summaryrefslogtreecommitdiffstats
path: root/sound
diff options
context:
space:
mode:
authorTakashi Iwai <tiwai@suse.de>2015-07-24 20:08:13 +0200
committerTakashi Iwai <tiwai@suse.de>2015-07-24 20:08:13 +0200
commit43cbf02e7ad51007af38f39c5b2abdc7a5d7f5aa (patch)
tree1057babea8807af3f4a3c44fd116b7bbe99eb733 /sound
parentALSA: hda - Add headset mic pin quirk for a Dell device (diff)
parentMerge remote-tracking branches 'asoc/fix/sgtl5000', 'asoc/fix/topology' and '... (diff)
downloadlinux-43cbf02e7ad51007af38f39c5b2abdc7a5d7f5aa.tar.xz
linux-43cbf02e7ad51007af38f39c5b2abdc7a5d7f5aa.zip
Merge tag 'asoc-fix-v4.2-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v4.2 A lot of small fixes here, a few to the core: - Fix for binding DAPM stream widgets on devices with prefixes assigned to them - Minor fixes for the newly added topology interfaces - Locking and memory leak fixes for DAPM - Driver specific fixes
Diffstat (limited to 'sound')
-rw-r--r--sound/core/hrtimer.c9
-rw-r--r--sound/core/memalloc.c2
-rw-r--r--sound/drivers/pcsp/pcsp.c17
-rw-r--r--sound/pci/asihpi/hpioctl.c1
-rw-r--r--sound/pci/hda/hda_intel.c2
-rw-r--r--sound/soc/codecs/pcm1681.c2
-rw-r--r--sound/soc/codecs/rt5645.c3
-rw-r--r--sound/soc/codecs/sgtl5000.h2
-rw-r--r--sound/soc/codecs/ssm4567.c8
-rw-r--r--sound/soc/fsl/fsl_ssi.c2
-rw-r--r--sound/soc/intel/Makefile2
-rw-r--r--sound/soc/intel/atom/sst/sst_drv_interface.c14
-rw-r--r--sound/soc/intel/boards/cht_bsw_max98090_ti.c4
-rw-r--r--sound/soc/mediatek/mt8173-max98090.c17
-rw-r--r--sound/soc/mediatek/mt8173-rt5650-rt5676.c19
-rw-r--r--sound/soc/mediatek/mtk-afe-pcm.c2
-rw-r--r--sound/soc/omap/omap-hdmi-audio.c12
-rw-r--r--sound/soc/sh/migor.c3
-rw-r--r--sound/soc/soc-core.c1
-rw-r--r--sound/soc/soc-dapm.c35
-rw-r--r--sound/soc/soc-topology.c23
-rw-r--r--sound/soc/zte/zx296702-i2s.c4
-rw-r--r--sound/soc/zte/zx296702-spdif.c4
23 files changed, 106 insertions, 82 deletions
diff --git a/sound/core/hrtimer.c b/sound/core/hrtimer.c
index 886be7da989d..f845ecf7e172 100644
--- a/sound/core/hrtimer.c
+++ b/sound/core/hrtimer.c
@@ -121,16 +121,9 @@ static struct snd_timer *mytimer;
static int __init snd_hrtimer_init(void)
{
struct snd_timer *timer;
- struct timespec tp;
int err;
- hrtimer_get_res(CLOCK_MONOTONIC, &tp);
- if (tp.tv_sec > 0 || !tp.tv_nsec) {
- pr_err("snd-hrtimer: Invalid resolution %u.%09u",
- (unsigned)tp.tv_sec, (unsigned)tp.tv_nsec);
- return -EINVAL;
- }
- resolution = tp.tv_nsec;
+ resolution = hrtimer_resolution;
/* Create a new timer and set up the fields */
err = snd_timer_global_new("hrtimer", SNDRV_TIMER_GLOBAL_HRTIMER,
diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c
index 082509eb805d..f05cb6a8cbe0 100644
--- a/sound/core/memalloc.c
+++ b/sound/core/memalloc.c
@@ -124,7 +124,7 @@ static void snd_malloc_dev_iram(struct snd_dma_buffer *dmab, size_t size)
dmab->addr = 0;
if (dev->of_node)
- pool = of_get_named_gen_pool(dev->of_node, "iram", 0);
+ pool = of_gen_pool_get(dev->of_node, "iram", 0);
if (!pool)
return;
diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c
index d9647bd84d0f..27e25bb78c97 100644
--- a/sound/drivers/pcsp/pcsp.c
+++ b/sound/drivers/pcsp/pcsp.c
