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-rw-r--r--sound/core/pcm_dmaengine.c11
-rw-r--r--sound/core/pcm_iec958.c65
-rw-r--r--sound/core/timer.c24
-rw-r--r--sound/core/timer_compat.c30
-rw-r--r--sound/firewire/dice/dice-stream.c14
-rw-r--r--sound/hda/ext/hdac_ext_bus.c3
-rw-r--r--sound/hda/ext/hdac_ext_controller.c66
-rw-r--r--sound/hda/ext/hdac_ext_stream.c5
-rw-r--r--sound/hda/hdac_device.c10
-rw-r--r--sound/hda/hdac_i915.c77
-rw-r--r--sound/hda/hdac_regmap.c40
-rw-r--r--sound/hda/local.h10
-rw-r--r--sound/isa/sscape.c2
-rw-r--r--sound/pci/hda/hda_generic.c6
-rw-r--r--sound/pci/hda/hda_intel.c63
-rw-r--r--sound/pci/hda/hda_sysfs.c8
-rw-r--r--sound/pci/hda/patch_cirrus.c14
-rw-r--r--sound/pci/hda/patch_hdmi.c19
-rw-r--r--sound/pci/hda/patch_realtek.c44
-rw-r--r--sound/pci/pcxhr/pcxhr_core.c1
-rw-r--r--sound/soc/atmel/atmel_ssc_dai.c4
-rw-r--r--sound/soc/au1x/dbdma2.c4
-rw-r--r--sound/soc/bcm/bcm2835-i2s.c27
-rw-r--r--sound/soc/codecs/Kconfig38
-rw-r--r--sound/soc/codecs/Makefile11
-rw-r--r--sound/soc/codecs/ak4642.c7
-rw-r--r--sound/soc/codecs/arizona.c44
-rw-r--r--sound/soc/codecs/arizona.h2
-rw-r--r--sound/soc/codecs/cs35l32.c17
-rw-r--r--sound/soc/codecs/cs42l56.c2
-rw-r--r--sound/soc/codecs/cs47l24.c49
-rw-r--r--sound/soc/codecs/da7213.c99
-rw-r--r--sound/soc/codecs/da7213.h35
-rw-r--r--sound/soc/codecs/da7218.c32
-rw-r--r--sound/soc/codecs/da7218.h21
-rw-r--r--sound/soc/codecs/da7219.c38
-rw-r--r--sound/soc/codecs/da7219.h20
-rw-r--r--sound/soc/codecs/es8328.c194
-rw-r--r--sound/soc/codecs/es8328.h23
-rw-r--r--sound/soc/codecs/hdac_hdmi.c288
-rw-r--r--sound/soc/codecs/hdmi-codec.c432
-rw-r--r--sound/soc/codecs/max98371.c441
-rw-r--r--sound/soc/codecs/max98371.h67
-rw-r--r--sound/soc/codecs/nau8825.c126
-rw-r--r--sound/soc/codecs/pcm5102a.c69
-rw-r--r--sound/soc/codecs/rt298.c70
-rw-r--r--sound/soc/codecs/rt298.h2
-rw-r--r--sound/soc/codecs/rt5640.c2
-rw-r--r--sound/soc/codecs/rt5640.h36
-rw-r--r--sound/soc/codecs/rt5645.c16
-rw-r--r--sound/soc/codecs/tas571x.c141
-rw-r--r--sound/soc/codecs/tas571x.h22
-rw-r--r--sound/soc/codecs/tas5720.c620
-rw-r--r--sound/soc/codecs/tas5720.h90
-rw-r--r--sound/soc/codecs/tlv320aic32x4-i2c.c74
-rw-r--r--sound/soc/codecs/tlv320aic32x4-spi.c76
-rw-r--r--sound/soc/codecs/tlv320aic32x4.c279
-rw-r--r--sound/soc/codecs/tlv320aic32x4.h7
-rw-r--r--sound/soc/codecs/twl6040.c16
-rw-r--r--sound/soc/codecs/wm5102.c9
-rw-r--r--sound/soc/codecs/wm5110.c8
-rw-r--r--sound/soc/codecs/wm8962.c2
-rw-r--r--sound/soc/codecs/wm8997.c2
-rw-r--r--sound/soc/codecs/wm8998.c2
-rw-r--r--sound/soc/codecs/wm_adsp.c292
-rw-r--r--sound/soc/codecs/wm_adsp.h1
-rw-r--r--sound/soc/davinci/Kconfig6
-rw-r--r--sound/soc/davinci/davinci-i2s.c80
-rw-r--r--sound/soc/davinci/davinci-mcasp.c92
-rw-r--r--sound/soc/davinci/davinci-mcasp.h5
-rw-r--r--sound/soc/dwc/designware_i2s.c14
-rw-r--r--sound/soc/fsl/fsl_sai.c24
-rw-r--r--sound/soc/fsl/fsl_ssi.c81
-rw-r--r--sound/soc/fsl/imx-pcm-fiq.c2
-rw-r--r--sound/soc/generic/simple-card.c1
-rw-r--r--sound/soc/intel/Kconfig17
-rw-r--r--sound/soc/intel/atom/sst-atom-controls.c2
-rw-r--r--sound/soc/intel/boards/Makefile2
-rw-r--r--sound/soc/intel/boards/broadwell.c2
-rw-r--r--sound/soc/intel/boards/bxt_rt298.c353
-rw-r--r--sound/soc/intel/boards/bytcr_rt5640.c2
-rw-r--r--sound/soc/intel/boards/bytcr_rt5651.c2
-rw-r--r--sound/soc/intel/boards/cht_bsw_max98090_ti.c2
-rw-r--r--sound/soc/intel/boards/cht_bsw_rt5645.c2
-rw-r--r--sound/soc/intel/boards/cht_bsw_rt5672.c2
-rw-r--r--sound/soc/intel/boards/haswell.c2
-rw-r--r--sound/soc/intel/boards/skl_nau88l25_max98357a.c87
-rw-r--r--sound/soc/intel/boards/skl_nau88l25_ssm4567.c86
-rw-r--r--sound/soc/intel/boards/skl_rt286.c59
-rw-r--r--sound/soc/intel/common/sst-acpi.h9
-rw-r--r--sound/soc/intel/haswell/sst-haswell-ipc.c2
-rw-r--r--sound/soc/intel/haswell/sst-haswell-pcm.c2
-rw-r--r--sound/soc/intel/skylake/Makefile2
-rw-r--r--sound/soc/intel/skylake/bxt-sst.c328
-rw-r--r--sound/soc/intel/skylake/skl-messages.c122
-rw-r--r--sound/soc/intel/skylake/skl-nhlt.c15
-rw-r--r--sound/soc/intel/skylake/skl-pcm.c119
-rw-r--r--sound/soc/intel/skylake/skl-sst-dsp.c3
-rw-r--r--sound/soc/intel/skylake/skl-sst-dsp.h15
-rw-r--r--sound/soc/intel/skylake/skl-sst.c13
-rw-r--r--sound/soc/intel/skylake/skl-topology.c69
-rw-r--r--sound/soc/intel/skylake/skl-topology.h10
-rw-r--r--sound/soc/intel/skylake/skl-tplg-interface.h2
-rw-r--r--sound/soc/intel/skylake/skl.c71
-rw-r--r--sound/soc/intel/skylake/skl.h6
-rw-r--r--sound/soc/kirkwood/Kconfig1
-rw-r--r--sound/soc/mediatek/Kconfig1
-rw-r--r--sound/soc/mediatek/mt8173-rt5650-rt5676.c27
-rw-r--r--sound/soc/mediatek/mt8173-rt5650.c50
-rw-r--r--sound/soc/mediatek/mtk-afe-pcm.c2
-rw-r--r--sound/soc/omap/mcbsp.c8
-rw-r--r--sound/soc/omap/omap-pcm.c2
-rw-r--r--sound/soc/pxa/brownstone.c1
-rw-r--r--sound/soc/pxa/mioa701_wm9713.c1
-rw-r--r--sound/soc/pxa/mmp-pcm.c1
-rw-r--r--sound/soc/pxa/mmp-sspa.c1
-rw-r--r--sound/soc/pxa/palm27x.c1
-rw-r--r--sound/soc/pxa/pxa-ssp.c1
-rw-r--r--sound/soc/pxa/pxa2xx-ac97.c1
-rw-r--r--sound/soc/pxa/pxa2xx-pcm.c1
-rw-r--r--sound/soc/rockchip/rockchip_i2s.c87
-rw-r--r--sound/soc/sh/rcar/adg.c8
-rw-r--r--sound/soc/sh/rcar/dma.c12
-rw-r--r--sound/soc/sh/rcar/rsnd.h13
-rw-r--r--sound/soc/sh/rcar/src.c4
-rw-r--r--sound/soc/soc-core.c14
-rw-r--r--sound/soc/soc-dapm.c7
-rw-r--r--sound/soc/soc-generic-dmaengine-pcm.c57
-rw-r--r--sound/soc/soc-topology.c48
-rw-r--r--sound/usb/Kconfig4
-rw-r--r--sound/usb/Makefile2
-rw-r--r--sound/usb/card.c14
-rw-r--r--sound/usb/card.h3
-rw-r--r--sound/usb/media.c318
-rw-r--r--sound/usb/media.h72
-rw-r--r--sound/usb/mixer.h3
-rw-r--r--sound/usb/mixer_maps.c14
-rw-r--r--sound/usb/pcm.c28
-rw-r--r--sound/usb/quirks-table.h1
-rw-r--r--sound/usb/quirks.c9
-rw-r--r--sound/usb/stream.c8
-rw-r--r--sound/usb/usbaudio.h6
142 files changed, 5546 insertions, 1367 deletions
diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c
index 697c166acf05..8eb58c709b14 100644
--- a/sound/core/pcm_dmaengine.c
+++ b/sound/core/pcm_dmaengine.c
@@ -106,8 +106,9 @@ EXPORT_SYMBOL_GPL(snd_hwparams_to_dma_slave_config);
* direction of the substream. If the substream is a playback stream the dst
* fields will be initialized, if it is a capture stream the src fields will be
* initialized. The {dst,src}_addr_width field will only be initialized if the
- * addr_width field of the DAI DMA data struct is not equal to
- * DMA_SLAVE_BUSWIDTH_UNDEFINED.
+ * SND_DMAENGINE_PCM_DAI_FLAG_PACK flag is set or if the addr_width field of
+ * the DAI DMA data struct is not equal to DMA_SLAVE_BUSWIDTH_UNDEFINED. If
+ * both conditions are met the latter takes priority.
*/
void snd_dmaengine_pcm_set_config_from_dai_data(
const struct snd_pcm_substream *substream,
@@ -117,11 +118,17 @@ void snd_dmaengine_pcm_set_config_from_dai_data(
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
slave_config->dst_addr = dma_data->addr;
slave_config->dst_maxburst = dma_data->maxburst;
+ if (dma_data->flags & SND_DMAENGINE_PCM_DAI_FLAG_PACK)
+ slave_config->dst_addr_width =
+ DMA_SLAVE_BUSWIDTH_UNDEFINED;
if (dma_data->addr_width != DMA_SLAVE_BUSWIDTH_UNDEFINED)
slave_config->dst_addr_width = dma_data->addr_width;
} else {
slave_config->src_addr = dma_data->addr;
slave_config->src_maxburst = dma_data->maxburst;
+ if (dma_data->flags & SND_DMAENGINE_PCM_DAI_FLAG_PACK)
+ slave_config->src_addr_width =
+ DMA_SLAVE_BUSWIDTH_UNDEFINED;
if (dma_data->addr_width != DMA_SLAVE_BUSWIDTH_UNDEFINED)
slave_config->src_addr_width = dma_data->addr_width;
}
diff --git a/sound/core/pcm_iec958.c b/sound/core/pcm_iec958.c
index 36b2d7aca1bd..5e6aed64f451 100644
--- a/sound/core/pcm_iec958.c
+++ b/sound/core/pcm_iec958.c
@@ -9,30 +9,18 @@
#include <linux/types.h>
#include <sound/asoundef.h>
#include <sound/pcm.h>
+#include <sound/pcm_params.h>
#include <sound/pcm_iec958.h>
-/**
- * snd_pcm_create_iec958_consumer - create consumer format IEC958 channel status
- * @runtime: pcm runtime structure with ->rate filled in
- * @cs: channel status buffer, at least four bytes
- * @len: length of channel status buffer
- *
- * Create the consumer format channel status data in @cs of maximum size
- * @len corresponding to the parameters of the PCM runtime @runtime.
- *
- * Drivers may wish to tweak the contents of the buffer after creation.
- *
- * Returns: length of buffer, or negative error code if something failed.
- */
-int snd_pcm_create_iec958_consumer(struct snd_pcm_runtime *runtime, u8 *cs,
- size_t len)
+static int create_iec958_consumer(uint rate, uint sample_width,
+ u8 *cs, size_t len)
{
unsigned int fs, ws;
if (len < 4)
return -EINVAL;
- switch (runtime->rate) {
+ switch (rate) {
case 32000:
fs = IEC958_AES3_CON_FS_32000;
break;
@@ -59,7 +47,7 @@ int snd_pcm_create_iec958_consumer(struct snd_pcm_runtime *runtime, u8 *cs,
}
if (len > 4) {
- switch (snd_pcm_format_width(runtime->format)) {
+ switch (sample_width) {
case 16:
ws = IEC958_AES4_CON_WORDLEN_20_16;
break;
@@ -71,6 +59,7 @@ int snd_pcm_create_iec958_consumer(struct snd_pcm_runtime *runtime, u8 *cs,
IEC958_AES4_CON_MAX_WORDLEN_24;
break;
case 24:
+ case 32: /* Assume 24-bit width for 32-bit samples. */
ws = IEC958_AES4_CON_WORDLEN_24_20 |
IEC958_AES4_CON_MAX_WORDLEN_24;
break;
@@ -92,4 +81,46 @@ int snd_pcm_create_iec958_consumer(struct snd_pcm_runtime *runtime, u8 *cs,
return len;
}
+
+/**
+ * snd_pcm_create_iec958_consumer - create consumer format IEC958 channel status
+ * @runtime: pcm runtime structure with ->rate filled in
+ * @cs: channel status buffer, at least four bytes
+ * @len: length of channel status buffer
+ *
+ * Create the consumer format channel status data in @cs of maximum size
+ * @len corresponding to the parameters of the PCM runtime @runtime.
+ *
+ * Drivers may wish to tweak the contents of the buffer after creation.
+ *
+ * Returns: length of buffer, or negative error code if something failed.
+ */
+int snd_pcm_create_iec958_consumer(struct snd_pcm_runtime *runtime, u8 *cs,
+ size_t len)
+{
+ return create_iec958_consumer(runtime->rate,
+ snd_pcm_format_width(runtime->format),
+ cs, len);
+}
EXPORT_SYMBOL(snd_pcm_create_iec958_consumer);
+
+/**
+ * snd_pcm_create_iec958_consumer_hw_params - create IEC958 channel status
+ * @hw_params: the hw_params instance for extracting rate and sample format
+ * @cs: channel status buffer, at least four bytes
+ * @len: length of channel status buffer
+ *
+ * Create the consumer format channel status data in @cs of maximum size
+ * @len corresponding to the parameters of the PCM runtime @runtime.
+ *
+ * Drivers may wish to tweak the contents of the buffer after creation.
+ *
+ * Returns: length of buffer, or negative error code if something failed.
+ */
+int snd_pcm_create_iec958_consumer_hw_params(struct snd_pcm_hw_params *params,
+ u8 *cs, size_t len)
+{
+ return create_iec958_consumer(params_rate(params), params_width(params),
+ cs, len);
+}
+EXPORT_SYMBOL(snd_pcm_create_iec958_consumer_hw_params);
diff --git a/sound/core/timer.c b/sound/core/timer.c
index aa1b15c155d1..6469bedda2f3 100644
--- a/sound/core/timer.c
+++ b/sound/core/timer.c
@@ -1019,8 +1019,8 @@ static int snd_timer_s_start(struct snd_timer * timer)
njiff += timer->sticks - priv->correction;
priv->correction = 0;
}
- priv->last_expires = priv->tlist.expires = njiff;
- add_timer(&priv->tlist);
+ priv->last_expires = njiff;
+ mod_timer(&priv->tlist, njiff);
return 0;
}
@@ -1502,17 +1502,13 @@ static int snd_timer_user_ginfo(struct file *file,
return err;
}
-static int snd_timer_user_gparams(struct file *file,
- struct snd_timer_gparams __user *_gparams)
+static int timer_set_gparams(struct snd_timer_gparams *gparams)
{
- struct snd_timer_gparams gparams;
struct snd_timer *t;
int err;
- if (copy_from_user(&gparams, _gparams, sizeof(gparams)))
- return -EFAULT;
mutex_lock(&register_mutex);
- t = snd_timer_find(&gparams.tid);
+ t = snd_timer_find(&gparams->tid);
if (!t) {
err = -ENODEV;
goto _error;
@@ -1525,12 +1521,22 @@ static int snd_timer_user_gparams(struct file *file,
err = -ENOSYS;
goto _error;
}
- err = t->hw.set_period(t, gparams.period_num, gparams.period_den);
+ err = t->hw.set_period(t, gparams->period_num, gparams->period_den);
_error:
mutex_unlock(&register_mutex);
return err;
}
+static int snd_timer_user_gparams(struct file *file,
+ struct snd_timer_gparams __user *_gparams)
+{
+ struct snd_timer_gparams gparams;
+
+ if (copy_from_user(&gparams, _gparams, sizeof(gparams)))
+ return -EFAULT;
+ return timer_set_gparams(&gparams);
+}
+
static int snd_timer_user_gstatus(struct file *file,
struct snd_timer_gstatus __user *_gstatus)
{
diff --git a/sound/core/timer_compat.c b/sound/core/timer_compat.c
index 2e908225d754..6a437eb66115 100644
--- a/sound/core/timer_compat.c
+++ b/sound/core/timer_compat.c
@@ -22,6 +22,19 @@
#include <linux/compat.h>
+/*
+ * ILP32/LP64 has different size for 'long' type. Additionally, the size
+ * of storage alignment differs depending on architectures. Here, '__packed'
+ * qualifier is used so that the size of this structure is multiple of 4 and
+ * it fits to any architectures with 32 bit storage alignment.
+ */
+struct snd_timer_gparams32 {
+ struct snd_timer_id tid;
+ u32 period_num;
+ u32 period_den;
+ unsigned char reserved[32];
+} __packed;
+
struct snd_timer_info32 {
u32 flags;
s32 card;
@@ -32,6 +45,19 @@ struct snd_timer_info32 {
unsigned char reserved[64];
};
+static int snd_timer_user_gparams_compat(struct file *file,
+ struct snd_timer_gparams32 __user *user)
+{
+ struct snd_timer_gparams gparams;
+
+ if (copy_from_user(&gparams.tid, &user->tid, sizeof(gparams.tid)) ||
+ get_user(gparams.period_num, &user->period_num) ||
+ get_user(gparams.period_den, &user->period_den))
+ return -EFAULT;
+
+ return timer_set_gparams(&gparams);
+}
+
static int snd_timer_user_info_compat(struct file *file,
struct snd_timer_info32 __user *_info)
{
@@ -99,6 +125,7 @@ static int snd_timer_user_status_compat(struct file *file,
*/
enum {
+ SNDRV_TIMER_IOCTL_GPARAMS32 = _IOW('T', 0x04, struct snd_timer_gparams32),
SNDRV_TIMER_IOCTL_INFO32 = _IOR('T', 0x11, struct snd_timer_info32),
SNDRV_TIMER_IOCTL_STATUS32 = _IOW('T', 0x14, struct snd_timer_status32),
#ifdef CONFIG_X86_X32
@@ -114,7 +141,6 @@ static long snd_timer_user_ioctl_compat(struct file *file, unsigned int cmd, uns
case SNDRV_TIMER_IOCTL_PVERSION:
case SNDRV_TIMER_IOCTL_TREAD:
case SNDRV_TIMER_IOCTL_GINFO:
- case SNDRV_TIMER_IOCTL_GPARAMS:
case SNDRV_TIMER_IOCTL_GSTATUS:
case SNDRV_TIMER_IOCTL_SELECT:
case SNDRV_TIMER_IOCTL_PARAMS:
@@ -128,6 +154,8 @@ static long snd_timer_user_ioctl_compat(struct file *file, unsigned int cmd, uns
case SNDRV_TIMER_IOCTL_PAUSE_OLD:
case SNDRV_TIMER_IOCTL_NEXT_DEVICE:
return snd_timer_user_ioctl(file, cmd, (unsigned long)argp);
+ case SNDRV_TIMER_IOCTL_GPARAMS32:
+ return snd_timer_user_gparams_compat(file, argp);
case SNDRV_TIMER_IOCTL_INFO32:
return snd_timer_user_info_compat(file, argp);
case SNDRV_TIMER_IOCTL_STATUS32:
diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c
index 845d5e5884a4..ec4db3a514fc 100644
--- a/sound/firewire/dice/dice-stream.c
+++ b/sound/firewire/dice/dice-stream.c
@@ -446,18 +446,12 @@ end:
void snd_dice_stream_destroy_duplex(struct snd_dice *dice)
{
- struct reg_params tx_params, rx_params;
-
- snd_dice_transaction_clear_enable(dice);
+ unsigned int i;
- if (get_register_params(dice, &tx_params, &rx_params) == 0) {
- stop_streams(dice, AMDTP_IN_STREAM, &tx_params);
- stop_streams(dice, AMDTP_OUT_STREAM, &rx_params);
+ for (i = 0; i < MAX_STREAMS; i++) {
+ destroy_stream(dice, AMDTP_IN_STREAM, i);
+ destroy_stream(dice, AMDTP_OUT_STREAM, i);
}
-
- release_resources(dice);
-
- dice->substreams_counter = 0;
}
void snd_dice_stream_update_duplex(struct snd_dice *dice)
diff --git a/sound/hda/ext/hdac_ext_bus.c b/sound/hda/ext/hdac_ext_bus.c
index 2433f7c81472..3b7ae24900fd 100644
--- a/sound/hda/ext/hdac_ext_bus.c
+++ b/sound/hda/ext/hdac_ext_bus.c
@@ -105,6 +105,9 @@ int snd_hdac_ext_bus_init(struct hdac_ext_bus *ebus, struct device *dev,
INIT_LIST_HEAD(&ebus->hlink_list);
ebus->idx = idx++;
+ mutex_init(&ebus->lock);
+ ebus->cmd_dma_state = true;
+
return 0;
}
EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_init);
diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c
index 548cc1e4114b..860f8cad6602 100644
--- a/sound/hda/ext/hdac_ext_controller.c
+++ b/sound/hda/ext/hdac_ext_controller.c
@@ -186,6 +186,9 @@ int snd_hdac_ext_bus_get_ml_capabilities(struct hdac_ext_bus *ebus)
hlink->lcaps = readl(hlink->ml_addr + AZX_REG_ML_LCAP);
hlink->lsdiid = readw(hlink->ml_addr + AZX_REG_ML_LSDIID);
+ /* since link in On, update the ref */
+ hlink->ref_count = 1;
+
list_add_tail(&hlink->list, &ebus->hlink_list);
}
@@ -327,3 +330,66 @@ int snd_hdac_ext_bus_link_power_down_all(struct hdac_ext_bus *ebus)
return 0;
}
EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_power_down_all);
+
+int snd_hdac_ext_bus_link_get(struct hdac_ext_bus *ebus,
+ struct hdac_ext_link *link)
+{
+ int ret = 0;
+
+ mutex_lock(&ebus->lock);
+
+ /*
+ * if we move from 0 to 1, count will be 1 so power up this link
+ * as well, also check the dma status and trigger that
+ */
+ if (++link->ref_count == 1) {
+ if (!ebus->cmd_dma_state) {
+ snd_hdac_bus_init_cmd_io(&ebus->bus);
+ ebus->cmd_dma_state = true;
+ }
+
+ ret = snd_hdac_ext_bus_link_power_up(link);
+ }
+
+ mutex_unlock(&ebus->lock);
+ return ret;
+}
+EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_get);
+
+int snd_hdac_ext_bus_link_put(struct hdac_ext_bus *ebus,
+ struct hdac_ext_link *link)
+{
+ int ret = 0;
+ struct hdac_ext_link *hlink;
+ bool link_up = false;
+
+ mutex_lock(&ebus->lock);
+
+ /*
+ * if we move from 1 to 0, count will be 0
+ * so power down this link as well
+ */
+ if (--link->ref_count == 0) {
+ ret = snd_hdac_ext_bus_link_power_down(link);
+
+ /*
+ * now check if all links are off, if so turn off
+ * cmd dma as well
+ */
+ list_for_each_entry(hlink, &ebus->hlink_list, list) {
+ if (hlink->ref_count) {
+ link_up = true;
+ break;
+ }
+ }
+
+ if (!link_up) {
+ snd_hdac_bus_stop_cmd_io(&ebus->bus);
+ ebus->cmd_dma_state = false;
+ }
+ }
+
+ mutex_unlock(&ebus->lock);
+ return ret;
+}
+EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_put);
diff --git a/sound/hda/ext/hdac_ext_stream.c b/sound/hda/ext/hdac_ext_stream.c
index 023cc4cad5c1..626f3bb24c55 100644
--- a/sound/hda/ext/hdac_ext_stream.c
+++ b/sound/hda/ext/hdac_ext_stream.c
@@ -104,12 +104,11 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_init_all);
*/
void snd_hdac_stream_free_all(struct hdac_ext_bus *ebus)
{
- struct hdac_stream *s;
+ struct hdac_stream *s, *_s;
struct hdac_ext_stream *stream;
struct hdac_bus *bus = ebus_to_hbus(ebus);
- while (!list_empty(&bus->stream_list)) {
- s = list_first_entry(&bus->stream_list, struct hdac_stream, list);
+ list_for_each_entry_safe(s, _s, &bus->stream_list, list) {
stream = stream_to_hdac_ext_stream(s);
snd_hdac_ext_stream_decouple(ebus, stream, false);
list_del(&s->list);
diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c
index d1a4d6973330..03c9872c31cf 100644
--- a/sound/hda/hdac_device.c
+++ b/sound/hda/hdac_device.c
@@ -299,13 +299,11 @@ EXPORT_SYMBOL_GPL(_snd_hdac_read_parm);
int snd_hdac_read_parm_uncached(struct hdac_device *codec, hda_nid_t nid,
int parm)
{
- int val;
+ unsigned int cmd, val;
- if (codec->regmap)
- regcache_cache_bypass(codec->regmap, true);
- val = snd_hdac_read_parm(codec, nid, parm);
- if (codec->regmap)
- regcache_cache_bypass(codec->regmap, false);
+ cmd = snd_hdac_regmap_encode_verb(nid, AC_VERB_PARAMETERS) | parm;
+ if (snd_hdac_regmap_read_raw_uncached(codec, cmd, &val) < 0)
+ return -1;
return val;
}
EXPORT_SYMBOL_GPL(snd_hdac_read_parm_uncached);
diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c
index fb96aead8257..607bbeaebddf 100644
--- a/sound/hda/hdac_i915.c
+++ b/sound/hda/hdac_i915.c
@@ -20,6 +20,7 @@
#include <sound/core.h>
#include <sound/hdaudio.h>
#include <sound/hda_i915.h>
+#include <sound/hda_register.h>
static struct i915_audio_component *hdac_acomp;
@@ -97,26 +98,65 @@ int snd_hdac_display_power(struct hdac_bus *bus, bool enable)
}
EXPORT_SYMBOL_GPL(snd_hdac_display_power);
+#define CONTROLLER_IN_GPU(pci) (((pci)->device == 0x0a0c) || \
+ ((pci)->device == 0x0c0c) || \
+ ((pci)->device == 0x0d0c) || \
+ ((pci)->device == 0x160c))
+
/**
- * snd_hdac_get_display_clk - Get CDCLK in kHz
+ * snd_hdac_i915_set_bclk - Reprogram BCLK for HSW/BDW
* @bus: HDA core bus
*
- * This function is supposed to be used only by a HD-audio controller
- * driver that needs the interaction with i915 graphics.
+ * Intel HSW/BDW display HDA controller is in GPU. Both its power and link BCLK
+ * depends on GPU. Two Extended Mode registers EM4 (M value) and EM5 (N Value)
+ * are used to convert CDClk (Core Display Clock) to 24MHz BCLK:
+ * BCLK = CDCLK * M / N
+ * The values will be lost when the display power well is disabled and need to
+ * be restored to avoid abnormal playback speed.
*
- * This function queries CDCLK value in kHz from the graphics driver and
- * returns the value. A negative code is returned in error.
+ * Call this function at initializing and changing power well, as well as
+ * at ELD notifier for the hotplug.
*/
-int snd_hdac_get_display_clk(struct hdac_bus *bus)
+void snd_hdac_i915_set_bclk(struct hdac_bus *bus)
{
struct i915_audio_component *acomp = bus->audio_component;
+ struct pci_dev *pci = to_pci_dev(bus->dev);
+ int cdclk_freq;
+ unsigned int bclk_m, bclk_n;
+
+ if (!acomp || !acomp->ops || !acomp->ops->get_cdclk_freq)
+ return; /* only for i915 binding */
+ if (!CONTROLLER_IN_GPU(pci))
+ return; /* only HSW/BDW */
+
+ cdclk_freq = acomp->ops->get_cdclk_freq(acomp->dev);
+ switch (cdclk_freq) {
+ case 337500:
+ bclk_m = 16;
+ bclk_n = 225;
+ break;
+
+ case 450000:
+ default: /* default CDCLK 450MHz */
+ bclk_m = 4;
+ bclk_n = 75;
+ break;
+
+ case 540000:
+ bclk_m = 4;
+ bclk_n = 90;
+ break;
+
+ case 675000:
+ bclk_m = 8;
+ bclk_n = 225;
+ break;
+ }
- if (!acomp || !acomp->ops)
- return -ENODEV;
-
- return acomp->ops->get_cdclk_freq(acomp->dev);
+ snd_hdac_chip_writew(bus, HSW_EM4, bclk_m);
+ snd_hdac_chip_writew(bus, HSW_EM5, bclk_n);
}
-EXPORT_SYMBOL_GPL(snd_hdac_get_display_clk);
+EXPORT_SYMBOL_GPL(snd_hdac_i915_set_bclk);
/* There is a fixed mapping between audio pin node and display port
* on current Intel platforms:
@@ -267,6 +307,18 @@ int snd_hdac_i915_register_notifier(const struct i915_audio_component_audio_ops
}
EXPORT_SYMBOL_GPL(snd_hdac_i915_register_notifier);
+/* check whether intel graphics is present */
+static bool i915_gfx_present(void)
+{
+ static struct pci_device_id ids[] = {
+ { PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_ANY_ID),
+ .class = PCI_BASE_CLASS_DISPLAY << 16,
+ .class_mask = 0xff << 16 },
+ {}
+ };
+ return pci_dev_present(ids);
+}
+
/**
* snd_hdac_i915_init - Initialize i915 audio component
* @bus: HDA core bus
@@ -286,6 +338,9 @@ int snd_hdac_i915_init(struct hdac_bus *bus)
struct i915_audio_component *acomp;
int ret;
+ if (!i915_gfx_present())
+ return -ENODEV;
+
acomp = kzalloc(sizeof(*acomp), GFP_KERNEL);
if (!acomp)
return -ENOMEM;
diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c
index bdbcd6b75ff6..87041ddd29cb 100644
--- a/sound/hda/hdac_regmap.c
+++ b/sound/hda/hdac_regmap.c
@@ -453,14 +453,30 @@ int snd_hdac_regmap_write_raw(struct hdac_device *codec, unsigned int reg,
EXPORT_SYMBOL_GPL(snd_hdac_regmap_write_raw);
static int reg_raw_read(struct hdac_device *codec, unsigned int reg,
- unsigned int *val)
+ unsigned int *val, bool uncached)
{
- if (!codec->regmap)
+ if (uncached || !codec->regmap)
return hda_reg_read(codec, reg, val);
else
return regmap_read(codec->regmap, reg, val);
}
+static int __snd_hdac_regmap_read_raw(struct hdac_device *codec,
+ unsigned int reg, unsigned int *val,
+ bool uncached)
+{
+ int err;
+
+ err = reg_raw_read(codec, reg, val, uncached);
+ if (err == -EAGAIN) {
+ err = snd_hdac_power_up_pm(codec);
+ if (!err)
+ err = reg_raw_read(codec, reg, val, uncached);
+ snd_hdac_power_down_pm(codec);
+ }
+ return err;
+}
+
/**
* snd_hdac_regmap_read_raw - read a pseudo register with power mgmt
* @codec: the codec object
@@ -472,19 +488,19 @@ static int reg_raw_read(struct hdac_device *codec, unsigned int reg,
int snd_hdac_regmap_read_raw(struct hdac_device *codec, unsigned int reg,
unsigned int *val)
{
- int err;
-
- err = reg_raw_read(codec, reg, val);
- if (err == -EAGAIN) {
- err = snd_hdac_power_up_pm(codec);
- if (!err)
- err = reg_raw_read(codec, reg, val);
- snd_hdac_power_down_pm(codec);
- }
- return err;
+ return __snd_hdac_regmap_read_raw(codec, reg, val, false);
}
EXPORT_SYMBOL_GPL(snd_hdac_regmap_read_raw);
+/* Works like snd_hdac_regmap_read_raw(), but this doesn't read from the
+ * cache but always via hda verbs.
+ */
+int snd_hdac_regmap_read_raw_uncached(struct hdac_device *codec,
+ unsigned int reg, unsigned int *val)
+{
+ return __snd_hdac_regmap_read_raw(codec, reg, val, true);
+}
+
/**
* snd_hdac_regmap_update_raw - update a pseudo register with power mgmt
* @codec: the codec object
diff --git a/sound/hda/local.h b/sound/hda/local.h
index d692f417ddc0..0d5bb159d538 100644
--- a/sound/hda/local.h
+++ b/sound/hda/local.h
@@ -16,6 +16,16 @@ static inline int get_wcaps_type(unsigned int wcaps)
return (wcaps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
}
+static inline unsigned int get_wcaps_channels(u32 wcaps)
+{
+ unsigned int chans;
+
+ chans = (wcaps & AC_WCAP_CHAN_CNT_EXT) >> 13;
+ chans = (chans + 1) * 2;
+
+ return chans;
+}
+
extern const struct attribute_group *hdac_dev_attr_groups[];
int hda_widget_sysfs_init(struct hdac_device *codec);
void hda_widget_sysfs_exit(struct hdac_device *codec);
diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c
index 7b248cdf06e2..fdcfa29e2205 100644
--- a/sound/isa/sscape.c
+++ b/sound/isa/sscape.c
@@ -591,7 +591,7 @@ static int sscape_upload_microcode(struct snd_card *card, int version)
}
err = upload_dma_data(sscape, init_fw->data, init_fw->size);
if (err == 0)
- snd_printk(KERN_INFO "sscape: MIDI firmware loaded %d KBs\n",
+ snd_printk(KERN_INFO "sscape: MIDI firmware loaded %zu KBs\n",
init_fw->size >> 10);
release_firmware(init_fw);
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 7ca5b89f088a..dfaf1a93fb8a 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -826,7 +826,7 @@ static hda_nid_t path_power_update(struct hda_codec *codec,
bool allow_powerdown)
{
hda_nid_t nid, changed = 0;
- int i, state;
+ int i, state, power;
for (i = 0; i < path->depth; i++) {
nid = path->path[i];
@@ -838,7 +838,9 @@ static hda_nid_t path_power_update(struct hda_codec *codec,
state = AC_PWRST_D0;
else
state = AC_PWRST_D3;
- if (!snd_hda_check_power_state(codec, nid, state)) {
+ power = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_POWER_STATE, 0);
+ if (power != (state | (state << 4))) {
snd_hda_codec_write(codec, nid, 0,
AC_VERB_SET_POWER_STATE, state);
changed = nid;
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 2624cfe98884..9a0d1445ca5c 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -857,50 +857,6 @@ static int param_set_xint(const char *val, const struct kernel_param *kp)
#define azx_del_card_list(chip) /* NOP */
#endif /* CONFIG_PM */
-/* Intel HSW/BDW display HDA controller is in GPU. Both its power and link BCLK
- * depends on GPU. Two Extended Mode registers EM4 (M value) and EM5 (N Value)
- * are used to convert CDClk (Core Display Clock) to 24MHz BCLK:
- * BCLK = CDCLK * M / N
- * The values will be lost when the display power well is disabled and need to
- * be restored to avoid abnormal playback speed.
- */
-static void haswell_set_bclk(struct hda_intel *hda)
-{
- struct azx *chip = &hda->chip;
- int cdclk_freq;
- unsigned int bclk_m, bclk_n;
-
- if (!hda->need_i915_power)
- return;
-
- cdclk_freq = snd_hdac_get_display_clk(azx_bus(chip));
- switch (cdclk_freq) {
- case 337500:
- bclk_m = 16;
- bclk_n = 225;
- break;
-
- case 450000:
- default: /* default CDCLK 450MHz */
- bclk_m = 4;
- bclk_n = 75;
- break;
-
- case 540000:
- bclk_m = 4;
- bclk_n = 90;
- break;
-
- case 675000:
- bclk_m = 8;
- bclk_n = 225;
- break;
- }
-
- azx_writew(chip, HSW_EM4, bclk_m);
- azx_writew(chip, HSW_EM5, bclk_n);
-}
-
#if defined(CONFIG_PM_SLEEP) || defined(SUPPORT_VGA_SWITCHEROO)
/*
* power management
@@ -958,7 +914,7 @@ static int azx_resume(struct device *dev)
if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL
&& hda->need_i915_power) {
snd_hdac_display_power(azx_bus(chip), true);
- haswell_set_bclk(hda);
+ snd_hdac_i915_set_bclk(azx_bus(chip));
}
if (chip->msi)
if (pci_enable_msi(pci) < 0)
@@ -1058,7 +1014,7 @@ static int azx_runtime_resume(struct device *dev)
bus = azx_bus(chip);
if (hda->need_i915_power) {
snd_hdac_display_power(bus, true);
- haswell_set_bclk(hda);
+ snd_hdac_i915_set_bclk(bus);
} else {
/* toggle codec wakeup bit for STATESTS read */
snd_hdac_set_codec_wakeup(bus, true);
@@ -1796,12 +1752,8 @@ static int azx_first_init(struct azx *chip)
/* initialize chip */
azx_init_pci(chip);
- if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) {
- struct hda_intel *hda;
-
- hda = container_of(chip, struct hda_intel, chip);
- haswell_set_bclk(hda);
- }
+ if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL)
+ snd_hdac_i915_set_bclk(bus);
hda_intel_init_chip(chip, (probe_only[dev] & 2) == 0);
@@ -2232,6 +2184,9 @@ static const struct pci_device_id azx_ids[] = {
/* Broxton-P(Apollolake) */
{ PCI_DEVICE(0x8086, 0x5a98),
.driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_BROXTON },
+ /* Broxton-T */
+ { PCI_DEVICE(0x8086, 0x1a98),
+ .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_BROXTON },
/* Haswell */
{ PCI_DEVICE(0x8086, 0x0a0c),
.driver_data = AZX_DRIVER_HDMI | AZX_DCAPS_INTEL_HASWELL },
@@ -2361,6 +2316,10 @@ static const struct pci_device_id azx_ids[] = {
.driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS },
{ PCI_DEVICE(0x1002, 0xaae8),
.driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS },
+ { PCI_DEVICE(0x1002, 0xaae0),
+ .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS },
+ { PCI_DEVICE(0x1002, 0xaaf0),
+ .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS },
/* VIA VT8251/VT8237A */
{ PCI_DEVICE(0x1106, 0x3288), .driver_data = AZX_DRIVER_VIA },
/* VIA GFX VT7122/VX900 */
diff --git a/sound/pci/hda/hda_sysfs.c b/sound/pci/hda/hda_sysfs.c
index 64e0d1d81ca5..9739fce9e032 100644
--- a/sound/pci/hda/hda_sysfs.c
+++ b/sound/pci/hda/hda_sysfs.c
@@ -141,14 +141,6 @@ static int reconfig_codec(struct hda_codec *codec)
err = snd_hda_codec_configure(codec);
if (err < 0)
goto error;
- /* rebuild PCMs */
- err = snd_hda_codec_build_pcms(codec);
- if (err < 0)
- goto error;
- /* rebuild mixers */
- err = snd_hda_codec_build_controls(codec);
- if (err < 0)
- goto error;
err = snd_card_register(codec->card);
error:
snd_hda_power_down(codec);
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index a47e8ae0eb30..80bbadc83721 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -361,6 +361,7 @@ static int cs_parse_auto_config(struct hda_codec *codec)
{
struct cs_spec *spec = codec->spec;
int err;
+ int i;
err = snd_hda_parse_pin_defcfg(codec, &spec->gen.autocfg, NULL, 0);
if (err < 0)
@@ -370,6 +371,19 @@ static int cs_parse_auto_config(struct hda_codec *codec)
if (err < 0)
return err;
+ /* keep the ADCs powered up when it's dynamically switchable */
+ if (spec->gen.dyn_adc_switch) {
+ unsigned int done = 0;
+ for (i = 0; i < spec->gen.input_mux.num_items; i++) {
+ int idx = spec->gen.dyn_adc_idx[i];
+ if (done & (1 << idx))
+ continue;
+ snd_hda_gen_fix_pin_power(codec,
+ spec->gen.adc_nids[idx]);
+ done |= 1 << idx;
+ }
+ }
+
return 0;
}
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 5af372d01834..a010d704e0e2 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1396,7 +1396,6 @@ static bool hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin,
struct hda_codec *codec = per_pin->codec;
struct hdmi_spec *spec = codec->spec;
struct hdmi_eld *eld = &spec->temp_eld;
- struct hdmi_eld *pin_eld = &per_pin->sink_eld;
hda_nid_t pin_nid = per_pin->pin_nid;
/*
* Always execute a GetPinSense verb here, even when called from
@@ -1413,15 +1412,15 @@ static bool hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin,
present = snd_hda_pin_sense(codec, pin_nid);
mutex_lock(&per_pin->lock);
- pin_eld->monitor_present = !!(present & AC_PINSENSE_PRESENCE);
- if (pin_eld->monitor_present)
+ eld->monitor_present = !!(present & AC_PINSENSE_PRESENCE);
+ if (eld->monitor_present)
eld->eld_valid = !!(present & AC_PINSENSE_ELDV);
else
eld->eld_valid = false;
codec_dbg(codec,
"HDMI status: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n",
- codec->addr, pin_nid, pin_eld->monitor_present, eld->eld_valid);
+ codec->addr, pin_nid, eld->monitor_present, eld->eld_valid);
if (eld->eld_valid) {
if (spec->ops.pin_get_eld(codec, pin_nid, eld->eld_buffer,
@@ -1441,7 +1440,7 @@ static bool hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin,
else
update_eld(codec, per_pin, eld);
- ret = !repoll || !pin_eld->monitor_present || pin_eld->eld_valid;
+ ret = !repoll || !eld->monitor_present || eld->eld_valid;
jack = snd_hda_jack_tbl_get(codec, pin_nid);
if (jack)
@@ -1859,6 +1858,8 @@ static void hdmi_set_chmap(struct hdac_device *hdac, int pcm_idx,
struct hdmi_spec *spec = codec->spec;
struct hdmi_spec_per_pin *per_pin = pcm_idx_to_pin(spec, pcm_idx);
+ if (!per_pin)
+ return;
mutex_lock(&per_pin->lock);
per_pin->chmap_set = true;
memcpy(per_pin->chmap, chmap, ARRAY_SIZE(per_pin->chmap));
@@ -2231,6 +2232,7 @@ static void intel_pin_eld_notify(void *audio_ptr, int port)
if (atomic_read(&(codec)->core.in_pm))
return;
+ snd_hdac_i915_set_bclk(&codec->bus->core);
check_presence_and_report(codec, pin_nid);
}
@@ -3399,6 +3401,9 @@ static int patch_atihdmi(struct hda_codec *codec)
spec->ops.pin_hbr_setup = atihdmi_pin_hbr_setup;
spec->ops.setup_stream = atihdmi_setup_stream;
+ spec->chmap.ops.pin_get_slot_channel = atihdmi_pin_get_slot_channel;
+ spec->chmap.ops.pin_set_slot_channel = atihdmi_pin_set_slot_channel;
+
if (!has_amd_full_remap_support(codec)) {
/* override to ATI/AMD-specific versions with pairwise mapping */
spec->chmap.ops.chmap_cea_alloc_validate_get_type =
@@ -3406,10 +3411,6 @@ static int patch_atihdmi(struct hda_codec *codec)
spec->chmap.ops.cea_alloc_to_tlv_chmap =
atihdmi_paired_cea_alloc_to_tlv_chmap;
spec->chmap.ops.chmap_validate = atihdmi_paired_chmap_validate;
- spec->chmap.ops.pin_get_slot_channel =
- atihdmi_pin_get_slot_channel;
- spec->chmap.ops.pin_set_slot_channel =
- atihdmi_pin_set_slot_channel;
}
/* ATI/AMD converters do not advertise all of their capabilities */
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 4f5ca0b9ce27..4918ffa5ba68 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4759,6 +4759,8 @@ enum {
ALC255_FIXUP_DELL_SPK_NOISE,
ALC225_FIXUP_DELL1_MIC_NO_PRESENCE,
ALC280_FIXUP_HP_HEADSET_MIC,
+ ALC221_FIXUP_HP_FRONT_MIC,
+ ALC292_FIXUP_TPT460,
};
static const struct hda_fixup alc269_fixups[] = {
@@ -5401,6 +5403,19 @@ static const struct hda_fixup alc269_fixups[] = {
.chained = true,
.chain_id = ALC269_FIXUP_HEADSET_MIC,
},
+ [ALC221_FIXUP_HP_FRONT_MIC] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x19, 0x02a19020 }, /* Front Mic */
+ { }
+ },
+ },
+ [ALC292_FIXUP_TPT460] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc_fixup_tpt440_dock,
+ .chained = true,
+ .chain_id = ALC293_FIXUP_LENOVO_SPK_NOISE,
+ },
};
static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -5434,6 +5449,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1028, 0x064a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x064b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x0665, "Dell XPS 13", ALC288_FIXUP_DELL_XPS_13),
+ SND_PCI_QUIRK(0x1028, 0x0669, "Dell Optiplex 9020m", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x069a, "Dell Vostro 5480", ALC290_FIXUP_SUBWOOFER_HSJACK),
SND_PCI_QUIRK(0x1028, 0x06c7, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x06d9, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE),
@@ -5506,6 +5522,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x103c, 0x2336, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
SND_PCI_QUIRK(0x103c, 0x2337, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
SND_PCI_QUIRK(0x103c, 0x221c, "HP EliteBook 755 G2", ALC280_FIXUP_HP_HEADSET_MIC),
+ SND_PCI_QUIRK(0x103c, 0x8256, "HP", ALC221_FIXUP_HP_FRONT_MIC),
SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300),
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),
@@ -5554,7 +5571,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x2218, "Thinkpad X1 Carbon 2nd", ALC292_FIXUP_TPT440_DOCK),
SND_PCI_QUIRK(0x17aa, 0x2223, "ThinkPad T550", ALC292_FIXUP_TPT440_DOCK),
SND_PCI_QUIRK(0x17aa, 0x2226, "ThinkPad X250", ALC292_FIXUP_TPT440_DOCK),
- SND_PCI_QUIRK(0x17aa, 0x2233, "Thinkpad", ALC293_FIXUP_LENOVO_SPK_NOISE),
+ SND_PCI_QUIRK(0x17aa, 0x2233, "Thinkpad", ALC292_FIXUP_TPT460),
SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY),
SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY),
SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI),
@@ -5567,6 +5584,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x5034, "Thinkpad T450", ALC292_FIXUP_TPT440_DOCK),
SND_PCI_QUIRK(0x17aa, 0x5036, "Thinkpad T450s", ALC292_FIXUP_TPT440_DOCK),
SND_PCI_QUIRK(0x17aa, 0x503c, "Thinkpad L450", ALC292_FIXUP_TPT440_DOCK),
+ SND_PCI_QUIRK(0x17aa, 0x504a, "ThinkPad X260", ALC292_FIXUP_TPT440_DOCK),
SND_PCI_QUIRK(0x17aa, 0x504b, "Thinkpad", ALC293_FIXUP_LENOVO_SPK_NOISE),
SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K),
@@ -5649,6 +5667,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
{.id = ALC283_FIXUP_SENSE_COMBO_JACK, .name = "alc283-sense-combo"},
{.id = ALC292_FIXUP_TPT440_DOCK, .name = "tpt440-dock"},
{.id = ALC292_FIXUP_TPT440, .name = "tpt440"},
+ {.id = ALC292_FIXUP_TPT460, .name = "tpt460"},
{}
};
#define ALC225_STANDARD_PINS \
@@ -6406,6 +6425,8 @@ enum {
ALC668_FIXUP_AUTO_MUTE,
ALC668_FIXUP_DELL_DISABLE_AAMIX,
ALC668_FIXUP_DELL_XPS13,
+ ALC662_FIXUP_ASUS_Nx50,
+ ALC668_FIXUP_ASUS_Nx51,
};
static const struct hda_fixup alc662_fixups[] = {
@@ -6646,6 +6667,21 @@ static const struct hda_fixup alc662_fixups[] = {
.type = HDA_FIXUP_FUNC,
.v.func = alc_fixup_bass_chmap,
},
+ [ALC662_FIXUP_ASUS_Nx50] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc_fixup_auto_mute_via_amp,
+ .chained = true,
+ .chain_id = ALC662_FIXUP_BASS_1A
+ },
+ [ALC668_FIXUP_ASUS_Nx51] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ {0x1a, 0x90170151}, /* bass speaker */
+ {}
+ },
+ .chained = true,
+ .chain_id = ALC662_FIXUP_BASS_CHMAP,
+ },
};
static const struct snd_pci_quirk alc662_fixup_tbl[] = {
@@ -6668,10 +6704,14 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = {
SND_PCI_QUIRK(0x1028, 0x0698, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x069f, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800),
- SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_BASS_1A),
+ SND_PCI_QUIRK(0x1043, 0x1080, "Asus UX501VW", ALC668_FIXUP_HEADSET_MODE),
+ SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_ASUS_Nx50),
SND_PCI_QUIRK(0x1043, 0x13df, "Asus N550JX", ALC662_FIXUP_BASS_1A),
+ SND_PCI_QUIRK(0x1043, 0x129d, "Asus N750", ALC662_FIXUP_ASUS_Nx50),
SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_BASS_MODE4_CHMAP),
SND_PCI_QUIRK(0x1043, 0x15a7, "ASUS UX51VZH", ALC662_FIXUP_BASS_16),
+ SND_PCI_QUIRK(0x1043, 0x177d, "ASUS N551", ALC668_FIXUP_ASUS_Nx51),
+ SND_PCI_QUIRK(0x1043, 0x17bd, "ASUS N751", ALC668_FIXUP_ASUS_Nx51),
SND_PCI_QUIRK(0x1043, 0x1b73, "ASUS N55SF", ALC662_FIXUP_BASS_16),
SND_PCI_QUIRK(0x1043, 0x1bf3, "ASUS N76VZ", ALC662_FIXUP_BASS_MODE4_CHMAP),
SND_PCI_QUIRK(0x1043, 0x8469, "ASUS mobo", ALC662_FIXUP_NO_JACK_DETECT),
diff --git a/sound/pci/pcxhr/pcxhr_core.c b/sound/pci/pcxhr/pcxhr_core.c
index c5194f5b150a..d7e71f309299 100644
--- a/sound/pci/pcxhr/pcxhr_core.c
+++ b/sound/pci/pcxhr/pcxhr_core.c
@@ -1341,5 +1341,6 @@ irqreturn_t pcxhr_threaded_irq(int irq, void *dev_id)
}
pcxhr_msg_thread(mgr);
+ mutex_unlock(&mgr->lock);
return IRQ_HANDLED;
}
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index 276897033639..1267e1af0fae 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -652,7 +652,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
rcmr = SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period)
| SSC_BF(RCMR_STTDLY, 1)
| SSC_BF(RCMR_START, SSC_START_RISING_RF)
- | SSC_BF(RCMR_CKI, SSC_CKI_FALLING)
+ | SSC_BF(RCMR_CKI, SSC_CKI_RISING)
| SSC_BF(RCMR_CKO, SSC_CKO_NONE)
| SSC_BF(RCMR_CKS, SSC_CKS_DIV);
@@ -692,7 +692,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
rcmr = SSC_BF(RCMR_PERIOD, 0)
| SSC_BF(RCMR_STTDLY, START_DELAY)
| SSC_BF(RCMR_START, SSC_START_RISING_RF)
- | SSC_BF(RCMR_CKI, SSC_CKI_FALLING)
+ | SSC_BF(RCMR_CKI, SSC_CKI_RISING)
| SSC_BF(RCMR_CKO, SSC_CKO_NONE)
| SSC_BF(RCMR_CKS, ssc->clk_from_rk_pin ?
SSC_CKS_PIN : SSC_CKS_CLOCK);
diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
index 5741c0aa6c03..b5d1caa04d8e 100644
--- a/sound/soc/au1x/dbdma2.c
+++ b/sound/soc/au1x/dbdma2.c
@@ -206,8 +206,8 @@ static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream,
stype = substream->stream;
pcd = to_dmadata(substream);
- DBG("runtime->dma_area = 0x%08lx dma_addr_t = 0x%08lx dma_size = %d "
- "runtime->min_align %d\n",
+ DBG("runtime->dma_area = 0x%08lx dma_addr_t = 0x%08lx dma_size = %zu "
+ "runtime->min_align %lu\n",
(unsigned long)runtime->dma_area,
(unsigned long)runtime->dma_addr, runtime->dma_bytes,
runtime->min_align);
diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c
index 1c1f2210387b..6ba20498202e 100644
--- a/sound/soc/bcm/bcm2835-i2s.c
+++ b/sound/soc/bcm/bcm2835-i2s.c
@@ -259,6 +259,9 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream,
case SNDRV_PCM_FORMAT_S16_LE:
data_length = 16;
break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ data_length = 24;
+ break;
case SNDRV_PCM_FORMAT_S32_LE:
data_length = 32;
break;
@@ -273,13 +276,20 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream,
/* otherwise calculate a fitting block ratio */
bclk_ratio = 2 * data_length;
- /* set target clock rate*/
- clk_set_rate(dev->clk, sampling_rate * bclk_ratio);
+ /* Clock should only be set up here if CPU is clock master */
+ switch (dev->fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ case SND_SOC_DAIFMT_CBS_CFM:
+ clk_set_rate(dev->clk, sampling_rate * bclk_ratio);
+ break;
+ default:
+ break;
+ }
/* Setup the frame format */
format = BCM2835_I2S_CHEN;
- if (data_length > 24)
+ if (data_length >= 24)
format |= BCM2835_I2S_CHWEX;
format |= BCM2835_I2S_CHWID((data_length-8)&0xf);
@@ -570,6 +580,7 @@ static struct snd_soc_dai_driver bcm2835_i2s_dai = {
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_192000,
.formats = SNDRV_PCM_FMTBIT_S16_LE
+ | SNDRV_PCM_FMTBIT_S24_LE
| SNDRV_PCM_FMTBIT_S32_LE
},
.capture = {
@@ -577,6 +588,7 @@ static struct snd_soc_dai_driver bcm2835_i2s_dai = {
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_192000,
.formats = SNDRV_PCM_FMTBIT_S16_LE
+ | SNDRV_PCM_FMTBIT_S24_LE
| SNDRV_PCM_FMTBIT_S32_LE
},
.ops = &bcm2835_i2s_dai_ops,
@@ -678,6 +690,15 @@ static int bcm2835_i2s_probe(struct platform_device *pdev)
dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].maxburst = 2;
dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].maxburst = 2;
+ /*
+ * Set the PACK flag to enable S16_LE support (2 S16_LE values
+ * packed into 32-bit transfers).
