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-rw-r--r--sound/soc/codecs/arizona.c6
-rw-r--r--sound/soc/codecs/cs4265.c12
-rw-r--r--sound/soc/codecs/da732x.h2
-rw-r--r--sound/soc/codecs/pcm512x.c4
-rw-r--r--sound/soc/codecs/rt5640.c1
-rw-r--r--sound/soc/davinci/davinci-mcasp.c14
-rw-r--r--sound/soc/fsl/Kconfig1
-rw-r--r--sound/soc/fsl/fsl_esai.c2
-rw-r--r--sound/soc/generic/simple-card.c8
-rw-r--r--sound/soc/intel/sst-acpi.c4
-rw-r--r--sound/soc/intel/sst-baytrail-ipc.c10
-rw-r--r--sound/soc/intel/sst-baytrail-ipc.h1
-rw-r--r--sound/soc/intel/sst-baytrail-pcm.c43
-rw-r--r--sound/soc/omap/omap-twl4030.c2
-rw-r--r--sound/soc/pxa/pxa-ssp.c4
-rw-r--r--sound/soc/sh/rcar/gen.c2
-rw-r--r--sound/soc/soc-core.c2
-rw-r--r--sound/soc/soc-dapm.c12
-rw-r--r--sound/soc/tegra/tegra_asoc_utils.h2
19 files changed, 63 insertions, 69 deletions
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index bd41ee4da078..2c71f16bd661 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -1278,6 +1278,8 @@ static int arizona_hw_params(struct snd_pcm_substream *substream,
else
rates = &arizona_48k_bclk_rates[0];
+ wl = snd_pcm_format_width(params_format(params));
+
if (tdm_slots) {
arizona_aif_dbg(dai, "Configuring for %d %d bit TDM slots\n",
tdm_slots, tdm_width);
@@ -1285,6 +1287,7 @@ static int arizona_hw_params(struct snd_pcm_substream *substream,
channels = tdm_slots;
} else {
bclk_target = snd_soc_params_to_bclk(params);
+ tdm_width = wl;
}
if (chan_limit && chan_limit < channels) {
@@ -1319,8 +1322,7 @@ static int arizona_hw_params(struct snd_pcm_substream *substream,
arizona_aif_dbg(dai, "BCLK %dHz LRCLK %dHz\n",
rates[bclk], rates[bclk] / lrclk);
- wl = snd_pcm_format_width(params_format(params));
- frame = wl << ARIZONA_AIF1TX_WL_SHIFT | wl;
+ frame = wl << ARIZONA_AIF1TX_WL_SHIFT | tdm_width;
reconfig = arizona_aif_cfg_changed(codec, base, bclk, lrclk, frame);
diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c
index a20b30ca52c0..98523209f739 100644
--- a/sound/soc/codecs/cs4265.c
+++ b/sound/soc/codecs/cs4265.c
@@ -282,10 +282,10 @@ static const struct cs4265_clk_para clk_map_table[] = {
/*64k*/
{8192000, 64000, 1, 0},
- {1228800, 64000, 1, 1},
- {1693440, 64000, 1, 2},
- {2457600, 64000, 1, 3},
- {3276800, 64000, 1, 4},
+ {12288000, 64000, 1, 1},
+ {16934400, 64000, 1, 2},
+ {24576000, 64000, 1, 3},
+ {32768000, 64000, 1, 4},
/* 88.2k */
{11289600, 88200, 1, 0},
@@ -435,10 +435,10 @@ static int cs4265_pcm_hw_params(struct snd_pcm_substream *substream,
index = cs4265_get_clk_index(cs4265->sysclk, params_rate(params));
if (index >= 0) {
snd_soc_update_bits(codec, CS4265_ADC_CTL,
- CS4265_ADC_FM, clk_map_table[index].fm_mode);
+ CS4265_ADC_FM, clk_map_table[index].fm_mode << 6);
snd_soc_update_bits(codec, CS4265_MCLK_FREQ,
CS4265_MCLK_FREQ_MASK,
- clk_map_table[index].mclkdiv);
+ clk_map_table[index].mclkdiv << 4);
} else {
dev_err(codec->dev, "can't get correct mclk\n");
diff --git a/sound/soc/codecs/da732x.h b/sound/soc/codecs/da732x.h
index 1dceafeec415..f586cbd30b77 100644
--- a/sound/soc/codecs/da732x.h
+++ b/sound/soc/codecs/da732x.h
@@ -11,7 +11,7 @@
*/
#ifndef __DA732X_H_
-#define __DA732X_H
+#define __DA732X_H_
