (ret) { + platform_device_put(nuc900evb_asoc_dev); + nuc900evb_asoc_dev = NULL; + } + +out: + return ret; +} + +static void __exit nuc900evb_audio_exit(void) +{ + platform_device_unregister(nuc900evb_asoc_dev); +} + +module_init(nuc900evb_audio_init); +module_exit(nuc900evb_audio_exit); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("NUC900 Series ASoC audio support"); +MODULE_AUTHOR("Wan ZongShun"); diff --git a/sound/soc/nuc900/nuc900-audio.h b/sound/soc/nuc900/nuc900-audio.h new file mode 100644 index 000000000000..3038f519729f --- /dev/null +++ b/sound/soc/nuc900/nuc900-audio.h @@ -0,0 +1,117 @@ +/* + * Copyright (c) 2010 Nuvoton technology corporation. + * + * Wan ZongShun <mcuos.com@gmail.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation;version 2 of the License. + * + */ + +#ifndef _NUC900_AUDIO_H +#define _NUC900_AUDIO_H + +#include <linux/io.h> + +/* Audio Control Registers */ +#define ACTL_CON 0x00 +#define ACTL_RESET 0x04 +#define ACTL_RDSTB 0x08 +#define ACTL_RDST_LENGTH 0x0C +#define ACTL_RDSTC 0x10 +#define ACTL_RSR 0x14 +#define ACTL_PDSTB 0x18 +#define ACTL_PDST_LENGTH 0x1C +#define ACTL_PDSTC 0x20 +#define ACTL_PSR 0x24 +#define ACTL_IISCON 0x28 +#define ACTL_ACCON 0x2C +#define ACTL_ACOS0 0x30 +#define ACTL_ACOS1 0x34 +#define ACTL_ACOS2 0x38 +#define ACTL_ACIS0 0x3C +#define ACTL_ACIS1 0x40 +#define ACTL_ACIS2 0x44 +#define ACTL_COUNTER 0x48 + +/* bit definition of REG_ACTL_CON register */ +#define R_DMA_IRQ 0x1000 +#define T_DMA_IRQ 0x0800 +#define IIS_AC_PIN_SEL 0x0100 +#define FIFO_TH 0x0080 +#define ADC_EN 0x0010 +#define M80_EN 0x0008 +#define ACLINK_EN 0x0004 +#define IIS_EN 0x0002 + +/* bit definition of REG_ACTL_RESET register */ +#define W5691_PLAY 0x20000 +#define ACTL_RESET_BIT 0x10000 +#define RECORD_RIGHT_CHNNEL 0x08000 +#define RECORD_LEFT_CHNNEL 0x04000 +#define PLAY_RIGHT_CHNNEL 0x02000 +#define PLAY_LEFT_CHNNEL 0x01000 +#define DAC_PLAY 0x00800 +#define ADC_RECORD 0x00400 +#define M80_PLAY 0x00200 +#define AC_RECORD 0x00100 +#define AC_PLAY 0x00080 +#define IIS_RECORD 0x00040 +#define IIS_PLAY 0x00020 +#define DAC_RESET 0x00010 +#define ADC_RESET 0x00008 +#define M80_RESET 0x00004 +#define AC_RESET 0x00002 +#define IIS_RESET 0x00001 + +/* bit definition of REG_ACTL_ACCON register */ +#define AC_BCLK_PU_EN 0x20 +#define AC_R_FINISH 0x10 +#define AC_W_FINISH 0x08 +#define AC_W_RES 0x04 +#define AC_C_RES 0x02 + +/* bit definition of ACTL_RSR register */ +#define R_FIFO_EMPTY 0x04 +#define R_DMA_END_IRQ 0x02 +#define R_DMA_MIDDLE_IRQ 0x01 + +/* bit definition of ACTL_PSR register */ +#define P_FIFO_EMPTY 0x04 +#define P_DMA_END_IRQ 0x02 +#define P_DMA_MIDDLE_IRQ 0x01 + +/* bit definition of ACTL_ACOS0 register */ +#define SLOT1_VALID 0x01 +#define SLOT2_VALID 0x02 +#define SLOT3_VALID 0x04 +#define SLOT4_VALID 0x08 +#define VALID_FRAME 0x10 + +/* bit definition of ACTL_ACOS1 register */ +#define R_WB 0x80 + +#define CODEC_READY 0x10 +#define RESET_PRSR 0x00 +#define AUDIO_WRITE(addr, val) __raw_writel(val, addr) +#define AUDIO_READ(addr) __raw_readl(addr) + +struct nuc900_audio { + void __iomem *mmio; + spinlock_t lock; + dma_addr_t dma_addr[2]; + unsigned long buffersize[2]; + unsigned long irq_num; + struct snd_pcm_substream *substream; + struct resource *res; + struct clk *clk; + struct device *dev; + +}; + +extern struct nuc900_audio *nuc900_ac97_data; +extern struct snd_soc_dai nuc900_ac97_dai; +extern struct snd_soc_platform nuc900_soc_platform; + +#endif /*end _NUC900_AUDIO_H */ diff --git a/sound/soc/nuc900/nuc900-pcm.c b/sound/soc/nuc900/nuc900-pcm.c new file mode 100644 index 000000000000..e81e803b3a63 --- /dev/null +++ b/sound/soc/nuc900/nuc900-pcm.c @@ -0,0 +1,354 @@ +/* + * Copyright (c) 2010 Nuvoton technology corporation. + * + * Wan ZongShun <mcuos.com@gmail.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation;version 2 of the License. + * + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/io.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/dma-mapping.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <mach/hardware.h> + +#include "nuc900-audio.h" + +static const struct snd_pcm_hardware nuc900_pcm_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = 4*1024, + .period_bytes_min = 1*1024, + .period_bytes_max = 4*1024, + .periods_min = 1, + .periods_max = 1024, +}; + +static int nuc900_dma_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct nuc900_audio *nuc900_audio = runtime->private_data; + unsigned long flags; + int ret = 0; + + spin_lock_irqsave(&nuc900_audio->lock, flags); + + ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); + if (ret < 0) + return ret; + + nuc900_audio->substream = substream; + nuc900_audio->dma_addr[substream->stream] = runtime->dma_addr; + nuc900_audio->buffersize[substream->stream] = + params_buffer_bytes(params); + + spin_unlock_irqrestore(&nuc900_audio->lock, flags); + + return ret; +} + +static void nuc900_update_dma_register(struct snd_pcm_substream *substream, + dma_addr_t dma_addr, size_t count) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct nuc900_audio *nuc900_audio = runtime->private_data; + void __iomem *mmio_addr, *mmio_len; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + mmio_addr = nuc900_audio->mmio + ACTL_PDSTB; + mmio_len = nuc900_audio->mmio + ACTL_PDST_LENGTH; + } else { + mmio_addr = nuc900_audio->mmio + ACTL_RDSTB; + mmio_len = nuc900_audio->mmio + ACTL_RDST_LENGTH; + } + + AUDIO_WRITE(mmio_addr, dma_addr); + AUDIO_WRITE(mmio_len, count); +} + +static void nuc900_dma_start(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct nuc900_audio *nuc900_audio = runtime->private_data; + unsigned long val; + + val = AUDIO_READ(nuc900_audio->mmio + ACTL_CON); + val |= (T_DMA_IRQ | R_DMA_IRQ); + AUDIO_WRITE(nuc900_audio->mmio + ACTL_CON, val); +} + +static void nuc900_dma_stop(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct nuc900_audio *nuc900_audio = runtime->private_data; + unsigned long val; + + val = AUDIO_READ(nuc900_audio->mmio + ACTL_CON); + val &= ~(T_DMA_IRQ | R_DMA_IRQ); + AUDIO_WRITE(nuc900_audio->mmio + ACTL_CON, val); +} + +static irqreturn_t nuc900_dma_interrupt(int irq, void *dev_id) +{ + struct snd_pcm_substream *substream = dev_id; + struct nuc900_audio *nuc900_audio = substream->runtime->private_data; + unsigned long val; + + spin_lock(&nuc900_audio->lock); + + val = AUDIO_READ(nuc900_audio->mmio + ACTL_CON); + + if (val & R_DMA_IRQ) { + AUDIO_WRITE(nuc900_audio->mmio + ACTL_CON, val | R_DMA_IRQ); + + val = AUDIO_READ(nuc900_audio->mmio + ACTL_RSR); + + if (val & R_DMA_MIDDLE_IRQ) { + val |= R_DMA_MIDDLE_IRQ; + AUDIO_WRITE(nuc900_audio->mmio + ACTL_RSR, val); + } + + if (val & R_DMA_END_IRQ) { + val |= R_DMA_END_IRQ; + AUDIO_WRITE(nuc900_audio->mmio + ACTL_RSR, val); + } + } else if (val & T_DMA_IRQ) { + AUDIO_WRITE(nuc900_audio->mmio + ACTL_CON, val | T_DMA_IRQ); + + val = AUDIO_READ(nuc900_audio->mmio + ACTL_PSR); + + if (val & P_DMA_MIDDLE_IRQ) { + val |= P_DMA_MIDDLE_IRQ; + AUDIO_WRITE(nuc900_audio->mmio + ACTL_PSR, val); + } + + if (val & P_DMA_END_IRQ) { + val |= P_DMA_END_IRQ; + AUDIO_WRITE(nuc900_audio->mmio + ACTL_PSR, val); + } + } else { + dev_err(nuc900_audio->dev, "Wrong DMA interrupt status!\n"); + spin_unlock(&nuc900_audio->lock); + return IRQ_HANDLED; + } + + spin_unlock(&nuc900_audio->lock); + + snd_pcm_period_elapsed(substream); + + return IRQ_HANDLED; +} + +static int nuc900_dma_hw_free(struct snd_pcm_substream *substream) +{ + snd_pcm_lib_free_pages(substream); + return 0; +} + +static int nuc900_dma_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct nuc900_audio *nuc900_audio = runtime->private_data; + unsigned long flags, val; + + spin_lock_irqsave(&nuc900_audio->lock, flags); + + nuc900_update_dma_register(substream, + nuc900_audio->dma_addr[substream->stream], + nuc900_audio->buffersize[substream->stream]); + + val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET); + + switch (runtime->channels) { + case 1: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + val &= ~(PLAY_LEFT_CHNNEL | PLAY_RIGHT_CHNNEL); + val |= PLAY_RIGHT_CHNNEL; + } else { + val &= ~(RECORD_LEFT_CHNNEL | RECORD_RIGHT_CHNNEL); + val |= RECORD_RIGHT_CHNNEL; + } + AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val); + break; + case 2: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + val |= (PLAY_LEFT_CHNNEL | PLAY_RIGHT_CHNNEL); + else + val |= (RECORD_LEFT_CHNNEL | RECORD_RIGHT_CHNNEL); + AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val); + break; + default: + return -EINVAL; + } + spin_unlock_irqrestore(&nuc900_audio->lock, flags); + return 0; +} + +static int nuc900_dma_trigger(struct snd_pcm_substream *substream, int cmd) +{ + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + nuc900_dma_start(substream); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + nuc900_dma_stop(substream); + break; + + default: + ret = -EINVAL; + break; + } + + return ret; +} + +int nuc900_dma_getposition(struct snd_pcm_substream *substream, + dma_addr_t *src, dma_addr_t *dst) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct nuc900_audio *nuc900_audio = runtime->private_data; + + if (src != NULL) + *src = AUDIO_READ(nuc900_audio->mmio + ACTL_PDSTC); + + if (dst != NULL) + *dst = AUDIO_READ(nuc900_audio->mmio + ACTL_RDSTC); + + return 0; +} + +static snd_pcm_uframes_t nuc900_dma_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + dma_addr_t src, dst; + unsigned long res; + + nuc900_dma_getposition(substream, &src, &dst); + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + res = dst - runtime->dma_addr; + else + res = src - runtime->dma_addr; + + return bytes_to_frames(substream->runtime, res); +} + +static int nuc900_dma_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct nuc900_audio *nuc900_audio; + + snd_soc_set_runtime_hwparams(substream, &nuc900_pcm_hardware); + + nuc900_audio = nuc900_ac97_data; + + if (request_irq(nuc900_audio->irq_num, nuc900_dma_interrupt, + IRQF_DISABLED, "nuc900-dma", substream)) + return -EBUSY; + + runtime->private_data = nuc900_audio; + + return 0; +} + +static int nuc900_dma_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct nuc900_audio *nuc900_audio = runtime->private_data; + + free_irq(nuc900_audio->irq_num, substream); + + return 0; +} + +static int nuc900_dma_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + return dma_mmap_writecombine(substream->pcm->card->dev, vma, + runtime->dma_area, + runtime->dma_addr, + runtime->dma_bytes); +} + +static struct snd_pcm_ops nuc900_dma_ops = { + .open = nuc900_dma_open, + .close = nuc900_dma_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = nuc900_dma_hw_params, + .hw_free = nuc900_dma_hw_free, + .prepare = nuc900_dma_prepare, + .trigger = nuc900_dma_trigger, + .pointer = nuc900_dma_pointer, + .mmap = nuc900_dma_mmap, +}; + +static void nuc900_dma_free_dma_buffers(struct snd_pcm *pcm) +{ + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static u64 nuc900_pcm_dmamask = DMA_BIT_MASK(32); +static int nuc900_dma_new(struct snd_card *card, + struct snd_soc_dai *dai, struct snd_pcm *pcm) +{ + if (!card->dev->dma_mask) + card->dev->dma_mask = &nuc900_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); + + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + card->dev, 4 * 1024, (4 * 1024) - 1); + + return 0; +} + +struct snd_soc_platform nuc900_soc_platform = { + .name = "nuc900-dma", + .pcm_ops = &nuc900_dma_ops, + .pcm_new = nuc900_dma_new, + .pcm_free = nuc900_dma_free_dma_buffers, +} +EXPORT_SYMBOL_GPL(nuc900_soc_platform); + +static int __init nuc900_soc_platform_init(void) +{ + return snd_soc_register_platform(&nuc900_soc_platform); +} + +static void __exit nuc900_soc_platform_exit(void) +{ + snd_soc_unregister_platform(&nuc900_soc_platform); +} + +module_init(nuc900_soc_platform_init); +module_exit(nuc900_soc_platform_exit); + +MODULE_AUTHOR("Wan ZongShun, <mcuos.com@gmail.com>"); +MODULE_DESCRIPTION("nuc900 Audio DMA module"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 6f44cb4d30b8..86f213905e2c 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -59,6 +59,7 @@ struct omap_mcbsp_data { int configured; unsigned int in_freq; int clk_div; + int wlen; }; #define to_mcbsp(priv) container_of((priv), struct omap_mcbsp_data, bus_id) @@ -154,20 +155,51 @@ static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); + struct omap_pcm_dma_data *dma_data; int dma_op_mode = omap_mcbsp_get_dma_op_mode(mcbsp_data->bus_id); - int samples; + int words; + + dma_data = snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); /* TODO: Currently, MODE_ELEMENT == MODE_FRAME */ if (dma_op_mode == MCBSP_DMA_MODE_THRESHOLD) - samples = snd_pcm_lib_period_bytes(substream) >> 1; + /* + * Configure McBSP threshold based on either: + * packet_size, when the sDMA is in packet mode, or + * based on the period size. + */ + if (dma_data->packet_size) + words = dma_data->packet_size; + else + words = snd_pcm_lib_period_bytes(substream) / + (mcbsp_data->wlen / 8); else - samples = 1; + words = 1; /* Configure McBSP internal buffer usage */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - omap_mcbsp_set_tx_threshold(mcbsp_data->bus_id, samples - 1); + omap_mcbsp_set_tx_threshold(mcbsp_data->bus_id, words); else - omap_mcbsp_set_rx_threshold(mcbsp_data->bus_id, samples - 1); + omap_mcbsp_set_rx_threshold(mcbsp_data->bus_id, words); +} + +static int omap_mcbsp_hwrule_min_buffersize(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_interval *buffer_size = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_BUFFER_SIZE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + struct omap_mcbsp_data *mcbsp_data = rule->private; + struct snd_interval frames; + int size; + + snd_interval_any(&frames); + size = omap_mcbsp_get_fifo_size(mcbsp_data->bus_id); + + frames.min = size / channels->min; + frames.integer = 1; + return snd_interval_refine(buffer_size, &frames); } static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream, @@ -182,33 +214,35 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream, if (!cpu_dai->active) err = omap_mcbsp_request(bus_id); + /* + * OMAP3 McBSP FIFO is word structured. + * McBSP2 has 1024 + 256 = 1280 word long buffer, + * McBSP1,3,4,5 has 128 word long buffer + * This means that the size of the FIFO depends on the sample format. + * For example on McBSP3: + * 16bit samples: size is 128 * 2 = 256 bytes + * 32bit samples: size is 128 * 4 = 512 bytes + * It is simpler to place constraint for buffer and period based on + * channels. + * McBSP3 as example again (16 or 32 bit samples): + * 1 channel (mono): size is 128 frames (128 words) + * 2 channels (stereo): size is 128 / 2 = 64 frames (2 * 64 words) + * 4 channels: size is 128 / 4 = 32 frames (4 * 32 words) + */ if (cpu_is_omap343x()) { - int dma_op_mode = omap_mcbsp_get_dma_op_mode(bus_id); - int max_period; - /* - * McBSP2 in OMAP3 has 1024 * 32-bit internal audio buffer. - * Set constraint for minimum buffer size to the same than FIFO - * size in order to avoid underruns in playback startup because - * HW is keeping the DMA request active until FIFO is filled. - */ - if (bus_id == 1) - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_BUFFER_BYTES, - 4096, UINT_MAX); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - max_period = omap_mcbsp_get_max_tx_threshold(bus_id); - else - max_period = omap_mcbsp_get_max_rx_threshold(bus_id); - - max_period++; - max_period <<= 1; - - if (dma_op_mode == MCBSP_DMA_MODE_THRESHOLD) - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_PERIOD_BYTES, - 32, max_period); + * Rule for the buffer size. We should not allow + * smaller buffer than the FIFO size to avoid underruns + */ + snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + omap_mcbsp_hwrule_min_buffersize, + mcbsp_data, + SNDRV_PCM_HW_PARAM_BUFFER_SIZE, -1); + + /* Make sure, that the period size is always even */ + snd_pcm_hw_constraint_step(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 2); } return err; @@ -289,11 +323,14 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; - int dma, bus_id = mcbsp_data->bus_id, id = cpu_dai->id; + struct omap_pcm_dma_data *dma_data; + int dma, bus_id = mcbsp_data->bus_id; int wlen, channels, wpf, sync_mode = OMAP_DMA_SYNC_ELEMENT; + int pkt_size = 0; unsigned long port; unsigned int format, div, framesize, master; + dma_data = &omap_mcbsp_dai_dma_params[cpu_dai->id][substream->stream]; if (cpu_class_is_omap1()) { dma = omap1_dma_reqs[bus_id][substream->stream]; port = omap1_mcbsp_port[bus_id][substream->stream]; @@ -306,35 +343,74 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, } else if (cpu_is_omap343x()) { dma = omap24xx_dma_reqs[bus_id][substream->stream]; port = omap34xx_mcbsp_port[bus_id][substream->stream]; - omap_mcbsp_dai_dma_params[id][substream->stream].set_threshold = - omap_mcbsp_set_threshold; - /* TODO: Currently, MODE_ELEMENT == MODE_FRAME */ - if (omap_mcbsp_get_dma_op_mode(bus_id) == - MCBSP_DMA_MODE_THRESHOLD) - sync_mode = OMAP_DMA_SYNC_FRAME; } else { return -ENODEV; } - omap_mcbsp_dai_dma_params[id][substream->stream].name = - substream->stream ? "Audio Capture" : "Audio Playback"; - omap_mcbsp_dai_dma_params[id][substream->stream].dma_req = dma; - omap_mcbsp_dai_dma_params[id][substream->stream].port_addr = port; - omap_mcbsp_dai_dma_params[id][substream->stream].sync_mode = sync_mode; switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: - omap_mcbsp_dai_dma_params[id][substream->stream].data_type = - OMAP_DMA_DATA_TYPE_S16; + dma_data->data_type = OMAP_DMA_DATA_TYPE_S16; + wlen = 16; break; case SNDRV_PCM_FORMAT_S32_LE: - omap_mcbsp_dai_dma_params[id][substream->stream].data_type = - OMAP_DMA_DATA_TYPE_S32; + dma_data->data_type = OMAP_DMA_DATA_TYPE_S32; + wlen = 32; break; default: return -EINVAL; } + if (cpu_is_omap343x()) { + dma_data->set_threshold = omap_mcbsp_set_threshold; + /* TODO: Currently, MODE_ELEMENT == MODE_FRAME */ + if (omap_mcbsp_get_dma_op_mode(bus_id) == + MCBSP_DMA_MODE_THRESHOLD) { + int period_words, max_thrsh; + + period_words = params_period_bytes(params) / (wlen / 8); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + max_thrsh = omap_mcbsp_get_max_tx_threshold( + mcbsp_data->bus_id); + else + max_thrsh = omap_mcbsp_get_max_rx_threshold( + mcbsp_data->bus_id); + /* + * If the period contains less or equal number of words, + * we are using the original threshold mode setup: + * McBSP threshold = sDMA frame size = period_size + * Otherwise we switch to sDMA packet mode: + * McBSP threshold = sDMA packet size + * sDMA frame size = period size + */ + if (period_words > max_thrsh) { + int divider = 0; + + /* + * Look for the biggest threshold value, which + * divides the period size evenly. + */ + divider = period_words / max_thrsh; + if (period_words % max_thrsh) + divider++; + while (period_words % divider && + divider < period_words) + divider++; + if (divider == period_words) + return -EINVAL; + + pkt_size = period_words / divider; + sync_mode = OMAP_DMA_SYNC_PACKET; + } else { + sync_mode = OMAP_DMA_SYNC_FRAME; + } + } + } - snd_soc_dai_set_dma_data(cpu_dai, substream, - &omap_mcbsp_dai_dma_params[id][substream->stream]); + dma_data->name = substream->stream ? "Audio Capture" : "Audio Playback"; + dma_data->dma_req = dma; + dma_data->port_addr = port; + dma_data->sync_mode = sync_mode; + dma_data->packet_size = pkt_size; + + snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); if (mcbsp_data->configured) { /* McBSP already configured by another stream */ @@ -360,7 +436,6 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: /* Set word lengths */ - wlen = 16; regs->rcr2 |= RWDLEN2(OMAP_MCBSP_WORD_16); regs->rcr1 |= RWDLEN1(OMAP_MCBSP_WORD_16); regs->xcr2 |= XWDLEN2(OMAP_MCBSP_WORD_16); @@ -368,7 +443,6 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, break; case SNDRV_PCM_FORMAT_S32_LE: /* Set word lengths */ - wlen = 32; regs->rcr2 |= RWDLEN2(OMAP_MCBSP_WORD_32); regs->rcr1 |= RWDLEN1(OMAP_MCBSP_WORD_32); regs->xcr2 |= XWDLEN2(OMAP_MCBSP_WORD_32); @@ -409,6 +483,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, } omap_mcbsp_config(bus_id, &mcbsp_data->regs); + mcbsp_data->wlen = wlen; mcbsp_data->configured = 1; return 0; diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index 87ce842fa2e8..9eecac135bbb 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -43,12 +43,14 @@ static struct regulator *omap3pandora_dac_reg; -static int omap3pandora_cmn_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, unsigned int fmt) +static int omap3pandora_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS; int ret; /* Set codec DAI configuration */ @@ -91,24 +93,6 @@ static int omap3pandora_cmn_hw_params(struct snd_pcm_substream *substream, return 0; } -static int omap3pandora_out_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - return omap3pandora_cmn_hw_params(substream, params, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_IB_NF | - SND_SOC_DAIFMT_CBS_CFS); -} - -static int omap3pandora_in_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - return omap3pandora_cmn_hw_params(substream, params, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); -} - static int omap3pandora_dac_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { @@ -231,12 +215,8 @@ static int omap3pandora_in_init(struct snd_soc_codec *codec) return snd_soc_dapm_sync(codec); } -static struct snd_soc_ops omap3pandora_out_ops = { - .hw_params = omap3pandora_out_hw_params, -}; - -static struct snd_soc_ops omap3pandora_in_ops = { - .hw_params = omap3pandora_in_hw_params, +static struct snd_soc_ops omap3pandora_ops = { + .hw_params = omap3pandora_hw_params, }; /* Digital audio interface glue - connects codec <--> CPU */ @@ -246,14 +226,14 @@ static struct snd_soc_dai_link omap3pandora_dai[] = { .stream_name = "HiFi Out", .cpu_dai = &omap_mcbsp_dai[0], .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], - .ops = &omap3pandora_out_ops, + .ops = &omap3pandora_ops, .init = omap3pandora_out_init, }, { .name = "TWL4030", .stream_name = "Line/Mic In", .cpu_dai = &omap_mcbsp_dai[1], .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], - .ops = &omap3pandora_in_ops, + .ops = &omap3pandora_ops, .init = omap3pandora_in_init, } }; diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 47d831ef2dbb..88052d29617f 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -27,6 +27,7 @@ #include <linux/gpio.h> #include <linux/platform_device.h> #include <sound/core.h> +#include <sound/jack.h> #include <sound/pcm.h> #include <sound/soc.h> #include <sound/soc-dapm.h> @@ -37,14 +38,22 @@ #include "omap-pcm.h" #include "../codecs/tlv320aic3x.h" +#define RX51_TVOUT_SEL_GPIO 40 +#define RX51_JACK_DETECT_GPIO 177 /* * REVISIT: TWL4030 GPIO base in RX-51. Now statically defined to 192. This * gpio is reserved in arch/arm/mach-omap2/board-rx51-peripherals.c */ #define RX51_SPEAKER_AMP_TWL_GPIO (192 + 7) +enum { + RX51_JACK_DISABLED, + RX51_JACK_TVOUT, /* tv-out */ +}; + static int rx51_spk_func; static int rx51_dmic_func; +static int rx51_jack_func; static void rx51_ext_control(struct snd_soc_codec *codec) { @@ -57,6 +66,9 @@ static void rx51_ext_control(struct snd_soc_codec *codec) else snd_soc_dapm_disable_pin(codec, "DMic"); + gpio_set_value(RX51_TVOUT_SEL_GPIO, + rx51_jack_func == RX51_JACK_TVOUT); + snd_soc_dapm_sync(codec); } @@ -162,6 +174,40 @@ static int rx51_set_input(struct snd_kcontrol *kcontrol, return 1; } +static int rx51_get_jack(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = rx51_jack_func; + + return 0; +} + +static int rx51_set_jack(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (rx51_jack_func == ucontrol->value.integer.value[0]) + return 0; + + rx51_jack_func = ucontrol->value.integer.value[0]; + rx51_ext_control(codec); + + return 1; +} + +static struct snd_soc_jack rx51_av_jack; + +static struct snd_soc_jack_gpio rx51_av_jack_gpios[] = { + { + .gpio = RX51_JACK_DETECT_GPIO, + .name = "avdet-gpio", + .report = SND_JACK_VIDEOOUT, + .invert = 1, + .debounce_time = 200, + }, +}; + static const struct snd_soc_dapm_widget aic34_dapm_widgets[] = { SND_SOC_DAPM_SPK("Ext Spk", rx51_spk_event), SND_SOC_DAPM_MIC("DMic", NULL), @@ -177,10 +223,12 @@ static const struct snd_soc_dapm_route audio_map[] = { static const char *spk_function[] = {"Off", "On"}; static const char *input_function[] = {"ADC", "Digital Mic"}; +static const char *jack_function[] = {"Off", "TV-OUT"}; static const struct soc_enum rx51_enum[] = { SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(spk_function), spk_function), SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(input_function), input_function), + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(jack_function), jack_function), }; static const struct snd_kcontrol_new aic34_rx51_controls[] = { @@ -188,10 +236,13 @@ static const struct snd_kcontrol_new aic34_rx51_controls[] = { rx51_get_spk, rx51_set_spk), SOC_ENUM_EXT("Input Select", rx51_enum[1], rx51_get_input, rx51_set_input), + SOC_ENUM_EXT("Jack Function", rx51_enum[2], + rx51_get_jack, rx51_set_jack), }; static int rx51_aic34_init(struct snd_soc_codec *codec) { + struct snd_soc_card *card = codec->socdev->card; int err; /* Set up NC codec pins */ @@ -214,7 +265,16 @@ static int rx51_aic34_init(struct snd_soc_codec *codec) snd_soc_dapm_sync(codec); - return 0; + /* AV jack detection */ + err = snd_soc_jack_new(card, "AV Jack", + SND_JACK_VIDEOOUT, &rx51_av_jack); + if (err) + return err; + err = snd_soc_jack_add_gpios(&rx51_av_jack, + ARRAY_SIZE(rx51_av_jack_gpios), + rx51_av_jack_gpios); + + return err; } /* Digital audio interface glue - connects codec <--> CPU */ @@ -259,6 +319,11 @@ static int __init rx51_soc_init(void) if (!machine_is_nokia_rx51()) return -ENODEV; + err = gpio_request(RX51_TVOUT_SEL_GPIO, "tvout_sel"); + if (err) + goto err_gpio_tvout_sel; + gpio_direction_output(RX51_TVOUT_SEL_GPIO, 0); + rx51_snd_device = platform_device_alloc("soc-audio", -1); if (!rx51_snd_device) { err = -ENOMEM; @@ -277,13 +342,19 @@ static int __init rx51_soc_init(void) err2: platform_device_put(rx51_snd_device); err1: + gpio_free(RX51_TVOUT_SEL_GPIO); +err_gpio_tvout_sel: return err; } static void __exit rx51_soc_exit(void) { + snd_soc_jack_free_gpios(&rx51_av_jack, ARRAY_SIZE(rx51_av_jack_gpios), + rx51_av_jack_gpios); + platform_device_unregister(rx51_snd_device); + gpio_free(RX51_TVOUT_SEL_GPIO); } module_init(rx51_soc_init); diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 2a7cc222d098..213963ac3c28 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -1,6 +1,6 @@ config SND_S3C24XX_SOC tristate "SoC Audio for the Samsung S3CXXXX chips" - depends on ARCH_S3C2410 || ARCH_S3C64XX + depends on ARCH_S3C2410 || ARCH_S3C64XX || ARCH_S5PC100 || ARCH_S5PV210 select S3C64XX_DMA if ARCH_S3C64XX help Say Y or M if you want to add support for codecs attached to @@ -120,8 +120,14 @@ config SND_S3C24XX_SOC_SIMTEC_HERMES config SND_SOC_SMDK_WM9713 tristate "SoC AC97 Audio support for SMDK with WM9713" - depends on SND_S3C24XX_SOC && MACH_SMDK6410 + depends on SND_S3C24XX_SOC && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110) select SND_SOC_WM9713 select SND_S3C_SOC_AC97 help Sat Y if you want to add support for SoC audio on the SMDK. + +config SND_S3C64XX_SOC_SMARTQ + tristate "SoC I2S Audio support for SmartQ board" + depends on SND_S3C24XX_SOC && MACH_SMARTQ + select SND_S3C64XX_SOC_I2S + select SND_SOC_WM8750 diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 81d8dc503f87..50172c385d90 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -29,6 +29,7 @@ snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o snd-soc-smdk-wm9713-objs := smdk_wm9713.o +snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -41,3 +42,4 @@ obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o obj-$(CONFIG_SND_SOC_SMDK_WM9713) += snd-soc-smdk-wm9713.o +obj-$(CONFIG_SND_S3C64XX_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o diff --git a/sound/soc/s3c24xx/neo1973_gta02_wm8753.c b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c index 209c25994c7e..4719558289d4 100644 --- a/sound/soc/s3c24xx/neo1973_gta02_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c @@ -182,7 +182,7 @@ static int neo1973_gta02_voice_hw_params( if (ret < 0) return ret; - /* configue and enable PLL for 12.288MHz output */ + /* configure and enable PLL for 12.288MHz output */ ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, iis_clkrate / 4, 12288000); if (ret < 0) diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 0cb4f86f6d1e..4ac620988e7c 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -201,7 +201,7 @@ static int neo1973_voice_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - /* configue and enable PLL for 12.288MHz output */ + /* configure and enable PLL for 12.288MHz output */ ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, iis_clkrate / 4, 12288000); if (ret < 0) diff --git a/sound/soc/s3c24xx/s3c-ac97.c b/sound/soc/s3c24xx/s3c-ac97.c index ecf4fd04ae96..31f6d45b6384 100644 --- a/sound/soc/s3c24xx/s3c-ac97.c +++ b/sound/soc/s3c24xx/s3c-ac97.c @@ -31,7 +31,6 @@ #define AC_CMD_DATA(x) (x & 0xffff) struct s3c_ac97_info { - unsigned state; struct clk *ac97_clk; void __iomem *regs; struct mutex lock; diff --git a/sound/soc/s3c24xx/s3c-dma.c b/sound/soc/s3c24xx/s3c-dma.c index 1b61c23ff300..f1b1bc4bacfb 100644 --- a/sound/soc/s3c24xx/s3c-dma.c +++ b/sound/soc/s3c24xx/s3c-dma.c @@ -94,8 +94,7 @@ static void s3c_dma_enqueue(struct snd_pcm_substream *substream) if ((pos + len) > prtd->dma_end) { len = prtd->dma_end - pos; - pr_debug(KERN_DEBUG "%s: corrected dma len %ld\n", - __func__, len); + pr_debug("%s: corrected dma len %ld\n", __func__, len); } ret = s3c2410_dma_enqueue(prtd->params->channel, diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index 13311c8cf965..64376b2aac73 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -32,7 +32,8 @@ #undef S3C_IIS_V2_SUPPORTED -#if defined(CONFIG_CPU_S3C2412) || defined(CONFIG_CPU_S3C2413) +#if defined(CONFIG_CPU_S3C2412) || defined(CONFIG_CPU_S3C2413) \ + || defined(CONFIG_CPU_S5PV210) #define S3C_IIS_V2_SUPPORTED #endif diff --git a/sound/soc/s3c24xx/smartq_wm8987.c b/sound/soc/s3c24xx/smartq_wm8987.c new file mode 100644 index 000000000000..b480348140b0 --- /dev/null +++ b/sound/soc/s3c24xx/smartq_wm8987.c @@ -0,0 +1,295 @@ +/* sound/soc/s3c24xx/smartq_wm8987.c + * + * Copyright 2010 Maurus Cuelenaere <mcuelenaere@gmail.com> + * + * Based on smdk6410_wm8987.c + * Copyright 2007 Wolfson Microelectronics PLC. - linux@wolfsonmicro.com + * Graeme Gregory - graeme.gregory@wolfsonmicro.com + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/gpio.h> + +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc-dapm.h> +#include <sound/jack.h> + +#include <asm/mach-types.h> + +#include "s3c-dma.h" +#include "s3c64xx-i2s.h" + +#include "../codecs/wm8750.h" + +/* + * WM8987 is register compatible with WM8750, so using that as base driver. + */ + +static struct snd_soc_card snd_soc_smartq; + +static int smartq_hifi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct s3c_i2sv2_rate_calc div; + unsigned int clk = 0; + int ret; + + s3c_i2sv2_iis_calc_rate(&div, NULL, params_rate(params), + s3c_i2sv2_get_clock(cpu_dai)); + + switch (params_rate(params)) { + case 8000: + case 16000: + case 32000: + case 48000: + case 96000: + clk = 12288000; + break; + case 11025: + case 22050: + case 44100: + case 88200: + clk = 11289600; + break; + } + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8750_SYSCLK, clk, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* set MCLK division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_RCLK, div.fs_div); + if (ret < 0) + return ret; + + /* set prescaler division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_PRESCALER, + div.clk_div - 1); + if (ret < 0) + return ret; + + return 0; +} + +/* + * SmartQ WM8987 HiFi DAI operations. + */ +static struct snd_soc_ops smartq_hifi_ops = { + .hw_params = smartq_hifi_hw_params, +}; + +static struct snd_soc_jack smartq_jack; + +static struct snd_soc_jack_pin smartq_jack_pins[] = { + /* Disable speaker when headphone is plugged in */ + { + .pin = "Internal Speaker", + .mask = SND_JACK_HEADPHONE, + }, +}; + +static struct snd_soc_jack_gpio smartq_jack_gpios[] = { + { + .gpio = S3C64XX_GPL(12), + .name = "headphone detect", + .report = SND_JACK_HEADPHONE, + .debounce_time = 200, + }, +}; + +static const struct snd_kcontrol_new wm8987_smartq_controls[] = { + SOC_DAPM_PIN_SWITCH("Internal Speaker"), + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Internal Mic"), +}; + +static int smartq_speaker_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, + int event) +{ + gpio_set_value(S3C64XX_GPK(12), SND_SOC_DAPM_EVENT_OFF(event)); + + return 0; +} + +static const struct snd_soc_dapm_widget wm8987_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Internal Speaker", smartq_speaker_event), + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Internal Mic", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + {"Headphone Jack", NULL, "LOUT2"}, + {"Headphone Jack", NULL, "ROUT2"}, + + {"Internal Speaker", NULL, "LOUT2"}, + {"Internal Speaker", NULL, "ROUT2"}, + + {"Mic Bias", NULL, "Internal Mic"}, + {"LINPUT2", NULL, "Mic Bias"}, +}; + +static int smartq_wm8987_init(struct snd_soc_codec *codec) +{ + int err = 0; + + /* Add SmartQ specific widgets */ + snd_soc_dapm_new_controls(codec, wm8987_dapm_widgets, + ARRAY_SIZE(wm8987_dapm_widgets)); + + /* add SmartQ specific controls */ + err = snd_soc_add_controls(codec, wm8987_smartq_controls, + ARRAY_SIZE(wm8987_smartq_controls)); + + if (err < 0) + return err; + + /* setup SmartQ specific audio path */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + /* set endpoints to not connected */ + snd_soc_dapm_nc_pin(codec, "LINPUT1"); + snd_soc_dapm_nc_pin(codec, "RINPUT1"); + snd_soc_dapm_nc_pin(codec, "OUT3"); + snd_soc_dapm_nc_pin(codec, "ROUT1"); + + /* set endpoints to default off mode */ + snd_soc_dapm_enable_pin(codec, "Internal Speaker"); + snd_soc_dapm_enable_pin(codec, "Internal Mic"); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + + err = snd_soc_dapm_sync(codec); + if (err) + return err; + + /* Headphone jack detection */ + err = snd_soc_jack_new(&snd_soc_smartq, "Headphone Jack", + SND_JACK_HEADPHONE, &smartq_jack); + if (err) + return err; + + err = snd_soc_jack_add_pins(&smartq_jack, ARRAY_SIZE(smartq_jack_pins), + smartq_jack_pins); + if (err) + return err; + + err = snd_soc_jack_add_gpios(&smartq_jack, + ARRAY_SIZE(smartq_jack_gpios), + smartq_jack_gpios); + + return err; +} + +static struct snd_soc_dai_link smartq_dai[] = { + { + .name = "wm8987", + .stream_name = "SmartQ Hi-Fi", + .cpu_dai = &s3c64xx_i2s_dai[0], + .codec_dai = &wm8750_dai, + .init = smartq_wm8987_init, + .ops = &smartq_hifi_ops, + }, +}; + +static struct snd_soc_card snd_soc_smartq = { + .name = "SmartQ", + .platform = &s3c24xx_soc_platform, + .dai_link = smartq_dai, + .num_links = ARRAY_SIZE(smartq_dai), +}; + +static struct snd_soc_device smartq_snd_devdata = { + .card = &snd_soc_smartq, + .codec_dev = &soc_codec_dev_wm8750, +}; + +static struct platform_device *smartq_snd_device; + +static int __init smartq_init(void) +{ + int ret; + + if (!machine_is_smartq7() && !machine_is_smartq5()) { + pr_info("Only SmartQ is supported by this ASoC driver\n"); + return -ENODEV; + } + + smartq_snd_device = platform_device_alloc("soc-audio", -1); + if (!smartq_snd_device) + return -ENOMEM; + + platform_set_drvdata(smartq_snd_device, &smartq_snd_devdata); + smartq_snd_devdata.dev = &smartq_snd_device->dev; + + ret = platform_device_add(smartq_snd_device); + if (ret) { + platform_device_put(smartq_snd_device); + return ret; + } + + /* Initialise GPIOs used by amplifiers */ + ret = gpio_request(S3C64XX_GPK(12), "amplifiers shutdown"); + if (ret) { + dev_err(&smartq_snd_device->dev, "Failed to register GPK12\n"); + goto err_unregister_device; + } + + /* Disable amplifiers */ + ret = gpio_direction_output(S3C64XX_GPK(12), 1); + if (ret) { + dev_err(&smartq_snd_device->dev, "Failed to configure GPK12\n"); + goto err_free_gpio_amp_shut; + } + + return 0; + +err_free_gpio_amp_shut: + gpio_free(S3C64XX_GPK(12)); +err_unregister_device: + platform_device_unregister(smartq_snd_device); + + return ret; +} + +static void __exit smartq_exit(void) +{ + snd_soc_jack_free_gpios(&smartq_jack, ARRAY_SIZE(smartq_jack_gpios), + smartq_jack_gpios); + + platform_device_unregister(smartq_snd_device); +} + +module_init(smartq_init); +module_exit(smartq_exit); + +/* Module information */ +MODULE_AUTHOR("Maurus Cuelenaere <mcuelenaere@gmail.com>"); +MODULE_DESCRIPTION("ALSA SoC SmartQ WM8987"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/smdk_wm9713.c b/sound/soc/s3c24xx/smdk_wm9713.c index 24fd39f38ccb..5527b9e88c98 100644 --- a/sound/soc/s3c24xx/smdk_wm9713.c +++ b/sound/soc/s3c24xx/smdk_wm9713.c @@ -25,6 +25,9 @@ static struct snd_soc_card smdk; * Default CFG switch settings to use this driver: * * SMDK6410: Set CFG1 1-3 On, CFG2 1-4 Off + * SMDKC100: Set CFG6 1-3 On, CFG7 1 On + * SMDKC110: Set CFGB10 1-2 Off, CFGB12 1-3 On + * SMDKV210: Set CFGB10 1-2 Off, CFGB12 1-3 On */ /* diff --git a/sound/soc/s6000/s6000-i2s.c b/sound/soc/s6000/s6000-i2s.c index 5b9ac1759bd2..59e3fa7bcb05 100644 --- a/sound/soc/s6000/s6000-i2s.c +++ b/sound/soc/s6000/s6000-i2s.c @@ -451,16 +451,15 @@ static int __devinit s6000_i2s_probe(struct platform_device *pdev) goto err_release_none; } - region = request_mem_region(scbmem->start, - scbmem->end - scbmem->start + 1, - pdev->name); + region = request_mem_region(scbmem->start, resource_size(scbmem), + pdev->name); if (!region) { dev_err(&pdev->dev, "I2S SCB region already claimed\n"); ret = -EBUSY; goto err_release_none; } - mmio = ioremap(scbmem->start, scbmem->end - scbmem->start + 1); + mmio = ioremap(scbmem->start, resource_size(scbmem)); if (!mmio) { dev_err(&pdev->dev, "can't ioremap SCB region\n"); ret = -ENOMEM; @@ -474,9 +473,8 @@ static int __devinit s6000_i2s_probe(struct platform_device *pdev) goto err_release_map; } - region = request_mem_region(sifmem->start, - sifmem->end - sifmem->start + 1, - pdev->name); + region = request_mem_region(sifmem->start, resource_size(sifmem), + pdev->name); if (!region) { dev_err(&pdev->dev, "I2S SIF region already claimed\n"); ret = -EBUSY; @@ -490,8 +488,8 @@ static int __devinit s6000_i2s_probe(struct platform_device *pdev) goto err_release_sif; } - region = request_mem_region(dma1->start, dma1->end - dma1->start + 1, - pdev->name); + region = request_mem_region(dma1->start, resource_size(dma1), + pdev->name); if (!region) { dev_err(&pdev->dev, "I2S DMA region already claimed\n"); ret = -EBUSY; @@ -500,9 +498,8 @@ static int __devinit s6000_i2s_probe(struct platform_device *pdev) dma2 = platform_get_resource(pdev, IORESOURCE_DMA, 1); if (dma2) { - region = request_mem_region(dma2->start, - dma2->end - dma2->start + 1, - pdev->name); + region = request_mem_region(dma2->start, resource_size(dma2), + pdev->name); if (!region) { dev_err(&pdev->dev, "I2S DMA region already claimed\n"); @@ -561,15 +558,15 @@ err_release_dev: kfree(dev); err_release_dma2: if (dma2) - release_mem_region(dma2->start, dma2->end - dma2->start + 1); + release_mem_region(dma2->start, resource_size(dma2)); err_release_dma1: - release_mem_region(dma1->start, dma1->end - dma1->start + 1); + release_mem_region(dma1->start, resource_size(dma1)); err_release_sif: - release_mem_region(sifmem->start, (sifmem->end - sifmem->start) + 1); + release_mem_region(sifmem->start, resource_size(sifmem)); err_release_map: iounmap(mmio); err_release_scb: - release_mem_region(scbmem->start, (scbmem->end - scbmem->start) + 1); + release_mem_region(scbmem->start, resource_size(scbmem)); err_release_none: return ret; } @@ -590,19 +587,18 @@ static void __devexit s6000_i2s_remove(struct platform_device *pdev) kfree(dev); region = platform_get_resource(pdev, IORESOURCE_DMA, 0); - release_mem_region(region->start, region->end - region->start + 1); + release_mem_region(region->start, resource_size(region)); region = platform_get_resource(pdev, IORESOURCE_DMA, 1); if (region) - release_mem_region(region->start, - region->end - region->start + 1); + release_mem_region(region->start, resource_size(region)); region = platform_get_resource(pdev, IORESOURCE_MEM, 0); - release_mem_region(region->start, (region->end - region->start) + 1); + release_mem_region(region->start, resource_size(region)); iounmap(mmio); region = platform_get_resource(pdev, IORESOURCE_IO, 0); - release_mem_region(region->start, (region->end - region->start) + 1); + release_mem_region(region->start, resource_size(region)); } static struct platform_driver s6000_i2s_driver = { diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index a1d14bc3c76f..52d7e8ed9c1f 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -48,7 +48,7 @@ config SND_SH7760_AC97 config SND_FSI_AK4642 bool "FSI-AK4642 sound support" - depends on SND_SOC_SH4_FSI + depends on SND_SOC_SH4_FSI && I2C_SH_MOBILE select SND_SOC_AK4642 help This option enables generic sound support for the @@ -56,7 +56,7 @@ config SND_FSI_AK4642 config SND_FSI_DA7210 bool "FSI-DA7210 sound support" - depends on SND_SOC_SH4_FSI + depends on SND_SOC_SH4_FSI && I2C_SH_MOBILE select SND_SOC_DA7210 help This option enables generic sound support for the diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c index be018542314e..dad575a22622 100644 --- a/sound/soc/sh/fsi-ak4642.c +++ b/sound/soc/sh/fsi-ak4642.c @@ -9,16 +9,7 @@ * for more details. */ -#include <linux/module.h> -#include <linux/moduleparam.h> #include <linux/platform_device.h> -#include <linux/i2c.h> -#include <linux/io.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> - #include <sound/sh_fsi.h> #include <../sound/soc/codecs/ak4642.h> @@ -38,7 +29,7 @@ static int fsi_ak4642_dai_init(struct snd_soc_codec *codec) static struct snd_soc_dai_link fsi_dai_link = { .name = "AK4642", .stream_name = "AK4642", - .cpu_dai = &fsi_soc_dai[0], /* fsi */ + .cpu_dai = &fsi_soc_dai[FSI_PORT_A], .codec_dai = &ak4642_dai, .init = fsi_ak4642_dai_init, .ops = NULL, @@ -62,7 +53,7 @@ static int __init fsi_ak4642_init(void) { int ret = -ENOMEM; - fsi_snd_device = platform_device_alloc("soc-audio", -1); + fsi_snd_device = platform_device_alloc("soc-audio", FSI_PORT_A); if (!fsi_snd_device) goto out; diff --git a/sound/soc/sh/fsi-da7210.c b/sound/soc/sh/fsi-da7210.c index 33b4d177f466..121bbb07bb03 100644 --- a/sound/soc/sh/fsi-da7210.c +++ b/sound/soc/sh/fsi-da7210.c @@ -10,16 +10,7 @@ * option) any later version. */ -#include <linux/interrupt.h> #include <linux/platform_device.h> -#include <linux/io.h> -#include <linux/i2c.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> - #include <sound/sh_fsi.h> #include "../codecs/da7210.h" @@ -33,7 +24,7 @@ static int fsi_da7210_init(struct snd_soc_codec *codec) static struct snd_soc_dai_link fsi_da7210_dai = { .name = "DA7210", .stream_name = "DA7210", - .cpu_dai = &fsi_soc_dai[1], /* FSI B */ + .cpu_dai = &fsi_soc_dai[FSI_PORT_B], .codec_dai = &da7210_dai, .init = fsi_da7210_init, }; @@ -56,7 +47,7 @@ static int __init fsi_da7210_sound_init(void) { int ret; - fsi_da7210_snd_device = platform_device_alloc("soc-audio", -1); + fsi_da7210_snd_device = platform_device_alloc("soc-audio", FSI_PORT_B); if (!fsi_da7210_snd_device) return -ENOMEM; diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index ec4acac49ebd..58c6bec642de 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -12,21 +12,12 @@ * published by the Free Software Foundation. */ -#include <linux/init.h> -#include <linux/module.h> -#include <linux/platform_device.h> #include <linux/delay.h> -#include <linux/list.h> #include <linux/pm_runtime.h> #include <linux/io.h> #include <linux/slab.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/initval.h> #include <sound/soc.h> -#include <sound/pcm_params.h> #include <sound/sh_fsi.h> -#include <asm/atomic.h> #define DO_FMT 0x0000 #define DOFF_CTL 0x0004 @@ -39,9 +30,11 @@ #define DIDT 0x0020 #define DODT 0x0024 #define MUTE_ST 0x0028 -#define REG_END MUTE_ST - +#define OUT_SEL 0x0030 +#define REG_END OUT_SEL +#define A_MST_CTLR 0x0180 +#define B_MST_CTLR 0x01A0 #define CPU_INT_ST 0x01F4 #define CPU_IEMSK 0x01F8 #define CPU_IMSK 0x01FC @@ -52,18 +45,18 @@ #define CLK_RST 0x0210 #define SOFT_RST 0x0214 #define FIFO_SZ 0x0218 -#define MREG_START CPU_INT_ST +#define MREG_START A_MST_CTLR #define MREG_END FIFO_SZ /* DO_FMT */ /* DI_FMT */ -#define CR_FMT(param) ((param) << 4) -# define CR_MONO 0x0 -# define CR_MONO_D 0x1 -# define CR_PCM 0x2 -# define CR_I2S 0x3 -# define CR_TDM 0x4 -# define CR_TDM_D 0x5 +#define CR_MONO (0x0 << 4) +#define CR_MONO_D (0x1 << 4) +#define CR_PCM (0x2 << 4) +#define CR_I2S (0x3 << 4) +#define CR_TDM (0x4 << 4) +#define CR_TDM_D (0x5 << 4) +#define CR_SPDIF 0x00100120 /* DOFF_CTL */ /* DIFF_CTL */ @@ -75,6 +68,14 @@ #define ERR_UNDER 0x00000001 #define ST_ERR (ERR_OVER | ERR_UNDER) +/* CKG1 */ +#define ACKMD_MASK 0x00007000 +#define BPFMD_MASK 0x00000700 + +/* A/B MST_CTLR */ +#define BP (1 << 4) /* Fix the signal of Biphase output */ +#define SE (1 << 0) /* Fix the master clock */ + /* CLK_RST */ #define B_CLK 0x00000010 #define A_CLK 0x00000001 @@ -119,9 +120,13 @@ struct fsi_priv { int period_len; int buffer_len; int periods; + + u32 mst_ctrl; }; -struct fsi_regs { +struct fsi_core { + int ver; + u32 int_st; u32 iemsk; u32 imsk; @@ -132,7 +137,7 @@ struct fsi_master { int irq; struct fsi_priv fsia; struct fsi_priv fsib; - struct fsi_regs *regs; + struct fsi_core *core; struct sh_fsi_platform_info *info; spinlock_t lock; }; @@ -169,24 +174,30 @@ static void __fsi_reg_mask_set(u32 reg, u32 mask, u32 data) static void fsi_reg_write(struct fsi_priv *fsi, u32 reg, u32 data) { - if (reg > REG_END) + if (reg > REG_END) { + pr_err("fsi: register access err (%s)\n", __func__); return; + } __fsi_reg_write((u32)(fsi->base + reg), data); } static u32 fsi_reg_read(struct fsi_priv *fsi, u32 reg) { - if (reg > REG_END) + if (reg > REG_END) { + pr_err("fsi: register access err (%s)\n", __func__); return 0; + } return __fsi_reg_read((u32)(fsi->base + reg)); } static void fsi_reg_mask_set(struct fsi_priv *fsi, u32 reg, u32 mask, u32 data) { - if (reg > REG_END) + if (reg > REG_END) { + pr_err("fsi: register access err (%s)\n", __func__); return; + } __fsi_reg_mask_set((u32)(fsi->base + reg), mask, data); } @@ -196,8 +207,10 @@ static void fsi_master_write(struct fsi_master *master, u32 reg, u32 data) unsigned long flags; if ((reg < MREG_START) || - (reg > MREG_END)) + (reg > MREG_END)) { + pr_err("fsi: register access err (%s)\n", __func__); return; + } spin_lock_irqsave(&master->lock, flags); __fsi_reg_write((u32)(master->base + reg), data); @@ -210,8 +223,10 @@ static u32 fsi_master_read(struct fsi_master *master, u32 reg) unsigned long flags; if ((reg < MREG_START) || - (reg > MREG_END)) + (reg > MREG_END)) { + pr_err("fsi: register access err (%s)\n", __func__); return 0; + } spin_lock_irqsave(&master->lock, flags); ret = __fsi_reg_read((u32)(master->base + reg)); @@ -226,8 +241,10 @@ static void fsi_master_mask_set(struct fsi_master *master, unsigned long flags; if ((reg < MREG_START) || - (reg > MREG_END)) + (reg > MREG_END)) { + pr_err("fsi: register access err (%s)\n", __func__); return; + } spin_lock_irqsave(&master->lock, flags); __fsi_reg_mask_set((u32)(master->base + reg), mask, data); @@ -349,8 +366,8 @@ static void fsi_irq_enable(struct fsi_priv *fsi, int is_play) u32 data = fsi_port_ab_io_bit(fsi, is_play); struct fsi_master *master = fsi_get_master(fsi); - fsi_master_mask_set(master, master->regs->imsk, data, data); - fsi_master_mask_set(master, master->regs->iemsk, data, data); + fsi_master_mask_set(master, master->core->imsk, data, data); + fsi_master_mask_set(master, master->core->iemsk, data, data); } static void fsi_irq_disable(struct fsi_priv *fsi, int is_play) @@ -358,18 +375,18 @@ static void fsi_irq_disable(struct fsi_priv *fsi, int is_play) u32 data = fsi_port_ab_io_bit(fsi, is_play); struct fsi_master *master = fsi_get_master(fsi); - fsi_master_mask_set(master, master->regs->imsk, data, 0); - fsi_master_mask_set(master, master->regs->iemsk, data, 0); + fsi_master_mask_set(master, master->core->imsk, data, 0); + fsi_master_mask_set(master, master->core->iemsk, data, 0); } static u32 fsi_irq_get_status(struct fsi_master *master) { - return fsi_master_read(master, master->regs->int_st); + return fsi_master_read(master, master->core->int_st); } static void fsi_irq_clear_all_status(struct fsi_master *master) { - fsi_master_write(master, master->regs->int_st, 0x0000000); + fsi_master_write(master, master->core->int_st, 0); } static void fsi_irq_clear_status(struct fsi_priv *fsi) @@ -381,7 +398,30 @@ static void fsi_irq_clear_status(struct fsi_priv *fsi) data |= fsi_port_ab_io_bit(fsi, 1); /* clear interrupt factor */ - fsi_master_mask_set(master, master->regs->int_st, data, 0); + fsi_master_mask_set(master, master->core->int_st, data, 0); +} + +/************************************************************************ + + + SPDIF master clock function + +These functions are used later FSI2 +************************************************************************/ +static void fsi_spdif_clk_ctrl(struct fsi_priv *fsi, int enable) +{ + struct fsi_master *master = fsi_get_master(fsi); + u32 val = BP | SE; + + if (master->core->ver < 2) { + pr_err("fsi: register access err (%s)\n", __func__); + return; + } + + if (enable) + fsi_master_mask_set(master, fsi->mst_ctrl, val, val); + else + fsi_master_mask_set(master, fsi->mst_ctrl, val, 0); } /************************************************************************ @@ -662,8 +702,8 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct fsi_priv *fsi = fsi_get_priv(substream); - const char *msg; u32 flags = fsi_get_info_flags(fsi); + struct fsi_master *master = fsi_get_master(fsi); u32 fmt; u32 reg; u32 data; @@ -700,36 +740,40 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream, fmt = is_play ? SH_FSI_GET_OFMT(flags) : SH_FSI_GET_IFMT(flags); switch (fmt) { case SH_FSI_FMT_MONO: - msg = "MONO"; - data = CR_FMT(CR_MONO); + data = CR_MONO; fsi->chan = 1; break; case SH_FSI_FMT_MONO_DELAY: - msg = "MONO Delay"; - data = CR_FMT(CR_MONO_D); + data = CR_MONO_D; fsi->chan = 1; break; case SH_FSI_FMT_PCM: - msg = "PCM"; - data = CR_FMT(CR_PCM); + data = CR_PCM; fsi->chan = 2; break; case SH_FSI_FMT_I2S: - msg = "I2S"; - data = CR_FMT(CR_I2S); + data = CR_I2S; fsi->chan = 2; break; case SH_FSI_FMT_TDM: - msg = "TDM"; fsi->chan = is_play ? SH_FSI_GET_CH_O(flags) : SH_FSI_GET_CH_I(flags); - data = CR_FMT(CR_TDM) | (fsi->chan - 1); + data = CR_TDM | (fsi->chan - 1); break; case SH_FSI_FMT_TDM_DELAY: - msg = "TDM Delay"; fsi->chan = is_play ? SH_FSI_GET_CH_O(flags) : SH_FSI_GET_CH_I(flags); - data = CR_FMT(CR_TDM_D) | (fsi->chan - 1); + data = CR_TDM_D | (fsi->chan - 1); + break; + case SH_FSI_FMT_SPDIF: + if (master->core->ver < 2) { + dev_err(dai->dev, "This FSI can not use SPDIF\n"); + return -EINVAL; + } + data = CR_SPDIF; + fsi->chan = 2; + fsi_spdif_clk_ctrl(fsi, 1); + fsi_reg_mask_set(fsi, OUT_SEL, 0x0010, 0x0010); break; default: dev_err(dai->dev, "unknown format.