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

[PATCH] ASoC: da732x: Drop no-op remove function
by Uwe Kleine-König 27 May '22

27 May '22

A remove callback that just returns 0 is equivalent to no callback at all as can be seen in i2c_device_remove(). So simplify accordingly. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/soc/codecs/da732x.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index 42d6a3fc3af5..267936015cb8 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -1547,11 +1547,6 @@ static int da732x_i2c_probe(struct i2c_client *i2c, return ret; } -static int da732x_i2c_remove(struct i2c_client *client) -{ - return 0; -} - static const struct i2c_device_id da732x_i2c_id[] = { { "da7320", 0}, { } @@ -1563,7 +1558,6 @@ static struct i2c_driver da732x_i2c_driver = { .name = "da7320", }, .probe = da732x_i2c_probe, - .remove = da732x_i2c_remove, .id_table = da732x_i2c_id, }; base-commit: 4b0986a3613c92f4ec1bdc7f60ec66fea135991f -- 2.36.1

2 1

[PATCH] ASoC: fsl_xcvr:Fix unbalanced pm_runtime_enable in fsl_xcvr_probe
by zhangqilong 27 May '22

27 May '22

Add missing pm_runtime_disable() when probe error out. It could avoid pm_runtime implementation complains when removing and probing again the driver. Fix:c590fa80b3928 ("ASoC: fsl_xcvr: register platform component before registering cpu dai") Signed-off-by: Zhang Qilong <zhangqilong3(a)huawei.com> --- sound/soc/fsl/fsl_xcvr.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index d0556c79fdb1..907435efe121 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -1228,6 +1228,7 @@ static int fsl_xcvr_probe(struct platform_device *pdev) */ ret = devm_snd_dmaengine_pcm_register(dev, NULL, 0); if (ret) { + pm_runtime_disable(dev); dev_err(dev, "failed to pcm register\n"); return ret; } @@ -1235,6 +1236,7 @@ static int fsl_xcvr_probe(struct platform_device *pdev) ret = devm_snd_soc_register_component(dev, &fsl_xcvr_comp, &fsl_xcvr_dai, 1); if (ret) { + pm_runtime_disable(dev); dev_err(dev, "failed to register component %s\n", fsl_xcvr_comp.name); } -- 2.31.1

2 1

[PATCH] ASoC: lm49453: Drop no-op remove function
by Uwe Kleine-König 27 May '22

27 May '22

A remove callback that just returns 0 is equivalent to no callback at all as can be seen in i2c_device_remove(). So simplify accordingly. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/soc/codecs/lm49453.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index 973d781b4b6a..86cead5022c0 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -1443,11 +1443,6 @@ static int lm49453_i2c_probe(struct i2c_client *i2c, return ret; } -static int lm49453_i2c_remove(struct i2c_client *client) -{ - return 0; -} - static const struct i2c_device_id lm49453_i2c_id[] = { { "lm49453", 0 }, { } @@ -1459,7 +1454,6 @@ static struct i2c_driver lm49453_i2c_driver = { .name = "lm49453", }, .probe = lm49453_i2c_probe, - .remove = lm49453_i2c_remove, .id_table = lm49453_i2c_id, }; base-commit: 4b0986a3613c92f4ec1bdc7f60ec66fea135991f -- 2.36.1

1 1

[PATCH] ASoC: Intel: common: fix typo for tplg naming
by David Lin 26 May '22

26 May '22

Correct typo form sof-adl-mx98360a-nau8825.tplg to sof-adl-max98360a-nau8825.tplg. The reason is tplg naming without naming limitaion of length. It will be consistency with sof topology generation. Signed-off-by: David Lin <CTLIN0(a)nuvoton.com> --- sound/soc/intel/common/soc-acpi-intel-adl-match.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index e6e52c7b6803..c1385161cdc8 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -453,7 +453,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .drv_name = "adl_mx98360a_nau8825", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &adl_max98360a_amp, - .sof_tplg_filename = "sof-adl-mx98360a-nau8825.tplg", + .sof_tplg_filename = "sof-adl-max98360a-nau8825.tplg", }, { .id = "RTL5682", -- 2.25.1

3 2

[PATCH] ASoC: rt5640: Do not manipulate pin "Platform Clock" if the "Platform Clock" is not in the DAPM
by Oder Chiou 26 May '22

