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authorMark Brown <broonie@linaro.org>2013-09-11 11:17:15 +0100
committerMark Brown <broonie@linaro.org>2013-09-11 11:17:15 +0100
commitc34c0d7684b8b79666da6b1bc37fc330cd0dd216 (patch)
treec2bc72d67862df770af45a88814759baa0744d2c
parent29dc5dd229dc3130b51df0932e59946fc09d3bd4 (diff)
parent4345adf92db760ca1a54061ce284aaa2e7d0791e (diff)
Merge remote-tracking branch 'asoc/fix/fsl' into asoc-linus
-rw-r--r--Documentation/devicetree/bindings/sound/imx-audio-spdif.txt34
-rw-r--r--Documentation/devicetree/bindings/sound/mvebu-audio.txt29
-rw-r--r--Documentation/sound/alsa/HD-Audio-Models.txt1
-rw-r--r--Documentation/sound/alsa/HD-Audio.txt2
-rw-r--r--MAINTAINERS11
-rw-r--r--arch/arm/plat-samsung/s3c-dma-ops.c13
-rw-r--r--include/sound/core.h8
-rw-r--r--include/sound/soc-dapm.h2
-rw-r--r--include/sound/soc.h8
-rw-r--r--include/uapi/sound/hdspm.h2
-rw-r--r--sound/core/pcm_lib.c4
-rw-r--r--sound/drivers/dummy.c2
-rw-r--r--sound/firewire/speakers.c4
-rw-r--r--sound/isa/gus/interwave.c3
-rw-r--r--sound/oss/dmabuf.c3
-rw-r--r--sound/pci/hda/Kconfig9
-rw-r--r--sound/pci/hda/hda_codec.c64
-rw-r--r--sound/pci/hda/hda_codec.h21
-rw-r--r--sound/pci/hda/hda_generic.c79
-rw-r--r--sound/pci/hda/hda_generic.h1
-rw-r--r--sound/pci/hda/hda_hwdep.c6
-rw-r--r--sound/pci/hda/hda_intel.c34
-rw-r--r--sound/pci/hda/hda_jack.c22
-rw-r--r--sound/pci/hda/hda_jack.h13
-rw-r--r--sound/pci/hda/hda_proc.c33
-rw-r--r--sound/pci/hda/patch_analog.c4528
-rw-r--r--sound/pci/hda/patch_conexant.c79
-rw-r--r--sound/pci/hda/patch_hdmi.c9
-rw-r--r--sound/pci/hda/patch_realtek.c190
-rw-r--r--sound/pci/hda/patch_sigmatel.c14
-rw-r--r--sound/pci/hda/patch_via.c2
-rw-r--r--sound/pci/rme96.c307
-rw-r--r--sound/pci/rme9652/hdspm.c779
-rw-r--r--sound/soc/cirrus/ep93xx-i2s.c2
-rw-r--r--sound/soc/codecs/dmic.c17
-rw-r--r--sound/soc/codecs/rt5640.c217
-rw-r--r--sound/soc/codecs/rt5640.h12
-rw-r--r--sound/soc/codecs/ssm2602.c3
-rw-r--r--sound/soc/codecs/tlv320aic32x4.c22
-rw-r--r--sound/soc/codecs/wm8904.c1
-rw-r--r--sound/soc/codecs/wm8962.c2
-rw-r--r--sound/soc/dwc/designware_i2s.c5
-rw-r--r--sound/soc/fsl/Kconfig11
-rw-r--r--sound/soc/fsl/Makefile2
-rw-r--r--sound/soc/fsl/fsl_spdif.c29
-rw-r--r--sound/soc/fsl/fsl_ssi.c1
-rw-r--r--sound/soc/fsl/imx-audmux.c3
-rw-r--r--sound/soc/fsl/imx-spdif.c148
-rw-r--r--sound/soc/generic/simple-card.c2
-rw-r--r--sound/soc/kirkwood/Kconfig4
-rw-r--r--sound/soc/kirkwood/kirkwood-i2s.c26
-rw-r--r--sound/soc/mxs/mxs-sgtl5000.c2
-rw-r--r--sound/soc/omap/mcbsp.c2
-rw-r--r--sound/soc/samsung/dma.c7
-rw-r--r--sound/soc/sh/fsi.c51
-rw-r--r--sound/soc/soc-core.c17
-rw-r--r--sound/soc/soc-dapm.c11
-rw-r--r--sound/soc/soc-jack.c2
-rw-r--r--sound/soc/soc-pcm.c10
-rw-r--r--sound/usb/6fire/firmware.c4
-rw-r--r--sound/usb/endpoint.c3
-rw-r--r--sound/usb/pcm.c243
-rw-r--r--sound/usb/usx2y/usbusx2y.c8
63 files changed, 2170 insertions, 5013 deletions
diff --git a/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt b/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt
new file mode 100644
index 000000000000..7d13479f9c3c
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt
@@ -0,0 +1,34 @@
+Freescale i.MX audio complex with S/PDIF transceiver
+
+Required properties:
+
+ - compatible : "fsl,imx-audio-spdif"
+
+ - model : The user-visible name of this sound complex
+
+ - spdif-controller : The phandle of the i.MX S/PDIF controller
+
+
+Optional properties:
+
+ - spdif-out : This is a boolean property. If present, the transmitting
+ function of S/PDIF will be enabled, indicating there's a physical
+ S/PDIF out connector/jack on the board or it's connecting to some
+ other IP block, such as an HDMI encoder/display-controller.
+
+ - spdif-in : This is a boolean property. If present, the receiving
+ function of S/PDIF will be enabled, indicating there's a physical
+ S/PDIF in connector/jack on the board.
+
+* Note: At least one of these two properties should be set in the DT binding.
+
+
+Example:
+
+sound-spdif {
+ compatible = "fsl,imx-audio-spdif";
+ model = "imx-spdif";
+ spdif-controller = <&spdif>;
+ spdif-out;
+ spdif-in;
+};
diff --git a/Documentation/devicetree/bindings/sound/mvebu-audio.txt b/Documentation/devicetree/bindings/sound/mvebu-audio.txt
new file mode 100644
index 000000000000..7e5fd37c1b3f
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/mvebu-audio.txt
@@ -0,0 +1,29 @@
+* mvebu (Kirkwood, Dove, Armada 370) audio controller
+
+Required properties:
+
+- compatible: "marvell,mvebu-audio"
+
+- reg: physical base address of the controller and length of memory mapped
+ region.
+
+- interrupts: list of two irq numbers.
+ The first irq is used for data flow and the second one is used for errors.
+
+- clocks: one or two phandles.
+ The first one is mandatory and defines the internal clock.
+ The second one is optional and defines an external clock.
+
+- clock-names: names associated to the clocks:
+ "internal" for the internal clock
+ "extclk" for the external clock
+
+Example:
+
+i2s1: audio-controller@b4000 {
+ compatible = "marvell,mvebu-audio";
+ reg = <0xb4000 0x2210>;
+ interrupts = <21>, <22>;
+ clocks = <&gate_clk 13>;
+ clock-names = "internal";
+};
diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt
index 809d72b8eff1..a46ddb85e83a 100644
--- a/Documentation/sound/alsa/HD-Audio-Models.txt
+++ b/Documentation/sound/alsa/HD-Audio-Models.txt
@@ -244,6 +244,7 @@ STAC9227/9228/9229/927x
5stack-no-fp D965 5stack without front panel
dell-3stack Dell Dimension E520
dell-bios Fixes with Dell BIOS setup
+ dell-bios-amic Fixes with Dell BIOS setup including analog mic
volknob Fixes with volume-knob widget 0x24
auto BIOS setup (default)
diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt
index c3c912d023cc..42a0a39b77e6 100644
--- a/Documentation/sound/alsa/HD-Audio.txt
+++ b/Documentation/sound/alsa/HD-Audio.txt
@@ -454,6 +454,8 @@ The generic parser supports the following hints:
- need_dac_fix (bool): limits the DACs depending on the channel count
- primary_hp (bool): probe headphone jacks as the primary outputs;
default true
+- multi_io (bool): try probing multi-I/O config (e.g. shared
+ line-in/surround, mic/clfe jacks)
- multi_cap_vol (bool): provide multiple capture volumes
- inv_dmic_split (bool): provide split internal mic volume/switch for
phase-inverted digital mics
diff --git a/MAINTAINERS b/MAINTAINERS
index b5e09128898f..a77b9440d87d 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -7676,6 +7676,17 @@ F: include/sound/
F: include/uapi/sound/
F: sound/
+SOUND - COMPRESSED AUDIO
+M: Vinod Koul <vinod.koul@intel.com>
+L: alsa-devel@alsa-project.org (moderated for non-subscribers)
+T: git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git
+S: Supported
+F: Documentation/sound/alsa/compress_offload.txt
+F: include/sound/compress_driver.h
+F: include/uapi/sound/compress_*
+F: sound/core/compress_offload.c
+F: sound/soc/soc-compress.c
+
SOUND - SOC LAYER / DYNAMIC AUDIO POWER MANAGEMENT (ASoC)
M: Liam Girdwood <lgirdwood@gmail.com>
M: Mark Brown <broonie@kernel.org>
diff --git a/arch/arm/plat-samsung/s3c-dma-ops.c b/arch/arm/plat-samsung/s3c-dma-ops.c
index 0cc40aea3f5a..98b10ba67dc7 100644
--- a/arch/arm/plat-samsung/s3c-dma-ops.c
+++ b/arch/arm/plat-samsung/s3c-dma-ops.c
@@ -82,7 +82,8 @@ static int s3c_dma_config(unsigned ch, struct samsung_dma_config *param)
static int s3c_dma_prepare(unsigned ch, struct samsung_dma_prep *param)
{
struct cb_data *data;
- int len = (param->cap == DMA_CYCLIC) ? param->period : param->len;
+ dma_addr_t pos = param->buf;
+ dma_addr_t end = param->buf + param->len;
list_for_each_entry(data, &dma_list, node)
if (data->ch == ch)
@@ -94,7 +95,15 @@ static int s3c_dma_prepare(unsigned ch, struct samsung_dma_prep *param)
data->fp_param = param->fp_param;
}
- s3c2410_dma_enqueue(ch, (void *)data, param->buf, len);
+ if (param->cap != DMA_CYCLIC) {
+ s3c2410_dma_enqueue(ch, (void *)data, param->buf, param->len);
+ return 0;
+ }
+
+ while (pos < end) {
+ s3c2410_dma_enqueue(ch, (void *)data, pos, param->period);
+ pos += param->period;
+ }
return 0;
}
diff --git a/include/sound/core.h b/include/sound/core.h
index c586617cfa0d..2a14f1f02d4f 100644
--- a/include/sound/core.h
+++ b/include/sound/core.h
@@ -27,6 +27,7 @@
#include <linux/rwsem.h> /* struct rw_semaphore */
#include <linux/pm.h> /* pm_message_t */
#include <linux/stringify.h>
+#include <linux/printk.h>
/* number of supported soundcards */
#ifdef CONFIG_SND_DYNAMIC_MINORS
@@ -376,6 +377,11 @@ void __snd_printk(unsigned int level, const char *file, int line,
#define snd_BUG() WARN(1, "BUG?\n")
/**
+ * Suppress high rates of output when CONFIG_SND_DEBUG is enabled.
+ */
+#define snd_printd_ratelimit() printk_ratelimit()
+
+/**
* snd_BUG_ON - debugging check macro
* @cond: condition to evaluate
*
@@ -398,6 +404,8 @@ static inline void _snd_printd(int level, const char *format, ...) {}
unlikely(__ret_warn_on); \
})
+static inline bool snd_printd_ratelimit(void) { return false; }
+
#endif /* CONFIG_SND_DEBUG */
#ifdef CONFIG_SND_DEBUG_VERBOSE
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index c728d28ae9a5..27a72d5d4b00 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -413,7 +413,7 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card,
struct snd_soc_dapm_widget *sink);
/* dapm path setup */
-int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm);
+int snd_soc_dapm_new_widgets(struct snd_soc_card *card);
void snd_soc_dapm_free(struct snd_soc_dapm_context *dapm);
int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm,
const struct snd_soc_dapm_route *route, int num);
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 8e2ad52078b6..d22cb0a06feb 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -697,7 +697,6 @@ struct snd_soc_codec {
unsigned int probed:1; /* Codec has been probed */
unsigned int ac97_registered:1; /* Codec has been AC97 registered */
unsigned int ac97_created:1; /* Codec has been created by SoC */
- unsigned int sysfs_registered:1; /* codec has been sysfs registered */
unsigned int cache_init:1; /* codec cache has been initialized */
unsigned int using_regmap:1; /* using regmap access */
u32 cache_only; /* Suppress writes to hardware */
@@ -705,7 +704,6 @@ struct snd_soc_codec {
/* codec IO */
void *control_data; /* codec control (i2c/3wire) data */
- enum snd_soc_control_type control_type;
hw_write_t hw_write;
unsigned int (*hw_read)(struct snd_soc_codec *, unsigned int);
unsigned int (*read)(struct snd_soc_codec *, unsigned int);
@@ -724,7 +722,6 @@ struct snd_soc_codec {
#ifdef CONFIG_DEBUG_FS
struct dentry *debugfs_codec_root;
struct dentry *debugfs_reg;
- struct dentry *debugfs_dapm;
#endif
};
@@ -849,7 +846,6 @@ struct snd_soc_platform {
#ifdef CONFIG_DEBUG_FS
struct dentry *debugfs_platform_root;
- struct dentry *debugfs_dapm;
#endif
};
@@ -934,6 +930,10 @@ struct snd_soc_dai_link {
/* machine stream operations */
const struct snd_soc_ops *ops;
const struct snd_soc_compr_ops *compr_ops;
+
+ /* For unidirectional dai links */
+ bool playback_only;
+ bool capture_only;
};
struct snd_soc_codec_conf {
diff --git a/include/uapi/sound/hdspm.h b/include/uapi/sound/hdspm.h
index 1f59ea2a4a76..d956c3593f65 100644
--- a/include/uapi/sound/hdspm.h
+++ b/include/uapi/sound/hdspm.h
@@ -111,7 +111,7 @@ struct hdspm_ltc {
enum hdspm_ltc_input_format input_format;
};
-#define SNDRV_HDSPM_IOCTL_GET_LTC _IOR('H', 0x46, struct hdspm_mixer_ioctl)
+#define SNDRV_HDSPM_IOCTL_GET_LTC _IOR('H', 0x46, struct hdspm_ltc)
/**
* The status data reflects the device's current state
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 82bb029d4414..6e03b465e44e 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -184,7 +184,7 @@ static void xrun(struct snd_pcm_substream *substream)
do { \
if (xrun_debug(substream, XRUN_DEBUG_BASIC)) { \
xrun_log_show(substream); \
- if (printk_ratelimit()) { \
+ if (snd_printd_ratelimit()) { \
snd_printd("PCM: " fmt, ##args); \
} \
dump_stack_on_xrun(substream); \
@@ -342,7 +342,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream,
return -EPIPE;
}
if (pos >= runtime->buffer_size) {
- if (printk_ratelimit()) {
+ if (snd_printd_ratelimit()) {
char name[16];
snd_pcm_debug_name(substream, name, sizeof(name));
xrun_log_show(substream);
diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c
index 11048cc744d0..915b4d7fbb23 100644
--- a/sound/drivers/dummy.c
+++ b/sound/drivers/dummy.c
@@ -1022,7 +1022,7 @@ static void dummy_proc_write(struct snd_info_entry *entry,
if (i >= ARRAY_SIZE(fields))
continue;
snd_info_get_str(item, ptr, sizeof(item));
- if (strict_strtoull(item, 0, &val))
+ if (kstrtoull(item, 0, &val))
continue;
if (fields[i].size == sizeof(int))
*get_dummy_int_ptr(dummy, fields[i].offset) = val;
diff --git a/sound/firewire/speakers.c b/sound/firewire/speakers.c
index 2c6386503940..fe9e6e2f2c5b 100644
--- a/sound/firewire/speakers.c
+++ b/sound/firewire/speakers.c
@@ -49,7 +49,6 @@ struct fwspk {
struct snd_card *card;
struct fw_unit *unit;
const struct device_info *device_info;
- struct snd_pcm_substream *pcm;
struct mutex mutex;
struct cmp_connection connection;
struct amdtp_out_stream stream;
@@ -363,8 +362,7 @@ static int fwspk_create_pcm(struct fwspk *fwspk)
return err;
pcm->private_data = fwspk;
strcpy(pcm->name, fwspk->device_info->short_name);
- fwspk->pcm = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
- fwspk->pcm->ops = &ops;
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &ops);
return 0;
}
diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c
index 9942691cc0ca..afef0d738078 100644
--- a/sound/isa/gus/interwave.c
+++ b/sound/isa/gus/interwave.c
@@ -443,8 +443,7 @@ static void snd_interwave_detect_memory(struct snd_gus_card *gus)
for (i = 0; i < 8; ++i)
iwave[i] = snd_gf1_peek(gus, bank_pos + i);
#ifdef CONFIG_SND_DEBUG_ROM
- printk(KERN_DEBUG "ROM at 0x%06x = %*phC\n", bank_pos,
- 8, iwave);
+ printk(KERN_DEBUG "ROM at 0x%06x = %8phC\n", bank_pos, iwave);
#endif
if (strncmp(iwave, "INTRWAVE", 8))
continue; /* first check */
diff --git a/sound/oss/dmabuf.c b/sound/oss/dmabuf.c
index a59c88818f48..461d94cfecbe 100644
--- a/sound/oss/dmabuf.c
+++ b/sound/oss/dmabuf.c
@@ -557,7 +557,6 @@ int DMAbuf_getrdbuffer(int dev, char **buf, int *len, int dontblock)
unsigned long flags;
int err = 0, n = 0;
struct dma_buffparms *dmap = adev->dmap_in;
- int go;
if (!(adev->open_mode & OPEN_READ))
return -EIO;
@@ -584,7 +583,7 @@ int DMAbuf_getrdbuffer(int dev, char **buf, int *len, int dontblock)
spin_unlock_irqrestore(&dmap->lock,flags);
return -EAGAIN;
}
- if ((go = adev->go))
+ if (adev->go)
timeout = dmabuf_timeout(dmap);
spin_unlock_irqrestore(&dmap->lock,flags);
diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig
index 59c5e9c03d53..8de66ccd7279 100644
--- a/sound/pci/hda/Kconfig
+++ b/sound/pci/hda/Kconfig
@@ -152,14 +152,9 @@ config SND_HDA_CODEC_HDMI
This module is automatically loaded at probing.
config SND_HDA_I915
- bool "Build Display HD-audio controller/codec power well support for i915 cards"
+ bool
+ default y
depends on DRM_I915
- help
- Say Y here to include full HDMI and DisplayPort HD-audio controller/codec
- power-well support for Intel Haswell graphics cards based on the i915 driver.
-
- Note that this option must be enabled for Intel Haswell C+ stepping machines, otherwise
- the GPU audio controller/codecs will not be initialized or damaged when exit from S3 mode.
config SND_HDA_CODEC_CIRRUS
bool "Build Cirrus Logic codec support"
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 8a005f0e5ca4..5b6c4e3c92ca 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -666,6 +666,64 @@ int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux,
}
EXPORT_SYMBOL_HDA(snd_hda_get_conn_index);
+
+/* return DEVLIST_LEN parameter of the given widget */
+static unsigned int get_num_devices(struct hda_codec *codec, hda_nid_t nid)
+{
+ unsigned int wcaps = get_wcaps(codec, nid);
+ unsigned int parm;
+
+ if (!codec->dp_mst || !(wcaps & AC_WCAP_DIGITAL) ||
+ get_wcaps_type(wcaps) != AC_WID_PIN)
+ return 0;
+
+ parm = snd_hda_param_read(codec, nid, AC_PAR_DEVLIST_LEN);
+ if (parm == -1 && codec->bus->rirb_error)
+ parm = 0;
+ return parm & AC_DEV_LIST_LEN_MASK;
+}
+
+/**
+ * snd_hda_get_devices - copy device list without cache
+ * @codec: the HDA codec
+ * @nid: NID of the pin to parse
+ * @dev_list: device list array
+ * @max_devices: max. number of devices to store
+ *
+ * Copy the device list. This info is dynamic and so not cached.
+ * Currently called only from hda_proc.c, so not exported.
+ */
+int snd_hda_get_devices(struct hda_codec *codec, hda_nid_t nid,
+ u8 *dev_list, int max_devices)
+{
+ unsigned int parm;
+ int i, dev_len, devices;
+
+ parm = get_num_devices(codec, nid);
+ if (!parm) /* not multi-stream capable */
+ return 0;
+
+ dev_len = parm + 1;
+ dev_len = dev_len < max_devices ? dev_len : max_devices;
+
+ devices = 0;
+ while (devices < dev_len) {
+ parm = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_DEVICE_LIST, devices);
+ if (parm == -1 && codec->bus->rirb_error)
+ break;
+
+ for (i = 0; i < 8; i++) {
+ dev_list[devices] = (u8)parm;
+ parm >>= 4;
+ devices++;
+ if (devices >= dev_len)
+ break;
+ }
+ }
+ return devices;
+}
+
/**
* snd_hda_queue_unsol_event - add an unsolicited event to queue
* @bus: the BUS
@@ -1216,11 +1274,13 @@ static void hda_jackpoll_work(struct work_struct *work)
{
struct hda_codec *codec =
container_of(work, struct hda_codec, jackpoll_work.work);
- if (!codec->jackpoll_interval)
- return;
snd_hda_jack_set_dirty_all(codec);
snd_hda_jack_poll_all(codec);
+
+ if (!codec->jackpoll_interval)
+ return;
+
queue_delayed_work(codec->bus->workq, &codec->jackpoll_work,
codec->jackpoll_interval);
}
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 701c2e069b10..7aa9870040c1 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -94,6 +94,8 @@ enum {
#define AC_VERB_GET_HDMI_DIP_XMIT 0x0f32
#define AC_VERB_GET_HDMI_CP_CTRL 0x0f33
#define AC_VERB_GET_HDMI_CHAN_SLOT 0x0f34
+#define AC_VERB_GET_DEVICE_SEL 0xf35
+#define AC_VERB_GET_DEVICE_LIST 0xf36
/*
* SET verbs
@@ -133,6 +135,7 @@ enum {
#define AC_VERB_SET_HDMI_DIP_XMIT 0x732
#define AC_VERB_SET_HDMI_CP_CTRL 0x733
#define AC_VERB_SET_HDMI_CHAN_SLOT 0x734
+#define AC_VERB_SET_DEVICE_SEL 0x735
/*
* Parameter IDs
@@ -154,6 +157,7 @@ enum {
#define AC_PAR_GPIO_CAP 0x11
#define AC_PAR_AMP_OUT_CAP 0x12
#define AC_PAR_VOL_KNB_CAP 0x13
+#define AC_PAR_DEVLIST_LEN 0x15
#define AC_PAR_HDMI_LPCM_CAP 0x20
/*
@@ -251,6 +255,11 @@ enum {
#define AC_UNSOL_RES_TAG_SHIFT 26
#define AC_UNSOL_RES_SUBTAG (0x1f<<21)
#define AC_UNSOL_RES_SUBTAG_SHIFT 21
+#define AC_UNSOL_RES_DE (0x3f<<15) /* Device Entry
+ * (for DP1.2 MST)
+ */
+#define AC_UNSOL_RES_DE_SHIFT 15
+#define AC_UNSOL_RES_IA (1<<2) /* Inactive (for DP1.2 MST) */
#define AC_UNSOL_RES_ELDV (1<<1) /* ELD Data valid (for HDMI) */
#define AC_UNSOL_RES_PD (1<<0) /* pinsense detect */
#define AC_UNSOL_RES_CP_STATE (1<<1) /* content protection */
@@ -352,6 +361,10 @@ enum {
#define AC_LPCMCAP_44K (1<<30) /* 44.1kHz support */
#define AC_LPCMCAP_44K_MS (1<<31) /* 44.1kHz-multiplies support */
+/* Display pin's device list length */
+#define AC_DEV_LIST_LEN_MASK 0x3f
+#define AC_MAX_DEV_LIST_LEN 64
+
/*
* Control Parameters
*/
@@ -460,6 +473,11 @@ enum {
#define AC_DEFCFG_PORT_CONN (0x3<<30)
#define AC_DEFCFG_PORT_CONN_SHIFT 30
+/* Display pin's device list entry */
+#define AC_DE_PD (1<<0)
+#define AC_DE_ELDV (1<<1)
+#define AC_DE_IA (1<<2)
+
/* device device types (0x0-0xf) */
enum {
AC_JACK_LINE_OUT,
@@ -885,6 +903,7 @@ struct hda_codec {
unsigned int pcm_format_first:1; /* PCM format must be set first */
unsigned int epss:1; /* supporting EPSS? */
unsigned int cached_write:1; /* write only to caches */
+ unsigned int dp_mst:1; /* support DP1.2 Multi-stream transport */
#ifdef CONFIG_PM
unsigned int power_on :1; /* current (global) power-state */
unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */
@@ -972,6 +991,8 @@ int snd_hda_override_conn_list(struct hda_codec *codec, hda_nid_t nid, int nums,
const hda_nid_t *list);
int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux,
hda_nid_t nid, int recursive);
+int snd_hda_get_devices(struct hda_codec *codec, hda_nid_t nid,
+ u8 *dev_list, int max_devices);
int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid,
u32 *ratesp, u64 *formatsp, unsigned int *bpsp);
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index e3c7ba8d7582..ac41e9cdc976 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -142,6 +142,9 @@ static void parse_user_hints(struct hda_codec *codec)
val = snd_hda_get_bool_hint(codec, "primary_hp");
if (val >= 0)
spec->no_primary_hp = !val;
+ val = snd_hda_get_bool_hint(codec, "multi_io");
+ if (val >= 0)
+ spec->no_multi_io = !val;
val = snd_hda_get_bool_hint(codec, "multi_cap_vol");
if (val >= 0)
spec->multi_cap_vol = !!val;
@@ -813,6 +816,8 @@ static void resume_path_from_idx(struct hda_codec *codec, int path_idx)
static int hda_gen_mixer_mute_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol);
+static int hda_gen_bind_mute_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
enum {
HDA_CTL_WIDGET_VOL,
@@ -830,7 +835,13 @@ static const struct snd_kcontrol_new control_templates[] = {
.put = hda_gen_mixer_mute_put, /* replaced */
.private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0),
},
- HDA_BIND_MUTE(NULL, 0, 0, 0),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .info = snd_hda_mixer_amp_switch_info,
+ .get = snd_hda_mixer_bind_switch_get,
+ .put = hda_gen_bind_mute_put, /* replaced */
+ .private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0),
+ },
};
/* add dynamic controls from template */
@@ -937,8 +948,8 @@ static int add_stereo_sw(struct hda_codec *codec, const char *pfx,
}
/* playback mute control with the software mute bit check */
-static int hda_gen_mixer_mute_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
+static void sync_auto_mute_bits(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct hda_gen_spec *spec = codec->spec;
@@ -949,10 +960,22 @@ static int hda_gen_mixer_mute_put(struct snd_kcontrol *kcontrol,
ucontrol->value.integer.value[0] &= enabled;
ucontrol->value.integer.value[1] &= enabled;
}
+}
+static int hda_gen_mixer_mute_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ sync_auto_mute_bits(kcontrol, ucontrol);
return snd_hda_mixer_amp_switch_put(kcontrol, ucontrol);
}
+static int hda_gen_bind_mute_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ sync_auto_mute_bits(kcontrol, ucontrol);
+ return snd_hda_mixer_bind_switch_put(kcontrol, ucontrol);
+}
+
/* any ctl assigned to the path with the given index? */
static bool path_has_mixer(struct hda_codec *codec, int path_idx, int ctl_type)
{
@@ -1541,7 +1564,8 @@ static int fill_and_eval_dacs(struct hda_codec *codec,
cfg->speaker_pins,
spec->multiout.extra_out_nid,
spec->speaker_paths);
- if (fill_mio_first && cfg->line_outs == 1 &&
+ if (!spec->no_multi_io &&
+ fill_mio_first && cfg->line_outs == 1 &&
cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) {
err = fill_multi_ios(codec, cfg->line_out_pins[0], true);
if (!err)
@@ -1554,7 +1578,7 @@ static int fill_and_eval_dacs(struct hda_codec *codec,
spec->private_dac_nids, spec->out_paths,
spec->main_out_badness);
- if (fill_mio_first &&
+ if (!spec->no_multi_io && fill_mio_first &&
cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) {
/* try to fill multi-io first */
err = fill_multi_ios(codec, cfg->line_out_pins[0], false);
@@ -1582,7 +1606,8 @@ static int fill_and_eval_dacs(struct hda_codec *codec,
return err;
badness += err;
}
- if (cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) {
+ if (!spec->no_multi_io &&
+ cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) {
err = fill_multi_ios(codec, cfg->line_out_pins[0], false);
if (err < 0)
return err;
@@ -1600,7 +1625,8 @@ static int fill_and_eval_dacs(struct hda_codec *codec,
check_aamix_out_path(codec, spec->speaker_paths[0]);
}
- if (cfg->hp_outs && cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
+ if (!spec->no_multi_io &&
+ cfg->hp_outs && cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
if (count_multiio_pins(codec, cfg->hp_pins[0]) >= 2)
spec->multi_ios = 1; /* give badness */
@@ -3724,7 +3750,8 @@ static int mux_select(struct hda_codec *codec, unsigned int adc_idx,
/* check each pin in the given array; returns true if any of them is plugged */
static bool detect_jacks(struct hda_codec *codec, int num_pins, hda_nid_t *pins)
{
- int i, present = 0;
+ int i;
+ bool present = false;
for (i = 0; i < num_pins; i++) {
hda_nid_t nid = pins[i];
@@ -3733,14 +3760,15 @@ static bool detect_jacks(struct hda_codec *codec, int num_pins, hda_nid_t *pins)
/* don't detect pins retasked as inputs */
if (snd_hda_codec_get_pin_target(codec, nid) & AC_PINCTL_IN_EN)
continue;
- present |= snd_hda_jack_detect(codec, nid);
+ if (snd_hda_jack_detect_state(codec, nid) == HDA_JACK_PRESENT)
+ present = true;
}
return present;
}
/* standard HP/line-out auto-mute helper */
static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins,
- bool mute)
+ int *paths, bool mute)
{
struct hda_gen_spec *spec = codec->spec;
int i;
@@ -3752,10 +3780,19 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins,
break;
if (spec->auto_mute_via_amp) {
+ struct nid_path *path;
+ hda_nid_t mute_nid;
+
+ path = snd_hda_get_path_from_idx(codec, paths[i]);
+ if (!path)
+ continue;
+ mute_nid = get_amp_nid_(path->ctls[NID_PATH_MUTE_CTL]);
+ if (!mute_nid)
+ continue;
if (mute)
- spec->mute_bits |= (1ULL << nid);
+ spec->mute_bits |= (1ULL << mute_nid);
else
- spec->mute_bits &= ~(1ULL << nid);
+ spec->mute_bits &= ~(1ULL << mute_nid);
set_pin_eapd(codec, nid, !mute);
continue;
}
@@ -3786,14 +3823,19 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins,
void snd_hda_gen_update_outputs(struct hda_codec *codec)
{
struct hda_gen_spec *spec = codec->spec;
+ int *paths;
int on;
/* Control HP pins/amps depending on master_mute state;
* in general, HP pins/amps control should be enabled in all cases,
* but currently set only for master_mute, just to be safe
*/
+ if (spec->autocfg.line_out_type == AUTO_PIN_HP_OUT)
+ paths = spec->out_paths;
+ else
+ paths = spec->hp_paths;
do_automute(codec, ARRAY_SIZE(spec->autocfg.hp_pins),
- spec->autocfg.hp_pins, spec->master_mute);
+ spec->autocfg.hp_pins, paths, spec->master_mute);
if (!spec->automute_speaker)
on = 0;
@@ -3801,8 +3843,12 @@ void snd_hda_gen_update_outputs(struct hda_codec *codec)
on = spec->hp_jack_present | spec->line_jack_present;
on |= spec->master_mute;
spec->speaker_muted = on;
+ if (spec->autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT)
+ paths = spec->out_paths;
+ else
+ paths = spec->speaker_paths;
do_automute(codec, ARRAY_SIZE(spec->autocfg.speaker_pins),
- spec->autocfg.speaker_pins, on);
+ spec->autocfg.speaker_pins, paths, on);
/* toggle line-out mutes if needed, too */
/* if LO is a copy of either HP or Speaker, don't need to handle it */
@@ -3815,8 +3861,9 @@ void snd_hda_gen_update_outputs(struct hda_codec *codec)
on = spec->hp_jack_present;
on |= spec->master_mute;
spec->line_out_muted = on;
+ paths = spec->out_paths;
do_automute(codec, ARRAY_SIZE(spec->autocfg.line_out_pins),
- spec->autocfg.line_out_pins, on);
+ spec->autocfg.line_out_pins, paths, on);
}
EXPORT_SYMBOL_HDA(snd_hda_gen_update_outputs);
@@ -3887,7 +3934,7 @@ void snd_hda_gen_mic_autoswitch(struct hda_codec *codec, struct hda_jack_tbl *ja
/* don't detect pins retasked as outputs */
if (snd_hda_codec_get_pin_target(codec, pin) & AC_PINCTL_OUT_EN)
continue;
- if (snd_hda_jack_detect(codec, pin)) {
+ if (snd_hda_jack_detect_state(codec, pin) == HDA_JACK_PRESENT) {
mux_select(codec, 0, spec->am_entry[i].idx);
return;
}
diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h
index e199a852388b..48d44026705b 100644
--- a/sound/pci/hda/hda_generic.h
+++ b/sound/pci/hda/hda_generic.h
@@ -220,6 +220,7 @@ struct hda_gen_spec {
unsigned int hp_mic:1; /* Allow HP as a mic-in */
unsigned int suppress_hp_mic_detect:1; /* Don't detect HP/mic */
unsigned int no_primary_hp:1; /* Don't prefer HP pins to speaker pins */
+ unsigned int no_multi_io:1; /* Don't try multi I/O config */
unsigned int multi_cap_vol:1; /* allow multiple capture xxx volumes */
unsigned int inv_dmic_split:1; /* inverted dmic w/a for conexant */
unsigned int own_eapd_ctl:1; /* set EAPD by own function */
diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c
index ce67608734b5..fe0bda19de15 100644
--- a/sound/pci/hda/hda_hwdep.c
+++ b/sound/pci/hda/hda_hwdep.c
@@ -295,7 +295,7 @@ static ssize_t type##_store(struct device *dev, \
struct snd_hwdep *hwdep = dev_get_drvdata(dev); \
struct hda_codec *codec = hwdep->private_data; \
unsigned long val; \
- int err = strict_strtoul(buf, 0, &val); \
+ int err = kstrtoul(buf, 0, &val); \
if (err < 0) \
return err; \
codec->type = val; \
@@ -654,7 +654,7 @@ int snd_hda_get_int_hint(struct hda_codec *codec, const char *key, int *valp)
p = snd_hda_get_hint(codec, key);
if (!p)
ret = -ENOENT;
- else if (strict_strtoul(p, 0, &val))
+ else if (kstrtoul(p, 0, &val))
ret = -EINVAL;
else {
*valp = val;
@@ -751,7 +751,7 @@ static void parse_##name##_mode(char *buf, struct hda_bus *bus, \
struct hda_codec **codecp) \
{ \
unsigned long val; \
- if (!strict_strtoul(buf, 0, &val)) \
+ if (!kstrtoul(buf, 0, &val)) \
(*codecp)->name = val; \
}
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 8860dd529520..c6c98298ac39 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1160,7 +1160,7 @@ static int azx_reset(struct azx *chip, int full_reset)
goto __skip;
/* clear STATESTS */
- azx_writeb(chip, STATESTS, STATESTS_INT_MASK);
+ azx_writew(chip, STATESTS, STATESTS_INT_MASK);
/* reset controller */
azx_enter_link_reset(chip);
@@ -1242,7 +1242,7 @@ static void azx_int_clear(struct azx *chip)
}
/* clear STATESTS */
- azx_writeb(chip, STATESTS, STATESTS_INT_MASK);
+ azx_writew(chip, STATESTS, STATESTS_INT_MASK);
/* clear rirb status */
azx_writeb(chip, RIRBSTS, RIRB_INT_MASK);
@@ -1451,8 +1451,8 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id)
#if 0
/* clear state status int */
- if (azx_readb(chip, STATESTS) & 0x04)
- azx_writeb(chip, STATESTS, 0x04);
+ if (azx_readw(chip, STATESTS) & 0x04)
+ azx_writew(chip, STATESTS, 0x04);
#endif
spin_unlock(&chip->reg_lock);
@@ -2971,6 +2971,10 @@ static int azx_runtime_suspend(struct device *dev)
struct snd_card *card = dev_get_drvdata(dev);
struct azx *chip = card->private_data;
+ /* enable controller wake up event */
+ azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) |
+ STATESTS_INT_MASK);
+
azx_stop_chip(chip);
azx_enter_link_reset(chip);
azx_clear_irq_pending(chip);
@@ -2983,11 +2987,31 @@ static int azx_runtime_resume(struct device *dev)
{
struct snd_card *card = dev_get_drvdata(dev);
struct azx *chip = card->private_data;
+ struct hda_bus *bus;
+ struct hda_codec *codec;
+ int status;
if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL)
hda_display_power(true);
+
+ /* Read STATESTS before controller reset */
+ status = azx_readw(chip, STATESTS);
+
azx_init_pci(chip);
azx_init_chip(chip, 1);
+
+ bus = chip->bus;
+ if (status && bus) {
+ list_for_each_entry(codec, &bus->codec_list, list)
+ if (status & (1 << codec->addr))
+ queue_delayed_work(codec->bus->workq,
+ &codec->jackpoll_work, codec->jackpoll_interval);
+ }
+
+ /* disable controller Wake Up event*/
+ azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) &
+ ~STATESTS_INT_MASK);
+
return 0;
}
@@ -3831,11 +3855,13 @@ static int azx_probe_continue(struct azx *chip)
/* Request power well for Haswell HDA controller and codec */
if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) {
+#ifdef CONFIG_SND_HDA_I915
err = hda_i915_init();
if (err < 0) {
snd_printk(KERN_ERR SFX "Error request power-well from i915\n");
goto out_free;
}
+#endif
hda_display_power(true);
}
diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c
index 3fd2973183e2..05b3e3e9108f 100644
--- a/sound/pci/hda/hda_jack.c
+++ b/sound/pci/hda/hda_jack.c
@@ -194,18 +194,24 @@ u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid)
EXPORT_SYMBOL_HDA(snd_hda_pin_sense);
/**
- * snd_hda_jack_detect - query pin Presence Detect status
+ * snd_hda_jack_detect_state - query pin Presence Detect status
* @codec: the CODEC to sense
* @nid: the pin NID to sense
*
- * Query and return the pin's Presence Detect status.
+ * Query and return the pin's Presence Detect status, as either
+ * HDA_JACK_NOT_PRESENT, HDA_JACK_PRESENT or HDA_JACK_PHANTOM.
*/
-int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid)
+int snd_hda_jack_detect_state(struct hda_codec *codec, hda_nid_t nid)
{
- u32 sense = snd_hda_pin_sense(codec, nid);
- return get_jack_plug_state(sense);
+ struct hda_jack_tbl *jack = snd_hda_jack_tbl_get(codec, nid);
+ if (jack && jack->phantom_jack)
+ return HDA_JACK_PHANTOM;
+ else if (snd_hda_pin_sense(codec, nid) & AC_PINSENSE_PRESENCE)
+ return HDA_JACK_PRESENT;
+ else
+ return HDA_JACK_NOT_PRESENT;
}
-EXPORT_SYMBOL_HDA(snd_hda_jack_detect);
+EXPORT_SYMBOL_HDA(snd_hda_jack_detect_state);
/**
* snd_hda_jack_detect_enable - enable the jack-detection
@@ -247,8 +253,8 @@ EXPORT_SYMBOL_HDA(snd_hda_jack_detect_enable);
int snd_hda_jack_set_gating_jack(struct hda_codec *codec, hda_nid_t gated_nid,
hda_nid_t gating_nid)
{
- struct hda_jack_tbl *gated = snd_hda_jack_tbl_get(codec, gated_nid);
- struct hda_jack_tbl *gating = snd_hda_jack_tbl_get(codec, gating_nid);
+ struct hda_jack_tbl *gated = snd_hda_jack_tbl_new(codec, gated_nid);
+ struct hda_jack_tbl *gating = snd_hda_jack_tbl_new(codec, gating_nid);
if (!gated || !gating)
return -EINVAL;
diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h
index ec12abd45263..379420c44eef 100644
--- a/sound/pci/hda/hda_jack.h
+++ b/sound/pci/hda/hda_jack.h
@@ -75,7 +75,18 @@ int snd_hda_jack_set_gating_jack(struct hda_codec *codec, hda_nid_t gated_nid,
hda_nid_t gating_nid);
u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid);
-int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid);
+
+/* the jack state returned from snd_hda_jack_detect_state() */
+enum {
+ HDA_JACK_NOT_PRESENT, HDA_JACK_PRESENT, HDA_JACK_PHANTOM,
+};
+
+int snd_hda_jack_detect_state(struct hda_codec *codec, hda_nid_t nid);
+
+static inline bool snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid)
+{
+ return snd_hda_jack_detect_state(codec, nid) != HDA_JACK_NOT_PRESENT;
+}
bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid);
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index 9760f001916d..a8cb22eec89e 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -582,6 +582,36 @@ static void print_gpio(struct snd_info_buffer *buffer,
print_nid_array(buffer, codec, nid, &codec->nids);
}
+static void print_device_list(struct snd_info_buffer *buffer,
+ struct hda_codec *codec, hda_nid_t nid)
+{
+ int i, curr = -1;
+ u8 dev_list[AC_MAX_DEV_LIST_LEN];
+ int devlist_len;
+
+ devlist_len = snd_hda_get_devices(codec, nid, dev_list,
+ AC_MAX_DEV_LIST_LEN);
+ snd_iprintf(buffer, " Devices: %d\n", devlist_len);
+ if (devlist_len <= 0)
+ return;
+
+ curr = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_DEVICE_SEL, 0);
+
+ for (i = 0; i < devlist_len; i++) {
+ if (i == curr)
+ snd_iprintf(buffer, " *");
+ else
+ snd_iprintf(buffer, " ");
+
+ snd_iprintf(buffer,
+ "Dev %02d: PD = %d, ELDV = %d, IA = %d\n", i,
+ !!(dev_list[i] & AC_DE_PD),
+ !!(dev_list[i] & AC_DE_ELDV),
+ !!(dev_list[i] & AC_DE_IA));
+ }
+}
+
static void print_codec_info(struct snd_info_entry *entry,
struct snd_info_buffer *buffer)
{
@@ -751,6 +781,9 @@ static void print_codec_info(struct snd_info_entry *entry,
(wid_caps & AC_WCAP_DELAY) >>
AC_WCAP_DELAY_SHIFT);
+ if (wid_type == AC_WID_PIN && codec->dp_mst)
+ print_device_list(buffer, codec, nid);
+
if (wid_caps & AC_WCAP_CONN_LIST)
print_conn_list(buffer, codec, nid, wid_type,
conn, conn_len);
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index d97f0d61a15b..0cbdd87dde6d 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -32,7 +32,6 @@
#include "hda_jack.h"
#include "hda_generic.h"
-#define ENABLE_AD_STATIC_QUIRKS
struct ad198x_spec {
struct hda_gen_spec gen;
@@ -43,114 +42,8 @@ struct ad198x_spec {
hda_nid_t eapd_nid;
unsigned int beep_amp; /* beep amp value, set via set_beep_amp() */
-
-#ifdef ENABLE_AD_STATIC_QUIRKS
- const struct snd_kcontrol_new *mixers[6];
- int num_mixers;
- const struct hda_verb *init_verbs[6]; /* initialization verbs
- * don't forget NULL termination!
