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authorMark Brown <broonie@kernel.org>2019-01-18 19:14:36 +0000
committerMark Brown <broonie@kernel.org>2019-01-18 19:14:36 +0000
commitf557d39a3cb8ea892dada232c51479ae3628cdee (patch)
tree028e1cb24c8e62663b78689ce7c91b2fdb86551c
parente412fcb0db5c44a3450ca678b281ea9332e6bf82 (diff)
parent4cb79ef9c6c4413427cd70afbb1f3bc01e9b7abf (diff)
Merge tag 'asoc-fix-v5.0-rc2' into asoc-5.1
ASoC: Fixes for v5.0 Quite a big batch of fixes here. There's a couple of things going on, the main one is that we found some issues with not deferring probe when we should, causing us to skip some driver initialization. The fixes for this then in turn exposed some issues with how we were searching for components which had previously gone unnoticed due to the original issue. There's also been the normal driver specific stuff and there's been what looks like several batches of automated scanning for issues which have generated quite a large set of smaller fixes for potential crashes and missed error handling.
-rw-r--r--include/sound/soc.h6
-rw-r--r--sound/soc/amd/raven/acp3x-pcm-dma.c6
-rw-r--r--sound/soc/codecs/rt5514-spi.c2
-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/atom/sst-mfld-platform-pcm.c8
-rw-r--r--sound/soc/intel/boards/glk_rt5682_max98357a.c45
-rw-r--r--sound/soc/soc-core.c34
-rw-r--r--sound/soc/soc-dapm.c10
10 files changed, 87 insertions, 76 deletions
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 8ec1de856ee7..e665f111b0d2 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -985,6 +985,12 @@ struct snd_soc_dai_link {
/* Do not create a PCM for this DAI link (Backend link) */
unsigned int ignore:1;
+ /*
+ * This driver uses legacy platform naming. Set by the core, machine
+ * drivers should not modify this value.
+ */
+ unsigned int legacy_platform:1;
+
struct list_head list; /* DAI link list of the soc card */
struct snd_soc_dobj dobj; /* For topology */
};
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/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.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 521663d8b585..e1bfba62fc08 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -850,6 +850,10 @@ 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/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
index aefa5ce4cb59..b0873fea23ab 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/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c
index 0739e3a75083..f6597c216fa8 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/soc-core.c b/sound/soc/soc-core.c
index 98bb05f6ed56..890d6c9c2752 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",