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[PATCH 1/2] ASoC: bindings: fsl-asoc-card: Add compatible string for MQS
by Shengjiu Wang 17 Jun '20

17 Jun '20

Add compatible string "fsl,imx-audio-mqs" for MQS Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Documentation/devicetree/bindings/sound/fsl-asoc-card.txt | 2 ++ 1 file changed, 2 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt index c60a5732d29c..6417c5dffa9a 100644 --- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt +++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt @@ -34,6 +34,8 @@ The compatible list for this generic sound card currently: "fsl,imx-audio-wm8960" + "fsl,imx-audio-mqs" + Required properties: - compatible : Contains one of entries in the compatible list. -- 2.21.0

3 3

[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

2 1

[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

3 4

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

1 0

[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

1 0

[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

1 0

[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

1 0

[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

2 1

[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

3 2

[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

2 3

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

1 0

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

1 0

[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

1 0

[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

1 0

[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

3 4

[PATCH 0/3] ASoC: fsl_easrc: Fix several warnings
by Shengjiu Wang 16 Jun '20

16 Jun '20

Fix several warnings with "make W=1" Shengjiu Wang (3): ASoC: fsl_easrc: Fix -Wmissing-prototypes warning ASoC: fsl_easrc: Fix -Wunused-but-set-variable ASoC: fsl_easrc: Fix "Function parameter not described" warnings sound/soc/fsl/fsl_easrc.c | 42 +++++++++++++++++---------------------- 1 file changed, 18 insertions(+), 24 deletions(-) -- 2.21.0

3 7

[PATCH v2 0/2] ASoC: mediatek: mt6358: support DMIC one-wire mode
by Jiaxin Yu 16 Jun '20

16 Jun '20

v2 changes: 1. Uses a DT property to select DMIC mode instead of a mixer control. v1 changes: 1. Uses a mixer control to select DMIC mode. 2. patchwork list: https://patchwork.kernel.org/patch/11578309 Jiaxin Yu (2): ASoC: mediatek: mt6358: support DMIC one-wire mode ASoC: dt-bindings: mediatek: mt6358: add dmic-mode property Documentation/devicetree/bindings/sound/mt6358.txt | 6 ++++++ sound/soc/codecs/mt6358.c | 23 +++++++++++++++++++++- 2 files changed, 28 insertions(+), 1 deletion(-) -- 1.8.1.1.dirty

3 4

[PATCH] ASoC: img-parallel-out: Fix a reference count leak
by wu000273＠umn.edu 16 Jun '20

16 Jun '20

From: Qiushi Wu <wu000273(a)umn.edu> pm_runtime_get_sync() increments the runtime PM usage counter even when it returns an error code, causing incorrect ref count if pm_runtime_put_noidle() is not called in error handling paths. Thus call pm_runtime_put_noidle() if pm_runtime_get_sync() fails. Signed-off-by: Qiushi Wu <wu000273(a)umn.edu> --- sound/soc/img/img-parallel-out.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/img/img-parallel-out.c b/sound/soc/img/img-parallel-out.c index 5ddbe3a31c2e..4da49a42e854 100644 --- a/sound/soc/img/img-parallel-out.c +++ b/sound/soc/img/img-parallel-out.c @@ -163,8 +163,10 @@ static int img_prl_out_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) } ret = pm_runtime_get_sync(prl->dev); - if (ret < 0) + if (ret < 0) { + pm_runtime_put_noidle(prl->dev); return ret; + } reg = img_prl_out_readl(prl, IMG_PRL_OUT_CTL); reg = (reg & ~IMG_PRL_OUT_CTL_EDGE_MASK) | control_set; -- 2.17.1

2 1

[V7 PATCH] dt-bindings: Added device tree binding for max98390
by Steve Lee 16 Jun '20

16 Jun '20

Add DT binding of max98390 amplifier driver. Signed-off-by: Steve Lee <steves.lee(a)maximintegrated.com> --- Changed since V6: * Re-confirm yaml dt binding check * Add minimum and maximum value for each temperature_calib and r0_calib * Add maxim prefix for naming. Changed since V5: * Change txt to yaml and fix up the examples. Changed since V4: * No changes. Changed since V3: * No changes. Changed since V2: * No changes. Changed since V1: * Modified sample text in example .../bindings/sound/maxim,max98390.yaml | 51 +++++++++++++++++++ 1 file changed, 51 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/maxim,max98390.yaml diff --git a/Documentation/devicetree/bindings/sound/maxim,max98390.yaml b/Documentation/devicetree/bindings/sound/maxim,max98390.yaml new file mode 100644 index 000000000000..e5ac35280da3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max98390.yaml @@ -0,0 +1,51 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/maxim,max98390.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Maxim Integrated MAX98390 Speaker Amplifier with Integrated Dynamic Speaker Management + +maintainers: + - Steve Lee <steves.lee(a)maximintegrated.com> + +properties: + compatible: + const: maxim,max98390 + + reg: + maxItems: 1 + description: I2C address of the device. + + maxim,temperature_calib: + allOf: + - $ref: /schemas/types.yaml#/definitions/uint32 + description: The calculated temperature data was measured while doing the calibration. + minimum: 0 + maximum: 65535 + + maxim,r0_calib: + allOf: + - $ref: /schemas/types.yaml#/definitions/uint32 + description: This is r0 calibration data which was measured in factory mode. + minimum: 1 + maximum: 8388607 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + max98390: amplifier@38 { + compatible = "maxim,max98390"; + reg = <0x38>; + maxim,temperature_calib = <1024>; + maxim,r0_calib = <100232>; + }; + }; -- 2.17.1

3 2

[PATCH 1/2] ASoC: bindings: wm8960: Add property for headphone detection
by Shengjiu Wang 16 Jun '20

16 Jun '20

Add two properties for headphone detection. wlf,hp-cfg: A list of headphone jack detect configuration register values wlf,gpio-cfg: A list of GPIO configuration register values Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Documentation/devicetree/bindings/sound/wm8960.txt | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/wm8960.txt b/Documentation/devicetree/bindings/sound/wm8960.txt index 6d29ac3750ee..85d3b287108c 100644 --- a/Documentation/devicetree/bindings/sound/wm8960.txt +++ b/Documentation/devicetree/bindings/sound/wm8960.txt @@ -21,6 +21,17 @@ Optional properties: enabled and disabled together with HP_L and HP_R pins in response to jack detect events. + - wlf,hp-cfg: A list of headphone jack detect configuration register values. + The list must be 3 entries long. + hp-cfg[0]: HPSEL[1:0] of R48 (Additional Control 4). + hp-cfg[1]: {HPSWEN:HPSWPOL} of R24 (Additional Control 2). + hp-cfg[2]: {TOCLKSEL:TOEN} of R23 (Additional Control 1). + + - wlf,gpio-cfg: A list of GPIO configuration register values. + The list must be 2 entries long. + gpio-cfg[0]: ALRCGPIO of R9 (Audio interface) + gpio-cfg[1]: {GPIOPOL:GPIOSEL[2:0]} of R48 (Additional Control 4). + Example: wm8960: codec@1a { -- 2.21.0

3 3

[PATCH 0/2] ASoC: max98357a: support MAX98360A in OF
by Tzung-Bi Shih 16 Jun '20

16 Jun '20

Commit 1a0f2433d738 ("ASoC: max98357a: Add ACPI HID MAX98360A") supports MAX98360A in ACPI world. This series supports MAX98360A in OF world. Tzung-Bi Shih (2): ASoC: max98357a: add compatible string for MAX98360A ASoC: dt-bindings: add compatible string for MAX98360A .../devicetree/bindings/sound/max98357a.txt | 12 +++++++++--- sound/soc/codecs/max98357a.c | 1 + 2 files changed, 10 insertions(+), 3 deletions(-) -- 2.27.0.278.ge193c7cf3a9-goog

3 4

[PATCH v3 0/3] ASoC: Intel: Add KeemBay ASoC platform driver
by Sia Jee Heng 16 Jun '20

16 Jun '20

The below series of patches support the KeemBay ASoC platform driver. The platform driver initialize the i2s to capture and playback the pcm data on the ARM. The i2s is running in polling mode. There is no DSP in the KeemBay SoC. Users are rely on the Gstreamer plugin to perform Audio preprocessing. Audio graph card is used to connect the platform driver with the tlv320aic3204 codec. Change History: v3: - Adjusted header format. - Use Audio graph card instead of custom sound card. - Use if-else instead of conditional operator. - Enabled .set_fmt to configure master clock. v2: - Corrected I2S naming for DT binding. v1: - Initial version. Sia Jee Heng (3): ASoC: Intel: Add KeemBay platform driver ASoC: Intel: Add makefiles and kconfig changes for KeemBay dt-bindings: sound: Add documentation for KeemBay i2s .../bindings/sound/intel,keembay-i2s.yaml | 68 +++ sound/soc/intel/Kconfig | 7 + sound/soc/intel/Makefile | 1 + sound/soc/intel/keembay/Makefile | 4 + sound/soc/intel/keembay/kmb_platform.c | 667 +++++++++++++++++++++ sound/soc/intel/keembay/kmb_platform.h | 145 +++++ 6 files changed, 892 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/intel,keembay-i2s.yaml create mode 100644 sound/soc/intel/keembay/Makefile create mode 100644 sound/soc/intel/keembay/kmb_platform.c create mode 100644 sound/soc/intel/keembay/kmb_platform.h -- 1.9.1

4 8

[PATCH 1/5] ASoC: rt5682: Enable Vref2 under using PLL2
by derek.fang＠realtek.com 16 Jun '20

16 Jun '20

From: "derek.fang" <derek.fang(a)realtek.com> Enable Vref2 under long term using PLL2 to avoid clock unstable. Signed-off-by: derek.fang <derek.fang(a)realtek.com> --- sound/soc/codecs/rt5682.c | 24 +++++++++++++++++------- 1 file changed, 17 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index d324512..ca8e1f2 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -960,13 +960,12 @@ int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert) rt5682_enable_push_button_irq(component, false); snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_LOW); - if (snd_soc_dapm_get_pin_status(dapm, "MICBIAS")) + if (!snd_soc_dapm_get_pin_status(dapm, "MICBIAS")) snd_soc_component_update_bits(component, - RT5682_PWR_ANLG_1, RT5682_PWR_VREF2, 0); - else + RT5682_PWR_ANLG_1, RT5682_PWR_MB, 0); + if (!snd_soc_dapm_get_pin_status(dapm, "Vref2")) snd_soc_component_update_bits(component, - RT5682_PWR_ANLG_1, - RT5682_PWR_VREF2 | RT5682_PWR_MB, 0); + RT5682_PWR_ANLG_1, RT5682_PWR_VREF2, 0); snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3, RT5682_PWR_CBJ, 0); @@ -1601,8 +1600,7 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { 0, set_filter_clk, SND_SOC_DAPM_PRE_PMU), SND_SOC_DAPM_SUPPLY("Vref1", RT5682_PWR_ANLG_1, RT5682_PWR_VREF1_BIT, 0, rt5682_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), - SND_SOC_DAPM_SUPPLY("Vref2", RT5682_PWR_ANLG_1, RT5682_PWR_VREF2_BIT, 0, - NULL, 0), + SND_SOC_DAPM_SUPPLY("Vref2", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS", SND_SOC_NOPM, 0, 0, NULL, 0), /* ASRC */ @@ -2485,6 +2483,15 @@ static int rt5682_wclk_prepare(struct clk_hw *hw) snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS"); snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, RT5682_PWR_MB, RT5682_PWR_MB); + + snd_soc_dapm_force_enable_pin_unlocked(dapm, "Vref2"); + snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, + RT5682_PWR_VREF2 | RT5682_PWR_FV2, + RT5682_PWR_VREF2); + usleep_range(55000, 60000); + snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, + RT5682_PWR_FV2, RT5682_PWR_FV2); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "I2S1"); snd_soc_dapm_force_enable_pin_unlocked(dapm, "PLL2F"); snd_soc_dapm_force_enable_pin_unlocked(dapm, "PLL2B"); @@ -2510,9 +2517,12 @@ static void rt5682_wclk_unprepare(struct clk_hw *hw) snd_soc_dapm_mutex_lock(dapm); snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Vref2"); if (!rt5682->jack_type) snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, + RT5682_PWR_VREF2 | RT5682_PWR_FV2 | RT5682_PWR_MB, 0); + snd_soc_dapm_disable_pin_unlocked(dapm, "I2S1"); snd_soc_dapm_disable_pin_unlocked(dapm, "PLL2F"); snd_soc_dapm_disable_pin_unlocked(dapm, "PLL2B"); -- 2.7.4

2 6

[PATCH v5 0/3] ASoC: Intel: Add KeemBay ASoC platform driver
by Sia Jee Heng 16 Jun '20

16 Jun '20

The below series of patches support the KeemBay ASoC platform driver. The platform driver initialize the i2s to capture and playback the pcm data on the ARM. The i2s is running in polling mode. There is no DSP in the KeemBay SoC. Users are rely on the Gstreamer plugin to perform Audio preprocessing. Audio graph card is used to connect the platform driver with the tlv320aic3204 codec. Change History: v5: - Remove OF dependency from Kconfig as OF is shifted to audio graph card. v4: - Reduce if-otology at the tx/rx function. - Fix indentation. - specify .rate directly v3: - Adjusted header format. - Use Audio graph card instead of custom sound card. - Use if-else instead of conditional operator. - Enabled .set_fmt to configure master clock. v2: - Corrected I2S naming for DT binding. v1: - Initial version. Sia Jee Heng (3): ASoC: Intel: Add KeemBay platform driver ASoC: Intel: Add makefiles and kconfig changes for KeemBay dt-bindings: sound: Add documentation for KeemBay i2s .../bindings/sound/intel,keembay-i2s.yaml | 68 +++ sound/soc/intel/Kconfig | 7 + sound/soc/intel/Makefile | 1 + sound/soc/intel/keembay/Makefile | 4 + sound/soc/intel/keembay/kmb_platform.c | 654 +++++++++++++++++++++ sound/soc/intel/keembay/kmb_platform.h | 145 +++++ 6 files changed, 879 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/intel,keembay-i2s.yaml create mode 100644 sound/soc/intel/keembay/Makefile create mode 100644 sound/soc/intel/keembay/kmb_platform.c create mode 100644 sound/soc/intel/keembay/kmb_platform.h -- 1.9.1

2 4

[v3] ASoC: AMD: Use mixer control to switch between DMICs
by Akshu Agrawal 16 Jun '20

16 Jun '20

Having mixer control to switch between DMICs prevents user to initiate capture simultaneously on both the DMIcs. Earlier 2 separate devices, one for each DMIC, gave an option of using them simultaneously, which is not supported. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- v2: Modified "Front Mic" to "DMIC Switch" v3: Changed to using of Mux sound/soc/amd/acp3x-rt5682-max9836.c | 58 +++++++++++++++------------- 1 file changed, 31 insertions(+), 27 deletions(-) diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index e499c00e0c66..f745b42dfd23 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -188,25 +188,27 @@ static int acp3x_ec_dmic0_startup(struct snd_pcm_substream *substream) machine->cap_i2s_instance = I2S_BT_INSTANCE; snd_soc_dai_set_bclk_ratio(codec_dai, 64); - if (dmic_sel) - gpiod_set_value(dmic_sel, 0); return rt5682_clk_enable(substream); } -static int acp3x_ec_dmic1_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_card *card = rtd->card; - struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); - struct acp3x_platform_info *machine = snd_soc_card_get_drvdata(card); +static int dmic_switch; - machine->cap_i2s_instance = I2S_BT_INSTANCE; - snd_soc_dai_set_bclk_ratio(codec_dai, 64); - if (dmic_sel) - gpiod_set_value(dmic_sel, 1); +static int dmic_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = dmic_switch; + return 0; +} - return rt5682_clk_enable(substream); +static int dmic_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + if (dmic_sel) { + dmic_switch = ucontrol->value.integer.value[0]; + gpiod_set_value(dmic_sel, dmic_switch); + } + return 0; } static void rt5682_shutdown(struct snd_pcm_substream *substream) @@ -229,11 +231,6 @@ static const struct snd_soc_ops acp3x_ec_cap0_ops = { .shutdown = rt5682_shutdown, }; -static const struct snd_soc_ops acp3x_ec_cap1_ops = { - .startup = acp3x_ec_dmic1_startup, - .shutdown = rt5682_shutdown, -}; - SND_SOC_DAILINK_DEF(acp3x_i2s, DAILINK_COMP_ARRAY(COMP_CPU("acp3x_i2s_playcap.0"))); SND_SOC_DAILINK_DEF(acp3x_bt, @@ -279,21 +276,26 @@ static struct snd_soc_dai_link acp3x_dai_5682_98357[] = { .ops = &acp3x_ec_cap0_ops, SND_SOC_DAILINK_REG(acp3x_bt, cros_ec, platform), }, - { - .name = "acp3x-ec-dmic1-capture", - .stream_name = "Capture DMIC1", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBS_CFS, - .dpcm_capture = 1, - .ops = &acp3x_ec_cap1_ops, - SND_SOC_DAILINK_REG(acp3x_bt, cros_ec, platform), - }, }; +static const char * const dmic_mux_text[] = { + "Front Mic", + "Rear Mic", +}; + +static SOC_ENUM_SINGLE_DECL( + acp3x_dmic_enum, SND_SOC_NOPM, 0, dmic_mux_text); + +static const struct snd_kcontrol_new acp3x_dmic_mux_control = + SOC_DAPM_ENUM_EXT("DMIC Select Mux", acp3x_dmic_enum, + dmic_get, dmic_set); + static const struct snd_soc_dapm_widget acp3x_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_SPK("Spk", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MUX("Dmic Mux", SND_SOC_NOPM, 0, 0, + &acp3x_dmic_mux_control), }; static const struct snd_soc_dapm_route acp3x_audio_route[] = { @@ -301,6 +303,8 @@ static const struct snd_soc_dapm_route acp3x_audio_route[] = { {"Headphone Jack", NULL, "HPOR"}, {"IN1P", NULL, "Headset Mic"}, {"Spk", NULL, "Speaker"}, + {"Dmic Mux", "Front Mic", "DMIC"}, + {"Dmic Mux", "Rear Mic", "DMIC"}, }; static const struct snd_kcontrol_new acp3x_mc_controls[] = { -- 2.20.1

2 1

[PATCH 0/3] ASoC: improve core dmesg logs and verbosity
by Pierre-Louis Bossart 16 Jun '20

16 Jun '20

Try to both reduce useless verbosity and keep useful error reports. Pierre-Louis Bossart (3): ASoC: soc-core: reduce verbosity of BE override message ASoC: soc-pcm: improve error messages in soc_pcm_new() ASoC: soc-pcm/compress: reduce verbosity on mapping ok messages sound/soc/soc-compress.c | 4 ++-- sound/soc/soc-core.c | 4 ++-- sound/soc/soc-pcm.c | 13 +++++++------ 3 files changed, 11 insertions(+), 10 deletions(-) -- 2.20.1

2 4

[PATCH] ASoC: img: Fix a reference count leak in img_i2s_in_set_fmt
by wu000273＠umn.edu 16 Jun '20

16 Jun '20

From: Qiushi Wu <wu000273(a)umn.edu> pm_runtime_get_sync() increments the runtime PM usage counter even when it returns an error code, causing incorrect ref count if pm_runtime_put_noidle() is not called in error handling paths. Thus call pm_runtime_put_noidle() if pm_runtime_get_sync() fails. Signed-off-by: Qiushi Wu <wu000273(a)umn.edu> --- sound/soc/img/img-i2s-in.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/img/img-i2s-in.c b/sound/soc/img/img-i2s-in.c index e30b66b94bf6..0843235d73c9 100644 --- a/sound/soc/img/img-i2s-in.c +++ b/sound/soc/img/img-i2s-in.c @@ -343,8 +343,10 @@ static int img_i2s_in_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) chan_control_mask = IMG_I2S_IN_CH_CTL_CLK_TRANS_MASK; ret = pm_runtime_get_sync(i2s->dev); - if (ret < 0) + if (ret < 0) { + pm_runtime_put_noidle(i2s->dev); return ret; + } for (i = 0; i < i2s->active_channels; i++) img_i2s_in_ch_disable(i2s, i); -- 2.17.1

2 1

[PATCH v2 00/12] ASoC: soc-component: collect component functions
by Kuninori Morimoto 16 Jun '20

16 Jun '20

Hi Mark We have soc-component.c now, but still many component related functions are implemented many place. This patch-set collect these into soc-component.c. v1 -> v2 - remove soc-compress.c exchange (But I have plan to repost it) - fixup loop break issue on some functions - direct return on some functions Link: https://lore.kernel.org/r/87a71nzhy2.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (12): ASoC: soc-component: add soc_component_pin() and share code ASoC: soc-component: move snd_soc_component_xxx_regmap() to soc-component ASoC: soc-component: move snd_soc_component_initialize() to soc-component.c ASoC: soc-component: add soc_component_err() ASoC: soc-component: add snd_soc_pcm_component_prepare() ASoC: soc-component: add snd_soc_pcm_component_hw_params() ASoC: soc-component: add snd_soc_pcm_component_hw_free() ASoC: soc-component: add snd_soc_pcm_component_trigger() ASoC: soc-component: add snd_soc_component_init() ASoC: soc-component: merge soc-io.c into soc-component.c ASoC: soc-component: merge soc_pcm_trigger_start/stop() ASoC: soc-component: tidyup Copyright include/sound/soc-component.h | 29 +- sound/soc/Makefile | 2 +- sound/soc/soc-component.c | 666 ++++++++++++++++++++++++---------- sound/soc/soc-core.c | 102 +----- sound/soc/soc-io.c | 202 ----------- sound/soc/soc-pcm.c | 114 ++---- 6 files changed, 531 insertions(+), 584 deletions(-) delete mode 100644 sound/soc/soc-io.c -- 2.17.1

3 14

[PATCH 1/3] dt-bindings: sound: Document wm8994 endpoints
by Jonathan Bakker 16 Jun '20

16 Jun '20

The wm8994 exposes several inputs and outputs that can be used by machine drivers in their routing. Add them to the documention so they don't have been duplicated in any machine drivers bindings. Signed-off-by: Jonathan Bakker <xc-racer2(a)live.ca> --- .../devicetree/bindings/sound/wm8994.txt | 23 +++++++++++++++++++ 1 file changed, 23 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/wm8994.txt b/Documentation/devicetree/bindings/sound/wm8994.txt index 367b58ce1bb9..8fa947509c10 100644 --- a/Documentation/devicetree/bindings/sound/wm8994.txt +++ b/Documentation/devicetree/bindings/sound/wm8994.txt @@ -68,6 +68,29 @@ Optional properties: - wlf,csnaddr-pd : If present enable the internal pull-down resistor on the CS/ADDR pin. +Pins on the device (for linking into audio routes): + + * IN1LN + * IN1LP + * IN2LN + * IN2LP:VXRN + * IN1RN + * IN1RP + * IN2RN + * IN2RP:VXRP + * SPKOUTLP + * SPKOUTLN + * SPKOUTRP + * SPKOUTRN + * HPOUT1L + * HPOUT1R + * HPOUT2P + * HPOUT2N + * LINEOUT1P + * LINEOUT1N + * LINEOUT2P + * LINEOUT2N + Example: wm8994: codec@1a { -- 2.20.1

2 2

[PATCH 0/4] Fix potential issues with x86 PCM SG-buffer
by Takashi Iwai 15 Jun '20

15 Jun '20

Hi, it turned out that x86 SG-pages that are used for PCM buffers aren't safe on x86 in all cases, and we need some limitation and workarounds for non-standard configurations. This patch set is an attempt to paper over it in a minimalistic way. Takashi === Takashi Iwai (4): ALSA: pcm: Use dma_mmap_coherent() on x86, too ALSA: memalloc: Initialize all fields of snd_dma_buffer properly ALSA: memalloc: Make SG-buffer helper usable for continuous buffer, too ALSA: pcm: Use SG-buffer only when direct DMA is available include/sound/memalloc.h | 9 ++++++++- sound/core/memalloc.c | 7 +++---- sound/core/pcm_memory.c | 13 +++++++++++++ sound/core/pcm_native.c | 2 -- sound/core/sgbuf.c | 3 +++ 5 files changed, 27 insertions(+), 7 deletions(-) -- 2.16.4

1 4

[PATCH] ASoC: soc-pcm: fix checks for multi-cpu FE dailinks
by Pierre-Louis Bossart 15 Jun '20

15 Jun '20

soc_dpcm_fe_runtime_update() is called for all dailinks, and we want to first discard all back-ends, then deal with front-ends. The existing code first reports an error with multi-cpu front-ends, and that check needs to be moved after we know that we are dealing with a front-end. BugLink: https://github.com/thesofproject/linux/issues/1970 Fixes: 6e1276a5e613d ('ASoC: Return error if the function does not support multi-cpu') Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- sound/soc/soc-pcm.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 2c114b4542ce..c517064f5391 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2630,15 +2630,15 @@ static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new) int count, paths; int ret; + if (!fe->dai_link->dynamic) + return 0; + if (fe->num_cpus > 1) { dev_err(fe->dev, "%s doesn't support Multi CPU yet\n", __func__); return -EINVAL; } - if (!fe->dai_link->dynamic) - return 0; - /* only check active links */ if (!snd_soc_dai_active(asoc_rtd_to_cpu(fe, 0))) return 0; -- 2.20.1

2 1

[PATCH 1/2] ASoC: rt1015: Update rt1015 default register value according to spec modification.
by jack.yu＠realtek.com 15 Jun '20

15 Jun '20

From: Jack Yu <jack.yu(a)realtek.com> Update rt1015 default register value according to spec modification. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt1015.c | 124 +++++++++++++++++++++++--------------- sound/soc/codecs/rt1015.h | 15 ++++- 2 files changed, 89 insertions(+), 50 deletions(-) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 67e2e944d21b..2cccb310fa96 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -34,30 +34,32 @@ static const struct reg_default rt1015_reg[] = { { 0x0000, 0x0000 }, { 0x0004, 0xa000 }, { 0x0006, 0x0003 }, - { 0x000a, 0x0802 }, - { 0x000c, 0x0020 }, + { 0x000a, 0x081e }, + { 0x000c, 0x0006 }, { 0x000e, 0x0000 }, { 0x0010, 0x0000 }, { 0x0012, 0x0000 }, + { 0x0014, 0x0000 }, + { 0x0016, 0x0000 }, + { 0x0018, 0x0000 }, { 0x0020, 0x8000 }, - { 0x0022, 0x471b }, - { 0x006a, 0x0000 }, - { 0x006c, 0x4020 }, + { 0x0022, 0x8043 }, { 0x0076, 0x0000 }, { 0x0078, 0x0000 }, - { 0x007a, 0x0000 }, + { 0x007a, 0x0002 }, { 0x007c, 0x10ec }, { 0x007d, 0x1015 }, { 0x00f0, 0x5000 }, - { 0x00f2, 0x0774 }, - { 0x00f3, 0x8400 }, + { 0x00f2, 0x004c }, + { 0x00f3, 0xecfe }, { 0x00f4, 0x0000 }, + { 0x00f6, 0x0400 }, { 0x0100, 0x0028 }, { 0x0102, 0xff02 }, - { 0x0104, 0x8232 }, + { 0x0104, 0xa213 }, { 0x0106, 0x200c }, - { 0x010c, 0x002f }, - { 0x010e, 0xc000 }, + { 0x010c, 0x0000 }, + { 0x010e, 0x0058 }, { 0x0111, 0x0200 }, { 0x0112, 0x0400 }, { 0x0114, 0x0022 }, @@ -65,38 +67,46 @@ static const struct reg_default rt1015_reg[] = { { 0x0118, 0x0000 }, { 0x011a, 0x0123 }, { 0x011c, 0x4567 }, - { 0x0300, 0xdddd }, - { 0x0302, 0x0000 }, - { 0x0311, 0x9330 }, - { 0x0313, 0x0000 }, - { 0x0314, 0x0000 }, + { 0x0300, 0x203d }, + { 0x0302, 0x001e }, + { 0x0311, 0x0000 }, + { 0x0313, 0x6014 }, + { 0x0314, 0x00a2 }, { 0x031a, 0x00a0 }, { 0x031c, 0x001f }, { 0x031d, 0xffff }, { 0x031e, 0x0000 }, { 0x031f, 0x0000 }, + { 0x0320, 0x0000 }, { 0x0321, 0x0000 }, - { 0x0322, 0x0000 }, - { 0x0328, 0x0000 }, - { 0x0329, 0x0000 }, - { 0x032a, 0x0000 }, - { 0x032b, 0x0000 }, - { 0x032c, 0x0000 }, - { 0x032d, 0x0000 }, - { 0x032e, 0x030e }, - { 0x0330, 0x0080 }, + { 0x0322, 0xd7df }, + { 0x0328, 0x10b2 }, + { 0x0329, 0x0175 }, + { 0x032a, 0x36ad }, + { 0x032b, 0x7e55 }, + { 0x032c, 0x0520 }, + { 0x032d, 0xaa00 }, + { 0x032e, 0x570e }, + { 0x0330, 0xe180 }, { 0x0332, 0x0034 }, - { 0x0334, 0x0000 }, - { 0x0336, 0x0000 }, + { 0x0334, 0x0001 }, + { 0x0336, 0x0010 }, + { 0x0338, 0x0000 }, + { 0x04fa, 0x0030 }, + { 0x04fc, 0x35c8 }, + { 0x04fe, 0x0800 }, + { 0x0500, 0x0400 }, + { 0x0502, 0x1000 }, + { 0x0504, 0x0000 }, { 0x0506, 0x04ff }, - { 0x0508, 0x0030 }, - { 0x050a, 0x0018 }, - { 0x0519, 0x307f }, - { 0x051a, 0xffff }, - { 0x051b, 0x4000 }, + { 0x0508, 0x0010 }, + { 0x050a, 0x001a }, + { 0x0519, 0x1c68 }, + { 0x051a, 0x0ccc }, + { 0x051b, 0x0666 }, { 0x051d, 0x0000 }, { 0x051f, 0x0000 }, - { 0x0536, 0x1000 }, + { 0x0536, 0x061c }, { 0x0538, 0x0000 }, { 0x053a, 0x0000 }, { 0x053c, 0x0000 }, @@ -110,19 +120,18 @@ static const struct reg_default rt1015_reg[] = { { 0x0544, 0x0000 }, { 0x0568, 0x0000 }, { 0x056a, 0x0000 }, - { 0x1000, 0x0000 }, - { 0x1002, 0x6505 }, + { 0x1000, 0x0040 }, + { 0x1002, 0x5405 }, { 0x1006, 0x5515 }, - { 0x1007, 0x003f }, - { 0x1009, 0x770f }, - { 0x100a, 0x01ff }, - { 0x100c, 0x0000 }, + { 0x1007, 0x05f7 }, + { 0x1009, 0x0b0a }, + { 0x100a, 0x00ef }, { 0x100d, 0x0003 }, { 0x1010, 0xa433 }, { 0x1020, 0x0000 }, - { 0x1200, 0x3d02 }, - { 0x1202, 0x0813 }, - { 0x1204, 0x0211 }, + { 0x1200, 0x5a01 }, + { 0x1202, 0x6524 }, + { 0x1204, 0x1f00 }, { 0x1206, 0x0000 }, { 0x1208, 0x0000 }, { 0x120a, 0x0000 }, @@ -130,16 +139,16 @@ static const struct reg_default rt1015_reg[] = { { 0x120e, 0x0000 }, { 0x1210, 0x0000 }, { 0x1212, 0x0000 }, - { 0x1300, 0x0701 }, - { 0x1302, 0x12f9 }, - { 0x1304, 0x3405 }, + { 0x1300, 0x10a1 }, + { 0x1302, 0x12ff }, + { 0x1304, 0x0400 }, { 0x1305, 0x0844 }, - { 0x1306, 0x1611 }, + { 0x1306, 0x4611 }, { 0x1308, 0x555e }, { 0x130a, 0x0000 }, - { 0x130c, 0x2400}, - { 0x130e, 0x7700 }, - { 0x130f, 0x0000 }, + { 0x130c, 0x2000 }, + { 0x130e, 0x0100 }, + { 0x130f, 0x0001 }, { 0x1310, 0x0000 }, { 0x1312, 0x0000 }, { 0x1314, 0x0000 }, @@ -209,6 +218,9 @@ static bool rt1015_volatile_register(struct device *dev, unsigned int reg) case RT1015_DC_CALIB_CLSD7: case RT1015_DC_CALIB_CLSD8: case RT1015_S_BST_TIMING_INTER1: + case RT1015_OSCK_STA: + case RT1015_MONO_DYNA_CTRL1: + case RT1015_MONO_DYNA_CTRL5: return true; default: @@ -224,6 +236,12 @@ static bool rt1015_readable_register(struct device *dev, unsigned int reg) case RT1015_CLK3: case RT1015_PLL1: case RT1015_PLL2: + case RT1015_DUM_RW1: + case RT1015_DUM_RW2: + case RT1015_DUM_RW3: + case RT1015_DUM_RW4: + case RT1015_DUM_RW5: + case RT1015_DUM_RW6: case RT1015_CLK_DET: case RT1015_SIL_DET: case RT1015_CUSTOMER_ID: @@ -235,6 +253,7 @@ static bool rt1015_readable_register(struct device *dev, unsigned int reg) case RT1015_PAD_DRV2: case RT1015_GAT_BOOST: case RT1015_PRO_ALT: + case RT1015_OSCK_STA: case RT1015_MAN_I2C: case RT1015_DAC1: case RT1015_DAC2: @@ -272,6 +291,13 @@ static bool rt1015_readable_register(struct device *dev, unsigned int reg) case RT1015_SMART_BST_CTRL2: case RT1015_ANA_CTRL1: case RT1015_ANA_CTRL2: + case RT1015_PWR_STATE_CTRL: + case RT1015_MONO_DYNA_CTRL: + case RT1015_MONO_DYNA_CTRL1: + case RT1015_MONO_DYNA_CTRL2: + case RT1015_MONO_DYNA_CTRL3: + case RT1015_MONO_DYNA_CTRL4: + case RT1015_MONO_DYNA_CTRL5: case RT1015_SPK_VOL: case RT1015_SHORT_DETTOP1: case RT1015_SHORT_DETTOP2: diff --git a/sound/soc/codecs/rt1015.h b/sound/soc/codecs/rt1015.h index 6fbe802082c4..8169962935a5 100644 --- a/sound/soc/codecs/rt1015.h +++ b/sound/soc/codecs/rt1015.h @@ -21,6 +21,12 @@ #define RT1015_CLK3 0x0006 #define RT1015_PLL1 0x000a #define RT1015_PLL2 0x000c +#define RT1015_DUM_RW1 0x000e +#define RT1015_DUM_RW2 0x0010 +#define RT1015_DUM_RW3 0x0012 +#define RT1015_DUM_RW4 0x0014 +#define RT1015_DUM_RW5 0x0016 +#define RT1015_DUM_RW6 0x0018 #define RT1015_CLK_DET 0x0020 #define RT1015_SIL_DET 0x0022 #define RT1015_CUSTOMER_ID 0x0076 @@ -32,6 +38,7 @@ #define RT1015_PAD_DRV2 0x00f2 #define RT1015_GAT_BOOST 0x00f3 #define RT1015_PRO_ALT 0x00f4 +#define RT1015_OSCK_STA 0x00f6 #define RT1015_MAN_I2C 0x0100 #define RT1015_DAC1 0x0102 #define RT1015_DAC2 0x0104 @@ -70,7 +77,13 @@ #define RT1015_ANA_CTRL1 0x0334 #define RT1015_ANA_CTRL2 0x0336 #define RT1015_PWR_STATE_CTRL 0x0338 -#define RT1015_SPK_VOL 0x0506 +#define RT1015_MONO_DYNA_CTRL 0x04fa +#define RT1015_MONO_DYNA_CTRL1 0x04fc +#define RT1015_MONO_DYNA_CTRL2 0x04fe +#define RT1015_MONO_DYNA_CTRL3 0x0500 +#define RT1015_MONO_DYNA_CTRL4 0x0502 +#define RT1015_MONO_DYNA_CTRL5 0x0504 +#define RT1015_SPK_VOL 0x0506 #define RT1015_SHORT_DETTOP1 0x0508 #define RT1015_SHORT_DETTOP2 0x050a #define RT1015_SPK_DC_DETECT1 0x0519 -- 2.25.1

2 4

[PATCH v3] ASoC: SOF: Intel: hda: Clear RIRB status before reading WP
by Brent Lu 15 Jun '20

15 Jun '20

Port commit 6d011d5057ff ("ALSA: hda: Clear RIRB status before reading WP") from legacy HDA driver to fix the get response timeout issue. Current SOF driver does not suffer from this issue because sync write is enabled in hda_init. The issue will come back if the sync write is disabled for some reason. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/sof/intel/hda-stream.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index 7f65dcc..1bda14c 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -653,11 +653,16 @@ irqreturn_t hda_dsp_stream_threaded_handler(int irq, void *context) if (status & AZX_INT_CTRL_EN) { rirb_status = snd_hdac_chip_readb(bus, RIRBSTS); if (rirb_status & RIRB_INT_MASK) { + /* + * Clearing the interrupt status here ensures + * that no interrupt gets masked after the RIRB + * wp is read in snd_hdac_bus_update_rirb. + */ + snd_hdac_chip_writeb(bus, RIRBSTS, + RIRB_INT_MASK); active = true; if (rirb_status & RIRB_INT_RESPONSE) snd_hdac_bus_update_rirb(bus); - snd_hdac_chip_writeb(bus, RIRBSTS, - RIRB_INT_MASK); } } #endif -- 2.7.4

5 4

[PATCH 1/2] ASoc: q6afe: add support to get port direction
by Srinivas Kandagatla 15 Jun '20

15 Jun '20

This patch adds support to q6afe_is_rx_port() to get direction of DSP BE dai port, this is useful for setting dailink directions correctly. Fixes: c25e295cd77b (ASoC: qcom: Add support to parse common audio device nodes") Reported-by: John Stultz <john.stultz(a)linaro.org> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/qdsp6/q6afe.c | 8 ++++++++ sound/soc/qcom/qdsp6/q6afe.h | 1 + 2 files changed, 9 insertions(+) diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c index e0945f7a58c8..0ce4eb60f984 100644 --- a/sound/soc/qcom/qdsp6/q6afe.c +++ b/sound/soc/qcom/qdsp6/q6afe.c @@ -800,6 +800,14 @@ int q6afe_get_port_id(int index) } EXPORT_SYMBOL_GPL(q6afe_get_port_id); +int q6afe_is_rx_port(int index) +{ + if (index < 0 || index >= AFE_PORT_MAX) + return -EINVAL; + + return port_maps[index].is_rx; +} +EXPORT_SYMBOL_GPL(q6afe_is_rx_port); static int afe_apr_send_pkt(struct q6afe *afe, struct apr_pkt *pkt, struct q6afe_port *port) { diff --git a/sound/soc/qcom/qdsp6/q6afe.h b/sound/soc/qcom/qdsp6/q6afe.h index c7ed5422baff..1a0f80a14afe 100644 --- a/sound/soc/qcom/qdsp6/q6afe.h +++ b/sound/soc/qcom/qdsp6/q6afe.h @@ -198,6 +198,7 @@ int q6afe_port_start(struct q6afe_port *port); int q6afe_port_stop(struct q6afe_port *port); void q6afe_port_put(struct q6afe_port *port); int q6afe_get_port_id(int index); +int q6afe_is_rx_port(int index); void q6afe_hdmi_port_prepare(struct q6afe_port *port, struct q6afe_hdmi_cfg *cfg); void q6afe_slim_port_prepare(struct q6afe_port *port, -- 2.21.0

4 5

[PATCH 0/2] ASoC: topology: fix use-after-free when removing components
by Pierre-Louis Bossart 15 Jun '20

15 Jun '20

This patchset fixes a memory allocation issue and removes a 100% reproducible use-after-free report thrown by KASAN in automated module removal tests across multiple platforms. All the credit goes to Bard Liao for root-causing the issue. DAIs may be registered at the same time as a component, or when the topology is loaded. This two-step registration causes the memory for topology-based DAIs to allocated last, and conversely to be released first by devres, before the component is released and the DAIs removed from the component DAI list with snd_soc_unregister_dais(). When we remove a component, by the time we walk through its dai list to unregister all dais, the dais allocated by the topology have been freed already by devres and the list is corrupted with pointers that are no longer valid. The suggestion is to add an explicit devm_ based registration for topology-based dais, so that each dai is cleanly removed from the component dai list in the release operation before devres releases the allocated memory. Pierre-Louis Bossart (2): ASoC: soc-devres: add devm_snd_soc_register_dai() ASoC: soc-topology: use devm_snd_soc_register_dai() include/sound/soc.h | 4 ++++ sound/soc/soc-devres.c | 37 +++++++++++++++++++++++++++++++++++++ sound/soc/soc-topology.c | 3 +-- 3 files changed, 42 insertions(+), 2 deletions(-) -- 2.20.1

2 3

alsa-gobject: seq/timer: add constructure for status object
by GitHub pull_request - opened 15 Jun '20

15 Jun '20

alsa-project/alsa-gobject pull request #39 was opened from takaswie: In patchset at #38, I forgot to add constructors for issued status object because no function call returns the instance of object. This patchset adds the constructors. Request URL : https://github.com/alsa-project/alsa-gobject/pull/39 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/39.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH] ALSA: usb-audio: Set 48 kHz rate for Rodecaster
by Christopher Swenson 15 Jun '20

15 Jun '20

>From 22581d37726a17ed9c400b8c9ff8ade52c17f65a Mon Sep 17 00:00:00 2001 From: Christopher Swenson <swenson(a)swenson.io> Date: Sun, 14 Jun 2020 15:19:41 -0700 Subject: [PATCH] ALSA: usb-audio: Set 48 kHz rate for Rodecaster Like the Line6 devices, the Rode Rodecaster Pro does not support UAC2_CS_RANGE and only supports a sample rate of 48 kHz. Tested against a Rode Rodecaster Pro. Tested-by: Christopher Swenson <swenson(a)swenson.io> Signed-off-by: Christopher Swenson <swenson(a)swenson.io> --- sound/usb/format.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/usb/format.c b/sound/usb/format.c index 5ffb457cc88c..1b28d01d1f4c 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -394,8 +394,9 @@ static int parse_uac2_sample_rate_range(struct snd_usb_audio *chip, return nr_rates; } -/* Line6 Helix series don't support the UAC2_CS_RANGE usb function - * call. Return a static table of known clock rates. +/* Line6 Helix series and the Rode Rodecaster Pro don't support the + * UAC2_CS_RANGE usb function call. Return a static table of known + * clock rates. */ static int line6_parse_audio_format_rates_quirk(struct snd_usb_audio *chip, struct audioformat *fp) @@ -408,6 +409,7 @@ static int line6_parse_audio_format_rates_quirk(struct snd_usb_audio *chip, case USB_ID(0x0e41, 0x4248): /* Line6 Helix >= fw 2.82 */ case USB_ID(0x0e41, 0x4249): /* Line6 Helix Rack >= fw 2.82 */ case USB_ID(0x0e41, 0x424a): /* Line6 Helix LT >= fw 2.82 */ + case USB_ID(0x19f7, 0x0011): /* Rode Rodecaster Pro */ return set_fixed_rate(fp, 48000, SNDRV_PCM_RATE_48000); } -- 2.26.0

2 1

[patch for-5.8] dma-pool: decouple DMA_REMAP from DMA_COHERENT_POOL
by David Rientjes 15 Jun '20

15 Jun '20

DMA_REMAP is an unnecessary requirement for AMD SEV, which requires DMA_COHERENT_POOL, so avoid selecting it when it is otherwise unnecessary. The only other requirement for DMA coherent pools is DMA_DIRECT_REMAP, so ensure that properly selects the config option when needed. Fixes: 82fef0ad811f ("x86/mm: unencrypted non-blocking DMA allocations use coherent pools") Suggested-by: Christoph Hellwig <hch(a)lst.de> Signed-off-by: David Rientjes <rientjes(a)google.com> --- kernel/dma/Kconfig | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/kernel/dma/Kconfig b/kernel/dma/Kconfig --- a/kernel/dma/Kconfig +++ b/kernel/dma/Kconfig @@ -73,18 +73,18 @@ config SWIOTLB config DMA_NONCOHERENT_MMAP bool +config DMA_COHERENT_POOL + bool + config DMA_REMAP + bool depends on MMU select GENERIC_ALLOCATOR select DMA_NONCOHERENT_MMAP - bool - -config DMA_COHERENT_POOL - bool - select DMA_REMAP config DMA_DIRECT_REMAP bool + select DMA_REMAP select DMA_COHERENT_POOL config DMA_CMA

3 2

alsa-gobject: seq/timer: use (inout) annotation for API to retrieve status
by GitHub pull_request - opened 15 Jun '20

15 Jun '20

alsa-project/alsa-gobject pull request #38 was opened from takaswie: It's probable to retrieve status for something several times in runtime, and it's convenient for the runtime to use pre-allocated instance for the status. Request URL : https://github.com/alsa-project/alsa-gobject/pull/38 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/38.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

sof-hda-dsp: add initial kcontrol values
by GitHub pull_request - opened 15 Jun '20

15 Jun '20

alsa-project/alsa-ucm-conf pull request #35 was opened from libinyang: This patch adds the initial values of sof-hda-dsp card. Signed-off-by: Libin Yang <libin.yang(a)intel.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/35 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/35.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 3/3] ASoC: samsung: Add driver for Aries boards
by Jonathan Bakker 14 Jun '20

14 Jun '20

Samsung Aries boards have a WM8994 codec connected to the Samsung I2S controller, the BT codec, and the cellular modem. Jack detection is done by a combination of an ADC, GPIOs, and an extcon device for the USB dock. There is also a GPIO for selection between the Mic path and the TV out path on the headphone jack. There are two main variants, one with an FM radio and where the modem is the master and one without a radio and the modem is the slave. Signed-off-by: Jonathan Bakker <xc-racer2(a)live.ca> --- sound/soc/samsung/Kconfig | 13 + sound/soc/samsung/Makefile | 2 + sound/soc/samsung/aries_wm8994.c | 695 +++++++++++++++++++++++++++++++ 3 files changed, 710 insertions(+) create mode 100644 sound/soc/samsung/aries_wm8994.c diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 112911dc271b..505087d2a50d 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -212,4 +212,17 @@ config SND_SOC_SAMSUNG_TM2_WM5110 help Say Y if you want to add support for SoC audio on the TM2 board. +config SND_SOC_SAMSUNG_ARIES_WM8994 + tristate "SoC I2S Audio support for WM8994 on Aries" + depends on SND_SOC_SAMSUNG && MFD_WM8994 && IIO && EXTCON + select SND_SOC_BT_SCO + select SND_SOC_WM8994 + select SND_SAMSUNG_I2S + help + Say Y if you want to add support for SoC audio on Aries boards, + which has a WM8994 codec connected to a BT codec, a cellular + modem, and the Samsung I2S controller. Jack detection is done + via ADC, GPIOs, and an extcon device. Switching between the Mic + and TV-Out path is also handled. + endif #SND_SOC_SAMSUNG diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 8f5dfe20b9f1..22259f7818f0 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -41,6 +41,7 @@ snd-soc-bells-objs := bells.o snd-soc-odroid-objs := odroid.o snd-soc-arndale-objs := arndale.o snd-soc-tm2-wm5110-objs := tm2_wm5110.o +snd-soc-aries-wm8994-objs := aries_wm8994.o obj-$(CONFIG_SND_SOC_SAMSUNG_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_SOC_SAMSUNG_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -64,3 +65,4 @@ obj-$(CONFIG_SND_SOC_BELLS) += snd-soc-bells.o obj-$(CONFIG_SND_SOC_ODROID) += snd-soc-odroid.o obj-$(CONFIG_SND_SOC_ARNDALE) += snd-soc-arndale.o obj-$(CONFIG_SND_SOC_SAMSUNG_TM2_WM5110) += snd-soc-tm2-wm5110.o +obj-$(CONFIG_SND_SOC_SAMSUNG_ARIES_WM8994) += snd-soc-aries-wm8994.o diff --git a/sound/soc/samsung/aries_wm8994.c b/sound/soc/samsung/aries_wm8994.c new file mode 100644 index 000000000000..8579c87dcae8 --- /dev/null +++ b/sound/soc/samsung/aries_wm8994.c @@ -0,0 +1,695 @@ +// SPDX-License-Identifier: GPL-2.0+ +#include <linux/extcon.h> +#include <linux/iio/consumer.h> +#include <linux/iio/iio.h> +#include <linux/input-event-codes.h> +#include <linux/mfd/wm8994/registers.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_device.h> +#include <linux/of_gpio.h> +#include <linux/regulator/consumer.h> +#include <sound/jack.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "i2s.h" +#include "../codecs/wm8994.h" + +#define ARIES_MCLK1_FREQ 24000000 + +struct aries_wm8994_variant { + unsigned int modem_dai_fmt; + bool has_fm_radio; +}; + +struct aries_wm8994_data { + struct extcon_dev *usb_extcon; + struct regulator *reg_main_micbias; + struct regulator *reg_headset_micbias; + struct gpio_desc *gpio_headset_detect; + struct gpio_desc *gpio_headset_key; + struct gpio_desc *gpio_earpath_sel; + struct iio_channel *adc; + const struct aries_wm8994_variant *variant; +}; + +/* USB dock */ +static struct snd_soc_jack aries_dock; + +static struct snd_soc_jack_pin dock_pins[] = { + { + .pin = "LINE", + .mask = SND_JACK_LINEOUT, + }, +}; + +static int aries_extcon_notifier(struct notifier_block *this, + unsigned long connected, void *_cmd) +{ + if (connected) + snd_soc_jack_report(&aries_dock, SND_JACK_LINEOUT, + SND_JACK_LINEOUT); + else + snd_soc_jack_report(&aries_dock, 0, SND_JACK_LINEOUT); + + return NOTIFY_DONE; +} + +static struct notifier_block aries_extcon_notifier_block = { + .notifier_call = aries_extcon_notifier, +}; + +/* Headset jack */ +static struct snd_soc_jack aries_headset; + +static struct snd_soc_jack_pin jack_pins[] = { + { + .pin = "HP", + .mask = SND_JACK_HEADPHONE, + }, { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + +static struct snd_soc_jack_zone headset_zones[] = { + { + .min_mv = 0, + .max_mv = 241, + .jack_type = SND_JACK_HEADPHONE, + }, { + .min_mv = 242, + .max_mv = 2980, + .jack_type = SND_JACK_HEADSET, + }, { + .min_mv = 2981, + .max_mv = UINT_MAX, + .jack_type = SND_JACK_HEADPHONE, + }, +}; + +static irqreturn_t headset_det_irq_thread(int irq, void *data) +{ + struct aries_wm8994_data *priv = (struct aries_wm8994_data *) data; + int ret = 0; + int time_left_ms = 300; + int adc; + + while (time_left_ms > 0) { + if (!gpiod_get_value(priv->gpio_headset_detect)) { + snd_soc_jack_report(&aries_headset, 0, + SND_JACK_HEADSET); + gpiod_set_value(priv->gpio_earpath_sel, 0); + return IRQ_HANDLED; + } + msleep(20); + time_left_ms -= 20; + } + + /* Temporarily enable micbias and earpath selector */ + ret = regulator_enable(priv->reg_headset_micbias); + if (ret) + pr_err("%s failed to enable micbias: %d", __func__, ret); + + gpiod_set_value(priv->gpio_earpath_sel, 1); + + ret = iio_read_channel_processed(priv->adc, &adc); + if (ret < 0) { + /* failed to read ADC, so assume headphone */ + pr_err("%s failed to read ADC, assuming headphones", __func__); + snd_soc_jack_report(&aries_headset, SND_JACK_HEADPHONE, + SND_JACK_HEADSET); + } else { + snd_soc_jack_report(&aries_headset, + snd_soc_jack_get_type(&aries_headset, adc), + SND_JACK_HEADSET); + } + + ret = regulator_disable(priv->reg_headset_micbias); + if (ret) + pr_err("%s failed disable micbias: %d", __func__, ret); + + /* Disable earpath selector when no mic connected */ + if (!(aries_headset.status & SND_JACK_MICROPHONE)) + gpiod_set_value(priv->gpio_earpath_sel, 0); + + return IRQ_HANDLED; +} + +static int headset_button_check(void *data) +{ + struct aries_wm8994_data *priv = (struct aries_wm8994_data *) data; + + /* Filter out keypresses when 4 pole jack not detected */ + if (gpiod_get_value_cansleep(priv->gpio_headset_key) && + aries_headset.status & SND_JACK_MICROPHONE) + return SND_JACK_BTN_0; + + return 0; +} + +static struct snd_soc_jack_gpio headset_button_gpio[] = { + { + .name = "Media Button", + .report = SND_JACK_BTN_0, + .debounce_time = 30, + .jack_status_check = headset_button_check, + }, +}; + +static int aries_spk_cfg(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_card *card = w->dapm->card; + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_component *component; + int ret = 0; + + rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]); + component = asoc_rtd_to_codec(rtd, 0)->component; + + /** + * We have an odd setup - the SPKMODE pin is pulled up so + * we only have access to the left side SPK configs, + * but SPKOUTR isn't bridged so when playing back in + * stereo, we only get the left hand channel. The only + * option we're left with is to force the AIF into mono + * mode. + */ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + ret = snd_soc_component_update_bits(component, + WM8994_AIF1_DAC1_FILTERS_1, + WM8994_AIF1DAC1_MONO, WM8994_AIF1DAC1_MONO); + break; + case SND_SOC_DAPM_PRE_PMD: + ret = snd_soc_component_update_bits(component, + WM8994_AIF1_DAC1_FILTERS_1, + WM8994_AIF1DAC1_MONO, 0); + break; + } + + return ret; +} + +static int aries_main_bias(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_card *card = w->dapm->card; + struct aries_wm8994_data *priv = snd_soc_card_get_drvdata(card); + int ret = 0; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + ret = regulator_enable(priv->reg_main_micbias); + break; + case SND_SOC_DAPM_POST_PMD: + ret = regulator_disable(priv->reg_main_micbias); + break; + } + + return ret; +} + +static int aries_headset_bias(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_card *card = w->dapm->card; + struct aries_wm8994_data *priv = snd_soc_card_get_drvdata(card); + int ret = 0; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + ret = regulator_enable(priv->reg_headset_micbias); + break; + case SND_SOC_DAPM_POST_PMD: + ret = regulator_disable(priv->reg_headset_micbias); + break; + } + + return ret; +} + +static const struct snd_kcontrol_new aries_controls[] = { + SOC_DAPM_PIN_SWITCH("Modem In"), + SOC_DAPM_PIN_SWITCH("Modem Out"), +}; + +static const struct snd_soc_dapm_widget aries_dapm_widgets[] = { + SND_SOC_DAPM_HP("HP", NULL), + + SND_SOC_DAPM_SPK("SPK", aries_spk_cfg), + SND_SOC_DAPM_SPK("RCV", NULL), + + SND_SOC_DAPM_LINE("LINE", NULL), + + SND_SOC_DAPM_MIC("Main Mic", aries_main_bias), + SND_SOC_DAPM_MIC("Headset Mic", aries_headset_bias), + + SND_SOC_DAPM_MIC("Bluetooth Mic", NULL), + SND_SOC_DAPM_SPK("Bluetooth SPK", NULL), + + SND_SOC_DAPM_LINE("Modem In", NULL), + SND_SOC_DAPM_LINE("Modem Out", NULL), + + /* This must be last as it is conditionally not used */ + SND_SOC_DAPM_LINE("FM In", NULL), +}; + +static int aries_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + unsigned int pll_out; + int ret; + + /* AIF1CLK should be >=3MHz for optimal performance */ + if (params_width(params) == 24) + pll_out = params_rate(params) * 384; + else if (params_rate(params) == 8000 || params_rate(params) == 11025) + pll_out = params_rate(params) * 512; + else + pll_out = params_rate(params) * 256; + + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, WM8994_FLL_SRC_MCLK1, + ARIES_MCLK1_FREQ, pll_out); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL1, + pll_out, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static int aries_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + int ret; + + /* Switch sysclk to MCLK1 */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_MCLK1, + ARIES_MCLK1_FREQ, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* Stop PLL */ + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, WM8994_FLL_SRC_MCLK1, + ARIES_MCLK1_FREQ, 0); + if (ret < 0) + return ret; + + return 0; +} + +/* + * Main DAI operations + */ +static struct snd_soc_ops aries_ops = { + .hw_params = aries_hw_params, + .hw_free = aries_hw_free, +}; + +static int aries_baseband_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + unsigned int pll_out; + int ret; + + pll_out = 8000 * 512; + + /* Set the codec FLL */ + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL2, WM8994_FLL_SRC_MCLK1, + ARIES_MCLK1_FREQ, pll_out); + if (ret < 0) + return ret; + + /* Set the codec system clock */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL2, + pll_out, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static int aries_late_probe(struct snd_soc_card *card) +{ + struct aries_wm8994_data *priv = snd_soc_card_get_drvdata(card); + int ret, irq; + + ret = snd_soc_card_jack_new(card, "Dock", SND_JACK_LINEOUT, + &aries_dock, dock_pins, ARRAY_SIZE(dock_pins)); + if (ret) + return ret; + + ret = devm_extcon_register_notifier(card->dev, + priv->usb_extcon, EXTCON_JACK_LINE_OUT, + &aries_extcon_notifier_block); + if (ret) + return ret; + + if (extcon_get_state(priv->usb_extcon, + EXTCON_JACK_LINE_OUT) > 0) + snd_soc_jack_report(&aries_dock, SND_JACK_LINEOUT, + SND_JACK_LINEOUT); + else + snd_soc_jack_report(&aries_dock, 0, SND_JACK_LINEOUT); + + ret = snd_soc_card_jack_new(card, "Headset", + SND_JACK_HEADSET | SND_JACK_BTN_0, + &aries_headset, + jack_pins, ARRAY_SIZE(jack_pins)); + if (ret) + return ret; + + ret = snd_soc_jack_add_zones(&aries_headset, ARRAY_SIZE(headset_zones), + headset_zones); + if (ret) + return ret; + + irq = gpiod_to_irq(priv->gpio_headset_detect); + if (irq < 0) { + dev_err(card->dev, "Failed to map headset detect gpio to irq"); + return -EINVAL; + } + + ret = devm_request_threaded_irq(card->dev, irq, NULL, + headset_det_irq_thread, + IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | + IRQF_ONESHOT, "headset_detect", priv); + if (ret) { + dev_err(card->dev, "Failed to request headset detect irq"); + return ret; + } + + headset_button_gpio[0].data = priv; + headset_button_gpio[0].desc = priv->gpio_headset_key; + + snd_jack_set_key(aries_headset.jack, SND_JACK_BTN_0, KEY_MEDIA); + + return snd_soc_jack_add_gpios(&aries_headset, + ARRAY_SIZE(headset_button_gpio), headset_button_gpio); +} + +static const struct snd_soc_pcm_stream baseband_params = { + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rate_min = 8000, + .rate_max = 8000, + .channels_min = 1, + .channels_max = 1, +}; + +static const struct snd_soc_pcm_stream bluetooth_params = { + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rate_min = 8000, + .rate_max = 8000, + .channels_min = 1, + .channels_max = 2, +}; + +static const struct snd_soc_dapm_widget aries_modem_widgets[] = { + SND_SOC_DAPM_INPUT("Modem RX"), + SND_SOC_DAPM_OUTPUT("Modem TX"), +}; + +static const struct snd_soc_dapm_route aries_modem_routes[] = { + { "Modem Capture", NULL, "Modem RX" }, + { "Modem TX", NULL, "Modem Playback" }, +}; + +static const struct snd_soc_component_driver aries_component = { + .name = "aries-audio", + .dapm_widgets = aries_modem_widgets, + .num_dapm_widgets = ARRAY_SIZE(aries_modem_widgets), + .dapm_routes = aries_modem_routes, + .num_dapm_routes = ARRAY_SIZE(aries_modem_routes), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static struct snd_soc_dai_driver aries_ext_dai[] = { + { + .name = "Voice call", + .playback = { + .stream_name = "Modem Playback", + .channels_min = 1, + .channels_max = 1, + .rate_min = 8000, + .rate_max = 8000, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "Modem Capture", + .channels_min = 1, + .channels_max = 1, + .rate_min = 8000, + .rate_max = 8000, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + }, +}; + +SND_SOC_DAILINK_DEFS(aif1, + DAILINK_COMP_ARRAY(COMP_CPU(SAMSUNG_I2S_DAI)), + DAILINK_COMP_ARRAY(COMP_CODEC(NULL, "wm8994-aif1")), + DAILINK_COMP_ARRAY(COMP_EMPTY())); + +SND_SOC_DAILINK_DEFS(baseband, + DAILINK_COMP_ARRAY(COMP_CPU("Voice call")), + DAILINK_COMP_ARRAY(COMP_CODEC(NULL, "wm8994-aif2"))); + +SND_SOC_DAILINK_DEFS(bluetooth, + DAILINK_COMP_ARRAY(COMP_CPU("bt-sco-pcm")), + DAILINK_COMP_ARRAY(COMP_CODEC(NULL, "wm8994-aif3"))); + +static struct snd_soc_dai_link aries_dai[] = { + { + .name = "WM8994 AIF1", + .stream_name = "HiFi", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, + .ops = &aries_ops, + SND_SOC_DAILINK_REG(aif1), + }, + { + .name = "WM8994 AIF2", + .stream_name = "Baseband", + .init = &aries_baseband_init, + .params = &baseband_params, + .ignore_suspend = 1, + SND_SOC_DAILINK_REG(baseband), + }, + { + .name = "WM8994 AIF3", + .stream_name = "Bluetooth", + .params = &bluetooth_params, + .ignore_suspend = 1, + SND_SOC_DAILINK_REG(bluetooth), + }, +}; + +static struct snd_soc_card aries_card = { + .name = "ARIES", + .owner = THIS_MODULE, + .dai_link = aries_dai, + .num_links = ARRAY_SIZE(aries_dai), + .controls = aries_controls, + .num_controls = ARRAY_SIZE(aries_controls), + .dapm_widgets = aries_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(aries_dapm_widgets), + .late_probe = aries_late_probe, +}; + +static const struct aries_wm8994_variant fascinate4g_variant = { + .modem_dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS + | SND_SOC_DAIFMT_IB_NF, + .has_fm_radio = false, +}; + +static const struct aries_wm8994_variant aries_variant = { + .modem_dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM + | SND_SOC_DAIFMT_IB_NF, + .has_fm_radio = true, +}; + +static const struct of_device_id samsung_wm8994_of_match[] = { + { + .compatible = "samsung,fascinate4g-wm8994", + .data = &fascinate4g_variant, + }, + { + .compatible = "samsung,aries-wm8994", + .data = &aries_variant, + }, + { /* sentinel */ }, +}; +MODULE_DEVICE_TABLE(of, samsung_wm8994_of_match); + +static int aries_audio_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct device_node *cpu, *codec, *extcon_np; + struct device *dev = &pdev->dev; + struct snd_soc_card *card = &aries_card; + struct aries_wm8994_data *priv; + struct snd_soc_dai_link *dai_link; + const struct of_device_id *match; + int ret, i; + + if (!np) + return -EINVAL; + + card->dev = dev; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + snd_soc_card_set_drvdata(card, priv); + + match = of_match_node(samsung_wm8994_of_match, np); + priv->variant = match->data; + + /* Remove FM widget if not present */ + if (!priv->variant->has_fm_radio) + card->num_dapm_widgets--; + + priv->reg_main_micbias = devm_regulator_get(dev, "main-micbias"); + if (IS_ERR(priv->reg_main_micbias)) { + dev_err(dev, "Failed to get main micbias regulator\n"); + return PTR_ERR(priv->reg_main_micbias); + } + + priv->reg_headset_micbias = devm_regulator_get(dev, "headset-micbias"); + if (IS_ERR(priv->reg_headset_micbias)) { + dev_err(dev, "Failed to get headset micbias regulator\n"); + return PTR_ERR(priv->reg_headset_micbias); + } + + priv->gpio_earpath_sel = devm_gpiod_get(dev, "earpath-sel", + GPIOD_OUT_LOW); + if (IS_ERR(priv->gpio_earpath_sel)) { + dev_err(dev, "Failed to get earpath selector gpio"); + return PTR_ERR(priv->gpio_earpath_sel); + } + + extcon_np = of_parse_phandle(np, "extcon", 0); + priv->usb_extcon = extcon_find_edev_by_node(extcon_np); + if (IS_ERR(priv->usb_extcon)) { + if (PTR_ERR(priv->usb_extcon) != -EPROBE_DEFER) + dev_err(dev, "Failed to get extcon device"); + return PTR_ERR(priv->usb_extcon); + } + of_node_put(extcon_np); + + priv->adc = devm_iio_channel_get(dev, "headset-detect"); + if (IS_ERR(priv->adc)) { + if (PTR_ERR(priv->adc) != -EPROBE_DEFER) + dev_err(dev, "Failed to get ADC channel"); + return PTR_ERR(priv->adc); + } + if (priv->adc->channel->type != IIO_VOLTAGE) + return -EINVAL; + + priv->gpio_headset_key = devm_gpiod_get(dev, "headset-key", + GPIOD_IN); + if (IS_ERR(priv->gpio_headset_key)) { + dev_err(dev, "Failed to get headset key gpio"); + return PTR_ERR(priv->gpio_headset_key); + } + + priv->gpio_headset_detect = devm_gpiod_get(dev, + "headset-detect", GPIOD_IN); + if (IS_ERR(priv->gpio_headset_detect)) { + dev_err(dev, "Failed to get headset detect gpio"); + return PTR_ERR(priv->gpio_headset_detect); + } + + /* Update card-name if provided through DT, else use default name */ + snd_soc_of_parse_card_name(card, "model"); + + ret = snd_soc_of_parse_audio_routing(card, "samsung,audio-routing"); + if (ret < 0) { + dev_err(dev, "Audio routing invalid/unspecified\n"); + return ret; + } + + aries_dai[1].dai_fmt = priv->variant->modem_dai_fmt; + + cpu = of_get_child_by_name(dev->of_node, "cpu"); + if (!cpu) + return -EINVAL; + + codec = of_get_child_by_name(dev->of_node, "codec"); + if (!codec) + return -EINVAL; + + for_each_card_prelinks(card, i, dai_link) { + dai_link->codecs->of_node = of_parse_phandle(codec, + "sound-dai", 0); + if (!dai_link->codecs->of_node) { + ret = -EINVAL; + goto out; + } + } + + /* Set CPU and platform of_node for main DAI */ + aries_dai[0].cpus->of_node = of_parse_phandle(cpu, + "sound-dai", 0); + if (!aries_dai[0].cpus->of_node) { + ret = -EINVAL; + goto out; + } + + aries_dai[0].platforms->of_node = aries_dai[0].cpus->of_node; + + /* Set CPU of_node for BT DAI */ + aries_dai[2].cpus->of_node = of_parse_phandle(cpu, + "sound-dai", 1); + if (!aries_dai[2].cpus->of_node) { + ret = -EINVAL; + goto out; + } + + ret = devm_snd_soc_register_component(dev, &aries_component, + aries_ext_dai, ARRAY_SIZE(aries_ext_dai)); + if (ret < 0) { + dev_err(dev, "Failed to register component: %d\n", ret); + goto out; + } + + ret = devm_snd_soc_register_card(dev, card); + if (ret) + dev_err(dev, "snd_soc_register_card() failed:%d\n", ret); + +out: + of_node_put(cpu); + of_node_put(codec); + + return ret; +} + +static struct platform_driver aries_audio_driver = { + .driver = { + .name = "aries-audio-wm8994", + .of_match_table = of_match_ptr(samsung_wm8994_of_match), + .pm = &snd_soc_pm_ops, + }, + .probe = aries_audio_probe, +}; + +module_platform_driver(aries_audio_driver); + +MODULE_DESCRIPTION("ALSA SoC ARIES WM8994"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:aries-audio-wm8994"); -- 2.20.1