- */
- unsigned int num_init_verbs;
-
- /* playback */
- struct hda_multi_out multiout; /* playback set-up
- * max_channels, dacs must be set
- * dig_out_nid and hp_nid are optional
- */
- unsigned int cur_eapd;
- unsigned int need_dac_fix;
-
- /* capture */
- unsigned int num_adc_nids;
- const hda_nid_t *adc_nids;
- hda_nid_t dig_in_nid; /* digital-in NID; optional */
-
- /* capture source */
- const struct hda_input_mux *input_mux;
- const hda_nid_t *capsrc_nids;
- unsigned int cur_mux[3];
-
- /* channel model */
- const struct hda_channel_mode *channel_mode;
- int num_channel_mode;
-
- /* PCM information */
- struct hda_pcm pcm_rec[3]; /* used in alc_build_pcms() */
-
- unsigned int spdif_route;
-
- unsigned int jack_present: 1;
- unsigned int inv_jack_detect: 1;/* inverted jack-detection */
- unsigned int analog_beep: 1; /* analog beep input present */
- unsigned int avoid_init_slave_vol:1;
-
-#ifdef CONFIG_PM
- struct hda_loopback_check loopback;
-#endif
- /* for virtual master */
- hda_nid_t vmaster_nid;
- const char * const *slave_vols;
- const char * const *slave_sws;
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-};
-
-#ifdef ENABLE_AD_STATIC_QUIRKS
-/*
- * input MUX handling (common part)
- */
-static int ad198x_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct ad198x_spec *spec = codec->spec;
-
- return snd_hda_input_mux_info(spec->input_mux, uinfo);
-}
-
-static int ad198x_mux_enum_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct ad198x_spec *spec = codec->spec;
- unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
-
- ucontrol->value.enumerated.item[0] = spec->cur_mux[adc_idx];
- return 0;
-}
-
-static int ad198x_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct ad198x_spec *spec = codec->spec;
- unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
-
- return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol,
- spec->capsrc_nids[adc_idx],
- &spec->cur_mux[adc_idx]);
-}
-
-/*
- * initialization (common callbacks)
- */
-static int ad198x_init(struct hda_codec *codec)
-{
- struct ad198x_spec *spec = codec->spec;
- int i;
-
- for (i = 0; i < spec->num_init_verbs; i++)
- snd_hda_sequence_write(codec, spec->init_verbs[i]);
- return 0;
-}
-
-static const char * const ad_slave_pfxs[] = {
- "Front", "Surround", "Center", "LFE", "Side",
- "Headphone", "Mono", "Speaker", "IEC958",
- NULL
};
-static const char * const ad1988_6stack_fp_slave_pfxs[] = {
- "Front", "Surround", "Center", "LFE", "Side", "IEC958",
- NULL
-};
-#endif /* ENABLE_AD_STATIC_QUIRKS */
#ifdef CONFIG_SND_HDA_INPUT_BEEP
/* additional beep mixers; the actual parameters are overwritten at build */
@@ -160,12 +53,6 @@ static const struct snd_kcontrol_new ad_beep_mixer[] = {
{ } /* end */
};
-static const struct snd_kcontrol_new ad_beep2_mixer[] = {
- HDA_CODEC_VOLUME("Digital Beep Playback Volume", 0, 0, HDA_OUTPUT),
- HDA_CODEC_MUTE_BEEP("Digital Beep Playback Switch", 0, 0, HDA_OUTPUT),
- { } /* end */
-};
-
#define set_beep_amp(spec, nid, idx, dir) \
((spec)->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 1, idx, dir)) /* mono */
#else
@@ -181,8 +68,7 @@ static int create_beep_ctls(struct hda_codec *codec)
if (!spec->beep_amp)
return 0;
- knew = spec->analog_beep ? ad_beep2_mixer : ad_beep_mixer;
- for ( ; knew->name; knew++) {
+ for (knew = ad_beep_mixer ; knew->name; knew++) {
int err;
struct snd_kcontrol *kctl;
kctl = snd_ctl_new1(knew, codec);
@@ -199,268 +85,6 @@ static int create_beep_ctls(struct hda_codec *codec)
#define create_beep_ctls(codec) 0
#endif
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static int ad198x_build_controls(struct hda_codec *codec)
-{
- struct ad198x_spec *spec = codec->spec;
- struct snd_kcontrol *kctl;
- unsigned int i;
- int err;
-
- for (i = 0; i < spec->num_mixers; i++) {
- err = snd_hda_add_new_ctls(codec, spec->mixers[i]);
- if (err < 0)
- return err;
- }
- if (spec->multiout.dig_out_nid) {
- err = snd_hda_create_spdif_out_ctls(codec,
- spec->multiout.dig_out_nid,
- spec->multiout.dig_out_nid);
- if (err < 0)
- return err;
- err = snd_hda_create_spdif_share_sw(codec,
- &spec->multiout);
- if (err < 0)
- return err;
- spec->multiout.share_spdif = 1;
- }
- if (spec->dig_in_nid) {
- err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid);
- if (err < 0)
- return err;
- }
-
- /* create beep controls if needed */
- err = create_beep_ctls(codec);
- if (err < 0)
- return err;
-
- /* if we have no master control, let's create it */
- if (!snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) {
- unsigned int vmaster_tlv[4];
- snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid,
- HDA_OUTPUT, vmaster_tlv);
- err = __snd_hda_add_vmaster(codec, "Master Playback Volume",
- vmaster_tlv,
- (spec->slave_vols ?
- spec->slave_vols : ad_slave_pfxs),
- "Playback Volume",
- !spec->avoid_init_slave_vol, NULL);
- if (err < 0)
- return err;
- }
- if (!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) {
- err = snd_hda_add_vmaster(codec, "Master Playback Switch",
- NULL,
- (spec->slave_sws ?
- spec->slave_sws : ad_slave_pfxs),
- "Playback Switch");
- if (err < 0)
- return err;
- }
-
- /* assign Capture Source enums to NID */
- kctl = snd_hda_find_mixer_ctl(codec, "Capture Source");
- if (!kctl)
- kctl = snd_hda_find_mixer_ctl(codec, "Input Source");
- for (i = 0; kctl && i < kctl->count; i++) {
- err = snd_hda_add_nid(codec, kctl, i, spec->capsrc_nids[i]);
- if (err < 0)
- return err;
- }
-
- /* assign IEC958 enums to NID */
- kctl = snd_hda_find_mixer_ctl(codec,
- SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source");
- if (kctl) {
- err = snd_hda_add_nid(codec, kctl, 0,
- spec->multiout.dig_out_nid);
- if (err < 0)
- return err;
- }
-
- return 0;
-}
-
-#ifdef CONFIG_PM
-static int ad198x_check_power_status(struct hda_codec *codec, hda_nid_t nid)
-{
- struct ad198x_spec *spec = codec->spec;
- return snd_hda_check_amp_list_power(codec, &spec->loopback, nid);
-}
-#endif
-
-/*
- * Analog playback callbacks
- */
-static int ad198x_playback_pcm_open(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- struct snd_pcm_substream *substream)
-{
- struct ad198x_spec *spec = codec->spec;
- return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream,
- hinfo);
-}
-
-static int ad198x_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- unsigned int stream_tag,
- unsigned int format,
- struct snd_pcm_substream *substream)
-{
- struct ad198x_spec *spec = codec->spec;
- return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, stream_tag,
- format, substream);
-}
-
-static int ad198x_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- struct snd_pcm_substream *substream)
-{
- struct ad198x_spec *spec = codec->spec;
- return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout);
-}
-
-/*
- * Digital out
- */
-static int ad198x_dig_playback_pcm_open(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- struct snd_pcm_substream *substream)
-{
- struct ad198x_spec *spec = codec->spec;
- return snd_hda_multi_out_dig_open(codec, &spec->multiout);
-}
-
-static int ad198x_dig_playback_pcm_close(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- struct snd_pcm_substream *substream)
-{
- struct ad198x_spec *spec = codec->spec;
- return snd_hda_multi_out_dig_close(codec, &spec->multiout);
-}
-
-static int ad198x_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- unsigned int stream_tag,
- unsigned int format,
- struct snd_pcm_substream *substream)
-{
- struct ad198x_spec *spec = codec->spec;
- return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, stream_tag,
- format, substream);
-}
-
-static int ad198x_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- struct snd_pcm_substream *substream)
-{
- struct ad198x_spec *spec = codec->spec;
- return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout);
-}
-
-/*
- * Analog capture
- */
-static int ad198x_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 ad198x_spec *spec = codec->spec;
- snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number],
- stream_tag, 0, format);
- return 0;
-}
-
-static int ad198x_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- struct snd_pcm_substream *substream)
-{
- struct ad198x_spec *spec = codec->spec;
- snd_hda_codec_cleanup_stream(codec, spec->adc_nids[substream->number]);
- return 0;
-}
-
-/*
- */
-static const struct hda_pcm_stream ad198x_pcm_analog_playback = {
- .substreams = 1,
- .channels_min = 2,
- .channels_max = 6, /* changed later */
- .nid = 0, /* fill later */
- .ops = {
- .open = ad198x_playback_pcm_open,
- .prepare = ad198x_playback_pcm_prepare,
- .cleanup = ad198x_playback_pcm_cleanup,
- },
-};
-
-static const struct hda_pcm_stream ad198x_pcm_analog_capture = {
- .substreams = 1,
- .channels_min = 2,
- .channels_max = 2,
- .nid = 0, /* fill later */
- .ops = {
- .prepare = ad198x_capture_pcm_prepare,
- .cleanup = ad198x_capture_pcm_cleanup
- },
-};
-
-static const struct hda_pcm_stream ad198x_pcm_digital_playback = {
- .substreams = 1,
- .channels_min = 2,
- .channels_max = 2,
- .nid = 0, /* fill later */
- .ops = {
- .open = ad198x_dig_playback_pcm_open,
- .close = ad198x_dig_playback_pcm_close,
- .prepare = ad198x_dig_playback_pcm_prepare,
- .cleanup = ad198x_dig_playback_pcm_cleanup
- },
-};
-
-static const struct hda_pcm_stream ad198x_pcm_digital_capture = {
- .substreams = 1,
- .channels_min = 2,
- .channels_max = 2,
- /* NID is set in alc_build_pcms */
-};
-
-static int ad198x_build_pcms(struct hda_codec *codec)
-{
- struct ad198x_spec *spec = codec->spec;
- struct hda_pcm *info = spec->pcm_rec;
-
- codec->num_pcms = 1;
- codec->pcm_info = info;
-
- info->name = "AD198x Analog";
- info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ad198x_pcm_analog_playback;
- info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = spec->multiout.max_channels;
- info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dac_nids[0];
- info->stream[SNDRV_PCM_STREAM_CAPTURE] = ad198x_pcm_analog_capture;
- info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = spec->num_adc_nids;
- info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0];
-
- if (spec->multiout.dig_out_nid) {
- info++;
- codec->num_pcms++;
- codec->spdif_status_reset = 1;
- info->name = "AD198x Digital";
- info->pcm_type = HDA_PCM_TYPE_SPDIF;
- info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ad198x_pcm_digital_playback;
- info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dig_out_nid;
- if (spec->dig_in_nid) {
- info->stream[SNDRV_PCM_STREAM_CAPTURE] = ad198x_pcm_digital_capture;
- info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in_nid;
- }
- }
-
- return 0;
-}
-#endif /* ENABLE_AD_STATIC_QUIRKS */
static void ad198x_power_eapd_write(struct hda_codec *codec, hda_nid_t front,
hda_nid_t hp)
@@ -507,18 +131,6 @@ static void ad198x_shutup(struct hda_codec *codec)
ad198x_power_eapd(codec);
}
-static void ad198x_free(struct hda_codec *codec)
-{
- struct ad198x_spec *spec = codec->spec;
-
- if (!spec)
- return;
-
- snd_hda_gen_spec_free(&spec->gen);
- kfree(spec);
- snd_hda_detach_beep_device(codec);
-}
-
#ifdef CONFIG_PM
static int ad198x_suspend(struct hda_codec *codec)
{
@@ -527,65 +139,6 @@ static int ad198x_suspend(struct hda_codec *codec)
}
#endif
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static const struct hda_codec_ops ad198x_patch_ops = {
- .build_controls = ad198x_build_controls,
- .build_pcms = ad198x_build_pcms,
- .init = ad198x_init,
- .free = ad198x_free,
-#ifdef CONFIG_PM
- .check_power_status = ad198x_check_power_status,
- .suspend = ad198x_suspend,
-#endif
- .reboot_notify = ad198x_shutup,
-};
-
-
-/*
- * EAPD control
- * the private value = nid
- */
-#define ad198x_eapd_info snd_ctl_boolean_mono_info
-
-static int ad198x_eapd_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct ad198x_spec *spec = codec->spec;
- if (codec->inv_eapd)
- ucontrol->value.integer.value[0] = ! spec->cur_eapd;
- else
- ucontrol->value.integer.value[0] = spec->cur_eapd;
- return 0;
-}
-
-static int ad198x_eapd_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct ad198x_spec *spec = codec->spec;
- hda_nid_t nid = kcontrol->private_value & 0xff;
- unsigned int eapd;
- eapd = !!ucontrol->value.integer.value[0];
- if (codec->inv_eapd)
- eapd = !eapd;
- if (eapd == spec->cur_eapd)
- return 0;
- spec->cur_eapd = eapd;
- snd_hda_codec_write_cache(codec, nid,
- 0, AC_VERB_SET_EAPD_BTLENABLE,
- eapd ? 0x02 : 0x00);
- return 1;
-}
-
-static int ad198x_ch_mode_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo);
-static int ad198x_ch_mode_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol);
-static int ad198x_ch_mode_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol);
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
/*
* Automatic parse of I/O pins from the BIOS configuration
@@ -646,537 +199,6 @@ static int ad198x_parse_auto_config(struct hda_codec *codec)
* AD1986A specific
*/
-#ifdef ENABLE_AD_STATIC_QUIRKS
-#define AD1986A_SPDIF_OUT 0x02
-#define AD1986A_FRONT_DAC 0x03
-#define AD1986A_SURR_DAC 0x04
-#define AD1986A_CLFE_DAC 0x05
-#define AD1986A_ADC 0x06
-
-static const hda_nid_t ad1986a_dac_nids[3] = {
- AD1986A_FRONT_DAC, AD1986A_SURR_DAC, AD1986A_CLFE_DAC
-};
-static const hda_nid_t ad1986a_adc_nids[1] = { AD1986A_ADC };
-static const hda_nid_t ad1986a_capsrc_nids[1] = { 0x12 };
-
-static const struct hda_input_mux ad1986a_capture_source = {
- .num_items = 7,
- .items = {
- { "Mic", 0x0 },
- { "CD", 0x1 },
- { "Aux", 0x3 },
- { "Line", 0x4 },
- { "Mix", 0x5 },
- { "Mono", 0x6 },
- { "Phone", 0x7 },
- },
-};
-
-
-static const struct hda_bind_ctls ad1986a_bind_pcm_vol = {
- .ops = &snd_hda_bind_vol,
- .values = {
- HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT),
- HDA_COMPOSE_AMP_VAL(AD1986A_SURR_DAC, 3, 0, HDA_OUTPUT),
- HDA_COMPOSE_AMP_VAL(AD1986A_CLFE_DAC, 3, 0, HDA_OUTPUT),
- 0
- },
-};
-
-static const struct hda_bind_ctls ad1986a_bind_pcm_sw = {
- .ops = &snd_hda_bind_sw,
- .values = {
- HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT),
- HDA_COMPOSE_AMP_VAL(AD1986A_SURR_DAC, 3, 0, HDA_OUTPUT),
- HDA_COMPOSE_AMP_VAL(AD1986A_CLFE_DAC, 3, 0, HDA_OUTPUT),
- 0
- },
-};
-
-/*
- * mixers
- */
-static const struct snd_kcontrol_new ad1986a_mixers[] = {
- /*
- * bind volumes/mutes of 3 DACs as a single PCM control for simplicity
- */
- HDA_BIND_VOL("PCM Playback Volume", &ad1986a_bind_pcm_vol),
- HDA_BIND_SW("PCM Playback Switch", &ad1986a_bind_pcm_sw),
- HDA_CODEC_VOLUME("Front Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x1c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Surround Playback Switch", 0x1c, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x1d, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x1d, 2, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x1d, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x1d, 2, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x1a, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x17, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x17, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Aux Playback Volume", 0x16, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Aux Playback Switch", 0x16, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x0f, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Mono Playback Volume", 0x1e, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Mono Playback Switch", 0x1e, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Capture Source",
- .info = ad198x_mux_enum_info,
- .get = ad198x_mux_enum_get,
- .put = ad198x_mux_enum_put,
- },
- HDA_CODEC_MUTE("Stereo Downmix Switch", 0x09, 0x0, HDA_OUTPUT),
- { } /* end */
-};
-
-/* additional mixers for 3stack mode */
-static const struct snd_kcontrol_new ad1986a_3st_mixers[] = {
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Channel Mode",
- .info = ad198x_ch_mode_info,
- .get = ad198x_ch_mode_get,
- .put = ad198x_ch_mode_put,
- },
- { } /* end */
-};
-
-/* laptop model - 2ch only */
-static const hda_nid_t ad1986a_laptop_dac_nids[1] = { AD1986A_FRONT_DAC };
-
-/* master controls both pins 0x1a and 0x1b */
-static const struct hda_bind_ctls ad1986a_laptop_master_vol = {
- .ops = &snd_hda_bind_vol,
- .values = {
- HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT),
- HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT),
- 0,
- },
-};
-
-static const struct hda_bind_ctls ad1986a_laptop_master_sw = {
- .ops = &snd_hda_bind_sw,
- .values = {
- HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT),
- HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT),
- 0,
- },
-};
-
-static const struct snd_kcontrol_new ad1986a_laptop_mixers[] = {
- HDA_CODEC_VOLUME("PCM Playback Volume", 0x03, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT),
- HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol),
- HDA_BIND_SW("Master Playback Switch", &ad1986a_laptop_master_sw),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x17, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x17, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Aux Playback Volume", 0x16, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Aux Playback Switch", 0x16, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x0f, 0x0, HDA_OUTPUT),
- /*
- HDA_CODEC_VOLUME("Mono Playback Volume", 0x1e, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Mono Playback Switch", 0x1e, 0x0, HDA_OUTPUT), */
- HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Capture Source",
- .info = ad198x_mux_enum_info,
- .get = ad198x_mux_enum_get,
- .put = ad198x_mux_enum_put,
- },
- { } /* end */
-};
-
-/* laptop-eapd model - 2ch only */
-
-static const struct hda_input_mux ad1986a_laptop_eapd_capture_source = {
- .num_items = 3,
- .items = {
- { "Mic", 0x0 },
- { "Internal Mic", 0x4 },
- { "Mix", 0x5 },
- },
-};
-
-static const struct hda_input_mux ad1986a_automic_capture_source = {
- .num_items = 2,
- .items = {
- { "Mic", 0x0 },
- { "Mix", 0x5 },
- },
-};
-
-static const struct snd_kcontrol_new ad1986a_laptop_master_mixers[] = {
- HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol),
- HDA_BIND_SW("Master Playback Switch", &ad1986a_laptop_master_sw),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new ad1986a_laptop_eapd_mixers[] = {
- HDA_CODEC_VOLUME("PCM Playback Volume", 0x03, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x0f, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Capture Source",
- .info = ad198x_mux_enum_info,
- .get = ad198x_mux_enum_get,
- .put = ad198x_mux_enum_put,
- },
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "External Amplifier",
- .subdevice = HDA_SUBDEV_NID_FLAG | 0x1b,
- .info = ad198x_eapd_info,
- .get = ad198x_eapd_get,
- .put = ad198x_eapd_put,
- .private_value = 0x1b, /* port-D */
- },
- { } /* end */
-};
-
-static const struct snd_kcontrol_new ad1986a_laptop_intmic_mixers[] = {
- HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x17, 0, HDA_OUTPUT),
- { } /* end */
-};
-
-/* re-connect the mic boost input according to the jack sensing */
-static void ad1986a_automic(struct hda_codec *codec)
-{
- unsigned int present;
- present = snd_hda_jack_detect(codec, 0x1f);
- /* 0 = 0x1f, 2 = 0x1d, 4 = mixed */
- snd_hda_codec_write(codec, 0x0f, 0, AC_VERB_SET_CONNECT_SEL,
- present ? 0 : 2);
-}
-
-#define AD1986A_MIC_EVENT 0x36
-
-static void ad1986a_automic_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- if ((res >> 26) != AD1986A_MIC_EVENT)
- return;
- ad1986a_automic(codec);
-}
-
-static int ad1986a_automic_init(struct hda_codec *codec)
-{
- ad198x_init(codec);
- ad1986a_automic(codec);
- return 0;
-}
-
-/* laptop-automute - 2ch only */
-
-static void ad1986a_update_hp(struct hda_codec *codec)
-{
- struct ad198x_spec *spec = codec->spec;
- unsigned int mute;
-
- if (spec->jack_present)
- mute = HDA_AMP_MUTE; /* mute internal speaker */
- else
- /* unmute internal speaker if necessary */
- mute = snd_hda_codec_amp_read(codec, 0x1a, 0, HDA_OUTPUT, 0);
- snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, mute);
-}
-
-static void ad1986a_hp_automute(struct hda_codec *codec)
-{
- struct ad198x_spec *spec = codec->spec;
-
- spec->jack_present = snd_hda_jack_detect(codec, 0x1a);
- if (spec->inv_jack_detect)
- spec->jack_present = !spec->jack_present;
- ad1986a_update_hp(codec);
-}
-
-#define AD1986A_HP_EVENT 0x37
-
-static void ad1986a_hp_unsol_event(struct hda_codec *codec, unsigned int res)
-{
- if ((res >> 26) != AD1986A_HP_EVENT)
- return;
- ad1986a_hp_automute(codec);
-}
-
-static int ad1986a_hp_init(struct hda_codec *codec)
-{
- ad198x_init(codec);
- ad1986a_hp_automute(codec);
- return 0;
-}
-
-/* bind hp and internal speaker mute (with plug check) */
-static int ad1986a_hp_master_sw_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- int change = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol);
- if (change)
- ad1986a_update_hp(codec);
- return change;
-}
-
-static const struct snd_kcontrol_new ad1986a_automute_master_mixers[] = {
- HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Master Playback Switch",
- .subdevice = HDA_SUBDEV_AMP_FLAG,
- .info = snd_hda_mixer_amp_switch_info,
- .get = snd_hda_mixer_amp_switch_get,
- .put = ad1986a_hp_master_sw_put,
- .private_value = HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT),
- },
- { } /* end */
-};
-
-
-/*
- * initialization verbs
- */
-static const struct hda_verb ad1986a_init_verbs[] = {
- /* Front, Surround, CLFE DAC; mute as default */
- {0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x05, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- /* Downmix - off */
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- /* HP, Line-Out, Surround, CLFE selectors */
- {0x0a, AC_VERB_SET_CONNECT_SEL, 0x0},
- {0x0b, AC_VERB_SET_CONNECT_SEL, 0x0},
- {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0},
- {0x0d, AC_VERB_SET_CONNECT_SEL, 0x0},
- /* Mono selector */
- {0x0e, AC_VERB_SET_CONNECT_SEL, 0x0},
- /* Mic selector: Mic 1/2 pin */
- {0x0f, AC_VERB_SET_CONNECT_SEL, 0x0},
- /* Line-in selector: Line-in */
- {0x10, AC_VERB_SET_CONNECT_SEL, 0x0},
- /* Mic 1/2 swap */
- {0x11, AC_VERB_SET_CONNECT_SEL, 0x0},
- /* Record selector: mic */
- {0x12, AC_VERB_SET_CONNECT_SEL, 0x0},
- /* Mic, Phone, CD, Aux, Line-In amp; mute as default */
- {0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- /* PC beep */
- {0x18, AC_VERB_SET_CONNECT_SEL, 0x0},
- /* HP, Line-Out, Surround, CLFE, Mono pins; mute as default */
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- /* HP Pin */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
- /* Front, Surround, CLFE Pins */
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
- {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
- /* Mono Pin */
- {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
- /* Mic Pin */
- {0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
- /* Line, Aux, CD, Beep-In Pin */
- {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
- {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
- {0x22, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
- {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
- {0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
- { } /* end */
-};
-
-static const struct hda_verb ad1986a_ch2_init[] = {
- /* Surround out -> Line In */
- { 0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
- /* Line-in selectors */
- { 0x10, AC_VERB_SET_CONNECT_SEL, 0x1 },
- /* CLFE -> Mic in */
- { 0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
- /* Mic selector, mix C/LFE (backmic) and Mic (frontmic) */
- { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x4 },
- { } /* end */
-};
-
-static const struct hda_verb ad1986a_ch4_init[] = {
- /* Surround out -> Surround */
- { 0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x10, AC_VERB_SET_CONNECT_SEL, 0x0 },
- /* CLFE -> Mic in */
- { 0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
- { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x4 },
- { } /* end */
-};
-
-static const struct hda_verb ad1986a_ch6_init[] = {
- /* Surround out -> Surround out */
- { 0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x10, AC_VERB_SET_CONNECT_SEL, 0x0 },
- /* CLFE -> CLFE */
- { 0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x0 },
- { } /* end */
-};
-
-static const struct hda_channel_mode ad1986a_modes[3] = {
- { 2, ad1986a_ch2_init },
- { 4, ad1986a_ch4_init },
- { 6, ad1986a_ch6_init },
-};
-
-/* eapd initialization */
-static const struct hda_verb ad1986a_eapd_init_verbs[] = {
- {0x1b, AC_VERB_SET_EAPD_BTLENABLE, 0x00 },
- {}
-};
-
-static const struct hda_verb ad1986a_automic_verbs[] = {
- {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- /*{0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},*/
- {0x0f, AC_VERB_SET_CONNECT_SEL, 0x0},
- {0x1f, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1986A_MIC_EVENT},
- {}
-};
-
-/* Ultra initialization */
-static const struct hda_verb ad1986a_ultra_init[] = {
- /* eapd initialization */
- { 0x1b, AC_VERB_SET_EAPD_BTLENABLE, 0x00 },
- /* CLFE -> Mic in */
- { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2 },
- { 0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
- { 0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080 },
- { } /* end */
-};
-
-/* pin sensing on HP jack */
-static const struct hda_verb ad1986a_hp_init_verbs[] = {
- {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1986A_HP_EVENT},
- {}
-};
-
-static void ad1986a_samsung_p50_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- switch (res >> 26) {
- case AD1986A_HP_EVENT:
- ad1986a_hp_automute(codec);
- break;
- case AD1986A_MIC_EVENT:
- ad1986a_automic(codec);
- break;
- }
-}
-
-static int ad1986a_samsung_p50_init(struct hda_codec *codec)
-{
- ad198x_init(codec);
- ad1986a_hp_automute(codec);
- ad1986a_automic(codec);
- return 0;
-}
-
-
-/* models */
-enum {
- AD1986A_AUTO,
- AD1986A_6STACK,
- AD1986A_3STACK,
- AD1986A_LAPTOP,
- AD1986A_LAPTOP_EAPD,
- AD1986A_LAPTOP_AUTOMUTE,
- AD1986A_ULTRA,
- AD1986A_SAMSUNG,
- AD1986A_SAMSUNG_P50,
- AD1986A_MODELS
-};
-
-static const char * const ad1986a_models[AD1986A_MODELS] = {
- [AD1986A_AUTO] = "auto",
- [AD1986A_6STACK] = "6stack",
- [AD1986A_3STACK] = "3stack",
- [AD1986A_LAPTOP] = "laptop",
- [AD1986A_LAPTOP_EAPD] = "laptop-eapd",
- [AD1986A_LAPTOP_AUTOMUTE] = "laptop-automute",
- [AD1986A_ULTRA] = "ultra",
- [AD1986A_SAMSUNG] = "samsung",
- [AD1986A_SAMSUNG_P50] = "samsung-p50",
-};
-
-static const struct snd_pci_quirk ad1986a_cfg_tbl[] = {
- SND_PCI_QUIRK(0x103c, 0x30af, "HP B2800", AD1986A_LAPTOP_EAPD),
- SND_PCI_QUIRK(0x1043, 0x1153, "ASUS M9", AD1986A_LAPTOP_EAPD),
- SND_PCI_QUIRK(0x1043, 0x11f7, "ASUS U5A", AD1986A_LAPTOP_EAPD),
- SND_PCI_QUIRK(0x1043, 0x1213, "ASUS A6J", AD1986A_LAPTOP_EAPD),
- SND_PCI_QUIRK(0x1043, 0x1263, "ASUS U5F", AD1986A_LAPTOP_EAPD),
- SND_PCI_QUIRK(0x1043, 0x1297, "ASUS Z62F", AD1986A_LAPTOP_EAPD),
- SND_PCI_QUIRK(0x1043, 0x12b3, "ASUS V1j", AD1986A_LAPTOP_EAPD),
- SND_PCI_QUIRK(0x1043, 0x1302, "ASUS W3j", AD1986A_LAPTOP_EAPD),
- SND_PCI_QUIRK(0x1043, 0x1443, "ASUS VX1", AD1986A_LAPTOP),
- SND_PCI_QUIRK(0x1043, 0x1447, "ASUS A8J", AD1986A_3STACK),
- SND_PCI_QUIRK(0x1043, 0x817f, "ASUS P5", AD1986A_3STACK),
- SND_PCI_QUIRK(0x1043, 0x818f, "ASUS P5", AD1986A_LAPTOP),
- SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS P5", AD1986A_3STACK),
- SND_PCI_QUIRK(0x1043, 0x81cb, "ASUS M2N", AD1986A_3STACK),
- SND_PCI_QUIRK(0x1043, 0x8234, "ASUS M2N", AD1986A_3STACK),
- SND_PCI_QUIRK(0x10de, 0xcb84, "ASUS A8N-VM", AD1986A_3STACK),
- SND_PCI_QUIRK(0x1179, 0xff40, "Toshiba Satellite L40-10Q", AD1986A_3STACK),
- SND_PCI_QUIRK(0x144d, 0xb03c, "Samsung R55", AD1986A_3STACK),
- SND_PCI_QUIRK(0x144d, 0xc01e, "FSC V2060", AD1986A_LAPTOP),
- SND_PCI_QUIRK(0x144d, 0xc024, "Samsung P50", AD1986A_SAMSUNG_P50),
- SND_PCI_QUIRK(0x144d, 0xc027, "Samsung Q1", AD1986A_ULTRA),
- SND_PCI_QUIRK_MASK(0x144d, 0xff00, 0xc000, "Samsung", AD1986A_SAMSUNG),
- SND_PCI_QUIRK(0x144d, 0xc504, "Samsung Q35", AD1986A_3STACK),
- SND_PCI_QUIRK(0x17aa, 0x1011, "Lenovo M55", AD1986A_LAPTOP),
- SND_PCI_QUIRK(0x17aa, 0x1017, "Lenovo A60", AD1986A_3STACK),
- SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo N100", AD1986A_LAPTOP_AUTOMUTE),
- SND_PCI_QUIRK(0x17c0, 0x2017, "Samsung M50", AD1986A_LAPTOP),
- {}
-};
-
-#ifdef CONFIG_PM
-static const struct hda_amp_list ad1986a_loopbacks[] = {
- { 0x13, HDA_OUTPUT, 0 }, /* Mic */
- { 0x14, HDA_OUTPUT, 0 }, /* Phone */
- { 0x15, HDA_OUTPUT, 0 }, /* CD */
- { 0x16, HDA_OUTPUT, 0 }, /* Aux */
- { 0x17, HDA_OUTPUT, 0 }, /* Line */
- { } /* end */
-};
-#endif
-
-static int is_jack_available(struct hda_codec *codec, hda_nid_t nid)
-{
- unsigned int conf = snd_hda_codec_get_pincfg(codec, nid);
- return get_defcfg_connect(conf) != AC_JACK_PORT_NONE;
-}
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
static int alloc_ad_spec(struct hda_codec *codec)
{
struct ad198x_spec *spec;
@@ -1203,6 +225,11 @@ static void ad_fixup_inv_jack_detect(struct hda_codec *codec,
enum {
AD1986A_FIXUP_INV_JACK_DETECT,
+ AD1986A_FIXUP_ULTRA,
+ AD1986A_FIXUP_SAMSUNG,
+ AD1986A_FIXUP_3STACK,
+ AD1986A_FIXUP_LAPTOP,
+ AD1986A_FIXUP_LAPTOP_IMIC,
};
static const struct hda_fixup ad1986a_fixups[] = {
@@ -1210,16 +237,86 @@ static const struct hda_fixup ad1986a_fixups[] = {
.type = HDA_FIXUP_FUNC,
.v.func = ad_fixup_inv_jack_detect,
},
+ [AD1986A_FIXUP_ULTRA] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x1b, 0x90170110 }, /* speaker */
+ { 0x1d, 0x90a7013e }, /* int mic */
+ {}
+ },
+ },
+ [AD1986A_FIXUP_SAMSUNG] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x1b, 0x90170110 }, /* speaker */
+ { 0x1d, 0x90a7013e }, /* int mic */
+ { 0x20, 0x411111f0 }, /* N/A */
+ { 0x24, 0x411111f0 }, /* N/A */
+ {}
+ },
+ },
+ [AD1986A_FIXUP_3STACK] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x1a, 0x02214021 }, /* headphone */
+ { 0x1b, 0x01014011 }, /* front */
+ { 0x1c, 0x01013012 }, /* surround */
+ { 0x1d, 0x01019015 }, /* clfe */
+ { 0x1e, 0x411111f0 }, /* N/A */
+ { 0x1f, 0x02a190f0 }, /* mic */
+ { 0x20, 0x018130f0 }, /* line-in */
+ {}
+ },
+ },
+ [AD1986A_FIXUP_LAPTOP] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x1a, 0x02214021 }, /* headphone */
+ { 0x1b, 0x90170110 }, /* speaker */
+ { 0x1c, 0x411111f0 }, /* N/A */
+ { 0x1d, 0x411111f0 }, /* N/A */
+ { 0x1e, 0x411111f0 }, /* N/A */
+ { 0x1f, 0x02a191f0 }, /* mic */
+ { 0x20, 0x411111f0 }, /* N/A */
+ {}
+ },
+ },
+ [AD1986A_FIXUP_LAPTOP_IMIC] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x1d, 0x90a7013e }, /* int mic */
+ {}
+ },
+ .chained_before = 1,
+ .chain_id = AD1986A_FIXUP_LAPTOP,
+ },
};
static const struct snd_pci_quirk ad1986a_fixup_tbl[] = {
+ SND_PCI_QUIRK(0x103c, 0x30af, "HP B2800", AD1986A_FIXUP_LAPTOP_IMIC),
+ SND_PCI_QUIRK_MASK(0x1043, 0xff00, 0x8100, "ASUS P5", AD1986A_FIXUP_3STACK),
+ SND_PCI_QUIRK_MASK(0x1043, 0xff00, 0x8200, "ASUS M2", AD1986A_FIXUP_3STACK),
+ SND_PCI_QUIRK(0x10de, 0xcb84, "ASUS A8N-VM", AD1986A_FIXUP_3STACK),
+ SND_PCI_QUIRK(0x144d, 0xc01e, "FSC V2060", AD1986A_FIXUP_LAPTOP),
+ SND_PCI_QUIRK_MASK(0x144d, 0xff00, 0xc000, "Samsung", AD1986A_FIXUP_SAMSUNG),
+ SND_PCI_QUIRK(0x144d, 0xc027, "Samsung Q1", AD1986A_FIXUP_ULTRA),
SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo N100", AD1986A_FIXUP_INV_JACK_DETECT),
+ SND_PCI_QUIRK(0x17aa, 0x1011, "Lenovo M55", AD1986A_FIXUP_3STACK),
+ SND_PCI_QUIRK(0x17aa, 0x1017, "Lenovo A60", AD1986A_FIXUP_3STACK),
+ {}
+};
+
+static const struct hda_model_fixup ad1986a_fixup_models[] = {
+ { .id = AD1986A_FIXUP_3STACK, .name = "3stack" },
+ { .id = AD1986A_FIXUP_LAPTOP, .name = "laptop" },
+ { .id = AD1986A_FIXUP_LAPTOP_IMIC, .name = "laptop-imic" },
+ { .id = AD1986A_FIXUP_LAPTOP_IMIC, .name = "laptop-eapd" }, /* alias */
{}
};
/*
*/
-static int ad1986a_parse_auto_config(struct hda_codec *codec)
+static int patch_ad1986a(struct hda_codec *codec)
{
int err;
struct ad198x_spec *spec;
@@ -1244,7 +341,8 @@ static int ad1986a_parse_auto_config(struct hda_codec *codec)
*/
spec->gen.multiout.no_share_stream = 1;
- snd_hda_pick_fixup(codec, NULL, ad1986a_fixup_tbl, ad1986a_fixups);
+ snd_hda_pick_fixup(codec, ad1986a_fixup_models, ad1986a_fixup_tbl,
+ ad1986a_fixups);
snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
err = ad198x_parse_auto_config(codec);
@@ -1258,330 +356,11 @@ static int ad1986a_parse_auto_config(struct hda_codec *codec)
return 0;
}
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static int patch_ad1986a(struct hda_codec *codec)
-{
- struct ad198x_spec *spec;
- int err, board_config;
-
- board_config = snd_hda_check_board_config(codec, AD1986A_MODELS,
- ad1986a_models,
- ad1986a_cfg_tbl);
- if (board_config < 0) {
- printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
- codec->chip_name);
- board_config = AD1986A_AUTO;
- }
-
- if (board_config == AD1986A_AUTO)
- return ad1986a_parse_auto_config(codec);
-
- err = alloc_ad_spec(codec);
- if (err < 0)
- return err;
- spec = codec->spec;
-
- err = snd_hda_attach_beep_device(codec, 0x19);
- if (err < 0) {
- ad198x_free(codec);
- return err;
- }
- set_beep_amp(spec, 0x18, 0, HDA_OUTPUT);
-
- spec->multiout.max_channels = 6;
- spec->multiout.num_dacs = ARRAY_SIZE(ad1986a_dac_nids);
- spec->multiout.dac_nids = ad1986a_dac_nids;
- spec->multiout.dig_out_nid = AD1986A_SPDIF_OUT;
- spec->num_adc_nids = 1;
- spec->adc_nids = ad1986a_adc_nids;
- spec->capsrc_nids = ad1986a_capsrc_nids;
- spec->input_mux = &ad1986a_capture_source;
- spec->num_mixers = 1;
- spec->mixers[0] = ad1986a_mixers;
- spec->num_init_verbs = 1;
- spec->init_verbs[0] = ad1986a_init_verbs;
-#ifdef CONFIG_PM
- spec->loopback.amplist = ad1986a_loopbacks;
-#endif
- spec->vmaster_nid = 0x1b;
- codec->inv_eapd = 1; /* AD1986A has the inverted EAPD implementation */
-
- codec->patch_ops = ad198x_patch_ops;
-
- /* override some parameters */
- switch (board_config) {
- case AD1986A_3STACK:
- spec->num_mixers = 2;
- spec->mixers[1] = ad1986a_3st_mixers;
- spec->num_init_verbs = 2;
- spec->init_verbs[1] = ad1986a_ch2_init;
- spec->channel_mode = ad1986a_modes;
- spec->num_channel_mode = ARRAY_SIZE(ad1986a_modes);
- spec->need_dac_fix = 1;
- spec->multiout.max_channels = 2;
- spec->multiout.num_dacs = 1;
- break;
- case AD1986A_LAPTOP:
- spec->mixers[0] = ad1986a_laptop_mixers;
- spec->multiout.max_channels = 2;
- spec->multiout.num_dacs = 1;
- spec->multiout.dac_nids = ad1986a_laptop_dac_nids;
- break;
- case AD1986A_LAPTOP_EAPD:
- spec->num_mixers = 3;
- spec->mixers[0] = ad1986a_laptop_master_mixers;
- spec->mixers[1] = ad1986a_laptop_eapd_mixers;
- spec->mixers[2] = ad1986a_laptop_intmic_mixers;
- spec->num_init_verbs = 2;
- spec->init_verbs[1] = ad1986a_eapd_init_verbs;
- spec->multiout.max_channels = 2;
- spec->multiout.num_dacs = 1;
- spec->multiout.dac_nids = ad1986a_laptop_dac_nids;
- if (!is_jack_available(codec, 0x25))
- spec->multiout.dig_out_nid = 0;
- spec->input_mux = &ad1986a_laptop_eapd_capture_source;
- break;
- case AD1986A_SAMSUNG:
- spec->num_mixers = 2;
- spec->mixers[0] = ad1986a_laptop_master_mixers;
- spec->mixers[1] = ad1986a_laptop_eapd_mixers;
- spec->num_init_verbs = 3;
- spec->init_verbs[1] = ad1986a_eapd_init_verbs;
- spec->init_verbs[2] = ad1986a_automic_verbs;
- spec->multiout.max_channels = 2;
- spec->multiout.num_dacs = 1;
- spec->multiout.dac_nids = ad1986a_laptop_dac_nids;
- if (!is_jack_available(codec, 0x25))
- spec->multiout.dig_out_nid = 0;
- spec->input_mux = &ad1986a_automic_capture_source;
- codec->patch_ops.unsol_event = ad1986a_automic_unsol_event;
- codec->patch_ops.init = ad1986a_automic_init;
- break;
- case AD1986A_SAMSUNG_P50:
- spec->num_mixers = 2;
- spec->mixers[0] = ad1986a_automute_master_mixers;
- spec->mixers[1] = ad1986a_laptop_eapd_mixers;
- spec->num_init_verbs = 4;
- spec->init_verbs[1] = ad1986a_eapd_init_verbs;
- spec->init_verbs[2] = ad1986a_automic_verbs;
- spec->init_verbs[3] = ad1986a_hp_init_verbs;
- spec->multiout.max_channels = 2;
- spec->multiout.num_dacs = 1;
- spec->multiout.dac_nids = ad1986a_laptop_dac_nids;
- if (!is_jack_available(codec, 0x25))
- spec->multiout.dig_out_nid = 0;
- spec->input_mux = &ad1986a_automic_capture_source;
- codec->patch_ops.unsol_event = ad1986a_samsung_p50_unsol_event;
- codec->patch_ops.init = ad1986a_samsung_p50_init;
- break;
- case AD1986A_LAPTOP_AUTOMUTE:
- spec->num_mixers = 3;
- spec->mixers[0] = ad1986a_automute_master_mixers;
- spec->mixers[1] = ad1986a_laptop_eapd_mixers;
- spec->mixers[2] = ad1986a_laptop_intmic_mixers;
- spec->num_init_verbs = 3;
- spec->init_verbs[1] = ad1986a_eapd_init_verbs;
- spec->init_verbs[2] = ad1986a_hp_init_verbs;
- spec->multiout.max_channels = 2;
- spec->multiout.num_dacs = 1;
- spec->multiout.dac_nids = ad1986a_laptop_dac_nids;
- if (!is_jack_available(codec, 0x25))
- spec->multiout.dig_out_nid = 0;
- spec->input_mux = &ad1986a_laptop_eapd_capture_source;
- codec->patch_ops.unsol_event = ad1986a_hp_unsol_event;
- codec->patch_ops.init = ad1986a_hp_init;
- /* Lenovo N100 seems to report the reversed bit
- * for HP jack-sensing
- */
- spec->inv_jack_detect = 1;
- break;
- case AD1986A_ULTRA:
- spec->mixers[0] = ad1986a_laptop_eapd_mixers;
- spec->num_init_verbs = 2;
- spec->init_verbs[1] = ad1986a_ultra_init;
- spec->multiout.max_channels = 2;
- spec->multiout.num_dacs = 1;
- spec->multiout.dac_nids = ad1986a_laptop_dac_nids;
- spec->multiout.dig_out_nid = 0;
- break;
- }
-
- /* AD1986A has a hardware problem that it can't share a stream
- * with multiple output pins. The copy of front to surrounds
- * causes noisy or silent outputs at a certain timing, e.g.