+ */
+ dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].flags =
+ SND_DMAENGINE_PCM_DAI_FLAG_PACK;
+ dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].flags =
+ SND_DMAENGINE_PCM_DAI_FLAG_PACK;
+
/* BCLK ratio - use default */
dev->bclk_ratio = 0;
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index f7f64bd019a5..75f571289154 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -79,6 +79,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_MAX98090 if I2C
select SND_SOC_MAX98095 if I2C
select SND_SOC_MAX98357A if GPIOLIB
+ select SND_SOC_MAX98371 if I2C
select SND_SOC_MAX9867 if I2C
select SND_SOC_MAX98925 if I2C
select SND_SOC_MAX98926 if I2C
@@ -88,12 +89,14 @@ config SND_SOC_ALL_CODECS
select SND_SOC_MC13783 if MFD_MC13XXX
select SND_SOC_ML26124 if I2C
select SND_SOC_NAU8825 if I2C
+ select SND_SOC_HDMI_CODEC
select SND_SOC_PCM1681 if I2C
select SND_SOC_PCM179X_I2C if I2C
select SND_SOC_PCM179X_SPI if SPI_MASTER
select SND_SOC_PCM3008
select SND_SOC_PCM3168A_I2C if I2C
select SND_SOC_PCM3168A_SPI if SPI_MASTER
+ select SND_SOC_PCM5102A
select SND_SOC_PCM512x_I2C if I2C
select SND_SOC_PCM512x_SPI if SPI_MASTER
select SND_SOC_RT286 if I2C
@@ -124,12 +127,14 @@ config SND_SOC_ALL_CODECS
select SND_SOC_TAS2552 if I2C
select SND_SOC_TAS5086 if I2C
select SND_SOC_TAS571X if I2C
+ select SND_SOC_TAS5720 if I2C
select SND_SOC_TFA9879 if I2C
select SND_SOC_TLV320AIC23_I2C if I2C
select SND_SOC_TLV320AIC23_SPI if SPI_MASTER
select SND_SOC_TLV320AIC26 if SPI_MASTER
select SND_SOC_TLV320AIC31XX if I2C
- select SND_SOC_TLV320AIC32X4 if I2C
+ select SND_SOC_TLV320AIC32X4_I2C if I2C
+ select SND_SOC_TLV320AIC32X4_SPI if SPI_MASTER
select SND_SOC_TLV320AIC3X if I2C
select SND_SOC_TPA6130A2 if I2C
select SND_SOC_TLV320DAC33 if I2C
@@ -477,6 +482,11 @@ config SND_SOC_DA9055
config SND_SOC_DMIC
tristate
+config SND_SOC_HDMI_CODEC
+ tristate
+ select SND_PCM_ELD
+ select SND_PCM_IEC958
+
config SND_SOC_ES8328
tristate "Everest Semi ES8328 CODEC"
@@ -522,6 +532,9 @@ config SND_SOC_MAX98095
config SND_SOC_MAX98357A
tristate
+config SND_SOC_MAX98371
+ tristate
+
config SND_SOC_MAX9867
tristate
@@ -575,6 +588,9 @@ config SND_SOC_PCM3168A_SPI
select SND_SOC_PCM3168A
select REGMAP_SPI
+config SND_SOC_PCM5102A
+ tristate
+
config SND_SOC_PCM512x
tristate
@@ -629,6 +645,7 @@ config SND_SOC_RT5514
config SND_SOC_RT5616
tristate "Realtek RT5616 CODEC"
+ depends on I2C
config SND_SOC_RT5631
tristate "Realtek ALC5631/RT5631 CODEC"
@@ -737,8 +754,15 @@ config SND_SOC_TAS5086
depends on I2C
config SND_SOC_TAS571X
- tristate "Texas Instruments TAS5711/TAS5717/TAS5719 power amplifiers"
+ tristate "Texas Instruments TAS5711/TAS5717/TAS5719/TAS5721 power amplifiers"
+ depends on I2C
+
+config SND_SOC_TAS5720
+ tristate "Texas Instruments TAS5720 Mono Audio amplifier"
depends on I2C
+ help
+ Enable support for Texas Instruments TAS5720L/M high-efficiency mono
+ Class-D audio power amplifiers.
config SND_SOC_TFA9879
tristate "NXP Semiconductors TFA9879 amplifier"
@@ -769,6 +793,16 @@ config SND_SOC_TLV320AIC31XX
config SND_SOC_TLV320AIC32X4
tristate
+config SND_SOC_TLV320AIC32X4_I2C
+ tristate
+ depends on I2C
+ select SND_SOC_TLV320AIC32X4
+
+config SND_SOC_TLV320AIC32X4_SPI
+ tristate
+ depends on SPI_MASTER
+ select SND_SOC_TLV320AIC32X4
+
config SND_SOC_TLV320AIC3X
tristate "Texas Instruments TLV320AIC3x CODECs"
depends on I2C
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 12a1fe22eaa1..0f548fd34ca3 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -74,6 +74,7 @@ snd-soc-max98088-objs := max98088.o
snd-soc-max98090-objs := max98090.o
snd-soc-max98095-objs := max98095.o
snd-soc-max98357a-objs := max98357a.o
+snd-soc-max98371-objs := max98371.o
snd-soc-max9867-objs := max9867.o
snd-soc-max98925-objs := max98925.o
snd-soc-max98926-objs := max98926.o
@@ -81,6 +82,7 @@ snd-soc-max9850-objs := max9850.o
snd-soc-mc13783-objs := mc13783.o
snd-soc-ml26124-objs := ml26124.o
snd-soc-nau8825-objs := nau8825.o
+snd-soc-hdmi-codec-objs := hdmi-codec.o
snd-soc-pcm1681-objs := pcm1681.o
snd-soc-pcm179x-codec-objs := pcm179x.o
snd-soc-pcm179x-i2c-objs := pcm179x-i2c.o
@@ -89,6 +91,7 @@ snd-soc-pcm3008-objs := pcm3008.o
snd-soc-pcm3168a-objs := pcm3168a.o
snd-soc-pcm3168a-i2c-objs := pcm3168a-i2c.o
snd-soc-pcm3168a-spi-objs := pcm3168a-spi.o
+snd-soc-pcm5102a-objs := pcm5102a.o
snd-soc-pcm512x-objs := pcm512x.o
snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o
snd-soc-pcm512x-spi-objs := pcm512x-spi.o
@@ -129,6 +132,7 @@ snd-soc-stac9766-objs := stac9766.o
snd-soc-sti-sas-objs := sti-sas.o
snd-soc-tas5086-objs := tas5086.o
snd-soc-tas571x-objs := tas571x.o
+snd-soc-tas5720-objs := tas5720.o
snd-soc-tfa9879-objs := tfa9879.o
snd-soc-tlv320aic23-objs := tlv320aic23.o
snd-soc-tlv320aic23-i2c-objs := tlv320aic23-i2c.o
@@ -136,6 +140,8 @@ snd-soc-tlv320aic23-spi-objs := tlv320aic23-spi.o
snd-soc-tlv320aic26-objs := tlv320aic26.o
snd-soc-tlv320aic31xx-objs := tlv320aic31xx.o
snd-soc-tlv320aic32x4-objs := tlv320aic32x4.o
+snd-soc-tlv320aic32x4-i2c-objs := tlv320aic32x4-i2c.o
+snd-soc-tlv320aic32x4-spi-objs := tlv320aic32x4-spi.o
snd-soc-tlv320aic3x-objs := tlv320aic3x.o
snd-soc-tlv320dac33-objs := tlv320dac33.o
snd-soc-ts3a227e-objs := ts3a227e.o
@@ -290,6 +296,7 @@ obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o
obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o
obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o
obj-$(CONFIG_SND_SOC_NAU8825) += snd-soc-nau8825.o
+obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o
obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o
obj-$(CONFIG_SND_SOC_PCM179X) += snd-soc-pcm179x-codec.o
obj-$(CONFIG_SND_SOC_PCM179X_I2C) += snd-soc-pcm179x-i2c.o
@@ -298,6 +305,7 @@ obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o
obj-$(CONFIG_SND_SOC_PCM3168A) += snd-soc-pcm3168a.o
obj-$(CONFIG_SND_SOC_PCM3168A_I2C) += snd-soc-pcm3168a-i2c.o
obj-$(CONFIG_SND_SOC_PCM3168A_SPI) += snd-soc-pcm3168a-spi.o
+obj-$(CONFIG_SND_SOC_PCM5102A) += snd-soc-pcm5102a.o
obj-$(CONFIG_SND_SOC_PCM512x) += snd-soc-pcm512x.o
obj-$(CONFIG_SND_SOC_PCM512x_I2C) += snd-soc-pcm512x-i2c.o
obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o
@@ -335,6 +343,7 @@ obj-$(CONFIG_SND_SOC_STI_SAS) += snd-soc-sti-sas.o
obj-$(CONFIG_SND_SOC_TAS2552) += snd-soc-tas2552.o
obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o
obj-$(CONFIG_SND_SOC_TAS571X) += snd-soc-tas571x.o
+obj-$(CONFIG_SND_SOC_TAS5720) += snd-soc-tas5720.o
obj-$(CONFIG_SND_SOC_TFA9879) += snd-soc-tfa9879.o
obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o
obj-$(CONFIG_SND_SOC_TLV320AIC23_I2C) += snd-soc-tlv320aic23-i2c.o
@@ -342,6 +351,8 @@ obj-$(CONFIG_SND_SOC_TLV320AIC23_SPI) += snd-soc-tlv320aic23-spi.o
obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o
obj-$(CONFIG_SND_SOC_TLV320AIC31XX) += snd-soc-tlv320aic31xx.o
obj-$(CONFIG_SND_SOC_TLV320AIC32X4) += snd-soc-tlv320aic32x4.o
+obj-$(CONFIG_SND_SOC_TLV320AIC32X4_I2C) += snd-soc-tlv320aic32x4-i2c.o
+obj-$(CONFIG_SND_SOC_TLV320AIC32X4_SPI) += snd-soc-tlv320aic32x4-spi.o
obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o
obj-$(CONFIG_SND_SOC_TLV320DAC33) += snd-soc-tlv320dac33.o
obj-$(CONFIG_SND_SOC_TS3A227E) += snd-soc-ts3a227e.o
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index cda27c22812a..4d8b9e49e8d6 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -560,6 +560,7 @@ static const struct regmap_config ak4642_regmap = {
.max_register = FIL1_3,
.reg_defaults = ak4642_reg,
.num_reg_defaults = NUM_AK4642_REG_DEFAULTS,
+ .cache_type = REGCACHE_RBTREE,
};
static const struct regmap_config ak4643_regmap = {
@@ -568,6 +569,7 @@ static const struct regmap_config ak4643_regmap = {
.max_register = SPK_MS,
.reg_defaults = ak4643_reg,
.num_reg_defaults = ARRAY_SIZE(ak4643_reg),
+ .cache_type = REGCACHE_RBTREE,
};
static const struct regmap_config ak4648_regmap = {
@@ -576,6 +578,7 @@ static const struct regmap_config ak4648_regmap = {
.max_register = EQ_FBEQE,
.reg_defaults = ak4648_reg,
.num_reg_defaults = ARRAY_SIZE(ak4648_reg),
+ .cache_type = REGCACHE_RBTREE,
};
static const struct ak4642_drvdata ak4642_drvdata = {
@@ -608,9 +611,7 @@ static struct clk *ak4642_of_parse_mcko(struct device *dev)
of_property_read_string(np, "clock-output-names", &clk_name);
- clk = clk_register_fixed_rate(dev, clk_name, parent_clk_name,
- (parent_clk_name) ? 0 : CLK_IS_ROOT,
- rate);
+ clk = clk_register_fixed_rate(dev, clk_name, parent_clk_name, 0, rate);
if (!IS_ERR(clk))
of_clk_add_provider(np, of_clk_src_simple_get, clk);
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index 92d22a018d68..664a8c044ffb 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -221,6 +221,8 @@ int arizona_init_spk(struct snd_soc_codec *codec)
switch (arizona->type) {
case WM8997:
+ case CS47L24:
+ case WM1831:
break;
default:
ret = snd_soc_dapm_new_controls(dapm, &arizona_spkr, 1);
@@ -249,6 +251,18 @@ int arizona_init_spk(struct snd_soc_codec *codec)
}
EXPORT_SYMBOL_GPL(arizona_init_spk);
+int arizona_free_spk(struct snd_soc_codec *codec)
+{
+ struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+ struct arizona *arizona = priv->arizona;
+
+ arizona_free_irq(arizona, ARIZONA_IRQ_SPK_OVERHEAT_WARN, arizona);
+ arizona_free_irq(arizona, ARIZONA_IRQ_SPK_OVERHEAT, arizona);
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(arizona_free_spk);
+
static const struct snd_soc_dapm_route arizona_mono_routes[] = {
{ "OUT1R", NULL, "OUT1L" },
{ "OUT2R", NULL, "OUT2L" },
@@ -1122,7 +1136,6 @@ int arizona_anc_ev(struct snd_soc_dapm_widget *w,
int event)
{
struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
- unsigned int mask = 0x3 << w->shift;
unsigned int val;
switch (event) {
@@ -1136,7 +1149,7 @@ int arizona_anc_ev(struct snd_soc_dapm_widget *w,
return 0;
}
- snd_soc_update_bits(codec, ARIZONA_CLOCK_CONTROL, mask, val);
+ snd_soc_write(codec, ARIZONA_CLOCK_CONTROL, val);
return 0;
}
@@ -2035,7 +2048,21 @@ static int arizona_calc_fratio(struct arizona_fll *fll,
init_ratio, Fref, refdiv);
while (div <= ARIZONA_FLL_MAX_REFDIV) {
- for (ratio = init_ratio; ratio <= ARIZONA_FLL_MAX_FRATIO;
+ /* start from init_ratio because this may already give a
+ * fractional N.K
+ */
+ for (ratio = init_ratio; ratio > 0; ratio--) {
+ if (target % (ratio * Fref)) {
+ cfg->refdiv = refdiv;
+ cfg->fratio = ratio - 1;
+ arizona_fll_dbg(fll,
+ "pseudo: found fref=%u refdiv=%d(%d) ratio=%d\n",
+ Fref, refdiv, div, ratio);
+ return ratio;
+ }
+ }
+
+ for (ratio = init_ratio + 1; ratio <= ARIZONA_FLL_MAX_FRATIO;
ratio++) {
if ((ARIZONA_FLL_VCO_CORNER / 2) /
(fll->vco_mult * ratio) < Fref) {
@@ -2061,17 +2088,6 @@ static int arizona_calc_fratio(struct arizona_fll *fll,
}
}
- for (ratio = init_ratio - 1; ratio > 0; ratio--) {
- if (target % (ratio * Fref)) {
- cfg->refdiv = refdiv;
- cfg->fratio = ratio - 1;
- arizona_fll_dbg(fll,
- "pseudo: found fref=%u refdiv=%d(%d) ratio=%d\n",
- Fref, refdiv, div, ratio);
- return ratio;
- }
- }
-
div *= 2;
Fref /= 2;
refdiv++;
diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h
index 1ea8e4ecf8d4..ce0531b8c632 100644
--- a/sound/soc/codecs/arizona.h
+++ b/sound/soc/codecs/arizona.h
@@ -307,6 +307,8 @@ extern int arizona_init_spk(struct snd_soc_codec *codec);
extern int arizona_init_gpio(struct snd_soc_codec *codec);
extern int arizona_init_mono(struct snd_soc_codec *codec);
+extern int arizona_free_spk(struct snd_soc_codec *codec);
+
extern int arizona_init_dai(struct arizona_priv *priv, int dai);
int arizona_set_output_mode(struct snd_soc_codec *codec, int output,
diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c
index 44c30fe3e315..287d13740be4 100644
--- a/sound/soc/codecs/cs35l32.c
+++ b/sound/soc/codecs/cs35l32.c
@@ -274,7 +274,9 @@ static int cs35l32_handle_of_data(struct i2c_client *i2c_client,
if (of_property_read_u32(np, "cirrus,sdout-share", &val) >= 0)
pdata->sdout_share = val;
- of_property_read_u32(np, "cirrus,boost-manager", &val);
+ if (of_property_read_u32(np, "cirrus,boost-manager", &val))
+ val = -1u;
+
switch (val) {
case CS35L32_BOOST_MGR_AUTO:
case CS35L32_BOOST_MGR_AUTO_AUDIO:
@@ -282,13 +284,15 @@ static int cs35l32_handle_of_data(struct i2c_client *i2c_client,
case CS35L32_BOOST_MGR_FIXED:
pdata->boost_mng = val;
break;
+ case -1u:
default:
dev_err(&i2c_client->dev,
"Wrong cirrus,boost-manager DT value %d\n", val);
pdata->boost_mng = CS35L32_BOOST_MGR_BYPASS;
}
- of_property_read_u32(np, "cirrus,sdout-datacfg", &val);
+ if (of_property_read_u32(np, "cirrus,sdout-datacfg", &val))
+ val = -1u;
switch (val) {
case CS35L32_DATA_CFG_LR_VP:
case CS35L32_DATA_CFG_LR_STAT:
@@ -296,13 +300,15 @@ static int cs35l32_handle_of_data(struct i2c_client *i2c_client,
case CS35L32_DATA_CFG_LR_VPSTAT:
pdata->sdout_datacfg = val;
break;
+ case -1u:
default:
dev_err(&i2c_client->dev,
"Wrong cirrus,sdout-datacfg DT value %d\n", val);
pdata->sdout_datacfg = CS35L32_DATA_CFG_LR;
}
- of_property_read_u32(np, "cirrus,battery-threshold", &val);
+ if (of_property_read_u32(np, "cirrus,battery-threshold", &val))
+ val = -1u;
switch (val) {
case CS35L32_BATT_THRESH_3_1V:
case CS35L32_BATT_THRESH_3_2V:
@@ -310,13 +316,15 @@ static int cs35l32_handle_of_data(struct i2c_client *i2c_client,
case CS35L32_BATT_THRESH_3_4V:
pdata->batt_thresh = val;
break;
+ case -1u:
default:
dev_err(&i2c_client->dev,
"Wrong cirrus,battery-threshold DT value %d\n", val);
pdata->batt_thresh = CS35L32_BATT_THRESH_3_3V;
}
- of_property_read_u32(np, "cirrus,battery-recovery", &val);
+ if (of_property_read_u32(np, "cirrus,battery-recovery", &val))
+ val = -1u;
switch (val) {
case CS35L32_BATT_RECOV_3_1V:
case CS35L32_BATT_RECOV_3_2V:
@@ -326,6 +334,7 @@ static int cs35l32_handle_of_data(struct i2c_client *i2c_client,
case CS35L32_BATT_RECOV_3_6V:
pdata->batt_recov = val;
break;
+ case -1u:
default:
dev_err(&i2c_client->dev,
"Wrong cirrus,battery-recovery DT value %d\n", val);
diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c
index 7cd5f769bb61..eec1ff853b98 100644
--- a/sound/soc/codecs/cs42l56.c
+++ b/sound/soc/codecs/cs42l56.c
@@ -56,7 +56,7 @@ struct cs42l56_private {
u8 iface;
u8 iface_fmt;
u8 iface_inv;
-#if defined(CONFIG_INPUT) || defined(CONFIG_INPUT_MODULE)
+#if IS_ENABLED(CONFIG_INPUT)
struct input_dev *beep;
struct work_struct beep_work;
int beep_rate;
diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c
index 576087bda330..5ec5a682d186 100644
--- a/sound/soc/codecs/cs47l24.c
+++ b/sound/soc/codecs/cs47l24.c
@@ -807,6 +807,9 @@ static const struct snd_soc_dapm_route cs47l24_dapm_routes[] = {
{ "IN2L PGA", NULL, "IN2L" },
{ "IN2R PGA", NULL, "IN2R" },
+ { "Audio Trace DSP", NULL, "DSP2" },
+ { "Audio Trace DSP", NULL, "SYSCLK" },
+
ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"),
ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"),
@@ -1016,6 +1019,27 @@ static struct snd_soc_dai_driver cs47l24_dai[] = {
.formats = CS47L24_FORMATS,
},
},
+ {
+ .name = "cs47l24-cpu-trace",
+ .capture = {
+ .stream_name = "Audio Trace CPU",
+ .channels_min = 1,
+ .channels_max = 6,
+ .rates = CS47L24_RATES,
+ .formats = CS47L24_FORMATS,
+ },
+ .compress_new = snd_soc_new_compress,
+ },
+ {
+ .name = "cs47l24-dsp-trace",
+ .capture = {
+ .stream_name = "Audio Trace DSP",
+ .channels_min = 1,
+ .channels_max = 6,
+ .rates = CS47L24_RATES,
+ .formats = CS47L24_FORMATS,
+ },
+ },
};
static int cs47l24_open(struct snd_compr_stream *stream)
@@ -1027,6 +1051,8 @@ static int cs47l24_open(struct snd_compr_stream *stream)
if (strcmp(rtd->codec_dai->name, "cs47l24-dsp-voicectrl") == 0) {
n_adsp = 2;
+ } else if (strcmp(rtd->codec_dai->name, "cs47l24-dsp-trace") == 0) {
+ n_adsp = 1;
} else {
dev_err(arizona->dev,
"No suitable compressed stream for DAI '%s'\n",
@@ -1041,10 +1067,16 @@ static irqreturn_t cs47l24_adsp2_irq(int irq, void *data)
{
struct cs47l24_priv *priv = data;
struct arizona *arizona = priv->core.arizona;
- int ret;
+ int serviced = 0;
+ int i, ret;
+
+ for (i = 1; i <= 2; ++i) {
+ ret = wm_adsp_compr_handle_irq(&priv->core.adsp[i]);
+ if (ret != -ENODEV)
+ serviced++;
+ }
- ret = wm_adsp_compr_handle_irq(&priv->core.adsp[2]);
- if (ret == -ENODEV) {
+ if (!serviced) {
dev_err(arizona->dev, "Spurious compressed data IRQ\n");
return IRQ_NONE;
}
@@ -1108,6 +1140,9 @@ static int cs47l24_codec_remove(struct snd_soc_codec *codec)
priv->core.arizona->dapm = NULL;
arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, priv);
+
+ arizona_free_spk(codec);
+
return 0;
}
@@ -1157,6 +1192,7 @@ static struct snd_compr_ops cs47l24_compr_ops = {
static struct snd_soc_platform_driver cs47l24_compr_platform = {
.compr_ops = &cs47l24_compr_ops,
};
+
static int cs47l24_probe(struct platform_device *pdev)
{
struct arizona *arizona = dev_get_drvdata(pdev->dev.parent);
@@ -1225,9 +1261,9 @@ static int cs47l24_probe(struct platform_device *pdev)
dev_err(&pdev->dev, "Failed to register platform: %d\n", ret);
return ret;
}
+
ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_cs47l24,
cs47l24_dai, ARRAY_SIZE(cs47l24_dai));
-
if (ret < 0) {
dev_err(&pdev->dev, "Failed to register codec: %d\n", ret);
snd_soc_unregister_platform(&pdev->dev);
@@ -1238,10 +1274,15 @@ static int cs47l24_probe(struct platform_device *pdev)
static int cs47l24_remove(struct platform_device *pdev)
{
+ struct cs47l24_priv *cs47l24 = platform_get_drvdata(pdev);
+
snd_soc_unregister_platform(&pdev->dev);
snd_soc_unregister_codec(&pdev->dev);
pm_runtime_disable(&pdev->dev);
+ wm_adsp2_remove(&cs47l24->core.adsp[1]);
+ wm_adsp2_remove(&cs47l24->core.adsp[2]);
+
return 0;
}
diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c
index 7278f93460c1..e5527bc570ae 100644
--- a/sound/soc/codecs/da7213.c
+++ b/sound/soc/codecs/da7213.c
@@ -726,6 +726,68 @@ static const struct snd_kcontrol_new da7213_dapm_mixoutr_controls[] = {
/*
+ * DAPM Events
+ */
+
+static int da7213_dai_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+ struct da7213_priv *da7213 = snd_soc_codec_get_drvdata(codec);
+ u8 pll_ctrl, pll_status;
+ int i = 0;
+ bool srm_lock = false;
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ /* Enable DAI clks for master mode */
+ if (da7213->master)
+ snd_soc_update_bits(codec, DA7213_DAI_CLK_MODE,
+ DA7213_DAI_CLK_EN_MASK,
+ DA7213_DAI_CLK_EN_MASK);
+
+ /* PC synchronised to DAI */
+ snd_soc_update_bits(codec, DA7213_PC_COUNT,
+ DA7213_PC_FREERUN_MASK, 0);
+
+ /* Slave mode, if SRM not enabled no need for status checks */
+ pll_ctrl = snd_soc_read(codec, DA7213_PLL_CTRL);
+ if (!(pll_ctrl & DA7213_PLL_SRM_EN))
+ return 0;
+
+ /* Check SRM has locked */
+ do {
+ pll_status = snd_soc_read(codec, DA7213_PLL_STATUS);
+ if (pll_status & DA7219_PLL_SRM_LOCK) {
+ srm_lock = true;
+ } else {
+ ++i;
+ msleep(50);
+ }
+ } while ((i < DA7213_SRM_CHECK_RETRIES) & (!srm_lock));
+
+ if (!srm_lock)
+ dev_warn(codec->dev, "SRM failed to lock\n");
+
+ return 0;
+ case SND_SOC_DAPM_POST_PMD:
+ /* PC free-running */
+ snd_soc_update_bits(codec, DA7213_PC_COUNT,
+ DA7213_PC_FREERUN_MASK,
+ DA7213_PC_FREERUN_MASK);
+
+ /* Disable DAI clks if in master mode */
+ if (da7213->master)
+ snd_soc_update_bits(codec, DA7213_DAI_CLK_MODE,
+ DA7213_DAI_CLK_EN_MASK, 0);
+ return 0;
+ default:
+ return -EINVAL;
+ }
+}
+
+
+/*
* DAPM widgets
*/
@@ -736,7 +798,8 @@ static const struct snd_soc_dapm_widget da7213_dapm_widgets[] = {
/* Use a supply here as this controls both input & output DAIs */
SND_SOC_DAPM_SUPPLY("DAI", DA7213_DAI_CTRL, DA7213_DAI_EN_SHIFT,
- DA7213_NO_INVERT, NULL, 0),
+ DA7213_NO_INVERT, da7213_dai_event,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
/*
* Input
@@ -1143,11 +1206,9 @@ static int da7213_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
/* Set master/slave mode */
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBM_CFM:
- dai_clk_mode |= DA7213_DAI_CLK_EN_MASTER_MODE;
da7213->master = true;
break;
case SND_SOC_DAIFMT_CBS_CFS:
- dai_clk_mode |= DA7213_DAI_CLK_EN_SLAVE_MODE;
da7213->master = false;
break;
default:
@@ -1281,28 +1342,28 @@ static int da7213_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
pll_ctrl = 0;
/* Workout input divider based on MCLK rate */
- if ((da7213->mclk_rate == 32768) && (source == DA7213_SYSCLK_PLL)) {
+ if (da7213->mclk_rate == 32768) {
/* 32KHz PLL Mode */
- indiv_bits = DA7213_PLL_INDIV_10_20_MHZ;
- indiv = DA7213_PLL_INDIV_10_20_MHZ_VAL;
+ indiv_bits = DA7213_PLL_INDIV_9_TO_18_MHZ;
+ indiv = DA7213_PLL_INDIV_9_TO_18_MHZ_VAL;
freq_ref = 3750000;
pll_ctrl |= DA7213_PLL_32K_MODE;
} else {
/* 5 - 54MHz MCLK */
if (da7213->mclk_rate < 5000000) {
goto pll_err;
- } else if (da7213->mclk_rate <= 10000000) {
- indiv_bits = DA7213_PLL_INDIV_5_10_MHZ;
- indiv = DA7213_PLL_INDIV_5_10_MHZ_VAL;
- } else if (da7213->mclk_rate <= 20000000) {
- indiv_bits = DA7213_PLL_INDIV_10_20_MHZ;
- indiv = DA7213_PLL_INDIV_10_20_MHZ_VAL;
- } else if (da7213->mclk_rate <= 40000000) {
- indiv_bits = DA7213_PLL_INDIV_20_40_MHZ;
- indiv = DA7213_PLL_INDIV_20_40_MHZ_VAL;
+ } else if (da7213->mclk_rate <= 9000000) {
+ indiv_bits = DA7213_PLL_INDIV_5_TO_9_MHZ;
+ indiv = DA7213_PLL_INDIV_5_TO_9_MHZ_VAL;
+ } else if (da7213->mclk_rate <= 18000000) {
+ indiv_bits = DA7213_PLL_INDIV_9_TO_18_MHZ;
+ indiv = DA7213_PLL_INDIV_9_TO_18_MHZ_VAL;
+ } else if (da7213->mclk_rate <= 36000000) {
+ indiv_bits = DA7213_PLL_INDIV_18_TO_36_MHZ;
+ indiv = DA7213_PLL_INDIV_18_TO_36_MHZ_VAL;
} else if (da7213->mclk_rate <= 54000000) {
- indiv_bits = DA7213_PLL_INDIV_40_54_MHZ;
- indiv = DA7213_PLL_INDIV_40_54_MHZ_VAL;
+ indiv_bits = DA7213_PLL_INDIV_36_TO_54_MHZ;
+ indiv = DA7213_PLL_INDIV_36_TO_54_MHZ_VAL;
} else {
goto pll_err;
}
@@ -1547,6 +1608,10 @@ static int da7213_probe(struct snd_soc_codec *codec)
/* Default to using SRM for slave mode */
da7213->srm_en = true;
+ /* Default PC counter to free-running */
+ snd_soc_update_bits(codec, DA7213_PC_COUNT, DA7213_PC_FREERUN_MASK,
+ DA7213_PC_FREERUN_MASK);
+
/* Enable all Gain Ramps */
snd_soc_update_bits(codec, DA7213_AUX_L_CTRL,
DA7213_GAIN_RAMP_EN, DA7213_GAIN_RAMP_EN);
diff --git a/sound/soc/codecs/da7213.h b/sound/soc/codecs/da7213.h
index 030fd691b076..fbb7a356a501 100644
--- a/sound/soc/codecs/da7213.h
+++ b/sound/soc/codecs/da7213.h
@@ -142,6 +142,9 @@
* Bit fields
*/
+/* DA7213_PLL_STATUS = 0x03 */
+#define DA7219_PLL_SRM_LOCK (0x1 << 1)
+
/* DA7213_SR = 0x22 */
#define DA7213_SR_8000 (0x1 << 0)
#define DA7213_SR_11025 (0x2 << 0)
@@ -160,10 +163,10 @@
#define DA7213_VMID_EN (0x1 << 7)
/* DA7213_PLL_CTRL = 0x27 */
-#define DA7213_PLL_INDIV_5_10_MHZ (0x0 << 2)
-#define DA7213_PLL_INDIV_10_20_MHZ (0x1 << 2)
-#define DA7213_PLL_INDIV_20_40_MHZ (0x2 << 2)
-#define DA7213_PLL_INDIV_40_54_MHZ (0x3 << 2)
+#define DA7213_PLL_INDIV_5_TO_9_MHZ (0x0 << 2)
+#define DA7213_PLL_INDIV_9_TO_18_MHZ (0x1 << 2)
+#define DA7213_PLL_INDIV_18_TO_36_MHZ (0x2 << 2)
+#define DA7213_PLL_INDIV_36_TO_54_MHZ (0x3 << 2)
#define DA7213_PLL_INDIV_MASK (0x3 << 2)
#define DA7213_PLL_MCLK_SQR_EN (0x1 << 4)
#define DA7213_PLL_32K_MODE (0x1 << 5)
@@ -178,8 +181,6 @@
#define DA7213_DAI_BCLKS_PER_WCLK_MASK (0x3 << 0)
#define DA7213_DAI_CLK_POL_INV (0x1 << 2)
#define DA7213_DAI_WCLK_POL_INV (0x1 << 3)
-#define DA7213_DAI_CLK_EN_SLAVE_MODE (0x0 << 7)
-#define DA7213_DAI_CLK_EN_MASTER_MODE (0x1 << 7)
#define DA7213_DAI_CLK_EN_MASK (0x1 << 7)
/* DA7213_DAI_CTRL = 0x29 */
@@ -412,6 +413,9 @@
#define DA7213_DMIC_CLK_RATE_SHIFT 2
#define DA7213_DMIC_CLK_RATE_MASK (0x1 << 2)
+/* DA7213_PC_COUNT = 0x94 */
+#define DA7213_PC_FREERUN_MASK (0x1 << 0)
+
/* DA7213_DIG_CTRL = 0x99 */
#define DA7213_DAC_L_INV_SHIFT 3
#define DA7213_DAC_R_INV_SHIFT 7
@@ -495,15 +499,16 @@
#define DA7213_ALC_AVG_ITERATIONS 5
/* PLL related */
-#define DA7213_SYSCLK_MCLK 0
-#define DA7213_SYSCLK_PLL 1
-#define DA7213_PLL_FREQ_OUT_90316800 90316800
-#define DA7213_PLL_FREQ_OUT_98304000 98304000
-#define DA7213_PLL_FREQ_OUT_94310400 94310400
-#define DA7213_PLL_INDIV_5_10_MHZ_VAL 2
-#define DA7213_PLL_INDIV_10_20_MHZ_VAL 4
-#define DA7213_PLL_INDIV_20_40_MHZ_VAL 8
-#define DA7213_PLL_INDIV_40_54_MHZ_VAL 16
+#define DA7213_SYSCLK_MCLK 0
+#define DA7213_SYSCLK_PLL 1
+#define DA7213_PLL_FREQ_OUT_90316800 90316800
+#define DA7213_PLL_FREQ_OUT_98304000 98304000
+#define DA7213_PLL_FREQ_OUT_94310400 94310400
+#define DA7213_PLL_INDIV_5_TO_9_MHZ_VAL 2
+#define DA7213_PLL_INDIV_9_TO_18_MHZ_VAL 4
+#define DA7213_PLL_INDIV_18_TO_36_MHZ_VAL 8
+#define DA7213_PLL_INDIV_36_TO_54_MHZ_VAL 16
+#define DA7213_SRM_CHECK_RETRIES 8
enum da7213_clk_src {
DA7213_CLKSRC_MCLK = 0,
diff --git a/sound/soc/codecs/da7218.c b/sound/soc/codecs/da7218.c
index 93575f251866..99ce23e113bf 100644
--- a/sound/soc/codecs/da7218.c
+++ b/sound/soc/codecs/da7218.c
@@ -1868,27 +1868,27 @@ static int da7218_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
/* Verify 32KHz, 2MHz - 54MHz MCLK provided, and set input divider */
if (da7218->mclk_rate == 32768) {
- indiv_bits = DA7218_PLL_INDIV_2_5_MHZ;
- indiv = DA7218_PLL_INDIV_2_10_MHZ_VAL;
+ indiv_bits = DA7218_PLL_INDIV_9_TO_18_MHZ;
+ indiv = DA7218_PLL_INDIV_9_TO_18_MHZ_VAL;
} else if (da7218->mclk_rate < 2000000) {
dev_err(codec->dev, "PLL input clock %d below valid range\n",
da7218->mclk_rate);
return -EINVAL;
- } else if (da7218->mclk_rate <= 5000000) {
- indiv_bits = DA7218_PLL_INDIV_2_5_MHZ;
- indiv = DA7218_PLL_INDIV_2_10_MHZ_VAL;
- } else if (da7218->mclk_rate <= 10000000) {
- indiv_bits = DA7218_PLL_INDIV_5_10_MHZ;
- indiv = DA7218_PLL_INDIV_2_10_MHZ_VAL;
- } else if (da7218->mclk_rate <= 20000000) {
- indiv_bits = DA7218_PLL_INDIV_10_20_MHZ;
- indiv = DA7218_PLL_INDIV_10_20_MHZ_VAL;
- } else if (da7218->mclk_rate <= 40000000) {
- indiv_bits = DA7218_PLL_INDIV_20_40_MHZ;
- indiv = DA7218_PLL_INDIV_20_40_MHZ_VAL;
+ } else if (da7218->mclk_rate <= 4500000) {
+ indiv_bits = DA7218_PLL_INDIV_2_TO_4_5_MHZ;
+ indiv = DA7218_PLL_INDIV_2_TO_4_5_MHZ_VAL;
+ } else if (da7218->mclk_rate <= 9000000) {
+ indiv_bits = DA7218_PLL_INDIV_4_5_TO_9_MHZ;
+ indiv = DA7218_PLL_INDIV_4_5_TO_9_MHZ_VAL;
+ } else if (da7218->mclk_rate <= 18000000) {
+ indiv_bits = DA7218_PLL_INDIV_9_TO_18_MHZ;
+ indiv = DA7218_PLL_INDIV_9_TO_18_MHZ_VAL;
+ } else if (da7218->mclk_rate <= 36000000) {
+ indiv_bits = DA7218_PLL_INDIV_18_TO_36_MHZ;
+ indiv = DA7218_PLL_INDIV_18_TO_36_MHZ_VAL;
} else if (da7218->mclk_rate <= 54000000) {
- indiv_bits = DA7218_PLL_INDIV_40_54_MHZ;
- indiv = DA7218_PLL_INDIV_40_54_MHZ_VAL;
+ indiv_bits = DA7218_PLL_INDIV_36_TO_54_MHZ;
+ indiv = DA7218_PLL_INDIV_36_TO_54_MHZ_VAL;
} else {
dev_err(codec->dev, "PLL input clock %d above valid range\n",
da7218->mclk_rate);
diff --git a/sound/soc/codecs/da7218.h b/sound/soc/codecs/da7218.h
index c2c59049a2ad..477cd37723cf 100644
--- a/sound/soc/codecs/da7218.h
+++ b/sound/soc/codecs/da7218.h
@@ -876,15 +876,11 @@
/* DA7218_PLL_CTRL = 0x91 */
#define DA7218_PLL_INDIV_SHIFT 0
#define DA7218_PLL_INDIV_MASK (0x7 << 0)
-#define DA7218_PLL_INDIV_2_5_MHZ (0x0 << 0)
-#define DA7218_PLL_INDIV_5_10_MHZ (0x1 << 0)
-#define DA7218_PLL_INDIV_10_20_MHZ (0x2 << 0)
-#define DA7218_PLL_INDIV_20_40_MHZ (0x3 << 0)
-#define DA7218_PLL_INDIV_40_54_MHZ (0x4 << 0)
-#define DA7218_PLL_INDIV_2_10_MHZ_VAL 2
-#define DA7218_PLL_INDIV_10_20_MHZ_VAL 4
-#define DA7218_PLL_INDIV_20_40_MHZ_VAL 8
-#define DA7218_PLL_INDIV_40_54_MHZ_VAL 16
+#define DA7218_PLL_INDIV_2_TO_4_5_MHZ (0x0 << 0)
+#define DA7218_PLL_INDIV_4_5_TO_9_MHZ (0x1 << 0)
+#define DA7218_PLL_INDIV_9_TO_18_MHZ (0x2 << 0)
+#define DA7218_PLL_INDIV_18_TO_36_MHZ (0x3 << 0)
+#define DA7218_PLL_INDIV_36_TO_54_MHZ (0x4 << 0)
#define DA7218_PLL_MCLK_SQR_EN_SHIFT 4
#define DA7218_PLL_MCLK_SQR_EN_MASK (0x1 << 4)
#define DA7218_PLL_MODE_SHIFT 6
@@ -1336,6 +1332,13 @@
#define DA7218_PLL_FREQ_OUT_90316 90316800
#define DA7218_PLL_FREQ_OUT_98304 98304000
+/* PLL Frequency Dividers */
+#define DA7218_PLL_INDIV_2_TO_4_5_MHZ_VAL 1
+#define DA7218_PLL_INDIV_4_5_TO_9_MHZ_VAL 2
+#define DA7218_PLL_INDIV_9_TO_18_MHZ_VAL 4
+#define DA7218_PLL_INDIV_18_TO_36_MHZ_VAL 8
+#define DA7218_PLL_INDIV_36_TO_54_MHZ_VAL 16
+
/* ALC Calibration */
#define DA7218_ALC_CALIB_DELAY_MIN 2500
#define DA7218_ALC_CALIB_DELAY_MAX 5000
diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c
index 81c0708b85c1..5c93899f1f0e 100644
--- a/sound/soc/codecs/da7219.c
+++ b/sound/soc/codecs/da7219.c
@@ -11,6 +11,7 @@
* option) any later version.
*/
+#include <linux/acpi.h>
#include <linux/clk.h>
#include <linux/i2c.h>
#include <linux/of_device.h>
@@ -1025,7 +1026,7 @@ static int da7219_set_dai_sysclk(struct snd_soc_dai *codec_dai,
if ((da7219->clk_src == clk_id) && (da7219->mclk_rate == freq))
return 0;
- if (((freq < 2000000) && (freq != 32768)) || (freq > 54000000)) {
+ if ((freq < 2000000) || (freq > 54000000)) {
dev_err(codec_dai->dev, "Unsupported MCLK value %d\n",
freq);
return -EINVAL;
@@ -1079,21 +1080,21 @@ static int da7219_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
dev_err(codec->dev, "PLL input clock %d below valid range\n",
da7219->mclk_rate);
return -EINVAL;
- } else if (da7219->mclk_rate <= 5000000) {
- indiv_bits = DA7219_PLL_INDIV_2_5_MHZ;
- indiv = DA7219_PLL_INDIV_2_5_MHZ_VAL;
- } else if (da7219->mclk_rate <= 10000000) {
- indiv_bits = DA7219_PLL_INDIV_5_10_MHZ;
- indiv = DA7219_PLL_INDIV_5_10_MHZ_VAL;
- } else if (da7219->mclk_rate <= 20000000) {
- indiv_bits = DA7219_PLL_INDIV_10_20_MHZ;
- indiv = DA7219_PLL_INDIV_10_20_MHZ_VAL;
- } else if (da7219->mclk_rate <= 40000000) {
- indiv_bits = DA7219_PLL_INDIV_20_40_MHZ;
- indiv = DA7219_PLL_INDIV_20_40_MHZ_VAL;
+ } else if (da7219->mclk_rate <= 4500000) {
+ indiv_bits = DA7219_PLL_INDIV_2_TO_4_5_MHZ;
+ indiv = DA7219_PLL_INDIV_2_TO_4_5_MHZ_VAL;
+ } else if (da7219->mclk_rate <= 9000000) {
+ indiv_bits = DA7219_PLL_INDIV_4_5_TO_9_MHZ;
+ indiv = DA7219_PLL_INDIV_4_5_TO_9_MHZ_VAL;
+ } else if (da7219->mclk_rate <= 18000000) {
+ indiv_bits = DA7219_PLL_INDIV_9_TO_18_MHZ;
+ indiv = DA7219_PLL_INDIV_9_TO_18_MHZ_VAL;
+ } else if (da7219->mclk_rate <= 36000000) {
+ indiv_bits = DA7219_PLL_INDIV_18_TO_36_MHZ;
+ indiv = DA7219_PLL_INDIV_18_TO_36_MHZ_VAL;
} else if (da7219->mclk_rate <= 54000000) {
- indiv_bits = DA7219_PLL_INDIV_40_54_MHZ;
- indiv = DA7219_PLL_INDIV_40_54_MHZ_VAL;
+ indiv_bits = DA7219_PLL_INDIV_36_TO_54_MHZ;
+ indiv = DA7219_PLL_INDIV_36_TO_54_MHZ_VAL;
} else {
dev_err(codec->dev, "PLL input clock %d above valid range\n",
da7219->mclk_rate);
@@ -1426,6 +1427,12 @@ static const struct of_device_id da7219_of_match[] = {
};
MODULE_DEVICE_TABLE(of, da7219_of_match);
+static const struct acpi_device_id da7219_acpi_match[] = {
+ { .id = "DLGS7219", },
+ { }
+};
+MODULE_DEVICE_TABLE(acpi, da7219_acpi_match);
+
static enum da7219_micbias_voltage
da7219_of_micbias_lvl(struct snd_soc_codec *codec, u32 val)
{
@@ -1955,6 +1962,7 @@ static struct i2c_driver da7219_i2c_driver = {
.driver = {
.name = "da7219",
.of_match_table = of_match_ptr(da7219_of_match),
+ .acpi_match_table = ACPI_PTR(da7219_acpi_match),
},
.probe = da7219_i2c_probe,
.remove = da7219_i2c_remove,
diff --git a/sound/soc/codecs/da7219.h b/sound/soc/codecs/da7219.h
index 5a787e738084..ff2a2f02ce40 100644
--- a/sound/soc/codecs/da7219.h
+++ b/sound/soc/codecs/da7219.h
@@ -194,11 +194,11 @@
/* DA7219_PLL_CTRL = 0x20 */
#define DA7219_PLL_INDIV_SHIFT 2
#define DA7219_PLL_INDIV_MASK (0x7 << 2)
-#define DA7219_PLL_INDIV_2_5_MHZ (0x0 << 2)
-#define DA7219_PLL_INDIV_5_10_MHZ (0x1 << 2)
-#define DA7219_PLL_INDIV_10_20_MHZ (0x2 << 2)
-#define DA7219_PLL_INDIV_20_40_MHZ (0x3 << 2)
-#define DA7219_PLL_INDIV_40_54_MHZ (0x4 << 2)
+#define DA7219_PLL_INDIV_2_TO_4_5_MHZ (0x0 << 2)
+#define DA7219_PLL_INDIV_4_5_TO_9_MHZ (0x1 << 2)
+#define DA7219_PLL_INDIV_9_TO_18_MHZ (0x2 << 2)
+#define DA7219_PLL_INDIV_18_TO_36_MHZ (0x3 << 2)
+#define DA7219_PLL_INDIV_36_TO_54_MHZ (0x4 << 2)
#define DA7219_PLL_MCLK_SQR_EN_SHIFT 5
#define DA7219_PLL_MCLK_SQR_EN_MASK (0x1 << 5)
#define DA7219_PLL_MODE_SHIFT 6
@@ -761,11 +761,11 @@
#define DA7219_PLL_FREQ_OUT_98304 98304000
/* PLL Frequency Dividers */
-#define DA7219_PLL_INDIV_2_5_MHZ_VAL 1
-#define DA7219_PLL_INDIV_5_10_MHZ_VAL 2
-#define DA7219_PLL_INDIV_10_20_MHZ_VAL 4
-#define DA7219_PLL_INDIV_20_40_MHZ_VAL 8
-#define DA7219_PLL_INDIV_40_54_MHZ_VAL 16
+#define DA7219_PLL_INDIV_2_TO_4_5_MHZ_VAL 1
+#define DA7219_PLL_INDIV_4_5_TO_9_MHZ_VAL 2
+#define DA7219_PLL_INDIV_9_TO_18_MHZ_VAL 4
+#define DA7219_PLL_INDIV_18_TO_36_MHZ_VAL 8
+#define DA7219_PLL_INDIV_36_TO_54_MHZ_VAL 16
/* SRM */
#define DA7219_SRM_CHECK_RETRIES 8
diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c
index afa6c5db9dcc..2086d7107622 100644
--- a/sound/soc/codecs/es8328.c
+++ b/sound/soc/codecs/es8328.c
@@ -26,18 +26,30 @@
#include <sound/tlv.h>
#include "es8328.h"
-#define ES8328_SYSCLK_RATE_1X 11289600
-#define ES8328_SYSCLK_RATE_2X 22579200
+static const unsigned int rates_12288[] = {
+ 8000, 12000, 16000, 24000, 32000, 48000, 96000,
+};
-/* Run the codec at 22.5792 or 11.2896 MHz to support these rates */
-static struct {
- int rate;
- u8 ratio;
-} mclk_ratios[] = {
- { 8000, 9 },
- {11025, 7 },
- {22050, 4 },
- {44100, 2 },
+static const int ratios_12288[] = {
+ 10, 7, 6, 4, 3, 2, 0,
+};
+
+static const struct snd_pcm_hw_constraint_list constraints_12288 = {
+ .count = ARRAY_SIZE(rates_12288),
+ .list = rates_12288,
+};
+
+static const unsigned int rates_11289[] = {
+ 8018, 11025, 22050, 44100, 88200,
+};
+
+static const int ratios_11289[] = {
+ 9, 7, 4, 2, 0,
+};
+
+static const struct snd_pcm_hw_constraint_list constraints_11289 = {
+ .count = ARRAY_SIZE(rates_11289),
+ .list = rates_11289,
};
/* regulator supplies for sgtl5000, VDDD is an optional external supply */
@@ -57,16 +69,28 @@ static const char * const supply_names[ES8328_SUPPLY_NUM] = {
"HPVDD",
};
-#define ES8328_RATES (SNDRV_PCM_RATE_44100 | \
+#define ES8328_RATES (SNDRV_PCM_RATE_96000 | \
+ SNDRV_PCM_RATE_48000 | \
+ SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_32000 | \
SNDRV_PCM_RATE_22050 | \
- SNDRV_PCM_RATE_11025)
-#define ES8328_FORMATS (SNDRV_PCM_FMTBIT_S16_LE)
+ SNDRV_PCM_RATE_16000 | \
+ SNDRV_PCM_RATE_11025 | \
+ SNDRV_PCM_RATE_8000)
+#define ES8328_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S18_3LE | \
+ SNDRV_PCM_FMTBIT_S20_3LE | \
+ SNDRV_PCM_FMTBIT_S24_LE | \
+ SNDRV_PCM_FMTBIT_S32_LE)
struct es8328_priv {
struct regmap *regmap;
struct clk *clk;
int playback_fs;
bool deemph;
+ int mclkdiv2;
+ const struct snd_pcm_hw_constraint_list *sysclk_constraints;
+ const int *mclk_ratios;
struct regulator_bulk_data supplies[ES8328_SUPPLY_NUM];
};
@@ -439,54 +463,131 @@ static int es8328_mute(struct snd_soc_dai *dai, int mute)
mute ? ES8328_DACCONTROL3_DACMUTE : 0);
}
+static int es8328_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
+
+ if (es8328->sysclk_constraints)
+ snd_pcm_hw_constraint_list(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ es8328->sysclk_constraints);
+
+ return 0;
+}
+
static int es8328_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct snd_soc_codec *codec = dai->codec;
struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
- int clk_rate;
int i;
int reg;
- u8 ratio;
+ int wl;
+ int ratio;
+
+ if (!es8328->sysclk_constraints) {
+ dev_err(codec->dev, "No MCLK configured\n");
+ return -EINVAL;
+ }
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
reg = ES8328_DACCONTROL2;
else
reg = ES8328_ADCCONTROL5;
- clk_rate = clk_get_rate(es8328->clk);
+ for (i = 0; i < es8328->sysclk_constraints->count; i++)
+ if (es8328->sysclk_constraints->list[i] == params_rate(params))
+ break;
- if ((clk_rate != ES8328_SYSCLK_RATE_1X) &&
- (clk_rate != ES8328_SYSCLK_RATE_2X)) {
- dev_err(codec->dev,
- "%s: clock is running at %d Hz, not %d or %d Hz\n",
- __func__, clk_rate,
- ES8328_SYSCLK_RATE_1X, ES8328_SYSCLK_RATE_2X);
+ if (i == es8328->sysclk_constraints->count) {
+ dev_err(codec->dev, "LRCLK %d unsupported with current clock\n",
+ params_rate(params));
return -EINVAL;
}
- /* find master mode MCLK to sampling frequency ratio */
- ratio = mclk_ratios[0].rate;
- for (i = 1; i < ARRAY_SIZE(mclk_ratios); i++)
- if (params_rate(params) <= mclk_ratios[i].rate)
- ratio = mclk_ratios[i].ratio;
+ ratio = es8328->mclk_ratios[i];
+ snd_soc_update_bits(codec, ES8328_MASTERMODE,
+ ES8328_MASTERMODE_MCLKDIV2,
+ es8328->mclkdiv2 ? ES8328_MASTERMODE_MCLKDIV2 : 0);
+
+ switch (params_width(params)) {
+ case 16:
+ wl = 3;
+ break;
+ case 18:
+ wl = 2;
+ break;
+ case 20:
+ wl = 1;
+ break;
+ case 24:
+ wl = 0;
+ break;
+ case 32:
+ wl = 4;
+ break;
+ default:
+ return -EINVAL;
+ }
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ snd_soc_update_bits(codec, ES8328_DACCONTROL1,
+ ES8328_DACCONTROL1_DACWL_MASK,
+ wl << ES8328_DACCONTROL1_DACWL_SHIFT);
+
es8328->playback_fs = params_rate(params);
es8328_set_deemph(codec);
- }
+ } else
+ snd_soc_update_bits(codec, ES8328_ADCCONTROL4,
+ ES8328_ADCCONTROL4_ADCWL_MASK,
+ wl << ES8328_ADCCONTROL4_ADCWL_SHIFT);
return snd_soc_update_bits(codec, reg, ES8328_RATEMASK, ratio);
}
+static int es8328_set_sysclk(struct snd_soc_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
+ int mclkdiv2 = 0;
+
+ switch (freq) {
+ case 0:
+ es8328->sysclk_constraints = NULL;
+ es8328->mclk_ratios = NULL;
+ break;
+ case 22579200:
+ mclkdiv2 = 1;
+ /* fallthru */
+ case 11289600:
+ es8328->sysclk_constraints = &constraints_11289;
+ es8328->mclk_ratios = ratios_11289;
+ break;
+ case 24576000:
+ mclkdiv2 = 1;
+ /* fallthru */
+ case 12288000:
+ es8328->sysclk_constraints = &constraints_12288;
+ es8328->mclk_ratios = ratios_12288;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ es8328->mclkdiv2 = mclkdiv2;
+ return 0;
+}
+
static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
- struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
- int clk_rate;
- u8 mode = ES8328_DACCONTROL1_DACWL_16;
+ u8 dac_mode = 0;
+ u8 adc_mode = 0;
/* set master/slave audio interface */
if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBM_CFM)
@@ -495,13 +596,16 @@ static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai,
/* interface format */
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
- mode |= ES8328_DACCONTROL1_DACFORMAT_I2S;
+ dac_mode |= ES8328_DACCONTROL1_DACFORMAT_I2S;
+ adc_mode |= ES8328_ADCCONTROL4_ADCFORMAT_I2S;
break;
case SND_SOC_DAIFMT_RIGHT_J:
- mode |= ES8328_DACCONTROL1_DACFORMAT_RJUST;
+ dac_mode |= ES8328_DACCONTROL1_DACFORMAT_RJUST;
+ adc_mode |= ES8328_ADCCONTROL4_ADCFORMAT_RJUST;
break;
case SND_SOC_DAIFMT_LEFT_J:
- mode |= ES8328_DACCONTROL1_DACFORMAT_LJUST;
+ dac_mode |= ES8328_DACCONTROL1_DACFORMAT_LJUST;
+ adc_mode |= ES8328_ADCCONTROL4_ADCFORMAT_LJUST;
break;
default:
return -EINVAL;
@@ -511,18 +615,14 @@ static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai,
if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF)
return -EINVAL;
- snd_soc_write(codec, ES8328_DACCONTROL1, mode);
- snd_soc_write(codec, ES8328_ADCCONTROL4, mode);
+ snd_soc_update_bits(codec, ES8328_DACCONTROL1,
+ ES8328_DACCONTROL1_DACFORMAT_MASK, dac_mode);
+ snd_soc_update_bits(codec, ES8328_ADCCONTROL4,
+ ES8328_ADCCONTROL4_ADCFORMAT_MASK, adc_mode);
/* Master serial port mode, with BCLK generated automatically */
- clk_rate = clk_get_rate(es8328->clk);
- if (clk_rate == ES8328_SYSCLK_RATE_1X)
- snd_soc_write(codec, ES8328_MASTERMODE,
- ES8328_MASTERMODE_MSC);
- else
- snd_soc_write(codec, ES8328_MASTERMODE,
- ES8328_MASTERMODE_MCLKDIV2 |
- ES8328_MASTERMODE_MSC);
+ snd_soc_update_bits(codec, ES8328_MASTERMODE,
+ ES8328_MASTERMODE_MSC, ES8328_MASTERMODE_MSC);
return 0;
}
@@ -579,8 +679,10 @@ static int es8328_set_bias_level(struct snd_soc_codec *codec,
}
static const struct snd_soc_dai_ops es8328_dai_ops = {
+ .startup = es8328_startup,
.hw_params = es8328_hw_params,
.digital_mute = es8328_mute,
+ .set_sysclk = es8328_set_sysclk,
.set_fmt = es8328_set_dai_fmt,
};
@@ -601,6 +703,7 @@ static struct snd_soc_dai_driver es8328_dai = {
.formats = ES8328_FORMATS,
},
.ops = &es8328_dai_ops,
+ .symmetric_rates = 1,
};
static int es8328_suspend(struct snd_soc_codec *codec)
@@ -708,6 +811,7 @@ const struct regmap_config es8328_regmap_config = {
.val_bits = 8,
.max_register = ES8328_REG_MAX,
.cache_type = REGCACHE_RBTREE,
+ .use_single_rw = true,
};
EXPORT_SYMBOL_GPL(es8328_regmap_config);
diff --git a/sound/soc/codecs/es8328.h b/sound/soc/codecs/es8328.h
index 156c748c89c7..1a736e72a929 100644
--- a/sound/soc/codecs/es8328.h
+++ b/sound/soc/codecs/es8328.h
@@ -22,7 +22,7 @@ int es8328_probe(struct device *dev, struct regmap *regmap);
#define ES8328_CONTROL1_VMIDSEL_50k (1 << 0)
#define ES8328_CONTROL1_VMIDSEL_500k (2 << 0)
#define ES8328_CONTROL1_VMIDSEL_5k (3 << 0)
-#define ES8328_CONTROL1_VMIDSEL_MASK (7 << 0)
+#define ES8328_CONTROL1_VMIDSEL_MASK (3 << 0)
#define ES8328_CONTROL1_ENREF (1 << 2)
#define ES8328_CONTROL1_SEQEN (1 << 3)
#define ES8328_CONTROL1_SAMEFS (1 << 4)
@@ -84,7 +84,20 @@ int es8328_probe(struct device *dev, struct regmap *regmap);
#define ES8328_ADCCONTROL1 0x09
#define ES8328_ADCCONTROL2 0x0a
#define ES8328_ADCCONTROL3 0x0b
+
#define ES8328_ADCCONTROL4 0x0c
+#define ES8328_ADCCONTROL4_ADCFORMAT_MASK (3 << 0)
+#define ES8328_ADCCONTROL4_ADCFORMAT_I2S (0 << 0)
+#define ES8328_ADCCONTROL4_ADCFORMAT_LJUST (1 << 0)
+#define ES8328_ADCCONTROL4_ADCFORMAT_RJUST (2 << 0)
+#define ES8328_ADCCONTROL4_ADCFORMAT_PCM (3 << 0)
+#define ES8328_ADCCONTROL4_ADCWL_SHIFT 2
+#define ES8328_ADCCONTROL4_ADCWL_MASK (7 << 2)
+#define ES8328_ADCCONTROL4_ADCLRP_I2S_POL_NORMAL (0 << 5)
+#define ES8328_ADCCONTROL4_ADCLRP_I2S_POL_INV (1 << 5)
+#define ES8328_ADCCONTROL4_ADCLRP_PCM_MSB_CLK2 (0 << 5)
+#define ES8328_ADCCONTROL4_ADCLRP_PCM_MSB_CLK1 (1 << 5)
+
#define ES8328_ADCCONTROL5 0x0d
#define ES8328_ADCCONTROL5_RATEMASK (0x1f << 0)
@@ -109,15 +122,13 @@ int es8328_probe(struct device *dev, struct regmap *regmap);
#define ES8328_ADCCONTROL14 0x16
#define ES8328_DACCONTROL1 0x17
+#define ES8328_DACCONTROL1_DACFORMAT_MASK (3 << 1)
#define ES8328_DACCONTROL1_DACFORMAT_I2S (0 << 1)
#define ES8328_DACCONTROL1_DACFORMAT_LJUST (1 << 1)
#define ES8328_DACCONTROL1_DACFORMAT_RJUST (2 << 1)
#define ES8328_DACCONTROL1_DACFORMAT_PCM (3 << 1)
-#define ES8328_DACCONTROL1_DACWL_24 (0 << 3)
-#define ES8328_DACCONTROL1_DACWL_20 (1 << 3)
-#define ES8328_DACCONTROL1_DACWL_18 (2 << 3)
-#define ES8328_DACCONTROL1_DACWL_16 (3 << 3)
-#define ES8328_DACCONTROL1_DACWL_32 (4 << 3)
+#define ES8328_DACCONTROL1_DACWL_SHIFT 3
+#define ES8328_DACCONTROL1_DACWL_MASK (7 << 3)
#define ES8328_DACCONTROL1_DACLRP_I2S_POL_NORMAL (0 << 6)
#define ES8328_DACCONTROL1_DACLRP_I2S_POL_INV (1 << 6)
#define ES8328_DACCONTROL1_DACLRP_PCM_MSB_CLK2 (0 << 6)
diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c
index 26f9459cb3bc..181cd3bf0b92 100644
--- a/sound/soc/codecs/hdac_hdmi.c
+++ b/sound/soc/codecs/hdac_hdmi.c
@@ -29,6 +29,7 @@
#include <sound/hdaudio_ext.h>
#include <sound/hda_i915.h>
#include <sound/pcm_drm_eld.h>
+#include <sound/hda_chmap.h>
#include "../../hda/local.h"
#include "hdac_hdmi.h"
@@ -60,11 +61,17 @@ struct hdac_hdmi_cvt {
struct hdac_hdmi_cvt_params params;
};
+/* Currently only spk_alloc, more to be added */
+struct hdac_hdmi_parsed_eld {
+ u8 spk_alloc;
+};
+
struct hdac_hdmi_eld {
bool monitor_present;
bool eld_valid;
int eld_size;
char eld_buffer[ELD_MAX_SIZE];
+ struct hdac_hdmi_parsed_eld info;
};
struct hdac_hdmi_pin {
@@ -76,6 +83,10 @@ struct hdac_hdmi_pin {
struct hdac_ext_device *edev;
int repoll_count;
struct delayed_work work;
+ struct mutex lock;
+ bool chmap_set;
+ unsigned char chmap[8]; /* ALSA API channel-map */
+ int channels; /* current number of channels */
};
struct hdac_hdmi_pcm {
@@ -100,8 +111,22 @@ struct hdac_hdmi_priv {
int num_pin;
int num_cvt;
struct mutex pin_mutex;
+ struct hdac_chmap chmap;
};
+static struct hdac_hdmi_pcm *get_hdmi_pcm_from_id(struct hdac_hdmi_priv *hdmi,
+ int pcm_idx)
+{
+ struct hdac_hdmi_pcm *pcm;
+
+ list_for_each_entry(pcm, &hdmi->pcm_list, head) {
+ if (pcm->pcm_id == pcm_idx)
+ return pcm;
+ }
+
+ return NULL;
+}
+
static inline struct hdac_ext_device *to_hda_ext_device(struct device *dev)
{
struct hdac_device *hdac = dev_to_hdac_dev(dev);
@@ -278,26 +303,31 @@ static int hdac_hdmi_setup_audio_infoframe(struct hdac_ext_device *hdac,
int i;
const u8 *eld_buf;
u8 conn_type;
- int channels = 2;
+ int channels, ca;
list_for_each_entry(pin, &hdmi->pin_list, head) {
if (pin->nid == pin_nid)
break;
}
+ ca = snd_hdac_channel_allocation(&hdac->hdac, pin->eld.info.spk_alloc,
+ pin->channels, pin->chmap_set, true, pin->chmap);
+
+ channels = snd_hdac_get_active_channels(ca);
+ hdmi->chmap.ops.set_channel_count(&hdac->hdac, cvt_nid, channels);
+
+ snd_hdac_setup_channel_mapping(&hdmi->chmap, pin->nid, false, ca,
+ pin->channels, pin->chmap, pin->chmap_set);
+
eld_buf = pin->eld.eld_buffer;
conn_type = drm_eld_get_conn_type(eld_buf);
- /* setup channel count */
- snd_hdac_codec_write(&hdac->hdac, cvt_nid, 0,
- AC_VERB_SET_CVT_CHAN_COUNT, channels - 1);
-
switch (conn_type) {
case DRM_ELD_CONN_TYPE_HDMI:
hdmi_audio_infoframe_init(&frame);
- /* Default stereo for now */
frame.channels = channels;
+ frame.channel_allocation = ca;
ret = hdmi_audio_infoframe_pack(&frame, buffer, sizeof(buffer));
if (ret < 0)
@@ -311,7 +341,7 @@ static int hdac_hdmi_setup_audio_infoframe(struct hdac_ext_device *hdac,
dp_ai.len = 0x1b;
dp_ai.ver = 0x11 << 2;
dp_ai.CC02_CT47 = channels - 1;
- dp_ai.CA = 0;
+ dp_ai.CA = ca;
dip = (u8 *)&dp_ai;
break;
@@ -370,17 +400,23 @@ static int hdac_hdmi_playback_prepare(struct snd_pcm_substream *substream,
struct hdac_ext_device *hdac = snd_soc_dai_get_drvdata(dai);
struct hdac_hdmi_priv *hdmi = hdac->private_data;
struct hdac_hdmi_dai_pin_map *dai_map;
+ struct hdac_hdmi_pin *pin;
struct hdac_ext_dma_params *dd;
int ret;
dai_map = &hdmi->dai_map[dai->id];
+ pin = dai_map->pin;
dd = (struct hdac_ext_dma_params *)snd_soc_dai_get_dma_data(dai, substream);
dev_dbg(&hdac->hdac.dev, "stream tag from cpu dai %d format in cvt 0x%x\n",
dd->stream_tag, dd->format);
+ mutex_lock(&pin->lock);
+ pin->channels = substream->runtime->channels;
+
ret = hdac_hdmi_setup_audio_infoframe(hdac, dai_map->cvt->nid,
dai_map->pin->nid);
+ mutex_unlock(&pin->lock);
if (ret < 0)
return ret;
@@ -640,6 +676,12 @@ static void hdac_hdmi_pcm_close(struct snd_pcm_substream *substream,
snd_hdac_codec_write(&hdac->hdac, dai_map->pin->nid, 0,
AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE);
+ mutex_lock(&dai_map->pin->lock);
+ dai_map->pin->chmap_set = false;
+ memset(dai_map->pin->chmap, 0, sizeof(dai_map->pin->chmap));
+ dai_map->pin->channels = 0;
+ mutex_unlock(&dai_map->pin->lock);
+
dai_map->pin = NULL;
}
}
@@ -647,10 +689,19 @@ static void hdac_hdmi_pcm_close(struct snd_pcm_substream *substream,
static int
hdac_hdmi_query_cvt_params(struct hdac_device *hdac, struct hdac_hdmi_cvt *cvt)
{
+ unsigned int chans;
+ struct hdac_ext_device *edev = to_ehdac_device(hdac);
+ struct hdac_hdmi_priv *hdmi = edev->private_data;
int err;
- /* Only stereo supported as of now */
- cvt->params.channels_min = cvt->params.channels_max = 2;
+ chans = get_wcaps(hdac, cvt->nid);
+ chans = get_wcaps_channels(chans);
+
+ cvt->params.channels_min = 2;
+
+ cvt->params.channels_max = chans;
+ if (chans > hdmi->chmap.channels_max)
+ hdmi->chmap.channels_max = chans;
err = snd_hdac_query_supported_pcm(hdac, cvt->nid,
&cvt->params.rates,
@@ -1008,6 +1059,12 @@ static int hdac_hdmi_add_cvt(struct hdac_ext_device *edev, hda_nid_t nid)
return hdac_hdmi_query_cvt_params(&edev->hdac, cvt);
}
+static void hdac_hdmi_parse_eld(struct hdac_ext_device *edev,
+ struct hdac_hdmi_pin *pin)
+{
+ pin->eld.info.spk_alloc = pin->eld.eld_buffer[DRM_ELD_SPEAKER];
+}
+
static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin, int repoll)
{
struct hdac_ext_device *edev = pin->edev;
@@ -1065,6 +1122,7 @@ static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin, int repoll)
snd_jack_report(pcm->jack, SND_JACK_AVOUT);
}
+ hdac_hdmi_parse_eld(edev, pin);
print_hex_dump_bytes("ELD: ", DUMP_PREFIX_OFFSET,
pin->eld.eld_buffer, pin->eld.eld_size);
@@ -1123,6 +1181,7 @@ static int hdac_hdmi_add_pin(struct hdac_ext_device *edev, hda_nid_t nid)
hdmi->num_pin++;
pin->edev = edev;
+ mutex_init(&pin->lock);
INIT_DELAYED_WORK(&pin->work, hdac_hdmi_repoll_eld);
return 0;
@@ -1342,6 +1401,19 @@ static struct i915_audio_component_audio_ops aops = {
.pin_eld_notify = hdac_hdmi_eld_notify_cb,
};
+static struct snd_pcm *hdac_hdmi_get_pcm_from_id(struct snd_soc_card *card,
+ int device)
+{
+ struct snd_soc_pcm_runtime *rtd;
+
+ list_for_each_entry(rtd, &card->rtd_list, list) {
+ if (rtd->pcm && (rtd->pcm->device == device))
+ return rtd->pcm;
+ }
+
+ return NULL;
+}
+
int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int device)
{
char jack_name[NAME_SIZE];
@@ -1351,6 +1423,8 @@ int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int device)
snd_soc_component_get_dapm(&codec->component);
struct hdac_hdmi_priv *hdmi = edev->private_data;
struct hdac_hdmi_pcm *pcm;
+ struct snd_pcm *snd_pcm;
+ int err;
/*
* this is a new PCM device, create new pcm and
@@ -1362,6 +1436,18 @@ int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int device)
pcm->pcm_id = device;
pcm->cvt = hdmi->dai_map[dai->id].cvt;
+ snd_pcm = hdac_hdmi_get_pcm_from_id(dai->component->card, device);
+ if (snd_pcm) {
+ err = snd_hdac_add_chmap_ctls(snd_pcm, device, &hdmi->chmap);
+ if (err < 0) {
+ dev_err(&edev->hdac.dev,
+ "chmap control add failed with err: %d for pcm: %d\n",
+ err, device);
+ kfree(pcm);
+ return err;
+ }
+ }
+
list_add_tail(&pcm->head, &hdmi->pcm_list);
sprintf(jack_name, "HDMI/DP, pcm=%d Jack", device);
@@ -1378,10 +1464,18 @@ static int hdmi_codec_probe(struct snd_soc_codec *codec)
struct snd_soc_dapm_context *dapm =
snd_soc_component_get_dapm(&codec->component);
struct hdac_hdmi_pin *pin;
+ struct hdac_ext_link *hlink = NULL;
int ret;
edev->scodec = codec;
+ /*
+ * hold the ref while we probe, also no need to drop the ref on
+ * exit, we call pm_runtime_suspend() so that will do for us
+ */
+ hlink = snd_hdac_ext_bus_get_link(edev->ebus, dev_name(&edev->hdac.dev));
+ snd_hdac_ext_bus_link_get(edev->ebus, hlink);
+
ret = create_fill_widget_route_map(dapm);
if (ret < 0)
return ret;
@@ -1420,32 +1514,39 @@ static int hdmi_codec_remove(struct snd_soc_codec *codec)
}
#ifdef CONFIG_PM
-static int hdmi_codec_resume(struct snd_soc_codec *codec)
+static int hdmi_codec_prepare(struct device *dev)
{
- struct hdac_ext_device *edev = snd_soc_codec_get_drvdata(codec);
+ struct hdac_ext_device *edev = to_hda_ext_device(dev);
+ struct hdac_device *hdac = &edev->hdac;
+
+ pm_runtime_get_sync(&edev->hdac.dev);
+
+ /*
+ * Power down afg.