@@ -42,16 +42,13 @@ struct snd_pcsp pcsp_chip;
static int snd_pcsp_create(struct snd_card *card)
{
static struct snd_device_ops ops = { };
- struct timespec tp;
- int err;
- int div, min_div, order;
-
- hrtimer_get_res(CLOCK_MONOTONIC, &tp);
+ unsigned int resolution = hrtimer_resolution;
+ int err, div, min_div, order;
if (!nopcm) {
- if (tp.tv_sec || tp.tv_nsec > PCSP_MAX_PERIOD_NS) {
+ if (resolution > PCSP_MAX_PERIOD_NS) {
printk(KERN_ERR "PCSP: Timer resolution is not sufficient "
- "(%linS)\n", tp.tv_nsec);
+ "(%unS)\n", resolution);
printk(KERN_ERR "PCSP: Make sure you have HPET and ACPI "
"enabled.\n");
printk(KERN_ERR "PCSP: Turned into nopcm mode.\n");
@@ -59,13 +56,13 @@ static int snd_pcsp_create(struct snd_card *card)
}
}
- if (loops_per_jiffy >= PCSP_MIN_LPJ && tp.tv_nsec <= PCSP_MIN_PERIOD_NS)
+ if (loops_per_jiffy >= PCSP_MIN_LPJ && resolution <= PCSP_MIN_PERIOD_NS)
min_div = MIN_DIV;
else
min_div = MAX_DIV;
#if PCSP_DEBUG
- printk(KERN_DEBUG "PCSP: lpj=%li, min_div=%i, res=%li\n",
- loops_per_jiffy, min_div, tp.tv_nsec);
+ printk(KERN_DEBUG "PCSP: lpj=%li, min_div=%i, res=%u\n",
+ loops_per_jiffy, min_div, resolution);
#endif
div = MAX_DIV / min_div;
diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c
index 6610bd096fc9..d17937b92331 100644
--- a/sound/pci/asihpi/hpioctl.c
+++ b/sound/pci/asihpi/hpioctl.c
@@ -32,6 +32,7 @@
#include <linux/pci.h>
#include <linux/stringify.h>
#include <linux/module.h>
+#include <linux/vmalloc.h>
#ifdef MODULE_FIRMWARE
MODULE_FIRMWARE("asihpi/dsp5000.bin");
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index f581b12211e8..735bdcb04ce8 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -171,7 +171,7 @@ MODULE_PARM_DESC(beep_mode, "Select HDA Beep registration mode "
#ifdef CONFIG_PM
static int param_set_xint(const char *val, const struct kernel_param *kp);
-static struct kernel_param_ops param_ops_xint = {
+static const struct kernel_param_ops param_ops_xint = {
.set = param_set_xint,
.get = param_get_int,
};
diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c
index 477e13d30971..e7ba557979cb 100644
--- a/sound/soc/codecs/pcm1681.c
+++ b/sound/soc/codecs/pcm1681.c
@@ -102,7 +102,7 @@ static int pcm1681_set_deemph(struct snd_soc_codec *codec)
if (val != -1) {
regmap_update_bits(priv->regmap, PCM1681_DEEMPH_CONTROL,
- PCM1681_DEEMPH_RATE_MASK, val);
+ PCM1681_DEEMPH_RATE_MASK, val << 3);
enable = 1;
} else
enable = 0;
diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index 9ce311e088fc..e9cc3aae5366 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -2943,6 +2943,9 @@ static int rt5645_irq_detection(struct rt5645_priv *rt5645)
{
int val, btn_type, gpio_state = 0, report = 0;
+ if (!rt5645->codec)
+ return -EINVAL;
+
switch (rt5645->pdata.jd_mode) {
case 0: /* Not using rt5645 JD */
if (rt5645->gpiod_hp_det) {
diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h
index bd7a344bf8c5..1c317de26176 100644
--- a/sound/soc/codecs/sgtl5000.h
+++ b/sound/soc/codecs/sgtl5000.h
@@ -275,7 +275,7 @@
#define SGTL5000_BIAS_CTRL_MASK 0x000e
#define SGTL5000_BIAS_CTRL_SHIFT 1
#define SGTL5000_BIAS_CTRL_WIDTH 3
-#define SGTL5000_SMALL_POP 0