- * changing the volume.
- * So, let's disable the shared stream.
- */
- spec->multiout.no_share_stream = 1;
-
- codec->no_trigger_sense = 1;
- codec->no_sticky_stream = 1;
-
- return 0;
-}
-#else /* ENABLE_AD_STATIC_QUIRKS */
-#define patch_ad1986a ad1986a_parse_auto_config
-#endif /* ENABLE_AD_STATIC_QUIRKS */
/*
* AD1983 specific
*/
-#ifdef ENABLE_AD_STATIC_QUIRKS
-#define AD1983_SPDIF_OUT 0x02
-#define AD1983_DAC 0x03
-#define AD1983_ADC 0x04
-
-static const hda_nid_t ad1983_dac_nids[1] = { AD1983_DAC };
-static const hda_nid_t ad1983_adc_nids[1] = { AD1983_ADC };
-static const hda_nid_t ad1983_capsrc_nids[1] = { 0x15 };
-
-static const struct hda_input_mux ad1983_capture_source = {
- .num_items = 4,
- .items = {
- { "Mic", 0x0 },
- { "Line", 0x1 },
- { "Mix", 0x2 },
- { "Mix Mono", 0x3 },
- },
-};
-
-/*
- * SPDIF playback route
- */
-static int ad1983_spdif_route_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
-{
- static const char * const texts[] = { "PCM", "ADC" };
-
- uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
- uinfo->count = 1;
- uinfo->value.enumerated.items = 2;
- if (uinfo->value.enumerated.item > 1)
- uinfo->value.enumerated.item = 1;
- strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
- return 0;
-}
-
-static int ad1983_spdif_route_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct ad198x_spec *spec = codec->spec;
-
- ucontrol->value.enumerated.item[0] = spec->spdif_route;
- return 0;
-}
-
-static int ad1983_spdif_route_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct ad198x_spec *spec = codec->spec;
-
- if (ucontrol->value.enumerated.item[0] > 1)
- return -EINVAL;
- if (spec->spdif_route != ucontrol->value.enumerated.item[0]) {
- spec->spdif_route = ucontrol->value.enumerated.item[0];
- snd_hda_codec_write_cache(codec, spec->multiout.dig_out_nid, 0,
- AC_VERB_SET_CONNECT_SEL,
- spec->spdif_route);
- return 1;
- }
- return 0;
-}
-
-static const struct snd_kcontrol_new ad1983_mixers[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x05, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Front Playback Switch", 0x05, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x06, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x06, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x07, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x07, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("PCM Playback Volume", 0x11, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("PCM Playback Switch", 0x11, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x12, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x13, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x13, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Capture Source",
- .info = ad198x_mux_enum_info,
- .get = ad198x_mux_enum_get,
- .put = ad198x_mux_enum_put,
- },
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
- .info = ad1983_spdif_route_info,
- .get = ad1983_spdif_route_get,
- .put = ad1983_spdif_route_put,
- },
- { } /* end */
-};
-
-static const struct hda_verb ad1983_init_verbs[] = {
- /* Front, HP, Mono; mute as default */
- {0x05, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x06, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- /* Beep, PCM, Mic, Line-In: mute */
- {0x10, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x11, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x12, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- /* Front, HP selectors; from Mix */
- {0x05, AC_VERB_SET_CONNECT_SEL, 0x01},
- {0x06, AC_VERB_SET_CONNECT_SEL, 0x01},
- /* Mono selector; from Mix */
- {0x0b, AC_VERB_SET_CONNECT_SEL, 0x03},
- /* Mic selector; Mic */
- {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0},
- /* Line-in selector: Line-in */
- {0x0d, AC_VERB_SET_CONNECT_SEL, 0x0},
- /* Mic boost: 0dB */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
- /* Record selector: mic */
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x0},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- /* SPDIF route: PCM */
- {0x02, AC_VERB_SET_CONNECT_SEL, 0x0},
- /* Front Pin */
- {0x05, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
- /* HP Pin */
- {0x06, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
- /* Mono Pin */
- {0x07, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
- /* Mic Pin */
- {0x08, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
- /* Line Pin */
- {0x09, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
- { } /* end */
-};
-
-#ifdef CONFIG_PM
-static const struct hda_amp_list ad1983_loopbacks[] = {
- { 0x12, HDA_OUTPUT, 0 }, /* Mic */
- { 0x13, HDA_OUTPUT, 0 }, /* Line */
- { } /* end */
-};
-#endif
-
-/* models */
-enum {
- AD1983_AUTO,
- AD1983_BASIC,
- AD1983_MODELS
-};
-
-static const char * const ad1983_models[AD1983_MODELS] = {
- [AD1983_AUTO] = "auto",
- [AD1983_BASIC] = "basic",
-};
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
-
/*
* SPDIF mux control for AD1983 auto-parser
*/
@@ -1656,7 +435,7 @@ static int ad1983_add_spdif_mux_ctl(struct hda_codec *codec)
return 0;
}
-static int ad1983_parse_auto_config(struct hda_codec *codec)
+static int patch_ad1983(struct hda_codec *codec)
{
struct ad198x_spec *spec;
int err;
@@ -1681,432 +460,11 @@ static int ad1983_parse_auto_config(struct hda_codec *codec)
return err;
}
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static int patch_ad1983(struct hda_codec *codec)
-{
- struct ad198x_spec *spec;
- int board_config;
- int err;
-
- board_config = snd_hda_check_board_config(codec, AD1983_MODELS,
- ad1983_models, NULL);
- if (board_config < 0) {
- printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
- codec->chip_name);
- board_config = AD1983_AUTO;
- }
-
- if (board_config == AD1983_AUTO)
- return ad1983_parse_auto_config(codec);
-
- err = alloc_ad_spec(codec);
- if (err < 0)
- return err;
- spec = codec->spec;
-
- err = snd_hda_attach_beep_device(codec, 0x10);
- if (err < 0) {
- ad198x_free(codec);
- return err;
- }
- set_beep_amp(spec, 0x10, 0, HDA_OUTPUT);
-
- spec->multiout.max_channels = 2;
- spec->multiout.num_dacs = ARRAY_SIZE(ad1983_dac_nids);
- spec->multiout.dac_nids = ad1983_dac_nids;
- spec->multiout.dig_out_nid = AD1983_SPDIF_OUT;
- spec->num_adc_nids = 1;
- spec->adc_nids = ad1983_adc_nids;
- spec->capsrc_nids = ad1983_capsrc_nids;
- spec->input_mux = &ad1983_capture_source;
- spec->num_mixers = 1;
- spec->mixers[0] = ad1983_mixers;
- spec->num_init_verbs = 1;
- spec->init_verbs[0] = ad1983_init_verbs;
- spec->spdif_route = 0;
-#ifdef CONFIG_PM
- spec->loopback.amplist = ad1983_loopbacks;
-#endif
- spec->vmaster_nid = 0x05;
-
- codec->patch_ops = ad198x_patch_ops;
-
- codec->no_trigger_sense = 1;
- codec->no_sticky_stream = 1;
-
- return 0;
-}
-#else /* ENABLE_AD_STATIC_QUIRKS */
-#define patch_ad1983 ad1983_parse_auto_config
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
/*
* AD1981 HD specific
*/
-#ifdef ENABLE_AD_STATIC_QUIRKS
-#define AD1981_SPDIF_OUT 0x02
-#define AD1981_DAC 0x03
-#define AD1981_ADC 0x04
-
-static const hda_nid_t ad1981_dac_nids[1] = { AD1981_DAC };
-static const hda_nid_t ad1981_adc_nids[1] = { AD1981_ADC };
-static const hda_nid_t ad1981_capsrc_nids[1] = { 0x15 };
-
-/* 0x0c, 0x09, 0x0e, 0x0f, 0x19, 0x05, 0x18, 0x17 */
-static const struct hda_input_mux ad1981_capture_source = {
- .num_items = 7,
- .items = {
- { "Front Mic", 0x0 },
- { "Line", 0x1 },
- { "Mix", 0x2 },
- { "Mix Mono", 0x3 },
- { "CD", 0x4 },
- { "Mic", 0x6 },
- { "Aux", 0x7 },
- },
-};
-
-static const struct snd_kcontrol_new ad1981_mixers[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x05, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Front Playback Switch", 0x05, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x06, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x06, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x07, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x07, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("PCM Playback Volume", 0x11, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("PCM Playback Switch", 0x11, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x12, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x13, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x13, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Aux Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Aux Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x1c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x1c, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x1d, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x1d, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x08, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Capture Source",
- .info = ad198x_mux_enum_info,
- .get = ad198x_mux_enum_get,
- .put = ad198x_mux_enum_put,
- },
- /* identical with AD1983 */
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
- .info = ad1983_spdif_route_info,
- .get = ad1983_spdif_route_get,
- .put = ad1983_spdif_route_put,
- },
- { } /* end */
-};
-
-static const struct hda_verb ad1981_init_verbs[] = {
- /* Front, HP, Mono; mute as default */
- {0x05, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x06, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- /* Beep, PCM, Front Mic, Line, Rear Mic, Aux, CD-In: mute */
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x11, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x12, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- /* Front, HP selectors; from Mix */
- {0x05, AC_VERB_SET_CONNECT_SEL, 0x01},
- {0x06, AC_VERB_SET_CONNECT_SEL, 0x01},
- /* Mono selector; from Mix */
- {0x0b, AC_VERB_SET_CONNECT_SEL, 0x03},
- /* Mic Mixer; select Front Mic */
- {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
- {0x1f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- /* Mic boost: 0dB */
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- /* Record selector: Front mic */
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x0},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- /* SPDIF route: PCM */
- {0x02, AC_VERB_SET_CONNECT_SEL, 0x0},
- /* Front Pin */
- {0x05, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
- /* HP Pin */
- {0x06, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
- /* Mono Pin */
- {0x07, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
- /* Front & Rear Mic Pins */
- {0x08, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
- /* Line Pin */
- {0x09, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
- /* Digital Beep */
- {0x0d, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* Line-Out as Input: disabled */
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- { } /* end */
-};
-
-#ifdef CONFIG_PM
-static const struct hda_amp_list ad1981_loopbacks[] = {
- { 0x12, HDA_OUTPUT, 0 }, /* Front Mic */
- { 0x13, HDA_OUTPUT, 0 }, /* Line */
- { 0x1b, HDA_OUTPUT, 0 }, /* Aux */
- { 0x1c, HDA_OUTPUT, 0 }, /* Mic */
- { 0x1d, HDA_OUTPUT, 0 }, /* CD */
- { } /* end */
-};
-#endif
-
-/*
- * Patch for HP nx6320
- *
- * nx6320 uses EAPD in the reverse way - EAPD-on means the internal
- * speaker output enabled _and_ mute-LED off.
- */
-
-#define AD1981_HP_EVENT 0x37
-#define AD1981_MIC_EVENT 0x38
-
-static const struct hda_verb ad1981_hp_init_verbs[] = {
- {0x05, AC_VERB_SET_EAPD_BTLENABLE, 0x00 }, /* default off */
- /* pin sensing on HP and Mic jacks */
- {0x06, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1981_HP_EVENT},
- {0x08, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1981_MIC_EVENT},
- {}
-};
-
-/* turn on/off EAPD (+ mute HP) as a master switch */
-static int ad1981_hp_master_sw_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct ad198x_spec *spec = codec->spec;
-
- if (! ad198x_eapd_put(kcontrol, ucontrol))
- return 0;
- /* change speaker pin appropriately */
- snd_hda_set_pin_ctl(codec, 0x05, spec->cur_eapd ? PIN_OUT : 0);
- /* toggle HP mute appropriately */
- snd_hda_codec_amp_stereo(codec, 0x06, HDA_OUTPUT, 0,
- HDA_AMP_MUTE,
- spec->cur_eapd ? 0 : HDA_AMP_MUTE);
- return 1;
-}
-
-/* bind volumes of both NID 0x05 and 0x06 */
-static const struct hda_bind_ctls ad1981_hp_bind_master_vol = {
- .ops = &snd_hda_bind_vol,
- .values = {
- HDA_COMPOSE_AMP_VAL(0x05, 3, 0, HDA_OUTPUT),
- HDA_COMPOSE_AMP_VAL(0x06, 3, 0, HDA_OUTPUT),
- 0
- },
-};
-
-/* mute internal speaker if HP is plugged */
-static void ad1981_hp_automute(struct hda_codec *codec)
-{
- unsigned int present;
-
- present = snd_hda_jack_detect(codec, 0x06);
- snd_hda_codec_amp_stereo(codec, 0x05, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
-}
-
-/* toggle input of built-in and mic jack appropriately */
-static void ad1981_hp_automic(struct hda_codec *codec)
-{
- static const struct hda_verb mic_jack_on[] = {
- {0x1f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
- {}
- };
- static const struct hda_verb mic_jack_off[] = {
- {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- {0x1f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
- {}
- };
- unsigned int present;
-
- present = snd_hda_jack_detect(codec, 0x08);
- if (present)
- snd_hda_sequence_write(codec, mic_jack_on);
- else
- snd_hda_sequence_write(codec, mic_jack_off);
-}
-
-/* unsolicited event for HP jack sensing */
-static void ad1981_hp_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- res >>= 26;
- switch (res) {
- case AD1981_HP_EVENT:
- ad1981_hp_automute(codec);
- break;
- case AD1981_MIC_EVENT:
- ad1981_hp_automic(codec);
- break;
- }
-}
-
-static const struct hda_input_mux ad1981_hp_capture_source = {
- .num_items = 3,
- .items = {
- { "Mic", 0x0 },
- { "Dock Mic", 0x1 },
- { "Mix", 0x2 },
- },
-};
-
-static const struct snd_kcontrol_new ad1981_hp_mixers[] = {
- HDA_BIND_VOL("Master Playback Volume", &ad1981_hp_bind_master_vol),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .subdevice = HDA_SUBDEV_NID_FLAG | 0x05,
- .name = "Master Playback Switch",
- .info = ad198x_eapd_info,
- .get = ad198x_eapd_get,
- .put = ad1981_hp_master_sw_put,
- .private_value = 0x05,
- },
- HDA_CODEC_VOLUME("PCM Playback Volume", 0x11, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("PCM Playback Switch", 0x11, 0x0, HDA_OUTPUT),
-#if 0
- /* FIXME: analog mic/line loopback doesn't work with my tests...
- * (although recording is OK)
- */
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x12, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x1c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x1c, 0x0, HDA_OUTPUT),
- /* FIXME: does this laptop have analog CD connection? */
- HDA_CODEC_VOLUME("CD Playback Volume", 0x1d, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x1d, 0x0, HDA_OUTPUT),
-#endif
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x08, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x18, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Capture Source",
- .info = ad198x_mux_enum_info,
- .get = ad198x_mux_enum_get,
- .put = ad198x_mux_enum_put,
- },
- { } /* end */
-};
-
-/* initialize jack-sensing, too */
-static int ad1981_hp_init(struct hda_codec *codec)
-{
- ad198x_init(codec);
- ad1981_hp_automute(codec);
- ad1981_hp_automic(codec);
- return 0;
-}
-
-/* configuration for Toshiba Laptops */
-static const struct hda_verb ad1981_toshiba_init_verbs[] = {
- {0x05, AC_VERB_SET_EAPD_BTLENABLE, 0x01 }, /* default on */
- /* pin sensing on HP and Mic jacks */
- {0x06, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1981_HP_EVENT},
- {0x08, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1981_MIC_EVENT},
- {}
-};
-
-static const struct snd_kcontrol_new ad1981_toshiba_mixers[] = {
- HDA_CODEC_VOLUME("Amp Volume", 0x1a, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Amp Switch", 0x1a, 0x0, HDA_OUTPUT),
- { }
-};
-
-/* configuration for Lenovo Thinkpad T60 */
-static const struct snd_kcontrol_new ad1981_thinkpad_mixers[] = {
- HDA_CODEC_VOLUME("Master Playback Volume", 0x05, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Master Playback Switch", 0x05, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("PCM Playback Volume", 0x11, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("PCM Playback Switch", 0x11, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x12, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x1d, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x1d, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x08, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Capture Source",
- .info = ad198x_mux_enum_info,
- .get = ad198x_mux_enum_get,
- .put = ad198x_mux_enum_put,
- },
- /* identical with AD1983 */
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
- .info = ad1983_spdif_route_info,
- .get = ad1983_spdif_route_get,
- .put = ad1983_spdif_route_put,
- },
- { } /* end */
-};
-
-static const struct hda_input_mux ad1981_thinkpad_capture_source = {
- .num_items = 3,
- .items = {
- { "Mic", 0x0 },
- { "Mix", 0x2 },
- { "CD", 0x4 },
- },
-};
-
-/* models */
-enum {
- AD1981_AUTO,
- AD1981_BASIC,
- AD1981_HP,
- AD1981_THINKPAD,
- AD1981_TOSHIBA,
- AD1981_MODELS
-};
-
-static const char * const ad1981_models[AD1981_MODELS] = {
- [AD1981_AUTO] = "auto",
- [AD1981_HP] = "hp",
- [AD1981_THINKPAD] = "thinkpad",
- [AD1981_BASIC] = "basic",
- [AD1981_TOSHIBA] = "toshiba"
-};
-
-static const struct snd_pci_quirk ad1981_cfg_tbl[] = {
- SND_PCI_QUIRK(0x1014, 0x0597, "Lenovo Z60", AD1981_THINKPAD),
- SND_PCI_QUIRK(0x1014, 0x05b7, "Lenovo Z60m", AD1981_THINKPAD),
- /* All HP models */
- SND_PCI_QUIRK_VENDOR(0x103c, "HP nx", AD1981_HP),
- SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba U205", AD1981_TOSHIBA),
- /* Lenovo Thinkpad T60/X60/Z6xx */
- SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo Thinkpad", AD1981_THINKPAD),
- /* HP nx6320 (reversed SSID, H/W bug) */
- SND_PCI_QUIRK(0x30b0, 0x103c, "HP nx6320", AD1981_HP),
- {}
-};
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
-
/* follow EAPD via vmaster hook */
static void ad_vmaster_eapd_hook(void *private_data, int enabled)
{
@@ -2172,7 +530,7 @@ static const struct snd_pci_quirk ad1981_fixup_tbl[] = {
{}
};
-static int ad1981_parse_auto_config(struct hda_codec *codec)
+static int patch_ad1981(struct hda_codec *codec)
{
struct ad198x_spec *spec;
int err;
@@ -2205,110 +563,6 @@ static int ad1981_parse_auto_config(struct hda_codec *codec)
return err;
}
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static int patch_ad1981(struct hda_codec *codec)
-{
- struct ad198x_spec *spec;
- int err, board_config;
-
- board_config = snd_hda_check_board_config(codec, AD1981_MODELS,
- ad1981_models,
- ad1981_cfg_tbl);
- if (board_config < 0) {
- printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
- codec->chip_name);
- board_config = AD1981_AUTO;
- }
-
- if (board_config == AD1981_AUTO)
- return ad1981_parse_auto_config(codec);
-
- err = alloc_ad_spec(codec);
- if (err < 0)
- return -ENOMEM;
- spec = codec->spec;
-
- err = snd_hda_attach_beep_device(codec, 0x10);
- if (err < 0) {
- ad198x_free(codec);
- return err;
- }
- set_beep_amp(spec, 0x0d, 0, HDA_OUTPUT);
-
- spec->multiout.max_channels = 2;
- spec->multiout.num_dacs = ARRAY_SIZE(ad1981_dac_nids);
- spec->multiout.dac_nids = ad1981_dac_nids;
- spec->multiout.dig_out_nid = AD1981_SPDIF_OUT;
- spec->num_adc_nids = 1;
- spec->adc_nids = ad1981_adc_nids;
- spec->capsrc_nids = ad1981_capsrc_nids;
- spec->input_mux = &ad1981_capture_source;
- spec->num_mixers = 1;
- spec->mixers[0] = ad1981_mixers;
- spec->num_init_verbs = 1;
- spec->init_verbs[0] = ad1981_init_verbs;
- spec->spdif_route = 0;
-#ifdef CONFIG_PM
- spec->loopback.amplist = ad1981_loopbacks;
-#endif
- spec->vmaster_nid = 0x05;
-
- codec->patch_ops = ad198x_patch_ops;
-
- /* override some parameters */
- switch (board_config) {
- case AD1981_HP:
- spec->mixers[0] = ad1981_hp_mixers;
- spec->num_init_verbs = 2;
- spec->init_verbs[1] = ad1981_hp_init_verbs;
- if (!is_jack_available(codec, 0x0a))
- spec->multiout.dig_out_nid = 0;
- spec->input_mux = &ad1981_hp_capture_source;
-
- codec->patch_ops.init = ad1981_hp_init;
- codec->patch_ops.unsol_event = ad1981_hp_unsol_event;
- /* set the upper-limit for mixer amp to 0dB for avoiding the
- * possible damage by overloading
- */
- snd_hda_override_amp_caps(codec, 0x11, HDA_INPUT,
- (0x17 << AC_AMPCAP_OFFSET_SHIFT) |
- (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) |
- (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) |
- (1 << AC_AMPCAP_MUTE_SHIFT));
- break;
- case AD1981_THINKPAD:
- spec->mixers[0] = ad1981_thinkpad_mixers;
- spec->input_mux = &ad1981_thinkpad_capture_source;
- /* set the upper-limit for mixer amp to 0dB for avoiding the
- * possible damage by overloading
- */
- snd_hda_override_amp_caps(codec, 0x11, HDA_INPUT,
- (0x17 << AC_AMPCAP_OFFSET_SHIFT) |
- (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) |
- (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) |
- (1 << AC_AMPCAP_MUTE_SHIFT));
- break;
- case AD1981_TOSHIBA:
- spec->mixers[0] = ad1981_hp_mixers;
- spec->mixers[1] = ad1981_toshiba_mixers;
- spec->num_init_verbs = 2;
- spec->init_verbs[1] = ad1981_toshiba_init_verbs;
- spec->multiout.dig_out_nid = 0;
- spec->input_mux = &ad1981_hp_capture_source;
- codec->patch_ops.init = ad1981_hp_init;
- codec->patch_ops.unsol_event = ad1981_hp_unsol_event;
- break;
- }
-
- codec->no_trigger_sense = 1;
- codec->no_sticky_stream = 1;
-
- return 0;
-}
-#else /* ENABLE_AD_STATIC_QUIRKS */
-#define patch_ad1981 ad1981_parse_auto_config
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
/*
* AD1988
@@ -2395,90 +649,7 @@ static int patch_ad1981(struct hda_codec *codec)
* E/F quad mic array
*/
-
#ifdef ENABLE_AD_STATIC_QUIRKS
-/* models */
-enum {
- AD1988_AUTO,
- AD1988_6STACK,
- AD1988_6STACK_DIG,
- AD1988_3STACK,
- AD1988_3STACK_DIG,
- AD1988_LAPTOP,
- AD1988_LAPTOP_DIG,
- AD1988_MODEL_LAST,
-};
-
-/* reivision id to check workarounds */
-#define AD1988A_REV2 0x100200
-
-#define is_rev2(codec) \
- ((codec)->vendor_id == 0x11d41988 && \
- (codec)->revision_id == AD1988A_REV2)
-
-/*
- * mixers
- */
-
-static const hda_nid_t ad1988_6stack_dac_nids[4] = {
- 0x04, 0x06, 0x05, 0x0a
-};
-
-static const hda_nid_t ad1988_3stack_dac_nids[3] = {
- 0x04, 0x05, 0x0a
-};
-
-/* for AD1988A revision-2, DAC2-4 are swapped */
-static const hda_nid_t ad1988_6stack_dac_nids_rev2[4] = {
- 0x04, 0x05, 0x0a, 0x06
-};
-
-static const hda_nid_t ad1988_alt_dac_nid[1] = {
- 0x03
-};
-
-static const hda_nid_t ad1988_3stack_dac_nids_rev2[3] = {
- 0x04, 0x0a, 0x06
-};
-
-static const hda_nid_t ad1988_adc_nids[3] = {
- 0x08, 0x09, 0x0f
-};
-
-static const hda_nid_t ad1988_capsrc_nids[3] = {
- 0x0c, 0x0d, 0x0e
-};
-
-#define AD1988_SPDIF_OUT 0x02
-#define AD1988_SPDIF_OUT_HDMI 0x0b
-#define AD1988_SPDIF_IN 0x07
-
-static const hda_nid_t ad1989b_slave_dig_outs[] = {
- AD1988_SPDIF_OUT, AD1988_SPDIF_OUT_HDMI, 0
-};
-
-static const struct hda_input_mux ad1988_6stack_capture_source = {
- .num_items = 5,
- .items = {
- { "Front Mic", 0x1 }, /* port-B */
- { "Line", 0x2 }, /* port-C */
- { "Mic", 0x4 }, /* port-E */
- { "CD", 0x5 },
- { "Mix", 0x9 },
- },
-};
-
-static const struct hda_input_mux ad1988_laptop_capture_source = {
- .num_items = 3,
- .items = {
- { "Mic/Line", 0x1 }, /* port-B */
- { "CD", 0x5 },
- { "Mix", 0x9 },
- },
-};
-
-/*
- */
static int ad198x_ch_mode_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
@@ -2509,569 +680,6 @@ static int ad198x_ch_mode_put(struct snd_kcontrol *kcontrol,
spec->multiout.num_dacs = spec->multiout.max_channels / 2;
return err;
}
-
-/* 6-stack mode */
-static const struct snd_kcontrol_new ad1988_6stack_mixers1[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x06, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x05, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x05, 2, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Side Playback Volume", 0x0a, 0x0, HDA_OUTPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new ad1988_6stack_mixers1_rev2[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x05, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0a, 2, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Side Playback Volume", 0x06, 0x0, HDA_OUTPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new ad1988_6stack_mixers2[] = {
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x29, 2, HDA_INPUT),
- HDA_BIND_MUTE("Surround Playback Switch", 0x2a, 2, HDA_INPUT),
- HDA_BIND_MUTE_MONO("Center Playback Switch", 0x27, 1, 2, HDA_INPUT),
- HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x27, 2, 2, HDA_INPUT),
- HDA_BIND_MUTE("Side Playback Switch", 0x28, 2, HDA_INPUT),
- HDA_BIND_MUTE("Headphone Playback Switch", 0x22, 2, HDA_INPUT),
- HDA_BIND_MUTE("Mono Playback Switch", 0x1e, 2, HDA_INPUT),
-
- HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x6, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x6, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x1, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x4, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x4, HDA_INPUT),
-
- HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT),
-
- HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x39, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x3c, 0x0, HDA_OUTPUT),
- { } /* end */
-};
-
-/* 3-stack mode */
-static const struct snd_kcontrol_new ad1988_3stack_mixers1[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x0a, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x05, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x05, 2, 0x0, HDA_OUTPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new ad1988_3stack_mixers1_rev2[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x0a, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x06, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x06, 2, 0x0, HDA_OUTPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new ad1988_3stack_mixers2[] = {
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x29, 2, HDA_INPUT),
- HDA_BIND_MUTE("Surround Playback Switch", 0x2c, 2, HDA_INPUT),
- HDA_BIND_MUTE_MONO("Center Playback Switch", 0x26, 1, 2, HDA_INPUT),
- HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x26, 2, 2, HDA_INPUT),
- HDA_BIND_MUTE("Headphone Playback Switch", 0x22, 2, HDA_INPUT),
- HDA_BIND_MUTE("Mono Playback Switch", 0x1e, 2, HDA_INPUT),
-
- HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x6, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x6, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x1, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x4, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x4, HDA_INPUT),
-
- HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT),
-
- HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x39, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x3c, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Channel Mode",
- .info = ad198x_ch_mode_info,
- .get = ad198x_ch_mode_get,
- .put = ad198x_ch_mode_put,
- },
-
- { } /* end */
-};
-
-/* laptop mode */
-static const struct snd_kcontrol_new ad1988_laptop_mixers[] = {
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("PCM Playback Switch", 0x29, 0x0, HDA_INPUT),
- HDA_BIND_MUTE("Mono Playback Switch", 0x1e, 2, HDA_INPUT),
-
- HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x6, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x6, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x1, HDA_INPUT),
-
- HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT),
-
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x39, 0x0, HDA_OUTPUT),
-
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "External Amplifier",
- .subdevice = HDA_SUBDEV_NID_FLAG | 0x12,
- .info = ad198x_eapd_info,
- .get = ad198x_eapd_get,
- .put = ad198x_eapd_put,
- .private_value = 0x12, /* port-D */
- },
-
- { } /* end */
-};
-
-/* capture */
-static const struct snd_kcontrol_new ad1988_capture_mixers[] = {
- HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_IDX("Capture Volume", 2, 0x0e, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 2, 0x0e, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- /* The multiple "Capture Source" controls confuse alsamixer
- * So call somewhat different..
- */
- /* .name = "Capture Source", */
- .name = "Input Source",
- .count = 3,
- .info = ad198x_mux_enum_info,
- .get = ad198x_mux_enum_get,
- .put = ad198x_mux_enum_put,
- },
- { } /* end */
-};
-
-static int ad1988_spdif_playback_source_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- static const char * const texts[] = {
- "PCM", "ADC1", "ADC2", "ADC3"
- };
- uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
- uinfo->count = 1;
- uinfo->value.enumerated.items = 4;
- if (uinfo->value.enumerated.item >= 4)
- uinfo->value.enumerated.item = 3;
- strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
- return 0;
-}
-
-static int ad1988_spdif_playback_source_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- unsigned int sel;
-
- sel = snd_hda_codec_read(codec, 0x1d, 0, AC_VERB_GET_AMP_GAIN_MUTE,
- AC_AMP_GET_INPUT);
- if (!(sel & 0x80))
- ucontrol->value.enumerated.item[0] = 0;
- else {
- sel = snd_hda_codec_read(codec, 0x0b, 0,
- AC_VERB_GET_CONNECT_SEL, 0);
- if (sel < 3)
- sel++;
- else
- sel = 0;
- ucontrol->value.enumerated.item[0] = sel;
- }
- return 0;
-}
-
-static int ad1988_spdif_playback_source_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- unsigned int val, sel;
- int change;
-
- val = ucontrol->value.enumerated.item[0];
- if (val > 3)
- return -EINVAL;
- if (!val) {
- sel = snd_hda_codec_read(codec, 0x1d, 0,
- AC_VERB_GET_AMP_GAIN_MUTE,
- AC_AMP_GET_INPUT);
- change = sel & 0x80;
- if (change) {
- snd_hda_codec_write_cache(codec, 0x1d, 0,
- AC_VERB_SET_AMP_GAIN_MUTE,
- AMP_IN_UNMUTE(0));
- snd_hda_codec_write_cache(codec, 0x1d, 0,
- AC_VERB_SET_AMP_GAIN_MUTE,
- AMP_IN_MUTE(1));
- }
- } else {
- sel = snd_hda_codec_read(codec, 0x1d, 0,
- AC_VERB_GET_AMP_GAIN_MUTE,
- AC_AMP_GET_INPUT | 0x01);
- change = sel & 0x80;
- if (change) {
- snd_hda_codec_write_cache(codec, 0x1d, 0,
- AC_VERB_SET_AMP_GAIN_MUTE,
- AMP_IN_MUTE(0));
- snd_hda_codec_write_cache(codec, 0x1d, 0,
- AC_VERB_SET_AMP_GAIN_MUTE,
- AMP_IN_UNMUTE(1));
- }
- sel = snd_hda_codec_read(codec, 0x0b, 0,
- AC_VERB_GET_CONNECT_SEL, 0) + 1;
- change |= sel != val;
- if (change)
- snd_hda_codec_write_cache(codec, 0x0b, 0,
- AC_VERB_SET_CONNECT_SEL,
- val - 1);
- }
- return change;
-}
-
-static const struct snd_kcontrol_new ad1988_spdif_out_mixers[] = {
- HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "IEC958 Playback Source",
- .subdevice = HDA_SUBDEV_NID_FLAG | 0x1b,
- .info = ad1988_spdif_playback_source_info,
- .get = ad1988_spdif_playback_source_get,
- .put = ad1988_spdif_playback_source_put,
- },
- { } /* end */
-};
-
-static const struct snd_kcontrol_new ad1988_spdif_in_mixers[] = {
- HDA_CODEC_VOLUME("IEC958 Capture Volume", 0x1c, 0x0, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new ad1989_spdif_out_mixers[] = {
- HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("HDMI Playback Volume", 0x1d, 0x0, HDA_OUTPUT),
- { } /* end */
-};
-
-/*
- * initialization verbs
- */
-
-/*
- * for 6-stack (+dig)
- */
-static const struct hda_verb ad1988_6stack_init_verbs[] = {
- /* Front, Surround, CLFE, side DAC; unmute as default */
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Port-A front headphon path */
- {0x37, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC0:03h */
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- /* Port-D line-out path */
- {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- /* Port-F surround path */
- {0x2a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x2a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- /* Port-G CLFE path */
- {0x27, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x27, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- /* Port-H side path */
- {0x28, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x28, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x25, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- /* Mono out path */
- {0x36, AC_VERB_SET_CONNECT_SEL, 0x1}, /* DAC1:04h */
- {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb01f}, /* unmute, 0dB */
- /* Port-B front mic-in path */
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x39, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- /* Port-C line-in path */
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x3a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x33, AC_VERB_SET_CONNECT_SEL, 0x0},
- /* Port-E mic-in path */
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x3c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x34, AC_VERB_SET_CONNECT_SEL, 0x0},
- /* Analog CD Input */
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- /* Analog Mix output amp */
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */
-
- { }
-};
-
-static const struct hda_verb ad1988_6stack_fp_init_verbs[] = {
- /* Headphone; unmute as default */
- {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Port-A front headphon path */
- {0x37, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC0:03h */
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-
- { }
-};
-
-static const struct hda_verb ad1988_capture_init_verbs[] = {
- /* mute analog mix */
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
- /* select ADCs - front-mic */
- {0x0c, AC_VERB_SET_CONNECT_SEL, 0x1},
- {0x0d, AC_VERB_SET_CONNECT_SEL, 0x1},
- {0x0e, AC_VERB_SET_CONNECT_SEL, 0x1},
-
- { }
-};
-
-static const struct hda_verb ad1988_spdif_init_verbs[] = {
- /* SPDIF out sel */
- {0x02, AC_VERB_SET_CONNECT_SEL, 0x0}, /* PCM */
- {0x0b, AC_VERB_SET_CONNECT_SEL, 0x0}, /* ADC1 */
- {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* SPDIF out pin */
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
-
- { }
-};
-
-static const struct hda_verb ad1988_spdif_in_init_verbs[] = {
- /* unmute SPDIF input pin */
- {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- { }
-};
-
-/* AD1989 has no ADC -> SPDIF route */
-static const struct hda_verb ad1989_spdif_init_verbs[] = {
- /* SPDIF-1 out pin */
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
- /* SPDIF-2/HDMI out pin */
- {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
- { }
-};
-
-/*
- * verbs for 3stack (+dig)
- */
-static const struct hda_verb ad1988_3stack_ch2_init[] = {
- /* set port-C to line-in */
- { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
- { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
- /* set port-E to mic-in */
- { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
- { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
- { } /* end */
-};
-
-static const struct hda_verb ad1988_3stack_ch6_init[] = {
- /* set port-C to surround out */
- { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- /* set port-E to CLFE out */
- { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { } /* end */
-};
-
-static const struct hda_channel_mode ad1988_3stack_modes[2] = {
- { 2, ad1988_3stack_ch2_init },
- { 6, ad1988_3stack_ch6_init },
-};
-
-static const struct hda_verb ad1988_3stack_init_verbs[] = {
- /* Front, Surround, CLFE, side DAC; unmute as default */
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Port-A front headphon path */
- {0x37, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC0:03h */
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- /* Port-D line-out path */
- {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- /* Mono out path */
- {0x36, AC_VERB_SET_CONNECT_SEL, 0x1}, /* DAC1:04h */
- {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb01f}, /* unmute, 0dB */
- /* Port-B front mic-in path */
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x39, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- /* Port-C line-in/surround path - 6ch mode as default */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x3a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x31, AC_VERB_SET_CONNECT_SEL, 0x0}, /* output sel: DAC 0x05 */
- {0x33, AC_VERB_SET_CONNECT_SEL, 0x0},
- /* Port-E mic-in/CLFE path - 6ch mode as default */
- {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x3c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x32, AC_VERB_SET_CONNECT_SEL, 0x1}, /* output sel: DAC 0x0a */
- {0x34, AC_VERB_SET_CONNECT_SEL, 0x0},
- /* mute analog mix */
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
- /* select ADCs - front-mic */
- {0x0c, AC_VERB_SET_CONNECT_SEL, 0x1},
- {0x0d, AC_VERB_SET_CONNECT_SEL, 0x1},
- {0x0e, AC_VERB_SET_CONNECT_SEL, 0x1},
- /* Analog Mix output amp */
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */
- { }
-};
-
-/*
- * verbs for laptop mode (+dig)
- */
-static const struct hda_verb ad1988_laptop_hp_on[] = {
- /* unmute port-A and mute port-D */
- { 0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { 0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
- { } /* end */
-};
-static const struct hda_verb ad1988_laptop_hp_off[] = {
- /* mute port-A and unmute port-D */
- { 0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
- { 0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { } /* end */
-};
-
-#define AD1988_HP_EVENT 0x01
-
-static const struct hda_verb ad1988_laptop_init_verbs[] = {
- /* Front, Surround, CLFE, side DAC; unmute as default */
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Port-A front headphon path */
- {0x37, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC0:03h */
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- /* unsolicited event for pin-sense */
- {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1988_HP_EVENT },
- /* Port-D line-out path + EAPD */
- {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x12, AC_VERB_SET_EAPD_BTLENABLE, 0x00}, /* EAPD-off */
- /* Mono out path */
- {0x36, AC_VERB_SET_CONNECT_SEL, 0x1}, /* DAC1:04h */
- {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb01f}, /* unmute, 0dB */
- /* Port-B mic-in path */
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x39, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- /* Port-C docking station - try to output */
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x3a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x33, AC_VERB_SET_CONNECT_SEL, 0x0},
- /* mute analog mix */
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
- /* select ADCs - mic */
- {0x0c, AC_VERB_SET_CONNECT_SEL, 0x1},
- {0x0d, AC_VERB_SET_CONNECT_SEL, 0x1},
- {0x0e, AC_VERB_SET_CONNECT_SEL, 0x1},
- /* Analog Mix output amp */
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */
- { }
-};
-
-static void ad1988_laptop_unsol_event(struct hda_codec *codec, unsigned int res)
-{
- if ((res >> 26) != AD1988_HP_EVENT)
- return;
- if (snd_hda_jack_detect(codec, 0x11))
- snd_hda_sequence_write(codec, ad1988_laptop_hp_on);
- else
- snd_hda_sequence_write(codec, ad1988_laptop_hp_off);
-}
-
-#ifdef CONFIG_PM
-static const struct hda_amp_list ad1988_loopbacks[] = {
- { 0x20, HDA_INPUT, 0 }, /* Front Mic */
- { 0x20, HDA_INPUT, 1 }, /* Line */
- { 0x20, HDA_INPUT, 4 }, /* Mic */
- { 0x20, HDA_INPUT, 6 }, /* CD */
- { } /* end */
-};
-#endif
#endif /* ENABLE_AD_STATIC_QUIRKS */
static int ad1988_auto_smux_enum_info(struct snd_kcontrol *kcontrol,
@@ -3220,7 +828,34 @@ static int ad1988_add_spdif_mux_ctl(struct hda_codec *codec)
/*
*/
-static int ad1988_parse_auto_config(struct hda_codec *codec)
+enum {
+ AD1988_FIXUP_6STACK_DIG,
+};
+
+static const struct hda_fixup ad1988_fixups[] = {
+ [AD1988_FIXUP_6STACK_DIG] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x11, 0x02214130 }, /* front-hp */
+ { 0x12, 0x01014010 }, /* line-out */
+ { 0x14, 0x02a19122 }, /* front-mic */
+ { 0x15, 0x01813021 }, /* line-in */
+ { 0x16, 0x01011012 }, /* line-out */
+ { 0x17, 0x01a19020 }, /* mic */
+ { 0x1b, 0x0145f1f0 }, /* SPDIF */
+ { 0x24, 0x01016011 }, /* line-out */
+ { 0x25, 0x01012013 }, /* line-out */
+ { }
+ }
+ },
+};
+
+static const struct hda_model_fixup ad1988_fixup_models[] = {
+ { .id = AD1988_FIXUP_6STACK_DIG, .name = "6stack-dig" },
+ {}
+};
+
+static int patch_ad1988(struct hda_codec *codec)
{
struct ad198x_spec *spec;
int err;
@@ -3234,12 +869,19 @@ static int ad1988_parse_auto_config(struct hda_codec *codec)
spec->gen.mixer_merge_nid = 0x21;
spec->gen.beep_nid = 0x10;
set_beep_amp(spec, 0x10, 0, HDA_OUTPUT);
+
+ snd_hda_pick_fixup(codec, ad1988_fixup_models, NULL, ad1988_fixups);
+ snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
+
err = ad198x_parse_auto_config(codec);
if (err < 0)
goto error;
err = ad1988_add_spdif_mux_ctl(codec);
if (err < 0)
goto error;
+
+ snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE);
+
return 0;
error:
@@ -3247,169 +889,6 @@ static int ad1988_parse_auto_config(struct hda_codec *codec)
return err;
}
-/*
- */
-
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static const char * const ad1988_models[AD1988_MODEL_LAST] = {
- [AD1988_6STACK] = "6stack",
- [AD1988_6STACK_DIG] = "6stack-dig",
- [AD1988_3STACK] = "3stack",
- [AD1988_3STACK_DIG] = "3stack-dig",
- [AD1988_LAPTOP] = "laptop",
- [AD1988_LAPTOP_DIG] = "laptop-dig",
- [AD1988_AUTO] = "auto",
-};
-
-static const struct snd_pci_quirk ad1988_cfg_tbl[] = {
- SND_PCI_QUIRK(0x1043, 0x81ec, "Asus P5B-DLX", AD1988_6STACK_DIG),
- SND_PCI_QUIRK(0x1043, 0x81f6, "Asus M2N-SLI", AD1988_6STACK_DIG),
- SND_PCI_QUIRK(0x1043, 0x8277, "Asus P5K-E/WIFI-AP", AD1988_6STACK_DIG),
- SND_PCI_QUIRK(0x1043, 0x82c0, "Asus M3N-HT Deluxe", AD1988_6STACK_DIG),
- SND_PCI_QUIRK(0x1043, 0x8311, "Asus P5Q-Premium/Pro", AD1988_6STACK_DIG),
- {}
-};
-
-static int patch_ad1988(struct hda_codec *codec)
-{
- struct ad198x_spec *spec;
- int err, board_config;
-
- board_config = snd_hda_check_board_config(codec, AD1988_MODEL_LAST,
- ad1988_models, ad1988_cfg_tbl);
- if (board_config < 0) {
- printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
- codec->chip_name);
- board_config = AD1988_AUTO;
- }
-
- if (board_config == AD1988_AUTO)
- return ad1988_parse_auto_config(codec);
-
- err = alloc_ad_spec(codec);
- if (err < 0)
- return err;
- spec = codec->spec;
-
- if (is_rev2(codec))
- snd_printk(KERN_INFO "patch_analog: AD1988A rev.2 is detected, enable workarounds\n");
-
- err = snd_hda_attach_beep_device(codec, 0x10);
- if (err < 0) {
- ad198x_free(codec);
- return err;
- }
- set_beep_amp(spec, 0x10, 0, HDA_OUTPUT);
-
- if (!spec->multiout.hp_nid)
- spec->multiout.hp_nid = ad1988_alt_dac_nid[0];
- switch (board_config) {
- case AD1988_6STACK:
- case AD1988_6STACK_DIG:
- spec->multiout.max_channels = 8;
- spec->multiout.num_dacs = 4;
- if (is_rev2(codec))
- spec->multiout.dac_nids = ad1988_6stack_dac_nids_rev2;
- else
- spec->multiout.dac_nids = ad1988_6stack_dac_nids;
- spec->input_mux = &ad1988_6stack_capture_source;
- spec->num_mixers = 2;
- if (is_rev2(codec))
- spec->mixers[0] = ad1988_6stack_mixers1_rev2;
- else
- spec->mixers[0] = ad1988_6stack_mixers1;
- spec->mixers[1] = ad1988_6stack_mixers2;
- spec->num_init_verbs = 1;
- spec->init_verbs[0] = ad1988_6stack_init_verbs;
- if (board_config == AD1988_6STACK_DIG) {
- spec->multiout.dig_out_nid = AD1988_SPDIF_OUT;
- spec->dig_in_nid = AD1988_SPDIF_IN;
- }
- break;
- case AD1988_3STACK:
- case AD1988_3STACK_DIG:
- spec->multiout.max_channels = 6;
- spec->multiout.num_dacs = 3;
- if (is_rev2(codec))
- spec->multiout.dac_nids = ad1988_3stack_dac_nids_rev2;
- else
- spec->multiout.dac_nids = ad1988_3stack_dac_nids;
- spec->input_mux = &ad1988_6stack_capture_source;
- spec->channel_mode = ad1988_3stack_modes;
- spec->num_channel_mode = ARRAY_SIZE(ad1988_3stack_modes);
- spec->num_mixers = 2;
- if (is_rev2(codec))
- spec->mixers[0] = ad1988_3stack_mixers1_rev2;
- else
- spec->mixers[0] = ad1988_3stack_mixers1;
- spec->mixers[1] = ad1988_3stack_mixers2;
- spec->num_init_verbs = 1;
- spec->init_verbs[0] = ad1988_3stack_init_verbs;
- if (board_config == AD1988_3STACK_DIG)
- spec->multiout.dig_out_nid = AD1988_SPDIF_OUT;
- break;
- case AD1988_LAPTOP:
- case AD1988_LAPTOP_DIG:
- spec->multiout.max_channels = 2;
- spec->multiout.num_dacs = 1;
- spec->multiout.dac_nids = ad1988_3stack_dac_nids;
- spec->input_mux = &ad1988_laptop_capture_source;
- spec->num_mixers = 1;
- spec->mixers[0] = ad1988_laptop_mixers;
- codec->inv_eapd = 1; /* inverted EAPD */
- spec->num_init_verbs = 1;
- spec->init_verbs[0] = ad1988_laptop_init_verbs;
- if (board_config == AD1988_LAPTOP_DIG)
- spec->multiout.dig_out_nid = AD1988_SPDIF_OUT;
- break;
- }
-
- spec->num_adc_nids = ARRAY_SIZE(ad1988_adc_nids);
- spec->adc_nids = ad1988_adc_nids;
- spec->capsrc_nids = ad1988_capsrc_nids;
- spec->mixers[spec->num_mixers++] = ad1988_capture_mixers;
- spec->init_verbs[spec->num_init_verbs++] = ad1988_capture_init_verbs;
- if (spec->multiout.dig_out_nid) {
- if (codec->vendor_id >= 0x11d4989a) {
- spec->mixers[spec->num_mixers++] =
- ad1989_spdif_out_mixers;
- spec->init_verbs[spec->num_init_verbs++] =
- ad1989_spdif_init_verbs;
- codec->slave_dig_outs = ad1989b_slave_dig_outs;
- } else {
- spec->mixers[spec->num_mixers++] =
- ad1988_spdif_out_mixers;
- spec->init_verbs[spec->num_init_verbs++] =
- ad1988_spdif_init_verbs;
- }
- }
- if (spec->dig_in_nid && codec->vendor_id < 0x11d4989a) {
- spec->mixers[spec->num_mixers++] = ad1988_spdif_in_mixers;
- spec->init_verbs[spec->num_init_verbs++] =
- ad1988_spdif_in_init_verbs;
- }
-
- codec->patch_ops = ad198x_patch_ops;
- switch (board_config) {
- case AD1988_LAPTOP:
- case AD1988_LAPTOP_DIG:
- codec->patch_ops.unsol_event = ad1988_laptop_unsol_event;
- break;
- }
-#ifdef CONFIG_PM
- spec->loopback.amplist = ad1988_loopbacks;
-#endif
- spec->vmaster_nid = 0x04;
-
- codec->no_trigger_sense = 1;
- codec->no_sticky_stream = 1;
-
- return 0;
-}
-#else /* ENABLE_AD_STATIC_QUIRKS */
-#define patch_ad1988 ad1988_parse_auto_config
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
/*
* AD1884 / AD1984
@@ -3423,167 +902,19 @@ static int patch_ad1988(struct hda_codec *codec)
*
* AD1984 = AD1884 + two digital mic-ins
*
- * FIXME:
- * For simplicity, we share the single DAC for both HP and line-outs
- * right now. The inidividual playbacks could be easily implemented,
- * but no build-up framework is given, so far.
