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authorTakashi Iwai <tiwai@suse.de>2020-11-05 18:19:32 +0100
committerTakashi Iwai <tiwai@suse.de>2020-11-05 18:19:32 +0100
commita6c96672a64f4f0e1bac9f37b5bb57d8ab551b4b (patch)
tree5c10278fcab319140d55b0e6faa726667c7f1c35 /sound
parent0938ecae432e7ac8b01080c35dd81d50a1e43033 (diff)
parentf9d7c6eb23f7e55e7a0ca5451da06909bdfdd0e4 (diff)
downloadlwn-a6c96672a64f4f0e1bac9f37b5bb57d8ab551b4b.tar.gz
lwn-a6c96672a64f4f0e1bac9f37b5bb57d8ab551b4b.zip
Merge tag 'asoc-fix-v5.10-rc2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.10 A batch of driver specific fixes that have come up since the merge window, nothing particularly major here but all good to have.
Diffstat (limited to 'sound')
-rw-r--r--sound/soc/atmel/mchp-spdiftx.c1
-rw-r--r--sound/soc/codecs/cs42l51.c22
-rw-r--r--sound/soc/codecs/wcd9335.c2
-rw-r--r--sound/soc/codecs/wcd934x.c2
-rw-r--r--sound/soc/codecs/wsa881x.c2
-rw-r--r--sound/soc/intel/Kconfig18
-rw-r--r--sound/soc/intel/atom/Makefile2
-rw-r--r--sound/soc/intel/atom/sst/Makefile6
-rw-r--r--sound/soc/intel/boards/kbl_rt5663_max98927.c39
-rw-r--r--sound/soc/intel/catpt/dsp.c9
-rw-r--r--sound/soc/intel/catpt/pcm.c10
-rw-r--r--sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c31
-rw-r--r--sound/soc/qcom/lpass-cpu.c14
-rw-r--r--sound/soc/qcom/lpass-sc7180.c2
-rw-r--r--sound/soc/qcom/sdm845.c2
-rw-r--r--sound/soc/sof/loader.c5
16 files changed, 119 insertions, 48 deletions
diff --git a/sound/soc/atmel/mchp-spdiftx.c b/sound/soc/atmel/mchp-spdiftx.c
index 82c1eecd2528..3bd350afb743 100644
--- a/sound/soc/atmel/mchp-spdiftx.c
+++ b/sound/soc/atmel/mchp-spdiftx.c
@@ -487,7 +487,6 @@ static int mchp_spdiftx_hw_params(struct snd_pcm_substream *substream,
}
mchp_spdiftx_channel_status_write(dev);
spin_unlock_irqrestore(&ctrl->lock, flags);
- mr |= SPDIFTX_MR_VALID1 | SPDIFTX_MR_VALID2;
if (dev->gclk_enabled) {
clk_disable_unprepare(dev->gclk);
diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c
index 097c4e8d9950..c61b17dc2af8 100644
--- a/sound/soc/codecs/cs42l51.c
+++ b/sound/soc/codecs/cs42l51.c
@@ -254,8 +254,28 @@ static const struct snd_soc_dapm_widget cs42l51_dapm_widgets[] = {
&cs42l51_adcr_mux_controls),
};
+static int mclk_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm);
+ struct cs42l51_private *cs42l51 = snd_soc_component_get_drvdata(comp);
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ return clk_prepare_enable(cs42l51->mclk_handle);
+ case SND_SOC_DAPM_POST_PMD:
+ /* Delay mclk shutdown to fulfill power-down sequence requirements */
+ msleep(20);
+ clk_disable_unprepare(cs42l51->mclk_handle);
+ break;
+ }
+
+ return 0;
+}
+
static const struct snd_soc_dapm_widget cs42l51_dapm_mclk_widgets[] = {
- SND_SOC_DAPM_CLOCK_SUPPLY("MCLK")
+ SND_SOC_DAPM_SUPPLY("MCLK", SND_SOC_NOPM, 0, 0, mclk_event,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
};
static const struct snd_soc_dapm_route cs42l51_routes[] = {
diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c
