March 2021 - Alsa-devel - mailman.alsa-project.org

aplay: avoid any further PCM writing if in abort
by GitHub pull_request - opened 19 Mar '21

19 Mar '21

alsa-project/alsa-utils pull request #81 was opened from aditpape: Terminating stream with CTRL-C will set in_aborting flag which is used to leave any write/read loop on the ALSA device. After ending the read/write loop aplay tries to drain the stream which is not required and can also lead to malfunctions: -If user interrupts a blocked/non responsive PCM (e.g. usb uac2 gadget which does not consume data due to stream stopped by host) it will successfully terminate the write loop but will hang again in drain call. This would require to hit CTRL-C again to unblock which should be avoided. Aplay currently anyhow allows signal handler to get invoked only once. Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> Request URL : https://github.com/alsa-project/alsa-utils/pull/81 Patch URL : https://github.com/alsa-project/alsa-utils/pull/81.patch Repository URL: https://github.com/alsa-project/alsa-utils

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[PATCH] ALSA: hda: Fix spelling mistakes
by huangjianghui 19 Mar '21

19 Mar '21

Signed-off-by: huangjianghui <huangjianghui(a)uniontech.com> --- sound/hda/hdac_stream.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c index a6ed3dc35f7e..1eb8563db2df 100644 --- a/sound/hda/hdac_stream.c +++ b/sound/hda/hdac_stream.c @@ -618,7 +618,7 @@ void snd_hdac_stream_sync_trigger(struct hdac_stream *azx_dev, bool set, EXPORT_SYMBOL_GPL(snd_hdac_stream_sync_trigger); /** - * snd_hdac_stream_sync - sync with start/strop trigger operation + * snd_hdac_stream_sync - sync with start/stop trigger operation * @azx_dev: HD-audio core stream (master stream) * @start: true = start, false = stop * @streams: bit flags of streams to sync -- 2.20.1

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[PATCH] ALSA: hda/realtek: add more delay time before determine_headset_type
by Hui Wang 19 Mar '21

19 Mar '21

We found a recording issue on the headset-mic recently, sometimes users plug a headset and select headset-mic from UI, but can't record any sound from headset-mic. The root cause is the determine_headset_type() returns a wrong type, e.g. users plug a ctia type headset, but that function returns omtp type. In the past, determine_headset_type() worked well because the internal mic is connected to the codec, so the "Input Source" or "Capture Source" is internal mic by default when users plug a headset, the determine_headset_type() will not be called unless users select headset-mic from UI, when users select headset-mic, the plugging action already finished and the headset is completely plugged into the jack, so determine_headset_type() could return a correct type. But more and more machines connect the internal mic to the PCH now, and the "Input Source" is headset mic by default, when users plug a headset, the determine_headset_type() will be called immediately, if the headset is not completely plugged in, it will return a wrong type. Here add 2s delay before calling determine_headset_type(), and since there is a pop-up dialogue when users do plugging action, to avoid freezing the UI, use the deleyed_work to call that function. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 31 ++++++++++++++++++++++++------- 1 file changed, 24 insertions(+), 7 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b47504fa8dfd..1f6fc8addf3e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -127,6 +127,8 @@ struct alc_spec { unsigned int coef0; struct input_dev *kb_dev; u8 alc_mute_keycode_map[1]; + struct hda_codec *codec; + struct delayed_work headset_set_work; }; /* @@ -1160,6 +1162,7 @@ static int alc_alloc_spec(struct hda_codec *codec, hda_nid_t mixer_nid) kfree(spec); return err; } + spec->codec = codec; return 0; } @@ -5363,6 +5366,21 @@ static void alc_determine_headset_type(struct hda_codec *codec) spec->current_headset_type = is_ctia ? ALC_HEADSET_TYPE_CTIA : ALC_HEADSET_TYPE_OMTP; } +static void alc_headset_check_and_set(struct work_struct *work) +{ + struct alc_spec *spec = container_of(work, struct alc_spec, + headset_set_work.work); + struct hda_codec *codec = spec->codec; + + if (spec->current_headset_type == ALC_HEADSET_TYPE_UNKNOWN) + alc_determine_headset_type(codec); + if (spec->current_headset_type == ALC_HEADSET_TYPE_CTIA) + alc_headset_mode_ctia(codec); + else if (spec->current_headset_type == ALC_HEADSET_TYPE_OMTP) + alc_headset_mode_omtp(codec); + spec->gen.hp_jack_present = true; +} + static void alc_update_headset_mode(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -5386,6 +5404,7 @@ static void alc_update_headset_mode(struct hda_codec *codec) return; } + cancel_delayed_work_sync(&spec->headset_set_work); switch (new_headset_mode) { case ALC_HEADSET_MODE_UNPLUGGED: alc_headset_mode_unplugged(codec); @@ -5394,13 +5413,7 @@ static void alc_update_headset_mode(struct hda_codec *codec) spec->gen.hp_jack_present = false; break; case ALC_HEADSET_MODE_HEADSET: - if (spec->current_headset_type == ALC_HEADSET_TYPE_UNKNOWN) - alc_determine_headset_type(codec); - if (spec->current_headset_type == ALC_HEADSET_TYPE_CTIA) - alc_headset_mode_ctia(codec); - else if (spec->current_headset_type == ALC_HEADSET_TYPE_OMTP) - alc_headset_mode_omtp(codec); - spec->gen.hp_jack_present = true; + schedule_delayed_work(&spec->headset_set_work, msecs_to_jiffies(2000)); break; case ALC_HEADSET_MODE_MIC: alc_headset_mode_mic_in(codec, hp_pin, spec->headphone_mic_pin); @@ -5466,6 +5479,7 @@ static void alc_fixup_headset_mode(struct hda_codec *codec, spec->parse_flags |= HDA_PINCFG_HEADSET_MIC | HDA_PINCFG_HEADPHONE_MIC; break; case HDA_FIXUP_ACT_PROBE: + INIT_DELAYED_WORK(&spec->headset_set_work, alc_headset_check_and_set); alc_probe_headset_mode(codec); break; case HDA_FIXUP_ACT_INIT: @@ -5475,6 +5489,9 @@ static void alc_fixup_headset_mode(struct hda_codec *codec, } alc_update_headset_mode(codec); break; + case HDA_FIXUP_ACT_FREE: + cancel_delayed_work_sync(&spec->headset_set_work); + break; } } -- 2.25.1