+ * codec_read is preferred over codec_write to set the power state.
+ * This way verb is send to set the power state and response
+ * is received. So setting power state is ensured without using loop
+ * to read the state.
+ */
+ snd_hdac_codec_read(hdac, hdac->afg, 0, AC_VERB_SET_POWER_STATE,
+ AC_PWRST_D3);
+
+ return 0;
+}
+
+static void hdmi_codec_complete(struct device *dev)
+{
+ struct hdac_ext_device *edev = to_hda_ext_device(dev);
struct hdac_hdmi_priv *hdmi = edev->private_data;
struct hdac_hdmi_pin *pin;
struct hdac_device *hdac = &edev->hdac;
- struct hdac_bus *bus = hdac->bus;
- int err;
- unsigned long timeout;
-
- hdac_hdmi_skl_enable_all_pins(&edev->hdac);
- hdac_hdmi_skl_enable_dp12(&edev->hdac);
/* Power up afg */
- if (!snd_hdac_check_power_state(hdac, hdac->afg, AC_PWRST_D0)) {
+ snd_hdac_codec_read(hdac, hdac->afg, 0, AC_VERB_SET_POWER_STATE,
+ AC_PWRST_D0);
- snd_hdac_codec_write(hdac, hdac->afg, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
-
- /* Wait till power state is set to D0 */
- timeout = jiffies + msecs_to_jiffies(1000);
- while (!snd_hdac_check_power_state(hdac, hdac->afg, AC_PWRST_D0)
- && time_before(jiffies, timeout)) {
- msleep(50);
- }
- }
+ hdac_hdmi_skl_enable_all_pins(&edev->hdac);
+ hdac_hdmi_skl_enable_dp12(&edev->hdac);
/*
* As the ELD notify callback request is not entertained while the
@@ -1455,44 +1556,96 @@ static int hdmi_codec_resume(struct snd_soc_codec *codec)
list_for_each_entry(pin, &hdmi->pin_list, head)
hdac_hdmi_present_sense(pin, 1);
- /*
- * Codec power is turned ON during controller resume.
- * Turn it OFF here
- */
- err = snd_hdac_display_power(bus, false);
- if (err < 0) {
- dev_err(bus->dev,
- "Cannot turn OFF display power on i915, err: %d\n",
- err);
- return err;
- }
-
- return 0;
+ pm_runtime_put_sync(&edev->hdac.dev);
}
#else
-#define hdmi_codec_resume NULL
+#define hdmi_codec_prepare NULL
+#define hdmi_codec_complete NULL
#endif
static struct snd_soc_codec_driver hdmi_hda_codec = {
.probe = hdmi_codec_probe,
.remove = hdmi_codec_remove,
- .resume = hdmi_codec_resume,
.idle_bias_off = true,
};
+static void hdac_hdmi_get_chmap(struct hdac_device *hdac, int pcm_idx,
+ unsigned char *chmap)
+{
+ struct hdac_ext_device *edev = to_ehdac_device(hdac);
+ struct hdac_hdmi_priv *hdmi = edev->private_data;
+ struct hdac_hdmi_pcm *pcm = get_hdmi_pcm_from_id(hdmi, pcm_idx);
+ struct hdac_hdmi_pin *pin = pcm->pin;
+
+ /* chmap is already set to 0 in caller */
+ if (!pin)
+ return;
+
+ memcpy(chmap, pin->chmap, ARRAY_SIZE(pin->chmap));
+}
+
+static void hdac_hdmi_set_chmap(struct hdac_device *hdac, int pcm_idx,
+ unsigned char *chmap, int prepared)
+{
+ struct hdac_ext_device *edev = to_ehdac_device(hdac);
+ struct hdac_hdmi_priv *hdmi = edev->private_data;
+ struct hdac_hdmi_pcm *pcm = get_hdmi_pcm_from_id(hdmi, pcm_idx);
+ struct hdac_hdmi_pin *pin = pcm->pin;
+
+ mutex_lock(&pin->lock);
+ pin->chmap_set = true;
+ memcpy(pin->chmap, chmap, ARRAY_SIZE(pin->chmap));
+ if (prepared)
+ hdac_hdmi_setup_audio_infoframe(edev, pcm->cvt->nid, pin->nid);
+ mutex_unlock(&pin->lock);
+}
+
+static bool is_hdac_hdmi_pcm_attached(struct hdac_device *hdac, int pcm_idx)
+{
+ struct hdac_ext_device *edev = to_ehdac_device(hdac);
+ struct hdac_hdmi_priv *hdmi = edev->private_data;
+ struct hdac_hdmi_pcm *pcm = get_hdmi_pcm_from_id(hdmi, pcm_idx);
+ struct hdac_hdmi_pin *pin = pcm->pin;
+
+ return pin ? true:false;
+}
+
+static int hdac_hdmi_get_spk_alloc(struct hdac_device *hdac, int pcm_idx)
+{
+ struct hdac_ext_device *edev = to_ehdac_device(hdac);
+ struct hdac_hdmi_priv *hdmi = edev->private_data;
+ struct hdac_hdmi_pcm *pcm = get_hdmi_pcm_from_id(hdmi, pcm_idx);
+ struct hdac_hdmi_pin *pin = pcm->pin;
+
+ if (!pin || !pin->eld.eld_valid)
+ return 0;
+
+ return pin->eld.info.spk_alloc;
+}
+
static int hdac_hdmi_dev_probe(struct hdac_ext_device *edev)
{
struct hdac_device *codec = &edev->hdac;
struct hdac_hdmi_priv *hdmi_priv;
struct snd_soc_dai_driver *hdmi_dais = NULL;
+ struct hdac_ext_link *hlink = NULL;
int num_dais = 0;
int ret = 0;
+ /* hold the ref while we probe */
+ hlink = snd_hdac_ext_bus_get_link(edev->ebus, dev_name(&edev->hdac.dev));
+ snd_hdac_ext_bus_link_get(edev->ebus, hlink);
+
hdmi_priv = devm_kzalloc(&codec->dev, sizeof(*hdmi_priv), GFP_KERNEL);
if (hdmi_priv == NULL)
return -ENOMEM;
edev->private_data = hdmi_priv;
+ snd_hdac_register_chmap_ops(codec, &hdmi_priv->chmap);
+ hdmi_priv->chmap.ops.get_chmap = hdac_hdmi_get_chmap;
+ hdmi_priv->chmap.ops.set_chmap = hdac_hdmi_set_chmap;
+ hdmi_priv->chmap.ops.is_pcm_attached = is_hdac_hdmi_pcm_attached;
+ hdmi_priv->chmap.ops.get_spk_alloc = hdac_hdmi_get_spk_alloc;
dev_set_drvdata(&codec->dev, edev);
@@ -1521,8 +1674,12 @@ static int hdac_hdmi_dev_probe(struct hdac_ext_device *edev)
}
/* ASoC specific initialization */
- return snd_soc_register_codec(&codec->dev, &hdmi_hda_codec,
- hdmi_dais, num_dais);
+ ret = snd_soc_register_codec(&codec->dev, &hdmi_hda_codec,
+ hdmi_dais, num_dais);
+
+ snd_hdac_ext_bus_link_put(edev->ebus, hlink);
+
+ return ret;
}
static int hdac_hdmi_dev_remove(struct hdac_ext_device *edev)
@@ -1561,7 +1718,8 @@ static int hdac_hdmi_runtime_suspend(struct device *dev)
struct hdac_ext_device *edev = to_hda_ext_device(dev);
struct hdac_device *hdac = &edev->hdac;
struct hdac_bus *bus = hdac->bus;
- unsigned long timeout;
+ struct hdac_ext_bus *ebus = hbus_to_ebus(bus);
+ struct hdac_ext_link *hlink = NULL;
int err;
dev_dbg(dev, "Enter: %s\n", __func__);
@@ -1570,26 +1728,24 @@ static int hdac_hdmi_runtime_suspend(struct device *dev)
if (!bus)
return 0;
- /* Power down afg */
- if (!snd_hdac_check_power_state(hdac, hdac->afg, AC_PWRST_D3)) {
- snd_hdac_codec_write(hdac, hdac->afg, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
-
- /* Wait till power state is set to D3 */
- timeout = jiffies + msecs_to_jiffies(1000);
- while (!snd_hdac_check_power_state(hdac, hdac->afg, AC_PWRST_D3)
- && time_before(jiffies, timeout)) {
-
- msleep(50);
- }
- }
-
+ /*
+ * Power down afg.
+ * codec_read is preferred over codec_write to set the power state.
+ * This way verb is send to set the power state and response
+ * is received. So setting power state is ensured without using loop
+ * to read the state.
+ */
+ snd_hdac_codec_read(hdac, hdac->afg, 0, AC_VERB_SET_POWER_STATE,
+ AC_PWRST_D3);
err = snd_hdac_display_power(bus, false);
if (err < 0) {
dev_err(bus->dev, "Cannot turn on display power on i915\n");
return err;
}
+ hlink = snd_hdac_ext_bus_get_link(ebus, dev_name(dev));
+ snd_hdac_ext_bus_link_put(ebus, hlink);
+
return 0;
}
@@ -1598,6 +1754,8 @@ static int hdac_hdmi_runtime_resume(struct device *dev)
struct hdac_ext_device *edev = to_hda_ext_device(dev);
struct hdac_device *hdac = &edev->hdac;
struct hdac_bus *bus = hdac->bus;
+ struct hdac_ext_bus *ebus = hbus_to_ebus(bus);
+ struct hdac_ext_link *hlink = NULL;
int err;
dev_dbg(dev, "Enter: %s\n", __func__);
@@ -1606,6 +1764,9 @@ static int hdac_hdmi_runtime_resume(struct device *dev)
if (!bus)
return 0;
+ hlink = snd_hdac_ext_bus_get_link(ebus, dev_name(dev));
+ snd_hdac_ext_bus_link_get(ebus, hlink);
+
err = snd_hdac_display_power(bus, true);
if (err < 0) {
dev_err(bus->dev, "Cannot turn on display power on i915\n");
@@ -1616,9 +1777,8 @@ static int hdac_hdmi_runtime_resume(struct device *dev)
hdac_hdmi_skl_enable_dp12(&edev->hdac);
/* Power up afg */
- if (!snd_hdac_check_power_state(hdac, hdac->afg, AC_PWRST_D0))
- snd_hdac_codec_write(hdac, hdac->afg, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
+ snd_hdac_codec_read(hdac, hdac->afg, 0, AC_VERB_SET_POWER_STATE,
+ AC_PWRST_D0);
return 0;
}
@@ -1629,6 +1789,8 @@ static int hdac_hdmi_runtime_resume(struct device *dev)
static const struct dev_pm_ops hdac_hdmi_pm = {
SET_RUNTIME_PM_OPS(hdac_hdmi_runtime_suspend, hdac_hdmi_runtime_resume, NULL)
+ .prepare = hdmi_codec_prepare,
+ .complete = hdmi_codec_complete,
};
static const struct hda_device_id hdmi_list[] = {
diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c
new file mode 100644
index 000000000000..8e36e883e453
--- /dev/null
+++ b/sound/soc/codecs/hdmi-codec.c
@@ -0,0 +1,432 @@
+/*
+ * ALSA SoC codec for HDMI encoder drivers
+ * Copyright (C) 2015 Texas Instruments Incorporated - http://www.ti.com/
+ * Author: Jyri Sarha <jsarha@ti.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ */
+#include <linux/module.h>
+#include <linux/string.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/pcm_drm_eld.h>
+#include <sound/hdmi-codec.h>
+#include <sound/pcm_iec958.h>
+
+#include <drm/drm_crtc.h> /* This is only to get MAX_ELD_BYTES */
+
+struct hdmi_codec_priv {
+ struct hdmi_codec_pdata hcd;
+ struct snd_soc_dai_driver *daidrv;
+ struct hdmi_codec_daifmt daifmt[2];
+ struct mutex current_stream_lock;
+ struct snd_pcm_substream *current_stream;
+ struct snd_pcm_hw_constraint_list ratec;
+ uint8_t eld[MAX_ELD_BYTES];
+};
+
+static const struct snd_soc_dapm_widget hdmi_widgets[] = {
+ SND_SOC_DAPM_OUTPUT("TX"),
+};
+
+static const struct snd_soc_dapm_route hdmi_routes[] = {
+ { "TX", NULL, "Playback" },
+};
+
+enum {
+ DAI_ID_I2S = 0,
+ DAI_ID_SPDIF,
+};
+
+static int hdmi_eld_ctl_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct snd_soc_component *component = snd_kcontrol_chip(kcontrol);
+ struct hdmi_codec_priv *hcp = snd_soc_component_get_drvdata(component);
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES;
+ uinfo->count = sizeof(hcp->eld);
+
+ return 0;
+}
+
+static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *component = snd_kcontrol_chip(kcontrol);
+ struct hdmi_codec_priv *hcp = snd_soc_component_get_drvdata(component);
+
+ memcpy(ucontrol->value.bytes.data, hcp->eld, sizeof(hcp->eld));
+
+ return 0;
+}
+
+static const struct snd_kcontrol_new hdmi_controls[] = {
+ {
+ .access = SNDRV_CTL_ELEM_ACCESS_READ |
+ SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = "ELD",
+ .info = hdmi_eld_ctl_info,
+ .get = hdmi_eld_ctl_get,
+ },
+};
+
+static int hdmi_codec_new_stream(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai);
+ int ret = 0;
+
+ mutex_lock(&hcp->current_stream_lock);
+ if (!hcp->current_stream) {
+ hcp->current_stream = substream;
+ } else if (hcp->current_stream != substream) {
+ dev_err(dai->dev, "Only one simultaneous stream supported!\n");
+ ret = -EINVAL;
+ }
+ mutex_unlock(&hcp->current_stream_lock);
+
+ return ret;
+}
+
+static int hdmi_codec_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai);
+ int ret = 0;
+
+ dev_dbg(dai->dev, "%s()\n", __func__);
+
+ ret = hdmi_codec_new_stream(substream, dai);
+ if (ret)
+ return ret;
+
+ if (hcp->hcd.ops->audio_startup) {
+ ret = hcp->hcd.ops->audio_startup(dai->dev->parent);
+ if (ret) {
+ mutex_lock(&hcp->current_stream_lock);
+ hcp->current_stream = NULL;
+ mutex_unlock(&hcp->current_stream_lock);
+ return ret;
+ }
+ }
+
+ if (hcp->hcd.ops->get_eld) {
+ ret = hcp->hcd.ops->get_eld(dai->dev->parent, hcp->eld,
+ sizeof(hcp->eld));
+
+ if (!ret) {
+ ret = snd_pcm_hw_constraint_eld(substream->runtime,
+ hcp->eld);
+ if (ret)
+ return ret;
+ }
+ }
+ return 0;
+}
+
+static void hdmi_codec_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai);
+
+ dev_dbg(dai->dev, "%s()\n", __func__);
+
+ WARN_ON(hcp->current_stream != substream);
+
+ hcp->hcd.ops->audio_shutdown(dai->dev->parent);
+
+ mutex_lock(&hcp->current_stream_lock);
+ hcp->current_stream = NULL;
+ mutex_unlock(&hcp->current_stream_lock);
+}
+
+static int hdmi_codec_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai);
+ struct hdmi_codec_params hp = {
+ .iec = {
+ .status = { 0 },
+ .subcode = { 0 },
+ .pad = 0,
+ .dig_subframe = { 0 },
+ }
+ };
+ int ret;
+
+ dev_dbg(dai->dev, "%s() width %d rate %d channels %d\n", __func__,
+ params_width(params), params_rate(params),
+ params_channels(params));
+
+ if (params_width(params) > 24)
+ params->msbits = 24;
+
+ ret = snd_pcm_create_iec958_consumer_hw_params(params, hp.iec.status,
+ sizeof(hp.iec.status));
+ if (ret < 0) {
+ dev_err(dai->dev, "Creating IEC958 channel status failed %d\n",
+ ret);
+ return ret;
+ }
+
+ ret = hdmi_codec_new_stream(substream, dai);
+ if (ret)
+ return ret;
+
+ hdmi_audio_infoframe_init(&hp.cea);
+ hp.cea.channels = params_channels(params);
+ hp.cea.coding_type = HDMI_AUDIO_CODING_TYPE_STREAM;
+ hp.cea.sample_size = HDMI_AUDIO_SAMPLE_SIZE_STREAM;
+ hp.cea.sample_frequency = HDMI_AUDIO_SAMPLE_FREQUENCY_STREAM;
+
+ hp.sample_width = params_width(params);
+ hp.sample_rate = params_rate(params);
+ hp.channels = params_channels(params);
+
+ return hcp->hcd.ops->hw_params(dai->dev->parent, &hcp->daifmt[dai->id],
+ &hp);
+}
+
+static int hdmi_codec_set_fmt(struct snd_soc_dai *dai,
+ unsigned int fmt)
+{
+ struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai);
+ struct hdmi_codec_daifmt cf = { 0 };
+ int ret = 0;
+
+ dev_dbg(dai->dev, "%s()\n", __func__);
+
+ if (dai->id == DAI_ID_SPDIF) {
+ cf.fmt = HDMI_SPDIF;
+ } else {
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ cf.bit_clk_master = 1;
+ cf.frame_clk_master = 1;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFM:
+ cf.frame_clk_master = 1;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFS:
+ cf.bit_clk_master = 1;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ cf.frame_clk_inv = 1;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ cf.bit_clk_inv = 1;
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ cf.frame_clk_inv = 1;
+ cf.bit_clk_inv = 1;
+ break;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ cf.fmt = HDMI_I2S;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ cf.fmt = HDMI_DSP_A;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ cf.fmt = HDMI_DSP_B;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ cf.fmt = HDMI_RIGHT_J;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ cf.fmt = HDMI_LEFT_J;
+ break;
+ case SND_SOC_DAIFMT_AC97:
+ cf.fmt = HDMI_AC97;
+ break;
+ default:
+ dev_err(dai->dev, "Invalid DAI interface format\n");
+ return -EINVAL;
+ }
+ }
+
+ hcp->daifmt[dai->id] = cf;
+
+ return ret;
+}
+
+static int hdmi_codec_digital_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai);
+
+ dev_dbg(dai->dev, "%s()\n", __func__);
+
+ if (hcp->hcd.ops->digital_mute)
+ return hcp->hcd.ops->digital_mute(dai->dev->parent, mute);
+
+ return 0;
+}
+
+static const struct snd_soc_dai_ops hdmi_dai_ops = {
+ .startup = hdmi_codec_startup,
+ .shutdown = hdmi_codec_shutdown,
+ .hw_params = hdmi_codec_hw_params,
+ .set_fmt = hdmi_codec_set_fmt,
+ .digital_mute = hdmi_codec_digital_mute,
+};
+
+
+#define HDMI_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\
+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\
+ SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 |\
+ SNDRV_PCM_RATE_192000)
+
+#define SPDIF_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |\
+ SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE |\
+ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE |\
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE)
+
+/*
+ * This list is only for formats allowed on the I2S bus. So there is
+ * some formats listed that are not supported by HDMI interface. For
+ * instance allowing the 32-bit formats enables 24-precision with CPU
+ * DAIs that do not support 24-bit formats. If the extra formats cause
+ * problems, we should add the video side driver an option to disable
+ * them.
+ */
+#define I2S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |\
+ SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE |\
+ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE |\
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE |\
+ SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE)
+
+static struct snd_soc_dai_driver hdmi_i2s_dai = {
+ .name = "i2s-hifi",
+ .id = DAI_ID_I2S,
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 8,
+ .rates = HDMI_RATES,
+ .formats = I2S_FORMATS,
+ .sig_bits = 24,
+ },
+ .ops = &hdmi_dai_ops,
+};
+
+static const struct snd_soc_dai_driver hdmi_spdif_dai = {
+ .name = "spdif-hifi",
+ .id = DAI_ID_SPDIF,
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = HDMI_RATES,
+ .formats = SPDIF_FORMATS,
+ },
+ .ops = &hdmi_dai_ops,
+};
+
+static struct snd_soc_codec_driver hdmi_codec = {
+ .controls = hdmi_controls,
+ .num_controls = ARRAY_SIZE(hdmi_controls),
+ .dapm_widgets = hdmi_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(hdmi_widgets),
+ .dapm_routes = hdmi_routes,
+ .num_dapm_routes = ARRAY_SIZE(hdmi_routes),
+};
+
+static int hdmi_codec_probe(struct platform_device *pdev)
+{
+ struct hdmi_codec_pdata *hcd = pdev->dev.platform_data;
+ struct device *dev = &pdev->dev;
+ struct hdmi_codec_priv *hcp;
+ int dai_count, i = 0;
+ int ret;
+
+ dev_dbg(dev, "%s()\n", __func__);
+
+ if (!hcd) {
+ dev_err(dev, "%s: No plalform data\n", __func__);
+ return -EINVAL;
+ }
+
+ dai_count = hcd->i2s + hcd->spdif;
+ if (dai_count < 1 || !hcd->ops || !hcd->ops->hw_params ||
+ !hcd->ops->audio_shutdown) {
+ dev_err(dev, "%s: Invalid parameters\n", __func__);
+ return -EINVAL;
+ }
+
+ hcp = devm_kzalloc(dev, sizeof(*hcp), GFP_KERNEL);
+ if (!hcp)
+ return -ENOMEM;
+
+ hcp->hcd = *hcd;
+ mutex_init(&hcp->current_stream_lock);
+
+ hcp->daidrv = devm_kzalloc(dev, dai_count * sizeof(*hcp->daidrv),
+ GFP_KERNEL);
+ if (!hcp->daidrv)
+ return -ENOMEM;
+
+ if (hcd->i2s) {
+ hcp->daidrv[i] = hdmi_i2s_dai;
+ hcp->daidrv[i].playback.channels_max =
+ hcd->max_i2s_channels;
+ i++;
+ }
+
+ if (hcd->spdif)
+ hcp->daidrv[i] = hdmi_spdif_dai;
+
+ ret = snd_soc_register_codec(dev, &hdmi_codec, hcp->daidrv,
+ dai_count);
+ if (ret) {
+ dev_err(dev, "%s: snd_soc_register_codec() failed (%d)\n",
+ __func__, ret);
+ return ret;
+ }
+
+ dev_set_drvdata(dev, hcp);
+ return 0;
+}
+
+static int hdmi_codec_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_codec(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver hdmi_codec_driver = {
+ .driver = {
+ .name = HDMI_CODEC_DRV_NAME,
+ },
+ .probe = hdmi_codec_probe,
+ .remove = hdmi_codec_remove,
+};
+
+module_platform_driver(hdmi_codec_driver);
+
+MODULE_AUTHOR("Jyri Sarha <jsarha@ti.com>");
+MODULE_DESCRIPTION("HDMI Audio Codec Driver");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:" HDMI_CODEC_DRV_NAME);
diff --git a/sound/soc/codecs/max98371.c b/sound/soc/codecs/max98371.c
new file mode 100644
index 000000000000..cf0a39bb631a
--- /dev/null
+++ b/sound/soc/codecs/max98371.c
@@ -0,0 +1,441 @@
+/*
+ * max98371.c -- ALSA SoC Stereo MAX98371 driver
+ *
+ * Copyright 2015-16 Maxim Integrated Products
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/i2c.h>
+#include <linux/module.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+#include "max98371.h"
+
+static const char *const monomix_text[] = {
+ "Left", "Right", "LeftRightDiv2",
+};
+
+static const char *const hpf_cutoff_txt[] = {
+ "Disable", "DC Block", "50Hz",
+ "100Hz", "200Hz", "400Hz", "800Hz",
+};
+
+static SOC_ENUM_SINGLE_DECL(max98371_monomix, MAX98371_MONOMIX_CFG, 0,
+ monomix_text);
+
+static SOC_ENUM_SINGLE_DECL(max98371_hpf_cutoff, MAX98371_HPF, 0,
+ hpf_cutoff_txt);
+
+static const DECLARE_TLV_DB_RANGE(max98371_dht_min_gain,
+ 0, 1, TLV_DB_SCALE_ITEM(537, 66, 0),
+ 2, 3, TLV_DB_SCALE_ITEM(677, 82, 0),
+ 4, 5, TLV_DB_SCALE_ITEM(852, 104, 0),
+ 6, 7, TLV_DB_SCALE_ITEM(1072, 131, 0),
+ 8, 9, TLV_DB_SCALE_ITEM(1350, 165, 0),
+ 10, 11, TLV_DB_SCALE_ITEM(1699, 101, 0),
+);
+
+static const DECLARE_TLV_DB_RANGE(max98371_dht_max_gain,
+ 0, 1, TLV_DB_SCALE_ITEM(537, 66, 0),
+ 2, 3, TLV_DB_SCALE_ITEM(677, 82, 0),
+ 4, 5, TLV_DB_SCALE_ITEM(852, 104, 0),
+ 6, 7, TLV_DB_SCALE_ITEM(1072, 131, 0),
+ 8, 9, TLV_DB_SCALE_ITEM(1350, 165, 0),
+ 10, 11, TLV_DB_SCALE_ITEM(1699, 208, 0),
+);
+
+static const DECLARE_TLV_DB_RANGE(max98371_dht_rot_gain,
+ 0, 1, TLV_DB_SCALE_ITEM(-50, -50, 0),
+ 2, 6, TLV_DB_SCALE_ITEM(-100, -100, 0),
+ 7, 8, TLV_DB_SCALE_ITEM(-800, -200, 0),
+ 9, 11, TLV_DB_SCALE_ITEM(-1200, -300, 0),
+ 12, 13, TLV_DB_SCALE_ITEM(-2000, -200, 0),
+ 14, 15, TLV_DB_SCALE_ITEM(-2500, -500, 0),
+);
+
+static const struct reg_default max98371_reg[] = {
+ { 0x01, 0x00 },
+ { 0x02, 0x00 },
+ { 0x03, 0x00 },
+ { 0x04, 0x00 },
+ { 0x05, 0x00 },
+ { 0x06, 0x00 },
+ { 0x07, 0x00 },
+ { 0x08, 0x00 },
+ { 0x09, 0x00 },
+ { 0x0A, 0x00 },
+ { 0x10, 0x06 },
+ { 0x11, 0x08 },
+ { 0x14, 0x80 },
+ { 0x15, 0x00 },
+ { 0x16, 0x00 },
+ { 0x18, 0x00 },
+ { 0x19, 0x00 },
+ { 0x1C, 0x00 },
+ { 0x1D, 0x00 },
+ { 0x1E, 0x00 },
+ { 0x1F, 0x00 },
+ { 0x20, 0x00 },
+ { 0x21, 0x00 },
+ { 0x22, 0x00 },
+ { 0x23, 0x00 },
+ { 0x24, 0x00 },
+ { 0x25, 0x00 },
+ { 0x26, 0x00 },
+ { 0x27, 0x00 },
+ { 0x28, 0x00 },
+ { 0x29, 0x00 },
+ { 0x2A, 0x00 },
+ { 0x2B, 0x00 },
+ { 0x2C, 0x00 },
+ { 0x2D, 0x00 },
+ { 0x2E, 0x0B },
+ { 0x31, 0x00 },
+ { 0x32, 0x18 },
+ { 0x33, 0x00 },
+ { 0x34, 0x00 },
+ { 0x36, 0x00 },
+ { 0x37, 0x00 },
+ { 0x38, 0x00 },
+ { 0x39, 0x00 },
+ { 0x3A, 0x00 },
+ { 0x3B, 0x00 },
+ { 0x3C, 0x00 },
+ { 0x3D, 0x00 },
+ { 0x3E, 0x00 },
+ { 0x3F, 0x00 },
+ { 0x40, 0x00 },
+ { 0x41, 0x00 },
+ { 0x42, 0x00 },
+ { 0x43, 0x00 },
+ { 0x4A, 0x00 },
+ { 0x4B, 0x00 },
+ { 0x4C, 0x00 },
+ { 0x4D, 0x00 },
+ { 0x4E, 0x00 },
+ { 0x50, 0x00 },
+ { 0x51, 0x00 },
+ { 0x55, 0x00 },
+ { 0x58, 0x00 },
+ { 0x59, 0x00 },
+ { 0x5C, 0x00 },
+ { 0xFF, 0x43 },
+};
+
+static bool max98371_volatile_register(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case MAX98371_IRQ_CLEAR1:
+ case MAX98371_IRQ_CLEAR2:
+ case MAX98371_IRQ_CLEAR3:
+ case MAX98371_VERSION:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static bool max98371_readable_register(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case MAX98371_SOFT_RESET:
+ return false;
+ default:
+ return true;
+ }
+};
+
+static const DECLARE_TLV_DB_RANGE(max98371_gain_tlv,
+ 0, 7, TLV_DB_SCALE_ITEM(0, 50, 0),
+ 8, 10, TLV_DB_SCALE_ITEM(400, 100, 0)
+);
+
+static const DECLARE_TLV_DB_RANGE(max98371_noload_gain_tlv,
+ 0, 11, TLV_DB_SCALE_ITEM(950, 100, 0),
+);
+
+static const DECLARE_TLV_DB_SCALE(digital_tlv, -6300, 50, 1);
+
+static const struct snd_kcontrol_new max98371_snd_controls[] = {
+ SOC_SINGLE_TLV("Speaker Volume", MAX98371_GAIN,
+ MAX98371_GAIN_SHIFT, (1<<MAX98371_GAIN_WIDTH)-1, 0,
+ max98371_gain_tlv),
+ SOC_SINGLE_TLV("Digital Volume", MAX98371_DIGITAL_GAIN, 0,
+ (1<<MAX98371_DIGITAL_GAIN_WIDTH)-1, 1, digital_tlv),
+ SOC_SINGLE_TLV("Speaker DHT Max Volume", MAX98371_GAIN,
+ 0, (1<<MAX98371_DHT_MAX_WIDTH)-1, 0,
+ max98371_dht_max_gain),
+ SOC_SINGLE_TLV("Speaker DHT Min Volume", MAX98371_DHT_GAIN,
+ 0, (1<<MAX98371_DHT_GAIN_WIDTH)-1, 0,
+ max98371_dht_min_gain),
+ SOC_SINGLE_TLV("Speaker DHT Rotation Volume", MAX98371_DHT_GAIN,
+ 0, (1<<MAX98371_DHT_ROT_WIDTH)-1, 0,
+ max98371_dht_rot_gain),
+ SOC_SINGLE("DHT Attack Step", MAX98371_DHT, MAX98371_DHT_STEP, 3, 0),
+ SOC_SINGLE("DHT Attack Rate", MAX98371_DHT, 0, 7, 0),
+ SOC_ENUM("Monomix Select", max98371_monomix),
+ SOC_ENUM("HPF Cutoff", max98371_hpf_cutoff),
+};
+
+static int max98371_dai_set_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct max98371_priv *max98371 = snd_soc_codec_get_drvdata(codec);
+ unsigned int val = 0;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ default:
+ dev_err(codec->dev, "DAI clock mode unsupported");
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ val |= 0;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ val |= MAX98371_DAI_RIGHT;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ val |= MAX98371_DAI_LEFT;
+ break;
+ default:
+ dev_err(codec->dev, "DAI wrong mode unsupported");
+ return -EINVAL;
+ }
+ regmap_update_bits(max98371->regmap, MAX98371_FMT,
+ MAX98371_FMT_MODE_MASK, val);
+ return 0;
+}
+
+static int max98371_dai_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct max98371_priv *max98371 = snd_soc_codec_get_drvdata(codec);
+ int blr_clk_ratio, ch_size, channels = params_channels(params);
+ int rate = params_rate(params);
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S8:
+ regmap_update_bits(max98371->regmap, MAX98371_FMT,
+ MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_16);
+ ch_size = 8;
+ break;
+ case SNDRV_PCM_FORMAT_S16_LE:
+ regmap_update_bits(max98371->regmap, MAX98371_FMT,
+ MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_16);
+ ch_size = 16;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ regmap_update_bits(max98371->regmap, MAX98371_FMT,
+ MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_32);
+ ch_size = 24;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ regmap_update_bits(max98371->regmap, MAX98371_FMT,
+ MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_32);
+ ch_size = 32;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* BCLK/LRCLK ratio calculation */
+ blr_clk_ratio = channels * ch_size;
+ switch (blr_clk_ratio) {
+ case 32:
+ regmap_update_bits(max98371->regmap,
+ MAX98371_DAI_CLK,
+ MAX98371_DAI_BSEL_MASK, MAX98371_DAI_BSEL_32);
+ break;
+ case 48:
+ regmap_update_bits(max98371->regmap,
+ MAX98371_DAI_CLK,
+ MAX98371_DAI_BSEL_MASK, MAX98371_DAI_BSEL_48);
+ break;
+ case 64:
+ regmap_update_bits(max98371->regmap,
+ MAX98371_DAI_CLK,
+ MAX98371_DAI_BSEL_MASK, MAX98371_DAI_BSEL_64);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (rate) {
+ case 32000:
+ regmap_update_bits(max98371->regmap,
+ MAX98371_SPK_SR,
+ MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_32);
+ break;
+ case 44100:
+ regmap_update_bits(max98371->regmap,
+ MAX98371_SPK_SR,
+ MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_44);
+ break;
+ case 48000:
+ regmap_update_bits(max98371->regmap,
+ MAX98371_SPK_SR,
+ MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_48);
+ break;
+ case 88200:
+ regmap_update_bits(max98371->regmap,
+ MAX98371_SPK_SR,
+ MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_88);
+ break;
+ case 96000:
+ regmap_update_bits(max98371->regmap,
+ MAX98371_SPK_SR,
+ MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_96);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* enabling both the RX channels*/
+ regmap_update_bits(max98371->regmap, MAX98371_MONOMIX_SRC,
+ MAX98371_MONOMIX_SRC_MASK, MONOMIX_RX_0_1);
+ regmap_update_bits(max98371->regmap, MAX98371_DAI_CHANNEL,
+ MAX98371_CHANNEL_MASK, MAX98371_CHANNEL_MASK);
+ return 0;
+}
+
+static const struct snd_soc_dapm_widget max98371_dapm_widgets[] = {
+ SND_SOC_DAPM_DAC("DAC", NULL, MAX98371_SPK_ENABLE, 0, 0),
+ SND_SOC_DAPM_SUPPLY("Global Enable", MAX98371_GLOBAL_ENABLE,
+ 0, 0, NULL, 0),
+ SND_SOC_DAPM_OUTPUT("SPK_OUT"),
+};
+
+static const struct snd_soc_dapm_route max98371_audio_map[] = {
+ {"DAC", NULL, "HiFi Playback"},
+ {"SPK_OUT", NULL, "DAC"},
+ {"SPK_OUT", NULL, "Global Enable"},
+};
+
+#define MAX98371_RATES SNDRV_PCM_RATE_8000_48000
+#define MAX98371_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | \
+ SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE)
+
+static const struct snd_soc_dai_ops max98371_dai_ops = {
+ .set_fmt = max98371_dai_set_fmt,
+ .hw_params = max98371_dai_hw_params,
+};
+
+static struct snd_soc_dai_driver max98371_dai[] = {
+ {
+ .name = "max98371-aif1",
+ .playback = {
+ .stream_name = "HiFi Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = MAX98371_FORMATS,
+ },
+ .ops = &max98371_dai_ops,
+ }
+};
+
+static const struct snd_soc_codec_driver max98371_codec = {
+ .controls = max98371_snd_controls,
+ .num_controls = ARRAY_SIZE(max98371_snd_controls),
+ .dapm_routes = max98371_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(max98371_audio_map),
+ .dapm_widgets = max98371_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(max98371_dapm_widgets),
+};
+
+static const struct regmap_config max98371_regmap = {
+ .reg_bits = 8,
+ .val_bits = 8,
+ .max_register = MAX98371_VERSION,
+ .reg_defaults = max98371_reg,
+ .num_reg_defaults = ARRAY_SIZE(max98371_reg),
+ .volatile_reg = max98371_volatile_register,
+ .readable_reg = max98371_readable_register,
+ .cache_type = REGCACHE_RBTREE,
+};
+
+static int max98371_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct max98371_priv *max98371;
+ int ret, reg;
+
+ max98371 = devm_kzalloc(&i2c->dev,
+ sizeof(*max98371), GFP_KERNEL);
+ if (!max98371)
+ return -ENOMEM;
+
+ i2c_set_clientdata(i2c, max98371);
+ max98371->regmap = devm_regmap_init_i2c(i2c, &max98371_regmap);
+ if (IS_ERR(max98371->regmap)) {
+ ret = PTR_ERR(max98371->regmap);
+ dev_err(&i2c->dev,
+ "Failed to allocate regmap: %d\n", ret);
+ return ret;
+ }
+
+ ret = regmap_read(max98371->regmap, MAX98371_VERSION, &reg);
+ if (ret < 0) {
+ dev_info(&i2c->dev, "device error %d\n", ret);
+ return ret;
+ }
+ dev_info(&i2c->dev, "device version %x\n", reg);
+
+ ret = snd_soc_register_codec(&i2c->dev, &max98371_codec,
+ max98371_dai, ARRAY_SIZE(max98371_dai));
+ if (ret < 0) {
+ dev_err(&i2c->dev, "Failed to register codec: %d\n", ret);
+ return ret;
+ }
+ return ret;
+}
+
+static int max98371_i2c_remove(struct i2c_client *client)
+{
+ snd_soc_unregister_codec(&client->dev);
+ return 0;
+}
+
+static const struct i2c_device_id max98371_i2c_id[] = {
+ { "max98371", 0 },
+};
+
+MODULE_DEVICE_TABLE(i2c, max98371_i2c_id);
+
+static const struct of_device_id max98371_of_match[] = {
+ { .compatible = "maxim,max98371", },
+ { }
+};
+MODULE_DEVICE_TABLE(of, max98371_of_match);
+
+static struct i2c_driver max98371_i2c_driver = {
+ .driver = {
+ .name = "max98371",
+ .owner = THIS_MODULE,
+ .pm = NULL,
+ .of_match_table = of_match_ptr(max98371_of_match),
+ },
+ .probe = max98371_i2c_probe,
+ .remove = max98371_i2c_remove,
+ .id_table = max98371_i2c_id,
+};
+
+module_i2c_driver(max98371_i2c_driver);
+
+MODULE_AUTHOR("anish kumar <yesanishhere@gmail.com>");
+MODULE_DESCRIPTION("ALSA SoC MAX98371 driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/max98371.h b/sound/soc/codecs/max98371.h
new file mode 100644
index 000000000000..9f6330964d98
--- /dev/null
+++ b/sound/soc/codecs/max98371.h
@@ -0,0 +1,67 @@
+/*
+ * max98371.h -- MAX98371 ALSA SoC Audio driver
+ *
+ * Copyright 2011-2012 Maxim Integrated Products
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _MAX98371_H
+#define _MAX98371_H
+
+#define MAX98371_IRQ_CLEAR1 0x01
+#define MAX98371_IRQ_CLEAR2 0x02
+#define MAX98371_IRQ_CLEAR3 0x03
+#define MAX98371_DAI_CLK 0x10
+#define MAX98371_DAI_BSEL_MASK 0xF
+#define MAX98371_DAI_BSEL_32 2
+#define MAX98371_DAI_BSEL_48 3
+#define MAX98371_DAI_BSEL_64 4
+#define MAX98371_SPK_SR 0x11
+#define MAX98371_SPK_SR_MASK 0xF
+#define MAX98371_SPK_SR_32 6
+#define MAX98371_SPK_SR_44 7
+#define MAX98371_SPK_SR_48 8
+#define MAX98371_SPK_SR_88 10
+#define MAX98371_SPK_SR_96 11
+#define MAX98371_DAI_CHANNEL 0x15
+#define MAX98371_CHANNEL_MASK 0x3
+#define MAX98371_MONOMIX_SRC 0x18
+#define MAX98371_MONOMIX_CFG 0x19
+#define MAX98371_HPF 0x1C
+#define MAX98371_MONOMIX_SRC_MASK 0xFF
+#define MONOMIX_RX_0_1 ((0x1)<<(4))
+#define M98371_DAI_CHANNEL_I2S 0x3
+#define MAX98371_DIGITAL_GAIN 0x2D
+#define MAX98371_DIGITAL_GAIN_WIDTH 0x7
+#define MAX98371_GAIN 0x2E
+#define MAX98371_GAIN_SHIFT 0x4
+#define MAX98371_GAIN_WIDTH 0x4
+#define MAX98371_DHT_MAX_WIDTH 4
+#define MAX98371_FMT 0x14
+#define MAX98371_CHANSZ_WIDTH 6
+#define MAX98371_FMT_MASK ((0x3)<<(MAX98371_CHANSZ_WIDTH))
+#define MAX98371_FMT_MODE_MASK ((0x7)<<(3))
+#define MAX98371_DAI_LEFT ((0x1)<<(3))
+#define MAX98371_DAI_RIGHT ((0x2)<<(3))
+#define MAX98371_DAI_CHANSZ_16 ((1)<<(MAX98371_CHANSZ_WIDTH))
+#define MAX98371_DAI_CHANSZ_24 ((2)<<(MAX98371_CHANSZ_WIDTH))
+#define MAX98371_DAI_CHANSZ_32 ((3)<<(MAX98371_CHANSZ_WIDTH))
+#define MAX98371_DHT 0x32
+#define MAX98371_DHT_STEP 0x3
+#define MAX98371_DHT_GAIN 0x31
+#define MAX98371_DHT_GAIN_WIDTH 0x4
+#define MAX98371_DHT_ROT_WIDTH 0x4
+#define MAX98371_SPK_ENABLE 0x4A
+#define MAX98371_GLOBAL_ENABLE 0x50
+#define MAX98371_SOFT_RESET 0x51
+#define MAX98371_VERSION 0xFF
+
+
+struct max98371_priv {
+ struct regmap *regmap;
+ struct snd_soc_codec *codec;
+};
+#endif
diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c
index 1c8729984c2b..683769f0f246 100644
--- a/sound/soc/codecs/nau8825.c
+++ b/sound/soc/codecs/nau8825.c
@@ -343,9 +343,12 @@ static const struct snd_soc_dapm_widget nau8825_dapm_widgets[] = {
SND_SOC_DAPM_SUPPLY("ADC Power", NAU8825_REG_ANALOG_ADC_2, 6, 0, NULL,
0),
- /* ADC for button press detection */
- SND_SOC_DAPM_ADC("SAR", NULL, NAU8825_REG_SAR_CTRL,
- NAU8825_SAR_ADC_EN_SFT, 0),
+ /* ADC for button press detection. A dapm supply widget is used to
+ * prevent dapm_power_widgets keeping the codec at SND_SOC_BIAS_ON
+ * during suspend.
