diff options
Diffstat (limited to 'sound')
34 files changed, 290 insertions, 88 deletions
diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 4490a699030b..555df64d46ff 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -529,7 +529,8 @@ static int snd_compress_check_input(struct snd_compr_params *params) { /* first let's check the buffer parameter's */ if (params->buffer.fragment_size == 0 || - params->buffer.fragments > INT_MAX / params->buffer.fragment_size) + params->buffer.fragments > INT_MAX / params->buffer.fragment_size || + params->buffer.fragments == 0) return -EINVAL; /* now codec parameters */ diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 6bda8f6c5f84..cdff5f976480 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -25,6 +25,7 @@ #include <linux/time.h> #include <linux/mutex.h> #include <linux/device.h> +#include <linux/nospec.h> #include <sound/core.h> #include <sound/minors.h> #include <sound/pcm.h> @@ -125,6 +126,7 @@ static int snd_pcm_control_ioctl(struct snd_card *card, return -EFAULT; if (stream < 0 || stream > 1) return -EINVAL; + stream = array_index_nospec(stream, 2); if (get_user(subdevice, &info->subdevice)) return -EFAULT; mutex_lock(®ister_mutex); diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 79018697b477..3586ab41dec4 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -35,6 +35,7 @@ #include <sound/timer.h> #include <sound/minors.h> #include <linux/uio.h> +#include <linux/delay.h> /* * Compatibility @@ -78,12 +79,12 @@ static DECLARE_RWSEM(snd_pcm_link_rwsem); * and this may lead to a deadlock when the code path takes read sem * twice (e.g. one in snd_pcm_action_nonatomic() and another in * snd_pcm_stream_lock()). As a (suboptimal) workaround, let writer to - * spin until it gets the lock. + * sleep until all the readers are completed without blocking by writer. */ -static inline void down_write_nonblock(struct rw_semaphore *lock) +static inline void down_write_nonfifo(struct rw_semaphore *lock) { while (!down_write_trylock(lock)) - cond_resched(); + msleep(1); } /** @@ -1825,7 +1826,7 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd) res = -ENOMEM; goto _nolock; } - down_write_nonblock(&snd_pcm_link_rwsem); + down_write_nonfifo(&snd_pcm_link_rwsem); write_lock_irq(&snd_pcm_link_rwlock); if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN || substream->runtime->status->state != substream1->runtime->status->state || @@ -1872,7 +1873,7 @@ static int snd_pcm_unlink(struct snd_pcm_substream *substream) struct snd_pcm_substream *s; int res = 0; - down_write_nonblock(&snd_pcm_link_rwsem); + down_write_nonfifo(&snd_pcm_link_rwsem); write_lock_irq(&snd_pcm_link_rwlock); if (!snd_pcm_stream_linked(substream)) { res = -EALREADY; @@ -2224,7 +2225,8 @@ int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream) static void pcm_release_private(struct snd_pcm_substream *substream) { - snd_pcm_unlink(substream); + if (snd_pcm_stream_linked(substream)) + snd_pcm_unlink(substream); } void snd_pcm_release_substream(struct snd_pcm_substream *substream) diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index ab894ed1ff67..8557e54d2659 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -40,6 +40,7 @@ config SND_OXFW * Mackie(Loud) U.420/U.420d * TASCAM FireOne * Stanton Controllers & Systems 1 Deck/Mixer + * APOGEE duet FireWire To compile this driver as a module, choose M here: the module will be called snd-oxfw. diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c index d0dfa822266b..a205b93fd9ac 100644 --- a/sound/firewire/bebob/bebob.c +++ b/sound/firewire/bebob/bebob.c @@ -434,7 +434,7 @@ static const struct ieee1394_device_id bebob_id_table[] = { /* Apogee Electronics, DA/AD/DD-16X (X-FireWire card) */ SND_BEBOB_DEV_ENTRY(VEN_APOGEE, 0x00010048, &spec_normal), /* Apogee Electronics, Ensemble */ - SND_BEBOB_DEV_ENTRY(VEN_APOGEE, 0x00001eee, &spec_normal), + SND_BEBOB_DEV_ENTRY(VEN_APOGEE, 0x01eeee, &spec_normal), /* ESI, Quatafire610 */ SND_BEBOB_DEV_ENTRY(VEN_ESI, 0x00010064, &spec_normal), /* AcousticReality, eARMasterOne */ @@ -474,7 +474,19 @@ static const struct ieee1394_device_id bebob_id_table[] = { /* Focusrite, SaffirePro 26 I/O */ SND_BEBOB_DEV_ENTRY(VEN_FOCUSRITE, 0x00000003, &saffirepro_26_spec), /* Focusrite, SaffirePro 10 I/O */ - SND_BEBOB_DEV_ENTRY(VEN_FOCUSRITE, 0x00000006, &saffirepro_10_spec), + { + // The combination of vendor_id and model_id is the same as the + // same as the one of Liquid Saffire 56. + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID | + IEEE1394_MATCH_SPECIFIER_ID | + IEEE1394_MATCH_VERSION, + .vendor_id = VEN_FOCUSRITE, + .model_id = 0x000006, + .specifier_id = 0x00a02d, + .version = 0x010001, + .driver_data = (kernel_ulong_t)&saffirepro_10_spec, + }, /* Focusrite, Saffire(no label and LE) */ SND_BEBOB_DEV_ENTRY(VEN_FOCUSRITE, MODEL_FOCUSRITE_SAFFIRE_BOTH, &saffire_spec), diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index 696b6cf35003..b0395c4209ab 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -20,6 +20,7 @@ #define VENDOR_LACIE 0x00d04b #define VENDOR_TASCAM 0x00022e #define OUI_STANTON 0x001260 +#define OUI_APOGEE 0x0003db #define MODEL_SATELLITE 0x00200f @@ -441,6 +442,13 @@ static const struct ieee1394_device_id oxfw_id_table[] = { .vendor_id = OUI_STANTON, .model_id = 0x002000, }, + // APOGEE, duet FireWire + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_APOGEE, + .model_id = 0x01dddd, + }, { } }; MODULE_DEVICE_TABLE(ieee1394, oxfw_id_table); diff --git a/sound/pci/cs46xx/dsp_spos.c b/sound/pci/cs46xx/dsp_spos.c index 4a0cbd2241d8..3191666ac129 100644 --- a/sound/pci/cs46xx/dsp_spos.c +++ b/sound/pci/cs46xx/dsp_spos.c @@ -899,6 +899,9 @@ int cs46xx_dsp_proc_done (struct snd_cs46xx *chip) struct dsp_spos_instance * ins = chip->dsp_spos_instance; int i; + if (!ins) + return 0; + snd_info_free_entry(ins->proc_sym_info_entry); ins->proc_sym_info_entry = NULL; diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index 50b216fc369f..5d422d65e62b 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -36,6 +36,7 @@ #include <linux/init.h> #include <linux/mutex.h> #include <linux/moduleparam.h> +#include <linux/nospec.h> #include <sound/core.h> #include <sound/tlv.h> @@ -1000,6 +1001,8 @@ static int snd_emu10k1_ipcm_poke(struct snd_emu10k1 *emu, if (ipcm->substream >= EMU10K1_FX8010_PCM_COUNT) return -EINVAL; + ipcm->substream = array_index_nospec(ipcm->substream, + EMU10K1_FX8010_PCM_COUNT); if (ipcm->channels > 32) return -EINVAL; pcm = &emu->fx8010.pcm[ipcm->substream]; @@ -1046,6 +1049,8 @@ static int snd_emu10k1_ipcm_peek(struct snd_emu10k1 *emu, if (ipcm->substream >= EMU10K1_FX8010_PCM_COUNT) return -EINVAL; + ipcm->substream = array_index_nospec(ipcm->substream, + EMU10K1_FX8010_PCM_COUNT); pcm = &emu->fx8010.pcm[ipcm->substream]; mutex_lock(&emu->fx8010.lock); spin_lock_irq(&emu->reg_lock); diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index 6efadbfb3fe3..7ea201c05e5d 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -109,7 +109,8 @@ static int hda_codec_driver_probe(struct device *dev) err = snd_hda_codec_build_controls(codec); if (err < 0) goto error_module; - if (codec->card->registered) { + /* only register after the bus probe finished; otherwise it's racy */ + if (!codec->bus->bus_probing && codec->card->registered) { err = snd_card_register(codec->card); if (err < 0) goto error_module; diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index c6b046ddefdd..1b5e217d1bb2 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3004,6 +3004,7 @@ static void hda_call_codec_resume(struct hda_codec *codec) hda_jackpoll_work(&codec->jackpoll_work.work); else snd_hda_jack_report_sync(codec); + codec->core.dev.power.power_state = PMSG_ON; atomic_dec(&codec->core.in_pm); } @@ -3036,10 +3037,62 @@ static int hda_codec_runtime_resume(struct device *dev) } #endif /* CONFIG_PM */ +#ifdef CONFIG_PM_SLEEP +static int hda_codec_force_resume(struct device *dev) +{ + int ret; + + /* The get/put pair below enforces the runtime resume even if the + * device hasn't been used at suspend time. This trick is needed to + * update the jack state change during the sleep. + */ + pm_runtime_get_noresume(dev); + ret = pm_runtime_force_resume(dev); + pm_runtime_put(dev); + return ret; +} + +static int hda_codec_pm_suspend(struct device *dev) +{ + dev->power.power_state = PMSG_SUSPEND; + return