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-rw-r--r--Documentation/devicetree/bindings/sound/tas2562.yaml2
-rw-r--r--Documentation/devicetree/bindings/sound/ti,tas2781.yaml78
-rw-r--r--include/sound/tas2781.h9
-rw-r--r--sound/drivers/aloop.c23
-rw-r--r--sound/pci/hda/hda_generic.c1
-rw-r--r--sound/pci/hda/patch_hdmi.c6
-rw-r--r--sound/pci/hda/patch_realtek.c3
-rw-r--r--sound/pci/oxygen/oxygen_mixer.c2
-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
-rw-r--r--sound/soc/generic/audio-graph-card2.c2
-rw-r--r--sound/soc/intel/boards/bxt_da7219_max98357a.c6
-rw-r--r--sound/soc/intel/boards/bxt_rt298.c3
-rw-r--r--sound/soc/mediatek/common/mtk-dsp-sof-common.c2
-rw-r--r--sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c3
-rw-r--r--sound/soc/mediatek/mt8195/mt8195-afe-pcm.c33
-rw-r--r--sound/soc/mediatek/mt8195/mt8195-mt6359.c41
-rw-r--r--sound/soc/sof/ipc3-dtrace.c3
-rw-r--r--sound/soc/sof/ipc4-loader.c11
-rw-r--r--sound/soc/sof/ipc4-pcm.c4
-rw-r--r--sound/usb/mixer_scarlett2.c42
22 files changed, 209 insertions, 149 deletions
diff --git a/Documentation/devicetree/bindings/sound/tas2562.yaml b/Documentation/devicetree/bindings/sound/tas2562.yaml
index f01c0dde0cf7..d28c102c0ce7 100644
--- a/Documentation/devicetree/bindings/sound/tas2562.yaml
+++ b/Documentation/devicetree/bindings/sound/tas2562.yaml
@@ -18,7 +18,6 @@ description: |
Specifications about the audio amplifier can be found at:
https://www.ti.com/lit/gpn/tas2562
- https://www.ti.com/lit/gpn/tas2563
https://www.ti.com/lit/gpn/tas2564
https://www.ti.com/lit/gpn/tas2110
@@ -29,7 +28,6 @@ properties:
compatible:
enum:
- ti,tas2562
- - ti,tas2563
- ti,tas2564
- ti,tas2110
diff --git a/Documentation/devicetree/bindings/sound/ti,tas2781.yaml b/Documentation/devicetree/bindings/sound/ti,tas2781.yaml
index a69e6c223308..976238689249 100644
--- a/Documentation/devicetree/bindings/sound/ti,tas2781.yaml
+++ b/Documentation/devicetree/bindings/sound/ti,tas2781.yaml
@@ -5,36 +5,46 @@
$id: http://devicetree.org/schemas/sound/ti,tas2781.yaml#
$schema: http://devicetree.org/meta-schemas/core.yaml#
-title: Texas Instruments TAS2781 SmartAMP
+title: Texas Instruments TAS2563/TAS2781 SmartAMP
maintainers:
- Shenghao Ding <shenghao-ding@ti.com>
-description:
- The TAS2781 is a mono, digital input Class-D audio amplifier
- optimized for efficiently driving high peak power into small
- loudspeakers. An integrated on-chip DSP supports Texas Instruments
- Smart Amp speaker protection algorithm. The integrated speaker
- voltage and current sense provides for real time
+description: |
+ The TAS2563/TAS2781 is a mono, digital input Class-D audio
+ amplifier optimized for efficiently driving high peak power into
+ small loudspeakers. An integrated on-chip DSP supports Texas
+ Instruments Smart Amp speaker protection algorithm. The
+ integrated speaker voltage and current sense provides for real time
monitoring of loudspeaker behavior.
-allOf:
- - $ref: dai-common.yaml#
+ Specifications about the audio amplifier can be found at:
+ https://www.ti.com/lit/gpn/tas2563
+ https://www.ti.com/lit/gpn/tas2781
properties:
compatible:
- enum:
- - ti,tas2781
+ description: |
+ ti,tas2563: 6.1-W Boosted Class-D Audio Amplifier With Integrated
+ DSP and IV Sense, 16/20/24/32bit stereo I2S or multichannel TDM.