index f2d9d52ee171..4d2b1ec7c03b 100644
--- a/sound/soc/codecs/wcd9335.c
+++ b/sound/soc/codecs/wcd9335.c
@@ -618,7 +618,7 @@ static const char * const sb_tx8_mux_text[] = {
"ZERO", "RX_MIX_TX8", "DEC8", "DEC8_192"
};
-static const DECLARE_TLV_DB_SCALE(digital_gain, 0, 1, 0);
+static const DECLARE_TLV_DB_SCALE(digital_gain, -8400, 100, -8400);
static const DECLARE_TLV_DB_SCALE(line_gain, 0, 7, 1);
static const DECLARE_TLV_DB_SCALE(analog_gain, 0, 25, 1);
static const DECLARE_TLV_DB_SCALE(ear_pa_gain, 0, 150, 0);
diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c
index 35697b072367..40f682f5dab8 100644
--- a/sound/soc/codecs/wcd934x.c
+++ b/sound/soc/codecs/wcd934x.c
@@ -551,7 +551,7 @@ struct wcd_iir_filter_ctl {
struct soc_bytes_ext bytes_ext;
};
-static const DECLARE_TLV_DB_SCALE(digital_gain, 0, 1, 0);
+static const DECLARE_TLV_DB_SCALE(digital_gain, -8400, 100, -8400);
static const DECLARE_TLV_DB_SCALE(line_gain, 0, 7, 1);
static const DECLARE_TLV_DB_SCALE(analog_gain, 0, 25, 1);
static const DECLARE_TLV_DB_SCALE(ear_pa_gain, 0, 150, 0);
diff --git a/sound/soc/codecs/wsa881x.c b/sound/soc/codecs/wsa881x.c
index 68e774e69c85..4530b74f5921 100644
--- a/sound/soc/codecs/wsa881x.c
+++ b/sound/soc/codecs/wsa881x.c
@@ -1026,6 +1026,8 @@ static struct snd_soc_dai_driver wsa881x_dais[] = {
.id = 0,
.playback = {
.stream_name = "SPKR Playback",
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
.rate_max = 48000,
.rate_min = 48000,
.channels_min = 1,
diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig
index d5bae5d1ab6f..a5b446d5af19 100644
--- a/sound/soc/intel/Kconfig
+++ b/sound/soc/intel/Kconfig
@@ -15,22 +15,6 @@ config SND_SOC_INTEL_SST_TOPLEVEL
if SND_SOC_INTEL_SST_TOPLEVEL
-config SND_SST_IPC
- tristate
- # This option controls the IPC core for HiFi2 platforms
-
-config SND_SST_IPC_PCI
- tristate
- select SND_SST_IPC
- # This option controls the PCI-based IPC for HiFi2 platforms
- # (Medfield, Merrifield).
-
-config SND_SST_IPC_ACPI
- tristate
- select SND_SST_IPC
- # This option controls the ACPI-based IPC for HiFi2 platforms
- # (Baytrail, Cherrytrail)
-
config SND_SOC_INTEL_SST
tristate
@@ -57,7 +41,6 @@ config SND_SST_ATOM_HIFI2_PLATFORM
config SND_SST_ATOM_HIFI2_PLATFORM_PCI
tristate "PCI HiFi2 (Merrifield) Platforms"
depends on X86 && PCI
- select SND_SST_IPC_PCI
select SND_SST_ATOM_HIFI2_PLATFORM
help
If you have a Intel Merrifield/Edison platform, then
@@ -70,7 +53,6 @@ config SND_SST_ATOM_HIFI2_PLATFORM_ACPI
tristate "ACPI HiFi2 (Baytrail, Cherrytrail) Platforms"
default ACPI
depends on X86 && ACPI && PCI
- select SND_SST_IPC_ACPI
select SND_SST_ATOM_HIFI2_PLATFORM
select SND_SOC_ACPI_INTEL_MATCH
select IOSF_MBI
diff --git a/sound/soc/intel/atom/Makefile b/sound/soc/intel/atom/Makefile
index a9326d5ec44c..c66f03f5d8d6 100644
--- a/sound/soc/intel/atom/Makefile
+++ b/sound/soc/intel/atom/Makefile
@@ -6,4 +6,4 @@ snd-soc-sst-atom-hifi2-platform-objs := sst-mfld-platform-pcm.o \