pm_runtime_force_suspend(dev); +} + +static int hda_codec_pm_resume(struct device *dev) +{ + dev->power.power_state = PMSG_RESUME; + return hda_codec_force_resume(dev); +} + +static int hda_codec_pm_freeze(struct device *dev) +{ + dev->power.power_state = PMSG_FREEZE; + return pm_runtime_force_suspend(dev); +} + +static int hda_codec_pm_thaw(struct device *dev) +{ + dev->power.power_state = PMSG_THAW; + return hda_codec_force_resume(dev); +} + +static int hda_codec_pm_restore(struct device *dev) +{ + dev->power.power_state = PMSG_RESTORE; + return hda_codec_force_resume(dev); +} +#endif /* CONFIG_PM_SLEEP */ + /* referred in hda_bind.c */ const struct dev_pm_ops hda_codec_driver_pm = { - SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, - pm_runtime_force_resume) +#ifdef CONFIG_PM_SLEEP + .suspend = hda_codec_pm_suspend, + .resume = hda_codec_pm_resume, + .freeze = hda_codec_pm_freeze, + .thaw = hda_codec_pm_thaw, + .poweroff = hda_codec_pm_suspend, + .restore = hda_codec_pm_restore, +#endif /* CONFIG_PM_SLEEP */ SET_RUNTIME_PM_OPS(hda_codec_runtime_suspend, hda_codec_runtime_resume, NULL) }; diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 776dffa88aee..171e11be938d 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -68,6 +68,7 @@ struct hda_bus { unsigned int response_reset:1; /* controller was reset */ unsigned int in_reset:1; /* during reset operation */ unsigned int no_response_fallback:1; /* don't fallback at RIRB error */ + unsigned int bus_probing :1; /* during probing process */ int primary_dig_out_type; /* primary digital out PCM type */ unsigned int mixer_assigned; /* codec addr for mixer name */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 3557e3943ad5..789eca17fc60 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2089,6 +2089,7 @@ static int azx_probe_continue(struct azx *chip) int val; int err; + to_hda_bus(bus)->bus_probing = 1; hda->probe_continued = 1; /* Request display power well for the HDA controller or codec. For @@ -2189,6 +2190,7 @@ i915_power_fail: if (err < 0) hda->init_failed = 1; complete_all(&hda->probe_wait); + to_hda_bus(bus)->bus_probing = 0; return err; } @@ -2352,6 +2354,10 @@ static const struct pci_device_id azx_ids[] = { /* AMD Hudson */ { PCI_DEVICE(0x1022, 0x780d), .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB }, + /* AMD Stoney */ + { PCI_DEVICE(0x1022, 0x157a), + .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB | + AZX_DCAPS_PM_RUNTIME }, /* AMD Raven */ { PCI_DEVICE(0x1022, 0x15e3), .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB | diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 0621920f7617..e85fb04ec7be 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -249,10 +249,12 @@ static int hda_tegra_suspend(struct device *dev) struct snd_card *card = dev_get_drvdata(dev); struct azx *chip = card->private_data; struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip); + struct hdac_bus *bus = azx_bus(chip); snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); azx_stop_chip(chip); + synchronize_irq(bus->irq); azx_enter_link_reset(chip); hda_tegra_disable_clocks(hda); diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index d392e867e9ab..447b3a8a83c3 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -853,6 +853,8 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x103c, 0x8079, "HP EliteBook 840 G3", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x807C, "HP EliteBook 820 G3", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x80FD, "HP ProBook 640 G2", CXT_FIXUP_HP_DOCK), + SND_PCI_QUIRK(0x103c, 0x828c, "HP EliteBook 840 G4", CXT_FIXUP_HP_DOCK), + SND_PCI_QUIRK(0x103c, 0x83b2, "HP EliteBook 840 G5", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x83b3, "HP EliteBook 830 G5", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x83d3, "HP ProBook 640 G4", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x8174, "HP Spectre x360", CXT_FIXUP_HP_SPECTRE), diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6c2668b4e3bc..0fc05ebdf81a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4489,9 +4489,18 @@ static void alc_fixup_tpt470_dock(struct hda_codec *codec, { 0x19, 0x21a11010 }, /* dock