+ */
+ SND_SOC_DAPM_SUPPLY("SAR", NAU8825_REG_SAR_CTRL,
+ NAU8825_SAR_ADC_EN_SFT, 0, NULL, 0),
SND_SOC_DAPM_PGA_S("ADACL", 2, NAU8825_REG_RDAC, 12, 0, NULL, 0),
SND_SOC_DAPM_PGA_S("ADACR", 2, NAU8825_REG_RDAC, 13, 0, NULL, 0),
@@ -607,6 +610,16 @@ static bool nau8825_is_jack_inserted(struct regmap *regmap)
static void nau8825_restart_jack_detection(struct regmap *regmap)
{
+ /* Chip needs one FSCLK cycle in order to generate interrupts,
+ * as we cannot guarantee one will be provided by the system. Turning
+ * master mode on then off enables us to generate that FSCLK cycle
+ * with a minimum of contention on the clock bus.
+ */
+ regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2,
+ NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_MASTER);
+ regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2,
+ NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_SLAVE);
+
/* this will restart the entire jack detection process including MIC/GND
* switching and create interrupts. We have to go from 0 to 1 and back
* to 0 to restart.
@@ -728,7 +741,10 @@ static irqreturn_t nau8825_interrupt(int irq, void *data)
struct regmap *regmap = nau8825->regmap;
int active_irq, clear_irq = 0, event = 0, event_mask = 0;
- regmap_read(regmap, NAU8825_REG_IRQ_STATUS, &active_irq);
+ if (regmap_read(regmap, NAU8825_REG_IRQ_STATUS, &active_irq)) {
+ dev_err(nau8825->dev, "failed to read irq status\n");
+ return IRQ_NONE;
+ }
if ((active_irq & NAU8825_JACK_EJECTION_IRQ_MASK) ==
NAU8825_JACK_EJECTION_DETECTED) {
@@ -1141,33 +1157,74 @@ static int nau8825_set_bias_level(struct snd_soc_codec *codec,
return ret;
}
}
-
- ret = regcache_sync(nau8825->regmap);
- if (ret) {
- dev_err(codec->dev,
- "Failed to sync cache: %d\n", ret);
- return ret;
- }
}
-
break;
case SND_SOC_BIAS_OFF:
if (nau8825->mclk_freq)
clk_disable_unprepare(nau8825->mclk);
-
- regcache_mark_dirty(nau8825->regmap);
break;
}
return 0;
}
+#ifdef CONFIG_PM
+static int nau8825_suspend(struct snd_soc_codec *codec)
+{
+ struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec);
+
+ disable_irq(nau8825->irq);
+ regcache_cache_only(nau8825->regmap, true);
+ regcache_mark_dirty(nau8825->regmap);
+
+ return 0;
+}
+
+static int nau8825_resume(struct snd_soc_codec *codec)
+{
+ struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec);
+
+ /* The chip may lose power and reset in S3. regcache_sync restores
+ * register values including configurations for sysclk, irq, and
+ * jack/button detection.
+ */
+ regcache_cache_only(nau8825->regmap, false);
+ regcache_sync(nau8825->regmap);
+
+ /* Check the jack plug status directly. If the headset is unplugged
+ * during S3 when the chip has no power, there will be no jack
+ * detection irq even after the nau8825_restart_jack_detection below,
+ * because the chip just thinks no headset has ever been plugged in.
+ */
+ if (!nau8825_is_jack_inserted(nau8825->regmap)) {
+ nau8825_eject_jack(nau8825);
+ snd_soc_jack_report(nau8825->jack, 0, SND_JACK_HEADSET);
+ }
+
+ enable_irq(nau8825->irq);
+
+ /* Run jack detection to check the type (OMTP or CTIA) of the headset
+ * if there is one. This handles the case where a different type of
+ * headset is plugged in during S3. This triggers an IRQ iff a headset
+ * is already plugged in.
+ */
+ nau8825_restart_jack_detection(nau8825->regmap);
+
+ return 0;
+}
+#else
+#define nau8825_suspend NULL
+#define nau8825_resume NULL
+#endif
+
static struct snd_soc_codec_driver nau8825_codec_driver = {
.probe = nau8825_codec_probe,
.set_sysclk = nau8825_set_sysclk,
.set_pll = nau8825_set_pll,
.set_bias_level = nau8825_set_bias_level,
.suspend_bias_off = true,
+ .suspend = nau8825_suspend,
+ .resume = nau8825_resume,
.controls = nau8825_controls,
.num_controls = ARRAY_SIZE(nau8825_controls),
@@ -1277,16 +1334,6 @@ static int nau8825_setup_irq(struct nau8825 *nau8825)
regmap_update_bits(regmap, NAU8825_REG_ENA_CTRL,
NAU8825_ENABLE_DACR, NAU8825_ENABLE_DACR);
- /* Chip needs one FSCLK cycle in order to generate interrupts,
- * as we cannot guarantee one will be provided by the system. Turning
- * master mode on then off enables us to generate that FSCLK cycle
- * with a minimum of contention on the clock bus.
- */
- regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2,
- NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_MASTER);
- regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2,
- NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_SLAVE);
-
ret = devm_request_threaded_irq(nau8825->dev, nau8825->irq, NULL,
nau8825_interrupt, IRQF_TRIGGER_LOW | IRQF_ONESHOT,
"nau8825", nau8825);
@@ -1354,36 +1401,6 @@ static int nau8825_i2c_remove(struct i2c_client *client)
return 0;
}
-#ifdef CONFIG_PM_SLEEP
-static int nau8825_suspend(struct device *dev)
-{
- struct i2c_client *client = to_i2c_client(dev);
- struct nau8825 *nau8825 = dev_get_drvdata(dev);
-
- disable_irq(client->irq);
- regcache_cache_only(nau8825->regmap, true);
- regcache_mark_dirty(nau8825->regmap);
-
- return 0;
-}
-
-static int nau8825_resume(struct device *dev)
-{
- struct i2c_client *client = to_i2c_client(dev);
- struct nau8825 *nau8825 = dev_get_drvdata(dev);
-
- regcache_cache_only(nau8825->regmap, false);
- regcache_sync(nau8825->regmap);
- enable_irq(client->irq);
-
- return 0;
-}
-#endif
-
-static const struct dev_pm_ops nau8825_pm = {
- SET_SYSTEM_SLEEP_PM_OPS(nau8825_suspend, nau8825_resume)
-};
-
static const struct i2c_device_id nau8825_i2c_ids[] = {
{ "nau8825", 0 },
{ }
@@ -1410,7 +1427,6 @@ static struct i2c_driver nau8825_driver = {
.name = "nau8825",
.of_match_table = of_match_ptr(nau8825_of_ids),
.acpi_match_table = ACPI_PTR(nau8825_acpi_match),
- .pm = &nau8825_pm,
},
.probe = nau8825_i2c_probe,
.remove = nau8825_i2c_remove,
diff --git a/sound/soc/codecs/pcm5102a.c b/sound/soc/codecs/pcm5102a.c
new file mode 100644
index 000000000000..ed515677409b
--- /dev/null
+++ b/sound/soc/codecs/pcm5102a.c
@@ -0,0 +1,69 @@
+/*
+ * Driver for the PCM5102A codec
+ *
+ * Author: Florian Meier <florian.meier@koalo.de>
+ * Copyright 2013
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+
+#include <sound/soc.h>
+
+static struct snd_soc_dai_driver pcm5102a_dai = {
+ .name = "pcm5102a-hifi",
+ .playback = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE |
+ SNDRV_PCM_FMTBIT_S32_LE
+ },
+};
+
+static struct snd_soc_codec_driver soc_codec_dev_pcm5102a;
+
+static int pcm5102a_probe(struct platform_device *pdev)
+{
+ return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_pcm5102a,
+ &pcm5102a_dai, 1);
+}
+
+static int pcm5102a_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_codec(&pdev->dev);
+ return 0;
+}
+
+static const struct of_device_id pcm5102a_of_match[] = {
+ { .compatible = "ti,pcm5102a", },
+ { }
+};
+MODULE_DEVICE_TABLE(of, pcm5102a_of_match);
+
+static struct platform_driver pcm5102a_codec_driver = {
+ .probe = pcm5102a_probe,
+ .remove = pcm5102a_remove,
+ .driver = {
+ .name = "pcm5102a-codec",
+ .owner = THIS_MODULE,
+ .of_match_table = pcm5102a_of_match,
+ },
+};
+
+module_platform_driver(pcm5102a_codec_driver);
+
+MODULE_DESCRIPTION("ASoC PCM5102A codec driver");
+MODULE_AUTHOR("Florian Meier <florian.meier@koalo.de>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c
index f0e6c06e89ac..f80cfe4d2ef2 100644
--- a/sound/soc/codecs/rt298.c
+++ b/sound/soc/codecs/rt298.c
@@ -17,6 +17,7 @@
#include <linux/i2c.h>
#include <linux/platform_device.h>
#include <linux/spi/spi.h>
+#include <linux/dmi.h>
#include <linux/acpi.h>
#include <sound/core.h>
#include <sound/pcm.h>
@@ -275,6 +276,8 @@ static int rt298_jack_detect(struct rt298_priv *rt298, bool *hp, bool *mic)
} else {
*mic = false;
regmap_write(rt298->regmap, RT298_SET_MIC1, 0x20);
+ regmap_update_bits(rt298->regmap,
+ RT298_CBJ_CTRL1, 0x0400, 0x0000);
}
} else {
regmap_read(rt298->regmap, RT298_GET_HP_SENSE, &buf);
@@ -481,6 +484,26 @@ static int rt298_adc_event(struct snd_soc_dapm_widget *w,
snd_soc_update_bits(codec,
VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, nid, 0),
0x7080, 0x7000);
+ /* If MCLK doesn't exist, reset AD filter */
+ if (!(snd_soc_read(codec, RT298_VAD_CTRL) & 0x200)) {
+ pr_info("NO MCLK\n");
+ switch (nid) {
+ case RT298_ADC_IN1:
+ snd_soc_update_bits(codec,
+ RT298_D_FILTER_CTRL, 0x2, 0x2);
+ mdelay(10);
+ snd_soc_update_bits(codec,
+ RT298_D_FILTER_CTRL, 0x2, 0x0);
+ break;
+ case RT298_ADC_IN2:
+ snd_soc_update_bits(codec,
+ RT298_D_FILTER_CTRL, 0x4, 0x4);
+ mdelay(10);
+ snd_soc_update_bits(codec,
+ RT298_D_FILTER_CTRL, 0x4, 0x0);
+ break;
+ }
+ }
break;
case SND_SOC_DAPM_PRE_PMD:
snd_soc_update_bits(codec,
@@ -519,30 +542,12 @@ static int rt298_mic1_event(struct snd_soc_dapm_widget *w,
return 0;
}
-static int rt298_vref_event(struct snd_soc_dapm_widget *w,
- struct snd_kcontrol *kcontrol, int event)
-{
- struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
-
- switch (event) {
- case SND_SOC_DAPM_PRE_PMU:
- snd_soc_update_bits(codec,
- RT298_CBJ_CTRL1, 0x0400, 0x0000);
- mdelay(50);
- break;
- default:
- return 0;
- }
-
- return 0;
-}
-
static const struct snd_soc_dapm_widget rt298_dapm_widgets[] = {
SND_SOC_DAPM_SUPPLY_S("HV", 1, RT298_POWER_CTRL1,
12, 1, NULL, 0),
SND_SOC_DAPM_SUPPLY("VREF", RT298_POWER_CTRL1,
- 0, 1, rt298_vref_event, SND_SOC_DAPM_PRE_PMU),
+ 0, 1, NULL, 0),
SND_SOC_DAPM_SUPPLY_S("BG_MBIAS", 1, RT298_POWER_CTRL2,
1, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY_S("LDO1", 1, RT298_POWER_CTRL2,
@@ -933,18 +938,9 @@ static int rt298_set_bias_level(struct snd_soc_codec *codec,
}
break;
- case SND_SOC_BIAS_ON:
- mdelay(30);
- snd_soc_update_bits(codec,
- RT298_CBJ_CTRL1, 0x0400, 0x0400);
-
- break;
-
case SND_SOC_BIAS_STANDBY:
snd_soc_write(codec,
RT298_SET_AUDIO_POWER, AC_PWRST_D3);
- snd_soc_update_bits(codec,
- RT298_CBJ_CTRL1, 0x0400, 0x0000);
break;
default:
@@ -1132,6 +1128,17 @@ static const struct acpi_device_id rt298_acpi_match[] = {
};
MODULE_DEVICE_TABLE(acpi, rt298_acpi_match);
+static const struct dmi_system_id force_combo_jack_table[] = {
+ {
+ .ident = "Intel Broxton P",
+ .matches = {
+ DMI_MATCH(DMI_SYS_VENDOR, "Intel Corp"),
+ DMI_MATCH(DMI_PRODUCT_NAME, "Broxton P")
+ }
+ },
+ { }
+};
+
static int rt298_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
@@ -1184,11 +1191,16 @@ static int rt298_i2c_probe(struct i2c_client *i2c,
/* enable jack combo mode on supported devices */
acpiid = acpi_match_device(dev->driver->acpi_match_table, dev);
- if (acpiid) {
+ if (acpiid && acpiid->driver_data) {
rt298->pdata = *(struct rt298_platform_data *)
acpiid->driver_data;
}
+ if (dmi_check_system(force_combo_jack_table)) {
+ rt298->pdata.cbj_en = true;
+ rt298->pdata.gpio2_en = false;
+ }
+
/* VREF Charging */
regmap_update_bits(rt298->regmap, 0x04, 0x80, 0x80);
regmap_update_bits(rt298->regmap, 0x1b, 0x860, 0x860);
diff --git a/sound/soc/codecs/rt298.h b/sound/soc/codecs/rt298.h
index d66f8847b676..3638f3d61209 100644
--- a/sound/soc/codecs/rt298.h
+++ b/sound/soc/codecs/rt298.h
@@ -137,6 +137,7 @@
#define RT298_A_BIAS_CTRL2 0x02
#define RT298_POWER_CTRL1 0x03
#define RT298_A_BIAS_CTRL3 0x04
+#define RT298_D_FILTER_CTRL 0x05
#define RT298_POWER_CTRL2 0x08
#define RT298_I2S_CTRL1 0x09
#define RT298_I2S_CTRL2 0x0a
@@ -148,6 +149,7 @@
#define RT298_IRQ_CTRL 0x33
#define RT298_WIND_FILTER_CTRL 0x46
#define RT298_PLL_CTRL1 0x49
+#define RT298_VAD_CTRL 0x4e
#define RT298_CBJ_CTRL1 0x4f
#define RT298_CBJ_CTRL2 0x50
#define RT298_PLL_CTRL 0x63
diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index e8b5ba04417a..09e8988bbb2d 100644
--- a/sound/soc/codecs/rt5640.c
+++ b/sound/soc/codecs/rt5640.c
@@ -359,7 +359,7 @@ static const DECLARE_TLV_DB_RANGE(bst_tlv,
/* Interface data select */
static const char * const rt5640_data_select[] = {
- "Normal", "left copy to right", "right copy to left", "Swap"};
+ "Normal", "Swap", "left copy to right", "right copy to left"};
static SOC_ENUM_SINGLE_DECL(rt5640_if1_dac_enum, RT5640_DIG_INF_DATA,
RT5640_IF1_DAC_SEL_SFT, rt5640_data_select);
diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h
index 1761c3a98b76..58b664b06c16 100644
--- a/sound/soc/codecs/rt5640.h
+++ b/sound/soc/codecs/rt5640.h
@@ -443,39 +443,39 @@
#define RT5640_IF1_DAC_SEL_MASK (0x3 << 14)
#define RT5640_IF1_DAC_SEL_SFT 14
#define RT5640_IF1_DAC_SEL_NOR (0x0 << 14)
-#define RT5640_IF1_DAC_SEL_L2R (0x1 << 14)
-#define RT5640_IF1_DAC_SEL_R2L (0x2 << 14)
-#define RT5640_IF1_DAC_SEL_SWAP (0x3 << 14)
+#define RT5640_IF1_DAC_SEL_SWAP (0x1 << 14)
+#define RT5640_IF1_DAC_SEL_L2R (0x2 << 14)
+#define RT5640_IF1_DAC_SEL_R2L (0x3 << 14)
#define RT5640_IF1_ADC_SEL_MASK (0x3 << 12)
#define RT5640_IF1_ADC_SEL_SFT 12
#define RT5640_IF1_ADC_SEL_NOR (0x0 << 12)
-#define RT5640_IF1_ADC_SEL_L2R (0x1 << 12)
-#define RT5640_IF1_ADC_SEL_R2L (0x2 << 12)
-#define RT5640_IF1_ADC_SEL_SWAP (0x3 << 12)
+#define RT5640_IF1_ADC_SEL_SWAP (0x1 << 12)
+#define RT5640_IF1_ADC_SEL_L2R (0x2 << 12)
+#define RT5640_IF1_ADC_SEL_R2L (0x3 << 12)
#define RT5640_IF2_DAC_SEL_MASK (0x3 << 10)
#define RT5640_IF2_DAC_SEL_SFT 10
#define RT5640_IF2_DAC_SEL_NOR (0x0 << 10)
-#define RT5640_IF2_DAC_SEL_L2R (0x1 << 10)
-#define RT5640_IF2_DAC_SEL_R2L (0x2 << 10)
-#define RT5640_IF2_DAC_SEL_SWAP (0x3 << 10)
+#define RT5640_IF2_DAC_SEL_SWAP (0x1 << 10)
+#define RT5640_IF2_DAC_SEL_L2R (0x2 << 10)
+#define RT5640_IF2_DAC_SEL_R2L (0x3 << 10)
#define RT5640_IF2_ADC_SEL_MASK (0x3 << 8)
#define RT5640_IF2_ADC_SEL_SFT 8
#define RT5640_IF2_ADC_SEL_NOR (0x0 << 8)
-#define RT5640_IF2_ADC_SEL_L2R (0x1 << 8)
-#define RT5640_IF2_ADC_SEL_R2L (0x2 << 8)
-#define RT5640_IF2_ADC_SEL_SWAP (0x3 << 8)
+#define RT5640_IF2_ADC_SEL_SWAP (0x1 << 8)
+#define RT5640_IF2_ADC_SEL_L2R (0x2 << 8)
+#define RT5640_IF2_ADC_SEL_R2L (0x3 << 8)
#define RT5640_IF3_DAC_SEL_MASK (0x3 << 6)
#define RT5640_IF3_DAC_SEL_SFT 6
#define RT5640_IF3_DAC_SEL_NOR (0x0 << 6)
-#define RT5640_IF3_DAC_SEL_L2R (0x1 << 6)
-#define RT5640_IF3_DAC_SEL_R2L (0x2 << 6)
-#define RT5640_IF3_DAC_SEL_SWAP (0x3 << 6)
+#define RT5640_IF3_DAC_SEL_SWAP (0x1 << 6)
+#define RT5640_IF3_DAC_SEL_L2R (0x2 << 6)
+#define RT5640_IF3_DAC_SEL_R2L (0x3 << 6)
#define RT5640_IF3_ADC_SEL_MASK (0x3 << 4)
#define RT5640_IF3_ADC_SEL_SFT 4
#define RT5640_IF3_ADC_SEL_NOR (0x0 << 4)
-#define RT5640_IF3_ADC_SEL_L2R (0x1 << 4)
-#define RT5640_IF3_ADC_SEL_R2L (0x2 << 4)
-#define RT5640_IF3_ADC_SEL_SWAP (0x3 << 4)
+#define RT5640_IF3_ADC_SEL_SWAP (0x1 << 4)
+#define RT5640_IF3_ADC_SEL_L2R (0x2 << 4)
+#define RT5640_IF3_ADC_SEL_R2L (0x3 << 4)
/* REC Left Mixer Control 1 (0x3b) */
#define RT5640_G_HP_L_RM_L_MASK (0x7 << 13)
diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index 7af5e7380d61..3c6594da6c9c 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -3286,10 +3286,8 @@ static void rt5645_jack_detect_work(struct work_struct *work)
if (btn_type == 0)/* button release */
report = rt5645->jack_type;
else {
- if (rt5645->pdata.jd_invert) {
- mod_timer(&rt5645->btn_check_timer,
- msecs_to_jiffies(100));
- }
+ mod_timer(&rt5645->btn_check_timer,
+ msecs_to_jiffies(100));
}
break;
@@ -3557,6 +3555,12 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = {
DMI_MATCH(DMI_SYS_VENDOR, "GOOGLE"),
},
},
+ {
+ .ident = "Google Setzer",
+ .matches = {
+ DMI_MATCH(DMI_PRODUCT_NAME, "Setzer"),
+ },
+ },
{ }
};
@@ -3810,9 +3814,9 @@ static int rt5645_i2c_probe(struct i2c_client *i2c,
if (rt5645->pdata.jd_invert) {
regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2,
RT5645_JD_1_1_MASK, RT5645_JD_1_1_INV);
- setup_timer(&rt5645->btn_check_timer,
- rt5645_btn_check_callback, (unsigned long)rt5645);
}
+ setup_timer(&rt5645->btn_check_timer,
+ rt5645_btn_check_callback, (unsigned long)rt5645);
INIT_DELAYED_WORK(&rt5645->jack_detect_work, rt5645_jack_detect_work);
INIT_DELAYED_WORK(&rt5645->rcclock_work, rt5645_rcclock_work);
diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c
index 39307ad41a34..b8d19b77bde9 100644
--- a/sound/soc/codecs/tas571x.c
+++ b/sound/soc/codecs/tas571x.c
@@ -4,6 +4,9 @@
* Copyright (C) 2015 Google, Inc.
* Copyright (c) 2013 Daniel Mack <zonque@gmail.com>
*
+ * TAS5721 support:
+ * Copyright (C) 2016 Petr Kulhavy, Barix AG <petr@barix.com>
+ *
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
@@ -57,6 +60,10 @@ static int tas571x_register_size(struct tas571x_private *priv, unsigned int reg)
case TAS571X_CH1_VOL_REG:
case TAS571X_CH2_VOL_REG:
return priv->chip->vol_reg_size;
+ case TAS571X_INPUT_MUX_REG:
+ case TAS571X_CH4_SRC_SELECT_REG:
+ case TAS571X_PWM_MUX_REG:
+ return 4;
default:
return 1;
}
@@ -167,6 +174,23 @@ static int tas571x_hw_params(struct snd_pcm_substream *substream,
TAS571X_SDI_FMT_MASK, val);
}
+static int tas571x_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u8 sysctl2;
+ int ret;
+
+ sysctl2 = mute ? TAS571X_SYS_CTRL_2_SDN_MASK : 0;
+
+ ret = snd_soc_update_bits(codec,
+ TAS571X_SYS_CTRL_2_REG,
+ TAS571X_SYS_CTRL_2_SDN_MASK,
+ sysctl2);
+ usleep_range(1000, 2000);
+
+ return ret;
+}
+
static int tas571x_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
@@ -214,6 +238,7 @@ static int tas571x_set_bias_level(struct snd_soc_codec *codec,
static const struct snd_soc_dai_ops tas571x_dai_ops = {
.set_fmt = tas571x_set_dai_fmt,
.hw_params = tas571x_hw_params,
+ .digital_mute = tas571x_mute,
};
static const char *const tas5711_supply_names[] = {
@@ -241,6 +266,26 @@ static const struct snd_kcontrol_new tas5711_controls[] = {
1, 1),
};
+static const struct regmap_range tas571x_readonly_regs_range[] = {
+ regmap_reg_range(TAS571X_CLK_CTRL_REG, TAS571X_DEV_ID_REG),
+};
+
+static const struct regmap_range tas571x_volatile_regs_range[] = {
+ regmap_reg_range(TAS571X_CLK_CTRL_REG, TAS571X_ERR_STATUS_REG),
+ regmap_reg_range(TAS571X_OSC_TRIM_REG, TAS571X_OSC_TRIM_REG),
+};
+
+static const struct regmap_access_table tas571x_write_regs = {
+ .no_ranges = tas571x_readonly_regs_range,
+ .n_no_ranges = ARRAY_SIZE(tas571x_readonly_regs_range),
+};
+
+static const struct regmap_access_table tas571x_volatile_regs = {
+ .yes_ranges = tas571x_volatile_regs_range,
+ .n_yes_ranges = ARRAY_SIZE(tas571x_volatile_regs_range),
+
+};
+
static const struct reg_default tas5711_reg_defaults[] = {
{ 0x04, 0x05 },
{ 0x05, 0x40 },
@@ -260,6 +305,8 @@ static const struct regmap_config tas5711_regmap_config = {
.reg_defaults = tas5711_reg_defaults,
.num_reg_defaults = ARRAY_SIZE(tas5711_reg_defaults),
.cache_type = REGCACHE_RBTREE,
+ .wr_table = &tas571x_write_regs,
+ .volatile_table = &tas571x_volatile_regs,
};
static const struct tas571x_chip tas5711_chip = {
@@ -314,6 +361,8 @@ static const struct regmap_config tas5717_regmap_config = {
.reg_defaults = tas5717_reg_defaults,
.num_reg_defaults = ARRAY_SIZE(tas5717_reg_defaults),
.cache_type = REGCACHE_RBTREE,
+ .wr_table = &tas571x_write_regs,
+ .volatile_table = &tas571x_volatile_regs,
};
/* This entry is reused for tas5719 as the software interface is identical. */
@@ -326,6 +375,77 @@ static const struct tas571x_chip tas5717_chip = {
.vol_reg_size = 2,
};
+static const char *const tas5721_supply_names[] = {
+ "AVDD",
+ "DVDD",
+ "DRVDD",
+ "PVDD",
+};
+
+static const struct snd_kcontrol_new tas5721_controls[] = {
+ SOC_SINGLE_TLV("Master Volume",
+ TAS571X_MVOL_REG,
+ 0, 0xff, 1, tas5711_volume_tlv),
+ SOC_DOUBLE_R_TLV("Speaker Volume",
+ TAS571X_CH1_VOL_REG,
+ TAS571X_CH2_VOL_REG,
+ 0, 0xff, 1, tas5711_volume_tlv),
+ SOC_DOUBLE("Speaker Switch",
+ TAS571X_SOFT_MUTE_REG,
+ TAS571X_SOFT_MUTE_CH1_SHIFT, TAS571X_SOFT_MUTE_CH2_SHIFT,
+ 1, 1),
+};
+
+static const struct reg_default tas5721_reg_defaults[] = {
+ {TAS571X_CLK_CTRL_REG, 0x6c},
+ {TAS571X_DEV_ID_REG, 0x00},
+ {TAS571X_ERR_STATUS_REG, 0x00},
+ {TAS571X_SYS_CTRL_1_REG, 0xa0},
+ {TAS571X_SDI_REG, 0x05},
+ {TAS571X_SYS_CTRL_2_REG, 0x40},
+ {TAS571X_SOFT_MUTE_REG, 0x00},
+ {TAS571X_MVOL_REG, 0xff},
+ {TAS571X_CH1_VOL_REG, 0x30},
+ {TAS571X_CH2_VOL_REG, 0x30},
+ {TAS571X_CH3_VOL_REG, 0x30},
+ {TAS571X_VOL_CFG_REG, 0x91},
+ {TAS571X_MODULATION_LIMIT_REG, 0x02},
+ {TAS571X_IC_DELAY_CH1_REG, 0xac},
+ {TAS571X_IC_DELAY_CH2_REG, 0x54},
+ {TAS571X_IC_DELAY_CH3_REG, 0xac},
+ {TAS571X_IC_DELAY_CH4_REG, 0x54},
+ {TAS571X_PWM_CH_SDN_GROUP_REG, 0x30},
+ {TAS571X_START_STOP_PERIOD_REG, 0x0f},
+ {TAS571X_OSC_TRIM_REG, 0x82},
+ {TAS571X_BKND_ERR_REG, 0x02},
+ {TAS571X_INPUT_MUX_REG, 0x17772},
+ {TAS571X_CH4_SRC_SELECT_REG, 0x4303},
+ {TAS571X_PWM_MUX_REG, 0x1021345},
+};
+
+static const struct regmap_config tas5721_regmap_config = {
+ .reg_bits = 8,
+ .val_bits = 32,
+ .max_register = 0xff,
+ .reg_read = tas571x_reg_read,
+ .reg_write = tas571x_reg_write,
+ .reg_defaults = tas5721_reg_defaults,
+ .num_reg_defaults = ARRAY_SIZE(tas5721_reg_defaults),
+ .cache_type = REGCACHE_RBTREE,
+ .wr_table = &tas571x_write_regs,
+ .volatile_table = &tas571x_volatile_regs,
+};
+
+
+static const struct tas571x_chip tas5721_chip = {
+ .supply_names = tas5721_supply_names,
+ .num_supply_names = ARRAY_SIZE(tas5721_supply_names),
+ .controls = tas5711_controls,
+ .num_controls = ARRAY_SIZE(tas5711_controls),
+ .regmap_config = &tas5721_regmap_config,
+ .vol_reg_size = 1,
+};
+
static const struct snd_soc_dapm_widget tas571x_dapm_widgets[] = {
SND_SOC_DAPM_DAC("DACL", NULL, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_DAC("DACR", NULL, SND_SOC_NOPM, 0, 0),
@@ -386,11 +506,10 @@ static int tas571x_i2c_probe(struct i2c_client *client,
i2c_set_clientdata(client, priv);
of_id = of_match_device(tas571x_of_match, dev);
- if (!of_id) {
- dev_err(dev, "Unknown device type\n");
- return -EINVAL;
- }
- priv->chip = of_id->data;
+ if (of_id)
+ priv->chip = of_id->data;
+ else
+ priv->chip = (void *) id->driver_data;
priv->mclk = devm_clk_get(dev, "mclk");
if (IS_ERR(priv->mclk) && PTR_ERR(priv->mclk) != -ENOENT) {
@@ -445,10 +564,6 @@ static int tas571x_i2c_probe(struct i2c_client *client,
if (ret)
return ret;
- ret = regmap_update_bits(priv->regmap, TAS571X_SYS_CTRL_2_REG,
- TAS571X_SYS_CTRL_2_SDN_MASK, 0);
- if (ret)
- return ret;
memcpy(&priv->codec_driver, &tas571x_codec, sizeof(priv->codec_driver));
priv->codec_driver.controls = priv->chip->controls;
@@ -486,14 +601,16 @@ static const struct of_device_id tas571x_of_match[] = {
{ .compatible = "ti,tas5711", .data = &tas5711_chip, },
{ .compatible = "ti,tas5717", .data = &tas5717_chip, },
{ .compatible = "ti,tas5719", .data = &tas5717_chip, },
+ { .compatible = "ti,tas5721", .data = &tas5721_chip, },
{ }
};
MODULE_DEVICE_TABLE(of, tas571x_of_match);
static const struct i2c_device_id tas571x_i2c_id[] = {
- { "tas5711", 0 },
- { "tas5717", 0 },
- { "tas5719", 0 },
+ { "tas5711", (kernel_ulong_t) &tas5711_chip },
+ { "tas5717", (kernel_ulong_t) &tas5717_chip },
+ { "tas5719", (kernel_ulong_t) &tas5717_chip },
+ { "tas5721", (kernel_ulong_t) &tas5721_chip },
{ }
};
MODULE_DEVICE_TABLE(i2c, tas571x_i2c_id);
diff --git a/sound/soc/codecs/tas571x.h b/sound/soc/codecs/tas571x.h
index 0aee471232cd..cf800c364f0f 100644
--- a/sound/soc/codecs/tas571x.h
+++ b/sound/soc/codecs/tas571x.h
@@ -13,6 +13,10 @@
#define _TAS571X_H
/* device registers */
+#define TAS571X_CLK_CTRL_REG 0x00
+#define TAS571X_DEV_ID_REG 0x01
+#define TAS571X_ERR_STATUS_REG 0x02
+#define TAS571X_SYS_CTRL_1_REG 0x03
#define TAS571X_SDI_REG 0x04
#define TAS571X_SDI_FMT_MASK 0x0f
@@ -27,7 +31,25 @@
#define TAS571X_MVOL_REG 0x07
#define TAS571X_CH1_VOL_REG 0x08
#define TAS571X_CH2_VOL_REG 0x09
+#define TAS571X_CH3_VOL_REG 0x0a
+#define TAS571X_VOL_CFG_REG 0x0e
+#define TAS571X_MODULATION_LIMIT_REG 0x10
+#define TAS571X_IC_DELAY_CH1_REG 0x11
+#define TAS571X_IC_DELAY_CH2_REG 0x12
+#define TAS571X_IC_DELAY_CH3_REG 0x13
+#define TAS571X_IC_DELAY_CH4_REG 0x14
+#define TAS571X_PWM_CH_SDN_GROUP_REG 0x19 /* N/A on TAS5717, TAS5719 */
+#define TAS571X_PWM_CH1_SDN_MASK (1<<0)
+#define TAS571X_PWM_CH2_SDN_SHIFT (1<<1)
+#define TAS571X_PWM_CH3_SDN_SHIFT (1<<2)
+#define TAS571X_PWM_CH4_SDN_SHIFT (1<<3)
+
+#define TAS571X_START_STOP_PERIOD_REG 0x1a
#define TAS571X_OSC_TRIM_REG 0x1b
+#define TAS571X_BKND_ERR_REG 0x1c
+#define TAS571X_INPUT_MUX_REG 0x20
+#define TAS571X_CH4_SRC_SELECT_REG 0x21
+#define TAS571X_PWM_MUX_REG 0x25
#endif /* _TAS571X_H */
diff --git a/sound/soc/codecs/tas5720.c b/sound/soc/codecs/tas5720.c
new file mode 100644
index 000000000000..f54fb46b77c2
--- /dev/null
+++ b/sound/soc/codecs/tas5720.c
@@ -0,0 +1,620 @@
+/*
+ * tas5720.c - ALSA SoC Texas Instruments TAS5720 Mono Audio Amplifier
+ *
+ * Copyright (C)2015-2016 Texas Instruments Incorporated - http://www.ti.com
+ *
+ * Author: Andreas Dannenberg <dannenberg@ti.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ */
+
+#include <linux/module.h>
+#include <linux/errno.h>
+#include <linux/device.h>
+#include <linux/i2c.h>
+#include <linux/pm_runtime.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+#include <linux/regulator/consumer.h>
+#include <linux/delay.h>
+
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/tlv.h>
+
+#include "tas5720.h"
+
+/* Define how often to check (and clear) the fault status register (in ms) */
+#define TAS5720_FAULT_CHECK_INTERVAL 200
+
+static const char * const tas5720_supply_names[] = {
+ "dvdd", /* Digital power supply. Connect to 3.3-V supply. */
+ "pvdd", /* Class-D amp and analog power supply (connected). */
+};
+
+#define TAS5720_NUM_SUPPLIES ARRAY_SIZE(tas5720_supply_names)
+
+struct tas5720_data {
+ struct snd_soc_codec *codec;
+ struct regmap *regmap;
+ struct i2c_client *tas5720_client;
+ struct regulator_bulk_data supplies[TAS5720_NUM_SUPPLIES];
+ struct delayed_work fault_check_work;
+ unsigned int last_fault;
+};
+
+static int tas5720_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ unsigned int rate = params_rate(params);
+ bool ssz_ds;
+ int ret;
+
+ switch (rate) {
+ case 44100:
+ case 48000:
+ ssz_ds = false;
+ break;
+ case 88200:
+ case 96000:
+ ssz_ds = true;
+ break;
+ default:
+ dev_err(codec->dev, "unsupported sample rate: %u\n", rate);
+ return -EINVAL;
+ }
+
+ ret = snd_soc_update_bits(codec, TAS5720_DIGITAL_CTRL1_REG,
+ TAS5720_SSZ_DS, ssz_ds);
+ if (ret < 0) {
+ dev_err(codec->dev, "error setting sample rate: %d\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static int tas5720_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u8 serial_format;
+ int ret;
+
+ if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) {
+ dev_vdbg(codec->dev, "DAI Format master is not found\n");
+ return -EINVAL;
+ }
+
+ switch (fmt & (SND_SOC_DAIFMT_FORMAT_MASK |
+ SND_SOC_DAIFMT_INV_MASK)) {
+ case (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF):
+ /* 1st data bit occur one BCLK cycle after the frame sync */
+ serial_format = TAS5720_SAIF_I2S;
+ break;
+ case (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF):
+ /*
+ * Note that although the TAS5720 does not have a dedicated DSP
+ * mode it doesn't care about the LRCLK duty cycle during TDM
+ * operation. Therefore we can use the device's I2S mode with
+ * its delaying of the 1st data bit to receive DSP_A formatted
+ * data. See device datasheet for additional details.
+ */
+ serial_format = TAS5720_SAIF_I2S;
+ break;
+ case (SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF):
+ /*
+ * Similar to DSP_A, we can use the fact that the TAS5720 does
+ * not care about the LRCLK duty cycle during TDM to receive
+ * DSP_B formatted data in LEFTJ mode (no delaying of the 1st
+ * data bit).
+ */
+ serial_format = TAS5720_SAIF_LEFTJ;
+ break;
+ case (SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF):
+ /* No delay after the frame sync */
+ serial_format = TAS5720_SAIF_LEFTJ;
+ break;
+ default:
+ dev_vdbg(codec->dev, "DAI Format is not found\n");
+ return -EINVAL;
+ }
+
+ ret = snd_soc_update_bits(codec, TAS5720_DIGITAL_CTRL1_REG,
+ TAS5720_SAIF_FORMAT_MASK,
+ serial_format);
+ if (ret < 0) {
+ dev_err(codec->dev, "error setting SAIF format: %d\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static int tas5720_set_dai_tdm_slot(struct snd_soc_dai *dai,
+ unsigned int tx_mask, unsigned int rx_mask,
+ int slots, int slot_width)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ unsigned int first_slot;
+ int ret;
+
+ if (!tx_mask) {
+ dev_err(codec->dev, "tx masks must not be 0\n");
+ return -EINVAL;
+ }
+
+ /*
+ * Determine the first slot that is being requested. We will only
+ * use the first slot that is found since the TAS5720 is a mono
+ * amplifier.
+ */
+ first_slot = __ffs(tx_mask);
+
+ if (first_slot > 7) {
+ dev_err(codec->dev, "slot selection out of bounds (%u)\n",
+ first_slot);
+ return -EINVAL;
+ }
+
+ /* Enable manual TDM slot selection (instead of I2C ID based) */
+ ret = snd_soc_update_bits(codec, TAS5720_DIGITAL_CTRL1_REG,
+ TAS5720_TDM_CFG_SRC, TAS5720_TDM_CFG_SRC);
+ if (ret < 0)
+ goto error_snd_soc_update_bits;
+
+ /* Configure the TDM slot to process audio from */
+ ret = snd_soc_update_bits(codec, TAS5720_DIGITAL_CTRL2_REG,
+ TAS5720_TDM_SLOT_SEL_MASK, first_slot);
+ if (ret < 0)
+ goto error_snd_soc_update_bits;
+
+ return 0;
+
+error_snd_soc_update_bits:
+ dev_err(codec->dev, "error configuring TDM mode: %d\n", ret);
+ return ret;
+}
+
+static int tas5720_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ int ret;
+
+ ret = snd_soc_update_bits(codec, TAS5720_DIGITAL_CTRL2_REG,
+ TAS5720_MUTE, mute ? TAS5720_MUTE : 0);
+ if (ret < 0) {
+ dev_err(codec->dev, "error (un-)muting device: %d\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static void tas5720_fault_check_work(struct work_struct *work)
+{
+ struct tas5720_data *tas5720 = container_of(work, struct tas5720_data,
+ fault_check_work.work);
+ struct device *dev = tas5720->codec->dev;
+ unsigned int curr_fault;
+ int ret;
+
+ ret = regmap_read(tas5720->regmap, TAS5720_FAULT_REG, &curr_fault);
+ if (ret < 0) {
+ dev_err(dev, "failed to read FAULT register: %d\n", ret);
+ goto out;
+ }
+
+ /* Check/handle all errors except SAIF clock errors */
+ curr_fault &= TAS5720_OCE | TAS5720_DCE | TAS5720_OTE;
+
+ /*
+ * Only flag errors once for a given occurrence. This is needed as
+ * the TAS5720 will take time clearing the fault condition internally
+ * during which we don't want to bombard the system with the same
+ * error message over and over.
+ */
+ if ((curr_fault & TAS5720_OCE) && !(tas5720->last_fault & TAS5720_OCE))
+ dev_crit(dev, "experienced an over current hardware fault\n");
+
+ if ((curr_fault & TAS5720_DCE) && !(tas5720->last_fault & TAS5720_DCE))
+ dev_crit(dev, "experienced a DC detection fault\n");
+
+ if ((curr_fault & TAS5720_OTE) && !(tas5720->last_fault & TAS5720_OTE))
+ dev_crit(dev, "experienced an over temperature fault\n");
+
+ /* Store current fault value so we can detect any changes next time */
+ tas5720->last_fault = curr_fault;
+
+ if (!curr_fault)
+ goto out;
+
+ /*
+ * Periodically toggle SDZ (shutdown bit) H->L->H to clear any latching
+ * faults as long as a fault condition persists. Always going through
+ * the full sequence no matter the first return value to minimizes
+ * chances for the device to end up in shutdown mode.
+ */
+ ret = regmap_write_bits(tas5720->regmap, TAS5720_POWER_CTRL_REG,
+ TAS5720_SDZ, 0);
+ if (ret < 0)
+ dev_err(dev, "failed to write POWER_CTRL register: %d\n", ret);
+
+ ret = regmap_write_bits(tas5720->regmap, TAS5720_POWER_CTRL_REG,
+ TAS5720_SDZ, TAS5720_SDZ);
+ if (ret < 0)
+ dev_err(dev, "failed to write POWER_CTRL register: %d\n", ret);
+
+out:
+ /* Schedule the next fault check at the specified interval */
+ schedule_delayed_work(&tas5720->fault_check_work,
+ msecs_to_jiffies(TAS5720_FAULT_CHECK_INTERVAL));
+}
+
+static int tas5720_codec_probe(struct snd_soc_codec *codec)
+{
+ struct tas5720_data *tas5720 = snd_soc_codec_get_drvdata(codec);
+ unsigned int device_id;
+ int ret;
+
+ tas5720->codec = codec;
+
+ ret = regulator_bulk_enable(ARRAY_SIZE(tas5720->supplies),
+ tas5720->supplies);
+ if (ret != 0) {
+ dev_err(codec->dev, "failed to enable supplies: %d\n", ret);
+ return ret;
+ }
+
+ ret = regmap_read(tas5720->regmap, TAS5720_DEVICE_ID_REG, &device_id);
+ if (ret < 0) {
+ dev_err(codec->dev, "failed to read device ID register: %d\n",
+ ret);
+ goto probe_fail;
+ }
+
+ if (device_id != TAS5720_DEVICE_ID) {
+ dev_err(codec->dev, "wrong device ID. expected: %u read: %u\n",
+ TAS5720_DEVICE_ID, device_id);
+ ret = -ENODEV;
+ goto probe_fail;
+ }
+
+ /* Set device to mute */
+ ret = snd_soc_update_bits(codec, TAS5720_DIGITAL_CTRL2_REG,
+ TAS5720_MUTE, TAS5720_MUTE);
+ if (ret < 0)
+ goto error_snd_soc_update_bits;
+
+ /*
+ * Enter shutdown mode - our default when not playing audio - to
+ * minimize current consumption. On the TAS5720 there is no real down
+ * side doing so as all device registers are preserved and the wakeup
+ * of the codec is rather quick which we do using a dapm widget.
+ */
+ ret = snd_soc_update_bits(codec, TAS5720_POWER_CTRL_REG,
+ TAS5720_SDZ, 0);
+ if (ret < 0)
+ goto error_snd_soc_update_bits;
+
+ INIT_DELAYED_WORK(&tas5720->fault_check_work, tas5720_fault_check_work);
+
+ return 0;
+
+error_snd_soc_update_bits:
+ dev_err(codec->dev, "error configuring device registers: %d\n", ret);
+
+probe_fail:
+ regulator_bulk_disable(ARRAY_SIZE(tas5720->supplies),
+ tas5720->supplies);
+ return ret;
+}
+
+static int tas5720_codec_remove(struct snd_soc_codec *codec)
+{
+ struct tas5720_data *tas5720 = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
+ cancel_delayed_work_sync(&tas5720->fault_check_work);
+
+ ret = regulator_bulk_disable(ARRAY_SIZE(tas5720->supplies),
+ tas5720->supplies);
+ if (ret < 0)
+ dev_err(codec->dev, "failed to disable supplies: %d\n", ret);
+
+ return ret;
+};
+
+static int tas5720_dac_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+ struct tas5720_data *tas5720 = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
+ if (event & SND_SOC_DAPM_POST_PMU) {
+ /* Take TAS5720 out of shutdown mode */
+ ret = snd_soc_update_bits(codec, TAS5720_POWER_CTRL_REG,
+ TAS5720_SDZ, TAS5720_SDZ);
+ if (ret < 0) {
+ dev_err(codec->dev, "error waking codec: %d\n", ret);
+ return ret;
+ }
+
+ /*
+ * Observe codec shutdown-to-active time. The datasheet only
+ * lists a nominal value however just use-it as-is without
+ * additional padding to minimize the delay introduced in
+ * starting to play audio (actually there is other setup done
+ * by the ASoC framework that will provide additional delays,
+ * so we should always be safe).
+ */
+ msleep(25);
+
+ /* Turn on TAS5720 periodic fault checking/handling */
+ tas5720->last_fault = 0;
+ schedule_delayed_work(&tas5720->fault_check_work,
+ msecs_to_jiffies(TAS5720_FAULT_CHECK_INTERVAL));
+ } else if (event & SND_SOC_DAPM_PRE_PMD) {
+ /* Disable TAS5720 periodic fault checking/handling */
+ cancel_delayed_work_sync(&tas5720->fault_check_work);
+
+ /* Place TAS5720 in shutdown mode to minimize current draw */
+ ret = snd_soc_update_bits(codec, TAS5720_POWER_CTRL_REG,
+ TAS5720_SDZ, 0);
+ if (ret < 0) {
+ dev_err(codec->dev, "error shutting down codec: %d\n",
+ ret);
+ return ret;
+ }
+ }
+
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int tas5720_suspend(struct snd_soc_codec *codec)
+{
+ struct tas5720_data *tas5720 = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
+ regcache_cache_only(tas5720->regmap, true);
+ regcache_mark_dirty(tas5720->regmap);
+
+ ret = regulator_bulk_disable(ARRAY_SIZE(tas5720->supplies),
+ tas5720->supplies);
+ if (ret < 0)
+ dev_err(codec->dev, "failed to disable supplies: %d\n", ret);
+
+ return ret;
+}
+
+static int tas5720_resume(struct snd_soc_codec *codec)
+{
+ struct tas5720_data *tas5720 = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
+ ret = regulator_bulk_enable(ARRAY_SIZE(tas5720->supplies),
+ tas5720->supplies);
+ if (ret < 0) {
+ dev_err(codec->dev, "failed to enable supplies: %d\n", ret);
+ return ret;
+ }
+
+ regcache_cache_only(tas5720->regmap, false);
+
+ ret = regcache_sync(tas5720->regmap);
+ if (ret < 0) {
+ dev_err(codec->dev, "failed to sync regcache: %d\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+#else
+#define tas5720_suspend NULL
+#define tas5720_resume NULL
+#endif
+
+static bool tas5720_is_volatile_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case TAS5720_DEVICE_ID_REG:
+ case TAS5720_FAULT_REG:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static const struct regmap_config tas5720_regmap_config = {
+ .reg_bits = 8,
+ .val_bits = 8,
+
+ .max_register = TAS5720_MAX_REG,
+ .cache_type = REGCACHE_RBTREE,
+ .volatile_reg = tas5720_is_volatile_reg,
+};
+
+/*
+ * DAC analog gain. There are four discrete values to select from, ranging
+ * from 19.2 dB to 26.3dB.
+ */
+static const DECLARE_TLV_DB_RANGE(dac_analog_tlv,
+ 0x0, 0x0, TLV_DB_SCALE_ITEM(1920, 0, 0),
+ 0x1, 0x1, TLV_DB_SCALE_ITEM(2070, 0, 0),
+ 0x2, 0x2, TLV_DB_SCALE_ITEM(2350, 0, 0),
+ 0x3, 0x3, TLV_DB_SCALE_ITEM(2630, 0, 0),
+);
+
+/*
+ * DAC digital volumes. From -103.5 to 24 dB in 0.5 dB steps. Note that
+ * setting the gain below -100 dB (register value <0x7) is effectively a MUTE
+ * as per device datasheet.
+ */
+static DECLARE_TLV_DB_SCALE(dac_tlv, -10350, 50, 0);
+
+static const struct snd_kcontrol_new tas5720_snd_controls[] = {
+ SOC_SINGLE_TLV("Speaker Driver Playback Volume",
+ TAS5720_VOLUME_CTRL_REG, 0, 0xff, 0, dac_tlv),
+ SOC_SINGLE_TLV("Speaker Driver Analog Gain", TAS5720_ANALOG_CTRL_REG,
+ TAS5720_ANALOG_GAIN_SHIFT, 3, 0, dac_analog_tlv),
+};
+
+static const struct snd_soc_dapm_widget tas5720_dapm_widgets[] = {
+ SND_SOC_DAPM_AIF_IN("DAC IN", "Playback", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_DAC_E("DAC", NULL, SND_SOC_NOPM, 0, 0, tas5720_dac_event,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+ SND_SOC_DAPM_OUTPUT("OUT")
+};
+
+static const struct snd_soc_dapm_route tas5720_audio_map[] = {
+ { "DAC", NULL, "DAC IN" },
+ { "OUT", NULL, "DAC" },
+};
+
+static struct snd_soc_codec_driver soc_codec_dev_tas5720 = {
+ .probe = tas5720_codec_probe,
+ .remove = tas5720_codec_remove,
+ .suspend = tas5720_suspend,
+ .resume = tas5720_resume,
+
+ .controls = tas5720_snd_controls,
+ .num_controls = ARRAY_SIZE(tas5720_snd_controls),
+ .dapm_widgets = tas5720_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(tas5720_dapm_widgets),
+ .dapm_routes = tas5720_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(tas5720_audio_map),
+};
+
+/* PCM rates supported by the TAS5720 driver */
+#define TAS5720_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\
+ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
+
+/* Formats supported by TAS5720 driver */
+#define TAS5720_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE |\
+ SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE)
+
+static struct snd_soc_dai_ops tas5720_speaker_dai_ops = {
+ .hw_params = tas5720_hw_params,
+ .set_fmt = tas5720_set_dai_fmt,
+ .set_tdm_slot = tas5720_set_dai_tdm_slot,
+ .digital_mute = tas5720_mute,
+};
+
+/*
+ * TAS5720 DAI structure
+ *
+ * Note that were are advertising .playback.channels_max = 2 despite this being
+ * a mono amplifier. The reason for that is that some serial ports such as TI's
+ * McASP module have a minimum number of channels (2) that they can output.
+ * Advertising more channels than we have will allow us to interface with such
+ * a serial port without really any negative side effects as the TAS5720 will
+ * simply ignore any extra channel(s) asides from the one channel that is
+ * configured to be played back.