+
+ ti,tas2781: 24-V Class-D Amplifier with Real Time Integrated Speaker
+ Protection and Audio Processing, 16/20/24/32bit stereo I2S or
+ multichannel TDM.
+ oneOf:
+ - items:
+ - enum:
+ - ti,tas2563
+ - const: ti,tas2781
+ - enum:
+ - ti,tas2781
reg:
description:
- I2C address, in multiple tas2781s case, all the i2c address
+ I2C address, in multiple-AMP case, all the i2c address
aggregate as one Audio Device to support multiple audio slots.
maxItems: 8
minItems: 1
- items:
- minimum: 0x38
- maximum: 0x3f
reset-gpios:
maxItems: 1
@@ -49,6 +59,44 @@ required:
- compatible
- reg
+allOf:
+ - $ref: dai-common.yaml#
+ - if:
+ properties:
+ compatible:
+ contains:
+ enum:
+ - ti,tas2563
+ then:
+ properties:
+ reg:
+ description:
+ I2C address, in multiple-AMP case, all the i2c address
+ aggregate as one Audio Device to support multiple audio slots.
+ maxItems: 4
+ minItems: 1
+ items:
+ minimum: 0x4c
+ maximum: 0x4f
+
+ - if:
+ properties:
+ compatible:
+ contains:
+ enum:
+ - ti,tas2781
+ then:
+ properties:
+ reg:
+ description:
+ I2C address, in multiple-AMP case, all the i2c address
+ aggregate as one Audio Device to support multiple audio slots.
+ maxItems: 8
+ minItems: 1
+ items:
+ minimum: 0x38
+ maximum: 0x3f
+
additionalProperties: false
examples:
diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h
index 0a86ab8d47b9..b00d65417c31 100644
--- a/include/sound/tas2781.h
+++ b/include/sound/tas2781.h
@@ -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>
@@ -60,7 +60,8 @@
#define TASDEVICE_CMD_FIELD_W 0x4
enum audio_device {
- TAS2781 = 0,
+ TAS2563,
+ TAS2781,
};
enum device_catlog_id {
diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c
index e87dc67f33c6..1c65e0a3b13c 100644
--- a/sound/drivers/aloop.c
+++ b/sound/drivers/aloop.c
@@ -322,6 +322,17 @@ static int loopback_snd_timer_close_cable(struct loopback_pcm *dpcm)
return 0;
}
+static bool is_access_interleaved(snd_pcm_access_t access)
+{
+ switch (access) {
+ case SNDRV_PCM_ACCESS_MMAP_INTERLEAVED:
+ case SNDRV_PCM_ACCESS_RW_INTERLEAVED:
+ return true;
+ default:
+ return false;
+ }
+};
+
static int loopback_check_format(struct loopback_cable *cable, int stream)
{
struct snd_pcm_runtime *runtime, *cruntime;
@@ -341,7 +352,8 @@ static int loopback_check_format(struct loopback_cable *cable, int stream)
check = runtime->format != cruntime->format ||
runtime->rate != cruntime->rate ||
runtime->channels != cruntime->channels ||
- runtime->access != cruntime->access;
+ is_access_interleaved(runtime->access) !=
+ is_access_interleaved(cruntime->access);
if (!check)
return 0;
if (stream == SNDRV_PCM_STREAM_CAPTURE) {
@@ -369,7 +381,8 @@ static int loopback_check_format(struct loopback_cable *cable, int stream)
&setup->channels_id);
setup->channels = runtime->channels;
}
- if (setup->access != runtime->access) {
+ if (is_access_interleaved(setup->access) !=
+ is_access_interleaved(runtime->access)) {
snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
&setup->access_id);
setup->access = runtime->access;
@@ -584,8 +597,7 @@ static void copy_play_buf(struct loopback_pcm *play,
size = play->pcm_buffer_size - src_off;
if (dst_off + size > capt->pcm_buffer_size)
size = capt->pcm_buffer_size - dst_off;
- if (runtime->access == SNDRV_PCM_ACCESS_RW_NONINTERLEAVED ||