- */
-
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static const hda_nid_t ad1884_dac_nids[1] = {
- 0x04,
-};
-
-static const hda_nid_t ad1884_adc_nids[2] = {
- 0x08, 0x09,
-};
-
-static const hda_nid_t ad1884_capsrc_nids[2] = {
- 0x0c, 0x0d,
-};
-
-#define AD1884_SPDIF_OUT 0x02
-
-static const struct hda_input_mux ad1884_capture_source = {
- .num_items = 4,
- .items = {
- { "Front Mic", 0x0 },
- { "Mic", 0x1 },
- { "CD", 0x2 },
- { "Mix", 0x3 },
- },
-};
-
-static const struct snd_kcontrol_new ad1884_base_mixers[] = {
- HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT),
- /* HDA_CODEC_VOLUME_IDX("PCM Playback Volume", 1, 0x03, 0x0, HDA_OUTPUT), */
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Front Playback Switch", 0x12, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x13, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x15, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x14, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- /* The multiple "Capture Source" controls confuse alsamixer
- * So call somewhat different..
- */
- /* .name = "Capture Source", */
- .name = "Input Source",
- .count = 2,
- .info = ad198x_mux_enum_info,
- .get = ad198x_mux_enum_get,
- .put = ad198x_mux_enum_put,
- },
- /* SPDIF controls */
- HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
- /* identical with ad1983 */
- .info = ad1983_spdif_route_info,
- .get = ad1983_spdif_route_get,
- .put = ad1983_spdif_route_put,
- },
- { } /* end */
-};
-
-static const struct snd_kcontrol_new ad1984_dmic_mixers[] = {
- HDA_CODEC_VOLUME("Digital Mic Capture Volume", 0x05, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Digital Mic Capture Switch", 0x05, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME_IDX("Digital Mic Capture Volume", 1, 0x06, 0x0,
- HDA_INPUT),
- HDA_CODEC_MUTE_IDX("Digital Mic Capture Switch", 1, 0x06, 0x0,
- HDA_INPUT),
- { } /* end */
-};
-
-/*
- * initialization verbs
+ * AD1883 / AD1884A / AD1984A / AD1984B
+ *
+ * port-B (0x14) - front mic-in
+ * port-E (0x1c) - rear mic-in
+ * port-F (0x16) - CD / ext out
+ * port-C (0x15) - rear line-in
+ * port-D (0x12) - rear line-out
+ * port-A (0x11) - front hp-out
+ *
+ * AD1984A = AD1884A + digital-mic
+ * AD1883 = equivalent with AD1984A
+ * AD1984B = AD1984A + extra SPDIF-out
*/
-static const struct hda_verb ad1884_init_verbs[] = {
- /* DACs; mute as default */
- {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- /* Port-A (HP) mixer */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- /* Port-A pin */
- {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* HP selector - select DAC2 */
- {0x22, AC_VERB_SET_CONNECT_SEL, 0x1},
- /* Port-D (Line-out) mixer */
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- /* Port-D pin */
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Mono-out mixer */
- {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- /* Mono-out pin */
- {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Mono selector */
- {0x0e, AC_VERB_SET_CONNECT_SEL, 0x1},
- /* Port-B (front mic) pin */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- /* Port-C (rear mic) pin */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- /* Analog mixer; mute as default */
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- /* Analog Mix output amp */
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */
- /* SPDIF output selector */
- {0x02, AC_VERB_SET_CONNECT_SEL, 0x0}, /* PCM */
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
- { } /* end */
-};
-
-#ifdef CONFIG_PM
-static const struct hda_amp_list ad1884_loopbacks[] = {
- { 0x20, HDA_INPUT, 0 }, /* Front Mic */
- { 0x20, HDA_INPUT, 1 }, /* Mic */
- { 0x20, HDA_INPUT, 2 }, /* CD */
- { 0x20, HDA_INPUT, 4 }, /* Docking */
- { } /* end */
-};
-#endif
-
-static const char * const ad1884_slave_vols[] = {
- "PCM", "Mic", "Mono", "Front Mic", "Mic", "CD",
- "Internal Mic", "Dock Mic", /* "Beep", */ "IEC958",
- NULL
-};
-
-enum {
- AD1884_AUTO,
- AD1884_BASIC,
- AD1884_MODELS
-};
-
-static const char * const ad1884_models[AD1884_MODELS] = {
- [AD1884_AUTO] = "auto",
- [AD1884_BASIC] = "basic",
-};
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
/* set the upper-limit for mixer amp to 0dB for avoiding the possible
* damage by overloading
@@ -3599,14 +930,34 @@ static void ad1884_fixup_amp_override(struct hda_codec *codec,
(1 << AC_AMPCAP_MUTE_SHIFT));
}
+/* toggle GPIO1 according to the mute state */
+static void ad1884_vmaster_hp_gpio_hook(void *private_data, int enabled)
+{
+ struct hda_codec *codec = private_data;
+ struct ad198x_spec *spec = codec->spec;
+
+ if (spec->eapd_nid)
+ ad_vmaster_eapd_hook(private_data, enabled);
+ snd_hda_codec_update_cache(codec, 0x01, 0,
+ AC_VERB_SET_GPIO_DATA,
+ enabled ? 0x00 : 0x02);
+}
+
static void ad1884_fixup_hp_eapd(struct hda_codec *codec,
const struct hda_fixup *fix, int action)
{
struct ad198x_spec *spec = codec->spec;
+ static const struct hda_verb gpio_init_verbs[] = {
+ {0x01, AC_VERB_SET_GPIO_MASK, 0x02},
+ {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02},
+ {0x01, AC_VERB_SET_GPIO_DATA, 0x02},
+ {},
+ };
switch (action) {
case HDA_FIXUP_ACT_PRE_PROBE:
- spec->gen.vmaster_mute.hook = ad_vmaster_eapd_hook;
+ spec->gen.vmaster_mute.hook = ad1884_vmaster_hp_gpio_hook;
+ snd_hda_sequence_write_cache(codec, gpio_init_verbs);
break;
case HDA_FIXUP_ACT_PROBE:
if (spec->gen.autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT)
@@ -3617,9 +968,18 @@ static void ad1884_fixup_hp_eapd(struct hda_codec *codec,
}
}
+/* set magic COEFs for dmic */
+static const struct hda_verb ad1884_dmic_init_verbs[] = {
+ {0x01, AC_VERB_SET_COEF_INDEX, 0x13f7},
+ {0x01, AC_VERB_SET_PROC_COEF, 0x08},
+ {}
+};
+
enum {
AD1884_FIXUP_AMP_OVERRIDE,
AD1884_FIXUP_HP_EAPD,
+ AD1884_FIXUP_DMIC_COEF,
+ AD1884_FIXUP_HP_TOUCHSMART,
};
static const struct hda_fixup ad1884_fixups[] = {
@@ -3633,15 +993,27 @@ static const struct hda_fixup ad1884_fixups[] = {
.chained = true,
.chain_id = AD1884_FIXUP_AMP_OVERRIDE,
},
+ [AD1884_FIXUP_DMIC_COEF] = {
+ .type = HDA_FIXUP_VERBS,
+ .v.verbs = ad1884_dmic_init_verbs,
+ },
+ [AD1884_FIXUP_HP_TOUCHSMART] = {
+ .type = HDA_FIXUP_VERBS,
+ .v.verbs = ad1884_dmic_init_verbs,
+ .chained = true,
+ .chain_id = AD1884_FIXUP_HP_EAPD,
+ },
};
static const struct snd_pci_quirk ad1884_fixup_tbl[] = {
+ SND_PCI_QUIRK(0x103c, 0x2a82, "HP Touchsmart", AD1884_FIXUP_HP_TOUCHSMART),
SND_PCI_QUIRK_VENDOR(0x103c, "HP", AD1884_FIXUP_HP_EAPD),
+ SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo Thinkpad", AD1884_FIXUP_DMIC_COEF),
{}
};
-static int ad1884_parse_auto_config(struct hda_codec *codec)
+static int patch_ad1884(struct hda_codec *codec)
{
struct ad198x_spec *spec;
int err;
@@ -3674,1170 +1046,6 @@ static int ad1884_parse_auto_config(struct hda_codec *codec)
return err;
}
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static int patch_ad1884_basic(struct hda_codec *codec)
-{
- struct ad198x_spec *spec;
- int err;
-
- err = alloc_ad_spec(codec);
- if (err < 0)
- return err;
- spec = codec->spec;
-
- err = snd_hda_attach_beep_device(codec, 0x10);
- if (err < 0) {
- ad198x_free(codec);
- return err;
- }
- set_beep_amp(spec, 0x10, 0, HDA_OUTPUT);
-
- spec->multiout.max_channels = 2;
- spec->multiout.num_dacs = ARRAY_SIZE(ad1884_dac_nids);
- spec->multiout.dac_nids = ad1884_dac_nids;
- spec->multiout.dig_out_nid = AD1884_SPDIF_OUT;
- spec->num_adc_nids = ARRAY_SIZE(ad1884_adc_nids);
- spec->adc_nids = ad1884_adc_nids;
- spec->capsrc_nids = ad1884_capsrc_nids;
- spec->input_mux = &ad1884_capture_source;
- spec->num_mixers = 1;
- spec->mixers[0] = ad1884_base_mixers;
- spec->num_init_verbs = 1;
- spec->init_verbs[0] = ad1884_init_verbs;
- spec->spdif_route = 0;
-#ifdef CONFIG_PM
- spec->loopback.amplist = ad1884_loopbacks;
-#endif
- spec->vmaster_nid = 0x04;
- /* we need to cover all playback volumes */
- spec->slave_vols = ad1884_slave_vols;
- /* slaves may contain input volumes, so we can't raise to 0dB blindly */
- spec->avoid_init_slave_vol = 1;
-
- codec->patch_ops = ad198x_patch_ops;
-
- codec->no_trigger_sense = 1;
- codec->no_sticky_stream = 1;
-
- return 0;
-}
-
-static int patch_ad1884(struct hda_codec *codec)
-{
- int board_config;
-
- board_config = snd_hda_check_board_config(codec, AD1884_MODELS,
- ad1884_models, NULL);
- if (board_config < 0) {
- printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
- codec->chip_name);
- board_config = AD1884_AUTO;
- }
-
- if (board_config == AD1884_AUTO)
- return ad1884_parse_auto_config(codec);
- else
- return patch_ad1884_basic(codec);
-}
-#else /* ENABLE_AD_STATIC_QUIRKS */
-#define patch_ad1884 ad1884_parse_auto_config
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
-
-#ifdef ENABLE_AD_STATIC_QUIRKS
-/*
- * Lenovo Thinkpad T61/X61
- */
-static const struct hda_input_mux ad1984_thinkpad_capture_source = {
- .num_items = 4,
- .items = {
- { "Mic", 0x0 },
- { "Internal Mic", 0x1 },
- { "Mix", 0x3 },
- { "Dock Mic", 0x4 },
- },
-};
-
-
-/*
- * Dell Precision T3400
- */
-static const struct hda_input_mux ad1984_dell_desktop_capture_source = {
- .num_items = 3,
- .items = {
- { "Front Mic", 0x0 },
- { "Line-In", 0x1 },
- { "Mix", 0x3 },
- },
-};
-
-
-static const struct snd_kcontrol_new ad1984_thinkpad_mixers[] = {
- HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT),
- /* HDA_CODEC_VOLUME_IDX("PCM Playback Volume", 1, 0x03, 0x0, HDA_OUTPUT), */
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Speaker Playback Switch", 0x12, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x20, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x20, 0x01, HDA_INPUT),
- HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT),
- HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT),
- HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x20, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x20, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x14, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x15, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Dock Mic Boost Volume", 0x25, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- /* The multiple "Capture Source" controls confuse alsamixer
- * So call somewhat different..
- */
- /* .name = "Capture Source", */
- .name = "Input Source",
- .count = 2,
- .info = ad198x_mux_enum_info,
- .get = ad198x_mux_enum_get,
- .put = ad198x_mux_enum_put,
- },
- /* SPDIF controls */
- HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
- /* identical with ad1983 */
- .info = ad1983_spdif_route_info,
- .get = ad1983_spdif_route_get,
- .put = ad1983_spdif_route_put,
- },
- { } /* end */
-};
-
-/* additional verbs */
-static const struct hda_verb ad1984_thinkpad_init_verbs[] = {
- /* Port-E (docking station mic) pin */
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* docking mic boost */
- {0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- /* Analog PC Beeper - allow firmware/ACPI beeps */
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3) | 0x1a},
- /* Analog mixer - docking mic; mute as default */
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- /* enable EAPD bit */
- {0x12, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
- { } /* end */
-};
-
-/*
- * Dell Precision T3400
- */
-static const struct snd_kcontrol_new ad1984_dell_desktop_mixers[] = {
- HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Speaker Playback Switch", 0x12, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x13, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
- HDA_CODEC_VOLUME("Line-In Playback Volume", 0x20, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Line-In Playback Switch", 0x20, 0x01, HDA_INPUT),
- HDA_CODEC_VOLUME("Line-In Boost Volume", 0x15, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x14, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- /* The multiple "Capture Source" controls confuse alsamixer
- * So call somewhat different..
- */
- /* .name = "Capture Source", */
- .name = "Input Source",
- .count = 2,
- .info = ad198x_mux_enum_info,
- .get = ad198x_mux_enum_get,
- .put = ad198x_mux_enum_put,
- },
- { } /* end */
-};
-
-/* Digial MIC ADC NID 0x05 + 0x06 */
-static int ad1984_pcm_dmic_prepare(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- unsigned int stream_tag,
- unsigned int format,
- struct snd_pcm_substream *substream)
-{
- snd_hda_codec_setup_stream(codec, 0x05 + substream->number,
- stream_tag, 0, format);
- return 0;
-}
-
-static int ad1984_pcm_dmic_cleanup(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- struct snd_pcm_substream *substream)
-{
- snd_hda_codec_cleanup_stream(codec, 0x05 + substream->number);
- return 0;
-}
-
-static const struct hda_pcm_stream ad1984_pcm_dmic_capture = {
- .substreams = 2,
- .channels_min = 2,
- .channels_max = 2,
- .nid = 0x05,
- .ops = {
- .prepare = ad1984_pcm_dmic_prepare,
- .cleanup = ad1984_pcm_dmic_cleanup
- },
-};
-
-static int ad1984_build_pcms(struct hda_codec *codec)
-{
- struct ad198x_spec *spec = codec->spec;
- struct hda_pcm *info;
- int err;
-
- err = ad198x_build_pcms(codec);
- if (err < 0)
- return err;
-
- info = spec->pcm_rec + codec->num_pcms;
- codec->num_pcms++;
- info->name = "AD1984 Digital Mic";
- info->stream[SNDRV_PCM_STREAM_CAPTURE] = ad1984_pcm_dmic_capture;
- return 0;
-}
-
-/* models */
-enum {
- AD1984_AUTO,
- AD1984_BASIC,
- AD1984_THINKPAD,
- AD1984_DELL_DESKTOP,
- AD1984_MODELS
-};
-
-static const char * const ad1984_models[AD1984_MODELS] = {
- [AD1984_AUTO] = "auto",
- [AD1984_BASIC] = "basic",
- [AD1984_THINKPAD] = "thinkpad",
- [AD1984_DELL_DESKTOP] = "dell_desktop",
-};
-
-static const 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),
- {}
-};
-
-static int patch_ad1984(struct hda_codec *codec)
-{
- struct ad198x_spec *spec;
- int board_config, err;
-
- board_config = snd_hda_check_board_config(codec, AD1984_MODELS,
- ad1984_models, ad1984_cfg_tbl);
- if (board_config < 0) {
- printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
- codec->chip_name);
- board_config = AD1984_AUTO;
- }
-
- if (board_config == AD1984_AUTO)
- return ad1884_parse_auto_config(codec);
-
- err = patch_ad1884_basic(codec);
- if (err < 0)
- return err;
- spec = codec->spec;
-
- switch (board_config) {
- case AD1984_BASIC:
- /* additional digital mics */
- spec->mixers[spec->num_mixers++] = ad1984_dmic_mixers;
- codec->patch_ops.build_pcms = ad1984_build_pcms;
- break;
- case AD1984_THINKPAD:
- 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;
- spec->analog_beep = 1;
- break;
- case AD1984_DELL_DESKTOP:
- spec->multiout.dig_out_nid = 0;
- spec->input_mux = &ad1984_dell_desktop_capture_source;
- spec->mixers[0] = ad1984_dell_desktop_mixers;
- break;
- }
- return 0;
-}
-#else /* ENABLE_AD_STATIC_QUIRKS */
-#define patch_ad1984 ad1884_parse_auto_config
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
-
-/*
- * AD1883 / AD1884A / AD1984A / AD1984B
- *
- * port-B (0x14) - front mic-in
- * port-E (0x1c) - rear mic-in
- * port-F (0x16) - CD / ext out
- * port-C (0x15) - rear line-in
- * port-D (0x12) - rear line-out
- * port-A (0x11) - front hp-out
- *
- * AD1984A = AD1884A + digital-mic
- * AD1883 = equivalent with AD1984A
- * AD1984B = AD1984A + extra SPDIF-out
- *
- * FIXME:
- * We share the single DAC for both HP and line-outs (see AD1884/1984).
- */
-
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static const hda_nid_t ad1884a_dac_nids[1] = {
- 0x03,
-};
-
-#define ad1884a_adc_nids ad1884_adc_nids
-#define ad1884a_capsrc_nids ad1884_capsrc_nids
-
-#define AD1884A_SPDIF_OUT 0x02
-
-static const struct hda_input_mux ad1884a_capture_source = {
- .num_items = 5,
- .items = {
- { "Front Mic", 0x0 },
- { "Mic", 0x4 },
- { "Line", 0x1 },
- { "CD", 0x2 },
- { "Mix", 0x3 },
- },
-};
-
-static const struct snd_kcontrol_new ad1884a_base_mixers[] = {
- HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Front Playback Switch", 0x12, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x13, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
- HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x01, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x14, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Boost Volume", 0x15, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x25, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- /* The multiple "Capture Source" controls confuse alsamixer
- * So call somewhat different..
- */
- /* .name = "Capture Source", */
- .name = "Input Source",
- .count = 2,
- .info = ad198x_mux_enum_info,
- .get = ad198x_mux_enum_get,
- .put = ad198x_mux_enum_put,
- },
- /* SPDIF controls */
- HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
- /* identical with ad1983 */
- .info = ad1983_spdif_route_info,
- .get = ad1983_spdif_route_get,
- .put = ad1983_spdif_route_put,
- },
- { } /* end */
-};
-
-/*
- * initialization verbs
- */
-static const struct hda_verb ad1884a_init_verbs[] = {
- /* DACs; unmute as default */
- {0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
- /* Port-A (HP) mixer - route only from analog mixer */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- /* Port-A pin */
- {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Port-D (Line-out) mixer - route only from analog mixer */
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- /* Port-D pin */
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Mono-out mixer - route only from analog mixer */
- {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- /* Mono-out pin */
- {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Port-B (front mic) pin */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- /* Port-C (rear line-in) pin */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- /* Port-E (rear mic) pin */
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* no boost */
- /* Port-F (CD) pin */
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Analog mixer; mute as default */
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, /* aux */
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
- /* Analog Mix output amp */
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* capture sources */
- {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x0d, AC_VERB_SET_CONNECT_SEL, 0x0},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* SPDIF output amp */
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
- { } /* end */
-};
-
-#ifdef CONFIG_PM
-static const struct hda_amp_list ad1884a_loopbacks[] = {
- { 0x20, HDA_INPUT, 0 }, /* Front Mic */
- { 0x20, HDA_INPUT, 1 }, /* Mic */
- { 0x20, HDA_INPUT, 2 }, /* CD */
- { 0x20, HDA_INPUT, 4 }, /* Docking */
- { } /* end */
-};
-#endif
-
-/*
- * Laptop model
- *
- * Port A: Headphone jack
- * Port B: MIC jack
- * Port C: Internal MIC
- * Port D: Dock Line Out (if enabled)
- * Port E: Dock Line In (if enabled)
- * Port F: Internal speakers
- */
-
-static int ad1884a_mobile_master_sw_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- int ret = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol);
- int mute = (!ucontrol->value.integer.value[0] &&
- !ucontrol->value.integer.value[1]);
- /* toggle GPIO1 according to the mute state */
- snd_hda_codec_write_cache(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
- mute ? 0x02 : 0x0);
- return ret;
-}
-
-static const struct snd_kcontrol_new ad1884a_laptop_mixers[] = {
- HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Master Playback Switch",
- .subdevice = HDA_SUBDEV_AMP_FLAG,
- .info = snd_hda_mixer_amp_switch_info,
- .get = snd_hda_mixer_amp_switch_get,
- .put = ad1884a_mobile_master_sw_put,
- .private_value = HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
- },
- HDA_CODEC_MUTE("Dock Playback Switch", 0x12, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
- HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x20, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x20, 0x01, HDA_INPUT),
- HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x20, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x20, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x14, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x15, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Dock Mic Boost Volume", 0x25, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new ad1884a_mobile_mixers[] = {
- HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
- /*HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Master Playback Switch",
- .subdevice = HDA_SUBDEV_AMP_FLAG,
- .info = snd_hda_mixer_amp_switch_info,
- .get = snd_hda_mixer_amp_switch_get,
- .put = ad1884a_mobile_master_sw_put,
- .private_value = HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
- },
- HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
- HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Capture Volume", 0x14, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Capture Volume", 0x15, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
- { } /* end */
-};
-
-/* mute internal speaker if HP is plugged */
-static void ad1884a_hp_automute(struct hda_codec *codec)
-{
- unsigned int present;
-
- present = snd_hda_jack_detect(codec, 0x11);
- snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
- snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_EAPD_BTLENABLE,
- present ? 0x00 : 0x02);
-}
-
-/* switch to external mic if plugged */
-static void ad1884a_hp_automic(struct hda_codec *codec)
-{
- unsigned int present;
-
- present = snd_hda_jack_detect(codec, 0x14);
- snd_hda_codec_write(codec, 0x0c, 0, AC_VERB_SET_CONNECT_SEL,
- present ? 0 : 1);
-}
-
-#define AD1884A_HP_EVENT 0x37
-#define AD1884A_MIC_EVENT 0x36
-
-/* unsolicited event for HP jack sensing */
-static void ad1884a_hp_unsol_event(struct hda_codec *codec, unsigned int res)
-{
- switch (res >> 26) {
- case AD1884A_HP_EVENT:
- ad1884a_hp_automute(codec);
- break;
- case AD1884A_MIC_EVENT:
- ad1884a_hp_automic(codec);
- break;
- }
-}
-
-/* initialize jack-sensing, too */
-static int ad1884a_hp_init(struct hda_codec *codec)
-{
- ad198x_init(codec);
- ad1884a_hp_automute(codec);
- ad1884a_hp_automic(codec);
- return 0;
-}
-
-/* mute internal speaker if HP or docking HP is plugged */
-static void ad1884a_laptop_automute(struct hda_codec *codec)
-{
- unsigned int present;
-
- present = snd_hda_jack_detect(codec, 0x11);
- if (!present)
- present = snd_hda_jack_detect(codec, 0x12);
- snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
- snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_EAPD_BTLENABLE,
- present ? 0x00 : 0x02);
-}
-
-/* switch to external mic if plugged */
-static void ad1884a_laptop_automic(struct hda_codec *codec)
-{
- unsigned int idx;
-
- if (snd_hda_jack_detect(codec, 0x14))
- idx = 0;
- else if (snd_hda_jack_detect(codec, 0x1c))
- idx = 4;
- else
- idx = 1;
- snd_hda_codec_write(codec, 0x0c, 0, AC_VERB_SET_CONNECT_SEL, idx);
-}
-
-/* unsolicited event for HP jack sensing */
-static void ad1884a_laptop_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- switch (res >> 26) {
- case AD1884A_HP_EVENT:
- ad1884a_laptop_automute(codec);
- break;
- case AD1884A_MIC_EVENT:
- ad1884a_laptop_automic(codec);
- break;
- }
-}
-
-/* initialize jack-sensing, too */
-static int ad1884a_laptop_init(struct hda_codec *codec)
-{
- ad198x_init(codec);
- ad1884a_laptop_automute(codec);
- ad1884a_laptop_automic(codec);
- return 0;
-}
-
-/* additional verbs for laptop model */
-static const struct hda_verb ad1884a_laptop_verbs[] = {
- /* Port-A (HP) pin - always unmuted */
- {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Port-F (int speaker) mixer - route only from analog mixer */
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- /* Port-F (int speaker) pin */
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* required for compaq 6530s/6531s speaker output */
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- /* Port-C pin - internal mic-in */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */
- /* Port-D (docking line-out) pin - default unmuted */
- {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* analog mix */
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- /* unsolicited event for pin-sense */
- {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
- {0x12, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
- {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT},
- {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT},
- /* allow to touch GPIO1 (for mute control) */
- {0x01, AC_VERB_SET_GPIO_MASK, 0x02},
- {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02},
- {0x01, AC_VERB_SET_GPIO_DATA, 0x02}, /* first muted */
- { } /* end */
-};
-
-static const struct hda_verb ad1884a_mobile_verbs[] = {
- /* DACs; unmute as default */
- {0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
- /* Port-A (HP) mixer - route only from analog mixer */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- /* Port-A pin */
- {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- /* Port-A (HP) pin - always unmuted */
- {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Port-B (mic jack) pin */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */
- /* Port-C (int mic) pin */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */
- /* Port-F (int speaker) mixer - route only from analog mixer */
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- /* Port-F pin */
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Analog mixer; mute as default */
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
- /* Analog Mix output amp */
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* capture sources */
- /* {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0}, */ /* set via unsol */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x0d, AC_VERB_SET_CONNECT_SEL, 0x0},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* unsolicited event for pin-sense */
- {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
- {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT},
- /* allow to touch GPIO1 (for mute control) */
- {0x01, AC_VERB_SET_GPIO_MASK, 0x02},
- {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02},
- {0x01, AC_VERB_SET_GPIO_DATA, 0x02}, /* first muted */
- { } /* end */
-};
-
-/*
- * Thinkpad X300
- * 0x11 - HP
- * 0x12 - speaker
- * 0x14 - mic-in
- * 0x17 - built-in mic
- */
-
-static const struct hda_verb ad1984a_thinkpad_verbs[] = {
- /* HP unmute */
- {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* analog mix */
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- /* turn on EAPD */
- {0x12, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
- /* unsolicited event for pin-sense */
- {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
- /* internal mic - dmic */
- {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- /* set magic COEFs for dmic */
- {0x01, AC_VERB_SET_COEF_INDEX, 0x13f7},
- {0x01, AC_VERB_SET_PROC_COEF, 0x08},
- { } /* end */
-};
-
-static const struct snd_kcontrol_new ad1984a_thinkpad_mixers[] = {
- HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
- HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x14, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x17, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Capture Source",
- .info = ad198x_mux_enum_info,
- .get = ad198x_mux_enum_get,
- .put = ad198x_mux_enum_put,
- },
- { } /* end */
-};
-
-static const struct hda_input_mux ad1984a_thinkpad_capture_source = {
- .num_items = 3,
- .items = {
- { "Mic", 0x0 },
- { "Internal Mic", 0x5 },
- { "Mix", 0x3 },
- },
-};
-
-/* mute internal speaker if HP is plugged */
-static void ad1984a_thinkpad_automute(struct hda_codec *codec)
-{
- unsigned int present;
-
- present = snd_hda_jack_detect(codec, 0x11);
- snd_hda_codec_amp_stereo(codec, 0x12, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
-}
-
-/* unsolicited event for HP jack sensing */
-static void ad1984a_thinkpad_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- if ((res >> 26) != AD1884A_HP_EVENT)
- return;
- ad1984a_thinkpad_automute(codec);
-}
-
-/* initialize jack-sensing, too */
-static int ad1984a_thinkpad_init(struct hda_codec *codec)
-{
- ad198x_init(codec);
- ad1984a_thinkpad_automute(codec);
- return 0;
-}
-
-/*
- * Precision R5500
- * 0x12 - HP/line-out
- * 0x13 - speaker (mono)
- * 0x15 - mic-in
- */
-
-static const struct hda_verb ad1984a_precision_verbs[] = {
- /* Unmute main output path */
- {0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE + 0x1f}, /* 0dB */
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5) + 0x17}, /* 0dB */
- /* Analog mixer; mute as default */
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- /* Select mic as input */
- {0x0c, AC_VERB_SET_CONNECT_SEL, 0x1},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE + 0x27}, /* 0dB */
- /* Configure as mic */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */
- /* HP unmute */
- {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* turn on EAPD */
- {0x13, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
- /* unsolicited event for pin-sense */
- {0x12, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
- { } /* end */
-};
-
-static const struct snd_kcontrol_new ad1984a_precision_mixers[] = {
- HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
- HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x15, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Front Playback Switch", 0x12, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Speaker Playback Volume", 0x13, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
- { } /* end */
-};
-
-
-/* mute internal speaker if HP is plugged */
-static void ad1984a_precision_automute(struct hda_codec *codec)
-{
- unsigned int present;
-
- present = snd_hda_jack_detect(codec, 0x12);
- snd_hda_codec_amp_stereo(codec, 0x13, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
-}
-
-
-/* unsolicited event for HP jack sensing */
-static void ad1984a_precision_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- if ((res >> 26) != AD1884A_HP_EVENT)
- return;
- ad1984a_precision_automute(codec);
-}
-
-/* initialize jack-sensing, too */
-static int ad1984a_precision_init(struct hda_codec *codec)
-{
- ad198x_init(codec);
- ad1984a_precision_automute(codec);
- return 0;
-}
-
-
-/*
- * HP Touchsmart
- * port-A (0x11) - front hp-out
- * port-B (0x14) - unused
- * port-C (0x15) - unused
- * port-D (0x12) - rear line out
- * port-E (0x1c) - front mic-in
- * port-F (0x16) - Internal speakers
- * digital-mic (0x17) - Internal mic
- */
-
-static const struct hda_verb ad1984a_touchsmart_verbs[] = {
- /* DACs; unmute as default */
- {0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
- /* Port-A (HP) mixer - route only from analog mixer */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- /* Port-A pin */
- {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- /* Port-A (HP) pin - always unmuted */
- {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Port-E (int speaker) mixer - route only from analog mixer */
- {0x25, AC_VERB_SET_AMP_GAIN_MUTE, 0x03},
- /* Port-E pin */
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- /* Port-F (int speaker) mixer - route only from analog mixer */
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- /* Port-F pin */
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Analog mixer; mute as default */
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
- /* Analog Mix output amp */
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* capture sources */
- /* {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0}, */ /* set via unsol */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x0d, AC_VERB_SET_CONNECT_SEL, 0x0},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* unsolicited event for pin-sense */
- {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
- {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT},
- /* allow to touch GPIO1 (for mute control) */
- {0x01, AC_VERB_SET_GPIO_MASK, 0x02},
- {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02},
- {0x01, AC_VERB_SET_GPIO_DATA, 0x02}, /* first muted */
- /* internal mic - dmic */
- {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- /* set magic COEFs for dmic */
- {0x01, AC_VERB_SET_COEF_INDEX, 0x13f7},
- {0x01, AC_VERB_SET_PROC_COEF, 0x08},
- { } /* end */
-};
-
-static const struct snd_kcontrol_new ad1984a_touchsmart_mixers[] = {
- HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
-/* HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .subdevice = HDA_SUBDEV_AMP_FLAG,
- .name = "Master Playback Switch",
- .info = snd_hda_mixer_amp_switch_info,
- .get = snd_hda_mixer_amp_switch_get,
- .put = ad1884a_mobile_master_sw_put,
- .private_value = HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
- },
- HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
- HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
- HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x25, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x17, 0x0, HDA_INPUT),
- { } /* end */
-};
-
-/* switch to external mic if plugged */
-static void ad1984a_touchsmart_automic(struct hda_codec *codec)
-{
- if (snd_hda_jack_detect(codec, 0x1c))
- snd_hda_codec_write(codec, 0x0c, 0,
- AC_VERB_SET_CONNECT_SEL, 0x4);
- else
- snd_hda_codec_write(codec, 0x0c, 0,
- AC_VERB_SET_CONNECT_SEL, 0x5);
-}
-
-
-/* unsolicited event for HP jack sensing */
-static void ad1984a_touchsmart_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- switch (res >> 26) {
- case AD1884A_HP_EVENT:
- ad1884a_hp_automute(codec);
- break;
- case AD1884A_MIC_EVENT:
- ad1984a_touchsmart_automic(codec);
- break;
- }
-}
-
-/* initialize jack-sensing, too */
-static int ad1984a_touchsmart_init(struct hda_codec *codec)
-{
- ad198x_init(codec);
- ad1884a_hp_automute(codec);
- ad1984a_touchsmart_automic(codec);
- return 0;
-}
-
-
-/*
- */
-
-enum {
- AD1884A_AUTO,
- AD1884A_DESKTOP,
- AD1884A_LAPTOP,
- AD1884A_MOBILE,
- AD1884A_THINKPAD,
- AD1984A_TOUCHSMART,
- AD1984A_PRECISION,
- AD1884A_MODELS
-};
-
-static const char * const ad1884a_models[AD1884A_MODELS] = {
- [AD1884A_AUTO] = "auto",
- [AD1884A_DESKTOP] = "desktop",
- [AD1884A_LAPTOP] = "laptop",
- [AD1884A_MOBILE] = "mobile",
- [AD1884A_THINKPAD] = "thinkpad",
- [AD1984A_TOUCHSMART] = "touchsmart",
- [AD1984A_PRECISION] = "precision",
-};
-
-static const struct snd_pci_quirk ad1884a_cfg_tbl[] = {
- SND_PCI_QUIRK(0x1028, 0x04ac, "Precision R5500", AD1984A_PRECISION),
- SND_PCI_QUIRK(0x103c, 0x3030, "HP", AD1884A_MOBILE),
- SND_PCI_QUIRK(0x103c, 0x3037, "HP 2230s", AD1884A_LAPTOP),
- SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE),
- SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x3070, "HP", AD1884A_MOBILE),
- SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x30d0, "HP laptop", AD1884A_LAPTOP),
- SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x30e0, "HP laptop", AD1884A_LAPTOP),
- SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3600, "HP laptop", AD1884A_LAPTOP),
- SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x7010, "HP laptop", AD1884A_MOBILE),
- SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD),
- SND_PCI_QUIRK(0x103c, 0x2a82, "Touchsmart", AD1984A_TOUCHSMART),
- {}
-};
-
-static int patch_ad1884a(struct hda_codec *codec)
-{
- struct ad198x_spec *spec;
- int err, board_config;
-
- board_config = snd_hda_check_board_config(codec, AD1884A_MODELS,
- ad1884a_models,
- ad1884a_cfg_tbl);
- if (board_config < 0) {
- printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
- codec->chip_name);
- board_config = AD1884A_AUTO;
- }
-
- if (board_config == AD1884A_AUTO)
- return ad1884_parse_auto_config(codec);
-
- err = alloc_ad_spec(codec);
- if (err < 0)
- return err;
- spec = codec->spec;
-
- err = snd_hda_attach_beep_device(codec, 0x10);
- if (err < 0) {
- ad198x_free(codec);
- return err;
- }
- set_beep_amp(spec, 0x10, 0, HDA_OUTPUT);
-
- spec->multiout.max_channels = 2;
- spec->multiout.num_dacs = ARRAY_SIZE(ad1884a_dac_nids);
- spec->multiout.dac_nids = ad1884a_dac_nids;
- spec->multiout.dig_out_nid = AD1884A_SPDIF_OUT;
- spec->num_adc_nids = ARRAY_SIZE(ad1884a_adc_nids);
- spec->adc_nids = ad1884a_adc_nids;
- spec->capsrc_nids = ad1884a_capsrc_nids;
- spec->input_mux = &ad1884a_capture_source;
- spec->num_mixers = 1;
- spec->mixers[0] = ad1884a_base_mixers;
- spec->num_init_verbs = 1;
- spec->init_verbs[0] = ad1884a_init_verbs;
- spec->spdif_route = 0;
-#ifdef CONFIG_PM
- spec->loopback.amplist = ad1884a_loopbacks;
-#endif
- codec->patch_ops = ad198x_patch_ops;
-
- /* override some parameters */
- switch (board_config) {
- case AD1884A_LAPTOP:
- spec->mixers[0] = ad1884a_laptop_mixers;
- spec->init_verbs[spec->num_init_verbs++] = ad1884a_laptop_verbs;
- spec->multiout.dig_out_nid = 0;
- codec->patch_ops.unsol_event = ad1884a_laptop_unsol_event;
- codec->patch_ops.init = ad1884a_laptop_init;
- /* set the upper-limit for mixer amp to 0dB for avoiding the
- * possible damage by overloading
- */
- snd_hda_override_amp_caps(codec, 0x20, HDA_INPUT,
- (0x17 << AC_AMPCAP_OFFSET_SHIFT) |
- (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) |
- (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) |
- (1 << AC_AMPCAP_MUTE_SHIFT));
- break;
- case AD1884A_MOBILE:
- spec->mixers[0] = ad1884a_mobile_mixers;
- spec->init_verbs[0] = ad1884a_mobile_verbs;
- spec->multiout.dig_out_nid = 0;
- codec->patch_ops.unsol_event = ad1884a_hp_unsol_event;
- codec->patch_ops.init = ad1884a_hp_init;
- /* set the upper-limit for mixer amp to 0dB for avoiding the
- * possible damage by overloading
- */
- snd_hda_override_amp_caps(codec, 0x20, HDA_INPUT,
- (0x17 << AC_AMPCAP_OFFSET_SHIFT) |
- (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) |
- (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) |
- (1 << AC_AMPCAP_MUTE_SHIFT));
- break;
- case AD1884A_THINKPAD:
- spec->mixers[0] = ad1984a_thinkpad_mixers;
- spec->init_verbs[spec->num_init_verbs++] =
- ad1984a_thinkpad_verbs;
- spec->multiout.dig_out_nid = 0;
- spec->input_mux = &ad1984a_thinkpad_capture_source;
- codec->patch_ops.unsol_event = ad1984a_thinkpad_unsol_event;
- codec->patch_ops.init = ad1984a_thinkpad_init;
- break;
- case AD1984A_PRECISION:
- spec->mixers[0] = ad1984a_precision_mixers;
- spec->init_verbs[spec->num_init_verbs++] =
- ad1984a_precision_verbs;
- spec->multiout.dig_out_nid = 0;
- codec->patch_ops.unsol_event = ad1984a_precision_unsol_event;
- codec->patch_ops.init = ad1984a_precision_init;
- break;
- case AD1984A_TOUCHSMART:
- spec->mixers[0] = ad1984a_touchsmart_mixers;
- spec->init_verbs[0] = ad1984a_touchsmart_verbs;
- spec->multiout.dig_out_nid = 0;
- codec->patch_ops.unsol_event = ad1984a_touchsmart_unsol_event;
- codec->patch_ops.init = ad1984a_touchsmart_init;
- /* set the upper-limit for mixer amp to 0dB for avoiding the
- * possible damage by overloading
- */
- snd_hda_override_amp_caps(codec, 0x20, HDA_INPUT,
- (0x17 << AC_AMPCAP_OFFSET_SHIFT) |
- (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) |
- (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) |
- (1 << AC_AMPCAP_MUTE_SHIFT));
- break;
- }
-
- codec->no_trigger_sense = 1;
- codec->no_sticky_stream = 1;
-
- return 0;
-}
-#else /* ENABLE_AD_STATIC_QUIRKS */
-#define patch_ad1884a ad1884_parse_auto_config
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
-
/*
* AD1882 / AD1882A
*
@@ -4850,299 +1058,7 @@ static int patch_ad1884a(struct hda_codec *codec)
* port-G - rear clfe-out (6stack)
*/
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static const hda_nid_t ad1882_dac_nids[3] = {
- 0x04, 0x03, 0x05
-};
-
-static const hda_nid_t ad1882_adc_nids[2] = {
- 0x08, 0x09,
-};
-
-static const hda_nid_t ad1882_capsrc_nids[2] = {
- 0x0c, 0x0d,
-};
-
-#define AD1882_SPDIF_OUT 0x02
-
-/* list: 0x11, 0x39, 0x3a, 0x18, 0x3c, 0x3b, 0x12, 0x20 */
-static const struct hda_input_mux ad1882_capture_source = {
- .num_items = 5,
- .items = {
- { "Front Mic", 0x1 },
- { "Mic", 0x4 },
- { "Line", 0x2 },
- { "CD", 0x3 },
- { "Mix", 0x7 },
- },
-};
-
-/* list: 0x11, 0x39, 0x3a, 0x3c, 0x18, 0x1f, 0x12, 0x20 */
-static const struct hda_input_mux ad1882a_capture_source = {
- .num_items = 5,
- .items = {
- { "Front Mic", 0x1 },
- { "Mic", 0x4},
- { "Line", 0x2 },
- { "Digital Mic", 0x06 },
- { "Mix", 0x7 },
- },
-};
-
-static const struct snd_kcontrol_new ad1882_base_mixers[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x05, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x05, 2, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Front Playback Switch", 0x12, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x13, 1, 0x0, HDA_OUTPUT),
-
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x3c, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x39, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Line-In Boost Volume", 0x3a, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- /* The multiple "Capture Source" controls confuse alsamixer
- * So call somewhat different..