\n"); @@ -737,12 +781,6 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream, } fsi_reg_write(fsi, reg, data); - /* - * clear clk reset if master mode - */ - if (is_master) - fsi_clk_ctrl(fsi, 1); - /* irq clear */ fsi_irq_disable(fsi, is_play); fsi_irq_clear_status(fsi); @@ -789,10 +827,93 @@ static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } +static int fsi_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct fsi_priv *fsi = fsi_get_priv(substream); + struct fsi_master *master = fsi_get_master(fsi); + int (*set_rate)(int is_porta, int rate) = master->info->set_rate; + int fsi_ver = master->core->ver; + int is_play = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + int ret; + + /* if slave mode, set_rate is not needed */ + if (!fsi_is_master_mode(fsi, is_play)) + return 0; + + /* it is error if no set_rate */ + if (!set_rate) + return -EIO; + + ret = set_rate(fsi_is_port_a(fsi), params_rate(params)); + if (ret > 0) { + u32 data = 0; + + switch (ret & SH_FSI_ACKMD_MASK) { + default: + /* FALL THROUGH */ + case SH_FSI_ACKMD_512: + data |= (0x0 << 12); + break; + case SH_FSI_ACKMD_256: + data |= (0x1 << 12); + break; + case SH_FSI_ACKMD_128: + data |= (0x2 << 12); + break; + case SH_FSI_ACKMD_64: + data |= (0x3 << 12); + break; + case SH_FSI_ACKMD_32: + if (fsi_ver < 2) + dev_err(dai->dev, "unsupported ACKMD\n"); + else + data |= (0x4 << 12); + break; + } + + switch (ret & SH_FSI_BPFMD_MASK) { + default: + /* FALL THROUGH */ + case SH_FSI_BPFMD_32: + data |= (0x0 << 8); + break; + case SH_FSI_BPFMD_64: + data |= (0x1 << 8); + break; + case SH_FSI_BPFMD_128: + data |= (0x2 << 8); + break; + case SH_FSI_BPFMD_256: + data |= (0x3 << 8); + break; + case SH_FSI_BPFMD_512: + data |= (0x4 << 8); + break; + case SH_FSI_BPFMD_16: + if (fsi_ver < 2) + dev_err(dai->dev, "unsupported ACKMD\n"); + else + data |= (0x7 << 8); + break; + } + + fsi_reg_mask_set(fsi, CKG1, (ACKMD_MASK | BPFMD_MASK) , data); + udelay(10); + fsi_clk_ctrl(fsi, 1); + ret = 0; + } + + return ret; + +} + static struct snd_soc_dai_ops fsi_dai_ops = { .startup = fsi_dai_startup, .shutdown = fsi_dai_shutdown, .trigger = fsi_dai_trigger, + .hw_params = fsi_dai_hw_params, }; /************************************************************************ @@ -965,11 +1086,6 @@ static int fsi_probe(struct platform_device *pdev) unsigned int irq; int ret; - if (0 != pdev->id) { - dev_err(&pdev->dev, "current fsi support id 0 only now\n"); - return -ENODEV; - } - id_entry = pdev->id_entry; if (!id_entry) { dev_err(&pdev->dev, "unknown fsi device\n"); @@ -998,14 +1114,21 @@ static int fsi_probe(struct platform_device *pdev) goto exit_kfree; } + /* master setting */ master->irq = irq; master->info = pdev->dev.platform_data; + master->core = (struct fsi_core *)id_entry->driver_data; + spin_lock_init(&master->lock); + + /* FSI A setting */ master->fsia.base = master->base; master->fsia.master = master; + master->fsia.mst_ctrl = A_MST_CTLR; + + /* FSI B setting */ master->fsib.base = master->base + 0x40; master->fsib.master = master; - master->regs = (struct fsi_regs *)id_entry->driver_data; - spin_lock_init(&master->lock); + master->fsib.mst_ctrl = B_MST_CTLR; pm_runtime_enable(&pdev->dev); pm_runtime_resume(&pdev->dev); @@ -1085,21 +1208,27 @@ static struct dev_pm_ops fsi_pm_ops = { .runtime_resume = fsi_runtime_nop, }; -static struct fsi_regs fsi_regs = { +static struct fsi_core fsi1_core = { + .ver = 1, + + /* Interrupt */ .int_st = INT_ST, .iemsk = IEMSK, .imsk = IMSK, }; -static struct fsi_regs fsi2_regs = { +static struct fsi_core fsi2_core = { + .ver = 2, + + /* Interrupt */ .int_st = CPU_INT_ST, .iemsk = CPU_IEMSK, .imsk = CPU_IMSK, }; static struct platform_device_id fsi_id_table[] = { - { "sh_fsi", (kernel_ulong_t)&fsi_regs }, - { "sh_fsi2", (kernel_ulong_t)&fsi2_regs }, + { "sh_fsi", (kernel_ulong_t)&fsi1_core }, + { "sh_fsi2", (kernel_ulong_t)&fsi2_core }, }; static struct platform_driver fsi_driver = { diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c index b823a5c9b9bc..87e2b7fcbf17 100644 --- a/sound/soc/sh/migor.c +++ b/sound/soc/sh/migor.c @@ -12,6 +12,7 @@ #include <linux/firmware.h> #include <linux/module.h> +#include <asm/clkdev.h> #include <asm/clock.h> #include <cpu/sh7722.h> @@ -40,12 +41,12 @@ static struct clk_ops siumckb_clk_ops = { }; static struct clk siumckb_clk = { - .name = "siumckb_clk", - .id = -1, .ops = &siumckb_clk_ops, .rate = 0, /* initialised at run-time */ }; +static struct clk_lookup *siumckb_lookup; + static int migor_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -180,6 +181,13 @@ static int __init migor_init(void) if (ret < 0) return ret; + siumckb_lookup = clkdev_alloc(&siumckb_clk, "siumckb_clk", NULL); + if (!siumckb_lookup) { + ret = -ENOMEM; + goto eclkdevalloc; + } + clkdev_add(siumckb_lookup); + /* Port number used on this machine: port B */ migor_snd_device = platform_device_alloc("soc-audio", 1); if (!migor_snd_device) { @@ -200,12 +208,15 @@ static int __init migor_init(void) epdevadd: platform_device_put(migor_snd_device); epdevalloc: + clkdev_drop(siumckb_lookup); +eclkdevalloc: clk_unregister(&siumckb_clk); return ret; } static void __exit migor_exit(void) { + clkdev_drop(siumckb_lookup); clk_unregister(&siumckb_clk); platform_device_unregister(migor_snd_device); } diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 472af38188c1..f6b0d2829ea9 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -203,8 +203,9 @@ static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg, data[1] = (value >> 8) & 0xff; data[2] = value & 0xff; - if (!snd_soc_codec_volatile_register(codec, reg)) - reg_cache[reg] = value; + if (!snd_soc_codec_volatile_register(codec, reg) + && reg < codec->reg_cache_size) + reg_cache[reg] = value; if (codec->cache_only) { codec->cache_sync = 1; @@ -340,7 +341,7 @@ static unsigned int snd_soc_16_8_read_i2c(struct snd_soc_codec *codec, static unsigned int snd_soc_16_8_read(struct snd_soc_codec *codec, unsigned int reg) { - u16 *cache = codec->reg_cache; + u8 *cache = codec->reg_cache; reg &= 0xff; if (reg >= codec->reg_cache_size) @@ -351,7 +352,7 @@ static unsigned int snd_soc_16_8_read(struct snd_soc_codec *codec, static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { - u16 *cache = codec->reg_cache; + u8 *cache = codec->reg_cache; u8 data[3]; int ret; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 998569d60330..acc91daa1c55 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -84,7 +84,7 @@ static int run_delayed_work(struct delayed_work *dwork) /* codec register dump */ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) { - int i, step = 1, count = 0; + int ret, i, step = 1, count = 0; if (!codec->reg_cache_size) return 0; @@ -101,12 +101,24 @@ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) if (count >= PAGE_SIZE - 1) break; - if (codec->display_register) + if (codec->display_register) { count += codec->display_register(codec, buf + count, PAGE_SIZE - count, i); - else - count += snprintf(buf + count, PAGE_SIZE - count, - "%4x", codec->read(codec, i)); + } else { + /* If the read fails it's almost certainly due to + * the register being volatile and the device being + * powered off. + */ + ret = codec->read(codec, i); + if (ret >= 0) + count += snprintf(buf + count, + PAGE_SIZE - count, + "%4x", ret); + else + count += snprintf(buf + count, + PAGE_SIZE - count, + "<no data: %d>", ret); + } if (count >= PAGE_SIZE - 1) break; @@ -239,7 +251,7 @@ static void soc_init_codec_debugfs(struct snd_soc_codec *codec) printk(KERN_WARNING "ASoC: Failed to create codec register debugfs file\n"); - codec->debugfs_pop_time = debugfs_create_u32("dapm_pop_time", 0744, + codec->debugfs_pop_time = debugfs_create_u32("dapm_pop_time", 0644, codec->debugfs_codec_root, &codec->pop_time); if (!codec->debugfs_pop_time) @@ -1307,7 +1319,7 @@ cpu_dai_err: } /* - * Attempt to initialise any uninitalised cards. Must be called with + * Attempt to initialise any uninitialised cards. Must be called with * client_mutex. */ static void snd_soc_instantiate_cards(void) @@ -2353,6 +2365,99 @@ int snd_soc_limit_volume(struct snd_soc_codec *codec, EXPORT_SYMBOL_GPL(snd_soc_limit_volume); /** + * snd_soc_info_volsw_2r_sx - double with tlv and variable data size + * mixer info callback + * @kcontrol: mixer control + * @uinfo: control element information + * + * Returns 0 for success. + */ +int snd_soc_info_volsw_2r_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int max = mc->max; + int min = mc->min; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = max-min; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_info_volsw_2r_sx); + +/** + * snd_soc_get_volsw_2r_sx - double with tlv and variable data size + * mixer get callback + * @kcontrol: mixer control + * @uinfo: control element information + * + * Returns 0 for success. + */ +int snd_soc_get_volsw_2r_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int mask = (1<<mc->shift)-1; + int min = mc->min; + int val = snd_soc_read(codec, mc->reg) & mask; + int valr = snd_soc_read(codec, mc->rreg) & mask; + + ucontrol->value.integer.value[0] = ((val & 0xff)-min) & mask; + ucontrol->value.integer.value[1] = ((valr & 0xff)-min) & mask; + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_get_volsw_2r_sx); + +/** + * snd_soc_put_volsw_2r_sx - double with tlv and variable data size + * mixer put callback + * @kcontrol: mixer control + * @uinfo: control element information + * + * Returns 0 for success. + */ +int snd_soc_put_volsw_2r_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int mask = (1<<mc->shift)-1; + int min = mc->min; + int ret; + unsigned int val, valr, oval, ovalr; + + val = ((ucontrol->value.integer.value[0]+min) & 0xff); + val &= mask; + valr = ((ucontrol->value.integer.value[1]+min) & 0xff); + valr &= mask; + + oval = snd_soc_read(codec, mc->reg) & mask; + ovalr = snd_soc_read(codec, mc->rreg) & mask; + + ret = 0; + if (oval != val) { + ret = snd_soc_write(codec, mc->reg, val); + if (ret < 0) + return ret; + } + if (ovalr != valr) { + ret = snd_soc_write(codec, mc->rreg, valr); + if (ret < 0) + return ret; + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_put_volsw_2r_sx); + +/** * snd_soc_dai_set_sysclk - configure DAI system or master clock. * @dai: DAI * @clk_id: DAI specific clock ID diff --git a/sound/sound_core.c b/sound/sound_core.c index 7c2d677a2df5..cb61317df509 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -576,8 +576,6 @@ static int soundcore_open(struct inode *inode, struct file *file) struct sound_unit *s; const struct file_operations *new_fops = NULL; - lock_kernel (); - chain=unit&0x0F; if(chain==4 || chain==5) /* dsp/audio/dsp16 */ { @@ -630,18 +628,19 @@ static int soundcore_open(struct inode *inode, struct file *file) const struct file_operations *old_fops = file->f_op; file->f_op = new_fops; spin_unlock(&sound_loader_lock); - if(file->f_op->open) + + if (file->f_op->open) err = file->f_op->open(inode,file); + if (err) { fops_put(file->f_op); file->f_op = fops_get(old_fops); } + fops_put(old_fops); - unlock_kernel(); return err; } spin_unlock(&sound_loader_lock); - unlock_kernel(); return -ENODEV; } diff --git a/sound/sparc/amd7930.c b/sound/sparc/amd7930.c index 71221fd20944..f8bcfc30f800 100644 --- a/sound/sparc/amd7930.c +++ b/sound/sparc/amd7930.c @@ -336,7 +336,7 @@ struct snd_amd7930 { int pgain; int mgain; - struct of_device *op; + struct platform_device *op; unsigned int irq; struct snd_amd7930 *next; }; @@ -906,7 +906,7 @@ static int __devinit snd_amd7930_mixer(struct snd_amd7930 *amd) static int snd_amd7930_free(struct snd_amd7930 *amd) { - struct of_device *op = amd->op; + struct platform_device *op = amd->op; amd7930_idle(amd); @@ -934,7 +934,7 @@ static struct snd_device_ops snd_amd7930_dev_ops = { }; static int __devinit snd_amd7930_create(struct snd_card *card, - struct of_device *op, + struct platform_device *op, int irq, int dev, struct snd_amd7930 **ramd) { @@ -1002,7 +1002,7 @@ static int __devinit snd_amd7930_create(struct snd_card *card, return 0; } -static int __devinit amd7930_sbus_probe(struct of_device *op, const struct of_device_id *match) +static int __devinit amd7930_sbus_probe(struct platform_device *op, const struct of_device_id *match) { struct resource *rp = &op->resource[0]; static int dev_num; @@ -1010,7 +1010,7 @@ static int __devinit amd7930_sbus_probe(struct of_device *op, const struct of_de struct snd_amd7930 *amd; int err, irq; - irq = op->irqs[0]; + irq = op->archdata.irqs[0]; if (dev_num >= SNDRV_CARDS) return -ENODEV; @@ -1075,7 +1075,7 @@ static struct of_platform_driver amd7930_sbus_driver = { static int __init amd7930_init(void) { - return of_register_driver(&amd7930_sbus_driver, &of_bus_type); + return of_register_platform_driver(&amd7930_sbus_driver); } static void __exit amd7930_exit(void) @@ -1092,7 +1092,7 @@ static void __exit amd7930_exit(void) amd7930_list = NULL; - of_unregister_driver(&amd7930_sbus_driver); + of_unregister_platform_driver(&amd7930_sbus_driver); } module_init(amd7930_init); diff --git a/sound/sparc/cs4231.c b/sound/sparc/cs4231.c index fb4c6f2f29e5..c276086c3b57 100644 --- a/sound/sparc/cs4231.c +++ b/sound/sparc/cs4231.c @@ -111,7 +111,7 @@ struct snd_cs4231 { struct mutex mce_mutex; /* mutex for mce register */ struct mutex open_mutex; /* mutex for ALSA open/close */ - struct of_device *op; + struct platform_device *op; unsigned int irq[2]; unsigned int regs_size; struct snd_cs4231 *next; @@ -1771,7 +1771,7 @@ static unsigned int sbus_dma_addr(struct cs4231_dma_control *dma_cont) static int snd_cs4231_sbus_free(struct snd_cs4231 *chip) { - struct of_device *op = chip->op; + struct platform_device *op = chip->op; if (chip->irq[0]) free_irq(chip->irq[0], chip); @@ -1794,7 +1794,7 @@ static struct snd_device_ops snd_cs4231_sbus_dev_ops = { }; static int __devinit snd_cs4231_sbus_create(struct snd_card *card, - struct of_device *op, + struct platform_device *op, int dev) { struct snd_cs4231 *chip = card->private_data; @@ -1832,14 +1832,14 @@ static int __devinit snd_cs4231_sbus_create(struct snd_card *card, chip->c_dma.request = sbus_dma_request; chip->c_dma.address = sbus_dma_addr; - if (request_irq(op->irqs[0], snd_cs4231_sbus_interrupt, + if (request_irq(op->archdata.irqs[0], snd_cs4231_sbus_interrupt, IRQF_SHARED, "cs4231", chip)) { snd_printdd("cs4231-%d: Unable to grab SBUS IRQ %d\n", - dev, op->irqs[0]); + dev, op->archdata.irqs[0]); snd_cs4231_sbus_free(chip); return -EBUSY; } - chip->irq[0] = op->irqs[0]; + chip->irq[0] = op->archdata.irqs[0]; if (snd_cs4231_probe(chip) < 0) { snd_cs4231_sbus_free(chip); @@ -1856,7 +1856,7 @@ static int __devinit snd_cs4231_sbus_create(struct snd_card *card, return 0; } -static int __devinit cs4231_sbus_probe(struct of_device *op, const struct of_device_id *match) +static int __devinit cs4231_sbus_probe(struct platform_device *op, const struct of_device_id *match) { struct resource *rp = &op->resource[0]; struct snd_card *card; @@ -1870,7 +1870,7 @@ static int __devinit cs4231_sbus_probe(struct of_device *op, const struct of_dev card->shortname, rp->flags & 0xffL, (unsigned long long)rp->start, - op->irqs[0]); + op->archdata.irqs[0]); err = snd_cs4231_sbus_create(card, op, dev); if (err < 0) { @@ -1931,7 +1931,7 @@ static unsigned int _ebus_dma_addr(struct cs4231_dma_control *dma_cont) static int snd_cs4231_ebus_free(struct snd_cs4231 *chip) { - struct of_device *op = chip->op; + struct platform_device *op = chip->op; if (chip->c_dma.ebus_info.regs) { ebus_dma_unregister(&chip->c_dma.ebus_info); @@ -1960,7 +1960,7 @@ static struct snd_device_ops snd_cs4231_ebus_dev_ops = { }; static int __devinit snd_cs4231_ebus_create(struct snd_card *card, - struct of_device *op, + struct platform_device *op, int dev) { struct snd_cs4231 *chip = card->private_data; @@ -1979,12 +1979,12 @@ static int __devinit snd_cs4231_ebus_create(struct snd_card *card, chip->c_dma.ebus_info.flags = EBUS_DMA_FLAG_USE_EBDMA_HANDLER; chip->c_dma.ebus_info.callback = snd_cs4231_ebus_capture_callback; chip->c_dma.ebus_info.client_cookie = chip; - chip->c_dma.ebus_info.irq = op->irqs[0]; + chip->c_dma.ebus_info.irq = op->archdata.irqs[0]; strcpy(chip->p_dma.ebus_info.name, "cs4231(play)"); chip->p_dma.ebus_info.flags = EBUS_DMA_FLAG_USE_EBDMA_HANDLER; chip->p_dma.ebus_info.callback = snd_cs4231_ebus_play_callback; chip->p_dma.ebus_info.client_cookie = chip; - chip->p_dma.ebus_info.irq = op->irqs[1]; + chip->p_dma.ebus_info.irq = op->archdata.irqs[1]; chip->p_dma.prepare = _ebus_dma_prepare; chip->p_dma.enable = _ebus_dma_enable; @@ -2048,7 +2048,7 @@ static int __devinit snd_cs4231_ebus_create(struct snd_card *card, return 0; } -static int __devinit cs4231_ebus_probe(struct of_device *op, const struct of_device_id *match) +static int __devinit cs4231_ebus_probe(struct platform_device *op, const struct of_device_id *match) { struct snd_card *card; int err; @@ -2060,7 +2060,7 @@ static int __devinit cs4231_ebus_probe(struct of_device *op, const struct of_dev sprintf(card->longname, "%s at 0x%llx, irq %d", card->shortname, op->resource[0].start, - op->irqs[0]); + op->archdata.irqs[0]); err = snd_cs4231_ebus_create(card, op, dev); if (err < 0) { @@ -2072,7 +2072,7 @@ static int __devinit cs4231_ebus_probe(struct of_device *op, const struct of_dev } #endif -static int __devinit cs4231_probe(struct of_device *op, const struct of_device_id *match) +static int __devinit cs4231_probe(struct platform_device *op, const struct of_device_id *match) { #ifdef EBUS_SUPPORT if (!strcmp(op->dev.of_node->parent->name, "ebus")) @@ -2086,7 +2086,7 @@ static int __devinit cs4231_probe(struct of_device *op, const struct of_device_i return -ENODEV; } -static int __devexit cs4231_remove(struct of_device *op) +static int __devexit cs4231_remove(struct platform_device *op) { struct snd_cs4231 *chip = dev_get_drvdata(&op->dev); @@ -2120,12 +2120,12 @@ static struct of_platform_driver cs4231_driver = { static int __init cs4231_init(void) { - return of_register_driver(&cs4231_driver, &of_bus_type); + return of_register_platform_driver(&cs4231_driver); } static void __exit cs4231_exit(void) { - of_unregister_driver(&cs4231_driver); + of_unregister_platform_driver(&cs4231_driver); } module_init(cs4231_init); diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index 1557bf132e73..39cd5d69d051 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -299,7 +299,7 @@ struct dbri_streaminfo { /* This structure holds the information for both chips (DBRI & CS4215) */ struct snd_dbri { int regs_size, irq; /* Needed for unload */ - struct of_device *op; /* OF device info */ + struct platform_device *op; /* OF device info */ spinlock_t lock; struct dbri_dma *dma; /* Pointer to our DMA block */ @@ -2523,7 +2523,7 @@ static void __devinit snd_dbri_proc(struct snd_card *card) static void snd_dbri_free(struct snd_dbri *dbri); static int __devinit snd_dbri_create(struct snd_card *card, - struct of_device *op, + struct platform_device *op, int irq, int dev) { struct snd_dbri *dbri = card->private_data; @@ -2592,7 +2592,7 @@ static void snd_dbri_free(struct snd_dbri *dbri) (void *)dbri->dma, dbri->dma_dvma); } -static int __devinit dbri_probe(struct of_device *op, const struct of_device_id *match) +static int __devinit dbri_probe(struct platform_device *op, const struct of_device_id *match) { struct snd_dbri *dbri; struct resource *rp; @@ -2608,7 +2608,7 @@ static int __devinit dbri_probe(struct of_device *op, const struct of_device_id return -ENOENT; } - irq = op->irqs[0]; + irq = op->archdata.irqs[0]; if (irq <= 0) { printk(KERN_ERR "DBRI-%d: No IRQ.