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authorTakashi Iwai <tiwai@suse.de>2019-09-23 20:16:13 +0200
committerTakashi Iwai <tiwai@suse.de>2019-09-23 20:16:13 +0200
commit5b8b8f764d333d5a69f2fad61b86406dfca2d261 (patch)
treeac68d367eda4f3a6d3780aba9532d6f71ce491ed /sound
parentALSA: hda - Add laptop imic fixup for ASUS M9V laptop (diff)
parentASoC: ti: fix SND_SOC_DM365_VOICE_CODEC dependencies (diff)
downloadlinux-dev-5b8b8f764d333d5a69f2fad61b86406dfca2d261.tar.xz
linux-dev-5b8b8f764d333d5a69f2fad61b86406dfca2d261.zip
Merge tag 'asoc-fix-v5.4-rc1' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.4 A small smattering of ASoC fixes for v5.4 - nothing too exciting here, all small standalone things.
Diffstat (limited to 'sound')
-rw-r--r--sound/soc/atmel/atmel_ssc_dai.c12
-rw-r--r--sound/soc/atmel/atmel_ssc_dai.h1
-rw-r--r--sound/soc/codecs/pcm3168a.c3
-rw-r--r--sound/soc/fsl/fsl_sai.c15
-rw-r--r--sound/soc/fsl/fsl_sai.h1
-rw-r--r--sound/soc/sh/rcar/ssi.c10
-rw-r--r--sound/soc/soc-core.c2
-rw-r--r--sound/soc/ti/Kconfig11
8 files changed, 34 insertions, 21 deletions
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index 48e9eef34c0f..ca603397651c 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -116,19 +116,16 @@ static struct atmel_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = {
static struct atmel_ssc_info ssc_info[NUM_SSC_DEVICES] = {
{
.name = "ssc0",
- .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock),
.dir_mask = SSC_DIR_MASK_UNUSED,
.initialized = 0,
},
{
.name = "ssc1",
- .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock),
.dir_mask = SSC_DIR_MASK_UNUSED,
.initialized = 0,
},
{
.name = "ssc2",
- .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock),
.dir_mask = SSC_DIR_MASK_UNUSED,
.initialized = 0,
},
@@ -317,13 +314,10 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream,
snd_soc_dai_set_dma_data(dai, substream, dma_params);
- spin_lock_irq(&ssc_p->lock);
- if (ssc_p->dir_mask & dir_mask) {
- spin_unlock_irq(&ssc_p->lock);
+ if (ssc_p->dir_mask & dir_mask)
return -EBUSY;
- }
+
ssc_p->dir_mask |= dir_mask;
- spin_unlock_irq(&ssc_p->lock);
return 0;
}
@@ -355,7 +349,6 @@ static void atmel_ssc_shutdown(struct snd_pcm_substream *substream,
dir_mask = 1 << dir;
- spin_lock_irq(&ssc_p->lock);
ssc_p->dir_mask &= ~dir_mask;
if (!ssc_p->dir_mask) {
if (ssc_p->initialized) {
@@ -369,7 +362,6 @@ static void atmel_ssc_shutdown(struct snd_pcm_substream *substream,
ssc_p->cmr_div = ssc_p->tcmr_period = ssc_p->rcmr_period = 0;
ssc_p->forced_divider = 0;
}
- spin_unlock_irq(&ssc_p->lock);
/* Shutdown the SSC clock. */
pr_debug("atmel_ssc_dai: Stopping clock\n");
diff --git a/sound/soc/atmel/atmel_ssc_dai.h b/sound/soc/atmel/atmel_ssc_dai.h
index ae764cb541c7..3470b966e449 100644
--- a/sound/soc/atmel/atmel_ssc_dai.h
+++ b/sound/soc/atmel/atmel_ssc_dai.h
@@ -93,7 +93,6 @@ struct atmel_ssc_state {
struct atmel_ssc_info {
char *name;
struct ssc_device *ssc;
- spinlock_t lock; /* lock for dir_mask */
unsigned short dir_mask; /* 0=unused, 1=playback, 2=capture */
unsigned short initialized; /* true if SSC has been initialized */
unsigned short daifmt;
diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c
index 50ed86d45c26..88b75695fbf7 100644
--- a/sound/soc/codecs/pcm3168a.c
+++ b/sound/soc/codecs/pcm3168a.c
@@ -21,8 +21,7 @@
#define PCM3168A_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
SNDRV_PCM_FMTBIT_S24_3LE | \
- SNDRV_PCM_FMTBIT_S24_LE | \
- SNDRV_PCM_FMTBIT_S32_LE)
+ SNDRV_PCM_FMTBIT_S24_LE)
#define PCM3168A_FMT_I2S 0x0
#define PCM3168A_FMT_LEFT_J 0x1
