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-rw-r--r--sound/soc/amd/raven/acp3x-pcm-dma.c6
-rw-r--r--sound/soc/codecs/hdac_hdmi.c116
-rw-r--r--sound/soc/codecs/pcm512x.c11
-rw-r--r--sound/soc/codecs/rt274.c5
-rw-r--r--sound/soc/codecs/rt5514-spi.c2
-rw-r--r--sound/soc/codecs/rt5682.c1
-rw-r--r--sound/soc/codecs/rt5682.h24
-rw-r--r--sound/soc/codecs/tlv320aic32x4.c4
-rw-r--r--sound/soc/fsl/imx-audmux.c24
-rw-r--r--sound/soc/intel/Kconfig2
-rw-r--r--sound/soc/intel/atom/sst-mfld-platform-pcm.c8
-rw-r--r--sound/soc/intel/boards/broadwell.c2
-rw-r--r--sound/soc/intel/boards/glk_rt5682_max98357a.c45
-rw-r--r--sound/soc/intel/boards/haswell.c2
-rw-r--r--sound/soc/intel/skylake/skl.c13
-rw-r--r--sound/soc/qcom/qdsp6/q6asm-dai.c17
-rw-r--r--sound/soc/qcom/sdm845.c31
-rw-r--r--sound/soc/sh/dma-sh7760.c2
-rw-r--r--sound/soc/soc-core.c34
-rw-r--r--sound/soc/soc-dapm.c10
-rw-r--r--sound/soc/ti/davinci-mcasp.c136
-rw-r--r--sound/soc/xilinx/Kconfig2
-rw-r--r--sound/soc/xilinx/xlnx_i2s.c15
23 files changed, 217 insertions, 295 deletions
diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c
index 022a8912c8a2..3d58338fa3cf 100644
--- a/sound/soc/amd/raven/acp3x-pcm-dma.c
+++ b/sound/soc/amd/raven/acp3x-pcm-dma.c
@@ -611,14 +611,16 @@ static int acp3x_audio_probe(struct platform_device *pdev)
}
irqflags = *((unsigned int *)(pdev->dev.platform_data));
- adata = devm_kzalloc(&pdev->dev, sizeof(struct i2s_dev_data),
- GFP_KERNEL);
res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
if (!res) {
dev_err(&pdev->dev, "IORESOURCE_IRQ FAILED\n");
return -ENODEV;
}
+ adata = devm_kzalloc(&pdev->dev, sizeof(*adata), GFP_KERNEL);
+ if (!adata)
+ return -ENOMEM;
+
adata->acp3x_base = devm_ioremap(&pdev->dev, res->start,
resource_size(res));
diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c
index 3ab2949c1dfa..b19d7a3e7a2c 100644
--- a/sound/soc/codecs/hdac_hdmi.c
+++ b/sound/soc/codecs/hdac_hdmi.c
@@ -1890,51 +1890,31 @@ static void hdmi_codec_remove(struct snd_soc_component *component)
pm_runtime_disable(&hdev->dev);
}
-#ifdef CONFIG_PM
-static int hdmi_codec_prepare(struct device *dev)
-{
- struct hdac_device *hdev = dev_to_hdac_dev(dev);
-
- pm_runtime_get_sync(&hdev->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(hdev, hdev->afg, 0, AC_VERB_SET_POWER_STATE,
- AC_PWRST_D3);
-
- return 0;
-}
-
-static void hdmi_codec_complete(struct device *dev)
+#ifdef CONFIG_PM_SLEEP
+static int hdmi_codec_resume(struct device *dev)
{
struct hdac_device *hdev = dev_to_hdac_dev(dev);
struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev);
+ int ret;
- /* Power up afg */
- snd_hdac_codec_read(hdev, hdev->afg, 0, AC_VERB_SET_POWER_STATE,
- AC_PWRST_D0);
-
- hdac_hdmi_skl_enable_all_pins(hdev);
- hdac_hdmi_skl_enable_dp12(hdev);
-
+ ret = pm_runtime_force_resume(dev);
+ if (ret < 0)
+ return ret;
/*
* As the ELD notify callback request is not entertained while the
* device is in suspend state. Need to manually check detection of
* all pins here. pin capablity change is not support, so use the
* already set pin caps.
+ *
+ * NOTE: this is safe to call even if the codec doesn't actually resume.
+ * The pin check involves only with DRM audio component hooks, so it
+ * works even if the HD-audio side is still dreaming peacefully.