- */
- /* .name = "Capture Source", */
- .name = "Input Source",
- .count = 2,
- .info = ad198x_mux_enum_info,
- .get = ad198x_mux_enum_get,
- .put = ad198x_mux_enum_put,
- },
- /* SPDIF controls */
- HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
- /* identical with ad1983 */
- .info = ad1983_spdif_route_info,
- .get = ad1983_spdif_route_get,
- .put = ad1983_spdif_route_put,
- },
- { } /* end */
-};
-
-static const struct snd_kcontrol_new ad1882_loopback_mixers[] = {
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x06, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x06, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new ad1882a_loopback_mixers[] = {
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x01, HDA_INPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x06, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x06, HDA_INPUT),
- HDA_CODEC_VOLUME("Digital Mic Boost Volume", 0x1f, 0x0, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new ad1882_3stack_mixers[] = {
- HDA_CODEC_MUTE("Surround Playback Switch", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x17, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x17, 2, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Channel Mode",
- .info = ad198x_ch_mode_info,
- .get = ad198x_ch_mode_get,
- .put = ad198x_ch_mode_put,
- },
- { } /* end */
-};
-
-/* simple auto-mute control for AD1882 3-stack board */
-#define AD1882_HP_EVENT 0x01
-
-static void ad1882_3stack_automute(struct hda_codec *codec)
-{
- bool mute = snd_hda_jack_detect(codec, 0x11);
- snd_hda_codec_write(codec, 0x12, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
- mute ? 0 : PIN_OUT);
-}
-
-static int ad1882_3stack_automute_init(struct hda_codec *codec)
-{
- ad198x_init(codec);
- ad1882_3stack_automute(codec);
- return 0;
-}
-
-static void ad1882_3stack_unsol_event(struct hda_codec *codec, unsigned int res)
-{
- switch (res >> 26) {
- case AD1882_HP_EVENT:
- ad1882_3stack_automute(codec);
- break;
- }
-}
-
-static const struct snd_kcontrol_new ad1882_6stack_mixers[] = {
- HDA_CODEC_MUTE("Surround Playback Switch", 0x16, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x24, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x24, 2, 0x0, HDA_OUTPUT),
- { } /* end */
-};
-
-static const struct hda_verb ad1882_ch2_init[] = {
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- { } /* end */
-};
-
-static const struct hda_verb ad1882_ch4_init[] = {
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- { } /* end */
-};
-
-static const struct hda_verb ad1882_ch6_init[] = {
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- { } /* end */
-};
-
-static const struct hda_channel_mode ad1882_modes[3] = {
- { 2, ad1882_ch2_init },
- { 4, ad1882_ch4_init },
- { 6, ad1882_ch6_init },
-};
-
-/*
- * initialization verbs
- */
-static const struct hda_verb ad1882_init_verbs[] = {
- /* DACs; mute as default */
- {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- /* Port-A (HP) mixer */
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- /* Port-A pin */
- {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* HP selector - select DAC2 */
- {0x37, AC_VERB_SET_CONNECT_SEL, 0x1},
- /* Port-D (Line-out) mixer */
- {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- /* Port-D pin */
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Mono-out mixer */
- {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- /* Mono-out pin */
- {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Port-B (front mic) pin */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x39, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* boost */
- /* Port-C (line-in) pin */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x3a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* boost */
- /* Port-C mixer - mute as input */
- {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* Port-E (mic-in) pin */
- {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x3c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* boost */
- /* Port-E mixer - mute as input */
- {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* Port-F (surround) */
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Port-G (CLFE) */
- {0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Analog mixer; mute as default */
- /* list: 0x39, 0x3a, 0x11, 0x12, 0x3c, 0x3b, 0x18, 0x1a */
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
- /* Analog Mix output amp */
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */
- /* SPDIF output selector */
- {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
- {0x02, AC_VERB_SET_CONNECT_SEL, 0x0}, /* PCM */
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
- { } /* end */
-};
-
-static const struct hda_verb ad1882_3stack_automute_verbs[] = {
- {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1882_HP_EVENT},
- { } /* end */
-};
-
-#ifdef CONFIG_PM
-static const struct hda_amp_list ad1882_loopbacks[] = {
- { 0x20, HDA_INPUT, 0 }, /* Front Mic */
- { 0x20, HDA_INPUT, 1 }, /* Mic */
- { 0x20, HDA_INPUT, 4 }, /* Line */
- { 0x20, HDA_INPUT, 6 }, /* CD */
- { } /* end */
-};
-#endif
-
-/* models */
-enum {
- AD1882_AUTO,
- AD1882_3STACK,
- AD1882_6STACK,
- AD1882_3STACK_AUTOMUTE,
- AD1882_MODELS
-};
-
-static const char * const ad1882_models[AD1986A_MODELS] = {
- [AD1882_AUTO] = "auto",
- [AD1882_3STACK] = "3stack",
- [AD1882_6STACK] = "6stack",
- [AD1882_3STACK_AUTOMUTE] = "3stack-automute",
-};
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
-static int ad1882_parse_auto_config(struct hda_codec *codec)
+static int patch_ad1882(struct hda_codec *codec)
{
struct ad198x_spec *spec;
int err;
@@ -5169,110 +1085,20 @@ static int ad1882_parse_auto_config(struct hda_codec *codec)
return err;
}
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static int patch_ad1882(struct hda_codec *codec)
-{
- struct ad198x_spec *spec;
- int err, board_config;
-
- board_config = snd_hda_check_board_config(codec, AD1882_MODELS,
- ad1882_models, NULL);
- if (board_config < 0) {
- printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
- codec->chip_name);
- board_config = AD1882_AUTO;
- }
-
- if (board_config == AD1882_AUTO)
- return ad1882_parse_auto_config(codec);
-
- err = alloc_ad_spec(codec);
- if (err < 0)
- return err;
- spec = codec->spec;
-
- err = snd_hda_attach_beep_device(codec, 0x10);
- if (err < 0) {
- ad198x_free(codec);
- return err;
- }
- set_beep_amp(spec, 0x10, 0, HDA_OUTPUT);
-
- spec->multiout.max_channels = 6;
- spec->multiout.num_dacs = 3;
- spec->multiout.dac_nids = ad1882_dac_nids;
- spec->multiout.dig_out_nid = AD1882_SPDIF_OUT;
- spec->num_adc_nids = ARRAY_SIZE(ad1882_adc_nids);
- spec->adc_nids = ad1882_adc_nids;
- spec->capsrc_nids = ad1882_capsrc_nids;
- if (codec->vendor_id == 0x11d41882)
- spec->input_mux = &ad1882_capture_source;
- else
- spec->input_mux = &ad1882a_capture_source;
- spec->num_mixers = 2;
- spec->mixers[0] = ad1882_base_mixers;
- if (codec->vendor_id == 0x11d41882)
- spec->mixers[1] = ad1882_loopback_mixers;
- else
- spec->mixers[1] = ad1882a_loopback_mixers;
- spec->num_init_verbs = 1;
- spec->init_verbs[0] = ad1882_init_verbs;
- spec->spdif_route = 0;
-#ifdef CONFIG_PM
- spec->loopback.amplist = ad1882_loopbacks;
-#endif
- spec->vmaster_nid = 0x04;
-
- codec->patch_ops = ad198x_patch_ops;
-
- /* override some parameters */
- switch (board_config) {
- default:
- case AD1882_3STACK:
- case AD1882_3STACK_AUTOMUTE:
- spec->num_mixers = 3;
- spec->mixers[2] = ad1882_3stack_mixers;
- spec->channel_mode = ad1882_modes;
- spec->num_channel_mode = ARRAY_SIZE(ad1882_modes);
- spec->need_dac_fix = 1;
- spec->multiout.max_channels = 2;
- spec->multiout.num_dacs = 1;
- if (board_config != AD1882_3STACK) {
- spec->init_verbs[spec->num_init_verbs++] =
- ad1882_3stack_automute_verbs;
- codec->patch_ops.unsol_event = ad1882_3stack_unsol_event;
- codec->patch_ops.init = ad1882_3stack_automute_init;
- }
- break;
- case AD1882_6STACK:
- spec->num_mixers = 3;
- spec->mixers[2] = ad1882_6stack_mixers;
- break;
- }
-
- codec->no_trigger_sense = 1;
- codec->no_sticky_stream = 1;
-
- return 0;
-}
-#else /* ENABLE_AD_STATIC_QUIRKS */
-#define patch_ad1882 ad1882_parse_auto_config
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
/*
* patch entries
*/
static const struct hda_codec_preset snd_hda_preset_analog[] = {
- { .id = 0x11d4184a, .name = "AD1884A", .patch = patch_ad1884a },
+ { .id = 0x11d4184a, .name = "AD1884A", .patch = patch_ad1884 },
{ .id = 0x11d41882, .name = "AD1882", .patch = patch_ad1882 },
- { .id = 0x11d41883, .name = "AD1883", .patch = patch_ad1884a },
+ { .id = 0x11d41883, .name = "AD1883", .patch = patch_ad1884 },
{ .id = 0x11d41884, .name = "AD1884", .patch = patch_ad1884 },
- { .id = 0x11d4194a, .name = "AD1984A", .patch = patch_ad1884a },
- { .id = 0x11d4194b, .name = "AD1984B", .patch = patch_ad1884a },
+ { .id = 0x11d4194a, .name = "AD1984A", .patch = patch_ad1884 },
+ { .id = 0x11d4194b, .name = "AD1984B", .patch = patch_ad1884 },
{ .id = 0x11d41981, .name = "AD1981", .patch = patch_ad1981 },
{ .id = 0x11d41983, .name = "AD1983", .patch = patch_ad1983 },
- { .id = 0x11d41984, .name = "AD1984", .patch = patch_ad1984 },
+ { .id = 0x11d41984, .name = "AD1984", .patch = patch_ad1884 },
{ .id = 0x11d41986, .name = "AD1986A", .patch = patch_ad1986a },
{ .id = 0x11d41988, .name = "AD1988", .patch = patch_ad1988 },
{ .id = 0x11d4198b, .name = "AD1988B", .patch = patch_ad1988 },
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index de00ce166470..4edd2d0f9a3c 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -66,6 +66,8 @@ struct conexant_spec {
hda_nid_t eapds[4];
bool dynamic_eapd;
+ unsigned int parse_flags; /* flag for snd_hda_parse_pin_defcfg() */
+
#ifdef ENABLE_CXT_STATIC_QUIRKS
const struct snd_kcontrol_new *mixers[5];
int num_mixers;
@@ -3200,6 +3202,9 @@ static int cx_auto_init(struct hda_codec *codec)
snd_hda_gen_init(codec);
if (!spec->dynamic_eapd)
cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, true);
+
+ snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_INIT);
+
return 0;
}
@@ -3224,6 +3229,8 @@ enum {
CXT_PINCFG_LEMOTE_A1205,
CXT_FIXUP_STEREO_DMIC,
CXT_FIXUP_INC_MIC_BOOST,
+ CXT_FIXUP_HEADPHONE_MIC_PIN,
+ CXT_FIXUP_HEADPHONE_MIC,
};
static void cxt_fixup_stereo_dmic(struct hda_codec *codec,
@@ -3246,6 +3253,59 @@ static void cxt5066_increase_mic_boost(struct hda_codec *codec,
(0 << AC_AMPCAP_MUTE_SHIFT));
}
+static void cxt_update_headset_mode(struct hda_codec *codec)
+{
+ /* The verbs used in this function were tested on a Conexant CX20751/2 codec. */
+ int i;
+ bool mic_mode = false;
+ struct conexant_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->gen.autocfg;
+
+ hda_nid_t mux_pin = spec->gen.imux_pins[spec->gen.cur_mux[0]];
+
+ for (i = 0; i < cfg->num_inputs; i++)
+ if (cfg->inputs[i].pin == mux_pin) {
+ mic_mode = !!cfg->inputs[i].is_headphone_mic;
+ break;
+ }
+
+ if (mic_mode) {
+ snd_hda_codec_write_cache(codec, 0x1c, 0, 0x410, 0x7c); /* enable merged mode for analog int-mic */
+ spec->gen.hp_jack_present = false;
+ } else {
+ snd_hda_codec_write_cache(codec, 0x1c, 0, 0x410, 0x54); /* disable merged mode for analog int-mic */
+ spec->gen.hp_jack_present = snd_hda_jack_detect(codec, spec->gen.autocfg.hp_pins[0]);
+ }
+
+ snd_hda_gen_update_outputs(codec);
+}
+
+static void cxt_update_headset_mode_hook(struct hda_codec *codec,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ cxt_update_headset_mode(codec);
+}
+
+static void cxt_fixup_headphone_mic(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ struct conexant_spec *spec = codec->spec;
+
+ switch (action) {
+ case HDA_FIXUP_ACT_PRE_PROBE:
+ spec->parse_flags |= HDA_PINCFG_HEADPHONE_MIC;
+ break;
+ case HDA_FIXUP_ACT_PROBE:
+ spec->gen.cap_sync_hook = cxt_update_headset_mode_hook;
+ spec->gen.automute_hook = cxt_update_headset_mode;
+ break;
+ case HDA_FIXUP_ACT_INIT:
+ cxt_update_headset_mode(codec);
+ break;
+ }
+}
+
+
/* ThinkPad X200 & co with cxt5051 */
static const struct hda_pintbl cxt_pincfg_lenovo_x200[] = {
{ 0x16, 0x042140ff }, /* HP (seq# overridden) */
@@ -3302,6 +3362,19 @@ static const struct hda_fixup cxt_fixups[] = {
.type = HDA_FIXUP_FUNC,
.v.func = cxt5066_increase_mic_boost,
},
+ [CXT_FIXUP_HEADPHONE_MIC_PIN] = {
+ .type = HDA_FIXUP_PINS,
+ .chained = true,
+ .chain_id = CXT_FIXUP_HEADPHONE_MIC,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x18, 0x03a1913d }, /* use as headphone mic, without its own jack detect */
+ { }
+ }
+ },
+ [CXT_FIXUP_HEADPHONE_MIC] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = cxt_fixup_headphone_mic,
+ },
};
static const struct snd_pci_quirk cxt5051_fixups[] = {
@@ -3311,6 +3384,7 @@ static const struct snd_pci_quirk cxt5051_fixups[] = {
static const struct snd_pci_quirk cxt5066_fixups[] = {
SND_PCI_QUIRK(0x1025, 0x0543, "Acer Aspire One 522", CXT_FIXUP_STEREO_DMIC),
+ SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN),
SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410),
SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T410", CXT_PINCFG_LENOVO_TP410),
SND_PCI_QUIRK(0x17aa, 0x215f, "Lenovo T510", CXT_PINCFG_LENOVO_TP410),
@@ -3395,7 +3469,8 @@ static int patch_conexant_auto(struct hda_codec *codec)
snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
- err = snd_hda_parse_pin_defcfg(codec, &spec->gen.autocfg, NULL, 0);
+ err = snd_hda_parse_pin_defcfg(codec, &spec->gen.autocfg, NULL,
+ spec->parse_flags);
if (err < 0)
goto error;
@@ -3416,6 +3491,8 @@ static int patch_conexant_auto(struct hda_codec *codec)
codec->bus->allow_bus_reset = 1;
}
+ snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE);
+
return 0;
error:
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 9f3586276871..24d82d6c3464 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -959,6 +959,7 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res)
int pin_nid;
int pin_idx;
struct hda_jack_tbl *jack;
+ int dev_entry = (res & AC_UNSOL_RES_DE) >> AC_UNSOL_RES_DE_SHIFT;
jack = snd_hda_jack_tbl_get_from_tag(codec, tag);
if (!jack)
@@ -967,8 +968,8 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res)
jack->jack_dirty = 1;
_snd_printd(SND_PR_VERBOSE,
- "HDMI hot plug event: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n",
- codec->addr, pin_nid,
+ "HDMI hot plug event: Codec=%d Pin=%d Device=%d Inactive=%d Presence_Detect=%d ELD_Valid=%d\n",
+ codec->addr, pin_nid, dev_entry, !!(res & AC_UNSOL_RES_IA),
!!(res & AC_UNSOL_RES_PD), !!(res & AC_UNSOL_RES_ELDV));
pin_idx = pin_nid_to_pin_index(spec, pin_nid);
@@ -1992,8 +1993,10 @@ static int patch_generic_hdmi(struct hda_codec *codec)
return -EINVAL;
}
codec->patch_ops = generic_hdmi_patch_ops;
- if (codec->vendor_id == 0x80862807)
+ if (codec->vendor_id == 0x80862807) {
codec->patch_ops.set_power_state = haswell_set_power_state;
+ codec->dp_mst = true;
+ }
generic_hdmi_init_per_pins(codec);
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 389db4c2801b..9e9378cde8fa 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -282,6 +282,7 @@ static void alc_eapd_shutup(struct hda_codec *codec)
{
alc_auto_setup_eapd(codec, false);
msleep(200);
+ snd_hda_shutup_pins(codec);
}
/* generic EAPD initialization */
@@ -826,7 +827,8 @@ static inline void alc_shutup(struct hda_codec *codec)
if (spec && spec->shutup)
spec->shutup(codec);
- snd_hda_shutup_pins(codec);
+ else
+ snd_hda_shutup_pins(codec);
}
#define alc_free snd_hda_gen_free
@@ -1853,8 +1855,10 @@ static void alc882_fixup_no_primary_hp(struct hda_codec *codec,
const struct hda_fixup *fix, int action)
{
struct alc_spec *spec = codec->spec;
- if (action == HDA_FIXUP_ACT_PRE_PROBE)
+ if (action == HDA_FIXUP_ACT_PRE_PROBE) {
spec->gen.no_primary_hp = 1;
+ spec->gen.no_multi_io = 1;
+ }
}
static const struct hda_fixup alc882_fixups[] = {
@@ -2533,6 +2537,7 @@ enum {
ALC269_TYPE_ALC269VD,
ALC269_TYPE_ALC280,
ALC269_TYPE_ALC282,
+ ALC269_TYPE_ALC283,
ALC269_TYPE_ALC284,
ALC269_TYPE_ALC286,
};
@@ -2558,6 +2563,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
case ALC269_TYPE_ALC269VB:
case ALC269_TYPE_ALC269VD:
case ALC269_TYPE_ALC282:
+ case ALC269_TYPE_ALC283:
case ALC269_TYPE_ALC286:
ssids = alc269_ssids;
break;
@@ -2583,15 +2589,81 @@ static void alc269_shutup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
- if (spec->codec_variant != ALC269_TYPE_ALC269VB)
- return;
-
if (spec->codec_variant == ALC269_TYPE_ALC269VB)
alc269vb_toggle_power_output(codec, 0);
if (spec->codec_variant == ALC269_TYPE_ALC269VB &&
(alc_get_coef0(codec) & 0x00ff) == 0x018) {
msleep(150);
}
+ snd_hda_shutup_pins(codec);
+}
+
+static void alc283_init(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0];
+ bool hp_pin_sense;
+ int val;
+
+ if (!hp_pin)
+ return;
+ hp_pin_sense = snd_hda_jack_detect(codec, hp_pin);
+
+ /* Index 0x43 Direct Drive HP AMP LPM Control 1 */
+ /* Headphone capless set to high power mode */
+ alc_write_coef_idx(codec, 0x43, 0x9004);
+
+ snd_hda_codec_write(codec, hp_pin, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE);
+
+ if (hp_pin_sense)
+ msleep(85);
+
+ snd_hda_codec_write(codec, hp_pin, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+
+ if (hp_pin_sense)
+ msleep(85);
+ /* Index 0x46 Combo jack auto switch control 2 */
+ /* 3k pull low control for Headset jack. */
+ val = alc_read_coef_idx(codec, 0x46);
+ alc_write_coef_idx(codec, 0x46, val & ~(3 << 12));
+ /* Headphone capless set to normal mode */
+ alc_write_coef_idx(codec, 0x43, 0x9614);
+}
+
+static void alc283_shutup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0];
+ bool hp_pin_sense;
+ int val;
+
+ if (!hp_pin) {
+ alc269_shutup(codec);
+ return;
+ }
+
+ hp_pin_sense = snd_hda_jack_detect(codec, hp_pin);
+
+ alc_write_coef_idx(codec, 0x43, 0x9004);
+
+ snd_hda_codec_write(codec, hp_pin, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE);
+
+ if (hp_pin_sense)
+ msleep(85);
+
+ snd_hda_codec_write(codec, hp_pin, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0);
+
+ val = alc_read_coef_idx(codec, 0x46);
+ alc_write_coef_idx(codec, 0x46, val | (3 << 12));
+
+ if (hp_pin_sense)
+ msleep(85);
+ snd_hda_shutup_pins(codec);
+ alc_write_coef_idx(codec, 0x43, 0x9614);
}
static void alc5505_coef_set(struct hda_codec *codec, unsigned int index_reg,
@@ -2722,6 +2794,7 @@ static int alc269_resume(struct hda_codec *codec)
hda_call_check_power_status(codec, 0x01);
if (spec->has_alc5505_dsp)
alc5505_dsp_resume(codec);
+
return 0;
}
#endif /* CONFIG_PM */
@@ -3261,6 +3334,28 @@ static void alc_fixup_headset_mode_alc668(struct hda_codec *codec,
alc_fixup_headset_mode(codec, fix, action);
}
+/* Returns the nid of the external mic input pin, or 0 if it cannot be found. */
+static int find_ext_mic_pin(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->gen.autocfg;
+ hda_nid_t nid;
+ unsigned int defcfg;
+ int i;
+
+ for (i = 0; i < cfg->num_inputs; i++) {
+ if (cfg->inputs[i].type != AUTO_PIN_MIC)
+ continue;
+ nid = cfg->inputs[i].pin;
+ defcfg = snd_hda_codec_get_pincfg(codec, nid);
+ if (snd_hda_get_input_pin_attr(defcfg) == INPUT_PIN_ATTR_INT)
+ continue;
+ return nid;
+ }
+
+ return 0;
+}
+
static void alc271_hp_gate_mic_jack(struct hda_codec *codec,
const struct hda_fixup *fix,
int action)
@@ -3268,11 +3363,12 @@ static void alc271_hp_gate_mic_jack(struct hda_codec *codec,
struct alc_spec *spec = codec->spec;
if (action == HDA_FIXUP_ACT_PROBE) {
- if (snd_BUG_ON(!spec->gen.am_entry[1].pin ||
- !spec->gen.autocfg.hp_pins[0]))
+ int mic_pin = find_ext_mic_pin(codec);
+ int hp_pin = spec->gen.autocfg.hp_pins[0];
+
+ if (snd_BUG_ON(!mic_pin || !hp_pin))
return;
- snd_hda_jack_set_gating_jack(codec, spec->gen.am_entry[1].pin,
- spec->gen.autocfg.hp_pins[0]);
+ snd_hda_jack_set_gating_jack(codec, mic_pin, hp_pin);
}
}
@@ -3308,6 +3404,45 @@ static void alc269_fixup_limit_int_mic_boost(struct hda_codec *codec,
}
}
+static void alc283_hp_automute_hook(struct hda_codec *codec,
+ struct hda_jack_tbl *jack)
+{
+ struct alc_spec *spec = codec->spec;
+ int vref;
+
+ msleep(200);
+ snd_hda_gen_hp_automute(codec, jack);
+
+ vref = spec->gen.hp_jack_present ? PIN_VREF80 : 0;
+
+ msleep(600);
+ snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+ vref);
+}
+
+static void alc283_chromebook_caps(struct hda_codec *codec)
+{
+ snd_hda_override_wcaps(codec, 0x03, 0);
+}
+
+static void alc283_fixup_chromebook(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ struct alc_spec *spec = codec->spec;
+ int val;
+
+ switch (action) {
+ case HDA_FIXUP_ACT_PRE_PROBE:
+ alc283_chromebook_caps(codec);
+ spec->gen.hp_automute_hook = alc283_hp_automute_hook;
+ /* MIC2-VREF control */
+ /* Set to manual mode */
+ val = alc_read_coef_idx(codec, 0x06);
+ alc_write_coef_idx(codec, 0x06, val & ~0x000c);
+ break;
+ }
+}
+
enum {
ALC269_FIXUP_SONY_VAIO,
ALC275_FIXUP_SONY_VAIO_GPIO2,
@@ -3344,6 +3479,7 @@ enum {
ALC269_FIXUP_ACER_AC700,
ALC269_FIXUP_LIMIT_INT_MIC_BOOST,
ALC269VB_FIXUP_ORDISSIMO_EVE2,
+ ALC283_FIXUP_CHROME_BOOK,
};
static const struct hda_fixup alc269_fixups[] = {
@@ -3595,11 +3731,20 @@ static const struct hda_fixup alc269_fixups[] = {
{ }
},
},
+ [ALC283_FIXUP_CHROME_BOOK] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc283_fixup_chromebook,
+ },
};
static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x029b, "Acer 1810TZ", ALC269_FIXUP_INV_DMIC),
SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC),
+ SND_PCI_QUIRK(0x1025, 0x047c, "Acer AC700", ALC269_FIXUP_ACER_AC700),
+ SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK),
+ SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK),
+ SND_PCI_QUIRK_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC),
+ SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
SND_PCI_QUIRK(0x1028, 0x05bd, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x05be, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x05c4, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
@@ -3637,6 +3782,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x103c, 0x18e6, "HP", ALC269_FIXUP_HP_GPIO_LED),
SND_PCI_QUIRK(0x103c, 0x1973, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1),
SND_PCI_QUIRK(0x103c, 0x1983, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1),
+ SND_PCI_QUIRK(0x103c, 0x21ed, "HP Falco Chromebook", ALC283_FIXUP_CHROME_BOOK),
SND_PCI_QUIRK_VENDOR(0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED),
SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
@@ -3655,11 +3801,6 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO),
- SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
- SND_PCI_QUIRK(0x1025, 0x047c, "Acer AC700", ALC269_FIXUP_ACER_AC700),
- SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK),
- SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK),
- SND_PCI_QUIRK_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC),
SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook", ALC269_FIXUP_LIFEBOOK),
SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE),
SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE),
@@ -3670,8 +3811,16 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x21fa, "Thinkpad X230", ALC269_FIXUP_LENOVO_DOCK),
SND_PCI_QUIRK(0x17aa, 0x21f3, "Thinkpad T430", ALC269_FIXUP_LENOVO_DOCK),
SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK),
- SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK),
SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK),
+ SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK),
+ SND_PCI_QUIRK(0x17aa, 0x220c, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+ SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+ SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+ SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+ SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+ SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+ SND_PCI_QUIRK(0x17aa, 0x5026, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+ SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K),
SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD),
SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */
@@ -3840,11 +3989,15 @@ static int patch_alc269(struct hda_codec *codec)
case 0x10ec0290:
spec->codec_variant = ALC269_TYPE_ALC280;
break;
- case 0x10ec0233:
case 0x10ec0282:
- case 0x10ec0283:
spec->codec_variant = ALC269_TYPE_ALC282;
break;
+ case 0x10ec0233:
+ case 0x10ec0283:
+ spec->codec_variant = ALC269_TYPE_ALC283;
+ spec->shutup = alc283_shutup;
+ spec->init_hook = alc283_init;
+ break;
case 0x10ec0284:
case 0x10ec0292:
spec->codec_variant = ALC269_TYPE_ALC284;
@@ -3872,7 +4025,8 @@ static int patch_alc269(struct hda_codec *codec)
codec->patch_ops.suspend = alc269_suspend;
codec->patch_ops.resume = alc269_resume;
#endif
- spec->shutup = alc269_shutup;
+ if (!spec->shutup)
+ spec->shutup = alc269_shutup;
snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE);
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 6d1924c19abf..fba0cef1c47f 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -158,6 +158,7 @@ enum {
STAC_D965_VERBS,
STAC_DELL_3ST,
STAC_DELL_BIOS,
+ STAC_DELL_BIOS_AMIC,
STAC_DELL_BIOS_SPDIF,
STAC_927X_DELL_DMIC,
STAC_927X_VOLKNOB,
@@ -3231,8 +3232,6 @@ static const struct hda_fixup stac927x_fixups[] = {
[STAC_DELL_BIOS] = {
.type = HDA_FIXUP_PINS,
.v.pins = (const struct hda_pintbl[]) {
- /* configure the analog microphone on some laptops */
- { 0x0c, 0x90a79130 },
/* correct the front output jack as a hp out */
{ 0x0f, 0x0221101f },
/* correct the front input jack as a mic */
@@ -3242,6 +3241,16 @@ static const struct hda_fixup stac927x_fixups[] = {
.chained = true,
.chain_id = STAC_927X_DELL_DMIC,
},
+ [STAC_DELL_BIOS_AMIC] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ /* configure the analog microphone on some laptops */
+ { 0x0c, 0x90a79130 },
+ {}
+ },
+ .chained = true,
+ .chain_id = STAC_DELL_BIOS,
+ },
[STAC_DELL_BIOS_SPDIF] = {
.type = HDA_FIXUP_PINS,
.v.pins = (const struct hda_pintbl[]) {
@@ -3270,6 +3279,7 @@ static const struct hda_model_fixup stac927x_models[] = {
{ .id = STAC_D965_5ST_NO_FP, .name = "5stack-no-fp" },
{ .id = STAC_DELL_3ST, .name = "dell-3stack" },
{ .id = STAC_DELL_BIOS, .name = "dell-bios" },
+ { .id = STAC_DELL_BIOS_AMIC, .name = "dell-bios-amic" },
{ .id = STAC_927X_VOLKNOB, .name = "volknob" },
{}
};
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index e2481baddc70..0bc20ef5687a 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -207,9 +207,9 @@ static void vt1708_stop_hp_work(struct hda_codec *codec)
return;
if (spec->hp_work_active) {
snd_hda_codec_write(codec, 0x1, 0, 0xf81, 1);
+ codec->jackpoll_interval = 0;
cancel_delayed_work_sync(&codec->jackpoll_work);
spec->hp_work_active = false;
- codec->jackpoll_interval = 0;
}
}
diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c
index 2a8ad9d1a2ae..bb9ebc5543d7 100644
--- a/sound/pci/rme96.c
+++ b/sound/pci/rme96.c
@@ -28,6 +28,7 @@
#include <linux/interrupt.h>
#include <linux/pci.h>
#include <linux/module.h>
+#include <linux/vmalloc.h>
#include <sound/core.h>
#include <sound/info.h>
@@ -198,6 +199,31 @@ MODULE_PARM_DESC(enable, "Enable RME Digi96 soundcard.");
#define RME96_AD1852_VOL_BITS 14
#define RME96_AD1855_VOL_BITS 10
+/* Defines for snd_rme96_trigger */
+#define RME96_TB_START_PLAYBACK 1
+#define RME96_TB_START_CAPTURE 2
+#define RME96_TB_STOP_PLAYBACK 4
+#define RME96_TB_STOP_CAPTURE 8
+#define RME96_TB_RESET_PLAYPOS 16
+#define RME96_TB_RESET_CAPTUREPOS 32
+#define RME96_TB_CLEAR_PLAYBACK_IRQ 64
+#define RME96_TB_CLEAR_CAPTURE_IRQ 128
+#define RME96_RESUME_PLAYBACK (RME96_TB_START_PLAYBACK)
+#define RME96_RESUME_CAPTURE (RME96_TB_START_CAPTURE)
+#define RME96_RESUME_BOTH (RME96_RESUME_PLAYBACK \
+ | RME96_RESUME_CAPTURE)
+#define RME96_START_PLAYBACK (RME96_TB_START_PLAYBACK \
+ | RME96_TB_RESET_PLAYPOS)
+#define RME96_START_CAPTURE (RME96_TB_START_CAPTURE \
+ | RME96_TB_RESET_CAPTUREPOS)
+#define RME96_START_BOTH (RME96_START_PLAYBACK \
+ | RME96_START_CAPTURE)
+#define RME96_STOP_PLAYBACK (RME96_TB_STOP_PLAYBACK \
+ | RME96_TB_CLEAR_PLAYBACK_IRQ)
+#define RME96_STOP_CAPTURE (RME96_TB_STOP_CAPTURE \
+ | RME96_TB_CLEAR_CAPTURE_IRQ)
+#define RME96_STOP_BOTH (RME96_STOP_PLAYBACK \
+ | RME96_STOP_CAPTURE)
struct rme96 {
spinlock_t lock;
@@ -214,6 +240,13 @@ struct rme96 {
u8 rev; /* card revision number */
+#ifdef CONFIG_PM
+ u32 playback_pointer;
+ u32 capture_pointer;
+ void *playback_suspend_buffer;
+ void *capture_suspend_buffer;
+#endif
+
struct snd_pcm_substream *playback_substream;
struct snd_pcm_substream *capture_substream;
@@ -344,6 +377,8 @@ static struct snd_pcm_hardware snd_rme96_playback_spdif_info =
{
.info = (SNDRV_PCM_INFO_MMAP_IOMEM |
SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_SYNC_START |
+ SNDRV_PCM_INFO_RESUME |
SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_PAUSE),
.formats = (SNDRV_PCM_FMTBIT_S16_LE |
@@ -373,6 +408,8 @@ static struct snd_pcm_hardware snd_rme96_capture_spdif_info =
{
.info = (SNDRV_PCM_INFO_MMAP_IOMEM |
SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_SYNC_START |
+ SNDRV_PCM_INFO_RESUME |
SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_PAUSE),
.formats = (SNDRV_PCM_FMTBIT_S16_LE |
@@ -402,6 +439,8 @@ static struct snd_pcm_hardware snd_rme96_playback_adat_info =
{
.info = (SNDRV_PCM_INFO_MMAP_IOMEM |
SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_SYNC_START |
+ SNDRV_PCM_INFO_RESUME |
SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_PAUSE),
.formats = (SNDRV_PCM_FMTBIT_S16_LE |
@@ -427,6 +466,8 @@ static struct snd_pcm_hardware snd_rme96_capture_adat_info =
{
.info = (SNDRV_PCM_INFO_MMAP_IOMEM |
SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_SYNC_START |
+ SNDRV_PCM_INFO_RESUME |
SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_PAUSE),
.formats = (SNDRV_PCM_FMTBIT_S16_LE |
@@ -1045,54 +1086,35 @@ snd_rme96_capture_hw_params(struct snd_pcm_substream *substream,
}
static void
-snd_rme96_playback_start(struct rme96 *rme96,
- int from_pause)
+snd_rme96_trigger(struct rme96 *rme96,
+ int op)
{
- if (!from_pause) {
+ if (op & RME96_TB_RESET_PLAYPOS)
writel(0, rme96->iobase + RME96_IO_RESET_PLAY_POS);
- }
-
- rme96->wcreg |= RME96_WCR_START;
- writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER);
-}
-
-static void
-snd_rme96_capture_start(struct rme96 *rme96,
- int from_pause)
-{
- if (!from_pause) {
+ if (op & RME96_TB_RESET_CAPTUREPOS)
writel(0, rme96->iobase + RME96_IO_RESET_REC_POS);
- }
-
- rme96->wcreg |= RME96_WCR_START_2;
+ if (op & RME96_TB_CLEAR_PLAYBACK_IRQ) {
+ rme96->rcreg = readl(rme96->iobase + RME96_IO_CONTROL_REGISTER);
+ if (rme96->rcreg & RME96_RCR_IRQ)
+ writel(0, rme96->iobase + RME96_IO_CONFIRM_PLAY_IRQ);
+ }
+ if (op & RME96_TB_CLEAR_CAPTURE_IRQ) {
+ rme96->rcreg = readl(rme96->iobase + RME96_IO_CONTROL_REGISTER);
+ if (rme96->rcreg & RME96_RCR_IRQ_2)
+ writel(0, rme96->iobase + RME96_IO_CONFIRM_REC_IRQ);
+ }
+ if (op & RME96_TB_START_PLAYBACK)
+ rme96->wcreg |= RME96_WCR_START;
+ if (op & RME96_TB_STOP_PLAYBACK)
+ rme96->wcreg &= ~RME96_WCR_START;
+ if (op & RME96_TB_START_CAPTURE)
+ rme96->wcreg |= RME96_WCR_START_2;
+ if (op & RME96_TB_STOP_CAPTURE)
+ rme96->wcreg &= ~RME96_WCR_START_2;
writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER);
}
-static void
-snd_rme96_playback_stop(struct rme96 *rme96)
-{
- /*
- * Check if there is an unconfirmed IRQ, if so confirm it, or else
- * the hardware will not stop generating interrupts
- */
- rme96->rcreg = readl(rme96->iobase + RME96_IO_CONTROL_REGISTER);
- if (rme96->rcreg & RME96_RCR_IRQ) {
- writel(0, rme96->iobase + RME96_IO_CONFIRM_PLAY_IRQ);
- }
- rme96->wcreg &= ~RME96_WCR_START;
- writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER);
-}
-static void
-snd_rme96_capture_stop(struct rme96 *rme96)
-{
- rme96->rcreg = readl(rme96->iobase + RME96_IO_CONTROL_REGISTER);
- if (rme96->rcreg & RME96_RCR_IRQ_2) {
- writel(0, rme96->iobase + RME96_IO_CONFIRM_REC_IRQ);
- }
- rme96->wcreg &= ~RME96_WCR_START_2;
- writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER);
-}
static irqreturn_t
snd_rme96_interrupt(int irq,
@@ -1155,6 +1177,7 @@ snd_rme96_playback_spdif_open(struct snd_pcm_substream *substream)
struct rme96 *rme96 = snd_pcm_substream_chip(substream);
struct snd_pcm_runtime *runtime = substream->runtime;
+ snd_pcm_set_sync(substream);
spin_lock_irq(&rme96->lock);
if (rme96->playback_substream != NULL) {
spin_unlock_irq(&rme96->lock);
@@ -1191,6 +1214,7 @@ snd_rme96_capture_spdif_open(struct snd_pcm_substream *substream)
struct rme96 *rme96 = snd_pcm_substream_chip(substream);
struct snd_pcm_runtime *runtime = substream->runtime;
+ snd_pcm_set_sync(substream);
runtime->hw = snd_rme96_capture_spdif_info;
if (snd_rme96_getinputtype(rme96) != RME96_INPUT_ANALOG &&
(rate = snd_rme96_capture_getrate(rme96, &isadat)) > 0)
@@ -1222,6 +1246,7 @@ snd_rme96_playback_adat_open(struct snd_pcm_substream *substream)
struct rme96 *rme96 = snd_pcm_substream_chip(substream);
struct snd_pcm_runtime *runtime = substream->runtime;
+ snd_pcm_set_sync(substream);
spin_lock_irq(&rme96->lock);
if (rme96->playback_substream != NULL) {
spin_unlock_irq(&rme96->lock);
@@ -1253,6 +1278,7 @@ snd_rme96_capture_adat_open(struct snd_pcm_substream *substream)
struct rme96 *rme96 = snd_pcm_substream_chip(substream);
struct snd_pcm_runtime *runtime = substream->runtime;
+ snd_pcm_set_sync(substream);
runtime->hw = snd_rme96_capture_adat_info;
if (snd_rme96_getinputtype(rme96) == RME96_INPUT_ANALOG) {
/* makes no sense to use analog input. Note that analog
@@ -1288,7 +1314,7 @@ snd_rme96_playback_close(struct snd_pcm_substream *substream)
spin_lock_irq(&rme96->lock);
if (RME96_ISPLAYING(rme96)) {
- snd_rme96_playback_stop(rme96);
+ snd_rme96_trigger(rme96, RME96_STOP_PLAYBACK);
}
rme96->playback_substream = NULL;
rme96->playback_periodsize = 0;
@@ -1309,7 +1335,7 @@ snd_rme96_capture_close(struct snd_pcm_substream *substream)
spin_lock_irq(&rme96->lock);
if (RME96_ISRECORDING(rme96)) {
- snd_rme96_capture_stop(rme96);
+ snd_rme96_trigger(rme96, RME96_STOP_CAPTURE);
}
rme96->capture_substream = NULL;
rme96->capture_periodsize = 0;
@@ -1324,7 +1350,7 @@ snd_rme96_playback_prepare(struct snd_pcm_substream *substream)
spin_lock_irq(&rme96->lock);
if (RME96_ISPLAYING(rme96)) {
- snd_rme96_playback_stop(rme96);
+ snd_rme96_trigger(rme96, RME96_STOP_PLAYBACK);
}
writel(0, rme96->iobase + RME96_IO_RESET_PLAY_POS);
spin_unlock_irq(&rme96->lock);
@@ -1338,7 +1364,7 @@ snd_rme96_capture_prepare(struct snd_pcm_substream *substream)
spin_lock_irq(&rme96->lock);
if (RME96_ISRECORDING(rme96)) {
- snd_rme96_capture_stop(rme96);
+ snd_rme96_trigger(rme96, RME96_STOP_CAPTURE);
}
writel(0, rme96->iobase + RME96_IO_RESET_REC_POS);
spin_unlock_irq(&rme96->lock);
@@ -1350,41 +1376,55 @@ snd_rme96_playback_trigger(struct snd_pcm_substream *substream,
int cmd)
{
struct rme96 *rme96 = snd_pcm_substream_chip(substream);
+ struct snd_pcm_substream *s;
+ bool sync;
+
+ snd_pcm_group_for_each_entry(s, substream) {
+ if (snd_pcm_substream_chip(s) == rme96)
+ snd_pcm_trigger_done(s, substream);
+ }
+
+ sync = (rme96->playback_substream && rme96->capture_substream) &&
+ (rme96->playback_substream->group ==
+ rme96->capture_substream->group);
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
if (!RME96_ISPLAYING(rme96)) {
- if (substream != rme96->playback_substream) {
+ if (substream != rme96->playback_substream)
return -EBUSY;
- }
- snd_rme96_playback_start(rme96, 0);
+ snd_rme96_trigger(rme96, sync ? RME96_START_BOTH
+ : RME96_START_PLAYBACK);
}
break;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_STOP:
if (RME96_ISPLAYING(rme96)) {
- if (substream != rme96->playback_substream) {
+ if (substream != rme96->playback_substream)
return -EBUSY;
- }
- snd_rme96_playback_stop(rme96);
+ snd_rme96_trigger(rme96, sync ? RME96_STOP_BOTH
+ : RME96_STOP_PLAYBACK);
}
break;
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
- if (RME96_ISPLAYING(rme96)) {
- snd_rme96_playback_stop(rme96);
- }
+ if (RME96_ISPLAYING(rme96))
+ snd_rme96_trigger(rme96, sync ? RME96_STOP_BOTH
+ : RME96_STOP_PLAYBACK);
break;
+ case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- if (!RME96_ISPLAYING(rme96)) {
- snd_rme96_playback_start(rme96, 1);
- }
+ if (!RME96_ISPLAYING(rme96))
+ snd_rme96_trigger(rme96, sync ? RME96_RESUME_BOTH
+ : RME96_RESUME_PLAYBACK);
break;
-
+
default:
return -EINVAL;
}
+
return 0;
}
@@ -1393,38 +1433,51 @@ snd_rme96_capture_trigger(struct snd_pcm_substream *substream,
int cmd)
{
struct rme96 *rme96 = snd_pcm_substream_chip(substream);
+ struct snd_pcm_substream *s;
+ bool sync;
+
+ snd_pcm_group_for_each_entry(s, substream) {
+ if (snd_pcm_substream_chip(s) == rme96)
+ snd_pcm_trigger_done(s, substream);
+ }
+
+ sync = (rme96->playback_substream && rme96->capture_substream) &&
+ (rme96->playback_substream->group ==
+ rme96->capture_substream->group);
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
if (!RME96_ISRECORDING(rme96)) {
- if (substream != rme96->capture_substream) {
+ if (substream != rme96->capture_substream)
return -EBUSY;
- }
- snd_rme96_capture_start(rme96, 0);
+ snd_rme96_trigger(rme96, sync ? RME96_START_BOTH
+ : RME96_START_CAPTURE);
}
break;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_STOP:
if (RME96_ISRECORDING(rme96)) {
- if (substream != rme96->capture_substream) {
+ if (substream != rme96->capture_substream)
return -EBUSY;
- }
- snd_rme96_capture_stop(rme96);
+ snd_rme96_trigger(rme96, sync ? RME96_STOP_BOTH
+ : RME96_STOP_CAPTURE);
}
break;
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
- if (RME96_ISRECORDING(rme96)) {
- snd_rme96_capture_stop(rme96);
- }
+ if (RME96_ISRECORDING(rme96))
+ snd_rme96_trigger(rme96, sync ? RME96_STOP_BOTH
+ : RME96_STOP_CAPTURE);
break;
+ case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- if (!RME96_ISRECORDING(rme96)) {
- snd_rme96_capture_start(rme96, 1);
- }
+ if (!RME96_ISRECORDING(rme96))
+ snd_rme96_trigger(rme96, sync ? RME96_RESUME_BOTH
+ : RME96_RESUME_CAPTURE);
break;
-
+
default:
return -EINVAL;
}
@@ -1505,8 +1558,7 @@ snd_rme96_free(void *private_data)
return;
}
if (rme96->irq >= 0) {
- snd_rme96_playback_stop(rme96);
- snd_rme96_capture_stop(rme96);
+ snd_rme96_trigger(rme96, RME96_STOP_BOTH);
rme96->areg &= ~RME96_AR_DAC_EN;
writel(rme96->areg, rme96->iobase + RME96_IO_ADDITIONAL_REG);
free_irq(rme96->irq, (void *)rme96);
@@ -1520,6 +1572,10 @@ snd_rme96_free(void *private_data)
pci_release_regions(rme96->pci);
rme96->port = 0;
}
+#ifdef CONFIG_PM
+ vfree(rme96->playback_suspend_buffer);
+ vfree(rme96->capture_suspend_buffer);
+#endif
pci_disable_device(rme96->pci);
}
@@ -1606,8 +1662,7 @@ snd_rme96_create(struct rme96 *rme96)
rme96->capture_periodsize = 0;
/* make sure playback/capture is stopped, if by some reason active */
- snd_rme96_playback_stop(rme96);
- snd_rme96_capture_stop(rme96);
+ snd_rme96_trigger(rme96, RME96_STOP_BOTH);
/* set default values in registers */
rme96->wcreg =
@@ -2319,6 +2374,87 @@ snd_rme96_create_switches(struct snd_card *card,
* Card initialisation
*/
+#ifdef CONFIG_PM
+
+static int
+snd_rme96_suspend(struct pci_dev *pci,
+ pm_message_t state)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct rme96 *rme96 = card->private_data;
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+ snd_pcm_suspend(rme96->playback_substream);
+ snd_pcm_suspend(rme96->capture_substream);
+
+ /* save capture & playback pointers */
+ rme96->playback_pointer = readl(rme96->iobase + RME96_IO_GET_PLAY_POS)
+ & RME96_RCR_AUDIO_ADDR_MASK;
+ rme96->capture_pointer = readl(rme96->iobase + RME96_IO_GET_REC_POS)
+ & RME96_RCR_AUDIO_ADDR_MASK;
+
+ /* save playback and capture buffers */
+ memcpy_fromio(rme96->playback_suspend_buffer,
+ rme96->iobase + RME96_IO_PLAY_BUFFER, RME96_BUFFER_SIZE);
+ memcpy_fromio(rme96->capture_suspend_buffer,
+ rme96->iobase + RME96_IO_REC_BUFFER, RME96_BUFFER_SIZE);
+
+ /* disable the DAC */
+ rme96->areg &= ~RME96_AR_DAC_EN;
+ writel(rme96->areg, rme96->iobase + RME96_IO_ADDITIONAL_REG);
+
+ pci_disable_device(pci);
+ pci_save_state(pci);
+
+ return 0;
+}
+
+static int
+snd_rme96_resume(struct pci_dev *pci)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct rme96 *rme96 = card->private_data;
+
+ pci_restore_state(pci);
+ if (pci_enable_device(pci) < 0) {
+ printk(KERN_ERR "rme96: pci_enable_device failed, disabling device\n");
+ snd_card_disconnect(card);
+ return -EIO;
+ }
+
+ /* reset playback and record buffer pointers */
+ writel(0, rme96->iobase + RME96_IO_SET_PLAY_POS
+ + rme96->playback_pointer);
+ writel(0, rme96->iobase + RME96_IO_SET_REC_POS
+ + rme96->capture_pointer);
+
+ /* restore playback and capture buffers */
+ memcpy_toio(rme96->iobase + RME96_IO_PLAY_BUFFER,
+ rme96->playback_suspend_buffer, RME96_BUFFER_SIZE);
+ memcpy_toio(rme96->iobase + RME96_IO_REC_BUFFER,
+ rme96->capture_suspend_buffer, RME96_BUFFER_SIZE);
+
+ /* reset the ADC */
+ writel(rme96->areg | RME96_AR_PD2,
+ rme96->iobase + RME96_IO_ADDITIONAL_REG);
+ writel(rme96->areg, rme96->iobase + RME96_IO_ADDITIONAL_REG);
+
+ /* reset and enable DAC, restore analog volume */
+ snd_rme96_reset_dac(rme96);
+ rme96->areg |= RME96_AR_DAC_EN;
+ writel(rme96->areg, rme96->iobase + RME96_IO_ADDITIONAL_REG);
+ if (RME96_HAS_ANALOG_OUT(rme96)) {
+ usleep_range(3000, 10000);
+ snd_rme96_apply_dac_volume(rme96);
+ }
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+
+ return 0;
+}
+
+#endif
+
static void snd_rme96_card_free(struct snd_card *card)
{
snd_rme96_free(card->private_data);
@@ -2355,6 +2491,23 @@ snd_rme96_probe(struct pci_dev *pci,
return err;
}
+#ifdef CONFIG_PM
+ rme96->playback_suspend_buffer = vmalloc(RME96_BUFFER_SIZE);
+ if (!rme96->playback_suspend_buffer) {
+ snd_printk(KERN_ERR
+ "Failed to allocate playback suspend buffer!\n");
+ snd_card_free(card);
+ return -ENOMEM;
+ }
+ rme96->capture_suspend_buffer = vmalloc(RME96_BUFFER_SIZE);
+ if (!rme96->capture_suspend_buffer) {
+ snd_printk(KERN_ERR
+ "Failed to allocate capture suspend buffer!\n");
+ snd_card_free(card);
+ return -ENOMEM;
+ }
+#endif
+
strcpy(card->driver, "Digi96");
switch (rme96->pci->device) {
case PCI_DEVICE_ID_RME_DIGI96:
@@ -2397,6 +2550,10 @@ static struct pci_driver rme96_driver = {
.id_table = snd_rme96_ids,
.probe = snd_rme96_probe,
.remove = snd_rme96_remove,
+#ifdef CONFIG_PM
+ .suspend = snd_rme96_suspend,
+ .resume = snd_rme96_resume,
+#endif
};
module_pci_driver(rme96_driver);
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index bd501931ee23..3cde55b753e2 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -38,6 +38,97 @@
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*
*/
+
+/* ************* Register Documentation *******************************************************
+ *
+ * Work in progress! Documentation is based on the code in this file.