mic */ { } }; + /* Assure the speaker pin to be coupled with DAC NID 0x03; otherwise + * the speaker output becomes too low by some reason on Thinkpads with + * ALC298 codec + */ + static hda_nid_t preferred_pairs[] = { + 0x14, 0x03, 0x17, 0x02, 0x21, 0x02, + 0 + }; struct alc_spec *spec = codec->spec; if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->gen.preferred_dacs = preferred_pairs; spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP; snd_hda_apply_pincfgs(codec, pincfgs); } else if (action == HDA_FIXUP_ACT_INIT) { @@ -4832,6 +4841,13 @@ static void alc280_fixup_hp_9480m(struct hda_codec *codec, } } +static void alc_fixup_disable_mic_vref(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) + snd_hda_codec_set_pin_target(codec, 0x19, PIN_VREFHIZ); +} + /* for hda_fixup_thinkpad_acpi() */ #include "thinkpad_helper.c" @@ -4938,6 +4954,7 @@ enum { ALC293_FIXUP_LENOVO_SPK_NOISE, ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY, ALC255_FIXUP_DELL_SPK_NOISE, + ALC225_FIXUP_DISABLE_MIC_VREF, ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, ALC295_FIXUP_DISABLE_DAC3, ALC280_FIXUP_HP_HEADSET_MIC, @@ -5596,6 +5613,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE }, + [ALC225_FIXUP_DISABLE_MIC_VREF] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_disable_mic_vref, + .chained = true, + .chain_id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE + }, [ALC225_FIXUP_DELL1_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -5605,7 +5628,7 @@ static const struct hda_fixup alc269_fixups[] = { {} }, .chained = true, - .chain_id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE + .chain_id = ALC225_FIXUP_DISABLE_MIC_VREF }, [ALC280_FIXUP_HP_HEADSET_MIC] = { .type = HDA_FIXUP_FUNC, diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index b94fc6357139..b044dea3c815 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -30,6 +30,7 @@ #include <linux/math64.h> #include <linux/vmalloc.h> #include <linux/io.h> +#include <linux/nospec.h> #include <sound/core.h> #include <sound/control.h> @@ -4065,15 +4066,16 @@ static int snd_hdsp_channel_info(struct snd_pcm_substream *substream, struct snd_pcm_channel_info *info) { struct hdsp *hdsp = snd_pcm_substream_chip(substream); - int mapped_channel; + unsigned int channel = info->channel; - if (snd_BUG_ON(info->channel >= hdsp->max_channels)) + if (snd_BUG_ON(channel >= hdsp->max_channels)) return -EINVAL; + channel = array_index_nospec(channel, hdsp->max_channels); - if ((mapped_channel = hdsp->channel_map[info->channel]) < 0) + if (hdsp->channel_map[channel] < 0) return -EINVAL; - info->offset = mapped_channel * HDSP_CHANNEL_BUFFER_BYTES; + info->offset = hdsp->channel_map[channel] * HDSP_CHANNEL_BUFFER_BYTES; info->first = 0; info->step = 32; return 0; diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c index 09103aab0cb2..7d410e39d1a0 100644 --- a/sound/soc/codecs/rt5514-spi.c +++ b/sound/soc/codecs/rt5514-spi.c @@ -253,6 +253,8 @@ static int rt5514_spi_pcm_probe(struct snd_soc_platform *platform) rt5514_dsp = devm_kzalloc(platform->dev, sizeof(*rt5514_dsp), GFP_KERNEL); + if (!rt5514_dsp) + return -ENOMEM; rt5514_dsp->dev = &rt5514_spi->dev; mutex_init(&rt5514_dsp->dma_lock); diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 0ba0bf13c3a9..887af25f7367 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -281,7 +281,7 @@ config SND_SOC_PHYCORE_AC97 config SND_SOC_EUKREA_TLV320 tristate "Eukrea TLV320" - depends on ARCH_MXC && I2C + depends on ARCH_MXC && !ARM64 && I2C select SND_SOC_TLV320AIC23_I2C select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index 17766f8b09a8..f01f7be66b37 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -88,49 +88,49 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf, if (!buf) return -ENOMEM; - ret = snprintf(buf, PAGE_SIZE, "PDCR: %08x\nPTCR: %08x\n", + ret = scnprintf(buf, PAGE_SIZE, "PDCR: %08x\nPTCR: %08x\n", pdcr, ptcr); if (ptcr & IMX_AUDMUX_V2_PTCR_TFSDIR) - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "TxFS output from %s, ", audmux_port_string((ptcr >> 27) & 0x7)); else - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "TxFS input, "); if (ptcr & IMX_AUDMUX_V2_PTCR_TCLKDIR) - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "TxClk output from %s", audmux_port_string((ptcr >> 22) & 0x7)); else - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "TxClk input"); - ret += snprintf(buf + ret, PAGE_SIZE - ret, "\n"); + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "\n"); if (ptcr & IMX_AUDMUX_V2_PTCR_SYN) { - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "Port is symmetric"); } else { if (ptcr & IMX_AUDMUX_V2_PTCR_RFSDIR) - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "RxFS output from %s, ", audmux_port_string((ptcr >> 17) & 0x7)); else - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "RxFS input, "); if (ptcr & IMX_AUDMUX_V2_PTCR_RCLKDIR) - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "RxClk output from %s", audmux_port_string((ptcr >> 12) & 0x7)); else - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "RxClk input"); } - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "\nData received from %s\n", audmux_port_string((pdcr >> 13) & 0x7)); diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index f5a8050351b5..e83e314a76a5 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -399,7 +399,13 @@ static int sst_media_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); + int ret; + + ret = + snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(params)); + if (ret) + return ret; memset(substream->runtime->dma_area, 0, params_buffer_bytes(params)); return 0; } diff --git a/sound/soc/intel/atom/sst/sst_loader.c b/sound/soc/intel/atom/sst/sst_loader.c index 33917146d9c4..054b1d514e8a 100644 --- a/sound/soc/intel/atom/sst/sst_loader.c +++ b/sound/soc/intel/atom/sst/sst_loader.c @@ -354,14 +354,14 @@ static int sst_request_fw(struct intel_sst_drv *sst) const struct firmware *fw; retval = request_firmware(&fw, sst->firmware_name, sst->dev); - if (fw == NULL) { - dev_err(sst->dev, "fw is returning as null\n"); - return -EINVAL; - } if (retval) { dev_err(sst->dev, "request fw failed %d\n", retval); return retval; } + if (fw == NULL) { + dev_err(sst->dev, "fw is returning as null\n"); + return -EINVAL; + } mutex_lock(&sst->sst_lock); retval = sst_cache_and_parse_fw(sst, fw); mutex_unlock(&sst->sst_lock); diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c index 7486a0022fde..993d2c105ae1 100644 --- a/sound/soc/intel/boards/broadwell.c +++ b/sound/soc/intel/boards/broadwell.c @@ -191,7 +191,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { .stream_name = "Loopback", .cpu_dai_name = "Loopback Pin", .platform_name = "haswell-pcm-audio", - .dynamic = 0, + .dynamic = 1, .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c index 863f1d5e2a2c..11d0cc2b0e39 100644 --- a/sound/soc/intel/boards/haswell.c +++ b/sound/soc/intel/boards/haswell.c @@ -145,7 +145,7 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = { .stream_name = "Loopback", .cpu_dai_name = "Loopback Pin", .platform_name = "haswell-pcm-audio", - .dynamic = 0, + .dynamic = 1, .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index 89fe95e877db..07af30017b48 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -36,6 +36,8 @@ #include "../codecs/twl6040.h" struct abe_twl6040 { + struct snd_soc_card card; + struct snd_soc_dai_link dai_links[2]; int jack_detection; /* board can detect jack events */ int mclk_freq; /* MCLK frequency speed for twl6040 */ }; @@ -208,40 +210,10 @@ static int omap_abe_dmic_init(struct snd_soc_pcm_runtime *rtd) ARRAY_SIZE(dmic_audio_map)); } -/* Digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link abe_twl6040_dai_links[] = { - { - .name = "TWL6040", - .stream_name = "TWL6040", - .codec_dai_name = "twl6040-legacy", - .codec_name = "twl6040-codec", - .init = omap_abe_twl6040_init, - .ops = &omap_abe_ops, - }, - { - .name = "DMIC", - .stream_name = "DMIC Capture", - .codec_dai_name = "dmic-hifi", - .codec_name = "dmic-codec", - .init = omap_abe_dmic_init, - .ops = &omap_abe_dmic_ops, - }, -}; - -/* Audio machine driver */ -static struct snd_soc_card omap_abe_card = { - .owner = THIS_MODULE, - - .dapm_widgets = twl6040_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(twl6040_dapm_widgets), - .dapm_routes = audio_map, - .num_dapm_routes = ARRAY_SIZE(audio_map), -}; - static int omap_abe_probe(struct platform_device *pdev) { struct device_node *node = pdev->dev.of_node; - struct snd_soc_card *card = &omap_abe_card; + struct snd_soc_card *card; struct device_node *dai_node; struct abe_twl6040 *priv; int num_links = 0; @@ -252,12 +224,18 @@ static int omap_abe_probe(struct platform_device *pdev) return -ENODEV; } - card->dev = &pdev->dev; - priv = devm_kzalloc(&pdev->dev, sizeof(struct abe_twl6040), GFP_KERNEL); if (priv == NULL) return -ENOMEM; + card = &priv->card; + card->dev = &pdev->dev; + card->owner = THIS_MODULE; + card->dapm_widgets = twl6040_dapm_widgets; + card->num_dapm_widgets = ARRAY_SIZE(twl6040_dapm_widgets); + card->dapm_routes = audio_map; + card->num_dapm_routes = ARRAY_SIZE(audio_map); + if (snd_soc_of_parse_card_name(card, "ti,model")) { dev_err(&pdev->dev, "Card name is not provided\n"); return -ENODEV; @@ -274,14 +252,27 @@ static int omap_abe_probe(struct platform_device *pdev) dev_err(&pdev->dev, "McPDM node is not provided\n"); return -EINVAL; } - abe_twl6040_dai_links[0].cpu_of_node = dai_node; - abe_twl6040_dai_links[0].platform_of_node = dai_node; + + priv->dai_links[0].name = "DMIC"; + priv->dai_links[0].stream_name = "TWL6040"; + priv->dai_links[0].cpu_of_node = dai_node; + priv->dai_links[0].platform_of_node = dai_node; + priv->dai_links[0].codec_dai_name = "twl6040-legacy"; + priv->dai_links[0].codec_name = "twl6040-codec"; + priv->dai_links[0].init = omap_abe_twl6040_init; + priv->dai_links[0].ops = &omap_abe_ops; dai_node = of_parse_phandle(node, "ti,dmic", 0); if (dai_node) { num_links = 2; - abe_twl6040_dai_links[1].cpu_of_node = dai_node; - abe_twl6040_dai_links[1].platform_of_node = dai_node; + priv->dai_links[1].name = "TWL6040"; + priv->dai_links[1].stream_name = "DMIC Capture"; + priv->dai_links[1].cpu_of_node = dai_node; + priv->dai_links[1].platform_of_node = dai_node; + priv->dai_links[1].codec_dai_name = "dmic-hifi"; + priv->dai_links[1].codec_name = "dmic-codec"; + priv->dai_links[1].init = omap_abe_dmic_init; + priv->dai_links[1].ops = &omap_abe_dmic_ops; } else { num_links = 1; } @@ -300,7 +291,7 @@ static int omap_abe_probe(struct platform_device *pdev) return -ENODEV; } - card->dai_link = abe_twl6040_dai_links; + card->dai_link = priv->dai_links; card->num_links = num_links; snd_soc_card_set_drvdata(card, priv); diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c index 09db2aec12a3..776e809a8aab 100644 --- a/sound/soc/omap/omap-dmic.c +++ b/sound/soc/omap/omap-dmic.c @@ -48,6 +48,8 @@ struct omap_dmic { struct device *dev; void __iomem *io_base; struct clk *fclk; + struct pm_qos_request pm_qos_req; + int latency; int fclk_freq; int out_freq; int clk_div; @@ -124,6 +126,8 @@ static void omap_dmic_dai_shutdown(struct snd_pcm_substream *substream, mutex_lock(&dmic->mutex); + pm_qos_remove_request(&dmic->pm_qos_req); + if (!dai->active) dmic->active = 0; @@ -226,6 +230,8 @@ static int omap_dmic_dai_hw_params(struct snd_pcm_substream *substream, /* packet size is threshold * channels */ dma_data = snd_soc_dai_get_dma_data(dai, substream); dma_data->maxburst = dmic->threshold * channels; + dmic->latency = (OMAP_DMIC_THRES_MAX - dmic->threshold) * USEC_PER_SEC / + params_rate(params); return 0; } @@ -236,6 +242,9 @@ static int omap_dmic_dai_prepare(struct snd_pcm_substream *substream, struct omap_dmic *dmic = snd_soc_dai_get_drvdata(dai); u32 ctrl; + if (pm_qos_request_active(&dmic->pm_qos_req)) + pm_qos_update_request(&dmic->pm_qos_req, dmic->latency); + /* Configure uplink threshold */ omap_dmic_write(dmic, OMAP_DMIC_FIFO_CTRL_REG, dmic->threshold); diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index 64609c77a79d..44ffeb71cd1d 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -54,6 +54,8 @@ struct omap_mcpdm { unsigned long phys_base; void __iomem *io_base; int irq; + struct pm_qos_request pm_qos_req; + int latency[2]; struct mutex mutex; @@ -277,6 +279,9 @@ static void omap_mcpdm_dai_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); + int tx = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + int stream1 = tx ? SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE; + int