26 May '22

The pin "Platform Clock" was only used by the Intel Byt CR platform. In the others, the error log will be informed. The patch will set the flag to avoid the pin "Platform Clock" manipulated by the other platforms. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> Reported-by: Sameer Pujar <spujar(a)nvidia.com> --- sound/soc/codecs/rt5640.c | 11 +++++++++-- sound/soc/codecs/rt5640.h | 2 ++ sound/soc/intel/boards/bytcr_rt5640.c | 2 ++ 3 files changed, 13 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 12da2bea1a7b..69c80d80ed9d 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2094,12 +2094,14 @@ EXPORT_SYMBOL_GPL(rt5640_sel_asrc_clk_src); void rt5640_enable_micbias1_for_ovcd(struct snd_soc_component *component) { struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + struct rt5640_priv *rt5640 = snd_soc_component_get_drvdata(component); snd_soc_dapm_mutex_lock(dapm); snd_soc_dapm_force_enable_pin_unlocked(dapm, "LDO2"); snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS1"); /* OVCD is unreliable when used with RCCLK as sysclk-source */ - snd_soc_dapm_force_enable_pin_unlocked(dapm, "Platform Clock"); + if (rt5640->use_platform_clock) + snd_soc_dapm_force_enable_pin_unlocked(dapm, "Platform Clock"); snd_soc_dapm_sync_unlocked(dapm); snd_soc_dapm_mutex_unlock(dapm); } @@ -2108,9 +2110,11 @@ EXPORT_SYMBOL_GPL(rt5640_enable_micbias1_for_ovcd); void rt5640_disable_micbias1_for_ovcd(struct snd_soc_component *component) { struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + struct rt5640_priv *rt5640 = snd_soc_component_get_drvdata(component); snd_soc_dapm_mutex_lock(dapm); - snd_soc_dapm_disable_pin_unlocked(dapm, "Platform Clock"); + if (rt5640->use_platform_clock) + snd_soc_dapm_disable_pin_unlocked(dapm, "Platform Clock"); snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS1"); snd_soc_dapm_disable_pin_unlocked(dapm, "LDO2"); snd_soc_dapm_sync_unlocked(dapm); @@ -2535,6 +2539,9 @@ static void rt5640_enable_jack_detect(struct snd_soc_component *component, rt5640->jd_gpio_irq_requested = true; } + if (jack_data && jack_data->use_platform_clock) + rt5640->use_platform_clock = jack_data->use_platform_clock; + ret = request_irq(rt5640->irq, rt5640_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "rt5640", rt5640); diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index 9e49b9a0ccaa..505c93514051 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -2155,11 +2155,13 @@ struct rt5640_priv { bool jd_inverted; unsigned int ovcd_th; unsigned int ovcd_sf; + bool use_platform_clock; }; struct rt5640_set_jack_data { int codec_irq_override; struct gpio_desc *jd_gpio; + bool use_platform_clock; }; int rt5640_dmic_enable(struct snd_soc_component *component, diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 7b948a219177..ed9fa1728722 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -1191,12 +1191,14 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) { struct snd_soc_card *card = runtime->card; struct byt_rt5640_private *priv = snd_soc_card_get_drvdata(card); + struct rt5640_set_jack_data *jack_data = &priv->jack_data; struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component; const struct snd_soc_dapm_route *custom_map = NULL; int num_routes = 0; int ret; card->dapm.idle_bias_off = true; + jack_data->use_platform_clock = true; /* Start with RC clk for jack-detect (we disable MCLK below) */ if (byt_rt5640_quirk & BYT_RT5640_MCLK_EN) -- 2.35.3

3 2

Update mapping for the new Gigabyte Aorus Pro WiFi rev 1.2
by GitHub pull_request - edited 26 May '22

26 May '22

alsa-project/alsa-ucm-conf pull request #156 was edited from 13r0ck: Requires kernel patch from: https://bugzilla.kernel.org/attachment.cgi?id=301036&action=diff This doesn't fix the front mic for some reason (any ideas please?)~~, as well I am not sure if this will cause a regression for the 1.1 board~~ Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/156 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/156.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ALSA: usb-audio: Cancel pending work at closing a MIDI substream
by Takashi Iwai 25 May '22

25 May '22

At closing a USB MIDI output substream, there might be still a pending work, which would eventually access the rawmidi runtime object that is being released. For fixing the race, make sure to cancel the pending work at closing. Reported-by: syzbot+6912c9592caca7ca0e7d(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/000000000000e7e75005dfd07cf6@google.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/midi.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 7c6ca2b433a5..344fbeadf161 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -1145,6 +1145,9 @@ static int snd_usbmidi_output_open(struct snd_rawmidi_substream *substream) static int snd_usbmidi_output_close(struct snd_rawmidi_substream *substream) { + struct usbmidi_out_port *port = substream->runtime->private_data; + + cancel_work_sync(&port->ep->work); return substream_open(substream, 0, 0); } -- 2.35.3