1 0

[PATCH 2/3] dt-bindings: sound: Add bindings for Samsung Aries audio complex
by Jonathan Bakker 14 Jun '20

14 Jun '20

The audio system on S5PV210 Aries boards have a wm8994 codec connected to the Samsung I2S0 DAI. Jack detection is done via GPIOs, an ADC, and an extcon device (fsa9480). There are two main variants, one with an FM radio and where the wm8994 is the master for the modem audio and the other without FM and the modem is the master. Signed-off-by: Jonathan Bakker <xc-racer2(a)live.ca> --- .../bindings/sound/samsung,aries-wm8994.yaml | 147 ++++++++++++++++++ 1 file changed, 147 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml diff --git a/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml b/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml new file mode 100644 index 000000000000..902a0b66628e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml @@ -0,0 +1,147 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung,aries-wm8994.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Samsung Aries audio complex with WM8994 codec + +maintainers: + - Jonathan Bakker <xc-racer2(a)live.ca> + +properties: + compatible: + oneOf: + - const: samsung,aries-wm8994 + description: With FM radio and modem master + + - const: samsung,fascinate4g-wm8994 + description: Without FM radio and modem slave + + model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex. + + cpu: + type: object + properties: + sound-dai: + minItems: 2 + maxItems: 2 + $ref: /schemas/types.yaml#/definitions/phandle-array + description: | + phandles to the I2S controller and bluetooth codec, + in that order + + codec: + type: object + properties: + sound-dai: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: phandle to the WM8994 CODEC + + samsung,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + List of the connections between audio + components; each entry is a pair of strings, the first being the + connection's sink, the second being the connection's source; + valid names for sources and sinks are the WM8994's pins (as + documented in its binding), and the jacks on the board - + For samsung,aries-wm8994: HP, SPK, RCV, LINE, Main Mic, Headset Mic, + or FM In + For samsung,fascinate4g-wm8994: HP, SPK, RCV, LINE, Main Mic, + or HeadsetMic + + extcon: + description: Extcon phandle for dock detection + + main-micbias-supply: + description: Supply for the micbias on the main mic + + headset-micbias-supply: + description: Supply for the micbias on the headset mic + + earpath-sel-gpios: + description: GPIO for switching between tv-out and mic paths + + headset-detect-gpios: + description: GPIO for detection of headset insertion + + headset-key-gpios: + description: GPIO for detection of headset key press + + io-channels: + maxItems: 1 + description: IO channel to read micbias voltage for headset detection + + io-channel-names: + const: headset-detect + +required: + - compatible + - model + - cpu + - codec + - samsung,audio-routing + - extcon + - main-micbias-supply + - headset-micbias-supply + - earpath-sel-gpios + - headset-detect-gpios + - headset-key-gpios + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + sound { + compatible = "samsung,fascinate4g-wm8994"; + + model = "Fascinate4G"; + + extcon = <&fsa9480>; + + main-micbias-supply = <&main_micbias_reg>; + headset-micbias-supply = <&headset_micbias_reg>; + + earpath-sel-gpios = <&gpj2 6 GPIO_ACTIVE_HIGH>; + + io-channels = <&adc 3>; + io-channel-names = "headset-detect"; + headset-detect-gpios = <&gph0 6 GPIO_ACTIVE_HIGH>; + headset-key-gpios = <&gph3 6 GPIO_ACTIVE_HIGH>; + + samsung,audio-routing = + "HP", "HPOUT1L", + "HP", "HPOUT1R", + + "SPK", "SPKOUTLN", + "SPK", "SPKOUTLP", + + "RCV", "HPOUT2N", + "RCV", "HPOUT2P", + + "LINE", "LINEOUT2N", + "LINE", "LINEOUT2P", + + "IN1LP", "Main Mic", + "IN1LN", "Main Mic", + + "IN1RP", "Headset Mic", + "IN1RN", "Headset Mic"; + + pinctrl-names = "default"; + pinctrl-0 = <&headset_det &earpath_sel>; + + cpu { + sound-dai = <&i2s0>, <&bt_codec>; + }; + + codec { + sound-dai = <&wm8994>; + }; + }; + -- 2.20.1

1 0

[PATCH 0/3] ASoC: samsung: Add machine driver for Aries boards
by Jonathan Bakker 14 Jun '20

14 Jun '20

This is a machine driver for Aries boards which are based on Samsung's S5PV210 platform. These are the first-gen Galaxy S devices. It consists of a WM8994 codec connected to I2S0 along with two codec-to-codec links to the modem and the bluetooth. --------- | | Modem | | --------- | AIF2 | ---------- --------- | | AIF1 | | I2S ------- wm8994 | | | | ---------- --------- | AIF3 | --------- | | BT | | --------- Jack detection is relatively complex with various GPIOs, an ADC, and an extcon device all involved. One oddity is that the headphone jack is shared between the mic and the analog tv-out and is controlled by a GPIO (earpath-sel) and the proper path should be selected based on the jack detection. There is also an oddity with how the speaker is connected. It's a mono speaker and the spkmode pin is pulled up, but the connections aren't bridged so if stereo playback is attempted, only the one channel will play. Therefore, when the speaker is enabled the entire AIF should be forced into mono mode. UCM2 config files are available at https://github.com/xc-racer99/aries-hw-files/tree/ucm2/usr/share/alsa/ucm2 Jonathan Bakker (3): dt-bindings: sound: Document wm8994 endpoints dt-bindings: sound: Add bindings for Samsung Aries audio complex ASoC: samsung: Add driver for Aries boards .../bindings/sound/samsung,aries-wm8994.yaml | 147 ++++ .../devicetree/bindings/sound/wm8994.txt | 23 + sound/soc/samsung/Kconfig | 13 + sound/soc/samsung/Makefile | 2 + sound/soc/samsung/aries_wm8994.c | 695 ++++++++++++++++++ 5 files changed, 880 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml create mode 100644 sound/soc/samsung/aries_wm8994.c -- 2.20.1

1 0

alsa usb problem
by GitHub issues - opened 14 Jun '20

14 Jun '20

alsa-project/alsa-lib issue #59 was opened from texstar: Plugging in a usb microphone causes pulse audio to crash with alsa-lib 1.2.3. Issue URL : https://github.com/alsa-project/alsa-lib/issues/59 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Re: next-0519 on thinkpad x60: sound related? window manager crash
by Alex Xu (Hello71) 14 Jun '20

14 Jun '20

Excerpts from Takashi Iwai's message of June 9, 2020 11:12 am: > On Tue, 09 Jun 2020 13:47:33 +0200, > Christoph Hellwig wrote: >> >> Alex, can you try this patch? > > Also could you check whether just papering over the memset() call > alone avoids the crash like below? For PulseAudio and dmix/dsnoop, > it's the only code path that accesses the vmapped buffer, I believe. > > If this works more or less, I'll cook a more comprehensive fix. > > > thanks, > > Takashi > > --- a/sound/core/pcm_native.c > +++ b/sound/core/pcm_native.c > @@ -754,9 +754,11 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, > while (runtime->boundary * 2 <= LONG_MAX - runtime->buffer_size) > runtime->boundary *= 2; > > +#if 0 > /* clear the buffer for avoiding possible kernel info leaks */ > if (runtime->dma_area && !substream->ops->copy_user) > memset(runtime->dma_area, 0, runtime->dma_bytes); > +#endif > > snd_pcm_timer_resolution_change(substream); > snd_pcm_set_state(substream, SNDRV_PCM_STATE_SETUP); > Sorry, this patch doesn't work for me with SME off using abfbb29297c2. David's newest submitted patch works for me, which I already replied to separately. Thanks, Alex.

2 5

[PATCH] ALSA: firewire-motu: wait for notification when changing clock configuration for protocol v3
by Takashi Sakamoto 14 Jun '20

14 Jun '20

This patch is for development period of v5.9 kernel. It costs expensive to change clock configuration for models of protocol version 3. In current implementation, speculative strategy is used to finish the operation; just waiting for 4 seconds. As long as I investigate, when accepting and changing clock status actually, the device sends notification with mask. This commit uses wise way to wait for the notification after changing sampling clock rate during 4 seconds. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/motu/motu-protocol-v3.c | 16 +++++++++++++--- 1 file changed, 13 insertions(+), 3 deletions(-) diff --git a/sound/firewire/motu/motu-protocol-v3.c b/sound/firewire/motu/motu-protocol-v3.c index 01a47ac7bb2d..4e6b0e449ee4 100644 --- a/sound/firewire/motu/motu-protocol-v3.c +++ b/sound/firewire/motu/motu-protocol-v3.c @@ -24,6 +24,9 @@ #define V3_NO_ADAT_OPT_OUT_IFACE_A 0x00040000 #define V3_NO_ADAT_OPT_OUT_IFACE_B 0x00400000 +#define V3_MSG_FLAG_CLK_CHANGED 0x00000002 +#define V3_CLK_WAIT_MSEC 4000 + int snd_motu_protocol_v3_get_clock_rate(struct snd_motu *motu, unsigned int *rate) { @@ -79,9 +82,16 @@ int snd_motu_protocol_v3_set_clock_rate(struct snd_motu *motu, return err; if (need_to_wait) { - /* Cost expensive. */ - if (msleep_interruptible(4000) > 0) - return -EINTR; + int result; + + motu->msg = 0; + result = wait_event_interruptible_timeout(motu->hwdep_wait, + motu->msg & V3_MSG_FLAG_CLK_CHANGED, + msecs_to_jiffies(V3_CLK_WAIT_MSEC)); + if (result < 0) + return result; + if (result == 0) + return -ETIMEDOUT; } return 0; -- 2.25.1

1 0

alsa-gobject: add Python 3 sample scripts
by GitHub pull_request - opened 14 Jun '20

14 Jun '20

alsa-project/alsa-gobject pull request #37 was opened from takaswie: PyGObject is a loader for GObject Introspection in Python 3 runtime. This patchset adds Python 3 sample scripts to use ALSACtl, ALSAHwdep, ALSARawmidi, ALSATimer, and ALSASeq loaded by PyGObject. Request URL : https://github.com/alsa-project/alsa-gobject/pull/37 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/37.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

alsa-gobject: seq: fix port creation API
by GitHub pull_request - opened 14 Jun '20

14 Jun '20

alsa-project/alsa-gobject pull request #36 was opened from takaswie: ALSASeq.UserClient.create_port() has nullable argument for the numerical ID of port to use SNDRV_SEQ_PORT_FLG_GIVEN_PORT flag properly. However, the nullable is not inconvenient for PyGObject and causes SIGSEGV. This patchset adds an alternative API, ALSASeq.UserClient.create_port_at() for the flag, then ALSASeq.UserClient.create_port() hs no argument for explicit port ID. Additionally, this patchset includes some fixes for the other parts. Request URL : https://github.com/alsa-project/alsa-gobject/pull/36 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/36.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH] ALSA: usb-audio: add quirk for Denon DCD-1500RE
by Yick W. Tse 14 Jun '20

14 Jun '20

fix error "clock source 41 is not valid, cannot use" [] New USB device found, idVendor=154e, idProduct=1002, bcdDevice= 1.00 [] New USB device strings: Mfr=1, Product=2, SerialNumber=0 [] Product: DCD-1500RE [] Manufacturer: D & M Holdings Inc. [] [] clock source 41 is not valid, cannot use [] usbcore: registered new interface driver snd-usb-audio Signed-off-by: Yick W. Tse <y_w_tse(a)yahoo.com.hk> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index bca0179a0ef8..c495e720e2f1 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1532,6 +1532,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) static bool is_itf_usb_dsd_dac(unsigned int id) { switch (id) { + case USB_ID(0x154e, 0x1002): /* Denon DCD-1500RE */ case USB_ID(0x154e, 0x1003): /* Denon DA-300USB */ case USB_ID(0x154e, 0x3005): /* Marantz HD-DAC1 */ case USB_ID(0x154e, 0x3006): /* Marantz SA-14S1 */ --

2 1

Cannot get Lenovo P520/Realtek ALC662 (rev 3) to produce sound
by Jan Pokorny 14 Jun '20

14 Jun '20

Hello list, this time within wider circle (and with more experience so far). I am seeking for help with shiny new Lenovo P520 machine that only lacks one thing for me to be entirely happy it. I believe this is relatively a new batch -- intentionally mentioning it since the topic of this very machine appears to be a settled case already: https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/commit/?… With newest BIOS, running the most recent kernel from Fedora Rawhide (but I did try 5.5.10 kernel from Fedora 30 as well), booted via UEFI (incl. secure boot), the only way to get it to produce any sound whatsoever is to manually load snd-pcsp [*1]. With that done, virtual terminal starts to beep as expected (as in printf '\a'), I am able to run "beep" program, and even "aplay -D hw:pcsp /usr/share/sound/alsa/*.wav" is producing the respective sounds, despite suffering from surrounding noise[*]. That validates that basic function of the built-in speaker works just fine (and previously, it was validated under Windows that it worked just fine). As the linked commit indicates, the proper model parameter to snd-hda-audio module is dual-codecs. I've tried that, and a handful of other models, to no avail. I considered the possibility the problem is just that I cannot work with "mute" state properly, so tried everything I could (incl. alsaunmute), again, without any change. Next dimension to the matrix was attaching external speaker to the front connector. Nothing has helped. Comparing outputs from alsa-info.sh for both states, "no model specified" and "dual-codecs" specified, there was no observable change other than this sole "model" parameter not/being (null). I am not sure whether this means this particular predefined quirk was applied properly, or the same observation is to be expected in general. Note that said quirk is, IIUIC, normally hooked onto the presence of "Subsystem: Lenovo Device 1036" (as in from lspci), and it is what this machine assuredly satisfies. I am attaching the output from the plain/default case, and looking forward to being able to move at least a bit with this. I'd be fine, for the time being, to use the sketched "PC speaker" workaround, if only it wasn't spoiled with so much noise/hum (no audio connector was used at that time). But that won't solve the urging general teleconferencing problem of these days, apparently. [*1] at a matter of fact, even it should play no role here, there was a change between this 5.7.0.rc7 kernel and the other tested, 5.5.10, in that the respective (another, IIRC) module got loaded automatically in the latter/older, whereas it needed to be loaded manually with the former/newer [*2] perhaps resembled https://bugzilla.redhat.com/show_bug.cgi?id=1660581 that I found when trying to do an extensive search around similar problems Thanks -- Jan

3 6

dlmisc.c: fix uclibc build
by GitHub pull_request - opened 13 Jun '20

13 Jun '20

alsa-project/alsa-lib pull request #58 was opened from bkuhls: RTLD_DL_LINKMAP & RTLD_DI_ORIGIN, are unsupported on uClibc: https://cgit.uclibc-ng.org/cgi/cgit/uclibc-ng.git/tree/include/dlfcn.h#n106 This patch adds detection for uClibc because uClibc also defines __GLIBC__: https://lists.gnu.org/archive/html/bug-gnulib/2010-11/msg00280.html Request URL : https://github.com/alsa-project/alsa-lib/pull/58 Patch URL : https://github.com/alsa-project/alsa-lib/pull/58.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

alsa-gobject: obsolete function with two arguments
by GitHub pull_request - opened 13 Jun '20

13 Jun '20

alsa-project/alsa-gobject pull request #35 was opened from takaswie: According to guideline of GObject Introspection, API with multiple out parameters is not preferrable[1]. This patchset replaces such functions with more suitable ones. [1] https://gi.readthedocs.io/en/latest/writingbindableapis.html#multiple-out-p… Request URL : https://github.com/alsa-project/alsa-gobject/pull/35 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/35.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

alsa-gobject: misc fixes and documentation for system calls
by GitHub pull_request - opened 13 Jun '20

13 Jun '20

alsa-project/alsa-gobject pull request #34 was opened from takaswie: This patchset includes misc fixes and documentation for system calls for each methods. Request URL : https://github.com/alsa-project/alsa-gobject/pull/34 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/34.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH] ASoC: fsl_ssi: Fix bclk calculation for mono channel
by Shengjiu Wang 12 Jun '20

12 Jun '20

For mono channel, ssi will switch to normal mode. In normal 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. 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> --- sound/soc/fsl/fsl_ssi.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index bad89b0d129e..e347776590f7 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -695,6 +695,11 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, /* Generate bit clock based on the slot number and slot width */ freq = slots * slot_width * params_rate(hw_params); + /* The slot_width is not fixed to 32 for normal mode */ + if (params_channels(hw_params) == 1) + freq = (slots <= 1 ? 2 : slots) * params_width(hw_params) * + params_rate(hw_params); + /* Don't apply it to any non-baudclk circumstance */ if (IS_ERR(ssi->baudclk)) return -EINVAL; -- 2.21.0

2 1

[alsa-devel] [PATCH 0/2 v4] ALSA: pcm: anonymous dup implementation
by Jaroslav Kysela 12 Jun '20

12 Jun '20

This patchset contains the anonymous dup implementation with permissions checking for the ALSA's PCM interface in kernel to enable the restricted DMA sound buffer sharing for the restricted tasks. The code was tested through qemu and it seems to be pretty stable. The initial tinyalsa implementation can be found here: https://github.com/perexg/tinyalsa/commits/anondup v2 of the patches: - change clone parameter to subdevice number for the pcm attach - change SNDRV_PCM_PERM_MAX to SNDRV_PCM_PERM_MASK - the tinyalsa implementation was a little updated (restructured) v3 of the patches: - group integer declarations in snd_pcm_anonymous_dup() - replaced substream->pcm with pcm in snd_pcm_anonymous_dup() - added SNDRV_PCM_PERM_RW check for read/write/readv/writev syscalls v4 of the patches: - more simple restriction control (only two modes - full/buffer) - the tinyalsa implementation follows this change Cc: Phil Burk <philburk(a)google.com> Cc: Zach Riggle <riggle(a)google.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Leo Yan <leo.yan(a)linaro.org> Cc: Baolin Wang <baolin.wang(a)linaro.org> Jaroslav Kysela (2): ALSA: pcm: implement the anonymous dup (inode file descriptor) ALSA: pcm: implement the mmap buffer mode for the anonymous dup include/sound/pcm.h | 10 +++-- include/uapi/sound/asound.h | 6 ++- sound/core/oss/pcm_oss.c | 2 +- sound/core/pcm.c | 13 +++--- sound/core/pcm_compat.c | 1 + sound/core/pcm_native.c | 97 ++++++++++++++++++++++++++++++++++++++++++--- 6 files changed, 110 insertions(+), 19 deletions(-) -- 2.13.6

4 17

[PATCH v3 1/2] dt-bindings: tas2562: Convert the tas2562 binding to yaml
by Dan Murphy 12 Jun '20

12 Jun '20

Convert the TAS2562 text file to yaml format. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- .../devicetree/bindings/sound/tas2562.txt | 34 ---------- .../devicetree/bindings/sound/tas2562.yaml | 65 +++++++++++++++++++ 2 files changed, 65 insertions(+), 34 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/tas2562.txt create mode 100644 Documentation/devicetree/bindings/sound/tas2562.yaml diff --git a/Documentation/devicetree/bindings/sound/tas2562.txt b/Documentation/devicetree/bindings/sound/tas2562.txt deleted file mode 100644 index 94796b547184..000000000000 --- a/Documentation/devicetree/bindings/sound/tas2562.txt +++ /dev/null @@ -1,34 +0,0 @@ -Texas Instruments TAS2562 Smart PA - -The TAS2562 is a mono, digital input Class-D audio amplifier optimized for -efficiently driving high peak power into small loudspeakers. -Integrated speaker voltage and current sense provides for -real time monitoring of loudspeaker behavior. - -Required properties: - - #address-cells - Should be <1>. - - #size-cells - Should be <0>. - - compatible: - Should contain "ti,tas2562", "ti,tas2563". - - reg: - The i2c address. Should be 0x4c, 0x4d, 0x4e or 0x4f. - - ti,imon-slot-no:- TDM TX current sense time slot. - -Optional properties: -- interrupt-parent: phandle to the interrupt controller which provides - the interrupt. -- interrupts: (GPIO) interrupt to which the chip is connected. -- shut-down: GPIO used to control the state of the device. - -Examples: -tas2562@4c { - #address-cells = <1>; - #size-cells = <0>; - compatible = "ti,tas2562"; - reg = <0x4c>; - - interrupt-parent = <&gpio1>; - interrupts = <14>; - - shut-down = <&gpio1 15 0>; - ti,imon-slot-no = <0>; -}; - diff --git a/Documentation/devicetree/bindings/sound/tas2562.yaml b/Documentation/devicetree/bindings/sound/tas2562.yaml new file mode 100644 index 000000000000..bb51fd88c41a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2562.yaml @@ -0,0 +1,65 @@ +# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause) +# Copyright (C) 2019 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: "http://devicetree.org/schemas/sound/tas2562.yaml#" +$schema: "http://devicetree.org/meta-schemas/core.yaml#" + +title: Texas Instruments TAS2562 Smart PA + +maintainers: + - Dan Murphy <dmurphy(a)ti.com> + +description: | + The TAS2562 is a mono, digital input Class-D audio amplifier optimized for + efficiently driving high peak power into small loudspeakers. + Integrated speaker voltage and current sense provides for + real time monitoring of loudspeaker behavior. + +properties: + compatible: + enum: + - ti,tas2562 + - ti,tas2563 + + reg: + maxItems: 1 + description: | + I2C address of the device can be one of these 0x4c, 0x4d, 0x4e or 0x4f + + shutdown-gpio: + description: GPIO used to control the state of the device. + + interrupts: + maxItems: 1 + + ti,imon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + description: TDM TX current sense time slot. + + '#sound-dai-cells': + const: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c0 { + #address-cells = <1>; + #size-cells = <0>; + codec: codec@4c { + compatible = "ti,tas2562"; + reg = <0x4c>; + #sound-dai-cells = <1>; + interrupt-parent = <&gpio1>; + interrupts = <14>; + shutdown-gpio = <&gpio1 15 0>; + ti,imon-slot-no = <0>; + }; + }; + -- 2.26.2

2 7

[PATCH] ASoC: rt5682: Register clocks even when mclk is NULL
by Akshu Agrawal 12 Jun '20

12 Jun '20

Fixes kernel crash on platforms which do not have mclk exposed in CCF framework. For these platforms have mclk as NULL and continue to register clocks. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/codecs/rt5682.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index d3245123101d..7c8b8d839db8 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -2829,12 +2829,11 @@ static int rt5682_probe(struct snd_soc_component *component) return ret; } rt5682->mclk = NULL; - } else { - /* Register CCF DAI clock control */ - ret = rt5682_register_dai_clks(component); - if (ret) - return ret; } + /* Register CCF DAI clock control */ + ret = rt5682_register_dai_clks(component); + if (ret) + return ret; /* Initial setup for CCF */ rt5682->lrck[RT5682_AIF1] = CLK_48; #endif -- 2.20.1

3 2

[PATCH v2 0/6] Enable Greybus Audio codec driver
by Vaibhav Agarwal 12 Jun '20

12 Jun '20

The existing GB Audio codec driver is dependent on MSM8994 Audio driver. During the development stage, this dependency was configured due to various changes involved in MSM Audio driver to enable addtional codec card and some of the changes proposed in mainline ASoC framework. However, these are not the real dependencies and some of them can be easily removed. The folowing patch series includes the changes to resolve unnecessary depedencies and make the codec driver functional with the latest kernel. Patch 1,2: Incudes jack framework related changes. Patch 3,4,5: Resolves compilation error observed with the latest kernel and also provides helper APIs required to allow dynamic addition/removal of modules. Patch 6: Finally provides config options and related Makefile changes to enable GB Codec driver. Thanks to Alexandre for raising the headsup [1] and motivating me to provide the necessary changes. [1] https://lore.kernel.org/lkml/20200507212912.599433-1-alexandre.belloni@boot… Changes from v1 - Include the changes for the review comments suggested by Dan - Rebase to latest staging-next Vaibhav Agarwal (6): staging: greybus: audio: Update snd_jack FW usage as per new APIs staging: greybus: audio: Maintain jack list within GB Audio module staging: greybus: audio: Resolve compilation errors for GB codec module staging: greybus: audio: Resolve compilation error in topology parser staging: greybus: audio: Add helper APIs for dynamic audio modules staging: greybus: audio: Enable GB codec, audio module compilation. drivers/staging/greybus/Kconfig | 14 +- drivers/staging/greybus/Makefile | 6 +- drivers/staging/greybus/audio_codec.c | 178 +++++++++++--------- drivers/staging/greybus/audio_codec.h | 12 +- drivers/staging/greybus/audio_helper.c | 197 +++++++++++++++++++++++ drivers/staging/greybus/audio_helper.h | 17 ++ drivers/staging/greybus/audio_module.c | 15 +- drivers/staging/greybus/audio_topology.c | 128 +++++++-------- 8 files changed, 412 insertions(+), 155 deletions(-) create mode 100644 drivers/staging/greybus/audio_helper.c create mode 100644 drivers/staging/greybus/audio_helper.h base-commit: af7b4801030c07637840191c69eb666917e4135d -- 2.26.2

4 16

[RFC PATCH v3 0/4] Reuse the dma channel if available in Back-End
by Shengjiu Wang 12 Jun '20

12 Jun '20

Reuse the dma channel if available in Back-End Shengjiu Wang (4): ASoC: soc-card: export snd_soc_lookup_component_nolocked ASoC: dmaengine_pcm: export soc_component_to_pcm ASoC: fsl_asrc_dma: Reuse the dma channel if available in Back-End ASoC: fsl_asrc_dma: Fix data copying speed issue with EDMA changes in v3: - update according to Nicolin's comments - split previous 0003 patch to two patches changes in v2: - update according to Mark's comments and split the patch include/sound/dmaengine_pcm.h | 11 +++++++ include/sound/soc.h | 2 ++ sound/soc/fsl/fsl_asrc_common.h | 2 ++ sound/soc/fsl/fsl_asrc_dma.c | 47 +++++++++++++++++++-------- sound/soc/soc-core.c | 3 +- sound/soc/soc-generic-dmaengine-pcm.c | 12 ------- 6 files changed, 50 insertions(+), 27 deletions(-) -- 2.21.0

3 9

alsa-gobject: timer: user_instance: add API to choose the type of event to read
by GitHub pull_request - opened 12 Jun '20

12 Jun '20

alsa-project/alsa-gobject pull request #33 was opened from takaswie: Currently ALSATimer.UserInstance has any attach API to decide the type of event to read. In design of ALSA Timer core, the attach API has two functionality; to attach or to detach with error for absent timer device. Current implementation has inconvenient for the functionality due to execution to choose the type of event in advance. The execution fails when the instance is already attached. This commit adds a new API to choose the type of event and split the call from any attach API. Request URL : https://github.com/alsa-project/alsa-gobject/pull/33 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/33.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH] Add implicit feedback quirk for SSL2+.
by Laurence Tratt 12 Jun '20

12 Jun '20

This uses the same quirk as the Motu M2 and M4 to ensure the driver uses the audio interface's clock. Tested on an SSL2+. 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 8a05dcb13..84c0ae431 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, 0x0002): /* Solid State Logic SSL2+ */ ep = 0x81; ifnum = 2; goto add_sync_ep_from_ifnum; -- 2.27.0

2 1

Re: [PATCH v2] ASoC: SOF: Intel: hda: unsolicited RIRB response
by Kai Vehmanen 12 Jun '20

12 Jun '20

Hi Brent, On Fri, 12 Jun 2020, Brent Lu wrote: > Port commit 6d011d5057ff ("ALSA: hda: Clear RIRB status before reading > WP") from legacy HDA driver to fix the get response timeout issue. > Current SOF driver does not suffer from this issue because sync write > is enabled in hda_init. The issue will come back if the sync write is > disabled for some reason. better to align the logic, so I'm ok to take this patch to SOF. Can you fix the summary though: - ASoC: SOF: Intel: hda: unsolicited RIRB response + ASoC: SOF: Intel: hda: Clear RIRB status before reading WP Br, Kai

2 1

Re: Re: [PATCH v5 0/3] ASoC: Intel: Add KeemBay ASoC platform driver
by Sia, Jee Heng 12 Jun '20

12 Jun '20

Hi, Please merge the patches if there is no further comments. Thanks Regards Jee Heng

2 1

[PATCH] Add implicit feedback quirk for SSL2+.
by Laurence Tratt 12 Jun '20

12 Jun '20

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 8a05dcb13..84c0ae431 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, 0x0002): /* Solid State Logic SSL2+ */ ep = 0x81; ifnum = 2; goto add_sync_ep_from_ifnum; -- 2.27.0

2 1

[PATCH] ASoC: SOF: Intel: hda: unsolicited RIRB response
by Brent Lu 12 Jun '20

12 Jun '20

The loop implementation could not solve the unsolicited response issue because the RIRBSTS is cleared after leaving the snd_hdac_bus_update_rirb() function. So the next loop will fail the status test against the RIRB_INT_MASK and skip all the RIRB handling stuff. On the other hand, there alwasy could be unsolicited responses in the last loop regardless the number of loops. Clear the RIRB interrupt before handling it so unsolicited response could trigger another RIRB interrupt to handle it later. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/sof/intel/hda-stream.c | 48 +++++++++++++++++----------------------- 1 file changed, 20 insertions(+), 28 deletions(-) diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index 7f65dcc..d21ac42 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -589,11 +589,10 @@ hda_dsp_set_bytes_transferred(struct hdac_stream *hstream, u64 buffer_size) hstream->curr_pos += num_bytes; } -static bool hda_dsp_stream_check(struct hdac_bus *bus, u32 status) +static void hda_dsp_stream_check(struct hdac_bus *bus, u32 status) { struct sof_intel_hda_dev *sof_hda = bus_to_sof_hda(bus); struct hdac_stream *s; - bool active = false; u32 sd_status; list_for_each_entry(s, &bus->stream_list, list) { @@ -605,7 +604,6 @@ static bool hda_dsp_stream_check(struct hdac_bus *bus, u32 status) snd_hdac_stream_writeb(s, SD_STS, sd_status); - active = true; if ((!s->substream && !s->cstream) || !s->running || (sd_status & SOF_HDA_CL_DMA_SD_INT_COMPLETE) == 0) @@ -621,8 +619,6 @@ static bool hda_dsp_stream_check(struct hdac_bus *bus, u32 status) } } } - - return active; } irqreturn_t hda_dsp_stream_threaded_handler(int irq, void *context) @@ -632,37 +628,33 @@ irqreturn_t hda_dsp_stream_threaded_handler(int irq, void *context) #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) u32 rirb_status; #endif - bool active; u32 status; - int i; - /* - * Loop 10 times to handle missed interrupts caused by - * unsolicited responses from the codec - */ - for (i = 0, active = true; i < 10 && active; i++) { - spin_lock_irq(&bus->reg_lock); + spin_lock_irq(&bus->reg_lock); - status = snd_hdac_chip_readl(bus, INTSTS); + status = snd_hdac_chip_readl(bus, INTSTS); - /* check streams */ - active = hda_dsp_stream_check(bus, status); + /* check streams */ + hda_dsp_stream_check(bus, status); - /* check and clear RIRB interrupt */ + /* check and clear RIRB interrupt */ #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) - if (status & AZX_INT_CTRL_EN) { - rirb_status = snd_hdac_chip_readb(bus, RIRBSTS); - if (rirb_status & RIRB_INT_MASK) { - active = true; - if (rirb_status & RIRB_INT_RESPONSE) - snd_hdac_bus_update_rirb(bus); - snd_hdac_chip_writeb(bus, RIRBSTS, - RIRB_INT_MASK); - } + if (status & AZX_INT_CTRL_EN) { + rirb_status = snd_hdac_chip_readb(bus, RIRBSTS); + if (rirb_status & RIRB_INT_MASK) { + /* + * clear current interrupt before reading RIRBWP + * so unsolicited response could trigger another + * interrupt + */ + snd_hdac_chip_writeb(bus, RIRBSTS, RIRB_INT_MASK); + + if (rirb_status & RIRB_INT_RESPONSE) + snd_hdac_bus_update_rirb(bus); } -#endif - spin_unlock_irq(&bus->reg_lock); } +#endif + spin_unlock_irq(&bus->reg_lock); return IRQ_HANDLED; } -- 2.7.4

5 9

[PATCH v2] ASoC: SOF: Intel: hda: unsolicited RIRB response
by Brent Lu 12 Jun '20

12 Jun '20

Port commit 6d011d5057ff ("ALSA: hda: Clear RIRB status before reading WP") from legacy HDA driver to fix the get response timeout issue. Current SOF driver does not suffer from this issue because sync write is enabled in hda_init. The issue will come back if the sync write is disabled for some reason. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/sof/intel/hda-stream.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index 7f65dcc..1bda14c 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -653,11 +653,16 @@ irqreturn_t hda_dsp_stream_threaded_handler(int irq, void *context) if (status & AZX_INT_CTRL_EN) { rirb_status = snd_hdac_chip_readb(bus, RIRBSTS); if (rirb_status & RIRB_INT_MASK) { + /* + * Clearing the interrupt status here ensures + * that no interrupt gets masked after the RIRB + * wp is read in snd_hdac_bus_update_rirb. + */ + snd_hdac_chip_writeb(bus, RIRBSTS, + RIRB_INT_MASK); active = true; if (rirb_status & RIRB_INT_RESPONSE) snd_hdac_bus_update_rirb(bus); - snd_hdac_chip_writeb(bus, RIRBSTS, - RIRB_INT_MASK); } } #endif -- 2.7.4

1 0

[RFC PATCH v2 0/3] ASoC: fsl_asrc_dma: Reuse the dma channel if available in Back-End
by Shengjiu Wang 12 Jun '20

12 Jun '20

Reuse the dma channel if available in Back-End Shengjiu Wang (3): ASoC: soc-card: export snd_soc_lookup_component_nolocked ASoC: dmaengine_pcm: export soc_component_to_pcm ASoC: fsl_asrc_dma: Reuse the dma channel if available in Back-End changes in v2: - update according to Mark's comments and split the patch include/sound/dmaengine_pcm.h | 11 ++++++ include/sound/soc.h | 2 ++ sound/soc/fsl/fsl_asrc_common.h | 2 ++ sound/soc/fsl/fsl_asrc_dma.c | 52 ++++++++++++++++++++------- sound/soc/soc-core.c | 3 +- sound/soc/soc-generic-dmaengine-pcm.c | 12 ------- 6 files changed, 57 insertions(+), 25 deletions(-) -- 2.21.0

3 6

alsa-gobject: ctl: obsolete abstract class for snd_ctl_elem_info structure
by GitHub pull_request - opened 12 Jun '20

12 Jun '20

alsa-project/alsa-gobject pull request #32 was opened from takaswie: In UAPI of Linux sound subsystem, snd_ctl_elem_info structure includes union member. Current implementation of ALSACtl uses abstract class (ALSACtlElemInfo) and concrete class (ALSACtlElemInfoBool and so on) to represent variation corresponding to each of union member. However, the design pattern is not convenient for some language bindings. For example, in Rust language, class-based inheritance is not supported. Gtk-rs project produces mechanism to upcast/downcast between super/sub classes and handles GObject class system. In my opinion, the code to upcast/downcast is a bit inconvenient to handle derivatives. This patchset obsoletes usage of abstract class for ALSACtl. The target is ALSACtlElemInfo. Some APIs are added into the object to retrieve data for integer/integer64/enumerated element type, then the object becomes concrete class. Request URL : https://github.com/alsa-project/alsa-gobject/pull/32 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/32.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH] ALSA: hda: Add NVIDIA codec IDs 9a & 9d through a0 to patch table
by Aaron Plattner 11 Jun '20

11 Jun '20

Signed-off-by: Aaron Plattner <aplattner(a)nvidia.com> --- sound/pci/hda/patch_hdmi.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index fbd7cc6026d8..e2b21ef5d7d1 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -4145,6 +4145,11 @@ HDA_CODEC_ENTRY(0x10de0095, "GPU 95 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de0097, "GPU 97 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de0098, "GPU 98 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de0099, "GPU 99 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de009a, "GPU 9a HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de009d, "GPU 9d HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de009e, "GPU 9e HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de009f, "GPU 9f HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de00a0, "GPU a0 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de8001, "MCP73 HDMI", patch_nvhdmi_2ch), HDA_CODEC_ENTRY(0x10de8067, "MCP67/68 HDMI", patch_nvhdmi_2ch), HDA_CODEC_ENTRY(0x11069f80, "VX900 HDMI/DP", patch_via_hdmi), -- 2.27.0

2 5

[PATCH 0/8] ASoC: qcom: q6asm: few fixes and enhancements.
by Srinivas Kandagatla 11 Jun '20

11 Jun '20

While trying out gapless playback, I found few issues with the existing code This patchset has few fixes and some enhancement to the code to handle multiple streams per asm session and also handle buffers that are not aligned to period sizes. I will send Gapless support patches once compressed gapless state machine is fixed. Thanks, srini Srinivas Kandagatla (8): ASoC: q6asm: add command opcode to timeout error report ASoC: q6asm: handle EOS correctly ASoC: q6asm: rename misleading session id variable ASoC: q6asm: make commands specific to streams ASoC: q6asm: use flags directly from asm-dai ASoC: q6asm: add length to write command token ASoC: q6asm-dai: check available buffer size before sending ASoC: q6asm: allow to specify buffer offset in q6asm_write sound/soc/qcom/qdsp6/q6asm-dai.c | 100 ++++++++++++++-------- sound/soc/qcom/qdsp6/q6asm.c | 138 ++++++++++++++++++------------- sound/soc/qcom/qdsp6/q6asm.h | 42 ++++++---- 3 files changed, 172 insertions(+), 108 deletions(-) -- 2.21.0

2 9

[PATCH] dt-bindings: Fix more incorrect 'reg' property sizes in examples
by Rob Herring 11 Jun '20

11 Jun '20

The examples template is a 'simple-bus' with a size of 1 cell for had between 2 and 4 cells which really only errors on I2C or SPI type devices with a single cell. The easiest fix in most cases is to change the 'reg' property to 1 cell for address and size. Cc: "Heiko Stübner" <heiko(a)sntech.de> Cc: Ezequiel Garcia <ezequiel(a)collabora.com> Cc: Mauro Carvalho Chehab <mchehab(a)kernel.org> Cc: Philipp Zabel <p.zabel(a)pengutronix.de> Cc: Miquel Raynal <miquel.raynal(a)bootlin.com> Cc: Richard Weinberger <richard(a)nod.at> Cc: "David S. Miller" <davem(a)davemloft.net> Cc: Bjorn Andersson <bjorn.andersson(a)linaro.org> Cc: Kishon Vijay Abraham I <kishon(a)ti.com> Cc: Vinod Koul <vkoul(a)kernel.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: linux-rockchip(a)lists.infradead.org Cc: linux-media(a)vger.kernel.org Cc: linux-mtd(a)lists.infradead.org Cc: netdev(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org Signed-off-by: Rob Herring <robh(a)kernel.org> --- Documentation/devicetree/bindings/bus/baikal,bt1-apb.yaml | 4 ++-- Documentation/devicetree/bindings/bus/baikal,bt1-axi.yaml | 4 ++-- .../devicetree/bindings/display/rockchip/rockchip-vop.yaml | 4 ++-- Documentation/devicetree/bindings/media/rockchip,vdec.yaml | 2 +- Documentation/devicetree/bindings/media/rockchip-vpu.yaml | 2 +- .../devicetree/bindings/mtd/arasan,nand-controller.yaml | 2 +- Documentation/devicetree/bindings/net/ti,k3-am654-cpts.yaml | 2 +- .../devicetree/bindings/phy/qcom,usb-snps-femto-v2.yaml | 2 +- Documentation/devicetree/bindings/sound/fsl,easrc.yaml | 2 +- 9 files changed, 12 insertions(+), 12 deletions(-) diff --git a/Documentation/devicetree/bindings/bus/baikal,bt1-apb.yaml b/Documentation/devicetree/bindings/bus/baikal,bt1-apb.yaml index d6a3b71ea835..68b0131a31d0 100644 --- a/Documentation/devicetree/bindings/bus/baikal,bt1-apb.yaml +++ b/Documentation/devicetree/bindings/bus/baikal,bt1-apb.yaml @@ -71,8 +71,8 @@ examples: bus@1f059000 { compatible = "baikal,bt1-apb", "simple-bus"; - reg = <0 0x1f059000 0 0x1000>, - <0 0x1d000000 0 0x2040000>; + reg = <0x1f059000 0x1000>, + <0x1d000000 0x2040000>; reg-names = "ehb", "nodev"; #address-cells = <1>; #size-cells = <1>; diff --git a/Documentation/devicetree/bindings/bus/baikal,bt1-axi.yaml b/Documentation/devicetree/bindings/bus/baikal,bt1-axi.yaml index 203bc0e5346b..29e1aaea132b 100644 --- a/Documentation/devicetree/bindings/bus/baikal,bt1-axi.yaml +++ b/Documentation/devicetree/bindings/bus/baikal,bt1-axi.yaml @@ -85,8 +85,8 @@ examples: bus@1f05a000 { compatible = "baikal,bt1-axi", "simple-bus"; - reg = <0 0x1f05a000 0 0x1000>, - <0 0x1f04d110 0 0x8>; + reg = <0x1f05a000 0x1000>, + <0x1f04d110 0x8>; reg-names = "qos", "ehb"; #address-cells = <1>; #size-cells = <1>; diff --git a/Documentation/devicetree/bindings/display/rockchip/rockchip-vop.yaml b/Documentation/devicetree/bindings/display/rockchip/rockchip-vop.yaml index 1695e3e4bcec..ed8148e26e24 100644 --- a/Documentation/devicetree/bindings/display/rockchip/rockchip-vop.yaml +++ b/Documentation/devicetree/bindings/display/rockchip/rockchip-vop.yaml @@ -106,8 +106,8 @@ examples: #include <dt-bindings/power/rk3288-power.h> vopb: vopb@ff930000 { compatible = "rockchip,rk3288-vop"; - reg = <0x0 0xff930000 0x0 0x19c>, - <0x0 0xff931000 0x0 0x1000>; + reg = <0xff930000 0x19c>, + <0xff931000 0x1000>; interrupts = <GIC_SPI 15 IRQ_TYPE_LEVEL_HIGH>; clocks = <&cru ACLK_VOP0>, <&cru DCLK_VOP0>, diff --git a/Documentation/devicetree/bindings/media/rockchip,vdec.yaml b/Documentation/devicetree/bindings/media/rockchip,vdec.yaml index 0c68cdad9a31..8d35c327018b 100644 --- a/Documentation/devicetree/bindings/media/rockchip,vdec.yaml +++ b/Documentation/devicetree/bindings/media/rockchip,vdec.yaml @@ -61,7 +61,7 @@ examples: vdec: video-codec@ff660000 { compatible = "rockchip,rk3399-vdec"; - reg = <0x0 0xff660000 0x0 0x400>; + reg = <0xff660000 0x400>; interrupts = <GIC_SPI 116 IRQ_TYPE_LEVEL_HIGH 0>; clocks = <&cru ACLK_VDU>, <&cru HCLK_VDU>, <&cru SCLK_VDU_CA>, <&cru SCLK_VDU_CORE>; diff --git a/Documentation/devicetree/bindings/media/rockchip-vpu.yaml b/Documentation/devicetree/bindings/media/rockchip-vpu.yaml index 27df18ad6a81..2b629456d75f 100644 --- a/Documentation/devicetree/bindings/media/rockchip-vpu.yaml +++ b/Documentation/devicetree/bindings/media/rockchip-vpu.yaml @@ -66,7 +66,7 @@ examples: vpu: video-codec@ff9a0000 { compatible = "rockchip,rk3288-vpu"; - reg = <0x0 0xff9a0000 0x0 0x800>; + reg = <0xff9a0000 0x800>; interrupts = <GIC_SPI 9 IRQ_TYPE_LEVEL_HIGH>, <GIC_SPI 10 IRQ_TYPE_LEVEL_HIGH>; interrupt-names = "vepu", "vdpu"; diff --git a/Documentation/devicetree/bindings/mtd/arasan,nand-controller.yaml b/Documentation/devicetree/bindings/mtd/arasan,nand-controller.yaml index db8f115a13ec..cb9794edff24 100644 --- a/Documentation/devicetree/bindings/mtd/arasan,nand-controller.yaml +++ b/Documentation/devicetree/bindings/mtd/arasan,nand-controller.yaml @@ -53,7 +53,7 @@ examples: - | nfc: nand-controller@ff100000 { compatible = "xlnx,zynqmp-nand-controller", "arasan,nfc-v3p10"; - reg = <0x0 0xff100000 0x0 0x1000>; + reg = <0xff100000 0x1000>; clock-names = "controller", "bus"; clocks = <&clk200>, <&clk100>; interrupt-parent = <&gic>; diff --git a/Documentation/devicetree/bindings/net/ti,k3-am654-cpts.yaml b/Documentation/devicetree/bindings/net/ti,k3-am654-cpts.yaml index af608f2ecfdf..9b7117920d90 100644 --- a/Documentation/devicetree/bindings/net/ti,k3-am654-cpts.yaml +++ b/Documentation/devicetree/bindings/net/ti,k3-am654-cpts.yaml @@ -121,7 +121,7 @@ examples: cpts@310d0000 { compatible = "ti,am65-cpts"; - reg = <0x0 0x310d0000 0x0 0x400>; + reg = <0x310d0000 0x400>; reg-names = "cpts"; clocks = <&main_cpts_mux>; clock-names = "cpts"; diff --git a/Documentation/devicetree/bindings/phy/qcom,usb-snps-femto-v2.yaml b/Documentation/devicetree/bindings/phy/qcom,usb-snps-femto-v2.yaml index 574f890fab1d..4949a2851532 100644 --- a/Documentation/devicetree/bindings/phy/qcom,usb-snps-femto-v2.yaml +++ b/Documentation/devicetree/bindings/phy/qcom,usb-snps-femto-v2.yaml @@ -65,7 +65,7 @@ examples: #include <dt-bindings/clock/qcom,gcc-sm8150.h> phy@88e2000 { compatible = "qcom,sm8150-usb-hs-phy"; - reg = <0 0x088e2000 0 0x400>; + reg = <0x088e2000 0x400>; #phy-cells = <0>; clocks = <&rpmhcc RPMH_CXO_CLK>; diff --git a/Documentation/devicetree/bindings/sound/fsl,easrc.yaml b/Documentation/devicetree/bindings/sound/fsl,easrc.yaml index 9dd57a974b28..32d547af9ce7 100644 --- a/Documentation/devicetree/bindings/sound/fsl,easrc.yaml +++ b/Documentation/devicetree/bindings/sound/fsl,easrc.yaml @@ -80,7 +80,7 @@ examples: easrc: easrc@300c0000 { compatible = "fsl,imx8mn-easrc"; - reg = <0x0 0x300c0000 0x0 0x10000>; + reg = <0x300c0000 0x10000>; interrupts = <0x0 122 0x4>; clocks = <&clk IMX8MN_CLK_ASRC_ROOT>; clock-names = "mem"; -- 2.25.1

2 1

next-0519 on thinkpad x60: sound related? window manager crash
by Pavel Machek 11 Jun '20