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Very unstable and broken after audio device switching, reboots, multimedia app usage etc etc etc
by GitHub issues - edited 19 Mar '21

19 Mar '21

alsa-project/alsa-lib issue #127 was edited from oliiix: To me this seems to be a huge issue for years now. Running Ubuntu 20.10 very latest. I'm using an Astro A50 and two HDMI Displays for playing audio plus there's two more outputs I'm not really using. But as soon as the computer ran a little long (yes, that's actually enough to cause problems, probably due to certain apps like Unreal engine and games!), came out of sleep or I switch playback devices (which I'm doing very often and that shouldnt be ANY issue), things get seriously mixed up. Suddenly the routing goes all wrong (i select one output device, another plays instead, or just no sound at all). Sometimes "no sound" problem gets fixed by just switching devices about 10-20! times until sound is back (it either works or doesn't but that it starts working again after ten times switching devices is very weird behavior). Reinstalling as well as resetting alsa doesn't do anything to remedy no sound, restarting computer doesn't fix sound problems, only switching devices ten times does! Alsa Mixer doesn't show any controls for my A50, yet it often works even if some profiles are to low volume and the one that has good volume there I can't hear audio that comes from behind me (I have the no sound issue also on my HDMI outputs, just saying). Pavucontrol on the other hand lists my A50 but doesn't show one of my HDMI displays. I mean what is going wrong here (for years)??? I mean I found forum posts about these issues that reach back years in time -.- When will we have proper working audio where audio device switching on linux is possible without any problems and the sound doesn't just "give up" after some time? maybe someone should create one all including audio software system structure instead of several packages that are unable to work together properly... can't be that two apps (pavucontrol and pulseaudio) that work on top on alsa show different devices and settings to begin with Issue URL : https://github.com/alsa-project/alsa-lib/issues/127 Repository URL: https://github.com/alsa-project/alsa-lib

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Very unstable and broken after audio device switching, reboots, multimedia app usage etc etc etc
by GitHub issues - opened 19 Mar '21