+ */
+static struct snd_soc_dai_driver tas5720_dai[] = {
+ {
+ .name = "tas5720-amplifier",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = TAS5720_RATES,
+ .formats = TAS5720_FORMATS,
+ },
+ .ops = &tas5720_speaker_dai_ops,
+ },
+};
+
+static int tas5720_probe(struct i2c_client *client,
+ const struct i2c_device_id *id)
+{
+ struct device *dev = &client->dev;
+ struct tas5720_data *data;
+ int ret;
+ int i;
+
+ data = devm_kzalloc(dev, sizeof(*data), GFP_KERNEL);
+ if (!data)
+ return -ENOMEM;
+
+ data->tas5720_client = client;
+ data->regmap = devm_regmap_init_i2c(client, &tas5720_regmap_config);
+ if (IS_ERR(data->regmap)) {
+ ret = PTR_ERR(data->regmap);
+ dev_err(dev, "failed to allocate register map: %d\n", ret);
+ return ret;
+ }
+
+ for (i = 0; i < ARRAY_SIZE(data->supplies); i++)
+ data->supplies[i].supply = tas5720_supply_names[i];
+
+ ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(data->supplies),
+ data->supplies);
+ if (ret != 0) {
+ dev_err(dev, "failed to request supplies: %d\n", ret);
+ return ret;
+ }
+
+ dev_set_drvdata(dev, data);
+
+ ret = snd_soc_register_codec(&client->dev,
+ &soc_codec_dev_tas5720,
+ tas5720_dai, ARRAY_SIZE(tas5720_dai));
+ if (ret < 0) {
+ dev_err(dev, "failed to register codec: %d\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static int tas5720_remove(struct i2c_client *client)
+{
+ struct device *dev = &client->dev;
+
+ snd_soc_unregister_codec(dev);
+
+ return 0;
+}
+
+static const struct i2c_device_id tas5720_id[] = {
+ { "tas5720", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, tas5720_id);
+
+#if IS_ENABLED(CONFIG_OF)
+static const struct of_device_id tas5720_of_match[] = {
+ { .compatible = "ti,tas5720", },
+ { },
+};
+MODULE_DEVICE_TABLE(of, tas5720_of_match);
+#endif
+
+static struct i2c_driver tas5720_i2c_driver = {
+ .driver = {
+ .name = "tas5720",
+ .of_match_table = of_match_ptr(tas5720_of_match),
+ },
+ .probe = tas5720_probe,
+ .remove = tas5720_remove,
+ .id_table = tas5720_id,
+};
+
+module_i2c_driver(tas5720_i2c_driver);
+
+MODULE_AUTHOR("Andreas Dannenberg <dannenberg@ti.com>");
+MODULE_DESCRIPTION("TAS5720 Audio amplifier driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tas5720.h b/sound/soc/codecs/tas5720.h
new file mode 100644
index 000000000000..3d077c779b12
--- /dev/null
+++ b/sound/soc/codecs/tas5720.h
@@ -0,0 +1,90 @@
+/*
+ * tas5720.h - ALSA SoC Texas Instruments TAS5720 Mono Audio Amplifier
+ *
+ * Copyright (C)2015-2016 Texas Instruments Incorporated - http://www.ti.com
+ *
+ * Author: Andreas Dannenberg <dannenberg@ti.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ */
+
+#ifndef __TAS5720_H__
+#define __TAS5720_H__
+
+/* Register Address Map */
+#define TAS5720_DEVICE_ID_REG 0x00
+#define TAS5720_POWER_CTRL_REG 0x01
+#define TAS5720_DIGITAL_CTRL1_REG 0x02
+#define TAS5720_DIGITAL_CTRL2_REG 0x03
+#define TAS5720_VOLUME_CTRL_REG 0x04
+#define TAS5720_ANALOG_CTRL_REG 0x06
+#define TAS5720_FAULT_REG 0x08
+#define TAS5720_DIGITAL_CLIP2_REG 0x10
+#define TAS5720_DIGITAL_CLIP1_REG 0x11
+#define TAS5720_MAX_REG TAS5720_DIGITAL_CLIP1_REG
+
+/* TAS5720_DEVICE_ID_REG */
+#define TAS5720_DEVICE_ID 0x01
+
+/* TAS5720_POWER_CTRL_REG */
+#define TAS5720_DIG_CLIP_MASK GENMASK(7, 2)
+#define TAS5720_SLEEP BIT(1)
+#define TAS5720_SDZ BIT(0)
+
+/* TAS5720_DIGITAL_CTRL1_REG */
+#define TAS5720_HPF_BYPASS BIT(7)
+#define TAS5720_TDM_CFG_SRC BIT(6)
+#define TAS5720_SSZ_DS BIT(3)
+#define TAS5720_SAIF_RIGHTJ_24BIT (0x0)
+#define TAS5720_SAIF_RIGHTJ_20BIT (0x1)
+#define TAS5720_SAIF_RIGHTJ_18BIT (0x2)
+#define TAS5720_SAIF_RIGHTJ_16BIT (0x3)
+#define TAS5720_SAIF_I2S (0x4)
+#define TAS5720_SAIF_LEFTJ (0x5)
+#define TAS5720_SAIF_FORMAT_MASK GENMASK(2, 0)
+
+/* TAS5720_DIGITAL_CTRL2_REG */
+#define TAS5720_MUTE BIT(4)
+#define TAS5720_TDM_SLOT_SEL_MASK GENMASK(2, 0)
+
+/* TAS5720_ANALOG_CTRL_REG */
+#define TAS5720_PWM_RATE_6_3_FSYNC (0x0 << 4)
+#define TAS5720_PWM_RATE_8_4_FSYNC (0x1 << 4)
+#define TAS5720_PWM_RATE_10_5_FSYNC (0x2 << 4)
+#define TAS5720_PWM_RATE_12_6_FSYNC (0x3 << 4)
+#define TAS5720_PWM_RATE_14_7_FSYNC (0x4 << 4)
+#define TAS5720_PWM_RATE_16_8_FSYNC (0x5 << 4)
+#define TAS5720_PWM_RATE_20_10_FSYNC (0x6 << 4)
+#define TAS5720_PWM_RATE_24_12_FSYNC (0x7 << 4)
+#define TAS5720_PWM_RATE_MASK GENMASK(6, 4)
+#define TAS5720_ANALOG_GAIN_19_2DBV (0x0 << 2)
+#define TAS5720_ANALOG_GAIN_20_7DBV (0x1 << 2)
+#define TAS5720_ANALOG_GAIN_23_5DBV (0x2 << 2)
+#define TAS5720_ANALOG_GAIN_26_3DBV (0x3 << 2)
+#define TAS5720_ANALOG_GAIN_MASK GENMASK(3, 2)
+#define TAS5720_ANALOG_GAIN_SHIFT (0x2)
+
+/* TAS5720_FAULT_REG */
+#define TAS5720_OC_THRESH_100PCT (0x0 << 4)
+#define TAS5720_OC_THRESH_75PCT (0x1 << 4)
+#define TAS5720_OC_THRESH_50PCT (0x2 << 4)
+#define TAS5720_OC_THRESH_25PCT (0x3 << 4)
+#define TAS5720_OC_THRESH_MASK GENMASK(5, 4)
+#define TAS5720_CLKE BIT(3)
+#define TAS5720_OCE BIT(2)
+#define TAS5720_DCE BIT(1)
+#define TAS5720_OTE BIT(0)
+#define TAS5720_FAULT_MASK GENMASK(3, 0)
+
+/* TAS5720_DIGITAL_CLIP1_REG */
+#define TAS5720_CLIP1_MASK GENMASK(7, 2)
+#define TAS5720_CLIP1_SHIFT (0x2)
+
+#endif /* __TAS5720_H__ */
diff --git a/sound/soc/codecs/tlv320aic32x4-i2c.c b/sound/soc/codecs/tlv320aic32x4-i2c.c
new file mode 100644
index 000000000000..59606cf3008f
--- /dev/null
+++ b/sound/soc/codecs/tlv320aic32x4-i2c.c
@@ -0,0 +1,74 @@
+/*
+ * linux/sound/soc/codecs/tlv320aic32x4-i2c.c
+ *
+ * Copyright 2011 NW Digital Radio
+ *
+ * Author: Jeremy McDermond <nh6z@nh6z.net>
+ *
+ * Based on sound/soc/codecs/wm8974 and TI driver for kernel 2.6.27.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+#include <linux/i2c.h>
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/regmap.h>
+#include <sound/soc.h>
+
+#include "tlv320aic32x4.h"
+
+static int aic32x4_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct regmap *regmap;
+ struct regmap_config config;
+
+ config = aic32x4_regmap_config;
+ config.reg_bits = 8;
+ config.val_bits = 8;
+
+ regmap = devm_regmap_init_i2c(i2c, &config);
+ return aic32x4_probe(&i2c->dev, regmap);
+}
+
+static int aic32x4_i2c_remove(struct i2c_client *i2c)
+{
+ return aic32x4_remove(&i2c->dev);
+}
+
+static const struct i2c_device_id aic32x4_i2c_id[] = {
+ { "tlv320aic32x4", 0 },
+ { /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(i2c, aic32x4_i2c_id);
+
+static const struct of_device_id aic32x4_of_id[] = {
+ { .compatible = "ti,tlv320aic32x4", },
+ { /* senitel */ }
+};
+MODULE_DEVICE_TABLE(of, aic32x4_of_id);
+
+static struct i2c_driver aic32x4_i2c_driver = {
+ .driver = {
+ .name = "tlv320aic32x4",
+ .of_match_table = aic32x4_of_id,
+ },
+ .probe = aic32x4_i2c_probe,
+ .remove = aic32x4_i2c_remove,
+ .id_table = aic32x4_i2c_id,
+};
+
+module_i2c_driver(aic32x4_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC TLV320AIC32x4 codec driver I2C");
+MODULE_AUTHOR("Jeremy McDermond <nh6z@nh6z.net>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tlv320aic32x4-spi.c b/sound/soc/codecs/tlv320aic32x4-spi.c
new file mode 100644
index 000000000000..724fcdd491b2
--- /dev/null
+++ b/sound/soc/codecs/tlv320aic32x4-spi.c
@@ -0,0 +1,76 @@
+/*
+ * linux/sound/soc/codecs/tlv320aic32x4-spi.c
+ *
+ * Copyright 2011 NW Digital Radio
+ *
+ * Author: Jeremy McDermond <nh6z@nh6z.net>
+ *
+ * Based on sound/soc/codecs/wm8974 and TI driver for kernel 2.6.27.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+#include <linux/spi/spi.h>
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/regmap.h>
+#include <sound/soc.h>
+
+#include "tlv320aic32x4.h"
+
+static int aic32x4_spi_probe(struct spi_device *spi)
+{
+ struct regmap *regmap;
+ struct regmap_config config;
+
+ config = aic32x4_regmap_config;
+ config.reg_bits = 7;
+ config.pad_bits = 1;
+ config.val_bits = 8;
+ config.read_flag_mask = 0x01;
+
+ regmap = devm_regmap_init_spi(spi, &config);
+ return aic32x4_probe(&spi->dev, regmap);
+}
+
+static int aic32x4_spi_remove(struct spi_device *spi)
+{
+ return aic32x4_remove(&spi->dev);
+}
+
+static const struct spi_device_id aic32x4_spi_id[] = {
+ { "tlv320aic32x4", 0 },
+ { /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(spi, aic32x4_spi_id);
+
+static const struct of_device_id aic32x4_of_id[] = {
+ { .compatible = "ti,tlv320aic32x4", },
+ { /* senitel */ }
+};
+MODULE_DEVICE_TABLE(of, aic32x4_of_id);
+
+static struct spi_driver aic32x4_spi_driver = {
+ .driver = {
+ .name = "tlv320aic32x4",
+ .owner = THIS_MODULE,
+ .of_match_table = aic32x4_of_id,
+ },
+ .probe = aic32x4_spi_probe,
+ .remove = aic32x4_spi_remove,
+ .id_table = aic32x4_spi_id,
+};
+
+module_spi_driver(aic32x4_spi_driver);
+
+MODULE_DESCRIPTION("ASoC TLV320AIC32x4 codec driver SPI");
+MODULE_AUTHOR("Jeremy McDermond <nh6z@nh6z.net>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index f2d3191961e1..85d4978d0384 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -30,7 +30,6 @@
#include <linux/pm.h>
#include <linux/gpio.h>
#include <linux/of_gpio.h>
-#include <linux/i2c.h>
#include <linux/cdev.h>
#include <linux/slab.h>
#include <linux/clk.h>
@@ -160,7 +159,10 @@ static const struct aic32x4_rate_divs aic32x4_divs[] = {
/* 48k rate */
{AIC32X4_FREQ_12000000, 48000, 1, 8, 1920, 128, 2, 8, 128, 2, 8, 4},
{AIC32X4_FREQ_24000000, 48000, 2, 8, 1920, 128, 8, 2, 64, 8, 4, 4},
- {AIC32X4_FREQ_25000000, 48000, 2, 7, 8643, 128, 8, 2, 64, 8, 4, 4}
+ {AIC32X4_FREQ_25000000, 48000, 2, 7, 8643, 128, 8, 2, 64, 8, 4, 4},
+
+ /* 96k rate */
+ {AIC32X4_FREQ_25000000, 96000, 2, 7, 8643, 64, 4, 4, 64, 4, 4, 1},
};
static const struct snd_kcontrol_new hpl_output_mixer_controls[] = {
@@ -181,16 +183,71 @@ static const struct snd_kcontrol_new lor_output_mixer_controls[] = {
SOC_DAPM_SINGLE("R_DAC Switch", AIC32X4_LORROUTE, 3, 1, 0),
};
-static const struct snd_kcontrol_new left_input_mixer_controls[] = {
- SOC_DAPM_SINGLE("IN1_L P Switch", AIC32X4_LMICPGAPIN, 6, 1, 0),
- SOC_DAPM_SINGLE("IN2_L P Switch", AIC32X4_LMICPGAPIN, 4, 1, 0),
- SOC_DAPM_SINGLE("IN3_L P Switch", AIC32X4_LMICPGAPIN, 2, 1, 0),
+static const char * const resistor_text[] = {
+ "Off", "10 kOhm", "20 kOhm", "40 kOhm",
};
-static const struct snd_kcontrol_new right_input_mixer_controls[] = {
- SOC_DAPM_SINGLE("IN1_R P Switch", AIC32X4_RMICPGAPIN, 6, 1, 0),
- SOC_DAPM_SINGLE("IN2_R P Switch", AIC32X4_RMICPGAPIN, 4, 1, 0),
- SOC_DAPM_SINGLE("IN3_R P Switch", AIC32X4_RMICPGAPIN, 2, 1, 0),
+/* Left mixer pins */
+static SOC_ENUM_SINGLE_DECL(in1l_lpga_p_enum, AIC32X4_LMICPGAPIN, 6, resistor_text);
+static SOC_ENUM_SINGLE_DECL(in2l_lpga_p_enum, AIC32X4_LMICPGAPIN, 4, resistor_text);
+static SOC_ENUM_SINGLE_DECL(in3l_lpga_p_enum, AIC32X4_LMICPGAPIN, 2, resistor_text);
+static SOC_ENUM_SINGLE_DECL(in1r_lpga_p_enum, AIC32X4_LMICPGAPIN, 0, resistor_text);
+
+static SOC_ENUM_SINGLE_DECL(cml_lpga_n_enum, AIC32X4_LMICPGANIN, 6, resistor_text);
+static SOC_ENUM_SINGLE_DECL(in2r_lpga_n_enum, AIC32X4_LMICPGANIN, 4, resistor_text);
+static SOC_ENUM_SINGLE_DECL(in3r_lpga_n_enum, AIC32X4_LMICPGANIN, 2, resistor_text);
+
+static const struct snd_kcontrol_new in1l_to_lmixer_controls[] = {
+ SOC_DAPM_ENUM("IN1_L L+ Switch", in1l_lpga_p_enum),
+};
+static const struct snd_kcontrol_new in2l_to_lmixer_controls[] = {
+ SOC_DAPM_ENUM("IN2_L L+ Switch", in2l_lpga_p_enum),
+};
+static const struct snd_kcontrol_new in3l_to_lmixer_controls[] = {
+ SOC_DAPM_ENUM("IN3_L L+ Switch", in3l_lpga_p_enum),
+};
+static const struct snd_kcontrol_new in1r_to_lmixer_controls[] = {
+ SOC_DAPM_ENUM("IN1_R L+ Switch", in1r_lpga_p_enum),
+};
+static const struct snd_kcontrol_new cml_to_lmixer_controls[] = {
+ SOC_DAPM_ENUM("CM_L L- Switch", cml_lpga_n_enum),
+};
+static const struct snd_kcontrol_new in2r_to_lmixer_controls[] = {
+ SOC_DAPM_ENUM("IN2_R L- Switch", in2r_lpga_n_enum),
+};
+static const struct snd_kcontrol_new in3r_to_lmixer_controls[] = {
+ SOC_DAPM_ENUM("IN3_R L- Switch", in3r_lpga_n_enum),
+};
+
+/* Right mixer pins */
+static SOC_ENUM_SINGLE_DECL(in1r_rpga_p_enum, AIC32X4_RMICPGAPIN, 6, resistor_text);
+static SOC_ENUM_SINGLE_DECL(in2r_rpga_p_enum, AIC32X4_RMICPGAPIN, 4, resistor_text);
+static SOC_ENUM_SINGLE_DECL(in3r_rpga_p_enum, AIC32X4_RMICPGAPIN, 2, resistor_text);
+static SOC_ENUM_SINGLE_DECL(in2l_rpga_p_enum, AIC32X4_RMICPGAPIN, 0, resistor_text);
+static SOC_ENUM_SINGLE_DECL(cmr_rpga_n_enum, AIC32X4_RMICPGANIN, 6, resistor_text);
+static SOC_ENUM_SINGLE_DECL(in1l_rpga_n_enum, AIC32X4_RMICPGANIN, 4, resistor_text);
+static SOC_ENUM_SINGLE_DECL(in3l_rpga_n_enum, AIC32X4_RMICPGANIN, 2, resistor_text);
+
+static const struct snd_kcontrol_new in1r_to_rmixer_controls[] = {
+ SOC_DAPM_ENUM("IN1_R R+ Switch", in1r_rpga_p_enum),
+};
+static const struct snd_kcontrol_new in2r_to_rmixer_controls[] = {
+ SOC_DAPM_ENUM("IN2_R R+ Switch", in2r_rpga_p_enum),
+};
+static const struct snd_kcontrol_new in3r_to_rmixer_controls[] = {
+ SOC_DAPM_ENUM("IN3_R R+ Switch", in3r_rpga_p_enum),
+};
+static const struct snd_kcontrol_new in2l_to_rmixer_controls[] = {
+ SOC_DAPM_ENUM("IN2_L R+ Switch", in2l_rpga_p_enum),
+};
+static const struct snd_kcontrol_new cmr_to_rmixer_controls[] = {
+ SOC_DAPM_ENUM("CM_R R- Switch", cmr_rpga_n_enum),
+};
+static const struct snd_kcontrol_new in1l_to_rmixer_controls[] = {
+ SOC_DAPM_ENUM("IN1_L R- Switch", in1l_rpga_n_enum),
+};
+static const struct snd_kcontrol_new in3l_to_rmixer_controls[] = {
+ SOC_DAPM_ENUM("IN3_L R- Switch", in3l_rpga_n_enum),
};
static const struct snd_soc_dapm_widget aic32x4_dapm_widgets[] = {
@@ -214,14 +271,39 @@ static const struct snd_soc_dapm_widget aic32x4_dapm_widgets[] = {
&lor_output_mixer_controls[0],
ARRAY_SIZE(lor_output_mixer_controls)),
SND_SOC_DAPM_PGA("LOR Power", AIC32X4_OUTPWRCTL, 2, 0, NULL, 0),
- SND_SOC_DAPM_MIXER("Left Input Mixer", SND_SOC_NOPM, 0, 0,
- &left_input_mixer_controls[0],
- ARRAY_SIZE(left_input_mixer_controls)),
- SND_SOC_DAPM_MIXER("Right Input Mixer", SND_SOC_NOPM, 0, 0,
- &right_input_mixer_controls[0],
- ARRAY_SIZE(right_input_mixer_controls)),
- SND_SOC_DAPM_ADC("Left ADC", "Left Capture", AIC32X4_ADCSETUP, 7, 0),
+
SND_SOC_DAPM_ADC("Right ADC", "Right Capture", AIC32X4_ADCSETUP, 6, 0),
+ SND_SOC_DAPM_MUX("IN1_R to Right Mixer Positive Resistor", SND_SOC_NOPM, 0, 0,
+ in1r_to_rmixer_controls),
+ SND_SOC_DAPM_MUX("IN2_R to Right Mixer Positive Resistor", SND_SOC_NOPM, 0, 0,
+ in2r_to_rmixer_controls),
+ SND_SOC_DAPM_MUX("IN3_R to Right Mixer Positive Resistor", SND_SOC_NOPM, 0, 0,
+ in3r_to_rmixer_controls),
+ SND_SOC_DAPM_MUX("IN2_L to Right Mixer Positive Resistor", SND_SOC_NOPM, 0, 0,
+ in2l_to_rmixer_controls),
+ SND_SOC_DAPM_MUX("CM_R to Right Mixer Negative Resistor", SND_SOC_NOPM, 0, 0,
+ cmr_to_rmixer_controls),
+ SND_SOC_DAPM_MUX("IN1_L to Right Mixer Negative Resistor", SND_SOC_NOPM, 0, 0,
+ in1l_to_rmixer_controls),
+ SND_SOC_DAPM_MUX("IN3_L to Right Mixer Negative Resistor", SND_SOC_NOPM, 0, 0,
+ in3l_to_rmixer_controls),
+
+ SND_SOC_DAPM_ADC("Left ADC", "Left Capture", AIC32X4_ADCSETUP, 7, 0),
+ SND_SOC_DAPM_MUX("IN1_L to Left Mixer Positive Resistor", SND_SOC_NOPM, 0, 0,
+ in1l_to_lmixer_controls),
+ SND_SOC_DAPM_MUX("IN2_L to Left Mixer Positive Resistor", SND_SOC_NOPM, 0, 0,
+ in2l_to_lmixer_controls),
+ SND_SOC_DAPM_MUX("IN3_L to Left Mixer Positive Resistor", SND_SOC_NOPM, 0, 0,
+ in3l_to_lmixer_controls),
+ SND_SOC_DAPM_MUX("IN1_R to Left Mixer Positive Resistor", SND_SOC_NOPM, 0, 0,
+ in1r_to_lmixer_controls),
+ SND_SOC_DAPM_MUX("CM_L to Left Mixer Negative Resistor", SND_SOC_NOPM, 0, 0,
+ cml_to_lmixer_controls),
+ SND_SOC_DAPM_MUX("IN2_R to Left Mixer Negative Resistor", SND_SOC_NOPM, 0, 0,
+ in2r_to_lmixer_controls),
+ SND_SOC_DAPM_MUX("IN3_R to Left Mixer Negative Resistor", SND_SOC_NOPM, 0, 0,
+ in3r_to_lmixer_controls),
+
SND_SOC_DAPM_MICBIAS("Mic Bias", AIC32X4_MICBIAS, 6, 0),
SND_SOC_DAPM_OUTPUT("HPL"),
@@ -261,19 +343,77 @@ static const struct snd_soc_dapm_route aic32x4_dapm_routes[] = {
{"LOR Power", NULL, "LOR Output Mixer"},
{"LOR", NULL, "LOR Power"},
- /* Left input */
- {"Left Input Mixer", "IN1_L P Switch", "IN1_L"},
- {"Left Input Mixer", "IN2_L P Switch", "IN2_L"},
- {"Left Input Mixer", "IN3_L P Switch", "IN3_L"},
-
- {"Left ADC", NULL, "Left Input Mixer"},
-
/* Right Input */
- {"Right Input Mixer", "IN1_R P Switch", "IN1_R"},
- {"Right Input Mixer", "IN2_R P Switch", "IN2_R"},
- {"Right Input Mixer", "IN3_R P Switch", "IN3_R"},
-
- {"Right ADC", NULL, "Right Input Mixer"},
+ {"Right ADC", NULL, "IN1_R to Right Mixer Positive Resistor"},
+ {"IN1_R to Right Mixer Positive Resistor", "10 kOhm", "IN1_R"},
+ {"IN1_R to Right Mixer Positive Resistor", "20 kOhm", "IN1_R"},
+ {"IN1_R to Right Mixer Positive Resistor", "40 kOhm", "IN1_R"},
+
+ {"Right ADC", NULL, "IN2_R to Right Mixer Positive Resistor"},
+ {"IN2_R to Right Mixer Positive Resistor", "10 kOhm", "IN2_R"},
+ {"IN2_R to Right Mixer Positive Resistor", "20 kOhm", "IN2_R"},
+ {"IN2_R to Right Mixer Positive Resistor", "40 kOhm", "IN2_R"},
+
+ {"Right ADC", NULL, "IN3_R to Right Mixer Positive Resistor"},
+ {"IN3_R to Right Mixer Positive Resistor", "10 kOhm", "IN3_R"},
+ {"IN3_R to Right Mixer Positive Resistor", "20 kOhm", "IN3_R"},
+ {"IN3_R to Right Mixer Positive Resistor", "40 kOhm", "IN3_R"},
+
+ {"Right ADC", NULL, "IN2_L to Right Mixer Positive Resistor"},
+ {"IN2_L to Right Mixer Positive Resistor", "10 kOhm", "IN2_L"},
+ {"IN2_L to Right Mixer Positive Resistor", "20 kOhm", "IN2_L"},
+ {"IN2_L to Right Mixer Positive Resistor", "40 kOhm", "IN2_L"},
+
+ {"Right ADC", NULL, "CM_R to Right Mixer Negative Resistor"},
+ {"CM_R to Right Mixer Negative Resistor", "10 kOhm", "CM_R"},
+ {"CM_R to Right Mixer Negative Resistor", "20 kOhm", "CM_R"},
+ {"CM_R to Right Mixer Negative Resistor", "40 kOhm", "CM_R"},
+
+ {"Right ADC", NULL, "IN1_L to Right Mixer Negative Resistor"},
+ {"IN1_L to Right Mixer Negative Resistor", "10 kOhm", "IN1_L"},
+ {"IN1_L to Right Mixer Negative Resistor", "20 kOhm", "IN1_L"},
+ {"IN1_L to Right Mixer Negative Resistor", "40 kOhm", "IN1_L"},
+
+ {"Right ADC", NULL, "IN3_L to Right Mixer Negative Resistor"},
+ {"IN3_L to Right Mixer Negative Resistor", "10 kOhm", "IN3_L"},
+ {"IN3_L to Right Mixer Negative Resistor", "20 kOhm", "IN3_L"},
+ {"IN3_L to Right Mixer Negative Resistor", "40 kOhm", "IN3_L"},
+
+ /* Left Input */
+ {"Left ADC", NULL, "IN1_L to Left Mixer Positive Resistor"},
+ {"IN1_L to Left Mixer Positive Resistor", "10 kOhm", "IN1_L"},
+ {"IN1_L to Left Mixer Positive Resistor", "20 kOhm", "IN1_L"},
+ {"IN1_L to Left Mixer Positive Resistor", "40 kOhm", "IN1_L"},
+
+ {"Left ADC", NULL, "IN2_L to Left Mixer Positive Resistor"},
+ {"IN2_L to Left Mixer Positive Resistor", "10 kOhm", "IN2_L"},
+ {"IN2_L to Left Mixer Positive Resistor", "20 kOhm", "IN2_L"},
+ {"IN2_L to Left Mixer Positive Resistor", "40 kOhm", "IN2_L"},
+
+ {"Left ADC", NULL, "IN3_L to Left Mixer Positive Resistor"},
+ {"IN3_L to Left Mixer Positive Resistor", "10 kOhm", "IN3_L"},
+ {"IN3_L to Left Mixer Positive Resistor", "20 kOhm", "IN3_L"},
+ {"IN3_L to Left Mixer Positive Resistor", "40 kOhm", "IN3_L"},
+
+ {"Left ADC", NULL, "IN1_R to Left Mixer Positive Resistor"},
+ {"IN1_R to Left Mixer Positive Resistor", "10 kOhm", "IN1_R"},
+ {"IN1_R to Left Mixer Positive Resistor", "20 kOhm", "IN1_R"},
+ {"IN1_R to Left Mixer Positive Resistor", "40 kOhm", "IN1_R"},
+
+ {"Left ADC", NULL, "CM_L to Left Mixer Negative Resistor"},
+ {"CM_L to Left Mixer Negative Resistor", "10 kOhm", "CM_L"},
+ {"CM_L to Left Mixer Negative Resistor", "20 kOhm", "CM_L"},
+ {"CM_L to Left Mixer Negative Resistor", "40 kOhm", "CM_L"},
+
+ {"Left ADC", NULL, "IN2_R to Left Mixer Negative Resistor"},
+ {"IN2_R to Left Mixer Negative Resistor", "10 kOhm", "IN2_R"},
+ {"IN2_R to Left Mixer Negative Resistor", "20 kOhm", "IN2_R"},
+ {"IN2_R to Left Mixer Negative Resistor", "40 kOhm", "IN2_R"},
+
+ {"Left ADC", NULL, "IN3_R to Left Mixer Negative Resistor"},
+ {"IN3_R to Left Mixer Negative Resistor", "10 kOhm", "IN3_R"},
+ {"IN3_R to Left Mixer Negative Resistor", "20 kOhm", "IN3_R"},
+ {"IN3_R to Left Mixer Negative Resistor", "40 kOhm", "IN3_R"},
};
static const struct regmap_range_cfg aic32x4_regmap_pages[] = {
@@ -287,14 +427,12 @@ static const struct regmap_range_cfg aic32x4_regmap_pages[] = {
},
};
-static const struct regmap_config aic32x4_regmap = {
- .reg_bits = 8,
- .val_bits = 8,
-
+const struct regmap_config aic32x4_regmap_config = {
.max_register = AIC32X4_RMICPGAVOL,
.ranges = aic32x4_regmap_pages,
.num_ranges = ARRAY_SIZE(aic32x4_regmap_pages),
};
+EXPORT_SYMBOL(aic32x4_regmap_config);
static inline int aic32x4_get_divs(int mclk, int rate)
{
@@ -567,7 +705,7 @@ static int aic32x4_set_bias_level(struct snd_soc_codec *codec,
return 0;
}
-#define AIC32X4_RATES SNDRV_PCM_RATE_8000_48000
+#define AIC32X4_RATES SNDRV_PCM_RATE_8000_96000
#define AIC32X4_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \
| SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
@@ -596,7 +734,7 @@ static struct snd_soc_dai_driver aic32x4_dai = {
.symmetric_rates = 1,
};
-static int aic32x4_probe(struct snd_soc_codec *codec)
+static int aic32x4_codec_probe(struct snd_soc_codec *codec)
{
struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec);
u32 tmp_reg;
@@ -655,7 +793,7 @@ static int aic32x4_probe(struct snd_soc_codec *codec)
}
static struct snd_soc_codec_driver soc_codec_dev_aic32x4 = {
- .probe = aic32x4_probe,
+ .probe = aic32x4_codec_probe,
.set_bias_level = aic32x4_set_bias_level,
.suspend_bias_off = true,
@@ -777,24 +915,22 @@ error_ldo:
return ret;
}
-static int aic32x4_i2c_probe(struct i2c_client *i2c,
- const struct i2c_device_id *id)
+int aic32x4_probe(struct device *dev, struct regmap *regmap)
{
- struct aic32x4_pdata *pdata = i2c->dev.platform_data;
struct aic32x4_priv *aic32x4;
- struct device_node *np = i2c->dev.of_node;
+ struct aic32x4_pdata *pdata = dev->platform_data;
+ struct device_node *np = dev->of_node;
int ret;
- aic32x4 = devm_kzalloc(&i2c->dev, sizeof(struct aic32x4_priv),
+ if (IS_ERR(regmap))
+ return PTR_ERR(regmap);
+
+ aic32x4 = devm_kzalloc(dev, sizeof(struct aic32x4_priv),
GFP_KERNEL);
if (aic32x4 == NULL)
return -ENOMEM;
- aic32x4->regmap = devm_regmap_init_i2c(i2c, &aic32x4_regmap);
- if (IS_ERR(aic32x4->regmap))
- return PTR_ERR(aic32x4->regmap);
-
- i2c_set_clientdata(i2c, aic32x4);
+ dev_set_drvdata(dev, aic32x4);
if (pdata) {
aic32x4->power_cfg = pdata->power_cfg;
@@ -804,7 +940,7 @@ static int aic32x4_i2c_probe(struct i2c_client *i2c,
} else if (np) {
ret = aic32x4_parse_dt(aic32x4, np);
if (ret) {
- dev_err(&i2c->dev, "Failed to parse DT node\n");
+ dev_err(dev, "Failed to parse DT node\n");
return ret;
}
} else {
@@ -814,71 +950,48 @@ static int aic32x4_i2c_probe(struct i2c_client *i2c,
aic32x4->rstn_gpio = -1;
}
- aic32x4->mclk = devm_clk_get(&i2c->dev, "mclk");
+ aic32x4->mclk = devm_clk_get(dev, "mclk");
if (IS_ERR(aic32x4->mclk)) {
- dev_err(&i2c->dev, "Failed getting the mclk. The current implementation does not support the usage of this codec without mclk\n");
+ dev_err(dev, "Failed getting the mclk. The current implementation does not support the usage of this codec without mclk\n");
return PTR_ERR(aic32x4->mclk);
}
if (gpio_is_valid(aic32x4->rstn_gpio)) {
- ret = devm_gpio_request_one(&i2c->dev, aic32x4->rstn_gpio,
+ ret = devm_gpio_request_one(dev, aic32x4->rstn_gpio,
GPIOF_OUT_INIT_LOW, "tlv320aic32x4 rstn");
if (ret != 0)
return ret;
}
- ret = aic32x4_setup_regulators(&i2c->dev, aic32x4);
+ ret = aic32x4_setup_regulators(dev, aic32x4);
if (ret) {
- dev_err(&i2c->dev, "Failed to setup regulators\n");
+ dev_err(dev, "Failed to setup regulators\n");
return ret;
}
- ret = snd_soc_register_codec(&i2c->dev,
+ ret = snd_soc_register_codec(dev,
&soc_codec_dev_aic32x4, &aic32x4_dai, 1);
if (ret) {
- dev_err(&i2c->dev, "Failed to register codec\n");
+ dev_err(dev, "Failed to register codec\n");
aic32x4_disable_regulators(aic32x4);
return ret;
}
- i2c_set_clientdata(i2c, aic32x4);
-
return 0;
}
+EXPORT_SYMBOL(aic32x4_probe);
-static int aic32x4_i2c_remove(struct i2c_client *client)
+int aic32x4_remove(struct device *dev)
{
- struct aic32x4_priv *aic32x4 = i2c_get_clientdata(client);
+ struct aic32x4_priv *aic32x4 = dev_get_drvdata(dev);
aic32x4_disable_regulators(aic32x4);
- snd_soc_unregister_codec(&client->dev);
+ snd_soc_unregister_codec(dev);
+
return 0;
}
-
-static const struct i2c_device_id aic32x4_i2c_id[] = {
- { "tlv320aic32x4", 0 },
- { }
-};
-MODULE_DEVICE_TABLE(i2c, aic32x4_i2c_id);
-
-static const struct of_device_id aic32x4_of_id[] = {
- { .compatible = "ti,tlv320aic32x4", },
- { /* senitel */ }
-};
-MODULE_DEVICE_TABLE(of, aic32x4_of_id);
-
-static struct i2c_driver aic32x4_i2c_driver = {
- .driver = {
- .name = "tlv320aic32x4",
- .of_match_table = aic32x4_of_id,
- },
- .probe = aic32x4_i2c_probe,
- .remove = aic32x4_i2c_remove,
- .id_table = aic32x4_i2c_id,
-};
-
-module_i2c_driver(aic32x4_i2c_driver);
+EXPORT_SYMBOL(aic32x4_remove);
MODULE_DESCRIPTION("ASoC tlv320aic32x4 codec driver");
MODULE_AUTHOR("Javier Martin <javier.martin@vista-silicon.com>");
diff --git a/sound/soc/codecs/tlv320aic32x4.h b/sound/soc/codecs/tlv320aic32x4.h
index 995f033a855d..a197dd51addc 100644
--- a/sound/soc/codecs/tlv320aic32x4.h
+++ b/sound/soc/codecs/tlv320aic32x4.h
@@ -10,6 +10,13 @@
#ifndef _TLV320AIC32X4_H
#define _TLV320AIC32X4_H
+struct device;
+struct regmap_config;
+
+extern const struct regmap_config aic32x4_regmap_config;
+int aic32x4_probe(struct device *dev, struct regmap *regmap);
+int aic32x4_remove(struct device *dev);
+
/* tlv320aic32x4 register space (in decimal to match datasheet) */
#define AIC32X4_PAGE1 128
diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index bc3de2e844e6..1f7081043566 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -824,7 +824,7 @@ static int twl6040_set_bias_level(struct snd_soc_codec *codec,
{
struct twl6040 *twl6040 = codec->control_data;
struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
- int ret;
+ int ret = 0;
switch (level) {
case SND_SOC_BIAS_ON:
@@ -832,12 +832,16 @@ static int twl6040_set_bias_level(struct snd_soc_codec *codec,
case SND_SOC_BIAS_PREPARE:
break;
case SND_SOC_BIAS_STANDBY:
- if (priv->codec_powered)
+ if (priv->codec_powered) {
+ /* Select low power PLL in standby */
+ ret = twl6040_set_pll(twl6040, TWL6040_SYSCLK_SEL_LPPLL,
+ 32768, 19200000);
break;
+ }
ret = twl6040_power(twl6040, 1);
if (ret)
- return ret;
+ break;
priv->codec_powered = 1;
@@ -853,7 +857,7 @@ static int twl6040_set_bias_level(struct snd_soc_codec *codec,
break;
}
- return 0;
+ return ret;
}
static int twl6040_startup(struct snd_pcm_substream *substream,
@@ -983,9 +987,9 @@ static void twl6040_mute_path(struct snd_soc_codec *codec, enum twl6040_dai_id i
if (mute) {
/* Power down drivers and DACs */
hflctl &= ~(TWL6040_HFDACENA | TWL6040_HFPGAENA |
- TWL6040_HFDRVENA);
+ TWL6040_HFDRVENA | TWL6040_HFSWENA);
hfrctl &= ~(TWL6040_HFDACENA | TWL6040_HFPGAENA |
- TWL6040_HFDRVENA);
+ TWL6040_HFDRVENA | TWL6040_HFSWENA);
}
twl6040_reg_write(twl6040, TWL6040_REG_HFLCTL, hflctl);
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index a8b3e3f701f9..da60e3fe5ee7 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -1955,11 +1955,16 @@ err_adsp2_codec_probe:
static int wm5102_codec_remove(struct snd_soc_codec *codec)
{
struct wm5102_priv *priv = snd_soc_codec_get_drvdata(codec);
+ struct arizona *arizona = priv->core.arizona;
wm_adsp2_codec_remove(&priv->core.adsp[0], codec);
priv->core.arizona->dapm = NULL;
+ arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, priv);
+
+ arizona_free_spk(codec);
+
return 0;
}
@@ -2093,10 +2098,14 @@ static int wm5102_probe(struct platform_device *pdev)
static int wm5102_remove(struct platform_device *pdev)
{
+ struct wm5102_priv *wm5102 = platform_get_drvdata(pdev);
+
snd_soc_unregister_platform(&pdev->dev);
snd_soc_unregister_codec(&pdev->dev);
pm_runtime_disable(&pdev->dev);
+ wm_adsp2_remove(&wm5102->core.adsp[0]);
+
return 0;
}
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index 83ba70fe16e6..b5820e4d5471 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -2298,6 +2298,8 @@ static int wm5110_codec_remove(struct snd_soc_codec *codec)
arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, priv);
+ arizona_free_spk(codec);
+
return 0;
}
@@ -2435,10 +2437,16 @@ static int wm5110_probe(struct platform_device *pdev)
static int wm5110_remove(struct platform_device *pdev)
{
+ struct wm5110_priv *wm5110 = platform_get_drvdata(pdev);
+ int i;
+
snd_soc_unregister_platform(&pdev->dev);
snd_soc_unregister_codec(&pdev->dev);
pm_runtime_disable(&pdev->dev);
+ for (i = 0; i < WM5110_NUM_ADSP; i++)
+ wm_adsp2_remove(&wm5110->core.adsp[i]);
+
return 0;
}
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 88223608a33f..720a14e0687d 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -2471,7 +2471,7 @@ static void wm8962_configure_bclk(struct snd_soc_codec *codec)
break;
default:
dev_warn(codec->dev, "Unknown DSPCLK divisor read back\n");
- dspclk = wm8962->sysclk;
+ dspclk = wm8962->sysclk_rate;
}
dev_dbg(codec->dev, "DSPCLK is %dHz, BCLK %d\n", dspclk, wm8962->bclk);
diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c
index 52d766efe14f..6b0785b5a5c5 100644
--- a/sound/soc/codecs/wm8997.c
+++ b/sound/soc/codecs/wm8997.c
@@ -1072,6 +1072,8 @@ static int wm8997_codec_remove(struct snd_soc_codec *codec)
priv->core.arizona->dapm = NULL;
+ arizona_free_spk(codec);
+
return 0;
}
diff --git a/sound/soc/codecs/wm8998.c b/sound/soc/codecs/wm8998.c
index 012396074a8a..449f66636205 100644
--- a/sound/soc/codecs/wm8998.c
+++ b/sound/soc/codecs/wm8998.c
@@ -1324,6 +1324,8 @@ static int wm8998_codec_remove(struct snd_soc_codec *codec)
priv->core.arizona->dapm = NULL;
+ arizona_free_spk(codec);
+
return 0;
}
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index d3b1cb15e7f0..a07bd7c2c587 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -160,6 +160,8 @@
#define ADSP2_RAM_RDY_SHIFT 0
#define ADSP2_RAM_RDY_WIDTH 1
+#define ADSP_MAX_STD_CTRL_SIZE 512
+
struct wm_adsp_buf {
struct list_head list;
void *buf;
@@ -271,8 +273,11 @@ struct wm_adsp_buffer {
__be32 words_written[2]; /* total words written (64 bit) */
};
+struct wm_adsp_compr;
+
struct wm_adsp_compr_buf {
struct wm_adsp *dsp;
+ struct wm_adsp_compr *compr;
struct wm_adsp_buffer_region *regions;
u32 host_buf_ptr;
@@ -435,6 +440,7 @@ struct wm_coeff_ctl {
size_t len;
unsigned int set:1;
struct snd_kcontrol *kcontrol;
+ struct soc_bytes_ext bytes_ext;
unsigned int flags;
};
@@ -711,10 +717,17 @@ static void wm_adsp2_show_fw_status(struct wm_adsp *dsp)
be16_to_cpu(scratch[3]));
}
+static inline struct wm_coeff_ctl *bytes_ext_to_ctl(struct soc_bytes_ext *ext)
+{
+ return container_of(ext, struct wm_coeff_ctl, bytes_ext);
+}
+
static int wm_coeff_info(struct snd_kcontrol *kctl,
struct snd_ctl_elem_info *uinfo)
{
- struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kctl->private_value;
+ struct soc_bytes_ext *bytes_ext =
+ (struct soc_bytes_ext *)kctl->private_value;
+ struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext);
uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES;
uinfo->count = ctl->len;
@@ -763,7 +776,9 @@ static int wm_coeff_write_control(struct wm_coeff_ctl *ctl,
static int wm_coeff_put(struct snd_kcontrol *kctl,
struct snd_ctl_elem_value *ucontrol)
{
- struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kctl->private_value;
+ struct soc_bytes_ext *bytes_ext =
+ (struct soc_bytes_ext *)kctl->private_value;
+ struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext);
char *p = ucontrol->value.bytes.data;
int ret = 0;
@@ -780,6 +795,29 @@ static int wm_coeff_put(struct snd_kcontrol *kctl,
return ret;
}
+static int wm_coeff_tlv_put(struct snd_kcontrol *kctl,
+ const unsigned int __user *bytes, unsigned int size)
+{
+ struct soc_bytes_ext *bytes_ext =
+ (struct soc_bytes_ext *)kctl->private_value;
+ struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext);
+ int ret = 0;
+
+ mutex_lock(&ctl->dsp->pwr_lock);
+
+ if (copy_from_user(ctl->cache, bytes, size)) {
+ ret = -EFAULT;
+ } else {
+ ctl->set = 1;
+ if (ctl->enabled)
+ ret = wm_coeff_write_control(ctl, ctl->cache, size);
+ }
+
+ mutex_unlock(&ctl->dsp->pwr_lock);
+
+ return ret;
+}
+
static int wm_coeff_read_control(struct wm_coeff_ctl *ctl,
void *buf, size_t len)
{
@@ -822,7 +860,9 @@ static int wm_coeff_read_control(struct wm_coeff_ctl *ctl,
static int wm_coeff_get(struct snd_kcontrol *kctl,
struct snd_ctl_elem_value *ucontrol)
{
- struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kctl->private_value;
+ struct soc_bytes_ext *bytes_ext =
+ (struct soc_bytes_ext *)kctl->private_value;
+ struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext);
char *p = ucontrol->value.bytes.data;
int ret = 0;
@@ -845,12 +885,72 @@ static int wm_coeff_get(struct snd_kcontrol *kctl,
return ret;
}
+static int wm_coeff_tlv_get(struct snd_kcontrol *kctl,
+ unsigned int __user *bytes, unsigned int size)
+{
+ struct soc_bytes_ext *bytes_ext =
+ (struct soc_bytes_ext *)kctl->private_value;
+ struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext);
+ int ret = 0;
+
+ mutex_lock(&ctl->dsp->pwr_lock);
+
+ if (ctl->flags & WMFW_CTL_FLAG_VOLATILE) {
+ if (ctl->enabled)
+ ret = wm_coeff_read_control(ctl, ctl->cache, size);
+ else
+ ret = -EPERM;
+ } else {
+ if (!ctl->flags && ctl->enabled)
+ ret = wm_coeff_read_control(ctl, ctl->cache, size);
+ }
+
+ if (!ret && copy_to_user(bytes, ctl->cache, size))
+ ret = -EFAULT;
+
+ mutex_unlock(&ctl->dsp->pwr_lock);
+
+ return ret;
+}
+
struct wmfw_ctl_work {
struct wm_adsp *dsp;
struct wm_coeff_ctl *ctl;
struct work_struct work;
};
+static unsigned int wmfw_convert_flags(unsigned int in, unsigned int len)
+{
+ unsigned int out, rd, wr, vol;
+
+ if (len > ADSP_MAX_STD_CTRL_SIZE) {
+ rd = SNDRV_CTL_ELEM_ACCESS_TLV_READ;
+ wr = SNDRV_CTL_ELEM_ACCESS_TLV_WRITE;
+ vol = SNDRV_CTL_ELEM_ACCESS_VOLATILE;
+
+ out = SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK;
+ } else {
+ rd = SNDRV_CTL_ELEM_ACCESS_READ;
+ wr = SNDRV_CTL_ELEM_ACCESS_WRITE;
+ vol = SNDRV_CTL_ELEM_ACCESS_VOLATILE;
+
+ out = 0;
+ }
+
+ if (in) {
+ if (in & WMFW_CTL_FLAG_READABLE)
+ out |= rd;
+ if (in & WMFW_CTL_FLAG_WRITEABLE)
+ out |= wr;
+ if (in & WMFW_CTL_FLAG_VOLATILE)
+ out |= vol;
+ } else {
+ out |= rd | wr | vol;
+ }
+
+ return out;
+}
+
static int wmfw_add_ctl(struct wm_adsp *dsp, struct wm_coeff_ctl *ctl)
{
struct snd_kcontrol_new *kcontrol;
@@ -868,19 +968,15 @@ static int wmfw_add_ctl(struct wm_adsp *dsp, struct wm_coeff_ctl *ctl)
kcontrol->info = wm_coeff_info;
kcontrol->get = wm_coeff_get;
kcontrol->put = wm_coeff_put;
- kcontrol->private_value = (unsigned long)ctl;
+ kcontrol->iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ kcontrol->tlv.c = snd_soc_bytes_tlv_callback;
+ kcontrol->private_value = (unsigned long)&ctl->bytes_ext;
- if (ctl->flags) {
- if (ctl->flags & WMFW_CTL_FLAG_WRITEABLE)
- kcontrol->access |= SNDRV_CTL_ELEM_ACCESS_WRITE;
- if (ctl->flags & WMFW_CTL_FLAG_READABLE)
- kcontrol->access |= SNDRV_CTL_ELEM_ACCESS_READ;
- if (ctl->flags & WMFW_CTL_FLAG_VOLATILE)
- kcontrol->access |= SNDRV_CTL_ELEM_ACCESS_VOLATILE;
- } else {
- kcontrol->access = SNDRV_CTL_ELEM_ACCESS_READWRITE;
- kcontrol->access |= SNDRV_CTL_ELEM_ACCESS_VOLATILE;
- }
+ ctl->bytes_ext.max = ctl->len;
+ ctl->bytes_ext.get = wm_coeff_tlv_get;
+ ctl->bytes_ext.put = wm_coeff_tlv_put;
+
+ kcontrol->access = wmfw_convert_flags(ctl->flags, ctl->len);
ret = snd_soc_add_card_controls(dsp->card, kcontrol, 1);
if (ret < 0)
@@ -944,6 +1040,13 @@ static void wm_adsp_ctl_work(struct work_struct *work)
kfree(ctl_work);
}
+static void wm_adsp_free_ctl_blk(struct wm_coeff_ctl *ctl)
+{
+ kfree(ctl->cache);
+ kfree(ctl->name);
+ kfree(ctl);
+}
+
static int wm_adsp_create_control(struct wm_adsp *dsp,
const struct wm_adsp_alg_region *alg_region,
unsigned int offset, unsigned int len,
@@ -1032,11 +1135,6 @@ static int wm_adsp_create_control(struct wm_adsp *dsp,
ctl->flags = flags;
ctl->offset = offset;
- if (len > 512) {
- adsp_warn(dsp, "Truncating control %s from %d\n",
- ctl->name, len);
- len = 512;
- }
ctl->len = len;
ctl->cache = kzalloc(ctl->len, GFP_KERNEL);
if (!ctl->cache) {
@@ -1564,6 +1662,19 @@ static struct wm_adsp_alg_region *wm_adsp_create_region(struct wm_adsp *dsp,
return alg_region;
}
+static void wm_adsp_free_alg_regions(struct wm_adsp *dsp)
+{
+ struct wm_adsp_alg_region *alg_region;
+
+ while (!list_empty(&dsp->alg_regions)) {
+ alg_region = list_first_entry(&dsp->alg_regions,
+ struct wm_adsp_alg_region,
+ list);
+ list_del(&alg_region->list);
+ kfree(alg_region);
+ }
+}
+
static int wm_adsp1_setup_algs(struct wm_adsp *dsp)
{
struct wmfw_adsp1_id_hdr adsp1_id;
@@ -1994,7 +2105,6 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w,
struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec);
struct wm_adsp *dsp = &dsps[w->shift];
- struct wm_adsp_alg_region *alg_region;
struct wm_coeff_ctl *ctl;
int ret;
unsigned int val;
@@ -2074,13 +2184,8 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w,
list_for_each_entry(ctl, &dsp->ctl_list, list)
ctl->enabled = 0;
- while (!list_empty(&dsp->alg_regions)) {
- alg_region = list_first_entry(&dsp->alg_regions,
- struct wm_adsp_alg_region,
- list);
- list_del(&alg_region->list);
- kfree(alg_region);
- }
+
+ wm_adsp_free_alg_regions(dsp);
break;
default:
@@ -2222,7 +2327,6 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w,
struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec);
struct wm_adsp *dsp = &dsps[w->shift];
- struct wm_adsp_alg_region *alg_region;
struct wm_coeff_ctl *ctl;
int ret;
@@ -2240,9 +2344,13 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w,
if (ret != 0)
goto err;
+ mutex_lock(&dsp->pwr_lock);
+
if (wm_adsp_fw[dsp->fw].num_caps != 0)
ret = wm_adsp_buffer_init(dsp);
+ mutex_unlock(&dsp->pwr_lock);
+
break;
case SND_SOC_DAPM_PRE_PMD:
@@ -2269,13 +2377,7 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w,
list_for_each_entry(ctl, &dsp->ctl_list, list)
ctl->enabled = 0;
- while (!list_empty(&dsp->alg_regions)) {
- alg_region = list_first_entry(&dsp->alg_regions,
- struct wm_adsp_alg_region,
- list);
- list_del(&alg_region->list);
- kfree(alg_region);
- }
+ wm_adsp_free_alg_regions(dsp);
if (wm_adsp_fw[dsp->fw].num_caps != 0)
wm_adsp_buffer_free(dsp);
@@ -2340,6 +2442,54 @@ int wm_adsp2_init(struct wm_adsp *dsp)
}
EXPORT_SYMBOL_GPL(wm_adsp2_init);
+void wm_adsp2_remove(struct wm_adsp *dsp)
+{
+ struct wm_coeff_ctl *ctl;
+
+ while (!list_empty(&dsp->ctl_list)) {
+ ctl = list_first_entry(&dsp->ctl_list, struct wm_coeff_ctl,
+ list);
+ list_del(&ctl->list);
+ wm_adsp_free_ctl_blk(ctl);
+ }
+}
+EXPORT_SYMBOL_GPL(wm_adsp2_remove);
+
+static inline int wm_adsp_compr_attached(struct wm_adsp_compr *compr)
+{
+ return compr->buf != NULL;
+}
+
+static int wm_adsp_compr_attach(struct wm_adsp_compr *compr)
+{
+ /*
+ * Note this will be more complex once each DSP can support multiple
+ * streams
+ */
+ if (!compr->dsp->buffer)
+ return -EINVAL;
+
+ compr->buf = compr->dsp->buffer;
+ compr->buf->compr = compr;
+
+ return 0;
+}
+
+static void wm_adsp_compr_detach(struct wm_adsp_compr *compr)
+{
+ if (!compr)
+ return;
+
+ /* Wake the poll so it can see buffer is no longer attached */
+ if (compr->stream)
+ snd_compr_fragment_elapsed(compr->stream);
+
+ if (wm_adsp_compr_attached(compr)) {
+ compr->buf->compr = NULL;
+ compr->buf = NULL;
+ }
+}
+
int wm_adsp_compr_open(struct wm_adsp *dsp, struct snd_compr_stream *stream)
{
struct wm_adsp_compr *compr;
@@ -2393,6 +2543,7 @@ int wm_adsp_compr_free(struct snd_compr_stream *stream)
mutex_lock(&dsp->pwr_lock);
+ wm_adsp_compr_detach(compr);
dsp->compr = NULL;
kfree(compr->raw_buf);
@@ -2689,6 +2840,8 @@ err_buffer:
static int wm_adsp_buffer_free(struct wm_adsp *dsp)
{
if (dsp->buffer) {
+ wm_adsp_compr_detach(dsp->buffer->compr);
+
kfree(dsp->buffer->regions);
kfree(dsp->buffer);
@@ -2698,25 +2851,6 @@ static int wm_adsp_buffer_free(struct wm_adsp *dsp)
return 0;
}
-static inline int wm_adsp_compr_attached(struct wm_adsp_compr *compr)
-{
- return compr->buf != NULL;
-}
-
-static int wm_adsp_compr_attach(struct wm_adsp_compr *compr)
-{
- /*
- * Note this will be more complex once each DSP can support multiple
- * streams
- */
- if (!compr->dsp->buffer)
- return -EINVAL;
-
- compr->buf = compr->dsp->buffer;
-
- return 0;
-}
-
int wm_adsp_compr_trigger(struct snd_compr_stream *stream, int cmd)
{
struct wm_adsp_compr *compr = stream->runtime->private_data;
@@ -2805,21 +2939,41 @@ static int wm_adsp_buffer_update_avail(struct wm_adsp_compr_buf *buf)
avail += wm_adsp_buffer_size(buf);
adsp_dbg(buf->dsp, "readindex=0x%x, writeindex=0x%x, avail=%d\n",
- buf->read_index, write_index, avail);
+ buf->read_index, write_index, avail * WM_ADSP_DATA_WORD_SIZE);
buf->avail = avail;
return 0;
}
+static int wm_adsp_buffer_get_error(struct wm_adsp_compr_buf *buf)
+{
+ int ret;
+
+ ret = wm_adsp_buffer_read(buf, HOST_BUFFER_FIELD(error), &buf->error);
+ if (ret < 0) {
+ adsp_err(buf->dsp, "Failed to check buffer error: %d\n", ret);
+ return ret;
+ }
+ if (buf->error != 0) {
+ adsp_err(buf->dsp, "Buffer error occurred: %d\n", buf->error);
+ return -EIO;
+ }
+
+ return 0;
+}
+
int wm_adsp_compr_handle_irq(struct wm_adsp *dsp)
{
- struct wm_adsp_compr_buf *buf = dsp->buffer;
- struct wm_adsp_compr *compr = dsp->compr;
+ struct wm_adsp_compr_buf *buf;
+ struct wm_adsp_compr *compr;
int ret = 0;
mutex_lock(&dsp->pwr_lock);
+ buf = dsp->buffer;
+ compr = dsp->compr;
+
if (!buf) {
ret = -ENODEV;
goto out;
@@ -2827,16 +2981,9 @@ int wm_adsp_compr_handle_irq(struct wm_adsp *dsp)
adsp_dbg(dsp, "Handling buffer IRQ\n");
- ret = wm_adsp_buffer_read(buf, HOST_BUFFER_FIELD(error), &buf->error);
- if (ret < 0) {
- adsp_err(dsp, "Failed to check buffer error: %d\n", ret);
- goto out;
- }
- if (buf->error != 0) {
- adsp_err(dsp, "Buffer error occurred: %d\n", buf->error);
- ret = -EIO;
- goto out;
- }
+ ret = wm_adsp_buffer_get_error(buf);
+ if (ret < 0)
+ goto out_notify; /* Wake poll to report error */
ret = wm_adsp_buffer_read(buf, HOST_BUFFER_FIELD(irq_count),
&buf->irq_count);
@@ -2851,6 +2998,7 @@ int wm_adsp_compr_handle_irq(struct wm_adsp *dsp)
goto out;
}
+out_notify:
if (compr && compr->stream)
snd_compr_fragment_elapsed(compr->stream);
@@ -2879,14 +3027,16 @@ int wm_adsp_compr_pointer(struct snd_compr_stream *stream,
struct snd_compr_tstamp *tstamp)
{
struct wm_adsp_compr *compr = stream->runtime->private_data;
- struct wm_adsp_compr_buf *buf = compr->buf;
struct wm_adsp *dsp = compr->dsp;
+ struct wm_adsp_compr_buf *buf;
int ret = 0;
adsp_dbg(dsp, "Pointer request\n");
mutex_lock(&dsp->pwr_lock);
+ buf = compr->buf;
+
if (!compr->buf) {
ret = -ENXIO;
goto out;
@@ -2909,6 +3059,10 @@ int wm_adsp_compr_pointer(struct snd_compr_stream *stream,
* DSP to inform us once a whole fragment is available.