11 Jun '20

Hi! My window manager stopped responding. I was able to recover machine using sysrq-k. I started writing nice report, when session failed second time. And then third time on next attempt. Any ideas? I'll send this out before this locks up... Best regards, Pavel [ 2801.147411] sdhci-pci 0000:15:00.2: Will use DMA mode even though HW doesn't fully claim to support it. [ 2801.187449] sdhci-pci 0000:15:00.2: Will use DMA mode even though HW doesn't fully claim to support it. [ 2801.192260] usb 1-2: new high-speed USB device number 5 using ehci-pci [ 2801.240241] sdhci-pci 0000:15:00.2: Will use DMA mode even though HW doesn't fully claim to support it. [ 2801.300663] sdhci-pci 0000:15:00.2: Will use DMA mode even though HW doesn't fully claim to support it. [ 2801.352181] usb 1-2: New USB device found, idVendor=0525, idProduct=a4a1, bcdDevice= 5.07 [ 2801.352192] usb 1-2: New USB device strings: Mfr=1, Product=2, SerialNumber=0 [ 2801.352200] usb 1-2: Product: Ethernet Gadget [ 2801.352207] usb 1-2: Manufacturer: Linux 5.7.0-rc4-00046-g6d7c0f75a522 with musb-hdrc [ 2801.419872] e1000e 0000:02:00.0 eth1: NIC Link is Down [ 2801.428760] cdc_ether 1-2:1.0 usb0: register 'cdc_ether' at usb-0000:00:1d.7-2, CDC Ethernet Device, 72:ed:12:23:c9:c2 [ 2804.020289] wlan0: authenticate with 5c:f4:ab:10:d2:bb [ 2804.020451] wlan0: send auth to 5c:f4:ab:10:d2:bb (try 1/3) [ 2804.022385] wlan0: authenticated [ 2804.024243] wlan0: associate with 5c:f4:ab:10:d2:bb (try 1/3) [ 2804.026985] wlan0: RX AssocResp from 5c:f4:ab:10:d2:bb (capab=0x411 status=0 aid=2) [ 2804.028961] wlan0: associated [ 2874.520955] perf: interrupt took too long (2507 > 2500), lowering kernel.perf_event_max_sample_rate to 79750 [ 3730.016148] perf: interrupt took too long (3135 > 3133), lowering kernel.perf_event_max_sample_rate to 63750 [ 4274.984810] BUG: unable to handle page fault for address: f8600000 [ 4274.984821] #PF: supervisor write access in kernel mode [ 4274.984827] #PF: error_code(0x0002) - not-present page [ 4274.984833] *pdpt = 000000002c0b2001 *pde = 0000000000000000 [ 4274.984843] Oops: 0002 [#1] PREEMPT SMP PTI [ 4274.984853] CPU: 1 PID: 3351 Comm: marco Not tainted 5.7.0-rc6-next-20200519+ #115 [ 4274.984859] Hardware name: LENOVO 17097HU/17097HU, BIOS 7BETD8WW (2.19 ) 03/31/2011 [ 4274.984871] EIP: memset+0xb/0x20 [ 4274.984878] Code: f9 01 72 0b 8a 0e 88 0f 8d b4 26 00 00 00 00 8b 45 f0 83 c4 04 5b 5e 5f 5d c3 8d 74 26 00 90 55 89 e5 57 89 c7 53 89 c3 89 d0 <f3> aa 89 d8 5b 5f 5d c3 cc cc cc cc cc cc cc cc cc cc cc cc cc 89 [ 4274.984885] EAX: 00000000 EBX: f85fe000 ECX: 0001e000 EDX: 00000000 [ 4274.984892] ESI: ed158400 EDI: f8600000 EBP: edcc9e6c ESP: edcc9e64 [ 4274.984898] DS: 007b ES: 007b FS: 00d8 GS: 00e0 SS: 0068 EFLAGS: 00210246 [ 4274.984905] CR0: 80050033 CR2: f8600000 CR3: 2c114000 CR4: 000006b0 [ 4274.984910] Call Trace: [ 4274.984923] snd_pcm_hw_params+0x38d/0x400 [ 4274.984930] snd_pcm_ioctl+0x187/0xe80 [ 4274.984940] ? __fget_files+0x86/0xc0 [ 4274.984947] ? __fget_light+0x6b/0x80 [ 4274.984954] ? snd_pcm_status_user64+0x90/0x90 [ 4274.984962] ksys_ioctl+0x1cd/0x880 [ 4274.984971] ? ksys_mmap_pgoff+0x81/0xc0 [ 4274.984978] ? fput+0xd/0x10 [ 4274.984984] ? ksys_mmap_pgoff+0x8d/0xc0 [ 4274.984991] __ia32_sys_ioctl+0x10/0x12 [ 4274.985000] do_int80_syscall_32+0x3c/0x100 [ 4274.985010] entry_INT80_32+0x116/0x116 [ 4274.985016] EIP: 0xb7f17092 [ 4274.985023] Code: 00 00 00 e9 90 ff ff ff ff a3 24 00 00 00 68 30 00 00 00 e9 80 ff ff ff ff a3 e8 ff ff ff 66 90 00 00 00 00 00 00 00 00 cd 80 <c3> 8d b4 26 00 00 00 00 8d b6 00 00 00 00 8b 1c 24 c3 8d b4 26 00 [ 4274.985030] EAX: ffffffda EBX: 00000011 ECX: c25c4111 EDX: bf8d5280 [ 4274.985036] ESI: 08250880 EDI: bf8d5280 EBP: 082a4150 ESP: bf8d50a4 [ 4274.985042] DS: 007b ES: 007b FS: 0000 GS: 0033 SS: 007b EFLAGS: 00200292 [ 4274.985051] ? nmi+0xcc/0x2bc [ 4274.985055] Modules linked in: [ 4274.985063] CR2: 00000000f8600000 [ 4274.985072] ---[ end trace 61b0852711d6de1d ]--- [ 4274.985079] EIP: memset+0xb/0x20 [ 4274.985086] Code: f9 01 72 0b 8a 0e 88 0f 8d b4 26 00 00 00 00 8b 45 f0 83 c4 04 5b 5e 5f 5d c3 8d 74 26 00 90 55 89 e5 57 89 c7 53 89 c3 89 d0 <f3> aa 89 d8 5b 5f 5d c3 cc cc cc cc cc cc cc cc cc cc cc cc cc 89 [ 4274.985092] EAX: 00000000 EBX: f85fe000 ECX: 0001e000 EDX: 00000000 [ 4274.985099] ESI: ed158400 EDI: f8600000 EBP: edcc9e6c ESP: edcc9e64 [ 4274.985105] DS: 007b ES: 007b FS: 00d8 GS: 00e0 SS: 0068 EFLAGS: 00210246 [ 4274.985112] CR0: 80050033 CR2: f8600000 CR3: 2c114000 CR4: 000006b0 [ 4337.396551] sysrq: SAK [ 4337.397010] tty tty7: SAK: killed process 2963 (Xorg): by session [ 4337.397282] tty tty7: SAK: killed process 2963 (Xorg): by controlling tty [ 4337.397621] tty tty7: SAK: killed process 3484 (console-kit-dae): by fd#9 [ 4337.397934] tty tty7: SAK: killed process 3485 (console-kit-dae): by fd#9 [ 4337.397940] tty tty7: SAK: killed process 3486 (console-kit-dae): by fd#9 [ 4337.397945] tty tty7: SAK: killed process 3487 (console-kit-dae): by fd#9 [ 4337.397951] tty tty7: SAK: killed process 3488 (console-kit-dae): by fd#9 [ 4337.397956] tty tty7: SAK: killed process 3489 (console-kit-dae): by fd#9 [ 4337.397961] tty tty7: SAK: killed process 3490 (console-kit-dae): by fd#9 [ 4337.397967] tty tty7: SAK: killed process 3491 (console-kit-dae): by fd#9 [ 4337.397972] tty tty7: SAK: killed process 3492 (console-kit-dae): by fd#9 [ 4337.397978] tty tty7: SAK: killed process 3493 (console-kit-dae): by fd#9 [ 4337.397983] tty tty7: SAK: killed process 3494 (console-kit-dae): by fd#9 [ 4337.397989] tty tty7: SAK: killed process 3495 (console-kit-dae): by fd#9 [ 4337.397994] tty tty7: SAK: killed process 3496 (console-kit-dae): by fd#9 [ 4337.397999] tty tty7: SAK: killed process 3497 (console-kit-dae): by fd#9 [ 4337.398005] tty tty7: SAK: killed process 3498 (console-kit-dae): by fd#9 [ 4337.398010] tty tty7: SAK: killed process 3499 (console-kit-dae): by fd#9 [ 4337.398015] tty tty7: SAK: killed process 3500 (console-kit-dae): by fd#9 [ 4337.398021] tty tty7: SAK: killed process 3501 (console-kit-dae): by fd#9 [ 4337.398026] tty tty7: SAK: killed process 3502 (console-kit-dae): by fd#9 [ 4337.398032] tty tty7: SAK: killed process 3503 (console-kit-dae): by fd#9 [ 4337.398037] tty tty7: SAK: killed process 3504 (console-kit-dae): by fd#9 [ 4337.398042] tty tty7: SAK: killed process 3505 (console-kit-dae): by fd#9 [ 4337.398048] tty tty7: SAK: killed process 3506 (console-kit-dae): by fd#9 [ 4337.398053] tty tty7: SAK: killed process 3507 (console-kit-dae): by fd#9 [ 4337.398059] tty tty7: SAK: killed process 3508 (console-kit-dae): by fd#9 [ 4337.398064] tty tty7: SAK: killed process 3509 (console-kit-dae): by fd#9 [ 4337.398070] tty tty7: SAK: killed process 3510 (console-kit-dae): by fd#9 [ 4337.398075] tty tty7: SAK: killed process 3511 (console-kit-dae): by fd#9 [ 4337.398080] tty tty7: SAK: killed process 3512 (console-kit-dae): by fd#9 [ 4337.398086] tty tty7: SAK: killed process 3513 (console-kit-dae): by fd#9 [ 4337.398091] tty tty7: SAK: killed process 3514 (console-kit-dae): by fd#9 [ 4337.398097] tty tty7: SAK: killed process 3515 (console-kit-dae): by fd#9 [ 4337.398102] tty tty7: SAK: killed process 3516 (console-kit-dae): by fd#9 [ 4337.398107] tty tty7: SAK: killed process 3517 (console-kit-dae): by fd#9 [ 4337.398113] tty tty7: SAK: killed process 3518 (console-kit-dae): by fd#9 [ 4337.398118] tty tty7: SAK: killed process 3519 (console-kit-dae): by fd#9 [ 4337.398124] tty tty7: SAK: killed process 3520 (console-kit-dae): by fd#9 [ 4337.398129] tty tty7: SAK: killed process 3521 (console-kit-dae): by fd#9 [ 4337.398135] tty tty7: SAK: killed process 3522 (console-kit-dae): by fd#9 [ 4337.398141] tty tty7: SAK: killed process 3523 (console-kit-dae): by fd#9 [ 4337.398146] tty tty7: SAK: killed process 3524 (console-kit-dae): by fd#9 [ 4337.398151] tty tty7: SAK: killed process 3525 (console-kit-dae): by fd#9 [ 4337.398157] tty tty7: SAK: killed process 3526 (console-kit-dae): by fd#9 [ 4337.398162] tty tty7: SAK: killed process 3527 (console-kit-dae): by fd#9 [ 4337.398168] tty tty7: SAK: killed process 3528 (console-kit-dae): by fd#9 [ 4337.398173] tty tty7: SAK: killed process 3529 (console-kit-dae): by fd#9 [ 4337.398178] tty tty7: SAK: killed process 3530 (console-kit-dae): by fd#9 [ 4337.398184] tty tty7: SAK: killed process 3531 (console-kit-dae): by fd#9 [ 4337.398189] tty tty7: SAK: killed process 3532 (console-kit-dae): by fd#9 [ 4337.398194] tty tty7: SAK: killed process 3533 (console-kit-dae): by fd#9 [ 4337.398200] tty tty7: SAK: killed process 3534 (console-kit-dae): by fd#9 [ 4337.398205] tty tty7: SAK: killed process 3535 (console-kit-dae): by fd#9 [ 4337.398210] tty tty7: SAK: killed process 3536 (console-kit-dae): by fd#9 [ 4337.398216] tty tty7: SAK: killed process 3537 (console-kit-dae): by fd#9 [ 4337.398221] tty tty7: SAK: killed process 3538 (console-kit-dae): by fd#9 [ 4337.398227] tty tty7: SAK: killed process 3539 (console-kit-dae): by fd#9 [ 4337.398232] tty tty7: SAK: killed process 3540 (console-kit-dae): by fd#9 [ 4337.398237] tty tty7: SAK: killed process 3541 (console-kit-dae): by fd#9 [ 4337.398243] tty tty7: SAK: killed process 3542 (console-kit-dae): by fd#9 [ 4337.398248] tty tty7: SAK: killed process 3543 (console-kit-dae): by fd#9 [ 4337.398253] tty tty7: SAK: killed process 3544 (console-kit-dae): by fd#9 [ 4337.398260] tty tty7: SAK: killed process 3545 (console-kit-dae): by fd#9 [ 4337.398265] tty tty7: SAK: killed process 3546 (console-kit-dae): by fd#9 [ 4337.398270] tty tty7: SAK: killed process 3548 (gmain): by fd#9 [ 4337.398276] tty tty7: SAK: killed process 3549 (gdbus): by fd#9 [ 4337.743402] wlan0: deauthenticating from 5c:f4:ab:10:d2:bb by local choice (Reason: 3=DEAUTH_LEAVING) [ 4349.750345] traps: clock-applet[7160] trap int3 ip:b71bafc0 sp:bffdd3b0 error:0 in libglib-2.0.so.0.5000.3[b716e000+12a000] [ 4349.751834] traps: mateweather-app[7164] trap int3 ip:b729ffc0 sp:bfc45bf0 error:0 in libglib-2.0.so.0.5000.3[b7253000+12a000] [ 4352.181317] wlan0: authenticate with 5c:f4:ab:10:d2:bb [ 4352.183330] wlan0: send auth to 5c:f4:ab:10:d2:bb (try 1/3) [ 4352.187927] wlan0: authenticated [ 4352.192203] wlan0: associate with 5c:f4:ab:10:d2:bb (try 1/3) [ 4352.194932] wlan0: RX AssocResp from 5c:f4:ab:10:d2:bb (capab=0x411 status=0 aid=2) [ 4352.199395] wlan0: associated [ 4688.523006] sysrq: SAK [ 4688.523318] tty tty7: SAK: killed process 6791 (Xorg): by session [ 4688.523683] tty tty7: SAK: killed process 6791 (Xorg): by controlling tty [ 4688.523930] tty tty7: SAK: killed process 7017 (console-kit-dae): by fd#9 [ 4688.524330] tty tty7: SAK: killed process 7018 (console-kit-dae): by fd#9 [ 4688.524337] tty tty7: SAK: killed process 7019 (console-kit-dae): by fd#9 [ 4688.524342] tty tty7: SAK: killed process 7020 (console-kit-dae): by fd#9 [ 4688.524347] tty tty7: SAK: killed process 7021 (console-kit-dae): by fd#9 [ 4688.524353] tty tty7: SAK: killed process 7022 (console-kit-dae): by fd#9 [ 4688.524358] tty tty7: SAK: killed process 7023 (console-kit-dae): by fd#9 [ 4688.524363] tty tty7: SAK: killed process 7024 (console-kit-dae): by fd#9 [ 4688.524369] tty tty7: SAK: killed process 7025 (console-kit-dae): by fd#9 [ 4688.524374] tty tty7: SAK: killed process 7026 (console-kit-dae): by fd#9 [ 4688.524380] tty tty7: SAK: killed process 7027 (console-kit-dae): by fd#9 [ 4688.524385] tty tty7: SAK: killed process 7028 (console-kit-dae): by fd#9 [ 4688.524391] tty tty7: SAK: killed process 7029 (console-kit-dae): by fd#9 [ 4688.524396] tty tty7: SAK: killed process 7030 (console-kit-dae): by fd#9 [ 4688.524402] tty tty7: SAK: killed process 7031 (console-kit-dae): by fd#9 [ 4688.524407] tty tty7: SAK: killed process 7032 (console-kit-dae): by fd#9 [ 4688.524412] tty tty7: SAK: killed process 7033 (console-kit-dae): by fd#9 [ 4688.524419] tty tty7: SAK: killed process 7034 (console-kit-dae): by fd#9 [ 4688.524424] tty tty7: SAK: killed process 7035 (console-kit-dae): by fd#9 [ 4688.524429] tty tty7: SAK: killed process 7036 (console-kit-dae): by fd#9 [ 4688.524435] tty tty7: SAK: killed process 7037 (console-kit-dae): by fd#9 [ 4688.524440] tty tty7: SAK: killed process 7038 (console-kit-dae): by fd#9 [ 4688.524446] tty tty7: SAK: killed process 7039 (console-kit-dae): by fd#9 [ 4688.524451] tty tty7: SAK: killed process 7040 (console-kit-dae): by fd#9 [ 4688.524456] tty tty7: SAK: killed process 7041 (console-kit-dae): by fd#9 [ 4688.524462] tty tty7: SAK: killed process 7042 (console-kit-dae): by fd#9 [ 4688.524467] tty tty7: SAK: killed process 7043 (console-kit-dae): by fd#9 [ 4688.524473] tty tty7: SAK: killed process 7044 (console-kit-dae): by fd#9 [ 4688.524478] tty tty7: SAK: killed process 7045 (console-kit-dae): by fd#9 [ 4688.524483] tty tty7: SAK: killed process 7046 (console-kit-dae): by fd#9 [ 4688.524489] tty tty7: SAK: killed process 7047 (console-kit-dae): by fd#9 [ 4688.524494] tty tty7: SAK: killed process 7048 (console-kit-dae): by fd#9 [ 4688.524500] tty tty7: SAK: killed process 7049 (console-kit-dae): by fd#9 [ 4688.524505] tty tty7: SAK: killed process 7050 (console-kit-dae): by fd#9 [ 4688.524511] tty tty7: SAK: killed process 7051 (console-kit-dae): by fd#9 [ 4688.524516] tty tty7: SAK: killed process 7052 (console-kit-dae): by fd#9 [ 4688.524521] tty tty7: SAK: killed process 7053 (console-kit-dae): by fd#9 [ 4688.524527] tty tty7: SAK: killed process 7054 (console-kit-dae): by fd#9 [ 4688.524532] tty tty7: SAK: killed process 7055 (console-kit-dae): by fd#9 [ 4688.524538] tty tty7: SAK: killed process 7056 (console-kit-dae): by fd#9 [ 4688.524543] tty tty7: SAK: killed process 7057 (console-kit-dae): by fd#9 [ 4688.524549] tty tty7: SAK: killed process 7058 (console-kit-dae): by fd#9 [ 4688.524554] tty tty7: SAK: killed process 7059 (console-kit-dae): by fd#9 [ 4688.524560] tty tty7: SAK: killed process 7060 (console-kit-dae): by fd#9 [ 4688.524565] tty tty7: SAK: killed process 7061 (console-kit-dae): by fd#9 [ 4688.524570] tty tty7: SAK: killed process 7062 (console-kit-dae): by fd#9 [ 4688.524576] tty tty7: SAK: killed process 7063 (console-kit-dae): by fd#9 [ 4688.524581] tty tty7: SAK: killed process 7064 (console-kit-dae): by fd#9 [ 4688.524587] tty tty7: SAK: killed process 7065 (console-kit-dae): by fd#9 [ 4688.524593] tty tty7: SAK: killed process 7066 (console-kit-dae): by fd#9 [ 4688.524598] tty tty7: SAK: killed process 7067 (console-kit-dae): by fd#9 [ 4688.524603] tty tty7: SAK: killed process 7068 (console-kit-dae): by fd#9 [ 4688.524609] tty tty7: SAK: killed process 7069 (console-kit-dae): by fd#9 [ 4688.524614] tty tty7: SAK: killed process 7070 (console-kit-dae): by fd#9 [ 4688.524620] tty tty7: SAK: killed process 7071 (console-kit-dae): by fd#9 [ 4688.524625] tty tty7: SAK: killed process 7072 (console-kit-dae): by fd#9 [ 4688.524631] tty tty7: SAK: killed process 7073 (console-kit-dae): by fd#9 [ 4688.524636] tty tty7: SAK: killed process 7074 (console-kit-dae): by fd#9 [ 4688.524642] tty tty7: SAK: killed process 7075 (console-kit-dae): by fd#9 [ 4688.524648] tty tty7: SAK: killed process 7076 (console-kit-dae): by fd#9 [ 4688.524653] tty tty7: SAK: killed process 7077 (console-kit-dae): by fd#9 [ 4688.524659] tty tty7: SAK: killed process 7078 (console-kit-dae): by fd#9 [ 4688.524664] tty tty7: SAK: killed process 7079 (console-kit-dae): by fd#9 [ 4688.524670] tty tty7: SAK: killed process 7083 (gmain): by fd#9 [ 4688.524675] tty tty7: SAK: killed process 7085 (gdbus): by fd#9 [ 4688.780517] wlan0: deauthenticating from 5c:f4:ab:10:d2:bb by local choice (Reason: 3=DEAUTH_LEAVING) [ 4698.601308] traps: clock-applet[7813] trap int3 ip:b71e9fc0 sp:bf901050 error:0 in libglib-2.0.so.0.5000.3[b719d000+12a000] [ 4698.676549] traps: mateweather-app[7814] trap int3 ip:b7260fc0 sp:bfcfb9d0 error:0 in libglib-2.0.so.0.5000.3[b7214000+12a000] [ 4700.456620] wlan0: authenticate with 5c:f4:ab:10:d2:bb [ 4700.456710] wlan0: send auth to 5c:f4:ab:10:d2:bb (try 1/3) [ 4700.459606] wlan0: authenticated [ 4700.460449] wlan0: associate with 5c:f4:ab:10:d2:bb (try 1/3) [ 4700.463153] wlan0: RX AssocResp from 5c:f4:ab:10:d2:bb (capab=0x411 status=0 aid=2) [ 4700.465007] wlan0: associated -- (english) http://www.livejournal.com/~pavelmachek (cesky, pictures) http://atrey.karlin.mff.cuni.cz/~pavel/picture/horses/blog.html

6 30

sof-hda-dsp: make Headphone Playback Switch conditional
by GitHub pull_request - opened 11 Jun '20

11 Jun '20

alsa-project/alsa-ucm-conf pull request #34 was opened from kv2019i: The Headphone Playback Switch control is not present in all HDA codecs. Allow the Headphones definition to work also on such systems. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/34 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/34.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Clock sync problem?
by Laurence Tratt 11 Jun '20

11 Jun '20

Hello, I've found an interesting issue with an SSL2+ audio interface -- it plays and records fine (and, for the money, the sound quality is excellent!), but recording slowly but surely drifts over time. For example, if I record a click track in Audacity (or Ardour), play it through the interface and record it, and then line up the start of the recorded click track with the original, the two tracks will be around 5ms out of alignment after 30s. Interestingly, this problem does not happen with OpenBSD, where the two tracks line up precisely. My guess is that the problem is to do with implicit feedback in endpoints. I wondered if the problem was the same as with the MOTU M4, but enabling that quirk for the SSL2+ introduced some crackle, but didn't change the drift. I'm attaching a) the alsa-info file b) output from OpenBSD's kernel with debugging in the USB audio driver turned on in case that gives some clues as to what the two OS's and drivers might be doing differently. If anyone has any suggestions for how this problem might be addressed, I would be very grateful! Laurie -- Personal http://tratt.net/laurie/ Software Development Team http://soft-dev.org/ https://github.com/ltratt http://twitter.com/laurencetratt

4 8

alsa-gobject: obsolete abstract class for `snd_timer_read` and `snd_timer_tread` structure
by GitHub pull_request - opened 11 Jun '20

11 Jun '20

alsa-project/alsa-gobject pull request #31 was opened from takaswie: In UAPI of Linux sound subsystem, two types of event is read from ALSA Timer character device; snd_timer_read and snd_timer_tread. Before attached to any timer device, an user instance can select which type of event is expected. ALSATimer.UserInstance uses one GObject signal to handle these events. These events are defined as derived object from abstract class, ALSATimer.EventData. However, the design pattern is not convenient for some language bindings. For example, in Rust language, class-based inheritance is not supported. Gtk-rs project produces mechanism to upcast/downcast between super/sub classes and handles GObject class system. In my opinion, the code to upcast/downcast is a bit complicated to handle derivatives. This patchset obsoletes usage of abstract class for ALSATimer.EventData. Instead, ALSATimer.Event is defined as concrete class and the instance of class is passed to the GObject signal. Applications use APIs to pick up expected type of data from the instance. Request URL : https://github.com/alsa-project/alsa-gobject/pull/31 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/31.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[RFC PATCH] ALSA: compress: Fix gapless playback state machine
by Srinivas Kandagatla 11 Jun '20

11 Jun '20

For gapless playback call to snd_compr_drain_notify() after partial drain should put the state to SNDRV_PCM_STATE_RUNNING rather than SNDRV_PCM_STATE_SETUP as the driver is ready to process the buffers for new track. With existing code, if we are playing 3 tracks in gapless, after partial drain finished on previous track 1 the state is set to SNDRV_PCM_STATE_SETUP which is then moved to SNDRV_PCM_STATE_PREPARED after data write. With this state calls to snd_compr_next_track() and few other calls will fail as they expect the state to be in SNDRV_PCM_STATE_RUNNING. Here is the sequence of events and state transitions: 1. set_params (Track 1), state = SNDRV_PCM_STATE_SETUP 2. set_metadata (Track 1), no state change, state = SNDRV_PCM_STATE_SETUP 3. fill and trigger start (Track 1), state = SNDRV_PCM_STATE_RUNNING 4. set_next_track (Track 2), state = SNDRV_PCM_STATE_RUNNING 5. partial_drain (Track 1), state = SNDRV_PCM_STATE_SETUP 6 snd_compr_drain_notify (Track 1), state = SNDRV_PCM_STATE_SETUP 7. fill data (Track 2), state = SNDRV_PCM_STATE_PREPARED 8. set_metadata (Track 3), no state change, state = SNDRV_PCM_STATE_PREPARED 9. set_next_track (Track 3), !! FAILURE as state != SNDRV_PCM_STATE_RUNNING Fixes: f44f2a5417b2 ("ALSA: compress: fix drain calls blocking other compress functions (v6)") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- I wonder who did gapless work on upstream so far? include/sound/compress_driver.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index 6ce8effa0b12..eabac33864c2 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -182,7 +182,7 @@ static inline void snd_compr_drain_notify(struct snd_compr_stream *stream) if (snd_BUG_ON(!stream)) return; - stream->runtime->state = SNDRV_PCM_STATE_SETUP; + stream->runtime->state = SNDRV_PCM_STATE_RUNNING; wake_up(&stream->runtime->sleep); } -- 2.21.0

5 11

[PATCH 2/2] ASoC: max98390: Update regmap readable reg and volatile
by Steve Lee 11 Jun '20

11 Jun '20

Update max98390_readable_register and max98390_volatile_reg Signed-off-by: Steve Lee <steves.lee(a)maximintegrated.com> --- sound/soc/codecs/max98390.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c index 57d15dbfcda3..b345e626956d 100644 --- a/sound/soc/codecs/max98390.c +++ b/sound/soc/codecs/max98390.c @@ -700,8 +700,8 @@ static bool max98390_readable_register(struct device *dev, unsigned int reg) case MAX98390_IRQ_CTRL ... MAX98390_WDOG_CTRL: case MAX98390_MEAS_ADC_THERM_WARN_THRESH ... MAX98390_BROWNOUT_INFINITE_HOLD: - case MAX98390_BROWNOUT_LVL_HOLD ... THERMAL_COILTEMP_RD_BACK_BYTE0: - case DSMIG_DEBUZZER_THRESHOLD ... MAX98390_R24FF_REV_ID: + case MAX98390_BROWNOUT_LVL_HOLD ... DSMIG_DEBUZZER_THRESHOLD: + case DSM_VOL_ENA ... MAX98390_R24FF_REV_ID: return true; default: return false; @@ -717,7 +717,7 @@ static bool max98390_volatile_reg(struct device *dev, unsigned int reg) case MAX98390_BROWNOUT_LOWEST_STATUS: case MAX98390_ENV_TRACK_BOOST_VOUT_READ: case DSM_STBASS_HPF_B0_BYTE0 ... DSM_DEBUZZER_ATTACK_TIME_BYTE2: - case THERMAL_RDC_RD_BACK_BYTE1 ... THERMAL_COILTEMP_RD_BACK_BYTE0: + case THERMAL_RDC_RD_BACK_BYTE1 ... DSMIG_DEBUZZER_THRESHOLD: case DSM_THERMAL_GAIN ... DSM_WBDRC_GAIN: return true; default: -- 2.17.1

1 0

[PATCH 1/2] ASoC: max98390: Add Amp init common setting func.
by Steve Lee 11 Jun '20

11 Jun '20

Add amp common init function to gather common init setting and finaize. - add max98390_init_regs func - move amp setting to max98390_init_regs func. - removed unneceary setting and finalize common register values. Signed-off-by: Steve Lee <steves.lee(a)maximintegrated.com> --- sound/soc/codecs/max98390.c | 26 ++++++++++++++++---------- 1 file changed, 16 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c index 0d63ebfbff2f..57d15dbfcda3 100644 --- a/sound/soc/codecs/max98390.c +++ b/sound/soc/codecs/max98390.c @@ -842,6 +842,20 @@ static int max98390_dsm_calibrate(struct snd_soc_component *component) return 0; } +static void max98390_init_regs(struct snd_soc_component *component) +{ + struct max98390_priv *max98390 = + snd_soc_component_get_drvdata(component); + + regmap_write(max98390->regmap, MAX98390_CLK_MON, 0x6f); + regmap_write(max98390->regmap, MAX98390_DAT_MON, 0x00); + regmap_write(max98390->regmap, MAX98390_PWR_GATE_CTL, 0x00); + regmap_write(max98390->regmap, MAX98390_PCM_RX_EN_A, 0x03); + regmap_write(max98390->regmap, MAX98390_ENV_TRACK_VOUT_HEADROOM, 0x0e); + regmap_write(max98390->regmap, MAX98390_BOOST_BYPASS1, 0x46); + regmap_write(max98390->regmap, MAX98390_FET_SCALING3, 0x03); +} + static int max98390_probe(struct snd_soc_component *component) { struct max98390_priv *max98390 = @@ -853,18 +867,10 @@ static int max98390_probe(struct snd_soc_component *component) /* Update dsm bin param */ max98390_dsm_init(component); - /* Amp Setting */ - regmap_write(max98390->regmap, MAX98390_CLK_MON, 0x6f); - regmap_write(max98390->regmap, MAX98390_PCM_RX_EN_A, 0x03); - regmap_write(max98390->regmap, MAX98390_PWR_GATE_CTL, 0x2d); - regmap_write(max98390->regmap, MAX98390_ENV_TRACK_VOUT_HEADROOM, 0x0e); - regmap_write(max98390->regmap, MAX98390_BOOST_BYPASS1, 0x46); - regmap_write(max98390->regmap, MAX98390_FET_SCALING3, 0x03); + /* Amp init setting */ + max98390_init_regs(component); /* Dsm Setting */ - regmap_write(max98390->regmap, DSM_VOL_CTRL, 0x94); - regmap_write(max98390->regmap, DSMIG_EN, 0x19); - regmap_write(max98390->regmap, MAX98390_R203A_AMP_EN, 0x80); if (max98390->ref_rdc_value) { regmap_write(max98390->regmap, DSM_TPROT_RECIP_RDC_ROOM_BYTE0, max98390->ref_rdc_value & 0x000000ff); -- 2.17.1

1 0

[PATCH v2 0/3] ASoC: ti: Add support for audio on J721e EVM
by Peter Ujfalusi 11 Jun '20

11 Jun '20

Hi, Changes since v1: - Fixed DT binding documentation errors - Rebased on ASoC head and updated the driver to compile and work This series adds support for the analog audio setup on the j721e EVM. The audio setup of the EVM is: Common Processor Board (CPB): McASP10 <-> pcm3168a Infotainment Expansion Board (IVI): McASP0 <-> 2x pcm3168a Both CPB and IVI wired in parallel serializer setup. The first patch adds the stream_name for McASP driver as it is needed in multicodec (and would be needed in DPCM) setup for proper DAPM handling. The second patch adds two DT schema, one for the cpb and one for the cpb+ivi card. Regards, Peter --- Peter Ujfalusi (3): ASoC: ti: davinci-mcasp: Specify stream_name for playback/capture bindings: sound: Add documentation for TI j721e EVM (CPB and IVI) ASoC: ti: Add custom machine driver for j721e EVM (CPB and IVI) .../bindings/sound/ti,j721e-cpb-audio.yaml | 93 ++ .../sound/ti,j721e-cpb-ivi-audio.yaml | 142 +++ sound/soc/ti/Kconfig | 8 + sound/soc/ti/Makefile | 2 + sound/soc/ti/davinci-mcasp.c | 3 + sound/soc/ti/j721e-evm.c | 867 ++++++++++++++++++ 6 files changed, 1115 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,j721e-cpb-audio.yaml create mode 100644 Documentation/devicetree/bindings/sound/ti,j721e-cpb-ivi-audio.yaml create mode 100644 sound/soc/ti/j721e-evm.c -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

2 8

alsa-gobject: obsolete abstract class for snd_seq_queue_timer structure
by GitHub pull_request - opened 11 Jun '20

11 Jun '20

alsa-project/alsa-gobject pull request #30 was opened from takaswie: In UAPI of Linux sound subsystem, `snd_seq_queue_timer` structure includes union member. Current implementation of ALSASeq uses abstract class (ALSASeqQueueTimer) and concrete class (ALSASeqQueueTimerAlsa) to represent variation corresponding to each of union member. However, the design pattern is not convenient for some language bindings. For example, in Rust language, class-based inheritance is not supported. Gtk-rs project produces mechanism to upcast/downcast between super/sub classes and handles GObject class system. In my opinion, the code to upcast/downcast is a bit complicated to handle derivatives. This patchset obsoletes usage of abstract class for ALSASeq. The target is ALSASeqQueueTimer. ALSASeqQueueTimerDataAlsa is newly defined to retrieve data for the case of ALSATimer and the inherited ALSASeqQueueTimerAlsa is deleted. Request URL : https://github.com/alsa-project/alsa-gobject/pull/30 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/30.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

alsa-gobject: seq: misc fixes (take 2)
by GitHub pull_request - opened 11 Jun '20

11 Jun '20

alsa-project/alsa-gobject pull request #29 was opened from takaswie: This patchset includes misc fixes for ALSASeq. Especially the execution of g-ir-scanner to generate ALSASeq-0.0 becomes to expect ALSATimer-0.0 is available. Although it is expected, it results in Python 3 gir loader (PyGObject) loading ALSATimer-0.0 before loading ALSASeq-0.0. Any test for ALSASeq-0.0 is going to fail because ALSATimer-0.0 is not available before installing. To solve the issue, the patchset includes the patch to change test procedure to arrange environment variables. Request URL : https://github.com/alsa-project/alsa-gobject/pull/29 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/29.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH 1/2] dt-bindings: ASoc: Fix tdm-slot documentation spelling error
by Dan Murphy 10 Jun '20

10 Jun '20

Fix the spelling of 'specified'. Also fix grammarical issue with the use of 'a' over 'an' Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- Documentation/devicetree/bindings/sound/tdm-slot.txt | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/tdm-slot.txt b/Documentation/devicetree/bindings/sound/tdm-slot.txt index 34cf70e2cbc4..4bb513ae62fc 100644 --- a/Documentation/devicetree/bindings/sound/tdm-slot.txt +++ b/Documentation/devicetree/bindings/sound/tdm-slot.txt @@ -14,8 +14,8 @@ For instance: dai-tdm-slot-tx-mask = <0 1>; dai-tdm-slot-rx-mask = <1 0>; -And for each spcified driver, there could be one .of_xlate_tdm_slot_mask() -to specify a explicit mapping of the channels and the slots. If it's absent +And for each specified driver, there could be one .of_xlate_tdm_slot_mask() +to specify an explicit mapping of the channels and the slots. If it's absent the default snd_soc_of_xlate_tdm_slot_mask() will be used to generating the tx and rx masks. -- 2.26.2

3 3

sof-hda-dsp: don't fail if Auto-Mute control is not present
by GitHub pull_request - opened 10 Jun '20

10 Jun '20

alsa-project/alsa-ucm-conf pull request #33 was opened from kv2019i: The "Auto-Mute Mode" control is not present in all HDA codecs. The generic SOF HDA UCM file should be robust enough to handle these cases as well. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/33 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/33.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

alsa-gobject: seq: misc fixes
by GitHub pull_request - opened 10 Jun '20

10 Jun '20

alsa-project/alsa-gobject pull request #28 was opened from takaswie: This patchset includes misc fixes for ALSASeq. Request URL : https://github.com/alsa-project/alsa-gobject/pull/28 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/28.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[GIT PULL] ASoC fixes for v5.8
by broonie＠kernel.org 10 Jun '20

10 Jun '20

The following changes since commit a6b675a89e51a1cdad0481b809b7840d3f86e4b5: ASoC: qcom: q6asm-dai: kCFI fix (2020-06-01 12:15:53 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.8 for you to fetch changes up to 44ce45f866c1606f03a411fb2d2c0db4626403a6: Merge series "ASoC: Fix dailink checks for DPCM" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: (2020-06-09 15:46:20 +0100) ---------------------------------------------------------------- ASoC: Fixes for v5.8 A small pile of fixes that came in during the merge window, the DPCM fixes from Pierre are the most notable thing here. ---------------------------------------------------------------- Bard Liao (1): ASoC: core: only convert non DPCM link to DPCM link Colin Ian King (1): ASoC: meson: fix memory leak of links if allocation of ldata fails Dan Murphy (1): dt-bindings: ASoc: Fix tdm-slot documentation spelling error Hans de Goede (2): ASoC: Intel: bytcr_rt5640: Add quirk for Toshiba Encore WT10-A tablet ASoC: rt5645: Add platform-data for Asus T101HA Mark Brown (1): Merge series "ASoC: Fix dailink checks for DPCM" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Oder Chiou (1): ASoC: rl6231: Modify the target DMIC clock rate Pavel Machek (CIP) (1): ASoC: meson: add missing free_irq() in error path Pierre-Louis Bossart (3): ASoC: soc-pcm: dpcm: fix playback/capture checks ASoC: Intel: boards: replace capture_only by dpcm_capture ASoC: SOF: nocodec: conditionally set dpcm_capture/dpcm_playback flags Shengjiu Wang (1): ASoC: fsl-asoc-card: Defer probe when fail to find codec device Steve Lee (1): ASoC: max98390: Fix potential crash during param fw loading Takashi Iwai (1): ASoC: max98390: Fix incorrect printf qualifier .../devicetree/bindings/sound/tdm-slot.txt | 4 +- sound/soc/codecs/max98390.c | 26 ++++++++++--- sound/soc/codecs/max98390.h | 3 +- sound/soc/codecs/rl6231.c | 4 +- sound/soc/codecs/rt5645.c | 14 +++++++ sound/soc/fsl/fsl-asoc-card.c | 2 +- sound/soc/intel/boards/bytcr_rt5640.c | 12 ++++++ sound/soc/intel/boards/glk_rt5682_max98357a.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 4 +- sound/soc/intel/boards/kbl_rt5663_max98927.c | 2 +- .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 2 +- sound/soc/meson/axg-fifo.c | 10 ++++- sound/soc/meson/meson-card-utils.c | 17 ++++++--- sound/soc/soc-core.c | 22 +++++++++-- sound/soc/soc-pcm.c | 44 +++++++++++++++++----- sound/soc/sof/nocodec.c | 6 ++- 16 files changed, 136 insertions(+), 38 deletions(-)

3 3

Re: [PATCH 2/5] soundwire: qcom: add support for mmio soundwire devices
by Srinivas Kandagatla 10 Jun '20

10 Jun '20

On 09/06/2020 12:04, Jonathan Marek wrote: > On 6/9/20 5:19 AM, Srinivas Kandagatla wrote: >> >> >> On 08/06/2020 21:43, Jonathan Marek wrote: >>> Adds support for qcom soundwire devices with memory mapped IO registers. >>> >>> Signed-off-by: Jonathan Marek <jonathan(a)marek.ca> >>> --- >> >> In general patch itself looks pretty trivial, but I would like to see >> what 1.5.1 controller provides in terms of error reporting of >> SoundWire slave register reads/writes. On WCD based controller we did >> not have a mechanism to report things like if the read is ignored or >> not. I was hoping that this version of controller would be able to >> report that. >> >> I will be nice to those patches if that is something which is >> supported in this version. >> >> --srini >> > > It does seem to support additional error reporting (it gets an error > during enumeration after finding the 2 WSA slaves). However the > downstream driver seems to disable this by setting BIT(31) in > FIFO_CFG_ADDR (the comment says "For SWR master version 1.5.1, continue > execute on command ignore"). Outside of the initial enumeration, it > doesn't seem to produce any extra errors (still relying on the timeout > mechanism to know if read/write is ignored). 1.5.* versions support reporting CMD_NACKED in FIFO_STATUS register, so you should use that instead of timeout mechanism which is used for 1.3.0 which did not have support for this. thanks, srini > >>> drivers/soundwire/qcom.c | 25 +++++++++++++++++++++++-- >>> 1 file changed, 23 insertions(+), 2 deletions(-) >>> >>> diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c >>> index f38d1fd3679f..628747df1c75 100644 >>> --- a/drivers/soundwire/qcom.c >>> +++ b/drivers/soundwire/qcom.c >>> @@ -90,6 +90,7 @@ struct qcom_swrm_ctrl { >>> struct sdw_bus bus; >>> struct device *dev; >>> struct regmap *regmap; >>> + void __iomem *mmio; >>> struct completion *comp; >>> struct work_struct slave_work; >>> /* read/write lock */ >>> @@ -154,6 +155,20 @@ static int qcom_swrm_ahb_reg_write(struct >>> qcom_swrm_ctrl *ctrl, >>> return SDW_CMD_OK; >>> } >>> +static int qcom_swrm_cpu_reg_read(struct qcom_swrm_ctrl *ctrl, int reg, >>> + u32 *val) >>> +{ >>> + *val = readl(ctrl->mmio + reg); >>> + return SDW_CMD_OK; >>> +} >>> + >>> +static int qcom_swrm_cpu_reg_write(struct qcom_swrm_ctrl *ctrl, int >>> reg, >>> + int val) >>> +{ >>> + writel(val, ctrl->mmio + reg); >>> + return SDW_CMD_OK; >>> +} >>> + >>> static int qcom_swrm_cmd_fifo_wr_cmd(struct qcom_swrm_ctrl *ctrl, >>> u8 cmd_data, >>> u8 dev_addr, u16 reg_addr) >>> { >>> @@ -746,6 +761,7 @@ static int qcom_swrm_probe(struct platform_device >>> *pdev) >>> struct sdw_master_prop *prop; >>> struct sdw_bus_params *params; >>> struct qcom_swrm_ctrl *ctrl; >>> + struct resource *res; >>> int ret; >>> u32 val; >>> @@ -760,8 +776,13 @@ static int qcom_swrm_probe(struct >>> platform_device *pdev) >>> if (!ctrl->regmap) >>> return -EINVAL; >>> } else { >>> - /* Only WCD based SoundWire controller is supported */ >>> - return -ENOTSUPP; >>> + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); >>> + >>> + ctrl->reg_read = qcom_swrm_cpu_reg_read; >>> + ctrl->reg_write = qcom_swrm_cpu_reg_write; >>> + ctrl->mmio = devm_ioremap_resource(dev, res); >>> + if (IS_ERR(ctrl->mmio)) >>> + return PTR_ERR(ctrl->mmio); >>> } >>> ctrl->irq = of_irq_get(dev->of_node, 0); >>>

1 0

[sound:topic/dma-fix 149/152] sound/core/sgbuf.c:63:11: warning: variable 'prot' set but not used
by kernel test robot 10 Jun '20

10 Jun '20

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/dma-fix head: cef734d7e27e6c8ecfabdc5596647ed3405ab43f commit: f3ed524c87951046aed8fde9db74c1d66c2e6000 [149/152] ALSA: pcm: Avoid vmap() for device SG-buffers config: i386-randconfig-s002-20200607 (attached as .config) compiler: gcc-9 (Debian 9.3.0-13) 9.3.0 reproduce: # apt-get install sparse # sparse version: v0.6.1-247-gcadbd124-dirty git checkout f3ed524c87951046aed8fde9db74c1d66c2e6000 # save the attached .config to linux build tree make W=1 C=1 ARCH=i386 CF='-fdiagnostic-prefix -D__CHECK_ENDIAN__' If you fix the issue, kindly add following tag as appropriate Reported-by: kernel test robot <lkp(a)intel.com> All warnings (new ones prefixed by >>, old ones prefixed by <<): sound/core/sgbuf.c: In function 'snd_malloc_sgbuf_pages': >> sound/core/sgbuf.c:63:11: warning: variable 'prot' set but not used [-Wunused-but-set-variable] 63 | pgprot_t prot = PAGE_KERNEL; | ^~~~ vim +/prot +63 sound/core/sgbuf.c 51e9f2e665bf2b Takashi Iwai 2008-07-30 52 f3ed524c879510 Takashi Iwai 2020-06-09 53 int snd_malloc_sgbuf_pages(struct device *device, ^1da177e4c3f41 Linus Torvalds 2005-04-16 54 size_t size, struct snd_dma_buffer *dmab, ^1da177e4c3f41 Linus Torvalds 2005-04-16 55 size_t *res_size) ^1da177e4c3f41 Linus Torvalds 2005-04-16 56 { ^1da177e4c3f41 Linus Torvalds 2005-04-16 57 struct snd_sg_buf *sgbuf; 51e9f2e665bf2b Takashi Iwai 2008-07-30 58 unsigned int i, pages, chunk, maxpages; ^1da177e4c3f41 Linus Torvalds 2005-04-16 59 struct snd_dma_buffer tmpb; 51e9f2e665bf2b Takashi Iwai 2008-07-30 60 struct snd_sg_page *table; 51e9f2e665bf2b Takashi Iwai 2008-07-30 61 struct page **pgtable; 42e748a0b3251c Takashi Iwai 2018-08-08 62 int type = SNDRV_DMA_TYPE_DEV; 42e748a0b3251c Takashi Iwai 2018-08-08 @63 pgprot_t prot = PAGE_KERNEL; ^1da177e4c3f41 Linus Torvalds 2005-04-16 64 ^1da177e4c3f41 Linus Torvalds 2005-04-16 65 dmab->area = NULL; ^1da177e4c3f41 Linus Torvalds 2005-04-16 66 dmab->addr = 0; 59feddb25f9d92 Panagiotis Issaris 2006-07-25 67 dmab->private_data = sgbuf = kzalloc(sizeof(*sgbuf), GFP_KERNEL); ^1da177e4c3f41 Linus Torvalds 2005-04-16 68 if (! sgbuf) f3ed524c879510 Takashi Iwai 2020-06-09 69 return -ENOMEM; 42e748a0b3251c Takashi Iwai 2018-08-08 70 if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_UC_SG) { 42e748a0b3251c Takashi Iwai 2018-08-08 71 type = SNDRV_DMA_TYPE_DEV_UC; 42e748a0b3251c Takashi Iwai 2018-08-08 72 #ifdef pgprot_noncached 42e748a0b3251c Takashi Iwai 2018-08-08 73 prot = pgprot_noncached(PAGE_KERNEL); 42e748a0b3251c Takashi Iwai 2018-08-08 74 #endif 42e748a0b3251c Takashi Iwai 2018-08-08 75 } ^1da177e4c3f41 Linus Torvalds 2005-04-16 76 sgbuf->dev = device; ^1da177e4c3f41 Linus Torvalds 2005-04-16 77 pages = snd_sgbuf_aligned_pages(size); ^1da177e4c3f41 Linus Torvalds 2005-04-16 78 sgbuf->tblsize = sgbuf_align_table(pages); 51e9f2e665bf2b Takashi Iwai 2008-07-30 79 table = kcalloc(sgbuf->tblsize, sizeof(*table), GFP_KERNEL); 51e9f2e665bf2b Takashi Iwai 2008-07-30 80 if (!table) ^1da177e4c3f41 Linus Torvalds 2005-04-16 81 goto _failed; 51e9f2e665bf2b Takashi Iwai 2008-07-30 82 sgbuf->table = table; 51e9f2e665bf2b Takashi Iwai 2008-07-30 83 pgtable = kcalloc(sgbuf->tblsize, sizeof(*pgtable), GFP_KERNEL); 51e9f2e665bf2b Takashi Iwai 2008-07-30 84 if (!pgtable) ^1da177e4c3f41 Linus Torvalds 2005-04-16 85 goto _failed; 51e9f2e665bf2b Takashi Iwai 2008-07-30 86 sgbuf->page_table = pgtable; ^1da177e4c3f41 Linus Torvalds 2005-04-16 87 51e9f2e665bf2b Takashi Iwai 2008-07-30 88 /* allocate pages */ 51e9f2e665bf2b Takashi Iwai 2008-07-30 89 maxpages = MAX_ALLOC_PAGES; 51e9f2e665bf2b Takashi Iwai 2008-07-30 90 while (pages > 0) { 51e9f2e665bf2b Takashi Iwai 2008-07-30 91 chunk = pages; 51e9f2e665bf2b Takashi Iwai 2008-07-30 92 /* don't be too eager to take a huge chunk */ 51e9f2e665bf2b Takashi Iwai 2008-07-30 93 if (chunk > maxpages) 51e9f2e665bf2b Takashi Iwai 2008-07-30 94 chunk = maxpages; 51e9f2e665bf2b Takashi Iwai 2008-07-30 95 chunk <<= PAGE_SHIFT; 42e748a0b3251c Takashi Iwai 2018-08-08 96 if (snd_dma_alloc_pages_fallback(type, device, 51e9f2e665bf2b Takashi Iwai 2008-07-30 97 chunk, &tmpb) < 0) { 51e9f2e665bf2b Takashi Iwai 2008-07-30 98 if (!sgbuf->pages) c810f9039f0406 Takashi Iwai 2012-08-03 99 goto _failed; 51e9f2e665bf2b Takashi Iwai 2008-07-30 100 if (!res_size) ^1da177e4c3f41 Linus Torvalds 2005-04-16 101 goto _failed; 51e9f2e665bf2b Takashi Iwai 2008-07-30 102 size = sgbuf->pages * PAGE_SIZE; ^1da177e4c3f41 Linus Torvalds 2005-04-16 103 break; ^1da177e4c3f41 Linus Torvalds 2005-04-16 104 } 51e9f2e665bf2b Takashi Iwai 2008-07-30 105 chunk = tmpb.bytes >> PAGE_SHIFT; 51e9f2e665bf2b Takashi Iwai 2008-07-30 106 for (i = 0; i < chunk; i++) { 51e9f2e665bf2b Takashi Iwai 2008-07-30 107 table->buf = tmpb.area; 51e9f2e665bf2b Takashi Iwai 2008-07-30 108 table->addr = tmpb.addr; 51e9f2e665bf2b Takashi Iwai 2008-07-30 109 if (!i) 51e9f2e665bf2b Takashi Iwai 2008-07-30 110 table->addr |= chunk; /* mark head */ 51e9f2e665bf2b Takashi Iwai 2008-07-30 111 table++; 51e9f2e665bf2b Takashi Iwai 2008-07-30 112 *pgtable++ = virt_to_page(tmpb.area); 51e9f2e665bf2b Takashi Iwai 2008-07-30 113 tmpb.area += PAGE_SIZE; 51e9f2e665bf2b Takashi Iwai 2008-07-30 114 tmpb.addr += PAGE_SIZE; 51e9f2e665bf2b Takashi Iwai 2008-07-30 115 } 51e9f2e665bf2b Takashi Iwai 2008-07-30 116 sgbuf->pages += chunk; 51e9f2e665bf2b Takashi Iwai 2008-07-30 117 pages -= chunk; 51e9f2e665bf2b Takashi Iwai 2008-07-30 118 if (chunk < maxpages) 51e9f2e665bf2b Takashi Iwai 2008-07-30 119 maxpages = chunk; ^1da177e4c3f41 Linus Torvalds 2005-04-16 120 } ^1da177e4c3f41 Linus Torvalds 2005-04-16 121 ^1da177e4c3f41 Linus Torvalds 2005-04-16 122 sgbuf->size = size; f3ed524c879510 Takashi Iwai 2020-06-09 123 dmab->addr = -1UL; /* some non-NULL value as validity */ 51e9f2e665bf2b Takashi Iwai 2008-07-30 124 if (res_size) 51e9f2e665bf2b Takashi Iwai 2008-07-30 125 *res_size = sgbuf->size; f3ed524c879510 Takashi Iwai 2020-06-09 126 return 0; ^1da177e4c3f41 Linus Torvalds 2005-04-16 127 ^1da177e4c3f41 Linus Torvalds 2005-04-16 128 _failed: ^1da177e4c3f41 Linus Torvalds 2005-04-16 129 snd_free_sgbuf_pages(dmab); /* free the table */ f3ed524c879510 Takashi Iwai 2020-06-09 130 return -ENOMEM; ^1da177e4c3f41 Linus Torvalds 2005-04-16 131 } 9d069dc00b02b8 Takashi Iwai 2012-09-20 132 :::::: The code at line 63 was first introduced by commit :::::: 42e748a0b3251cca0de2c269ca106884907eb289 ALSA: memalloc: Add non-cached buffer type :::::: TO: Takashi Iwai <tiwai(a)suse.de> :::::: CC: Takashi Iwai <tiwai(a)suse.de> --- 0-DAY CI Kernel Test Service, Intel Corporation https://lists.01.org/hyperkitty/list/kbuild-all@lists.01.org

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Re: [PATCH 3/5] soundwire: qcom: add v1.5.1 compatible
by Srinivas Kandagatla 10 Jun '20

10 Jun '20

On 09/06/2020 12:17, Jonathan Marek wrote: > On 6/9/20 1:26 AM, Vinod Koul wrote: >> On 08-06-20, 16:43, Jonathan Marek wrote: >>> Add a compatible string for HW version v1.5.1 on sm8250 SoCs. >> >> Please document this new compatible >> > > Does it really need to be documented? The documentation already says the > compatible should be "qcom,soundwire-v<MAJOR>.<MINOR>.<STEP>". It gives > "qcom,soundwire-v1.5.0" as an example, which is not actually a supported > compatible, so my understanding is we don't need to update the list of > examples with every possible compatible. checkpatch should have reported about this, and in future once we convert to yaml and list the compatible strings then dt_binding_check would fail too. So there is no harm in adding an additional compatible string for this new entry. --srini

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[sound:topic/dma-fix2 150/150] sound/core/pcm_memory.c:44:45: error: lvalue required as left operand of assignment
by kernel test robot 10 Jun '20

10 Jun '20

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/dma-fix2 head: de846f4c8c865ca0fe3952ee80ad165f4c64ea34 commit: de846f4c8c865ca0fe3952ee80ad165f4c64ea34 [150/150] ALSA: pcm: Use SG-buffer only when direct DMA is available config: nios2-allyesconfig (attached as .config) compiler: nios2-linux-gcc (GCC) 9.3.0 reproduce (this is a W=1 build): wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout de846f4c8c865ca0fe3952ee80ad165f4c64ea34 # save the attached .config to linux build tree COMPILER_INSTALL_PATH=$HOME/0day COMPILER=gcc-9.3.0 make.cross ARCH=nios2 If you fix the issue, kindly add following tag as appropriate Reported-by: kernel test robot <lkp(a)intel.com> All error/warnings (new ones prefixed by >>, old ones prefixed by <<): sound/core/pcm_memory.c: In function 'do_alloc_pages': >> sound/core/pcm_memory.c:44:45: error: lvalue required as left operand of assignment 44 | (type == SNDRV_DMA_TYPE_DEV_SG || type = SNDRV_DMA_TYPE_DEV_UC_SG) && | ^ >> sound/core/pcm_memory.c:45:6: error: implicit declaration of function 'dma_is_direct'; did you mean 'io_is_direct'? [-Werror=implicit-function-declaration] 45 | dma_is_direct(get_dma_ops(dev))) { | ^~~~~~~~~~~~~ | io_is_direct >> sound/core/pcm_memory.c:45:20: error: implicit declaration of function 'get_dma_ops' [-Werror=implicit-function-declaration] 45 | dma_is_direct(get_dma_ops(dev))) { | ^~~~~~~~~~~ In file included from include/linux/printk.h:326, from include/linux/kernel.h:15, from include/asm-generic/bug.h:19, from ./arch/nios2/include/generated/asm/bug.h:1, from include/linux/bug.h:5, from include/linux/io.h:11, from sound/core/pcm_memory.c:7: >> sound/core/pcm_memory.c:47:11: error: 'device' undeclared (first use in this function) 47 | dev_dbg(device, "Use continuous page allocatorn"); | ^~~~~~ include/linux/dynamic_debug.h:125:15: note: in definition of macro '__dynamic_func_call' 125 | func(&id, ##__VA_ARGS__); | ^~~~~~~~~~~ include/linux/dynamic_debug.h:157:2: note: in expansion of macro '_dynamic_func_call' 157 | _dynamic_func_call(fmt,__dynamic_dev_dbg, | ^~~~~~~~~~~~~~~~~~ include/linux/dev_printk.h:114:2: note: in expansion of macro 'dynamic_dev_dbg' 114 | dynamic_dev_dbg(dev, dev_fmt(fmt), ##__VA_ARGS__) | ^~~~~~~~~~~~~~~ >> sound/core/pcm_memory.c:47:3: note: in expansion of macro 'dev_dbg' 47 | dev_dbg(device, "Use continuous page allocatorn"); | ^~~~~~~ sound/core/pcm_memory.c:47:11: note: each undeclared identifier is reported only once for each function it appears in 47 | dev_dbg(device, "Use continuous page allocatorn"); | ^~~~~~ include/linux/dynamic_debug.h:125:15: note: in definition of macro '__dynamic_func_call' 125 | func(&id, ##__VA_ARGS__); | ^~~~~~~~~~~ include/linux/dynamic_debug.h:157:2: note: in expansion of macro '_dynamic_func_call' 157 | _dynamic_func_call(fmt,__dynamic_dev_dbg, | ^~~~~~~~~~~~~~~~~~ include/linux/dev_printk.h:114:2: note: in expansion of macro 'dynamic_dev_dbg' 114 | dynamic_dev_dbg(dev, dev_fmt(fmt), ##__VA_ARGS__) | ^~~~~~~~~~~~~~~ >> sound/core/pcm_memory.c:47:3: note: in expansion of macro 'dev_dbg' 47 | dev_dbg(device, "Use continuous page allocatorn"); | ^~~~~~~ cc1: some warnings being treated as errors vim +44 sound/core/pcm_memory.c 33 34 static int do_alloc_pages(struct snd_card *card, int type, struct device *dev, 35 size_t size, struct snd_dma_buffer *dmab) 36 { 37 int err; 38 39 if (max_alloc_per_card && 40 card->total_pcm_alloc_bytes + size > max_alloc_per_card) 41 return -ENOMEM; 42 43 if (IS_ENABLED(CONFIG_SND_DMA_SGBUF) && > 44 (type == SNDRV_DMA_TYPE_DEV_SG || type = SNDRV_DMA_TYPE_DEV_UC_SG) && > 45 dma_is_direct(get_dma_ops(dev))) { 46 /* mutate to continuous page allocation */ > 47 dev_dbg(device, "Use continuous page allocator\n"); 48 if (type == SNDRV_DMA_TYPE_DEV_SG) 49 type = SNDRV_DMA_TYPE_DEV; 50 else 51 type = SNDRV_DMA_TYPE_DEV_UC; 52 } 53 54 err = snd_dma_alloc_pages(type, dev, size, dmab); 55 if (!err) { 56 mutex_lock(&card->memory_mutex); 57 card->total_pcm_alloc_bytes += dmab->bytes; 58 mutex_unlock(&card->memory_mutex); 59 } 60 return err; 61 } 62 --- 0-DAY CI Kernel Test Service, Intel Corporation https://lists.01.org/hyperkitty/list/kbuild-all@lists.01.org

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Re: [PATCH 4/5] soundwire: qcom: avoid dependency on CONFIG_SLIMBUS
by Srinivas Kandagatla 10 Jun '20

10 Jun '20

On 09/06/2020 12:33, Jonathan Marek wrote: > On 6/9/20 5:52 AM, Srinivas Kandagatla wrote: >> >> >> On 08/06/2020 21:43, Jonathan Marek wrote: >>> The driver may be used without slimbus, so don't depend on slimbus. >>> >>> Signed-off-by: Jonathan Marek <jonathan(a)marek.ca> >>> --- >>> drivers/soundwire/Kconfig | 1 - >>> drivers/soundwire/qcom.c | 5 +++++ >>> 2 files changed, 5 insertions(+), 1 deletion(-) >>> >>> diff --git a/drivers/soundwire/Kconfig b/drivers/soundwire/Kconfig >>> index fa2b4ab92ed9..d121cf739090 100644 >>> --- a/drivers/soundwire/Kconfig >>> +++ b/drivers/soundwire/Kconfig >>> @@ -33,7 +33,6 @@ config SOUNDWIRE_INTEL >>> config SOUNDWIRE_QCOM >>> tristate "Qualcomm SoundWire Master driver" >>> - depends on SLIMBUS >>> depends on SND_SOC >> >> Why not move this to imply SLIMBUS this will give more flexibility! >> >> > > If you mean to change it to "select SLIMBUS", I'd prefer not to, because > this would increase code size unnecessarily in my kernel. imply is week select, which means that this driver can be built without SLIMBus selected. So removing reference to slimbus_bus is necessary in this case. On the other hand, SLIMBus is going to be used sm8250 for BT audio, so this would not be unnecessary. Also mostly these are build as modules, so not sure why kernel size will increase here! Am not 100% sure if SLIMBus will be kept on all SoCs, but keeping depends or imply in place would enforce or spell out some level of dependency on this. > >>> help >>> SoundWire Qualcomm Master driver. >>> diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c >>> index 14334442615f..ac81c64768ea 100644 >>> --- a/drivers/soundwire/qcom.c >>> +++ b/drivers/soundwire/qcom.c >>> @@ -769,13 +769,18 @@ static int qcom_swrm_probe(struct >>> platform_device *pdev) >>> if (!ctrl) >>> return -ENOMEM; >>> +#ifdef CONFIG_SLIMBUS >>> if (dev->parent->bus == &slimbus_bus) { >>> +#else >>> + if (false) { >>> +#endif >> >> May be you can do bit more cleanup here, which could endup like: >> >> >> ctrl->regmap = dev_get_regmap(dev->parent, NULL); >> if (!ctrl->regmap) { >> res = platform_get_resource(pdev, IORESOURCE_MEM, 0); >> if (res) { >> ctrl->mmio = devm_ioremap_resource(dev, res); >> if (IS_ERR(ctrl->mmio)) { >> dev_err(dev, "No valid mem resource found\n"); >> return PTR_ERR(ctrl->mmio); >> } >> >> ctrl->reg_read = qcom_swrm_cpu_reg_read; >> ctrl->reg_write = qcom_swrm_cpu_reg_write; >> } else { >> dev_err(dev, "No valid slim resource found\n"); >> return -EINVAL; >> } >> } else { >> ctrl->reg_read = qcom_swrm_ahb_reg_read; >> ctrl->reg_write = qcom_swrm_ahb_reg_write; >> } >> >> >> >> thanks, >> srini > > I don't think this is better, it feels more obfuscated, and I think its > possible we may end up with the mmio sdw having a parent with a regmap. > (it is not how I did things up in my upstream stack, but in downstream > the sdw nodes are inside the "macro" codec nodes) > > I understand the '#ifdef CONFIG_SLIMBUS'/'dev->parent->bus == > &slimbus_bus' is ugly, but at least its clear what's going on. Unless > you have a better suggestion? Other suggestion I had in my mind was to use compatible strings to get reg_read, reg_write callbacks + some flags like (if_type) populated. This can help looking up resources correctly. Thanks, srini > >>> ctrl->reg_read = qcom_swrm_ahb_reg_read; >>> ctrl->reg_write = qcom_swrm_ahb_reg_write; >>> ctrl->regmap = dev_get_regmap(dev->parent, NULL); >>> if (!ctrl->regmap) >>> return -EINVAL; >>> } else { >>> + >>> res = platform_get_resource(pdev, IORESOURCE_MEM, 0); >>> ctrl->reg_read = qcom_swrm_cpu_reg_read; >>>

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We were woken up with POLLOUT set
by Stuart Naylor 09 Jun '20

09 Jun '20

Yeah pulseaudio is doing the same on Raspbian with USB devices. Sent from Mail<https://go.microsoft.com/fwlink/?LinkId=550986> for Windows 10 From: alsa-devel-request(a)alsa-project.org<mailto:alsa-devel-request@alsa-project.org> Sent: 09 June 2020 22:40 To: alsa-devel(a)alsa-project.org<mailto:alsa-devel@alsa-project.org> Subject: Alsa-devel Digest, Vol 160, Issue 72 Send Alsa-devel mailing list submissions to alsa-devel(a)alsa-project.org To subscribe or unsubscribe via the World Wide Web, visit https://mailman.alsa-project.org/mailman/listinfo/alsa-devel or, via email, send a message with subject or body 'help' to alsa-devel-request(a)alsa-project.org You can reach the person managing the list at alsa-devel-owner(a)alsa-project.org When replying, please edit your Subject line so it is more specific than "Re: Contents of Alsa-devel digest..." Today's Topics: 1. Re: [RFC PATCH 0/2] TAS2563 DSP Firmware Loader (Dan Murphy) 2. Re: [RFC PATCH 0/2] TAS2563 DSP Firmware Loader (Mark Brown) 3. Re: [RFC PATCH 1/2] dt-bindings: tas2562: Add firmware support for tas2563 (Mark Brown) 4. Re: [RFC PATCH 1/2] dt-bindings: tas2562: Add firmware support for tas2563 (Dan Murphy) 5. We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Jun 09 14:31:33 elrond pulseaudio[11933]: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Jun 09 14:31:33 elrond pulseaudio[11933]: ALSA woke us up to write new data to the device, but there was actually nothing to write. (GitHub issues - opened) 6. We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Jun 09 14:31:33 elrond pulseaudio[11933]: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Jun 09 14:31:33 elrond pulseaudio[11933]: ALSA woke us up to write new data to the device, but there was actually nothing to write. (GitHub issues - edited) 7. ALSA journalctl error. (GitHub issues - edited) 8. ALSA journalctl Error (GitHub issues - edited) ---------------------------------------------------------------------- Message: 1 Date: Tue, 9 Jun 2020 13:07:50 -0500 From: Dan Murphy <dmurphy(a)ti.com> To: Mark Brown <broonie(a)kernel.org> Cc: robh(a)kernel.org, alsa-devel(a)alsa-project.org, devicetree(a)vger.kernel.org, linux-kernel(a)vger.kernel.org, tiwai(a)suse.com, lgirdwood(a)gmail.com Subject: Re: [RFC PATCH 0/2] TAS2563 DSP Firmware Loader Message-ID: <6d6aaed3-dac8-e1ec-436c-9b04273df2b3(a)ti.com> Content-Type: text/plain; charset="windows-1252"; format=flowed Mark On 6/9/20 12:52 PM, Mark Brown wrote: > On Tue, Jun 09, 2020 at 12:28:39PM -0500, Dan Murphy wrote: > >> These programs and configurations are selectable via files under the I2C dev >> node. There may be a better way to select this through ALSA controls but I was >> unable to find a good example of this. This is why this is an RFC patchset. > I think you can just use enums for most of this - what you want to do I > think is parse the firmware, build templates for the controls and then > add them with snd_soc_add_component_controls(). Userspace *should* cope > with controls being hotplugged. Yes this was my concern if userspace could cope with dynamic controls. Dan ------------------------------ Message: 2 Date: Tue, 9 Jun 2020 19:16:15 +0100 From: Mark Brown <broonie(a)kernel.org> To: Dan Murphy <dmurphy(a)ti.com> Cc: robh(a)kernel.org, alsa-devel(a)alsa-project.org, devicetree(a)vger.kernel.org, linux-kernel(a)vger.kernel.org, tiwai(a)suse.com, lgirdwood(a)gmail.com Subject: Re: [RFC PATCH 0/2] TAS2563 DSP Firmware Loader Message-ID: <20200609181615.GR4583(a)sirena.org.uk> Content-Type: text/plain; charset="us-ascii" On Tue, Jun 09, 2020 at 01:07:50PM -0500, Dan Murphy wrote: > On 6/9/20 12:52 PM, Mark Brown wrote: > > I think you can just use enums for most of this - what you want to do I > > think is parse the firmware, build templates for the controls and then > > add them with snd_soc_add_component_controls(). Userspace *should* cope > > with controls being hotplugged. > Yes this was my concern if userspace could cope with dynamic controls. Things like alsactl definitely do, and obviously anything that starts after the firmware loads will be fine too.