stream2 = tx ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; mutex_lock(&mcpdm->mutex); @@ -289,6 +294,14 @@ static void omap_mcpdm_dai_shutdown(struct snd_pcm_substream *substream, } } + if (mcpdm->latency[stream2]) + pm_qos_update_request(&mcpdm->pm_qos_req, + mcpdm->latency[stream2]); + else if (mcpdm->latency[stream1]) + pm_qos_remove_request(&mcpdm->pm_qos_req); + + mcpdm->latency[stream1] = 0; + mutex_unlock(&mcpdm->mutex); } @@ -300,7 +313,7 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream, int stream = substream->stream; struct snd_dmaengine_dai_dma_data *dma_data; u32 threshold; - int channels; + int channels, latency; int link_mask = 0; channels = params_channels(params); @@ -340,14 +353,25 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream, dma_data->maxburst = (MCPDM_DN_THRES_MAX - threshold) * channels; + latency = threshold; } else { /* If playback is not running assume a stereo stream to come */ if (!mcpdm->config[!stream].link_mask) mcpdm->config[!stream].link_mask = (0x3 << 3); dma_data->maxburst = threshold * channels; + latency = (MCPDM_DN_THRES_MAX - threshold); } + /* + * The DMA must act to a DMA request within latency time (usec) to avoid + * under/overflow + */ + mcpdm->latency[stream] = latency * USEC_PER_SEC / params_rate(params); + + if (!mcpdm->latency[stream]) + mcpdm->latency[stream] = 10; + /* Check if we need to restart McPDM with this stream */ if (mcpdm->config[stream].link_mask && mcpdm->config[stream].link_mask != link_mask) @@ -362,6 +386,20 @@ static int omap_mcpdm_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); + struct pm_qos_request *pm_qos_req = &mcpdm->pm_qos_req; + int tx = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + int stream1 = tx ? SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE; + int stream2 = tx ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; + int latency = mcpdm->latency[stream2]; + + /* Prevent omap hardware from hitting off between FIFO fills */ + if (!latency || mcpdm->latency[stream1] < latency) + latency = mcpdm->latency[stream1]; + + if (pm_qos_request_active(pm_qos_req)) + pm_qos_update_request(pm_qos_req, latency); + else if (latency) + pm_qos_add_request(pm_qos_req, PM_QOS_CPU_DMA_LATENCY, latency); if (!omap_mcpdm_active(mcpdm)) { omap_mcpdm_start(mcpdm); @@ -423,6 +461,9 @@ static int omap_mcpdm_remove(struct snd_soc_dai *dai) free_irq(mcpdm->irq, (void *)mcpdm); pm_runtime_disable(mcpdm->dev); + if (pm_qos_request_active(&mcpdm->pm_qos_req)) + pm_qos_remove_request(&mcpdm->pm_qos_req); + return 0; } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4e3de566809c..168559b5e9f3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2018,6 +2018,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) } card->instantiated = 1; + dapm_mark_endpoints_dirty(card); snd_soc_dapm_sync(&card->dapm); mutex_unlock(&card->mutex); mutex_unlock(&client_mutex); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 8bfc534e3b34..ab647f1fe11b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1976,19 +1976,19 @@ static ssize_t dapm_widget_power_read_file(struct file *file, out = is_connected_output_ep(w, NULL, NULL); } - ret = snprintf(buf, PAGE_SIZE, "%s: %s%s in %d out %d", + ret = scnprintf(buf, PAGE_SIZE, "%s: %s%s in %d out %d", w->name, w->power ? "On" : "Off", w->force ? " (forced)" : "", in, out); if (w->reg >= 0) - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, " - R%d(0x%x) mask 0x%x", w->reg, w->reg, w->mask << w->shift); - ret += snprintf(buf + ret, PAGE_SIZE - ret, "\n"); + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "\n"); if (w->sname) - ret += snprintf(buf + ret, PAGE_SIZE - ret, " stream %s %s\n", + ret += scnprintf(buf + ret, PAGE_SIZE - ret, " stream %s %s\n", w->sname, w->active ? "active" : "inactive"); @@ -2001,7 +2001,7 @@ static ssize_t dapm_widget_power_read_file(struct file *file, if (!p->connect) continue; - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, " %s \"%s\" \"%s\"\n", (rdir == SND_SOC_DAPM_DIR_IN) ? "in" : "out", p->name ? p->name : "static", diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index d6b48c796bfc..086fe4d27f60 