*/
if (buf->avail < wm_adsp_compr_frag_words(compr)) {
+ ret = wm_adsp_buffer_get_error(buf);
+ if (ret < 0)
+ goto out;
+
ret = wm_adsp_buffer_reenable_irq(buf);
if (ret < 0) {
adsp_err(dsp,
diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h
index b61cb57e600f..feb61e2c4bb4 100644
--- a/sound/soc/codecs/wm_adsp.h
+++ b/sound/soc/codecs/wm_adsp.h
@@ -92,6 +92,7 @@ extern const struct snd_kcontrol_new wm_adsp_fw_controls[];
int wm_adsp1_init(struct wm_adsp *dsp);
int wm_adsp2_init(struct wm_adsp *dsp);
+void wm_adsp2_remove(struct wm_adsp *dsp);
int wm_adsp2_codec_probe(struct wm_adsp *dsp, struct snd_soc_codec *codec);
int wm_adsp2_codec_remove(struct wm_adsp *dsp, struct snd_soc_codec *codec);
int wm_adsp1_event(struct snd_soc_dapm_widget *w,
diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig
index 50ca291cc225..6b732d8e5896 100644
--- a/sound/soc/davinci/Kconfig
+++ b/sound/soc/davinci/Kconfig
@@ -16,7 +16,11 @@ config SND_EDMA_SOC
- DRA7xx family
config SND_DAVINCI_SOC_I2S
- tristate
+ tristate "DaVinci Multichannel Buffered Serial Port (McBSP) support"
+ depends on SND_EDMA_SOC
+ help
+ Say Y or M here if you want to have support for McBSP IP found in
+ Texas Instruments DaVinci DA850 SoCs.
config SND_DAVINCI_SOC_MCASP
tristate "Multichannel Audio Serial Port (McASP) support"
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index ec98548a5fc9..384961651904 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -4,9 +4,15 @@
* Author: Vladimir Barinov, <vbarinov@embeddedalley.com>
* Copyright: (C) 2007 MontaVista Software, Inc., <source@mvista.com>
*
+ * DT support (c) 2016 Petr Kulhavy, Barix AG <petr@barix.com>
+ * based on davinci-mcasp.c DT support
+ *
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
+ *
+ * TODO:
+ * on DA850 implement HW FIFOs instead of DMA into DXR and DRR registers
*/
#include <linux/init.h>
@@ -650,13 +656,24 @@ static const struct snd_soc_component_driver davinci_i2s_component = {
static int davinci_i2s_probe(struct platform_device *pdev)
{
+ struct snd_dmaengine_dai_dma_data *dma_data;
struct davinci_mcbsp_dev *dev;
struct resource *mem, *res;
void __iomem *io_base;
int *dma;
int ret;
- mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ mem = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu");
+ if (!mem) {
+ dev_warn(&pdev->dev,
+ "\"mpu\" mem resource not found, using index 0\n");
+ mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ if (!mem) {
+ dev_err(&pdev->dev, "no mem resource?\n");
+ return -ENODEV;
+ }
+ }
+
io_base = devm_ioremap_resource(&pdev->dev, mem);
if (IS_ERR(io_base))
return PTR_ERR(io_base);
@@ -666,39 +683,43 @@ static int davinci_i2s_probe(struct platform_device *pdev)
if (!dev)
return -ENOMEM;
- dev->clk = clk_get(&pdev->dev, NULL);
- if (IS_ERR(dev->clk))
- return -ENODEV;
- clk_enable(dev->clk);
-
dev->base = io_base;
- dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].addr =
- (dma_addr_t)(mem->start + DAVINCI_MCBSP_DXR_REG);
+ /* setup DMA, first TX, then RX */
+ dma_data = &dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK];
+ dma_data->addr = (dma_addr_t)(mem->start + DAVINCI_MCBSP_DXR_REG);
- dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].addr =
- (dma_addr_t)(mem->start + DAVINCI_MCBSP_DRR_REG);
-
- /* first TX, then RX */
res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
- if (!res) {
- dev_err(&pdev->dev, "no DMA resource\n");
- ret = -ENXIO;
- goto err_release_clk;
+ if (res) {
+ dma = &dev->dma_request[SNDRV_PCM_STREAM_PLAYBACK];
+ *dma = res->start;
+ dma_data->filter_data = dma;
+ } else if (IS_ENABLED(CONFIG_OF) && pdev->dev.of_node) {
+ dma_data->filter_data = "tx";
+ } else {
+ dev_err(&pdev->dev, "Missing DMA tx resource\n");
+ return -ENODEV;
}
- dma = &dev->dma_request[SNDRV_PCM_STREAM_PLAYBACK];
- *dma = res->start;
- dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].filter_data = dma;
+
+ dma_data = &dev->dma_data[SNDRV_PCM_STREAM_CAPTURE];
+ dma_data->addr = (dma_addr_t)(mem->start + DAVINCI_MCBSP_DRR_REG);
res = platform_get_resource(pdev, IORESOURCE_DMA, 1);
- if (!res) {
- dev_err(&pdev->dev, "no DMA resource\n");
- ret = -ENXIO;
- goto err_release_clk;
+ if (res) {
+ dma = &dev->dma_request[SNDRV_PCM_STREAM_CAPTURE];
+ *dma = res->start;
+ dma_data->filter_data = dma;
+ } else if (IS_ENABLED(CONFIG_OF) && pdev->dev.of_node) {
+ dma_data->filter_data = "rx";
+ } else {
+ dev_err(&pdev->dev, "Missing DMA rx resource\n");
+ return -ENODEV;
}
- dma = &dev->dma_request[SNDRV_PCM_STREAM_CAPTURE];
- *dma = res->start;
- dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].filter_data = dma;
+
+ dev->clk = clk_get(&pdev->dev, NULL);
+ if (IS_ERR(dev->clk))
+ return -ENODEV;
+ clk_enable(dev->clk);
dev->dev = &pdev->dev;
dev_set_drvdata(&pdev->dev, dev);
@@ -737,11 +758,18 @@ static int davinci_i2s_remove(struct platform_device *pdev)
return 0;
}
+static const struct of_device_id davinci_i2s_match[] = {
+ { .compatible = "ti,da850-mcbsp" },
+ {},
+};
+MODULE_DEVICE_TABLE(of, davinci_i2s_match);
+
static struct platform_driver davinci_mcbsp_driver = {
.probe = davinci_i2s_probe,
.remove = davinci_i2s_remove,
.driver = {
.name = "davinci-mcbsp",
+ .of_match_table = of_match_ptr(davinci_i2s_match),
},
};
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index e1324989bd6b..0f66fda2c772 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -489,7 +489,7 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
mcasp_clr_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, AFSRE);
mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG,
- ACLKX | AHCLKX | AFSX | ACLKR | AHCLKR | AFSR);
+ ACLKX | AFSX | ACLKR | AHCLKR | AFSR);
mcasp->bclk_master = 0;
break;
default:
@@ -540,21 +540,19 @@ out:
return ret;
}
-static int __davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id,
+static int __davinci_mcasp_set_clkdiv(struct davinci_mcasp *mcasp, int div_id,
int div, bool explicit)
{
- struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai);
-
pm_runtime_get_sync(mcasp->dev);
switch (div_id) {
- case 0: /* MCLK divider */
+ case MCASP_CLKDIV_AUXCLK: /* MCLK divider */
mcasp_mod_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG,
AHCLKXDIV(div - 1), AHCLKXDIV_MASK);
mcasp_mod_bits(mcasp, DAVINCI_MCASP_AHCLKRCTL_REG,
AHCLKRDIV(div - 1), AHCLKRDIV_MASK);
break;
- case 1: /* BCLK divider */
+ case MCASP_CLKDIV_BCLK: /* BCLK divider */
mcasp_mod_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG,
ACLKXDIV(div - 1), ACLKXDIV_MASK);
mcasp_mod_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG,
@@ -563,7 +561,8 @@ static int __davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id,
mcasp->bclk_div = div;
break;
- case 2: /*
+ case MCASP_CLKDIV_BCLK_FS_RATIO:
+ /*
* BCLK/LRCLK ratio descries how many bit-clock cycles
* fit into one frame. The clock ratio is given for a
* full period of data (for I2S format both left and
@@ -591,7 +590,9 @@ static int __davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id,
static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id,
int div)
{
- return __davinci_mcasp_set_clkdiv(dai, div_id, div, 1);
+ struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai);
+
+ return __davinci_mcasp_set_clkdiv(mcasp, div_id, div, 1);
}
static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id,
@@ -999,27 +1000,53 @@ static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp,
}
static int davinci_mcasp_calc_clk_div(struct davinci_mcasp *mcasp,
- unsigned int bclk_freq,
- int *error_ppm)
+ unsigned int bclk_freq, bool set)
{
- int div = mcasp->sysclk_freq / bclk_freq;
- int rem = mcasp->sysclk_freq % bclk_freq;
+ int error_ppm;
+ unsigned int sysclk_freq = mcasp->sysclk_freq;
+ u32 reg = mcasp_get_reg(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG);
+ int div = sysclk_freq / bclk_freq;
+ int rem = sysclk_freq % bclk_freq;
+ int aux_div = 1;
+
+ if (div > (ACLKXDIV_MASK + 1)) {
+ if (reg & AHCLKXE) {
+ aux_div = div / (ACLKXDIV_MASK + 1);
+ if (div % (ACLKXDIV_MASK + 1))
+ aux_div++;
+
+ sysclk_freq /= aux_div;
+ div = sysclk_freq / bclk_freq;
+ rem = sysclk_freq % bclk_freq;
+ } else if (set) {
+ dev_warn(mcasp->dev, "Too fast reference clock (%u)\n",
+ sysclk_freq);
+ }
+ }
if (rem != 0) {
if (div == 0 ||
- ((mcasp->sysclk_freq / div) - bclk_freq) >
- (bclk_freq - (mcasp->sysclk_freq / (div+1)))) {
+ ((sysclk_freq / div) - bclk_freq) >
+ (bclk_freq - (sysclk_freq / (div+1)))) {
div++;
rem = rem - bclk_freq;
}
}
- if (error_ppm)
- *error_ppm =
- (div*1000000 + (int)div64_long(1000000LL*rem,
- (int)bclk_freq))
- /div - 1000000;
+ error_ppm = (div*1000000 + (int)div64_long(1000000LL*rem,
+ (int)bclk_freq)) / div - 1000000;
+
+ if (set) {
+ if (error_ppm)
+ dev_info(mcasp->dev, "Sample-rate is off by %d PPM\n",
+ error_ppm);
+
+ __davinci_mcasp_set_clkdiv(mcasp, MCASP_CLKDIV_BCLK, div, 0);
+ if (reg & AHCLKXE)
+ __davinci_mcasp_set_clkdiv(mcasp, MCASP_CLKDIV_AUXCLK,
+ aux_div, 0);
+ }
- return div;
+ return error_ppm;
}
static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
@@ -1044,18 +1071,11 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
int slots = mcasp->tdm_slots;
int rate = params_rate(params);
int sbits = params_width(params);
- int ppm, div;
if (mcasp->slot_width)
sbits = mcasp->slot_width;
- div = davinci_mcasp_calc_clk_div(mcasp, rate*sbits*slots,
- &ppm);
- if (ppm)
- dev_info(mcasp->dev, "Sample-rate is off by %d PPM\n",
- ppm);
-
- __davinci_mcasp_set_clkdiv(cpu_dai, 1, div, 0);
+ davinci_mcasp_calc_clk_div(mcasp, rate * sbits * slots, true);
}
ret = mcasp_common_hw_param(mcasp, substream->stream,
@@ -1166,7 +1186,8 @@ static int davinci_mcasp_hw_rule_rate(struct snd_pcm_hw_params *params,
davinci_mcasp_dai_rates[i];
int ppm;
- davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, &ppm);
+ ppm = davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq,
+ false);
if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) {
if (range.empty) {
range.min = davinci_mcasp_dai_rates[i];
@@ -1205,8 +1226,9 @@ static int davinci_mcasp_hw_rule_format(struct snd_pcm_hw_params *params,
if (rd->mcasp->slot_width)
sbits = rd->mcasp->slot_width;
- davinci_mcasp_calc_clk_div(rd->mcasp, sbits*slots*rate,
- &ppm);
+ ppm = davinci_mcasp_calc_clk_div(rd->mcasp,
+ sbits * slots * rate,
+ false);
if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) {
snd_mask_set(&nfmt, i);
count++;
@@ -1230,11 +1252,15 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
int i, dir;
int tdm_slots = mcasp->tdm_slots;
- if (mcasp->tdm_mask[substream->stream])
- tdm_slots = hweight32(mcasp->tdm_mask[substream->stream]);
+ /* Do not allow more then one stream per direction */
+ if (mcasp->substreams[substream->stream])
+ return -EBUSY;
mcasp->substreams[substream->stream] = substream;
+ if (mcasp->tdm_mask[substream->stream])
+ tdm_slots = hweight32(mcasp->tdm_mask[substream->stream]);
+
if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE)
return 0;
diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h
index a3be108a8c17..1e8787fb3fb7 100644
--- a/sound/soc/davinci/davinci-mcasp.h
+++ b/sound/soc/davinci/davinci-mcasp.h
@@ -306,4 +306,9 @@
#define NUMEVT(x) (((x) & 0xFF) << 8)
#define NUMDMA_MASK (0xFF)
+/* clock divider IDs */
+#define MCASP_CLKDIV_AUXCLK 0 /* HCLK divider from AUXCLK */
+#define MCASP_CLKDIV_BCLK 1 /* BCLK divider from HCLK */
+#define MCASP_CLKDIV_BCLK_FS_RATIO 2 /* to set BCLK FS ration */
+
#endif /* DAVINCI_MCASP_H */
diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c
index bff258d7bcea..0db69b7e9617 100644
--- a/sound/soc/dwc/designware_i2s.c
+++ b/sound/soc/dwc/designware_i2s.c
@@ -100,6 +100,7 @@ struct dw_i2s_dev {
struct device *dev;
u32 ccr;
u32 xfer_resolution;
+ u32 fifo_th;
/* data related to DMA transfers b/w i2s and DMAC */
union dw_i2s_snd_dma_data play_dma_data;
@@ -147,17 +148,18 @@ static inline void i2s_clear_irqs(struct dw_i2s_dev *dev, u32 stream)
static void i2s_start(struct dw_i2s_dev *dev,
struct snd_pcm_substream *substream)
{
+ struct i2s_clk_config_data *config = &dev->config;
u32 i, irq;
i2s_write_reg(dev->i2s_base, IER, 1);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- for (i = 0; i < 4; i++) {
+ for (i = 0; i < (config->chan_nr / 2); i++) {
irq = i2s_read_reg(dev->i2s_base, IMR(i));
i2s_write_reg(dev->i2s_base, IMR(i), irq & ~0x30);
}
i2s_write_reg(dev->i2s_base, ITER, 1);
} else {
- for (i = 0; i < 4; i++) {
+ for (i = 0; i < (config->chan_nr / 2); i++) {
irq = i2s_read_reg(dev->i2s_base, IMR(i));
i2s_write_reg(dev->i2s_base, IMR(i), irq & ~0x03);
}
@@ -231,14 +233,16 @@ static void dw_i2s_config(struct dw_i2s_dev *dev, int stream)
if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
i2s_write_reg(dev->i2s_base, TCR(ch_reg),
dev->xfer_resolution);
- i2s_write_reg(dev->i2s_base, TFCR(ch_reg), 0x02);
+ i2s_write_reg(dev->i2s_base, TFCR(ch_reg),
+ dev->fifo_th - 1);
irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg));
i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x30);
i2s_write_reg(dev->i2s_base, TER(ch_reg), 1);
} else {
i2s_write_reg(dev->i2s_base, RCR(ch_reg),
dev->xfer_resolution);
- i2s_write_reg(dev->i2s_base, RFCR(ch_reg), 0x07);
+ i2s_write_reg(dev->i2s_base, RFCR(ch_reg),
+ dev->fifo_th - 1);
irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg));
i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x03);
i2s_write_reg(dev->i2s_base, RER(ch_reg), 1);
@@ -498,6 +502,7 @@ static int dw_configure_dai(struct dw_i2s_dev *dev,
*/
u32 comp1 = i2s_read_reg(dev->i2s_base, dev->i2s_reg_comp1);
u32 comp2 = i2s_read_reg(dev->i2s_base, dev->i2s_reg_comp2);
+ u32 fifo_depth = 1 << (1 + COMP1_FIFO_DEPTH_GLOBAL(comp1));
u32 idx;
if (dev->capability & DWC_I2S_RECORD &&
@@ -536,6 +541,7 @@ static int dw_configure_dai(struct dw_i2s_dev *dev,
dev->capability |= DW_I2S_SLAVE;
}
+ dev->fifo_th = fifo_depth / 2;
return 0;
}
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index 0754df771e3b..2147994ab46f 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -21,6 +21,8 @@
#include <sound/core.h>
#include <sound/dmaengine_pcm.h>
#include <sound/pcm_params.h>
+#include <linux/mfd/syscon.h>
+#include <linux/mfd/syscon/imx6q-iomuxc-gpr.h>
#include "fsl_sai.h"
#include "imx-pcm.h"
@@ -786,10 +788,12 @@ static int fsl_sai_probe(struct platform_device *pdev)
{
struct device_node *np = pdev->dev.of_node;
struct fsl_sai *sai;
+ struct regmap *gpr;
struct resource *res;
void __iomem *base;
char tmp[8];
int irq, ret, i;
+ int index;
sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL);
if (!sai)
@@ -797,7 +801,8 @@ static int fsl_sai_probe(struct platform_device *pdev)
sai->pdev = pdev;
- if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai"))
+ if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai") ||
+ of_device_is_compatible(pdev->dev.of_node, "fsl,imx6ul-sai"))
sai->sai_on_imx = true;
sai->is_lsb_first = of_property_read_bool(np, "lsb-first");
@@ -877,6 +882,22 @@ static int fsl_sai_probe(struct platform_device *pdev)
fsl_sai_dai.symmetric_samplebits = 0;
}
+ if (of_find_property(np, "fsl,sai-mclk-direction-output", NULL) &&
+ of_device_is_compatible(pdev->dev.of_node, "fsl,imx6ul-sai")) {
+ gpr = syscon_regmap_lookup_by_compatible("fsl,imx6ul-iomuxc-gpr");
+ if (IS_ERR(gpr)) {
+ dev_err(&pdev->dev, "cannot find iomuxc registers\n");
+ return PTR_ERR(gpr);
+ }
+
+ index = of_alias_get_id(np, "sai");
+ if (index < 0)
+ return index;
+
+ regmap_update_bits(gpr, IOMUXC_GPR1, MCLK_DIR(index),
+ MCLK_DIR(index));
+ }
+
sai->dma_params_rx.addr = res->start + FSL_SAI_RDR;
sai->dma_params_tx.addr = res->start + FSL_SAI_TDR;
sai->dma_params_rx.maxburst = FSL_SAI_MAXBURST_RX;
@@ -898,6 +919,7 @@ static int fsl_sai_probe(struct platform_device *pdev)
static const struct of_device_id fsl_sai_ids[] = {
{ .compatible = "fsl,vf610-sai", },
{ .compatible = "fsl,imx6sx-sai", },
+ { .compatible = "fsl,imx6ul-sai", },
{ /* sentinel */ }
};
MODULE_DEVICE_TABLE(of, fsl_sai_ids);
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index ed8de1035cda..632ecc0e3956 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -137,6 +137,7 @@ static bool fsl_ssi_volatile_reg(struct device *dev, unsigned int reg)
case CCSR_SSI_SACDAT:
case CCSR_SSI_SATAG:
case CCSR_SSI_SACCST:
+ case CCSR_SSI_SOR:
return true;
default:
return false;
@@ -261,6 +262,7 @@ struct fsl_ssi_private {
struct fsl_ssi_dbg dbg_stats;
const struct fsl_ssi_soc_data *soc;
+ struct device *dev;
};
/*
@@ -400,6 +402,26 @@ static void fsl_ssi_rxtx_config(struct fsl_ssi_private *ssi_private,
}
/*
+ * Clear RX or TX FIFO to remove samples from the previous
+ * stream session which may be still present in the FIFO and
+ * may introduce bad samples and/or channel slipping.
+ *
+ * Note: The SOR is not documented in recent IMX datasheet, but
+ * is described in IMX51 reference manual at section 56.3.3.15.
+ */
+static void fsl_ssi_fifo_clear(struct fsl_ssi_private *ssi_private,
+ bool is_rx)
+{
+ if (is_rx) {
+ regmap_update_bits(ssi_private->regs, CCSR_SSI_SOR,
+ CCSR_SSI_SOR_RX_CLR, CCSR_SSI_SOR_RX_CLR);
+ } else {
+ regmap_update_bits(ssi_private->regs, CCSR_SSI_SOR,
+ CCSR_SSI_SOR_TX_CLR, CCSR_SSI_SOR_TX_CLR);
+ }
+}
+
+/*
* Calculate the bits that have to be disabled for the current stream that is
* getting disabled. This keeps the bits enabled that are necessary for the
* second stream to work if 'stream_active' is true.
@@ -474,9 +496,11 @@ static void fsl_ssi_config(struct fsl_ssi_private *ssi_private, bool enable,
* (online configuration)
*/
if (enable) {
- regmap_update_bits(regs, CCSR_SSI_SIER, vals->sier, vals->sier);
+ fsl_ssi_fifo_clear(ssi_private, vals->scr & CCSR_SSI_SCR_RE);
+
regmap_update_bits(regs, CCSR_SSI_SRCR, vals->srcr, vals->srcr);
regmap_update_bits(regs, CCSR_SSI_STCR, vals->stcr, vals->stcr);
+ regmap_update_bits(regs, CCSR_SSI_SIER, vals->sier, vals->sier);
} else {
u32 sier;
u32 srcr;
@@ -506,8 +530,40 @@ static void fsl_ssi_config(struct fsl_ssi_private *ssi_private, bool enable,
config_done:
/* Enabling of subunits is done after configuration */
- if (enable)
+ if (enable) {
+ if (ssi_private->use_dma && (vals->scr & CCSR_SSI_SCR_TE)) {
+ /*
+ * Be sure the Tx FIFO is filled when TE is set.
+ * Otherwise, there are some chances to start the
+ * playback with some void samples inserted first,
+ * generating a channel slip.
+ *
+ * First, SSIEN must be set, to let the FIFO be filled.
+ *
+ * Notes:
+ * - Limit this fix to the DMA case until FIQ cases can
+ * be tested.
+ * - Limit the length of the busy loop to not lock the
+ * system too long, even if 1-2 loops are sufficient
+ * in general.
+ */
+ int i;
+ int max_loop = 100;
+ regmap_update_bits(regs, CCSR_SSI_SCR,
+ CCSR_SSI_SCR_SSIEN, CCSR_SSI_SCR_SSIEN);
+ for (i = 0; i < max_loop; i++) {
+ u32 sfcsr;
+ regmap_read(regs, CCSR_SSI_SFCSR, &sfcsr);
+ if (CCSR_SSI_SFCSR_TFCNT0(sfcsr))
+ break;
+ }
+ if (i == max_loop) {
+ dev_err(ssi_private->dev,
+ "Timeout waiting TX FIFO filling\n");
+ }
+ }
regmap_update_bits(regs, CCSR_SSI_SCR, vals->scr, vals->scr);
+ }
}
@@ -670,6 +726,15 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream,
if (IS_ERR(ssi_private->baudclk))
return -EINVAL;
+ /*
+ * Hardware limitation: The bclk rate must be
+ * never greater than 1/5 IPG clock rate
+ */
+ if (freq * 5 > clk_get_rate(ssi_private->clk)) {
+ dev_err(cpu_dai->dev, "bitclk > ipgclk/5\n");
+ return -EINVAL;
+ }
+
baudclk_is_used = ssi_private->baudclk_streams & ~(BIT(substream->stream));
/* It should be already enough to divide clock by setting pm alone */
@@ -686,13 +751,6 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream,
else
clkrate = clk_round_rate(ssi_private->baudclk, tmprate);
- /*
- * Hardware limitation: The bclk rate must be
- * never greater than 1/5 IPG clock rate
- */
- if (clkrate * 5 > clk_get_rate(ssi_private->clk))
- continue;
-
clkrate /= factor;
afreq = clkrate / (i + 1);
@@ -1158,14 +1216,14 @@ static struct snd_soc_dai_driver fsl_ssi_dai_template = {
.playback = {
.stream_name = "CPU-Playback",
.channels_min = 1,
- .channels_max = 2,
+ .channels_max = 32,
.rates = FSLSSI_I2S_RATES,
.formats = FSLSSI_I2S_FORMATS,
},
.capture = {
.stream_name = "CPU-Capture",
.channels_min = 1,
- .channels_max = 2,
+ .channels_max = 32,
.rates = FSLSSI_I2S_RATES,
.formats = FSLSSI_I2S_FORMATS,
},
@@ -1402,6 +1460,7 @@ static int fsl_ssi_probe(struct platform_device *pdev)
}
ssi_private->soc = of_id->data;
+ ssi_private->dev = &pdev->dev;
sprop = of_get_property(np, "fsl,mode", NULL);
if (sprop) {
diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c
index e63cd5ecfd8f..dac6688540dc 100644
--- a/sound/soc/fsl/imx-pcm-fiq.c
+++ b/sound/soc/fsl/imx-pcm-fiq.c
@@ -220,7 +220,7 @@ static int snd_imx_pcm_mmap(struct snd_pcm_substream *substream,
ret = dma_mmap_wc(substream->pcm->card->dev, vma, runtime->dma_area,
runtime->dma_addr, runtime->dma_bytes);
- pr_debug("%s: ret: %d %p %pad 0x%08x\n", __func__, ret,
+ pr_debug("%s: ret: %d %p %pad 0x%08zx\n", __func__, ret,
runtime->dma_area,
&runtime->dma_addr,
runtime->dma_bytes);
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 2389ab47e25f..466492b7d4f5 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -643,6 +643,7 @@ MODULE_DEVICE_TABLE(of, asoc_simple_of_match);
static struct platform_driver asoc_simple_card = {
.driver = {
.name = "asoc-simple-card",
+ .pm = &snd_soc_pm_ops,
.of_match_table = asoc_simple_of_match,
},
.probe = asoc_simple_card_probe,
diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig
index b3e6c2300457..91c15abb625e 100644
--- a/sound/soc/intel/Kconfig
+++ b/sound/soc/intel/Kconfig
@@ -58,6 +58,21 @@ config SND_SOC_INTEL_HASWELL_MACH
Say Y if you have such a device
If unsure select "N".
+config SND_SOC_INTEL_BXT_RT298_MACH
+ tristate "ASoC Audio driver for Broxton with RT298 I2S mode"
+ depends on X86 && ACPI && I2C
+ select SND_SOC_INTEL_SST
+ select SND_SOC_INTEL_SKYLAKE
+ select SND_SOC_RT298
+ select SND_SOC_DMIC
+ select SND_SOC_HDAC_HDMI
+ select SND_HDA_DSP_LOADER
+ help
+ This adds support for ASoC machine driver for Broxton platforms
+ with RT286 I2S audio codec.
+ Say Y if you have such a device
+ If unsure select "N".
+
config SND_SOC_INTEL_BYT_RT5640_MACH
tristate "ASoC Audio driver for Intel Baytrail with RT5640 codec"
depends on X86_INTEL_LPSS && I2C
@@ -162,8 +177,8 @@ config SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH
config SND_SOC_INTEL_SKYLAKE
tristate
select SND_HDA_EXT_CORE
+ select SND_HDA_DSP_LOADER
select SND_SOC_TOPOLOGY
- select SND_HDA_I915
select SND_SOC_INTEL_SST
config SND_SOC_INTEL_SKL_RT286_MACH
diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c
index b97e6adcf1b2..98720a93de8a 100644
--- a/sound/soc/intel/atom/sst-atom-controls.c
+++ b/sound/soc/intel/atom/sst-atom-controls.c
@@ -195,7 +195,7 @@ static int sst_check_and_send_slot_map(struct sst_data *drv, struct snd_kcontrol
if (e->w && e->w->power)
ret = sst_send_slot_map(drv);
- else
+ else if (!e->w)
dev_err(&drv->pdev->dev, "Slot control: %s doesn't have DAPM widget!!!\n",
kcontrol->id.name);
return ret;
diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile
index 3310c0f9c356..a8506774f510 100644
--- a/sound/soc/intel/boards/Makefile
+++ b/sound/soc/intel/boards/Makefile
@@ -2,6 +2,7 @@ snd-soc-sst-haswell-objs := haswell.o
snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o
snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o
snd-soc-sst-broadwell-objs := broadwell.o
+snd-soc-sst-bxt-rt298-objs := bxt_rt298.o
snd-soc-sst-bytcr-rt5640-objs := bytcr_rt5640.o
snd-soc-sst-bytcr-rt5651-objs := bytcr_rt5651.o
snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o
@@ -14,6 +15,7 @@ snd-soc-skl_nau88l25_ssm4567-objs := skl_nau88l25_ssm4567.o
obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o
obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o
obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o
+obj-$(CONFIG_SND_SOC_INTEL_BXT_RT298_MACH) += snd-soc-sst-bxt-rt298.o
obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o
obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-rt5640.o
obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5651_MACH) += snd-soc-sst-bytcr-rt5651.o
diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c
index 3f8a1e10bed0..7486a0022fde 100644
--- a/sound/soc/intel/boards/broadwell.c
+++ b/sound/soc/intel/boards/broadwell.c
@@ -201,7 +201,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = {
{
/* SSP0 - Codec */
.name = "Codec",
- .be_id = 0,
+ .id = 0,
.cpu_dai_name = "snd-soc-dummy-dai",
.platform_name = "snd-soc-dummy",
.no_pcm = 1,
diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c
new file mode 100644
index 000000000000..f4787515c0ed
--- /dev/null
+++ b/sound/soc/intel/boards/bxt_rt298.c
@@ -0,0 +1,353 @@
+/*
+ * Intel Broxton-P I2S Machine Driver
+ *
+ * Copyright (C) 2014-2016, Intel Corporation. All rights reserved.
+ *
+ * Modified from:
+ * Intel Skylake I2S Machine driver
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License version
+ * 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include <sound/pcm_params.h>
+#include "../../codecs/hdac_hdmi.h"
+#include "../../codecs/rt298.h"
+
+static struct snd_soc_jack broxton_headset;
+/* Headset jack detection DAPM pins */
+
+enum {
+ BXT_DPCM_AUDIO_PB = 0,
+ BXT_DPCM_AUDIO_CP,
+ BXT_DPCM_AUDIO_REF_CP,
+ BXT_DPCM_AUDIO_HDMI1_PB,
+ BXT_DPCM_AUDIO_HDMI2_PB,
+ BXT_DPCM_AUDIO_HDMI3_PB,
+};
+
+static struct snd_soc_jack_pin broxton_headset_pins[] = {
+ {
+ .pin = "Mic Jack",
+ .mask = SND_JACK_MICROPHONE,
+ },
+ {
+ .pin = "Headphone Jack",
+ .mask = SND_JACK_HEADPHONE,
+ },
+};
+
+static const struct snd_kcontrol_new broxton_controls[] = {
+ SOC_DAPM_PIN_SWITCH("Speaker"),
+ SOC_DAPM_PIN_SWITCH("Headphone Jack"),
+ SOC_DAPM_PIN_SWITCH("Mic Jack"),
+};
+
+static const struct snd_soc_dapm_widget broxton_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_SPK("Speaker", NULL),
+ SND_SOC_DAPM_MIC("Mic Jack", NULL),
+ SND_SOC_DAPM_MIC("DMIC2", NULL),
+ SND_SOC_DAPM_MIC("SoC DMIC", NULL),
+ SND_SOC_DAPM_SPK("HDMI1", NULL),
+ SND_SOC_DAPM_SPK("HDMI2", NULL),
+ SND_SOC_DAPM_SPK("HDMI3", NULL),
+};
+
+static const struct snd_soc_dapm_route broxton_rt298_map[] = {
+ /* speaker */
+ {"Speaker", NULL, "SPOR"},
+ {"Speaker", NULL, "SPOL"},
+
+ /* HP jack connectors - unknown if we have jack detect */
+ {"Headphone Jack", NULL, "HPO Pin"},
+
+ /* other jacks */
+ {"MIC1", NULL, "Mic Jack"},
+
+ /* digital mics */
+ {"DMIC1 Pin", NULL, "DMIC2"},
+ {"DMic", NULL, "SoC DMIC"},
+
+ {"HDMI1", NULL, "hif5 Output"},
+ {"HDMI2", NULL, "hif6 Output"},
+ {"HDMI3", NULL, "hif7 Output"},
+
+ /* CODEC BE connections */
+ { "AIF1 Playback", NULL, "ssp5 Tx"},
+ { "ssp5 Tx", NULL, "codec0_out"},
+
+ { "codec0_in", NULL, "ssp5 Rx" },
+ { "ssp5 Rx", NULL, "AIF1 Capture" },
+
+ { "dmic01_hifi", NULL, "DMIC01 Rx" },
+ { "DMIC01 Rx", NULL, "Capture" },
+
+ { "hifi3", NULL, "iDisp3 Tx"},
+ { "iDisp3 Tx", NULL, "iDisp3_out"},
+ { "hifi2", NULL, "iDisp2 Tx"},
+ { "iDisp2 Tx", NULL, "iDisp2_out"},
+ { "hifi1", NULL, "iDisp1 Tx"},
+ { "iDisp1 Tx", NULL, "iDisp1_out"},
+
+};
+
+static int broxton_rt298_codec_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_codec *codec = rtd->codec;
+ int ret = 0;
+
+ ret = snd_soc_card_jack_new(rtd->card, "Headset",
+ SND_JACK_HEADSET | SND_JACK_BTN_0,
+ &broxton_headset,
+ broxton_headset_pins, ARRAY_SIZE(broxton_headset_pins));
+
+ if (ret)
+ return ret;
+
+ rt298_mic_detect(codec, &broxton_headset);
+ return 0;
+}
+
+static int broxton_hdmi_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_dai *dai = rtd->codec_dai;
+
+ return hdac_hdmi_jack_init(dai, BXT_DPCM_AUDIO_HDMI1_PB + dai->id);
+}
+
+static int broxton_ssp5_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+ struct snd_interval *channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+ struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+
+ /* The ADSP will covert the FE rate to 48k, stereo */
+ rate->min = rate->max = 48000;
+ channels->min = channels->max = 2;
+
+ /* set SSP5 to 24 bit */
+ snd_mask_none(fmt);
+ snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE);
+
+ return 0;
+}
+
+static int broxton_rt298_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ int ret;
+
+ ret = snd_soc_dai_set_sysclk(codec_dai, RT298_SCLK_S_PLL,
+ 19200000, SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ dev_err(rtd->dev, "can't set codec sysclk configuration\n");
+ return ret;
+ }
+
+ return ret;
+}
+
+static struct snd_soc_ops broxton_rt298_ops = {
+ .hw_params = broxton_rt298_hw_params,
+};
+
+/* broxton digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link broxton_rt298_dais[] = {
+ /* Front End DAI links */
+ [BXT_DPCM_AUDIO_PB]
+ {
+ .name = "Bxt Audio Port",
+ .stream_name = "Audio",
+ .cpu_dai_name = "System Pin",
+ .platform_name = "0000:00:0e.0",
+ .nonatomic = 1,
+ .dynamic = 1,
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .dpcm_playback = 1,
+ },
+ [BXT_DPCM_AUDIO_CP]
+ {
+ .name = "Bxt Audio Capture Port",
+ .stream_name = "Audio Record",
+ .cpu_dai_name = "System Pin",
+ .platform_name = "0000:00:0e.0",
+ .nonatomic = 1,
+ .dynamic = 1,
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .dpcm_capture = 1,
+ },
+ [BXT_DPCM_AUDIO_REF_CP]
+ {
+ .name = "Bxt Audio Reference cap",
+ .stream_name = "refcap",
+ .cpu_dai_name = "Reference Pin",
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .platform_name = "0000:00:0e.0",
+ .init = NULL,
+ .dpcm_capture = 1,
+ .nonatomic = 1,
+ .dynamic = 1,
+ },
+ [BXT_DPCM_AUDIO_HDMI1_PB]
+ {
+ .name = "Bxt HDMI Port1",
+ .stream_name = "Hdmi1",
+ .cpu_dai_name = "HDMI1 Pin",
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .platform_name = "0000:00:0e.0",
+ .dpcm_playback = 1,
+ .init = NULL,
+ .nonatomic = 1,
+ .dynamic = 1,
+ },
+ [BXT_DPCM_AUDIO_HDMI2_PB]
+ {
+ .name = "Bxt HDMI Port2",
+ .stream_name = "Hdmi2",
+ .cpu_dai_name = "HDMI2 Pin",
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .platform_name = "0000:00:0e.0",
+ .dpcm_playback = 1,
+ .init = NULL,
+ .nonatomic = 1,
+ .dynamic = 1,
+ },
+ [BXT_DPCM_AUDIO_HDMI3_PB]
+ {
+ .name = "Bxt HDMI Port3",
+ .stream_name = "Hdmi3",
+ .cpu_dai_name = "HDMI3 Pin",
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .platform_name = "0000:00:0e.0",
+ .dpcm_playback = 1,
+ .init = NULL,
+ .nonatomic = 1,
+ .dynamic = 1,
+ },
+ /* Back End DAI links */
+ {
+ /* SSP5 - Codec */
+ .name = "SSP5-Codec",
+ .id = 0,
+ .cpu_dai_name = "SSP5 Pin",
+ .platform_name = "0000:00:0e.0",
+ .no_pcm = 1,
+ .codec_name = "i2c-INT343A:00",
+ .codec_dai_name = "rt298-aif1",
+ .init = broxton_rt298_codec_init,
+ .dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS,
+ .ignore_pmdown_time = 1,
+ .be_hw_params_fixup = broxton_ssp5_fixup,
+ .ops = &broxton_rt298_ops,
+ .dpcm_playback = 1,
+ .dpcm_capture = 1,
+ },
+ {
+ .name = "dmic01",
+ .id = 1,
+ .cpu_dai_name = "DMIC01 Pin",
+ .codec_name = "dmic-codec",
+ .codec_dai_name = "dmic-hifi",
+ .platform_name = "0000:00:0e.0",
+ .ignore_suspend = 1,
+ .dpcm_capture = 1,
+ .no_pcm = 1,
+ },
+ {
+ .name = "iDisp1",
+ .id = 3,
+ .cpu_dai_name = "iDisp1 Pin",
+ .codec_name = "ehdaudio0D2",
+ .codec_dai_name = "intel-hdmi-hifi1",
+ .platform_name = "0000:00:0e.0",
+ .init = broxton_hdmi_init,
+ .dpcm_playback = 1,
+ .no_pcm = 1,
+ },
+ {
+ .name = "iDisp2",
+ .id = 4,
+ .cpu_dai_name = "iDisp2 Pin",
+ .codec_name = "ehdaudio0D2",
+ .codec_dai_name = "intel-hdmi-hifi2",
+ .platform_name = "0000:00:0e.0",
+ .init = broxton_hdmi_init,
+ .dpcm_playback = 1,
+ .no_pcm = 1,
+ },
+ {
+ .name = "iDisp3",
+ .id = 5,
+ .cpu_dai_name = "iDisp3 Pin",
+ .codec_name = "ehdaudio0D2",
+ .codec_dai_name = "intel-hdmi-hifi3",
+ .platform_name = "0000:00:0e.0",
+ .init = broxton_hdmi_init,
+ .dpcm_playback = 1,
+ .no_pcm = 1,
+ },
+};
+
+/* broxton audio machine driver for SPT + RT298S */
+static struct snd_soc_card broxton_rt298 = {
+ .name = "broxton-rt298",
+ .owner = THIS_MODULE,
+ .dai_link = broxton_rt298_dais,
+ .num_links = ARRAY_SIZE(broxton_rt298_dais),
+ .controls = broxton_controls,
+ .num_controls = ARRAY_SIZE(broxton_controls),
+ .dapm_widgets = broxton_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(broxton_widgets),
+ .dapm_routes = broxton_rt298_map,
+ .num_dapm_routes = ARRAY_SIZE(broxton_rt298_map),
+ .fully_routed = true,
+};
+
+static int broxton_audio_probe(struct platform_device *pdev)
+{
+ broxton_rt298.dev = &pdev->dev;
+
+ return devm_snd_soc_register_card(&pdev->dev, &broxton_rt298);
+}
+
+static struct platform_driver broxton_audio = {
+ .probe = broxton_audio_probe,
+ .driver = {
+ .name = "bxt_alc298s_i2s",
+ },
+};
+module_platform_driver(broxton_audio)
+
+/* Module information */
+MODULE_AUTHOR("Ramesh Babu <Ramesh.Babu@intel.com>");
+MODULE_AUTHOR("Senthilnathan Veppur <senthilnathanx.veppur@intel.com>");
+MODULE_DESCRIPTION("Intel SST Audio for Broxton");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:bxt_alc298s_i2s");
diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c
index 032a2e753f0b..88efb62439ba 100644
--- a/sound/soc/intel/boards/bytcr_rt5640.c
+++ b/sound/soc/intel/boards/bytcr_rt5640.c
@@ -304,7 +304,7 @@ static struct snd_soc_dai_link byt_rt5640_dais[] = {
/* back ends */
{
.name = "SSP2-Codec",
- .be_id = 1,
+ .id = 1,
.cpu_dai_name = "ssp2-port",
.platform_name = "sst-mfld-platform",
.no_pcm = 1,
diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c
index 1c95ccc886c4..35f591eab3c9 100644
--- a/sound/soc/intel/boards/bytcr_rt5651.c
+++ b/sound/soc/intel/boards/bytcr_rt5651.c
@@ -267,7 +267,7 @@ static struct snd_soc_dai_link byt_rt5651_dais[] = {
/* back ends */
{
.name = "SSP2-Codec",
- .be_id = 1,
+ .id = 1,
.cpu_dai_name = "ssp2-port",
.platform_name = "sst-mfld-platform",
.no_pcm = 1,
diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c
index e609f089593a..6260df6bd49c 100644
--- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c
+++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c
@@ -255,7 +255,7 @@ static struct snd_soc_dai_link cht_dailink[] = {
/* back ends */
{
.name = "SSP2-Codec",
- .be_id = 1,
+ .id = 1,
.cpu_dai_name = "ssp2-port",
.platform_name = "sst-mfld-platform",
.no_pcm = 1,
diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c
index 2a6f80843bc9..0618a7f1025b 100644
--- a/sound/soc/intel/boards/cht_bsw_rt5645.c
+++ b/sound/soc/intel/boards/cht_bsw_rt5645.c
@@ -295,7 +295,7 @@ static struct snd_soc_dai_link cht_dailink[] = {
/* back ends */
{
.name = "SSP2-Codec",
- .be_id = 1,
+ .id = 1,
.cpu_dai_name = "ssp2-port",
.platform_name = "sst-mfld-platform",
.no_pcm = 1,
diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c
index 2e5347f8f96c..df9d254baa18 100644
--- a/sound/soc/intel/boards/cht_bsw_rt5672.c
+++ b/sound/soc/intel/boards/cht_bsw_rt5672.c
@@ -273,7 +273,7 @@ static struct snd_soc_dai_link cht_dailink[] = {
{
/* SSP2 - Codec */
.name = "SSP2-Codec",
- .be_id = 1,
+ .id = 1,
.cpu_dai_name = "ssp2-port",
.platform_name = "sst-mfld-platform",
.no_pcm = 1,
diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c
index 22558572cb9c..863f1d5e2a2c 100644
--- a/sound/soc/intel/boards/haswell.c
+++ b/sound/soc/intel/boards/haswell.c
@@ -156,7 +156,7 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = {
{
/* SSP0 - Codec */
.name = "Codec",
- .be_id = 0,
+ .id = 0,
.cpu_dai_name = "snd-soc-dummy-dai",
.platform_name = "snd-soc-dummy",
.no_pcm = 1,
diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c
index 72176b79a18d..d2808652b974 100644
--- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c
+++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c
@@ -30,6 +30,16 @@
static struct snd_soc_jack skylake_headset;
static struct snd_soc_card skylake_audio_card;
+struct skl_hdmi_pcm {
+ struct list_head head;
+ struct snd_soc_dai *codec_dai;
+ int device;
+};
+
+struct skl_nau8825_private {
+ struct list_head hdmi_pcm_list;
+};
+
enum {
SKL_DPCM_AUDIO_PB = 0,
SKL_DPCM_AUDIO_CP,
@@ -192,23 +202,56 @@ static int skylake_nau8825_codec_init(struct snd_soc_pcm_runtime *rtd)
static int skylake_hdmi1_init(struct snd_soc_pcm_runtime *rtd)
{
+ struct skl_nau8825_private *ctx = snd_soc_card_get_drvdata(rtd->card);
struct snd_soc_dai *dai = rtd->codec_dai;
+ struct skl_hdmi_pcm *pcm;
+
+ pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
+ if (!pcm)
+ return -ENOMEM;
+
+ pcm->device = SKL_DPCM_AUDIO_HDMI1_PB;
+ pcm->codec_dai = dai;
- return hdac_hdmi_jack_init(dai, SKL_DPCM_AUDIO_HDMI1_PB);
+ list_add_tail(&pcm->head, &ctx->hdmi_pcm_list);
+
+ return 0;
}
static int skylake_hdmi2_init(struct snd_soc_pcm_runtime *rtd)
{
+ struct skl_nau8825_private *ctx = snd_soc_card_get_drvdata(rtd->card);
struct snd_soc_dai *dai = rtd->codec_dai;
+ struct skl_hdmi_pcm *pcm;
+
+ pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
+ if (!pcm)
+ return -ENOMEM;
+
+ pcm->device = SKL_DPCM_AUDIO_HDMI2_PB;
+ pcm->codec_dai = dai;
- return hdac_hdmi_jack_init(dai, SKL_DPCM_AUDIO_HDMI2_PB);
+ list_add_tail(&pcm->head, &ctx->hdmi_pcm_list);
+
+ return 0;
}
static int skylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd)
{
+ struct skl_nau8825_private *ctx = snd_soc_card_get_drvdata(rtd->card);
struct snd_soc_dai *dai = rtd->codec_dai;
+ struct skl_hdmi_pcm *pcm;
+
+ pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
+ if (!pcm)
+ return -ENOMEM;
- return hdac_hdmi_jack_init(dai, SKL_DPCM_AUDIO_HDMI3_PB);
+ pcm->device = SKL_DPCM_AUDIO_HDMI3_PB;
+ pcm->codec_dai = dai;
+
+ list_add_tail(&pcm->head, &ctx->hdmi_pcm_list);
+
+ return 0;
}
static int skylake_nau8825_fe_init(struct snd_soc_pcm_runtime *rtd)
@@ -391,7 +434,6 @@ static struct snd_soc_dai_link skylake_dais[] = {
.platform_name = "0000:00:1f.3",
.init = NULL,
.dpcm_capture = 1,
- .ignore_suspend = 1,
.nonatomic = 1,
.dynamic = 1,
.ops = &skylaye_refcap_ops,
@@ -456,7 +498,7 @@ static struct snd_soc_dai_link skylake_dais[] = {
{
/* SSP0 - Codec */
.name = "SSP0-Codec",
- .be_id = 0,
+ .id = 0,
.cpu_dai_name = "SSP0 Pin",
.platform_name = "0000:00:1f.3",
.no_pcm = 1,
@@ -472,7 +514,7 @@ static struct snd_soc_dai_link skylake_dais[] = {
{
/* SSP1 - Codec */
.name = "SSP1-Codec",
- .be_id = 1,
+ .id = 1,
.cpu_dai_name = "SSP1 Pin",
.platform_name = "0000:00:1f.3",
.no_pcm = 1,
@@ -489,7 +531,7 @@ static struct snd_soc_dai_link skylake_dais[] = {
},
{
.name = "dmic01",
- .be_id = 2,
+ .id = 2,
.cpu_dai_name = "DMIC01 Pin",
.codec_name = "dmic-codec",
.codec_dai_name = "dmic-hifi",
@@ -501,7 +543,7 @@ static struct snd_soc_dai_link skylake_dais[] = {
},
{
.name = "iDisp1",
- .be_id = 3,
+ .id = 3,
.cpu_dai_name = "iDisp1 Pin",
.codec_name = "ehdaudio0D2",
.codec_dai_name = "intel-hdmi-hifi1",
@@ -512,7 +554,7 @@ static struct snd_soc_dai_link skylake_dais[] = {
},
{
.name = "iDisp2",
- .be_id = 4,
+ .id = 4,
.cpu_dai_name = "iDisp2 Pin",
.codec_name = "ehdaudio0D2",
.codec_dai_name = "intel-hdmi-hifi2",
@@ -523,7 +565,7 @@ static struct snd_soc_dai_link skylake_dais[] = {
},
{
.name = "iDisp3",
- .be_id = 5,
+ .id = 5,
.cpu_dai_name = "iDisp3 Pin",
.codec_name = "ehdaudio0D2",
.codec_dai_name = "intel-hdmi-hifi3",
@@ -534,6 +576,21 @@ static struct snd_soc_dai_link skylake_dais[] = {
},
};
+static int skylake_card_late_probe(struct snd_soc_card *card)
+{
+ struct skl_nau8825_private *ctx = snd_soc_card_get_drvdata(card);
+ struct skl_hdmi_pcm *pcm;
+ int err;
+
+ list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) {
+ err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device);
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
/* skylake audio machine driver for SPT + NAU88L25 */
static struct snd_soc_card skylake_audio_card = {
.name = "sklnau8825max",
@@ -547,11 +604,21 @@ static struct snd_soc_card skylake_audio_card = {
.dapm_routes = skylake_map,
.num_dapm_routes = ARRAY_SIZE(skylake_map),
.fully_routed = true,
+ .late_probe = skylake_card_late_probe,
};
static int skylake_audio_probe(struct platform_device *pdev)
{
+ struct skl_nau8825_private *ctx;
+
+ ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC);
+ if (!ctx)
+ return -ENOMEM;
+
+ INIT_LIST_HEAD(&ctx->hdmi_pcm_list);
+
skylake_audio_card.dev = &pdev->dev;
+ snd_soc_card_set_drvdata(&skylake_audio_card, ctx);
return devm_snd_soc_register_card(&pdev->dev, &skylake_audio_card);
}
diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c
index 5f1ca99ae9b0..e19aa99c4f72 100644
--- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c
+++ b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c
@@ -34,6 +34,15 @@
static struct snd_soc_jack skylake_headset;
static struct snd_soc_card skylake_audio_card;
+struct skl_hdmi_pcm {
+ struct list_head head;
+ struct snd_soc_dai *codec_dai;
+ int device;
+};
+
+struct skl_nau88125_private {
+ struct list_head hdmi_pcm_list;
+};
enum {
SKL_DPCM_AUDIO_PB = 0,
SKL_DPCM_AUDIO_CP,
@@ -222,24 +231,57 @@ static int skylake_nau8825_codec_init(struct snd_soc_pcm_runtime *rtd)
static int skylake_hdmi1_init(struct snd_soc_pcm_runtime *rtd)
{
+ struct skl_nau88125_private *ctx = snd_soc_card_get_drvdata(rtd->card);