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ALSA journalctl Error
by GitHub issues - edited 09 Jun '20

09 Jun '20

alsa-project/alsa-lib issue #57 was edited from danieloppenlander: I ran across an error in my journalctl on Ubuntu 20.04. It said to report a bug. Here is my alsa-info.sh output: http://alsa-project.org/db/?f=6b505aa5138f2a974037ab5c3a89e1605807df97 Here is the journalctl message: ``` pulseaudio[11933]: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. pulseaudio[11933]: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. pulseaudio[11933]: ALSA woke us up to write new data to the device, but there was actually nothing to write. ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/57 Repository URL: https://github.com/alsa-project/alsa-lib

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ALSA journalctl Error
by GitHub issues - edited 09 Jun '20

09 Jun '20

alsa-project/alsa-lib issue #57 was edited from danieloppenlander: I ran across an error in my journalctl on Ubuntu 20.04. It said to report a bug. Here is my alsa-info.sh output: http://alsa-project.org/db/?f=6b505aa5138f2a974037ab5c3a89e1605807df97 Here is the journalctl message: ``` We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Jun 09 14:31:33 elrond pulseaudio[11933]: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Jun 09 14:31:33 elrond pulseaudio[11933]: ALSA woke us up to write new data to the device, but there was actually nothing to write. ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/57 Repository URL: https://github.com/alsa-project/alsa-lib

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ALSA journalctl error.
by GitHub issues - edited 09 Jun '20

09 Jun '20

alsa-project/alsa-lib issue #57 was edited from danieloppenlander: I ran across an error in my journalctl on Ubuntu 20.04. It said to report a bug. Here is my alsa-info.sh output: http://alsa-project.org/db/?f=6b505aa5138f2a974037ab5c3a89e1605807df97 Here is the journalctl message: ``` We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Jun 09 14:31:33 elrond pulseaudio[11933]: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Jun 09 14:31:33 elrond pulseaudio[11933]: ALSA woke us up to write new data to the device, but there was actually nothing to write. ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/57 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Jun 09 14:31:33 elrond pulseaudio[11933]: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Jun 09 14:31:33 elrond pulseaudio[11933]: ALSA woke us up to write new data to the device, but there was actually nothing to write.
by GitHub issues - edited 09 Jun '20

09 Jun '20

alsa-project/alsa-lib issue #57 was edited from danieloppenlander: I ran across an error in my journalctl on Ubuntu 20.04. It said to report a bug. Here is my alsa-info.sh output: http://alsa-project.org/db/?f=6b505aa5138f2a974037ab5c3a89e1605807df97 Here is the journalctl message: ``` We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Jun 09 14:31:33 elrond pulseaudio[11933]: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Jun 09 14:31:33 elrond pulseaudio[11933]: ALSA woke us up to write new data to the device, but there was actually nothing to write. ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/57 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Jun 09 14:31:33 elrond pulseaudio[11933]: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Jun 09 14:31:33 elrond pulseaudio[11933]: ALSA woke us up to write new data to the device, but there was actually nothing to write.
by GitHub issues - opened 09 Jun '20

09 Jun '20

alsa-project/alsa-lib issue #57 was opened from danieloppenlander: I ran across an error in my journalctl on Ubuntu 20.04. It said to report a bug. Here is my alsa-info.sh output: http://alsa-project.org/db/?f=6b505aa5138f2a974037ab5c3a89e1605807df97 Issue URL : https://github.com/alsa-project/alsa-lib/issues/57 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 1/2] ASoC: Intel: bytcr_rt5640: Add quirk for Toshiba Encore WT10-A tablet
by Hans de Goede 09 Jun '20

09 Jun '20

The Toshiba Encore WT10-A tablet almost fully works with the default settings for Bay Trail CR devices. The only issue is that it uses a digital mic. connected the the DMIC1 input instead of an analog mic. Add a quirk for this model using the default settings with the input-map replaced with BYT_RT5640_DMIC1_MAP. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/bytcr_rt5640.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index fbfd53874b47..5c1a5e2aff6f 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -754,6 +754,18 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_JD_NOT_INV | BYT_RT5640_MCLK_EN), }, + { /* Toshiba Encore WT10-A */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "TOSHIBA"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "TOSHIBA WT10-A-103"), + }, + .driver_data = (void *)(BYT_RT5640_DMIC1_MAP | + BYT_RT5640_JD_SRC_JD1_IN4P | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_SSP0_AIF2 | + BYT_RT5640_MCLK_EN), + }, { /* Catch-all for generic Insyde tablets, must be last */ .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Insyde"), -- 2.26.2

3 5

[PATCH v2] dt-bindings: tas2562: Convert the tas2562 binding to yaml
by Dan Murphy 09 Jun '20

09 Jun '20

Convert the TAS2562 text file to yaml format. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- .../devicetree/bindings/sound/tas2562.txt | 34 ---------- .../devicetree/bindings/sound/tas2562.yaml | 65 +++++++++++++++++++ 2 files changed, 65 insertions(+), 34 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/tas2562.txt create mode 100644 Documentation/devicetree/bindings/sound/tas2562.yaml diff --git a/Documentation/devicetree/bindings/sound/tas2562.txt b/Documentation/devicetree/bindings/sound/tas2562.txt deleted file mode 100644 index 94796b547184..000000000000 --- a/Documentation/devicetree/bindings/sound/tas2562.txt +++ /dev/null @@ -1,34 +0,0 @@ -Texas Instruments TAS2562 Smart PA - -The TAS2562 is a mono, digital input Class-D audio amplifier optimized for -efficiently driving high peak power into small loudspeakers. -Integrated speaker voltage and current sense provides for -real time monitoring of loudspeaker behavior. - -Required properties: - - #address-cells - Should be <1>. - - #size-cells - Should be <0>. - - compatible: - Should contain "ti,tas2562", "ti,tas2563". - - reg: - The i2c address. Should be 0x4c, 0x4d, 0x4e or 0x4f. - - ti,imon-slot-no:- TDM TX current sense time slot. - -Optional properties: -- interrupt-parent: phandle to the interrupt controller which provides - the interrupt. -- interrupts: (GPIO) interrupt to which the chip is connected. -- shut-down: GPIO used to control the state of the device. - -Examples: -tas2562@4c { - #address-cells = <1>; - #size-cells = <0>; - compatible = "ti,tas2562"; - reg = <0x4c>; - - interrupt-parent = <&gpio1>; - interrupts = <14>; - - shut-down = <&gpio1 15 0>; - ti,imon-slot-no = <0>; -}; - diff --git a/Documentation/devicetree/bindings/sound/tas2562.yaml b/Documentation/devicetree/bindings/sound/tas2562.yaml new file mode 100644 index 000000000000..c6168aa32954 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2562.yaml @@ -0,0 +1,65 @@ +# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause) +# Copyright (C) 2019 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: "http://devicetree.org/schemas/sound/tas2562.yaml#" +$schema: "http://devicetree.org/meta-schemas/core.yaml#" + +title: Texas Instruments TAS2562 Smart PA + +maintainers: + - Dan Murphy <dmurphy(a)ti.com> + +description: | + The TAS2562 is a mono, digital input Class-D audio amplifier optimized for + efficiently driving high peak power into small loudspeakers. + Integrated speaker voltage and current sense provides for + real time monitoring of loudspeaker behavior. + +properties: + compatible: + enum: + - ti,tas2562 + - ti,tas2563 + + reg: + maxItems: 1 + description: | + I2C address of the device can be one of these 0x4c, 0x4d, 0x4e or 0x4f + + shut-down: + description: GPIO used to control the state of the device. + + interrupts: + maxItems: 1 + + ti,imon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + description: TDM TX current sense time slot. + + '#sound-dai-cells': + const: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c0 { + #address-cells = <1>; + #size-cells = <0>; + codec: codec@4c { + compatible = "ti,tas2562"; + reg = <0x4c>; + #sound-dai-cells = <1>; + interrupt-parent = <&gpio1>; + interrupts = <14>; + shut-down = <&gpio1 15 0>; + ti,imon-slot-no = <0>; + }; + }; + -- 2.26.2

1 0

[RFC PATCH] ASoC: fsl_asrc_dma: Fix warning "Cannot create DMA dma:tx symlink"
by Shengjiu Wang 09 Jun '20

09 Jun '20

The issue log is: [ 48.021506] CPU: 0 PID: 664 Comm: aplay Not tainted 5.7.0-rc1-13120-g12b434cbbea0 #343 [ 48.031063] Hardware name: Freescale i.MX6 Quad/DualLite (Device Tree) [ 48.037638] [<c0110dd8>] (unwind_backtrace) from [<c010b8ec>] (show_stack+0x10/0x14) [ 48.045413] [<c010b8ec>] (show_stack) from [<c0557fc0>] (dump_stack+0xe4/0x118) [ 48.052757] [<c0557fc0>] (dump_stack) from [<c032aeb4>] (sysfs_warn_dup+0x50/0x64) [ 48.060357] [<c032aeb4>] (sysfs_warn_dup) from [<c032b1a8>] (sysfs_do_create_link_sd+0xc8/0xd4) [ 48.069086] [<c032b1a8>] (sysfs_do_create_link_sd) from [<c05dc668>] (dma_request_chan+0xb0/0x210) [ 48.078068] [<c05dc668>] (dma_request_chan) from [<c05dc7d0>] (dma_request_slave_channel+0x8/0x14) [ 48.087060] [<c05dc7d0>] (dma_request_slave_channel) from [<c09d5cd4>] (fsl_asrc_dma_hw_params+0x1dc/0x434) [ 48.096831] [<c09d5cd4>] (fsl_asrc_dma_hw_params) from [<c09c143c>] (soc_pcm_hw_params+0x4b0/0x650) [ 48.105903] [<c09c143c>] (soc_pcm_hw_params) from [<c09c36a8>] (dpcm_fe_dai_hw_params+0x70/0xe4) [ 48.114715] [<c09c36a8>] (dpcm_fe_dai_hw_params) from [<c099b274>] (snd_pcm_hw_params+0x158/0x418) [ 48.123701] [<c099b274>] (snd_pcm_hw_params) from [<c099c5a0>] (snd_pcm_ioctl+0x734/0x183c) [ 48.132079] [<c099c5a0>] (snd_pcm_ioctl) from [<c029ff94>] (ksys_ioctl+0x2ac/0xb98) [ 48.139765] [<c029ff94>] (ksys_ioctl) from [<c0100080>] (ret_fast_syscall+0x0/0x28) [ 48.147440] Exception stack(0xed3c5fa8 to 0xed3c5ff0) [ 48.152515] 5fa0: bec28670 00e92870 00000004 c25c4111 bec28670 0002000f [ 48.160716] 5fc0: bec28670 00e92870 00e92820 00000036 0000bb80 00000000 0002c2f8 00000003 [ 48.168913] 5fe0: b6f4d3fc bec2853c b6ee7655 b6e22cf8 [ 48.174236] fsl-esai-dai 2024000.esai: Cannot create DMA dma:tx symlink The dma channel is already requested by Back-End cpu dai driver, if fsl_asrc_dma requests dma chan with same dma:tx symlink, then this warning comes out. The warning is added by commit 71723a96b8b1 ("dmaengine: Create symlinks between DMA channels and slaves") commit bad83565eafe ("dmaengine: Cleanups for the slave <-> channel symlink support") As the dma channel is requested by Back-End, we need to reuse the channel and avoid to request a new one, then the issue can be fixed. In order to get the dma channel which is already requested in Back-End. we export two functions (snd_soc_lookup_component_nolocked and soc_component_to_pcm), if we can get the dma channel, then reuse it. if can't, then request a new one. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- include/sound/dmaengine_pcm.h | 11 ++++++ include/sound/soc.h | 2 ++ sound/soc/fsl/fsl_asrc_common.h | 2 ++ sound/soc/fsl/fsl_asrc_dma.c | 49 +++++++++++++++++++++------ sound/soc/soc-core.c | 3 +- sound/soc/soc-generic-dmaengine-pcm.c | 12 ------- 6 files changed, 55 insertions(+), 24 deletions(-) diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index b65220685920..8c5e38180fb0 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -161,4 +161,15 @@ int snd_dmaengine_pcm_prepare_slave_config(struct snd_pcm_substream *substream, #define SND_DMAENGINE_PCM_DRV_NAME "snd_dmaengine_pcm" +struct dmaengine_pcm { + struct dma_chan *chan[SNDRV_PCM_STREAM_LAST + 1]; + const struct snd_dmaengine_pcm_config *config; + struct snd_soc_component component; + unsigned int flags; +}; + +static inline struct dmaengine_pcm *soc_component_to_pcm(struct snd_soc_component *p) +{ + return container_of(p, struct dmaengine_pcm, component); +} #endif diff --git a/include/sound/soc.h b/include/sound/soc.h index 74868436ac79..565612a8d690 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -444,6 +444,8 @@ int devm_snd_soc_register_component(struct device *dev, const struct snd_soc_component_driver *component_driver, struct snd_soc_dai_driver *dai_drv, int num_dai); void snd_soc_unregister_component(struct device *dev); +struct snd_soc_component *snd_soc_lookup_component_nolocked(struct device *dev, + const char *driver_name); struct snd_soc_component *snd_soc_lookup_component(struct device *dev, const char *driver_name); diff --git a/sound/soc/fsl/fsl_asrc_common.h b/sound/soc/fsl/fsl_asrc_common.h index 77665b15c8db..09512bc79b80 100644 --- a/sound/soc/fsl/fsl_asrc_common.h +++ b/sound/soc/fsl/fsl_asrc_common.h @@ -32,6 +32,7 @@ enum asrc_pair_index { * @dma_chan: inputer and output DMA channels * @dma_data: private dma data * @pos: hardware pointer position + * @req_dma_chan_dev_to_dev: flag for release dev_to_dev chan * @private: pair private area */ struct fsl_asrc_pair { @@ -45,6 +46,7 @@ struct fsl_asrc_pair { struct dma_chan *dma_chan[2]; struct imx_dma_data dma_data; unsigned int pos; + bool req_dma_chan_dev_to_dev; void *private; }; diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index d6a3fc5f87e5..3ad862225326 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -135,6 +135,7 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, struct snd_dmaengine_dai_dma_data *dma_params_be = NULL; struct snd_pcm_runtime *runtime = substream->runtime; struct fsl_asrc_pair *pair = runtime->private_data; + struct snd_soc_component *component_be = NULL; struct fsl_asrc *asrc = pair->asrc; struct dma_slave_config config_fe, config_be; enum asrc_pair_index index = pair->index; @@ -142,7 +143,7 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, int stream = substream->stream; struct imx_dma_data *tmp_data; struct snd_soc_dpcm *dpcm; - struct dma_chan *tmp_chan; + struct dma_chan *tmp_chan = NULL, *tmp_chan_new = NULL; struct device *dev_be; u8 dir = tx ? OUT : IN; dma_cap_mask_t mask; @@ -160,6 +161,7 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, substream_be = snd_soc_dpcm_get_substream(be, stream); dma_params_be = snd_soc_dai_get_dma_data(dai, substream_be); dev_be = dai->dev; + component_be = snd_soc_lookup_component_nolocked(dev_be, SND_DMAENGINE_PCM_DRV_NAME); break; } @@ -205,10 +207,16 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, */ if (!asrc->use_edma) { /* Get DMA request of Back-End */ - tmp_chan = dma_request_slave_channel(dev_be, tx ? "tx" : "rx"); + if (component_be) + tmp_chan = soc_component_to_pcm(component_be)->chan[substream->stream]; + if (!tmp_chan) { + tmp_chan_new = dma_request_slave_channel(dev_be, tx ? "tx" : "rx"); + tmp_chan = tmp_chan_new; + } tmp_data = tmp_chan->private; pair->dma_data.dma_request = tmp_data->dma_request; - dma_release_channel(tmp_chan); + if (tmp_chan_new) + dma_release_channel(tmp_chan_new); /* Get DMA request of Front-End */ tmp_chan = asrc->get_dma_channel(pair, dir); @@ -220,9 +228,26 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, pair->dma_chan[dir] = dma_request_channel(mask, filter, &pair->dma_data); + pair->req_dma_chan_dev_to_dev = true; } else { - pair->dma_chan[dir] = - asrc->get_dma_channel(pair, dir); + /* With EDMA, there is two dma channels can be used for p2p, + * one is from ASRC, one is from another peripheral + * (ESAI or SAI) previously we select the dma channel of ASRC, + * but find an issue for ideal ratio case, there is no control + * for data copy speed, the speed is faster than sample + * frequency. + * + * So we switch to dma channel of peripheral (ESAI or SAI), + * that copy speed of DMA is controlled by data consumption + * speed in the peripheral FIFO. + */ + pair->req_dma_chan_dev_to_dev = false; + if (component_be) + pair->dma_chan[dir] = soc_component_to_pcm(component_be)->chan[substream->stream];; + if (!pair->dma_chan[dir]) { + pair->dma_chan[dir] = dma_request_slave_channel(dev_be, tx ? "tx" : "rx"); + pair->req_dma_chan_dev_to_dev = true; + } } if (!pair->dma_chan[dir]) { @@ -273,19 +298,21 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, static int fsl_asrc_dma_hw_free(struct snd_soc_component *component, struct snd_pcm_substream *substream) { + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; struct snd_pcm_runtime *runtime = substream->runtime; struct fsl_asrc_pair *pair = runtime->private_data; + u8 dir = tx ? OUT : IN; snd_pcm_set_runtime_buffer(substream, NULL); - if (pair->dma_chan[IN]) - dma_release_channel(pair->dma_chan[IN]); + if (pair->dma_chan[!dir]) + dma_release_channel(pair->dma_chan[!dir]); - if (pair->dma_chan[OUT]) - dma_release_channel(pair->dma_chan[OUT]); + if (pair->dma_chan[dir] && pair->req_dma_chan_dev_to_dev) + dma_release_channel(pair->dma_chan[dir]); - pair->dma_chan[IN] = NULL; - pair->dma_chan[OUT] = NULL; + pair->dma_chan[!dir] = NULL; + pair->dma_chan[dir] = NULL; return 0; } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b07eca2c6ccc..d4c73e86d058 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -310,7 +310,7 @@ struct snd_soc_component *snd_soc_rtdcom_lookup(struct snd_soc_pcm_runtime *rtd, } EXPORT_SYMBOL_GPL(snd_soc_rtdcom_lookup); -static struct snd_soc_component +struct snd_soc_component *snd_soc_lookup_component_nolocked(struct device *dev, const char *driver_name) { struct snd_soc_component *component; @@ -329,6 +329,7 @@ static struct snd_soc_component return found_component; } +EXPORT_SYMBOL_GPL(snd_soc_lookup_component_nolocked); struct snd_soc_component *snd_soc_lookup_component(struct device *dev, const char *driver_name) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index f728309a0833..80a4e71f2d95 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -21,18 +21,6 @@ */ #define SND_DMAENGINE_PCM_FLAG_NO_RESIDUE BIT(31) -struct dmaengine_pcm { - struct dma_chan *chan[SNDRV_PCM_STREAM_LAST + 1]; - const struct snd_dmaengine_pcm_config *config; - struct snd_soc_component component; - unsigned int flags; -}; - -static struct dmaengine_pcm *soc_component_to_pcm(struct snd_soc_component *p) -{ - return container_of(p, struct dmaengine_pcm, component); -} - static struct device *dmaengine_dma_dev(struct dmaengine_pcm *pcm, struct snd_pcm_substream *substream) { -- 2.21.0

2 1

[PATCH] soc/meson: add missing free_irq() in error path
by Pavel Machek 09 Jun '20

09 Jun '20

free_irq() is missing in case of error, fix that. Signed-off-by: Pavel Machek (CIP) <pavel(a)denx.de> diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c index 2e9b56b29d31..b2e867113226 100644 --- a/sound/soc/meson/axg-fifo.c +++ b/sound/soc/meson/axg-fifo.c @@ -249,7 +249,7 @@ int axg_fifo_pcm_open(struct snd_soc_component *component, /* Enable pclk to access registers and clock the fifo ip */ ret = clk_prepare_enable(fifo->pclk); if (ret) - return ret; + goto free_irq; /* Setup status2 so it reports the memory pointer */ regmap_update_bits(fifo->map, FIFO_CTRL1, @@ -269,8 +269,14 @@ int axg_fifo_pcm_open(struct snd_soc_component *component, /* Take memory arbitror out of reset */ ret = reset_control_deassert(fifo->arb); if (ret) - clk_disable_unprepare(fifo->pclk); + goto free_clk; + + return 0; +free_clk: + clk_disable_unprepare(fifo->pclk); +free_irq: + free_irq(fifo->irq, ss); return ret; } EXPORT_SYMBOL_GPL(axg_fifo_pcm_open); -- (english) http://www.livejournal.com/~pavelmachek (cesky, pictures) http://atrey.karlin.mff.cuni.cz/~pavel/picture/horses/blog.html

3 4

Re: [PATCH 4/5] soundwire: qcom: avoid dependency on CONFIG_SLIMBUS
by Srinivas Kandagatla 09 Jun '20

09 Jun '20

On 08/06/2020 21:43, Jonathan Marek wrote: > The driver may be used without slimbus, so don't depend on slimbus. > > Signed-off-by: Jonathan Marek <jonathan(a)marek.ca> > --- > drivers/soundwire/Kconfig | 1 - > drivers/soundwire/qcom.c | 5 +++++ > 2 files changed, 5 insertions(+), 1 deletion(-) > > diff --git a/drivers/soundwire/Kconfig b/drivers/soundwire/Kconfig > index fa2b4ab92ed9..d121cf739090 100644 > --- a/drivers/soundwire/Kconfig > +++ b/drivers/soundwire/Kconfig > @@ -33,7 +33,6 @@ config SOUNDWIRE_INTEL > > config SOUNDWIRE_QCOM > tristate "Qualcomm SoundWire Master driver" > - depends on SLIMBUS > depends on SND_SOC Why not move this to imply SLIMBUS this will give more flexibility! > help > SoundWire Qualcomm Master driver. > diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c > index 14334442615f..ac81c64768ea 100644 > --- a/drivers/soundwire/qcom.c > +++ b/drivers/soundwire/qcom.c > @@ -769,13 +769,18 @@ static int qcom_swrm_probe(struct platform_device *pdev) > if (!ctrl) > return -ENOMEM; > > +#ifdef CONFIG_SLIMBUS > if (dev->parent->bus == &slimbus_bus) { > +#else > + if (false) { > +#endif May be you can do bit more cleanup here, which could endup like: ctrl->regmap = dev_get_regmap(dev->parent, NULL); if (!ctrl->regmap) { res = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (res) { ctrl->mmio = devm_ioremap_resource(dev, res); if (IS_ERR(ctrl->mmio)) { dev_err(dev, "No valid mem resource found\n"); return PTR_ERR(ctrl->mmio); } ctrl->reg_read = qcom_swrm_cpu_reg_read; ctrl->reg_write = qcom_swrm_cpu_reg_write; } else { dev_err(dev, "No valid slim resource found\n"); return -EINVAL; } } else { ctrl->reg_read = qcom_swrm_ahb_reg_read; ctrl->reg_write = qcom_swrm_ahb_reg_write; } thanks, srini > ctrl->reg_read = qcom_swrm_ahb_reg_read; > ctrl->reg_write = qcom_swrm_ahb_reg_write; > ctrl->regmap = dev_get_regmap(dev->parent, NULL); > if (!ctrl->regmap) > return -EINVAL; > } else { > + > res = platform_get_resource(pdev, IORESOURCE_MEM, 0); > > ctrl->reg_read = qcom_swrm_cpu_reg_read; >

1 0

Re: [PATCH 0/5] soundwire: qcom: add mmio support
by Srinivas Kandagatla 09 Jun '20

09 Jun '20

Thanks Jonathan for the patches, On 08/06/2020 21:43, Jonathan Marek wrote: > This adds initial support for soundwire device on sm8250. > One thing off my list!! > Tested with the "wsa" sdw device, which is simpler than the others. WSA881x? did you test both enumeration and streaming? Are you planing to add any new WSA or WCD codec support for this SoC? thanks, srini > > Jonathan Marek (5): > soundwire: qcom: fix abh/ahb typo > soundwire: qcom: add support for mmio soundwire devices > soundwire: qcom: add v1.5.1 compatible > soundwire: qcom: avoid dependency on CONFIG_SLIMBUS > soundwire: qcom: enable CPU interrupts for mmio devices > > drivers/soundwire/Kconfig | 1 - > drivers/soundwire/qcom.c | 42 +++++++++++++++++++++++++++++++++++---- > 2 files changed, 38 insertions(+), 5 deletions(-) >

1 0

Re: [PATCH 2/5] soundwire: qcom: add support for mmio soundwire devices
by Srinivas Kandagatla 09 Jun '20

09 Jun '20

On 08/06/2020 21:43, Jonathan Marek wrote: > Adds support for qcom soundwire devices with memory mapped IO registers. > > Signed-off-by: Jonathan Marek <jonathan(a)marek.ca> > --- In general patch itself looks pretty trivial, but I would like to see what 1.5.1 controller provides in terms of error reporting of SoundWire slave register reads/writes. On WCD based controller we did not have a mechanism to report things like if the read is ignored or not. I was hoping that this version of controller would be able to report that. I will be nice to those patches if that is something which is supported in this version. --srini > drivers/soundwire/qcom.c | 25 +++++++++++++++++++++++-- > 1 file changed, 23 insertions(+), 2 deletions(-) > > diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c > index f38d1fd3679f..628747df1c75 100644 > --- a/drivers/soundwire/qcom.c > +++ b/drivers/soundwire/qcom.c > @@ -90,6 +90,7 @@ struct qcom_swrm_ctrl { > struct sdw_bus bus; > struct device *dev; > struct regmap *regmap; > + void __iomem *mmio; > struct completion *comp; > struct work_struct slave_work; > /* read/write lock */ > @@ -154,6 +155,20 @@ static int qcom_swrm_ahb_reg_write(struct qcom_swrm_ctrl *ctrl, > return SDW_CMD_OK; > } > > +static int qcom_swrm_cpu_reg_read(struct qcom_swrm_ctrl *ctrl, int reg, > + u32 *val) > +{ > + *val = readl(ctrl->mmio + reg); > + return SDW_CMD_OK; > +} > + > +static int qcom_swrm_cpu_reg_write(struct qcom_swrm_ctrl *ctrl, int reg, > + int val) > +{ > + writel(val, ctrl->mmio + reg); > + return SDW_CMD_OK; > +} > + > static int qcom_swrm_cmd_fifo_wr_cmd(struct qcom_swrm_ctrl *ctrl, u8 cmd_data, > u8 dev_addr, u16 reg_addr) > { > @@ -746,6 +761,7 @@ static int qcom_swrm_probe(struct platform_device *pdev) > struct sdw_master_prop *prop; > struct sdw_bus_params *params; > struct qcom_swrm_ctrl *ctrl; > + struct resource *res; > int ret; > u32 val; > > @@ -760,8 +776,13 @@ static int qcom_swrm_probe(struct platform_device *pdev) > if (!ctrl->regmap) > return -EINVAL; > } else { > - /* Only WCD based SoundWire controller is supported */ > - return -ENOTSUPP; > + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); > + > + ctrl->reg_read = qcom_swrm_cpu_reg_read; > + ctrl->reg_write = qcom_swrm_cpu_reg_write; > + ctrl->mmio = devm_ioremap_resource(dev, res); > + if (IS_ERR(ctrl->mmio)) > + return PTR_ERR(ctrl->mmio); > } > > ctrl->irq = of_irq_get(dev->of_node, 0); >

1 0

Re: [PATCH 2/5] soundwire: qcom: add support for mmio soundwire devices
by Vinod Koul 09 Jun '20

09 Jun '20

Hi Jonathan, On 08-06-20, 16:43, Jonathan Marek wrote: > Adds support for qcom soundwire devices with memory mapped IO registers. Please use 'SoundWire Master devices' instead :) > > Signed-off-by: Jonathan Marek <jonathan(a)marek.ca> > --- > drivers/soundwire/qcom.c | 25 +++++++++++++++++++++++-- > 1 file changed, 23 insertions(+), 2 deletions(-) > > diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c > index f38d1fd3679f..628747df1c75 100644 > --- a/drivers/soundwire/qcom.c > +++ b/drivers/soundwire/qcom.c > @@ -90,6 +90,7 @@ struct qcom_swrm_ctrl { > struct sdw_bus bus; > struct device *dev; > struct regmap *regmap; > + void __iomem *mmio; > struct completion *comp; > struct work_struct slave_work; > /* read/write lock */ > @@ -154,6 +155,20 @@ static int qcom_swrm_ahb_reg_write(struct qcom_swrm_ctrl *ctrl, > return SDW_CMD_OK; > } > > +static int qcom_swrm_cpu_reg_read(struct qcom_swrm_ctrl *ctrl, int reg, > + u32 *val) > +{ > + *val = readl(ctrl->mmio + reg); > + return SDW_CMD_OK; > +} > + > +static int qcom_swrm_cpu_reg_write(struct qcom_swrm_ctrl *ctrl, int reg, > + int val) > +{ > + writel(val, ctrl->mmio + reg); > + return SDW_CMD_OK; > +} this looks fine but regmap supports mmio also, so I am thinking we should remove these and set regmap (mmio/slim)... Srini..? -- ~Vinod

2 1

Re: [PATCH 1/5] soundwire: qcom: fix abh/ahb typo
by Srinivas Kandagatla 09 Jun '20

09 Jun '20

On 08/06/2020 21:43, Jonathan Marek wrote: > The function name qcom_swrm_abh_reg_read should say ahb, fix that. > > Signed-off-by: Jonathan Marek <jonathan(a)marek.ca> > --- Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> > drivers/soundwire/qcom.c | 4 ++-- > 1 file changed, 2 insertions(+), 2 deletions(-) > > diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c > index a1c2a44a3b4d..f38d1fd3679f 100644 > --- a/drivers/soundwire/qcom.c > +++ b/drivers/soundwire/qcom.c > @@ -114,7 +114,7 @@ struct qcom_swrm_ctrl { > > #define to_qcom_sdw(b) container_of(b, struct qcom_swrm_ctrl, bus) > > -static int qcom_swrm_abh_reg_read(struct qcom_swrm_ctrl *ctrl, int reg, > +static int qcom_swrm_ahb_reg_read(struct qcom_swrm_ctrl *ctrl, int reg, > u32 *val) > { > struct regmap *wcd_regmap = ctrl->regmap; > @@ -754,7 +754,7 @@ static int qcom_swrm_probe(struct platform_device *pdev) > return -ENOMEM; > > if (dev->parent->bus == &slimbus_bus) { > - ctrl->reg_read = qcom_swrm_abh_reg_read; > + ctrl->reg_read = qcom_swrm_ahb_reg_read; > ctrl->reg_write = qcom_swrm_ahb_reg_write; > ctrl->regmap = dev_get_regmap(dev->parent, NULL); > if (!ctrl->regmap) >

1 0

alsa-gobject: timer: add global method to retrieve current source of timestamp
by GitHub pull_request - opened 09 Jun '20

09 Jun '20

alsa-project/alsa-gobject pull request #27 was opened from takaswie: ALSA snd_timer kernel module has a parameter for source of timestamping. Applications cannot refer to the parameter via UAPI of Linux kernel, however it's possible via sysfs. This patch adds an API to refer to the parameter. Request URL : https://github.com/alsa-project/alsa-gobject/pull/27 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/27.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH 1/3] bytcr-rt5640: Fix DMIC1 not working when connected over AIF2
by Hans de Goede 09 Jun '20

09 Jun '20

The RT5640 codec has 2 ADC-s which are caled the Stereo ADC and the Mono ADC (even though the mono ADC seems to be stereo too?). The Stereo ADC is connected to the AIF1 interface of the codec and the Mono ADC to the AIF2 interface. Almost all X86 BYT/CHT design use the AIF1 interface, but a few models use the AIF2 interface. For this reason the IN?-InternalMic.conf bits not only select/enable the microphone input on the Stereo ADC, but also on the Mono ADC, so that the analog mic. inputs also work in devices using the AIF2 interface. Before this commit DigitalMics.conf was only setting up the DMIC input on the Stereo ADC and not on the Mono ADC, causing DMIC1 to not work when connected over AIF2. This commit adds selecting / enabling DMIC1 on the Mono ADC. This fixes the internal microphone not working on the Toshiba Encore WT10A. Note that for devices which use the AIF1 interface this is a no-op since the AIF1 interface is not connected to anything, there is no complete chain so DAPM will not enable any of the elements in the chain. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- ucm2/codecs/rt5640/DigitalMics.conf | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/ucm2/codecs/rt5640/DigitalMics.conf b/ucm2/codecs/rt5640/DigitalMics.conf index 503c4a9..fdd412f 100644 --- a/ucm2/codecs/rt5640/DigitalMics.conf +++ b/ucm2/codecs/rt5640/DigitalMics.conf @@ -6,6 +6,9 @@ SectionDevice."Mic" { ] EnableSequence [ + cset "name='Mono ADC MIXL ADC2 Switch' on" + cset "name='Mono ADC MIXR ADC2 Switch' on" + cset "name='Mono ADC Capture Switch' on" cset "name='Stereo ADC MIXL ADC2 Switch' on" cset "name='Stereo ADC MIXR ADC2 Switch' on" cset "name='Internal Mic Switch' on" @@ -13,6 +16,9 @@ SectionDevice."Mic" { ] DisableSequence [ + cset "name='Mono ADC MIXL ADC2 Switch' off" + cset "name='Mono ADC MIXR ADC2 Switch' off" + cset "name='Mono ADC Capture Switch' off" cset "name='Stereo ADC MIXL ADC2 Switch' off" cset "name='Stereo ADC MIXR ADC2 Switch' off" cset "name='Internal Mic Switch' off" -- 2.26.2

3 6

[PATCH alsa-lib] conf: USB-Audio: Add HP Thunderbolt Dock in the IEC958 blacklist
by Kai-Heng Feng 09 Jun '20

09 Jun '20

Both headset and speaker on HP Thunderbolt Dock don't support IEC958 interface. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- src/conf/cards/USB-Audio.conf | 2 ++ 1 file changed, 2 insertions(+) diff --git a/src/conf/cards/USB-Audio.conf b/src/conf/cards/USB-Audio.conf index 735079e3..01664abf 100644 --- a/src/conf/cards/USB-Audio.conf +++ b/src/conf/cards/USB-Audio.conf @@ -62,6 +62,8 @@ USB-Audio.pcm.iec958_device { "Scarlett 2i4 USB" 999 "Sennheiser USB headset" 999 "SWTOR Gaming Headset by Razer" 999 + "Thunderbolt Dock Audio Headset" 999 + "Thunderbolt Dock Audio Module" 999 "USB Device 0x46d_0x821" 999 "USB Device 0x46d_0x992" 999 "WD15 Dock" 999 -- 2.17.1

2 1

Renoir APU - Mic-mute Led status not updated for onboard DMIC
by Mukunda, Vijendar 09 Jun '20

09 Jun '20

Renoir APU has on board DMIC and HD Audio Endpoint combination. Mic-mute LED status not updated for On board DMIC i.e internal Mic when Mute is applied from OS in Ubuntu. When External Mic is connected(HD Audio Endpoint) , ACPI event got updated based on Mic mute status , able to toggle the Mic Led status in Ubuntu. We have found a thread which explains toggling Mic Led status for Lenovo thinkpads. Below is the link. https://askubuntu.com/questions/125367/enabling-mic-mute-button-and-light-o… In case of Internal Mic(DMIC), Mic-mute Led status is not updated as onboard DMIC doesn't have any volume control. We are not really sure how to update Mic mute status in this case. Help us in resolving this issue. Alsa info link: https://pastebin.ubuntu.com/p/4kprhDZYbg/ Thanks, Vijendar

1 0

Re: [PATCH 3/5] soundwire: qcom: add v1.5.1 compatible
by Vinod Koul 09 Jun '20

09 Jun '20

On 08-06-20, 16:43, Jonathan Marek wrote: > Add a compatible string for HW version v1.5.1 on sm8250 SoCs. Please document this new compatible -- ~Vinod

1 0

Update configs for Broadwell devices
by GitHub pull_request - opened 09 Jun '20

09 Jun '20

alsa-project/alsa-ucm-conf pull request #32 was opened from plbossart: use the SOF card test to reuse legacy configs. tests require https://github.com/alsa-project/alsa-tests/pull/9 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/32 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/32.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v4 0/3] ASoC: Intel: Add KeemBay ASoC platform driver
by Sia Jee Heng 09 Jun '20

09 Jun '20

The below series of patches support the KeemBay ASoC platform driver. The platform driver initialize the i2s to capture and playback the pcm data on the ARM. The i2s is running in polling mode. There is no DSP in the KeemBay SoC. Users are rely on the Gstreamer plugin to perform Audio preprocessing. Audio graph card is used to connect the platform driver with the tlv320aic3204 codec. Change History: v4: - Reduce if-otology at the tx/rx function. - Fix indentation. - specify .rate directly v3: - Adjusted header format. - Use Audio graph card instead of custom sound card. - Use if-else instead of conditional operator. - Enabled .set_fmt to configure master clock. v2: - Corrected I2S naming for DT binding. v1: - Initial version. Sia Jee Heng (3): ASoC: Intel: Add KeemBay platform driver ASoC: Intel: Add makefiles and kconfig changes for KeemBay dt-bindings: sound: Add documentation for KeemBay i2s .../bindings/sound/intel,keembay-i2s.yaml | 68 +++ sound/soc/intel/Kconfig | 7 + sound/soc/intel/Makefile | 1 + sound/soc/intel/keembay/Makefile | 4 + sound/soc/intel/keembay/kmb_platform.c | 654 +++++++++++++++++++++ sound/soc/intel/keembay/kmb_platform.h | 145 +++++ 6 files changed, 879 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/intel,keembay-i2s.yaml create mode 100644 sound/soc/intel/keembay/Makefile create mode 100644 sound/soc/intel/keembay/kmb_platform.c create mode 100644 sound/soc/intel/keembay/kmb_platform.h -- 1.9.1

3 5

alsa-gobject: doc: hwdep: fix missing content of introduction element
by GitHub pull_request - opened 09 Jun '20

09 Jun '20

alsa-project/alsa-gobject pull request #26 was opened from takaswie: A commit eb03ec0340b ("hwdep: fulfill documentation for ALSAHwdep") fulfilled documentation for ALSAHwDep but the content of introduction element is missing. This patch fixes it. Request URL : https://github.com/alsa-project/alsa-gobject/pull/26 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/26.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH AUTOSEL 5.6 324/606] ALSA: hda/realtek - Add new codec supported for ALC287
by Sasha Levin 09 Jun '20

09 Jun '20

From: Kailang Yang <kailang(a)realtek.com> [ Upstream commit 630e36126e420e1756378b3427b42711ce0b9ddd ] Enable new codec supported for ALC287. Signed-off-by: Kailang Yang <kailang(a)realtek.com> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/dcf5ce5507104d0589a917cbb71dc3c6@realtek.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 92c6e58c3862..e62d58872b6e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -384,6 +384,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0282: case 0x10ec0283: case 0x10ec0286: + case 0x10ec0287: case 0x10ec0288: case 0x10ec0285: case 0x10ec0298: @@ -8292,6 +8293,7 @@ static int patch_alc269(struct hda_codec *codec) case 0x10ec0215: case 0x10ec0245: case 0x10ec0285: + case 0x10ec0287: case 0x10ec0289: spec->codec_variant = ALC269_TYPE_ALC215; spec->shutup = alc225_shutup; @@ -9570,6 +9572,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0284, "ALC284", patch_alc269), HDA_CODEC_ENTRY(0x10ec0285, "ALC285", patch_alc269), HDA_CODEC_ENTRY(0x10ec0286, "ALC286", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0287, "ALC287", patch_alc269), HDA_CODEC_ENTRY(0x10ec0288, "ALC288", patch_alc269), HDA_CODEC_ENTRY(0x10ec0289, "ALC289", patch_alc269), HDA_CODEC_ENTRY(0x10ec0290, "ALC290", patch_alc269), -- 2.25.1

1 0

[PATCH AUTOSEL 5.6 323/606] ALSA: usb-audio: Quirks for Gigabyte TRX40 Aorus Master onboard audio
by Sasha Levin 09 Jun '20

09 Jun '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 7f5ad9c9003425175f46c94df380e8c9e558cfb5 ] Gigabyte TRX40 Aorus Master is equipped with two USB-audio devices, a Realtek ALC1220-VB codec (USB ID 0414:a001) and an ESS SABRE9218 DAC (USB ID 0414:a000). The latter serves solely for the headphone output on the front panel while the former serves for the rest I/Os (mostly for the I/Os in the rear panel but also including the front mic). Both chips do work more or less with the unmodified USB-audio driver, but there are a few glitches. The ALC1220-VB returns an error for an inquiry to some jacks, as already seen on other TRX40-based mobos. However this machine has a slightly incompatible configuration, hence the existing mapping cannot be used as is. Meanwhile the ESS chip seems working without any quirk. But since both audio devices don't provide any specific names, both cards appear as "USB-Audio", and it's quite confusing for users. This patch is an attempt to overcome those issues: - The specific mapping table for ALC1220-VB is provided, reducing the non-working nodes and renaming the badly chosen controls. The connector map isn't needed here unlike other TRX40 quirks. - For both USB IDs (0414:a000 and 0414:a001), provide specific card name strings, so that user-space can identify more easily; and more importantly, UCM profile can be applied to each. Reported-by: Linus Torvalds <torvalds(a)linux-foundation.org> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200526082810.29506-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/mixer_maps.c | 19 +++++++++++++++++++ sound/usb/quirks-table.h | 25 +++++++++++++++++++++++++ 2 files changed, 44 insertions(+) diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index bfdc6ad52785..9af7aa93f6fa 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -397,6 +397,21 @@ static const struct usbmix_connector_map trx40_mobo_connector_map[] = { {} }; +/* Rear panel + front mic on Gigabyte TRX40 Aorus Master with ALC1220-VB */ +static const struct usbmix_name_map aorus_master_alc1220vb_map[] = { + { 17, NULL }, /* OT, IEC958?, disabled */ + { 19, NULL, 12 }, /* FU, Input Gain Pad - broken response, disabled */ + { 16, "Line Out" }, /* OT */ + { 22, "Line Out Playback" }, /* FU */ + { 7, "Line" }, /* IT */ + { 19, "Line Capture" }, /* FU */ + { 8, "Mic" }, /* IT */ + { 20, "Mic Capture" }, /* FU */ + { 9, "Front Mic" }, /* IT */ + { 21, "Front Mic Capture" }, /* FU */ + {} +}; + /* * Control map entries */ @@ -526,6 +541,10 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = { .id = USB_ID(0x1b1c, 0x0a42), .map = corsair_virtuoso_map, }, + { /* Gigabyte TRX40 Aorus Master (rear panel + front mic) */ + .id = USB_ID(0x0414, 0xa001), + .map = aorus_master_alc1220vb_map, + }, { /* Gigabyte TRX40 Aorus Pro WiFi */ .id = USB_ID(0x0414, 0xa002), .map = trx40_mobo_map, diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index aa4c16ce0e57..bbae11605a4c 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3650,4 +3650,29 @@ ALC1220_VB_DESKTOP(0x0db0, 0x543d), /* MSI TRX40 */ ALC1220_VB_DESKTOP(0x26ce, 0x0a01), /* Asrock TRX40 Creator */ #undef ALC1220_VB_DESKTOP +/* Two entries for Gigabyte TRX40 Aorus Master: + * TRX40 Aorus Master has two USB-audio devices, one for the front headphone + * with ESS SABRE9218 DAC chip, while another for the rest I/O (the rear + * panel and the front mic) with Realtek ALC1220-VB. + * Here we provide two distinct names for making UCM profiles easier. + */ +{ + USB_DEVICE(0x0414, 0xa000), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Gigabyte", + .product_name = "Aorus Master Front Headphone", + .profile_name = "Gigabyte-Aorus-Master-Front-Headphone", + .ifnum = QUIRK_NO_INTERFACE + } +}, +{ + USB_DEVICE(0x0414, 0xa001), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Gigabyte", + .product_name = "Aorus Master Main Audio", + .profile_name = "Gigabyte-Aorus-Master-Main-Audio", + .ifnum = QUIRK_NO_INTERFACE + } +}, + #undef USB_DEVICE_VENDOR_SPEC -- 2.25.1

1 0

[PATCH AUTOSEL 5.6 320/606] ALSA: usb-audio: mixer: volume quirk for ESS Technology Asus USB DAC
by Sasha Levin 09 Jun '20

09 Jun '20

From: Chris Chiu <chiu(a)endlessm.com> [ Upstream commit 4020d1ccbe55bdf67b31d718d2400506eaf4b43f ] The Asus USB DAC is a USB type-C audio dongle for connecting to the headset and headphone. The volume minimum value -23040 which is 0xa600 in hexadecimal with the resolution value 1 indicates this should be endianness issue caused by the firmware bug. Add a volume quirk to fix the volume control problem. Also fixes this warning: Warning! Unlikely big volume range (=23040), cval->res is probably wrong. [5] FU [Headset Capture Volume] ch = 1, val = -23040/0/1 Warning! Unlikely big volume range (=23040), cval->res is probably wrong. [7] FU [Headset Playback Volume] ch = 1, val = -23040/0/1 Signed-off-by: Chris Chiu <chiu(a)endlessm.com> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200526062613.55401-1-chiu@endlessm.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/mixer.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 7a2961ad60de..68fefe55e5c0 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1171,6 +1171,14 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, cval->res = 384; } break; + case USB_ID(0x0495, 0x3042): /* ESS Technology Asus USB DAC */ + if ((strstr(kctl->id.name, "Playback Volume") != NULL) || + strstr(kctl->id.name, "Capture Volume") != NULL) { + cval->min >>= 8; + cval->max = 0; + cval->res = 1; + } + break; } } -- 2.25.1

1 0

[PATCH AUTOSEL 5.6 319/606] ALSA: hda/realtek - Add a model for Thinkpad T570 without DAC workaround
by Sasha Levin 09 Jun '20

09 Jun '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 399c01aa49e548c82d40f8161915a5941dd3c60e ] We fixed the regression of the speaker volume for some Thinkpad models (e.g. T570) by the commit 54947cd64c1b ("ALSA: hda/realtek - Fix speaker output regression on Thinkpad T570"). Essentially it fixes the DAC / pin pairing by a static table. It was confirmed and merged to stable kernel later. Now, interestingly, we got another regression report for the very same model (T570) about the similar problem, and the commit above was the culprit. That is, by some reason, there are devices that prefer the DAC1, and another device DAC2! Unfortunately those have the same ID and we have no idea what can differentiate, in this patch, a new fixup model "tpt470-dock-fix" is provided, so that users with such a machine can apply it manually. When model=tpt470-dock-fix option is passed to snd-hda-intel module, it avoids the fixed DAC pairing and the DAC1 is assigned to the speaker like the earlier versions. Fixes: 54947cd64c1b ("ALSA: hda/realtek - Fix speaker output regression on Thinkpad T570") BugLink: https://apibugzilla.suse.com/show_bug.cgi?id=1172017 Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200526062406.9799-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 36 +++++++++++++++++++++++++---------- 1 file changed, 26 insertions(+), 10 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 041d2a32059b..92c6e58c3862 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5484,18 +5484,9 @@ static void alc_fixup_tpt470_dock(struct hda_codec *codec, { 0x19, 0x21a11010 }, /* dock mic */ { } }; - /* Assure the speaker pin to be coupled with DAC NID 0x03; otherwise - * the speaker output becomes too low by some reason on Thinkpads with - * ALC298 codec - */ - static const hda_nid_t preferred_pairs[] = { - 0x14, 0x03, 0x17, 0x02, 0x21, 0x02, - 0 - }; struct alc_spec *spec = codec->spec; if (action == HDA_FIXUP_ACT_PRE_PROBE) { - spec->gen.preferred_dacs = preferred_pairs; spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP; snd_hda_apply_pincfgs(codec, pincfgs); } else if (action == HDA_FIXUP_ACT_INIT) { @@ -5508,6 +5499,23 @@ static void alc_fixup_tpt470_dock(struct hda_codec *codec, } } +static void alc_fixup_tpt470_dacs(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + /* Assure the speaker pin to be coupled with DAC NID 0x03; otherwise + * the speaker output becomes too low by some reason on Thinkpads with + * ALC298 codec + */ + static const hda_nid_t preferred_pairs[] = { + 0x14, 0x03, 0x17, 0x02, 0x21, 0x02, + 0 + }; + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) + spec->gen.preferred_dacs = preferred_pairs; +} + static void alc_shutup_dell_xps13(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -6063,6 +6071,7 @@ enum { ALC700_FIXUP_INTEL_REFERENCE, ALC274_FIXUP_DELL_BIND_DACS, ALC274_FIXUP_DELL_AIO_LINEOUT_VERB, + ALC298_FIXUP_TPT470_DOCK_FIX, ALC298_FIXUP_TPT470_DOCK, ALC255_FIXUP_DUMMY_LINEOUT_VERB, ALC255_FIXUP_DELL_HEADSET_MIC, @@ -6994,12 +7003,18 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC274_FIXUP_DELL_BIND_DACS }, - [ALC298_FIXUP_TPT470_DOCK] = { + [ALC298_FIXUP_TPT470_DOCK_FIX] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_tpt470_dock, .chained = true, .chain_id = ALC293_FIXUP_LENOVO_SPK_NOISE }, + [ALC298_FIXUP_TPT470_DOCK] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_tpt470_dacs, + .chained = true, + .chain_id = ALC298_FIXUP_TPT470_DOCK_FIX + }, [ALC255_FIXUP_DUMMY_LINEOUT_VERB] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -7638,6 +7653,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC292_FIXUP_TPT440_DOCK, .name = "tpt440-dock"}, {.id = ALC292_FIXUP_TPT440, .name = "tpt440"}, {.id = ALC292_FIXUP_TPT460, .name = "tpt460"}, + {.id = ALC298_FIXUP_TPT470_DOCK_FIX, .name = "tpt470-dock-fix"}, {.id = ALC298_FIXUP_TPT470_DOCK, .name = "tpt470-dock"}, {.id = ALC233_FIXUP_LENOVO_MULTI_CODECS, .name = "dual-codecs"}, {.id = ALC700_FIXUP_INTEL_REFERENCE, .name = "alc700-ref"}, -- 2.25.1

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[PATCH AUTOSEL 5.6 318/606] ALSA: hwdep: fix a left shifting 1 by 31 UB bug
by Sasha Levin 09 Jun '20