#include <sound/soc.h>
diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c
index 163ec3855fd4..0c8aefab404c 100644
--- a/sound/soc/codecs/pcm512x.c
+++ b/sound/soc/codecs/pcm512x.c
@@ -259,13 +259,13 @@ static const struct soc_enum pcm512x_veds =
pcm512x_ramp_step_text);
static const struct snd_kcontrol_new pcm512x_controls[] = {
-SOC_DOUBLE_R_TLV("Playback Digital Volume", PCM512x_DIGITAL_VOLUME_2,
+SOC_DOUBLE_R_TLV("Digital Playback Volume", PCM512x_DIGITAL_VOLUME_2,
PCM512x_DIGITAL_VOLUME_3, 0, 255, 1, digital_tlv),
SOC_DOUBLE_TLV("Playback Volume", PCM512x_ANALOG_GAIN_CTRL,
PCM512x_LAGN_SHIFT, PCM512x_RAGN_SHIFT, 1, 1, analog_tlv),
SOC_DOUBLE_TLV("Playback Boost Volume", PCM512x_ANALOG_GAIN_BOOST,
PCM512x_AGBL_SHIFT, PCM512x_AGBR_SHIFT, 1, 0, boost_tlv),
-SOC_DOUBLE("Playback Digital Switch", PCM512x_MUTE, PCM512x_RQML_SHIFT,
+SOC_DOUBLE("Digital Playback Switch", PCM512x_MUTE, PCM512x_RQML_SHIFT,
PCM512x_RQMR_SHIFT, 1, 1),
SOC_SINGLE("Deemphasis Switch", PCM512x_DSP, PCM512x_DEMP_SHIFT, 1, 1),
diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index 6bc6efdec550..f1ec6e6bd08a 100644
--- a/sound/soc/codecs/rt5640.c
+++ b/sound/soc/codecs/rt5640.c
@@ -2059,6 +2059,7 @@ static struct snd_soc_codec_driver soc_codec_dev_rt5640 = {
static const struct regmap_config rt5640_regmap = {
.reg_bits = 8,
.val_bits = 16,
+ .use_single_rw = true,
.max_register = RT5640_VENDOR_ID2 + 1 + (ARRAY_SIZE(rt5640_ranges) *
RT5640_PR_SPACING),
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index c28508da34cf..6a6b2ff7d7d7 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -403,7 +403,8 @@ out:
return ret;
}
-static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div)
+static int __davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id,
+ int div, bool explicit)
{
struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai);
@@ -420,7 +421,8 @@ static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div
ACLKXDIV(div - 1), ACLKXDIV_MASK);
mcasp_mod_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG,
ACLKRDIV(div - 1), ACLKRDIV_MASK);
- mcasp->bclk_div = div;
+ if (explicit)
+ mcasp->bclk_div = div;
break;
case 2: /* BCLK/LRCLK ratio */
@@ -434,6 +436,12 @@ static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div
return 0;
}
+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);
+}
+
static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id,
unsigned int freq, int dir)
{
@@ -738,7 +746,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
"Inaccurate BCLK: %u Hz / %u != %u Hz\n",
mcasp->sysclk_freq, div, bclk_freq);
}
- davinci_mcasp_set_clkdiv(cpu_dai, 1, div);
+ __davinci_mcasp_set_clkdiv(cpu_dai, 1, div, 0);
}
ret = mcasp_common_hw_param(mcasp, substream->stream,
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index f54a8fc99291..f3012b645b51 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -49,7 +49,6 @@ config SND_SOC_FSL_ESAI
tristate "Enhanced Serial Audio Interface (ESAI) module support"
select REGMAP_MMIO
select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n
- select SND_SOC_FSL_UTILS
help
Say Y if you want to add Enhanced Synchronous Audio Interface
(ESAI) support for the Freescale CPUs.