- runtime->access == SNDRV_PCM_ACCESS_MMAP_NONINTERLEAVED)
+ if (!is_access_interleaved(runtime->access))
copy_play_buf_part_n(play, capt, size, src_off, dst_off);
else
memcpy(dst + dst_off, src + src_off, size);
@@ -1544,8 +1556,7 @@ static int loopback_access_get(struct snd_kcontrol *kcontrol,
mutex_lock(&loopback->cable_lock);
access = loopback->setup[kcontrol->id.subdevice][kcontrol->id.device].access;
- ucontrol->value.enumerated.item[0] = access == SNDRV_PCM_ACCESS_RW_NONINTERLEAVED ||
- access == SNDRV_PCM_ACCESS_MMAP_NONINTERLEAVED;
+ ucontrol->value.enumerated.item[0] = !is_access_interleaved(access);
mutex_unlock(&loopback->cable_lock);
return 0;
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index bf685d01259d..de2a3d08c73c 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -3946,7 +3946,6 @@ static int create_mute_led_cdev(struct hda_codec *codec,
cdev->max_brightness = 1;
cdev->default_trigger = micmute ? "audio-micmute" : "audio-mute";
cdev->brightness_set_blocking = callback;
- cdev->brightness = ledtrig_audio_get(idx);
cdev->flags = LED_CORE_SUSPENDRESUME;
err = led_classdev_register(&codec->core.dev, cdev);
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 200779296a1b..495d63101186 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -2301,6 +2301,7 @@ static int generic_hdmi_build_pcms(struct hda_codec *codec)
codec_dbg(codec, "hdmi: pcm_num set to %d\n", pcm_num);
for (idx = 0; idx < pcm_num; idx++) {
+ struct hdmi_spec_per_cvt *per_cvt;
struct hda_pcm *info;
struct hda_pcm_stream *pstr;
@@ -2316,6 +2317,11 @@ static int generic_hdmi_build_pcms(struct hda_codec *codec)
pstr = &info->stream[SNDRV_PCM_STREAM_PLAYBACK];
pstr->substreams = 1;
pstr->ops = generic_ops;
+
+ per_cvt = get_cvt(spec, 0);
+ pstr->channels_min = per_cvt->channels_min;
+ pstr->channels_max = per_cvt->channels_max;
+
/* pcm number is less than pcm_rec array size */
if (spec->pcm_used >= ARRAY_SIZE(spec->pcm_rec))
break;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index b68c94757051..f6f16622f9cc 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -9861,6 +9861,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x103c, 0x87f5, "HP", ALC287_FIXUP_HP_GPIO_LED),
SND_PCI_QUIRK(0x103c, 0x87f6, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP),
SND_PCI_QUIRK(0x103c, 0x87f7, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP),
+ SND_PCI_QUIRK(0x103c, 0x87fe, "HP Laptop 15s-fq2xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2),
SND_PCI_QUIRK(0x103c, 0x8805, "HP ProBook 650 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED),
SND_PCI_QUIRK(0x103c, 0x880d, "HP EliteBook 830 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED),
SND_PCI_QUIRK(0x103c, 0x8811, "HP Spectre x360 15-eb1xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1),
@@ -9955,6 +9956,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x103c, 0x8c71, "HP EliteBook 845 G11", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED),
SND_PCI_QUIRK(0x103c, 0x8c72, "HP EliteBook 865 G11", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED),
SND_PCI_QUIRK(0x103c, 0x8c96, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF),
+ SND_PCI_QUIRK(0x103c, 0x8c97, "HP ZBook", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF),
SND_PCI_QUIRK(0x103c, 0x8ca4, "HP ZBook Fury", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED),
