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

[PATCH alsa-lib] control: ctlparse - use type-specific bound on element count
by John Keeping 16 Jun '20

16 Jun '20

Using a fixed bound of 128 means that too many values may be set for an INTEGER64 type and that any elements past 128 are out of reach for BYTE type controls. Derive the maximum number of elements from the type so that the full range is parsed for all types. Signed-off-by: John Keeping <john(a)metanate.com> --- src/control/ctlparse.c | 24 +++++++++++++++++++++++- 1 file changed, 23 insertions(+), 1 deletion(-) diff --git a/src/control/ctlparse.c b/src/control/ctlparse.c index 1ac31bb9..ee1e0602 100644 --- a/src/control/ctlparse.c +++ b/src/control/ctlparse.c @@ -304,6 +304,25 @@ static int get_ctl_enum_item_index(snd_ctl_t *handle, return -1; } +static unsigned int get_ctl_type_max_elements(snd_ctl_elem_type_t type) +{ + struct snd_ctl_elem_value value; + + switch (type) { + case SND_CTL_ELEM_TYPE_BOOLEAN: + case SND_CTL_ELEM_TYPE_INTEGER: + return ARRAY_SIZE(value.value.integer.value); + case SND_CTL_ELEM_TYPE_INTEGER64: + return ARRAY_SIZE(value.value.integer64.value); + case SND_CTL_ELEM_TYPE_ENUMERATED: + return ARRAY_SIZE(value.value.enumerated.item); + case SND_CTL_ELEM_TYPE_BYTES: + return ARRAY_SIZE(value.value.bytes.data); + default: + return 0; + } +} + /** * \brief parse ASCII string as CTL element value * \param handle CTL handle @@ -331,8 +350,11 @@ int snd_ctl_ascii_value_parse(snd_ctl_t *handle, type = snd_ctl_elem_info_get_type(info); count = snd_ctl_elem_info_get_count(info); snd_ctl_elem_value_set_id(dst, &myid); + + if (count > get_ctl_type_max_elements(type)) + count = get_ctl_type_max_elements(type); - for (idx = 0; idx < count && idx < 128 && ptr && *ptr; idx++) { + for (idx = 0; idx < count && ptr && *ptr; idx++) { if (*ptr == ',') goto skip; switch (type) { -- 2.27.0

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[RESENDx2][PATCH] sound: qcom: Kconfig: Tweak dependencies on SND_SOC_SDM845
by John Stultz 16 Jun '20

16 Jun '20

CROS_EC isn't strictly required for audio to work on other SDM845 platforms (like the Dragonboard 845c). So lets remove the dependency and select the related CROS_EC options if CROS_EC is already enabled. Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Cc: Rohit kumar <rohitkr(a)codeaurora.org> Cc: Patrick Lai <plai(a)codeaurora.org> Cc: Banajit Goswami <bgoswami(a)codeaurora.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: John Stultz <john.stultz(a)linaro.org> --- sound/soc/qcom/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index f51b28d1b94d..0434f2c7d746 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -99,12 +99,12 @@ config SND_SOC_MSM8996 config SND_SOC_SDM845 tristate "SoC Machine driver for SDM845 boards" - depends on QCOM_APR && CROS_EC && I2C && SOUNDWIRE + depends on QCOM_APR && I2C && SOUNDWIRE select SND_SOC_QDSP6 select SND_SOC_QCOM_COMMON select SND_SOC_RT5663 select SND_SOC_MAX98927 - select SND_SOC_CROS_EC_CODEC + select SND_SOC_CROS_EC_CODEC if CROS_EC help To add support for audio on Qualcomm Technologies Inc. SDM845 SoC-based systems. -- 2.17.1

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pcm: dmic: assembly: add x32 support.
by GitHub pull_request - opened 16 Jun '20

16 Jun '20

alsa-project/alsa-lib pull request #61 was opened from hvdijk: x32 is the x86_64 ABI that uses 32-bit pointers, so requires loading addresses into edi/esi/ebx rather than rdi/rsi/rbx. Note that instructions such as movl %eax, (%rdi) do not require updating, as loading an address into %edi zeroes the high bits, causing the full %rdi register to hold a valid address. Signed-off-by: Harald van Dijk <harald(a)gigawatt.nl> Without this, even aplay segfaults when building with `-mx32`. Request URL : https://github.com/alsa-project/alsa-lib/pull/61 Patch URL : https://github.com/alsa-project/alsa-lib/pull/61.patch Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH] soundwire: Replace 'objs' by 'y'
by Vinod Koul 16 Jun '20

16 Jun '20

`-objs` is designed for building host programs, change to `-y`, more straightforward for device drivers. See Documentation/kbuild/makefiles.rst Signed-off-by: Vinod Koul <vkoul(a)kernel.org> --- drivers/soundwire/Makefile | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/drivers/soundwire/Makefile b/drivers/soundwire/Makefile index b5871612613b..7c53ffae9f50 100644 --- a/drivers/soundwire/Makefile +++ b/drivers/soundwire/Makefile @@ -4,22 +4,22 @@ # #Bus Objs -soundwire-bus-objs := bus_type.o bus.o master.o slave.o mipi_disco.o stream.o \ +soundwire-bus-y := bus_type.o bus.o master.o slave.o mipi_disco.o stream.o \ sysfs_slave.o sysfs_slave_dpn.o obj-$(CONFIG_SOUNDWIRE) += soundwire-bus.o ifdef CONFIG_DEBUG_FS -soundwire-bus-objs += debugfs.o +soundwire-bus-y += debugfs.o endif #Cadence Objs -soundwire-cadence-objs := cadence_master.o +soundwire-cadence-y := cadence_master.o obj-$(CONFIG_SOUNDWIRE_CADENCE) += soundwire-cadence.o #Intel driver -soundwire-intel-objs := intel.o intel_init.o +soundwire-intel-y := intel.o intel_init.o obj-$(CONFIG_SOUNDWIRE_INTEL) += soundwire-intel.o #Qualcomm driver -soundwire-qcom-objs := qcom.o +soundwire-qcom-y := qcom.o obj-$(CONFIG_SOUNDWIRE_QCOM) += soundwire-qcom.o -- 2.26.2

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[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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