June 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH] ALSA: hda/realtek - Add quirk for MSI GE63 laptop
by Takashi Iwai 16 Jun '20

16 Jun '20

MSI GE63 laptop with ALC1220 codec requires the very same quirk (ALC1220_FIXUP_CLEVO_P950) as other MSI devices for the proper sound output. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208057 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d1b67d75f954..3ca93421e209 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2461,6 +2461,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1458, 0xa0b8, "Gigabyte AZ370-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), SND_PCI_QUIRK(0x1458, 0xa0cd, "Gigabyte X570 Aorus Master", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1458, 0xa0ce, "Gigabyte X570 Aorus Xtreme", ALC1220_FIXUP_CLEVO_P950), + SND_PCI_QUIRK(0x1462, 0x11f7, "MSI-GE63", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1228, "MSI-GP63", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1275, "MSI-GL63", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1276, "MSI-GL73", ALC1220_FIXUP_CLEVO_P950), -- 2.25.0

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[PATCH] ALSA: usb-audio: Fix potential use-after-free of streams
by Takashi Iwai 16 Jun '20

16 Jun '20

With the recent full-duplex support of implicit feedback streams, an endpoint can be still running after closing the capture stream as long as the playback stream with the sync-endpoint is running. In such a state, the URBs are still be handled and they may call retire_data_urb callback, which tries to transfer the data from the PCM buffer. Since the PCM stream gets closed, this may lead to use-after-free. This patch adds the proper clearance of the callback at stopping the capture stream for addressing the possible UAF above. Fixes: 10ce77e4817f ("ALSA: usb-audio: Add duplex sound support for USB devices using implicit feedback") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 84c0ae431936..a777d36c4f5a 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1787,6 +1787,7 @@ static int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream return 0; case SNDRV_PCM_TRIGGER_STOP: stop_endpoints(subs); + subs->data_endpoint->retire_data_urb = NULL; subs->running = 0; return 0; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: -- 2.25.0

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[PATCH v2] ASoC: rt1015: Flush DAC data before playback.
by jack.yu＠realtek.com 16 Jun '20

16 Jun '20

From: Jack Yu <jack.yu(a)realtek.com> Flush DAC data before playback. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt1015.c | 18 ++++++++++++++---- sound/soc/codecs/rt1015.h | 5 +++++ 2 files changed, 19 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 2cccb310fa96..1f61b4aa4353 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -493,7 +493,7 @@ static int rt1015_bypass_boost_put(struct snd_kcontrol *kcontrol, if (!rt1015->dac_is_used) { rt1015->bypass_boost = ucontrol->value.integer.value[0]; - if (rt1015->bypass_boost == 1) { + if (rt1015->bypass_boost == RT1015_Bypass_Boost) { snd_soc_component_write(component, RT1015_PWR4, 0x00b2); snd_soc_component_write(component, @@ -549,7 +549,7 @@ static int r1015_dac_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: rt1015->dac_is_used = 1; - if (rt1015->bypass_boost == 0) { + if (rt1015->bypass_boost == RT1015_Enable_Boost) { snd_soc_component_write(component, RT1015_SYS_RST1, 0x05f7); snd_soc_component_write(component, @@ -566,8 +566,17 @@ static int r1015_dac_event(struct snd_soc_dapm_widget *w, } break; + case SND_SOC_DAPM_POST_PMU: + if (rt1015->bypass_boost == RT1015_Bypass_Boost) { + regmap_write(rt1015->regmap, RT1015_MAN_I2C, 0x00a8); + regmap_write(rt1015->regmap, RT1015_SYS_RST1, 0x0597); + regmap_write(rt1015->regmap, RT1015_SYS_RST1, 0x05f7); + regmap_write(rt1015->regmap, RT1015_MAN_I2C, 0x0028); + } + break; + case SND_SOC_DAPM_POST_PMD: - if (rt1015->bypass_boost == 0) { + if (rt1015->bypass_boost == RT1015_Enable_Boost) { snd_soc_component_write(component, RT1015_PWR9, 0xa800); snd_soc_component_write(component, @@ -617,7 +626,8 @@ static const struct snd_soc_dapm_widget rt1015_dapm_widgets[] = { SND_SOC_DAPM_AIF_IN("AIFRX", "AIF Playback", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC_E("DAC", NULL, RT1015_PWR1, RT1015_PWR_DAC_BIT, 0, - r1015_dac_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + r1015_dac_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_OUTPUT("SPO"), }; diff --git a/sound/soc/codecs/rt1015.h b/sound/soc/codecs/rt1015.h index 8169962935a5..7bd159e8f958 100644 --- a/sound/soc/codecs/rt1015.h +++ b/sound/soc/codecs/rt1015.h @@ -368,6 +368,11 @@ enum { FIXED_ADAPTIVE, }; +enum { + RT1015_Enable_Boost = 0, + RT1015_Bypass_Boost, +}; + struct rt1015_priv { struct snd_soc_component *component; struct regmap *regmap; -- 2.25.1