SND_PCI_QUIRK(0x103c, 0x8ca7, "HP ZBook Fury", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED),
SND_PCI_QUIRK(0x103c, 0x8cf5, "HP ZBook Studio 16", ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED),
@@ -10231,6 +10233,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x3176, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC),
SND_PCI_QUIRK(0x17aa, 0x3178, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC),
SND_PCI_QUIRK(0x17aa, 0x31af, "ThinkCentre Station", ALC623_FIXUP_LENOVO_THINKSTATION_P340),
+ SND_PCI_QUIRK(0x17aa, 0x334b, "Lenovo ThinkCentre M70 Gen5", ALC283_FIXUP_HEADSET_MIC),
SND_PCI_QUIRK(0x17aa, 0x3801, "Lenovo Yoga9 14IAP7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN),
SND_PCI_QUIRK(0x17aa, 0x3802, "Lenovo Yoga DuetITL 2021", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS),
SND_PCI_QUIRK(0x17aa, 0x3813, "Legion 7i 15IMHG05", ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS),
diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c
index 46705ec77b48..eb3aca16359c 100644
--- a/sound/pci/oxygen/oxygen_mixer.c
+++ b/sound/pci/oxygen/oxygen_mixer.c
@@ -718,7 +718,7 @@ static int ac97_fp_rec_volume_put(struct snd_kcontrol *ctl,
oldreg = oxygen_read_ac97(chip, 1, AC97_REC_GAIN);
newreg = oldreg & ~0x0707;
newreg = newreg | (value->value.integer.value[0] & 7);
- newreg = newreg | ((value->value.integer.value[0] & 7) << 8);
+ newreg = newreg | ((value->value.integer.value[1] & 7) << 8);
change = newreg != oldreg;
if (change)
oxygen_write_ac97(chip, 1, AC97_REC_GAIN, newreg);
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" },
{},
};
diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c
index 9c94677f681a..62606e20be9a 100644
--- a/sound/soc/generic/audio-graph-card2.c
+++ b/sound/soc/generic/audio-graph-card2.c
@@ -556,7 +556,7 @@ static int graph_parse_node_multi_nm(struct snd_soc_dai_link *dai_link,
struct device_node *mcodec_port;
int codec_idx;
- if (*nm_idx >= nm_max)
+ if (*nm_idx > nm_max)
break;
mcpu_ep_n = of_get_next_child(mcpu_port, mcpu_ep_n);
diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c
index 816fad8c1ff0..540f7a29310a 100644
--- a/sound/soc/intel/boards/bxt_da7219_max98357a.c
+++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c
@@ -797,6 +797,9 @@ static int broxton_audio_probe(struct platform_device *pdev)
broxton_audio_card.name = "glkda7219max";
/* Fixup the SSP entries for geminilake */
for (i = 0; i < ARRAY_SIZE(broxton_dais); i++) {
+ if (!broxton_dais[i].codecs->dai_name)
+ continue;
+
/* MAXIM_CODEC is connected to SSP1. */
if (!strcmp(broxton_dais[i].codecs->dai_name,
BXT_MAXIM_CODEC_DAI)) {
@@ -822,6 +825,9 @@ static int broxton_audio_probe(struct platform_device *pdev)
broxton_audio_card.name = "cmlda7219max";
for (i = 0; i < ARRAY_SIZE(broxton_dais); i++) {
+ if (!broxton_dais[i].codecs->dai_name)
+ continue;
+
/* MAXIM_CODEC is connected to SSP1. */
if (!strcmp(broxton_dais[i].codecs->dai_name,
BXT_MAXIM_CODEC_DAI)) {
diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c
index 4631106f2a28..c0eb65c14aa9 100644
--- a/sound/soc/intel/boards/bxt_rt298.c
+++ b/sound/soc/intel/boards/bxt_rt298.c
@@ -604,7 +604,8 @@ static int broxton_audio_probe(struct platform_device *pdev)
int i;