09 Jun '20

From: Changming Liu <liu.changm(a)northeastern.edu> [ Upstream commit fb8cd6481ffd126f35e9e146a0dcf0c4e8899f2e ] The "info.index" variable can be 31 in "1 << info.index". This might trigger an undefined behavior since 1 is signed. Fix this by casting 1 to 1u just to be sure "1u << 31" is defined. Signed-off-by: Changming Liu <liu.changm(a)northeastern.edu> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/BL0PR06MB4548170B842CB055C9AF695DE5B00@BL0PR06MB4… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/hwdep.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/hwdep.c b/sound/core/hwdep.c index b412d3b3d5ff..21edb8ac95eb 100644 --- a/sound/core/hwdep.c +++ b/sound/core/hwdep.c @@ -216,12 +216,12 @@ static int snd_hwdep_dsp_load(struct snd_hwdep *hw, if (info.index >= 32) return -EINVAL; /* check whether the dsp was already loaded */ - if (hw->dsp_loaded & (1 << info.index)) + if (hw->dsp_loaded & (1u << info.index)) return -EBUSY; err = hw->ops.dsp_load(hw, &info); if (err < 0) return err; - hw->dsp_loaded |= (1 << info.index); + hw->dsp_loaded |= (1u << info.index); return 0; } -- 2.25.1

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[PATCH AUTOSEL 5.6 258/606] ALSA: usb-audio: add mapping for ASRock TRX40 Creator
by Sasha Levin 09 Jun '20

09 Jun '20

From: Andrew Oakley <andrew(a)adoakley.name> [ Upstream commit da7a8f1a8fc3e14c6dcc52b4098bddb8f20390be ] This is another TRX40 based motherboard with ALC1220-VB USB-audio that requires a static mapping table. This motherboard also has a PCI device which advertises no codecs. The PCI ID is 1022:1487 and PCI SSID is 1022:d102. As this is using the AMD vendor ID, don't blacklist for now in case other boards have a working audio device with the same ssid. alsa-info.sh report for this board: http://alsa-project.org/db/?f=0a742f89066527497b77ce16bca486daccf8a70c Signed-off-by: Andrew Oakley <andrew(a)adoakley.name> Link: https://lore.kernel.org/r/20200503141639.35519-1-andrew@adoakley.name Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/mixer_maps.c | 5 +++++ sound/usb/quirks-table.h | 1 + 2 files changed, 6 insertions(+) diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index 0260c750e156..bfdc6ad52785 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -549,6 +549,11 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = { .map = trx40_mobo_map, .connector_map = trx40_mobo_connector_map, }, + { /* Asrock TRX40 Creator */ + .id = USB_ID(0x26ce, 0x0a01), + .map = trx40_mobo_map, + .connector_map = trx40_mobo_connector_map, + }, { 0 } /* terminator */ }; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 8c2f5c23e1b4..aa4c16ce0e57 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3647,6 +3647,7 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), ALC1220_VB_DESKTOP(0x0414, 0xa002), /* Gigabyte TRX40 Aorus Pro WiFi */ ALC1220_VB_DESKTOP(0x0db0, 0x0d64), /* MSI TRX40 Creator */ ALC1220_VB_DESKTOP(0x0db0, 0x543d), /* MSI TRX40 */ +ALC1220_VB_DESKTOP(0x26ce, 0x0a01), /* Asrock TRX40 Creator */ #undef ALC1220_VB_DESKTOP #undef USB_DEVICE_VENDOR_SPEC -- 2.25.1

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[PATCH AUTOSEL 5.6 146/606] ALSA: hda/realtek - Add more fixup entries for Clevo machines
by Sasha Levin 09 Jun '20

09 Jun '20

From: PeiSen Hou <pshou(a)realtek.com> commit 259eb82475316672a5d682a94dc8bdd53cf8d8c3 upstream. A few known Clevo machines (PC50, PC70, X170) with ALC1220 codec need the existing quirk for pins for PB51 and co. Signed-off-by: PeiSen Hou <pshou(a)realtek.com> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200519065012.13119-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> --- sound/pci/hda/patch_realtek.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 23315b69ac38..041d2a32059b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2473,6 +2473,9 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x97e1, "Clevo P970[ER][CDFN]", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1558, 0x65d1, "Clevo PB51[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x67d1, "Clevo PB71[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), + SND_PCI_QUIRK(0x1558, 0x50d3, "Clevo PC50[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), + SND_PCI_QUIRK(0x1558, 0x70d1, "Clevo PC70[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), + SND_PCI_QUIRK(0x1558, 0x7714, "Clevo X170", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_FIXUP_EAPD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Y530", ALC882_FIXUP_LENOVO_Y530), -- 2.25.1

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[PATCH AUTOSEL 5.6 145/606] ALSA: hda/realtek - Fix silent output on Gigabyte X570 Aorus Xtreme
by Sasha Levin 09 Jun '20

09 Jun '20

From: Christian Lachner <gladiac(a)gmail.com> commit d9e8fe0cffbfdd18de96fa68ee2a8b667a0b046e upstream. The Gigabyte X570 Aorus Xtreme motherboard with ALC1220 codec requires a similar workaround for Clevo laptops to enforce the DAC/mixer connection path. Set up a quirk entry for that. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=205275 Signed-off-by: Christian Lachner <gladiac(a)gmail.com> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200518053844.42743-2-gladiac@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> --- 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 dc2302171a71..23315b69ac38 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2457,6 +2457,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3/Z87X-UD3H", ALC889_FIXUP_FRONT_HP_NO_PRESENCE), 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, 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), -- 2.25.1

1 0

[PATCH AUTOSEL 5.6 144/606] ALSA: pcm: fix incorrect hw_base increase
by Sasha Levin 09 Jun '20

09 Jun '20

From: Brent Lu <brent.lu(a)intel.com> commit e7513c5786f8b33f0c107b3759e433bc6cbb2efa upstream. There is a corner case that ALSA keeps increasing the hw_ptr but DMA already stop working/updating the position for a long time. In following log we can see the position returned from DMA driver does not move at all but the hw_ptr got increased at some point of time so snd_pcm_avail() will return a large number which seems to be a buffer underrun event from user space program point of view. The program thinks there is space in the buffer and fill more data. [ 418.510086] sound pcmC0D5p: pos 96 hw_ptr 96 appl_ptr 4096 avail 12368 [ 418.510149] sound pcmC0D5p: pos 96 hw_ptr 96 appl_ptr 6910 avail 9554 ... [ 418.681052] sound pcmC0D5p: pos 96 hw_ptr 96 appl_ptr 15102 avail 1362 [ 418.681130] sound pcmC0D5p: pos 96 hw_ptr 96 appl_ptr 16464 avail 0 [ 418.726515] sound pcmC0D5p: pos 96 hw_ptr 16464 appl_ptr 16464 avail 16368 This is because the hw_base will be increased by runtime->buffer_size frames unconditionally if the hw_ptr is not updated for over half of buffer time. As the hw_base increases, so does the hw_ptr increased by the same number. The avail value returned from snd_pcm_avail() could exceed the limit (buffer_size) easily becase the hw_ptr itself got increased by same buffer_size samples when the corner case happens. In following log, the buffer_size is 16368 samples but the avail is 21810 samples so CRAS server complains about it. [ 418.851755] sound pcmC0D5p: pos 96 hw_ptr 16464 appl_ptr 27390 avail 5442 [ 418.926491] sound pcmC0D5p: pos 96 hw_ptr 32832 appl_ptr 27390 avail 21810 cras_server[1907]: pcm_avail returned frames larger than buf_size: sof-glkda7219max: :0,5: 21810 > 16368 By updating runtime->hw_ptr_jiffies each time the HWSYNC is called, the hw_base will keep the same when buffer stall happens at long as the interval between each HWSYNC call is shorter than half of buffer time. Following is a log captured by a patched kernel. The hw_base/hw_ptr value is fixed in this corner case and user space program should be aware of the buffer stall and handle it. [ 293.525543] sound pcmC0D5p: pos 96 hw_ptr 96 appl_ptr 4096 avail 12368 [ 293.525606] sound pcmC0D5p: pos 96 hw_ptr 96 appl_ptr 6880 avail 9584 [ 293.525975] sound pcmC0D5p: pos 96 hw_ptr 96 appl_ptr 10976 avail 5488 [ 293.611178] sound pcmC0D5p: pos 96 hw_ptr 96 appl_ptr 15072 avail 1392 [ 293.696429] sound pcmC0D5p: pos 96 hw_ptr 96 appl_ptr 16464 avail 0 ... [ 381.139517] sound pcmC0D5p: pos 96 hw_ptr 96 appl_ptr 16464 avail 0 Signed-off-by: Brent Lu <brent.lu(a)intel.com> Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/1589776238-23877-1-git-send-email-brent.lu@intel.… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> --- sound/core/pcm_lib.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 872a852de75c..d531e1bc2b81 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -433,6 +433,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, no_delta_check: if (runtime->status->hw_ptr == new_hw_ptr) { + runtime->hw_ptr_jiffies = curr_jiffies; update_audio_tstamp(substream, &curr_tstamp, &audio_tstamp); return 0; } -- 2.25.1

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[PATCH AUTOSEL 5.6 143/606] ALSA: iec1712: Initialize STDSP24 properly when using the model=staudio option
by Sasha Levin 09 Jun '20

09 Jun '20

From: Scott Bahling <sbahling(a)suse.com> commit b0cb099062b0c18246c3a20caaab4c0afc303255 upstream. The ST Audio ADCIII is an STDSP24 card plus extension box. With commit e8a91ae18bdc ("ALSA: ice1712: Add support for STAudio ADCIII") we enabled the ADCIII ports using the model=staudio option but forgot this part to ensure the STDSP24 card is initialized properly. Fixes: e8a91ae18bdc ("ALSA: ice1712: Add support for STAudio ADCIII") Signed-off-by: Scott Bahling <sbahling(a)suse.com> Cc: <stable(a)vger.kernel.org> BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1048934 Link: https://lore.kernel.org/r/20200518175728.28766-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> --- sound/pci/ice1712/ice1712.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 884d0cdec08c..73e1e5400506 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -2332,7 +2332,8 @@ static int snd_ice1712_chip_init(struct snd_ice1712 *ice) pci_write_config_byte(ice->pci, 0x61, ice->eeprom.data[ICE_EEP1_ACLINK]); pci_write_config_byte(ice->pci, 0x62, ice->eeprom.data[ICE_EEP1_I2SID]); pci_write_config_byte(ice->pci, 0x63, ice->eeprom.data[ICE_EEP1_SPDIF]); - if (ice->eeprom.subvendor != ICE1712_SUBDEVICE_STDSP24) { + if (ice->eeprom.subvendor != ICE1712_SUBDEVICE_STDSP24 && + ice->eeprom.subvendor != ICE1712_SUBDEVICE_STAUDIO_ADCIII) { ice->gpio.write_mask = ice->eeprom.gpiomask; ice->gpio.direction = ice->eeprom.gpiodir; snd_ice1712_write(ice, ICE1712_IREG_GPIO_WRITE_MASK, -- 2.25.1

1 0

[PATCH AUTOSEL 5.6 140/606] ALSA: hda/realtek: Enable headset mic of ASUS UX581LV with ALC295
by Sasha Levin 09 Jun '20

09 Jun '20

From: Jian-Hong Pan <jian-hong(a)endlessm.com> [ Upstream commit 7900e81797613b92f855f9921392a7430cbdf88c ] The ASUS UX581LV laptop's audio (1043:19e1) with ALC295 can't detect the headset microphone until ALC295_FIXUP_ASUS_MIC_NO_PRESENCE quirk applied. Signed-off-by: Jian-Hong Pan <jian-hong(a)endlessm.com> Link: https://lore.kernel.org/r/20200512061525.133985-3-jian-hong@endlessm.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- 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 ece762d0c714..dc2302171a71 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7436,6 +7436,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x18f1, "Asus FX505DT", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x19ce, "ASUS B9450FA", ALC294_FIXUP_ASUS_HPE), + SND_PCI_QUIRK(0x1043, 0x19e1, "ASUS UX581LV", ALC295_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), SND_PCI_QUIRK(0x1043, 0x1a30, "ASUS X705UD", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1b11, "ASUS UX431DA", ALC294_FIXUP_ASUS_COEF_1B), -- 2.25.1

1 0

[PATCH AUTOSEL 5.6 139/606] ALSA: hda/realtek - Enable headset mic of ASUS UX550GE with ALC295
by Sasha Levin 09 Jun '20

09 Jun '20

From: Jian-Hong Pan <jian-hong(a)endlessm.com> [ Upstream commit ad97d667854c2fbce05a004e107f358ef4b04cf6 ] The ASUS laptop UX550GE with ALC295 can't detect the headset microphone until ALC295_FIXUP_ASUS_MIC_NO_PRESENCE quirk applied. Signed-off-by: Jian-Hong Pan <jian-hong(a)endlessm.com> Signed-off-by: Daniel Drake <drake(a)endlessm.com> Link: https://lore.kernel.org/r/20200512061525.133985-2-jian-hong@endlessm.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f92cd420e98d..ece762d0c714 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8050,6 +8050,10 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x12, 0x90a60130}, {0x17, 0x90170110}, {0x21, 0x03211020}), + SND_HDA_PIN_QUIRK(0x10ec0295, 0x1043, "ASUS", ALC295_FIXUP_ASUS_MIC_NO_PRESENCE, + {0x12, 0x90a60120}, + {0x17, 0x90170110}, + {0x21, 0x04211030}), SND_HDA_PIN_QUIRK(0x10ec0295, 0x1043, "ASUS", ALC295_FIXUP_ASUS_MIC_NO_PRESENCE, {0x12, 0x90a60130}, {0x17, 0x90170110}, -- 2.25.1

1 0

[PATCH AUTOSEL 5.6 138/606] ALSA: hda/realtek - Enable headset mic of ASUS GL503VM with ALC295
by Sasha Levin 09 Jun '20

09 Jun '20

From: Chris Chiu <chiu(a)endlessm.com> [ Upstream commit 9e43342b464f1de570a3ad8256ac77645749ef45 ] The ASUS laptop GL503VM with ALC295 can't detect the headset microphone. The headset microphone does not work until pin 0x19 is enabled for it. Signed-off-by: Chris Chiu <chiu(a)endlessm.com> Signed-off-by: Daniel Drake <drake(a)endlessm.com> Signed-off-by: Jian-Hong Pan <jian-hong(a)endlessm.com> Link: https://lore.kernel.org/r/20200512061525.133985-1-jian-hong@endlessm.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b377aca71cbf..f92cd420e98d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6096,6 +6096,7 @@ enum { ALC285_FIXUP_HP_MUTE_LED, ALC236_FIXUP_HP_MUTE_LED, ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, + ALC295_FIXUP_ASUS_MIC_NO_PRESENCE, }; static const struct hda_fixup alc269_fixups[] = { @@ -7259,6 +7260,15 @@ static const struct hda_fixup alc269_fixups[] = { { } }, }, + [ALC295_FIXUP_ASUS_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -8040,6 +8050,14 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x12, 0x90a60130}, {0x17, 0x90170110}, {0x21, 0x03211020}), + SND_HDA_PIN_QUIRK(0x10ec0295, 0x1043, "ASUS", ALC295_FIXUP_ASUS_MIC_NO_PRESENCE, + {0x12, 0x90a60130}, + {0x17, 0x90170110}, + {0x21, 0x03211020}), + SND_HDA_PIN_QUIRK(0x10ec0295, 0x1043, "ASUS", ALC295_FIXUP_ASUS_MIC_NO_PRESENCE, + {0x12, 0x90a60130}, + {0x17, 0x90170110}, + {0x21, 0x03211020}), SND_HDA_PIN_QUIRK(0x10ec0295, 0x1028, "Dell", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE, {0x14, 0x90170110}, {0x21, 0x04211020}), -- 2.25.1

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[PATCH AUTOSEL 5.6 137/606] ALSA: hda/realtek: Add quirk for Samsung Notebook
by Sasha Levin 09 Jun '20

09 Jun '20

From: Mike Pozulp <pozulp.kernel(a)gmail.com> [ Upstream commit 14425f1f521fdfe274a7bb390637c786432e08b4 ] Some models of the Samsung Notebook 9 have very quiet and distorted headphone output. This quirk changes the VREF value of the ALC298 codec NID 0x1a from default HIZ to new 100. [ adjusted to 5.7-base and rearranged in SSID order -- tiwai ] Signed-off-by: Mike Pozulp <pozulp.kernel(a)gmail.com> BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207423 Link: https://lore.kernel.org/r/20200510032838.1989130-1-pozulp.kernel@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 368ed3678fc2..b377aca71cbf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6095,6 +6095,7 @@ enum { ALC285_FIXUP_HP_GPIO_LED, ALC285_FIXUP_HP_MUTE_LED, ALC236_FIXUP_HP_MUTE_LED, + ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, }; static const struct hda_fixup alc269_fixups[] = { @@ -7251,6 +7252,13 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc236_fixup_hp_mute_led, }, + [ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc5 }, + { } + }, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7446,6 +7454,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x10ec, 0x10f2, "Intel Reference board", ALC700_FIXUP_INTEL_REFERENCE), SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-SZ6", ALC269_FIXUP_HEADSET_MODE), SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC), + SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), + SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xc740, "Samsung Ativ book 8 (NP870Z5G)", ALC269_FIXUP_ATIV_BOOK_8), SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC), -- 2.25.1

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[PATCH AUTOSEL 5.6 136/606] ALSA: hda/realtek - Add HP new mute led supported for ALC236
by Sasha Levin 09 Jun '20

09 Jun '20

From: Kailang Yang <kailang(a)realtek.com> [ Upstream commit 24164f434dc9c23cd34fca1e36acea9d0581bdde ] HP new platform has new mute led feature. COEF index 0x34 bit 5 to control playback mute led. COEF index 0x35 bit 2 and bit 3 to control Mic mute led. [ corrected typos by tiwai ] Signed-off-by: Kailang Yang <kailang(a)realtek.com> Link: https://lore.kernel.org/r/6741211598ba499687362ff2aa30626b@realtek.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 44 +++++++++++++++++++++++++++++++++++ 1 file changed, 44 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 44fbd5d2d89c..368ed3678fc2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4223,6 +4223,23 @@ static void alc285_fixup_hp_mute_led_coefbit(struct hda_codec *codec, } } +static void alc236_fixup_hp_mute_led_coefbit(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->mute_led_polarity = 0; + spec->mute_led_coef_idx = 0x34; + spec->mute_led_coefbit_mask = 1<<5; + spec->mute_led_coefbit_on = 0; + spec->mute_led_coefbit_off = 1<<5; + spec->gen.vmaster_mute.hook = alc_fixup_mute_led_coefbit_hook; + spec->gen.vmaster_mute_enum = 1; + } +} + /* turn on/off mic-mute LED per capture hook by coef bit */ static void alc_hp_cap_micmute_update(struct hda_codec *codec) { @@ -4250,6 +4267,20 @@ static void alc285_fixup_hp_coef_micmute_led(struct hda_codec *codec, } } +static void alc236_fixup_hp_coef_micmute_led(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->mic_led_coef_idx = 0x35; + spec->mic_led_coefbit_mask = 3<<2; + spec->mic_led_coefbit_on = 2<<2; + spec->mic_led_coefbit_off = 1<<2; + snd_hda_gen_add_micmute_led(codec, alc_hp_cap_micmute_update); + } +} + static void alc285_fixup_hp_mute_led(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -4257,6 +4288,13 @@ static void alc285_fixup_hp_mute_led(struct hda_codec *codec, alc285_fixup_hp_coef_micmute_led(codec, fix, action); } +static void alc236_fixup_hp_mute_led(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + alc236_fixup_hp_mute_led_coefbit(codec, fix, action); + alc236_fixup_hp_coef_micmute_led(codec, fix, action); +} + #if IS_REACHABLE(CONFIG_INPUT) static void gpio2_mic_hotkey_event(struct hda_codec *codec, struct hda_jack_callback *event) @@ -6056,6 +6094,7 @@ enum { ALC294_FIXUP_ASUS_COEF_1B, ALC285_FIXUP_HP_GPIO_LED, ALC285_FIXUP_HP_MUTE_LED, + ALC236_FIXUP_HP_MUTE_LED, }; static const struct hda_fixup alc269_fixups[] = { @@ -7208,6 +7247,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc285_fixup_hp_mute_led, }, + [ALC236_FIXUP_HP_MUTE_LED] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc236_fixup_hp_mute_led, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7354,6 +7397,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x84e7, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x8736, "HP", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x877a, "HP", ALC285_FIXUP_HP_MUTE_LED), + SND_PCI_QUIRK(0x103c, 0x877d, "HP", ALC236_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), -- 2.25.1

1 0

[PATCH AUTOSEL 5.6 135/606] ALSA: hda/realtek - Add supported new mute Led for HP
by Sasha Levin 09 Jun '20

09 Jun '20

From: Kailang Yang <kailang(a)realtek.com> [ Upstream commit 431e76c3edd76d84a0ed1eb81a286b2ddecc5ee4 ] HP Note Book supported new mute Led. Hardware PIN was not enough to meet old LED rule. JD2 to control playback mute led. GPO3 to control capture mute led. (ALC285 didn't control GPO3 via verb command) This two PIN just could control by COEF registers. [ corrected typos by tiwai ] Signed-off-by: Kailang Yang <kailang(a)realtek.com> Link: https://lore.kernel.org/r/6741211598ba499687362ff2aa30626b@realtek.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 81 +++++++++++++++++++++++++++++++++++ 1 file changed, 81 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d73c814358bf..44fbd5d2d89c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -86,6 +86,14 @@ struct alc_spec { unsigned int gpio_mute_led_mask; unsigned int gpio_mic_led_mask; + unsigned int mute_led_coef_idx; + unsigned int mute_led_coefbit_mask; + unsigned int mute_led_coefbit_on; + unsigned int mute_led_coefbit_off; + unsigned int mic_led_coef_idx; + unsigned int mic_led_coefbit_mask; + unsigned int mic_led_coefbit_on; + unsigned int mic_led_coefbit_off; hda_nid_t headset_mic_pin; hda_nid_t headphone_mic_pin; @@ -4182,6 +4190,73 @@ static void alc280_fixup_hp_gpio4(struct hda_codec *codec, } } +/* update mute-LED according to the speaker mute state via COEF bit */ +static void alc_fixup_mute_led_coefbit_hook(void *private_data, int enabled) +{ + struct hda_codec *codec = private_data; + struct alc_spec *spec = codec->spec; + + if (spec->mute_led_polarity) + enabled = !enabled; + + /* temporarily power up/down for setting COEF bit */ + enabled ? alc_update_coef_idx(codec, spec->mute_led_coef_idx, + spec->mute_led_coefbit_mask, spec->mute_led_coefbit_off) : + alc_update_coef_idx(codec, spec->mute_led_coef_idx, + spec->mute_led_coefbit_mask, spec->mute_led_coefbit_on); +} + +static void alc285_fixup_hp_mute_led_coefbit(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->mute_led_polarity = 0; + spec->mute_led_coef_idx = 0x0b; + spec->mute_led_coefbit_mask = 1<<3; + spec->mute_led_coefbit_on = 1<<3; + spec->mute_led_coefbit_off = 0; + spec->gen.vmaster_mute.hook = alc_fixup_mute_led_coefbit_hook; + spec->gen.vmaster_mute_enum = 1; + } +} + +/* turn on/off mic-mute LED per capture hook by coef bit */ +static void alc_hp_cap_micmute_update(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + if (spec->gen.micmute_led.led_value) + alc_update_coef_idx(codec, spec->mic_led_coef_idx, + spec->mic_led_coefbit_mask, spec->mic_led_coefbit_on); + else + alc_update_coef_idx(codec, spec->mic_led_coef_idx, + spec->mic_led_coefbit_mask, spec->mic_led_coefbit_off); +} + +static void alc285_fixup_hp_coef_micmute_led(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->mic_led_coef_idx = 0x19; + spec->mic_led_coefbit_mask = 1<<13; + spec->mic_led_coefbit_on = 1<<13; + spec->mic_led_coefbit_off = 0; + snd_hda_gen_add_micmute_led(codec, alc_hp_cap_micmute_update); + } +} + +static void alc285_fixup_hp_mute_led(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + alc285_fixup_hp_mute_led_coefbit(codec, fix, action); + alc285_fixup_hp_coef_micmute_led(codec, fix, action); +} + #if IS_REACHABLE(CONFIG_INPUT) static void gpio2_mic_hotkey_event(struct hda_codec *codec, struct hda_jack_callback *event) @@ -5980,6 +6055,7 @@ enum { ALC294_FIXUP_ASUS_HPE, ALC294_FIXUP_ASUS_COEF_1B, ALC285_FIXUP_HP_GPIO_LED, + ALC285_FIXUP_HP_MUTE_LED, }; static const struct hda_fixup alc269_fixups[] = { @@ -7128,6 +7204,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc285_fixup_hp_gpio_led, }, + [ALC285_FIXUP_HP_MUTE_LED] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_hp_mute_led, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7273,6 +7353,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8497, "HP Envy x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x84e7, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x8736, "HP", ALC285_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x877a, "HP", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), -- 2.25.1

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[PATCH AUTOSEL 5.6 054/606] Revert "ALSA: hda/realtek: Fix pop noise on ALC225"
by Sasha Levin 09 Jun '20

09 Jun '20

From: Kai-Heng Feng <kai.heng.feng(a)canonical.com> commit f41224efcf8aafe80ea47ac870c5e32f3209ffc8 upstream. This reverts commit 3b36b13d5e69d6f51ff1c55d1b404a74646c9757. Enable power save node breaks some systems with ACL225. Revert the patch and use a platform specific quirk for the original issue isntead. Fixes: 3b36b13d5e69 ("ALSA: hda/realtek: Fix pop noise on ALC225") BugLink: https://bugs.launchpad.net/bugs/1875916 Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> Link: https://lore.kernel.org/r/20200503152449.22761-1-kai.heng.feng@canonical.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> --- sound/pci/hda/patch_realtek.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d6bc39dce4cf..d73c814358bf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8121,8 +8121,6 @@ static int patch_alc269(struct hda_codec *codec) spec->gen.mixer_nid = 0; break; case 0x10ec0225: - codec->power_save_node = 1; - /* fall through */ case 0x10ec0295: case 0x10ec0299: spec->codec_variant = ALC269_TYPE_ALC225; -- 2.25.1

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[PATCH AUTOSEL 5.6 025/606] ALSA: usb-audio: Add control message quirk delay for Kingston HyperX headset
by Sasha Levin 09 Jun '20

09 Jun '20

From: Jesus Ramos <jesus-ramos(a)live.com> commit 073919e09ca445d4486968e3f851372ff44cf2b5 upstream. Kingston HyperX headset with 0951:16ad also needs the same quirk for delaying the frequency controls. Signed-off-by: Jesus Ramos <jesus-ramos(a)live.com> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/BY5PR19MB3634BA68C7CCA23D8DF428E796AF0@BY5PR19MB3… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> --- sound/usb/quirks.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 0686e056e39b..732580bdc6a4 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1592,13 +1592,14 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) msleep(20); - /* Zoom R16/24, Logitech H650e, Jabra 550a needs a tiny delay here, - * otherwise requests like get/set frequency return as failed despite - * actually succeeding. + /* Zoom R16/24, Logitech H650e, Jabra 550a, Kingston HyperX needs a tiny + * delay here, otherwise requests like get/set frequency return as + * failed despite actually succeeding. */ if ((chip->usb_id == USB_ID(0x1686, 0x00dd) || chip->usb_id == USB_ID(0x046d, 0x0a46) || - chip->usb_id == USB_ID(0x0b0e, 0x0349)) && + chip->usb_id == USB_ID(0x0b0e, 0x0349) || + chip->usb_id == USB_ID(0x0951, 0x16ad)) && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) usleep_range(1000, 2000); } -- 2.25.1

1 0

[PATCH AUTOSEL 5.6 024/606] ALSA: rawmidi: Fix racy buffer resize under concurrent accesses
by Sasha Levin 09 Jun '20

09 Jun '20

From: Takashi Iwai <tiwai(a)suse.de> commit c1f6e3c818dd734c30f6a7eeebf232ba2cf3181d upstream. The rawmidi core allows user to resize the runtime buffer via ioctl, and this may lead to UAF when performed during concurrent reads or writes: the read/write functions unlock the runtime lock temporarily during copying form/to user-space, and that's the race window. This patch fixes the hole by introducing a reference counter for the runtime buffer read/write access and returns -EBUSY error when the resize is performed concurrently against read/write. Note that the ref count field is a simple integer instead of refcount_t here, since the all contexts accessing the buffer is basically protected with a spinlock, hence we need no expensive atomic ops. Also, note that this busy check is needed only against read / write functions, and not in receive/transmit callbacks; the race can happen only at the spinlock hole mentioned in the above, while the whole function is protected for receive / transmit callbacks. Reported-by: butt3rflyh4ck <butterflyhuangxx(a)gmail.com> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/CAFcO6XMWpUVK_yzzCpp8_XP7+=oUpQvuBeCbMffEDkpe8jWr… Link: https://lore.kernel.org/r/s5heerw3r5z.wl-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> --- include/sound/rawmidi.h | 1 + sound/core/rawmidi.c | 31 +++++++++++++++++++++++++++---- 2 files changed, 28 insertions(+), 4 deletions(-) diff --git a/include/sound/rawmidi.h b/include/sound/rawmidi.h index a36b7227a15a..334842daa904 100644 --- a/include/sound/rawmidi.h +++ b/include/sound/rawmidi.h @@ -61,6 +61,7 @@ struct snd_rawmidi_runtime { size_t avail_min; /* min avail for wakeup */ size_t avail; /* max used buffer for wakeup */ size_t xruns; /* over/underruns counter */ + int buffer_ref; /* buffer reference count */ /* misc */ spinlock_t lock; wait_queue_head_t sleep; diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 20dd08e1f675..2a688b711a9a 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -120,6 +120,17 @@ static void snd_rawmidi_input_event_work(struct work_struct *work) runtime->event(runtime->substream); } +/* buffer refcount management: call with runtime->lock held */ +static inline void snd_rawmidi_buffer_ref(struct snd_rawmidi_runtime *runtime) +{ + runtime->buffer_ref++; +} + +static inline void snd_rawmidi_buffer_unref(struct snd_rawmidi_runtime *runtime) +{ + runtime->buffer_ref--; +} + static int snd_rawmidi_runtime_create(struct snd_rawmidi_substream *substream) { struct snd_rawmidi_runtime *runtime; @@ -669,6 +680,11 @@ static int resize_runtime_buffer(struct snd_rawmidi_runtime *runtime, if (!newbuf) return -ENOMEM; spin_lock_irq(&runtime->lock); + if (runtime->buffer_ref) { + spin_unlock_irq(&runtime->lock); + kvfree(newbuf); + return -EBUSY; + } oldbuf = runtime->buffer; runtime->buffer = newbuf; runtime->buffer_size = params->buffer_size; @@ -1019,8 +1035,10 @@ static long snd_rawmidi_kernel_read1(struct snd_rawmidi_substream *substream, long result = 0, count1; struct snd_rawmidi_runtime *runtime = substream->runtime; unsigned long appl_ptr; + int err = 0; spin_lock_irqsave(&runtime->lock, flags); + snd_rawmidi_buffer_ref(runtime); while (count > 0 && runtime->avail) { count1 = runtime->buffer_size - runtime->appl_ptr; if (count1 > count) @@ -1039,16 +1057,19 @@ static long snd_rawmidi_kernel_read1(struct snd_rawmidi_substream *substream, if (userbuf) { spin_unlock_irqrestore(&runtime->lock, flags); if (copy_to_user(userbuf + result, - runtime->buffer + appl_ptr, count1)) { - return result > 0 ? result : -EFAULT; - } + runtime->buffer + appl_ptr, count1)) + err = -EFAULT; spin_lock_irqsave(&runtime->lock, flags); + if (err) + goto out; } result += count1; count -= count1; } + out: + snd_rawmidi_buffer_unref(runtime); spin_unlock_irqrestore(&runtime->lock, flags); - return result; + return result > 0 ? result : err; } long snd_rawmidi_kernel_read(struct snd_rawmidi_substream *substream, @@ -1342,6 +1363,7 @@ static long snd_rawmidi_kernel_write1(struct snd_rawmidi_substream *substream, return -EAGAIN; } } + snd_rawmidi_buffer_ref(runtime); while (count > 0 && runtime->avail > 0) { count1 = runtime->buffer_size - runtime->appl_ptr; if (count1 > count) @@ -1373,6 +1395,7 @@ static long snd_rawmidi_kernel_write1(struct snd_rawmidi_substream *substream, } __end: count1 = runtime->avail < runtime->buffer_size; + snd_rawmidi_buffer_unref(runtime); spin_unlock_irqrestore(&runtime->lock, flags); if (count1) snd_rawmidi_output_trigger(substream, 1); -- 2.25.1

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[PATCH AUTOSEL 5.6 023/606] ALSA: hda/realtek - Add COEF workaround for ASUS ZenBook UX431DA
by Sasha Levin 09 Jun '20

09 Jun '20

From: Takashi Iwai <tiwai(a)suse.de> commit 1b94e59d30afecf18254ad413e953e7587645a20 upstream. ASUS ZenBook UX431DA requires an additional COEF setup when booted from the recent Windows 10, otherwise it produces the noisy output. The quirk turns on COEF 0x1b bit 10 that has been cleared supposedly due to the pop noise reduction. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207553 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Link: https://lore.kernel.org/r/20200512073203.14091-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> --- sound/pci/hda/patch_realtek.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e787792770be..d6bc39dce4cf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5978,6 +5978,7 @@ enum { ALC294_FIXUP_ASUS_DUAL_SPK, ALC285_FIXUP_THINKPAD_HEADSET_JACK, ALC294_FIXUP_ASUS_HPE, + ALC294_FIXUP_ASUS_COEF_1B, ALC285_FIXUP_HP_GPIO_LED, }; @@ -7112,6 +7113,17 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC294_FIXUP_ASUS_HEADSET_MIC }, + [ALC294_FIXUP_ASUS_COEF_1B] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + /* Set bit 10 to correct noisy output after reboot from + * Windows 10 (due to pop noise reduction?) + */ + { 0x20, AC_VERB_SET_COEF_INDEX, 0x1b }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x4e4b }, + { } + }, + }, [ALC285_FIXUP_HP_GPIO_LED] = { .type = HDA_FIXUP_FUNC, .v.func = alc285_fixup_hp_gpio_led, @@ -7283,6 +7295,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x19ce, "ASUS B9450FA", ALC294_FIXUP_ASUS_HPE), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), SND_PCI_QUIRK(0x1043, 0x1a30, "ASUS X705UD", ALC256_FIXUP_ASUS_MIC), + SND_PCI_QUIRK(0x1043, 0x1b11, "ASUS UX431DA", ALC294_FIXUP_ASUS_COEF_1B), SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1043, 0x1bbd, "ASUS Z550MA", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1c23, "Asus X55U", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), -- 2.25.1

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[PATCH AUTOSEL 5.6 022/606] ALSA: hda/realtek - Limit int mic boost for Thinkpad T530
by Sasha Levin 09 Jun '20

09 Jun '20

From: Takashi Iwai <tiwai(a)suse.de> commit b590b38ca305d6d7902ec7c4f7e273e0069f3bcc upstream. Lenovo Thinkpad T530 seems to have a sensitive internal mic capture that needs to limit the mic boost like a few other Thinkpad models. Although we may change the quirk for ALC269_FIXUP_LENOVO_DOCK, this hits way too many other laptop models, so let's add a new fixup model that limits the internal mic boost on top of the existing quirk and apply to only T530. BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1171293 Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200514160533.10337-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> --- sound/pci/hda/patch_realtek.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 611498270c5e..e787792770be 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5856,6 +5856,7 @@ enum { ALC269_FIXUP_HP_LINE1_MIC1_LED, ALC269_FIXUP_INV_DMIC, ALC269_FIXUP_LENOVO_DOCK, + ALC269_FIXUP_LENOVO_DOCK_LIMIT_BOOST, ALC269_FIXUP_NO_SHUTUP, ALC286_FIXUP_SONY_MIC_NO_PRESENCE, ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT, @@ -6175,6 +6176,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT }, + [ALC269_FIXUP_LENOVO_DOCK_LIMIT_BOOST] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_limit_int_mic_boost, + .chained = true, + .chain_id = ALC269_FIXUP_LENOVO_DOCK, + }, [ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_pincfg_no_hp_to_lineout, @@ -7317,7 +7324,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x21ca, "Thinkpad L412", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 15", ALC269_FIXUP_SKU_IGNORE), - SND_PCI_QUIRK(0x17aa, 0x21f6, "Thinkpad T530", ALC269_FIXUP_LENOVO_DOCK), + SND_PCI_QUIRK(0x17aa, 0x21f6, "Thinkpad T530", ALC269_FIXUP_LENOVO_DOCK_LIMIT_BOOST), SND_PCI_QUIRK(0x17aa, 0x21fa, "Thinkpad X230", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x21f3, "Thinkpad T430", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK), @@ -7456,6 +7463,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC269_FIXUP_HEADSET_MODE, .name = "headset-mode"}, {.id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC, .name = "headset-mode-no-hp-mic"}, {.id = ALC269_FIXUP_LENOVO_DOCK, .name = "lenovo-dock"}, + {.id = ALC269_FIXUP_LENOVO_DOCK_LIMIT_BOOST, .name = "lenovo-dock-limit-boost"}, {.id = ALC269_FIXUP_HP_GPIO_LED, .name = "hp-gpio-led"}, {.id = ALC269_FIXUP_HP_DOCK_GPIO_MIC1_LED, .name = "hp-dock-gpio-mic1-led"}, {.id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "dell-headset-multi"}, -- 2.25.1

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Re: [PATCH 2/5] soundwire: qcom: add support for mmio soundwire devices
by Pierre-Louis Bossart 08 Jun '20

08 Jun '20

On 6/8/20 3:43 PM, Jonathan Marek wrote: > Adds support for qcom soundwire devices with memory mapped IO registers. 'device' is an ambiguous term for SoundWire. Seems to me this is a SoundWire Master device directly accessed with mmio registers instead of over a SLIMbus link? > > Signed-off-by: Jonathan Marek <jonathan(a)marek.ca> > --- > drivers/soundwire/qcom.c | 25 +++++++++++++++++++++++-- > 1 file changed, 23 insertions(+), 2 deletions(-) > > diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c > index f38d1fd3679f..628747df1c75 100644 > --- a/drivers/soundwire/qcom.c > +++ b/drivers/soundwire/qcom.c > @@ -90,6 +90,7 @@ struct qcom_swrm_ctrl { > struct sdw_bus bus; > struct device *dev; > struct regmap *regmap; > + void __iomem *mmio; > struct completion *comp; > struct work_struct slave_work; > /* read/write lock */ > @@ -154,6 +155,20 @@ static int qcom_swrm_ahb_reg_write(struct qcom_swrm_ctrl *ctrl, > return SDW_CMD_OK; > } > > +static int qcom_swrm_cpu_reg_read(struct qcom_swrm_ctrl *ctrl, int reg, > + u32 *val) > +{ > + *val = readl(ctrl->mmio + reg); > + return SDW_CMD_OK; > +} > + > +static int qcom_swrm_cpu_reg_write(struct qcom_swrm_ctrl *ctrl, int reg, > + int val) > +{ > + writel(val, ctrl->mmio + reg); > + return SDW_CMD_OK; > +} > + > static int qcom_swrm_cmd_fifo_wr_cmd(struct qcom_swrm_ctrl *ctrl, u8 cmd_data, > u8 dev_addr, u16 reg_addr) > { > @@ -746,6 +761,7 @@ static int qcom_swrm_probe(struct platform_device *pdev) > struct sdw_master_prop *prop; > struct sdw_bus_params *params; > struct qcom_swrm_ctrl *ctrl; > + struct resource *res; > int ret; > u32 val; > > @@ -760,8 +776,13 @@ static int qcom_swrm_probe(struct platform_device *pdev) > if (!ctrl->regmap) > return -EINVAL; > } else { > - /* Only WCD based SoundWire controller is supported */ > - return -ENOTSUPP; I would move patch4 before this one, and add the functionality *after* removing the SLIMbus dependency. > + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); > + > + ctrl->reg_read = qcom_swrm_cpu_reg_read; > + ctrl->reg_write = qcom_swrm_cpu_reg_write; > + ctrl->mmio = devm_ioremap_resource(dev, res); > + if (IS_ERR(ctrl->mmio)) > + return PTR_ERR(ctrl->mmio); maybe deal with the resource checks before setting callbacks? There are quite a few drivers who do things this way: clk/qcom/common.c- res = platform_get_resource(pdev, IORESOURCE_MEM, 0); clk/qcom/common.c: base = devm_ioremap_resource(dev, res); -- > } > > ctrl->irq = of_irq_get(dev->of_node, 0); >

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Re: [PATCH 4/5] soundwire: qcom: avoid dependency on CONFIG_SLIMBUS
by Pierre-Louis Bossart 08 Jun '20

08 Jun '20

On 6/8/20 3:43 PM, Jonathan Marek wrote: > The driver may be used without slimbus, so don't depend on slimbus. > > Signed-off-by: Jonathan Marek <jonathan(a)marek.ca> > --- > drivers/soundwire/Kconfig | 1 - > drivers/soundwire/qcom.c | 5 +++++ > 2 files changed, 5 insertions(+), 1 deletion(-) > > diff --git a/drivers/soundwire/Kconfig b/drivers/soundwire/Kconfig > index fa2b4ab92ed9..d121cf739090 100644 > --- a/drivers/soundwire/Kconfig > +++ b/drivers/soundwire/Kconfig > @@ -33,7 +33,6 @@ config SOUNDWIRE_INTEL > > config SOUNDWIRE_QCOM > tristate "Qualcomm SoundWire Master driver" > - depends on SLIMBUS > depends on SND_SOC > help > SoundWire Qualcomm Master driver. > diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c > index 14334442615f..ac81c64768ea 100644 > --- a/drivers/soundwire/qcom.c > +++ b/drivers/soundwire/qcom.c > @@ -769,13 +769,18 @@ static int qcom_swrm_probe(struct platform_device *pdev) > if (!ctrl) > return -ENOMEM; > > +#ifdef CONFIG_SLIMBUS > if (dev->parent->bus == &slimbus_bus) { > +#else > + if (false) { > +#endif maybe: if (IS_ENABLED(CONFIG_SLIMBUS) && dev->parent->bus == &slimbus_bus) > ctrl->reg_read = qcom_swrm_ahb_reg_read; > ctrl->reg_write = qcom_swrm_ahb_reg_write; > ctrl->regmap = dev_get_regmap(dev->parent, NULL); > if (!ctrl->regmap) > return -EINVAL; > } else { > + > res = platform_get_resource(pdev, IORESOURCE_MEM, 0); > > ctrl->reg_read = qcom_swrm_cpu_reg_read; >

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1.2.3: alsa-libs is not LTO ready
by GitHub issues - edited 08 Jun '20

08 Jun '20

alsa-project/alsa-lib issue #56 was edited from kloczek: When build with LTO test suite is failing ``` [tkloczko@barrel alsa-lib-1.2.3]$ make check Making check in doc make[1]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/doc' Making check in pictures make[2]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/doc/pictures' make[2]: Nothing to be done for 'check'. make[2]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/doc/pictures' make[2]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/doc' make[2]: Nothing to be done for 'check-am'. make[2]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/doc' make[1]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/doc' Making check in include make[1]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/include' Making check in sound make[2]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/include/sound' Making check in uapi make[3]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/include/sound/uapi' make[3]: Nothing to be done for 'check'. make[3]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/include/sound/uapi' make[3]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/include/sound' make[3]: Nothing to be done for 'check-am'. make[3]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/include/sound' make[2]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/include/sound' make[2]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/include' make[2]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/include' make[1]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/include' Making check in src make[1]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src' Making check in control make[2]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/control' make[2]: Nothing to be done for 'check'. make[2]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/control' Making check in mixer make[2]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/mixer' make[2]: Nothing to be done for 'check'. make[2]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/mixer' Making check in pcm make[2]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/pcm' make[3]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/pcm' make[3]: Nothing to be done for 'check-am'. make[3]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/pcm' make[2]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/pcm' Making check in timer make[2]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/timer' make[2]: Nothing to be done for 'check'. make[2]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/timer' Making check in rawmidi make[2]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/rawmidi' make[2]: Nothing to be done for 'check'. make[2]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/rawmidi' Making check in hwdep make[2]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/hwdep' make[2]: Nothing to be done for 'check'. make[2]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/hwdep' Making check in seq make[2]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/seq' make[2]: Nothing to be done for 'check'. make[2]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/seq' Making check in ucm make[2]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/ucm' make[2]: Nothing to be done for 'check'. make[2]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/ucm' Making check in conf make[2]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/conf' Making check in cards make[3]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/conf/cards' make[3]: Nothing to be done for 'check'. make[3]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/conf/cards' Making check in pcm make[3]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/conf/pcm' make[3]: Nothing to be done for 'check'. make[3]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/conf/pcm' make[3]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/conf' make[3]: Nothing to be done for 'check-am'. make[3]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/conf' make[2]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/conf' make[2]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src' make[2]: Nothing to be done for 'check-am'. make[2]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src' make[1]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src' Making check in src/topology make[1]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/topology' make[1]: Nothing to be done for 'check'. make[1]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/topology' Making check in modules make[1]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/modules' Making check in mixer make[2]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/modules/mixer' Making check in simple make[3]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/modules/mixer/simple' make[3]: Nothing to be done for 'check'. make[3]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/modules/mixer/simple' make[3]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/modules/mixer' make[3]: Nothing to be done for 'check-am'. make[3]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/modules/mixer' make[2]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/modules/mixer' make[2]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/modules' make[2]: Nothing to be done for 'check-am'. make[2]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/modules' make[1]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/modules' Making check in aserver make[1]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/aserver' make[1]: Nothing to be done for 'check'. make[1]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/aserver' Making check in test make[1]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/test' Making check in . make[2]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/test' make control pcm pcm_min latency seq playmidi1 timer rawmidi midiloop oldapi queue_timer namehint client_event_filter chmap audio_time user-ctl-element-set pcm-multi-thread make[3]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/test' make[3]: 'control' is up to date. make[3]: 'pcm' is up to date. make[3]: 'pcm_min' is up to date. make[3]: 'latency' is up to date. make[3]: 'seq' is up to date. make[3]: 'playmidi1' is up to date. make[3]: 'timer' is up to date. make[3]: 'rawmidi' is up to date. make[3]: 'midiloop' is up to date. /bin/sh ../libtool --tag=CC --mode=link gcc -Wall -pipe -g -O2 -g -pipe -Wall -Werror=format-security -Wp,-D_FORTIFY_SOURCE=2 -Wp,-D_GLIBCXX_ASSERTIONS -fstack-protector-strong -grecord-gcc-switches -specs=/usr/lib/rpm/redhat/redhat-hardened-cc1 -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -m64 -mtune=generic -fasynchronous-unwind-tables -fstack-clash-protection -fcf-protection -flto=auto -flto-partition=none -Wl,-z,relro -Wl,--as-needed -Wl,-z,now -specs=/usr/lib/rpm/redhat/redhat-hardened-ld -flto=auto -flto-partition=none -fuse-linker-plugin -o oldapi oldapi.o ../src/libasound.la libtool: link: gcc -Wall -pipe -g -O2 -g -pipe -Wall -Werror=format-security -Wp,-D_FORTIFY_SOURCE=2 -Wp,-D_GLIBCXX_ASSERTIONS -fstack-protector-strong -grecord-gcc-switches -specs=/usr/lib/rpm/redhat/redhat-hardened-cc1 -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -m64 -mtune=generic -fasynchronous-unwind-tables -fstack-clash-protection -fcf-protection -flto=auto -flto-partition=none -Wl,-z -Wl,relro -Wl,--as-needed -Wl,-z -Wl,now -specs=/usr/lib/rpm/redhat/redhat-hardened-ld -flto=auto -flto-partition=none -fuse-linker-plugin -o .libs/oldapi oldapi.o ../src/.libs/libasound.so -lm -ldl -lpthread -lrt /usr/bin/ld: oldapi.lto.o: in function `main': /home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/test/oldapi.c:39: undefined reference to `snd_pcm_hw_params_get_access(a)ALSA_0.9' collect2: error: ld returned 1 exit status make[3]: *** [Makefile:529: oldapi] Error 1 make[3]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/test' make[2]: *** [Makefile:796: check-am] Error 2 make[2]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/test' make[1]: *** [Makefile:645: check-recursive] Error 1 make[1]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/test' ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/56 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

1.2.3: alsa-l;ibs is not LTO ready
by GitHub issues - opened 08 Jun '20

08 Jun '20

alsa-project/alsa-lib issue #56 was opened from kloczek: When build with LTO test suite is failing ``` [tkloczko@barrel alsa-lib-1.2.3]$ make check Making check in doc make[1]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/doc' Making check in pictures make[2]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/doc/pictures' make[2]: Nothing to be done for 'check'. make[2]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/doc/pictures' make[2]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/doc' make[2]: Nothing to be done for 'check-am'. make[2]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/doc' make[1]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/doc' Making check in include make[1]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/include' Making check in sound make[2]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/include/sound' Making check in uapi make[3]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/include/sound/uapi' make[3]: Nothing to be done for 'check'. make[3]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/include/sound/uapi' make[3]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/include/sound' make[3]: Nothing to be done for 'check-am'. make[3]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/include/sound' make[2]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/include/sound' make[2]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/include' make[2]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/include' make[1]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/include' Making check in src make[1]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src' Making check in control make[2]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/control' make[2]: Nothing to be done for 'check'. make[2]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/control' Making check in mixer make[2]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/mixer' make[2]: Nothing to be done for 'check'. make[2]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/mixer' Making check in pcm make[2]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/pcm' make[3]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/pcm' make[3]: Nothing to be done for 'check-am'. make[3]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/pcm' make[2]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/pcm' Making check in timer make[2]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/timer' make[2]: Nothing to be done for 'check'. make[2]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/timer' Making check in rawmidi make[2]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/rawmidi' make[2]: Nothing to be done for 'check'. make[2]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/rawmidi' Making check in hwdep make[2]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/hwdep' make[2]: Nothing to be done for 'check'. make[2]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/hwdep' Making check in seq make[2]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/seq' make[2]: Nothing to be done for 'check'. make[2]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/seq' Making check in ucm make[2]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/ucm' make[2]: Nothing to be done for 'check'. make[2]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/ucm' Making check in conf make[2]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/conf' Making check in cards make[3]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/conf/cards' make[3]: Nothing to be done for 'check'. make[3]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/conf/cards' Making check in pcm make[3]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/conf/pcm' make[3]: Nothing to be done for 'check'. make[3]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/conf/pcm' make[3]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/conf' make[3]: Nothing to be done for 'check-am'. make[3]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/conf' make[2]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/conf' make[2]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src' make[2]: Nothing to be done for 'check-am'. make[2]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src' make[1]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src' Making check in src/topology make[1]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/topology' make[1]: Nothing to be done for 'check'. make[1]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/src/topology' Making check in modules make[1]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/modules' Making check in mixer make[2]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/modules/mixer' Making check in simple make[3]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/modules/mixer/simple' make[3]: Nothing to be done for 'check'. make[3]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/modules/mixer/simple' make[3]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/modules/mixer' make[3]: Nothing to be done for 'check-am'. make[3]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/modules/mixer' make[2]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/modules/mixer' make[2]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/modules' make[2]: Nothing to be done for 'check-am'. make[2]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/modules' make[1]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/modules' Making check in aserver make[1]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/aserver' make[1]: Nothing to be done for 'check'. make[1]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/aserver' Making check in test make[1]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/test' Making check in . make[2]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/test' make control pcm pcm_min latency seq playmidi1 timer rawmidi midiloop oldapi queue_timer namehint client_event_filter chmap audio_time user-ctl-element-set pcm-multi-thread make[3]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/test' make[3]: 'control' is up to date. make[3]: 'pcm' is up to date. make[3]: 'pcm_min' is up to date. make[3]: 'latency' is up to date. make[3]: 'seq' is up to date. make[3]: 'playmidi1' is up to date. make[3]: 'timer' is up to date. make[3]: 'rawmidi' is up to date. make[3]: 'midiloop' is up to date. /bin/sh ../libtool --tag=CC --mode=link gcc -Wall -pipe -g -O2 -g -pipe -Wall -Werror=format-security -Wp,-D_FORTIFY_SOURCE=2 -Wp,-D_GLIBCXX_ASSERTIONS -fstack-protector-strong -grecord-gcc-switches -specs=/usr/lib/rpm/redhat/redhat-hardened-cc1 -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -m64 -mtune=generic -fasynchronous-unwind-tables -fstack-clash-protection -fcf-protection -flto=auto -flto-partition=none -Wl,-z,relro -Wl,--as-needed -Wl,-z,now -specs=/usr/lib/rpm/redhat/redhat-hardened-ld -flto=auto -flto-partition=none -fuse-linker-plugin -o oldapi oldapi.o ../src/libasound.la libtool: link: gcc -Wall -pipe -g -O2 -g -pipe -Wall -Werror=format-security -Wp,-D_FORTIFY_SOURCE=2 -Wp,-D_GLIBCXX_ASSERTIONS -fstack-protector-strong -grecord-gcc-switches -specs=/usr/lib/rpm/redhat/redhat-hardened-cc1 -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -m64 -mtune=generic -fasynchronous-unwind-tables -fstack-clash-protection -fcf-protection -flto=auto -flto-partition=none -Wl,-z -Wl,relro -Wl,--as-needed -Wl,-z -Wl,now -specs=/usr/lib/rpm/redhat/redhat-hardened-ld -flto=auto -flto-partition=none -fuse-linker-plugin -o .libs/oldapi oldapi.o ../src/.libs/libasound.so -lm -ldl -lpthread -lrt /usr/bin/ld: oldapi.lto.o: in function `main': /home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/test/oldapi.c:39: undefined reference to `snd_pcm_hw_params_get_access(a)ALSA_0.9' collect2: error: ld returned 1 exit status make[3]: *** [Makefile:529: oldapi] Error 1 make[3]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/test' make[2]: *** [Makefile:796: check-am] Error 2 make[2]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/test' make[1]: *** [Makefile:645: check-recursive] Error 1 make[1]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.3/test' ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/56 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 0/2] ALSA: pcm: stream linking locking fixes
by Michał Mirosław 08 Jun '20

08 Jun '20

Two patches fixing locking issues for SNDRV_PCM_IOCTL_LINK handling: first adds a check preventing linking a stream to itself, second quiets lockdep warning about nested locks. Michał Mirosław (2): ALSA: pcm: disallow linking stream to itself ALSA: pcm: fix snd_pcm_link() lockdep splat sound/core/pcm_native.c | 20 ++++++++++++++++++-- 1 file changed, 18 insertions(+), 2 deletions(-) -- 2.20.1

2 6

alsa-gobject: seq: introduce container of events
by GitHub pull_request - opened 08 Jun '20

08 Jun '20

alsa-project/alsa-gobject pull request #25 was opened from takaswie: Hi, This patchset adds a new GObject-derived object; ALSASeqEventCntr to obsolete usage of ALSASeqEvent, ALSASeqEventFixed, and ALSASeqEventVariable. At present, ALSASeqUserClient processes events per each. It brings recursive calls of scheduling API or recursive handling for emitted GObject signal for received events. It's inconvenient in a point of memory usage and signal frequency. The ALSASeqEventCntr includes a flat memory to represent a batch of events. The object includes many accessor methods to properties of event according to given index number. The object enables applications to handle a batch of events without the recursive calls or the recursive handling. This patchset obsoletes ALSASeqEvent, thus event structure is not produced itself. Applications can define its own event structure so that it's convenient in a point of chose programming language. It's probable to design collection in the languages with the object. Request URL : https://github.com/alsa-project/alsa-gobject/pull/25 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/25.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

Re: SOF 1.5 release on FTP server
by Jaroslav Kysela 08 Jun '20

08 Jun '20

Dne 01. 06. 20 v 10:14 Paul Dann napsal(a): > On Mon, 1 Jun 2020 at 08:53, Jaroslav Kysela <perex(a)perex.cz> wrote: >> I don't plan any new release on ALSA FTP. Intel releases signed firmware files >> on github now: >> >> https://github.com/thesofproject/sof-bin > > Hi Jaroslav; many thanks for the fast response. It does seem that the > Github version is missing a number of files that are present in your > FTP version. I'd just like to check this doesn't represent a loss of > functionality? My FTP packages also include all unsigned firmware files compiled with gcc. They are unusable for the most Intel platforms where only signed firmware files can be loaded to DSP (security). Those files are probably good for the development platforms, but they do not have much value for the general Linux distros at the moment. I assume that developers can compile own versions. BTW: Fixed topology files for v1.5 (generic sof hda): https://www.alsa-project.org/files/pub/misc/sof/sof-bin-topology-1.5-dmic-2… Use this on top of the sof-bin repo / v1.5 branch if you like. It seems that the fix in sof-bin is delayed. Jaroslav > > Many thanks again, > Paul > -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

4 3

[PATCH] ALSA: hda/realtek - add a pintbl quirk for several Lenovo machines
by Hui Wang 08 Jun '20

08 Jun '20

A couple of Lenovo ThinkCentre machines all have 2 front mics and they use the same codec alc623 and have the same pin config, so add a pintbl entry for those machines to apply the fixup ALC283_FIXUP_HEADSET_MIC. Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0aa778ff7f2b..6d73f8beadb6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8161,6 +8161,12 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { ALC225_STANDARD_PINS, {0x12, 0xb7a60130}, {0x17, 0x90170110}), + SND_HDA_PIN_QUIRK(0x10ec0623, 0x17aa, "Lenovo", ALC283_FIXUP_HEADSET_MIC, + {0x14, 0x01014010}, + {0x17, 0x90170120}, + {0x18, 0x02a11030}, + {0x19, 0x02a1103f}, + {0x21, 0x0221101f}), {} }; -- 2.17.1

2 1

[PATCH] ASoC: meson: fix memory leak of links if allocation of ldata fails
by Colin King 08 Jun '20