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[PATCH v3] ASoC: fsl_ssi: Fix bclk calculation for mono channel
by Shengjiu Wang 16 Jun '20

16 Jun '20

For mono channel, SSI will switch to Normal mode. In Normal mode and Network mode, the Word Length Control bits control the word length divider in clock generator, which is different with I2S Master mode (the word length is fixed to 32bit), it should be the value of params_width(hw_params). The condition "slots == 2" is not good for I2S Master mode, because for Network mode and Normal mode, the slots can also be 2. Then we need to use (ssi->i2s_net & SSI_SCR_I2S_MODE_MASK) to check if it is I2S Master mode. So we refine the formula for mono channel, otherwise there will be sound issue for S24_LE. Fixes: b0a7043d5c2c ("ASoC: fsl_ssi: Caculate bit clock rate using slot number and width") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v3 - update according to Nicolin's comments changes in v2 - refine patch for Network mode and Normal mode. sound/soc/fsl/fsl_ssi.c | 13 +++++++++---- 1 file changed, 9 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index bad89b0d129e..1a2fa7f18142 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -678,8 +678,9 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, struct regmap *regs = ssi->regs; u32 pm = 999, div2, psr, stccr, mask, afreq, factor, i; unsigned long clkrate, baudrate, tmprate; - unsigned int slots = params_channels(hw_params); - unsigned int slot_width = 32; + unsigned int channels = params_channels(hw_params); + unsigned int slot_width = params_width(hw_params); + unsigned int slots = 2; u64 sub, savesub = 100000; unsigned int freq; bool baudclk_is_used; @@ -688,10 +689,14 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, /* Override slots and slot_width if being specifically set... */ if (ssi->slots) slots = ssi->slots; - /* ...but keep 32 bits if slots is 2 -- I2S Master mode */ - if (ssi->slot_width && slots != 2) + if (ssi->slot_width) slot_width = ssi->slot_width; + /* ...but force 32 bits for stereo audio using I2S Master Mode */ + if (channels == 2 && + (ssi->i2s_net & SSI_SCR_I2S_MODE_MASK) == SSI_SCR_I2S_MODE_MASTER) + slot_width = 32; + /* Generate bit clock based on the slot number and slot width */ freq = slots * slot_width * params_rate(hw_params); -- 2.21.0

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[PATCH 1/2] ALSA: hda: Make codec controlled LED support more generic
by Kai-Heng Feng 16 Jun '20