struct snd_soc_dai *dai = rtd->codec_dai;
+ struct skl_hdmi_pcm *pcm;
+
+ pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
+ if (!pcm)
+ return -ENOMEM;
+
+ pcm->device = SKL_DPCM_AUDIO_HDMI1_PB;
+ pcm->codec_dai = dai;
+
+ list_add_tail(&pcm->head, &ctx->hdmi_pcm_list);
- return hdac_hdmi_jack_init(dai, SKL_DPCM_AUDIO_HDMI1_PB);
+ return 0;
}
static int skylake_hdmi2_init(struct snd_soc_pcm_runtime *rtd)
{
+ struct skl_nau88125_private *ctx = snd_soc_card_get_drvdata(rtd->card);
struct snd_soc_dai *dai = rtd->codec_dai;
+ struct skl_hdmi_pcm *pcm;
+
+ pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
+ if (!pcm)
+ return -ENOMEM;
+
+ pcm->device = SKL_DPCM_AUDIO_HDMI2_PB;
+ pcm->codec_dai = dai;
- return hdac_hdmi_jack_init(dai, SKL_DPCM_AUDIO_HDMI2_PB);
+ list_add_tail(&pcm->head, &ctx->hdmi_pcm_list);
+
+ return 0;
}
static int skylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd)
{
+ struct skl_nau88125_private *ctx = snd_soc_card_get_drvdata(rtd->card);
struct snd_soc_dai *dai = rtd->codec_dai;
+ struct skl_hdmi_pcm *pcm;
+
+ pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
+ if (!pcm)
+ return -ENOMEM;
- return hdac_hdmi_jack_init(dai, SKL_DPCM_AUDIO_HDMI3_PB);
+ pcm->device = SKL_DPCM_AUDIO_HDMI3_PB;
+ pcm->codec_dai = dai;
+
+ list_add_tail(&pcm->head, &ctx->hdmi_pcm_list);
+
+ return 0;
}
static int skylake_nau8825_fe_init(struct snd_soc_pcm_runtime *rtd)
@@ -440,7 +482,6 @@ static struct snd_soc_dai_link skylake_dais[] = {
.platform_name = "0000:00:1f.3",
.init = NULL,
.dpcm_capture = 1,
- .ignore_suspend = 1,
.nonatomic = 1,
.dynamic = 1,
.ops = &skylaye_refcap_ops,
@@ -505,7 +546,7 @@ static struct snd_soc_dai_link skylake_dais[] = {
{
/* SSP0 - Codec */
.name = "SSP0-Codec",
- .be_id = 0,
+ .id = 0,
.cpu_dai_name = "SSP0 Pin",
.platform_name = "0000:00:1f.3",
.no_pcm = 1,
@@ -523,7 +564,7 @@ static struct snd_soc_dai_link skylake_dais[] = {
{
/* SSP1 - Codec */
.name = "SSP1-Codec",
- .be_id = 1,
+ .id = 1,
.cpu_dai_name = "SSP1 Pin",
.platform_name = "0000:00:1f.3",
.no_pcm = 1,
@@ -540,7 +581,7 @@ static struct snd_soc_dai_link skylake_dais[] = {
},
{
.name = "dmic01",
- .be_id = 2,
+ .id = 2,
.cpu_dai_name = "DMIC01 Pin",
.codec_name = "dmic-codec",
.codec_dai_name = "dmic-hifi",
@@ -552,7 +593,7 @@ static struct snd_soc_dai_link skylake_dais[] = {
},
{
.name = "iDisp1",
- .be_id = 3,
+ .id = 3,
.cpu_dai_name = "iDisp1 Pin",
.codec_name = "ehdaudio0D2",
.codec_dai_name = "intel-hdmi-hifi1",
@@ -563,7 +604,7 @@ static struct snd_soc_dai_link skylake_dais[] = {
},
{
.name = "iDisp2",
- .be_id = 4,
+ .id = 4,
.cpu_dai_name = "iDisp2 Pin",
.codec_name = "ehdaudio0D2",
.codec_dai_name = "intel-hdmi-hifi2",
@@ -574,7 +615,7 @@ static struct snd_soc_dai_link skylake_dais[] = {
},
{
.name = "iDisp3",
- .be_id = 5,
+ .id = 5,
.cpu_dai_name = "iDisp3 Pin",
.codec_name = "ehdaudio0D2",
.codec_dai_name = "intel-hdmi-hifi3",
@@ -585,6 +626,21 @@ static struct snd_soc_dai_link skylake_dais[] = {
},
};
+static int skylake_card_late_probe(struct snd_soc_card *card)
+{
+ struct skl_nau88125_private *ctx = snd_soc_card_get_drvdata(card);
+ struct skl_hdmi_pcm *pcm;
+ int err;
+
+ list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) {
+ err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device);
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
/* skylake audio machine driver for SPT + NAU88L25 */
static struct snd_soc_card skylake_audio_card = {
.name = "sklnau8825adi",
@@ -600,11 +656,21 @@ static struct snd_soc_card skylake_audio_card = {
.codec_conf = ssm4567_codec_conf,
.num_configs = ARRAY_SIZE(ssm4567_codec_conf),
.fully_routed = true,
+ .late_probe = skylake_card_late_probe,
};
static int skylake_audio_probe(struct platform_device *pdev)
{
+ struct skl_nau88125_private *ctx;
+
+ ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC);
+ if (!ctx)
+ return -ENOMEM;
+
+ INIT_LIST_HEAD(&ctx->hdmi_pcm_list);
+
skylake_audio_card.dev = &pdev->dev;
+ snd_soc_card_set_drvdata(&skylake_audio_card, ctx);
return devm_snd_soc_register_card(&pdev->dev, &skylake_audio_card);
}
diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c
index 2016397a8e75..426b48233fdb 100644
--- a/sound/soc/intel/boards/skl_rt286.c
+++ b/sound/soc/intel/boards/skl_rt286.c
@@ -30,6 +30,16 @@
static struct snd_soc_jack skylake_headset;
+struct skl_hdmi_pcm {
+ struct list_head head;
+ struct snd_soc_dai *codec_dai;
+ int device;
+};
+
+struct skl_rt286_private {
+ struct list_head hdmi_pcm_list;
+};
+
enum {
SKL_DPCM_AUDIO_PB = 0,
SKL_DPCM_AUDIO_CP,
@@ -142,9 +152,20 @@ static int skylake_rt286_codec_init(struct snd_soc_pcm_runtime *rtd)
static int skylake_hdmi_init(struct snd_soc_pcm_runtime *rtd)
{
+ struct skl_rt286_private *ctx = snd_soc_card_get_drvdata(rtd->card);
struct snd_soc_dai *dai = rtd->codec_dai;
+ struct skl_hdmi_pcm *pcm;
+
+ pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
+ if (!pcm)
+ return -ENOMEM;
- return hdac_hdmi_jack_init(dai, SKL_DPCM_AUDIO_HDMI1_PB + dai->id);
+ pcm->device = SKL_DPCM_AUDIO_HDMI1_PB + dai->id;
+ pcm->codec_dai = dai;
+
+ list_add_tail(&pcm->head, &ctx->hdmi_pcm_list);
+
+ return 0;
}
static unsigned int rates[] = {
@@ -317,7 +338,6 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = {
.platform_name = "0000:00:1f.3",
.init = NULL,
.dpcm_capture = 1,
- .ignore_suspend = 1,
.nonatomic = 1,
.dynamic = 1,
},
@@ -375,7 +395,7 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = {
{
/* SSP0 - Codec */
.name = "SSP0-Codec",
- .be_id = 0,
+ .id = 0,
.cpu_dai_name = "SSP0 Pin",
.platform_name = "0000:00:1f.3",
.no_pcm = 1,
@@ -393,7 +413,7 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = {
},
{
.name = "dmic01",
- .be_id = 1,
+ .id = 1,
.cpu_dai_name = "DMIC01 Pin",
.codec_name = "dmic-codec",
.codec_dai_name = "dmic-hifi",
@@ -405,7 +425,7 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = {
},
{
.name = "iDisp1",
- .be_id = 2,
+ .id = 2,
.cpu_dai_name = "iDisp1 Pin",
.codec_name = "ehdaudio0D2",
.codec_dai_name = "intel-hdmi-hifi1",
@@ -416,7 +436,7 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = {
},
{
.name = "iDisp2",
- .be_id = 3,
+ .id = 3,
.cpu_dai_name = "iDisp2 Pin",
.codec_name = "ehdaudio0D2",
.codec_dai_name = "intel-hdmi-hifi2",
@@ -427,7 +447,7 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = {
},
{
.name = "iDisp3",
- .be_id = 4,
+ .id = 4,
.cpu_dai_name = "iDisp3 Pin",
.codec_name = "ehdaudio0D2",
.codec_dai_name = "intel-hdmi-hifi3",
@@ -438,6 +458,21 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = {
},
};
+static int skylake_card_late_probe(struct snd_soc_card *card)
+{
+ struct skl_rt286_private *ctx = snd_soc_card_get_drvdata(card);
+ struct skl_hdmi_pcm *pcm;
+ int err;
+
+ list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) {
+ err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device);
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
/* skylake audio machine driver for SPT + RT286S */
static struct snd_soc_card skylake_rt286 = {
.name = "skylake-rt286",
@@ -451,11 +486,21 @@ static struct snd_soc_card skylake_rt286 = {
.dapm_routes = skylake_rt286_map,
.num_dapm_routes = ARRAY_SIZE(skylake_rt286_map),
.fully_routed = true,
+ .late_probe = skylake_card_late_probe,
};
static int skylake_audio_probe(struct platform_device *pdev)
{
+ struct skl_rt286_private *ctx;
+
+ ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC);
+ if (!ctx)
+ return -ENOMEM;
+
+ INIT_LIST_HEAD(&ctx->hdmi_pcm_list);
+
skylake_rt286.dev = &pdev->dev;
+ snd_soc_card_set_drvdata(&skylake_rt286, ctx);
return devm_snd_soc_register_card(&pdev->dev, &skylake_rt286);
}
diff --git a/sound/soc/intel/common/sst-acpi.h b/sound/soc/intel/common/sst-acpi.h
index 4dcfb7e5ed70..8398cb227ba9 100644
--- a/sound/soc/intel/common/sst-acpi.h
+++ b/sound/soc/intel/common/sst-acpi.h
@@ -12,10 +12,19 @@
*
*/
+#include <linux/kconfig.h>
+#include <linux/stddef.h>
#include <linux/acpi.h>
/* translation fron HID to I2C name, needed for DAI codec_name */
+#if IS_ENABLED(CONFIG_ACPI)
const char *sst_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN]);
+#else
+inline const char *sst_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN])
+{
+ return NULL;
+}
+#endif
/* acpi match */
struct sst_acpi_mach *sst_acpi_find_machine(struct sst_acpi_mach *machines);
diff --git a/sound/soc/intel/haswell/sst-haswell-ipc.c b/sound/soc/intel/haswell/sst-haswell-ipc.c
index ac60f1301e21..91565229d074 100644
--- a/sound/soc/intel/haswell/sst-haswell-ipc.c
+++ b/sound/soc/intel/haswell/sst-haswell-ipc.c
@@ -1345,7 +1345,7 @@ int sst_hsw_stream_reset(struct sst_hsw *hsw, struct sst_hsw_stream *stream)
return 0;
/* wait for pause to complete before we reset the stream */
- while (stream->running && tries--)
+ while (stream->running && --tries)
msleep(1);
if (!tries) {
dev_err(hsw->dev, "error: reset stream %d still running\n",
diff --git a/sound/soc/intel/haswell/sst-haswell-pcm.c b/sound/soc/intel/haswell/sst-haswell-pcm.c
index 1aa819c7e09b..994256b39b9c 100644
--- a/sound/soc/intel/haswell/sst-haswell-pcm.c
+++ b/sound/soc/intel/haswell/sst-haswell-pcm.c
@@ -445,7 +445,7 @@ static int create_adsp_page_table(struct snd_pcm_substream *substream,
pages = snd_sgbuf_aligned_pages(size);
- dev_dbg(rtd->dev, "generating page table for %p size 0x%zu pages %d\n",
+ dev_dbg(rtd->dev, "generating page table for %p size 0x%zx pages %d\n",
dma_area, size, pages);
for (i = 0; i < pages; i++) {
diff --git a/sound/soc/intel/skylake/Makefile b/sound/soc/intel/skylake/Makefile
index 914b6dab9bea..c28f5d0e1d99 100644
--- a/sound/soc/intel/skylake/Makefile
+++ b/sound/soc/intel/skylake/Makefile
@@ -5,6 +5,6 @@ obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE) += snd-soc-skl.o
# Skylake IPC Support
snd-soc-skl-ipc-objs := skl-sst-ipc.o skl-sst-dsp.o skl-sst-cldma.o \
- skl-sst.o
+ skl-sst.o bxt-sst.o
obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE) += snd-soc-skl-ipc.o
diff --git a/sound/soc/intel/skylake/bxt-sst.c b/sound/soc/intel/skylake/bxt-sst.c
new file mode 100644
index 000000000000..965ce40ce752
--- /dev/null
+++ b/sound/soc/intel/skylake/bxt-sst.c
@@ -0,0 +1,328 @@
+/*
+ * bxt-sst.c - DSP library functions for BXT platform
+ *
+ * Copyright (C) 2015-16 Intel Corp
+ * Author:Rafal Redzimski <rafal.f.redzimski@intel.com>
+ * Jeeja KP <jeeja.kp@intel.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; version 2 of the License.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ */
+
+#include <linux/module.h>
+#include <linux/delay.h>
+#include <linux/firmware.h>
+#include <linux/device.h>
+
+#include "../common/sst-dsp.h"
+#include "../common/sst-dsp-priv.h"
+#include "skl-sst-ipc.h"
+
+#define BXT_BASEFW_TIMEOUT 3000
+#define BXT_INIT_TIMEOUT 500
+#define BXT_IPC_PURGE_FW 0x01004000
+
+#define BXT_ROM_INIT 0x5
+#define BXT_ADSP_SRAM0_BASE 0x80000
+
+/* Firmware status window */
+#define BXT_ADSP_FW_STATUS BXT_ADSP_SRAM0_BASE
+#define BXT_ADSP_ERROR_CODE (BXT_ADSP_FW_STATUS + 0x4)
+
+#define BXT_ADSP_SRAM1_BASE 0xA0000
+
+static unsigned int bxt_get_errorcode(struct sst_dsp *ctx)
+{
+ return sst_dsp_shim_read(ctx, BXT_ADSP_ERROR_CODE);
+}
+
+static int sst_bxt_prepare_fw(struct sst_dsp *ctx,
+ const void *fwdata, u32 fwsize)
+{
+ int stream_tag, ret, i;
+ u32 reg;
+
+ stream_tag = ctx->dsp_ops.prepare(ctx->dev, 0x40, fwsize, &ctx->dmab);
+ if (stream_tag < 0) {
+ dev_err(ctx->dev, "Failed to prepare DMA FW loading err: %x\n",
+ stream_tag);
+ return stream_tag;
+ }
+
+ ctx->dsp_ops.stream_tag = stream_tag;
+ memcpy(ctx->dmab.area, fwdata, fwsize);
+
+ /* Purge FW request */
+ sst_dsp_shim_write(ctx, SKL_ADSP_REG_HIPCI, SKL_ADSP_REG_HIPCI_BUSY |
+ BXT_IPC_PURGE_FW | (stream_tag - 1));
+
+ ret = skl_dsp_enable_core(ctx);
+ if (ret < 0) {
+ dev_err(ctx->dev, "Boot dsp core failed ret: %d\n", ret);
+ ret = -EIO;
+ goto base_fw_load_failed;
+ }
+
+ for (i = BXT_INIT_TIMEOUT; i > 0; --i) {
+ reg = sst_dsp_shim_read(ctx, SKL_ADSP_REG_HIPCIE);
+
+ if (reg & SKL_ADSP_REG_HIPCIE_DONE) {
+ sst_dsp_shim_update_bits_forced(ctx,
+ SKL_ADSP_REG_HIPCIE,
+ SKL_ADSP_REG_HIPCIE_DONE,
+ SKL_ADSP_REG_HIPCIE_DONE);
+ break;
+ }
+ mdelay(1);
+ }
+ if (!i) {
+ dev_info(ctx->dev, "Waiting for HIPCIE done, reg: 0x%x\n", reg);
+ sst_dsp_shim_update_bits(ctx, SKL_ADSP_REG_HIPCIE,
+ SKL_ADSP_REG_HIPCIE_DONE,
+ SKL_ADSP_REG_HIPCIE_DONE);
+ }
+
+ /* enable Interrupt */
+ skl_ipc_int_enable(ctx);
+ skl_ipc_op_int_enable(ctx);
+
+ for (i = BXT_INIT_TIMEOUT; i > 0; --i) {
+ if (SKL_FW_INIT ==
+ (sst_dsp_shim_read(ctx, BXT_ADSP_FW_STATUS) &
+ SKL_FW_STS_MASK)) {
+
+ dev_info(ctx->dev, "ROM loaded, continue FW loading\n");
+ break;
+ }
+ mdelay(1);
+ }
+ if (!i) {
+ dev_err(ctx->dev, "Timeout for ROM init, HIPCIE: 0x%x\n", reg);
+ ret = -EIO;
+ goto base_fw_load_failed;
+ }
+
+ return ret;
+
+base_fw_load_failed:
+ ctx->dsp_ops.cleanup(ctx->dev, &ctx->dmab, stream_tag);
+ skl_dsp_disable_core(ctx);
+ return ret;
+}
+
+static int sst_transfer_fw_host_dma(struct sst_dsp *ctx)
+{
+ int ret;
+
+ ctx->dsp_ops.trigger(ctx->dev, true, ctx->dsp_ops.stream_tag);
+ ret = sst_dsp_register_poll(ctx, BXT_ADSP_FW_STATUS, SKL_FW_STS_MASK,
+ BXT_ROM_INIT, BXT_BASEFW_TIMEOUT, "Firmware boot");
+
+ ctx->dsp_ops.trigger(ctx->dev, false, ctx->dsp_ops.stream_tag);
+ ctx->dsp_ops.cleanup(ctx->dev, &ctx->dmab, ctx->dsp_ops.stream_tag);
+
+ return ret;
+}
+
+static int bxt_load_base_firmware(struct sst_dsp *ctx)
+{
+ const struct firmware *fw = NULL;
+ struct skl_sst *skl = ctx->thread_context;
+ int ret;
+
+ ret = request_firmware(&fw, ctx->fw_name, ctx->dev);
+ if (ret < 0) {
+ dev_err(ctx->dev, "Request firmware failed %d\n", ret);
+ goto sst_load_base_firmware_failed;
+ }
+
+ ret = sst_bxt_prepare_fw(ctx, fw->data, fw->size);
+ /* Retry Enabling core and ROM load. Retry seemed to help */
+ if (ret < 0) {
+ ret = sst_bxt_prepare_fw(ctx, fw->data, fw->size);
+ if (ret < 0) {
+ dev_err(ctx->dev, "Core En/ROM load fail:%d\n", ret);
+ goto sst_load_base_firmware_failed;
+ }
+ }
+
+ ret = sst_transfer_fw_host_dma(ctx);
+ if (ret < 0) {
+ dev_err(ctx->dev, "Transfer firmware failed %d\n", ret);
+ dev_info(ctx->dev, "Error code=0x%x: FW status=0x%x\n",
+ sst_dsp_shim_read(ctx, BXT_ADSP_ERROR_CODE),
+ sst_dsp_shim_read(ctx, BXT_ADSP_FW_STATUS));
+
+ skl_dsp_disable_core(ctx);
+ } else {
+ dev_dbg(ctx->dev, "Firmware download successful\n");
+ ret = wait_event_timeout(skl->boot_wait, skl->boot_complete,
+ msecs_to_jiffies(SKL_IPC_BOOT_MSECS));
+ if (ret == 0) {
+ dev_err(ctx->dev, "DSP boot fail, FW Ready timeout\n");
+ skl_dsp_disable_core(ctx);
+ ret = -EIO;
+ } else {
+ skl_dsp_set_state_locked(ctx, SKL_DSP_RUNNING);
+ ret = 0;
+ }
+ }
+
+sst_load_base_firmware_failed:
+ release_firmware(fw);
+ return ret;
+}
+
+static int bxt_set_dsp_D0(struct sst_dsp *ctx)
+{
+ struct skl_sst *skl = ctx->thread_context;
+ int ret;
+
+ skl->boot_complete = false;
+
+ ret = skl_dsp_enable_core(ctx);
+ if (ret < 0) {
+ dev_err(ctx->dev, "enable dsp core failed ret: %d\n", ret);
+ return ret;
+ }
+
+ /* enable interrupt */
+ skl_ipc_int_enable(ctx);
+ skl_ipc_op_int_enable(ctx);
+
+ ret = wait_event_timeout(skl->boot_wait, skl->boot_complete,
+ msecs_to_jiffies(SKL_IPC_BOOT_MSECS));
+ if (ret == 0) {
+ dev_err(ctx->dev, "ipc: error DSP boot timeout\n");
+ dev_err(ctx->dev, "Error code=0x%x: FW status=0x%x\n",
+ sst_dsp_shim_read(ctx, BXT_ADSP_ERROR_CODE),
+ sst_dsp_shim_read(ctx, BXT_ADSP_FW_STATUS));
+ return -EIO;
+ }
+
+ skl_dsp_set_state_locked(ctx, SKL_DSP_RUNNING);
+ return 0;
+}
+
+static int bxt_set_dsp_D3(struct sst_dsp *ctx)
+{
+ struct skl_ipc_dxstate_info dx;
+ struct skl_sst *skl = ctx->thread_context;
+ int ret = 0;
+
+ if (!is_skl_dsp_running(ctx))
+ return ret;
+
+ dx.core_mask = SKL_DSP_CORE0_MASK;
+ dx.dx_mask = SKL_IPC_D3_MASK;
+
+ ret = skl_ipc_set_dx(&skl->ipc, SKL_INSTANCE_ID,
+ SKL_BASE_FW_MODULE_ID, &dx);
+ if (ret < 0) {
+ dev_err(ctx->dev, "Failed to set DSP to D3 state: %d\n", ret);
+ return ret;
+ }
+
+ ret = skl_dsp_disable_core(ctx);
+ if (ret < 0) {
+ dev_err(ctx->dev, "disbale dsp core failed: %d\n", ret);
+ ret = -EIO;
+ }
+
+ skl_dsp_set_state_locked(ctx, SKL_DSP_RESET);
+ return 0;
+}
+
+static struct skl_dsp_fw_ops bxt_fw_ops = {
+ .set_state_D0 = bxt_set_dsp_D0,
+ .set_state_D3 = bxt_set_dsp_D3,
+ .load_fw = bxt_load_base_firmware,
+ .get_fw_errcode = bxt_get_errorcode,
+};
+
+static struct sst_ops skl_ops = {
+ .irq_handler = skl_dsp_sst_interrupt,
+ .write = sst_shim32_write,
+ .read = sst_shim32_read,
+ .ram_read = sst_memcpy_fromio_32,
+ .ram_write = sst_memcpy_toio_32,
+ .free = skl_dsp_free,
+};
+
+static struct sst_dsp_device skl_dev = {
+ .thread = skl_dsp_irq_thread_handler,
+ .ops = &skl_ops,
+};
+
+int bxt_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq,
+ const char *fw_name, struct skl_dsp_loader_ops dsp_ops,
+ struct skl_sst **dsp)
+{
+ struct skl_sst *skl;
+ struct sst_dsp *sst;
+ int ret;
+
+ skl = devm_kzalloc(dev, sizeof(*skl), GFP_KERNEL);
+ if (skl == NULL)
+ return -ENOMEM;
+
+ skl->dev = dev;
+ skl_dev.thread_context = skl;
+
+ skl->dsp = skl_dsp_ctx_init(dev, &skl_dev, irq);
+ if (!skl->dsp) {
+ dev_err(skl->dev, "skl_dsp_ctx_init failed\n");
+ return -ENODEV;
+ }
+
+ sst = skl->dsp;
+ sst->fw_name = fw_name;
+ sst->dsp_ops = dsp_ops;
+ sst->fw_ops = bxt_fw_ops;
+ sst->addr.lpe = mmio_base;
+ sst->addr.shim = mmio_base;
+
+ sst_dsp_mailbox_init(sst, (BXT_ADSP_SRAM0_BASE + SKL_ADSP_W0_STAT_SZ),
+ SKL_ADSP_W0_UP_SZ, BXT_ADSP_SRAM1_BASE, SKL_ADSP_W1_SZ);
+
+ ret = skl_ipc_init(dev, skl);
+ if (ret)
+ return ret;
+
+ skl->boot_complete = false;
+ init_waitqueue_head(&skl->boot_wait);
+
+ ret = sst->fw_ops.load_fw(sst);
+ if (ret < 0) {
+ dev_err(dev, "Load base fw failed: %x", ret);
+ return ret;
+ }
+
+ if (dsp)
+ *dsp = skl;
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(bxt_sst_dsp_init);
+
+
+void bxt_sst_dsp_cleanup(struct device *dev, struct skl_sst *ctx)
+{
+ skl_ipc_free(&ctx->ipc);
+ ctx->dsp->cl_dev.ops.cl_cleanup_controller(ctx->dsp);
+
+ if (ctx->dsp->addr.lpe)
+ iounmap(ctx->dsp->addr.lpe);
+
+ ctx->dsp->ops->free(ctx->dsp);
+}
+EXPORT_SYMBOL_GPL(bxt_sst_dsp_cleanup);
+
+MODULE_LICENSE("GPL v2");
+MODULE_DESCRIPTION("Intel Broxton IPC driver");
diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c
index 79c5089b85d6..226db84ba20f 100644
--- a/sound/soc/intel/skylake/skl-messages.c
+++ b/sound/soc/intel/skylake/skl-messages.c
@@ -72,6 +72,105 @@ static void skl_dsp_enable_notification(struct skl_sst *ctx, bool enable)
skl_ipc_set_large_config(&ctx->ipc, &msg, (u32 *)&mask);
}
+static int skl_dsp_setup_spib(struct device *dev, unsigned int size,
+ int stream_tag, int enable)
+{
+ struct hdac_ext_bus *ebus = dev_get_drvdata(dev);
+ struct hdac_bus *bus = ebus_to_hbus(ebus);
+ struct hdac_stream *stream = snd_hdac_get_stream(bus,
+ SNDRV_PCM_STREAM_PLAYBACK, stream_tag);
+ struct hdac_ext_stream *estream;
+
+ if (!stream)
+ return -EINVAL;
+
+ estream = stream_to_hdac_ext_stream(stream);
+ /* enable/disable SPIB for this hdac stream */
+ snd_hdac_ext_stream_spbcap_enable(ebus, enable, stream->index);
+
+ /* set the spib value */
+ snd_hdac_ext_stream_set_spib(ebus, estream, size);
+
+ return 0;
+}
+
+static int skl_dsp_prepare(struct device *dev, unsigned int format,
+ unsigned int size, struct snd_dma_buffer *dmab)
+{
+ struct hdac_ext_bus *ebus = dev_get_drvdata(dev);
+ struct hdac_bus *bus = ebus_to_hbus(ebus);
+ struct hdac_ext_stream *estream;
+ struct hdac_stream *stream;
+ struct snd_pcm_substream substream;
+ int ret;
+
+ if (!bus)
+ return -ENODEV;
+
+ memset(&substream, 0, sizeof(substream));
+ substream.stream = SNDRV_PCM_STREAM_PLAYBACK;
+
+ estream = snd_hdac_ext_stream_assign(ebus, &substream,
+ HDAC_EXT_STREAM_TYPE_HOST);
+ if (!estream)
+ return -ENODEV;
+
+ stream = hdac_stream(estream);
+
+ /* assign decouple host dma channel */
+ ret = snd_hdac_dsp_prepare(stream, format, size, dmab);
+ if (ret < 0)
+ return ret;
+
+ skl_dsp_setup_spib(dev, size, stream->stream_tag, true);
+
+ return stream->stream_tag;
+}
+
+static int skl_dsp_trigger(struct device *dev, bool start, int stream_tag)
+{
+ struct hdac_ext_bus *ebus = dev_get_drvdata(dev);
+ struct hdac_stream *stream;
+ struct hdac_bus *bus = ebus_to_hbus(ebus);
+
+ if (!bus)
+ return -ENODEV;
+
+ stream = snd_hdac_get_stream(bus,
+ SNDRV_PCM_STREAM_PLAYBACK, stream_tag);
+ if (!stream)
+ return -EINVAL;
+
+ snd_hdac_dsp_trigger(stream, start);
+
+ return 0;
+}
+
+static int skl_dsp_cleanup(struct device *dev,
+ struct snd_dma_buffer *dmab, int stream_tag)
+{
+ struct hdac_ext_bus *ebus = dev_get_drvdata(dev);
+ struct hdac_stream *stream;
+ struct hdac_ext_stream *estream;
+ struct hdac_bus *bus = ebus_to_hbus(ebus);
+
+ if (!bus)
+ return -ENODEV;
+
+ stream = snd_hdac_get_stream(bus,
+ SNDRV_PCM_STREAM_PLAYBACK, stream_tag);
+ if (!stream)
+ return -EINVAL;
+
+ estream = stream_to_hdac_ext_stream(stream);
+ skl_dsp_setup_spib(dev, 0, stream_tag, false);
+ snd_hdac_ext_stream_release(estream, HDAC_EXT_STREAM_TYPE_HOST);
+
+ snd_hdac_dsp_cleanup(stream, dmab);
+
+ return 0;
+}
+
static struct skl_dsp_loader_ops skl_get_loader_ops(void)
{
struct skl_dsp_loader_ops loader_ops;
@@ -84,6 +183,21 @@ static struct skl_dsp_loader_ops skl_get_loader_ops(void)
return loader_ops;
};
+static struct skl_dsp_loader_ops bxt_get_loader_ops(void)
+{
+ struct skl_dsp_loader_ops loader_ops;
+
+ memset(&loader_ops, 0, sizeof(loader_ops));
+
+ loader_ops.alloc_dma_buf = skl_alloc_dma_buf;
+ loader_ops.free_dma_buf = skl_free_dma_buf;
+ loader_ops.prepare = skl_dsp_prepare;
+ loader_ops.trigger = skl_dsp_trigger;
+ loader_ops.cleanup = skl_dsp_cleanup;
+
+ return loader_ops;
+};
+
static const struct skl_dsp_ops dsp_ops[] = {
{
.id = 0x9d70,
@@ -91,6 +205,12 @@ static const struct skl_dsp_ops dsp_ops[] = {
.init = skl_sst_dsp_init,
.cleanup = skl_sst_dsp_cleanup
},
+ {
+ .id = 0x5a98,
+ .loader_ops = bxt_get_loader_ops,
+ .init = bxt_sst_dsp_init,
+ .cleanup = bxt_sst_dsp_cleanup
+ },
};
static int skl_get_dsp_ops(int pci_id)
@@ -744,7 +864,7 @@ int skl_init_module(struct skl_sst *ctx,
return ret;
}
mconfig->m_state = SKL_MODULE_INIT_DONE;
-
+ kfree(param_data);
return ret;
}
diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c
index 14d1916ea9f8..7d73648e5f9a 100644
--- a/sound/soc/intel/skylake/skl-nhlt.c
+++ b/sound/soc/intel/skylake/skl-nhlt.c
@@ -25,11 +25,12 @@ static u8 OSC_UUID[16] = {0x6E, 0x88, 0x9F, 0xA6, 0xEB, 0x6C, 0x94, 0x45,
#define DSDT_NHLT_PATH "\\_SB.PCI0.HDAS"
-void *skl_nhlt_init(struct device *dev)
+struct nhlt_acpi_table *skl_nhlt_init(struct device *dev)
{
acpi_handle handle;
union acpi_object *obj;
struct nhlt_resource_desc *nhlt_ptr = NULL;
+ struct nhlt_acpi_table *nhlt_table = NULL;
if (ACPI_FAILURE(acpi_get_handle(NULL, DSDT_NHLT_PATH, &handle))) {
dev_err(dev, "Requested NHLT device not found\n");
@@ -39,18 +40,20 @@ void *skl_nhlt_init(struct device *dev)
obj = acpi_evaluate_dsm(handle, OSC_UUID, 1, 1, NULL);
if (obj && obj->type == ACPI_TYPE_BUFFER) {
nhlt_ptr = (struct nhlt_resource_desc *)obj->buffer.pointer;
-
- return memremap(nhlt_ptr->min_addr, nhlt_ptr->length,
+ nhlt_table = (struct nhlt_acpi_table *)
+ memremap(nhlt_ptr->min_addr, nhlt_ptr->length,
MEMREMAP_WB);
+ ACPI_FREE(obj);
+ return nhlt_table;
}
dev_err(dev, "device specific method to extract NHLT blob failed\n");
return NULL;
}
-void skl_nhlt_free(void *addr)
+void skl_nhlt_free(struct nhlt_acpi_table *nhlt)
{
- memunmap(addr);
+ memunmap((void *) nhlt);
}
static struct nhlt_specific_cfg *skl_get_specific_cfg(
@@ -120,7 +123,7 @@ struct nhlt_specific_cfg
struct hdac_bus *bus = ebus_to_hbus(&skl->ebus);
struct device *dev = bus->dev;
struct nhlt_specific_cfg *sp_config;
- struct nhlt_acpi_table *nhlt = (struct nhlt_acpi_table *)skl->nhlt;
+ struct nhlt_acpi_table *nhlt = skl->nhlt;
u16 bps = (s_fmt == 16) ? 16 : 32;
u8 j;
diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c
index dab0900eef26..7c81b31748ff 100644
--- a/sound/soc/intel/skylake/skl-pcm.c
+++ b/sound/soc/intel/skylake/skl-pcm.c
@@ -51,7 +51,7 @@ static struct snd_pcm_hardware azx_pcm_hw = {
.rate_min = 8000,
.rate_max = 48000,
.channels_min = 1,
- .channels_max = HDA_QUAD,
+ .channels_max = 8,
.buffer_bytes_max = AZX_MAX_BUF_SIZE,
.period_bytes_min = 128,
.period_bytes_max = AZX_MAX_BUF_SIZE / 2,
@@ -213,7 +213,7 @@ static int skl_be_prepare(struct snd_pcm_substream *substream,
struct skl_sst *ctx = skl->skl_sst;
struct skl_module_cfg *mconfig;
- if ((dai->playback_active > 1) || (dai->capture_active > 1))
+ if (dai->playback_widget->power || dai->capture_widget->power)
return 0;
mconfig = skl_tplg_be_get_cpr_module(dai, substream->stream);
@@ -402,23 +402,33 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd,
struct skl_module_cfg *mconfig;
struct hdac_ext_bus *ebus = get_bus_ctx(substream);
struct hdac_ext_stream *stream = get_hdac_ext_stream(substream);
+ struct snd_soc_dapm_widget *w;
int ret;
mconfig = skl_tplg_fe_get_cpr_module(dai, substream->stream);
if (!mconfig)
return -EIO;
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ w = dai->playback_widget;
+ else
+ w = dai->capture_widget;
+
switch (cmd) {
case SNDRV_PCM_TRIGGER_RESUME:
- skl_pcm_prepare(substream, dai);
- /*
- * enable DMA Resume enable bit for the stream, set the dpib
- * & lpib position to resune before starting the DMA
- */
- snd_hdac_ext_stream_drsm_enable(ebus, true,
- hdac_stream(stream)->index);
- snd_hdac_ext_stream_set_dpibr(ebus, stream, stream->dpib);
- snd_hdac_ext_stream_set_lpib(stream, stream->lpib);
+ if (!w->ignore_suspend) {
+ skl_pcm_prepare(substream, dai);
+ /*
+ * enable DMA Resume enable bit for the stream, set the
+ * dpib & lpib position to resume before starting the
+ * DMA
+ */
+ snd_hdac_ext_stream_drsm_enable(ebus, true,
+ hdac_stream(stream)->index);
+ snd_hdac_ext_stream_set_dpibr(ebus, stream,
+ stream->dpib);
+ snd_hdac_ext_stream_set_lpib(stream, stream->lpib);
+ }
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
@@ -448,7 +458,7 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd,
return ret;
ret = skl_decoupled_trigger(substream, cmd);
- if (cmd == SNDRV_PCM_TRIGGER_SUSPEND) {
+ if ((cmd == SNDRV_PCM_TRIGGER_SUSPEND) && !w->ignore_suspend) {
/* save the dpib and lpib positions */
stream->dpib = readl(ebus->bus.remap_addr +
AZX_REG_VS_SDXDPIB_XBASE +
@@ -523,7 +533,6 @@ static int skl_link_pcm_prepare(struct snd_pcm_substream *substream,
if (!link)
return -EINVAL;
- snd_hdac_ext_bus_link_power_up(link);
snd_hdac_ext_link_stream_reset(link_dev);
snd_hdac_ext_link_stream_setup(link_dev, format_val);
@@ -682,7 +691,7 @@ static struct snd_soc_dai_driver skl_platform_dai[] = {
.playback = {
.stream_name = "HDMI1 Playback",
.channels_min = HDA_STEREO,
- .channels_max = HDA_STEREO,
+ .channels_max = 8,
.rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |
SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |
SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 |
@@ -697,7 +706,7 @@ static struct snd_soc_dai_driver skl_platform_dai[] = {
.playback = {
.stream_name = "HDMI2 Playback",
.channels_min = HDA_STEREO,
- .channels_max = HDA_STEREO,
+ .channels_max = 8,
.rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |
SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |
SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 |
@@ -712,7 +721,7 @@ static struct snd_soc_dai_driver skl_platform_dai[] = {
.playback = {
.stream_name = "HDMI3 Playback",
.channels_min = HDA_STEREO,
- .channels_max = HDA_STEREO,
+ .channels_max = 8,
.rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |
SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |
SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 |
@@ -760,12 +769,84 @@ static struct snd_soc_dai_driver skl_platform_dai[] = {
},
},
{
+ .name = "SSP2 Pin",
+ .ops = &skl_be_ssp_dai_ops,
+ .playback = {
+ .stream_name = "ssp2 Tx",
+ .channels_min = HDA_STEREO,
+ .channels_max = HDA_STEREO,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .capture = {
+ .stream_name = "ssp2 Rx",
+ .channels_min = HDA_STEREO,
+ .channels_max = HDA_STEREO,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+},
+{
+ .name = "SSP3 Pin",
+ .ops = &skl_be_ssp_dai_ops,
+ .playback = {
+ .stream_name = "ssp3 Tx",
+ .channels_min = HDA_STEREO,
+ .channels_max = HDA_STEREO,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .capture = {
+ .stream_name = "ssp3 Rx",
+ .channels_min = HDA_STEREO,
+ .channels_max = HDA_STEREO,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+},
+{
+ .name = "SSP4 Pin",
+ .ops = &skl_be_ssp_dai_ops,
+ .playback = {
+ .stream_name = "ssp4 Tx",
+ .channels_min = HDA_STEREO,
+ .channels_max = HDA_STEREO,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .capture = {
+ .stream_name = "ssp4 Rx",
+ .channels_min = HDA_STEREO,
+ .channels_max = HDA_STEREO,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+},
+{
+ .name = "SSP5 Pin",
+ .ops = &skl_be_ssp_dai_ops,
+ .playback = {
+ .stream_name = "ssp5 Tx",
+ .channels_min = HDA_STEREO,
+ .channels_max = HDA_STEREO,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .capture = {
+ .stream_name = "ssp5 Rx",
+ .channels_min = HDA_STEREO,
+ .channels_max = HDA_STEREO,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+},
+{
.name = "iDisp1 Pin",
.ops = &skl_link_dai_ops,
.playback = {
.stream_name = "iDisp1 Tx",
.channels_min = HDA_STEREO,
- .channels_max = HDA_STEREO,
+ .channels_max = 8,
.rates = SNDRV_PCM_RATE_8000|SNDRV_PCM_RATE_16000|SNDRV_PCM_RATE_48000,
.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE |
SNDRV_PCM_FMTBIT_S24_LE,
@@ -777,7 +858,7 @@ static struct snd_soc_dai_driver skl_platform_dai[] = {
.playback = {
.stream_name = "iDisp2 Tx",
.channels_min = HDA_STEREO,
- .channels_max = HDA_STEREO,
+ .channels_max = 8,
.rates = SNDRV_PCM_RATE_8000|SNDRV_PCM_RATE_16000|
SNDRV_PCM_RATE_48000,
.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE |
@@ -790,7 +871,7 @@ static struct snd_soc_dai_driver skl_platform_dai[] = {
.playback = {
.stream_name = "iDisp3 Tx",
.channels_min = HDA_STEREO,
- .channels_max = HDA_STEREO,
+ .channels_max = 8,
.rates = SNDRV_PCM_RATE_8000|SNDRV_PCM_RATE_16000|
SNDRV_PCM_RATE_48000,
.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE |
diff --git a/sound/soc/intel/skylake/skl-sst-dsp.c b/sound/soc/intel/skylake/skl-sst-dsp.c
index a5267e8a96e0..13c19855ee1a 100644
--- a/sound/soc/intel/skylake/skl-sst-dsp.c
+++ b/sound/soc/intel/skylake/skl-sst-dsp.c
@@ -336,6 +336,9 @@ void skl_dsp_free(struct sst_dsp *dsp)
skl_ipc_int_disable(dsp);
free_irq(dsp->irq, dsp);
+ skl_ipc_op_int_disable(dsp);
+ skl_ipc_int_disable(dsp);
+
skl_dsp_disable_core(dsp);
}
EXPORT_SYMBOL_GPL(skl_dsp_free);
diff --git a/sound/soc/intel/skylake/skl-sst-dsp.h b/sound/soc/intel/skylake/skl-sst-dsp.h
index b6e310d49dd6..deabe7308d3b 100644
--- a/sound/soc/intel/skylake/skl-sst-dsp.h
+++ b/sound/soc/intel/skylake/skl-sst-dsp.h
@@ -118,16 +118,25 @@ struct skl_dsp_fw_ops {
int (*set_state_D0)(struct sst_dsp *ctx);
int (*set_state_D3)(struct sst_dsp *ctx);
unsigned int (*get_fw_errcode)(struct sst_dsp *ctx);
- int (*load_mod)(struct sst_dsp *ctx, u16 mod_id, char *mod_name);
+ int (*load_mod)(struct sst_dsp *ctx, u16 mod_id, u8 *mod_name);
int (*unload_mod)(struct sst_dsp *ctx, u16 mod_id);
};
struct skl_dsp_loader_ops {
+ int stream_tag;
+
int (*alloc_dma_buf)(struct device *dev,
struct snd_dma_buffer *dmab, size_t size);
int (*free_dma_buf)(struct device *dev,
struct snd_dma_buffer *dmab);
+ int (*prepare)(struct device *dev, unsigned int format,
+ unsigned int byte_size,
+ struct snd_dma_buffer *bufp);
+ int (*trigger)(struct device *dev, bool start, int stream_tag);
+
+ int (*cleanup)(struct device *dev, struct snd_dma_buffer *dmab,
+ int stream_tag);
};
struct skl_load_module_info {
@@ -160,6 +169,10 @@ int skl_dsp_boot(struct sst_dsp *ctx);
int skl_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq,
const char *fw_name, struct skl_dsp_loader_ops dsp_ops,
struct skl_sst **dsp);
+int bxt_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq,
+ const char *fw_name, struct skl_dsp_loader_ops dsp_ops,
+ struct skl_sst **dsp);
void skl_sst_dsp_cleanup(struct device *dev, struct skl_sst *ctx);
+void bxt_sst_dsp_cleanup(struct device *dev, struct skl_sst *ctx);
#endif /*__SKL_SST_DSP_H__*/
diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c
index 348a734f8e24..13ec8d53b526 100644
--- a/sound/soc/intel/skylake/skl-sst.c
+++ b/sound/soc/intel/skylake/skl-sst.c
@@ -20,6 +20,7 @@
#include <linux/delay.h>
#include <linux/device.h>
#include <linux/err.h>
+#include <linux/uuid.h>
#include "../common/sst-dsp.h"
#include "../common/sst-dsp-priv.h"
#include "../common/sst-ipc.h"
@@ -304,14 +305,16 @@ static int skl_transfer_module(struct sst_dsp *ctx,
return ret;
}
-static int skl_load_module(struct sst_dsp *ctx, u16 mod_id, char *guid)
+static int skl_load_module(struct sst_dsp *ctx, u16 mod_id, u8 *guid)
{
struct skl_module_table *module_entry = NULL;
int ret = 0;
char mod_name[64]; /* guid str = 32 chars + 4 hyphens */
+ uuid_le *uuid_mod;
- snprintf(mod_name, sizeof(mod_name), "%s%s%s",
- "intel/dsp_fw_", guid, ".bin");
+ uuid_mod = (uuid_le *)guid;
+ snprintf(mod_name, sizeof(mod_name), "%s%pUL%s",
+ "intel/dsp_fw_", uuid_mod, ".bin");
module_entry = skl_module_get_from_id(ctx, mod_id);
if (module_entry == NULL) {
@@ -451,6 +454,10 @@ void skl_sst_dsp_cleanup(struct device *dev, struct skl_sst *ctx)
skl_clear_module_table(ctx->dsp);
skl_ipc_free(&ctx->ipc);
ctx->dsp->ops->free(ctx->dsp);
+ if (ctx->boot_complete) {
+ ctx->dsp->cl_dev.ops.cl_cleanup_controller(ctx->dsp);
+ skl_cldma_int_disable(ctx->dsp);
+ }
}
EXPORT_SYMBOL_GPL(skl_sst_dsp_cleanup);
diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c
index 545b4e77b8aa..3e036b0349b9 100644
--- a/sound/soc/intel/skylake/skl-topology.c
+++ b/sound/soc/intel/skylake/skl-topology.c
@@ -154,13 +154,32 @@ static void skl_dump_mconfig(struct skl_sst *ctx,
dev_dbg(ctx->dev, "ch_cfg = %d\n", mcfg->out_fmt[0].ch_cfg);
}
+static void skl_tplg_update_chmap(struct skl_module_fmt *fmt, int chs)
+{
+ int slot_map = 0xFFFFFFFF;
+ int start_slot = 0;
+ int i;
+
+ for (i = 0; i < chs; i++) {
+ /*
+ * For 2 channels with starting slot as 0, slot map will
+ * look like 0xFFFFFF10.
+ */
+ slot_map &= (~(0xF << (4 * i)) | (start_slot << (4 * i)));
+ start_slot++;
+ }
+ fmt->ch_map = slot_map;
+}
+
static void skl_tplg_update_params(struct skl_module_fmt *fmt,
struct skl_pipe_params *params, int fixup)
{
if (fixup & SKL_RATE_FIXUP_MASK)
fmt->s_freq = params->s_freq;
- if (fixup & SKL_CH_FIXUP_MASK)
+ if (fixup & SKL_CH_FIXUP_MASK) {
fmt->channels = params->ch;
+ skl_tplg_update_chmap(fmt, fmt->channels);
+ }
if (fixup & SKL_FMT_FIXUP_MASK) {
fmt->valid_bit_depth = skl_get_bit_depth(params->s_fmt);
@@ -239,6 +258,7 @@ static void skl_tplg_update_buffer_size(struct skl_sst *ctx,
{
int multiplier = 1;
struct skl_module_fmt *in_fmt, *out_fmt;
+ int in_rate, out_rate;
/* Since fixups is applied to pin 0 only, ibs, obs needs
@@ -249,15 +269,24 @@ static void skl_tplg_update_buffer_size(struct skl_sst *ctx,
if (mcfg->m_type == SKL_MODULE_TYPE_SRCINT)
multiplier = 5;
- mcfg->ibs = (in_fmt->s_freq / 1000) *
- (mcfg->in_fmt->channels) *
- (mcfg->in_fmt->bit_depth >> 3) *
- multiplier;
-
- mcfg->obs = (mcfg->out_fmt->s_freq / 1000) *
- (mcfg->out_fmt->channels) *
- (mcfg->out_fmt->bit_depth >> 3) *
- multiplier;
+
+ if (in_fmt->s_freq % 1000)
+ in_rate = (in_fmt->s_freq / 1000) + 1;
+ else
+ in_rate = (in_fmt->s_freq / 1000);
+
+ mcfg->ibs = in_rate * (mcfg->in_fmt->channels) *
+ (mcfg->in_fmt->bit_depth >> 3) *
+ multiplier;
+
+ if (mcfg->out_fmt->s_freq % 1000)
+ out_rate = (mcfg->out_fmt->s_freq / 1000) + 1;
+ else
+ out_rate = (mcfg->out_fmt->s_freq / 1000);
+
+ mcfg->obs = out_rate * (mcfg->out_fmt->channels) *
+ (mcfg->out_fmt->bit_depth >> 3) *
+ multiplier;
}
static int skl_tplg_update_be_blob(struct snd_soc_dapm_widget *w,
@@ -485,11 +514,15 @@ skl_tplg_init_pipe_modules(struct skl *skl, struct skl_pipe *pipe)
if (!skl_is_pipe_mcps_avail(skl, mconfig))
return -ENOMEM;
+ skl_tplg_alloc_pipe_mcps(skl, mconfig);
+
if (mconfig->is_loadable && ctx->dsp->fw_ops.load_mod) {
ret = ctx->dsp->fw_ops.load_mod(ctx->dsp,
mconfig->id.module_id, mconfig->guid);
if (ret < 0)
return ret;
+
+ mconfig->m_state = SKL_MODULE_LOADED;
}
/* update blob if blob is null for be with default value */
@@ -509,7 +542,6 @@ skl_tplg_init_pipe_modules(struct skl *skl, struct skl_pipe *pipe)
ret = skl_tplg_set_module_params(w, ctx);
if (ret < 0)
return ret;
- skl_tplg_alloc_pipe_mcps(skl, mconfig);
}
return 0;
@@ -524,7 +556,8 @@ static int skl_tplg_unload_pipe_modules(struct skl_sst *ctx,
list_for_each_entry(w_module, &pipe->w_list, node) {
mconfig = w_module->w->priv;
- if (mconfig->is_loadable && ctx->dsp->fw_ops.unload_mod)
+ if (mconfig->is_loadable && ctx->dsp->fw_ops.unload_mod &&
+ mconfig->m_state > SKL_MODULE_UNINIT)
return ctx->dsp->fw_ops.unload_mod(ctx->dsp,
mconfig->id.module_id);
}
@@ -558,6 +591,9 @@ static int skl_tplg_mixer_dapm_pre_pmu_event(struct snd_soc_dapm_widget *w,
if (!skl_is_pipe_mem_avail(skl, mconfig))
return -ENOMEM;
+ skl_tplg_alloc_pipe_mem(skl, mconfig);
+ skl_tplg_alloc_pipe_mcps(skl, mconfig);
+
/*
* Create a list of modules for pipe.
* This list contains modules from source to sink
@@ -601,9 +637,6 @@ static int skl_tplg_mixer_dapm_pre_pmu_event(struct snd_soc_dapm_widget *w,
src_module = dst_module;
}
- skl_tplg_alloc_pipe_mem(skl, mconfig);
- skl_tplg_alloc_pipe_mcps(skl, mconfig);
-
return 0;
}
@@ -1550,6 +1583,8 @@ static int skl_tplg_widget_load(struct snd_soc_component *cmpnt,
return -ENOMEM;
w->priv = mconfig;
+ memcpy(&mconfig->guid, &dfw_config->uuid, 16);
+
mconfig->id.module_id = dfw_config->module_id;
mconfig->id.instance_id = dfw_config->instance_id;
mconfig->mcps = dfw_config->max_mcps;
@@ -1579,10 +1614,6 @@ static int skl_tplg_widget_load(struct snd_soc_component *cmpnt,
mconfig->time_slot = dfw_config->time_slot;
mconfig->formats_config.caps_size = dfw_config->caps.caps_size;
- if (dfw_config->is_loadable)
- memcpy(mconfig->guid, dfw_config->uuid,
- ARRAY_SIZE(dfw_config->uuid));
-
mconfig->m_in_pin = devm_kzalloc(bus->dev, (mconfig->max_in_queue) *
sizeof(*mconfig->m_in_pin),
GFP_KERNEL);
diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h
index de3c401284d9..e4b399cd7868 100644
--- a/sound/soc/intel/skylake/skl-topology.h
+++ b/sound/soc/intel/skylake/skl-topology.h
@@ -274,14 +274,14 @@ struct skl_pipe {
enum skl_module_state {
SKL_MODULE_UNINIT = 0,
- SKL_MODULE_INIT_DONE = 1,
- SKL_MODULE_LOADED = 2,
- SKL_MODULE_UNLOADED = 3,
- SKL_MODULE_BIND_DONE = 4
+ SKL_MODULE_LOADED = 1,
+ SKL_MODULE_INIT_DONE = 2,
+ SKL_MODULE_BIND_DONE = 3,
+ SKL_MODULE_UNLOADED = 4,
};
struct skl_module_cfg {
- char guid[SKL_UUID_STR_SZ];
+ u8 guid[16];
struct skl_module_inst_id id;
u8 domain;
bool homogenous_inputs;
diff --git a/sound/soc/intel/skylake/skl-tplg-interface.h b/sound/soc/intel/skylake/skl-tplg-interface.h
index 1db88a63ac17..a32e5e9cc530 100644
--- a/sound/soc/intel/skylake/skl-tplg-interface.h
+++ b/sound/soc/intel/skylake/skl-tplg-interface.h
@@ -181,7 +181,7 @@ struct skl_dfw_pipe {
} __packed;
struct skl_dfw_module {
- char uuid[SKL_UUID_STR_SZ];
+ u8 uuid[16];
u16 module_id;
u16 instance_id;
diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c
index ab5e25aaeee3..06d8c263c68f 100644
--- a/sound/soc/intel/skylake/skl.c
+++ b/sound/soc/intel/skylake/skl.c
@@ -222,20 +222,36 @@ static int skl_suspend(struct device *dev)
struct hdac_ext_bus *ebus = pci_get_drvdata(pci);
struct skl *skl = ebus_to_skl(ebus);
struct hdac_bus *bus = ebus_to_hbus(ebus);
+ int ret = 0;
/*
* Do not suspend if streams which are marked ignore suspend are
* running, we need to save the state for these and continue
*/
if (skl->supend_active) {
+ /* turn off the links and stop the CORB/RIRB DMA if it is On */
snd_hdac_ext_bus_link_power_down_all(ebus);
+
+ if (ebus->cmd_dma_state)
+ snd_hdac_bus_stop_cmd_io(&ebus->bus);
+
enable_irq_wake(bus->irq);
pci_save_state(pci);
pci_disable_device(pci);
- return 0;
} else {
- return _skl_suspend(ebus);
+ ret = _skl_suspend(ebus);
+ if (ret < 0)
+ return ret;
+ }
+
+ if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) {
+ ret = snd_hdac_display_power(bus, false);
+ if (ret < 0)
+ dev_err(bus->dev,
+ "Cannot turn OFF display power on i915\n");
}
+
+ return ret;
}
static int skl_resume(struct device *dev)
@@ -244,6 +260,7 @@ static int skl_resume(struct device *dev)
struct hdac_ext_bus *ebus = pci_get_drvdata(pci);
struct skl *skl = ebus_to_skl(ebus);
struct hdac_bus *bus = ebus_to_hbus(ebus);
+ struct hdac_ext_link *hlink = NULL;
int ret;
/* Turned OFF in HDMI codec driver after codec reconfiguration */
@@ -265,8 +282,29 @@ static int skl_resume(struct device *dev)
ret = pci_enable_device(pci);
snd_hdac_ext_bus_link_power_up_all(ebus);
disable_irq_wake(bus->irq);
+ /*
+ * turn On the links which are On before active suspend
+ * and start the CORB/RIRB DMA if On before
+ * active suspend.