for (i = 0; i < ARRAY_SIZE(broxton_rt298_dais); i++) {
- if (!strncmp(card->dai_link[i].codecs->name, "i2c-INT343A:00",
+ if (card->dai_link[i].codecs->name &&
+ !strncmp(card->dai_link[i].codecs->name, "i2c-INT343A:00",
I2C_NAME_SIZE)) {
if (!strncmp(card->name, "broxton-rt298",
PLATFORM_NAME_SIZE)) {
diff --git a/sound/soc/mediatek/common/mtk-dsp-sof-common.c b/sound/soc/mediatek/common/mtk-dsp-sof-common.c
index f3894010f656..7ec8965a70c0 100644
--- a/sound/soc/mediatek/common/mtk-dsp-sof-common.c
+++ b/sound/soc/mediatek/common/mtk-dsp-sof-common.c
@@ -24,7 +24,7 @@ int mtk_sof_dai_link_fixup(struct snd_soc_pcm_runtime *rtd,
struct snd_soc_dai_link *sof_dai_link = NULL;
const struct sof_conn_stream *conn = &sof_priv->conn_streams[i];
- if (strcmp(rtd->dai_link->name, conn->normal_link))
+ if (conn->normal_link && strcmp(rtd->dai_link->name, conn->normal_link))
continue;
for_each_card_rtds(card, runtime) {
diff --git a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c
index 5bd6addd1450..bfcb2c486c39 100644
--- a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c
+++ b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c
@@ -1208,7 +1208,8 @@ static int mt8192_mt6359_dev_probe(struct platform_device *pdev)
dai_link->ignore = 0;
}
- if (strcmp(dai_link->codecs[0].dai_name, RT1015_CODEC_DAI) == 0)
+ if (dai_link->num_codecs && dai_link->codecs[0].dai_name &&
+ strcmp(dai_link->codecs[0].dai_name, RT1015_CODEC_DAI) == 0)
dai_link->ops = &mt8192_rt1015_i2s_ops;
if (!dai_link->platforms->name)
diff --git a/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c b/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c
index 1e33863c85ca..620d7ade1992 100644
--- a/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c
+++ b/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c
@@ -1795,10 +1795,6 @@ static const struct snd_kcontrol_new mt8195_memif_controls[] = {
MT8195_AFE_IRQ_28),
};
-static const struct snd_soc_component_driver mt8195_afe_pcm_dai_component = {
- .name = "mt8195-afe-pcm-dai",
-};
-
static const struct mtk_base_memif_data memif_data[MT8195_AFE_MEMIF_NUM] = {
[MT8195_AFE_MEMIF_DL2] = {
.name = "DL2",
@@ -3037,7 +3033,6 @@ static int mt8195_afe_pcm_dev_probe(struct platform_device *pdev)
struct device *dev = &pdev->dev;
struct reset_control *rstc;
int i, irq_id, ret;
- struct snd_soc_component *component;
ret = of_reserved_mem_device_init(dev);
if (ret)
@@ -3170,36 +3165,12 @@ static int mt8195_afe_pcm_dev_probe(struct platform_device *pdev)
/* register component */
ret = devm_snd_soc_register_component(dev, &mt8195_afe_component,
- NULL, 0);
+ afe->dai_drivers, afe->num_dai_drivers);
if (ret) {
dev_warn(dev, "err_platform\n");
goto err_pm_put;
}
- component = devm_kzalloc(dev, sizeof(*component), GFP_KERNEL);
- if (!component) {
- ret = -ENOMEM;
- goto err_pm_put;
- }
-
- ret = snd_soc_component_initialize(component,
- &mt8195_afe_pcm_dai_component,
- dev);
- if (ret)
- goto err_pm_put;
-
-#ifdef CONFIG_DEBUG_FS
- component->debugfs_prefix = "pcm";
-#endif
-
- ret = snd_soc_add_component(component,
- afe->dai_drivers,
- afe->num_dai_drivers);
- if (ret) {
- dev_warn(dev, "err_dai_component\n");
- goto err_pm_put;
- }
-
ret = regmap_multi_reg_write(afe->regmap, mt8195_afe_reg_defaults,
ARRAY_SIZE(mt8195_afe_reg_defaults));
if (ret)