+ *
+ * --------- HDSPM_controlRegister ---------
+ * :7654.3210:7654.3210:7654.3210:7654.3210: bit number per byte
+ * :||||.||||:||||.||||:||||.||||:||||.||||:
+ * :3322.2222:2222.1111:1111.1100:0000.0000: bit number
+ * :1098.7654:3210.9876:5432.1098:7654.3210: 0..31
+ * :||||.||||:||||.||||:||||.||||:||||.||||:
+ * :8421.8421:8421.8421:8421.8421:8421.8421: hex digit
+ * : . : . : . : x . : HDSPM_AudioInterruptEnable \_ setting both bits
+ * : . : . : . : . x: HDSPM_Start / enables audio IO
+ * : . : . : . : x. : HDSPM_ClockModeMaster - 1: Master, 0: Slave
+ * : . : . : . : .210 : HDSPM_LatencyMask - 3 Bit value for latency
+ * : . : . : . : . : 0:64, 1:128, 2:256, 3:512,
+ * : . : . : . : . : 4:1024, 5:2048, 6:4096, 7:8192
+ * :x . : . : . x:xx . : HDSPM_FrequencyMask
+ * : . : . : . :10 . : HDSPM_Frequency1|HDSPM_Frequency0: 1=32K,2=44.1K,3=48K,0=??
+ * : . : . : . x: . : <MADI> HDSPM_DoubleSpeed
+ * :x . : . : . : . : <MADI> HDSPM_QuadSpeed
+ * : . 3 : . 10: 2 . : . : HDSPM_SyncRefMask :
+ * : . : . x: . : . : HDSPM_SyncRef0
+ * : . : . x : . : . : HDSPM_SyncRef1
+ * : . : . : x . : . : <AES32> HDSPM_SyncRef2
+ * : . x : . : . : . : <AES32> HDSPM_SyncRef3
+ * : . : . 10: . : . : <MADI> sync ref: 0:WC, 1:Madi, 2:TCO, 3:SyncIn
+ * : . 3 : . 10: 2 . : . : <AES32> 0:WC, 1:AES1 ... 8:AES8, 9: TCO, 10:SyncIn?
+ * : . x : . : . : . : <MADIe> HDSPe_FLOAT_FORMAT
+ * : . : . : x . : . : <MADI> HDSPM_InputSelect0 : 0=optical,1=coax
+ * : . : . :x . : . : <MADI> HDSPM_InputSelect1
+ * : . : .x : . : . : <MADI> HDSPM_clr_tms
+ * : . : . : . x : . : <MADI> HDSPM_TX_64ch
+ * : . : . : . x : . : <AES32> HDSPM_Emphasis
+ * : . : . : .x : . : <MADI> HDSPM_AutoInp
+ * : . : . x : . : . : <MADI> HDSPM_SMUX
+ * : . : .x : . : . : <MADI> HDSPM_clr_tms
+ * : . : x. : . : . : <MADI> HDSPM_taxi_reset
+ * : . x: . : . : . : <MADI> HDSPM_LineOut
+ * : . x: . : . : . : <AES32> ??????????????????
+ * : . : x. : . : . : <AES32> HDSPM_WCK48
+ * : . : . : .x : . : <AES32> HDSPM_Dolby
+ * : . : x . : . : . : HDSPM_Midi0InterruptEnable
+ * : . :x . : . : . : HDSPM_Midi1InterruptEnable
+ * : . : x . : . : . : HDSPM_Midi2InterruptEnable
+ * : . x : . : . : . : <MADI> HDSPM_Midi3InterruptEnable
+ * : . x : . : . : . : <AES32> HDSPM_DS_DoubleWire
+ * : .x : . : . : . : <AES32> HDSPM_QS_DoubleWire
+ * : x. : . : . : . : <AES32> HDSPM_QS_QuadWire
+ * : . : . : . x : . : <AES32> HDSPM_Professional
+ * : x . : . : . : . : HDSPM_wclk_sel
+ * : . : . : . : . :
+ * :7654.3210:7654.3210:7654.3210:7654.3210: bit number per byte
+ * :||||.||||:||||.||||:||||.||||:||||.||||:
+ * :3322.2222:2222.1111:1111.1100:0000.0000: bit number
+ * :1098.7654:3210.9876:5432.1098:7654.3210: 0..31
+ * :||||.||||:||||.||||:||||.||||:||||.||||:
+ * :8421.8421:8421.8421:8421.8421:8421.8421:hex digit
+ *
+ *
+ *
+ * AIO / RayDAT only
+ *
+ * ------------ HDSPM_WR_SETTINGS ----------
+ * :3322.2222:2222.1111:1111.1100:0000.0000: bit number per byte
+ * :1098.7654:3210.9876:5432.1098:7654.3210:
+ * :||||.||||:||||.||||:||||.||||:||||.||||: bit number
+ * :7654.3210:7654.3210:7654.3210:7654.3210: 0..31
+ * :||||.||||:||||.||||:||||.||||:||||.||||:
+ * :8421.8421:8421.8421:8421.8421:8421.8421: hex digit
+ * : . : . : . : . x: HDSPM_c0Master 1: Master, 0: Slave
+ * : . : . : . : . x : HDSPM_c0_SyncRef0
+ * : . : . : . : . x : HDSPM_c0_SyncRef1
+ * : . : . : . : .x : HDSPM_c0_SyncRef2
+ * : . : . : . : x. : HDSPM_c0_SyncRef3
+ * : . : . : . : 3.210 : HDSPM_c0_SyncRefMask:
+ * : . : . : . : . : RayDat: 0:WC, 1:AES, 2:SPDIF, 3..6: ADAT1..4,
+ * : . : . : . : . : 9:TCO, 10:SyncIn
+ * : . : . : . : . : AIO: 0:WC, 1:AES, 2: SPDIF, 3: ATAT,
+ * : . : . : . : . : 9:TCO, 10:SyncIn
+ * : . : . : . : . :
+ * : . : . : . : . :
+ * :3322.2222:2222.1111:1111.1100:0000.0000: bit number per byte
+ * :1098.7654:3210.9876:5432.1098:7654.3210:
+ * :||||.||||:||||.||||:||||.||||:||||.||||: bit number
+ * :7654.3210:7654.3210:7654.3210:7654.3210: 0..31
+ * :||||.||||:||||.||||:||||.||||:||||.||||:
+ * :8421.8421:8421.8421:8421.8421:8421.8421: hex digit
+ *
+ */
#include <linux/init.h>
#include <linux/delay.h>
#include <linux/interrupt.h>
@@ -95,7 +186,7 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}");
#define HDSPM_controlRegister 64
#define HDSPM_interruptConfirmation 96
#define HDSPM_control2Reg 256 /* not in specs ???????? */
-#define HDSPM_freqReg 256 /* for AES32 */
+#define HDSPM_freqReg 256 /* for setting arbitrary clock values (DDS feature) */
#define HDSPM_midiDataOut0 352 /* just believe in old code */
#define HDSPM_midiDataOut1 356
#define HDSPM_eeprom_wr 384 /* for AES32 */
@@ -258,6 +349,25 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}");
#define HDSPM_wclk_sel (1<<30)
+/* additional control register bits for AIO*/
+#define HDSPM_c0_Wck48 0x20 /* also RayDAT */
+#define HDSPM_c0_Input0 0x1000
+#define HDSPM_c0_Input1 0x2000
+#define HDSPM_c0_Spdif_Opt 0x4000
+#define HDSPM_c0_Pro 0x8000
+#define HDSPM_c0_clr_tms 0x10000
+#define HDSPM_c0_AEB1 0x20000
+#define HDSPM_c0_AEB2 0x40000
+#define HDSPM_c0_LineOut 0x80000
+#define HDSPM_c0_AD_GAIN0 0x100000
+#define HDSPM_c0_AD_GAIN1 0x200000
+#define HDSPM_c0_DA_GAIN0 0x400000
+#define HDSPM_c0_DA_GAIN1 0x800000
+#define HDSPM_c0_PH_GAIN0 0x1000000
+#define HDSPM_c0_PH_GAIN1 0x2000000
+#define HDSPM_c0_Sym6db 0x4000000
+
+
/* --- bit helper defines */
#define HDSPM_LatencyMask (HDSPM_Latency0|HDSPM_Latency1|HDSPM_Latency2)
#define HDSPM_FrequencyMask (HDSPM_Frequency0|HDSPM_Frequency1|\
@@ -341,11 +451,11 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}");
#define HDSPM_madiLock (1<<3) /* MADI Locked =1, no=0 */
#define HDSPM_madiSync (1<<18) /* MADI is in sync */
-#define HDSPM_tcoLock 0x00000020 /* Optional TCO locked status FOR HDSPe MADI! */
-#define HDSPM_tcoSync 0x10000000 /* Optional TCO sync status */
+#define HDSPM_tcoLockMadi 0x00000020 /* Optional TCO locked status for HDSPe MADI*/
+#define HDSPM_tcoSync 0x10000000 /* Optional TCO sync status for HDSPe MADI and AES32!*/
-#define HDSPM_syncInLock 0x00010000 /* Sync In lock status FOR HDSPe MADI! */
-#define HDSPM_syncInSync 0x00020000 /* Sync In sync status FOR HDSPe MADI! */
+#define HDSPM_syncInLock 0x00010000 /* Sync In lock status for HDSPe MADI! */
+#define HDSPM_syncInSync 0x00020000 /* Sync In sync status for HDSPe MADI! */
#define HDSPM_BufferPositionMask 0x000FFC0 /* Bit 6..15 : h/w buffer pointer */
/* since 64byte accurate, last 6 bits are not used */
@@ -363,7 +473,7 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}");
* Interrupt
*/
#define HDSPM_tco_detect 0x08000000
-#define HDSPM_tco_lock 0x20000000
+#define HDSPM_tcoLockAes 0x20000000 /* Optional TCO locked status for HDSPe AES */
#define HDSPM_s2_tco_detect 0x00000040
#define HDSPM_s2_AEBO_D 0x00000080
@@ -461,7 +571,9 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}");
#define HDSPM_AES32_AUTOSYNC_FROM_AES6 6
#define HDSPM_AES32_AUTOSYNC_FROM_AES7 7
#define HDSPM_AES32_AUTOSYNC_FROM_AES8 8
-#define HDSPM_AES32_AUTOSYNC_FROM_NONE 9
+#define HDSPM_AES32_AUTOSYNC_FROM_TCO 9
+#define HDSPM_AES32_AUTOSYNC_FROM_SYNC_IN 10
+#define HDSPM_AES32_AUTOSYNC_FROM_NONE 11
/* status2 */
/* HDSPM_LockAES_bit is given by HDSPM_LockAES >> (AES# - 1) */
@@ -537,36 +649,39 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}");
/* names for speed modes */
static char *hdspm_speed_names[] = { "single", "double", "quad" };
-static char *texts_autosync_aes_tco[] = { "Word Clock",
+static const char *const texts_autosync_aes_tco[] = { "Word Clock",
"AES1", "AES2", "AES3", "AES4",
"AES5", "AES6", "AES7", "AES8",
- "TCO" };
-static char *texts_autosync_aes[] = { "Word Clock",
+ "TCO", "Sync In"
+};
+static const char *const texts_autosync_aes[] = { "Word Clock",
"AES1", "AES2", "AES3", "AES4",
- "AES5", "AES6", "AES7", "AES8" };
-static char *texts_autosync_madi_tco[] = { "Word Clock",
+ "AES5", "AES6", "AES7", "AES8",
+ "Sync In"
+};
+static const char *const texts_autosync_madi_tco[] = { "Word Clock",
"MADI", "TCO", "Sync In" };
-static char *texts_autosync_madi[] = { "Word Clock",
+static const char *const texts_autosync_madi[] = { "Word Clock",
"MADI", "Sync In" };
-static char *texts_autosync_raydat_tco[] = {
+static const char *const texts_autosync_raydat_tco[] = {
"Word Clock",
"ADAT 1", "ADAT 2", "ADAT 3", "ADAT 4",
"AES", "SPDIF", "TCO", "Sync In"
};
-static char *texts_autosync_raydat[] = {
+static const char *const texts_autosync_raydat[] = {
"Word Clock",
"ADAT 1", "ADAT 2", "ADAT 3", "ADAT 4",
"AES", "SPDIF", "Sync In"
};
-static char *texts_autosync_aio_tco[] = {
+static const char *const texts_autosync_aio_tco[] = {
"Word Clock",
"ADAT", "AES", "SPDIF", "TCO", "Sync In"
};
-static char *texts_autosync_aio[] = { "Word Clock",
+static const char *const texts_autosync_aio[] = { "Word Clock",
"ADAT", "AES", "SPDIF", "Sync In" };
-static char *texts_freq[] = {
+static const char *const texts_freq[] = {
"No Lock",
"32 kHz",
"44.1 kHz",
@@ -629,7 +744,8 @@ static char *texts_ports_aio_in_ss[] = {
"AES.L", "AES.R",
"SPDIF.L", "SPDIF.R",
"ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4", "ADAT.5", "ADAT.6",
- "ADAT.7", "ADAT.8"
+ "ADAT.7", "ADAT.8",
+ "AEB.1", "AEB.2", "AEB.3", "AEB.4"
};
static char *texts_ports_aio_out_ss[] = {
@@ -638,14 +754,16 @@ static char *texts_ports_aio_out_ss[] = {
"SPDIF.L", "SPDIF.R",
"ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4", "ADAT.5", "ADAT.6",
"ADAT.7", "ADAT.8",
- "Phone.L", "Phone.R"
+ "Phone.L", "Phone.R",
+ "AEB.1", "AEB.2", "AEB.3", "AEB.4"
};
static char *texts_ports_aio_in_ds[] = {
"Analogue.L", "Analogue.R",
"AES.L", "AES.R",
"SPDIF.L", "SPDIF.R",
- "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4"
+ "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4",
+ "AEB.1", "AEB.2", "AEB.3", "AEB.4"
};
static char *texts_ports_aio_out_ds[] = {
@@ -653,14 +771,16 @@ static char *texts_ports_aio_out_ds[] = {
"AES.L", "AES.R",
"SPDIF.L", "SPDIF.R",
"ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4",
- "Phone.L", "Phone.R"
+ "Phone.L", "Phone.R",
+ "AEB.1", "AEB.2", "AEB.3", "AEB.4"
};
static char *texts_ports_aio_in_qs[] = {
"Analogue.L", "Analogue.R",
"AES.L", "AES.R",
"SPDIF.L", "SPDIF.R",
- "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4"
+ "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4",
+ "AEB.1", "AEB.2", "AEB.3", "AEB.4"
};
static char *texts_ports_aio_out_qs[] = {
@@ -668,7 +788,8 @@ static char *texts_ports_aio_out_qs[] = {
"AES.L", "AES.R",
"SPDIF.L", "SPDIF.R",
"ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4",
- "Phone.L", "Phone.R"
+ "Phone.L", "Phone.R",
+ "AEB.1", "AEB.2", "AEB.3", "AEB.4"
};
static char *texts_ports_aes32[] = {
@@ -745,8 +866,8 @@ static char channel_map_aio_in_ss[HDSPM_MAX_CHANNELS] = {
8, 9, /* aes in, */
10, 11, /* spdif in */
12, 13, 14, 15, 16, 17, 18, 19, /* ADAT in */
- -1, -1,
- -1, -1, -1, -1, -1, -1, -1, -1,
+ 2, 3, 4, 5, /* AEB */
+ -1, -1, -1, -1, -1, -1,
-1, -1, -1, -1, -1, -1, -1, -1,
-1, -1, -1, -1, -1, -1, -1, -1,
-1, -1, -1, -1, -1, -1, -1, -1,
@@ -760,7 +881,8 @@ static char channel_map_aio_out_ss[HDSPM_MAX_CHANNELS] = {
10, 11, /* spdif out */
12, 13, 14, 15, 16, 17, 18, 19, /* ADAT out */
6, 7, /* phone out */
- -1, -1, -1, -1, -1, -1, -1, -1,
+ 2, 3, 4, 5, /* AEB */
+ -1, -1, -1, -1,
-1, -1, -1, -1, -1, -1, -1, -1,
-1, -1, -1, -1, -1, -1, -1, -1,
-1, -1, -1, -1, -1, -1, -1, -1,
@@ -773,7 +895,8 @@ static char channel_map_aio_in_ds[HDSPM_MAX_CHANNELS] = {
8, 9, /* aes in */
10, 11, /* spdif in */
12, 14, 16, 18, /* adat in */
- -1, -1, -1, -1, -1, -1,
+ 2, 3, 4, 5, /* AEB */
+ -1, -1,
-1, -1, -1, -1, -1, -1, -1, -1,
-1, -1, -1, -1, -1, -1, -1, -1,
-1, -1, -1, -1, -1, -1, -1, -1,
@@ -788,7 +911,7 @@ static char channel_map_aio_out_ds[HDSPM_MAX_CHANNELS] = {
10, 11, /* spdif out */
12, 14, 16, 18, /* adat out */
6, 7, /* phone out */
- -1, -1, -1, -1,
+ 2, 3, 4, 5, /* AEB */
-1, -1, -1, -1, -1, -1, -1, -1,
-1, -1, -1, -1, -1, -1, -1, -1,
-1, -1, -1, -1, -1, -1, -1, -1,
@@ -802,7 +925,8 @@ static char channel_map_aio_in_qs[HDSPM_MAX_CHANNELS] = {
8, 9, /* aes in */
10, 11, /* spdif in */
12, 16, /* adat in */
- -1, -1, -1, -1, -1, -1, -1, -1,
+ 2, 3, 4, 5, /* AEB */
+ -1, -1, -1, -1,
-1, -1, -1, -1, -1, -1, -1, -1,
-1, -1, -1, -1, -1, -1, -1, -1,
-1, -1, -1, -1, -1, -1, -1, -1,
@@ -817,7 +941,8 @@ static char channel_map_aio_out_qs[HDSPM_MAX_CHANNELS] = {
10, 11, /* spdif out */
12, 16, /* adat out */
6, 7, /* phone out */
- -1, -1, -1, -1, -1, -1,
+ 2, 3, 4, 5, /* AEB */
+ -1, -1,
-1, -1, -1, -1, -1, -1, -1, -1,
-1, -1, -1, -1, -1, -1, -1, -1,
-1, -1, -1, -1, -1, -1, -1, -1,
@@ -856,11 +981,11 @@ struct hdspm_midi {
};
struct hdspm_tco {
- int input;
- int framerate;
- int wordclock;
- int samplerate;
- int pull;
+ int input; /* 0: LTC, 1:Video, 2: WC*/
+ int framerate; /* 0=24, 1=25, 2=29.97, 3=29.97d, 4=30, 5=30d */
+ int wordclock; /* 0=1:1, 1=44.1->48, 2=48->44.1 */
+ int samplerate; /* 0=44.1, 1=48, 2= freq from app */
+ int pull; /* 0=0, 1=+0.1%, 2=-0.1%, 3=+4%, 4=-4%*/
int term; /* 0 = off, 1 = on */
};
@@ -879,7 +1004,7 @@ struct hdspm {
u32 control_register; /* cached value */
u32 control2_register; /* cached value */
- u32 settings_register;
+ u32 settings_register; /* cached value for AIO / RayDat (sync reference, master/slave) */
struct hdspm_midi midi[4];
struct tasklet_struct midi_tasklet;
@@ -941,7 +1066,7 @@ struct hdspm {
struct hdspm_tco *tco; /* NULL if no TCO detected */
- char **texts_autosync;
+ const char *const *texts_autosync;
int texts_autosync_items;
cycles_t last_interrupt;
@@ -976,12 +1101,24 @@ static inline void snd_hdspm_initialize_midi_flush(struct hdspm *hdspm);
static inline int hdspm_get_pll_freq(struct hdspm *hdspm);
static int hdspm_update_simple_mixer_controls(struct hdspm *hdspm);
static int hdspm_autosync_ref(struct hdspm *hdspm);
+static int hdspm_set_toggle_setting(struct hdspm *hdspm, u32 regmask, int out);
static int snd_hdspm_set_defaults(struct hdspm *hdspm);
static int hdspm_system_clock_mode(struct hdspm *hdspm);
static void hdspm_set_sgbuf(struct hdspm *hdspm,
struct snd_pcm_substream *substream,
unsigned int reg, int channels);
+static int hdspm_aes_sync_check(struct hdspm *hdspm, int idx);
+static int hdspm_wc_sync_check(struct hdspm *hdspm);
+static int hdspm_tco_sync_check(struct hdspm *hdspm);
+static int hdspm_sync_in_sync_check(struct hdspm *hdspm);
+
+static int hdspm_get_aes_sample_rate(struct hdspm *hdspm, int index);
+static int hdspm_get_tco_sample_rate(struct hdspm *hdspm);
+static int hdspm_get_wc_sample_rate(struct hdspm *hdspm);
+
+
+
static inline int HDSPM_bit2freq(int n)
{
static const int bit2freq_tab[] = {
@@ -992,6 +1129,12 @@ static inline int HDSPM_bit2freq(int n)
return bit2freq_tab[n];
}
+static bool hdspm_is_raydat_or_aio(struct hdspm *hdspm)
+{
+ return ((AIO == hdspm->io_type) || (RayDAT == hdspm->io_type));
+}
+
+
/* Write/read to/from HDSPM with Adresses in Bytes
not words but only 32Bit writes are allowed */
@@ -1107,14 +1250,11 @@ static int hdspm_rate_multiplier(struct hdspm *hdspm, int rate)
else if (hdspm->control_register &
HDSPM_DoubleSpeed)
return rate * 2;
- };
+ }
return rate;
}
-static int hdspm_tco_sync_check(struct hdspm *hdspm);
-static int hdspm_sync_in_sync_check(struct hdspm *hdspm);
-
-/* check for external sample rate */
+/* check for external sample rate, returns the sample rate in Hz*/
static int hdspm_external_sample_rate(struct hdspm *hdspm)
{
unsigned int status, status2, timecode;
@@ -1127,17 +1267,36 @@ static int hdspm_external_sample_rate(struct hdspm *hdspm)
timecode = hdspm_read(hdspm, HDSPM_timecodeRegister);
syncref = hdspm_autosync_ref(hdspm);
+ switch (syncref) {
+ case HDSPM_AES32_AUTOSYNC_FROM_WORD:
+ /* Check WC sync and get sample rate */
+ if (hdspm_wc_sync_check(hdspm))
+ return HDSPM_bit2freq(hdspm_get_wc_sample_rate(hdspm));
+ break;
- if (syncref == HDSPM_AES32_AUTOSYNC_FROM_WORD &&
- status & HDSPM_AES32_wcLock)
- return HDSPM_bit2freq((status >> HDSPM_AES32_wcFreq_bit) & 0xF);
+ case HDSPM_AES32_AUTOSYNC_FROM_AES1:
+ case HDSPM_AES32_AUTOSYNC_FROM_AES2:
+ case HDSPM_AES32_AUTOSYNC_FROM_AES3:
+ case HDSPM_AES32_AUTOSYNC_FROM_AES4:
+ case HDSPM_AES32_AUTOSYNC_FROM_AES5:
+ case HDSPM_AES32_AUTOSYNC_FROM_AES6:
+ case HDSPM_AES32_AUTOSYNC_FROM_AES7:
+ case HDSPM_AES32_AUTOSYNC_FROM_AES8:
+ /* Check AES sync and get sample rate */
+ if (hdspm_aes_sync_check(hdspm, syncref - HDSPM_AES32_AUTOSYNC_FROM_AES1))
+ return HDSPM_bit2freq(hdspm_get_aes_sample_rate(hdspm,
+ syncref - HDSPM_AES32_AUTOSYNC_FROM_AES1));
+ break;
- if (syncref >= HDSPM_AES32_AUTOSYNC_FROM_AES1 &&
- syncref <= HDSPM_AES32_AUTOSYNC_FROM_AES8 &&
- status2 & (HDSPM_LockAES >>
- (syncref - HDSPM_AES32_AUTOSYNC_FROM_AES1)))
- return HDSPM_bit2freq((timecode >> (4*(syncref-HDSPM_AES32_AUTOSYNC_FROM_AES1))) & 0xF);
- return 0;
+
+ case HDSPM_AES32_AUTOSYNC_FROM_TCO:
+ /* Check TCO sync and get sample rate */
+ if (hdspm_tco_sync_check(hdspm))
+ return HDSPM_bit2freq(hdspm_get_tco_sample_rate(hdspm));
+ break;
+ default:
+ return 0;
+ } /* end switch(syncref) */
break;
case MADIface:
@@ -2129,6 +2288,9 @@ static int hdspm_get_wc_sample_rate(struct hdspm *hdspm)
status = hdspm_read(hdspm, HDSPM_RD_STATUS_1);
return (status >> 16) & 0xF;
break;
+ case AES32:
+ status = hdspm_read(hdspm, HDSPM_statusRegister);
+ return (status >> HDSPM_AES32_wcFreq_bit) & 0xF;
default:
break;
}
@@ -2152,6 +2314,9 @@ static int hdspm_get_tco_sample_rate(struct hdspm *hdspm)
status = hdspm_read(hdspm, HDSPM_RD_STATUS_1);
return (status >> 20) & 0xF;
break;
+ case AES32:
+ status = hdspm_read(hdspm, HDSPM_statusRegister);
+ return (status >> 1) & 0xF;
default:
break;
}
@@ -2183,6 +2348,23 @@ static int hdspm_get_sync_in_sample_rate(struct hdspm *hdspm)
return 0;
}
+/**
+ * Returns the AES sample rate class for the given card.
+ **/
+static int hdspm_get_aes_sample_rate(struct hdspm *hdspm, int index)
+{
+ int timecode;
+
+ switch (hdspm->io_type) {
+ case AES32:
+ timecode = hdspm_read(hdspm, HDSPM_timecodeRegister);
+ return (timecode >> (4*index)) & 0xF;
+ break;
+ default:
+ break;
+ }
+ return 0;
+}
/**
* Returns the sample rate class for input source <idx> for
@@ -2196,15 +2378,23 @@ static int hdspm_get_s1_sample_rate(struct hdspm *hdspm, unsigned int idx)
}
#define ENUMERATED_CTL_INFO(info, texts) \
-{ \
- uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; \
- uinfo->count = 1; \
- uinfo->value.enumerated.items = ARRAY_SIZE(texts); \
- if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) \
- uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; \
- strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); \
-}
+ snd_ctl_enum_info(info, 1, ARRAY_SIZE(texts), texts)
+
+/* Helper function to query the external sample rate and return the
+ * corresponding enum to be returned to userspace.
+ */
+static int hdspm_external_rate_to_enum(struct hdspm *hdspm)
+{
+ int rate = hdspm_external_sample_rate(hdspm);
+ int i, selected_rate = 0;
+ for (i = 1; i < 10; i++)
+ if (HDSPM_bit2freq(i) == rate) {
+ selected_rate = i;
+ break;
+ }
+ return selected_rate;
+}
#define HDSPM_AUTOSYNC_SAMPLE_RATE(xname, xindex) \
@@ -2270,7 +2460,7 @@ static int snd_hdspm_get_autosync_sample_rate(struct snd_kcontrol *kcontrol,
default:
ucontrol->value.enumerated.item[0] =
hdspm_get_s1_sample_rate(hdspm,
- ucontrol->id.index-1);
+ kcontrol->private_value-1);
}
break;
@@ -2289,28 +2479,24 @@ static int snd_hdspm_get_autosync_sample_rate(struct snd_kcontrol *kcontrol,
ucontrol->value.enumerated.item[0] =
hdspm_get_sync_in_sample_rate(hdspm);
break;
+ case 11: /* External Rate */
+ ucontrol->value.enumerated.item[0] =
+ hdspm_external_rate_to_enum(hdspm);
+ break;
default: /* AES1 to AES8 */
ucontrol->value.enumerated.item[0] =
- hdspm_get_s1_sample_rate(hdspm,
- kcontrol->private_value-1);
+ hdspm_get_aes_sample_rate(hdspm,
+ kcontrol->private_value -
+ HDSPM_AES32_AUTOSYNC_FROM_AES1);
break;
}
break;
case MADI:
case MADIface:
- {
- int rate = hdspm_external_sample_rate(hdspm);
- int i, selected_rate = 0;
- for (i = 1; i < 10; i++)
- if (HDSPM_bit2freq(i) == rate) {
- selected_rate = i;
- break;
- }
- ucontrol->value.enumerated.item[0] = selected_rate;
- }
+ ucontrol->value.enumerated.item[0] =
+ hdspm_external_rate_to_enum(hdspm);
break;
-
default:
break;
}
@@ -2359,33 +2545,17 @@ static int hdspm_system_clock_mode(struct hdspm *hdspm)
**/
static void hdspm_set_system_clock_mode(struct hdspm *hdspm, int mode)
{
- switch (hdspm->io_type) {
- case AIO:
- case RayDAT:
- if (0 == mode)
- hdspm->settings_register |= HDSPM_c0Master;
- else
- hdspm->settings_register &= ~HDSPM_c0Master;
-
- hdspm_write(hdspm, HDSPM_WR_SETTINGS, hdspm->settings_register);
- break;
-
- default:
- if (0 == mode)
- hdspm->control_register |= HDSPM_ClockModeMaster;
- else
- hdspm->control_register &= ~HDSPM_ClockModeMaster;
-
- hdspm_write(hdspm, HDSPM_controlRegister,
- hdspm->control_register);
- }
+ hdspm_set_toggle_setting(hdspm,
+ (hdspm_is_raydat_or_aio(hdspm)) ?