08 Jun '20

From: Colin Ian King <colin.king(a)canonical.com> Currently if the allocation of ldata fails the error return path does not kfree the allocated links object. Fix this by adding an error exit return path that performs the necessary kfree'ing. Addresses-Coverity: ("Resource leak") Fixes: 7864a79f37b5 ("ASoC: meson: add axg sound card support") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/meson/meson-card-utils.c | 17 ++++++++++++----- 1 file changed, 12 insertions(+), 5 deletions(-) diff --git a/sound/soc/meson/meson-card-utils.c b/sound/soc/meson/meson-card-utils.c index 2ca8c98e204f..5a4a91c88734 100644 --- a/sound/soc/meson/meson-card-utils.c +++ b/sound/soc/meson/meson-card-utils.c @@ -49,19 +49,26 @@ int meson_card_reallocate_links(struct snd_soc_card *card, links = krealloc(priv->card.dai_link, num_links * sizeof(*priv->card.dai_link), GFP_KERNEL | __GFP_ZERO); + if (!links) + goto err_links; + ldata = krealloc(priv->link_data, num_links * sizeof(*priv->link_data), GFP_KERNEL | __GFP_ZERO); - - if (!links || !ldata) { - dev_err(priv->card.dev, "failed to allocate links\n"); - return -ENOMEM; - } + if (!ldata) + goto err_ldata; priv->card.dai_link = links; priv->link_data = ldata; priv->card.num_links = num_links; return 0; + +err_ldata: + kfree(links); +err_links: + dev_err(priv->card.dev, "failed to allocate links\n"); + return -ENOMEM; + } EXPORT_SYMBOL_GPL(meson_card_reallocate_links); -- 2.25.1

3 2

[PATCH] ALSA: usb-audio: Use the new macro for HP Dock rename quirks
by Takashi Iwai 08 Jun '20

08 Jun '20

Replace the open-code with the new QUIRK_DEVICE_PROFILE() macro for simplicity. Fixes: 0c5086f56999 ("ALSA: usb-audio: Add vendor, product and profile name for HP Thunderbolt Dock") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/quirks-table.h | 16 ++++------------ 1 file changed, 4 insertions(+), 12 deletions(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 90d65bfa733d..4ec491011b19 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -43,22 +43,14 @@ /* HP Thunderbolt Dock Audio Headset */ { USB_DEVICE(0x03f0, 0x0269), - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .vendor_name = "HP", - .product_name = "Thunderbolt Dock Audio Headset", - .profile_name = "HP-Thunderbolt-Dock-Audio-Headset", - .ifnum = QUIRK_NO_INTERFACE - } + QUIRK_DEVICE_PROFILE("HP", "Thunderbolt Dock Audio Headset", + "HP-Thunderbolt-Dock-Audio-Headset"), }, /* HP Thunderbolt Dock Audio Module */ { USB_DEVICE(0x03f0, 0x0567), - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .vendor_name = "HP", - .product_name = "Thunderbolt Dock Audio Module", - .profile_name = "HP-Thunderbolt-Dock-Audio-Module", - .ifnum = QUIRK_NO_INTERFACE - } + QUIRK_DEVICE_PROFILE("HP", "Thunderbolt Dock Audio Module", + "HP-Thunderbolt-Dock-Audio-Module"), }, /* FTDI devices */ { -- 2.16.4

2 5

[PATCH] ALSA: usb-audio: Add vendor, product and profile name for HP Thunderbolt Dock
by Kai-Heng Feng 08 Jun '20

08 Jun '20

The HP Thunderbolt Dock has two separate USB devices, one is for speaker and one is for headset. Add names for them so userspace can apply UCM settings. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/usb/quirks-table.h | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 6d6492195bdc..90d65bfa733d 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -40,6 +40,26 @@ .ifnum = QUIRK_NO_INTERFACE \ } +/* HP Thunderbolt Dock Audio Headset */ +{ + USB_DEVICE(0x03f0, 0x0269), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "HP", + .product_name = "Thunderbolt Dock Audio Headset", + .profile_name = "HP-Thunderbolt-Dock-Audio-Headset", + .ifnum = QUIRK_NO_INTERFACE + } +}, +/* HP Thunderbolt Dock Audio Module */ +{ + USB_DEVICE(0x03f0, 0x0567), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "HP", + .product_name = "Thunderbolt Dock Audio Module", + .profile_name = "HP-Thunderbolt-Dock-Audio-Module", + .ifnum = QUIRK_NO_INTERFACE + } +}, /* FTDI devices */ { USB_DEVICE(0x0403, 0xb8d8), -- 2.17.1

2 1

config for Renoir APU
by GitHub pull_request - edited 07 Jun '20

07 Jun '20

alsa-project/alsa-ucm-conf pull request #31 was edited from perexg: This is just a test config (not for merge). BEWARE: This configuration is not intended for the broad use at the moment. It will broke other HDA Intel hardware. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/31 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/31.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

config for Renoir APU
by GitHub pull_request - opened 07 Jun '20

07 Jun '20

alsa-project/alsa-ucm-conf pull request #31 was opened from perexg: This is just a test config (not for merge). Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/31 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/31.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ALSA control service in user space
by Takashi Sakamoto 07 Jun '20

07 Jun '20

Hi, Today I released hinawa-rs[1], Rust bindings to libhinawa v2.0[2]. Aug 2018 I sent a message[3] about new releases of libhinawa/hinawa-utils. In the releases, I used Python 3 to write tools for audio and music units on IEEE 1394 bus, supported by ALSA firewire stack. The attempt is successful to make my understandings further about vendor-dependent protocols. Earlier the message, Nov 2016 I had a presentation about ALSA device control service in user space[4]. The idea requires server program to perform some tasks below in user space: * register control element set into ALSA control core for existent card instance * dispatch control event from ALSA control applications * communicate with device to operate its functions such as volume control * dispatch any event from the device. The registered control element set can be handled transparently by ALSA control applications such as amixer. When the application operates the added control element, events are delivered to the server program. Then the server program communicates with the device for the operation. Now instead of Python 3, I select Rust language to write the server programs for audio and music units on IEEE 1394 bus. The hinawa-rs helps the third and fourth of the above tasks. For the first and second task, I wrote some libraries in alsa-gobject project[5]. The demonstration by Python 3 is public[6]. I already write alsa-gobject-rs which includes Rust bindings for the libraries. It's not public yet since I make no official releases of alsa-gobject itself. At the same time, I already write the prototype of server programs itself and it works well in my local. The code base already reaches 28,000 lines even if it supports the limited number of devices. I realized again that it was a better decision not to implement them in kernel land. As IEEE 1394 bus becomes legacy, the audio and music units is outdated. One of my motivation to continue the development is the idea of control service in use space itself since the idea is also available for USB Audio as well. When developers work for devices with vendor-dependent protocol, the idea helps the developer because user space applications can communicate with the device via USB character device. For example, I recently saw an attempt by Guy Sherman for US-4x4 Utils[7]. As a quick glance to his repository[8], vendor-dependent command is used to operate the device. For this kind of work, the idea is available as well to produce transparent control element set. Anyway I'm going to make official release of alsa-gobject as a next step toward the idea. [1] https://github.com/takaswie/hinawa-rs [2] https://github.com/takaswie/libhinawa [3] https://www.alsa-project.org/pipermail/alsa-devel/2018-August/139448.html [4] https://github.com/takaswie/presentations/blob/master/20161101/content.md [5] https://github.com/alsa-project/alsa-gobject [6] 'self.open(card_id)' should be 'self.open(card_id, 0)' now https://mailman.alsa-project.org/pipermail/alsa-devel/2019-December/159339.… [7] https://lists.linuxaudio.org/archives/linux-audio-dev/2020-May/037774.html [8] https://github.com/guysherman/tascam-util Regards Takashi Sakamoto

2 3

alsa-gobject: seq: fulfill documentation for ALSASeq
by GitHub pull_request - opened 06 Jun '20

06 Jun '20

alsa-project/alsa-gobject pull request #24 was opened from takaswie: This patch fulfills documentation for ALSASeq. Request URL : https://github.com/alsa-project/alsa-gobject/pull/24 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/24.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

alsa-gobject: hwdep/timer/rawmidi: fulfill documentation for ALSAHwdep/ALSATimer/ALSARawmidi
by GitHub pull_request - opened 06 Jun '20

06 Jun '20

alsa-project/alsa-gobject pull request #23 was opened from takaswie: This patchset fulfills documentation for ALSAHwdep/ALSATimer/ALSARawmidi, except for ALSASeq, including some documentation improvements for ALSACtl. Request URL : https://github.com/alsa-project/alsa-gobject/pull/23 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/23.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

detailed documentation for plug plugin?
by D. R. Evans 06 Jun '20

06 Jun '20

This is a follow-on to the topic started in e-mail <62e4a031-f175-70bc-099f-441fa0623a9f(a)gmail.com> on the alsa-users reflector; I also posted this e-mail on that reflector, but I suspect that there are more people here who would immediately understand the behaviour described below, and can point me to the right documentation. -------- I have a machine on which the following works for reading audio: device: default [which is actually hw:0,0,0] rate: 44100 [which is all that the raw device supports] And the following does not work: device: plug:default rate: 8000 Everything else is identical in the two scenarios. So I am trying to find detailed documentation about the "plug" plugin, to see why it doesn't seem to do what the minimal description I've been able to find suggests. That is, from what little I've been able to find, it seemed like the second scenario should work if the first one does: behind the scenes it would take the 44100 output from the default device and downsample to 8000, and make that available to the snd_pcm_readX() functions. But that is not what's happening, since the second scenario fails: snd_pcm_readX() blocks (even in non-blocking mode). Somewhere there has to be a detailed, technical description of precisely what the plug plugin does; can someone please provide a suggestion where to look? Doc -- Web: http://enginehousebooks.com/drevans

1 0

[RFC][PATCH] slimbus: core: Set fwnode for a device when setting of_node
by John Stultz 06 Jun '20

06 Jun '20

From: Saravana Kannan <saravanak(a)google.com> When setting the of_node for a newly created device, also set the fwnode. This allows fw_devlink to work for slimbus devices. Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Saravana Kannan <saravanak(a)google.com> Signed-off-by: John Stultz <john.stultz(a)linaro.org> --- drivers/slimbus/core.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/drivers/slimbus/core.c b/drivers/slimbus/core.c index 526e3215d8fe..44228a5b246d 100644 --- a/drivers/slimbus/core.c +++ b/drivers/slimbus/core.c @@ -163,8 +163,10 @@ static int slim_add_device(struct slim_controller *ctrl, INIT_LIST_HEAD(&sbdev->stream_list); spin_lock_init(&sbdev->stream_list_lock); - if (node) + if (node) { sbdev->dev.of_node = of_node_get(node); + sbdev->dev.fwnode = of_fwnode_handle(node); + } dev_set_name(&sbdev->dev, "%x:%x:%x:%x", sbdev->e_addr.manf_id, -- 2.17.1

2 2

[PATCH] ALSA: emu10k1: delete an unnecessary condition
by Dan Carpenter 05 Jun '20

05 Jun '20

The "val" variable is an unsigned int so it's always <= UINT_MAX. This check is always true so it can be removed. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/pci/emu10k1/emu10k1x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index ddb7c2ce3f7c9..def8161cde4c2 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -1040,7 +1040,7 @@ static void snd_emu10k1x_proc_reg_write(struct snd_info_entry *entry, if (sscanf(line, "%x %x %x", &reg, &channel_id, &val) != 3) continue; - if (reg < 0x49 && val <= 0xffffffff && channel_id <= 2) + if (reg < 0x49 && channel_id <= 2) snd_emu10k1x_ptr_write(emu, reg, channel_id, val); } } -- 2.26.2

2 1

[PATCH 0/2] alsa-lib: -ENODATA documentation
by Kai Vehmanen 05 Jun '20

05 Jun '20

Hello all, here's patch to document -ENODATA error as a PCM error code. We have recently merged patches to SOF firmware to use this error code: https://github.com/thesofproject/sof/pull/2918 Some additional background on the use-case: https://github.com/thesofproject/sof/issues/2564 In thread "Functionality of pcm_notify in snd-aloop?" https://mailman.alsa-project.org/pipermail/alsa-devel/2020-April/166974.html .. we discussed the possibility for ALSA to provide a mechanism for application to wait for the condition to change, but at least for the currently known usages, the synchronization responsibility will be in user-space (UCM file will describe the PCM dependencies). Anyways, this is orthogonal to documenting -ENODATA, so sending this now. Kai Vehmanen (2): pcm: add documentation for -ENODATA error code pcm: fix spelling in documentation for -EBADFD src/pcm/pcm.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) -- 2.26.2

2 3

[v2, 1/2] ASoC: mediatek: mt6358: support DMIC one-wire mode
by Jiaxin Yu 05 Jun '20

05 Jun '20

Supports DMIC one-wire mode. Uses a DT property "dmic-mode" to select. Signed-off-by: Jiaxin Yu <jiaxin.yu(a)mediatek.com> --- sound/soc/codecs/mt6358.c | 23 ++++++++++++++++++++++- 1 file changed, 22 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/mt6358.c b/sound/soc/codecs/mt6358.c index 1b830ea..1f39d59 100644 --- a/sound/soc/codecs/mt6358.c +++ b/sound/soc/codecs/mt6358.c @@ -95,6 +95,8 @@ struct mt6358_priv { struct regulator *avdd_reg; int wov_enabled; + + unsigned int dmic_one_wire_mode; }; int mt6358_set_mtkaif_protocol(struct snd_soc_component *cmpnt, @@ -1831,7 +1833,10 @@ static int mt6358_dmic_enable(struct mt6358_priv *priv) mt6358_mtkaif_tx_enable(priv); /* UL dmic setting */ - regmap_write(priv->regmap, MT6358_AFE_UL_SRC_CON0_H, 0x0080); + if (priv->dmic_one_wire_mode) + regmap_write(priv->regmap, MT6358_AFE_UL_SRC_CON0_H, 0x0400); + else + regmap_write(priv->regmap, MT6358_AFE_UL_SRC_CON0_H, 0x0080); /* UL turn on */ regmap_write(priv->regmap, MT6358_AFE_UL_SRC_CON0_L, 0x0003); @@ -2426,6 +2431,20 @@ static int mt6358_codec_probe(struct snd_soc_component *cmpnt) .num_dapm_routes = ARRAY_SIZE(mt6358_dapm_routes), }; +static void mt6358_parse_dt(struct mt6358_priv *priv) +{ + int ret; + struct device *dev = priv->dev; + + ret = of_property_read_u32(dev->of_node, "mediatek,dmic-mode", + &priv->dmic_one_wire_mode); + if (ret) { + dev_warn(priv->dev, "%s() failed to read dmic-mode\n", + __func__); + priv->dmic_one_wire_mode = 0; + } +} + static int mt6358_platform_driver_probe(struct platform_device *pdev) { struct mt6358_priv *priv; @@ -2445,6 +2464,8 @@ static int mt6358_platform_driver_probe(struct platform_device *pdev) if (IS_ERR(priv->regmap)) return PTR_ERR(priv->regmap); + mt6358_parse_dt(priv); + dev_info(priv->dev, "%s(), dev name %s\n", __func__, dev_name(&pdev->dev)); -- 1.8.1.1.dirty

1 1

[PATCH v2] ALSA: usb-audio: Manage auto-pm of all bundled interfaces
by Takashi Iwai 05 Jun '20

05 Jun '20

Currently USB-audio driver manages the auto-pm of the primary interface although a card may consist of multiple interfaces. This may leave the secondary and other interfaces left running unnecessarily after the auto-suspend. This patch allows the driver managing the auto-pm of all bundled interfaces per card. The chip->pm_intf field is extended as chip->intf[] to contain the array of assigned interfaces, and the runtime-PM is performed to all those interfaces. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- v1->v2: fix a typo that caused a build error sound/usb/card.c | 35 ++++++++++++++++++++++++++++++----- sound/usb/usbaudio.h | 4 +++- 2 files changed, 33 insertions(+), 6 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 359f7a04be1c..162bdd6eb4d4 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -634,7 +634,6 @@ static int usb_audio_probe(struct usb_interface *intf, id, &chip); if (err < 0) goto __error; - chip->pm_intf = intf; break; } else if (vid[i] != -1 || pid[i] != -1) { dev_info(&dev->dev, @@ -651,6 +650,13 @@ static int usb_audio_probe(struct usb_interface *intf, goto __error; } } + + if (chip->num_interfaces >= MAX_CARD_INTERFACES) { + dev_info(&dev->dev, "Too many interfaces assigned to the single USB-audio card\n"); + err = -EINVAL; + goto __error; + } + dev_set_drvdata(&dev->dev, chip); /* @@ -703,6 +709,7 @@ static int usb_audio_probe(struct usb_interface *intf, } usb_chip[chip->index] = chip; + chip->intf[chip->num_interfaces] = intf; chip->num_interfaces++; usb_set_intfdata(intf, chip); atomic_dec(&chip->active); @@ -818,19 +825,37 @@ void snd_usb_unlock_shutdown(struct snd_usb_audio *chip) int snd_usb_autoresume(struct snd_usb_audio *chip) { + int i, err; + if (atomic_read(&chip->shutdown)) return -EIO; - if (atomic_inc_return(&chip->active) == 1) - return usb_autopm_get_interface(chip->pm_intf); + if (atomic_inc_return(&chip->active) != 1) + return 0; + + for (i = 0; i < chip->num_interfaces; i++) { + err = usb_autopm_get_interface(chip->intf[i]); + if (err < 0) { + /* rollback */ + while (--i >= 0) + usb_autopm_put_interface(chip->intf[i]); + atomic_dec(&chip->active); + return err; + } + } return 0; } void snd_usb_autosuspend(struct snd_usb_audio *chip) { + int i; + if (atomic_read(&chip->shutdown)) return; - if (atomic_dec_and_test(&chip->active)) - usb_autopm_put_interface(chip->pm_intf); + if (!atomic_dec_and_test(&chip->active)) + return; + + for (i = 0; i < chip->num_interfaces; i++) + usb_autopm_put_interface(chip->intf[i]); } static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index e0ebfb25fbd5..b91c4c0807ec 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -19,11 +19,13 @@ struct media_device; struct media_intf_devnode; +#define MAX_CARD_INTERFACES 16 + struct snd_usb_audio { int index; struct usb_device *dev; struct snd_card *card; - struct usb_interface *pm_intf; + struct usb_interface *intf[MAX_CARD_INTERFACES]; u32 usb_id; struct mutex mutex; unsigned int system_suspend; -- 2.16.4

1 0

[sound:for-next 141/141] sound/usb/card.c:841:29: error: expected ';' after expression
by kernel test robot 04 Jun '20

04 Jun '20

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: 3398e5c7b0383a1795090d76545884091636644c commit: 3398e5c7b0383a1795090d76545884091636644c [141/141] ALSA: usb-audio: Manage auto-pm of all bundled interfaces config: x86_64-allyesconfig (attached as .config) compiler: clang version 11.0.0 (https://github.com/llvm/llvm-project ac47588bc4ff5927a01ed6fcd269ce86aba52a7c) reproduce (this is a W=1 build): wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross # install x86_64 cross compiling tool for clang build # apt-get install binutils-x86-64-linux-gnu git checkout 3398e5c7b0383a1795090d76545884091636644c # save the attached .config to linux build tree COMPILER_INSTALL_PATH=$HOME/0day COMPILER=clang make.cross ARCH=x86_64 If you fix the issue, kindly add following tag as appropriate Reported-by: kernel test robot <lkp(a)intel.com> All errors (new ones prefixed by >>, old ones prefixed by <<): >> sound/usb/card.c:841:29: error: expected ';' after expression atomic_dec(&chip->active)) ^ ; >> sound/usb/card.c:841:29: error: expected expression 2 errors generated. vim +841 sound/usb/card.c 825 826 int snd_usb_autoresume(struct snd_usb_audio *chip) 827 { 828 int i, err; 829 830 if (atomic_read(&chip->shutdown)) 831 return -EIO; 832 if (atomic_inc_return(&chip->active) != 1) 833 return 0; 834 835 for (i = 0; i < chip->num_interfaces; i++) { 836 err = usb_autopm_get_interface(chip->intf[i]); 837 if (err < 0) { 838 /* rollback */ 839 while (--i >= 0) 840 usb_autopm_put_interface(chip->intf[i]); > 841 atomic_dec(&chip->active)) 842 return err; 843 } 844 } 845 return 0; 846 } 847 --- 0-DAY CI Kernel Test Service, Intel Corporation https://lists.01.org/hyperkitty/list/kbuild-all@lists.01.org

1 0

[PATCH] sound/pci/hda: add sienna_cichlid audio asic id for sienna_cichlid up
by Alex Deucher 04 Jun '20

04 Jun '20

From: Hersen Wu <hersenxs.wu(a)amd.com> dp/hdmi ati hda is not shown in audio settings Signed-off-by: Hersen Wu <hersenxs.wu(a)amd.com> Reviewed-by: Alex Deucher <alexander.deucher(a)amd.com> Signed-off-by: Alex Deucher <alexander.deucher(a)amd.com> --- sound/pci/hda/hda_intel.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 92a042e34d3e..4188bc4bd4e2 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2643,6 +2643,9 @@ static const struct pci_device_id azx_ids[] = { { PCI_DEVICE(0x1002, 0xab38), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS | AZX_DCAPS_PM_RUNTIME }, + { PCI_DEVICE(0x1002, 0xab28), + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS | + AZX_DCAPS_PM_RUNTIME }, /* VIA VT8251/VT8237A */ { PCI_DEVICE(0x1106, 0x3288), .driver_data = AZX_DRIVER_VIA }, /* VIA GFX VT7122/VX900 */ -- 2.25.4

2 2

[v2 PATCH] ASoC: max98390: Fix potential crash during param fw loading
by Steve Lee 04 Jun '20

04 Jun '20

malformed firmware file can cause out-of-bound access and crash during dsm_param bin loading. - add MIN/MAX param size to avoid out-of-bound access. - read start addr and size of param and check bound. - add condition that fw->size > param_size + _PAYLOAD_OFFSET to confirm enough data. Signed-off-by: Steve Lee <steves.lee(a)maximintegrated.com> --- Change log v2: * add condtion that param_size + _PAYLOAD_OFFSET is less than fw->size to confirm enough data * remove unintended code sound/soc/codecs/max98390.c | 24 ++++++++++++++++++++---- sound/soc/codecs/max98390.h | 3 ++- 2 files changed, 22 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c index be7cd0aeb6a6..0d63ebfbff2f 100644 --- a/sound/soc/codecs/max98390.c +++ b/sound/soc/codecs/max98390.c @@ -754,6 +754,7 @@ static struct snd_soc_dai_driver max98390_dai[] = { static int max98390_dsm_init(struct snd_soc_component *component) { int ret; + int param_size, param_start_addr; char filename[128]; const char *vendor, *product; struct max98390_priv *max98390 = @@ -780,14 +781,29 @@ static int max98390_dsm_init(struct snd_soc_component *component) dev_dbg(component->dev, "max98390: param fw size %zd\n", fw->size); + if (fw->size < MAX98390_DSM_PARAM_MIN_SIZE) { + dev_err(component->dev, + "param fw is invalid.\n"); + goto err_alloc; + } dsm_param = (char *)fw->data; + param_start_addr = (dsm_param[0] & 0xff) | (dsm_param[1] & 0xff) << 8; + param_size = (dsm_param[2] & 0xff) | (dsm_param[3] & 0xff) << 8; + if (param_size > MAX98390_DSM_PARAM_MAX_SIZE || + param_start_addr < DSM_STBASS_HPF_B0_BYTE0 || + fw->size < param_size + MAX98390_DSM_PAYLOAD_OFFSET) { + dev_err(component->dev, + "param fw is invalid.\n"); + goto err_alloc; + } + regmap_write(max98390->regmap, MAX98390_R203A_AMP_EN, 0x80); dsm_param += MAX98390_DSM_PAYLOAD_OFFSET; - regmap_bulk_write(max98390->regmap, DSM_EQ_BQ1_B0_BYTE0, - dsm_param, - fw->size - MAX98390_DSM_PAYLOAD_OFFSET); - release_firmware(fw); + regmap_bulk_write(max98390->regmap, param_start_addr, + dsm_param, param_size); regmap_write(max98390->regmap, MAX98390_R23E1_DSP_GLOBAL_EN, 0x01); +err_alloc: + release_firmware(fw); err: return ret; } diff --git a/sound/soc/codecs/max98390.h b/sound/soc/codecs/max98390.h index f59cb114d957..5f444e7779b0 100644 --- a/sound/soc/codecs/max98390.h +++ b/sound/soc/codecs/max98390.h @@ -650,7 +650,8 @@ /* DSM register offset */ #define MAX98390_DSM_PAYLOAD_OFFSET 16 -#define MAX98390_DSM_PAYLOAD_OFFSET_2 495 +#define MAX98390_DSM_PARAM_MAX_SIZE 770 +#define MAX98390_DSM_PARAM_MIN_SIZE 670 struct max98390_priv { struct regmap *regmap; -- 2.17.1

2 1

[PATCH] ASoC: rl6231: Modify the target DMIC clock rate
by Oder Chiou 04 Jun '20

04 Jun '20

Some DMIC components will not work correctly in the clock rate 3.072MHz. We recommend the clock rate 1.536MHz in the gerenal case. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rl6231.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rl6231.c b/sound/soc/codecs/rl6231.c index 2586d1cafc0c..8c9daf32bab8 100644 --- a/sound/soc/codecs/rl6231.c +++ b/sound/soc/codecs/rl6231.c @@ -80,8 +80,8 @@ int rl6231_calc_dmic_clk(int rate) for (i = 0; i < ARRAY_SIZE(div); i++) { if ((div[i] % 3) == 0) continue; - /* find divider that gives DMIC frequency below 3.072MHz */ - if (3072000 * div[i] >= rate) + /* find divider that gives DMIC frequency below 1.536MHz */ + if (1536000 * div[i] >= rate) return i; } -- 2.25.1

2 1

[PATCH] ASoC: max98390: Fix incorrect printf qualifier
by Takashi Iwai 04 Jun '20

04 Jun '20

This patch addresses a compile warning: sound/soc/codecs/max98390.c:781:3: warning: format ‘%ld’ expects argument of type ‘long int’, but argument 4 has type ‘size_t {aka const unsigned int}’ [-Wformat=] Fixes: a6e3f4f34cdb ("ASoC: max98390: Added Amplifier Driver") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/codecs/max98390.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c index b9ce44dda886..be7cd0aeb6a6 100644 --- a/sound/soc/codecs/max98390.c +++ b/sound/soc/codecs/max98390.c @@ -778,7 +778,7 @@ static int max98390_dsm_init(struct snd_soc_component *component) } dev_dbg(component->dev, - "max98390: param fw size %ld\n", + "max98390: param fw size %zd\n", fw->size); dsm_param = (char *)fw->data; dsm_param += MAX98390_DSM_PAYLOAD_OFFSET; -- 2.16.4

2 2

[PATCH] ASoC: fsl-asoc-card: Defer probe when fail to find codec device
by Shengjiu Wang 04 Jun '20

04 Jun '20

Defer probe when fail to find codec device, because the codec device maybe probed later than machine driver. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl-asoc-card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index cf4feb835743..00be73900888 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -581,7 +581,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) if (!fsl_asoc_card_is_ac97(priv) && !codec_dev) { dev_err(&pdev->dev, "failed to find codec device\n"); - ret = -EINVAL; + ret = -EPROBE_DEFER; goto asrc_fail; } -- 2.21.0

2 1

Re: [PATCH] ASoC: max98390: fix build warning detected by -Wformat
by Mark Brown 04 Jun '20

04 Jun '20

On Thu, Jun 04, 2020 at 04:49:34PM +0800, Xiaoming Ni wrote: > Fix build warning: > sound/soc/codecs/max98390.c:781:3: warning: format '%ld' expects > argument of type 'long int', but argument 4 has type 'size_t {aka > const unsigned int}' [-Wformat=] Thanks but this was already fixed.

1 0

alsa-gobject: ctl: fulfill documentation for ALSACtl
by GitHub pull_request - opened 04 Jun '20

04 Jun '20

alsa-project/alsa-gobject pull request #22 was opened from takaswie: This patch fulfills alsactl documentation for GObject-derived objects and boxed objects. Request URL : https://github.com/alsa-project/alsa-gobject/pull/22 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/22.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

alsa-project alsa-ucm : Audio endpoint is not switching when external mic connected
by Mukunda, Vijendar 04 Jun '20

04 Jun '20

We recently up streamed ACP PDM driver for Renoir APU. Our ACP IP has an internal PDM Decoder block where DMIC is directly connected. There are no KControls defined for PDM Driver. Our platform supports on board DMIC and South bridge Azalia (HD Audio ) endpoint . By default, DMIC endpoint is used as audio input device. When Headset Mic jack connected, it's not switching to External Mic. How does audio endpoint will switch in gnome when external mic is connected? >From our understanding, UCM conf file for HDA driver is also required for configuring mixer controls for HD Audio capture device. Similarly when External Mic is removed , it has to switch to internal dmic . Need help in resolving this issue. Thanks, Vijendar

4 11

UCM file required for Multiple sound card scenario for Renoir APU
by GitHub issues - opened 04 Jun '20

04 Jun '20

alsa-project/alsa-ucm-conf issue #30 was opened from VijendarReddy: We recently upstreamed PDM Driver for Renoir APU. Renoir APU platform has Onboard DMIC + Southbridge AZ (HD Audio Endpoint) support. When External Mic connected, Device switching should happen from Internal DMIC to external Mic. Need UCM file to support Multiple sound cards for Renoir APU. Alsa info link: https://pastebin.ubuntu.com/p/4kprhDZYbg/ Created Ticket based on discussion on the below-mentioned thread. https://www.alsa-project.org/pipermail/alsa-devel/2020-June/168527.html Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/30 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 00/24] ASoC: soc-component: collect component functions
by Kuninori Morimoto 04 Jun '20

04 Jun '20

Hi Mark We have soc-component.c now, but still many component related functions are implemented many place. This patch-set collect these into soc-component.c. Kuninori Morimoto (24): ASoC: soc-component: add soc_component_pin() and share code ASoC: soc-component: move snd_soc_component_xxx_regmap() to soc-component ASoC: soc-component: move snd_soc_component_initialize() to soc-component.c ASoC: soc-component: add soc_component_err() ASoC: soc-component: add snd_soc_pcm_component_prepare() ASoC: soc-component: add snd_soc_pcm_component_hw_params() ASoC: soc-component: add snd_soc_pcm_component_hw_free() ASoC: soc-component: add snd_soc_pcm_component_trigger() ASoC: soc-component: add snd_soc_component_compr_open() ASoC: soc-component: add snd_soc_component_compr_free() ASoC: soc-component: add snd_soc_component_compr_trigger() ASoC: soc-component: add snd_soc_component_compr_set_params() ASoC: soc-component: add snd_soc_component_compr_get_params() ASoC: soc-component: add snd_soc_component_compr_get_caps() ASoC: soc-component: add snd_soc_component_compr_get_codec_caps() ASoC: soc-component: add snd_soc_component_compr_ack() ASoC: soc-component: add snd_soc_component_compr_pointer() ASoC: soc-component: add snd_soc_component_compr_copy() ASoC: soc-component: add snd_soc_component_compr_set_metadata() ASoC: soc-component: add snd_soc_component_compr_get_metadata() ASoC: soc-component: add snd_soc_component_init() ASoC: soc-component: merge soc-io.c into soc-component.c ASoC: soc-component: merge soc_pcm_trigger_start/stop() ASoC: soc-component: tidyup Copyright include/sound/soc-component.h | 51 +- sound/soc/Makefile | 2 +- sound/soc/soc-component.c | 912 ++++++++++++++++++++++++++-------- sound/soc/soc-compress.c | 223 ++------- sound/soc/soc-core.c | 102 +--- sound/soc/soc-io.c | 202 -------- sound/soc/soc-pcm.c | 115 ++--- 7 files changed, 830 insertions(+), 777 deletions(-) delete mode 100644 sound/soc/soc-io.c -- 2.17.1

3 36

[PATCH 1/2] ALSA: usb-audio: Fix inconsistent card PM state after resume
by Takashi Iwai 04 Jun '20

04 Jun '20

When a USB-audio interface gets runtime-suspended via auto-pm feature, the driver suspends all functionality and increment chip->num_suspended_intf. Later on, when the system gets suspended to S3, the driver increments chip->num_suspended_intf again, skips the device changes, and sets the card power state to SNDRV_CTL_POWER_D3hot. In return, when the system gets resumed from S3, the resume callback decrements chip->num_suspended_intf. Since this refcount is still not zero (it's been runtime-suspended), the whole resume is skipped. But there is a small pitfall here. The problem is that the driver doesn't restore the card power state after this resume call, leaving it as SNDRV_CTL_POWER_D3hot. So, even after the system resume finishes, the card instance still appears as if it were system-suspended, and this confuses many ioctl accesses that are blocked unexpectedly. In details, we have two issues behind the scene: one is that the card power state is changed only when the refcount becomes zero, and another is that the prior auto-suspend check is kept in a boolean flag. Although the latter problem is almost negligible since the auto-pm feature is imposed only on the primary interface, but this can be a potential problem on the devices with multiple interfaces. This patch addresses those issues by the following: - Replace chip->autosuspended boolean flag with chip->system_suspend counter - At the first system-suspend, chip->num_suspended_intf is recorded to chip->system_suspend - At system-resume, the card power state is restored when the chip->num_suspended_intf refcount reaches to chip->system_suspend, i.e. the state returns to the auto-suspended Also, the patch fixes yet another hidden problem by the code refactoring along with the fixes above: namely, when some resume procedure failed, the driver left chip->num_suspended_intf that was already decreased, and it might lead to the refcount unbalance. In the new code, the refcount decrement is done after the whole resume procedure, and the problem is avoided as well. Fixes: 0662292aec05 ("ALSA: usb-audio: Handle normal and auto-suspend equally") Reported-and-tested-by: Macpaul Lin <macpaul.lin(a)mediatek.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/card.c | 19 ++++++++++++------- sound/usb/usbaudio.h | 2 +- 2 files changed, 13 insertions(+), 8 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index fd6fd1726ea0..359f7a04be1c 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -843,9 +843,6 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) if (chip == (void *)-1L) return 0; - chip->autosuspended = !!PMSG_IS_AUTO(message); - if (!chip->autosuspended) - snd_power_change_state(chip->card, SNDRV_CTL_POWER_D3hot); if (!chip->num_suspended_intf++) { list_for_each_entry(as, &chip->pcm_list, list) { snd_usb_pcm_suspend(as); @@ -858,6 +855,11 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) snd_usb_mixer_suspend(mixer); } + if (!PMSG_IS_AUTO(message) && !chip->system_suspend) { + snd_power_change_state(chip->card, SNDRV_CTL_POWER_D3hot); + chip->system_suspend = chip->num_suspended_intf; + } + return 0; } @@ -871,10 +873,10 @@ static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume) if (chip == (void *)-1L) return 0; - if (--chip->num_suspended_intf) - return 0; atomic_inc(&chip->active); /* avoid autopm */ + if (chip->num_suspended_intf > 1) + goto out; list_for_each_entry(as, &chip->pcm_list, list) { err = snd_usb_pcm_resume(as); @@ -896,9 +898,12 @@ static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume) snd_usbmidi_resume(p); } - if (!chip->autosuspended) + out: + if (chip->num_suspended_intf == chip->system_suspend) { snd_power_change_state(chip->card, SNDRV_CTL_POWER_D0); - chip->autosuspended = 0; + chip->system_suspend = 0; + } + chip->num_suspended_intf--; err_out: atomic_dec(&chip->active); /* allow autopm after this point */ diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 1c892c7f14d7..e0ebfb25fbd5 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -26,7 +26,7 @@ struct snd_usb_audio { struct usb_interface *pm_intf; u32 usb_id; struct mutex mutex; - unsigned int autosuspended:1; + unsigned int system_suspend; atomic_t active; atomic_t shutdown; atomic_t usage_count; -- 2.25.0

2 3

[PATCH] ASoC: max98390: Fix potential crash during param fw loading
by Steve Lee 04 Jun '20

04 Jun '20

malformed firmware file can cause out-of-bound access and crash during dsm_param bin loading. - add MIN/MAX param size to avoid out-of-bound access. - read start addr and size of param and check bound. Signed-off-by: Steve Lee <steves.lee(a)maximintegrated.com> --- sound/soc/codecs/max98390.c | 23 ++++++++++++++++++----- sound/soc/codecs/max98390.h | 3 ++- 2 files changed, 20 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c index be7cd0aeb6a6..8c7db568fe64 100644 --- a/sound/soc/codecs/max98390.c +++ b/sound/soc/codecs/max98390.c @@ -754,6 +754,7 @@ static struct snd_soc_dai_driver max98390_dai[] = { static int max98390_dsm_init(struct snd_soc_component *component) { int ret; + int param_size, param_start_addr; char filename[128]; const char *vendor, *product; struct max98390_priv *max98390 = @@ -780,14 +781,27 @@ static int max98390_dsm_init(struct snd_soc_component *component) dev_dbg(component->dev, "max98390: param fw size %zd\n", fw->size); + if (fw->size < MAX98390_DSM_PARAM_MIN_SIZE) { + dev_err(component->dev, + "param fw is invalid.\n"); + goto err_alloc; + } dsm_param = (char *)fw->data; + param_start_addr = (dsm_param[0] & 0xff) | (dsm_param[1] & 0xff) << 8; + param_size = (dsm_param[2] & 0xff) | (dsm_param[3] & 0xff) << 8; + if (param_size > MAX98390_DSM_PARAM_MAX_SIZE || + param_start_addr < DSM_STBASS_HPF_B0_BYTE0) { + dev_err(component->dev, + "param fw is invalid.\n"); + goto err_alloc; + } dsm_param += MAX98390_DSM_PAYLOAD_OFFSET; - regmap_bulk_write(max98390->regmap, DSM_EQ_BQ1_B0_BYTE0, - dsm_param, - fw->size - MAX98390_DSM_PAYLOAD_OFFSET); - release_firmware(fw); + regmap_bulk_write(max98390->regmap, param_start_addr, + dsm_param, param_size); regmap_write(max98390->regmap, MAX98390_R23E1_DSP_GLOBAL_EN, 0x01); +err_alloc: + release_firmware(fw); err: return ret; } @@ -847,7 +861,6 @@ static int max98390_probe(struct snd_soc_component *component) /* Dsm Setting */ regmap_write(max98390->regmap, DSM_VOL_CTRL, 0x94); - regmap_write(max98390->regmap, DSMIG_EN, 0x19); regmap_write(max98390->regmap, MAX98390_R203A_AMP_EN, 0x80); if (max98390->ref_rdc_value) { regmap_write(max98390->regmap, DSM_TPROT_RECIP_RDC_ROOM_BYTE0, diff --git a/sound/soc/codecs/max98390.h b/sound/soc/codecs/max98390.h index f59cb114d957..5f444e7779b0 100644 --- a/sound/soc/codecs/max98390.h +++ b/sound/soc/codecs/max98390.h @@ -650,7 +650,8 @@ /* DSM register offset */ #define MAX98390_DSM_PAYLOAD_OFFSET 16 -#define MAX98390_DSM_PAYLOAD_OFFSET_2 495 +#define MAX98390_DSM_PARAM_MAX_SIZE 770 +#define MAX98390_DSM_PARAM_MIN_SIZE 670 struct max98390_priv { struct regmap *regmap; -- 2.17.1

3 6

[RESEND PATCH v1 0/6] Enable Greybus Audio codec driver
by Vaibhav Agarwal 03 Jun '20

03 Jun '20

[REQUEST] This patch series intends to "Enable Greybus Audio codec driver" existing in the staging tree. I have shared the original patch series with Greybus-Dev mailing list and as per the suggestion from Alexandre, I'm also interested to push Greybus Audio to sound soc tree. Thus, now I'm resending it to ASoc maintainers for review. Following is the top level SW design for GB Codec driver and GB Audio modules. +--------------+ +-------------->+GBA Module 1 | | +--------------+ +-----------------------+ | | | | | Greybus | +--------------+ | SND SOC | Audio +-------------->+GBA Module 2 | | Framework | Codec | +--------------+ | | Driver | | | | +-----------------------+ +--------------+ +-------------->+GBA Module N | +--------------+ Patch 5 contains the changes to provide helper APIs to link DAPM DAI widgets for the module added and remove/free component controls for the module removed dynamically. Eventually, I propose to extend snd_soc_xxx APIs for this purpose. Kindly share your inputs. [COVER LETTER] The existing GB Audio codec driver is dependent on MSM8994 Audio driver. During the development stage, this depdency was configured due to various changes involved in MSM Audio driver to enable addtional codec card and some of the changes proposed in mainline ASoC framework. However, these are not the real dependencies and some of them can be easily removed. The folowing patch series includes the changes to resolve unnecessary depedencies and make the codec driver functional with the latest kernel. Patch 1,2: Incudes jack framework related changes. Patch 3,4,5: Resolves compilation error observed with the latest kernel and also provides helper APIs required to allow dynamic addition/removal of modules. Patch 6: Finally provides config options and related Makefile changes to enable GB Codec driver. Thanks to Alexandre for raising the headsup [1] and motivating me to provide the necessary changes. [1] https://lore.kernel.org/lkml/20200507212912.599433-1-alexandre.belloni@boot… Vaibhav Agarwal (6): staging: greybus: audio: Update snd_jack FW usage as per new APIs staging: greybus: audio: Maintain jack list within GB Audio module staging: greybus: audio: Resolve compilation errors for GB codec module staging: greybus: audio: Resolve compilation error in topology parser staging: greybus: audio: Add helper APIs for dynamic audio modules staging: greybus: audio: Enable GB codec, audio module compilation. drivers/staging/greybus/Kconfig | 14 +- drivers/staging/greybus/Makefile | 6 +- drivers/staging/greybus/audio_codec.c | 187 +++++++++++++-------- drivers/staging/greybus/audio_codec.h | 12 +- drivers/staging/greybus/audio_helper.c | 197 +++++++++++++++++++++++ drivers/staging/greybus/audio_helper.h | 17 ++ drivers/staging/greybus/audio_module.c | 20 +-- drivers/staging/greybus/audio_topology.c | 130 +++++++-------- 8 files changed, 427 insertions(+), 156 deletions(-) create mode 100644 drivers/staging/greybus/audio_helper.c create mode 100644 drivers/staging/greybus/audio_helper.h base-commit: ae73e7784871ebe2c43da619b4a1e2c9ff81508d -- 2.26.2

2 13

[PATCH] fix snd_pcm_drain() excluding SETUP state from valid states
by sylvain.bertrand＠gmail.com 03 Jun '20

03 Jun '20

once draining is done, the pcm enters the SETUP state, which ought to be valid for snd_pcm_drain() signed-off-by: Sylvain BERTRAND <sylvain.bertrand(a)legeek.net> --- I missed this one in my previous patch because exiting with or without an error once draining is done was producing the same result. --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -1329,7 +1329,7 @@ int snd_pcm_drain(snd_pcm_t *pcm) SNDMSG("PCM not set up"); return -EIO; } - err = bad_pcm_state(pcm, P_STATE_RUNNABLE); + err = bad_pcm_state(pcm, P_STATE_RUNNABLE | P_STATE(SETUP)); if (err < 0) return err; /* lock handled in the callback */

2 6

[PATCH] sound: usb: pcm: fix incorrect power state when playing sound after PM_AUTO suspend
by Macpaul Lin 03 Jun '20

03 Jun '20

This patch fix incorrect power state changed by usb_audio_suspend() when CONFIG_PM is enabled. After receiving suspend PM message with auto flag, usb_audio_suspend() change card's power state to SNDRV_CTL_POWER_D3hot. Only when the other resume PM message with auto flag can change power state to SNDRV_CTL_POWER_D0 in __usb_audio_resume(). However, when system is not under auto suspend, resume PM message with auto flag might not be able to receive on time which cause the power state was incorrect. At this time, if a player starts to play sound, will cause snd_usb_pcm_open() to access the card and setup_hw_info() will resume the card. But even the card is back to work and all function normal, the power state is still in SNDRV_CTL_POWER_D3hot. Which cause the infinite loop happened in snd_power_wait() to check the power state. Thus the successive setting ioctl cannot be passed to card. Hence we suggest to change power state to SNDRV_CTL_POWER_D0 when card has been resumed successfully. Signed-off-by: Macpaul Lin <macpaul.lin(a)mediatek.com> --- sound/usb/pcm.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index a4e4064..d667ecb 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1322,6 +1322,17 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre if (err < 0) return err; + /* fix incorrect power state when resuming by open and later ioctls */ + if (IS_ENABLED(CONFIG_PM) && + snd_power_get_state(subs->stream->chip->card) + == SNDRV_CTL_POWER_D3hot) { + /* set these variables for power state correction */ + subs->stream->chip->autosuspended = 0; + subs->stream->chip->num_suspended_intf = 1; + dev_info(&subs->dev->dev, + "change power state from D3hot to D0\n"); + } + return snd_usb_autoresume(subs->stream->chip); } -- 1.7.9.5

2 9

[PATCH v2 0/2] powerpc: Remove support for ppc405/440 Xilinx platforms
by Michal Simek 03 Jun '20

03 Jun '20

Hi, recently we wanted to update xilinx intc driver and we found that function which we wanted to remove is still wired by ancient Xilinx PowerPC platforms. Here is the thread about it. https://lore.kernel.org/linux-next/48d3232d-0f1d-42ea-3109-f44bbabfa2e8@xil… I have been talking about it internally and there is no interest in these platforms and it is also orphan for quite a long time. None is really running/testing these platforms regularly that's why I think it makes sense to remove them also with drivers which are specific to this platform. U-Boot support was removed in 2017 without anybody complain about it https://github.com/Xilinx/u-boot-xlnx/commit/98f705c9cefdfdba62c069821bbba1… Based on current ppc/next. If anyone has any objection about it, please let me know. Thanks, Michal Changes in v2: - Remove also sound/drivers/pcm-indirect2.[ch] files Reported-by: Takashi Iwai <tiwai(a)suse.de> - Based on my chat with Arnd I removed arch/powerpc/xmon/ changes done in v1 to keep them the same as before. (kbuild reported some issues with it too) Michal Simek (2): sound: ac97: Remove sound driver for ancient platform powerpc: Remove Xilinx PPC405/PPC440 support Documentation/devicetree/bindings/xilinx.txt | 143 -- Documentation/powerpc/bootwrapper.rst | 28 +- MAINTAINERS | 6 - arch/powerpc/Kconfig.debug | 2 +- arch/powerpc/boot/Makefile | 7 +- arch/powerpc/boot/dts/Makefile | 1 - arch/powerpc/boot/dts/virtex440-ml507.dts | 406 ------ arch/powerpc/boot/dts/virtex440-ml510.dts | 466 ------- arch/powerpc/boot/ops.h | 1 - arch/powerpc/boot/serial.c | 5 - arch/powerpc/boot/uartlite.c | 79 -- arch/powerpc/boot/virtex.c | 97 -- arch/powerpc/boot/virtex405-head.S | 31 - arch/powerpc/boot/wrapper | 8 - arch/powerpc/configs/40x/virtex_defconfig | 75 - arch/powerpc/configs/44x/virtex5_defconfig | 74 - arch/powerpc/configs/ppc40x_defconfig | 8 - arch/powerpc/configs/ppc44x_defconfig | 8 - arch/powerpc/include/asm/xilinx_intc.h | 16 - arch/powerpc/include/asm/xilinx_pci.h | 21 - arch/powerpc/kernel/cputable.c | 39 - arch/powerpc/platforms/40x/Kconfig | 31 - arch/powerpc/platforms/40x/Makefile | 1 - arch/powerpc/platforms/40x/virtex.c | 54 - arch/powerpc/platforms/44x/Kconfig | 37 - arch/powerpc/platforms/44x/Makefile | 2 - arch/powerpc/platforms/44x/virtex.c | 60 - arch/powerpc/platforms/44x/virtex_ml510.c | 30 - arch/powerpc/platforms/Kconfig | 4 - arch/powerpc/sysdev/Makefile | 2 - arch/powerpc/sysdev/xilinx_intc.c | 88 -- arch/powerpc/sysdev/xilinx_pci.c | 132 -- drivers/char/Kconfig | 2 +- drivers/video/fbdev/Kconfig | 2 +- sound/drivers/Kconfig | 12 - sound/drivers/Makefile | 2 - sound/drivers/ml403-ac97cr.c | 1298 ------------------ sound/drivers/pcm-indirect2.c | 560 -------- sound/drivers/pcm-indirect2.h | 127 -- 39 files changed, 7 insertions(+), 3958 deletions(-) delete mode 100644 arch/powerpc/boot/dts/virtex440-ml507.dts delete mode 100644 arch/powerpc/boot/dts/virtex440-ml510.dts delete mode 100644 arch/powerpc/boot/uartlite.c delete mode 100644 arch/powerpc/boot/virtex.c delete mode 100644 arch/powerpc/boot/virtex405-head.S delete mode 100644 arch/powerpc/configs/40x/virtex_defconfig delete mode 100644 arch/powerpc/configs/44x/virtex5_defconfig delete mode 100644 arch/powerpc/include/asm/xilinx_intc.h delete mode 100644 arch/powerpc/include/asm/xilinx_pci.h delete mode 100644 arch/powerpc/platforms/40x/virtex.c delete mode 100644 arch/powerpc/platforms/44x/virtex.c delete mode 100644 arch/powerpc/platforms/44x/virtex_ml510.c delete mode 100644 arch/powerpc/sysdev/xilinx_intc.c delete mode 100644 arch/powerpc/sysdev/xilinx_pci.c delete mode 100644 sound/drivers/ml403-ac97cr.c delete mode 100644 sound/drivers/pcm-indirect2.c delete mode 100644 sound/drivers/pcm-indirect2.h -- 2.26.0

5 12

[PATCH] ALSA: es1688: Add the missed snd_card_free()
by Chuhong Yuan 03 Jun '20

03 Jun '20

snd_es968_pnp_detect() misses a snd_card_free() in a failed path. Add the missed function call to fix it. Fixes: a20971b201ac ("ALSA: Merge es1688 and es968 drivers") Signed-off-by: Chuhong Yuan <hslester96(a)gmail.com> --- sound/isa/es1688/es1688.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c index ff3a05ad99c0..64610571a5e1 100644 --- a/sound/isa/es1688/es1688.c +++ b/sound/isa/es1688/es1688.c @@ -267,8 +267,10 @@ static int snd_es968_pnp_detect(struct pnp_card_link *pcard, return error; } error = snd_es1688_probe(card, dev); - if (error < 0) + if (error < 0) { + snd_card_free(card); return error; + } pnp_set_card_drvdata(pcard, card); snd_es968_pnp_is_probed = 1; return 0; -- 2.26.2

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[GIT PULL] ASoC updates for v5.8
by Mark Brown 03 Jun '20