16 Jun '20

Currently, only HDA codec GPIO controlled LED class is supported, and only via platform specific quirk. There are systems that control LED via COEF instead of GPIO, and to support those systems, move the LED class registration to snd_hda_gen_add_micmute_led(), so all systems can facilitate the same interface. In addition to that, add LED_CORE_SUSPENDRESUME flag since some systems don't restore the LED properly after suspend. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/hda_generic.c | 28 ++++++++++++++++++++++++++++ sound/pci/hda/patch_realtek.c | 30 ------------------------------ 2 files changed, 28 insertions(+), 30 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index f4e9d9445e18..4242407734c0 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -4006,6 +4006,28 @@ static const struct snd_kcontrol_new micmute_led_mode_ctl = { * * Returns 0 if the hook is established or a negative error code. */ + +#if IS_REACHABLE(CONFIG_LEDS_TRIGGER_AUDIO) +static int micmute_led_set(struct led_classdev *led_cdev, + enum led_brightness brightness) +{ + struct hda_codec *codec = dev_to_hda_codec(led_cdev->dev->parent); + struct hda_gen_spec *spec = codec->spec; + + spec->micmute_led.led_mode = !brightness; + call_micmute_led_update(codec); + return 0; +} + +static struct led_classdev micmute_led_cdev = { + .name = "hda::micmute", + .max_brightness = 1, + .brightness_set_blocking = micmute_led_set, + .default_trigger = "audio-micmute", + .flags = LED_CORE_SUSPENDRESUME, +}; +#endif + int snd_hda_gen_add_micmute_led(struct hda_codec *codec, void (*hook)(struct hda_codec *)) { @@ -4019,6 +4041,12 @@ int snd_hda_gen_add_micmute_led(struct hda_codec *codec, spec->cap_sync_hook = update_micmute_led; if (!snd_hda_gen_add_kctl(spec, NULL, &micmute_led_mode_ctl)) return -ENOMEM; + +#if IS_REACHABLE(CONFIG_LEDS_TRIGGER_AUDIO) + micmute_led_cdev.brightness = ledtrig_audio_get(LED_AUDIO_MICMUTE); + if (devm_led_classdev_register(&codec->core.dev, &micmute_led_cdev)) + codec_warn(codec, "failed to register micmute LED\n"); +#endif return 0; } EXPORT_SYMBOL_GPL(snd_hda_gen_add_micmute_led); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6d73f8beadb6..cead44a6c6cd 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4109,26 +4109,6 @@ static void alc_gpio_micmute_update(struct hda_codec *codec) spec->gen.micmute_led.led_value); } -#if IS_REACHABLE(CONFIG_LEDS_TRIGGER_AUDIO) -static int micmute_led_set(struct led_classdev *led_cdev, - enum led_brightness brightness) -{ - struct hda_codec *codec = dev_to_hda_codec(led_cdev->dev->parent); - struct alc_spec *spec = codec->spec; - - alc_update_gpio_led(codec, spec->gpio_mic_led_mask, - spec->micmute_led_polarity, !!brightness); - return 0; -} - -static struct led_classdev micmute_led_cdev = { - .name = "hda::micmute", - .max_brightness = 1, - .brightness_set_blocking = micmute_led_set, - .default_trigger = "audio-micmute", -}; -#endif - /* setup mute and mic-mute GPIO bits, add hooks appropriately */ static void alc_fixup_hp_gpio_led(struct hda_codec *codec, int action, @@ -4136,9 +4116,6 @@ static void alc_fixup_hp_gpio_led(struct hda_codec *codec, unsigned int micmute_mask) { struct alc_spec *spec = codec->spec; -#if IS_REACHABLE(CONFIG_LEDS_TRIGGER_AUDIO) - int err; -#endif alc_fixup_gpio(codec, action, mute_mask | micmute_mask); @@ -4151,13 +4128,6 @@ static void alc_fixup_hp_gpio_led(struct hda_codec *codec, if (micmute_mask) { spec->gpio_mic_led_mask = micmute_mask; snd_hda_gen_add_micmute_led(codec, alc_gpio_micmute_update); - -#if IS_REACHABLE(CONFIG_LEDS_TRIGGER_AUDIO) - micmute_led_cdev.brightness = ledtrig_audio_get(LED_AUDIO_MICMUTE); - err = devm_led_classdev_register(&codec->core.dev, &micmute_led_cdev); - if (err) - codec_warn(codec, "failed to register micmute LED\n"); -#endif } } -- 2.17.1

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low volume libalsa 1.2.3
by GitHub issues - edited 16 Jun '20

16 Jun '20

alsa-project/alsa-lib issue #60 was edited from texstar: Sound is about 30% lower with alsa 1.2.3 than with alsa 1.2.2 using the same volume setting. Backing down to 1.2.2 to restore order as the natives are getting stabby. Issue URL : https://github.com/alsa-project/alsa-lib/issues/60 Repository URL: https://github.com/alsa-project/alsa-lib

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low volume libalsa 1.2.3
by GitHub issues - opened 16 Jun '20

16 Jun '20

alsa-project/alsa-lib issue #60 was opened from texstar: Sound is about 30% lower with alsa 1.2.3 than with alsa 1.2.2 using the same volume settings. Backing down to 1.2.2 to restore order as the natives are getting stabby. Issue URL : https://github.com/alsa-project/alsa-lib/issues/60 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH] Add implicit feedback quirk for SSL2.
by Laurence Tratt 16 Jun '20