1 0

[PATCH] ASoC: stm32: sai: Use of_device_get_match_data() to simplify code
by Tang Bin 25 May '22

25 May '22

Retrieve of match data, it's better and cleaner to use 'of_device_get_match_data' over 'of_match_device'. Signed-off-by: Tang Bin <tangbin(a)cmss.chinamobile.com> --- sound/soc/stm/stm32_sai_sub.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index dd636af81..d300605a2 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -1500,7 +1500,6 @@ static int stm32_sai_sub_parse_of(struct platform_device *pdev, static int stm32_sai_sub_probe(struct platform_device *pdev) { struct stm32_sai_sub_data *sai; - const struct of_device_id *of_id; const struct snd_dmaengine_pcm_config *conf = &stm32_sai_pcm_config; int ret; @@ -1508,10 +1507,9 @@ static int stm32_sai_sub_probe(struct platform_device *pdev) if (!sai) return -ENOMEM; - of_id = of_match_device(stm32_sai_sub_ids, &pdev->dev); - if (!of_id) + sai->id = (uintptr_t)of_device_get_match_data(&pdev->dev); + if (!sai->id) return -EINVAL; - sai->id = (uintptr_t)of_id->data; sai->pdev = pdev; mutex_init(&sai->ctrl_lock); -- 2.20.1.windows.1

4 6

[PATCH] ALSA: usb-audio: Add mixer mapping for Gigabyte B450/550 Mobos
by Takashi Iwai 25 May '22

25 May '22

This patch implements a static mapping for Gigabyte B450/550 Mobos so that the mixer elements appear reasonably and jack detections work properly. Reported-and-tested-by: Brock Szuszczewicz <brock(a)system76.com> BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=215988 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/mixer_maps.c | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index 7ef7a8abcc2b..3c795675f048 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -439,6 +439,31 @@ static const struct usbmix_name_map msi_mpg_x570s_carbon_max_wifi_alc4080_map[] {} }; +/* Gigabyte B450/550 Mobo */ +static const struct usbmix_name_map gigabyte_b450_map[] = { + { 24, NULL }, /* OT, IEC958?, disabled */ + { 21, "Speaker" }, /* OT */ + { 29, "Speaker Playback" }, /* FU */ + { 22, "Headphone" }, /* OT */ + { 30, "Headphone Playback" }, /* FU */ + { 11, "Line" }, /* IT */ + { 27, "Line Capture" }, /* FU */ + { 12, "Mic" }, /* IT */ + { 28, "Mic Capture" }, /* FU */ + { 9, "Front Mic" }, /* IT */ + { 25, "Front Mic Capture" }, /* FU */ + {} +}; + +static const struct usbmix_connector_map gigabyte_b450_connector_map[] = { + { 13, 21 }, /* Speaker */ + { 14, 22 }, /* Headphone */ + { 19, 11 }, /* Line */ + { 20, 12 }, /* Mic */ + { 17, 9 }, /* Front Mic */ + {} +}; + /* * Control map entries */ @@ -581,6 +606,11 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = { .map = trx40_mobo_map, .connector_map = trx40_mobo_connector_map, }, + { /* Gigabyte B450/550 Mobo */ + .id = USB_ID(0x0414, 0xa00d), + .map = gigabyte_b450_map, + .connector_map = gigabyte_b450_connector_map, + }, { /* ASUS ROG Zenith II */ .id = USB_ID(0x0b05, 0x1916), .map = asus_rog_map, -- 2.35.3

1 0

conf: Use ino64_t to save and compare inode numbers
by GitHub pull_request - opened 25 May '22

25 May '22

alsa-project/alsa-lib pull request #231 was opened from smcv: On 32-bit platforms when not using the large-file-support ABI, struct stat64 contains ino64_t which is 64-bit, while ino_t is only 32-bit. snd_config_update_r() checks whether a file has been replaced by saving the ino member of a struct stat64 and comparing it with a previously-saved inode number. On 32-bit platforms, assigning the 64-bit member of struct stat64 to a 32-bit member of struct finfo will truncate it modulo 1<<32, which could conceivably result in libasound not reloading configuration when it should (although the inode number space is large enough to make this failure mode highly unlikely). --- This is from source code inspection while looking at #223, I haven't seen a genuine user-observable bug result from this. cc @dos1, @perexg, @bertogg Request URL : https://github.com/alsa-project/alsa-lib/pull/231 Patch URL : https://github.com/alsa-project/alsa-lib/pull/231.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0