@@ -3224,8 +3195,6 @@ err_pm_put:
static void mt8195_afe_pcm_dev_remove(struct platform_device *pdev)
{
- snd_soc_unregister_component(&pdev->dev);
-
pm_runtime_disable(&pdev->dev);
if (!pm_runtime_status_suspended(&pdev->dev))
mt8195_afe_runtime_suspend(&pdev->dev);
diff --git a/sound/soc/mediatek/mt8195/mt8195-mt6359.c b/sound/soc/mediatek/mt8195/mt8195-mt6359.c
index 4feb9fb76967..53fd8a897b9d 100644
--- a/sound/soc/mediatek/mt8195/mt8195-mt6359.c
+++ b/sound/soc/mediatek/mt8195/mt8195-mt6359.c
@@ -934,12 +934,11 @@ SND_SOC_DAILINK_DEFS(ETDM1_IN_BE,
SND_SOC_DAILINK_DEFS(ETDM2_IN_BE,
DAILINK_COMP_ARRAY(COMP_CPU("ETDM2_IN")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()),
DAILINK_COMP_ARRAY(COMP_EMPTY()));
SND_SOC_DAILINK_DEFS(ETDM1_OUT_BE,
DAILINK_COMP_ARRAY(COMP_CPU("ETDM1_OUT")),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
DAILINK_COMP_ARRAY(COMP_EMPTY()));
SND_SOC_DAILINK_DEFS(ETDM2_OUT_BE,
@@ -1237,8 +1236,6 @@ static struct snd_soc_dai_link mt8195_mt6359_dai_links[] = {
SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBS_CFS,
.dpcm_capture = 1,
- .init = mt8195_rt5682_init,
- .ops = &mt8195_rt5682_etdm_ops,
.be_hw_params_fixup = mt8195_etdm_hw_params_fixup,
SND_SOC_DAILINK_REG(ETDM2_IN_BE),
},
@@ -1249,7 +1246,6 @@ static struct snd_soc_dai_link mt8195_mt6359_dai_links[] = {
SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBS_CFS,
.dpcm_playback = 1,
- .ops = &mt8195_rt5682_etdm_ops,
.be_hw_params_fixup = mt8195_etdm_hw_params_fixup,
SND_SOC_DAILINK_REG(ETDM1_OUT_BE),
},
@@ -1381,7 +1377,7 @@ static int mt8195_mt6359_dev_probe(struct platform_device *pdev)
struct snd_soc_dai_link *dai_link;
struct mtk_soc_card_data *soc_card_data;
struct mt8195_mt6359_priv *mach_priv;
- struct device_node *platform_node, *adsp_node, *dp_node, *hdmi_node;
+ struct device_node *platform_node, *adsp_node, *codec_node, *dp_node, *hdmi_node;
struct mt8195_card_data *card_data;
int is5682s = 0;
int init6359 = 0;
@@ -1401,8 +1397,12 @@ static int mt8195_mt6359_dev_probe(struct platform_device *pdev)
if (!card->name)
card->name = card_data->name;
- if (strstr(card->name, "_5682s"))
+ if (strstr(card->name, "_5682s")) {
+ codec_node = of_find_compatible_node(NULL, NULL, "realtek,rt5682s");
is5682s = 1;
+ } else
+ codec_node = of_find_compatible_node(NULL, NULL, "realtek,rt5682i");
+
soc_card_data = devm_kzalloc(&pdev->dev, sizeof(*card_data), GFP_KERNEL);
if (!soc_card_data)
return -ENOMEM;
@@ -1488,12 +1488,27 @@ static int mt8195_mt6359_dev_probe(struct platform_device *pdev)
dai_link->codecs->dai_name = "i2s-hifi";
dai_link->init = mt8195_hdmi_codec_init;
}
- } else if (strcmp(dai_link->name, "ETDM1_OUT_BE") == 0 ||
- strcmp(dai_link->name, "ETDM2_IN_BE") == 0) {
- dai_link->codecs->name =
- is5682s ? RT5682S_DEV0_NAME : RT5682_DEV0_NAME;
- dai_link->codecs->dai_name =
- is5682s ? RT5682S_CODEC_DAI : RT5682_CODEC_DAI;
+ } else if (strcmp(dai_link->name, "ETDM1_OUT_BE") == 0) {
+ if (!codec_node) {
+ dev_err(&pdev->dev, "Codec not found!\n");
+ } else {
+ dai_link->codecs->of_node = codec_node;
+ dai_link->codecs->name = NULL;
+ dai_link->codecs->dai_name =
+ is5682s ? RT5682S_CODEC_DAI : RT5682_CODEC_DAI;
+ dai_link->init = mt8195_rt5682_init;
+ dai_link->ops = &mt8195_rt5682_etdm_ops;
+ }
+ } else if (strcmp(dai_link->name, "ETDM2_IN_BE") == 0) {