03 Jun '20

The following changes since commit 9cb1fd0efd195590b828b9b865421ad345a4a145: Linux 5.7-rc7 (2020-05-24 15:32:54 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v5.8 for you to fetch changes up to 358c7c61fd04d324f83d7968daf8dd9a6ff86a9a: Merge remote-tracking branch 'asoc/for-5.8' into asoc-linus (2020-06-01 13:01:15 +0100) ---------------------------------------------------------------- ASoC: Updates for v5.8 This has been another very active release with a bunch of new drivers, lots of fixes everywhere and continued core improvements from Morimoto-san: - Lots of core cleanups and refactorings from Morimoto-san, factoring out common operations and making the card abstraction more solid. - Continued work on cleaning up and improving the Intel drivers, along with some new platform support for them. - Fixes to make the Marvell SSPA driver work upstream. - Support for AMD Renoir ACP, Dialog DA7212, Freescale EASRC and i.MX8M, Intel Elkhard Lake, Maxim MAX98390, Nuvoton NAU8812 and NAU8814 and Realtek RT1016. ---------------------------------------------------------------- Akshu Agrawal (2): ASoC: amd :High hw_level while simultaneous capture ASoC: amd: raven: Make the driver name consistent across files Amadeusz Sławiński (3): ASoC: topology: Remove unneeded semicolon ASoC: Intel: baytrail: Fix register access ASoC: Intel: Use readq to read 64 bit registers Andy Shevchenko (1): ASoC: Intel: Skylake: Replace guid_copy() with import_guid() Ard Biesheuvel (1): ASoC: cros_ec_codec: switch to library API for SHA-256 Arnd Bergmann (7): ASoC: component: suppress uninitialized-variable warning ASoC: codecs: wm97xx: fix ac97 dependency ASoC: fsl_easrc: mark PM functions __maybe_unused ASoC: rt5682: fix I2C/Soundwire dependencies ASoC: SOF: sort out Kconfig, again ASoC: cros_ec_codec: allocate shash_desc dynamically ASoC: rt5682: split i2c driver into separate module Artur Kloniecki (1): ASoC: SOF: Add XRUN flags field to struct sof_ipc_buffer. Bard Liao (9): ASoC: intel: sof_sdw: init all aggregated codecs ASoC: SOF: topology: fix: handle DAI widget connections properly with multiple CPU DAI's ASoC: SOF: align sof_ipc_dai_alh_params with FW ASoC: SOF: topology: Get ALH rate amd channels from topology ASoC: SOF: topology: fix: parse hda_tokens to &config->hda ASoC: SOF: topology: Get HDA rate and channels from topology ASoC: SOF: topology: send ipc for all found DAIs in sof_set_dai_config ASoC: topology: set component dai_index to ipc dai config dai_index ASoC: SOF: topology: replace sof_link_hda_process by sof_set_dai_config Brent Lu (3): ASoC: bdw-rt5677: add channel constraint ASoC: bdw-rt5650: add channel constraint ASoC: broadwell: add channel constraint Cezary Rojewski (2): ASoC: Intel: haswell: Power transition refactor ASoC: Intel: Skylake: Update description for HDaudio kconfig Charles Keepax (1): ASoC: dapm: Move dai_link widgets to runtime to fix use after free ChenTao (1): soc: fsl_asrc: Make some functions static Chris Garren (1): ASoC: adau7118: Mark the ADAU7118 reset register as volatile Christophe JAILLET (2): ASoC: ux500: mop500: Fix some refcounted resources issues ASoC: ti: omap-mcbsp: Fix an error handling path in 'asoc_mcbsp_probe()' Chuhong Yuan (1): ASoC: sta32x: add missed function calls in error paths Colin Ian King (1): ASoC: fsl_easrc: fix spelling mistake "prefitler" -> "prefilter" Dan Murphy (9): ASoC: tlv320adcx140: Fix mic gain registers ASoC: tlv320adcx140: Fix bias config values ASoC: tlv320adcx140: Add controls for PDM clk ASoC: tlv320adcx140: Add device tree property for PDM edges ASoC: tlv320adcx140: Configure PDM sampling edge dt-bindings: sound: tlv320adcx140: Add GPI config property ASoC: tlv320adcx140: Add support for configuring GPI pins ASoC: tlv320adcx140: Fix warnings when using W=1 ASoC: tlv320adcx140: Fix dt-binding-check issue Daniel Baluta (5): ASoC: SOF: imx: Add i.MX8M HW support ASoC: SOF: Add i.MX8MP device descriptor dt-bindings: dsp: fsl: Add fsl,imx8mp-dsp entry ASoC: SOF: Do nothing when DSP PM callbacks are not set ASoC: SOF: define INFO_ flags in dsp_ops Dinghao Liu (6): ASoC: ti: Fix runtime PM imbalance in omap2_mcbsp_set_clks_src ASoC: tas2552: Fix runtime PM imbalance in tas2552_component_probe ASoC: img-spdif-out: Fix runtime PM imbalance on error ASoC: img-spdif-in: Fix runtime PM imbalance on error ASoC: wm8962: Fix runtime PM imbalance on error ASoC: img-i2s-out: Fix runtime PM imbalance on error Dmitry Osipenko (2): ASoC: tegra: tegra_wm8903: Support nvidia, headset property ASoC: tegra-wm8903: Document new nvidia, headset property Fabio Estevam (2): ASoC: fsl_micfil: Remove unneeded ifdef's ASoC: fsl_micfil: Do not pass irq numbers in comments Geert Uytterhoeven (1): ASoC: Fix misspellings of "Analog Devices" Gustavo A. R. Silva (2): ASoC: soc-core: Replace zero-length array with flexible-array ASoC: Intel: Skylake: Replace zero-length array with flexible-array Hans de Goede (1): ASoC: Intel: bytcr_rt5640: Add quirk for Toshiba Encore WT8-A tablet Hui Wang (1): ASoC: intel/skl/hda - set autosuspend timeout for hda codecs Jack Yu (1): ASoC: rt1015: Add condition to prevent SoC providing bclk in ratio of 50 times of sample rate. Jaroslav Kysela (1): ASoC: snd-sof-intel-hda-common - add hda_model parameter and pass it to HDA codec driver Jaska Uimonen (2): ASoC: SOF: topology: stop parsing when all tokens have been found ASoC: SOF: topology: handle multiple sets of tuple arrays Jason Yan (9): ASoC: wm8994: remove wm1811_snd_controls and mixin_boost_tlv ASoC: wm8991: remove defined but not used 'wm8991_dapm_rxvoice_controls' ASoC: wm8990: remove some defined but unused symbols ASoC: wm8900: remove some defined but not used symbols ASoC: wcd934x: remove unnecessary comparisons to bool ASoC: wcd9335: remove unneeded semicolon ASoC: wcd934x: remove unneeded semicolon ASoC: soc-core: return true, false in snd_soc_volsw_is_stereo() ASoC: ti: remove comparison to bool in omap_mcbsp_dai_set_dai_fmt() Johan Jonker (1): ASoC: rockchip-i2s: add description for rk3308 John Stultz (1): ASoC: qcom: q6asm-dai: kCFI fix Kai Vehmanen (2): ASoC: intel/skl/hda - fix oops on systems without i915 audio codec ASoC: SOF: Intel: hda: fix generic hda codec support Kai-Heng Feng (1): ASoC: SOF: Update correct LED status at the first time usage of update_mute_led() Karol Trzcinski (11): ASoC: SOF: Mark get_ext* function ext_hdr arguments as const ASoC: SOF: Introduce offset in firmware data ASoC: SOF: Introduce extended manifest ASoC: SOF: ext_manifest: parse firmware version ASoC: SOF: ext_manifest: parse windows ASoC: SOF: ext_manifest: parse compiler version ASoC: SOF: loader: Adjust validation condition for fw_offset ASoC: SOF: Introduce extended manifest ASoC: SOF: ext_manifest: parse firmware version ASoC: SOF: ext_manifest: parse windows ASoC: SOF: ext_manifest: parse compiler version Keyon Jie (5): ASoC: Intel: sof-da7219-max98373: add DMIC widget and route ASoC: SOF: topology: add support to smart amplifier ASoC: SOF: Intel: BYT: harden IPC initialization and handling ASoC: topology: refine and log the header in the correct pass ASoC: topology: remove the redundant pass checks Krzysztof Kozlowski (1): dt-bindings: sound: wm8994: Correct required supplies based on actual implementaion Kuninori Morimoto (78): ASoC: soc: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: soc-core: set rtd->num_cpu/codec at soc_new_pcm_runtime() ASoC: soc-core: tidyup soc_new_pcm_runtime() rtd setups ASoC: soc-core: remove cpu_dai/codec_dai/cpu_dais/codec_dais soundwire: intel: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: soc-compress: add snd_compress_ops ASoC: codec: wm_adsp: use snd_compress_ops ASoC: uniphier: use snd_compress_ops ASoC: qcom: q6sp6: use snd_compress_ops ASoC: intel: atom: use snd_compress_ops ASoC: sof: use snd_compress_ops ASoC: sprd: use snd_compress_ops ASoC: soc-compress: remove snd_compr_ops ASoC: soc-dai: add soc_dai_err() ASoC: soc-dai: don't overwide dai->driver->ops ASoC: soc-dai: add snd_soc_pcm_dai_new() ASoC: soc-dai: add snd_soc_pcm_dai_prepare() ASoC: soc-dai: add snd_soc_pcm_dai_trigger() ASoC: soc-dai: add snd_soc_pcm_dai_bespoke_trigger() ASoC: soc-dai: add snd_soc_pcm_dai_probe() ASoC: soc-dai: add snd_soc_pcm_dai_remove() ASoC: soc-dai: add snd_soc_dai_compr_start() ASoC: soc-dai: add snd_soc_dai_compr_shutdown() ASoC: soc-dai: add snd_soc_dai_compr_trigger() ASoC: soc-dai: add snd_soc_dai_compr_set_params() ASoC: soc-dai: add snd_soc_dai_compr_get_params() ASoC: soc-dai: add snd_soc_dai_compr_ack() ASoC: soc-dai: add snd_soc_dai_compr_pointer() ASoC: soc-dai: add snd_soc_dai_compr_set_metadata() ASoC: soc-dai: add snd_soc_dai_compr_get_metadata() ASoC: dt-bindings: simple-card: switch to yaml base Documentation ASoC: soc-pcm: replace snd_soc_runtime_activate()/deactivate() to macro ASoC: soc-dai: add snd_soc_dai_action() ASoC: soc-dapm: use snd_soc_dai_activate()/deactivate() ASoC: soc-dai: add snd_soc_dai_active() ASoC: soc-component: add snd_soc_component_active() ASoC: soc-dai: add snd_soc_dai_stream_active() ASoC: use snd_soc_xxx_active() ASoC: atomel: use snd_soc_xxx_active() ASoC: bcm: use snd_soc_xxx_active() ASoC: cirrus: use snd_soc_xxx_active() ASoC: codecs: use snd_soc_xxx_active() ASoC: fsl: use snd_soc_xxx_active() ASoC: intel: use snd_soc_xxx_active() ASoC: jz4740: use snd_soc_xxx_active() ASoC: mediatek: use snd_soc_xxx_active() ASoC: meson: use snd_soc_xxx_active() ASoC: pxa: use snd_soc_xxx_active() ASoC: ti: use snd_soc_xxx_active() ASoC: uniphier: use snd_soc_xxx_active() ASoC: dwc: use snd_soc_xxx_active() ASoC: cleanup dai / component active code ASoC: add soc-link.c ASoC: soc-link: move soc_rtd_xxx() ASoC: soc-link: remove unneeded parameter from snd_soc_link_xxx() ASoC: soc-link: add snd_soc_link_be_hw_params_fixup() ASoC: soc-link: add snd_soc_link_compr_startup() ASoC: soc-link: add snd_soc_link_compr_shutdown() ASoC: soc-link: add snd_soc_link_compr_set_params() ASoC: soc.h: convert bool to bit field for snd_soc_card ASoC: add soc-card.c ASoC: soc-card: move snd_soc_card_get_kcontrol() to soc-card ASoC: soc-card: move snd_soc_card_jack_new() to soc-card ASoC: soc-card: move snd_soc_card_set/get_drvdata() to soc-card ASoC: soc-card: move snd_soc_card_get_codec_dai() to soc-card ASoC: soc-card: move snd_soc_card_subclass to soc-card ASoC: soc-card: add snd_soc_card_suspend_pre() ASoC: soc-card: add snd_soc_card_suspend_post() ASoC: soc-card: add snd_soc_card_resume_pre() ASoC: soc-card: add snd_soc_card_resume_post() ASoC: soc-card: add probed bit field to snd_soc_card ASoC: soc-card: add snd_soc_card_probe() ASoC: soc-card: add snd_soc_card_late_probe() ASoC: soc-card: add snd_soc_card_remove() ASoC: soc-card: add snd_soc_card_set_bias_level() ASoC: soc-card: add snd_soc_card_set_bias_level_post() ASoC: soc-card: add snd_soc_card_add_dai_link() ASoC: soc-card: add snd_soc_card_remove_dai_link() Lad Prabhakar (1): dt-bindings: ASoC: renesas,rsnd: Add r8a7742 support Libin Yang (6): ALSA: hda: Add ElkhartLake HDMI codec vid ASoC: SOF: Intel: add PCI ID for ElkhartLake ASoC: Intel: boards: support Elkhart Lake with rt5660 ASoC: intel: add depends on SND_SOC_SOF_HDA_AUDIO_CODEC for common hdmi ASoC: sof-sdw: remove CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC condition ASoC: sof_pcm512x: remove CONFIG_SND_HDA_CODEC_HDMI condition Lubomir Rintel (12): ASoC: mmp-sspa: Flip SNDRV_PCM_FMTBIT_S24_3LE on ASoC: mmp-sspa: Drop S20_3LE case ASoC: mmp-sspa: A trivial typo fix ASoC: mmp-sspa: Get rid of dma_params and phys_base ASoC: mmp-sspa: Add support for soc-generic-dmaengine-pcm ASoC: mmp-sspa: Remove the embedded struct ssp_device ASoC: mmp-sspa: Prepare/unprepare the clocks ASoC: mmp-sspa: Add support for the runtime power management ASoC: mmp-sspa: Set appropriate bus format for given bit width ASoC: mmp-sspa: Add Device Tree support ASoC: Add Marvell MMP SSPA binding ASoC: mmp-sspa: Fix the error handling in probe() Mac Chiang (1): ASoC: Intel: boards: split woofer and tweeter support Marcin Rajwa (2): ASoC: SOF: add a power_down_notify method ASoC: SOF: inform DSP that driver is going to be removed Mark Brown (37): Merge series "ASoC: remove rtd->cpu/codec_dai{s} v2" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "ASoC: Intel: machine drivers update for 5.8" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "Support headset on Tegra boards that use WM8903" from Dmitry Osipenko <digetx(a)gmail.com>: Merge series "ASoC: SOF: trivial code cleanups" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: SOF: adjust dmesg verbosity" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge branch 'asoc-5.7' into asoc-5.8 Merge series "Add support for SOF on i.MX8M" from Daniel Baluta <daniel.baluta(a)oss.nxp.com> Daniel Baluta <daniel.baluta(a)nxp.com>: Merge series "ASoC: SOF: topology and firmware IPC updates for 5.8" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: Fix dependency issues of SND_SOC" from Wei Li <liwei391(a)huawei.com>: Merge series "ASoC: add snd_compress_ops and replace" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "ASoC: Add new module driver for new ASRC" from Shengjiu Wang <shengjiu.wang(a)nxp.com>: Merge tag 'tegra-for-5.7-asoc' of git://git.kernel.org/.../tegra/linux into asoc-5.7 ASoC: SOF: Fix build ASoC: dmic: Allow GPIO operations to sleep ASoC: SOF: Add missing dependency on IMX_SCU Merge branch 'for-5.7' of https://git.kernel.org/.../broonie/sound into asoc-5.8 Merge series "add channel constraint for BDW machine drivers" from Brent Lu <brent.lu(a)intel.com>: Merge series "Kconfig updates for DMIC and SOF HDMI support" from Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com>: Merge series "ASoC: soc-dai: add snd_soc_dai_xxx()" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "SOF topology parsing updates" from Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com>: Merge series "ASoC: rt5682: apply some refactors" from Tzung-Bi Shih <tzungbi(a)google.com>: Merge branch 'for-5.7' of https://git.kernel.org/.../broonie/sound into asoc-5.8 Merge branch 'for-5.7' of https://git.kernel.org/.../broonie/sound into asoc-5.8 Merge series "ASoC: fsl_esai: Add support for imx8qm" from Shengjiu Wang <shengjiu.wang(a)nxp.com>: Merge series "ASoC: da7213: support for usage with simple-card" from Sebastian Reichel <sebastian.reichel(a)collabora.com>: Merge series "ASoC: Intel: minor fixes to machine drivers" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: codecs: rt*-sdw: memory leaks and simplifications" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: cleanup DAI/Component activity" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "ASoC: SOF: Intel and IMX updates for 5.8" from Kai Vehmanen <kai.vehmanen(a)linux.intel.com>: Merge series "Add Renoir ACP driver" from Vijendar Mukunda <Vijendar.Mukunda(a)amd.com>: Merge series "ASoC: SOF: extended manifest support for 5.8" from Kai Vehmanen <kai.vehmanen(a)linux.intel.com>: Merge series "ASoC: add soc-link" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "ASoC: SOF: finalize Baytrail/CherryTrail support" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC topology header parsing refinement" from Keyon Jie <yang.jie(a)linux.intel.com>: Merge series "Kconfig updates for DMIC and SOF HDMI support" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge remote-tracking branch 'asoc/for-5.7' into asoc-linus Merge remote-tracking branch 'asoc/for-5.8' into asoc-linus Mateusz Gorski (3): ASoC: Intel: Skylake: Add alternative topology binary name ASoC: Intel: Multiple I/O PCM format support for pipe ASoC: Intel: Skylake: Automatic DMIC format configuration according to information from NHLT Oder Chiou (4): ASoC: rt1308: Revise the devicetree file mode ASoC: rl6231: Add the K bypass for the PLL parameters ASoC: rt1016: Add the rt1016 support ASoC: rt1016: Add the devicetree file for the rt1016 Pan Xiuli (6): ASoC: SOF: add probe support extend data ASoC: SOF: add debug ABI version ASoC: SOF: change type char to uint8_t in info.h ASoC: SOF: change type char to uint8_t in trace.h ASoC: SOF: change type char to uint8_t in topology.h ASoC: SOF: make sof_ipc_cc_version to fixed length Paul Cercueil (1): ASoC: ingenic: Unconditionally depend on devicetree Pavel Dobias (7): ASoC: max9867: fix volume controls ASoC: max9867: fix ADC level control ASoC: max9867: don't use regmap defaults ASoC: max9867: add filter controls ASoC: max9867: add mono playback switch ASoC: max9867: add digital microphone controls ASoC: max9867: keep ADCs and DACs always on Payal Kshirsagar (2): ASoC: SOF: remove unneeded variables ASoC: SOF: Intel: hda: remove unnecessary parentheses Pierre-Louis Bossart (29): ASoC: codecs: rt1308-sdw: reduce verbosity ASoC: SOF: Intel: hda: reduce verbosity on SoundWire detection ASoC: SOF: Intel: hda: log number of microphones detected in NHLT tables ASoC: Intel: sof_sdw_hdmi: fix compilation issue in fallback mode ASoC: Intel: sof_sdw_hdmi: remove codec_dai use ASoC: Intel: sof_sdw_rt1308: remove codec dai use ASoC: Intel: sof_sdw_rt5682: remove codec_dai use ASoC: Intel: sof_sdw_rt700: remove codec_dai use ASoC: Intel: sof_sdw_rt711: remove codec_dai use ASoC: Intel: skl_hda_generic: remove rtd->codec_dai ASoC: SOF: imx: fix undefined reference issue ASoC: Intel: boards: add explicit dependency on GPIOLIB when DMIC is used ASoC: Intel: boards: sof-wm8804: support for Hifiberry Digiplus boards ASoC: SOF/Intel: clarify SPDX license with GPL-2.0-only ASoC: codecs: rt*-sdw: don't assign slave_ops ASoC: codecs: rt*-sdw: fix memory leak in set_sdw_stream() ASoC: codecs: rt1308-sdw: remove duplicate allocation ASoC: SOF: imx: make dsp_ops static ASoC: SOF: imx: make imx8m_dsp_ops static ASoC: Intel: soc-acpi: change machine driver name for WM8804 platforms ASoC: Intel: sof_sdw: fix typo in components string ASoC: Intel: sof_sdw_rt711: remap buttons ASoC: Intel: sof_sdw_rt700: remap buttons ASoC: Intel: cht_bsw_nau8824: remap BTN_0 as KEY_PLAYPAUSE ASoC: Intel: bytcr_rt5640/51: remove .ignore_suspend ASoC: Intel: byt/cht: add .pm_ops ASoC: SOF: Intel: BYT: mask BUSY or DONE interrupts in handler ASoC: reduce verbosity of error messages for sof-dai and sof-link ASoC: SOF: Intel: Baytrail: fix 'defined but not used' warnings Qiushi Wu (1): ASoC: fix incomplete error-handling in img_i2s_in_probe. Ranjani Sridharan (6): ASoC: soc-core: Add dynamic debug logs in soc_dai_link_sanity_check() ASoC: Intel: broadwell: Fix oops during module removal ASoC: SOF: Intel: byt: Add PM callbacks ASoC: SOF: pm: handle resume on legacy Intel platforms ASoC: SOF: ipc: ignore DSP replies received when they are not expected ASoC: SOF: Intel: BYT: add .remove op Ravulapati Vishnu vardhan rao (1): ASoC: amd: Removing unnecessary instance initialization Robert Jarzmik (1): ASoC: pxa: remove Compulab pxa2xx boards Rong Chen (1): ASoC: soc-compress: avoid false-positive Wuninitialized warning Sebastian Reichel (4): ASoC: da7213: Add da7212 DT compatible ASoC: da7213: Add regulator support ASoC: da7213: move set_sysclk to codec level ASoC: da7213: move set_pll to codec level Sebastiano Carlucci (1): ASoC: SOF: topology: Add support for DC Blocker Seppo Ingalsuo (3): ASoC: SOF: Intel: Fix typo in header file comment text ASoC: SOF: Intel: Change DMIC load IPC to fixed length ASoC: SOF: Intel: Rename deprecated DMIC IPC struct field Seven Lee (2): ASoc: nau8810: add AUX related dapm widgets and routes ASoC: nau8810: add I2C device and compatible ID Shengjiu Wang (16): ASoC: fsl_asrc: rename asrc_priv to asrc ASoC: dt-bindings: fsl_asrc: Add new property fsl, asrc-format ASoC: fsl-asoc-card: Support new property fsl, asrc-format ASoC: fsl_asrc: Support new property fsl,asrc-format ASoC: fsl_asrc: Move common definition to fsl_asrc_common ASoC: dt-bindings: fsl_easrc: Add document for EASRC ASoC: fsl_easrc: Add EASRC ASoC CPU DAI drivers ASoC: wm8962: set CLOCKING2 as non-volatile register ASoC: fsl_esai: Disable exception interrupt before scheduling tasklet ASoC: fsl_easrc: Check for null pointer before dereferencing "ctx" in fsl_easrc_hw_free() ASoC: wm8524: Add support S32_LE ASoC: wm8962: Use force clear for WM8962_SYSCLK_ENA after reset ASoC: fsl_esai: introduce SoC specific data ASoC: fsl_esai: Add new compatible string for imx8qm ASoC: fsl_asrc: Set ASR76K and ASR56K based on processing clock ASoC: fsl_asrc: Merge suspend/resume function to runtime_suspend/resume Sowjanya Komatineni (4): ASoC: tegra: Use device managed resource APIs to get the clock ASoC: nau8825: Change Tegra clk_out_2 provider to PMC ASoC: tegra: Add audio mclk parent configuration ASoC: tegra: Enable audio mclk during tegra_asoc_utils_init() Stephan Gerhold (3): ASoC: qdsp6: Suggest more generic node names dt-bindings: sound: lpass-cpu: Document DAI subnodes ASoC: qcom: lpass-cpu: Make I2S SD lines configurable Steve Lee (1): ASoC: max98390: Added Amplifier Driver Sven Van Asbroeck (2): dt-bindings: sound: add Microsemi ZL38060 binding ASoC: Add initial ZL38060 driver Tang Bin (10): ASoC: fsl_micfil: Omit superfluous error message in fsl_micfil_probe() ASoC: hisilicon: Use IS_ERR() instead of IS_ERR_OR_NULL() ASoC: mxs-saif: Avoid unnecessary check ASoC: hisilicon: Use the defined variable to clean code ASoC: mxs-saif: Fix unused assignment ASoC: qcom: Use the defined variable to simplify code ASoC: mediatek: Fix error handling ASoC: fsl_micfil: Fix indentation to put on one line affected code ASoC: fsl_micfil: Fix unused assignment in fsl_set_clock_params() ASoC: fsl: imx-audmix: Fix unused assignment to variable 'ret' Tzung-Bi Shih (8): ASoC: rt5682: simplify assertions ASoC: rt5682: fix space issues ASoC: rt5682: remove empty default case ASoC: rt5682: replace message printing from pr_() to dev_() ASoC: rt5682: remove duplicate rt5682_reset() calls ASoC: rt5682: remove unwanted btn_type assignment ASoC: mediatek: mt8183: fix error handling of platform_get_irq() ASoC: mediatek: mt8183-da7219: set headset button maps Vijendar Mukunda (17): ASoC: amd: add Renoir ACP3x IP register header ASoC: amd: add Renoir ACP PCI driver ASoC: amd: add acp init/de-init functions ASoC: amd: create acp3x pdm platform device ASoC: amd: add ACP3x PDM platform driver ASoC: amd: irq handler changes for ACP3x PDM dma driver ASoC: amd: add acp3x pdm driver dma ops ASoC: amd: add ACP PDM DMA driver dai ops ASoC: amd: add Renoir ACP PCI driver PM ops ASoC: amd: add ACP PDM DMA driver pm ops ASoC: amd: enable Renoir acp3x drivers build ASoC: amd: create platform devices for Renoir ASoC: amd: RN machine driver using dmic ASoC: amd: enable build for RN machine driver ASoC: amd: fix kernel warning ASoC: amd: refactoring dai_hw_params() callback ASoC: amd: return error when acp de-init fails Wei Li (2): ASoC: wm89xx: Fix build errors caused by I2C dependency ASoC: Fix wrong dependency of da7210 and wm8983 Wei Yongjun (4): ASoC: tegra: tegra_wm8903: Use devm_snd_soc_register_card() ASoC: rt5677: Use devm_snd_soc_register_component() ASoC: SOF: core: fix error return code in sof_probe_continue() ASoC: mmp-sspa: Fix return value check in asoc_mmp_sspa_probe() Wolfram Sang (1): ASoC: codecs: max9768: update contact email Xiyu Yang (2): ASoC: davinci-mcasp: Fix dma_chan refcnt leak when getting dma type ASoC: fsl_asrc_dma: Fix dma_chan leak when config DMA channel failed Yong Zhi (2): ASoC: Intel: sof_da7219_max98373: Add BE dailink for dmic16k ASoC: max98373: reorder max98373_reset() in resume Yongbo Zhang (1): SoC: rsnd: add interrupt support for SSI BUSIF buffer YueHaibing (2): ASoC: SOF: imx8: Fix randbuild error ASoC: SOF: Intel: Fix unused variable warning derek.fang (1): ASoC: rt1015: Enable class-D silence and clock detections kbuild test robot (1): ASoC: fix semicolon.cocci warnings randerwang (3): ASoC: Intel: sof_sdw: add amp number in components string for ucm ASoC: SOF: Intel: change trigger sequence to fix pop noise when stopping playback on sdw platforms ASoC: SOF: Intel: sdw: relax sdw machine select constraints shengjiu wang (1): ASoC: fsl_asrc: Fix -Wmissing-prototypes warning 朱灿灿 (1): Subject: [PATCH v2] ASoC: soc-pcm: fix BE dai not hw_free and shutdown during mixer update Documentation/devicetree/bindings/dsp/fsl,dsp.yaml | 2 + Documentation/devicetree/bindings/sound/da7213.txt | 8 +- .../devicetree/bindings/sound/fsl,asrc.txt | 4 + .../devicetree/bindings/sound/fsl,easrc.yaml | 101 + .../devicetree/bindings/sound/fsl,esai.txt | 1 + .../bindings/sound/marvell,mmp-sspa.yaml | 122 ++ .../devicetree/bindings/sound/nau8810.txt | 5 +- .../devicetree/bindings/sound/nau8825.txt | 2 +- .../bindings/sound/nvidia,tegra-audio-wm8903.txt | 1 + .../devicetree/bindings/sound/qcom,lpass-cpu.txt | 25 + .../devicetree/bindings/sound/qcom,q6adm.txt | 2 +- .../devicetree/bindings/sound/qcom,q6afe.txt | 46 +- .../devicetree/bindings/sound/qcom,q6asm.txt | 7 +- .../devicetree/bindings/sound/qcom,q6core.txt | 2 +- .../devicetree/bindings/sound/renesas,rsnd.txt | 1 + .../devicetree/bindings/sound/rockchip-i2s.yaml | 12 +- Documentation/devicetree/bindings/sound/rt1016.txt | 17 + Documentation/devicetree/bindings/sound/rt1308.txt | 0 .../devicetree/bindings/sound/simple-card.txt | 351 ---- .../devicetree/bindings/sound/simple-card.yaml | 484 +++++ .../devicetree/bindings/sound/tlv320adcx140.yaml | 51 + Documentation/devicetree/bindings/sound/wm8994.txt | 18 +- .../devicetree/bindings/sound/zl38060.yaml | 69 + drivers/soundwire/intel.c | 4 +- include/sound/soc-acpi.h | 2 +- include/sound/soc-card.h | 69 + include/sound/soc-component.h | 46 +- include/sound/soc-dai.h | 62 +- include/sound/soc-dapm.h | 2 +- include/sound/soc-link.h | 27 + include/sound/soc.h | 104 +- include/sound/sof.h | 5 +- include/sound/sof/channel_map.h | 2 +- include/sound/sof/control.h | 2 +- include/sound/sof/dai-imx.h | 2 +- include/sound/sof/dai-intel.h | 22 +- include/sound/sof/dai.h | 2 +- include/sound/sof/ext_manifest.h | 95 + include/sound/sof/header.h | 2 +- include/sound/sof/info.h | 28 +- include/sound/sof/pm.h | 2 +- include/sound/sof/stream.h | 2 +- include/sound/sof/topology.h | 20 +- include/sound/sof/trace.h | 4 +- include/sound/sof/xtensa.h | 2 +- include/uapi/sound/skl-tplg-interface.h | 2 + include/uapi/sound/sof/abi.h | 2 +- include/uapi/sound/sof/tokens.h | 8 + sound/hda/intel-nhlt.c | 2 +- sound/pci/hda/patch_hdmi.c | 1 + sound/soc/Makefile | 2 +- sound/soc/amd/Kconfig | 15 +- sound/soc/amd/Makefile | 1 + sound/soc/amd/raven/acp3x-i2s.c | 6 +- sound/soc/amd/raven/acp3x-pcm-dma.c | 49 +- sound/soc/amd/renoir/Makefile | 7 + sound/soc/amd/renoir/acp3x-pdm-dma.c | 524 +++++ sound/soc/amd/renoir/acp3x-rn.c | 77 + sound/soc/amd/renoir/rn-pci-acp3x.c | 344 ++++ sound/soc/amd/renoir/rn_acp3x.h | 88 + sound/soc/amd/renoir/rn_chip_offset_byte.h | 349 ++++ sound/soc/atmel/atmel_ssc_dai.c | 4 +- sound/soc/bcm/bcm2835-i2s.c | 6 +- sound/soc/bcm/cygnus-ssp.c | 4 +- sound/soc/cirrus/ep93xx-i2s.c | 4 +- sound/soc/codecs/Kconfig | 40 +- sound/soc/codecs/Makefile | 6 + sound/soc/codecs/ad1980.c | 2 +- sound/soc/codecs/ad73311.c | 2 +- sound/soc/codecs/adau7118-i2c.c | 7 + sound/soc/codecs/adav80x.c | 4 +- sound/soc/codecs/arizona.c | 2 +- sound/soc/codecs/cros_ec_codec.c | 22 +- sound/soc/codecs/cs4271.c | 4 +- sound/soc/codecs/cs47l15.c | 9 +- sound/soc/codecs/cs47l24.c | 8 +- sound/soc/codecs/cs47l35.c | 9 +- sound/soc/codecs/cs47l85.c | 9 +- sound/soc/codecs/cs47l90.c | 9 +- sound/soc/codecs/cs47l92.c | 9 +- sound/soc/codecs/da7213.c | 102 +- sound/soc/codecs/da7213.h | 9 + sound/soc/codecs/dmic.c | 4 +- sound/soc/codecs/jz4725b.c | 4 +- sound/soc/codecs/jz4740.c | 4 +- sound/soc/codecs/jz4770.c | 2 +- sound/soc/codecs/madera.c | 2 +- sound/soc/codecs/max9768.c | 2 +- sound/soc/codecs/max98090.c | 6 +- sound/soc/codecs/max98373.c | 2 +- sound/soc/codecs/max98390.c | 1040 ++++++++++ sound/soc/codecs/max98390.h | 663 ++++++ sound/soc/codecs/max9867.c | 95 +- sound/soc/codecs/max9867.h | 1 - sound/soc/codecs/nau8810.c | 39 +- sound/soc/codecs/nau8810.h | 8 + sound/soc/codecs/rl6231.c | 29 +- sound/soc/codecs/rl6231.h | 1 + sound/soc/codecs/rt1015.c | 29 +- sound/soc/codecs/rt1015.h | 1 + sound/soc/codecs/rt1016.c | 695 +++++++ sound/soc/codecs/rt1016.h | 232 +++ sound/soc/codecs/rt1308-sdw.c | 15 +- sound/soc/codecs/rt5677-spi.c | 12 +- sound/soc/codecs/rt5682-i2c.c | 306 +++ sound/soc/codecs/rt5682-sdw.c | 462 ++++- sound/soc/codecs/rt5682-sdw.h | 20 - sound/soc/codecs/rt5682.c | 888 +------- sound/soc/codecs/rt5682.h | 32 +- sound/soc/codecs/rt700-sdw.c | 3 - sound/soc/codecs/rt700.c | 3 + sound/soc/codecs/rt711-sdw.c | 3 - sound/soc/codecs/rt711.c | 3 + sound/soc/codecs/rt715-sdw.c | 3 - sound/soc/codecs/rt715.c | 3 + sound/soc/codecs/sta32x.c | 11 +- sound/soc/codecs/tas2552.c | 1 + sound/soc/codecs/tlv320adcx140.c | 96 +- sound/soc/codecs/tlv320adcx140.h | 11 + sound/soc/codecs/tlv320aic23.c | 2 +- sound/soc/codecs/tlv320dac33.c | 2 +- sound/soc/codecs/uda1380.c | 2 +- sound/soc/codecs/wcd9335.c | 44 +- sound/soc/codecs/wcd934x.c | 30 +- sound/soc/codecs/wl1273.c | 2 +- sound/soc/codecs/wm5102.c | 9 +- sound/soc/codecs/wm5110.c | 8 +- sound/soc/codecs/wm8524.c | 4 +- sound/soc/codecs/wm8711.c | 2 +- sound/soc/codecs/wm8753.c | 4 +- sound/soc/codecs/wm8782.c | 2 +- sound/soc/codecs/wm8900.c | 6 - sound/soc/codecs/wm8962.c | 9 +- sound/soc/codecs/wm8990.c | 98 - sound/soc/codecs/wm8991.c | 8 - sound/soc/codecs/wm8994.c | 8 - sound/soc/codecs/wm_adsp.c | 18 +- sound/soc/codecs/wm_adsp.h | 18 +- sound/soc/codecs/zl38060.c | 638 ++++++ sound/soc/dwc/dwc-i2s.c | 2 +- sound/soc/fsl/Kconfig | 11 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl-asoc-card.c | 24 +- sound/soc/fsl/fsl_asrc.c | 368 ++-- sound/soc/fsl/fsl_asrc.h | 74 +- sound/soc/fsl/fsl_asrc_common.h | 106 + sound/soc/fsl/fsl_asrc_dma.c | 55 +- sound/soc/fsl/fsl_audmix.c | 4 +- sound/soc/fsl/fsl_easrc.c | 2117 ++++++++++++++++++++ sound/soc/fsl/fsl_easrc.h | 651 ++++++ sound/soc/fsl/fsl_esai.c | 52 +- sound/soc/fsl/fsl_micfil.c | 17 +- sound/soc/fsl/fsl_spdif.c | 4 +- sound/soc/hisilicon/hi6210-i2s.c | 16 +- sound/soc/img/img-i2s-in.c | 1 + sound/soc/img/img-i2s-out.c | 8 +- sound/soc/img/img-spdif-in.c | 4 +- sound/soc/img/img-spdif-out.c | 4 +- sound/soc/intel/Kconfig | 8 +- sound/soc/intel/Makefile | 2 +- sound/soc/intel/atom/Makefile | 2 +- sound/soc/intel/atom/sst-atom-controls.h | 2 +- sound/soc/intel/atom/sst-mfld-platform-compress.c | 43 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 16 +- sound/soc/intel/atom/sst-mfld-platform.h | 2 +- sound/soc/intel/atom/sst/Makefile | 2 +- sound/soc/intel/baytrail/sst-baytrail-ipc.c | 4 +- sound/soc/intel/boards/Kconfig | 83 +- sound/soc/intel/boards/Makefile | 6 +- sound/soc/intel/boards/bdw-rt5650.c | 29 + sound/soc/intel/boards/bdw-rt5677.c | 26 + sound/soc/intel/boards/broadwell.c | 43 +- sound/soc/intel/boards/bytcht_cx2072x.c | 5 +- sound/soc/intel/boards/bytcht_da7213.c | 3 + sound/soc/intel/boards/bytcht_es8316.c | 3 + sound/soc/intel/boards/bytcr_rt5640.c | 19 +- sound/soc/intel/boards/bytcr_rt5651.c | 6 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 3 + sound/soc/intel/boards/cht_bsw_nau8824.c | 7 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 3 + sound/soc/intel/boards/cht_bsw_rt5672.c | 3 + sound/soc/intel/boards/cml_rt1011_rt5682.c | 224 ++- sound/soc/intel/boards/ehl_rt5660.c | 323 +++ sound/soc/intel/boards/glk_rt5682_max98357a.c | 2 +- sound/soc/intel/boards/hda_dsp_common.c | 2 +- sound/soc/intel/boards/hda_dsp_common.h | 2 +- sound/soc/intel/boards/kbl_da7219_max98357a.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 2 +- sound/soc/intel/boards/kbl_rt5660.c | 2 +- sound/soc/intel/boards/skl_hda_dsp_common.c | 2 +- sound/soc/intel/boards/skl_hda_dsp_common.h | 2 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 33 +- sound/soc/intel/boards/sof_da7219_max98373.c | 23 +- sound/soc/intel/boards/sof_maxim_common.c | 2 +- sound/soc/intel/boards/sof_maxim_common.h | 2 +- sound/soc/intel/boards/sof_pcm512x.c | 11 +- sound/soc/intel/boards/sof_rt5682.c | 2 +- sound/soc/intel/boards/sof_sdw.c | 65 +- sound/soc/intel/boards/sof_sdw_common.h | 2 +- sound/soc/intel/boards/sof_sdw_dmic.c | 2 +- sound/soc/intel/boards/sof_sdw_hdmi.c | 11 +- sound/soc/intel/boards/sof_sdw_rt1308.c | 4 +- sound/soc/intel/boards/sof_sdw_rt5682.c | 5 +- sound/soc/intel/boards/sof_sdw_rt700.c | 13 +- sound/soc/intel/boards/sof_sdw_rt711.c | 13 +- sound/soc/intel/boards/sof_sdw_rt715.c | 2 +- sound/soc/intel/boards/sof_wm8804.c | 298 +++ sound/soc/intel/common/Makefile | 2 +- sound/soc/intel/common/soc-acpi-intel-bxt-match.c | 4 +- sound/soc/intel/common/soc-acpi-intel-cfl-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-cml-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-cnl-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-ehl-match.c | 9 +- sound/soc/intel/common/soc-acpi-intel-glk-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-hda-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-icl-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-jsl-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-kbl-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-skl-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-tgl-match.c | 2 +- sound/soc/intel/common/soc-intel-quirks.h | 2 +- sound/soc/intel/common/sst-dsp.c | 9 +- sound/soc/intel/haswell/sst-haswell-dsp.c | 185 +- sound/soc/intel/skylake/Makefile | 2 +- sound/soc/intel/skylake/skl-i2s.h | 2 +- sound/soc/intel/skylake/skl-ssp-clk.c | 2 +- sound/soc/intel/skylake/skl-sst-utils.c | 2 +- sound/soc/intel/skylake/skl-topology.c | 184 +- sound/soc/intel/skylake/skl-topology.h | 5 +- sound/soc/intel/skylake/skl.h | 2 +- sound/soc/jz4740/Kconfig | 2 +- sound/soc/jz4740/jz4740-i2s.c | 12 +- sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 7 +- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 8 +- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 7 +- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 16 +- sound/soc/meson/axg-tdm-interface.c | 2 +- sound/soc/mxs/mxs-saif.c | 5 +- sound/soc/pxa/Kconfig | 25 +- sound/soc/pxa/mmp-sspa.c | 445 ++-- sound/soc/pxa/mmp-sspa.h | 32 +- sound/soc/pxa/pxa-ssp.c | 8 +- sound/soc/pxa/pxa2xx-i2s.c | 2 +- sound/soc/qcom/lpass-apq8016.c | 9 +- sound/soc/qcom/lpass-cpu.c | 235 ++- sound/soc/qcom/lpass-lpaif-reg.h | 30 +- sound/soc/qcom/lpass.h | 4 + sound/soc/qcom/qdsp6/q6asm-dai.c | 55 +- sound/soc/sh/rcar/gen.c | 8 + sound/soc/sh/rcar/rsnd.h | 9 + sound/soc/sh/rcar/ssi.c | 145 ++ sound/soc/soc-card.c | 225 +++ sound/soc/soc-compress.c | 281 ++- sound/soc/soc-core.c | 245 +-- sound/soc/soc-dai.c | 485 +++-- sound/soc/soc-dapm.c | 30 +- sound/soc/soc-generic-dmaengine-pcm.c | 6 +- sound/soc/soc-jack.c | 38 - sound/soc/soc-link.c | 155 ++ sound/soc/soc-pcm.c | 251 +-- sound/soc/soc-topology.c | 96 +- sound/soc/sof/Makefile | 2 +- sound/soc/sof/compress.c | 9 +- sound/soc/sof/compress.h | 9 +- sound/soc/sof/control.c | 6 +- sound/soc/sof/core.c | 9 +- sound/soc/sof/debug.c | 2 +- sound/soc/sof/imx/Kconfig | 34 +- sound/soc/sof/imx/Makefile | 4 +- sound/soc/sof/imx/imx8.c | 4 +- sound/soc/sof/imx/imx8m.c | 285 +++ sound/soc/sof/intel/Makefile | 2 +- sound/soc/sof/intel/apl.c | 2 +- sound/soc/sof/intel/bdw.c | 2 +- sound/soc/sof/intel/byt.c | 172 +- sound/soc/sof/intel/cnl.c | 2 +- sound/soc/sof/intel/hda-bus.c | 2 +- sound/soc/sof/intel/hda-codec.c | 58 +- sound/soc/sof/intel/hda-compress.c | 2 +- sound/soc/sof/intel/hda-ctrl.c | 2 +- sound/soc/sof/intel/hda-dai.c | 2 +- sound/soc/sof/intel/hda-dsp.c | 10 +- sound/soc/sof/intel/hda-ipc.c | 2 +- sound/soc/sof/intel/hda-ipc.h | 2 +- sound/soc/sof/intel/hda-loader.c | 11 +- sound/soc/sof/intel/hda-pcm.c | 2 +- sound/soc/sof/intel/hda-stream.c | 2 +- sound/soc/sof/intel/hda-trace.c | 2 +- sound/soc/sof/intel/hda.c | 28 +- sound/soc/sof/intel/hda.h | 2 +- sound/soc/sof/intel/intel-ipc.c | 2 +- sound/soc/sof/intel/shim.h | 2 +- sound/soc/sof/ipc.c | 11 +- sound/soc/sof/loader.c | 213 +- sound/soc/sof/nocodec.c | 8 +- sound/soc/sof/ops.c | 2 +- sound/soc/sof/ops.h | 2 +- sound/soc/sof/pcm.c | 18 +- sound/soc/sof/pm.c | 29 +- sound/soc/sof/probe.c | 2 +- sound/soc/sof/probe.h | 2 +- sound/soc/sof/sof-acpi-dev.c | 2 +- sound/soc/sof/sof-audio.c | 2 +- sound/soc/sof/sof-audio.h | 4 +- sound/soc/sof/sof-of-dev.c | 16 +- sound/soc/sof/sof-pci-dev.c | 4 +- sound/soc/sof/sof-priv.h | 7 +- sound/soc/sof/topology.c | 485 +++-- sound/soc/sof/trace.c | 2 +- sound/soc/sof/utils.c | 2 +- sound/soc/sof/xtensa/Makefile | 2 +- sound/soc/sof/xtensa/core.c | 2 +- sound/soc/sprd/sprd-pcm-compress.c | 49 +- sound/soc/sprd/sprd-pcm-dma.c | 2 +- sound/soc/sprd/sprd-pcm-dma.h | 2 +- sound/soc/tegra/tegra_alc5632.c | 7 +- sound/soc/tegra/tegra_asoc_utils.c | 113 +- sound/soc/tegra/tegra_asoc_utils.h | 1 - sound/soc/tegra/tegra_max98090.c | 22 +- sound/soc/tegra/tegra_rt5640.c | 22 +- sound/soc/tegra/tegra_rt5677.c | 7 +- sound/soc/tegra/tegra_sgtl5000.c | 7 +- sound/soc/tegra/tegra_wm8753.c | 22 +- sound/soc/tegra/tegra_wm8903.c | 42 +- sound/soc/tegra/tegra_wm9712.c | 8 +- sound/soc/tegra/trimslice.c | 18 +- sound/soc/ti/davinci-mcasp.c | 6 +- sound/soc/ti/omap-dmic.c | 4 +- sound/soc/ti/omap-mcbsp.c | 21 +- sound/soc/ti/omap-mcpdm.c | 8 +- sound/soc/uniphier/aio-compress.c | 45 +- sound/soc/uniphier/aio-cpu.c | 4 +- sound/soc/uniphier/aio-dma.c | 2 +- sound/soc/uniphier/aio.h | 2 +- sound/soc/ux500/mop500.c | 11 +- 335 files changed, 15208 insertions(+), 4136 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,easrc.yaml create mode 100644 Documentation/devicetree/bindings/sound/marvell,mmp-sspa.yaml create mode 100644 Documentation/devicetree/bindings/sound/rt1016.txt mode change 100755 => 100644 Documentation/devicetree/bindings/sound/rt1308.txt delete mode 100644 Documentation/devicetree/bindings/sound/simple-card.txt create mode 100644 Documentation/devicetree/bindings/sound/simple-card.yaml create mode 100644 Documentation/devicetree/bindings/sound/zl38060.yaml create mode 100644 include/sound/soc-card.h create mode 100644 include/sound/soc-link.h create mode 100644 include/sound/sof/ext_manifest.h create mode 100644 sound/soc/amd/renoir/Makefile create mode 100644 sound/soc/amd/renoir/acp3x-pdm-dma.c create mode 100644 sound/soc/amd/renoir/acp3x-rn.c create mode 100644 sound/soc/amd/renoir/rn-pci-acp3x.c create mode 100644 sound/soc/amd/renoir/rn_acp3x.h create mode 100644 sound/soc/amd/renoir/rn_chip_offset_byte.h create mode 100644 sound/soc/codecs/max98390.c create mode 100644 sound/soc/codecs/max98390.h create mode 100644 sound/soc/codecs/rt1016.c create mode 100644 sound/soc/codecs/rt1016.h create mode 100644 sound/soc/codecs/rt5682-i2c.c delete mode 100644 sound/soc/codecs/rt5682-sdw.h create mode 100644 sound/soc/codecs/zl38060.c create mode 100644 sound/soc/fsl/fsl_asrc_common.h create mode 100644 sound/soc/fsl/fsl_easrc.c create mode 100644 sound/soc/fsl/fsl_easrc.h create mode 100644 sound/soc/intel/boards/ehl_rt5660.c create mode 100644 sound/soc/intel/boards/sof_wm8804.c create mode 100644 sound/soc/soc-card.c create mode 100644 sound/soc/soc-link.c create mode 100644 sound/soc/sof/imx/imx8m.c

3 6

Cmi8888 iommu fix
by Tony Dijksterhuis 02 Jun '20

02 Jun '20

Hi everyone. I am using a pink faun i2s (CMI8888) card with a x470 board and amd ryzen cpu. >From the first day using this card i was not able to get or working. Al i got was stuttering sound. After many hours searching i found the thread below https://www.alsa-project.org/pipermail/alsa-devel/2013-December/069753.html Enabling iommu in my bios fixed indeed this issue. However i'm wondering was there ever being a kernel patch fixing this?. Regards Tony Dijksterhuis. Outlook voor Android downloaden<https://aka.ms/ghei36>

1 0

Subject: [PATCH v2] ASoC: soc-pcm: fix BE dai not hw_free and shutdown during mixer update
by 朱灿灿 02 Jun '20

02 Jun '20

FE state is SND_SOC_DPCM_STATE_PREPARE now, BE1 is used by FE. Later when new BE2 is added to FE by mixer update, it will call dpcm_run_update_startup() to update BE2's state, but unfortunately BE2 .prepare() meets error, it will disconnect all non started BE. This make BE1 dai skip .hw_free() and .shutdown(), and the BE1 users will never decrease to zero. Signed-off-by: zhucancan <zhucancan(a)vivo.com> --- re-format patch content v2 --- sound/soc/soc-pcm.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 1f302de44052..df34422bd0dd 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2730,12 +2730,12 @@ static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) close: dpcm_be_dai_shutdown(fe, stream); disconnect: - /* disconnect any non started BEs */ + /* disconnect any closed BEs */ spin_lock_irqsave(&fe->card->dpcm_lock, flags); for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; - if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) - dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; + if (be->dpcm[stream].state == SND_SOC_DPCM_STATE_CLOSE) + dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; } spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); -- 2.21.0

3 2

[PATCH] Add duplex sound support for USB devices using implicit feedback
by Erwin Burema 02 Jun '20

02 Jun '20

For USB sound devices using implicit feedback the endpoint used for this feedback should be able to be opened twice, once for required feedback and second time for audio data. This way these devices can be put in duplex audio mode. Since this only works if the settings of the endpoint don't change a check is included for this. This fixes bug 207023 ("MOTU M2 regression on duplex audio") and should also fix bug 103751 ("M-Audio Fast Track Ultra usb audio device will not operate full-duplex") Signed-off-by: Erwin Burema <e.burema(a)gmail.com> --- sound/usb/card.h | 1 + sound/usb/endpoint.c | 195 ++++++++++++++++++++++++++++++++++++++++++- sound/usb/pcm.c | 5 ++ 3 files changed, 197 insertions(+), 4 deletions(-) diff --git a/sound/usb/card.h b/sound/usb/card.h index 820e564656ed..d6219fba9699 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -108,6 +108,7 @@ struct snd_usb_endpoint { int iface, altsetting; int skip_packets; /* quirks for devices to ignore the first n packets in a stream */ + bool is_implicit_feedback; /* This endpoint is used as implicit feedback */ spinlock_t lock; struct list_head list; diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 50104f658ed4..9bea7d3f99f8 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -522,6 +522,8 @@ struct snd_usb_endpoint *snd_usb_add_endpoint(struct snd_usb_audio *chip, list_add_tail(&ep->list, &chip->ep_list); + ep->is_implicit_feedback = 0; + __exit_unlock: mutex_unlock(&chip->mutex); @@ -621,6 +623,178 @@ static void release_urbs(struct snd_usb_endpoint *ep, int force) ep->nurbs = 0; } +/* + * Check data endpoint for format differences + */ +static bool check_ep_params(struct snd_usb_endpoint *ep, + snd_pcm_format_t pcm_format, + unsigned int channels, + unsigned int period_bytes, + unsigned int frames_per_period, + unsigned int periods_per_buffer, + struct audioformat *fmt, + struct snd_usb_endpoint *sync_ep) +{ + unsigned int maxsize, minsize, packs_per_ms, max_packs_per_urb; + unsigned int max_packs_per_period, urbs_per_period, urb_packs; + unsigned int max_urbs; + int frame_bits = snd_pcm_format_physical_width(pcm_format) * channels; + int tx_length_quirk = (ep->chip->tx_length_quirk && + usb_pipeout(ep->pipe)); + bool ret = 1; + + if (pcm_format == SNDRV_PCM_FORMAT_DSD_U16_LE && fmt->dsd_dop) { + /* + * When operating in DSD DOP mode, the size of a sample frame + * in hardware differs from the actual physical format width + * because we need to make room for the DOP markers. + */ + frame_bits += channels << 3; + } + + ret = ret && (ep->datainterval == fmt->datainterval); + ret = ret && (ep->stride == frame_bits >> 3); + + switch (pcm_format) { + case SNDRV_PCM_FORMAT_U8: + ret = ret && (ep->silence_value == 0x80); + break; + case SNDRV_PCM_FORMAT_DSD_U8: + case SNDRV_PCM_FORMAT_DSD_U16_LE: + case SNDRV_PCM_FORMAT_DSD_U32_LE: + case SNDRV_PCM_FORMAT_DSD_U16_BE: + case SNDRV_PCM_FORMAT_DSD_U32_BE: + ret = ret && (ep->silence_value == 0x69); + break; + default: + ret = ret && (ep->silence_value == 0); + } + + /* assume max. frequency is 50% higher than nominal */ + ret = ret && (ep->freqmax == ep->freqn + (ep->freqn >> 1)); + /* Round up freqmax to nearest integer in order to calculate maximum + * packet size, which must represent a whole number of frames. + * This is accomplished by adding 0x0.ffff before converting the + * Q16.16 format into integer. + * In order to accurately calculate the maximum packet size when + * the data interval is more than 1 (i.e. ep->datainterval > 0), + * multiply by the data interval prior to rounding. For instance, + * a freqmax of 41 kHz will result in a max packet size of 6 (5.125) + * frames with a data interval of 1, but 11 (10.25) frames with a + * data interval of 2. + * (ep->freqmax << ep->datainterval overflows at 8.192 MHz for the + * maximum datainterval value of 3, at USB full speed, higher for + * USB high speed, noting that ep->freqmax is in units of + * frames per packet in Q16.16 format.) + */ + maxsize = (((ep->freqmax << ep->datainterval) + 0xffff) >> 16) * + (frame_bits >> 3); + if (tx_length_quirk) + maxsize += sizeof(__le32); /* Space for length descriptor */ + /* but wMaxPacketSize might reduce this */ + if (ep->maxpacksize && ep->maxpacksize < maxsize) { + /* whatever fits into a max. size packet */ + unsigned int data_maxsize = maxsize = ep->maxpacksize; + + if (tx_length_quirk) + /* Need to remove the length descriptor to calc freq */ + data_maxsize -= sizeof(__le32); + ret = ret && (ep->freqmax == (data_maxsize / (frame_bits >> 3)) + << (16 - ep->datainterval)); + } + + if (ep->fill_max) + ret = ret && (ep->curpacksize == ep->maxpacksize); + else + ret = ret && (ep->curpacksize == maxsize); + + if (snd_usb_get_speed(ep->chip->dev) != USB_SPEED_FULL) { + packs_per_ms = 8 >> ep->datainterval; + max_packs_per_urb = MAX_PACKS_HS; + } else { + packs_per_ms = 1; + max_packs_per_urb = MAX_PACKS; + } + if (sync_ep && !snd_usb_endpoint_implicit_feedback_sink(ep)) + max_packs_per_urb = min(max_packs_per_urb, + 1U << sync_ep->syncinterval); + max_packs_per_urb = max(1u, max_packs_per_urb >> ep->datainterval); + + /* + * Capture endpoints need to use small URBs because there's no way + * to tell in advance where the next period will end, and we don't + * want the next URB to complete much after the period ends. + * + * Playback endpoints with implicit sync much use the same parameters + * as their corresponding capture endpoint. + */ + if (usb_pipein(ep->pipe) || + snd_usb_endpoint_implicit_feedback_sink(ep)) { + + urb_packs = packs_per_ms; + /* + * Wireless devices can poll at a max rate of once per 4ms. + * For dataintervals less than 5, increase the packet count to + * allow the host controller to use bursting to fill in the + * gaps. + */ + if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_WIRELESS) { + int interval = ep->datainterval; + + while (interval < 5) { + urb_packs <<= 1; + ++interval; + } + } + /* make capture URBs <= 1 ms and smaller than a period */ + urb_packs = min(max_packs_per_urb, urb_packs); + while (urb_packs > 1 && urb_packs * maxsize >= period_bytes) + urb_packs >>= 1; + ret = ret && (ep->nurbs == MAX_URBS); + + /* + * Playback endpoints without implicit sync are adjusted so that + * a period fits as evenly as possible in the smallest number of + * URBs. The total number of URBs is adjusted to the size of the + * ALSA buffer, subject to the MAX_URBS and MAX_QUEUE limits. + */ + } else { + /* determine how small a packet can be */ + minsize = (ep->freqn >> (16 - ep->datainterval)) * + (frame_bits >> 3); + /* with sync from device, assume it can be 12% lower */ + if (sync_ep) + minsize -= minsize >> 3; + minsize = max(minsize, 1u); + + /* how many packets will contain an entire ALSA period? */ + max_packs_per_period = DIV_ROUND_UP(period_bytes, minsize); + + /* how many URBs will contain a period? */ + urbs_per_period = DIV_ROUND_UP(max_packs_per_period, + max_packs_per_urb); + /* how many packets are needed in each URB? */ + urb_packs = DIV_ROUND_UP(max_packs_per_period, urbs_per_period); + + /* limit the number of frames in a single URB */ + ret = ret && (ep->max_urb_frames == + DIV_ROUND_UP(frames_per_period, urbs_per_period)); + + /* try to use enough URBs to contain an entire ALSA buffer */ + max_urbs = min((unsigned) MAX_URBS, + MAX_QUEUE * packs_per_ms / urb_packs); + ret = ret && (ep->nurbs == min(max_urbs, + urbs_per_period * periods_per_buffer)); + } + + ret = ret && (ep->datainterval == fmt->datainterval); + ret = ret && (ep->maxpacksize == fmt->maxpacksize); + ret = ret && + (ep->fill_max == !!(fmt->attributes & UAC_EP_CS_ATTR_FILL_MAX)); + + return ret; +} + /* * configure a data endpoint */ @@ -886,10 +1060,23 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, int err; if (ep->use_count != 0) { - usb_audio_warn(ep->chip, - "Unable to change format on ep #%x: already in use\n", - ep->ep_num); - return -EBUSY; + bool check = ep->is_implicit_feedback && + check_ep_params(ep, pcm_format, + channels, period_bytes, + period_frames, buffer_periods, + fmt, sync_ep); + + if (!check) { + usb_audio_warn(ep->chip, + "Unable to change format on ep #%x: already in use\n", + ep->ep_num); + return -EBUSY; + } + + usb_audio_dbg(ep->chip, + "Ep #%x already in use as implicit feedback but format not changed\n", + ep->ep_num); + return 0; } /* release old buffers, if any */ diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index b50965ab3b3a..d61c2f1095b5 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -404,6 +404,8 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, if (!subs->sync_endpoint) return -EINVAL; + subs->sync_endpoint->is_implicit_feedback = 1; + subs->data_endpoint->sync_master = subs->sync_endpoint; return 1; @@ -502,12 +504,15 @@ static int set_sync_endpoint(struct snd_usb_substream *subs, implicit_fb ? SND_USB_ENDPOINT_TYPE_DATA : SND_USB_ENDPOINT_TYPE_SYNC); + if (!subs->sync_endpoint) { if (is_playback && attr == USB_ENDPOINT_SYNC_NONE) return 0; return -EINVAL; } + subs->sync_endpoint->is_implicit_feedback = implicit_fb; + subs->data_endpoint->sync_master = subs->sync_endpoint; return 0; -- 2.26.2

3 9

[PATCH] ALSA: core: Fix control device release issue
by 朱灿灿 02 Jun '20

02 Jun '20

From: zhucancan <zhucancan(a)vivo.com> Date: Tue, 2 Jun 2020 16:22:57 +0800 Subject: [PATCH] ALSA: core: Fix control device release issue We use snd_pcm_add_usr_ctls() in component's .pcm_new(), unfortunately snd_soc_dapm_add_routes() meets error during adding card->dapm_routes/of_dapm_routes, it will goto probe_end to call soc_cleanup_card_resources(). The commit dc82e52492f6 ("ALSA: core: Assure control device to be registered at last") will make pcm device release is prior to control device release, but the control device needs to use pcm pointor, which is already freed by pcm device release. [ 70.056000] Unable to handle kernel paging request at virtual address ffffff80f2d2c480 [ 70.066139] init: processing action (init.svc.media=*) from (/system/etc/init/mediaserver.rc:1) [ 70.076311] Mem abort info: [ 70.126679] ESR = 0x96000047 [ 70.126685] EC = 0x25: DABT (current EL), IL = 32 bits [ 70.126689] SET = 0, FnV = 0 [ 70.126693] EA = 0, S1PTW = 0 [ 70.126696] Data abort info: [ 70.126700] ISV = 0, ISS = 0x00000047 [ 70.126704] CM = 0, WnR = 1 [ 70.126710] swapper pgtable: 4k pages, 39-bit VAs, pgdp=00000000a216b000 [ 70.126715] [ffffff80f2d2c480] pgd=00000001ffb7f003, pud=00000001ffb7f003, pmd=00000001ff9e8003, pte=0068000172d2cf12 [ 70.126731] Internal error: Oops: 96000047 [#1] PREEMPT SMP [ 70.126821] CPU: 0 PID: 5 Comm: kworker/0:0 Tainted: G W O 5.4.12-qgki-debug-g4c7ca9f20 #10 [ 70.126840] Workqueue: events deferred_probe_work_func$2b500e852cead69fde05a2b98ceb2fa6.cfi_jt [ 70.126845] pstate: 80c00005 (Nzcv daif +PAN +UAO) [ 70.126854] pc : pcm_usr_ctl_private_free$165f4400ee6732c7c8c8157be0f72518+0x20/0x34 [ 70.126861] lr : snd_ctl_remove+0xf4/0x154 [ 70.126862] sp : ffffffc01007b750 [ 70.126865] x29: ffffffc01007b750 x28: ffffffd973c2bb2c [ 70.126868] x27: ffffff8107b741a0 x26: dead000000000100 [ 70.126874] x25: 0000000000000002 x24: ffffffd973c2bb8c [ 70.126877] x23: ffffffd975027c80 x22: ffffffd97415437f [ 70.126880] x21: 0000000000000001 x20: ffffffd973c37e10 [ 70.126883] x19: ffffff80f2d55d00 x18: ffffff80451de4d0 [ 70.126886] x17: ffffff80f2d58000 x16: 0000000000000400 [ 70.126889] x15: a6fe4e9bdc5a6b05 x14: 0000000000000000 [ 70.126892] x13: ffffffff03ab5408 x12: 0000000000000001 [ 70.126896] x11: 7079542070704120 x10: 0000000000000398 [ 70.126899] x9 : 0000000000000001 x8 : ffffff80f2d2c398 [ 70.126902] x7 : 0000000000000000 x6 : ffffff80f6b4d178 [ 70.126905] x5 : ffffffd97335fe8c x4 : 0000000000000000 [ 70.126908] x3 : ffffffc01007b650 x2 : ffffffc01007b768 [ 70.126911] x1 : 00000000ffffffff x0 : ffffff811ff41480 [ 70.126916] Call trace: [ 70.126920] pcm_usr_ctl_private_free$165f4400ee6732c7c8c8157be0f72518+0x20/0x34 [ 70.126923] snd_ctl_remove+0xf4/0x154 [ 70.126927] snd_ctl_dev_free$1b8efea49186b79e6e4a39cac5748f1c+0x3c/0x70 [ 70.126930] snd_device_free_all+0x234/0x2c8 [ 70.126934] release_card_device$fb7328386cac0bdf1b3f2f88da79521f+0x24/0xcc [ 70.126941] device_release$d2fdd8912beb71f9ccc7ca72fceb0522+0x48/0x118 [ 70.126949] kobject_cleanup+0x138/0x25c [ 70.126952] kobject_put+0x50/0x60 [ 70.126955] put_device+0x14/0x20 [ 70.126957] snd_card_free+0x60/0x98 [ 70.126963] soc_cleanup_card_resources+0x2c/0x420 [ 70.126966] snd_soc_bind_card+0xcc0/0xfa8 [ 70.126968] snd_soc_register_card+0x110/0x128 [ 70.126974] devm_snd_soc_register_card+0x4c/0x90 Fixes: dc82e52492f6 ("ALSA: core: Assure control device to be registered at last") Signed-off-by: zhucancan <zhucancan(a)vivo.com> --- sound/core/device.c | 8 -------- 1 file changed, 8 deletions(-) diff --git a/sound/core/device.c b/sound/core/device.c index bf0b04a7ee79..1aee0f5623a2 100644 --- a/sound/core/device.c +++ b/sound/core/device.c @@ -225,14 +225,6 @@ void snd_device_free_all(struct snd_card *card) if (snd_BUG_ON(!card)) return; - list_for_each_entry_safe_reverse(dev, next, &card->devices, list) { - /* exception: free ctl and lowlevel stuff later */ - if (dev->type == SNDRV_DEV_CONTROL || - dev->type == SNDRV_DEV_LOWLEVEL) - continue; - __snd_device_free(dev); - } - /* free all */ list_for_each_entry_safe_reverse(dev, next, &card->devices, list) __snd_device_free(dev); -- 2.21.0

2 1

Re: [PATCH] ALSA: core: Fix control device release issue
by Markus Elfring 02 Jun '20

02 Jun '20

> We use snd_pcm_add_usr_ctls() in component's .pcm_new(), > unfortunately snd_soc_dapm_add_routes() meets error during > adding card->dapm_routes/of_dapm_routes, it will goto probe_end > to call soc_cleanup_card_resources(). > > The commit dc82e52492f6 ("ALSA: core: Assure control device > to be registered at last") will make pcm device release is > prior to control device release, but the control device needs > to use pcm pointor, which is already freed by pcm device release. I suggest to improve this change description. Regards, Markus

1 0

Db845c
by GitHub pull_request - opened 02 Jun '20

02 Jun '20

alsa-project/alsa-ucm-conf pull request #29 was opened from Srinivas-Kandagatla: Hi Jaroslav Kysela, I removed Lenovo ucm2 configs for now, are you okay to pull DragonBoard DB845c ucm2 for now till we resolve in general naming issues with DT. All the drivers and DT are already in mainline since 5.6. Thanks, srini Changes sinc v2: - remove Lenovo ucm2 - remove cdev references. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/29 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/29.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] soundwire: clarify SPDX use of GPL-2.0
by Bard Liao 02 Jun '20