obj-$(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM) += snd-soc-sst-atom-hifi2-platform.o
# DSP driver
-obj-$(CONFIG_SND_SST_IPC) += sst/
+obj-$(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM) += sst/
diff --git a/sound/soc/intel/atom/sst/Makefile b/sound/soc/intel/atom/sst/Makefile
index f17c905df3e2..5761d30a5f9d 100644
--- a/sound/soc/intel/atom/sst/Makefile
+++ b/sound/soc/intel/atom/sst/Makefile
@@ -3,6 +3,6 @@ snd-intel-sst-core-objs := sst.o sst_ipc.o sst_stream.o sst_drv_interface.o sst_
snd-intel-sst-pci-objs += sst_pci.o
snd-intel-sst-acpi-objs += sst_acpi.o
-obj-$(CONFIG_SND_SST_IPC) += snd-intel-sst-core.o
-obj-$(CONFIG_SND_SST_IPC_PCI) += snd-intel-sst-pci.o
-obj-$(CONFIG_SND_SST_IPC_ACPI) += snd-intel-sst-acpi.o
+obj-$(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM) += snd-intel-sst-core.o
+obj-$(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM_PCI) += snd-intel-sst-pci.o
+obj-$(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM_ACPI) += snd-intel-sst-acpi.o
diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c
index 3ea4602dfb3e..9a4b3d0973f6 100644
--- a/sound/soc/intel/boards/kbl_rt5663_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c
@@ -401,17 +401,40 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
struct snd_interval *chan = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
- struct snd_soc_dpcm *dpcm = container_of(
- params, struct snd_soc_dpcm, hw_params);
- struct snd_soc_dai_link *fe_dai_link = dpcm->fe->dai_link;
- struct snd_soc_dai_link *be_dai_link = dpcm->be->dai_link;
+ struct snd_soc_dpcm *dpcm, *rtd_dpcm = NULL;
+
+ /*
+ * The following loop will be called only for playback stream
+ * In this platform, there is only one playback device on every SSP
+ */
+ for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_PLAYBACK, dpcm) {
+ rtd_dpcm = dpcm;
+ break;
+ }
+
+ /*
+ * This following loop will be called only for capture stream
+ * In this platform, there is only one capture device on every SSP
+ */
+ for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_CAPTURE, dpcm) {
+ rtd_dpcm = dpcm;
+ break;
+ }
+
+ if (!rtd_dpcm)
+ return -EINVAL;
+
+ /*
+ * The above 2 loops are mutually exclusive based on the stream direction,
+ * thus rtd_dpcm variable will never be overwritten
+ */
/*
* The ADSP will convert the FE rate to 48k, stereo, 24 bit
*/
- if (!strcmp(fe_dai_link->name, "Kbl Audio Port") ||
- !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") ||
- !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) {
+ if (!strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Port") ||
+ !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Headset Playback") ||
+ !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Capture Port")) {
rate->min = rate->max = 48000;
chan->min = chan->max = 2;
snd_mask_none(fmt);
@@ -421,7 +444,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
* The speaker on the SSP0 supports S16_LE and not S24_LE.
