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authorGravatar Linus Torvalds <torvalds@linux-foundation.org> 2024-01-19 12:30:29 -0800
committerGravatar Linus Torvalds <torvalds@linux-foundation.org> 2024-01-19 12:30:29 -0800
commita1fe5b6d0dce12893f40f0f3cc4e3885456155fb (patch)
tree522fac116264c34260c675aa3e0045bf2fe82fe2 /sound/soc/codecs
parentMerge tag 'drm-next-2024-01-19' of git://anongit.freedesktop.org/drm/drm (diff)
parentALSA: hda/realtek: Enable headset mic on Lenovo M70 Gen5 (diff)
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Merge tag 'sound-fix-6.8-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai: "A collection of small fixes: - Lots of ASoC SOF fixes and related reworks - ASoC TAS codec fixes including DT updates - A few HD-audio quirks and regression fixes - Minor fixes for aloop, oxygen and scarlett2 mixer" * tag 'sound-fix-6.8-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (23 commits) ALSA: hda/realtek: Enable headset mic on Lenovo M70 Gen5 ALSA: hda/realtek: Enable mute/micmute LEDs and limit mic boost on HP ZBook ALSA: hda/relatek: Enable Mute LED on HP Laptop 15s-fq2xxx ASoC: SOF: ipc4-loader: remove the CPC check warnings ASoC: SOF: ipc4-pcm: remove log message for LLP ALSA: hda: generic: Remove obsolete call to ledtrig_audio_get ALSA: scarlett2: Fix yet more -Wformat-truncation warnings ALSA: hda: Properly setup HDMI stream ASoC: audio-graph-card2: fix index check on graph_parse_node_multi_nm() ASoC: SOF: icp3-dtrace: Revert "Fix wrong kfree() usage" ALSA: oxygen: Fix right channel of capture volume mixer ALSA: aloop: Introduce a function to get if access is interleaved mode ASoC: mediatek: sof-common: Add NULL check for normal_link string ASoC: mediatek: mt8195: Remove afe-dai component and rework codec link ASoC: mediatek: mt8192: Check existence of dai_name before dereferencing ASoC: Intel: bxt_rt298: Fix kernel ops due to COMP_DUMMY change ASoC: Intel: bxt_da7219_max98357a: Fix kernel ops due to COMP_DUMMY change ASoC: codecs: rtq9128: Fix TDM enable and DAI format control flow ASoC: codecs: rtq9128: Fix PM_RUNTIME usage ASoC: tas2781: Add tas2563 into driver ...
Diffstat (limited to 'sound/soc/codecs')
-rw-r--r--sound/soc/codecs/rtq9128.c73
-rw-r--r--sound/soc/codecs/tas2562.c3
-rw-r--r--sound/soc/codecs/tas2781-i2c.c8
3 files changed, 46 insertions, 38 deletions
diff --git a/sound/soc/codecs/rtq9128.c b/sound/soc/codecs/rtq9128.c
index c22b047115cc..aa3eadecd974 100644
--- a/sound/soc/codecs/rtq9128.c
+++ b/sound/soc/codecs/rtq9128.c
@@ -59,6 +59,7 @@
struct rtq9128_data {
struct gpio_desc *enable;
+ unsigned int daifmt;
int tdm_slots;
int tdm_slot_width;
bool tdm_input_data2_select;
@@ -391,7 +392,11 @@ static int rtq9128_component_probe(struct snd_soc_component *comp)
unsigned int val;
int i, ret;
- pm_runtime_resume_and_get(comp->dev);
+ ret = pm_runtime_resume_and_get(comp->dev);
+ if (ret < 0) {
+ dev_err(comp->dev, "Failed to resume device (%d)\n", ret);
+ return ret;
+ }
val = snd_soc_component_read(comp, RTQ9128_REG_EFUSE_DATA);
@@ -437,10 +442,7 @@ static const struct snd_soc_component_driver rtq9128_comp_driver = {
static int rtq9128_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
{
struct rtq9128_data *data = snd_soc_dai_get_drvdata(dai);
- struct snd_soc_component *comp = dai->component;
struct device *dev = dai->dev;
- unsigned int audfmt, fmtval;
- int ret;
dev_dbg(dev, "%s: fmt 0x%8x\n", __func__, fmt);
@@ -450,35 +452,10 @@ static int rtq9128_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
return -EINVAL;
}
- fmtval = fmt & SND_SOC_DAIFMT_FORMAT_MASK;
- if (data->tdm_slots && fmtval != SND_SOC_DAIFMT_DSP_A && fmtval != SND_SOC_DAIFMT_DSP_B) {
- dev_err(dev, "TDM is used, format only support DSP_A or DSP_B\n");
- return -EINVAL;
- }
+ /* Store here and will be used in runtime hw_params for DAI format setting */
+ data->daifmt = fmt;
- switch (fmtval) {
- case SND_SOC_DAIFMT_I2S:
- audfmt = 8;
- break;