*/
hdac_hdmi_present_sense_all_pins(hdev, hdmi, false);
-
- pm_runtime_put_sync(&hdev->dev);
+ return 0;
}
#else
-#define hdmi_codec_prepare NULL
-#define hdmi_codec_complete NULL
+#define hdmi_codec_resume NULL
#endif
static const struct snd_soc_component_driver hdmi_hda_codec = {
@@ -2135,75 +2115,6 @@ static int hdac_hdmi_dev_remove(struct hdac_device *hdev)
}
#ifdef CONFIG_PM
-/*
- * Power management sequences
- * ==========================
- *
- * The following explains the PM handling of HDAC HDMI with its parent
- * device SKL and display power usage
- *
- * Probe
- * -----
- * In SKL probe,
- * 1. skl_probe_work() powers up the display (refcount++ -> 1)
- * 2. enumerates the codecs on the link
- * 3. powers down the display (refcount-- -> 0)
- *
- * In HDAC HDMI probe,
- * 1. hdac_hdmi_dev_probe() powers up the display (refcount++ -> 1)
- * 2. probe the codec
- * 3. put the HDAC HDMI device to runtime suspend
- * 4. hdac_hdmi_runtime_suspend() powers down the display (refcount-- -> 0)
- *
- * Once children are runtime suspended, SKL device also goes to runtime
- * suspend
- *
- * HDMI Playback
- * -------------
- * Open HDMI device,
- * 1. skl_runtime_resume() invoked
- * 2. hdac_hdmi_runtime_resume() powers up the display (refcount++ -> 1)
- *
- * Close HDMI device,
- * 1. hdac_hdmi_runtime_suspend() powers down the display (refcount-- -> 0)
- * 2. skl_runtime_suspend() invoked
- *
- * S0/S3 Cycle with playback in progress
- * -------------------------------------
- * When the device is opened for playback, the device is runtime active
- * already and the display refcount is 1 as explained above.
- *
- * Entering to S3,
- * 1. hdmi_codec_prepare() invoke the runtime resume of codec which just
- * increments the PM runtime usage count of the codec since the device
- * is in use already
- * 2. skl_suspend() powers down the display (refcount-- -> 0)
- *
- * Wakeup from S3,
- * 1. skl_resume() powers up the display (refcount++ -> 1)
- * 2. hdmi_codec_complete() invokes the runtime suspend of codec which just
- * decrements the PM runtime usage count of the codec since the device
- * is in use already
- *
- * Once playback is stopped, the display refcount is set to 0 as explained
- * above in the HDMI playback sequence. The PM handlings are designed in
- * such way that to balance the refcount of display power when the codec
- * device put to S3 while playback is going on.
- *
- * S0/S3 Cycle without playback in progress
- * ----------------------------------------
- * Entering to S3,
- * 1. hdmi_codec_prepare() invoke the runtime resume of codec
- * 2. skl_runtime_resume() invoked
- * 3. hdac_hdmi_runtime_resume() powers up the display (refcount++ -> 1)
- * 4. skl_suspend() powers down the display (refcount-- -> 0)
- *
- * Wakeup from S3,
- * 1. skl_resume() powers up the display (refcount++ -> 1)
- * 2. hdmi_codec_complete() invokes the runtime suspend of codec
- * 3. hdac_hdmi_runtime_suspend() powers down the display (refcount-- -> 0)
- * 4. skl_runtime_suspend() invoked
- */
static int hdac_hdmi_runtime_suspend(struct device *dev)
{
struct hdac_device *hdev = dev_to_hdac_dev(dev);
@@ -2277,8 +2188,7 @@ 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,
+ SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, hdmi_codec_resume)
};
static const struct hda_device_id hdmi_list[] = {
diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c
index 6cb1653be804..4cc24a5d5c31 100644
--- a/sound/soc/codecs/pcm512x.c
+++ b/sound/soc/codecs/pcm512x.c
@@ -1400,24 +1400,20 @@ static int pcm512x_digital_mute(struct snd_soc_dai *dai, int mute)
if (ret != 0) {
dev_err(component->dev,
"Failed to set digital mute: %d\n", ret);
- mutex_unlock(&pcm512x->mutex);
- return ret;
+ goto unlock;
}
regmap_read_poll_timeout(pcm512x->regmap,
PCM512x_ANALOG_MUTE_DET,
mute_det, (mute_det & 0x3) == 0,
200, 10000);
-
- mutex_unlock(&pcm512x->mutex);
} else {
pcm512x->mute &= ~0x1;
ret = pcm512x_update_mute(pcm512x);
if (ret != 0) {
dev_err(component->dev,
"Failed to update digital mute: %d\n", ret);
- mutex_unlock(&pcm512x->mutex);
- return ret;
+ goto unlock;
}
regmap_read_poll_timeout(pcm512x->regmap,
@@ -1428,9 +1424,10 @@ static int pcm512x_digital_mute(struct snd_soc_dai *dai, int mute)