+#define SGTL5000_SMALL_POP 1
/*
* SGTL5000_CHIP_MIC_CTRL
diff --git a/sound/soc/codecs/ssm4567.c b/sound/soc/codecs/ssm4567.c
index 938d2cb6d78b..84a4f5ad8064 100644
--- a/sound/soc/codecs/ssm4567.c
+++ b/sound/soc/codecs/ssm4567.c
@@ -315,7 +315,13 @@ static int ssm4567_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
if (invert_fclk)
ctrl1 |= SSM4567_SAI_CTRL_1_FSYNC;
- return regmap_write(ssm4567->regmap, SSM4567_REG_SAI_CTRL_1, ctrl1);
+ return regmap_update_bits(ssm4567->regmap, SSM4567_REG_SAI_CTRL_1,
+ SSM4567_SAI_CTRL_1_BCLK |
+ SSM4567_SAI_CTRL_1_FSYNC |
+ SSM4567_SAI_CTRL_1_LJ |
+ SSM4567_SAI_CTRL_1_TDM |
+ SSM4567_SAI_CTRL_1_PDM,
+ ctrl1);
}
static int ssm4567_set_power(struct ssm4567 *ssm4567, bool enable)
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index c7647e066cfd..c0b940e2019f 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -633,7 +633,7 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream,
sub *= 100000;
do_div(sub, freq);
- if (sub < savesub) {
+ if (sub < savesub && !(i == 0 && psr == 0 && div2 == 0)) {
baudrate = tmprate;
savesub = sub;
pm = i;
diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile
index 3853ec2ddbc7..6de5d5cd3280 100644
--- a/sound/soc/intel/Makefile
+++ b/sound/soc/intel/Makefile
@@ -7,4 +7,4 @@ obj-$(CONFIG_SND_SOC_INTEL_BAYTRAIL) += baytrail/
obj-$(CONFIG_SND_SST_MFLD_PLATFORM) += atom/
# Machine support
-obj-$(CONFIG_SND_SOC_INTEL_SST) += boards/
+obj-$(CONFIG_SND_SOC) += boards/
diff --git a/sound/soc/intel/atom/sst/sst_drv_interface.c b/sound/soc/intel/atom/sst/sst_drv_interface.c
index 620da1d1b9e3..0e0e4d9c021f 100644
--- a/sound/soc/intel/atom/sst/sst_drv_interface.c
+++ b/sound/soc/intel/atom/sst/sst_drv_interface.c
@@ -42,6 +42,11 @@
#define MIN_FRAGMENT_SIZE (50 * 1024)
#define MAX_FRAGMENT_SIZE (1024 * 1024)
#define SST_GET_BYTES_PER_SAMPLE(pcm_wd_sz) (((pcm_wd_sz + 15) >> 4) << 1)
+#ifdef CONFIG_PM
+#define GET_USAGE_COUNT(dev) (atomic_read(&dev->power.usage_count))
+#else
+#define GET_USAGE_COUNT(dev) 1
+#endif
int free_stream_context(struct intel_sst_drv *ctx, unsigned int str_id)
{
@@ -141,15 +146,9 @@ static int sst_power_control(struct device *dev, bool state)
int ret = 0;
int usage_count = 0;
-#ifdef CONFIG_PM
- usage_count = atomic_read(&dev->power.usage_count);
-#else
- usage_count = 1;
-#endif
-
if (state == true) {
ret = pm_runtime_get_sync(dev);
-
+ usage_count = GET_USAGE_COUNT(dev);
dev_dbg(ctx->dev, "Enable: pm usage count: %d\n", usage_count);
if (ret < 0) {
dev_err(ctx->dev, "Runtime get failed with err: %d\n", ret);
@@ -164,6 +163,7 @@ static int sst_power_control(struct device *dev, bool state)
}
}
} else {
+ usage_count = GET_USAGE_COUNT(dev);
dev_dbg(ctx->dev, "Disable: pm usage count: %d\n", usage_count);
return sst_pm_runtime_put(ctx);
}
diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c
index d604ee80eda4..70f832114a5a 100644
--- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c
+++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c
@@ -69,12 +69,12 @@ static const struct snd_soc_dapm_route cht_audio_map[] = {
{"Headphone", NULL, "HPR"},
{"Ext Spk", NULL, "SPKL"},
{"Ext Spk", NULL, "SPKR"},