+ HDSPM_c0Master : HDSPM_ClockModeMaster,
+ (0 == mode));
}
static int snd_hdspm_info_system_clock_mode(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
- static char *texts[] = { "Master", "AutoSync" };
+ static const char *const texts[] = { "Master", "AutoSync" };
ENUMERATED_CTL_INFO(uinfo, texts);
return 0;
}
@@ -2809,16 +2979,7 @@ static int snd_hdspm_info_pref_sync_ref(struct snd_kcontrol *kcontrol,
{
struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
- uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
- uinfo->count = 1;
- uinfo->value.enumerated.items = hdspm->texts_autosync_items;
-
- if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
- uinfo->value.enumerated.item =
- uinfo->value.enumerated.items - 1;
-
- strcpy(uinfo->value.enumerated.name,
- hdspm->texts_autosync[uinfo->value.enumerated.item]);
+ snd_ctl_enum_info(uinfo, 1, hdspm->texts_autosync_items, hdspm->texts_autosync);
return 0;
}
@@ -2873,19 +3034,20 @@ static int snd_hdspm_put_pref_sync_ref(struct snd_kcontrol *kcontrol,
static int hdspm_autosync_ref(struct hdspm *hdspm)
{
+ /* This looks at the autosync selected sync reference */
if (AES32 == hdspm->io_type) {
+
unsigned int status = hdspm_read(hdspm, HDSPM_statusRegister);
- unsigned int syncref =
- (status >> HDSPM_AES32_syncref_bit) & 0xF;
- if (syncref == 0)
- return HDSPM_AES32_AUTOSYNC_FROM_WORD;
- if (syncref <= 8)
+ unsigned int syncref = (status >> HDSPM_AES32_syncref_bit) & 0xF;
+ if ((syncref >= HDSPM_AES32_AUTOSYNC_FROM_WORD) &&
+ (syncref <= HDSPM_AES32_AUTOSYNC_FROM_SYNC_IN)) {
return syncref;
+ }
return HDSPM_AES32_AUTOSYNC_FROM_NONE;
+
} else if (MADI == hdspm->io_type) {
- /* This looks at the autosync selected sync reference */
- unsigned int status2 = hdspm_read(hdspm, HDSPM_statusRegister2);
+ unsigned int status2 = hdspm_read(hdspm, HDSPM_statusRegister2);
switch (status2 & HDSPM_SelSyncRefMask) {
case HDSPM_SelSyncRef_WORD:
return HDSPM_AUTOSYNC_FROM_WORD;
@@ -2898,7 +3060,7 @@ static int hdspm_autosync_ref(struct hdspm *hdspm)
case HDSPM_SelSyncRef_NVALID:
return HDSPM_AUTOSYNC_FROM_NONE;
default:
- return 0;
+ return HDSPM_AUTOSYNC_FROM_NONE;
}
}
@@ -2912,31 +3074,15 @@ static int snd_hdspm_info_autosync_ref(struct snd_kcontrol *kcontrol,
struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
if (AES32 == hdspm->io_type) {
- static char *texts[] = { "WordClock", "AES1", "AES2", "AES3",
- "AES4", "AES5", "AES6", "AES7", "AES8", "None"};
-
- uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
- uinfo->count = 1;
- uinfo->value.enumerated.items = 10;
- if (uinfo->value.enumerated.item >=
- uinfo->value.enumerated.items)
- uinfo->value.enumerated.item =
- uinfo->value.enumerated.items - 1;
- strcpy(uinfo->value.enumerated.name,
- texts[uinfo->value.enumerated.item]);
+ static const char *const texts[] = { "WordClock", "AES1", "AES2", "AES3",
+ "AES4", "AES5", "AES6", "AES7", "AES8", "TCO", "Sync In", "None"};
+
+ ENUMERATED_CTL_INFO(uinfo, texts);
} else if (MADI == hdspm->io_type) {
- static char *texts[] = {"Word Clock", "MADI", "TCO",
+ static const char *const texts[] = {"Word Clock", "MADI", "TCO",
"Sync In", "None" };
- uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
- uinfo->count = 1;
- uinfo->value.enumerated.items = 5;
- if (uinfo->value.enumerated.item >=
- uinfo->value.enumerated.items)
- uinfo->value.enumerated.item =
- uinfo->value.enumerated.items - 1;
- strcpy(uinfo->value.enumerated.name,
- texts[uinfo->value.enumerated.item]);
+ ENUMERATED_CTL_INFO(uinfo, texts);
}
return 0;
}
@@ -2964,7 +3110,7 @@ static int snd_hdspm_get_autosync_ref(struct snd_kcontrol *kcontrol,
static int snd_hdspm_info_tco_video_input_format(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
- static char *texts[] = {"No video", "NTSC", "PAL"};
+ static const char *const texts[] = {"No video", "NTSC", "PAL"};
ENUMERATED_CTL_INFO(uinfo, texts);
return 0;
}
@@ -3010,7 +3156,7 @@ static int snd_hdspm_get_tco_video_input_format(struct snd_kcontrol *kcontrol,
static int snd_hdspm_info_tco_ltc_frames(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
- static char *texts[] = {"No lock", "24 fps", "25 fps", "29.97 fps",
+ static const char *const texts[] = {"No lock", "24 fps", "25 fps", "29.97 fps",
"30 fps"};
ENUMERATED_CTL_INFO(uinfo, texts);
return 0;
@@ -3027,19 +3173,19 @@ static int hdspm_tco_ltc_frames(struct hdspm *hdspm)
HDSPM_TCO1_LTC_Format_MSB)) {
case 0:
/* 24 fps */
- ret = 1;
+ ret = fps_24;
break;
case HDSPM_TCO1_LTC_Format_LSB:
/* 25 fps */
- ret = 2;
+ ret = fps_25;
break;
case HDSPM_TCO1_LTC_Format_MSB:
- /* 25 fps */
- ret = 3;
+ /* 29.97 fps */
+ ret = fps_2997;
break;
default:
/* 30 fps */
- ret = 4;
+ ret = fps_30;
break;
}
}
@@ -3067,16 +3213,35 @@ static int snd_hdspm_get_tco_ltc_frames(struct snd_kcontrol *kcontrol,
static int hdspm_toggle_setting(struct hdspm *hdspm, u32 regmask)
{
- return (hdspm->control_register & regmask) ? 1 : 0;
+ u32 reg;
+
+ if (hdspm_is_raydat_or_aio(hdspm))
+ reg = hdspm->settings_register;
+ else
+ reg = hdspm->control_register;
+
+ return (reg & regmask) ? 1 : 0;
}
static int hdspm_set_toggle_setting(struct hdspm *hdspm, u32 regmask, int out)
{
+ u32 *reg;
+ u32 target_reg;
+
+ if (hdspm_is_raydat_or_aio(hdspm)) {
+ reg = &(hdspm->settings_register);
+ target_reg = HDSPM_WR_SETTINGS;
+ } else {
+ reg = &(hdspm->control_register);
+ target_reg = HDSPM_controlRegister;
+ }
+
if (out)
- hdspm->control_register |= regmask;
+ *reg |= regmask;
else
- hdspm->control_register &= ~regmask;
- hdspm_write(hdspm, HDSPM_controlRegister, hdspm->control_register);
+ *reg &= ~regmask;
+
+ hdspm_write(hdspm, target_reg, *reg);
return 0;
}
@@ -3141,7 +3306,7 @@ static int hdspm_set_input_select(struct hdspm * hdspm, int out)
static int snd_hdspm_info_input_select(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
- static char *texts[] = { "optical", "coaxial" };
+ static const char *const texts[] = { "optical", "coaxial" };
ENUMERATED_CTL_INFO(uinfo, texts);
return 0;
}
@@ -3203,7 +3368,7 @@ static int hdspm_set_ds_wire(struct hdspm * hdspm, int ds)
static int snd_hdspm_info_ds_wire(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
- static char *texts[] = { "Single", "Double" };
+ static const char *const texts[] = { "Single", "Double" };
ENUMERATED_CTL_INFO(uinfo, texts);
return 0;
}
@@ -3276,7 +3441,7 @@ static int hdspm_set_qs_wire(struct hdspm * hdspm, int mode)
static int snd_hdspm_info_qs_wire(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
- static char *texts[] = { "Single", "Double", "Quad" };
+ static const char *const texts[] = { "Single", "Double", "Quad" };
ENUMERATED_CTL_INFO(uinfo, texts);
return 0;
}
@@ -3313,6 +3478,84 @@ static int snd_hdspm_put_qs_wire(struct snd_kcontrol *kcontrol,
return change;
}
+#define HDSPM_CONTROL_TRISTATE(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .private_value = xindex, \
+ .info = snd_hdspm_info_tristate, \
+ .get = snd_hdspm_get_tristate, \
+ .put = snd_hdspm_put_tristate \
+}
+
+static int hdspm_tristate(struct hdspm *hdspm, u32 regmask)
+{
+ u32 reg = hdspm->settings_register & (regmask * 3);
+ return reg / regmask;
+}
+
+static int hdspm_set_tristate(struct hdspm *hdspm, int mode, u32 regmask)
+{
+ hdspm->settings_register &= ~(regmask * 3);
+ hdspm->settings_register |= (regmask * mode);
+ hdspm_write(hdspm, HDSPM_WR_SETTINGS, hdspm->settings_register);
+
+ return 0;
+}
+
+static int snd_hdspm_info_tristate(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ u32 regmask = kcontrol->private_value;
+
+ static const char *const texts_spdif[] = { "Optical", "Coaxial", "Internal" };
+ static const char *const texts_levels[] = { "Hi Gain", "+4 dBu", "-10 dBV" };
+
+ switch (regmask) {
+ case HDSPM_c0_Input0:
+ ENUMERATED_CTL_INFO(uinfo, texts_spdif);
+ break;
+ default:
+ ENUMERATED_CTL_INFO(uinfo, texts_levels);
+ break;
+ }
+ return 0;
+}
+
+static int snd_hdspm_get_tristate(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+ u32 regmask = kcontrol->private_value;
+
+ spin_lock_irq(&hdspm->lock);
+ ucontrol->value.enumerated.item[0] = hdspm_tristate(hdspm, regmask);
+ spin_unlock_irq(&hdspm->lock);
+ return 0;
+}
+
+static int snd_hdspm_put_tristate(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+ u32 regmask = kcontrol->private_value;
+ int change;
+ int val;
+
+ if (!snd_hdspm_use_is_exclusive(hdspm))
+ return -EBUSY;
+ val = ucontrol->value.integer.value[0];
+ if (val < 0)
+ val = 0;
+ if (val > 2)
+ val = 2;
+
+ spin_lock_irq(&hdspm->lock);
+ change = val != hdspm_tristate(hdspm, regmask);
+ hdspm_set_tristate(hdspm, val, regmask);
+ spin_unlock_irq(&hdspm->lock);
+ return change;
+}
+
#define HDSPM_MADI_SPEEDMODE(xname, xindex) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
.name = xname, \
@@ -3352,7 +3595,7 @@ static int hdspm_set_madi_speedmode(struct hdspm *hdspm, int mode)
static int snd_hdspm_info_madi_speedmode(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
- static char *texts[] = { "Single", "Double", "Quad" };
+ static const char *const texts[] = { "Single", "Double", "Quad" };
ENUMERATED_CTL_INFO(uinfo, texts);
return 0;
}
@@ -3587,7 +3830,7 @@ static int snd_hdspm_put_playback_mixer(struct snd_kcontrol *kcontrol,
static int snd_hdspm_info_sync_check(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
- static char *texts[] = { "No Lock", "Lock", "Sync", "N/A" };
+ static const char *const texts[] = { "No Lock", "Lock", "Sync", "N/A" };
ENUMERATED_CTL_INFO(uinfo, texts);
return 0;
}
@@ -3595,7 +3838,7 @@ static int snd_hdspm_info_sync_check(struct snd_kcontrol *kcontrol,
static int snd_hdspm_tco_info_lock_check(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
- static char *texts[] = { "No Lock", "Lock" };
+ static const char *const texts[] = { "No Lock", "Lock" };
ENUMERATED_CTL_INFO(uinfo, texts);
return 0;
}
@@ -3745,9 +3988,18 @@ static int hdspm_tco_sync_check(struct hdspm *hdspm)
if (hdspm->tco) {
switch (hdspm->io_type) {
case MADI:
+ status = hdspm_read(hdspm, HDSPM_statusRegister);
+ if (status & HDSPM_tcoLockMadi) {
+ if (status & HDSPM_tcoSync)
+ return 2;
+ else
+ return 1;
+ }
+ return 0;
+ break;
case AES32:
status = hdspm_read(hdspm, HDSPM_statusRegister);
- if (status & HDSPM_tcoLock) {
+ if (status & HDSPM_tcoLockAes) {
if (status & HDSPM_tcoSync)
return 2;
else
@@ -3807,7 +4059,8 @@ static int snd_hdspm_get_sync_check(struct snd_kcontrol *kcontrol,
case 5: /* SYNC IN */
val = hdspm_sync_in_sync_check(hdspm); break;
default:
- val = hdspm_s1_sync_check(hdspm, ucontrol->id.index-1);
+ val = hdspm_s1_sync_check(hdspm,
+ kcontrol->private_value-1);
}
break;
@@ -3975,7 +4228,8 @@ static void hdspm_tco_write(struct hdspm *hdspm)
static int snd_hdspm_info_tco_sample_rate(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
- static char *texts[] = { "44.1 kHz", "48 kHz" };
+ /* TODO freq from app could be supported here, see tco->samplerate */
+ static const char *const texts[] = { "44.1 kHz", "48 kHz" };
ENUMERATED_CTL_INFO(uinfo, texts);
return 0;
}
@@ -4021,7 +4275,8 @@ static int snd_hdspm_put_tco_sample_rate(struct snd_kcontrol *kcontrol,
static int snd_hdspm_info_tco_pull(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
- static char *texts[] = { "0", "+ 0.1 %", "- 0.1 %", "+ 4 %", "- 4 %" };
+ static const char *const texts[] = { "0", "+ 0.1 %", "- 0.1 %",
+ "+ 4 %", "- 4 %" };
ENUMERATED_CTL_INFO(uinfo, texts);
return 0;
}
@@ -4066,7 +4321,7 @@ static int snd_hdspm_put_tco_pull(struct snd_kcontrol *kcontrol,
static int snd_hdspm_info_tco_wck_conversion(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
- static char *texts[] = { "1:1", "44.1 -> 48", "48 -> 44.1" };
+ static const char *const texts[] = { "1:1", "44.1 -> 48", "48 -> 44.1" };
ENUMERATED_CTL_INFO(uinfo, texts);
return 0;
}
@@ -4112,7 +4367,7 @@ static int snd_hdspm_put_tco_wck_conversion(struct snd_kcontrol *kcontrol,
static int snd_hdspm_info_tco_frame_rate(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
- static char *texts[] = { "24 fps", "25 fps", "29.97fps",
+ static const char *const texts[] = { "24 fps", "25 fps", "29.97fps",
"29.97 dfps", "30 fps", "30 dfps" };
ENUMERATED_CTL_INFO(uinfo, texts);
return 0;
@@ -4159,7 +4414,7 @@ static int snd_hdspm_put_tco_frame_rate(struct snd_kcontrol *kcontrol,
static int snd_hdspm_info_tco_sync_source(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
- static char *texts[] = { "LTC", "Video", "WCK" };
+ static const char *const texts[] = { "LTC", "Video", "WCK" };
ENUMERATED_CTL_INFO(uinfo, texts);
return 0;
}
@@ -4284,7 +4539,6 @@ static struct snd_kcontrol_new snd_hdspm_controls_aio[] = {
HDSPM_INTERNAL_CLOCK("Internal Clock", 0),
HDSPM_SYSTEM_CLOCK_MODE("System Clock Mode", 0),
HDSPM_PREF_SYNC_REF("Preferred Sync Reference", 0),
- HDSPM_AUTOSYNC_REF("AutoSync Reference", 0),
HDSPM_SYSTEM_SAMPLE_RATE("System Sample Rate", 0),
HDSPM_AUTOSYNC_SAMPLE_RATE("External Rate", 0),
HDSPM_SYNC_CHECK("WC SyncCheck", 0),
@@ -4298,7 +4552,16 @@ static struct snd_kcontrol_new snd_hdspm_controls_aio[] = {
HDSPM_AUTOSYNC_SAMPLE_RATE("SPDIF Frequency", 2),
HDSPM_AUTOSYNC_SAMPLE_RATE("ADAT Frequency", 3),
HDSPM_AUTOSYNC_SAMPLE_RATE("TCO Frequency", 4),
- HDSPM_AUTOSYNC_SAMPLE_RATE("SYNC IN Frequency", 5)
+ HDSPM_AUTOSYNC_SAMPLE_RATE("SYNC IN Frequency", 5),
+ HDSPM_CONTROL_TRISTATE("S/PDIF Input", HDSPM_c0_Input0),
+ HDSPM_TOGGLE_SETTING("S/PDIF Out Optical", HDSPM_c0_Spdif_Opt),
+ HDSPM_TOGGLE_SETTING("S/PDIF Out Professional", HDSPM_c0_Pro),
+ HDSPM_TOGGLE_SETTING("ADAT internal (AEB/TEB)", HDSPM_c0_AEB1),
+ HDSPM_TOGGLE_SETTING("XLR Breakout Cable", HDSPM_c0_Sym6db),
+ HDSPM_TOGGLE_SETTING("Single Speed WordClock Out", HDSPM_c0_Wck48),
+ HDSPM_CONTROL_TRISTATE("Input Level", HDSPM_c0_AD_GAIN0),
+ HDSPM_CONTROL_TRISTATE("Output Level", HDSPM_c0_DA_GAIN0),
+ HDSPM_CONTROL_TRISTATE("Phones Level", HDSPM_c0_PH_GAIN0)
/*
HDSPM_INPUT_SELECT("Input Select", 0),
@@ -4335,7 +4598,9 @@ static struct snd_kcontrol_new snd_hdspm_controls_raydat[] = {
HDSPM_AUTOSYNC_SAMPLE_RATE("ADAT3 Frequency", 5),
HDSPM_AUTOSYNC_SAMPLE_RATE("ADAT4 Frequency", 6),
HDSPM_AUTOSYNC_SAMPLE_RATE("TCO Frequency", 7),
- HDSPM_AUTOSYNC_SAMPLE_RATE("SYNC IN Frequency", 8)
+ HDSPM_AUTOSYNC_SAMPLE_RATE("SYNC IN Frequency", 8),
+ HDSPM_TOGGLE_SETTING("S/PDIF Out Professional", HDSPM_c0_Pro),
+ HDSPM_TOGGLE_SETTING("Single Speed WordClock Out", HDSPM_c0_Wck48)
};
static struct snd_kcontrol_new snd_hdspm_controls_aes32[] = {
@@ -4345,7 +4610,7 @@ static struct snd_kcontrol_new snd_hdspm_controls_aes32[] = {
HDSPM_PREF_SYNC_REF("Preferred Sync Reference", 0),
HDSPM_AUTOSYNC_REF("AutoSync Reference", 0),
HDSPM_SYSTEM_SAMPLE_RATE("System Sample Rate", 0),
- HDSPM_AUTOSYNC_SAMPLE_RATE("External Rate", 0),
+ HDSPM_AUTOSYNC_SAMPLE_RATE("External Rate", 11),
HDSPM_SYNC_CHECK("WC Sync Check", 0),
HDSPM_SYNC_CHECK("AES1 Sync Check", 1),
HDSPM_SYNC_CHECK("AES2 Sync Check", 2),
@@ -4501,77 +4766,22 @@ static int snd_hdspm_create_controls(struct snd_card *card,
------------------------------------------------------------*/
static void
-snd_hdspm_proc_read_madi(struct snd_info_entry * entry,
- struct snd_info_buffer *buffer)
+snd_hdspm_proc_read_tco(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
{
struct hdspm *hdspm = entry->private_data;
- unsigned int status, status2, control, freq;
-
- char *pref_sync_ref;
- char *autosync_ref;
- char *system_clock_mode;
- char *insel;
- int x, x2;
-
- /* TCO stuff */
+ unsigned int status, control;
int a, ltc, frames, seconds, minutes, hours;
unsigned int period;
u64 freq_const = 0;
u32 rate;
+ snd_iprintf(buffer, "--- TCO ---\n");
+
status = hdspm_read(hdspm, HDSPM_statusRegister);
- status2 = hdspm_read(hdspm, HDSPM_statusRegister2);
control = hdspm->control_register;
- freq = hdspm_read(hdspm, HDSPM_timecodeRegister);
- snd_iprintf(buffer, "%s (Card #%d) Rev.%x Status2first3bits: %x\n",
- hdspm->card_name, hdspm->card->number + 1,
- hdspm->firmware_rev,
- (status2 & HDSPM_version0) |
- (status2 & HDSPM_version1) | (status2 &
- HDSPM_version2));
- snd_iprintf(buffer, "HW Serial: 0x%06x%06x\n",
- (hdspm_read(hdspm, HDSPM_midiStatusIn1)>>8) & 0xFFFFFF,
- hdspm->serial);
-
- snd_iprintf(buffer, "IRQ: %d Registers bus: 0x%lx VM: 0x%lx\n",
- hdspm->irq, hdspm->port, (unsigned long)hdspm->iobase);
-
- snd_iprintf(buffer, "--- System ---\n");
-
- snd_iprintf(buffer,
- "IRQ Pending: Audio=%d, MIDI0=%d, MIDI1=%d, IRQcount=%d\n",
- status & HDSPM_audioIRQPending,
- (status & HDSPM_midi0IRQPending) ? 1 : 0,
- (status & HDSPM_midi1IRQPending) ? 1 : 0,
- hdspm->irq_count);
- snd_iprintf(buffer,
- "HW pointer: id = %d, rawptr = %d (%d->%d) "
- "estimated= %ld (bytes)\n",
- ((status & HDSPM_BufferID) ? 1 : 0),
- (status & HDSPM_BufferPositionMask),
- (status & HDSPM_BufferPositionMask) %
- (2 * (int)hdspm->period_bytes),
- ((status & HDSPM_BufferPositionMask) - 64) %
- (2 * (int)hdspm->period_bytes),
- (long) hdspm_hw_pointer(hdspm) * 4);
-
- snd_iprintf(buffer,
- "MIDI FIFO: Out1=0x%x, Out2=0x%x, In1=0x%x, In2=0x%x \n",
- hdspm_read(hdspm, HDSPM_midiStatusOut0) & 0xFF,
- hdspm_read(hdspm, HDSPM_midiStatusOut1) & 0xFF,
- hdspm_read(hdspm, HDSPM_midiStatusIn0) & 0xFF,
- hdspm_read(hdspm, HDSPM_midiStatusIn1) & 0xFF);
- snd_iprintf(buffer,
- "MIDIoverMADI FIFO: In=0x%x, Out=0x%x \n",
- hdspm_read(hdspm, HDSPM_midiStatusIn2) & 0xFF,
- hdspm_read(hdspm, HDSPM_midiStatusOut2) & 0xFF);
- snd_iprintf(buffer,
- "Register: ctrl1=0x%x, ctrl2=0x%x, status1=0x%x, "
- "status2=0x%x\n",
- hdspm->control_register, hdspm->control2_register,
- status, status2);
if (status & HDSPM_tco_detect) {
snd_iprintf(buffer, "TCO module detected.\n");
a = hdspm_read(hdspm, HDSPM_RD_TCO+4);
@@ -4665,6 +4875,75 @@ snd_hdspm_proc_read_madi(struct snd_info_entry * entry,
} else {
snd_iprintf(buffer, "No TCO module detected.\n");
}
+}
+
+static void
+snd_hdspm_proc_read_madi(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct hdspm *hdspm = entry->private_data;
+ unsigned int status, status2, control, freq;
+
+ char *pref_sync_ref;
+ char *autosync_ref;
+ char *system_clock_mode;
+ char *insel;
+ int x, x2;
+
+ status = hdspm_read(hdspm, HDSPM_statusRegister);
+ status2 = hdspm_read(hdspm, HDSPM_statusRegister2);
+ control = hdspm->control_register;
+ freq = hdspm_read(hdspm, HDSPM_timecodeRegister);
+
+ snd_iprintf(buffer, "%s (Card #%d) Rev.%x Status2first3bits: %x\n",
+ hdspm->card_name, hdspm->card->number + 1,
+ hdspm->firmware_rev,
+ (status2 & HDSPM_version0) |
+ (status2 & HDSPM_version1) | (status2 &
+ HDSPM_version2));
+
+ snd_iprintf(buffer, "HW Serial: 0x%06x%06x\n",
+ (hdspm_read(hdspm, HDSPM_midiStatusIn1)>>8) & 0xFFFFFF,
+ hdspm->serial);
+
+ snd_iprintf(buffer, "IRQ: %d Registers bus: 0x%lx VM: 0x%lx\n",
+ hdspm->irq, hdspm->port, (unsigned long)hdspm->iobase);
+
+ snd_iprintf(buffer, "--- System ---\n");
+
+ snd_iprintf(buffer,
+ "IRQ Pending: Audio=%d, MIDI0=%d, MIDI1=%d, IRQcount=%d\n",
+ status & HDSPM_audioIRQPending,
+ (status & HDSPM_midi0IRQPending) ? 1 : 0,
+ (status & HDSPM_midi1IRQPending) ? 1 : 0,
+ hdspm->irq_count);
+ snd_iprintf(buffer,
+ "HW pointer: id = %d, rawptr = %d (%d->%d) "
+ "estimated= %ld (bytes)\n",
+ ((status & HDSPM_BufferID) ? 1 : 0),
+ (status & HDSPM_BufferPositionMask),
+ (status & HDSPM_BufferPositionMask) %
+ (2 * (int)hdspm->period_bytes),
+ ((status & HDSPM_BufferPositionMask) - 64) %
+ (2 * (int)hdspm->period_bytes),
+ (long) hdspm_hw_pointer(hdspm) * 4);
+
+ snd_iprintf(buffer,
+ "MIDI FIFO: Out1=0x%x, Out2=0x%x, In1=0x%x, In2=0x%x \n",
+ hdspm_read(hdspm, HDSPM_midiStatusOut0) & 0xFF,
+ hdspm_read(hdspm, HDSPM_midiStatusOut1) & 0xFF,
+ hdspm_read(hdspm, HDSPM_midiStatusIn0) & 0xFF,
+ hdspm_read(hdspm, HDSPM_midiStatusIn1) & 0xFF);
+ snd_iprintf(buffer,
+ "MIDIoverMADI FIFO: In=0x%x, Out=0x%x \n",
+ hdspm_read(hdspm, HDSPM_midiStatusIn2) & 0xFF,
+ hdspm_read(hdspm, HDSPM_midiStatusOut2) & 0xFF);
+ snd_iprintf(buffer,
+ "Register: ctrl1=0x%x, ctrl2=0x%x, status1=0x%x, "
+ "status2=0x%x\n",
+ hdspm->control_register, hdspm->control2_register,
+ status, status2);
+
snd_iprintf(buffer, "--- Settings ---\n");
@@ -4768,6 +5047,9 @@ snd_hdspm_proc_read_madi(struct snd_info_entry * entry,
(status & HDSPM_RX_64ch) ? "64 channels" :
"56 channels");
+ /* call readout function for TCO specific status */
+ snd_hdspm_proc_read_tco(entry, buffer);
+
snd_iprintf(buffer, "\n");
}
@@ -4909,11 +5191,18 @@ snd_hdspm_proc_read_aes32(struct snd_info_entry * entry,
autosync_ref = "AES7"; break;
case HDSPM_AES32_AUTOSYNC_FROM_AES8:
autosync_ref = "AES8"; break;
+ case HDSPM_AES32_AUTOSYNC_FROM_TCO:
+ autosync_ref = "TCO"; break;
+ case HDSPM_AES32_AUTOSYNC_FROM_SYNC_IN:
+ autosync_ref = "Sync In"; break;
default:
autosync_ref = "---"; break;
}
snd_iprintf(buffer, "AutoSync ref = %s\n", autosync_ref);
+ /* call readout function for TCO specific status */
+ snd_hdspm_proc_read_tco(entry, buffer);
+
snd_iprintf(buffer, "\n");
}
@@ -5097,7 +5386,7 @@ static int snd_hdspm_set_defaults(struct hdspm * hdspm)
case AES32:
hdspm->control_register =
- HDSPM_ClockModeMaster | /* Master Cloack Mode on */
+ HDSPM_ClockModeMaster | /* Master Clock Mode on */
hdspm_encode_latency(7) | /* latency max=8192samples */
HDSPM_SyncRef0 | /* AES1 is syncclock */
HDSPM_LineOut | /* Analog output in */
@@ -5123,9 +5412,8 @@ static int snd_hdspm_set_defaults(struct hdspm * hdspm)
all_in_all_mixer(hdspm, 0 * UNITY_GAIN);
- if (hdspm->io_type == AIO || hdspm->io_type == RayDAT) {
+ if (hdspm_is_raydat_or_aio(hdspm))
hdspm_write(hdspm, HDSPM_WR_SETTINGS, hdspm->settings_register);
- }
/* set a default rate so that the channel map is set up. */
hdspm_set_rate(hdspm, 48000, 1);
@@ -5371,6 +5659,16 @@ static int snd_hdspm_hw_params(struct snd_pcm_substream *substream,
*/
+ /* For AES cards, the float format bit is the same as the
+ * preferred sync reference. Since we don't want to break
+ * sync settings, we have to skip the remaining part of this
+ * function.
+ */
+ if (hdspm->io_type == AES32) {
+ return 0;
+ }
+
+
/* Switch to native float format if requested */
if (SNDRV_PCM_FORMAT_FLOAT_LE == params_format(params)) {
if (!(hdspm->control_register & HDSPe_FLOAT_FORMAT))
@@ -6013,7 +6311,7 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file,
ltc.format = fps_2997;
break;
default:
- ltc.format = 30;
+ ltc.format = fps_30;
break;
}
if (i & HDSPM_TCO1_set_drop_frame_flag) {
@@ -6479,10 +6777,6 @@ static int snd_hdspm_create(struct snd_card *card,
break;
case AIO:
- if (0 == (hdspm_read(hdspm, HDSPM_statusRegister2) & HDSPM_s2_AEBI_D)) {
- snd_printk(KERN_INFO "HDSPM: AEB input board found, but not supported\n");
- }
-
hdspm->ss_in_channels = AIO_IN_SS_CHANNELS;
hdspm->ds_in_channels = AIO_IN_DS_CHANNELS;
hdspm->qs_in_channels = AIO_IN_QS_CHANNELS;
@@ -6490,6 +6784,20 @@ static int snd_hdspm_create(struct snd_card *card,
hdspm->ds_out_channels = AIO_OUT_DS_CHANNELS;
hdspm->qs_out_channels = AIO_OUT_QS_CHANNELS;
+ if (0 == (hdspm_read(hdspm, HDSPM_statusRegister2) & HDSPM_s2_AEBI_D)) {
+ snd_printk(KERN_INFO "HDSPM: AEB input board found\n");
+ hdspm->ss_in_channels += 4;
+ hdspm->ds_in_channels += 4;
+ hdspm->qs_in_channels += 4;
+ }
+
+ if (0 == (hdspm_read(hdspm, HDSPM_statusRegister2) & HDSPM_s2_AEBO_D)) {
+ snd_printk(KERN_INFO "HDSPM: AEB output board found\n");
+ hdspm->ss_out_channels += 4;
+ hdspm->ds_out_channels += 4;
+ hdspm->qs_out_channels += 4;
+ }
+
hdspm->channel_map_out_ss = channel_map_aio_out_ss;
hdspm->channel_map_out_ds = channel_map_aio_out_ds;
hdspm->channel_map_out_qs = channel_map_aio_out_qs;
@@ -6558,6 +6866,7 @@ static int snd_hdspm_create(struct snd_card *card,
break;
case MADI:
+ case AES32:
if (hdspm_read(hdspm, HDSPM_statusRegister) & HDSPM_tco_detect) {
hdspm->midiPorts++;
hdspm->tco = kzalloc(sizeof(struct hdspm_tco),
@@ -6565,7 +6874,7 @@ static int snd_hdspm_create(struct snd_card *card,
if (NULL != hdspm->tco) {
hdspm_tco_write(hdspm);
}
- snd_printk(KERN_INFO "HDSPM: MADI TCO module found\n");
+ snd_printk(KERN_INFO "HDSPM: MADI/AES TCO module found\n");
} else {
hdspm->tco = NULL;
}
@@ -6580,10 +6889,12 @@ static int snd_hdspm_create(struct snd_card *card,
case AES32:
if (hdspm->tco) {
hdspm->texts_autosync = texts_autosync_aes_tco;
- hdspm->texts_autosync_items = 10;
+ hdspm->texts_autosync_items =
+ ARRAY_SIZE(texts_autosync_aes_tco);
} else {
hdspm->texts_autosync = texts_autosync_aes;
- hdspm->texts_autosync_items = 9;
+ hdspm->texts_autosync_items =
+ ARRAY_SIZE(texts_autosync_aes);
}
break;
diff --git a/sound/soc/cirrus/ep93xx-i2s.c b/sound/soc/cirrus/ep93xx-i2s.c
index f23f331e9a97..a57643d6402f 100644
--- a/sound/soc/cirrus/ep93xx-i2s.c
+++ b/sound/soc/cirrus/ep93xx-i2s.c
@@ -408,7 +408,6 @@ static int ep93xx_i2s_probe(struct platform_device *pdev)
return 0;
fail_put_lrclk:
- dev_set_drvdata(&pdev->dev, NULL);
clk_put(info->lrclk);
fail_put_sclk:
clk_put(info->sclk);
@@ -423,7 +422,6 @@ static int ep93xx_i2s_remove(struct platform_device *pdev)
struct ep93xx_i2s_info *info = dev_get_drvdata(&pdev->dev);
snd_soc_unregister_component(&pdev->dev);
- dev_set_drvdata(&pdev->dev, NULL);
clk_put(info->lrclk);
clk_put(info->sclk);
clk_put(info->mclk);
diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c
index 66967ba6f757..b2090b2a5e2d 100644
--- a/sound/soc/codecs/dmic.c
+++ b/sound/soc/codecs/dmic.c
@@ -50,20 +50,11 @@ static const struct snd_soc_dapm_route intercon[] = {
{"DMIC AIF", NULL, "DMic"},
};
-static int dmic_probe(struct snd_soc_codec *codec)
-{
- struct snd_soc_dapm_context *dapm = &codec->dapm;
-
- snd_soc_dapm_new_controls(dapm, dmic_dapm_widgets,
- ARRAY_SIZE(dmic_dapm_widgets));
- snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
- snd_soc_dapm_new_widgets(dapm);
-
- return 0;
-}
-
static struct snd_soc_codec_driver soc_dmic = {
- .probe = dmic_probe,
+ .dapm_widgets = dmic_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(dmic_dapm_widgets),
+ .dapm_routes = intercon,
+ .num_dapm_routes = ARRAY_SIZE(intercon),
};
static int dmic_dev_probe(struct platform_device *pdev)
diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index 4db7314baabc..c26a8f814b18 100644
--- a/sound/soc/codecs/rt5640.c
+++ b/sound/soc/codecs/rt5640.c
@@ -50,8 +50,6 @@ static const struct regmap_range_cfg rt5640_ranges[] = {
static struct reg_default init_list[] = {
{RT5640_PR_BASE + 0x3d, 0x3600},
- {RT5640_PR_BASE + 0x1c, 0x0D21},
- {RT5640_PR_BASE + 0x1b, 0x0000},
{RT5640_PR_BASE + 0x12, 0x0aa8},
{RT5640_PR_BASE + 0x14, 0x0aaa},
{RT5640_PR_BASE + 0x20, 0x6110},
@@ -384,15 +382,11 @@ static const SOC_ENUM_SINGLE_DECL(
static const struct snd_kcontrol_new rt5640_snd_controls[] = {
/* Speaker Output Volume */
- SOC_DOUBLE("Speaker Playback Switch", RT5640_SPK_VOL,
- RT5640_L_MUTE_SFT, RT5640_R_MUTE_SFT, 1, 1),
SOC_DOUBLE("Speaker Channel Switch", RT5640_SPK_VOL,
RT5640_VOL_L_SFT, RT5640_VOL_R_SFT, 1, 1),
SOC_DOUBLE_TLV("Speaker Playback Volume", RT5640_SPK_VOL,
RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 39, 1, out_vol_tlv),
/* Headphone Output Volume */
- SOC_DOUBLE("HP Playback Switch", RT5640_HP_VOL,
- RT5640_L_MUTE_SFT, RT5640_R_MUTE_SFT, 1, 1),
SOC_DOUBLE("HP Channel Switch", RT5640_HP_VOL,
RT5640_VOL_L_SFT, RT5640_VOL_R_SFT, 1, 1),
SOC_DOUBLE_TLV("HP Playback Volume", RT5640_HP_VOL,
@@ -737,6 +731,22 @@ static const struct snd_kcontrol_new rt5640_mono_mix[] = {
RT5640_M_BST1_MM_SFT, 1, 1),
};
+static const struct snd_kcontrol_new spk_l_enable_control =
+ SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5640_SPK_VOL,
+ RT5640_L_MUTE_SFT, 1, 1);
+
+static const struct snd_kcontrol_new spk_r_enable_control =
+ SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5640_SPK_VOL,
+ RT5640_R_MUTE_SFT, 1, 1);
+
+static const struct snd_kcontrol_new hp_l_enable_control =
+ SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5640_HP_VOL,
+ RT5640_L_MUTE_SFT, 1, 1);
+
+static const struct snd_kcontrol_new hp_r_enable_control =
+ SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5640_HP_VOL,
+ RT5640_R_MUTE_SFT, 1, 1);
+
/* Stereo ADC source */
static const char * const rt5640_stereo_adc1_src[] = {
"DIG MIX", "ADC"
@@ -868,33 +878,6 @@ static const SOC_ENUM_SINGLE_DECL(
static const struct snd_kcontrol_new rt5640_sdi_mux =
SOC_DAPM_ENUM("SDI select", rt5640_sdi_sel_enum);
-static int spk_event(struct snd_soc_dapm_widget *w,
- struct snd_kcontrol *kcontrol, int event)
-{
- struct snd_soc_codec *codec = w->codec;
- struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec);
-
- switch (event) {
- case SND_SOC_DAPM_POST_PMU:
- regmap_update_bits(rt5640->regmap, RT5640_PWR_DIG1,
- 0x0001, 0x0001);
- regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + 0x1c,
- 0xf000, 0xf000);
- break;
-
- case SND_SOC_DAPM_PRE_PMD:
- regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + 0x1c,
- 0xf000, 0x0000);
- regmap_update_bits(rt5640->regmap, RT5640_PWR_DIG1,
- 0x0001, 0x0000);
- break;
-
- default:
- return 0;
- }
- return 0;
-}
-
static int rt5640_set_dmic1_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
@@ -943,6 +926,117 @@ static int rt5640_set_dmic2_event(struct snd_soc_dapm_widget *w,
return 0;
}
+void hp_amp_power_on(struct snd_soc_codec *codec)
+{
+ struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec);
+
+ /* depop parameters */
+ regmap_update_bits(rt5640->regmap, RT5640_PR_BASE +
+ RT5640_CHPUMP_INT_REG1, 0x0700, 0x0200);
+ regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M2,
+ RT5640_DEPOP_MASK, RT5640_DEPOP_MAN);
+ regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1,
+ RT5640_HP_CP_MASK | RT5640_HP_SG_MASK | RT5640_HP_CB_MASK,
+ RT5640_HP_CP_PU | RT5640_HP_SG_DIS | RT5640_HP_CB_PU);
+ regmap_write(rt5640->regmap, RT5640_PR_BASE + RT5640_HP_DCC_INT1,
+ 0x9f00);
+ /* headphone amp power on */
+ regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1,
+ RT5640_PWR_FV1 | RT5640_PWR_FV2, 0);
+ regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1,
+ RT5640_PWR_HA,
+ RT5640_PWR_HA);
+ usleep_range(10000, 15000);
+ regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1,
+ RT5640_PWR_FV1 | RT5640_PWR_FV2 ,
+ RT5640_PWR_FV1 | RT5640_PWR_FV2);
+}
+
+static void rt5640_pmu_depop(struct snd_soc_codec *codec)
+{
+ struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec);
+
+ regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M2,
+ RT5640_DEPOP_MASK | RT5640_DIG_DP_MASK,
+ RT5640_DEPOP_AUTO | RT5640_DIG_DP_EN);
+ regmap_update_bits(rt5640->regmap, RT5640_CHARGE_PUMP,
+ RT5640_PM_HP_MASK, RT5640_PM_HP_HV);
+
+ regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M3,
+ RT5640_CP_FQ1_MASK | RT5640_CP_FQ2_MASK | RT5640_CP_FQ3_MASK,
+ (RT5640_CP_FQ_192_KHZ << RT5640_CP_FQ1_SFT) |
+ (RT5640_CP_FQ_12_KHZ << RT5640_CP_FQ2_SFT) |
+ (RT5640_CP_FQ_192_KHZ << RT5640_CP_FQ3_SFT));
+
+ regmap_write(rt5640->regmap, RT5640_PR_BASE +
+ RT5640_MAMP_INT_REG2, 0x1c00);
+ regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1,
+ RT5640_HP_CP_MASK | RT5640_HP_SG_MASK,
+ RT5640_HP_CP_PD | RT5640_HP_SG_EN);
+ regmap_update_bits(rt5640->regmap, RT5640_PR_BASE +
+ RT5640_CHPUMP_INT_REG1, 0x0700, 0x0400);
+}
+
+static int rt5640_hp_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec);
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ rt5640_pmu_depop(codec);
+ rt5640->hp_mute = 0;
+ break;
+
+ case SND_SOC_DAPM_PRE_PMD:
+ rt5640->hp_mute = 1;
+ usleep_range(70000, 75000);
+ break;
+
+ default:
+ return 0;
+ }
+
+ return 0;
+}
+
+static int rt5640_hp_power_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ hp_amp_power_on(codec);
+ break;
+ default:
+ return 0;
+ }
+
+ return 0;
+}
+
+static int rt5640_hp_post_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec);
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ if (!rt5640->hp_mute)
+ usleep_range(80000, 85000);
+
+ break;
+
+ default:
+ return 0;
+ }
+
+ return 0;
+}
+
static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = {
SND_SOC_DAPM_SUPPLY("PLL1", RT5640_PWR_ANLG2,
RT5640_PWR_PLL_BIT, 0, NULL, 0),
@@ -1132,15 +1226,28 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = {
rt5640_mono_mix, ARRAY_SIZE(rt5640_mono_mix)),
SND_SOC_DAPM_SUPPLY("Improve MONO Amp Drv", RT5640_PWR_ANLG1,
RT5640_PWR_MA_BIT, 0, NULL, 0),
- SND_SOC_DAPM_SUPPLY("Improve HP Amp Drv", RT5640_PWR_ANLG1,
- SND_SOC_NOPM, 0, NULL, 0),
- SND_SOC_DAPM_PGA("HP L Amp", RT5640_PWR_ANLG1,
+ SND_SOC_DAPM_SUPPLY_S("Improve HP Amp Drv", 1, SND_SOC_NOPM,
+ 0, 0, rt5640_hp_power_event, SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0,
+ rt5640_hp_event,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_SUPPLY("HP L Amp", RT5640_PWR_ANLG1,
RT5640_PWR_HP_L_BIT, 0, NULL, 0),
- SND_SOC_DAPM_PGA("HP R Amp", RT5640_PWR_ANLG1,
+ SND_SOC_DAPM_SUPPLY("HP R Amp", RT5640_PWR_ANLG1,
RT5640_PWR_HP_R_BIT, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("Improve SPK Amp Drv", RT5640_PWR_DIG1,
- SND_SOC_NOPM, 0, spk_event,
- SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+ RT5640_PWR_CLS_D_BIT, 0, NULL, 0),
+
+ /* Output Switch */
+ SND_SOC_DAPM_SWITCH("Speaker L Playback", SND_SOC_NOPM, 0, 0,
+ &spk_l_enable_control),
+ SND_SOC_DAPM_SWITCH("Speaker R Playback", SND_SOC_NOPM, 0, 0,
+ &spk_r_enable_control),
+ SND_SOC_DAPM_SWITCH("HP L Playback", SND_SOC_NOPM, 0, 0,
+ &hp_l_enable_control),
+ SND_SOC_DAPM_SWITCH("HP R Playback", SND_SOC_NOPM, 0, 0,
+ &hp_r_enable_control),
+ SND_SOC_DAPM_POST("HP Post", rt5640_hp_post_event),
/* Output Lines */
SND_SOC_DAPM_OUTPUT("SPOLP"),
SND_SOC_DAPM_OUTPUT("SPOLN"),
@@ -1381,9 +1488,11 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = {
{"HPO MIX L", "HPO MIX DAC2 Switch", "DAC L2"},
{"HPO MIX L", "HPO MIX DAC1 Switch", "DAC L1"},
{"HPO MIX L", "HPO MIX HPVOL Switch", "HPOVOL L"},
+ {"HPO MIX L", NULL, "HP L Amp"},
{"HPO MIX R", "HPO MIX DAC2 Switch", "DAC R2"},
{"HPO MIX R", "HPO MIX DAC1 Switch", "DAC R1"},
{"HPO MIX R", "HPO MIX HPVOL Switch", "HPOVOL R"},
+ {"HPO MIX R", NULL, "HP R Amp"},
{"LOUT MIX", "DAC L1 Switch", "DAC L1"},
{"LOUT MIX", "DAC R1 Switch", "DAC R1"},
@@ -1396,13 +1505,15 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = {
{"Mono MIX", "OUTVOL L Switch", "OUTVOL L"},
{"Mono MIX", "BST1 Switch", "BST1"},
- {"HP L Amp", NULL, "HPO MIX L"},
- {"HP R Amp", NULL, "HPO MIX R"},
+ {"HP Amp", NULL, "HPO MIX L"},
+ {"HP Amp", NULL, "HPO MIX R"},
- {"SPOLP", NULL, "SPOL MIX"},
- {"SPOLN", NULL, "SPOL MIX"},
- {"SPORP", NULL, "SPOR MIX"},
- {"SPORN", NULL, "SPOR MIX"},
+ {"Speaker L Playback", "Switch", "SPOL MIX"},
+ {"Speaker R Playback", "Switch", "SPOR MIX"},
+ {"SPOLP", NULL, "Speaker L Playback"},
+ {"SPOLN", NULL, "Speaker L Playback"},
+ {"SPORP", NULL, "Speaker R Playback"},
+ {"SPORN", NULL, "Speaker R Playback"},
{"SPOLP", NULL, "Improve SPK Amp Drv"},
{"SPOLN", NULL, "Improve SPK Amp Drv"},
@@ -1412,8 +1523,10 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = {
{"HPOL", NULL, "Improve HP Amp Drv"},
{"HPOR", NULL, "Improve HP Amp Drv"},
- {"HPOL", NULL, "HP L Amp"},
- {"HPOR", NULL, "HP R Amp"},
+ {"HP L Playback", "Switch", "HP Amp"},
+ {"HP R Playback", "Switch", "HP Amp"},
+ {"HPOL", NULL, "HP L Playback"},
+ {"HPOR", NULL, "HP R Playback"},
{"LOUTL", NULL, "LOUT MIX"},
{"LOUTR", NULL, "LOUT MIX"},
{"MONOP", NULL, "Mono MIX"},
@@ -1792,17 +1905,13 @@ static int rt5640_set_bias_level(struct snd_soc_codec *codec,
RT5640_PWR_BG | RT5640_PWR_VREF2,
RT5640_PWR_VREF1 | RT5640_PWR_MB |
RT5640_PWR_BG | RT5640_PWR_VREF2);
- mdelay(10);
+ usleep_range(10000, 15000);
snd_soc_update_bits(codec, RT5640_PWR_ANLG1,
RT5640_PWR_FV1 | RT5640_PWR_FV2,
RT5640_PWR_FV1 | RT5640_PWR_FV2);
regcache_sync(rt5640->regmap);
snd_soc_update_bits(codec, RT5640_DUMMY1,
0x0301, 0x0301);
- snd_soc_update_bits(codec, RT5640_DEPOP_M1,
- 0x001d, 0x0019);
- snd_soc_update_bits(codec, RT5640_DEPOP_M2,
- 0x2000, 0x2000);
snd_soc_update_bits(codec, RT5640_MICBIAS,
0x0030, 0x0030);
}
@@ -1846,8 +1955,6 @@ static int rt5640_probe(struct snd_soc_codec *codec)
rt5640_set_bias_level(codec, SND_SOC_BIAS_OFF);
snd_soc_update_bits(codec, RT5640_DUMMY1, 0x0301, 0x0301);
- snd_soc_update_bits(codec, RT5640_DEPOP_M1, 0x001d, 0x0019);
- snd_soc_update_bits(codec, RT5640_DEPOP_M2, 0x2000, 0x2000);
snd_soc_update_bits(codec, RT5640_MICBIAS, 0x0030, 0x0030);
snd_soc_update_bits(codec, RT5640_DSP_PATH2, 0xfc00, 0x0c00);
@@ -2069,6 +2176,8 @@ static int rt5640_i2c_probe(struct i2c_client *i2c,
regmap_update_bits(rt5640->regmap, RT5640_IN3_IN4,
RT5640_IN_DF2, RT5640_IN_DF2);
+ rt5640->hp_mute = 1;
+
ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5640,
rt5640_dai, ARRAY_SIZE(rt5640_dai));
if (ret < 0)
diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h
index c48286d7118f..5e8df25a13f3 100644
--- a/sound/soc/codecs/rt5640.h
+++ b/sound/soc/codecs/rt5640.h
@@ -145,6 +145,8 @@
/* Index of Codec Private Register definition */
+#define RT5640_CHPUMP_INT_REG1 0x24
+#define RT5640_MAMP_INT_REG2 0x37
#define RT5640_3D_SPK 0x63
#define RT5640_WND_1 0x6c
#define RT5640_WND_2 0x6d
@@ -153,6 +155,7 @@
#define RT5640_WND_5 0x70
#define RT5640_WND_8 0x73
#define RT5640_DIP_SPK_INF 0x75
+#define RT5640_HP_DCC_INT1 0x77
#define RT5640_EQ_BW_LOP 0xa0
#define RT5640_EQ_GN_LOP 0xa1
#define RT5640_EQ_FC_BP1 0xa2
@@ -1201,6 +1204,14 @@
#define RT5640_CP_FQ2_SFT 4
#define RT5640_CP_FQ3_MASK (0x7)
#define RT5640_CP_FQ3_SFT 0
+#define RT5640_CP_FQ_1_5_KHZ 0
+#define RT5640_CP_FQ_3_KHZ 1
+#define RT5640_CP_FQ_6_KHZ 2
+#define RT5640_CP_FQ_12_KHZ 3
+#define RT5640_CP_FQ_24_KHZ 4
+#define RT5640_CP_FQ_48_KHZ 5
+#define RT5640_CP_FQ_96_KHZ 6
+#define RT5640_CP_FQ_192_KHZ 7
/* HPOUT charge pump (0x91) */
#define RT5640_OSW_L_MASK (0x1 << 11)
@@ -2087,6 +2098,7 @@ struct rt5640_priv {
int pll_out;
int dmic_en;
+ bool hp_mute;
};
#endif
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index f8d30e5f6371..492644e67ace 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -561,8 +561,9 @@ static int ssm2602_suspend(struct snd_soc_codec *codec)
static int ssm2602_resume(struct snd_soc_codec *codec)
{
- snd_soc_cache_sync(codec);
+ struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
+ regcache_sync(ssm2602->regmap);
ssm2602_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
return 0;
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index 17df4e32feac..2ed57d4aa445 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -338,18 +338,6 @@ static inline int aic32x4_get_divs(int mclk, int rate)
return -EINVAL;
}
-static int aic32x4_add_widgets(struct snd_soc_codec *codec)
-{
- snd_soc_dapm_new_controls(&codec->dapm, aic32x4_dapm_widgets,
- ARRAY_SIZE(aic32x4_dapm_widgets));
-
- snd_soc_dapm_add_routes(&codec->dapm, aic32x4_dapm_routes,
- ARRAY_SIZE(aic32x4_dapm_routes));
-
- snd_soc_dapm_new_widgets(&codec->dapm);
- return 0;
-}
-
static int aic32x4_set_dai_sysclk(struct snd_soc_dai *codec_dai,
int clk_id, unsigned int freq, int dir)
{
@@ -683,9 +671,6 @@ static int aic32x4_probe(struct snd_soc_codec *codec)
}
aic32x4_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- snd_soc_add_codec_controls(codec, aic32x4_snd_controls,
- ARRAY_SIZE(aic32x4_snd_controls));
- aic32x4_add_widgets(codec);
/*
* Workaround: for an unknown reason, the ADC needs to be powered up
@@ -714,6 +699,13 @@ static struct snd_soc_codec_driver soc_codec_dev_aic32x4 = {
.suspend = aic32x4_suspend,
.resume = aic32x4_resume,
.set_bias_level = aic32x4_set_bias_level,
+
+ .controls = aic32x4_snd_controls,
+ .num_controls = ARRAY_SIZE(aic32x4_snd_controls),
+ .dapm_widgets = aic32x4_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(aic32x4_dapm_widgets),
+ .dapm_routes = aic32x4_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(aic32x4_dapm_routes),
};
static int aic32x4_i2c_probe(struct i2c_client *i2c,
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 91dfbfeda6f8..4dfa8dceeabf 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -1202,7 +1202,6 @@ static int wm8904_add_widgets(struct snd_soc_codec *codec)
break;
}
- snd_soc_dapm_new_widgets(dapm);
return 0;
}
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 36782f067cc5..11d80f3b6137 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -3174,7 +3174,7 @@ static ssize_t wm8962_beep_set(struct device *dev,
long int time;
int ret;
- ret = strict_strtol(buf, 10, &time);
+ ret = kstrtol(buf, 10, &time);
if (ret != 0)
return ret;
diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c
index 70eb37a5dd16..25c31f1655f6 100644
--- a/sound/soc/dwc/designware_i2s.c
+++ b/sound/soc/dwc/designware_i2s.c
@@ -421,13 +421,11 @@ static int dw_i2s_probe(struct platform_device *pdev)
dw_i2s_dai, 1);
if (ret != 0) {
dev_err(&pdev->dev, "not able to register dai\n");
- goto err_set_drvdata;
+ goto err_clk_disable;
}
return 0;
-err_set_drvdata:
- dev_set_drvdata(&pdev->dev, NULL);
err_clk_disable:
clk_disable(dev->clk);
err_clk_put:
@@ -440,7 +438,6 @@ static int dw_i2s_remove(struct platform_device *pdev)
struct dw_i2s_dev *dev = dev_get_drvdata(&pdev->dev);
snd_soc_unregister_component(&pdev->dev);
- dev_set_drvdata(&pdev->dev, NULL);
clk_put(dev->clk);
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index cd088cc8c866..b7ab71f2ccc1 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -193,6 +193,17 @@ config SND_SOC_IMX_SGTL5000
Say Y if you want to add support for SoC audio on an i.MX board with
a sgtl5000 codec.