200, 10000);
}
+unlock:
mutex_unlock(&pcm512x->mutex);
- return 0;
+ return ret;
}
static const struct snd_soc_dai_ops pcm512x_dai_ops = {
diff --git a/sound/soc/codecs/rt274.c b/sound/soc/codecs/rt274.c
index 0ef966d56bac..e2855ab9a2c6 100644
--- a/sound/soc/codecs/rt274.c
+++ b/sound/soc/codecs/rt274.c
@@ -1128,8 +1128,11 @@ static int rt274_i2c_probe(struct i2c_client *i2c,
return ret;
}
- regmap_read(rt274->regmap,
+ ret = regmap_read(rt274->regmap,
RT274_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &val);
+ if (ret)
+ return ret;
+
if (val != RT274_VENDOR_ID) {
dev_err(&i2c->dev,
"Device with ID register %#x is not rt274\n", val);
diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c
index 4d46f4567c3a..bec2eefa8b0f 100644
--- a/sound/soc/codecs/rt5514-spi.c
+++ b/sound/soc/codecs/rt5514-spi.c
@@ -280,6 +280,8 @@ static int rt5514_spi_pcm_probe(struct snd_soc_component *component)
rt5514_dsp = devm_kzalloc(component->dev, sizeof(*rt5514_dsp),
GFP_KERNEL);
+ if (!rt5514_dsp)
+ return -ENOMEM;
rt5514_dsp->dev = &rt5514_spi->dev;
mutex_init(&rt5514_dsp->dma_lock);
diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c
index 34cfaf8f6f34..89c43b26c379 100644
--- a/sound/soc/codecs/rt5682.c
+++ b/sound/soc/codecs/rt5682.c
@@ -2512,6 +2512,7 @@ static void rt5682_calibrate(struct rt5682_priv *rt5682)
regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0000);
regmap_write(rt5682->regmap, RT5682_CHOP_DAC, 0x2000);
regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x2005);
+ regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0xc0c4);
mutex_unlock(&rt5682->calibrate_mutex);
diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h
index d82a8301fd74..96944cff0ed7 100644
--- a/sound/soc/codecs/rt5682.h
+++ b/sound/soc/codecs/rt5682.h
@@ -849,18 +849,18 @@
#define RT5682_SCLK_SRC_PLL2 (0x2 << 13)
#define RT5682_SCLK_SRC_SDW (0x3 << 13)
#define RT5682_SCLK_SRC_RCCLK (0x4 << 13)
-#define RT5682_PLL1_SRC_MASK (0x3 << 10)
-#define RT5682_PLL1_SRC_SFT 10
-#define RT5682_PLL1_SRC_MCLK (0x0 << 10)
-#define RT5682_PLL1_SRC_BCLK1 (0x1 << 10)
-#define RT5682_PLL1_SRC_SDW (0x2 << 10)
-#define RT5682_PLL1_SRC_RC (0x3 << 10)
-#define RT5682_PLL2_SRC_MASK (0x3 << 8)
-#define RT5682_PLL2_SRC_SFT 8
-#define RT5682_PLL2_SRC_MCLK (0x0 << 8)
-#define RT5682_PLL2_SRC_BCLK1 (0x1 << 8)
-#define RT5682_PLL2_SRC_SDW (0x2 << 8)
-#define RT5682_PLL2_SRC_RC (0x3 << 8)
+#define RT5682_PLL2_SRC_MASK (0x3 << 10)
+#define RT5682_PLL2_SRC_SFT 10
+#define RT5682_PLL2_SRC_MCLK (0x0 << 10)
+#define RT5682_PLL2_SRC_BCLK1 (0x1 << 10)
+#define RT5682_PLL2_SRC_SDW (0x2 << 10)
+#define RT5682_PLL2_SRC_RC (0x3 << 10)
+#define RT5682_PLL1_SRC_MASK (0x3 << 8)
+#define RT5682_PLL1_SRC_SFT 8
+#define RT5682_PLL1_SRC_MCLK (0x0 << 8)
+#define RT5682_PLL1_SRC_BCLK1 (0x1 << 8)
+#define RT5682_PLL1_SRC_SDW (0x2 << 8)
+#define RT5682_PLL1_SRC_RC (0x3 << 8)
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index e2b5a11b16d1..f03195d2ab2e 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -822,6 +822,10 @@ static int aic32x4_set_bias_level(struct snd_soc_component *component,
case SND_SOC_BIAS_PREPARE:
break;
case SND_SOC_BIAS_STANDBY:
+ /* Initial cold start */
+ if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_OFF)
+ break;
+
/* Switch off BCLK_N Divider */
snd_soc_component_update_bits(component, AIC32X4_BCLKN,
AIC32X4_BCLKEN, 0);
diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c
index 392d5eef356d..99e07b01a2ce 100644
--- a/sound/soc/fsl/imx-audmux.c
+++ b/sound/soc/fsl/imx-audmux.c
@@ -86,49 +86,49 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf,
if (!buf)
return -ENOMEM;
- ret = snprintf(buf, PAGE_SIZE, "PDCR: %08x\nPTCR: %08x\n",
+ ret = scnprintf(buf, PAGE_SIZE, "PDCR: %08x\nPTCR: %08x\n",
pdcr, ptcr);
if (ptcr & IMX_AUDMUX_V2_PTCR_TFSDIR)
- ret += snprintf(buf + ret, PAGE_SIZE - ret,
+ ret += scnprintf(buf + ret, PAGE_SIZE - ret,
"TxFS output from %s, ",
audmux_port_string((ptcr >> 27) & 0x7));
else
- ret += snprintf(buf + ret, PAGE_SIZE - ret,
+ ret += scnprintf(buf + ret, PAGE_SIZE - ret,
"TxFS input, ");
if (ptcr & IMX_AUDMUX_V2_PTCR_TCLKDIR)
- ret += snprintf(buf + ret, PAGE_SIZE - ret,