- {"AIF1 Playback", NULL, "ssp2 Tx"},
+ {"HiFi Playback", NULL, "ssp2 Tx"},
{"ssp2 Tx", NULL, "codec_out0"},
{"ssp2 Tx", NULL, "codec_out1"},
{"codec_in0", NULL, "ssp2 Rx" },
{"codec_in1", NULL, "ssp2 Rx" },
- {"ssp2 Rx", NULL, "AIF1 Capture"},
+ {"ssp2 Rx", NULL, "HiFi Capture"},
};
static const struct snd_kcontrol_new cht_mc_controls[] = {
diff --git a/sound/soc/mediatek/mt8173-max98090.c b/sound/soc/mediatek/mt8173-max98090.c
index 4d44b5803e55..2d2536af141f 100644
--- a/sound/soc/mediatek/mt8173-max98090.c
+++ b/sound/soc/mediatek/mt8173-max98090.c
@@ -103,7 +103,6 @@ static struct snd_soc_dai_link mt8173_max98090_dais[] = {
.name = "MAX98090 Playback",
.stream_name = "MAX98090 Playback",
.cpu_dai_name = "DL1",
- .platform_name = "11220000.mt8173-afe-pcm",
.codec_name = "snd-soc-dummy",
.codec_dai_name = "snd-soc-dummy-dai",
.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
@@ -114,7 +113,6 @@ static struct snd_soc_dai_link mt8173_max98090_dais[] = {
.name = "MAX98090 Capture",
.stream_name = "MAX98090 Capture",
.cpu_dai_name = "VUL",
- .platform_name = "11220000.mt8173-afe-pcm",
.codec_name = "snd-soc-dummy",
.codec_dai_name = "snd-soc-dummy-dai",
.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
@@ -125,7 +123,6 @@ static struct snd_soc_dai_link mt8173_max98090_dais[] = {
{
.name = "Codec",
.cpu_dai_name = "I2S",
- .platform_name = "11220000.mt8173-afe-pcm",
.no_pcm = 1,
.codec_dai_name = "HiFi",
.init = mt8173_max98090_init,
@@ -152,9 +149,21 @@ static struct snd_soc_card mt8173_max98090_card = {
static int mt8173_max98090_dev_probe(struct platform_device *pdev)
{
struct snd_soc_card *card = &mt8173_max98090_card;
- struct device_node *codec_node;
+ struct device_node *codec_node, *platform_node;
int ret, i;
+ platform_node = of_parse_phandle(pdev->dev.of_node,
+ "mediatek,platform", 0);
+ if (!platform_node) {
+ dev_err(&pdev->dev, "Property 'platform' missing or invalid\n");
+ return -EINVAL;
+ }
+ for (i = 0; i < card->num_links; i++) {
+ if (mt8173_max98090_dais[i].platform_name)
+ continue;
+ mt8173_max98090_dais[i].platform_of_node = platform_node;
+ }
+
codec_node = of_parse_phandle(pdev->dev.of_node,
"mediatek,audio-codec", 0);
if (!codec_node) {
diff --git a/sound/soc/mediatek/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173-rt5650-rt5676.c
index 094055323059..6f52eca05e26 100644
--- a/sound/soc/mediatek/mt8173-rt5650-rt5676.c
+++ b/sound/soc/mediatek/mt8173-rt5650-rt5676.c
@@ -138,7 +138,6 @@ static struct snd_soc_dai_link mt8173_rt5650_rt5676_dais[] = {
.name = "rt5650_rt5676 Playback",
.stream_name = "rt5650_rt5676 Playback",
.cpu_dai_name = "DL1",
- .platform_name = "11220000.mt8173-afe-pcm",
.codec_name = "snd-soc-dummy",
.codec_dai_name = "snd-soc-dummy-dai",
.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
@@ -149,7 +148,6 @@ static struct snd_soc_dai_link mt8173_rt5650_rt5676_dais[] = {
.name = "rt5650_rt5676 Capture",
.stream_name = "rt5650_rt5676 Capture",
.cpu_dai_name = "VUL",
- .platform_name = "11220000.mt8173-afe-pcm",
.codec_name = "snd-soc-dummy",
.codec_dai_name = "snd-soc-dummy-dai",
.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