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index ef0b74693093..b517e4bc1b87 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -628,6 +628,16 @@ static int fsl_sai_startup(struct snd_pcm_substream *substream,
FSL_SAI_CR3_TRCE_MASK,
FSL_SAI_CR3_TRCE);
+ /*
+ * EDMA controller needs period size to be a multiple of
+ * tx/rx maxburst
+ */
+ if (sai->soc_data->use_edma)
+ snd_pcm_hw_constraint_step(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
+ tx ? sai->dma_params_tx.maxburst :
+ sai->dma_params_rx.maxburst);
+
ret = snd_pcm_hw_constraint_list(substream->runtime, 0,
SNDRV_PCM_HW_PARAM_RATE, &fsl_sai_rate_constraints);
@@ -1026,30 +1036,35 @@ static int fsl_sai_remove(struct platform_device *pdev)
static const struct fsl_sai_soc_data fsl_sai_vf610_data = {
.use_imx_pcm = false,
+ .use_edma = false,
.fifo_depth = 32,
.reg_offset = 0,
};
static const struct fsl_sai_soc_data fsl_sai_imx6sx_data = {
.use_imx_pcm = true,
+ .use_edma = false,
.fifo_depth = 32,
.reg_offset = 0,
};
static const struct fsl_sai_soc_data fsl_sai_imx7ulp_data = {
.use_imx_pcm = true,
+ .use_edma = false,
.fifo_depth = 16,
.reg_offset = 8,
};
static const struct fsl_sai_soc_data fsl_sai_imx8mq_data = {
.use_imx_pcm = true,
+ .use_edma = false,
.fifo_depth = 128,
.reg_offset = 8,
};
static const struct fsl_sai_soc_data fsl_sai_imx8qm_data = {
.use_imx_pcm = true,
+ .use_edma = true,
.fifo_depth = 64,
.reg_offset = 0,
};
diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h
index b12cb578f6d0..76b15deea80c 100644
--- a/sound/soc/fsl/fsl_sai.h
+++ b/sound/soc/fsl/fsl_sai.h
@@ -157,6 +157,7 @@
struct fsl_sai_soc_data {
bool use_imx_pcm;
+ bool use_edma;
unsigned int fifo_depth;
unsigned int reg_offset;
};
diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index f6a7466622ea..fc5d089868df 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -286,6 +286,11 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod,
if (rsnd_ssi_is_multi_slave(mod, io))
return 0;
+ if (rsnd_runtime_is_tdm_split(io))
+ chan = rsnd_io_converted_chan(io);
+
+ chan = rsnd_channel_normalization(chan);
+
if (ssi->usrcnt > 0) {
if (ssi->rate != rate) {
dev_err(dev, "SSI parent/child should use same rate\n");
@@ -300,11 +305,6 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod,
return 0;
}
- if (rsnd_runtime_is_tdm_split(io))
- chan = rsnd_io_converted_chan(io);
-
- chan = rsnd_channel_normalization(chan);
-
main_rate = rsnd_ssi_clk_query(rdai, rate, chan, &idx);
if (!main_rate) {
dev_err(dev, "unsupported clock rate\n");
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 35f48e9c5ead..88978a3036c4 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -978,7 +978,7 @@ static void soc_cleanup_component(struct snd_soc_component *component)
/* For framework level robustness */
snd_soc_component_set_jack(component, NULL, NULL);
- list_del(&component->card_list);
+ list_del_init(&component->card_list);
snd_soc_dapm_free(snd_soc_component_get_dapm(component));
soc_cleanup_component_debugfs(component);
component->card = NULL;
diff --git a/sound/soc/ti/Kconfig b/sound/soc/ti/Kconfig
index 87a9b9dd4e98..29f61053ab62 100644
--- a/sound/soc/ti/Kconfig
+++ b/sound/soc/ti/Kconfig
@@ -200,11 +200,18 @@ config SND_SOC_DM365_AIC3X_CODEC
config SND_SOC_DM365_VOICE_CODEC
bool "Voice Codec - CQ93VC"
- select MFD_DAVINCI_VOICECODEC
- select SND_SOC_CQ0093VC
help
Say Y if you want to add support for SoC On-chip voice codec
endchoice
+config SND_SOC_DM365_VOICE_CODEC_MODULE
+ def_tristate y
+ depends on SND_SOC_DM365_VOICE_CODEC && SND_SOC
+ select MFD_DAVINCI_VOICECODEC
+ select SND_SOC_CQ0093VC
+ help
+ The is an internal symbol needed to ensure that the codec
+ and MFD driver can be built as loadable modules if necessary.
+
endmenu