+ if (!codec_node) {
+ dev_err(&pdev->dev, "Codec not found!\n");
+ } else {
+ dai_link->codecs->of_node = codec_node;
+ dai_link->codecs->name = NULL;
+ dai_link->codecs->dai_name =
+ is5682s ? RT5682S_CODEC_DAI : RT5682_CODEC_DAI;
+ dai_link->ops = &mt8195_rt5682_etdm_ops;
+ }
} else if (strcmp(dai_link->name, "DL_SRC_BE") == 0 ||
strcmp(dai_link->name, "UL_SRC1_BE") == 0 ||
strcmp(dai_link->name, "UL_SRC2_BE") == 0) {
diff --git a/sound/soc/sof/ipc3-dtrace.c b/sound/soc/sof/ipc3-dtrace.c
index 93b189c2d2ee..0dca139322f3 100644
--- a/sound/soc/sof/ipc3-dtrace.c
+++ b/sound/soc/sof/ipc3-dtrace.c
@@ -137,7 +137,6 @@ static int trace_filter_parse(struct snd_sof_dev *sdev, char *string,
dev_err(sdev->dev,
"Parsing filter entry '%s' failed with %d\n",
entry, entry_len);
- kfree(*out);
return -EINVAL;
}
}
@@ -209,13 +208,13 @@ static ssize_t dfsentry_trace_filter_write(struct file *file, const char __user
ret = ipc3_trace_update_filter(sdev, num_elems, elems);
if (ret < 0) {
dev_err(sdev->dev, "Filter update failed: %d\n", ret);
- kfree(elems);
goto error;
}
}
ret = count;
error:
kfree(string);
+ kfree(elems);
return ret;
}
diff --git a/sound/soc/sof/ipc4-loader.c b/sound/soc/sof/ipc4-loader.c
index 3539b0a66e1b..c79479afa8d0 100644
--- a/sound/soc/sof/ipc4-loader.c
+++ b/sound/soc/sof/ipc4-loader.c
@@ -482,13 +482,10 @@ void sof_ipc4_update_cpc_from_manifest(struct snd_sof_dev *sdev,
msg = "No CPC match in the firmware file's manifest";
no_cpc:
- dev_warn(sdev->dev, "%s (UUID: %pUL): %s (ibs/obs: %u/%u)\n",
- fw_module->man4_module_entry.name,
- &fw_module->man4_module_entry.uuid, msg, basecfg->ibs,
- basecfg->obs);
- dev_warn_once(sdev->dev, "Please try to update the firmware.\n");
- dev_warn_once(sdev->dev, "If the issue persists, file a bug at\n");
- dev_warn_once(sdev->dev, "https://github.com/thesofproject/sof/issues/\n");
+ dev_dbg(sdev->dev, "%s (UUID: %pUL): %s (ibs/obs: %u/%u)\n",
+ fw_module->man4_module_entry.name,
+ &fw_module->man4_module_entry.uuid, msg, basecfg->ibs,
+ basecfg->obs);
}
const struct sof_ipc_fw_loader_ops ipc4_loader_ops = {
diff --git a/sound/soc/sof/ipc4-pcm.c b/sound/soc/sof/ipc4-pcm.c
index 39039a647cca..85d3f390e4b2 100644
--- a/sound/soc/sof/ipc4-pcm.c
+++ b/sound/soc/sof/ipc4-pcm.c
@@ -768,10 +768,8 @@ static void sof_ipc4_build_time_info(struct snd_sof_dev *sdev, struct snd_sof_pc
info->llp_offset = offsetof(struct sof_ipc4_fw_registers, llp_evad_reading_slot) +
sdev->fw_info_box.offset;
sof_mailbox_read(sdev, info->llp_offset, &llp_slot, sizeof(llp_slot));
- if (llp_slot.node_id != dai_copier->data.gtw_cfg.node_id) {
- dev_info(sdev->dev, "no llp found, fall back to default HDA path");
+ if (llp_slot.node_id != dai_copier->data.gtw_cfg.node_id)
info->llp_offset = 0;
- }
}
static int sof_ipc4_pcm_hw_params(struct snd_soc_component *component,
diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c
index 1de3ddc50eb6..6de605a601e5 100644
--- a/sound/usb/mixer_scarlett2.c
+++ b/sound/usb/mixer_scarlett2.c
@@ -5361,9 +5361,9 @@ static int scarlett2_add_line_out_ctls(struct usb_mixer_interface *mixer)
if (private->vol_sw_hw_switch[index])
scarlett2_vol_ctl_set_writable(mixer, i, 0);
- snprintf(s, sizeof(s),
- "Line Out %02d Volume Control Playback Enum",
- i + 1);
+ scnprintf(s, sizeof(s),
+ "Line Out %02d Volume Control Playback Enum",