* thus changing the mask here
*/
- if (!strcmp(be_dai_link->name, "SSP0-Codec"))
+ if (!strcmp(rtd_dpcm->be->dai_link->name, "SSP0-Codec"))
snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE);
return 0;
diff --git a/sound/soc/intel/catpt/dsp.c b/sound/soc/intel/catpt/dsp.c
index 7d2968571951..9e807b941732 100644
--- a/sound/soc/intel/catpt/dsp.c
+++ b/sound/soc/intel/catpt/dsp.c
@@ -267,9 +267,12 @@ static int catpt_dsp_select_lpclock(struct catpt_dev *cdev, bool lp, bool waiti)
reg, (reg & CATPT_ISD_DCPWM),
500, 10000);
if (ret) {
- dev_err(cdev->dev, "await WAITI timeout\n");
- mutex_unlock(&cdev->clk_mutex);
- return ret;
+ dev_warn(cdev->dev, "await WAITI timeout\n");
+ /* no signal - only high clock selection allowed */
+ if (lp) {
+ mutex_unlock(&cdev->clk_mutex);
+ return 0;
+ }
}
}
diff --git a/sound/soc/intel/catpt/pcm.c b/sound/soc/intel/catpt/pcm.c
index f78018c857b8..ba653ebea7d1 100644
--- a/sound/soc/intel/catpt/pcm.c
+++ b/sound/soc/intel/catpt/pcm.c
@@ -667,7 +667,17 @@ static int catpt_dai_pcm_new(struct snd_soc_pcm_runtime *rtm,
break;
}
+ /* see if this is a new configuration */
+ if (!memcmp(&cdev->devfmt[devfmt.iface], &devfmt, sizeof(devfmt)))
+ return 0;
+
+ pm_runtime_get_sync(cdev->dev);
+
ret = catpt_ipc_set_device_format(cdev, &devfmt);
+
+ pm_runtime_mark_last_busy(cdev->dev);
+ pm_runtime_put_autosuspend(cdev->dev);
+
if (ret)
return CATPT_IPC_ERROR(ret);
diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c
index c2c1eb16fcc0..26e7d9a7198f 100644
--- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c
+++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c
@@ -630,15 +630,34 @@ static struct snd_soc_codec_conf mt8183_da7219_rt1015_codec_conf[] = {
},
};
+static const struct snd_kcontrol_new mt8183_da7219_rt1015_snd_controls[] = {
+ SOC_DAPM_PIN_SWITCH("Left Spk"),
+ SOC_DAPM_PIN_SWITCH("Right Spk"),
+};
+
+static const
+struct snd_soc_dapm_widget mt8183_da7219_rt1015_dapm_widgets[] = {
+ SND_SOC_DAPM_SPK("Left Spk", NULL),
+ SND_SOC_DAPM_SPK("Right Spk", NULL),
+ SND_SOC_DAPM_PINCTRL("TDM_OUT_PINCTRL",
+ "aud_tdm_out_on", "aud_tdm_out_off"),
+};
+
+static const struct snd_soc_dapm_route mt8183_da7219_rt1015_dapm_routes[] = {
+ {"Left Spk", NULL, "Left SPO"},
+ {"Right Spk", NULL, "Right SPO"},
+ {"I2S Playback", NULL, "TDM_OUT_PINCTRL"},
+};
+
static struct snd_soc_card mt8183_da7219_rt1015_card = {
.name = "mt8183_da7219_rt1015",
.owner = THIS_MODULE,
- .controls = mt8183_da7219_max98357_snd_controls,
- .num_controls = ARRAY_SIZE(mt8183_da7219_max98357_snd_controls),
- .dapm_widgets = mt8183_da7219_max98357_dapm_widgets,
- .num_dapm_widgets = ARRAY_SIZE(mt8183_da7219_max98357_dapm_widgets),
- .dapm_routes = mt8183_da7219_max98357_dapm_routes,
- .num_dapm_routes = ARRAY_SIZE(mt8183_da7219_max98357_dapm_routes),
+ .controls = mt8183_da7219_rt1015_snd_controls,
+ .num_controls = ARRAY_SIZE(mt8183_da7219_rt1015_snd_controls),
+ .dapm_widgets = mt8183_da7219_rt1015_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(mt8183_da7219_rt1015_dapm_widgets),
+ .dapm_routes = mt8183_da7219_rt1015_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(mt8183_da7219_rt1015_dapm_routes),
.dai_link = mt8183_da7219_dai_links,
.num_links = ARRAY_SIZE(mt8183_da7219_dai_links),
.aux_dev = &mt8183_da7219_max98357_headset_dev,
diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c
index ba2aca301a9b..9d17c87445a9 100644