- case SND_SOC_DAIFMT_LEFT_J:
- audfmt = 9;
- break;
- case SND_SOC_DAIFMT_RIGHT_J:
- audfmt = 10;
- break;
- case SND_SOC_DAIFMT_DSP_A:
- audfmt = data->tdm_slots ? 12 : 11;
- break;
- case SND_SOC_DAIFMT_DSP_B:
- audfmt = data->tdm_slots ? 4 : 3;
- break;
- default:
- dev_err(dev, "Unsupported format 0x%8x\n", fmt);
- return -EINVAL;
- }
-
- ret = snd_soc_component_write_field(comp, RTQ9128_REG_I2S_OPT, RTQ9128_AUDFMT_MASK, audfmt);
- return ret < 0 ? ret : 0;
+ return 0;
}
static int rtq9128_dai_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
@@ -554,10 +531,38 @@ static int rtq9128_dai_hw_params(struct snd_pcm_substream *stream, struct snd_pc
unsigned int width, slot_width, bitrate, audbit, dolen;
struct snd_soc_component *comp = dai->component;
struct device *dev = dai->dev;
+ unsigned int fmtval, audfmt;
int ret;
dev_dbg(dev, "%s: width %d\n", __func__, params_width(param));
+ fmtval = FIELD_GET(SND_SOC_DAIFMT_FORMAT_MASK, data->daifmt);
+ if (data->tdm_slots && fmtval != SND_SOC_DAIFMT_DSP_A && fmtval != SND_SOC_DAIFMT_DSP_B) {
+ dev_err(dev, "TDM is used, format only support DSP_A or DSP_B\n");
+ return -EINVAL;
+ }
+
+ switch (fmtval) {
+ case SND_SOC_DAIFMT_I2S:
+ audfmt = 8;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ audfmt = 9;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ audfmt = 10;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ audfmt = data->tdm_slots ? 12 : 11;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ audfmt = data->tdm_slots ? 4 : 3;
+ break;
+ default:
+ dev_err(dev, "Unsupported format 0x%8x\n", fmtval);
+ return -EINVAL;
+ }
+
switch (width = params_width(param)) {
case 16:
audbit = 0;
@@ -611,6 +616,10 @@ static int rtq9128_dai_hw_params(struct snd_pcm_substream *stream, struct snd_pc
return -EINVAL;
}
+ ret = snd_soc_component_write_field(comp, RTQ9128_REG_I2S_OPT, RTQ9128_AUDFMT_MASK, audfmt);
+ if (ret < 0)
+ return ret;
+
ret = snd_soc_component_write_field(comp, RTQ9128_REG_I2S_OPT, RTQ9128_AUDBIT_MASK, audbit);
if (ret < 0)
return ret;
diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c
index 962c2cdfa017..54561ae598b8 100644
--- a/sound/soc/codecs/tas2562.c
+++ b/sound/soc/codecs/tas2562.c
@@ -59,7 +59,6 @@ struct tas2562_data {
enum tas256x_model {
TAS2562,
- TAS2563,
TAS2564,
TAS2110,
};
@@ -721,7 +720,6 @@ static int tas2562_parse_dt(struct tas2562_data *tas2562)
static const struct i2c_device_id tas2562_id[] = {
{ "tas2562", TAS2562 },
- { "tas2563", TAS2563 },
{ "tas2564", TAS2564 },
{ "tas2110", TAS2110 },
{ }
@@ -770,7 +768,6 @@ static int tas2562_probe(struct i2c_client *client)
#ifdef CONFIG_OF
static const struct of_device_id tas2562_of_match[] = {
{ .compatible = "ti,tas2562", },
- { .compatible = "ti,tas2563", },
{ .compatible = "ti,tas2564", },
{ .compatible = "ti,tas2110", },
{ },
diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c
index 917b1c15f71d..32913bd1a623 100644
--- a/sound/soc/codecs/tas2781-i2c.c
+++ b/sound/soc/codecs/tas2781-i2c.c
@@ -1,13 +1,13 @@
// SPDX-License-Identifier: GPL-2.0
//
-// ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier
+// ALSA SoC Texas Instruments TAS2563/TAS2781 Audio Smart Amplifier
//
// Copyright (C) 2022 - 2023 Texas Instruments Incorporated
// https://www.ti.com
//
-// The TAS2781 driver implements a flexible and configurable
+// The TAS2563/TAS2781 driver implements a flexible and configurable
// algo coefficient setting for one, two, or even multiple
-// TAS2781 chips.
+// TAS2563/TAS2781 chips.
//
// Author: Shenghao Ding <shenghao-ding@ti.com>
// Author: Kevin Lu <kevin-lu@ti.com>
@@ -32,6 +32,7 @@
#include <sound/tas2781-tlv.h>
static const struct i2c_device_id tasdevice_id[] = {
+ { "tas2563", TAS2563 },
{ "tas2781", TAS2781 },
{}
};
@@ -39,6 +40,7 @@ MODULE_DEVICE_TABLE(i2c, tasdevice_id);
#ifdef CONFIG_OF
static const struct of_device_id tasdevice_of_match[] = {
+ { .compatible = "ti,tas2563" },
{ .compatible = "ti,tas2781" },
{},
};