+ ret += scnprintf(buf + ret, PAGE_SIZE - ret,
"TxClk output from %s",
audmux_port_string((ptcr >> 22) & 0x7));
else
- ret += snprintf(buf + ret, PAGE_SIZE - ret,
+ ret += scnprintf(buf + ret, PAGE_SIZE - ret,
"TxClk input");
- ret += snprintf(buf + ret, PAGE_SIZE - ret, "\n");
+ ret += scnprintf(buf + ret, PAGE_SIZE - ret, "\n");
if (ptcr & IMX_AUDMUX_V2_PTCR_SYN) {
- ret += snprintf(buf + ret, PAGE_SIZE - ret,
+ ret += scnprintf(buf + ret, PAGE_SIZE - ret,
"Port is symmetric");
} else {
if (ptcr & IMX_AUDMUX_V2_PTCR_RFSDIR)
- ret += snprintf(buf + ret, PAGE_SIZE - ret,
+ ret += scnprintf(buf + ret, PAGE_SIZE - ret,
"RxFS output from %s, ",
audmux_port_string((ptcr >> 17) & 0x7));
else
- ret += snprintf(buf + ret, PAGE_SIZE - ret,
+ ret += scnprintf(buf + ret, PAGE_SIZE - ret,
"RxFS input, ");
if (ptcr & IMX_AUDMUX_V2_PTCR_RCLKDIR)
- ret += snprintf(buf + ret, PAGE_SIZE - ret,
+ ret += scnprintf(buf + ret, PAGE_SIZE - ret,
"RxClk output from %s",
audmux_port_string((ptcr >> 12) & 0x7));
else
- ret += snprintf(buf + ret, PAGE_SIZE - ret,
+ ret += scnprintf(buf + ret, PAGE_SIZE - ret,
"RxClk input");
}
- ret += snprintf(buf + ret, PAGE_SIZE - ret,
+ ret += scnprintf(buf + ret, PAGE_SIZE - ret,
"\nData received from %s\n",
audmux_port_string((pdcr >> 13) & 0x7));
diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig
index 99a62ba409df..bd9fd2035c55 100644
--- a/sound/soc/intel/Kconfig
+++ b/sound/soc/intel/Kconfig
@@ -91,7 +91,7 @@ config SND_SST_ATOM_HIFI2_PLATFORM_PCI
config SND_SST_ATOM_HIFI2_PLATFORM_ACPI
tristate "ACPI HiFi2 (Baytrail, Cherrytrail) Platforms"
default ACPI
- depends on X86 && ACPI
+ depends on X86 && ACPI && PCI
select SND_SST_IPC_ACPI
select SND_SST_ATOM_HIFI2_PLATFORM
select SND_SOC_ACPI_INTEL_MATCH
diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
index afc559866095..91a2436ce952 100644
--- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c
+++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
@@ -399,7 +399,13 @@ static int sst_media_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
- snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
+ int ret;
+
+ ret =
+ snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(params));
+ if (ret)
+ return ret;
memset(substream->runtime->dma_area, 0, params_buffer_bytes(params));
return 0;
}
diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c
index 68e6543e6cb0..99f2a0156ae8 100644
--- a/sound/soc/intel/boards/broadwell.c
+++ b/sound/soc/intel/boards/broadwell.c
@@ -192,7 +192,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = {
.stream_name = "Loopback",
.cpu_dai_name = "Loopback Pin",
.platform_name = "haswell-pcm-audio",
- .dynamic = 0,
+ .dynamic = 1,
.codec_name = "snd-soc-dummy",
.codec_dai_name = "snd-soc-dummy-dai",
.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c
index c74c4f17316f..8f83b182c4f9 100644
--- a/sound/soc/intel/boards/glk_rt5682_max98357a.c
+++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c
@@ -55,39 +55,6 @@ enum {
GLK_DPCM_AUDIO_HDMI3_PB,
};
-static int platform_clock_control(struct snd_soc_dapm_widget *w,
- struct snd_kcontrol *k, int event)
-{
- struct snd_soc_dapm_context *dapm = w->dapm;
- struct snd_soc_card *card = dapm->card;
- struct snd_soc_dai *codec_dai;
- int ret = 0;
-
- codec_dai = snd_soc_card_get_codec_dai(card, GLK_REALTEK_CODEC_DAI);
- if (!codec_dai) {
- dev_err(card->dev, "Codec dai not found; Unable to set/unset codec pll\n");
- return -EIO;
- }
-
- if (SND_SOC_DAPM_EVENT_OFF(event)) {
- ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, 0);
- if (ret)
- dev_err(card->dev, "failed to stop sysclk: %d\n", ret);
- } else if (SND_SOC_DAPM_EVENT_ON(event)) {
- ret = snd_soc_dai_set_pll(codec_dai, 0, RT5682_PLL1_S_MCLK,
- GLK_PLAT_CLK_FREQ, RT5682_PLL_FREQ);
- if (ret < 0) {
- dev_err(card->dev, "can't set codec pll: %d\n", ret);
- return ret;
- }
- }
-
- if (ret)
- dev_err(card->dev, "failed to start internal clk: %d\n", ret);
-
- return ret;
-}
-
static const struct snd_kcontrol_new geminilake_controls[] = {
SOC_DAPM_PIN_SWITCH("Headphone Jack"),
SOC_DAPM_PIN_SWITCH("Headset Mic"),
@@ -102,14 +69,10 @@ static const struct snd_soc_dapm_widget geminilake_widgets[] = {
SND_SOC_DAPM_SPK("HDMI1", NULL),