+ i + 1);
err = scarlett2_add_new_ctl(mixer,
&scarlett2_sw_hw_enum_ctl,
i, 1, s,
@@ -5406,8 +5406,8 @@ static int scarlett2_add_line_in_ctls(struct usb_mixer_interface *mixer)
/* Add input level (line/inst) controls */
for (i = 0; i < info->level_input_count; i++) {
- snprintf(s, sizeof(s), fmt, i + 1 + info->level_input_first,
- "Level", "Enum");
+ scnprintf(s, sizeof(s), fmt, i + 1 + info->level_input_first,
+ "Level", "Enum");
err = scarlett2_add_new_ctl(mixer, &scarlett2_level_enum_ctl,
i, 1, s, &private->level_ctls[i]);
if (err < 0)
@@ -5416,7 +5416,7 @@ static int scarlett2_add_line_in_ctls(struct usb_mixer_interface *mixer)
/* Add input pad controls */
for (i = 0; i < info->pad_input_count; i++) {
- snprintf(s, sizeof(s), fmt, i + 1, "Pad", "Switch");
+ scnprintf(s, sizeof(s), fmt, i + 1, "Pad", "Switch");
err = scarlett2_add_new_ctl(mixer, &scarlett2_pad_ctl,
i, 1, s, &private->pad_ctls[i]);
if (err < 0)
@@ -5425,8 +5425,8 @@ static int scarlett2_add_line_in_ctls(struct usb_mixer_interface *mixer)
/* Add input air controls */
for (i = 0; i < info->air_input_count; i++) {
- snprintf(s, sizeof(s), fmt, i + 1 + info->air_input_first,
- "Air", info->air_option ? "Enum" : "Switch");
+ scnprintf(s, sizeof(s), fmt, i + 1 + info->air_input_first,
+ "Air", info->air_option ? "Enum" : "Switch");
err = scarlett2_add_new_ctl(
mixer, &scarlett2_air_ctl[info->air_option],
i, 1, s, &private->air_ctls[i]);
@@ -5481,9 +5481,9 @@ static int scarlett2_add_line_in_ctls(struct usb_mixer_interface *mixer)
for (i = 0; i < info->gain_input_count; i++) {
if (i % 2) {
- snprintf(s, sizeof(s),
- "Line In %d-%d Link Capture Switch",
- i, i + 1);
+ scnprintf(s, sizeof(s),
+ "Line In %d-%d Link Capture Switch",
+ i, i + 1);
err = scarlett2_add_new_ctl(
mixer, &scarlett2_input_link_ctl,
i / 2, 1, s,
@@ -5492,30 +5492,30 @@ static int scarlett2_add_line_in_ctls(struct usb_mixer_interface *mixer)
return err;
}
- snprintf(s, sizeof(s), fmt, i + 1,
- "Gain", "Volume");
+ scnprintf(s, sizeof(s), fmt, i + 1,
+ "Gain", "Volume");
err = scarlett2_add_new_ctl(
mixer, &scarlett2_input_gain_ctl,
i, 1, s, &private->input_gain_ctls[i]);
if (err < 0)
return err;
- snprintf(s, sizeof(s), fmt, i + 1,
- "Autogain", "Switch");
+ scnprintf(s, sizeof(s), fmt, i + 1,
+ "Autogain", "Switch");
err = scarlett2_add_new_ctl(
mixer, &scarlett2_autogain_switch_ctl,
i, 1, s, &private->autogain_ctls[i]);
if (err < 0)
return err;
- snprintf(s, sizeof(s), fmt, i + 1,
- "Autogain Status", "Enum");
+ scnprintf(s, sizeof(s), fmt, i + 1,
+ "Autogain Status", "Enum");
err = scarlett2_add_new_ctl(
mixer, &scarlett2_autogain_status_ctl,
i, 1, s, &private->autogain_status_ctls[i]);
- snprintf(s, sizeof(s), fmt, i + 1,
- "Safe", "Switch");
+ scnprintf(s, sizeof(s), fmt, i + 1,
+ "Safe", "Switch");
err = scarlett2_add_new_ctl(
mixer, &scarlett2_safe_ctl,
i, 1, s, &private->safe_ctls[i]);
@@ -5902,8 +5902,8 @@ static int scarlett2_add_direct_monitor_ctls(struct usb_mixer_interface *mixer)
for (k = 0; k < private->num_mix_in; k++, index++) {
char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
- snprintf(name, sizeof(name), format,
- mix_type, 'A' + j, k + 1);
+ scnprintf(name, sizeof(name), format,
+ mix_type, 'A' + j, k + 1);
err = scarlett2_add_new_ctl(
mixer, &scarlett2_monitor_mix_ctl,