19 Mar '21

alsa-project/alsa-lib issue #127 was opened from oliiix: To me this seems to be a huge issue for years now. Running Ubuntu 20.10 very latest. I'm using an Astro A50 and two HDMI Displays for playing audio plus there's two more outputs I'm not really using. But as soon as the computer ran a little long (yes, that's actually enough to cause problems, probably due to certain apps like Unreal engine and games!), came out of sleep or I switch playback devices (which I'm doing very often and that shouldnt be ANY issue), things get seriously mixed up. Suddenly the routing goes all wrong (i select one output device, another plays instead, or just no sound at all). Sometimes "no sound" problem gets fixed by just switching devices about 10-20! times until sound is back (it either works or doesn't but that it starts working again after ten times switching devices is very weird behavior). Reinstalling as well as resetting alsa doesn't do anything to remedy no sound, restarting computer doesn't fix sound problems, only switching devices ten times does! Alsa Mixer doesn't show any controls for my A50, yet it often works even if some profiles are to low volume and the one that has good volume there I can't hear audio that comes from behind me (I have the no sound issue also on my HDMI outputs, just saying). Pavucontrol on the other hand lists my A50 but doesn't show one of my HDMI displays. I mean what is going wrong here (for years)??? I mean I found forum posts about these issues that reach back years in time -.- When will we have proper working audio where audio device switching on linux is possible without any problems and the sound doesn't just "give up" after some time? Issue URL : https://github.com/alsa-project/alsa-lib/issues/127 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ASoC: fsl_sai: remove reset code from dai_probe
by Shengjiu Wang 19 Mar '21

19 Mar '21

From: Viorel Suman <viorel.suman(a)nxp.com> SAI software reset is done in runtime resume, there is no need to do it in fsl_sai_dai_probe. Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 7 ------- 1 file changed, 7 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 6ef2ce348232..9e7893f91882 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -711,13 +711,6 @@ static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai) struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev); unsigned int ofs = sai->soc_data->reg_offset; - /* Software Reset for both Tx and Rx */ - regmap_write(sai->regmap, FSL_SAI_TCSR(ofs), FSL_SAI_CSR_SR); - regmap_write(sai->regmap, FSL_SAI_RCSR(ofs), FSL_SAI_CSR_SR); - /* Clear SR bit to finish the reset */ - regmap_write(sai->regmap, FSL_SAI_TCSR(ofs), 0); - regmap_write(sai->regmap, FSL_SAI_RCSR(ofs), 0); - regmap_update_bits(sai->regmap, FSL_SAI_TCR1(ofs), FSL_SAI_CR1_RFW_MASK(sai->soc_data->fifo_depth), sai->soc_data->fifo_depth - FSL_SAI_MAXBURST_TX); -- 2.27.0

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[PATCH] docs: devicetree: bindings: sound: Fix a typo in the file rt5682.txt
by Bhaskar Chowdhury 18 Mar '21

18 Mar '21

s/drving/driving/ Signed-off-by: Bhaskar Chowdhury <unixbhaskar(a)gmail.com> --- Documentation/devicetree/bindings/sound/rt5682.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/rt5682.txt b/Documentation/devicetree/bindings/sound/rt5682.txt index 9c5fadb6ac82..cd8c53d8497e 100644 --- a/Documentation/devicetree/bindings/sound/rt5682.txt +++ b/Documentation/devicetree/bindings/sound/rt5682.txt @@ -44,7 +44,7 @@ Optional properties: - realtek,dmic-delay-ms : Set the delay time (ms) for the requirement of the particular DMIC. -- realtek,dmic-clk-driving-high : Set the high drving of the DMIC clock out. +- realtek,dmic-clk-driving-high : Set the high driving of the DMIC clock out. Pins on the device (for linking into audio routes) for RT5682: -- 2.26.2

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[PATCH v3] ASoC: Intel: sof_rt5682: Add ALC1015Q-VB speaker amp support
by Brent Lu 18 Mar '21