16 Jun '20

As expected, this requires the same quirk as the SSL2+ in order for the clock to sync. This was suggested by, and tested on an SSL2, by Dmitry. Suggested-by: Dmitry <dpavlushko(a)gmail.com> Signed-off-by: Laurence Tratt <laurie(a)tratt.net> --- sound/usb/pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git sound/usb/pcm.c sound/usb/pcm.c index 84c0ae4319..dc1608bdf6 100644 --- sound/usb/pcm.c +++ sound/usb/pcm.c @@ -367,6 +367,7 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, ifnum = 0; goto add_sync_ep_from_ifnum; case USB_ID(0x07fd, 0x0008): /* MOTU M Series */ + case USB_ID(0x31e9, 0x0001): /* Solid State Logic SSL2 */ case USB_ID(0x31e9, 0x0002): /* Solid State Logic SSL2+ */ ep = 0x81; ifnum = 2; -- 2.27.0

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[PATCH] Add implicit feedback quirk for SSL2.
by Laurence Tratt 16 Jun '20

16 Jun '20

As expected, this requires the same quirk as the SSL2+ in order for the clock to sync. This was suggested by, and tested on an SSL2, by Dmitry. Signed-off-by: Dmitry <dpavlushko(a)gmail.com> Signed-off-by: Laurence Tratt <laurie(a)tratt.net> --- sound/usb/pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git sound/usb/pcm.c sound/usb/pcm.c index 84c0ae4319..dc1608bdf6 100644 --- sound/usb/pcm.c +++ sound/usb/pcm.c @@ -367,6 +367,7 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, ifnum = 0; goto add_sync_ep_from_ifnum; case USB_ID(0x07fd, 0x0008): /* MOTU M Series */ + case USB_ID(0x31e9, 0x0001): /* Solid State Logic SSL2 */ case USB_ID(0x31e9, 0x0002): /* Solid State Logic SSL2+ */ ep = 0x81; ifnum = 2; -- 2.27.0

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[PATCH v2] ASoC: fsl_ssi: Fix bclk calculation for mono channel
by Shengjiu Wang 16 Jun '20

16 Jun '20

For mono channel, SSI will switch to Normal mode. In Normal mode and Network mode, the Word Length Control bits control the word length divider in clock generator, which is different with I2S Master mode (the word length is fixed to 32bit), it should be the value of params_width(hw_params). The condition "slots == 2" is not good for I2S Master mode, because for Network mode and Normal mode, the slots can also be 2. Then we need to use (ssi->i2s_net & SSI_SCR_I2S_MODE_MASK) to check if it is I2S Master mode. So we refine the famula for mono channel, otherwise there will be sound issue for S24_LE. Fixes: b0a7043d5c2c ("ASoC: fsl_ssi: Caculate bit clock rate using slot number and width") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v2 - refine patch for Network mode and Normal mode. sound/soc/fsl/fsl_ssi.c | 15 +++++++++++---- 1 file changed, 11 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index bad89b0d129e..cbf67d132fda 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -678,7 +678,8 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, struct regmap *regs = ssi->regs; u32 pm = 999, div2, psr, stccr, mask, afreq, factor, i; unsigned long clkrate, baudrate, tmprate; - unsigned int slots = params_channels(hw_params); + unsigned int channels = params_channels(hw_params); + unsigned int slots; unsigned int slot_width = 32; u64 sub, savesub = 100000; unsigned int freq; @@ -688,9 +689,15 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, /* Override slots and slot_width if being specifically set... */ if (ssi->slots) slots = ssi->slots; - /* ...but keep 32 bits if slots is 2 -- I2S Master mode */ - if (ssi->slot_width && slots != 2) - slot_width = ssi->slot_width; + else + /* Apply two slots for mono channel, because DC = 2 */ + slots = (channels == 1) ? 2 : channels; + + /* ...but keep 32 bits if I2S Master mode */ + if ((ssi->i2s_net & SSI_SCR_I2S_MODE_MASK) != SSI_SCR_I2S_MODE_MASTER || + channels == 1) + slot_width = ssi->slot_width ? ssi->slot_width : + params_width(hw_params); /* Generate bit clock based on the slot number and slot width */ freq = slots * slot_width * params_rate(hw_params); -- 2.21.0

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