SND_SOC_DAPM_SPK("HDMI2", NULL),
SND_SOC_DAPM_SPK("HDMI3", NULL),
- SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0,
- platform_clock_control, SND_SOC_DAPM_PRE_PMU |
- SND_SOC_DAPM_POST_PMD),
};
static const struct snd_soc_dapm_route geminilake_map[] = {
/* HP jack connectors - unknown if we have jack detection */
- { "Headphone Jack", NULL, "Platform Clock" },
{ "Headphone Jack", NULL, "HPOL" },
{ "Headphone Jack", NULL, "HPOR" },
@@ -117,7 +80,6 @@ static const struct snd_soc_dapm_route geminilake_map[] = {
{ "Spk", NULL, "Speaker" },
/* other jacks */
- { "Headset Mic", NULL, "Platform Clock" },
{ "IN1P", NULL, "Headset Mic" },
/* digital mics */
@@ -177,6 +139,13 @@ static int geminilake_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd)
struct snd_soc_jack *jack;
int ret;
+ ret = snd_soc_dai_set_pll(codec_dai, 0, RT5682_PLL1_S_MCLK,
+ GLK_PLAT_CLK_FREQ, RT5682_PLL_FREQ);
+ if (ret < 0) {
+ dev_err(rtd->dev, "can't set codec pll: %d\n", ret);
+ return ret;
+ }
+
/* Configure sysclk for codec */
ret = snd_soc_dai_set_sysclk(codec_dai, RT5682_SCLK_S_PLL1,
RT5682_PLL_FREQ, SND_SOC_CLOCK_IN);
diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c
index eab1f439dd3f..a4022983a7ce 100644
--- a/sound/soc/intel/boards/haswell.c
+++ b/sound/soc/intel/boards/haswell.c
@@ -146,7 +146,7 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = {
.stream_name = "Loopback",
.cpu_dai_name = "Loopback Pin",
.platform_name = "haswell-pcm-audio",
- .dynamic = 0,
+ .dynamic = 1,
.codec_name = "snd-soc-dummy",
.codec_dai_name = "snd-soc-dummy-dai",
.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c
index 60c94836bf5b..4ed5b7e17d44 100644
--- a/sound/soc/intel/skylake/skl.c
+++ b/sound/soc/intel/skylake/skl.c
@@ -336,9 +336,6 @@ static int skl_suspend(struct device *dev)
skl->skl_sst->fw_loaded = false;
}
- if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI))
- snd_hdac_display_power(bus, HDA_CODEC_IDX_CONTROLLER, false);
-
return 0;
}
@@ -350,10 +347,6 @@ static int skl_resume(struct device *dev)
struct hdac_ext_link *hlink = NULL;
int ret;
- /* Turned OFF in HDMI codec driver after codec reconfiguration */
- if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI))
- snd_hdac_display_power(bus, HDA_CODEC_IDX_CONTROLLER, true);
-
/*
* resume only when we are not in suspend active, otherwise need to
* restore the device
@@ -446,8 +439,10 @@ static int skl_free(struct hdac_bus *bus)
snd_hdac_ext_bus_exit(bus);
cancel_work_sync(&skl->probe_work);
- if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI))
+ if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) {
+ snd_hdac_display_power(bus, HDA_CODEC_IDX_CONTROLLER, false);
snd_hdac_i915_exit(bus);
+ }
return 0;
}
@@ -814,7 +809,7 @@ static void skl_probe_work(struct work_struct *work)
err = skl_platform_register(bus->dev);
if (err < 0) {
dev_err(bus->dev, "platform register failed: %d\n", err);
- return;
+ goto out_err;
}
err = skl_machine_device_register(skl);
diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
index 5b986b74dd36..548eb4fa2da6 100644
--- a/sound/soc/qcom/qdsp6/q6asm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -570,10 +570,10 @@ static int q6asm_dai_compr_open(struct snd_compr_stream *stream)
prtd->audio_client = q6asm_audio_client_alloc(dev,
(q6asm_cb)compress_event_handler,
prtd, stream_id, LEGACY_PCM_MODE);
- if (!prtd->audio_client) {
+ if (IS_ERR(prtd->audio_client)) {
dev_err(dev, "Could not allocate memory\n");
- kfree(prtd);
- return -ENOMEM;
+ ret = PTR_ERR(prtd->audio_client);
+ goto free_prtd;
}
size = COMPR_PLAYBACK_MAX_FRAGMENT_SIZE *
@@ -582,7 +582,7 @@ static int q6asm_dai_compr_open(struct snd_compr_stream *stream)
&prtd->dma_buffer);
if (ret) {
dev_err(dev, "Cannot allocate buffer(s)\n");
- return ret;
+ goto free_client;
}
if (pdata->sid < 0)
@@ -595,6 +595,13 @@ static int q6asm_dai_compr_open(struct snd_compr_stream *stream)
runtime->private_data = prtd;
return 0;
+
+free_client:
+ q6asm_audio_client_free(prtd->audio_client);
+free_prtd:
+ kfree(prtd);
+
+ return ret;
}
static int q6asm_dai_compr_free(struct snd_compr_stream *stream)
@@ -874,7 +881,7 @@ static int of_q6asm_parse_dai_data(struct device *dev,
for_each_child_of_node(dev->of_node, node) {