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index 72d154e7dd03..a3b29ed84963 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -18,7 +18,6 @@
#include "fsl_esai.h"
#include "imx-pcm.h"
-#include "fsl_utils.h"
#define FSL_ESAI_RATES SNDRV_PCM_RATE_8000_192000
#define FSL_ESAI_FORMATS (SNDRV_PCM_FMTBIT_S8 | \
@@ -607,7 +606,6 @@ static struct snd_soc_dai_ops fsl_esai_dai_ops = {
.hw_params = fsl_esai_hw_params,
.set_sysclk = fsl_esai_set_dai_sysclk,
.set_fmt = fsl_esai_set_dai_fmt,
- .xlate_tdm_slot_mask = fsl_asoc_xlate_tdm_slot_mask,
.set_tdm_slot = fsl_esai_set_dai_tdm_slot,
};
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 159e517fa09a..cef7776b712c 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -481,12 +481,19 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
snd_soc_card_set_drvdata(&priv->snd_card, priv);
ret = devm_snd_soc_register_card(&pdev->dev, &priv->snd_card);
+ if (ret >= 0)
+ return ret;
err:
asoc_simple_card_unref(pdev);
return ret;
}
+static int asoc_simple_card_remove(struct platform_device *pdev)
+{
+ return asoc_simple_card_unref(pdev);
+}
+
static const struct of_device_id asoc_simple_of_match[] = {
{ .compatible = "simple-audio-card", },
{},
@@ -500,6 +507,7 @@ static struct platform_driver asoc_simple_card = {
.of_match_table = asoc_simple_of_match,
},
.probe = asoc_simple_card_probe,
+ .remove = asoc_simple_card_remove,
};
module_platform_driver(asoc_simple_card);
diff --git a/sound/soc/intel/sst-acpi.c b/sound/soc/intel/sst-acpi.c
index 42edc6f4fc4a..03d0a166b635 100644
--- a/sound/soc/intel/sst-acpi.c
+++ b/sound/soc/intel/sst-acpi.c
@@ -246,8 +246,8 @@ static struct sst_acpi_desc sst_acpi_broadwell_desc = {
};
static struct sst_acpi_mach baytrail_machines[] = {
- { "10EC5640", "byt-rt5640", "intel/fw_sst_0f28.bin-i2s_master" },
- { "193C9890", "byt-max98090", "intel/fw_sst_0f28.bin-i2s_master" },
+ { "10EC5640", "byt-rt5640", "intel/fw_sst_0f28.bin-48kHz_i2s_master" },
+ { "193C9890", "byt-max98090", "intel/fw_sst_0f28.bin-48kHz_i2s_master" },
{}
};
diff --git a/sound/soc/intel/sst-baytrail-ipc.c b/sound/soc/intel/sst-baytrail-ipc.c
index 67673a2c0f41..b4ad98c43e5c 100644
--- a/sound/soc/intel/sst-baytrail-ipc.c
+++ b/sound/soc/intel/sst-baytrail-ipc.c
@@ -817,7 +817,7 @@ static struct sst_dsp_device byt_dev = {
.ops = &sst_byt_ops,
};
-int sst_byt_dsp_suspend_noirq(struct device *dev, struct sst_pdata *pdata)
+int sst_byt_dsp_suspend_late(struct device *dev, struct sst_pdata *pdata)
{
struct sst_byt *byt = pdata->dsp;
@@ -826,14 +826,6 @@ int sst_byt_dsp_suspend_noirq(struct device *dev, struct sst_pdata *pdata)
sst_byt_drop_all(byt);
dev_dbg(byt->dev, "dsp in reset\n");
- return 0;
-}
-EXPORT_SYMBOL_GPL(sst_byt_dsp_suspend_noirq);
-
-int sst_byt_dsp_suspend_late(struct device *dev, struct sst_pdata *pdata)
-{
- struct sst_byt *byt = pdata->dsp;
-
dev_dbg(byt->dev, "free all blocks and unload fw\n");
sst_fw_unload(byt->fw);
diff --git a/sound/soc/intel/sst-baytrail-ipc.h b/sound/soc/intel/sst-baytrail-ipc.h
index 06a4d202689b..8faff6dcf25d 100644
--- a/sound/soc/intel/sst-baytrail-ipc.h
+++ b/sound/soc/intel/sst-baytrail-ipc.h
@@ -66,7 +66,6 @@ int sst_byt_get_dsp_position(struct sst_byt *byt,
int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata);
void sst_byt_dsp_free(struct device *dev, struct sst_pdata *pdata);