\n", dev); return -ENODEV; @@ -2662,7 +2662,7 @@ _err: return err; } -static int __devexit dbri_remove(struct of_device *op) +static int __devexit dbri_remove(struct platform_device *op) { struct snd_card *card = dev_get_drvdata(&op->dev); @@ -2699,12 +2699,12 @@ static struct of_platform_driver dbri_sbus_driver = { /* Probe for the dbri chip and then attach the driver. */ static int __init dbri_init(void) { - return of_register_driver(&dbri_sbus_driver, &of_bus_type); + return of_register_platform_driver(&dbri_sbus_driver); } static void __exit dbri_exit(void) { - of_unregister_driver(&dbri_sbus_driver); + of_unregister_platform_driver(&dbri_sbus_driver); } module_init(dbri_init); diff --git a/sound/synth/emux/emux_hwdep.c b/sound/synth/emux/emux_hwdep.c index ff0b2a8fd25b..5ae1eae9f6db 100644 --- a/sound/synth/emux/emux_hwdep.c +++ b/sound/synth/emux/emux_hwdep.c @@ -128,6 +128,9 @@ snd_emux_init_hwdep(struct snd_emux *emu) strcpy(hw->name, SNDRV_EMUX_HWDEP_NAME); hw->iface = SNDRV_HWDEP_IFACE_EMUX_WAVETABLE; hw->ops.ioctl = snd_emux_hwdep_ioctl; + /* The ioctl parameter types are compatible between 32- and + * 64-bit architectures, so use the same function. */ + hw->ops.ioctl_compat = snd_emux_hwdep_ioctl; hw->exclusive = 1; hw->private_data = emu; if ((err = snd_card_register(emu->card)) < 0) diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index 44d6d2ec964f..112984f4080f 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -65,6 +65,7 @@ config SND_USB_CAIAQ * Native Instruments Guitar Rig Session I/O * Native Instruments Guitar Rig mobile * Native Instruments Traktor Kontrol X1 + * Native Instruments Traktor Kontrol S4 To compile this driver as a module, choose M here: the module will be called snd-usb-caiaq. @@ -82,6 +83,7 @@ config SND_USB_CAIAQ_INPUT * Native Instruments Kore Controller * Native Instruments Kore Controller 2 * Native Instruments Audio Kontrol 1 + * Native Instruments Traktor Kontrol S4 config SND_USB_US122L tristate "Tascam US-122L USB driver" diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index 4328cad6c3a2..68b97477577b 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -111,7 +111,7 @@ static int stream_start(struct snd_usb_caiaqdev *dev) memset(dev->sub_capture, 0, sizeof(dev->sub_capture)); dev->input_panic = 0; dev->output_panic = 0; - dev->first_packet = 1; + dev->first_packet = 4; dev->streaming = 1; dev->warned = 0; @@ -169,7 +169,7 @@ static int snd_usb_caiaq_substream_close(struct snd_pcm_substream *substream) } static int snd_usb_caiaq_pcm_hw_params(struct snd_pcm_substream *sub, - struct snd_pcm_hw_params *hw_params) + struct snd_pcm_hw_params *hw_params) { debug("%s(%p)\n", __func__, sub); return snd_pcm_lib_malloc_pages(sub, params_buffer_bytes(hw_params)); @@ -189,7 +189,7 @@ static int snd_usb_caiaq_pcm_hw_free(struct snd_pcm_substream *sub) #endif static unsigned int rates[] = { 5512, 8000, 11025, 16000, 22050, 32000, 44100, - 48000, 64000, 88200, 96000, 176400, 192000 }; + 48000, 64000, 88200, 96000, 176400, 192000 }; static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) { @@ -201,12 +201,39 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) debug("%s(%p)\n", __func__, substream); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - dev->period_out_count[index] = BYTES_PER_SAMPLE + 1; - dev->audio_out_buf_pos[index] = BYTES_PER_SAMPLE + 1; + int out_pos; + + switch (dev->spec.data_alignment) { + case 0: + case 2: + out_pos = BYTES_PER_SAMPLE + 1; + break; + case 3: + default: + out_pos = 0; + break; + } + + dev->period_out_count[index] = out_pos; + dev->audio_out_buf_pos[index] = out_pos; } else { - int in_pos = (dev->spec.data_alignment == 2) ? 0 : 2; - dev->period_in_count[index] = BYTES_PER_SAMPLE + in_pos; - dev->audio_in_buf_pos[index] = BYTES_PER_SAMPLE + in_pos; + int in_pos; + + switch (dev->spec.data_alignment) { + case 0: + in_pos = BYTES_PER_SAMPLE + 2; + break; + case 2: + in_pos = BYTES_PER_SAMPLE; + break; + case 3: + default: + in_pos = 0; + break; + } + + dev->period_in_count[index] = in_pos; + dev->audio_in_buf_pos[index] = in_pos; } if (dev->streaming) @@ -221,7 +248,7 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) snd_pcm_limit_hw_rates(runtime); bytes_per_sample = BYTES_PER_SAMPLE; - if (dev->spec.data_alignment == 2) + if (dev->spec.data_alignment >= 2) bytes_per_sample++; bpp = ((runtime->rate / 8000) + CLOCK_DRIFT_TOLERANCE) @@ -253,6 +280,8 @@ static int snd_usb_caiaq_pcm_trigger(struct snd_pcm_substream *sub, int cmd) { struct snd_usb_caiaqdev *dev = snd_pcm_substream_chip(sub); + debug("%s(%p) cmd %d\n", __func__, sub, cmd); + switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: @@ -402,6 +431,61 @@ static void read_in_urb_mode2(struct snd_usb_caiaqdev *dev, } } +static void read_in_urb_mode3(struct snd_usb_caiaqdev *dev, + const struct urb *urb, + const struct usb_iso_packet_descriptor *iso) +{ + unsigned char *usb_buf = urb->transfer_buffer + iso->offset; + int stream, i; + + /* paranoia check */ + if (iso->actual_length % (BYTES_PER_SAMPLE_USB * CHANNELS_PER_STREAM)) + return; + + for (i = 0; i < iso->actual_length;) { + for (stream = 0; stream < dev->n_streams; stream++) { + struct snd_pcm_substream *sub = dev->sub_capture[stream]; + char *audio_buf = NULL; + int c, n, sz = 0; + + if (sub && !dev->input_panic) { + struct snd_pcm_runtime *rt = sub->runtime; + audio_buf = rt->dma_area; + sz = frames_to_bytes(rt, rt->buffer_size); + } + + for (c = 0; c < CHANNELS_PER_STREAM; c++) { + /* 3 audio data bytes, followed by 1 check byte */ + if (audio_buf) { + for (n = 0; n < BYTES_PER_SAMPLE; n++) { + audio_buf[dev->audio_in_buf_pos[stream]++] = usb_buf[i+n]; + + if (dev->audio_in_buf_pos[stream] == sz) + dev->audio_in_buf_pos[stream] = 0; + } + + dev->period_in_count[stream] += BYTES_PER_SAMPLE; + } + + i += BYTES_PER_SAMPLE; + + if (usb_buf[i] != ((stream << 1) | c) && + !dev->first_packet) { + if (!dev->input_panic) + printk(" EXPECTED: %02x got %02x, c %d, stream %d, i %d\n", + ((stream << 1) | c), usb_buf[i], c, stream, i); + dev->input_panic = 1; + } + + i++; + } + } + } + + if (dev->first_packet > 0) + dev->first_packet--; +} + static void read_in_urb(struct snd_usb_caiaqdev *dev, const struct urb *urb, const struct usb_iso_packet_descriptor *iso) @@ -419,6 +503,9 @@ static void read_in_urb(struct snd_usb_caiaqdev *dev, case 2: read_in_urb_mode2(dev, urb, iso); break; + case 3: + read_in_urb_mode3(dev, urb, iso); + break; } if ((dev->input_panic || dev->output_panic) && !dev->warned) { @@ -429,9 +516,9 @@ static void read_in_urb(struct snd_usb_caiaqdev *dev, } } -static void fill_out_urb(struct snd_usb_caiaqdev *dev, - struct urb *urb, - const struct usb_iso_packet_descriptor *iso) +static void fill_out_urb_mode_0(struct snd_usb_caiaqdev *dev, + struct urb *urb, + const struct usb_iso_packet_descriptor *iso) { unsigned char *usb_buf = urb->transfer_buffer + iso->offset; struct snd_pcm_substream *sub; @@ -457,9 +544,67 @@ static void fill_out_urb(struct snd_usb_caiaqdev *dev, /* fill in the check bytes */ if (dev->spec.data_alignment == 2 && i % (dev->n_streams * BYTES_PER_SAMPLE_USB) == - (dev->n_streams * CHANNELS_PER_STREAM)) - for (stream = 0; stream < dev->n_streams; stream++, i++) - usb_buf[i] = MAKE_CHECKBYTE(dev, stream, i); + (dev->n_streams * CHANNELS_PER_STREAM)) + for (stream = 0; stream < dev->n_streams; stream++, i++) + usb_buf[i] = MAKE_CHECKBYTE(dev, stream, i); + } +} + +static void fill_out_urb_mode_3(struct snd_usb_caiaqdev *dev, + struct urb *urb, + const struct usb_iso_packet_descriptor *iso) +{ + unsigned char *usb_buf = urb->transfer_buffer + iso->offset; + int stream, i; + + for (i = 0; i < iso->length;) { + for (stream = 0; stream < dev->n_streams; stream++) { + struct snd_pcm_substream *sub = dev->sub_playback[stream]; + char *audio_buf = NULL; + int c, n, sz = 0; + + if (sub) { + struct snd_pcm_runtime *rt = sub->runtime; + audio_buf = rt->dma_area; + sz = frames_to_bytes(rt, rt->buffer_size); + } + + for (c = 0; c < CHANNELS_PER_STREAM; c++) { + for (n = 0; n < BYTES_PER_SAMPLE; n++) { + if (audio_buf) { + usb_buf[i+n] = audio_buf[dev->audio_out_buf_pos[stream]++]; + + if (dev->audio_out_buf_pos[stream] == sz) + dev->audio_out_buf_pos[stream] = 0; + } else { + usb_buf[i+n] = 0; + } + } + + if (audio_buf) + dev->period_out_count[stream] += BYTES_PER_SAMPLE; + + i += BYTES_PER_SAMPLE; + + /* fill in the check byte pattern */ + usb_buf[i++] = (stream << 1) | c; + } + } + } +} + +static inline void fill_out_urb(struct snd_usb_caiaqdev *dev, + struct urb *urb, + const struct usb_iso_packet_descriptor *iso) +{ + switch (dev->spec.data_alignment) { + case 0: + case 2: + fill_out_urb_mode_0(dev, urb, iso); + break; + case 3: + fill_out_urb_mode_3(dev, urb, iso); + break; } } diff --git a/sound/usb/caiaq/control.c b/sound/usb/caiaq/control.c index 91c804cd2782..00e5d0a469e1 100644 --- a/sound/usb/caiaq/control.c +++ b/sound/usb/caiaq/control.c @@ -55,6 +55,10 @@ static int control_info(struct snd_kcontrol *kcontrol, case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1): maxval = 127; break; + + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLS4): + maxval = 31; + break; } if (is_intval) { @@ -93,6 +97,7 @@ static int control_put(struct snd_kcontrol *kcontrol, struct snd_usb_audio *chip = snd_kcontrol_chip(kcontrol); struct snd_usb_caiaqdev *dev = caiaqdev(chip->card); int pos = kcontrol->private_value; + int v = ucontrol->value.integer.value[0]; unsigned char cmd = EP1_CMD_WRITE_IO; if (dev->chip.usb_id == @@ -100,12 +105,27 @@ static int control_put(struct snd_kcontrol *kcontrol, cmd = EP1_CMD_DIMM_LEDS; if (pos & CNT_INTVAL) { - dev->control_state[pos & ~CNT_INTVAL] - = ucontrol->value.integer.value[0]; - snd_usb_caiaq_send_command(dev, cmd, - dev->control_state, sizeof(dev->control_state)); + int i = pos & ~CNT_INTVAL; + + dev->control_state[i] = v; + + if (dev->chip.usb_id == + USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLS4)) { + int actual_len; + + dev->ep8_out_buf[0] = i; + dev->ep8_out_buf[1] = v; + + usb_bulk_msg(dev->chip.dev, + usb_sndbulkpipe(dev->chip.dev, 8), + dev->ep8_out_buf, sizeof(dev->ep8_out_buf), + &actual_len, 200); + } else { + snd_usb_caiaq_send_command(dev, cmd, + dev->control_state, sizeof(dev->control_state)); + } } else { - if (ucontrol->value.integer.value[0]) + if (v) dev->control_state[pos / 8] |= 1 << (pos % 8); else dev->control_state[pos / 8] &= ~(1 << (pos % 8)); @@ -296,6 +316,179 @@ static struct caiaq_controller kontrolx1_controller[] = { { "LED Deck B: SYNC", 8 | CNT_INTVAL }, }; +static struct caiaq_controller kontrols4_controller[] = { + { "LED: Master: Quant", 10 | CNT_INTVAL }, + { "LED: Master: Headphone", 11 | CNT_INTVAL }, + { "LED: Master: Master", 12 | CNT_INTVAL }, + { "LED: Master: Snap", 14 | CNT_INTVAL }, + { "LED: Master: Warning", 15 | CNT_INTVAL }, + { "LED: Master: Master button", 112 | CNT_INTVAL }, + { "LED: Master: Snap button", 113 | CNT_INTVAL }, + { "LED: Master: Rec", 118 | CNT_INTVAL }, + { "LED: Master: Size", 119 | CNT_INTVAL }, + { "LED: Master: Quant button", 120 | CNT_INTVAL }, + { "LED: Master: Browser button", 121 | CNT_INTVAL }, + { "LED: Master: Play button", 126 | CNT_INTVAL }, + { "LED: Master: Undo button", 127 | CNT_INTVAL }, + + { "LED: Channel A: >", 4 | CNT_INTVAL }, + { "LED: Channel A: <", 5 | CNT_INTVAL }, + { "LED: Channel A: Meter 1", 97 | CNT_INTVAL }, + { "LED: Channel A: Meter 2", 98 | CNT_INTVAL }, + { "LED: Channel A: Meter 3", 99 | CNT_INTVAL }, + { "LED: Channel A: Meter 4", 100 | CNT_INTVAL }, + { "LED: Channel A: Meter 5", 101 | CNT_INTVAL }, + { "LED: Channel A: Meter 6", 102 | CNT_INTVAL }, + { "LED: Channel A: Meter clip", 103 | CNT_INTVAL }, + { "LED: Channel A: Active", 114 | CNT_INTVAL }, + { "LED: Channel A: Cue", 116 | CNT_INTVAL }, + { "LED: Channel A: FX1", 149 | CNT_INTVAL }, + { "LED: Channel A: FX2", 148 | CNT_INTVAL }, + + { "LED: Channel B: >", 2 | CNT_INTVAL }, + { "LED: Channel B: <", 3 | CNT_INTVAL }, + { "LED: Channel B: Meter 1", 89 | CNT_INTVAL }, + { "LED: Channel B: Meter 2", 90 | CNT_INTVAL }, + { "LED: Channel B: Meter 3", 91 | CNT_INTVAL }, + { "LED: Channel B: Meter 4", 92 | CNT_INTVAL }, + { "LED: Channel B: Meter 5", 93 | CNT_INTVAL }, + { "LED: Channel B: Meter 6", 94 | CNT_INTVAL }, + { "LED: Channel B: Meter clip", 95 | CNT_INTVAL }, + { "LED: Channel B: Active", 122 | CNT_INTVAL }, + { "LED: Channel B: Cue", 125 | CNT_INTVAL }, + { "LED: Channel B: FX1", 147 | CNT_INTVAL }, + { "LED: Channel B: FX2", 146 | CNT_INTVAL }, + + { "LED: Channel C: >", 6 | CNT_INTVAL }, + { "LED: Channel C: <", 7 | CNT_INTVAL }, + { "LED: Channel C: Meter 1", 105 | CNT_INTVAL }, + { "LED: Channel C: Meter 2", 106 | CNT_INTVAL }, + { "LED: Channel C: Meter 3", 107 | CNT_INTVAL }, + { "LED: Channel C: Meter 4", 108 | CNT_INTVAL }, + { "LED: Channel C: Meter 5", 109 | CNT_INTVAL }, + { "LED: Channel C: Meter 6", 110 | CNT_INTVAL }, + { "LED: Channel C: Meter clip", 111 | CNT_INTVAL }, + { "LED: Channel C: Active", 115 | CNT_INTVAL }, + { "LED: Channel C: Cue", 117 | CNT_INTVAL }, + { "LED: Channel C: FX1", 151 | CNT_INTVAL }, + { "LED: Channel C: FX2", 150 | CNT_INTVAL }, + + { "LED: Channel D: >", 0 | CNT_INTVAL }, + { "LED: Channel D: <", 1 | CNT_INTVAL }, + { "LED: Channel D: Meter 1", 81 | CNT_INTVAL }, + { "LED: Channel D: Meter 2", 82 | CNT_INTVAL }, + { "LED: Channel D: Meter 3", 83 | CNT_INTVAL }, + { "LED: Channel D: Meter 4", 84 | CNT_INTVAL }, + { "LED: Channel D: Meter 5", 85 | CNT_INTVAL }, + { "LED: Channel D: Meter 6", 86 | CNT_INTVAL }, + { "LED: Channel D: Meter clip", 87 | CNT_INTVAL }, + { "LED: Channel D: Active", 123 | CNT_INTVAL }, + { "LED: Channel D: Cue", 124 | CNT_INTVAL }, + { "LED: Channel D: FX1", 145 | CNT_INTVAL }, + { "LED: Channel D: FX2", 144 | CNT_INTVAL }, + + { "LED: Deck A: 1 (blue)", 22 | CNT_INTVAL }, + { "LED: Deck A: 1 (green)", 23 | CNT_INTVAL }, + { "LED: Deck A: 2 (blue)", 20 | CNT_INTVAL }, + { "LED: Deck A: 2 (green)", 21 | CNT_INTVAL }, + { "LED: Deck A: 3 (blue)", 18 | CNT_INTVAL }, + { "LED: Deck A: 3 (green)", 19 | CNT_INTVAL }, + { "LED: Deck A: 4 (blue)", 16 | CNT_INTVAL }, + { "LED: Deck A: 4 (green)", 17 | CNT_INTVAL }, + { "LED: Deck A: Load", 44 | CNT_INTVAL }, + { "LED: Deck A: Deck C button", 45 | CNT_INTVAL }, + { "LED: Deck A: In", 47 | CNT_INTVAL }, + { "LED: Deck A: Out", 46 | CNT_INTVAL }, + { "LED: Deck A: Shift", 24 | CNT_INTVAL }, + { "LED: Deck A: Sync", 27 | CNT_INTVAL }, + { "LED: Deck A: Cue", 26 | CNT_INTVAL }, + { "LED: Deck A: Play", 25 | CNT_INTVAL }, + { "LED: Deck A: Tempo up", 33 | CNT_INTVAL }, + { "LED: Deck A: Tempo down", 32 | CNT_INTVAL }, + { "LED: Deck A: Master", 34 | CNT_INTVAL }, + { "LED: Deck A: Keylock", 35 | CNT_INTVAL }, + { "LED: Deck A: Deck A", 37 | CNT_INTVAL }, + { "LED: Deck A: Deck C", 36 | CNT_INTVAL }, + { "LED: Deck A: Samples", 38 | CNT_INTVAL }, + { "LED: Deck A: On Air", 39 | CNT_INTVAL }, + { "LED: Deck A: Sample 1", 31 | CNT_INTVAL }, + { "LED: Deck A: Sample 2", 30 | CNT_INTVAL }, + { "LED: Deck A: Sample 3", 29 | CNT_INTVAL }, + { "LED: Deck A: Sample 4", 28 | CNT_INTVAL }, + { "LED: Deck A: Digit 1 - A", 55 | CNT_INTVAL }, + { "LED: Deck A: Digit 1 - B", 54 | CNT_INTVAL }, + { "LED: Deck A: Digit 1 - C", 53 | CNT_INTVAL }, + { "LED: Deck A: Digit 1 - D", 52 | CNT_INTVAL }, + { "LED: Deck A: Digit 1 - E", 51 | CNT_INTVAL }, + { "LED: Deck A: Digit 1 - F", 50 | CNT_INTVAL }, + { "LED: Deck A: Digit 1 - G", 49 | CNT_INTVAL }, + { "LED: Deck A: Digit 1 - dot", 48 | CNT_INTVAL }, + { "LED: Deck A: Digit 2 - A", 63 | CNT_INTVAL }, + { "LED: Deck A: Digit 2 - B", 62 | CNT_INTVAL }, + { "LED: Deck A: Digit 2 - C", 61 | CNT_INTVAL }, + { "LED: Deck A: Digit 2 - D", 60 | CNT_INTVAL }, + { "LED: Deck A: Digit 2 - E", 59 | CNT_INTVAL }, + { "LED: Deck A: Digit 2 - F", 58 | CNT_INTVAL }, + { "LED: Deck A: Digit 2 - G", 57 | CNT_INTVAL }, + { "LED: Deck A: Digit 2 - dot", 56 | CNT_INTVAL }, + + { "LED: Deck B: 1 (blue)", 78 | CNT_INTVAL }, + { "LED: Deck B: 1 (green)", 79 | CNT_INTVAL }, + { "LED: Deck B: 2 (blue)", 76 | CNT_INTVAL }, + { "LED: Deck B: 2 (green)", 77 | CNT_INTVAL }, + { "LED: Deck B: 3 (blue)", 74 | CNT_INTVAL }, + { "LED: Deck B: 3 (green)", 75 | CNT_INTVAL }, + { "LED: Deck B: 4 (blue)", 72 | CNT_INTVAL }, + { "LED: Deck B: 4 (green)", 73 | CNT_INTVAL }, + { "LED: Deck B: Load", 180 | CNT_INTVAL }, + { "LED: Deck B: Deck D button", 181 | CNT_INTVAL }, + { "LED: Deck B: In", 183 | CNT_INTVAL }, + { "LED: Deck B: Out", 182 | CNT_INTVAL }, + { "LED: Deck B: Shift", 64 | CNT_INTVAL }, + { "LED: Deck B: Sync", 67 | CNT_INTVAL }, + { "LED: Deck B: Cue", 66 | CNT_INTVAL }, + { "LED: Deck B: Play", 65 | CNT_INTVAL }, + { "LED: Deck B: Tempo up", 185 | CNT_INTVAL }, + { "LED: Deck B: Tempo down", 184 | CNT_INTVAL }, + { "LED: Deck B: Master", 186 | CNT_INTVAL }, + { "LED: Deck B: Keylock", 187 | CNT_INTVAL }, + { "LED: Deck B: Deck B", 189 | CNT_INTVAL }, + { "LED: Deck B: Deck D", 188 | CNT_INTVAL }, + { "LED: Deck B: Samples", 190 | CNT_INTVAL }, + { "LED: Deck B: On Air", 191 | CNT_INTVAL }, + { "LED: Deck B: Sample 1", 71 | CNT_INTVAL }, + { "LED: Deck B: Sample 2", 70 | CNT_INTVAL }, + { "LED: Deck B: Sample 3", 69 | CNT_INTVAL }, + { "LED: Deck B: Sample 4", 68 | CNT_INTVAL }, + { "LED: Deck B: Digit 1 - A", 175 | CNT_INTVAL }, + { "LED: Deck B: Digit 1 - B", 174 | CNT_INTVAL }, + { "LED: Deck B: Digit 1 - C", 173 | CNT_INTVAL }, + { "LED: Deck B: Digit 1 - D", 172 | CNT_INTVAL }, + { "LED: Deck B: Digit 1 - E", 171 | CNT_INTVAL }, + { "LED: Deck B: Digit 1 - F", 170 | CNT_INTVAL }, + { "LED: Deck B: Digit 1 - G", 169 | CNT_INTVAL }, + { "LED: Deck B: Digit 1 - dot", 168 | CNT_INTVAL }, + { "LED: Deck B: Digit 2 - A", 167 | CNT_INTVAL }, + { "LED: Deck B: Digit 2 - B", 166 | CNT_INTVAL }, + { "LED: Deck B: Digit 2 - C", 165 | CNT_INTVAL }, + { "LED: Deck B: Digit 2 - D", 164 | CNT_INTVAL }, + { "LED: Deck B: Digit 2 - E", 163 | CNT_INTVAL }, + { "LED: Deck B: Digit 2 - F", 162 | CNT_INTVAL }, + { "LED: Deck B: Digit 2 - G", 161 | CNT_INTVAL }, + { "LED: Deck B: Digit 2 - dot", 160 | CNT_INTVAL }, + + { "LED: FX1: dry/wet", 153 | CNT_INTVAL }, + { "LED: FX1: 1", 154 | CNT_INTVAL }, + { "LED: FX1: 2", 155 | CNT_INTVAL }, + { "LED: FX1: 3", 156 | CNT_INTVAL }, + { "LED: FX1: Mode", 157 | CNT_INTVAL }, + { "LED: FX2: dry/wet", 129 | CNT_INTVAL }, + { "LED: FX2: 1", 130 | CNT_INTVAL }, + { "LED: FX2: 2", 131 | CNT_INTVAL }, + { "LED: FX2: 3", 132 | CNT_INTVAL }, + { "LED: FX2: Mode", 133 | CNT_INTVAL }, +}; + static int __devinit add_controls(struct caiaq_controller *c, int num, struct snd_usb_caiaqdev *dev) { @@ -354,6 +547,11 @@ int __devinit snd_usb_caiaq_control_init(struct snd_usb_caiaqdev *dev) ret = add_controls(kontrolx1_controller, ARRAY_SIZE(kontrolx1_controller), dev); break; + + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLS4): + ret = add_controls(kontrols4_controller, + ARRAY_SIZE(kontrols4_controller), dev); + break; } return ret; diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index cdfb856bddd2..6480c3283c05 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -36,7 +36,7 @@ #include "input.h" MODULE_AUTHOR("Daniel Mack <daniel@caiaq.de>"); -MODULE_DESCRIPTION("caiaq USB audio, version 1.3.21"); +MODULE_DESCRIPTION("caiaq USB audio"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, RigKontrol3}," @@ -48,7 +48,8 @@ MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, Audio 8 DJ}," "{Native Instruments, Session I/O}," "{Native Instruments, GuitarRig mobile}" - "{Native Instruments, Traktor Kontrol X1}"); + "{Native Instruments, Traktor Kontrol X1}" + "{Native Instruments, Traktor Kontrol S4}"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-max */ static char* id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* Id for this card */ @@ -134,6 +135,11 @@ static struct usb_device_id snd_usb_id_table[] = { .idVendor = USB_VID_NATIVEINSTRUMENTS, .idProduct = USB_PID_TRAKTORKONTROLX1 }, + { + .match_flags = USB_DEVICE_ID_MATCH_DEVICE, + .idVendor = USB_VID_NATIVEINSTRUMENTS, + .idProduct = USB_PID_TRAKTORKONTROLS4 + }, { /* terminator */ } }; diff --git a/sound/usb/caiaq/device.h b/sound/usb/caiaq/device.h index f1117ecc84fd..e3d8a3efb35b 100644 --- a/sound/usb/caiaq/device.h +++ b/sound/usb/caiaq/device.h @@ -16,6 +16,7 @@ #define USB_PID_SESSIONIO 0x1915 #define USB_PID_GUITARRIGMOBILE 0x0d8d #define USB_PID_TRAKTORKONTROLX1 0x2305 +#define USB_PID_TRAKTORKONTROLS4 0xbaff #define EP1_BUFSIZE 64 #define EP4_BUFSIZE 512 @@ -99,13 +100,14 @@ struct snd_usb_caiaqdev { struct snd_pcm_substream *sub_capture[MAX_STREAMS]; /* Controls */ - unsigned char control_state[64]; + unsigned char control_state[256]; + unsigned char ep8_out_buf[2]; /* Linux input */ #ifdef CONFIG_SND_USB_CAIAQ_INPUT struct input_dev *input_dev; char phys[64]; /* physical device path */ - unsigned short keycode[64]; + unsigned short keycode[128]; struct urb *ep4_in_urb; unsigned char ep4_in_buf[EP4_BUFSIZE]; #endif diff --git a/sound/usb/caiaq/input.c b/sound/usb/caiaq/input.c index dcb620796d9e..4432ef7a70a9 100644 --- a/sound/usb/caiaq/input.c +++ b/sound/usb/caiaq/input.c @@ -67,7 +67,12 @@ static unsigned short keycode_kore[] = { KEY_BRL_DOT5 }; -#define KONTROLX1_INPUTS 40 +#define KONTROLX1_INPUTS (40) +#define KONTROLS4_BUTTONS (12 * 8) +#define KONTROLS4_AXIS (46) + +#define KONTROLS4_BUTTON(X) ((X) + BTN_MISC) +#define KONTROLS4_ABS(X) ((X) + ABS_HAT0X) #define DEG90 (range / 2) #define DEG180 (range) @@ -139,6 +144,13 @@ static unsigned int decode_erp(unsigned char a, unsigned char b) #undef HIGH_PEAK #undef LOW_PEAK +static inline void snd_caiaq_input_report_abs(struct snd_usb_caiaqdev *dev, + int axis, const unsigned char *buf, + int offset) +{ + input_report_abs(dev->input_dev, axis, + (buf[offset * 2] << 8) | buf[offset * 2 + 1]); +} static void snd_caiaq_input_read_analog(struct snd_usb_caiaqdev *dev, const unsigned char *buf, @@ -148,36 +160,30 @@ static void snd_caiaq_input_read_analog(struct snd_usb_caiaqdev *dev, switch (dev->chip.usb_id) { case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL2): - input_report_abs(input_dev, ABS_X, (buf[4] << 8) | buf[5]); - input_report_abs(input_dev, ABS_Y, (buf[0] << 8) | buf[1]); - input_report_abs(input_dev, ABS_Z, (buf[2] << 8) | buf[3]); - input_sync(input_dev); + snd_caiaq_input_report_abs(dev, ABS_X, buf, 2); + snd_caiaq_input_report_abs(dev, ABS_Y, buf, 0); + snd_caiaq_input_report_abs(dev, ABS_Z, buf, 1); break; case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL3): - input_report_abs(input_dev, ABS_X, (buf[0] << 8) | buf[1]); - input_report_abs(input_dev, ABS_Y, (buf[2] << 8) | buf[3]); - input_report_abs(input_dev, ABS_Z, (buf[4] << 8) | buf[5]); - input_sync(input_dev); - break; case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER2): - input_report_abs(input_dev, ABS_X, (buf[0] << 8) | buf[1]); - input_report_abs(input_dev, ABS_Y, (buf[2] << 8) | buf[3]); - input_report_abs(input_dev, ABS_Z, (buf[4] << 8) | buf[5]); - input_sync(input_dev); + snd_caiaq_input_report_abs(dev, ABS_X, buf, 0); + snd_caiaq_input_report_abs(dev, ABS_Y, buf, 1); + snd_caiaq_input_report_abs(dev, ABS_Z, buf, 2); break; case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1): - input_report_abs(input_dev, ABS_HAT0X, (buf[8] << 8) | buf[9]); - input_report_abs(input_dev, ABS_HAT0Y, (buf[4] << 8) | buf[5]); - input_report_abs(input_dev, ABS_HAT1X, (buf[12] << 8) | buf[13]); - input_report_abs(input_dev, ABS_HAT1Y, (buf[2] << 8) | buf[3]); - input_report_abs(input_dev, ABS_HAT2X, (buf[14] << 8) | buf[15]); - input_report_abs(input_dev, ABS_HAT2Y, (buf[0] << 8) | buf[1]); - input_report_abs(input_dev, ABS_HAT3X, (buf[10] << 8) | buf[11]); - input_report_abs(input_dev, ABS_HAT3Y, (buf[6] << 8) | buf[7]); - input_sync(input_dev); + snd_caiaq_input_report_abs(dev, ABS_HAT0X, buf, 4); + snd_caiaq_input_report_abs(dev, ABS_HAT0Y, buf, 2); + snd_caiaq_input_report_abs(dev, ABS_HAT1X, buf, 6); + snd_caiaq_input_report_abs(dev, ABS_HAT1Y, buf, 1); + snd_caiaq_input_report_abs(dev, ABS_HAT2X, buf, 7); + snd_caiaq_input_report_abs(dev, ABS_HAT2Y, buf, 0); + snd_caiaq_input_report_abs(dev, ABS_HAT3X, buf, 5); + snd_caiaq_input_report_abs(dev, ABS_HAT3Y, buf, 3); break; } + + input_sync(input_dev); } static void snd_caiaq_input_read_erp(struct snd_usb_caiaqdev *dev, @@ -250,6 +256,150 @@ static void snd_caiaq_input_read_io(struct snd_usb_caiaqdev *dev, input_sync(input_dev); } +#define TKS4_MSGBLOCK_SIZE 16 + +static void snd_usb_caiaq_tks4_dispatch(struct snd_usb_caiaqdev *dev, + const unsigned char *buf, + unsigned int len) +{ + while (len) { + unsigned int i, block_id = (buf[0] << 8) | buf[1]; + + switch (block_id) { + case 0: + /* buttons */ + for (i = 0; i < KONTROLS4_BUTTONS; i++) + input_report_key(dev->input_dev, KONTROLS4_BUTTON(i), + (buf[4 + (i / 8)] >> (i % 8)) & 1); + break; + + case 1: + /* left wheel */ + input_report_abs(dev->input_dev, KONTROLS4_ABS(36), buf[9] | ((buf[8] & 0x3) << 8)); + /* right wheel */ + input_report_abs(dev->input_dev, KONTROLS4_ABS(37), buf[13] | ((buf[12] & 0x3) << 8)); + + /* rotary encoders */ + input_report_abs(dev->input_dev, KONTROLS4_ABS(38), buf[3] & 0xf); + input_report_abs(dev->input_dev, KONTROLS4_ABS(39), buf[4] >> 4); + input_report_abs(dev->input_dev, KONTROLS4_ABS(40), buf[4] & 0xf); + input_report_abs(dev->input_dev, KONTROLS4_ABS(41), buf[5] >> 4); + input_report_abs(dev->input_dev, KONTROLS4_ABS(42), buf[5] & 0xf); + input_report_abs(dev->input_dev, KONTROLS4_ABS(43), buf[6] >> 4); + input_report_abs(dev->input_dev, KONTROLS4_ABS(44), buf[6] & 0xf); + input_report_abs(dev->input_dev, KONTROLS4_ABS(45), buf[7] >> 4); + input_report_abs(dev->input_dev, KONTROLS4_ABS(46), buf[7] & 0xf); + + break; + case 2: + /* Volume Fader Channel D */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(0), buf, 1); + /* Volume Fader Channel B */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(1), buf, 2); + /* Volume Fader Channel A */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(2), buf, 3); + /* Volume Fader Channel C */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(3), buf, 4); + /* Loop Volume */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(4), buf, 6); + /* Crossfader */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(7), buf, 7); + + break; + + case 3: + /* Tempo Fader R */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(6), buf, 3); + /* Tempo Fader L */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(5), buf, 4); + /* Mic Volume */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(8), buf, 6); + /* Cue Mix */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(9), buf, 7); + + break; + + case 4: + /* Wheel distance sensor L */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(10), buf, 1); + /* Wheel distance sensor R */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(11), buf, 2); + /* Channel D EQ - Filter */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(12), buf, 3); + /* Channel D EQ - Low */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(13), buf, 4); + /* Channel D EQ - Mid */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(14), buf, 5); + /* Channel D EQ - Hi */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(15), buf, 6); + /* FX2 - dry/wet */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(16), buf, 7); + + break; + + case 5: + /* FX2 - 1 */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(17), buf, 1); + /* FX2 - 2 */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(18), buf, 2); + /* FX2 - 3 */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(19), buf, 3); + /* Channel B EQ - Filter */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(20), buf, 4); + /* Channel B EQ - Low */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(21), buf, 5); + /* Channel B EQ - Mid */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(22), buf, 6); + /* Channel B EQ - Hi */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(23), buf, 7); + + break; + + case 6: + /* Channel A EQ - Filter */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(24), buf, 1); + /* Channel A EQ - Low */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(25), buf, 2); + /* Channel A EQ - Mid */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(26), buf, 3); + /* Channel A EQ - Hi */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(27), buf, 4); + /* Channel C EQ - Filter */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(28), buf, 5); + /* Channel C EQ - Low */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(29), buf, 6); + /* Channel C EQ - Mid */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(30), buf, 7); + + break; + + case 7: + /* Channel C EQ - Hi */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(31), buf, 1); + /* FX1 - wet/dry */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(32), buf, 2); + /* FX1 - 1 */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(33), buf, 3); + /* FX1 - 2 */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(34), buf, 4); + /* FX1 - 3 */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(35), buf, 5); + + break; + + default: + debug("%s(): bogus block (id %d)\n", + __func__, block_id); + return; + } + + len -= TKS4_MSGBLOCK_SIZE; + buf += TKS4_MSGBLOCK_SIZE; + } + + input_sync(dev->input_dev); +} + static void snd_usb_caiaq_ep4_reply_dispatch(struct urb *urb) { struct snd_usb_caiaqdev *dev = urb->context; @@ -259,11 +409,11 @@ static void snd_usb_caiaq_ep4_reply_dispatch(struct urb *urb) if (urb->status || !dev || urb != dev->ep4_in_urb) return; - if (urb->actual_length < 24) - goto requeue; - switch (dev->chip.usb_id) { case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1): + if (urb->actual_length < 24) + goto requeue; + if (buf[0] & 0x3) snd_caiaq_input_read_io(dev, buf + 1, 7); @@ -271,6 +421,10 @@ static void snd_usb_caiaq_ep4_reply_dispatch(struct urb *urb) snd_caiaq_input_read_analog(dev, buf + 8, 16); break; + + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLS4): + snd_usb_caiaq_tks4_dispatch(dev, buf, urb->actual_length); + break; } requeue: @@ -289,6 +443,7 @@ static int snd_usb_caiaq_input_open(struct input_dev *idev) switch (dev->chip.usb_id) { case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1): + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLS4): if (usb_submit_urb(dev->ep4_in_urb, GFP_KERNEL) != 0) return -EIO; break; @@ -306,6 +461,7 @@ static void snd_usb_caiaq_input_close(struct input_dev *idev) switch (dev->chip.usb_id) { case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1): + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLS4): usb_kill_urb(dev->ep4_in_urb); break; } @@ -456,6 +612,46 @@ int snd_usb_caiaq_input_init(struct snd_usb_caiaqdev *dev) snd_usb_caiaq_set_auto_msg(dev, 1, 10, 5); break; + + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLS4): + input->evbit[0] = BIT_MASK(EV_KEY) | BIT_MASK(EV_ABS); + BUILD_BUG_ON(sizeof(dev->keycode) < KONTROLS4_BUTTONS); + for (i = 0; i < KONTROLS4_BUTTONS; i++) + dev->keycode[i] = KONTROLS4_BUTTON(i); + input->keycodemax = KONTROLS4_BUTTONS; + + for (i = 0; i < KONTROLS4_AXIS; i++) { + int axis = KONTROLS4_ABS(i); + input->absbit[BIT_WORD(axis)] |= BIT_MASK(axis); + } + + /* 36 analog potentiometers and faders */ + for (i = 0; i < 36; i++) + input_set_abs_params(input, KONTROLS4_ABS(i), 0, 0xfff, 0, 10); + + /* 2 encoder wheels */ + input_set_abs_params(input, KONTROLS4_ABS(36), 0, 0x3ff, 0, 1); + input_set_abs_params(input, KONTROLS4_ABS(37), 0, 0x3ff, 0, 1); + + /* 9 rotary encoders */ + for (i = 0; i < 9; i++) + input_set_abs_params(input, KONTROLS4_ABS(38+i), 0, 0xf, 0, 1); + + dev->ep4_in_urb = usb_alloc_urb(0, GFP_KERNEL); + if (!dev->ep4_in_urb) { + ret = -ENOMEM; + goto exit_free_idev; + } + + usb_fill_bulk_urb(dev->ep4_in_urb, usb_dev, + usb_rcvbulkpipe(usb_dev, 0x4), + dev->ep4_in_buf, EP4_BUFSIZE, + snd_usb_caiaq_ep4_reply_dispatch, dev); + + snd_usb_caiaq_set_auto_msg(dev, 1, 10, 5); + + break; + default: /* no input methods supported on this device */ goto exit_free_idev; diff --git a/sound/usb/card.c b/sound/usb/card.c index 7a8ac1d81be7..800f7cb4f251 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -126,7 +126,7 @@ static void snd_usb_stream_disconnect(struct list_head *head) for (idx = 0; idx < 2; idx++) { subs = &as->substream[idx]; if (!subs->num_formats) - return; + continue; snd_usb_release_substream_urbs(subs, 1); subs->interface = -1; } @@ -216,8 +216,13 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) } switch (protocol) { + default: + snd_printdd(KERN_WARNING "unknown interface protocol %#02x, assuming v1\n", + protocol); + /* fall through */ + case UAC_VERSION_1: { - struct uac_ac_header_descriptor_v1 *h1 = control_header; + struct uac1_ac_header_descriptor *h1 = control_header; if (!h1->bInCollection) { snd_printk(KERN_INFO "skipping empty audio interface (v1)\n"); @@ -253,10 +258,6 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) break; } - - default: - snd_printk(KERN_ERR "unknown protocol version 0x%02x\n", protocol); - return -EINVAL; } return 