@@ -161,7 +159,6 @@ static struct snd_soc_dai_link mt8173_rt5650_rt5676_dais[] = {
{
.name = "Codec",
.cpu_dai_name = "I2S",
- .platform_name = "11220000.mt8173-afe-pcm",
.no_pcm = 1,
.codecs = mt8173_rt5650_rt5676_codecs,
.num_codecs = 2,
@@ -209,7 +206,21 @@ static struct snd_soc_card mt8173_rt5650_rt5676_card = {
static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev)
{
struct snd_soc_card *card = &mt8173_rt5650_rt5676_card;
- int ret;
+ struct device_node *platform_node;
+ int i, ret;
+
+ platform_node = of_parse_phandle(pdev->dev.of_node,
+ "mediatek,platform", 0);
+ if (!platform_node) {
+ dev_err(&pdev->dev, "Property 'platform' missing or invalid\n");
+ return -EINVAL;
+ }
+
+ for (i = 0; i < card->num_links; i++) {
+ if (mt8173_rt5650_rt5676_dais[i].platform_name)
+ continue;
+ mt8173_rt5650_rt5676_dais[i].platform_of_node = platform_node;
+ }
mt8173_rt5650_rt5676_codecs[0].of_node =
of_parse_phandle(pdev->dev.of_node, "mediatek,audio-codec", 0);
diff --git a/sound/soc/mediatek/mtk-afe-pcm.c b/sound/soc/mediatek/mtk-afe-pcm.c
index cc228db5fb76..9863da73dfe0 100644
--- a/sound/soc/mediatek/mtk-afe-pcm.c
+++ b/sound/soc/mediatek/mtk-afe-pcm.c
@@ -1199,6 +1199,8 @@ err_pm_disable:
static int mtk_afe_pcm_dev_remove(struct platform_device *pdev)
{
pm_runtime_disable(&pdev->dev);
+ if (!pm_runtime_status_suspended(&pdev->dev))
+ mtk_afe_runtime_suspend(&pdev->dev);
snd_soc_unregister_component(&pdev->dev);
snd_soc_unregister_platform(&pdev->dev);
return 0;
diff --git a/sound/soc/omap/omap-hdmi-audio.c b/sound/soc/omap/omap-hdmi-audio.c
index 4775da4c4db5..aeef25c0cb3d 100644
--- a/sound/soc/omap/omap-hdmi-audio.c
+++ b/sound/soc/omap/omap-hdmi-audio.c
@@ -210,16 +210,18 @@ static int hdmi_dai_hw_params(struct snd_pcm_substream *substream,
cea->db3 = 0; /* not used, all zeros */
- /*
- * The OMAP HDMI IP requires to use the 8-channel channel code when
- * transmitting more than two channels.
- */
if (params_channels(params) == 2)
cea->db4_ca = 0x0;
+ else if (params_channels(params) == 6)
+ cea->db4_ca = 0xb;
else
cea->db4_ca = 0x13;
- cea->db5_dminh_lsv = CEA861_AUDIO_INFOFRAME_DB5_DM_INH_PROHIBITED;
+ if (cea->db4_ca == 0x00)
+ cea->db5_dminh_lsv = CEA861_AUDIO_INFOFRAME_DB5_DM_INH_PERMITTED;
+ else
+ cea->db5_dminh_lsv = CEA861_AUDIO_INFOFRAME_DB5_DM_INH_PROHIBITED;
+
/* the expression is trivial but makes clear what we are doing */
cea->db5_dminh_lsv |= (0 & CEA861_AUDIO_INFOFRAME_DB5_LSV);
diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c
index 82f582344fe7..672bcd4c252b 100644
--- a/sound/soc/sh/migor.c
+++ b/sound/soc/sh/migor.c
@@ -162,12 +162,11 @@ static int __init migor_init(void)
if (ret < 0)
return ret;
- siumckb_lookup = clkdev_alloc(&siumckb_clk, "siumckb_clk", NULL);
+ siumckb_lookup = clkdev_create(&siumckb_clk, "siumckb_clk", NULL);
if (!siumckb_lookup) {
ret = -ENOMEM;
goto eclkdevalloc;
}
- clkdev_add(siumckb_lookup);
/* Port number used on this machine: port B */
migor_snd_device = platform_device_alloc("soc-audio", 1);
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 3a4a5c0e3f97..0e1e69c7abd5 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1716,6 +1716,7 @@ card_probe_error:
if (card->remove)