+ */
+ list_for_each_entry(hlink, &ebus->hlink_list, list) {
+ if (hlink->ref_count)
+ snd_hdac_ext_bus_link_power_up(hlink);
+ }
+
+ if (ebus->cmd_dma_state)
+ snd_hdac_bus_init_cmd_io(&ebus->bus);
} else {
ret = _skl_resume(ebus);
+
+ /* turn off the links which are off before suspend */
+ list_for_each_entry(hlink, &ebus->hlink_list, list) {
+ if (!hlink->ref_count)
+ snd_hdac_ext_bus_link_power_down(hlink);
+ }
+
+ if (!ebus->cmd_dma_state)
+ snd_hdac_bus_stop_cmd_io(&ebus->bus);
}
return ret;
@@ -316,17 +354,20 @@ static int skl_free(struct hdac_ext_bus *ebus)
if (bus->irq >= 0)
free_irq(bus->irq, (void *)bus);
- if (bus->remap_addr)
- iounmap(bus->remap_addr);
-
snd_hdac_bus_free_stream_pages(bus);
snd_hdac_stream_free_all(ebus);
snd_hdac_link_free_all(ebus);
+
+ if (bus->remap_addr)
+ iounmap(bus->remap_addr);
+
pci_release_regions(skl->pci);
pci_disable_device(skl->pci);
snd_hdac_ext_bus_exit(ebus);
+ if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI))
+ snd_hdac_i915_exit(&ebus->bus);
return 0;
}
@@ -599,6 +640,7 @@ static int skl_probe(struct pci_dev *pci,
struct skl *skl;
struct hdac_ext_bus *ebus = NULL;
struct hdac_bus *bus = NULL;
+ struct hdac_ext_link *hlink = NULL;
int err;
/* we use ext core ops, so provide NULL for ops here */
@@ -629,7 +671,7 @@ static int skl_probe(struct pci_dev *pci,
err = skl_machine_device_register(skl,
(void *)pci_id->driver_data);
if (err < 0)
- goto out_free;
+ goto out_nhlt_free;
err = skl_init_dsp(skl);
if (err < 0) {
@@ -665,6 +707,12 @@ static int skl_probe(struct pci_dev *pci,
}
}
+ /*
+ * we are done probling so decrement link counts
+ */
+ list_for_each_entry(hlink, &ebus->hlink_list, list)
+ snd_hdac_ext_bus_link_put(ebus, hlink);
+
/*configure PM */
pm_runtime_put_noidle(bus->dev);
pm_runtime_allow(bus->dev);
@@ -679,6 +727,8 @@ out_dsp_free:
skl_free_dsp(skl);
out_mach_free:
skl_machine_device_unregister(skl);
+out_nhlt_free:
+ skl_nhlt_free(skl->nhlt);
out_free:
skl->init_failed = 1;
skl_free(ebus);
@@ -719,16 +769,17 @@ static void skl_remove(struct pci_dev *pci)
if (skl->tplg)
release_firmware(skl->tplg);
- if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI))
- snd_hdac_i915_exit(&ebus->bus);
-
if (pci_dev_run_wake(pci))
pm_runtime_get_noresume(&pci->dev);
- pci_dev_put(pci);
+
+ /* codec removal, invoke bus_device_remove */
+ snd_hdac_ext_bus_device_remove(ebus);
+
skl_platform_unregister(&pci->dev);
skl_free_dsp(skl);
skl_machine_device_unregister(skl);
skl_dmic_device_unregister(skl);
+ skl_nhlt_free(skl->nhlt);
skl_free(ebus);
dev_set_drvdata(&pci->dev, NULL);
}
diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h
index 39e16fa7a92b..4b4b3876aea9 100644
--- a/sound/soc/intel/skylake/skl.h
+++ b/sound/soc/intel/skylake/skl.h
@@ -66,7 +66,7 @@ struct skl {
struct platform_device *dmic_dev;
struct platform_device *i2s_dev;
- void *nhlt; /* nhlt ptr */
+ struct nhlt_acpi_table *nhlt; /* nhlt ptr */
struct skl_sst *skl_sst; /* sst skl ctx */
struct skl_dsp_resource resource;
@@ -103,8 +103,8 @@ struct skl_dsp_ops {
int skl_platform_unregister(struct device *dev);
int skl_platform_register(struct device *dev);
-void *skl_nhlt_init(struct device *dev);
-void skl_nhlt_free(void *addr);
+struct nhlt_acpi_table *skl_nhlt_init(struct device *dev);
+void skl_nhlt_free(struct nhlt_acpi_table *addr);
struct nhlt_specific_cfg *skl_get_ep_blob(struct skl *skl, u32 instance,
u8 link_type, u8 s_fmt, u8 no_ch, u32 s_rate, u8 dirn);
diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig
index 132bb83f8e99..bc3c7b5ac752 100644
--- a/sound/soc/kirkwood/Kconfig
+++ b/sound/soc/kirkwood/Kconfig
@@ -1,6 +1,7 @@
config SND_KIRKWOOD_SOC
tristate "SoC Audio for the Marvell Kirkwood and Dove chips"
depends on ARCH_DOVE || ARCH_MVEBU || COMPILE_TEST
+ depends on HAS_DMA
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/mediatek/Kconfig b/sound/soc/mediatek/Kconfig
index f7e789e97fbc..3abf51c07851 100644
--- a/sound/soc/mediatek/Kconfig
+++ b/sound/soc/mediatek/Kconfig
@@ -43,6 +43,7 @@ config SND_SOC_MT8173_RT5650_RT5676
depends on SND_SOC_MEDIATEK && I2C
select SND_SOC_RT5645
select SND_SOC_RT5677
+ select SND_SOC_HDMI_CODEC
help
This adds ASoC driver for Mediatek MT8173 boards
with the RT5650 and RT5676 codecs.
diff --git a/sound/soc/mediatek/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173-rt5650-rt5676.c
index 5c4c58c69c51..bb593926c62d 100644
--- a/sound/soc/mediatek/mt8173-rt5650-rt5676.c
+++ b/sound/soc/mediatek/mt8173-rt5650-rt5676.c
@@ -134,7 +134,9 @@ static struct snd_soc_dai_link_component mt8173_rt5650_rt5676_codecs[] = {
enum {
DAI_LINK_PLAYBACK,
DAI_LINK_CAPTURE,
+ DAI_LINK_HDMI,
DAI_LINK_CODEC_I2S,
+ DAI_LINK_HDMI_I2S,
DAI_LINK_INTERCODEC
};
@@ -161,6 +163,16 @@ static struct snd_soc_dai_link mt8173_rt5650_rt5676_dais[] = {
.dynamic = 1,
.dpcm_capture = 1,
},
+ [DAI_LINK_HDMI] = {
+ .name = "HDMI",
+ .stream_name = "HDMI PCM",
+ .cpu_dai_name = "HDMI",
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .dynamic = 1,
+ .dpcm_playback = 1,
+ },
/* Back End DAI links */
[DAI_LINK_CODEC_I2S] = {
@@ -177,6 +189,13 @@ static struct snd_soc_dai_link mt8173_rt5650_rt5676_dais[] = {
.dpcm_playback = 1,
.dpcm_capture = 1,
},
+ [DAI_LINK_HDMI_I2S] = {
+ .name = "HDMI BE",
+ .cpu_dai_name = "HDMIO",
+ .no_pcm = 1,
+ .codec_dai_name = "i2s-hifi",
+ .dpcm_playback = 1,
+ },
/* rt5676 <-> rt5650 intercodec link: Sets rt5676 I2S2 as master */
[DAI_LINK_INTERCODEC] = {
.name = "rt5650_rt5676 intercodec",
@@ -251,6 +270,14 @@ static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev)
mt8173_rt5650_rt5676_dais[DAI_LINK_INTERCODEC].codec_of_node =
mt8173_rt5650_rt5676_codecs[1].of_node;
+ mt8173_rt5650_rt5676_dais[DAI_LINK_HDMI_I2S].codec_of_node =
+ of_parse_phandle(pdev->dev.of_node, "mediatek,audio-codec", 2);
+ if (!mt8173_rt5650_rt5676_dais[DAI_LINK_HDMI_I2S].codec_of_node) {
+ dev_err(&pdev->dev,
+ "Property 'audio-codec' missing or invalid\n");
+ return -EINVAL;
+ }
+
card->dev = &pdev->dev;
platform_set_drvdata(pdev, card);
diff --git a/sound/soc/mediatek/mt8173-rt5650.c b/sound/soc/mediatek/mt8173-rt5650.c
index bb09bb1b7f1c..a27a6673dbe3 100644
--- a/sound/soc/mediatek/mt8173-rt5650.c
+++ b/sound/soc/mediatek/mt8173-rt5650.c
@@ -85,12 +85,29 @@ static int mt8173_rt5650_init(struct snd_soc_pcm_runtime *runtime)
{
struct snd_soc_card *card = runtime->card;
struct snd_soc_codec *codec = runtime->codec_dais[0]->codec;
+ const char *codec_capture_dai = runtime->codec_dais[1]->name;
int ret;
rt5645_sel_asrc_clk_src(codec,
- RT5645_DA_STEREO_FILTER |
- RT5645_AD_STEREO_FILTER,
+ RT5645_DA_STEREO_FILTER,
RT5645_CLK_SEL_I2S1_ASRC);
+
+ if (!strcmp(codec_capture_dai, "rt5645-aif1")) {
+ rt5645_sel_asrc_clk_src(codec,
+ RT5645_AD_STEREO_FILTER,
+ RT5645_CLK_SEL_I2S1_ASRC);
+ } else if (!strcmp(codec_capture_dai, "rt5645-aif2")) {
+ rt5645_sel_asrc_clk_src(codec,
+ RT5645_AD_STEREO_FILTER,
+ RT5645_CLK_SEL_I2S2_ASRC);
+ } else {
+ dev_warn(card->dev,
+ "Only one dai codec found in DTS, enabled rt5645 AD filter\n");
+ rt5645_sel_asrc_clk_src(codec,
+ RT5645_AD_STEREO_FILTER,
+ RT5645_CLK_SEL_I2S1_ASRC);
+ }
+
/* enable jack detection */
ret = snd_soc_card_jack_new(card, "Headset Jack",
SND_JACK_HEADPHONE | SND_JACK_MICROPHONE |
@@ -110,6 +127,11 @@ static int mt8173_rt5650_init(struct snd_soc_pcm_runtime *runtime)
static struct snd_soc_dai_link_component mt8173_rt5650_codecs[] = {
{
+ /* Playback */
+ .dai_name = "rt5645-aif1",
+ },
+ {
+ /* Capture */
.dai_name = "rt5645-aif1",
},
};
@@ -149,7 +171,7 @@ static struct snd_soc_dai_link mt8173_rt5650_dais[] = {
.cpu_dai_name = "I2S",
.no_pcm = 1,
.codecs = mt8173_rt5650_codecs,
- .num_codecs = 1,
+ .num_codecs = 2,
.init = mt8173_rt5650_init,
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBS_CFS,
@@ -177,6 +199,8 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev)
{
struct snd_soc_card *card = &mt8173_rt5650_card;
struct device_node *platform_node;
+ struct device_node *np;
+ const char *codec_capture_dai;
int i, ret;
platform_node = of_parse_phandle(pdev->dev.of_node,
@@ -199,6 +223,26 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev)
"Property 'audio-codec' missing or invalid\n");
return -EINVAL;
}
+ mt8173_rt5650_codecs[1].of_node = mt8173_rt5650_codecs[0].of_node;
+
+ if (of_find_node_by_name(platform_node, "codec-capture")) {
+ np = of_get_child_by_name(pdev->dev.of_node, "codec-capture");
+ if (!np) {
+ dev_err(&pdev->dev,
+ "%s: Can't find codec-capture DT node\n",
+ __func__);
+ return -EINVAL;
+ }
+ ret = snd_soc_of_get_dai_name(np, &codec_capture_dai);
+ if (ret < 0) {
+ dev_err(&pdev->dev,
+ "%s codec_capture_dai name fail %d\n",
+ __func__, ret);
+ return ret;
+ }
+ mt8173_rt5650_codecs[1].dai_name = codec_capture_dai;
+ }
+
card->dev = &pdev->dev;
platform_set_drvdata(pdev, card);
diff --git a/sound/soc/mediatek/mtk-afe-pcm.c b/sound/soc/mediatek/mtk-afe-pcm.c
index f1c58a2c12fb..2b5df2ef51a3 100644
--- a/sound/soc/mediatek/mtk-afe-pcm.c
+++ b/sound/soc/mediatek/mtk-afe-pcm.c
@@ -123,6 +123,7 @@
#define AFE_TDM_CON1_WLEN_32BIT (0x2 << 8)
#define AFE_TDM_CON1_MSB_ALIGNED (0x1 << 4)
#define AFE_TDM_CON1_1_BCK_DELAY (0x1 << 3)
+#define AFE_TDM_CON1_LRCK_INV (0x1 << 2)
#define AFE_TDM_CON1_BCK_INV (0x1 << 1)
#define AFE_TDM_CON1_EN (0x1 << 0)
@@ -449,6 +450,7 @@ static int mtk_afe_hdmi_prepare(struct snd_pcm_substream *substream,
runtime->rate * runtime->channels * 32);
val = AFE_TDM_CON1_BCK_INV |
+ AFE_TDM_CON1_LRCK_INV |
AFE_TDM_CON1_1_BCK_DELAY |
AFE_TDM_CON1_MSB_ALIGNED | /* I2S mode */
AFE_TDM_CON1_WLEN_32BIT |
diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c
index c7563e230c7d..4a16e778966b 100644
--- a/sound/soc/omap/mcbsp.c
+++ b/sound/soc/omap/mcbsp.c
@@ -260,6 +260,10 @@ static void omap_st_on(struct omap_mcbsp *mcbsp)
if (mcbsp->pdata->enable_st_clock)
mcbsp->pdata->enable_st_clock(mcbsp->id, 1);
+ /* Disable Sidetone clock auto-gating for normal operation */
+ w = MCBSP_ST_READ(mcbsp, SYSCONFIG);
+ MCBSP_ST_WRITE(mcbsp, SYSCONFIG, w & ~(ST_AUTOIDLE));
+
/* Enable McBSP Sidetone */
w = MCBSP_READ(mcbsp, SSELCR);
MCBSP_WRITE(mcbsp, SSELCR, w | SIDETONEEN);
@@ -279,6 +283,10 @@ static void omap_st_off(struct omap_mcbsp *mcbsp)
w = MCBSP_READ(mcbsp, SSELCR);
MCBSP_WRITE(mcbsp, SSELCR, w & ~(SIDETONEEN));
+ /* Enable Sidetone clock auto-gating to reduce power consumption */
+ w = MCBSP_ST_READ(mcbsp, SYSCONFIG);
+ MCBSP_ST_WRITE(mcbsp, SYSCONFIG, w | ST_AUTOIDLE);
+
if (mcbsp->pdata->enable_st_clock)
mcbsp->pdata->enable_st_clock(mcbsp->id, 0);
}
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index 99381a27295b..a84f677234f0 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -82,6 +82,8 @@ static int omap_pcm_hw_params(struct snd_pcm_substream *substream,
struct dma_chan *chan;
int err = 0;
+ memset(&config, 0x00, sizeof(config));
+
dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
/* return if this is a bufferless transfer e.g.
diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c
index ec522e94b0e2..b6cb9950f05d 100644
--- a/sound/soc/pxa/brownstone.c
+++ b/sound/soc/pxa/brownstone.c
@@ -133,3 +133,4 @@ module_platform_driver(mmp_driver);
MODULE_AUTHOR("Leo Yan <leoy@marvell.com>");
MODULE_DESCRIPTION("ALSA SoC Brownstone");
MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:brownstone-audio");
diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c
index 5c8f9db50a47..d1661fa6ee08 100644
--- a/sound/soc/pxa/mioa701_wm9713.c
+++ b/sound/soc/pxa/mioa701_wm9713.c
@@ -207,3 +207,4 @@ module_platform_driver(mioa701_wm9713_driver);
MODULE_AUTHOR("Robert Jarzmik (rjarzmik@free.fr)");
MODULE_DESCRIPTION("ALSA SoC WM9713 MIO A701");
MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:mioa701-wm9713");
diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c
index 51e790d006f5..96df9b2d8fc4 100644
--- a/sound/soc/pxa/mmp-pcm.c
+++ b/sound/soc/pxa/mmp-pcm.c
@@ -248,3 +248,4 @@ module_platform_driver(mmp_pcm_driver);
MODULE_AUTHOR("Leo Yan <leoy@marvell.com>");
MODULE_DESCRIPTION("MMP Soc Audio DMA module");
MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:mmp-pcm-audio");
diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c
index eca60c29791a..ca8b23f8c525 100644
--- a/sound/soc/pxa/mmp-sspa.c
+++ b/sound/soc/pxa/mmp-sspa.c
@@ -482,3 +482,4 @@ module_platform_driver(asoc_mmp_sspa_driver);
MODULE_AUTHOR("Leo Yan <leoy@marvell.com>");
MODULE_DESCRIPTION("MMP SSPA SoC Interface");
MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:mmp-sspa-dai");
diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c
index 4e74d9573f03..bcc81e920a67 100644
--- a/sound/soc/pxa/palm27x.c
+++ b/sound/soc/pxa/palm27x.c
@@ -161,3 +161,4 @@ module_platform_driver(palm27x_wm9712_driver);
MODULE_AUTHOR("Marek Vasut <marek.vasut@gmail.com>");
MODULE_DESCRIPTION("ALSA SoC Palm T|X, T5 and LifeDrive");
MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:palm27x-asoc");
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index da03fad1b9cd..3cad990dad2c 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -833,3 +833,4 @@ module_platform_driver(asoc_ssp_driver);
MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
MODULE_DESCRIPTION("PXA SSP/PCM SoC Interface");
MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:pxa-ssp-dai");
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index f3de615aacd7..9615e6de1306 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -287,3 +287,4 @@ module_platform_driver(pxa2xx_ac97_driver);
MODULE_AUTHOR("Nicolas Pitre");
MODULE_DESCRIPTION("AC97 driver for the Intel PXA2xx chip");
MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:pxa2xx-ac97");
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index 9f390398d518..410d48b93031 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -117,3 +117,4 @@ module_platform_driver(pxa_pcm_driver);
MODULE_AUTHOR("Nicolas Pitre");
MODULE_DESCRIPTION("Intel PXA2xx PCM DMA module");
MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:pxa-pcm-audio");
diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c
index 2f8e20416bd3..574c6af28c06 100644
--- a/sound/soc/rockchip/rockchip_i2s.c
+++ b/sound/soc/rockchip/rockchip_i2s.c
@@ -34,6 +34,13 @@ struct rk_i2s_dev {
struct regmap *regmap;
+/*
+ * Used to indicate the tx/rx status.
+ * I2S controller hopes to start the tx and rx together,
+ * also to stop them when they are both try to stop.
+*/
+ bool tx_start;
+ bool rx_start;
bool is_master_mode;
};
@@ -75,29 +82,37 @@ static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on)
I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_ENABLE);
regmap_update_bits(i2s->regmap, I2S_XFER,
- I2S_XFER_TXS_START,
- I2S_XFER_TXS_START);
+ I2S_XFER_TXS_START | I2S_XFER_RXS_START,
+ I2S_XFER_TXS_START | I2S_XFER_RXS_START);
+
+ i2s->tx_start = true;
} else {
+ i2s->tx_start = false;
+
regmap_update_bits(i2s->regmap, I2S_DMACR,
I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_DISABLE);
- regmap_update_bits(i2s->regmap, I2S_XFER,
- I2S_XFER_TXS_START,
- I2S_XFER_TXS_STOP);
+ if (!i2s->rx_start) {
+ regmap_update_bits(i2s->regmap, I2S_XFER,
+ I2S_XFER_TXS_START |
+ I2S_XFER_RXS_START,
+ I2S_XFER_TXS_STOP |
+ I2S_XFER_RXS_STOP);
- regmap_update_bits(i2s->regmap, I2S_CLR,
- I2S_CLR_TXC,
- I2S_CLR_TXC);
+ regmap_update_bits(i2s->regmap, I2S_CLR,
+ I2S_CLR_TXC | I2S_CLR_RXC,
+ I2S_CLR_TXC | I2S_CLR_RXC);
- regmap_read(i2s->regmap, I2S_CLR, &val);
-
- /* Should wait for clear operation to finish */
- while (val & I2S_CLR_TXC) {
regmap_read(i2s->regmap, I2S_CLR, &val);
- retry--;
- if (!retry) {
- dev_warn(i2s->dev, "fail to clear\n");
- break;
+
+ /* Should wait for clear operation to finish */
+ while (val) {
+ regmap_read(i2s->regmap, I2S_CLR, &val);
+ retry--;
+ if (!retry) {
+ dev_warn(i2s->dev, "fail to clear\n");
+ break;
+ }
}
}
}
@@ -113,29 +128,37 @@ static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on)
I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_ENABLE);
regmap_update_bits(i2s->regmap, I2S_XFER,
- I2S_XFER_RXS_START,
- I2S_XFER_RXS_START);
+ I2S_XFER_TXS_START | I2S_XFER_RXS_START,
+ I2S_XFER_TXS_START | I2S_XFER_RXS_START);
+
+ i2s->rx_start = true;
} else {
+ i2s->rx_start = false;
+
regmap_update_bits(i2s->regmap, I2S_DMACR,
I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_DISABLE);
- regmap_update_bits(i2s->regmap, I2S_XFER,
- I2S_XFER_RXS_START,
- I2S_XFER_RXS_STOP);
+ if (!i2s->tx_start) {
+ regmap_update_bits(i2s->regmap, I2S_XFER,
+ I2S_XFER_TXS_START |
+ I2S_XFER_RXS_START,
+ I2S_XFER_TXS_STOP |
+ I2S_XFER_RXS_STOP);
- regmap_update_bits(i2s->regmap, I2S_CLR,
- I2S_CLR_RXC,
- I2S_CLR_RXC);
+ regmap_update_bits(i2s->regmap, I2S_CLR,
+ I2S_CLR_TXC | I2S_CLR_RXC,
+ I2S_CLR_TXC | I2S_CLR_RXC);
- regmap_read(i2s->regmap, I2S_CLR, &val);
-
- /* Should wait for clear operation to finish */
- while (val & I2S_CLR_RXC) {
regmap_read(i2s->regmap, I2S_CLR, &val);
- retry--;
- if (!retry) {
- dev_warn(i2s->dev, "fail to clear\n");
- break;
+
+ /* Should wait for clear operation to finish */
+ while (val) {
+ regmap_read(i2s->regmap, I2S_CLR, &val);
+ retry--;
+ if (!retry) {
+ dev_warn(i2s->dev, "fail to clear\n");
+ break;
+ }
}
}
}
diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c
index 606399de684d..49354d17ea55 100644
--- a/sound/soc/sh/rcar/adg.c
+++ b/sound/soc/sh/rcar/adg.c
@@ -492,9 +492,7 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv,
*/
if (!count) {
clk = clk_register_fixed_rate(dev, clkout_name[CLKOUT],
- parent_clk_name,
- (parent_clk_name) ?
- 0 : CLK_IS_ROOT, req_rate);
+ parent_clk_name, 0, req_rate);
if (!IS_ERR(clk)) {
adg->clkout[CLKOUT] = clk;
of_clk_add_provider(np, of_clk_src_simple_get, clk);
@@ -506,9 +504,7 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv,
else {
for (i = 0; i < CLKOUTMAX; i++) {
clk = clk_register_fixed_rate(dev, clkout_name[i],
- parent_clk_name,
- (parent_clk_name) ?
- 0 : CLK_IS_ROOT,
+ parent_clk_name, 0,
req_rate);
if (!IS_ERR(clk)) {
adg->onecell.clks = adg->clkout;
diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c
index 7658e8fd7bdc..6bc93cbb3049 100644
--- a/sound/soc/sh/rcar/dma.c
+++ b/sound/soc/sh/rcar/dma.c
@@ -316,11 +316,15 @@ static u32 rsnd_dmapp_get_id(struct rsnd_dai_stream *io,
size = ARRAY_SIZE(gen2_id_table_cmd);
}
- if (!entry)
- return 0xFF;
+ if ((!entry) || (size <= id)) {
+ struct device *dev = rsnd_priv_to_dev(rsnd_io_to_priv(io));
- if (size <= id)
- return 0xFF;
+ dev_err(dev, "unknown connection (%s[%d])\n",
+ rsnd_mod_name(mod), rsnd_mod_id(mod));
+
+ /* use non-prohibited SRS number as error */
+ return 0x00; /* SSI00 */
+ }
return entry[id];
}
diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h
index fc89a67258ca..a8f61d79333b 100644
--- a/sound/soc/sh/rcar/rsnd.h
+++ b/sound/soc/sh/rcar/rsnd.h
@@ -276,8 +276,9 @@ struct rsnd_mod {
/*
* status
*
- * 0xH0000CB0
+ * 0xH0000CBA
*
+ * A 0: probe 1: remove
* B 0: init 1: quit
* C 0: start 1: stop
*
@@ -287,19 +288,19 @@ struct rsnd_mod {
* H 0: fallback
* H 0: hw_params
*/
+#define __rsnd_mod_shift_probe 0
+#define __rsnd_mod_shift_remove 0
#define __rsnd_mod_shift_init 4
#define __rsnd_mod_shift_quit 4
#define __rsnd_mod_shift_start 8
#define __rsnd_mod_shift_stop 8
-#define __rsnd_mod_shift_probe 28 /* always called */
-#define __rsnd_mod_shift_remove 28 /* always called */
#define __rsnd_mod_shift_irq 28 /* always called */
#define __rsnd_mod_shift_pcm_new 28 /* always called */
#define __rsnd_mod_shift_fallback 28 /* always called */
#define __rsnd_mod_shift_hw_params 28 /* always called */
-#define __rsnd_mod_add_probe 0
-#define __rsnd_mod_add_remove 0
+#define __rsnd_mod_add_probe 1
+#define __rsnd_mod_add_remove -1
#define __rsnd_mod_add_init 1
#define __rsnd_mod_add_quit -1
#define __rsnd_mod_add_start 1
@@ -310,7 +311,7 @@ struct rsnd_mod {
#define __rsnd_mod_add_hw_params 0
#define __rsnd_mod_call_probe 0
-#define __rsnd_mod_call_remove 0
+#define __rsnd_mod_call_remove 1
#define __rsnd_mod_call_init 0
#define __rsnd_mod_call_quit 1
#define __rsnd_mod_call_start 0
diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c
index 15d6ffe8be74..e39f916d0f2f 100644
--- a/sound/soc/sh/rcar/src.c
+++ b/sound/soc/sh/rcar/src.c
@@ -572,6 +572,9 @@ int rsnd_src_probe(struct rsnd_priv *priv)
i = 0;
for_each_child_of_node(node, np) {
+ if (!of_device_is_available(np))
+ goto skip;
+
src = rsnd_src_get(priv, i);
snprintf(name, RSND_SRC_NAME_SIZE, "%s.%d",
@@ -595,6 +598,7 @@ int rsnd_src_probe(struct rsnd_priv *priv)
if (ret)
goto rsnd_src_probe_done;
+skip:
i++;
}
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index d2e62b159610..16369cad4803 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -930,7 +930,18 @@ static struct snd_soc_component *soc_find_component(
return NULL;
}
-static struct snd_soc_dai *snd_soc_find_dai(
+/**
+ * snd_soc_find_dai - Find a registered DAI
+ *
+ * @dlc: name of the DAI and optional component info to match
+ *
+ * This function will search all regsitered components and their DAIs to
+ * find the DAI of the same name. The component's of_node and name
+ * should also match if being specified.
+ *
+ * Return: pointer of DAI, or NULL if not found.
+ */
+struct snd_soc_dai *snd_soc_find_dai(
const struct snd_soc_dai_link_component *dlc)
{
struct snd_soc_component *component;
@@ -959,6 +970,7 @@ static struct snd_soc_dai *snd_soc_find_dai(
return NULL;
}
+EXPORT_SYMBOL_GPL(snd_soc_find_dai);
static bool soc_is_dai_link_bound(struct snd_soc_card *card,
struct snd_soc_dai_link *dai_link)
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 801ae1a81dfd..c4464858bf01 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -2188,6 +2188,13 @@ static ssize_t dapm_widget_show_component(struct snd_soc_component *cmpnt,
int count = 0;
char *state = "not set";
+ /* card won't be set for the dummy component, as a spot fix
+ * we're checking for that case specifically here but in future
+ * we will ensure that the dummy component looks like others.
+ */
+ if (!cmpnt->card)
+ return 0;
+
list_for_each_entry(w, &cmpnt->card->widgets, list) {
if (w->dapm != dapm)
continue;
diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
index 6fd1906af387..6cef3977507a 100644
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -163,31 +163,42 @@ static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substrea
}
/*
- * Prepare formats mask for valid/allowed sample types. If the dma does
- * not have support for the given physical word size, it needs to be
- * masked out so user space can not use the format which produces
- * corrupted audio.
- * In case the dma driver does not implement the slave_caps the default
- * assumption is that it supports 1, 2 and 4 bytes widths.
+ * If SND_DMAENGINE_PCM_DAI_FLAG_PACK is set keep
+ * hw.formats set to 0, meaning no restrictions are in place.
+ * In this case it's the responsibility of the DAI driver to
+ * provide the supported format information.
*/
- for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) {
- int bits = snd_pcm_format_physical_width(i);
-
- /* Enable only samples with DMA supported physical widths */
- switch (bits) {
- case 8:
- case 16:
- case 24:
- case 32:
- case 64:
- if (addr_widths & (1 << (bits / 8)))
- hw.formats |= (1LL << i);
- break;
- default:
- /* Unsupported types */
- break;
+ if (!(dma_data->flags & SND_DMAENGINE_PCM_DAI_FLAG_PACK))
+ /*
+ * Prepare formats mask for valid/allowed sample types. If the
+ * dma does not have support for the given physical word size,
+ * it needs to be masked out so user space can not use the
+ * format which produces corrupted audio.
+ * In case the dma driver does not implement the slave_caps the
+ * default assumption is that it supports 1, 2 and 4 bytes
+ * widths.
+ */
+ for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) {
+ int bits = snd_pcm_format_physical_width(i);
+
+ /*
+ * Enable only samples with DMA supported physical
+ * widths
+ */
+ switch (bits) {
+ case 8:
+ case 16:
+ case 24:
+ case 32:
+ case 64:
+ if (addr_widths & (1 << (bits / 8)))
+ hw.formats |= (1LL << i);
+ break;
+ default:
+ /* Unsupported types */
+ break;
+ }
}
- }
return snd_soc_set_runtime_hwparams(substream, &hw);
}
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index 1cf94d7fb9f4..ee7f15aa46fc 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -1023,6 +1023,11 @@ static int soc_tplg_kcontrol_elems_load(struct soc_tplg *tplg,
control_hdr = (struct snd_soc_tplg_ctl_hdr *)tplg->pos;
+ if (control_hdr->size != sizeof(*control_hdr)) {
+ dev_err(tplg->dev, "ASoC: invalid control size\n");
+ return -EINVAL;
+ }
+
switch (control_hdr->ops.info) {
case SND_SOC_TPLG_CTL_VOLSW:
case SND_SOC_TPLG_CTL_STROBE:
@@ -1476,6 +1481,8 @@ widget:
widget->dobj.type = SND_SOC_DOBJ_WIDGET;
widget->dobj.ops = tplg->ops;
widget->dobj.index = tplg->index;
+ kfree(template.sname);
+ kfree(template.name);
list_add(&widget->dobj.list, &tplg->comp->dobj_list);
return 0;
@@ -1499,10 +1506,17 @@ static int soc_tplg_dapm_widget_elems_load(struct soc_tplg *tplg,
for (i = 0; i < count; i++) {
widget = (struct snd_soc_tplg_dapm_widget *) tplg->pos;
+ if (widget->size != sizeof(*widget)) {
+ dev_err(tplg->dev, "ASoC: invalid widget size\n");
+ return -EINVAL;
+ }
+
ret = soc_tplg_dapm_widget_create(tplg, widget);
- if (ret < 0)
+ if (ret < 0) {
dev_err(tplg->dev, "ASoC: failed to load widget %s\n",
widget->name);
+ return ret;
+ }
}
return 0;
@@ -1586,6 +1600,7 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg,
return snd_soc_register_dai(tplg->comp, dai_drv);
}
+/* create the FE DAI link */
static int soc_tplg_link_create(struct soc_tplg *tplg,
struct snd_soc_tplg_pcm *pcm)
{
@@ -1598,6 +1613,16 @@ static int soc_tplg_link_create(struct soc_tplg *tplg,
link->name = pcm->pcm_name;
link->stream_name = pcm->pcm_name;
+ link->id = pcm->pcm_id;
+
+ link->cpu_dai_name = pcm->dai_name;
+ link->codec_name = "snd-soc-dummy";
+ link->codec_dai_name = "snd-soc-dummy-dai";
+
+ /* enable DPCM */
+ link->dynamic = 1;
+ link->dpcm_playback = pcm->playback;
+ link->dpcm_capture = pcm->capture;
/* pass control to component driver for optional further init */
ret = soc_tplg_dai_link_load(tplg, link);
@@ -1639,8 +1664,6 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg,
if (tplg->pass != SOC_TPLG_PASS_PCM_DAI)
return 0;
- pcm = (struct snd_soc_tplg_pcm *)tplg->pos;
-
if (soc_tplg_check_elem_count(tplg,
sizeof(struct snd_soc_tplg_pcm), count,
hdr->payload_size, "PCM DAI")) {
@@ -1650,7 +1673,13 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg,
}
/* create the FE DAIs and DAI links */
+ pcm = (struct snd_soc_tplg_pcm *)tplg->pos;
for (i = 0; i < count; i++) {
+ if (pcm->size != sizeof(*pcm)) {
+ dev_err(tplg->dev, "ASoC: invalid pcm size\n");
+ return -EINVAL;
+ }
+
soc_tplg_pcm_create(tplg, pcm);
pcm++;
}
@@ -1670,6 +1699,11 @@ static int soc_tplg_manifest_load(struct soc_tplg *tplg,
return 0;
manifest = (struct snd_soc_tplg_manifest *)tplg->pos;
+ if (manifest->size != sizeof(*manifest)) {
+ dev_err(tplg->dev, "ASoC: invalid manifest size\n");
+ return -EINVAL;
+ }
+
tplg->pos += sizeof(struct snd_soc_tplg_manifest);
if (tplg->comp && tplg->ops && tplg->ops->manifest)
@@ -1686,6 +1720,14 @@ static int soc_valid_header(struct soc_tplg *tplg,
if (soc_tplg_get_hdr_offset(tplg) >= tplg->fw->size)
return 0;
+ if (hdr->size != sizeof(*hdr)) {
+ dev_err(tplg->dev,
+ "ASoC: invalid header size for type %d at offset 0x%lx size 0x%zx.\n",
+ hdr->type, soc_tplg_get_hdr_offset(tplg),
+ tplg->fw->size);
+ return -EINVAL;
+ }
+
/* big endian firmware objects not supported atm */
if (hdr->magic == cpu_to_be32(SND_SOC_TPLG_MAGIC)) {
dev_err(tplg->dev,
diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig
index d14bf411515b..a452ad7cec40 100644
--- a/sound/usb/Kconfig
+++ b/sound/usb/Kconfig
@@ -15,7 +15,6 @@ config SND_USB_AUDIO
select SND_RAWMIDI
select SND_PCM
select BITREVERSE
- select SND_USB_AUDIO_USE_MEDIA_CONTROLLER if MEDIA_CONTROLLER && (MEDIA_SUPPORT=y || MEDIA_SUPPORT=SND_USB_AUDIO)
help
Say Y here to include support for USB audio and USB MIDI
devices.
@@ -23,9 +22,6 @@ config SND_USB_AUDIO
To compile this driver as a module, choose M here: the module
will be called snd-usb-audio.
-config SND_USB_AUDIO_USE_MEDIA_CONTROLLER
- bool
-
config SND_USB_UA101
tristate "Edirol UA-101/UA-1000 driver"
select SND_PCM
diff --git a/sound/usb/Makefile b/sound/usb/Makefile
index 8dca3c407f5a..2d2d122b069f 100644
--- a/sound/usb/Makefile
+++ b/sound/usb/Makefile
@@ -15,8 +15,6 @@ snd-usb-audio-objs := card.o \
quirks.o \
stream.o
-snd-usb-audio-$(CONFIG_SND_USB_AUDIO_USE_MEDIA_CONTROLLER) += media.o
-
snd-usbmidi-lib-objs := midi.o
# Toplevel Module Dependency
diff --git a/sound/usb/card.c b/sound/usb/card.c
index 63244bbba8c7..3fc63583a537 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -66,7 +66,6 @@
#include "format.h"
#include "power.h"
#include "stream.h"
-#include "media.h"
MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>");
MODULE_DESCRIPTION("USB Audio");
@@ -612,11 +611,6 @@ static int usb_audio_probe(struct usb_interface *intf,
if (err < 0)
goto __error;
- if (quirk->media_device) {
- /* don't want to fail when media_snd_device_create() fails */
- media_snd_device_create(chip, intf);
- }
-
usb_chip[chip->index] = chip;
chip->num_interfaces++;
usb_set_intfdata(intf, chip);
@@ -673,14 +667,6 @@ static void usb_audio_disconnect(struct usb_interface *intf)
list_for_each(p, &chip->midi_list) {
snd_usbmidi_disconnect(p);
}
- /*
- * Nice to check quirk && quirk->media_device
- * need some special handlings. Doesn't look like
- * we have access to quirk here
- * Acceses mixer_list
- */
- media_snd_device_delete(chip);
-
/* release mixer resources */
list_for_each_entry(mixer, &chip->mixer_list, list) {
snd_usb_mixer_disconnect(mixer);
diff --git a/sound/usb/card.h b/sound/usb/card.h
index 34a0898e2238..71778ca4b26a 100644
--- a/sound/usb/card.h
+++ b/sound/usb/card.h
@@ -105,8 +105,6 @@ struct snd_usb_endpoint {
struct list_head list;
};
-struct media_ctl;
-
struct snd_usb_substream {
struct snd_usb_stream *stream;
struct usb_device *dev;
@@ -158,7 +156,6 @@ struct snd_usb_substream {
} dsd_dop;
bool trigger_tstamp_pending_update; /* trigger timestamp being updated from initial estimate */
- struct media_ctl *media_ctl;
};
struct snd_usb_stream {
diff --git a/sound/usb/media.c b/sound/usb/media.c
deleted file mode 100644
index 93a50d01490c..000000000000
--- a/sound/usb/media.c
+++ /dev/null
@@ -1,318 +0,0 @@
-/*
- * media.c - Media Controller specific ALSA driver code
- *
- * Copyright (c) 2016 Shuah Khan <shuahkh@osg.samsung.com>
- * Copyright (c) 2016 Samsung Electronics Co., Ltd.
- *
- * This file is released under the GPLv2.
- */
-
-/*
- * This file adds Media Controller support to ALSA driver
- * to use the Media Controller API to share tuner with DVB
- * and V4L2 drivers that control media device. Media device
- * is created based on existing quirks framework. Using this
- * approach, the media controller API usage can be added for
- * a specific device.
-*/
-
-#include <linux/init.h>
-#include <linux/list.h>
-#include <linux/mutex.h>
-#include <linux/slab.h>
-#include <linux/usb.h>
-
-#include <sound/pcm.h>
-#include <sound/core.h>
-
-#include "usbaudio.h"
-#include "card.h"
-#include "mixer.h"
-#include "media.h"
-
-static int media_snd_enable_source(struct media_ctl *mctl)
-{
- if (mctl && mctl->media_dev->enable_source)
- return mctl->media_dev->enable_source(&mctl->media_entity,
- &mctl->media_pipe);
- return 0;
-}
-
-static void media_snd_disable_source(struct media_ctl *mctl)
-{
- if (mctl && mctl->media_dev->disable_source)
- mctl->media_dev->disable_source(&mctl->media_entity);
-}
-
-int media_snd_stream_init(struct snd_usb_substream *subs, struct snd_pcm *pcm,
- int stream)
-{
- struct media_device *mdev;
- struct media_ctl *mctl;
- struct device *pcm_dev = &pcm->streams[stream].dev;
- u32 intf_type;
- int ret = 0;
- u16 mixer_pad;
- struct media_entity *entity;
-
- mdev = subs->stream->chip->media_dev;
- if (!mdev)
- return -ENODEV;
-
- if (subs->media_ctl)
- return 0;
-
- /* allocate media_ctl */
- mctl = kzalloc(sizeof(*mctl), GFP_KERNEL);
- if (!mctl)
- return -ENOMEM;
-
- mctl->media_dev = mdev;
- if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
- intf_type = MEDIA_INTF_T_ALSA_PCM_PLAYBACK;
- mctl->media_entity.function = MEDIA_ENT_F_AUDIO_PLAYBACK;
- mctl->media_pad.flags = MEDIA_PAD_FL_SOURCE;
- mixer_pad = 1;
- } else {
- intf_type = MEDIA_INTF_T_ALSA_PCM_CAPTURE;
- mctl->media_entity.function = MEDIA_ENT_F_AUDIO_CAPTURE;
- mctl->media_pad.flags = MEDIA_PAD_FL_SINK;
- mixer_pad = 2;
- }
- mctl->media_entity.name = pcm->name;
- media_entity_pads_init(&mctl->media_entity, 1, &mctl->media_pad);
- ret = media_device_register_entity(mctl->media_dev,
- &mctl->media_entity);
- if (ret)
- goto free_mctl;
-
- mctl->intf_devnode = media_devnode_create(mdev, intf_type, 0,
- MAJOR(pcm_dev->devt),
- MINOR(pcm_dev->devt));
- if (!mctl->intf_devnode) {
- ret = -ENOMEM;
- goto unregister_entity;
- }
- mctl->intf_link = media_create_intf_link(&mctl->media_entity,
- &mctl->intf_devnode->intf,
- MEDIA_LNK_FL_ENABLED);
- if (!mctl->intf_link) {
- ret = -ENOMEM;
- goto devnode_remove;
- }
-
- /* create link between mixer and audio */
- media_device_for_each_entity(entity, mdev) {
- switch (entity->function) {
- case MEDIA_ENT_F_AUDIO_MIXER:
- ret = media_create_pad_link(entity, mixer_pad,
- &mctl->media_entity, 0,
- MEDIA_LNK_FL_ENABLED);
- if (ret)
- goto remove_intf_link;
- break;
- }
- }
-
- subs->media_ctl = mctl;
- return 0;
-
-remove_intf_link:
- media_remove_intf_link(mctl->intf_link);
-devnode_remove:
- media_devnode_remove(mctl->intf_devnode);
-unregister_entity:
- media_device_unregister_entity(&mctl->media_entity);
-free_mctl:
- kfree(mctl);
- return ret;
-}
-
-void media_snd_stream_delete(struct snd_usb_substream *subs)
-{
- struct media_ctl *mctl = subs->media_ctl;
-
- if (mctl && mctl->media_dev) {
- struct media_device *mdev;
-
- mdev = subs->stream->chip->media_dev;
- if (mdev && media_devnode_is_registered(&mdev->devnode)) {
- media_devnode_remove(mctl->intf_devnode);
- media_device_unregister_entity(&mctl->media_entity);
- media_entity_cleanup(&mctl->media_entity);
- }
- kfree(mctl);
- subs->media_ctl = NULL;
- }
-}
-
-int media_snd_start_pipeline(struct snd_usb_substream *subs)
-{
- struct media_ctl *mctl = subs->media_ctl;
-
- if (mctl)
- return media_snd_enable_source(mctl);
- return 0;
-}
-
-void media_snd_stop_pipeline(struct snd_usb_substream *subs)
-{
- struct media_ctl *mctl = subs->media_ctl;
-
- if (mctl)
- media_snd_disable_source(mctl);
-}
-
-int media_snd_mixer_init(struct snd_usb_audio *chip)
-{
- struct device *ctl_dev = &chip->card->ctl_dev;
- struct media_intf_devnode *ctl_intf;
- struct usb_mixer_interface *mixer;
- struct media_device *mdev = chip->media_dev;
- struct media_mixer_ctl *mctl;
- u32 intf_type = MEDIA_INTF_T_ALSA_CONTROL;
- int ret;
-
- if (!mdev)
- return -ENODEV;
-
- ctl_intf = chip->ctl_intf_media_devnode;
- if (!ctl_intf) {
- ctl_intf = media_devnode_create(mdev, intf_type, 0,
- MAJOR(ctl_dev->devt),
- MINOR(ctl_dev->devt));
- if (!ctl_intf)
- return -ENOMEM;
- chip->ctl_intf_media_devnode = ctl_intf;
- }
-
- list_for_each_entry(mixer, &chip->mixer_list, list) {
-
- if (mixer->media_mixer_ctl)
- continue;
-
- /* allocate media_mixer_ctl */
- mctl = kzalloc(sizeof(*mctl), GFP_KERNEL);
- if (!mctl)
- return -ENOMEM;
-
- mctl->media_dev = mdev;
- mctl->media_entity.function = MEDIA_ENT_F_AUDIO_MIXER;
- mctl->media_entity.name = chip->card->mixername;
- mctl->media_pad[0].flags = MEDIA_PAD_FL_SINK;
- mctl->media_pad[1].flags = MEDIA_PAD_FL_SOURCE;
- mctl->media_pad[2].flags = MEDIA_PAD_FL_SOURCE;
- media_entity_pads_init(&mctl->media_entity, MEDIA_MIXER_PAD_MAX,
- mctl->media_pad);
- ret = media_device_register_entity(mctl->media_dev,
- &mctl->media_entity);
- if (ret) {
- kfree(mctl);
- return ret;
- }
-
- mctl->intf_link = media_create_intf_link(&mctl->media_entity,
- &ctl_intf->intf,
- MEDIA_LNK_FL_ENABLED);
- if (!mctl->intf_link) {
- media_device_unregister_entity(&mctl->media_entity);
- media_entity_cleanup(&mctl->media_entity);
- kfree(mctl);
- return -ENOMEM;
- }
- mctl->intf_devnode = ctl_intf;
- mixer->media_mixer_ctl = mctl;
- }
- return 0;
-}
-
-static void media_snd_mixer_delete(struct snd_usb_audio *chip)
-{
- struct usb_mixer_interface *mixer;
- struct media_device *mdev = chip->media_dev;
-
- if (!mdev)
- return;
-
- list_for_each_entry(mixer, &chip->mixer_list, list) {
- struct media_mixer_ctl *mctl;
-
- mctl = mixer->media_mixer_ctl;
- if (!mixer->media_mixer_ctl)
- continue;
-
- if (media_devnode_is_registered(&mdev->devnode)) {
- media_device_unregister_entity(&mctl->media_entity);
- media_entity_cleanup(&mctl->media_entity);
- }
- kfree(mctl);
- mixer->media_mixer_ctl = NULL;
- }
- if (media_devnode_is_registered(&mdev->devnode))
- media_devnode_remove(chip->ctl_intf_media_devnode);
- chip->ctl_intf_media_devnode = NULL;
-}
-
-int media_snd_device_create(struct snd_usb_audio *chip,
- struct usb_interface *iface)
-{
- struct media_device *mdev;
- struct usb_device *usbdev = interface_to_usbdev(iface);
- int ret;
-
- mdev = media_device_get_devres(&usbdev->dev);
- if (!mdev)
- return -ENOMEM;
- if (!mdev->dev) {
- /* register media device */
- mdev->dev = &usbdev->dev;
- if (usbdev->product)
- strlcpy(mdev->model, usbdev->product,
- sizeof(mdev->model));
- if (usbdev->serial)
- strlcpy(mdev->serial, usbdev->serial,
- sizeof(mdev->serial));
- strcpy(mdev->bus_info, usbdev->devpath);
- mdev->hw_revision = le16_to_cpu(usbdev->descriptor.bcdDevice);
- media_device_init(mdev);
- }
- if (!media_devnode_is_registered(&mdev->devnode)) {
- ret = media_device_register(mdev);
- if (ret) {
- dev_err(&usbdev->dev,
- "Couldn't register media device. Error: %d\n",
- ret);
- return ret;
- }
- }
-
- /* save media device - avoid lookups */
- chip->media_dev = mdev;
-
- /* Create media entities for mixer and control dev */
- ret = media_snd_mixer_init(chip);
- if (ret) {
- dev_err(&usbdev->dev,
- "Couldn't create media mixer entities. Error: %d\n",
- ret);
-
- /* clear saved media_dev */
- chip->media_dev = NULL;
-
- return ret;
- }
- return 0;
-}
-
-void media_snd_device_delete(struct snd_usb_audio *chip)
-{
- struct media_device *mdev = chip->media_dev;
-
- media_snd_mixer_delete(chip);
-
- if (mdev) {
- if (media_devnode_is_registered(&mdev->devnode))
- media_device_unregister(mdev);
- chip->media_dev = NULL;
- }
-}
diff --git a/sound/usb/media.h b/sound/usb/media.h
deleted file mode 100644
index 1dcdcdc5f7aa..000000000000
--- a/sound/usb/media.h
+++ /dev/null
@@ -1,72 +0,0 @@
-/*
- * media.h - Media Controller specific ALSA driver code
- *
- * Copyright (c) 2016 Shuah Khan <shuahkh@osg.samsung.com>
- * Copyright (c) 2016 Samsung Electronics Co., Ltd.
- *
- * This file is released under the GPLv2.
- */
-
-/*
- * This file adds Media Controller support to ALSA driver
- * to use the Media Controller API to share tuner with DVB
- * and V4L2 drivers that control media device. Media device
- * is created based on existing quirks framework. Using this
- * approach, the media controller API usage can be added for
- * a specific device.
-*/
-#ifndef __MEDIA_H
-
-#ifdef CONFIG_SND_USB_AUDIO_USE_MEDIA_CONTROLLER
-
-#include <media/media-device.h>
-#include <media/media-entity.h>
-#include <sound/asound.h>
-
-struct media_ctl {
- struct media_device *media_dev;
- struct media_entity media_entity;
- struct media_intf_devnode *intf_devnode;
- struct media_link *intf_link;
- struct media_pad media_pad;
- struct media_pipeline media_pipe;
-};
-
-/*
- * One source pad each for SNDRV_PCM_STREAM_CAPTURE and
- * SNDRV_PCM_STREAM_PLAYBACK. One for sink pad to link
- * to AUDIO Source
-*/
-#define MEDIA_MIXER_PAD_MAX (SNDRV_PCM_STREAM_LAST + 2)
-
-struct media_mixer_ctl {
- struct media_device *media_dev;
- struct media_entity media_entity;
- struct media_intf_devnode *intf_devnode;
- struct media_link *intf_link;
- struct media_pad media_pad[MEDIA_MIXER_PAD_MAX];
- struct media_pipeline media_pipe;
-};
-
-int media_snd_device_create(struct snd_usb_audio *chip,
- struct usb_interface *iface);
-void media_snd_device_delete(struct snd_usb_audio *chip);
-int media_snd_stream_init(struct snd_usb_substream *subs, struct snd_pcm *pcm,
- int stream);
-void media_snd_stream_delete(struct snd_usb_substream *subs);
-int media_snd_start_pipeline(struct snd_usb_substream *subs);
-void media_snd_stop_pipeline(struct snd_usb_substream *subs);
-#else
-static inline int media_snd_device_create(struct snd_usb_audio *chip,
- struct usb_interface *iface)
- { return 0; }
-static inline void media_snd_device_delete(struct snd_usb_audio *chip) { }
-static inline int media_snd_stream_init(struct snd_usb_substream *subs,
- struct snd_pcm *pcm, int stream)
- { return 0; }
-static inline void media_snd_stream_delete(struct snd_usb_substream *subs) { }
-static inline int media_snd_start_pipeline(struct snd_usb_substream *subs)
- { return 0; }
-static inline void media_snd_stop_pipeline(struct snd_usb_substream *subs) { }
-#endif
-#endif /* __MEDIA_H */
diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h
index f3789446ab9c..3417ef347e40 100644
--- a/sound/usb/mixer.h
+++ b/sound/usb/mixer.h
@@ -3,8 +3,6 @@
#include <sound/info.h>
-struct media_mixer_ctl;
-
struct usb_mixer_interface {
struct snd_usb_audio *chip;
struct usb_host_interface *hostif;
@@ -24,7 +22,6 @@ struct usb_mixer_interface {
struct urb *rc_urb;
struct usb_ctrlrequest *rc_setup_packet;
u8 rc_buffer[6];
- struct media_mixer_ctl *media_mixer_ctl;
};
#define MAX_CHANNELS 16 /* max logical channels */
diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c
index ddca6547399b..1f8fb0d904e0 100644
--- a/sound/usb/mixer_maps.c
+++ b/sound/usb/mixer_maps.c
@@ -349,6 +349,16 @@ static struct usbmix_name_map bose_companion5_map[] = {
};
/*
+ * Dell usb dock with ALC4020 codec had a firmware problem where it got
+ * screwed up when zero volume is passed; just skip it as a workaround
+ */
+static const struct usbmix_name_map dell_alc4020_map[] = {
+ { 16, NULL },
+ { 19, NULL },
+ { 0 }
+};
+
+/*
* Control map entries
*/
@@ -431,6 +441,10 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = {
.map = aureon_51_2_map,
},
{
+ .id = USB_ID(0x0bda, 0x4014),
+ .map = dell_alc4020_map,
+ },
+ {
.id = USB_ID(0x0dba, 0x1000),
.map = mbox1_map,
},
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 0e4e0640c504..44d178ee9177 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -35,7 +35,6 @@
#include "pcm.h"
#include "clock.h"
#include "power.h"
-#include "media.h"
#define SUBSTREAM_FLAG_DATA_EP_STARTED 0
#define SUBSTREAM_FLAG_SYNC_EP_STARTED 1
@@ -718,14 +717,10 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream,
struct audioformat *fmt;
int ret;
- ret = media_snd_start_pipeline(subs);
- if (ret)
- return ret;
-
ret = snd_pcm_lib_alloc_vmalloc_buffer(substream,
params_buffer_bytes(hw_params));
if (ret < 0)
- goto err_ret;
+ return ret;
subs->pcm_format = params_format(hw_params);
subs->period_bytes = params_period_bytes(hw_params);
@@ -739,27 +734,22 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream,
dev_dbg(&subs->dev->dev,
"cannot set format: format = %#x, rate = %d, channels = %d\n",
subs->pcm_format, subs->cur_rate, subs->channels);
- ret = -EINVAL;
- goto err_ret;
+ return -EINVAL;
}
ret = snd_usb_lock_shutdown(subs->stream->chip);
if (ret < 0)
- goto err_ret;
+ return ret;
ret = set_format(subs, fmt);
snd_usb_unlock_shutdown(subs->stream->chip);
if (ret < 0)
- goto err_ret;
+ return ret;
subs->interface = fmt->iface;
subs->altset_idx = fmt->altset_idx;
subs->need_setup_ep = true;
return 0;
-
-err_ret:
- media_snd_stop_pipeline(subs);
- return ret;
}
/*
@@ -771,7 +761,6 @@ static int snd_usb_hw_free(struct snd_pcm_substream *substream)
{
struct snd_usb_substream *subs = substream->runtime->private_data;
- media_snd_stop_pipeline(subs);
subs->cur_audiofmt = NULL;
subs->cur_rate = 0;
subs->period_bytes = 0;
@@ -1232,7 +1221,6 @@ static int snd_usb_pcm_open(struct snd_pcm_substream *substream, int direction)
struct snd_usb_stream *as = snd_pcm_substream_chip(substream);
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_usb_substream *subs = &as->substream[direction];
- int ret;
subs->interface = -1;
subs->altset_idx = 0;
@@ -1246,12 +1234,7 @@ static int snd_usb_pcm_open(struct snd_pcm_substream *substream, int direction)
subs->dsd_dop.channel = 0;
subs->dsd_dop.marker = 1;
- ret = setup_hw_info(runtime, subs);
- if (ret == 0)
- ret = media_snd_stream_init(subs, as->pcm, direction);
- if (ret)
- snd_usb_autosuspend(subs->stream->chip);
- return ret;
+ return setup_hw_info(runtime, subs);
}
static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction)
@@ -1260,7 +1243,6 @@ static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction)
struct snd_usb_substream *subs = &as->substream[direction];
stop_endpoints(subs, true);
- media_snd_stop_pipeline(subs);
if (subs->interface >= 0 &&
!snd_usb_lock_shutdown(subs->stream->chip)) {
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 9d087b19c70c..c60a776e815d 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -2886,7 +2886,6 @@ YAMAHA_DEVICE(0x7010, "UB99"),
.product_name = pname, \
.ifnum = QUIRK_ANY_INTERFACE, \
.type = QUIRK_AUDIO_ALIGN_TRANSFER, \
- .media_device = 1, \
} \
}
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index fb62bce2435c..6adde457b602 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -150,6 +150,7 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip,
usb_audio_err(chip, "cannot memdup\n");
return -ENOMEM;
}
+ INIT_LIST_HEAD(&fp->list);
if (fp->nr_rates > MAX_NR_RATES) {
kfree(fp);
return -EINVAL;
@@ -193,6 +194,7 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip,
return 0;
error:
+ list_del(&fp->list); /* unlink for avoiding double-free */
kfree(fp);
kfree(rate_table);
return err;
@@ -469,6 +471,7 @@ static int create_uaxx_quirk(struct snd_usb_audio *chip,
fp->ep_attr = get_endpoint(alts, 0)->bmAttributes;
fp->datainterval = 0;
fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize);
+ INIT_LIST_HEAD(&fp->list);
switch (fp->maxpacksize) {
case 0x120:
@@ -492,6 +495,7 @@ static int create_uaxx_quirk(struct snd_usb_audio *chip,
? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
err = snd_usb_add_audio_stream(chip, stream, fp);
if (err < 0) {
+ list_del(&fp->list); /* unlink for avoiding double-free */
kfree(fp);
return err;
}
@@ -1130,9 +1134,14 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip)
case USB_ID(0x045E, 0x076F): /* MS Lifecam HD-6000 */
case USB_ID(0x045E, 0x0772): /* MS Lifecam Studio */
case USB_ID(0x045E, 0x0779): /* MS Lifecam HD-3000 */
+ case USB_ID(0x047F, 0x0415): /* Plantronics BT-300 */
case USB_ID(0x047F, 0xAA05): /* Plantronics DA45 */
case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */
+ case USB_ID(0x0556, 0x0014): /* Phoenix Audio TMX320VC */
case USB_ID(0x074D, 0x3553): /* Outlaw RR2150 (Micronas UAC3553B) */
+ case USB_ID(0x1de7, 0x0013): /* Phoenix Audio MT202exe */
+ case USB_ID(0x1de7, 0x0014): /* Phoenix Audio TMX320 */
+ case USB_ID(0x1de7, 0x0114): /* Phoenix Audio MT202pcs */
case USB_ID(0x21B4, 0x0081): /* AudioQuest DragonFly */
return true;
}
diff --git a/sound/usb/stream.c b/sound/usb/stream.c
index 51258a15f653..8e9548bc1f1a 100644
--- a/sound/usb/stream.c
+++ b/sound/usb/stream.c
@@ -36,7 +36,6 @@
#include "format.h"
#include "clock.h"
#include "stream.h"
-#include "media.h"
/*
* free a substream
@@ -53,7 +52,6 @@ static void free_substream(struct snd_usb_substream *subs)
kfree(fp);
}
kfree(subs->rate_list.list);
- media_snd_stream_delete(subs);
}
@@ -316,7 +314,9 @@ static struct snd_pcm_chmap_elem *convert_chmap(int channels, unsigned int bits,
/*
* add this endpoint to the chip instance.
* if a stream with the same endpoint already exists, append to it.
- * if not, create a new pcm stream.
+ * if not, create a new pcm stream. note, fp is added to the substream
+ * fmt_list and will be freed on the chip instance release. do not free
+ * fp or do remove it from the substream fmt_list to avoid double-free.
*/
int snd_usb_add_audio_stream(struct snd_usb_audio *chip,
int stream,
@@ -677,6 +677,7 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no)
* (fp->maxpacksize & 0x7ff);
fp->attributes = parse_uac_endpoint_attributes(chip, alts, protocol, iface_no);
fp->clock = clock;
+ INIT_LIST_HEAD(&fp->list);
/* some quirks for attributes here */
@@ -725,6 +726,7 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no)
dev_dbg(&dev->dev, "%u:%d: add audio endpoint %#x\n", iface_no, altno, fp->endpoint);
err = snd_usb_add_audio_stream(chip, stream, fp);
if (err < 0) {
+ list_del(&fp->list); /* unlink for avoiding double-free */
kfree(fp->rate_table);
kfree(fp->chmap);
kfree(fp);
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index a161c7c1b126..b665d85555cb 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -30,9 +30,6 @@
*
*/
-struct media_device;
-struct media_intf_devnode;
-
struct snd_usb_audio {
int index;
struct usb_device *dev;
@@ -63,8 +60,6 @@ struct snd_usb_audio {
bool autoclock; /* from the 'autoclock' module param */
struct usb_host_interface *ctrl_intf; /* the audio control interface */
- struct media_device *media_dev;
- struct media_intf_devnode *ctl_intf_media_devnode;
};
#define usb_audio_err(chip, fmt, args...) \
@@ -115,7 +110,6 @@ struct snd_usb_audio_quirk {
const char *product_name;
int16_t ifnum;
uint16_t type;
- bool media_device;
const void *data;
};