0; @@ -299,9 +300,13 @@ static int snd_usb_audio_create(struct usb_device *dev, int idx, *rchip = NULL; - if (snd_usb_get_speed(dev) != USB_SPEED_LOW && - snd_usb_get_speed(dev) != USB_SPEED_FULL && - snd_usb_get_speed(dev) != USB_SPEED_HIGH) { + switch (snd_usb_get_speed(dev)) { + case USB_SPEED_LOW: + case USB_SPEED_FULL: + case USB_SPEED_HIGH: + case USB_SPEED_SUPER: + break; + default: snd_printk(KERN_ERR "unknown device speed %d\n", snd_usb_get_speed(dev)); return -ENXIO; } @@ -377,11 +382,22 @@ static int snd_usb_audio_create(struct usb_device *dev, int idx, if (len < sizeof(card->longname)) usb_make_path(dev, card->longname + len, sizeof(card->longname) - len); - strlcat(card->longname, - snd_usb_get_speed(dev) == USB_SPEED_LOW ? ", low speed" : - snd_usb_get_speed(dev) == USB_SPEED_FULL ? ", full speed" : - ", high speed", - sizeof(card->longname)); + switch (snd_usb_get_speed(dev)) { + case USB_SPEED_LOW: + strlcat(card->longname, ", low speed", sizeof(card->longname)); + break; + case USB_SPEED_FULL: + strlcat(card->longname, ", full speed", sizeof(card->longname)); + break; + case USB_SPEED_HIGH: + strlcat(card->longname, ", high speed", sizeof(card->longname)); + break; + case USB_SPEED_SUPER: + strlcat(card->longname, ", super speed", sizeof(card->longname)); + break; + default: + break; + } snd_usb_audio_create_proc(chip); @@ -465,7 +481,13 @@ static void *snd_usb_audio_probe(struct usb_device *dev, goto __error; } - chip->ctrl_intf = alts; + /* + * For devices with more than one control interface, we assume the + * first contains the audio controls. We might need a more specific + * check here in the future. + */ + if (!chip->ctrl_intf) + chip->ctrl_intf = alts; if (err > 0) { /* create normal USB audio interfaces */ diff --git a/sound/usb/clock.c b/sound/usb/clock.c index b5855114667e..7754a1034545 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -19,33 +19,19 @@ #include <linux/bitops.h> #include <linux/init.h> -#include <linux/list.h> -#include <linux/slab.h> #include <linux/string.h> #include <linux/usb.h> -#include <linux/moduleparam.h> -#include <linux/mutex.h> #include <linux/usb/audio.h> #include <linux/usb/audio-v2.h> #include <sound/core.h> #include <sound/info.h> #include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/initval.h> #include "usbaudio.h" #include "card.h" -#include "midi.h" -#include "mixer.h" -#include "proc.h" -#include "quirks.h" -#include "endpoint.h" #include "helper.h" -#include "debug.h" -#include "pcm.h" -#include "urb.h" -#include "format.h" +#include "clock.h" static struct uac_clock_source_descriptor * snd_usb_find_clock_source(struct usb_host_interface *ctrl_iface, @@ -134,10 +120,7 @@ static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, int source_id) return !!data; } -/* Try to find the clock source ID of a given clock entity */ - static int __uac_clock_find_source(struct snd_usb_audio *chip, - struct usb_host_interface *host_iface, int entity_id, unsigned long *visited) { struct uac_clock_source_descriptor *source; @@ -154,11 +137,11 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, } /* first, see if the ID we're looking for is a clock source already */ - source = snd_usb_find_clock_source(host_iface, entity_id); + source = snd_usb_find_clock_source(chip->ctrl_intf, entity_id); if (source) return source->bClockID; - selector = snd_usb_find_clock_selector(host_iface, entity_id); + selector = snd_usb_find_clock_selector(chip->ctrl_intf, entity_id); if (selector) { int ret; @@ -168,6 +151,8 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, if (ret < 0) return ret; + /* Selector values are one-based */ + if (ret > selector->bNrInPins || ret < 1) { printk(KERN_ERR "%s(): selector reported illegal value, id %d, ret %d\n", @@ -176,27 +161,35 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, return -EINVAL; } - return __uac_clock_find_source(chip, host_iface, - selector->baCSourceID[ret-1], + return __uac_clock_find_source(chip, selector->baCSourceID[ret-1], visited); } /* FIXME: multipliers only act as pass-thru element for now */ - multiplier = snd_usb_find_clock_multiplier(host_iface, entity_id); + multiplier = snd_usb_find_clock_multiplier(chip->ctrl_intf, entity_id); if (multiplier) - return __uac_clock_find_source(chip, host_iface, - multiplier->bCSourceID, visited); + return __uac_clock_find_source(chip, multiplier->bCSourceID, + visited); return -EINVAL; } -int snd_usb_clock_find_source(struct snd_usb_audio *chip, - struct usb_host_interface *host_iface, - int entity_id) +/* + * For all kinds of sample rate settings and other device queries, + * the clock source (end-leaf) must be used. However, clock selectors, + * clock multipliers and sample rate converters may be specified as + * clock source input to terminal. This functions walks the clock path + * to its end and tries to find the source. + * + * The 'visited' bitfield is used internally to detect recursive loops. + * + * Returns the clock source UnitID (>=0) on success, or an error. + */ +int snd_usb_clock_find_source(struct snd_usb_audio *chip, int entity_id) { DECLARE_BITMAP(visited, 256); memset(visited, 0, sizeof(visited)); - return __uac_clock_find_source(chip, host_iface, entity_id, visited); + return __uac_clock_find_source(chip, entity_id, visited); } static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface, @@ -211,11 +204,8 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface, ep = get_endpoint(alts, 0)->bEndpointAddress; /* if endpoint doesn't have sampling rate control, bail out */ - if (!(fmt->attributes & UAC_EP_CS_ATTR_SAMPLE_RATE)) { - snd_printk(KERN_WARNING "%d:%d:%d: endpoint lacks sample rate attribute bit, cannot set.\n", - dev->devnum, iface, fmt->altsetting); + if (!(fmt->attributes & UAC_EP_CS_ATTR_SAMPLE_RATE)) return 0; - } data[0] = rate; data[1] = rate >> 8; @@ -254,12 +244,13 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, struct usb_device *dev = chip->dev; unsigned char data[4]; int err, crate; - int clock = snd_usb_clock_find_source(chip, chip->ctrl_intf, fmt->clock); + int clock = snd_usb_clock_find_source(chip, fmt->clock); if (clock < 0) return clock; if (!uac_clock_source_is_valid(chip, clock)) { + /* TODO: should we try to find valid clock setups by ourself? */ snd_printk(KERN_ERR "%d:%d:%d: clock source %d is not valid, cannot use\n", dev->devnum, iface, fmt->altsetting, clock); return -ENXIO; @@ -304,12 +295,11 @@ int snd_usb_init_sample_rate(struct snd_usb_audio *chip, int iface, switch (altsd->bInterfaceProtocol) { case UAC_VERSION_1: + default: return set_sample_rate_v1(chip, iface, alts, fmt, rate); case UAC_VERSION_2: return set_sample_rate_v2(chip, iface, alts, fmt, rate); } - - return -EINVAL; } diff --git a/sound/usb/clock.h b/sound/usb/clock.h index beb253684e2d..46630936d31f 100644 --- a/sound/usb/clock.h +++ b/sound/usb/clock.h @@ -5,8 +5,6 @@ int snd_usb_init_sample_rate(struct snd_usb_audio *chip, int iface, struct usb_host_interface *alts, struct audioformat *fmt, int rate); -int snd_usb_clock_find_source(struct snd_usb_audio *chip, - struct usb_host_interface *host_iface, - int entity_id); +int snd_usb_clock_find_source(struct snd_usb_audio *chip, int entity_id); #endif /* __USBAUDIO_CLOCK_H */ diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 6f6596cf2b19..b0ef9f501896 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -33,6 +33,7 @@ #include "pcm.h" #include "helper.h" #include "format.h" +#include "clock.h" /* * free a substream @@ -274,8 +275,14 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) /* get audio formats */ switch (protocol) { + default: + snd_printdd(KERN_WARNING "%d:%u:%d: unknown interface protocol %#02x, assuming v1\n", + dev->devnum, iface_no, altno, protocol); + protocol = UAC_VERSION_1; + /* fall through */ + case UAC_VERSION_1: { - struct uac_as_header_descriptor_v1 *as = + struct uac1_as_header_descriptor *as = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL); if (!as) { @@ -297,7 +304,7 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) case UAC_VERSION_2: { struct uac2_input_terminal_descriptor *input_term; struct uac2_output_terminal_descriptor *output_term; - struct uac_as_header_descriptor_v2 *as = + struct uac2_as_header_descriptor *as = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL); if (!as) { @@ -335,11 +342,6 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) dev->devnum, iface_no, altno, as->bTerminalLink); continue; } - - default: - snd_printk(KERN_ERR "%d:%u:%d : unknown interface protocol %04x\n", - dev->devnum, iface_no, altno, protocol); - continue; } /* get format type */ @@ -403,8 +405,6 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) break; case USB_ID(0x041e, 0x3020): /* Creative SB Audigy 2 NX */ case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */ - case USB_ID(0x0763, 0x2080): /* M-Audio Fast Track Ultra 8 */ - case USB_ID(0x0763, 0x2081): /* M-Audio Fast Track Ultra 8R */ /* doesn't set the sample rate attribute, but supports it */ fp->attributes |= UAC_EP_CS_ATTR_SAMPLE_RATE; break; diff --git a/sound/usb/format.c b/sound/usb/format.c index 30364aba79cc..69148212aa70 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -49,7 +49,8 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip, u64 pcm_formats; switch (protocol) { - case UAC_VERSION_1: { + case UAC_VERSION_1: + default: { struct uac_format_type_i_discrete_descriptor *fmt = _fmt; sample_width = fmt->bBitResolution; sample_bytes = fmt->bSubframeSize; @@ -64,9 +65,6 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip, format <<= 1; break; } - - default: - return -EINVAL; } pcm_formats = 0; @@ -264,13 +262,12 @@ static int parse_uac2_sample_rate_range(struct audioformat *fp, int nr_triplets, * on the audioformat table (audio class v2). */ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip, - struct audioformat *fp, - struct usb_host_interface *iface) + struct audioformat *fp) { struct usb_device *dev = chip->dev; unsigned char tmp[2], *data; int nr_triplets, data_size, ret = 0; - int clock = snd_usb_clock_find_source(chip, chip->ctrl_intf, fp->clock); + int clock = snd_usb_clock_find_source(chip, fp->clock); if (clock < 0) { snd_printk(KERN_ERR "%s(): unable to find clock source (clock %d)\n", @@ -385,13 +382,17 @@ static int parse_audio_format_i(struct snd_usb_audio *chip, * audio class v2 uses class specific EP0 range requests for that. */ switch (protocol) { + default: + snd_printdd(KERN_WARNING "%d:%u:%d : invalid protocol version %d, assuming v1\n", + chip->dev->devnum, fp->iface, fp->altsetting, protocol); + /* fall through */ case UAC_VERSION_1: fp->channels = fmt->bNrChannels; ret = parse_audio_format_rates_v1(chip, fp, (unsigned char *) fmt, 7); break; case UAC_VERSION_2: /* fp->channels is already set in this case */ - ret = parse_audio_format_rates_v2(chip, fp, iface); + ret = parse_audio_format_rates_v2(chip, fp); break; } @@ -435,6 +436,10 @@ static int parse_audio_format_ii(struct snd_usb_audio *chip, fp->channels = 1; switch (protocol) { + default: + snd_printdd(KERN_WARNING "%d:%u:%d : invalid protocol version %d, assuming v1\n", + chip->dev->devnum, fp->iface, fp->altsetting, protocol); + /* fall through */ case UAC_VERSION_1: { struct uac_format_type_ii_discrete_descriptor *fmt = _fmt; brate = le16_to_cpu(fmt->wMaxBitRate); @@ -450,7 +455,7 @@ static int parse_audio_format_ii(struct snd_usb_audio *chip, framesize = le16_to_cpu(fmt->wSamplesPerFrame); snd_printd(KERN_INFO "found format II with max.bitrate = %d, frame size=%d\n", brate, framesize); fp->frame_size = framesize; - ret = parse_audio_format_rates_v2(chip, fp, iface); + ret = parse_audio_format_rates_v2(chip, fp); break; } } diff --git a/sound/usb/helper.c b/sound/usb/helper.c index d48d6f8f6ac9..f280c1903c25 100644 --- a/sound/usb/helper.c +++ b/sound/usb/helper.c @@ -103,11 +103,16 @@ int snd_usb_ctl_msg(struct usb_device *dev, unsigned int pipe, __u8 request, unsigned char snd_usb_parse_datainterval(struct snd_usb_audio *chip, struct usb_host_interface *alts) { - if (snd_usb_get_speed(chip->dev) == USB_SPEED_HIGH && - get_endpoint(alts, 0)->bInterval >= 1 && - get_endpoint(alts, 0)->bInterval <= 4) - return get_endpoint(alts, 0)->bInterval - 1; - else - return 0; + switch (snd_usb_get_speed(chip->dev)) { + case USB_SPEED_HIGH: + case USB_SPEED_SUPER: + if (get_endpoint(alts, 0)->bInterval >= 1 && + get_endpoint(alts, 0)->bInterval <= 4) + return get_endpoint(alts, 0)->bInterval - 1; + break; + default: + break; + } + return 0; } diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 46785643c66d..156cd0716c42 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -434,7 +434,7 @@ static void snd_usbmidi_maudio_broken_running_status_input( u8 cin = buffer[i] & 0x0f; struct usbmidi_in_port *port = &ep->ports[cable]; int length; - + length = snd_usbmidi_cin_length[cin]; if (cin == 0xf && buffer[i + 1] >= 0xf8) ; /* realtime msg: no running status change */ @@ -628,13 +628,13 @@ static struct usb_protocol_ops snd_usbmidi_standard_ops = { static struct usb_protocol_ops snd_usbmidi_midiman_ops = { .input = snd_usbmidi_midiman_input, - .output = snd_usbmidi_standard_output, + .output = snd_usbmidi_standard_output, .output_packet = snd_usbmidi_output_midiman_packet, }; static struct usb_protocol_ops snd_usbmidi_maudio_broken_running_status_ops = { .input = snd_usbmidi_maudio_broken_running_status_input, - .output = snd_usbmidi_standard_output, + .output = snd_usbmidi_standard_output, .output_packet = snd_usbmidi_output_standard_packet, }; @@ -834,7 +834,14 @@ static void snd_usbmidi_us122l_output(struct snd_usb_midi_out_endpoint *ep, if (!ep->ports[0].active) return; - count = snd_usb_get_speed(ep->umidi->dev) == USB_SPEED_HIGH ? 1 : 2; + switch (snd_usb_get_speed(ep->umidi->dev)) { + case USB_SPEED_HIGH: + case USB_SPEED_SUPER: + count = 1; + break; + default: + count = 2; + } count = snd_rawmidi_transmit(ep->ports[0].substream, urb->transfer_buffer, count); @@ -1248,7 +1255,7 @@ static void snd_usbmidi_out_endpoint_delete(struct snd_usb_midi_out_endpoint *ep */ static int snd_usbmidi_out_endpoint_create(struct snd_usb_midi* umidi, struct snd_usb_midi_endpoint_info* ep_info, - struct snd_usb_midi_endpoint* rep) + struct snd_usb_midi_endpoint* rep) { struct snd_usb_midi_out_endpoint* ep; unsigned int i; @@ -1398,7 +1405,7 @@ static void snd_usbmidi_rawmidi_free(struct snd_rawmidi *rmidi) } static struct snd_rawmidi_substream *snd_usbmidi_find_substream(struct snd_usb_midi* umidi, - int stream, int number) + int stream, int number) { struct list_head* list; @@ -1811,7 +1818,7 @@ static int snd_usbmidi_detect_endpoints(struct snd_usb_midi* umidi, snd_usbmidi_switch_roland_altsetting(umidi); if (endpoint[0].out_ep || endpoint[0].in_ep) - return 0; + return 0; intf = umidi->iface; if (!intf || intf->num_altsetting < 1) @@ -1849,7 +1856,7 @@ static int snd_usbmidi_detect_per_port_endpoints(struct snd_usb_midi* umidi, struct snd_usb_midi_endpoint_info* endpoints) { int err, i; - + err = snd_usbmidi_detect_endpoints(umidi, endpoints, MIDI_MAX_ENDPOINTS); for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) { if (endpoints[i].out_ep) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 736d134cc03c..f2d74d654b3c 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -26,6 +26,22 @@ * */ +/* + * TODOs, for both the mixer and the streaming interfaces: + * + * - support for UAC2 effect units + * - support for graphical equalizers + * - RANGE and MEM set commands (UAC2) + * - RANGE and MEM interrupt dispatchers (UAC2) + * - audio channel clustering (UAC2) + * - audio sample rate converter units (UAC2) + * - proper handling of clock multipliers (UAC2) + * - dispatch clock change notifications (UAC2) + * - stop PCM streams which use a clock that became invalid + * - stop PCM streams which use a clock selector that has changed + * - parse available sample rates again when clock sources changed + */ + #include <linux/bitops.h> #include <linux/init.h> #include <linux/list.h> @@ -275,28 +291,28 @@ static int get_abs_value(struct usb_mixer_elem_info *cval, int val) static int get_ctl_value_v1(struct usb_mixer_elem_info *cval, int request, int validx, int *value_ret) { + struct snd_usb_audio *chip = cval->mixer->chip; unsigned char buf[2]; int val_len = cval->val_type >= USB_MIXER_S16 ? 