struct sst_dsp *sst_byt_get_dsp(struct sst_byt *byt);
-int sst_byt_dsp_suspend_noirq(struct device *dev, struct sst_pdata *pdata);
int sst_byt_dsp_suspend_late(struct device *dev, struct sst_pdata *pdata);
int sst_byt_dsp_boot(struct device *dev, struct sst_pdata *pdata);
int sst_byt_dsp_wait_for_ready(struct device *dev, struct sst_pdata *pdata);
diff --git a/sound/soc/intel/sst-baytrail-pcm.c b/sound/soc/intel/sst-baytrail-pcm.c
index 599401c0c655..eab1c7d85187 100644
--- a/sound/soc/intel/sst-baytrail-pcm.c
+++ b/sound/soc/intel/sst-baytrail-pcm.c
@@ -59,6 +59,9 @@ struct sst_byt_priv_data {
/* DAI data */
struct sst_byt_pcm_data pcm[BYT_PCM_COUNT];
+
+ /* flag indicating is stream context restore needed after suspend */
+ bool restore_stream;
};
/* this may get called several times by oss emulation */
@@ -184,7 +187,10 @@ static int sst_byt_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
sst_byt_stream_start(byt, pcm_data->stream, 0);
break;
case SNDRV_PCM_TRIGGER_RESUME:
- schedule_work(&pcm_data->work);
+ if (pdata->restore_stream == true)
+ schedule_work(&pcm_data->work);
+ else
+ sst_byt_stream_resume(byt, pcm_data->stream);
break;
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
sst_byt_stream_resume(byt, pcm_data->stream);
@@ -193,6 +199,7 @@ static int sst_byt_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
sst_byt_stream_stop(byt, pcm_data->stream);
break;
case SNDRV_PCM_TRIGGER_SUSPEND:
+ pdata->restore_stream = false;
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
sst_byt_stream_pause(byt, pcm_data->stream);
break;
@@ -404,26 +411,10 @@ static const struct snd_soc_component_driver byt_dai_component = {
};
#ifdef CONFIG_PM
-static int sst_byt_pcm_dev_suspend_noirq(struct device *dev)
-{
- struct sst_pdata *sst_pdata = dev_get_platdata(dev);
- int ret;
-
- dev_dbg(dev, "suspending noirq\n");
-
- /* at this point all streams will be stopped and context saved */
- ret = sst_byt_dsp_suspend_noirq(dev, sst_pdata);
- if (ret < 0) {
- dev_err(dev, "failed to suspend %d\n", ret);
- return ret;
- }
-
- return ret;
-}
-
static int sst_byt_pcm_dev_suspend_late(struct device *dev)
{
struct sst_pdata *sst_pdata = dev_get_platdata(dev);
+ struct sst_byt_priv_data *priv_data = dev_get_drvdata(dev);
int ret;
dev_dbg(dev, "suspending late\n");
@@ -434,34 +425,30 @@ static int sst_byt_pcm_dev_suspend_late(struct device *dev)
return ret;
}
+ priv_data->restore_stream = true;
+
return ret;
}
static int sst_byt_pcm_dev_resume_early(struct device *dev)
{
struct sst_pdata *sst_pdata = dev_get_platdata(dev);
+ int ret;
dev_dbg(dev, "resume early\n");
/* load fw and boot DSP */
- return sst_byt_dsp_boot(dev, sst_pdata);
-}
-
-static int sst_byt_pcm_dev_resume(struct device *dev)
-{
- struct sst_pdata *sst_pdata = dev_get_platdata(dev);
-
- dev_dbg(dev, "resume\n");
+ ret = sst_byt_dsp_boot(dev, sst_pdata);
+ if (ret)
+ return ret;
/* wait for FW to finish booting */
return sst_byt_dsp_wait_for_ready(dev, sst_pdata);
}
static const struct dev_pm_ops sst_byt_pm_ops = {
- .suspend_noirq = sst_byt_pcm_dev_suspend_noirq,
.suspend_late = sst_byt_pcm_dev_suspend_late,
.resume_early = sst_byt_pcm_dev_resume_early,
- .resume = sst_byt_pcm_dev_resume,
};
#define SST_BYT_PM_OPS (&sst_byt_pm_ops)
diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c
