diff options
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, |