+config SND_SOC_IMX_SPDIF
+ tristate "SoC Audio support for i.MX boards with S/PDIF"
+ select SND_SOC_IMX_PCM_DMA
+ select SND_SOC_FSL_SPDIF
+ select SND_SOC_SPDIF
+ select REGMAP_MMIO
+ help
+ SoC Audio support for i.MX boards with S/PDIF
+ Say Y if you want to add support for SoC audio on an i.MX board with
+ a S/DPDIF.
+
config SND_SOC_IMX_MC13783
tristate "SoC Audio support for I.MX boards with mc13783"
depends on MFD_MC13783 && ARM
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index 4b5970e014dd..8db705b0fdf9 100644
--- a/sound/soc/fsl/Makefile
+++ b/sound/soc/fsl/Makefile
@@ -45,6 +45,7 @@ snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o
snd-soc-wm1133-ev1-objs := wm1133-ev1.o
snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o
snd-soc-imx-wm8962-objs := imx-wm8962.o
+snd-soc-imx-spdif-objs := imx-spdif.o
snd-soc-imx-mc13783-objs := imx-mc13783.o
obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o
@@ -53,4 +54,5 @@ obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o
obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o
obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o
obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o
+obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o
obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o
diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c
index 42a43820d993..3920c3e849ce 100644
--- a/sound/soc/fsl/fsl_spdif.c
+++ b/sound/soc/fsl/fsl_spdif.c
@@ -411,8 +411,8 @@ static int spdif_set_sample_rate(struct snd_pcm_substream *substream,
return 0;
}
-int fsl_spdif_startup(struct snd_pcm_substream *substream,
- struct snd_soc_dai *cpu_dai)
+static int fsl_spdif_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *cpu_dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai);
@@ -546,7 +546,7 @@ static int fsl_spdif_trigger(struct snd_pcm_substream *substream,
return 0;
}
-struct snd_soc_dai_ops fsl_spdif_dai_ops = {
+static struct snd_soc_dai_ops fsl_spdif_dai_ops = {
.startup = fsl_spdif_startup,
.hw_params = fsl_spdif_hw_params,
.trigger = fsl_spdif_trigger,
@@ -555,7 +555,6 @@ struct snd_soc_dai_ops fsl_spdif_dai_ops = {
/*
- * ============================================
* FSL SPDIF IEC958 controller(mixer) functions
*
* Channel status get/put control
@@ -563,7 +562,6 @@ struct snd_soc_dai_ops fsl_spdif_dai_ops = {
* Valid bit value get control
* DPLL lock status get control
* User bit sync mode selection control
- * ============================================
*/
static int fsl_spdif_info(struct snd_kcontrol *kcontrol,
@@ -921,7 +919,7 @@ static int fsl_spdif_dai_probe(struct snd_soc_dai *dai)
return 0;
}
-struct snd_soc_dai_driver fsl_spdif_dai = {
+static struct snd_soc_dai_driver fsl_spdif_dai = {
.probe = &fsl_spdif_dai_probe,
.playback = {
.channels_min = 2,
@@ -942,11 +940,7 @@ static const struct snd_soc_component_driver fsl_spdif_component = {
.name = "fsl-spdif",
};
-/*
- * ================
- * FSL SPDIF REGMAP
- * ================
- */
+/* FSL SPDIF REGMAP */
static bool fsl_spdif_readable_reg(struct device *dev, unsigned int reg)
{
@@ -1077,9 +1071,9 @@ static int fsl_spdif_probe_txclk(struct fsl_spdif_priv *spdif_priv,
break;
}
- dev_dbg(&pdev->dev, "use rxtx%d as tx clock source for %dHz sample rate",
+ dev_dbg(&pdev->dev, "use rxtx%d as tx clock source for %dHz sample rate\n",
spdif_priv->txclk_src[index], rate[index]);
- dev_dbg(&pdev->dev, "use divisor %d for %dHz sample rate",
+ dev_dbg(&pdev->dev, "use divisor %d for %dHz sample rate\n",
spdif_priv->txclk_div[index], rate[index]);
return 0;
@@ -1119,10 +1113,8 @@ static int fsl_spdif_probe(struct platform_device *pdev)
}
regs = devm_ioremap_resource(&pdev->dev, res);
- if (IS_ERR(regs)) {
- dev_err(&pdev->dev, "could not map device resources\n");
+ if (IS_ERR(regs))
return PTR_ERR(regs);
- }
spdif_priv->regmap = devm_regmap_init_mmio_clk(&pdev->dev,
"core", regs, &fsl_spdif_regmap_config);
@@ -1184,7 +1176,7 @@ static int fsl_spdif_probe(struct platform_device *pdev)
&spdif_priv->cpu_dai_drv, 1);
if (ret) {
dev_err(&pdev->dev, "failed to register DAI: %d\n", ret);
- goto error_dev;
+ return ret;
}
ret = imx_pcm_dma_init(pdev);
@@ -1197,8 +1189,6 @@ static int fsl_spdif_probe(struct platform_device *pdev)
error_component:
snd_soc_unregister_component(&pdev->dev);
-error_dev:
- dev_set_drvdata(&pdev->dev, NULL);
return ret;
}
@@ -1207,7 +1197,6 @@ static int fsl_spdif_remove(struct platform_device *pdev)
{
imx_pcm_dma_exit(pdev);
snd_soc_unregister_component(&pdev->dev);
- dev_set_drvdata(&pdev->dev, NULL);
return 0;
}
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 5cf626c4dc96..c6b743978d5e 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -1114,7 +1114,6 @@ error_dai:
snd_soc_unregister_component(&pdev->dev);
error_dev:
- dev_set_drvdata(&pdev->dev, NULL);
device_remove_file(&pdev->dev, dev_attr);
error_clk:
diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c
index ab17381cc981..d3bf71a0ec56 100644
--- a/sound/soc/fsl/imx-audmux.c
+++ b/sound/soc/fsl/imx-audmux.c
@@ -335,7 +335,8 @@ static int imx_audmux_probe(struct platform_device *pdev)
if (audmux_type == IMX31_AUDMUX)
audmux_debugfs_init();
- imx_audmux_parse_dt_defaults(pdev, pdev->dev.of_node);
+ if (of_id)
+ imx_audmux_parse_dt_defaults(pdev, pdev->dev.of_node);
return 0;
}
diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c
new file mode 100644
index 000000000000..816013b0ebba
--- /dev/null
+++ b/sound/soc/fsl/imx-spdif.c
@@ -0,0 +1,148 @@
+/*
+ * Copyright (C) 2013 Freescale Semiconductor, Inc.
+ *
+ * The code contained herein is licensed under the GNU General Public
+ * License. You may obtain a copy of the GNU General Public License
+ * Version 2 or later at the following locations:
+ *
+ * http://www.opensource.org/licenses/gpl-license.html
+ * http://www.gnu.org/copyleft/gpl.html
+ */
+
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <sound/soc.h>
+
+struct imx_spdif_data {
+ struct snd_soc_dai_link dai[2];
+ struct snd_soc_card card;
+ struct platform_device *txdev;
+ struct platform_device *rxdev;
+};
+
+static int imx_spdif_audio_probe(struct platform_device *pdev)
+{
+ struct device_node *spdif_np, *np = pdev->dev.of_node;
+ struct imx_spdif_data *data;
+ int ret = 0, num_links = 0;
+
+ spdif_np = of_parse_phandle(np, "spdif-controller", 0);
+ if (!spdif_np) {
+ dev_err(&pdev->dev, "failed to find spdif-controller\n");
+ ret = -EINVAL;
+ goto end;
+ }
+
+ data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL);
+ if (!data) {
+ dev_err(&pdev->dev, "failed to allocate memory\n");
+ ret = -ENOMEM;
+ goto end;
+ }
+
+ if (of_property_read_bool(np, "spdif-out")) {
+ data->dai[num_links].name = "S/PDIF TX";
+ data->dai[num_links].stream_name = "S/PDIF PCM Playback";
+ data->dai[num_links].codec_dai_name = "dit-hifi";
+ data->dai[num_links].codec_name = "spdif-dit";
+ data->dai[num_links].cpu_of_node = spdif_np;
+ data->dai[num_links].platform_of_node = spdif_np;
+ num_links++;
+
+ data->txdev = platform_device_register_simple("spdif-dit", -1, NULL, 0);
+ if (IS_ERR(data->txdev)) {
+ ret = PTR_ERR(data->txdev);
+ dev_err(&pdev->dev, "register dit failed: %d\n", ret);
+ goto end;
+ }
+ }
+
+ if (of_property_read_bool(np, "spdif-in")) {
+ data->dai[num_links].name = "S/PDIF RX";
+ data->dai[num_links].stream_name = "S/PDIF PCM Capture";
+ data->dai[num_links].codec_dai_name = "dir-hifi";
+ data->dai[num_links].codec_name = "spdif-dir";
+ data->dai[num_links].cpu_of_node = spdif_np;
+ data->dai[num_links].platform_of_node = spdif_np;
+ num_links++;
+
+ data->rxdev = platform_device_register_simple("spdif-dir", -1, NULL, 0);
+ if (IS_ERR(data->rxdev)) {
+ ret = PTR_ERR(data->rxdev);
+ dev_err(&pdev->dev, "register dir failed: %d\n", ret);
+ goto error_dit;
+ }
+ }
+
+ if (!num_links) {
+ dev_err(&pdev->dev, "no enabled S/PDIF DAI link\n");
+ goto error_dir;
+ }
+
+ data->card.dev = &pdev->dev;
+ data->card.num_links = num_links;
+ data->card.dai_link = data->dai;
+
+ ret = snd_soc_of_parse_card_name(&data->card, "model");
+ if (ret)
+ goto error_dir;
+
+ ret = snd_soc_register_card(&data->card);
+ if (ret) {
+ dev_err(&pdev->dev, "snd_soc_register_card failed: %d\n", ret);
+ goto error_dir;
+ }
+
+ platform_set_drvdata(pdev, data);
+
+ goto end;
+
+error_dir:
+ if (data->rxdev)
+ platform_device_unregister(data->rxdev);
+error_dit:
+ if (data->txdev)
+ platform_device_unregister(data->txdev);
+end:
+ if (spdif_np)
+ of_node_put(spdif_np);
+
+ return ret;
+}
+
+static int imx_spdif_audio_remove(struct platform_device *pdev)
+{
+ struct imx_spdif_data *data = platform_get_drvdata(pdev);
+
+ if (data->rxdev)
+ platform_device_unregister(data->rxdev);
+ if (data->txdev)
+ platform_device_unregister(data->txdev);
+
+ snd_soc_unregister_card(&data->card);
+
+ return 0;
+}
+
+static const struct of_device_id imx_spdif_dt_ids[] = {
+ { .compatible = "fsl,imx-audio-spdif", },
+ { /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, imx_spdif_dt_ids);
+
+static struct platform_driver imx_spdif_driver = {
+ .driver = {
+ .name = "imx-spdif",
+ .owner = THIS_MODULE,
+ .of_match_table = imx_spdif_dt_ids,
+ },
+ .probe = imx_spdif_audio_probe,
+ .remove = imx_spdif_audio_remove,
+};
+
+module_platform_driver(imx_spdif_driver);
+
+MODULE_AUTHOR("Freescale Semiconductor, Inc.");
+MODULE_DESCRIPTION("Freescale i.MX S/PDIF machine driver");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:imx-spdif");
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 6cf8355a8542..8c49147db84c 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -105,6 +105,7 @@ static int asoc_simple_card_remove(struct platform_device *pdev)
static struct platform_driver asoc_simple_card = {
.driver = {
.name = "asoc-simple-card",
+ .owner = THIS_MODULE,
},
.probe = asoc_simple_card_probe,
.remove = asoc_simple_card_remove,
@@ -112,6 +113,7 @@ static struct platform_driver asoc_simple_card = {
module_platform_driver(asoc_simple_card);
+MODULE_ALIAS("platform:asoc-simple-card");
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("ASoC Simple Sound Card");
MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>");
diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig
index 9e1970c44e86..78ed4a42ad21 100644
--- a/sound/soc/kirkwood/Kconfig
+++ b/sound/soc/kirkwood/Kconfig
@@ -1,6 +1,6 @@
config SND_KIRKWOOD_SOC
- tristate "SoC Audio for the Marvell Kirkwood chip"
- depends on ARCH_KIRKWOOD || COMPILE_TEST
+ tristate "SoC Audio for the Marvell Kirkwood and Dove chips"
+ depends on ARCH_KIRKWOOD || ARCH_DOVE || COMPILE_TEST
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
diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c
index e5f3f7a9ea26..7fce340ab3ef 100644
--- a/sound/soc/kirkwood/kirkwood-i2s.c
+++ b/sound/soc/kirkwood/kirkwood-i2s.c
@@ -22,6 +22,8 @@
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <linux/platform_data/asoc-kirkwood.h>
+#include <linux/of.h>
+
#include "kirkwood.h"
#define DRV_NAME "mvebu-audio"
@@ -453,6 +455,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev)
struct snd_soc_dai_driver *soc_dai = &kirkwood_i2s_dai;
struct kirkwood_dma_data *priv;
struct resource *mem;
+ struct device_node *np = pdev->dev.of_node;
int err;
priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
@@ -473,14 +476,16 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev)
return -ENXIO;
}
- if (!data) {
- dev_err(&pdev->dev, "no platform data ?!\n");
+ if (np) {
+ priv->burst = 128; /* might be 32 or 128 */
+ } else if (data) {
+ priv->burst = data->burst;
+ } else {
+ dev_err(&pdev->dev, "no DT nor platform data ?!\n");
return -EINVAL;
}
- priv->burst = data->burst;
-
- priv->clk = devm_clk_get(&pdev->dev, NULL);
+ priv->clk = devm_clk_get(&pdev->dev, np ? "internal" : NULL);
if (IS_ERR(priv->clk)) {
dev_err(&pdev->dev, "no clock\n");
return PTR_ERR(priv->clk);
@@ -507,7 +512,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev)
priv->ctl_rec = KIRKWOOD_RECCTL_SIZE_24;
/* Select the burst size */
- if (data->burst == 32) {
+ if (priv->burst == 32) {
priv->ctl_play |= KIRKWOOD_PLAYCTL_BURST_32;
priv->ctl_rec |= KIRKWOOD_RECCTL_BURST_32;
} else {
@@ -552,12 +557,21 @@ static int kirkwood_i2s_dev_remove(struct platform_device *pdev)
return 0;
}
+#ifdef CONFIG_OF
+static struct of_device_id mvebu_audio_of_match[] = {
+ { .compatible = "marvell,mvebu-audio" },
+ { }
+};
+MODULE_DEVICE_TABLE(of, mvebu_audio_of_match);
+#endif
+
static struct platform_driver kirkwood_i2s_driver = {
.probe = kirkwood_i2s_dev_probe,
.remove = kirkwood_i2s_dev_remove,
.driver = {
.name = DRV_NAME,
.owner = THIS_MODULE,
+ .of_match_table = of_match_ptr(mvebu_audio_of_match),
},
};
diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c
index ce084eb10c49..4bb273786ff3 100644
--- a/sound/soc/mxs/mxs-sgtl5000.c
+++ b/sound/soc/mxs/mxs-sgtl5000.c
@@ -105,11 +105,13 @@ static struct snd_soc_dai_link mxs_sgtl5000_dai[] = {
.stream_name = "HiFi Playback",
.codec_dai_name = "sgtl5000",
.ops = &mxs_sgtl5000_hifi_ops,
+ .playback_only = true,
}, {
.name = "HiFi Rx",
.stream_name = "HiFi Capture",
.codec_dai_name = "sgtl5000",
.ops = &mxs_sgtl5000_hifi_ops,
+ .capture_only = true,
},
};
diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c
index 361e4c03646e..83433fdea32a 100644
--- a/sound/soc/omap/mcbsp.c
+++ b/sound/soc/omap/mcbsp.c
@@ -781,7 +781,7 @@ static ssize_t prop##_store(struct device *dev, \
unsigned long val; \
int status; \
\
- status = strict_strtoul(buf, 0, &val); \
+ status = kstrtoul(buf, 0, &val); \
if (status) \
return status; \
\
diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c
index a0c67f60f594..9338d11e9216 100644
--- a/sound/soc/samsung/dma.c
+++ b/sound/soc/samsung/dma.c
@@ -90,6 +90,13 @@ static void dma_enqueue(struct snd_pcm_substream *substream)
dma_info.period = prtd->dma_period;
dma_info.len = prtd->dma_period*limit;
+ if (dma_info.cap == DMA_CYCLIC) {
+ dma_info.buf = pos;
+ prtd->params->ops->prepare(prtd->params->ch, &dma_info);
+ prtd->dma_loaded += limit;
+ return;
+ }
+
while (prtd->dma_loaded < limit) {
pr_debug("dma_loaded: %d\n", prtd->dma_loaded);
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 30390260bb67..b33ca7cd085b 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -235,6 +235,8 @@ struct fsi_stream {
struct sh_dmae_slave slave; /* see fsi_handler_init() */
struct work_struct work;
dma_addr_t dma;
+ int loop_cnt;
+ int additional_pos;
};
struct fsi_clk {
@@ -1289,6 +1291,8 @@ static int fsi_dma_init(struct fsi_priv *fsi, struct fsi_stream *io)
io->bus_option = BUSOP_SET(24, PACKAGE_24BITBUS_BACK) |
BUSOP_SET(16, PACKAGE_16BITBUS_STREAM);
+ io->loop_cnt = 2; /* push 1st, 2nd period first, then 3rd, 4th... */
+ io->additional_pos = 0;
io->dma = dma_map_single(dai->dev, runtime->dma_area,
snd_pcm_lib_buffer_bytes(io->substream), dir);
return 0;
@@ -1305,11 +1309,15 @@ static int fsi_dma_quit(struct fsi_priv *fsi, struct fsi_stream *io)
return 0;
}
-static dma_addr_t fsi_dma_get_area(struct fsi_stream *io)
+static dma_addr_t fsi_dma_get_area(struct fsi_stream *io, int additional)
{
struct snd_pcm_runtime *runtime = io->substream->runtime;
+ int period = io->period_pos + additional;
- return io->dma + samples_to_bytes(runtime, io->buff_sample_pos);
+ if (period >= runtime->periods)
+ period = 0;
+
+ return io->dma + samples_to_bytes(runtime, period * io->period_samples);
}
static void fsi_dma_complete(void *data)
@@ -1321,7 +1329,7 @@ static void fsi_dma_complete(void *data)
enum dma_data_direction dir = fsi_stream_is_play(fsi, io) ?
DMA_TO_DEVICE : DMA_FROM_DEVICE;
- dma_sync_single_for_cpu(dai->dev, fsi_dma_get_area(io),
+ dma_sync_single_for_cpu(dai->dev, fsi_dma_get_area(io, 0),
samples_to_bytes(runtime, io->period_samples), dir);
io->buff_sample_pos += io->period_samples;
@@ -1347,7 +1355,7 @@ static void fsi_dma_do_work(struct work_struct *work)
struct snd_pcm_runtime *runtime;
enum dma_data_direction dir;
int is_play = fsi_stream_is_play(fsi, io);
- int len;
+ int len, i;
dma_addr_t buf;
if (!fsi_stream_is_working(fsi, io))
@@ -1357,26 +1365,33 @@ static void fsi_dma_do_work(struct work_struct *work)
runtime = io->substream->runtime;
dir = is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE;
len = samples_to_bytes(runtime, io->period_samples);
- buf = fsi_dma_get_area(io);
- dma_sync_single_for_device(dai->dev, buf, len, dir);
+ for (i = 0; i < io->loop_cnt; i++) {
+ buf = fsi_dma_get_area(io, io->additional_pos);
- desc = dmaengine_prep_slave_single(io->chan, buf, len, dir,
- DMA_PREP_INTERRUPT | DMA_CTRL_ACK);
- if (!desc) {
- dev_err(dai->dev, "dmaengine_prep_slave_sg() fail\n");
- return;
- }
+ dma_sync_single_for_device(dai->dev, buf, len, dir);
- desc->callback = fsi_dma_complete;
- desc->callback_param = io;
+ desc = dmaengine_prep_slave_single(io->chan, buf, len, dir,
+ DMA_PREP_INTERRUPT | DMA_CTRL_ACK);
+ if (!desc) {
+ dev_err(dai->dev, "dmaengine_prep_slave_sg() fail\n");
+ return;
+ }
- if (dmaengine_submit(desc) < 0) {
- dev_err(dai->dev, "tx_submit() fail\n");
- return;
+ desc->callback = fsi_dma_complete;
+ desc->callback_param = io;
+
+ if (dmaengine_submit(desc) < 0) {
+ dev_err(dai->dev, "tx_submit() fail\n");
+ return;
+ }
+
+ dma_async_issue_pending(io->chan);
+
+ io->additional_pos = 1;
}
- dma_async_issue_pending(io->chan);
+ io->loop_cnt = 1;
/*
* FIXME
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 528f8708221d..4d0561312f3b 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -203,7 +203,7 @@ static ssize_t pmdown_time_set(struct device *dev,
struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev);
int ret;
- ret = strict_strtol(buf, 10, &rtd->pmdown_time);
+ ret = kstrtol(buf, 10, &rtd->pmdown_time);
if (ret)
return ret;
@@ -248,6 +248,7 @@ static ssize_t codec_reg_write_file(struct file *file,
char *start = buf;
unsigned long reg, value;
struct snd_soc_codec *codec = file->private_data;
+ int ret;
buf_size = min(count, (sizeof(buf)-1));
if (copy_from_user(buf, user_buf, buf_size))
@@ -259,8 +260,9 @@ static ssize_t codec_reg_write_file(struct file *file,
reg = simple_strtoul(start, &start, 16);
while (*start == ' ')
start++;
- if (strict_strtoul(start, 16, &value))
- return -EINVAL;
+ ret = kstrtoul(start, 16, &value);
+ if (ret)
+ return ret;
/* Userspace has been fiddling around behind the kernel's back */
add_taint(TAINT_USER, LOCKDEP_NOW_UNRELIABLE);
@@ -1243,9 +1245,6 @@ static int soc_post_component_init(struct snd_soc_card *card,
}
rtd->card = card;
- /* Make sure all DAPM widgets are instantiated */
- snd_soc_dapm_new_widgets(&codec->dapm);
-
/* machine controls, routes and widgets are not prefixed */
temp = codec->name_prefix;
codec->name_prefix = NULL;
@@ -1741,8 +1740,6 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes,
card->num_dapm_routes);
- snd_soc_dapm_new_widgets(&card->dapm);
-
for (i = 0; i < card->num_links; i++) {
dai_link = &card->dai_link[i];
dai_fmt = dai_link->dai_fmt;
@@ -1821,12 +1818,12 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
}
}
- snd_soc_dapm_new_widgets(&card->dapm);
-
if (card->fully_routed)
list_for_each_entry(codec, &card->codec_dev_list, card_list)
snd_soc_dapm_auto_nc_codec_pins(codec);
+ snd_soc_dapm_new_widgets(card);
+
ret = snd_card_register(card->snd_card);
if (ret < 0) {
dev_err(card->dev, "ASoC: failed to register soundcard %d\n",
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index d84bd0f167b6..c17c14c394df 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -229,6 +229,8 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget,
template.id = snd_soc_dapm_kcontrol;
template.name = kcontrol->id.name;
+ data->value = template.on_val;
+
data->widget = snd_soc_dapm_new_control(widget->dapm,
&template);
if (!data->widget) {
@@ -2374,6 +2376,9 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm,
wsource->ext = 1;
}
+ dapm_mark_dirty(wsource, "Route added");
+ dapm_mark_dirty(wsink, "Route added");
+
/* connect static paths */
if (control == NULL) {
list_add(&path->list, &dapm->card->paths);
@@ -2436,9 +2441,6 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm,
return 0;
}
- dapm_mark_dirty(wsource, "Route added");
- dapm_mark_dirty(wsink, "Route added");
-
return 0;
err:
kfree(path);
@@ -2712,9 +2714,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_weak_routes);
*
* Returns 0 for success.
*/
-int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm)
+int snd_soc_dapm_new_widgets(struct snd_soc_card *card)
{
- struct snd_soc_card *card = dapm->card;
struct snd_soc_dapm_widget *w;
unsigned int val;
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index 7aa26b5178aa..71358e3b54d9 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -183,8 +183,6 @@ int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count,
list_add(&(pins[i].list), &jack->pins);
}
- snd_soc_dapm_new_widgets(&jack->codec->card->dapm);
-
/* Update to reflect the last reported status; canned jack
* implementations are likely to set their state before the
* card has an opportunity to associate pins.
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index fb70fbe26862..330c9a6b5cb5 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2020,6 +2020,16 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
capture = 1;
}
+ if (rtd->dai_link->playback_only) {
+ playback = 1;
+ capture = 0;
+ }
+
+ if (rtd->dai_link->capture_only) {
+ playback = 0;
+ capture = 1;
+ }
+
/* create the PCM */
if (rtd->dai_link->no_pcm) {
snprintf(new_name, sizeof(new_name), "(%s)",
diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c
index b9defcdeb7ef..780bf3f62d28 100644
--- a/sound/usb/6fire/firmware.c
+++ b/sound/usb/6fire/firmware.c
@@ -346,10 +346,10 @@ static int usb6fire_fw_check(u8 *version)
if (!memcmp(version, known_fw_versions + i, 2))
return 0;
- snd_printk(KERN_ERR PREFIX "invalid fimware version in device: %*ph. "
+ snd_printk(KERN_ERR PREFIX "invalid fimware version in device: %4ph. "
"please reconnect to power. if this failure "
"still happens, check your firmware installation.",
- 4, version);
+ version);
return -EINVAL;
}
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index 659950e5b94f..93e970f2b3c0 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -418,6 +418,9 @@ struct snd_usb_endpoint *snd_usb_add_endpoint(struct snd_usb_audio *chip,
struct snd_usb_endpoint *ep;
int is_playback = direction == SNDRV_PCM_STREAM_PLAYBACK;
+ if (WARN_ON(!alts))
+ return NULL;
+
mutex_lock(&chip->mutex);
list_for_each_entry(ep, &chip->ep_list, list) {
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 15b151ed4899..b375d58871e7 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -327,6 +327,137 @@ static int search_roland_implicit_fb(struct usb_device *dev, int ifnum,
return 0;
}
+static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs,
+ struct usb_device *dev,
+ struct usb_interface_descriptor *altsd,
+ unsigned int attr)
+{
+ struct usb_host_interface *alts;
+ struct usb_interface *iface;
+ unsigned int ep;
+
+ /* Implicit feedback sync EPs consumers are always playback EPs */
+ if (subs->direction != SNDRV_PCM_STREAM_PLAYBACK)
+ return 0;
+
+ switch (subs->stream->chip->usb_id) {
+ case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */
+ case USB_ID(0x0763, 0x2031): /* M-Audio Fast Track C600 */
+ ep = 0x81;
+ iface = usb_ifnum_to_if(dev, 3);
+
+ if (!iface || iface->num_altsetting == 0)
+ return -EINVAL;
+
+ alts = &iface->altsetting[1];
+ goto add_sync_ep;
+ break;
+ case USB_ID(0x0763, 0x2080): /* M-Audio FastTrack Ultra */
+ case USB_ID(0x0763, 0x2081):
+ ep = 0x81;
+ iface = usb_ifnum_to_if(dev, 2);
+
+ if (!iface || iface->num_altsetting == 0)
+ return -EINVAL;
+
+ alts = &iface->altsetting[1];
+ goto add_sync_ep;
+ }
+ if (attr == USB_ENDPOINT_SYNC_ASYNC &&
+ altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC &&
+ altsd->bInterfaceProtocol == 2 &&
+ altsd->bNumEndpoints == 1 &&
+ USB_ID_VENDOR(subs->stream->chip->usb_id) == 0x0582 /* Roland */ &&
+ search_roland_implicit_fb(dev, altsd->bInterfaceNumber + 1,
+ altsd->bAlternateSetting,
+ &alts, &ep) >= 0) {
+ goto add_sync_ep;
+ }
+
+ /* No quirk */
+ return 0;
+
+add_sync_ep:
+ subs->sync_endpoint = snd_usb_add_endpoint(subs->stream->chip,
+ alts, ep, !subs->direction,
+ SND_USB_ENDPOINT_TYPE_DATA);
+ if (!subs->sync_endpoint)
+ return -EINVAL;
+
+ subs->data_endpoint->sync_master = subs->sync_endpoint;
+
+ return 0;
+}
+
+static int set_sync_endpoint(struct snd_usb_substream *subs,
+ struct audioformat *fmt,
+ struct usb_device *dev,
+ struct usb_host_interface *alts,
+ struct usb_interface_descriptor *altsd)
+{
+ int is_playback = subs->direction == SNDRV_PCM_STREAM_PLAYBACK;
+ unsigned int ep, attr;
+ bool implicit_fb;
+ int err;
+
+ /* we need a sync pipe in async OUT or adaptive IN mode */
+ /* check the number of EP, since some devices have broken
+ * descriptors which fool us. if it has only one EP,
+ * assume it as adaptive-out or sync-in.
+ */
+ attr = fmt->ep_attr & USB_ENDPOINT_SYNCTYPE;
+
+ err = set_sync_ep_implicit_fb_quirk(subs, dev, altsd, attr);
+ if (err < 0)
+ return err;
+
+ if (altsd->bNumEndpoints < 2)
+ return 0;
+
+ if ((is_playback && attr != USB_ENDPOINT_SYNC_ASYNC) ||
+ (!is_playback && attr != USB_ENDPOINT_SYNC_ADAPTIVE))
+ return 0;
+
+ /* check sync-pipe endpoint */
+ /* ... and check descriptor size before accessing bSynchAddress
+ because there is a version of the SB Audigy 2 NX firmware lacking
+ the audio fields in the endpoint descriptors */
+ if ((get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_ISOC ||
+ (get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
+ get_endpoint(alts, 1)->bSynchAddress != 0)) {
+ snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. bmAttributes %02x, bLength %d, bSynchAddress %02x\n",
+ dev->devnum, fmt->iface, fmt->altsetting,
+ get_endpoint(alts, 1)->bmAttributes,
+ get_endpoint(alts, 1)->bLength,
+ get_endpoint(alts, 1)->bSynchAddress);
+ return -EINVAL;
+ }
+ ep = get_endpoint(alts, 1)->bEndpointAddress;
+ if (get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
+ ((is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress | USB_DIR_IN)) ||
+ (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)))) {
+ snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. is_playback %d, ep %02x, bSynchAddress %02x\n",
+ dev->devnum, fmt->iface, fmt->altsetting,
+ is_playback, ep, get_endpoint(alts, 0)->bSynchAddress);
+ return -EINVAL;
+ }
+
+ implicit_fb = (get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_USAGE_MASK)
+ == USB_ENDPOINT_USAGE_IMPLICIT_FB;
+
+ subs->sync_endpoint = snd_usb_add_endpoint(subs->stream->chip,
+ alts, ep, !subs->direction,
+ implicit_fb ?
+ SND_USB_ENDPOINT_TYPE_DATA :
+ SND_USB_ENDPOINT_TYPE_SYNC);
+ if (!subs->sync_endpoint)
+ return -EINVAL;
+
+ subs->data_endpoint->sync_master = subs->sync_endpoint;
+
+ return 0;
+}
+
/*
* find a matching format and set up the interface
*/
@@ -336,9 +467,7 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt)
struct usb_host_interface *alts;
struct usb_interface_descriptor *altsd;
struct usb_interface *iface;
- unsigned int ep, attr;
- int is_playback = subs->direction == SNDRV_PCM_STREAM_PLAYBACK;
- int err, implicit_fb = 0;
+ int err;
iface = usb_ifnum_to_if(dev, fmt->iface);
if (WARN_ON(!iface))
@@ -383,118 +512,22 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt)
subs->data_endpoint = snd_usb_add_endpoint(subs->stream->chip,
alts, fmt->endpoint, subs->direction,
SND_USB_ENDPOINT_TYPE_DATA);
+
if (!subs->data_endpoint)
return -EINVAL;
- /* we need a sync pipe in async OUT or adaptive IN mode */
- /* check the number of EP, since some devices have broken
- * descriptors which fool us. if it has only one EP,
- * assume it as adaptive-out or sync-in.
- */
- attr = fmt->ep_attr & USB_ENDPOINT_SYNCTYPE;
-
- switch (subs->stream->chip->usb_id) {
- case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */
- case USB_ID(0x0763, 0x2031): /* M-Audio Fast Track C600 */
- if (is_playback) {
- implicit_fb = 1;
- ep = 0x81;
- iface = usb_ifnum_to_if(dev, 3);
-
- if (!iface || iface->num_altsetting == 0)
- return -EINVAL;
-
- alts = &iface->altsetting[1];
- goto add_sync_ep;
- }
- break;
- case USB_ID(0x0763, 0x2080): /* M-Audio FastTrack Ultra */
- case USB_ID(0x0763, 0x2081):
- if (is_playback) {
- implicit_fb = 1;
- ep = 0x81;
- iface = usb_ifnum_to_if(dev, 2);
-
- if (!iface || iface->num_altsetting == 0)
- return -EINVAL;
-
- alts = &iface->altsetting[1];
- goto add_sync_ep;
- }
- }
- if (is_playback &&
- attr == USB_ENDPOINT_SYNC_ASYNC &&
- altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC &&
- altsd->bInterfaceProtocol == 2 &&
- altsd->bNumEndpoints == 1 &&
- USB_ID_VENDOR(subs->stream->chip->usb_id) == 0x0582 /* Roland */ &&
- search_roland_implicit_fb(dev, altsd->bInterfaceNumber + 1,
- altsd->bAlternateSetting,
- &alts, &ep) >= 0) {
- implicit_fb = 1;
- goto add_sync_ep;
- }
-
- if (((is_playback && attr == USB_ENDPOINT_SYNC_ASYNC) ||
- (!is_playback && attr == USB_ENDPOINT_SYNC_ADAPTIVE)) &&
- altsd->bNumEndpoints >= 2) {
- /* check sync-pipe endpoint */
- /* ... and check descriptor size before accessing bSynchAddress
- because there is a version of the SB Audigy 2 NX firmware lacking
- the audio fields in the endpoint descriptors */
- if ((get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_ISOC ||
- (get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
- get_endpoint(alts, 1)->bSynchAddress != 0 &&
- !implicit_fb)) {
- snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. bmAttributes %02x, bLength %d, bSynchAddress %02x\n",
- dev->devnum, fmt->iface, fmt->altsetting,
- get_endpoint(alts, 1)->bmAttributes,
- get_endpoint(alts, 1)->bLength,
- get_endpoint(alts, 1)->bSynchAddress);
- return -EINVAL;
- }
- ep = get_endpoint(alts, 1)->bEndpointAddress;
- if (!implicit_fb &&
- get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
- (( is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress | USB_DIR_IN)) ||
- (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)))) {
- snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. is_playback %d, ep %02x, bSynchAddress %02x\n",
- dev->devnum, fmt->iface, fmt->altsetting,
- is_playback, ep, get_endpoint(alts, 0)->bSynchAddress);
- return -EINVAL;
- }
-
- implicit_fb = (get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_USAGE_MASK)
- == USB_ENDPOINT_USAGE_IMPLICIT_FB;
-
-add_sync_ep:
- subs->sync_endpoint = snd_usb_add_endpoint(subs->stream->chip,
- alts, ep, !subs->direction,
- implicit_fb ?
- SND_USB_ENDPOINT_TYPE_DATA :
- SND_USB_ENDPOINT_TYPE_SYNC);
- if (!subs->sync_endpoint)
- return -EINVAL;
-
- subs->data_endpoint->sync_master = subs->sync_endpoint;
- }
+ err = set_sync_endpoint(subs, fmt, dev, alts, altsd);
+ if (err < 0)
+ return err;
- if ((err = snd_usb_init_pitch(subs->stream->chip, fmt->iface, alts, fmt)) < 0)
+ err = snd_usb_init_pitch(subs->stream->chip, fmt->iface, alts, fmt);
+ if (err < 0)
return err;
subs->cur_audiofmt = fmt;
snd_usb_set_format_quirk(subs, fmt);
-#if 0
- printk(KERN_DEBUG
- "setting done: format = %d, rate = %d..%d, channels = %d\n",
- fmt->format, fmt->rate_min, fmt->rate_max, fmt->channels);
- printk(KERN_DEBUG
- " datapipe = 0x%0x, syncpipe = 0x%0x\n",
- subs->datapipe, subs->syncpipe);
-#endif
-
return 0;
}
diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c
index 1f9bbd55553f..5a51b18c50fe 100644
--- a/sound/usb/usx2y/usbusx2y.c
+++ b/sound/usb/usx2y/usbusx2y.c
@@ -305,11 +305,9 @@ static void usX2Y_unlinkSeq(struct snd_usX2Y_AsyncSeq *S)
{
int i;
for (i = 0; i < URBS_AsyncSeq; ++i) {
- if (S[i].urb) {
- usb_kill_urb(S->urb[i]);
- usb_free_urb(S->urb[i]);
- S->urb[i] = NULL;
- }
+ usb_kill_urb(S->urb[i]);
+ usb_free_urb(S->urb[i]);
+ S->urb[i] = NULL;
}
kfree(S->buffer);
}