index f8a6adc2d81c..4336d1831485 100644
--- a/sound/soc/omap/omap-twl4030.c
+++ b/sound/soc/omap/omap-twl4030.c
@@ -260,7 +260,7 @@ static struct snd_soc_dai_link omap_twl4030_dai_links[] = {
.stream_name = "TWL4030 Voice",
.cpu_dai_name = "omap-mcbsp.3",
.codec_dai_name = "twl4030-voice",
- .platform_name = "omap-mcbsp.2",
+ .platform_name = "omap-mcbsp.3",
.codec_name = "twl4030-codec",
.dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF |
SND_SOC_DAIFMT_CBM_CFM,
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 0109f6c2334e..a8e097433074 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -765,9 +765,7 @@ static int pxa_ssp_remove(struct snd_soc_dai *dai)
SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 | \
SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
-#define PXA_SSP_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\
- SNDRV_PCM_FMTBIT_S24_LE | \
- SNDRV_PCM_FMTBIT_S32_LE)
+#define PXA_SSP_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE)
static const struct snd_soc_dai_ops pxa_ssp_dai_ops = {
.startup = pxa_ssp_startup,
diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c
index 3fdf3be7b99a..f95e7ab135e8 100644
--- a/sound/soc/sh/rcar/gen.c
+++ b/sound/soc/sh/rcar/gen.c
@@ -247,7 +247,7 @@ rsnd_gen2_dma_addr(struct rsnd_priv *priv,
};
/* it shouldn't happen */
- if (use_dvc & !use_src)
+ if (use_dvc && !use_src)
dev_err(dev, "DVC is selected without SRC\n");
/* use SSIU or SSI ? */
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index d4bfd4a9076f..889f4e3d35dc 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1325,7 +1325,7 @@ static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd,
device_initialize(rtd->dev);
rtd->dev->parent = rtd->card->dev;
rtd->dev->release = rtd_release;
- rtd->dev->init_name = name;
+ dev_set_name(rtd->dev, "%s", name);
dev_set_drvdata(rtd->dev, rtd);
mutex_init(&rtd->pcm_mutex);
INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients);
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 8348352dc2c6..177bd8639ef9 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -2860,12 +2860,14 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol,
struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol);
struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
unsigned int reg_val, val;
- int ret = 0;
- if (e->reg != SND_SOC_NOPM)
- ret = soc_dapm_read(dapm, e->reg, &reg_val);
- else
+ if (e->reg != SND_SOC_NOPM) {
+ int ret = soc_dapm_read(dapm, e->reg, &reg_val);
+ if (ret)
+ return ret;
+ } else {
reg_val = dapm_kcontrol_get_value(kcontrol);
+ }
val = (reg_val >> e->shift_l) & e->mask;
ucontrol->value.enumerated.item[0] = snd_soc_enum_val_to_item(e, val);
@@ -2875,7 +2877,7 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol,
ucontrol->value.enumerated.item[1] = val;
}
- return ret;
+ return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_double);
diff --git a/sound/soc/tegra/tegra_asoc_utils.h b/sound/soc/tegra/tegra_asoc_utils.h
index 9577121ce971..ca8037634100 100644
--- a/sound/soc/tegra/tegra_asoc_utils.h
+++ b/sound/soc/tegra/tegra_asoc_utils.h
@@ -21,7 +21,7 @@
*/
#ifndef __TEGRA_ASOC_UTILS_H__
-#define __TEGRA_ASOC_UTILS_H_
+#define __TEGRA_ASOC_UTILS_H__
struct clk;
struct device;
generated by cgit v1.2.3 (git 2.25.1) at 2024-04-27 17:37:17 +0000