--- a/sound/soc/qcom/lpass-cpu.c
+++ b/sound/soc/qcom/lpass-cpu.c
@@ -80,6 +80,12 @@ static int lpass_cpu_daiops_startup(struct snd_pcm_substream *substream,
dev_err(dai->dev, "error in enabling mi2s osr clk: %d\n", ret);
return ret;
}
+ ret = clk_prepare(drvdata->mi2s_bit_clk[dai->driver->id]);
+ if (ret) {
+ dev_err(dai->dev, "error in enabling mi2s bit clk: %d\n", ret);
+ clk_disable_unprepare(drvdata->mi2s_osr_clk[dai->driver->id]);
+ return ret;
+ }
return 0;
}
@@ -88,9 +94,8 @@ static void lpass_cpu_daiops_shutdown(struct snd_pcm_substream *substream,
{
struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai);
- clk_disable_unprepare(drvdata->mi2s_bit_clk[dai->driver->id]);
-
clk_disable_unprepare(drvdata->mi2s_osr_clk[dai->driver->id]);
+ clk_unprepare(drvdata->mi2s_bit_clk[dai->driver->id]);
}
static int lpass_cpu_daiops_hw_params(struct snd_pcm_substream *substream,
@@ -303,10 +308,10 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream,
dev_err(dai->dev, "error writing to i2sctl reg: %d\n",
ret);
- ret = clk_prepare_enable(drvdata->mi2s_bit_clk[id]);
+ ret = clk_enable(drvdata->mi2s_bit_clk[id]);
if (ret) {
dev_err(dai->dev, "error in enabling mi2s bit clk: %d\n", ret);
- clk_disable_unprepare(drvdata->mi2s_osr_clk[id]);
+ clk_disable(drvdata->mi2s_osr_clk[id]);
return ret;
}
@@ -324,6 +329,7 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream,
if (ret)
dev_err(dai->dev, "error writing to i2sctl reg: %d\n",
ret);
+ clk_disable(drvdata->mi2s_bit_clk[dai->driver->id]);
break;
}
diff --git a/sound/soc/qcom/lpass-sc7180.c b/sound/soc/qcom/lpass-sc7180.c
index c6292f9e613f..bc998d501600 100644
--- a/sound/soc/qcom/lpass-sc7180.c
+++ b/sound/soc/qcom/lpass-sc7180.c
@@ -188,7 +188,7 @@ static struct lpass_variant sc7180_data = {
.micmode = REG_FIELD_ID(0x1000, 4, 8, 3, 0x1000),
.micmono = REG_FIELD_ID(0x1000, 3, 3, 3, 0x1000),
.wssrc = REG_FIELD_ID(0x1000, 2, 2, 3, 0x1000),
- .bitwidth = REG_FIELD_ID(0x1000, 0, 0, 3, 0x1000),
+ .bitwidth = REG_FIELD_ID(0x1000, 0, 1, 3, 0x1000),
.rdma_dyncclk = REG_FIELD_ID(0xC000, 21, 21, 5, 0x1000),
.rdma_bursten = REG_FIELD_ID(0xC000, 20, 20, 5, 0x1000),
diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c
index ab1bf23c21a6..6c2760e27ea6 100644
--- a/sound/soc/qcom/sdm845.c
+++ b/sound/soc/qcom/sdm845.c
@@ -17,6 +17,7 @@
#include "qdsp6/q6afe.h"
#include "../codecs/rt5663.h"
+#define DRIVER_NAME "sdm845"
#define DEFAULT_SAMPLE_RATE_48K 48000
#define DEFAULT_MCLK_RATE 24576000
#define TDM_BCLK_RATE 6144000
@@ -552,6 +553,7 @@ static int sdm845_snd_platform_probe(struct platform_device *pdev)
if (!data)
return -ENOMEM;
+ card->driver_name = DRIVER_NAME;
card->dapm_widgets = sdm845_snd_widgets;
card->num_dapm_widgets = ARRAY_SIZE(sdm845_snd_widgets);
card->dev = dev;
diff --git a/sound/soc/sof/loader.c b/sound/soc/sof/loader.c
index 68ed454f7ddf..ba9ed66f98bc 100644
--- a/sound/soc/sof/loader.c
+++ b/sound/soc/sof/loader.c
@@ -118,6 +118,11 @@ int snd_sof_fw_parse_ext_data(struct snd_sof_dev *sdev, u32 bar, u32 offset)
case SOF_IPC_EXT_CC_INFO:
ret = get_cc_info(sdev, ext_hdr);
break;
+ case SOF_IPC_EXT_UNUSED:
+ case SOF_IPC_EXT_PROBE_INFO:
+ case SOF_IPC_EXT_USER_ABI_INFO:
+ /* They are supported but we don't do anything here */
+ break;
default:
dev_warn(sdev->dev, "warning: unknown ext header type %d size 0x%x\n",
ext_hdr->type, ext_hdr->hdr.size);