18 Mar '21

This patch adds jsl_rt5682_rt1015p which supports the RT5682 headset codec and ALC1015Q-VB speaker amplifier combination on JasperLake platform. This driver also supports ALC1015Q-CG if running in auto-mode. Following table shows the audio interface support of the two amplifiers. | ALC1015Q-CG | ALC1015Q-VB ===================================== I2C | Yes | No Auto-mode | 48K, 64fs | 16k, 32fs | 48k, 32fs | 48k, 64fs Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/sof_realtek_common.c | 105 ++++++++++++++++++ sound/soc/intel/boards/sof_realtek_common.h | 7 ++ sound/soc/intel/boards/sof_rt5682.c | 19 +++- .../intel/common/soc-acpi-intel-jsl-match.c | 13 +++ 5 files changed, 143 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index d1d28129a32b..58379393b8e4 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -457,6 +457,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH select SND_SOC_MAX98373_I2C select SND_SOC_RT1011 select SND_SOC_RT1015 + select SND_SOC_RT1015P select SND_SOC_RT5682_I2C select SND_SOC_DMIC select SND_SOC_HDAC_HDMI diff --git a/sound/soc/intel/boards/sof_realtek_common.c b/sound/soc/intel/boards/sof_realtek_common.c index f3cf73c620ba..2ec34f8df9e1 100644 --- a/sound/soc/intel/boards/sof_realtek_common.c +++ b/sound/soc/intel/boards/sof_realtek_common.c @@ -7,6 +7,7 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/soc-acpi.h> #include <sound/soc-dai.h> #include <sound/soc-dapm.h> #include <uapi/sound/asound.h> @@ -136,3 +137,107 @@ void sof_rt1011_codec_conf(struct snd_soc_card *card) card->codec_conf = rt1011_codec_confs; card->num_configs = ARRAY_SIZE(rt1011_codec_confs); } + +/* + * rt1015: i2c mode driver for ALC1015 and ALC1015Q + * rt1015p: auto-mode driver for ALC1015, ALC1015Q, and ALC1015Q-VB + * + * For stereo output, there are always two amplifiers on the board. + * However, the ACPI implements only one device instance (UID=0) if they + * are sharing the same enable pin. The code will detect the number of + * device instance and use corresponding DAPM structures for + * initialization. + */ +static const struct snd_soc_dapm_route rt1015p_1dev_dapm_routes[] = { + /* speaker */ + { "Left Spk", NULL, "Speaker" }, + { "Right Spk", NULL, "Speaker" }, +}; + +static const struct snd_soc_dapm_route rt1015p_2dev_dapm_routes[] = { + /* speaker */ + { "Left Spk", NULL, "Left Speaker" }, + { "Right Spk", NULL, "Right Speaker" }, +}; + +static struct snd_soc_codec_conf rt1015p_codec_confs[] = { + { + .dlc = COMP_CODEC_CONF(RT1015P_DEV0_NAME), + .name_prefix = "Left", + }, + { + .dlc = COMP_CODEC_CONF(RT1015P_DEV1_NAME), + .name_prefix = "Right", + }, +}; + +static struct snd_soc_dai_link_component rt1015p_dai_link_components[] = { + { + .name = RT1015P_DEV0_NAME, + .dai_name = RT1015P_CODEC_DAI, + }, + { + .name = RT1015P_DEV1_NAME, + .dai_name = RT1015P_CODEC_DAI, + }, +}; + +static int rt1015p_get_num_codecs(void) +{ + static int dev_num; + + if (dev_num) + return dev_num; + + if (!acpi_dev_present("RTL1015", "1", -1)) + dev_num = 1; + else + dev_num = 2; + + return dev_num; +} + +static int rt1015p_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + /* reserved for debugging purpose */ + + return 0; +} + +static const struct snd_soc_ops rt1015p_ops = { + .hw_params = rt1015p_hw_params, +}; + +static int rt1015p_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + if (rt1015p_get_num_codecs() == 1) + ret = snd_soc_dapm_add_routes(&card->dapm, rt1015p_1dev_dapm_routes, + ARRAY_SIZE(rt1015p_1dev_dapm_routes)); + else + ret = snd_soc_dapm_add_routes(&card->dapm, rt1015p_2dev_dapm_routes, + ARRAY_SIZE(rt1015p_2dev_dapm_routes)); + if (ret) + dev_err(rtd->dev, "Speaker map addition failed: %d\n", ret); + return ret; +} + +void sof_rt1015p_dai_link(struct snd_soc_dai_link *link) +{ + link->codecs = rt1015p_dai_link_components; + link->num_codecs = rt1015p_get_num_codecs(); + link->init = rt1015p_init; + link->ops = &rt1015p_ops; +} + +void sof_rt1015p_codec_conf(struct snd_soc_card *card) +{ + if (rt1015p_get_num_codecs() == 1) + return; + + card->codec_conf = rt1015p_codec_confs; + card->num_configs = ARRAY_SIZE(rt1015p_codec_confs); +} diff --git a/sound/soc/intel/boards/sof_realtek_common.h b/sound/soc/intel/boards/sof_realtek_common.h index 87cb3812b926..cb0b49b2855c 100644 --- a/sound/soc/intel/boards/sof_realtek_common.h +++ b/sound/soc/intel/boards/sof_realtek_common.h @@ -21,4 +21,11 @@ void sof_rt1011_dai_link(struct snd_soc_dai_link *link); void sof_rt1011_codec_conf(struct snd_soc_card *card); +#define RT1015P_CODEC_DAI "HiFi" +#define RT1015P_DEV0_NAME "RTL1015:00" +#define RT1015P_DEV1_NAME "RTL1015:01" + +void sof_rt1015p_dai_link(struct snd_soc_dai_link *link); +void sof_rt1015p_codec_conf(struct snd_soc_card *card); + #endif /* __SOF_REALTEK_COMMON_H */ diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 55505e207bc0..f4b898c1719f 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -45,8 +45,9 @@ #define SOF_RT1011_SPEAKER_AMP_PRESENT BIT(13) #define SOF_RT1015_SPEAKER_AMP_PRESENT BIT(14) #define SOF_RT1015_SPEAKER_AMP_100FS BIT(15) -#define SOF_MAX98373_SPEAKER_AMP_PRESENT BIT(16) -#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(17) +#define SOF_RT1015P_SPEAKER_AMP_PRESENT BIT(16) +#define SOF_MAX98373_SPEAKER_AMP_PRESENT BIT(17) +#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(18) /* Default: MCLK on, MCLK 19.2M, SSP0 */ static unsigned long sof_rt5682_quirk = SOF_RT5682_MCLK_EN | @@ -723,6 +724,8 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, links[id].num_codecs = ARRAY_SIZE(rt1015_components); links[id].init = speaker_codec_init_lr; links[id].ops = &sof_rt1015_ops; + } else if (sof_rt5682_quirk & SOF_RT1015P_SPEAKER_AMP_PRESENT) { + sof_rt1015p_dai_link(&links[id]); } else if (sof_rt5682_quirk & SOF_MAX98373_SPEAKER_AMP_PRESENT) { links[id].codecs = max_98373_components; @@ -851,6 +854,8 @@ static int sof_audio_probe(struct platform_device *pdev) sof_max98373_codec_conf(&sof_audio_card_rt5682); else if (sof_rt5682_quirk & SOF_RT1011_SPEAKER_AMP_PRESENT) sof_rt1011_codec_conf(&sof_audio_card_rt5682); + else if (sof_rt5682_quirk & SOF_RT1015P_SPEAKER_AMP_PRESENT) + sof_rt1015p_codec_conf(&sof_audio_card_rt5682); dai_links = sof_card_dai_links_create(&pdev->dev, ssp_codec, ssp_amp, dmic_be_num, hdmi_num); @@ -940,6 +945,15 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_SSP_AMP(1) | SOF_RT5682_NUM_HDMIDEV(4)), }, + { + .name = "jsl_rt5682_rt1015p", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_MCLK_24MHZ | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_RT1015P_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1)), + }, { } }; @@ -966,3 +980,4 @@ MODULE_ALIAS("platform:tgl_max98373_rt5682"); MODULE_ALIAS("platform:jsl_rt5682_max98360a"); MODULE_ALIAS("platform:cml_rt1015_rt5682"); MODULE_ALIAS("platform:tgl_rt1011_rt5682"); +MODULE_ALIAS("platform:jsl_rt5682_rt1015p"); diff --git a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c index 52238db0bcb5..73fe4f89a82d 100644 --- a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c @@ -19,6 +19,11 @@ static struct snd_soc_acpi_codecs rt1015_spk = { .codecs = {"10EC1015"} }; +static struct snd_soc_acpi_codecs rt1015p_spk = { + .num_codecs = 1, + .codecs = {"RTL1015"} +}; + static struct snd_soc_acpi_codecs mx98360a_spk = { .num_codecs = 1, .codecs = {"MX98360A"} @@ -52,6 +57,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_jsl_machines[] = { .quirk_data = &rt1015_spk, .sof_tplg_filename = "sof-jsl-rt5682-rt1015.tplg", }, + { + .id = "10EC5682", + .drv_name = "jsl_rt5682_rt1015p", + .sof_fw_filename = "sof-jsl.ri", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &rt1015p_spk, + .sof_tplg_filename = "sof-jsl-rt5682-rt1015.tplg", + }, { .id = "10EC5682", .drv_name = "jsl_rt5682_max98360a", -- 2.17.1