ret = of_property_read_u32(node, "reg", &id);
- if (ret || id > MAX_SESSIONS || id < 0) {
+ if (ret || id >= MAX_SESSIONS || id < 0) {
dev_err(dev, "valid dai id not found:%d\n", ret);
continue;
}
diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c
index 1db8ef668223..6f66a58e23ca 100644
--- a/sound/soc/qcom/sdm845.c
+++ b/sound/soc/qcom/sdm845.c
@@ -158,17 +158,24 @@ static int sdm845_snd_hw_params(struct snd_pcm_substream *substream,
return ret;
}
+static void sdm845_jack_free(struct snd_jack *jack)
+{
+ struct snd_soc_component *component = jack->private_data;
+
+ snd_soc_component_set_jack(component, NULL, NULL);
+}
+
static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_component *component;
- struct snd_soc_dai_link *dai_link = rtd->dai_link;
struct snd_soc_card *card = rtd->card;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
struct sdm845_snd_data *pdata = snd_soc_card_get_drvdata(card);
- int i, rval;
+ struct snd_jack *jack;
+ int rval;
if (!pdata->jack_setup) {
- struct snd_jack *jack;
-
rval = snd_soc_card_jack_new(card, "Headset Jack",
SND_JACK_HEADSET |
SND_JACK_HEADPHONE |
@@ -190,16 +197,22 @@ static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd)
pdata->jack_setup = true;
}
- for (i = 0 ; i < dai_link->num_codecs; i++) {
- struct snd_soc_dai *dai = rtd->codec_dais[i];
+ switch (cpu_dai->id) {
+ case PRIMARY_MI2S_RX:
+ jack = pdata->jack.jack;
+ component = codec_dai->component;
- component = dai->component;
- rval = snd_soc_component_set_jack(
- component, &pdata->jack, NULL);
+ jack->private_data = component;
+ jack->private_free = sdm845_jack_free;
+ rval = snd_soc_component_set_jack(component,
+ &pdata->jack, NULL);
if (rval != 0 && rval != -ENOTSUPP) {
dev_warn(card->dev, "Failed to set jack: %d\n", rval);
return rval;
}
+ break;
+ default:
+ break;
}
return 0;
diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c
index 922fb6aa3ed1..5aee11c94f2a 100644
--- a/sound/soc/sh/dma-sh7760.c
+++ b/sound/soc/sh/dma-sh7760.c
@@ -202,7 +202,7 @@ static int camelot_prepare(struct snd_pcm_substream *substream)
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct camelot_pcm *cam = &cam_pcm_data[rtd->cpu_dai->id];
- pr_debug("PCM data: addr 0x%08ulx len %d\n",
+ pr_debug("PCM data: addr 0x%08lx len %d\n",
(u32)runtime->dma_addr, runtime->dma_bytes);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 0462b3ec977a..aae450ba4f08 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -742,7 +742,7 @@ static struct snd_soc_component *soc_find_component(
if (of_node) {
if (component->dev->of_node == of_node)
return component;
- } else if (strcmp(component->name, name) == 0) {
+ } else if (name && strcmp(component->name, name) == 0) {
return component;
}
}
@@ -1034,17 +1034,18 @@ static int snd_soc_init_platform(struct snd_soc_card *card,
* this function should be removed in the future
*/
/* convert Legacy platform link */
- if (!platform) {
+ if (!platform || dai_link->legacy_platform) {
platform = devm_kzalloc(card->dev,
sizeof(struct snd_soc_dai_link_component),
GFP_KERNEL);
if (!platform)
return -ENOMEM;
- dai_link->platform = platform;
- platform->name = dai_link->platform_name;
- platform->of_node = dai_link->platform_of_node;
- platform->dai_name = NULL;
+ dai_link->platform = platform;
+ dai_link->legacy_platform = 1;
+ platform->name = dai_link->platform_name;
+ platform->of_node = dai_link->platform_of_node;
+ platform->dai_name = NULL;
}
/* if there's no platform we match on the empty platform */
@@ -1129,6 +1130,15 @@ static int soc_init_dai_link(struct snd_soc_card *card,
link->name);
return -EINVAL;
}
+
+ /*
+ * Defer card registartion if platform dai component is not added to
+ * component list.
+ */
+ if ((link->platform->of_node || link->platform->name) &&
+ !soc_find_component(link->platform->of_node, link->platform->name))
+ return -EPROBE_DEFER;
+
/*
* CPU device may be specified by either name or OF node, but
* can be left unspecified, and will be matched based on DAI
@@ -1140,6 +1150,15 @@ static int soc_init_dai_link(struct snd_soc_card *card,
link->name);
return -EINVAL;
}
+
+ /*
+ * Defer card registartion if cpu dai component is not added to
+ * component list.