card->remove(card);
+ snd_soc_dapm_free(&card->dapm);
soc_cleanup_card_debugfs(card);
snd_card_free(card->snd_card);
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index aa327c92480c..e0de8072c514 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -358,9 +358,10 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget,
data->widget =
snd_soc_dapm_new_control_unlocked(widget->dapm,
&template);
+ kfree(name);
if (!data->widget) {
ret = -ENOMEM;
- goto err_name;
+ goto err_data;
}
}
break;
@@ -389,11 +390,12 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget,
data->value = template.on_val;
- data->widget = snd_soc_dapm_new_control(widget->dapm,
- &template);
+ data->widget = snd_soc_dapm_new_control_unlocked(
+ widget->dapm, &template);
+ kfree(name);
if (!data->widget) {
ret = -ENOMEM;
- goto err_name;
+ goto err_data;
}
snd_soc_dapm_add_path(widget->dapm, data->widget,
@@ -408,8 +410,6 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget,
return 0;
-err_name:
- kfree(name);
err_data:
kfree(data);
return ret;
@@ -418,8 +418,6 @@ err_data:
static void dapm_kcontrol_free(struct snd_kcontrol *kctl)
{
struct dapm_kcontrol_data *data = snd_kcontrol_chip(kctl);
- if (data->widget)
- kfree(data->widget->name);
kfree(data->wlist);
kfree(data);
}
@@ -1952,6 +1950,7 @@ static ssize_t dapm_widget_power_read_file(struct file *file,
size_t count, loff_t *ppos)
{
struct snd_soc_dapm_widget *w = file->private_data;
+ struct snd_soc_card *card = w->dapm->card;
char *buf;
int in, out;
ssize_t ret;
@@ -1961,6 +1960,8 @@ static ssize_t dapm_widget_power_read_file(struct file *file,
if (!buf)
return -ENOMEM;
+ mutex_lock(&card->dapm_mutex);
+
/* Supply widgets are not handled by is_connected_{input,output}_ep() */
if (w->is_supply) {
in = 0;
@@ -2007,6 +2008,8 @@ static ssize_t dapm_widget_power_read_file(struct file *file,
p->sink->name);
}
+ mutex_unlock(&card->dapm_mutex);
+
ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret);
kfree(buf);
@@ -2281,11 +2284,15 @@ static ssize_t dapm_widget_show(struct device *dev,
struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev);
int i, count = 0;
+ mutex_lock(&rtd->card->dapm_mutex);
+
for (i = 0; i < rtd->num_codecs; i++) {
struct snd_soc_codec *codec = rtd->codec_dais[i]->codec;
count += dapm_widget_show_codec(codec, buf + count);
}
+ mutex_unlock(&rtd->card->dapm_mutex);
+
return count;
}
@@ -3334,16 +3341,10 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm,
}
prefix = soc_dapm_prefix(dapm);
- if (prefix) {
+ if (prefix)
w->name = kasprintf(GFP_KERNEL, "%s %s", prefix, widget->name);
- if (widget->sname)
- w->sname = kasprintf(GFP_KERNEL, "%s %s", prefix,
- widget->sname);
- } else {
+ else
w->name = kasprintf(GFP_KERNEL, "%s", widget->name);
- if (widget->sname)
- w->sname = kasprintf(GFP_KERNEL, "%s", widget->sname);
- }
if (w->name == NULL) {
kfree(w);
return NULL;
@@ -3792,7 +3793,7 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card)
break;
}
- if (!w->sname || !strstr(w->sname, dai_w->name))
+ if (!w->sname || !strstr(w->sname, dai_w->sname))
continue;
if (dai_w->id == snd_soc_dapm_dai_in) {