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[PATCH 00/23] ASoC: codecs: remove cppcheck warnings
by Pierre-Louis Bossart 18 Mar '21

18 Mar '21

Lots of small fixes in various codec drivers that should have no functional impact. Pierre-Louis Bossart (23): ASoC: ab8500-codec: remove useless structure ASoC: ad1836: remove useless return ASoC: adau1977: remove useless return ASoC: cros_ec_codec: remove null pointer dereference warning ASoC: cx2070x: remove useless assignment ASoC: cx2070x: remove duplicate else branch ASoC: da7219-aad: remove useless initialization ASoC: hdac_hdmi: remove useless initializations ASoC: hdac_hdmi: align function arguments ASoC: hdmi-codec: remove useless initialization ASoC: hdmi-codec: remove unused spk_mask member ASoC: max98090: remove useless assignment ASoC: mt6358: remove useless initializations ASoC: mt6359: remove useless assignment ASoC: nau8825: remove useless assignment ASoC: pcm1681: remove useless assignment ASoC: sigmadsp: align function prototype ASoC: sti-sas: remove unused struct members ASoC: tas2562: remove useless assignment ASoC: tas2562: remove warning on return value ASoC: tas2770: remove useless initialization ASoC: tlv320dac33: clarify expression ASoC: tscs454: remove useless test on PLL disable sound/soc/codecs/ab8500-codec.c | 7 ------- sound/soc/codecs/ad1836.c | 2 -- sound/soc/codecs/adau1977.c | 2 -- sound/soc/codecs/cros_ec_codec.c | 2 +- sound/soc/codecs/cx2072x.c | 11 ++++------- sound/soc/codecs/da7219-aad.c | 2 +- sound/soc/codecs/hdac_hdmi.c | 14 +++++++------- sound/soc/codecs/hdac_hdmi.h | 2 +- sound/soc/codecs/hdmi-codec.c | 3 +-- sound/soc/codecs/max98090.c | 2 +- sound/soc/codecs/mt6358.c | 4 ++-- sound/soc/codecs/mt6359.c | 4 ++-- sound/soc/codecs/nau8825.c | 2 +- sound/soc/codecs/pcm1681.c | 2 +- sound/soc/codecs/sigmadsp.h | 2 +- sound/soc/codecs/sti-sas.c | 3 --- sound/soc/codecs/tas2562.c | 3 +-- sound/soc/codecs/tas2770.c | 2 +- sound/soc/codecs/tlv320dac33.c | 2 +- sound/soc/codecs/tscs454.c | 7 ++++++- 20 files changed, 32 insertions(+), 46 deletions(-) -- 2.25.1

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[PATCH] ASoC: Intel: Skylake: skl-topology: fix -frame-larger-than
by Nick Desaulniers 18 Mar '21

18 Mar '21

Fixes: sound/soc/intel/skylake/skl-topology.c:3613:13: warning: stack frame size of 1304 bytes in function 'skl_tplg_complete' [-Wframe-larger-than=] struct snd_ctl_elem_value is 1224 bytes in my configuration. Heap allocate it, then free it within the current frame. Signed-off-by: Nick Desaulniers <nick.desaulniers(a)gmail.com> --- sound/soc/intel/skylake/skl-topology.c | 16 ++++++++++------ 1 file changed, 10 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index ae466cd59292..cdc916c91301 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -3613,10 +3613,15 @@ static int skl_manifest_load(struct snd_soc_component *cmpnt, int index, static void skl_tplg_complete(struct snd_soc_component *component) { struct snd_soc_dobj *dobj; - struct snd_soc_acpi_mach *mach = - dev_get_platdata(component->card->dev); + struct snd_soc_acpi_mach *mach; + struct snd_ctl_elem_value *val; int i; + val = kzalloc(sizeof(*val), GFP_KERNEL); + if (!val) + return; + + mach = dev_get_platdata(component->card->dev); list_for_each_entry(dobj, &component->dobj_list, list) { struct snd_kcontrol *kcontrol = dobj->control.kcontrol; struct soc_enum *se = @@ -3631,14 +3636,13 @@ static void skl_tplg_complete(struct snd_soc_component *component) sprintf(chan_text, "c%d", mach->mach_params.dmic_num); for (i = 0; i < se->items; i++) { - struct snd_ctl_elem_value val; - if (strstr(texts[i], chan_text)) { - val.value.enumerated.item[0] = i; - kcontrol->put(kcontrol, &val); + val->value.enumerated.item[0] = i; + kcontrol->put(kcontrol, val); } } } + kfree(val); } static struct snd_soc_tplg_ops skl_tplg_ops = { base-commit: 65f0d2414b7079556fbbcc070b3d1c9f9587606d prerequisite-patch-id: 4d05aad8c2b50c0c3b4447dd842abe8b1b840927 -- 2.25.1

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