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1989,6 +1989,7 @@ int snd_soc_tplg_component_load(struct snd_soc_component *comp, struct snd_soc_tplg_ops *ops, const struct firmware *fw, u32 id) { struct soc_tplg tplg; + int ret; /* setup parsing context */ memset(&tplg, 0, sizeof(tplg)); @@ -2002,7 +2003,12 @@ int snd_soc_tplg_component_load(struct snd_soc_component *comp, tplg.bytes_ext_ops = ops->bytes_ext_ops; tplg.bytes_ext_ops_count = ops->bytes_ext_ops_count; - return soc_tplg_load(&tplg); + ret = soc_tplg_load(&tplg); + /* free the created components if fail to load topology */ + if (ret) + snd_soc_tplg_component_remove(comp, SND_SOC_TPLG_INDEX_ALL); + + return ret; } EXPORT_SYMBOL_GPL(snd_soc_tplg_component_load); diff --git a/sound/synth/emux/emux_hwdep.c b/sound/synth/emux/emux_hwdep.c index e557946718a9..d9fcae071b47 100644 --- a/sound/synth/emux/emux_hwdep.c +++ b/sound/synth/emux/emux_hwdep.c @@ -22,9 +22,9 @@ #include <sound/core.h> #include <sound/hwdep.h> #include <linux/uaccess.h> +#include <linux/nospec.h> #include "emux_voice.h" - #define TMP_CLIENT_ID 0x1001 /* @@ -66,13 +66,16 @@ snd_emux_hwdep_misc_mode(struct snd_emux *emu, void __user *arg) return -EFAULT; if (info.mode < 0 || info.mode >= EMUX_MD_END) return -EINVAL; + info.mode = array_index_nospec(info.mode, EMUX_MD_END); if (info.port < 0) { for (i = 0; i < emu->num_ports; i++) emu->portptrs[i]->ctrls[info.mode] = info.value; } else { - if (info.port < emu->num_ports) + if (info.port < emu->num_ports) { + info.port = array_index_nospec(info.port, emu->num_ports); emu->portptrs[info.port]->ctrls[info.mode] = info.value; + } } return 0; } diff --git a/sound/usb/card.c b/sound/usb/card.c index 8906199a83e6..549b9b061694 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -644,9 +644,12 @@ static int usb_audio_probe(struct usb_interface *intf, __error: if (chip) { + /* chip->active is inside the chip->card object, + * decrement before memory is possibly returned. + */ + atomic_dec(&chip->active); if (!chip->num_interfaces) snd_card_free(chip->card); - atomic_dec(&chip->active); } mutex_unlock(®ister_mutex); return err; diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index db8404e31fae..64b90b8ec661 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1882,7 +1882,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, char *name) { struct uac_processing_unit_descriptor *desc = raw_desc; - int num_ins = desc->bNrInPins; + int num_ins; struct usb_mixer_elem_info *cval; struct snd_kcontrol *kctl; int i, err, nameid, type, len; @@ -1897,7 +1897,13 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, 0, NULL, default_value_info }; - if (desc->bLength < 13 || desc->bLength < 13 + num_ins || + if (desc->bLength < 13) { + usb_audio_err(state->chip, "invalid %s descriptor (id %d)\n", name, unitid); + return -EINVAL; + } + + num_ins = desc->bNrInPins; + if (desc->bLength < 13 + num_ins || desc->bLength < num_ins + uac_processing_unit_bControlSize(desc, state->mixer->protocol)) { usb_audio_err(state->chip, "invalid %s descriptor (id %d)\n", name, unitid); return -EINVAL; diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index e6ac7b9b4648..497bad9f2789 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -313,6 +313,9 @@ static int search_roland_implicit_fb(struct usb_device *dev, int ifnum, return 0; } +/* Setup an implicit feedback endpoint from a quirk. Returns 0 if no quirk + * applies. Returns 1 if a quirk was found. + */ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, struct usb_device *dev, struct usb_interface_descriptor *altsd, @@ -391,7 +394,7 @@ add_sync_ep: subs->data_endpoint->sync_master = subs->sync_endpoint; - return 0; + return 1; } static int set_sync_endpoint(struct snd_usb_substream *subs, @@ -430,6 +433,10 @@ static int set_sync_endpoint(struct snd_usb_substream *subs, if (err < 0) return err; + /* endpoint set by quirk */ + if (err > 0) + return 0; + if (altsd->bNumEndpoints < 2) return 0; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 15cbe2565703..d32727c74a16 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3321,6 +3321,9 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } } }, + { + .ifnum = -1 + }, } } }, |