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index d0960683c409..59ac211f8fe7 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -144,7 +144,7 @@ static const struct snd_soc_tplg_kcontrol_ops io_ops[] = {
{SND_SOC_TPLG_CTL_STROBE, snd_soc_get_strobe,
snd_soc_put_strobe, NULL},
{SND_SOC_TPLG_DAPM_CTL_VOLSW, snd_soc_dapm_get_volsw,
- snd_soc_dapm_put_volsw, NULL},
+ snd_soc_dapm_put_volsw, snd_soc_info_volsw},
{SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE, snd_soc_dapm_get_enum_double,
snd_soc_dapm_put_enum_double, snd_soc_info_enum_double},
{SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT, snd_soc_dapm_get_enum_double,
@@ -580,27 +580,26 @@ static int soc_tplg_init_kcontrol(struct soc_tplg *tplg,
}
static int soc_tplg_create_tlv(struct soc_tplg *tplg,
- struct snd_kcontrol_new *kc, u32 tlv_size)
+ struct snd_kcontrol_new *kc, struct snd_soc_tplg_ctl_tlv *tplg_tlv)
{
- struct snd_soc_tplg_ctl_tlv *tplg_tlv;
struct snd_ctl_tlv *tlv;
+ int size;
- if (tlv_size == 0)
+ if (tplg_tlv->count == 0)
return 0;
- tplg_tlv = (struct snd_soc_tplg_ctl_tlv *) tplg->pos;
- tplg->pos += tlv_size;
-
- tlv = kzalloc(sizeof(*tlv) + tlv_size, GFP_KERNEL);
+ size = ((tplg_tlv->count + (sizeof(unsigned int) - 1)) &
+ ~(sizeof(unsigned int) - 1));
+ tlv = kzalloc(sizeof(*tlv) + size, GFP_KERNEL);
if (tlv == NULL)
return -ENOMEM;
dev_dbg(tplg->dev, " created TLV type %d size %d bytes\n",
- tplg_tlv->numid, tplg_tlv->size);
+ tplg_tlv->numid, size);
tlv->numid = tplg_tlv->numid;
- tlv->length = tplg_tlv->size;
- memcpy(tlv->tlv, tplg_tlv + 1, tplg_tlv->size);
+ tlv->length = size;
+ memcpy(&tlv->tlv[0], tplg_tlv->data, size);
kc->tlv.p = (void *)tlv;
return 0;
@@ -773,7 +772,7 @@ static int soc_tplg_dmixer_create(struct soc_tplg *tplg, unsigned int count,
}
/* create any TLV data */
- soc_tplg_create_tlv(tplg, &kc, mc->hdr.tlv_size);
+ soc_tplg_create_tlv(tplg, &kc, &mc->tlv);
/* register control here */
err = soc_tplg_add_kcontrol(tplg, &kc,
diff --git a/sound/soc/zte/zx296702-i2s.c b/sound/soc/zte/zx296702-i2s.c
index 98d96e1b17e0..1930c42e1f55 100644
--- a/sound/soc/zte/zx296702-i2s.c
+++ b/sound/soc/zte/zx296702-i2s.c
@@ -393,9 +393,9 @@ static int zx_i2s_probe(struct platform_device *pdev)
res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
zx_i2s->mapbase = res->start;
zx_i2s->reg_base = devm_ioremap_resource(&pdev->dev, res);
- if (!zx_i2s->reg_base) {
+ if (IS_ERR(zx_i2s->reg_base)) {
dev_err(&pdev->dev, "ioremap failed!\n");
- return -EIO;
+ return PTR_ERR(zx_i2s->reg_base);
}
writel_relaxed(0, zx_i2s->reg_base + ZX_I2S_FIFO_CTRL);
diff --git a/sound/soc/zte/zx296702-spdif.c b/sound/soc/zte/zx296702-spdif.c
index 11a0e46a1156..26265ce4caca 100644
--- a/sound/soc/zte/zx296702-spdif.c
+++ b/sound/soc/zte/zx296702-spdif.c
@@ -322,9 +322,9 @@ static int zx_spdif_probe(struct platform_device *pdev)
res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
zx_spdif->mapbase = res->start;
zx_spdif->reg_base = devm_ioremap_resource(&pdev->dev, res);
- if (!zx_spdif->reg_base) {
+ if (IS_ERR(zx_spdif->reg_base)) {
dev_err(&pdev->dev, "ioremap failed!\n");
- return -EIO;
+ return PTR_ERR(zx_spdif->reg_base);
}
zx_spdif_dev_init(zx_spdif->reg_base);