+ */
+ if ((link->cpu_of_node || link->cpu_name) &&
+ !soc_find_component(link->cpu_of_node, link->cpu_name))
+ return -EPROBE_DEFER;
+
/*
* At least one of CPU DAI name or CPU device name/node must be
* specified
@@ -2739,15 +2758,18 @@ int snd_soc_register_card(struct snd_soc_card *card)
if (!card->name || !card->dev)
return -EINVAL;
+ mutex_lock(&client_mutex);
for_each_card_prelinks(card, i, link) {
ret = soc_init_dai_link(card, link);
if (ret) {
dev_err(card->dev, "ASoC: failed to init link %s\n",
link->name);
+ mutex_unlock(&client_mutex);
return ret;
}
}
+ mutex_unlock(&client_mutex);
dev_set_drvdata(card->dev, card);
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index a5178845065b..2c4c13419539 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -2019,19 +2019,19 @@ static ssize_t dapm_widget_power_read_file(struct file *file,
out = is_connected_output_ep(w, NULL, NULL);
}
- ret = snprintf(buf, PAGE_SIZE, "%s: %s%s in %d out %d",
+ ret = scnprintf(buf, PAGE_SIZE, "%s: %s%s in %d out %d",
w->name, w->power ? "On" : "Off",
w->force ? " (forced)" : "", in, out);
if (w->reg >= 0)
- ret += snprintf(buf + ret, PAGE_SIZE - ret,
+ ret += scnprintf(buf + ret, PAGE_SIZE - ret,
" - R%d(0x%x) mask 0x%x",
w->reg, w->reg, w->mask << w->shift);
- ret += snprintf(buf + ret, PAGE_SIZE - ret, "\n");
+ ret += scnprintf(buf + ret, PAGE_SIZE - ret, "\n");
if (w->sname)
- ret += snprintf(buf + ret, PAGE_SIZE - ret, " stream %s %s\n",
+ ret += scnprintf(buf + ret, PAGE_SIZE - ret, " stream %s %s\n",
w->sname,
w->active ? "active" : "inactive");
@@ -2044,7 +2044,7 @@ static ssize_t dapm_widget_power_read_file(struct file *file,
if (!p->connect)
continue;
- ret += snprintf(buf + ret, PAGE_SIZE - ret,
+ ret += scnprintf(buf + ret, PAGE_SIZE - ret,
" %s \"%s\" \"%s\"\n",
(rdir == SND_SOC_DAPM_DIR_IN) ? "in" : "out",
p->name ? p->name : "static",
diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c
index eeda6d5565bc..a10fcb5963c6 100644
--- a/sound/soc/ti/davinci-mcasp.c
+++ b/sound/soc/ti/davinci-mcasp.c
@@ -108,7 +108,7 @@ struct davinci_mcasp {
/* Used for comstraint setting on the second stream */
u32 channels;
-#ifdef CONFIG_PM_SLEEP
+#ifdef CONFIG_PM
struct davinci_mcasp_context context;
#endif
@@ -1486,74 +1486,6 @@ static int davinci_mcasp_dai_probe(struct snd_soc_dai *dai)
return 0;
}
-#ifdef CONFIG_PM_SLEEP
-static int davinci_mcasp_suspend(struct snd_soc_dai *dai)
-{
- struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai);
- struct davinci_mcasp_context *context = &mcasp->context;
- u32 reg;
- int i;
-
- context->pm_state = pm_runtime_active(mcasp->dev);
- if (!context->pm_state)
- pm_runtime_get_sync(mcasp->dev);
-
- for (i = 0; i < ARRAY_SIZE(context_regs); i++)
- context->config_regs[i] = mcasp_get_reg(mcasp, context_regs[i]);
-
- if (mcasp->txnumevt) {
- reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET;
- context->afifo_regs[0] = mcasp_get_reg(mcasp, reg);
- }
- if (mcasp->rxnumevt) {
- reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET;
- context->afifo_regs[1] = mcasp_get_reg(mcasp, reg);
- }
-
- for (i = 0; i < mcasp->num_serializer; i++)
- context->xrsr_regs[i] = mcasp_get_reg(mcasp,
- DAVINCI_MCASP_XRSRCTL_REG(i));
-
- pm_runtime_put_sync(mcasp->dev);
-
- return 0;
-}
-
-static int davinci_mcasp_resume(struct snd_soc_dai *dai)
-{
- struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai);
- struct davinci_mcasp_context *context = &mcasp->context;
- u32 reg;
- int i;
-
- pm_runtime_get_sync(mcasp->dev);
-
- for (i = 0; i < ARRAY_SIZE(context_regs); i++)
- mcasp_set_reg(mcasp, context_regs[i], context->config_regs[i]);
-
- if (mcasp->txnumevt) {
- reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET;
- mcasp_set_reg(mcasp, reg, context->afifo_regs[0]);
- }
- if (mcasp->rxnumevt) {
- reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET;
- mcasp_set_reg(mcasp, reg, context->afifo_regs[1]);
- }
-
- for (i = 0; i < mcasp->num_serializer; i++)
- mcasp_set_reg(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i),
- context->xrsr_regs[i]);