2 : 1; int timeout = 10; while (timeout-- > 0) { - if (snd_usb_ctl_msg(cval->mixer->chip->dev, - usb_rcvctrlpipe(cval->mixer->chip->dev, 0), - request, + if (snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), request, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, - validx, cval->mixer->ctrlif | (cval->id << 8), + validx, snd_usb_ctrl_intf(chip) | (cval->id << 8), buf, val_len, 100) >= val_len) { *value_ret = convert_signed_value(cval, snd_usb_combine_bytes(buf, val_len)); return 0; } } snd_printdd(KERN_ERR "cannot get ctl value: req = %#x, wValue = %#x, wIndex = %#x, type = %d\n", - request, validx, cval->mixer->ctrlif | (cval->id << 8), cval->val_type); + request, validx, snd_usb_ctrl_intf(chip) | (cval->id << 8), cval->val_type); return -EINVAL; } static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, int validx, int *value_ret) { + struct snd_usb_audio *chip = cval->mixer->chip; unsigned char buf[2 + 3*sizeof(__u16)]; /* enough space for one range */ unsigned char *val; int ret, size; @@ -312,16 +328,14 @@ static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, int v memset(buf, 0, sizeof(buf)); - ret = snd_usb_ctl_msg(cval->mixer->chip->dev, - usb_rcvctrlpipe(cval->mixer->chip->dev, 0), - bRequest, + ret = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), bRequest, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, - validx, cval->mixer->ctrlif | (cval->id << 8), + validx, snd_usb_ctrl_intf(chip) | (cval->id << 8), buf, size, 1000); if (ret < 0) { snd_printk(KERN_ERR "cannot get ctl value: req = %#x, wValue = %#x, wIndex = %#x, type = %d\n", - request, validx, cval->mixer->ctrlif | (cval->id << 8), cval->val_type); + request, validx, snd_usb_ctrl_intf(chip) | (cval->id << 8), cval->val_type); return ret; } @@ -397,6 +411,7 @@ static int get_cur_mix_value(struct usb_mixer_elem_info *cval, int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, int request, int validx, int value_set) { + struct snd_usb_audio *chip = cval->mixer->chip; unsigned char buf[2]; int val_len, timeout = 10; @@ -419,15 +434,14 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, buf[0] = value_set & 0xff; buf[1] = (value_set >> 8) & 0xff; while (timeout-- > 0) - if (snd_usb_ctl_msg(cval->mixer->chip->dev, - usb_sndctrlpipe(cval->mixer->chip->dev, 0), - request, + if (snd_usb_ctl_msg(chip->dev, + usb_sndctrlpipe(chip->dev, 0), request, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT, - validx, cval->mixer->ctrlif | (cval->id << 8), + validx, snd_usb_ctrl_intf(chip) | (cval->id << 8), buf, val_len, 100) >= 0) return 0; snd_printdd(KERN_ERR "cannot set ctl value: req = %#x, wValue = %#x, wIndex = %#x, type = %d, data = %#x/%#x\n", - request, validx, cval->mixer->ctrlif | (cval->id << 8), cval->val_type, buf[0], buf[1]); + request, validx, snd_usb_ctrl_intf(chip) | (cval->id << 8), cval->val_type, buf[0], buf[1]); return -EINVAL; } @@ -582,9 +596,9 @@ static int get_term_name(struct mixer_build *state, struct usb_audio_term *iterm switch (iterm->type >> 16) { case UAC_SELECTOR_UNIT: strcpy(name, "Selector"); return 8; - case UAC_PROCESSING_UNIT_V1: + case UAC1_PROCESSING_UNIT: strcpy(name, "Process Unit"); return 12; - case UAC_EXTENSION_UNIT_V1: + case UAC1_EXTENSION_UNIT: strcpy(name, "Ext Unit"); return 8; case UAC_MIXER_UNIT: strcpy(name, "Mixer"); return 5; @@ -672,8 +686,8 @@ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_ term->name = uac_selector_unit_iSelector(d); return 0; } - case UAC_PROCESSING_UNIT_V1: - case UAC_EXTENSION_UNIT_V1: { + case UAC1_PROCESSING_UNIT: + case UAC1_EXTENSION_UNIT: { struct uac_processing_unit_descriptor *d = p1; if (d->bNrInPins) { id = d->baSourceID[0]; @@ -767,7 +781,7 @@ static int get_min_max(struct usb_mixer_elem_info *cval, int default_min) if (get_ctl_value(cval, UAC_GET_MAX, (cval->control << 8) | minchn, &cval->max) < 0 || get_ctl_value(cval, UAC_GET_MIN, (cval->control << 8) | minchn, &cval->min) < 0) { snd_printd(KERN_ERR "%d:%d: cannot get min/max values for control %d (id %d)\n", - cval->id, cval->mixer->ctrlif, cval->control, cval->id); + cval->id, snd_usb_ctrl_intf(cval->mixer->chip), cval->control, cval->id); return -EINVAL; } if (get_ctl_value(cval, UAC_GET_RES, (cval->control << 8) | minchn, &cval->res) < 0) { @@ -1199,14 +1213,6 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void } } else { /* UAC_VERSION_2 */ for (i = 0; i < 30/2; i++) { - /* From the USB Audio spec v2.0: - bmaControls() is a (ch+1)-element array of 4-byte bitmaps, - each containing a set of bit pairs. If a Control is present, - it must be Host readable. If a certain Control is not - present then the bit pair must be set to 0b00. - If a Control is present but read-only, the bit pair must be - set to 0b01. If a Control is also Host programmable, the bit - pair must be set to 0b11. The value 0b10 is not allowed. */ unsigned int ch_bits = 0; unsigned int ch_read_only = 0; @@ -1634,9 +1640,10 @@ static int mixer_ctl_selector_info(struct snd_kcontrol *kcontrol, struct snd_ctl uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = cval->max; - if ((int)uinfo->value.enumerated.item >= cval->max) + if (uinfo->value.enumerated.item >= cval->max) uinfo->value.enumerated.item = cval->max - 1; - strcpy(uinfo->value.enumerated.name, itemlist[uinfo->value.enumerated.item]); + strlcpy(uinfo->value.enumerated.name, itemlist[uinfo->value.enumerated.item], + sizeof(uinfo->value.enumerated.name)); return 0; } @@ -1855,13 +1862,13 @@ static int parse_audio_unit(struct mixer_build *state, int unitid) return parse_audio_selector_unit(state, unitid, p1); case UAC_FEATURE_UNIT: return parse_audio_feature_unit(state, unitid, p1); - case UAC_PROCESSING_UNIT_V1: + case UAC1_PROCESSING_UNIT: /* UAC2_EFFECT_UNIT has the same value */ if (state->mixer->protocol == UAC_VERSION_1) return parse_audio_processing_unit(state, unitid, p1); else return 0; /* FIXME - effect units not implemented yet */ - case UAC_EXTENSION_UNIT_V1: + case UAC1_EXTENSION_UNIT: /* UAC2_PROCESSING_UNIT_V2 has the same value */ if (state->mixer->protocol == UAC_VERSION_1) return parse_audio_extension_unit(state, unitid, p1); @@ -1905,7 +1912,7 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) struct usb_host_interface *hostif; void *p; - hostif = &usb_ifnum_to_if(mixer->chip->dev, mixer->ctrlif)->altsetting[0]; + hostif = mixer->chip->ctrl_intf; memset(&state, 0, sizeof(state)); state.chip = mixer->chip; state.mixer = mixer; @@ -1925,7 +1932,7 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) p = NULL; while ((p = snd_usb_find_csint_desc(hostif->extra, hostif->extralen, p, UAC_OUTPUT_TERMINAL)) != NULL) { if (mixer->protocol == UAC_VERSION_1) { - struct uac_output_terminal_descriptor_v1 *desc = p; + struct uac1_output_terminal_descriptor *desc = p; if (desc->bLength < sizeof(*desc)) continue; /* invalid descriptor? */ @@ -1997,7 +2004,7 @@ static void snd_usb_mixer_proc_read(struct snd_info_entry *entry, list_for_each_entry(mixer, &chip->mixer_list, list) { snd_iprintf(buffer, "USB Mixer: usb_id=0x%08x, ctrlif=%i, ctlerr=%i\n", - chip->usb_id, mixer->ctrlif, + chip->usb_id, snd_usb_ctrl_intf(chip), mixer->ignore_ctl_error); snd_iprintf(buffer, "Card: %s\n", chip->card->longname); for (unitid = 0; unitid < MAX_ID_ELEMS; unitid++) { @@ -2115,7 +2122,7 @@ static int snd_usb_mixer_status_create(struct usb_mixer_interface *mixer) int buffer_length; unsigned int epnum; - hostif = &usb_ifnum_to_if(mixer->chip->dev, mixer->ctrlif)->altsetting[0]; + hostif = mixer->chip->ctrl_intf; /* we need one interrupt input endpoint */ if (get_iface_desc(hostif)->bNumEndpoints < 1) return 0; @@ -2158,7 +2165,6 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, if (!mixer) return -ENOMEM; mixer->chip = chip; - mixer->ctrlif = ctrlif; mixer->ignore_ctl_error = ignore_error; mixer->id_elems = kcalloc(MAX_ID_ELEMS, sizeof(*mixer->id_elems), GFP_KERNEL); @@ -2168,7 +2174,15 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, } host_iface = &usb_ifnum_to_if(chip->dev, ctrlif)->altsetting[0]; - mixer->protocol = get_iface_desc(host_iface)->bInterfaceProtocol; + switch (get_iface_desc(host_iface)->bInterfaceProtocol) { + case UAC_VERSION_1: + default: + mixer->protocol = UAC_VERSION_1; + break; + case UAC_VERSION_2: + mixer->protocol = UAC_VERSION_2; + break; + } if ((err = snd_usb_mixer_controls(mixer)) < 0 || (err = snd_usb_mixer_status_create(mixer)) < 0) diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index a7cf1007fbb0..26c636c5c93a 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -3,7 +3,6 @@ struct usb_mixer_interface { struct snd_usb_audio *chip; - unsigned int ctrlif; struct list_head list; unsigned int ignore_ctl_error; struct urb *urb; diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 456829882f40..f49756c1b837 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -173,13 +173,12 @@ int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface, switch (altsd->bInterfaceProtocol) { case UAC_VERSION_1: + default: return init_pitch_v1(chip, iface, alts, fmt); case UAC_VERSION_2: return init_pitch_v2(chip, iface, alts, fmt); } - - return -EINVAL; } /* @@ -467,7 +466,7 @@ static int hw_check_valid_format(struct snd_usb_substream *subs, return 0; } /* check whether the period time is >= the data packet interval */ - if (snd_usb_get_speed(subs->dev) == USB_SPEED_HIGH) { + if (snd_usb_get_speed(subs->dev) != USB_SPEED_FULL) { ptime = 125 * (1 << fp->datainterval); if (ptime > pt->max || (ptime == pt->max && pt->openmax)) { hwc_debug(" > check: ptime %u > max %u\n", ptime, pt->max); @@ -636,7 +635,7 @@ static int hw_rule_period_time(struct snd_pcm_hw_params *params, min_datainterval = min(min_datainterval, fp->datainterval); } if (min_datainterval == 0xff) { - hwc_debug(" --> get emtpy\n"); + hwc_debug(" --> get empty\n"); it->empty = 1; return -EINVAL; } @@ -735,7 +734,7 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre } param_period_time_if_needed = SNDRV_PCM_HW_PARAM_PERIOD_TIME; - if (snd_usb_get_speed(subs->dev) != USB_SPEED_HIGH) + if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL) /* full speed devices have fixed data packet interval */ ptmin = 1000; if (ptmin == 1000) diff --git a/sound/usb/pcm.h b/sound/usb/pcm.h index 1c931b68f3b5..ed3e283f618d 100644 --- a/sound/usb/pcm.h +++ b/sound/usb/pcm.h @@ -7,8 +7,5 @@ int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface, struct usb_host_interface *alts, struct audioformat *fmt); -int snd_usb_init_sample_rate(struct snd_usb_audio *chip, int iface, - struct usb_host_interface *alts, - struct audioformat *fmt, int rate); #endif /* __USBAUDIO_PCM_H */ diff --git a/sound/usb/proc.c b/sound/usb/proc.c index f5e3f356b95f..3c650ab3c91d 100644 --- a/sound/usb/proc.c +++ b/sound/usb/proc.c @@ -107,7 +107,7 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s } snd_iprintf(buffer, "\n"); } - if (snd_usb_get_speed(subs->dev) == USB_SPEED_HIGH) + if (snd_usb_get_speed(subs->dev) != USB_SPEED_FULL) snd_iprintf(buffer, " Data packet interval: %d us\n", 125 * (1 << fp->datainterval)); // snd_iprintf(buffer, " Max Packet Size = %d\n", fp->maxpacksize); diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index f8797f61a24b..682e3e06b07c 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -240,9 +240,21 @@ YAMAHA_DEVICE(0x104f, NULL), YAMAHA_DEVICE(0x1050, NULL), YAMAHA_DEVICE(0x1051, NULL), YAMAHA_DEVICE(0x1052, NULL), +YAMAHA_INTERFACE(0x1053, 0, NULL), +YAMAHA_INTERFACE(0x1054, 0, NULL), +YAMAHA_DEVICE(0x1055, NULL), +YAMAHA_DEVICE(0x1056, NULL), +YAMAHA_DEVICE(0x1057, NULL), +YAMAHA_DEVICE(0x1058, NULL), +YAMAHA_DEVICE(0x1059, NULL), +YAMAHA_DEVICE(0x105a, NULL), +YAMAHA_DEVICE(0x105b, NULL), +YAMAHA_DEVICE(0x105c, NULL), +YAMAHA_DEVICE(0x105d, NULL), YAMAHA_DEVICE(0x2000, "DGP-7"), YAMAHA_DEVICE(0x2001, "DGP-5"), YAMAHA_DEVICE(0x2002, NULL), +YAMAHA_DEVICE(0x2003, NULL), YAMAHA_DEVICE(0x5000, "CS1D"), YAMAHA_DEVICE(0x5001, "DSP1D"), YAMAHA_DEVICE(0x5002, "DME32"), @@ -1136,11 +1148,34 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { + /* has ID 0x0066 when not in "Advanced Driver" mode */ + USB_DEVICE(0x0582, 0x0064), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "EDIROL", */ + /* .product_name = "PCR-1", */ + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = -1 + } + } + } +}, +{ /* has ID 0x0067 when not in "Advanced Driver" mode */ USB_DEVICE(0x0582, 0x0065), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "EDIROL", - .product_name = "PCR-1", + /* .vendor_name = "EDIROL", */ + /* .product_name = "PCR-1", */ .ifnum = 0, .type = QUIRK_MIDI_FIXED_ENDPOINT, .data = & (const struct snd_usb_midi_endpoint_info) { @@ -1525,6 +1560,50 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + /* has ID 0x0110 when not in Advanced Driver mode */ + USB_DEVICE_VENDOR_SPEC(0x0582, 0x010f), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "Roland", */ + /* .product_name = "A-PRO", */ + .ifnum = 1, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0003, + .in_cables = 0x0007 + } + } +}, +{ + USB_DEVICE(0x0582, 0x0113), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "BOSS", */ + /* .product_name = "ME-25", */ + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0001, + .in_cables = 0x0001 + } + }, + { + .ifnum = -1 + } + } + } +}, /* Guillemot devices */ { @@ -1830,7 +1909,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), USB_DEVICE(0x0763, 0x2080), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { /* .vendor_name = "M-Audio", */ - /* .product_name = "Fast Track Ultra 8", */ + /* .product_name = "Fast Track Ultra", */ .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, .data = & (const struct snd_usb_audio_quirk[]) { @@ -1840,11 +1919,51 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x01, + .ep_attr = 0x09, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + } + } }, { .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x81, + .ep_attr = 0x05, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + } + } }, /* interface 3 (MIDI) is standard compliant */ { @@ -1867,11 +1986,51 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x01, + .ep_attr = 0x09, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + } + } }, { .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x81, + .ep_attr = 0x05, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + } + } }, /* interface 3 (MIDI) is standard compliant */ { @@ -2152,7 +2311,21 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - USB_DEVICE_VENDOR_SPEC(0x2040, 0x7201), + USB_DEVICE_VENDOR_SPEC(0x2040, 0x7240), + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Hauppauge", + .product_name = "HVR-850", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_AUDIO_ALIGN_TRANSFER, + } +}, +{ + USB_DEVICE_VENDOR_SPEC(0x2040, 0x7210), .match_flags = USB_DEVICE_ID_MATCH_DEVICE | USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, @@ -2166,7 +2339,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - USB_DEVICE_VENDOR_SPEC(0x2040, 0x7202), + USB_DEVICE_VENDOR_SPEC(0x2040, 0x7217), .match_flags = USB_DEVICE_ID_MATCH_DEVICE | USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, @@ -2180,7 +2353,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - USB_DEVICE_VENDOR_SPEC(0x2040, 0x7203), + USB_DEVICE_VENDOR_SPEC(0x2040, 0x721b), .match_flags = USB_DEVICE_ID_MATCH_DEVICE | USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, @@ -2194,7 +2367,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - USB_DEVICE_VENDOR_SPEC(0x2040, 0x7204), + USB_DEVICE_VENDOR_SPEC(0x2040, 0x721e), .match_flags = USB_DEVICE_ID_MATCH_DEVICE | USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, @@ -2208,7 +2381,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - USB_DEVICE_VENDOR_SPEC(0x2040, 0x7205), + USB_DEVICE_VENDOR_SPEC(0x2040, 0x721f), .match_flags = USB_DEVICE_ID_MATCH_DEVICE | USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, @@ -2222,7 +2395,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - USB_DEVICE_VENDOR_SPEC(0x2040, 0x7250), + USB_DEVICE_VENDOR_SPEC(0x2040, 0x7280), .match_flags = USB_DEVICE_ID_MATCH_DEVICE | USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, @@ -2236,7 +2409,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - USB_DEVICE_VENDOR_SPEC(0x2040, 0x7230), + USB_DEVICE_VENDOR_SPEC(0x0fd9, 0x0008), .match_flags = USB_DEVICE_ID_MATCH_DEVICE | USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, @@ -2244,7 +2417,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .vendor_name = "Hauppauge", - .product_name = "HVR-850", + .product_name = "HVR-950Q", .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_AUDIO_ALIGN_TRANSFER, } diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index b45e54c09ba2..9a9da09586a5 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -32,6 +32,7 @@ #include "helper.h" #include "endpoint.h" #include "pcm.h" +#include "clock.h" /* * handle the quirks for the contained interfaces diff --git a/sound/usb/urb.c b/sound/usb/urb.c index de607d4411ac..8deeaad10f10 100644 --- a/sound/usb/urb.c +++ b/sound/usb/urb.c @@ -244,7 +244,7 @@ int snd_usb_init_substream_urbs(struct snd_usb_substream *subs, else subs->curpacksize = maxsize; - if (snd_usb_get_speed(subs->dev) == USB_SPEED_HIGH) + if (snd_usb_get_speed(subs->dev) != USB_SPEED_FULL) packs_per_ms = 8 >> subs->datainterval; else packs_per_ms = 1; diff --git a/sound/usb/usx2y/usx2yhwdeppcm.c b/sound/usb/usx2y/usx2yhwdeppcm.c index 2a528e56afd5..287ef73b1237 100644 --- a/sound/usb/usx2y/usx2yhwdeppcm.c +++ b/sound/usb/usx2y/usx2yhwdeppcm.c @@ -36,9 +36,9 @@ plain usx2y alsa mode is able to achieve 64frames, 4periods, but only at the cost of easier triggered i.e. aeolus xruns (128 or 256frames, 2periods works but is useless cause of crackling). - + This is a first "proof of concept" implementation. - Later, funcionalities should migrate to more apropriate places: + Later, functionalities should migrate to more apropriate places: Userland: - The jackd could mmap its float-pcm buffers directly from alsa-lib. - alsa-lib could provide power of 2 period sized shaping combined with int/float @@ -54,7 +54,7 @@ #include <linux/gfp.h> #include "usbusx2yaudio.c" -#if defined(USX2Y_NRPACKS_VARIABLE) || (!defined(USX2Y_NRPACKS_VARIABLE) && USX2Y_NRPACKS == 1) +#if defined(USX2Y_NRPACKS_VARIABLE) || USX2Y_NRPACKS == 1 #include <sound/hwdep.h> |