-
- if (!context->pm_state)
- pm_runtime_put_sync(mcasp->dev);
-
- return 0;
-}
-#else
-#define davinci_mcasp_suspend NULL
-#define davinci_mcasp_resume NULL
-#endif
-
#define DAVINCI_MCASP_RATES SNDRV_PCM_RATE_8000_192000
#define DAVINCI_MCASP_PCM_FMTS (SNDRV_PCM_FMTBIT_S8 | \
@@ -1571,8 +1503,6 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = {
{
.name = "davinci-mcasp.0",
.probe = davinci_mcasp_dai_probe,
- .suspend = davinci_mcasp_suspend,
- .resume = davinci_mcasp_resume,
.playback = {
.channels_min = 1,
.channels_max = 32 * 16,
@@ -1976,7 +1906,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
}
mcasp->num_serializer = pdata->num_serializer;
-#ifdef CONFIG_PM_SLEEP
+#ifdef CONFIG_PM
mcasp->context.xrsr_regs = devm_kcalloc(&pdev->dev,
mcasp->num_serializer, sizeof(u32),
GFP_KERNEL);
@@ -2196,11 +2126,73 @@ static int davinci_mcasp_remove(struct platform_device *pdev)
return 0;
}
+#ifdef CONFIG_PM
+static int davinci_mcasp_runtime_suspend(struct device *dev)
+{
+ struct davinci_mcasp *mcasp = dev_get_drvdata(dev);
+ struct davinci_mcasp_context *context = &mcasp->context;
+ u32 reg;
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(context_regs); i++)
+ context->config_regs[i] = mcasp_get_reg(mcasp, context_regs[i]);
+
+ if (mcasp->txnumevt) {
+ reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET;
+ context->afifo_regs[0] = mcasp_get_reg(mcasp, reg);
+ }
+ if (mcasp->rxnumevt) {
+ reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET;
+ context->afifo_regs[1] = mcasp_get_reg(mcasp, reg);
+ }
+
+ for (i = 0; i < mcasp->num_serializer; i++)
+ context->xrsr_regs[i] = mcasp_get_reg(mcasp,
+ DAVINCI_MCASP_XRSRCTL_REG(i));
+
+ return 0;
+}
+
+static int davinci_mcasp_runtime_resume(struct device *dev)
+{
+ struct davinci_mcasp *mcasp = dev_get_drvdata(dev);
+ struct davinci_mcasp_context *context = &mcasp->context;
+ u32 reg;
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(context_regs); i++)
+ mcasp_set_reg(mcasp, context_regs[i], context->config_regs[i]);
+
+ if (mcasp->txnumevt) {
+ reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET;
+ mcasp_set_reg(mcasp, reg, context->afifo_regs[0]);
+ }
+ if (mcasp->rxnumevt) {
+ reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET;
+ mcasp_set_reg(mcasp, reg, context->afifo_regs[1]);
+ }
+
+ for (i = 0; i < mcasp->num_serializer; i++)
+ mcasp_set_reg(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i),
+ context->xrsr_regs[i]);
+
+ return 0;
+}
+
+#endif
+
+static const struct dev_pm_ops davinci_mcasp_pm_ops = {
+ SET_RUNTIME_PM_OPS(davinci_mcasp_runtime_suspend,
+ davinci_mcasp_runtime_resume,
+ NULL)
+};
+
static struct platform_driver davinci_mcasp_driver = {
.probe = davinci_mcasp_probe,
.remove = davinci_mcasp_remove,
.driver = {
.name = "davinci-mcasp",
+ .pm = &davinci_mcasp_pm_ops,
.of_match_table = mcasp_dt_ids,
},
};
diff --git a/sound/soc/xilinx/Kconfig b/sound/soc/xilinx/Kconfig
index 25e287feb58c..723a583a8d57 100644
--- a/sound/soc/xilinx/Kconfig
+++ b/sound/soc/xilinx/Kconfig
@@ -1,5 +1,5 @@
config SND_SOC_XILINX_I2S
- tristate "Audio support for the the Xilinx I2S"
+ tristate "Audio support for the Xilinx I2S"
help
Select this option to enable Xilinx I2S Audio. This enables
I2S playback and capture using xilinx soft IP. In transmitter
diff --git a/sound/soc/xilinx/xlnx_i2s.c b/sound/soc/xilinx/xlnx_i2s.c
index d4ae9eff41ce..8b353166ad44 100644
--- a/sound/soc/xilinx/xlnx_i2s.c
+++ b/sound/soc/xilinx/xlnx_i2s.c
@@ -1,12 +1,11 @@
// SPDX-License-Identifier: GPL-2.0
-/*
- * Xilinx ASoC I2S audio support
- *
- * Copyright (C) 2018 Xilinx, Inc.
- *
- * Author: Praveen Vuppala <praveenv@xilinx.com>
- * Author: Maruthi Srinivas Bayyavarapu <maruthis@xilinx.com>
- */
+//
+// Xilinx ASoC I2S audio support
+//
+// Copyright (C) 2018 Xilinx, Inc.
+//
+// Author: Praveen Vuppala <praveenv@xilinx.com>
+// Author: Maruthi Srinivas Bayyavarapu <maruthis@xilinx.com>
#include <linux/io.h>
#include <linux/module.h>
generated by cgit v1.2.3-58-ga151 (git 2.39.0) at 2024-11-22 20:09:58 +0000