02 Jun '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Change SPDX from GPL-2.0 to GPL-2.0-only for Intel-contributed code. This was explicit before the transition to SPDX and lost in translation. No change to the dual-license parts, the only clarification is to the GPL-2.0 term. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/bus.c | 2 +- drivers/soundwire/bus.h | 2 +- drivers/soundwire/bus_type.c | 2 +- drivers/soundwire/cadence_master.c | 2 +- drivers/soundwire/cadence_master.h | 2 +- drivers/soundwire/intel.c | 2 +- drivers/soundwire/intel.h | 2 +- drivers/soundwire/intel_init.c | 2 +- drivers/soundwire/mipi_disco.c | 2 +- drivers/soundwire/slave.c | 2 +- drivers/soundwire/stream.c | 2 +- include/linux/soundwire/sdw.h | 2 +- include/linux/soundwire/sdw_intel.h | 2 +- include/linux/soundwire/sdw_registers.h | 2 +- include/linux/soundwire/sdw_type.h | 2 +- 15 files changed, 15 insertions(+), 15 deletions(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index 24ba77226376..2289c2ac8c5a 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -1,4 +1,4 @@ -// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +// SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) // Copyright(c) 2015-17 Intel Corporation. #include <linux/acpi.h> diff --git a/drivers/soundwire/bus.h b/drivers/soundwire/bus.h index 82484f741168..697a5b371568 100644 --- a/drivers/soundwire/bus.h +++ b/drivers/soundwire/bus.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) */ +/* SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) */ /* Copyright(c) 2015-17 Intel Corporation. */ #ifndef __SDW_BUS_H diff --git a/drivers/soundwire/bus_type.c b/drivers/soundwire/bus_type.c index de9a671802b8..95a87d4e26c2 100644 --- a/drivers/soundwire/bus_type.c +++ b/drivers/soundwire/bus_type.c @@ -1,4 +1,4 @@ -// SPDX-License-Identifier: GPL-2.0 +// SPDX-License-Identifier: GPL-2.0-only // Copyright(c) 2015-17 Intel Corporation. #include <linux/module.h> diff --git a/drivers/soundwire/cadence_master.c b/drivers/soundwire/cadence_master.c index 9ea87538b9ef..fe238645ed2c 100644 --- a/drivers/soundwire/cadence_master.c +++ b/drivers/soundwire/cadence_master.c @@ -1,4 +1,4 @@ -// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +// SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) // Copyright(c) 2015-17 Intel Corporation. /* diff --git a/drivers/soundwire/cadence_master.h b/drivers/soundwire/cadence_master.h index b410656f8194..ea606b4fba76 100644 --- a/drivers/soundwire/cadence_master.h +++ b/drivers/soundwire/cadence_master.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) */ +/* SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) */ /* Copyright(c) 2015-17 Intel Corporation. */ #include <sound/soc.h> diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c index 4cfdd074e310..1eda63f488f5 100644 --- a/drivers/soundwire/intel.c +++ b/drivers/soundwire/intel.c @@ -1,4 +1,4 @@ -// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +// SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) // Copyright(c) 2015-17 Intel Corporation. /* diff --git a/drivers/soundwire/intel.h b/drivers/soundwire/intel.h index 38b7c125fb10..7b4af8018e1a 100644 --- a/drivers/soundwire/intel.h +++ b/drivers/soundwire/intel.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) */ +/* SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) */ /* Copyright(c) 2015-17 Intel Corporation. */ #ifndef __SDW_INTEL_LOCAL_H diff --git a/drivers/soundwire/intel_init.c b/drivers/soundwire/intel_init.c index d5d42795a48f..947345d5c960 100644 --- a/drivers/soundwire/intel_init.c +++ b/drivers/soundwire/intel_init.c @@ -1,4 +1,4 @@ -// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +// SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) // Copyright(c) 2015-17 Intel Corporation. /* diff --git a/drivers/soundwire/mipi_disco.c b/drivers/soundwire/mipi_disco.c index 4ae62b452b8c..dbd8fc6f4dad 100644 --- a/drivers/soundwire/mipi_disco.c +++ b/drivers/soundwire/mipi_disco.c @@ -1,4 +1,4 @@ -// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +// SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) // Copyright(c) 2015-17 Intel Corporation. /* diff --git a/drivers/soundwire/slave.c b/drivers/soundwire/slave.c index 0839445ee07b..8e8b62dff51c 100644 --- a/drivers/soundwire/slave.c +++ b/drivers/soundwire/slave.c @@ -1,4 +1,4 @@ -// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +// SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) // Copyright(c) 2015-17 Intel Corporation. #include <linux/acpi.h> diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index a9a72574b34a..296cea48bdcc 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -1,4 +1,4 @@ -// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +// SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) // Copyright(c) 2015-18 Intel Corporation. /* diff --git a/include/linux/soundwire/sdw.h b/include/linux/soundwire/sdw.h index 9c27a32df9bb..1fde044f5f3e 100644 --- a/include/linux/soundwire/sdw.h +++ b/include/linux/soundwire/sdw.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) */ +/* SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) */ /* Copyright(c) 2015-17 Intel Corporation. */ #ifndef __SOUNDWIRE_H diff --git a/include/linux/soundwire/sdw_intel.h b/include/linux/soundwire/sdw_intel.h index 979b41b5dcb4..96a8633339a1 100644 --- a/include/linux/soundwire/sdw_intel.h +++ b/include/linux/soundwire/sdw_intel.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) */ +/* SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) */ /* Copyright(c) 2015-17 Intel Corporation. */ #ifndef __SDW_INTEL_H diff --git a/include/linux/soundwire/sdw_registers.h b/include/linux/soundwire/sdw_registers.h index a686f7988156..06d895168b32 100644 --- a/include/linux/soundwire/sdw_registers.h +++ b/include/linux/soundwire/sdw_registers.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) */ +/* SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) */ /* Copyright(c) 2015-17 Intel Corporation. */ #ifndef __SDW_REGISTERS_H diff --git a/include/linux/soundwire/sdw_type.h b/include/linux/soundwire/sdw_type.h index 52eb66cd11bc..25ab91f5c6be 100644 --- a/include/linux/soundwire/sdw_type.h +++ b/include/linux/soundwire/sdw_type.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0-only */ /* Copyright(c) 2015-17 Intel Corporation. */ #ifndef __SOUNDWIRE_TYPES_H -- 2.17.1

3 3

Pioneer DJ DJM-900NXS2 support patches
by Dmitry Panchenko | d-Systems 01 Jun '20

01 Jun '20

Pioneer DJ DJM-900NXS2 is a widely used DJ mixer with 2 audio USB interfaces. Both have a MIDI controller, 10 playback and 12 capture channels. Audio endpoints are vendor-specific and 3 files need to be patched. All playback and capture channels work fine with all supported sample rates (44.1k, 48k, 96k). Patches are attached. -- Kind regards, Dmitry Panchenko d-Systems Ltd Lai 6, Tartu 51005, Estonia www.d-systems.ee

3 5

[RFC][PATCH] ASoC: qcom: q6asm-dai: kCFI fix
by John Stultz 01 Jun '20

01 Jun '20

Fixes the following kCFI crash seen on db845c, caused by the function prototypes not matching the callback function prototype. [ 82.585661] Unable to handle kernel NULL pointer dereference at virtual address 0000000000000001 [ 82.595387] Mem abort info: [ 82.599463] ESR = 0x96000005 [ 82.602658] EC = 0x25: DABT (current EL), IL = 32 bits [ 82.608177] SET = 0, FnV = 0 [ 82.611829] EA = 0, S1PTW = 0 [ 82.615369] Data abort info: [ 82.618751] ISV = 0, ISS = 0x00000005 [ 82.622641] CM = 0, WnR = 0 [ 82.625774] user pgtable: 4k pages, 39-bit VAs, pgdp=0000000174259000 [ 82.632292] [0000000000000001] pgd=0000000000000000, pud=0000000000000000 [ 82.639167] Internal error: Oops: 96000005 [#1] PREEMPT SMP [ 82.644795] Modules linked in: hci_uart btqca xhci_plat_hcd xhci_pci_renesas xhci_pci xhci_hcd wcn36xx wcnss_ctrl wcd934x vctrl_regulator ufs_qcom syscon_reboot_e [ 82.644927] qcom_apcs_ipc_mailbox q6asm_dai q6routing q6asm q6afe_dai q6adm q6afe q6core q6dsp_common pm8941_pwrkey pm8916_wdt platform_mhu pinctrl_spmi_mpp pine [ 82.812982] CPU: 3 PID: 240 Comm: kworker/u16:4 Tainted: G W 5.6.0-rc7-mainline-00960-g0c34353d11b9-dirty #1 [ 82.824201] Hardware name: Thundercomm Dragonboard 845c (DT) [ 82.829937] Workqueue: qcom_apr_rx apr_rxwq [apr] [ 82.834698] pstate: 80c00005 (Nzcv daif +PAN +UAO) [ 82.839553] pc : __cfi_check_fail+0x4/0x1c [q6asm_dai] [ 82.844754] lr : __cfi_check+0x3a8/0x3b0 [q6asm_dai] [ 82.849767] sp : ffffffc0105f3c20 [ 82.853123] x29: ffffffc0105f3c30 x28: 0000000000000020 [ 82.858489] x27: ffffff80f4588400 x26: ffffff80f458ec94 [ 82.863854] x25: ffffff80f458ece8 x24: ffffffe3670c7000 [ 82.869220] x23: ffffff8094bb7b34 x22: ffffffe367137000 [ 82.874585] x21: bd07909b332eada6 x20: 0000000000000001 [ 82.879950] x19: ffffffe36713863c x18: ffffff80f8df4430 [ 82.885316] x17: 0000000000000001 x16: ffffffe39d15e660 [ 82.890681] x15: 0000000000000001 x14: 0000000000000027 [ 82.896047] x13: 0000000000000000 x12: ffffffe39e6465a0 [ 82.901413] x11: 0000000000000051 x10: 000000000000ffff [ 82.906779] x9 : 000ffffffe366c19 x8 : c3c5f18762d1ceef [ 82.912145] x7 : 0000000000000000 x6 : ffffffc010877698 [ 82.917511] x5 : ffffffc0105f3c00 x4 : 0000000000000000 [ 82.922877] x3 : 0000000000000000 x2 : 0000000000000001 [ 82.928243] x1 : ffffffe36713863c x0 : 0000000000000001 [ 82.933610] Call trace: [ 82.936099] __cfi_check_fail+0x4/0x1c [q6asm_dai] [ 82.940955] q6asm_srvc_callback+0x22c/0x618 [q6asm] [ 82.945973] apr_rxwq+0x1a8/0x27c [apr] [ 82.949861] process_one_work+0x2e8/0x54c [ 82.953919] worker_thread+0x27c/0x4d4 [ 82.957715] kthread+0x144/0x154 [ 82.960985] ret_from_fork+0x10/0x18 [ 82.964603] Code: a8c37bfd f85f8e5e d65f03c0 b40000a0 (39400008) [ 82.970762] ---[ end trace 410accb839617143 ]--- [ 82.975429] Kernel panic - not syncing: Fatal exception 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: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Cc: Vinod Koul <vkoul(a)kernel.org> Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Cc: Stephan Gerhold <stephan(a)gerhold.net> Cc: Sami Tolvanen <samitolvanen(a)google.com> Cc: Todd Kjos <tkjos(a)google.com> Cc: Alistair Delva <adelva(a)google.com> Cc: Amit Pundir <amit.pundir(a)linaro.org> Cc: Sumit Semwal <sumit.semwal(a)linaro.org> Cc: alsa-devel(a)alsa-project.org Signed-off-by: John Stultz <john.stultz(a)linaro.org> --- sound/soc/qcom/qdsp6/q6asm-dai.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index 125af00bba53..4640804aab7f 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -176,7 +176,7 @@ static const struct snd_compr_codec_caps q6asm_compr_caps = { }; static void event_handler(uint32_t opcode, uint32_t token, - uint32_t *payload, void *priv) + void *payload, void *priv) { struct q6asm_dai_rtd *prtd = priv; struct snd_pcm_substream *substream = prtd->substream; @@ -490,7 +490,7 @@ static int q6asm_dai_hw_params(struct snd_soc_component *component, } static void compress_event_handler(uint32_t opcode, uint32_t token, - uint32_t *payload, void *priv) + void *payload, void *priv) { struct q6asm_dai_rtd *prtd = priv; struct snd_compr_stream *substream = prtd->cstream; -- 2.17.1

3 3

SOF 1.5 release on FTP server
by Paul Dann 01 Jun '20

01 Jun '20

Hi there, I'm eagerly awaiting the sof-firmware 1.5 builds to land on your server: https://www.alsa-project.org/files/pub/misc/sof/ Archlinux builds its sof-firmware package from there. SOF 1.5 was tagged on the 27th April on Github. I'm wondering if there are plans to release a build for this version to the same place? Many thanks, Paul

2 1

[V6 PATCH 2/2] ASoC: max98390: Added Amplifier Driver
by Steve Lee 01 Jun '20

01 Jun '20

This is the initial amplifier driver for max98390. Reported-by: kbuild test robot <lkp(a)intel.com> Signed-off-by: Steve Lee <steves.lee(a)maximintegrated.com> --- Changes since V5: * Fixed warning message reported by kbuild. * Finalize default register value. * Update missed AMP_EN register setting during dsm_init. * Modified SPDEX header style. Changes since V4: * Revert return calibration status in max98390_dsm_calib_get: There is no need to return calibration status in get function because there is mixer control lock protecting mixer control access. And set function of DSM calib finishes calibration synchronously. * Fix sprintf in max98390_dsm_init func: Use snprintf to avoid any unreasonable value from DMI. Changes since V3: * Add operation at max98390_dsm_calib_get: Add cal_running variable at max98390_priv structure Add return cal_running at max98390_dsm_calib_get Update cal_running value during calibration * Update max98390_dsm_init func Add dsm_param bin name can select upon vendor and product name. Changes since V2: * Removed warn massage in max98390_dsm_calib_get func and add comment. Changes since V1: * Updated according to Mark's comments. * Updated according to Tzung-Bi's comments. * Removed Enum structure and use TLV data for Volume control sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max98390.c | 1041 +++++++++++++++++++++++++++++++++++ sound/soc/codecs/max98390.h | 663 ++++++++++++++++++++++ 4 files changed, 1711 insertions(+) create mode 100644 sound/soc/codecs/max98390.c create mode 100644 sound/soc/codecs/max98390.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e60e0b6a689c..93f88cd9970a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -116,6 +116,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_MAX98926 imply SND_SOC_MAX98927 imply SND_SOC_MAX98373 + imply SND_SOC_MAX98390 imply SND_SOC_MAX9850 imply SND_SOC_MAX9860 imply SND_SOC_MAX9759 @@ -867,6 +868,10 @@ config SND_SOC_MAX98373 tristate "Maxim Integrated MAX98373 Speaker Amplifier" depends on I2C +config SND_SOC_MAX98390 + tristate "Maxim Integrated MAX98390 Speaker Amplifier" + depends on I2C + config SND_SOC_MAX9850 tristate depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 03533157cda6..e7c17f701529 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -115,6 +115,7 @@ snd-soc-max98925-objs := max98925.o snd-soc-max98926-objs := max98926.o snd-soc-max98927-objs := max98927.o snd-soc-max98373-objs := max98373.o +snd-soc-max98390-objs := max98390.o snd-soc-max9850-objs := max9850.o snd-soc-max9860-objs := max9860.o snd-soc-mc13783-objs := mc13783.o @@ -415,6 +416,7 @@ obj-$(CONFIG_SND_SOC_MAX98925) += snd-soc-max98925.o obj-$(CONFIG_SND_SOC_MAX98926) += snd-soc-max98926.o obj-$(CONFIG_SND_SOC_MAX98927) += snd-soc-max98927.o obj-$(CONFIG_SND_SOC_MAX98373) += snd-soc-max98373.o +obj-$(CONFIG_SND_SOC_MAX98390) += snd-soc-max98390.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MAX9860) += snd-soc-max9860.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c new file mode 100644 index 000000000000..dc02ee151b52 --- /dev/null +++ b/sound/soc/codecs/max98390.c @@ -0,0 +1,1041 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +// +// max98390.c -- MAX98390 ALSA Soc Audio driver +// +// Copyright (C) 2020 Maxim Integrated Products + +#include <linux/acpi.h> +#include <linux/cdev.h> +#include <linux/dmi.h> +#include <linux/firmware.h> +#include <linux/gpio.h> +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/of_gpio.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/time.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include "max98390.h" + +static struct reg_default max98390_reg_defaults[] = { + {MAX98390_INT_EN1, 0xf0}, + {MAX98390_INT_EN2, 0x00}, + {MAX98390_INT_EN3, 0x00}, + {MAX98390_INT_FLAG_CLR1, 0x00}, + {MAX98390_INT_FLAG_CLR2, 0x00}, + {MAX98390_INT_FLAG_CLR3, 0x00}, + {MAX98390_IRQ_CTRL, 0x01}, + {MAX98390_CLK_MON, 0x6d}, + {MAX98390_DAT_MON, 0x03}, + {MAX98390_WDOG_CTRL, 0x00}, + {MAX98390_WDOG_RST, 0x00}, + {MAX98390_MEAS_ADC_THERM_WARN_THRESH, 0x75}, + {MAX98390_MEAS_ADC_THERM_SHDN_THRESH, 0x8c}, + {MAX98390_MEAS_ADC_THERM_HYSTERESIS, 0x08}, + {MAX98390_PIN_CFG, 0x55}, + {MAX98390_PCM_RX_EN_A, 0x00}, + {MAX98390_PCM_RX_EN_B, 0x00}, + {MAX98390_PCM_TX_EN_A, 0x00}, + {MAX98390_PCM_TX_EN_B, 0x00}, + {MAX98390_PCM_TX_HIZ_CTRL_A, 0xff}, + {MAX98390_PCM_TX_HIZ_CTRL_B, 0xff}, + {MAX98390_PCM_CH_SRC_1, 0x00}, + {MAX98390_PCM_CH_SRC_2, 0x00}, + {MAX98390_PCM_CH_SRC_3, 0x00}, + {MAX98390_PCM_MODE_CFG, 0xc0}, + {MAX98390_PCM_MASTER_MODE, 0x1c}, + {MAX98390_PCM_CLK_SETUP, 0x44}, + {MAX98390_PCM_SR_SETUP, 0x08}, + {MAX98390_ICC_RX_EN_A, 0x00}, + {MAX98390_ICC_RX_EN_B, 0x00}, + {MAX98390_ICC_TX_EN_A, 0x00}, + {MAX98390_ICC_TX_EN_B, 0x00}, + {MAX98390_ICC_HIZ_MANUAL_MODE, 0x00}, + {MAX98390_ICC_TX_HIZ_EN_A, 0x00}, + {MAX98390_ICC_TX_HIZ_EN_B, 0x00}, + {MAX98390_ICC_LNK_EN, 0x00}, + {MAX98390_R2039_AMP_DSP_CFG, 0x0f}, + {MAX98390_R203A_AMP_EN, 0x81}, + {MAX98390_TONE_GEN_DC_CFG, 0x00}, + {MAX98390_SPK_SRC_SEL, 0x00}, + {MAX98390_SSM_CFG, 0x85}, + {MAX98390_MEAS_EN, 0x03}, + {MAX98390_MEAS_DSP_CFG, 0x0f}, + {MAX98390_BOOST_CTRL0, 0x1c}, + {MAX98390_BOOST_CTRL3, 0x01}, + {MAX98390_BOOST_CTRL1, 0x40}, + {MAX98390_MEAS_ADC_CFG, 0x07}, + {MAX98390_MEAS_ADC_BASE_MSB, 0x00}, + {MAX98390_MEAS_ADC_BASE_LSB, 0x23}, + {MAX98390_ADC_CH0_DIVIDE, 0x00}, + {MAX98390_ADC_CH1_DIVIDE, 0x00}, + {MAX98390_ADC_CH2_DIVIDE, 0x00}, + {MAX98390_ADC_CH0_FILT_CFG, 0x00}, + {MAX98390_ADC_CH1_FILT_CFG, 0x00}, + {MAX98390_ADC_CH2_FILT_CFG, 0x00}, + {MAX98390_PWR_GATE_CTL, 0x2c}, + {MAX98390_BROWNOUT_EN, 0x00}, + {MAX98390_BROWNOUT_INFINITE_HOLD, 0x00}, + {MAX98390_BROWNOUT_INFINITE_HOLD_CLR, 0x00}, + {MAX98390_BROWNOUT_LVL_HOLD, 0x00}, + {MAX98390_BROWNOUT_LVL1_THRESH, 0x00}, + {MAX98390_BROWNOUT_LVL2_THRESH, 0x00}, + {MAX98390_BROWNOUT_LVL3_THRESH, 0x00}, + {MAX98390_BROWNOUT_LVL4_THRESH, 0x00}, + {MAX98390_BROWNOUT_THRESH_HYSTERYSIS, 0x00}, + {MAX98390_BROWNOUT_AMP_LIMITER_ATK_REL, 0x1f}, + {MAX98390_BROWNOUT_AMP_GAIN_ATK_REL, 0x00}, + {MAX98390_BROWNOUT_AMP1_CLIP_MODE, 0x00}, + {MAX98390_BROWNOUT_LVL1_CUR_LIMIT, 0x00}, + {MAX98390_BROWNOUT_LVL1_AMP1_CTRL1, 0x00}, + {MAX98390_BROWNOUT_LVL1_AMP1_CTRL2, 0x00}, + {MAX98390_BROWNOUT_LVL1_AMP1_CTRL3, 0x00}, + {MAX98390_BROWNOUT_LVL2_CUR_LIMIT, 0x00}, + {MAX98390_BROWNOUT_LVL2_AMP1_CTRL1, 0x00}, + {MAX98390_BROWNOUT_LVL2_AMP1_CTRL2, 0x00}, + {MAX98390_BROWNOUT_LVL2_AMP1_CTRL3, 0x00}, + {MAX98390_BROWNOUT_LVL3_CUR_LIMIT, 0x00}, + {MAX98390_BROWNOUT_LVL3_AMP1_CTRL1, 0x00}, + {MAX98390_BROWNOUT_LVL3_AMP1_CTRL2, 0x00}, + {MAX98390_BROWNOUT_LVL3_AMP1_CTRL3, 0x00}, + {MAX98390_BROWNOUT_LVL4_CUR_LIMIT, 0x00}, + {MAX98390_BROWNOUT_LVL4_AMP1_CTRL1, 0x00}, + {MAX98390_BROWNOUT_LVL4_AMP1_CTRL2, 0x00}, + {MAX98390_BROWNOUT_LVL4_AMP1_CTRL3, 0x00}, + {MAX98390_BROWNOUT_ILIM_HLD, 0x00}, + {MAX98390_BROWNOUT_LIM_HLD, 0x00}, + {MAX98390_BROWNOUT_CLIP_HLD, 0x00}, + {MAX98390_BROWNOUT_GAIN_HLD, 0x00}, + {MAX98390_ENV_TRACK_VOUT_HEADROOM, 0x0f}, + {MAX98390_ENV_TRACK_BOOST_VOUT_DELAY, 0x80}, + {MAX98390_ENV_TRACK_REL_RATE, 0x07}, + {MAX98390_ENV_TRACK_HOLD_RATE, 0x07}, + {MAX98390_ENV_TRACK_CTRL, 0x01}, + {MAX98390_BOOST_BYPASS1, 0x49}, + {MAX98390_BOOST_BYPASS2, 0x2b}, + {MAX98390_BOOST_BYPASS3, 0x08}, + {MAX98390_FET_SCALING1, 0x00}, + {MAX98390_FET_SCALING2, 0x03}, + {MAX98390_FET_SCALING3, 0x00}, + {MAX98390_FET_SCALING4, 0x07}, + {MAX98390_SPK_SPEEDUP, 0x00}, + {DSMIG_WB_DRC_RELEASE_TIME_1, 0x00}, + {DSMIG_WB_DRC_RELEASE_TIME_2, 0x00}, + {DSMIG_WB_DRC_ATTACK_TIME_1, 0x00}, + {DSMIG_WB_DRC_ATTACK_TIME_2, 0x00}, + {DSMIG_WB_DRC_COMPRESSION_RATIO, 0x00}, + {DSMIG_WB_DRC_COMPRESSION_THRESHOLD, 0x00}, + {DSMIG_WB_DRC_MAKEUPGAIN, 0x00}, + {DSMIG_WB_DRC_NOISE_GATE_THRESHOLD, 0x00}, + {DSMIG_WBDRC_HPF_ENABLE, 0x00}, + {DSMIG_WB_DRC_TEST_SMOOTHER_OUT_EN, 0x00}, + {DSMIG_PPR_THRESHOLD, 0x00}, + {DSM_STEREO_BASS_CHANNEL_SELECT, 0x00}, + {DSM_TPROT_THRESHOLD_BYTE0, 0x00}, + {DSM_TPROT_THRESHOLD_BYTE1, 0x00}, + {DSM_TPROT_ROOM_TEMPERATURE_BYTE0, 0x00}, + {DSM_TPROT_ROOM_TEMPERATURE_BYTE1, 0x00}, + {DSM_TPROT_RECIP_RDC_ROOM_BYTE0, 0x00}, + {DSM_TPROT_RECIP_RDC_ROOM_BYTE1, 0x00}, + {DSM_TPROT_RECIP_RDC_ROOM_BYTE2, 0x00}, + {DSM_TPROT_RECIP_TCONST_BYTE0, 0x00}, + {DSM_TPROT_RECIP_TCONST_BYTE1, 0x00}, + {DSM_TPROT_RECIP_TCONST_BYTE2, 0x00}, + {DSM_THERMAL_ATTENUATION_SETTINGS, 0x00}, + {DSM_THERMAL_PILOT_TONE_ATTENUATION, 0x00}, + {DSM_TPROT_PG_TEMP_THRESH_BYTE0, 0x00}, + {DSM_TPROT_PG_TEMP_THRESH_BYTE1, 0x00}, + {DSMIG_DEBUZZER_THRESHOLD, 0x00}, + {DSMIG_DEBUZZER_ALPHA_COEF_TEST_ONLY, 0x08}, + {DSM_VOL_ENA, 0x20}, + {DSM_VOL_CTRL, 0xa0}, + {DSMIG_EN, 0x00}, + {MAX98390_R23E1_DSP_GLOBAL_EN, 0x00}, + {MAX98390_R23FF_GLOBAL_EN, 0x00}, +}; + +static int max98390_dsm_calibrate(struct snd_soc_component *component); + +static int max98390_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_component *component = codec_dai->component; + struct max98390_priv *max98390 = + snd_soc_component_get_drvdata(component); + unsigned int mode; + unsigned int format; + unsigned int invert = 0; + + dev_dbg(component->dev, "%s: fmt 0x%08X\n", __func__, fmt); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + mode = MAX98390_PCM_MASTER_MODE_SLAVE; + break; + case SND_SOC_DAIFMT_CBM_CFM: + max98390->master = true; + mode = MAX98390_PCM_MASTER_MODE_MASTER; + break; + default: + dev_err(component->dev, "DAI clock mode unsupported\n"); + return -EINVAL; + } + + regmap_update_bits(max98390->regmap, + MAX98390_PCM_MASTER_MODE, + MAX98390_PCM_MASTER_MODE_MASK, + mode); + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + invert = MAX98390_PCM_MODE_CFG_PCM_BCLKEDGE; + break; + default: + dev_err(component->dev, "DAI invert mode unsupported\n"); + return -EINVAL; + } + + regmap_update_bits(max98390->regmap, + MAX98390_PCM_MODE_CFG, + MAX98390_PCM_MODE_CFG_PCM_BCLKEDGE, + invert); + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + format = MAX98390_PCM_FORMAT_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + format = MAX98390_PCM_FORMAT_LJ; + break; + case SND_SOC_DAIFMT_DSP_A: + format = MAX98390_PCM_FORMAT_TDM_MODE1; + break; + case SND_SOC_DAIFMT_DSP_B: + format = MAX98390_PCM_FORMAT_TDM_MODE0; + break; + default: + return -EINVAL; + } + + regmap_update_bits(max98390->regmap, + MAX98390_PCM_MODE_CFG, + MAX98390_PCM_MODE_CFG_FORMAT_MASK, + format << MAX98390_PCM_MODE_CFG_FORMAT_SHIFT); + + return 0; +} + +static int max98390_get_bclk_sel(int bclk) +{ + int i; + /* BCLKs per LRCLK */ + static int bclk_sel_table[] = { + 32, 48, 64, 96, 128, 192, 256, 320, 384, 512, + }; + /* match BCLKs per LRCLK */ + for (i = 0; i < ARRAY_SIZE(bclk_sel_table); i++) { + if (bclk_sel_table[i] == bclk) + return i + 2; + } + return 0; +} + +static int max98390_set_clock(struct snd_soc_component *component, + struct snd_pcm_hw_params *params) +{ + struct max98390_priv *max98390 = + snd_soc_component_get_drvdata(component); + /* codec MCLK rate in master mode */ + static int rate_table[] = { + 5644800, 6000000, 6144000, 6500000, + 9600000, 11289600, 12000000, 12288000, + 13000000, 19200000, + }; + /* BCLK/LRCLK ratio calculation */ + int blr_clk_ratio = params_channels(params) + * snd_pcm_format_width(params_format(params)); + int value; + + if (max98390->master) { + int i; + /* match rate to closest value */ + for (i = 0; i < ARRAY_SIZE(rate_table); i++) { + if (rate_table[i] >= max98390->sysclk) + break; + } + if (i == ARRAY_SIZE(rate_table)) { + dev_err(component->dev, "failed to find proper clock rate.\n"); + return -EINVAL; + } + + regmap_update_bits(max98390->regmap, + MAX98390_PCM_MASTER_MODE, + MAX98390_PCM_MASTER_MODE_MCLK_MASK, + i << MAX98390_PCM_MASTER_MODE_MCLK_RATE_SHIFT); + } + + if (!max98390->tdm_mode) { + /* BCLK configuration */ + value = max98390_get_bclk_sel(blr_clk_ratio); + if (!value) { + dev_err(component->dev, "format unsupported %d\n", + params_format(params)); + return -EINVAL; + } + + regmap_update_bits(max98390->regmap, + MAX98390_PCM_CLK_SETUP, + MAX98390_PCM_CLK_SETUP_BSEL_MASK, + value); + } + return 0; +} + +static int max98390_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = + dai->component; + struct max98390_priv *max98390 = + snd_soc_component_get_drvdata(component); + + unsigned int sampling_rate; + unsigned int chan_sz; + + /* pcm mode configuration */ + switch (snd_pcm_format_width(params_format(params))) { + case 16: + chan_sz = MAX98390_PCM_MODE_CFG_CHANSZ_16; + break; + case 24: + chan_sz = MAX98390_PCM_MODE_CFG_CHANSZ_24; + break; + case 32: + chan_sz = MAX98390_PCM_MODE_CFG_CHANSZ_32; + break; + default: + dev_err(component->dev, "format unsupported %d\n", + params_format(params)); + goto err; + } + + regmap_update_bits(max98390->regmap, + MAX98390_PCM_MODE_CFG, + MAX98390_PCM_MODE_CFG_CHANSZ_MASK, chan_sz); + + dev_dbg(component->dev, "format supported %d", + params_format(params)); + + /* sampling rate configuration */ + switch (params_rate(params)) { + case 8000: + sampling_rate = MAX98390_PCM_SR_SET1_SR_8000; + break; + case 11025: + sampling_rate = MAX98390_PCM_SR_SET1_SR_11025; + break; + case 12000: + sampling_rate = MAX98390_PCM_SR_SET1_SR_12000; + break; + case 16000: + sampling_rate = MAX98390_PCM_SR_SET1_SR_16000; + break; + case 22050: + sampling_rate = MAX98390_PCM_SR_SET1_SR_22050; + break; + case 24000: + sampling_rate = MAX98390_PCM_SR_SET1_SR_24000; + break; + case 32000: + sampling_rate = MAX98390_PCM_SR_SET1_SR_32000; + break; + case 44100: + sampling_rate = MAX98390_PCM_SR_SET1_SR_44100; + break; + case 48000: + sampling_rate = MAX98390_PCM_SR_SET1_SR_48000; + break; + default: + dev_err(component->dev, "rate %d not supported\n", + params_rate(params)); + goto err; + } + + /* set DAI_SR to correct LRCLK frequency */ + regmap_update_bits(max98390->regmap, + MAX98390_PCM_SR_SETUP, + MAX98390_PCM_SR_SET1_SR_MASK, + sampling_rate); + + return max98390_set_clock(component, params); +err: + return -EINVAL; +} + +static int max98390_dai_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width) +{ + struct snd_soc_component *component = dai->component; + struct max98390_priv *max98390 = + snd_soc_component_get_drvdata(component); + + int bsel; + unsigned int chan_sz; + + if (!tx_mask && !rx_mask && !slots && !slot_width) + max98390->tdm_mode = false; + else + max98390->tdm_mode = true; + + dev_dbg(component->dev, + "Tdm mode : %d\n", max98390->tdm_mode); + + /* BCLK configuration */ + bsel = max98390_get_bclk_sel(slots * slot_width); + if (!bsel) { + dev_err(component->dev, "BCLK %d not supported\n", + slots * slot_width); + return -EINVAL; + } + + regmap_update_bits(max98390->regmap, + MAX98390_PCM_CLK_SETUP, + MAX98390_PCM_CLK_SETUP_BSEL_MASK, + bsel); + + /* Channel size configuration */ + switch (slot_width) { + case 16: + chan_sz = MAX98390_PCM_MODE_CFG_CHANSZ_16; + break; + case 24: + chan_sz = MAX98390_PCM_MODE_CFG_CHANSZ_24; + break; + case 32: + chan_sz = MAX98390_PCM_MODE_CFG_CHANSZ_32; + break; + default: + dev_err(component->dev, "format unsupported %d\n", + slot_width); + return -EINVAL; + } + + regmap_update_bits(max98390->regmap, + MAX98390_PCM_MODE_CFG, + MAX98390_PCM_MODE_CFG_CHANSZ_MASK, chan_sz); + + /* Rx slot configuration */ + regmap_write(max98390->regmap, + MAX98390_PCM_RX_EN_A, + rx_mask & 0xFF); + regmap_write(max98390->regmap, + MAX98390_PCM_RX_EN_B, + (rx_mask & 0xFF00) >> 8); + + /* Tx slot Hi-Z configuration */ + regmap_write(max98390->regmap, + MAX98390_PCM_TX_HIZ_CTRL_A, + ~tx_mask & 0xFF); + regmap_write(max98390->regmap, + MAX98390_PCM_TX_HIZ_CTRL_B, + (~tx_mask & 0xFF00) >> 8); + + return 0; +} + +static int max98390_dai_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_component *component = dai->component; + struct max98390_priv *max98390 = + snd_soc_component_get_drvdata(component); + + max98390->sysclk = freq; + return 0; +} + +static const struct snd_soc_dai_ops max98390_dai_ops = { + .set_sysclk = max98390_dai_set_sysclk, + .set_fmt = max98390_dai_set_fmt, + .hw_params = max98390_dai_hw_params, + .set_tdm_slot = max98390_dai_tdm_slot, +}; + +static int max98390_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct max98390_priv *max98390 = + snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_update_bits(max98390->regmap, + MAX98390_R203A_AMP_EN, + MAX98390_AMP_EN_MASK, 1); + regmap_update_bits(max98390->regmap, + MAX98390_R23FF_GLOBAL_EN, + MAX98390_GLOBAL_EN_MASK, 1); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(max98390->regmap, + MAX98390_R23FF_GLOBAL_EN, + MAX98390_GLOBAL_EN_MASK, 0); + regmap_update_bits(max98390->regmap, + MAX98390_R203A_AMP_EN, + MAX98390_AMP_EN_MASK, 0); + break; + } + return 0; +} + +static const char * const max98390_switch_text[] = { + "Left", "Right", "LeftRight"}; + +static const char * const max98390_boost_voltage_text[] = { + "6.5V", "6.625V", "6.75V", "6.875V", "7V", "7.125V", "7.25V", "7.375V", + "7.5V", "7.625V", "7.75V", "7.875V", "8V", "8.125V", "8.25V", "8.375V", + "8.5V", "8.625V", "8.75V", "8.875V", "9V", "9.125V", "9.25V", "9.375V", + "9.5V", "9.625V", "9.75V", "9.875V", "10V" +}; + +static SOC_ENUM_SINGLE_DECL(max98390_boost_voltage, + MAX98390_BOOST_CTRL0, 0, + max98390_boost_voltage_text); + +static DECLARE_TLV_DB_SCALE(max98390_spk_tlv, 300, 300, 0); +static DECLARE_TLV_DB_SCALE(max98390_digital_tlv, -8000, 50, 0); + +static const char * const max98390_current_limit_text[] = { + "0.00A", "0.50A", "1.00A", "1.05A", "1.10A", "1.15A", "1.20A", "1.25A", + "1.30A", "1.35A", "1.40A", "1.45A", "1.50A", "1.55A", "1.60A", "1.65A", + "1.70A", "1.75A", "1.80A", "1.85A", "1.90A", "1.95A", "2.00A", "2.05A", + "2.10A", "2.15A", "2.20A", "2.25A", "2.30A", "2.35A", "2.40A", "2.45A", + "2.50A", "2.55A", "2.60A", "2.65A", "2.70A", "2.75A", "2.80A", "2.85A", + "2.90A", "2.95A", "3.00A", "3.05A", "3.10A", "3.15A", "3.20A", "3.25A", + "3.30A", "3.35A", "3.40A", "3.45A", "3.50A", "3.55A", "3.60A", "3.65A", + "3.70A", "3.75A", "3.80A", "3.85A", "3.90A", "3.95A", "4.00A", "4.05A", + "4.10A" +}; + +static SOC_ENUM_SINGLE_DECL(max98390_current_limit, + MAX98390_BOOST_CTRL1, 0, + max98390_current_limit_text); + +static int max98390_ref_rdc_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct max98390_priv *max98390 = + snd_soc_component_get_drvdata(component); + + max98390->ref_rdc_value = ucontrol->value.integer.value[0]; + + regmap_write(max98390->regmap, DSM_TPROT_RECIP_RDC_ROOM_BYTE0, + max98390->ref_rdc_value & 0x000000ff); + regmap_write(max98390->regmap, DSM_TPROT_RECIP_RDC_ROOM_BYTE1, + (max98390->ref_rdc_value >> 8) & 0x000000ff); + regmap_write(max98390->regmap, DSM_TPROT_RECIP_RDC_ROOM_BYTE2, + (max98390->ref_rdc_value >> 16) & 0x000000ff); + + return 0; +} + +static int max98390_ref_rdc_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct max98390_priv *max98390 = + snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = max98390->ref_rdc_value; + + return 0; +} + +static int max98390_ambient_temp_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct max98390_priv *max98390 = + snd_soc_component_get_drvdata(component); + + max98390->ambient_temp_value = ucontrol->value.integer.value[0]; + + regmap_write(max98390->regmap, DSM_TPROT_ROOM_TEMPERATURE_BYTE1, + (max98390->ambient_temp_value >> 8) & 0x000000ff); + regmap_write(max98390->regmap, DSM_TPROT_ROOM_TEMPERATURE_BYTE0, + (max98390->ambient_temp_value) & 0x000000ff); + + return 0; +} + +static int max98390_ambient_temp_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct max98390_priv *max98390 = + snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = max98390->ambient_temp_value; + + return 0; +} + +static int max98390_adaptive_rdc_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + + dev_warn(component->dev, "Put adaptive rdc not supported\n"); + + return 0; +} + +static int max98390_adaptive_rdc_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int rdc, rdc0; + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct max98390_priv *max98390 = + snd_soc_component_get_drvdata(component); + + regmap_read(max98390->regmap, THERMAL_RDC_RD_BACK_BYTE1, &rdc); + regmap_read(max98390->regmap, THERMAL_RDC_RD_BACK_BYTE0, &rdc0); + ucontrol->value.integer.value[0] = rdc0 | rdc << 8; + + return 0; +} + +static int max98390_dsm_calib_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + /* Do nothing */ + return 0; +} + +static int max98390_dsm_calib_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + + max98390_dsm_calibrate(component); + + return 0; +} + +static const struct snd_kcontrol_new max98390_snd_controls[] = { + SOC_SINGLE_TLV("Digital Volume", DSM_VOL_CTRL, + 0, 184, 0, + max98390_digital_tlv), + SOC_SINGLE_TLV("Speaker Volume", MAX98390_R203D_SPK_GAIN, + 0, 6, 0, + max98390_spk_tlv), + SOC_SINGLE("Ramp Up Bypass Switch", MAX98390_R2039_AMP_DSP_CFG, + MAX98390_AMP_DSP_CFG_RMP_UP_SHIFT, 1, 0), + SOC_SINGLE("Ramp Down Bypass Switch", MAX98390_R2039_AMP_DSP_CFG, + MAX98390_AMP_DSP_CFG_RMP_DN_SHIFT, 1, 0), + SOC_SINGLE("Boost Clock Phase", MAX98390_BOOST_CTRL3, + MAX98390_BOOST_CLK_PHASE_CFG_SHIFT, 3, 0), + SOC_ENUM("Boost Output Voltage", max98390_boost_voltage), + SOC_ENUM("Current Limit", max98390_current_limit), + SOC_SINGLE_EXT("DSM Rdc", SND_SOC_NOPM, 0, 0xffffff, 0, + max98390_ref_rdc_get, max98390_ref_rdc_put), + SOC_SINGLE_EXT("DSM Ambient Temp", SND_SOC_NOPM, 0, 0xffff, 0, + max98390_ambient_temp_get, max98390_ambient_temp_put), + SOC_SINGLE_EXT("DSM Adaptive Rdc", SND_SOC_NOPM, 0, 0xffff, 0, + max98390_adaptive_rdc_get, max98390_adaptive_rdc_put), + SOC_SINGLE_EXT("DSM Calibration", SND_SOC_NOPM, 0, 1, 0, + max98390_dsm_calib_get, max98390_dsm_calib_put), +}; + +static const struct soc_enum dai_sel_enum = + SOC_ENUM_SINGLE(MAX98390_PCM_CH_SRC_1, + MAX98390_PCM_RX_CH_SRC_SHIFT, + 3, max98390_switch_text); + +static const struct snd_kcontrol_new max98390_dai_controls = + SOC_DAPM_ENUM("DAI Sel", dai_sel_enum); + +static const struct snd_soc_dapm_widget max98390_dapm_widgets[] = { + SND_SOC_DAPM_DAC_E("Amp Enable", "HiFi Playback", + MAX98390_R203A_AMP_EN, 0, 0, max98390_dac_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX("DAI Sel Mux", SND_SOC_NOPM, 0, 0, + &max98390_dai_controls), + SND_SOC_DAPM_OUTPUT("BE_OUT"), +}; + +static const struct snd_soc_dapm_route max98390_audio_map[] = { + /* Plabyack */ + {"DAI Sel Mux", "Left", "Amp Enable"}, + {"DAI Sel Mux", "Right", "Amp Enable"}, + {"DAI Sel Mux", "LeftRight", "Amp Enable"}, + {"BE_OUT", NULL, "DAI Sel Mux"}, +}; + +static bool max98390_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98390_SOFTWARE_RESET ... MAX98390_INT_EN3: + case MAX98390_IRQ_CTRL ... MAX98390_WDOG_CTRL: + case MAX98390_MEAS_ADC_THERM_WARN_THRESH + ... MAX98390_BROWNOUT_INFINITE_HOLD: + case MAX98390_BROWNOUT_LVL_HOLD ... THERMAL_COILTEMP_RD_BACK_BYTE0: + case DSMIG_DEBUZZER_THRESHOLD ... MAX98390_R24FF_REV_ID: + return true; + default: + return false; + } +}; + +static bool max98390_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98390_SOFTWARE_RESET ... MAX98390_INT_EN3: + case MAX98390_MEAS_ADC_CH0_READ ... MAX98390_MEAS_ADC_CH2_READ: + case MAX98390_PWR_GATE_STATUS ... MAX98390_BROWNOUT_STATUS: + case MAX98390_BROWNOUT_LOWEST_STATUS: + case MAX98390_ENV_TRACK_BOOST_VOUT_READ: + case DSM_STBASS_HPF_B0_BYTE0 ... DSM_DEBUZZER_ATTACK_TIME_BYTE2: + case THERMAL_RDC_RD_BACK_BYTE1 ... THERMAL_COILTEMP_RD_BACK_BYTE0: + case DSM_THERMAL_GAIN ... DSM_WBDRC_GAIN: + return true; + default: + return false; + } +} + +#define MAX98390_RATES SNDRV_PCM_RATE_8000_48000 + +#define MAX98390_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver max98390_dai[] = { + { + .name = "max98390-aif1", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = MAX98390_RATES, + .formats = MAX98390_FORMATS, + }, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 1, + .channels_max = 2, + .rates = MAX98390_RATES, + .formats = MAX98390_FORMATS, + }, + .ops = &max98390_dai_ops, + } +}; + +static int max98390_dsm_init(struct snd_soc_component *component) +{ + int ret; + char filename[128]; + const char *vendor, *product; + struct max98390_priv *max98390 = + snd_soc_component_get_drvdata(component); + const struct firmware *fw; + char *dsm_param; + + vendor = dmi_get_system_info(DMI_SYS_VENDOR); + product = dmi_get_system_info(DMI_PRODUCT_NAME); + + if (vendor && product) { + snprintf(filename, sizeof(filename), "dsm_param_%s_%s.bin", + vendor, product); + } else { + sprintf(filename, "dsm_param.bin"); + } + ret = request_firmware(&fw, filename, component->dev); + if (ret) { + ret = request_firmware(&fw, "dsm_param.bin", component->dev); + if (ret) + goto err; + } + + regmap_write(max98390->regmap, MAX98390_R203A_AMP_EN, 0x80); + + dev_dbg(component->dev, + "max98390: param fw size %zu\n", + fw->size); + dsm_param = (char *)fw->data; + dsm_param += MAX98390_DSM_PAYLOAD_OFFSET; + regmap_bulk_write(max98390->regmap, DSM_EQ_BQ1_B0_BYTE0, + dsm_param, + fw->size - MAX98390_DSM_PAYLOAD_OFFSET); + release_firmware(fw); + regmap_write(max98390->regmap, MAX98390_R23E1_DSP_GLOBAL_EN, 0x01); + +err: + return ret; +} + +static int max98390_dsm_calibrate(struct snd_soc_component *component) +{ + unsigned int rdc, rdc_cal_result, temp; + unsigned int rdc_integer, rdc_factor; + struct max98390_priv *max98390 = + snd_soc_component_get_drvdata(component); + + regmap_write(max98390->regmap, MAX98390_R203A_AMP_EN, 0x81); + regmap_write(max98390->regmap, MAX98390_R23FF_GLOBAL_EN, 0x01); + + regmap_read(max98390->regmap, + THERMAL_RDC_RD_BACK_BYTE1, &rdc); + regmap_read(max98390->regmap, + THERMAL_RDC_RD_BACK_BYTE0, &rdc_cal_result); + rdc_cal_result |= (rdc << 8) & 0x0000FFFF; + if (rdc_cal_result) + max98390->ref_rdc_value = 268435456U / rdc_cal_result; + + regmap_read(max98390->regmap, MAX98390_MEAS_ADC_CH2_READ, &temp); + max98390->ambient_temp_value = temp * 52 - 1188; + + rdc_integer = rdc_cal_result * 937 / 65536; + rdc_factor = ((rdc_cal_result * 937 * 100) / 65536) + - (rdc_integer * 100); + + dev_info(component->dev, "rdc resistance about %d.%02d ohm, reg=0x%X temp reg=0x%X\n", + rdc_integer, rdc_factor, rdc_cal_result, temp); + + regmap_write(max98390->regmap, MAX98390_R23FF_GLOBAL_EN, 0x00); + regmap_write(max98390->regmap, MAX98390_R203A_AMP_EN, 0x80); + + return 0; +} + +static int max98390_probe(struct snd_soc_component *component) +{ + struct max98390_priv *max98390 = + snd_soc_component_get_drvdata(component); + + regmap_write(max98390->regmap, MAX98390_SOFTWARE_RESET, 0x01); + /* Sleep reset settle time */ + msleep(20); + /* Update dsm bin param */ + max98390_dsm_init(component); + + /* Amp Setting */ + regmap_write(max98390->regmap, MAX98390_CLK_MON, 0x6f); + regmap_write(max98390->regmap, MAX98390_DAT_MON, 0x0b); + regmap_write(max98390->regmap, MAX98390_PCM_RX_EN_A, 0x03); + regmap_write(max98390->regmap, MAX98390_PWR_GATE_CTL, 0x2d); + regmap_write(max98390->regmap, MAX98390_ENV_TRACK_VOUT_HEADROOM, 0x0e); + regmap_write(max98390->regmap, MAX98390_BOOST_BYPASS1, 0x46); + regmap_write(max98390->regmap, MAX98390_FET_SCALING3, 0x03); + + /* Dsm Setting */ + regmap_write(max98390->regmap, DSM_VOL_CTRL, 0x94); + regmap_write(max98390->regmap, DSMIG_EN, 0x09); + regmap_write(max98390->regmap, MAX98390_R203A_AMP_EN, 0x80); + if (max98390->ref_rdc_value) { + regmap_write(max98390->regmap, DSM_TPROT_RECIP_RDC_ROOM_BYTE0, + max98390->ref_rdc_value & 0x000000ff); + regmap_write(max98390->regmap, DSM_TPROT_RECIP_RDC_ROOM_BYTE1, + (max98390->ref_rdc_value >> 8) & 0x000000ff); + regmap_write(max98390->regmap, DSM_TPROT_RECIP_RDC_ROOM_BYTE2, + (max98390->ref_rdc_value >> 16) & 0x000000ff); + } + if (max98390->ambient_temp_value) { + regmap_write(max98390->regmap, DSM_TPROT_ROOM_TEMPERATURE_BYTE1, + (max98390->ambient_temp_value >> 8) & 0x000000ff); + regmap_write(max98390->regmap, DSM_TPROT_ROOM_TEMPERATURE_BYTE0, + (max98390->ambient_temp_value) & 0x000000ff); + } + + return 0; +} + +#ifdef CONFIG_PM_SLEEP +static int max98390_suspend(struct device *dev) +{ + struct max98390_priv *max98390 = dev_get_drvdata(dev); + + dev_dbg(dev, "%s:Enter\n", __func__); + + regcache_cache_only(max98390->regmap, true); + regcache_mark_dirty(max98390->regmap); + + return 0; +} + +static int max98390_resume(struct device *dev) +{ + struct max98390_priv *max98390 = dev_get_drvdata(dev); + + dev_dbg(dev, "%s:Enter\n", __func__); + + regcache_cache_only(max98390->regmap, false); + regcache_sync(max98390->regmap); + + return 0; +} +#endif + +static const struct dev_pm_ops max98390_pm = { + SET_SYSTEM_SLEEP_PM_OPS(max98390_suspend, max98390_resume) +}; + +static const struct snd_soc_component_driver soc_codec_dev_max98390 = { + .probe = max98390_probe, + .controls = max98390_snd_controls, + .num_controls = ARRAY_SIZE(max98390_snd_controls), + .dapm_widgets = max98390_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max98390_dapm_widgets), + .dapm_routes = max98390_audio_map, + .num_dapm_routes = ARRAY_SIZE(max98390_audio_map), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static const struct regmap_config max98390_regmap = { + .reg_bits = 16, + .val_bits = 8, + .max_register = MAX98390_R24FF_REV_ID, + .reg_defaults = max98390_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(max98390_reg_defaults), + .readable_reg = max98390_readable_register, + .volatile_reg = max98390_volatile_reg, + .cache_type = REGCACHE_RBTREE, +}; + +#ifdef CONFIG_OF +static const struct of_device_id max98390_dt_ids[] = { + { .compatible = "maxim,max98390", }, + { } +}; +MODULE_DEVICE_TABLE(of, max98390_dt_ids); +#endif + +static int max98390_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + int ret = 0; + int reg = 0; + + struct max98390_priv *max98390 = NULL; + struct i2c_adapter *adapter = to_i2c_adapter(i2c->dev.parent); + + ret = i2c_check_functionality(adapter, + I2C_FUNC_SMBUS_BYTE + | I2C_FUNC_SMBUS_BYTE_DATA); + if (!ret) { + dev_err(&i2c->dev, "I2C check functionality failed\n"); + return -ENXIO; + } + + max98390 = devm_kzalloc(&i2c->dev, sizeof(*max98390), GFP_KERNEL); + if (!max98390) { + ret = -ENOMEM; + return ret; + } + i2c_set_clientdata(i2c, max98390); + + ret = device_property_read_u32(&i2c->dev, "maxim,temperature_calib", + &max98390->ambient_temp_value); + if (ret) { + dev_info(&i2c->dev, + "no optional property 'temperature_calib' found, default:\n"); + } + ret = device_property_read_u32(&i2c->dev, "maxim,r0_calib", + &max98390->ref_rdc_value); + if (ret) { + dev_info(&i2c->dev, + "no optional property 'r0_calib' found, default:\n"); + } + + dev_info(&i2c->dev, + "%s: r0_calib: 0x%x,temperature_calib: 0x%x", + __func__, max98390->ref_rdc_value, + max98390->ambient_temp_value); + + /* regmap initialization */ + max98390->regmap = devm_regmap_init_i2c(i2c, &max98390_regmap); + if (IS_ERR(max98390->regmap)) { + ret = PTR_ERR(max98390->regmap); + dev_err(&i2c->dev, + "Failed to allocate regmap: %d\n", ret); + return ret; + } + + /* Check Revision ID */ + ret = regmap_read(max98390->regmap, + MAX98390_R24FF_REV_ID, &reg); + if (ret) { + dev_err(&i2c->dev, + "ret=%d, Failed to read: 0x%02X\n", + ret, MAX98390_R24FF_REV_ID); + return ret; + } + dev_info(&i2c->dev, "MAX98390 revisionID: 0x%02X\n", reg); + + ret = devm_snd_soc_register_component(&i2c->dev, + &soc_codec_dev_max98390, + max98390_dai, ARRAY_SIZE(max98390_dai)); + + return ret; +} + +static const struct i2c_device_id max98390_i2c_id[] = { + { "max98390", 0}, + {}, +}; + +MODULE_DEVICE_TABLE(i2c, max98390_i2c_id); + +#if defined(CONFIG_OF) +static const struct of_device_id max98390_of_match[] = { + { .compatible = "maxim,max98390", }, + {} +}; +MODULE_DEVICE_TABLE(of, max98390_of_match); +#endif + +#ifdef CONFIG_ACPI +static const struct acpi_device_id max98390_acpi_match[] = { + { "MX98390", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, max98390_acpi_match); +#endif + +static struct i2c_driver max98390_i2c_driver = { + .driver = { + .name = "max98390", + .of_match_table = of_match_ptr(max98390_of_match), + .acpi_match_table = ACPI_PTR(max98390_acpi_match), + .pm = &max98390_pm, + }, + .probe = max98390_i2c_probe, + .id_table = max98390_i2c_id, +}; + +module_i2c_driver(max98390_i2c_driver) + +MODULE_DESCRIPTION("ALSA SoC MAX98390 driver"); +MODULE_AUTHOR("Steve Lee <steves.lee(a)maximintegrated.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/max98390.h b/sound/soc/codecs/max98390.h new file mode 100644 index 000000000000..f59cb114d957 --- /dev/null +++ b/sound/soc/codecs/max98390.h @@ -0,0 +1,663 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Copyright (c) 2020, Maxim Integrated. + */ + +#ifndef _MAX98390_H +#define _MAX98390_H + +/* MAX98390 Register Address */ +#define MAX98390_SOFTWARE_RESET 0x2000 +#define MAX98390_INT_RAW1 0x2002 +#define MAX98390_INT_RAW2 0x2003 +#define MAX98390_INT_RAW3 0x2004 +#define MAX98390_INT_STATE1 0x2005 +#define MAX98390_INT_STATE2 0x2006 +#define MAX98390_INT_STATE3 0x2007 +#define MAX98390_INT_FLAG1 0x2008 +#define MAX98390_INT_FLAG2 0x2009 +#define MAX98390_INT_FLAG3 0x200a +#define MAX98390_INT_EN1 0x200b +#define MAX98390_INT_EN2 0x200c +#define MAX98390_INT_EN3 0x200d +#define MAX98390_INT_FLAG_CLR1 0x200e +#define MAX98390_INT_FLAG_CLR2 0x200f +#define MAX98390_INT_FLAG_CLR3 0x2010 +#define MAX98390_IRQ_CTRL 0x2011 +#define MAX98390_CLK_MON 0x2012 +#define MAX98390_DAT_MON 0x2014 +#define MAX98390_WDOG_CTRL 0x2015 +#define MAX98390_WDOG_RST 0x2016 +#define MAX98390_MEAS_ADC_THERM_WARN_THRESH 0x2017 +#define MAX98390_MEAS_ADC_THERM_SHDN_THRESH 0x2018 +#define MAX98390_MEAS_ADC_THERM_HYSTERESIS 0x2019 +#define MAX98390_PIN_CFG 0x201a +#define MAX98390_PCM_RX_EN_A 0x201b +#define MAX98390_PCM_RX_EN_B 0x201c +#define MAX98390_PCM_TX_EN_A 0x201d +#define MAX98390_PCM_TX_EN_B 0x201e +#define MAX98390_PCM_TX_HIZ_CTRL_A 0x201f +#define MAX98390_PCM_TX_HIZ_CTRL_B 0x2020 +#define MAX98390_PCM_CH_SRC_1 0x2021 +#define MAX98390_PCM_CH_SRC_2 0x2022 +#define MAX98390_PCM_CH_SRC_3 0x2023 +#define MAX98390_PCM_MODE_CFG 0x2024 +#define MAX98390_PCM_MASTER_MODE 0x2025 +#define MAX98390_PCM_CLK_SETUP 0x2026 +#define MAX98390_PCM_SR_SETUP 0x2027 +#define MAX98390_ICC_RX_EN_A 0x202c +#define MAX98390_ICC_RX_EN_B 0x202d +#define MAX98390_ICC_TX_EN_A 0x202e +#define MAX98390_ICC_TX_EN_B 0x202f +#define MAX98390_ICC_HIZ_MANUAL_MODE 0x2030 +#define MAX98390_ICC_TX_HIZ_EN_A 0x2031 +#define MAX98390_ICC_TX_HIZ_EN_B 0x2032 +#define MAX98390_ICC_LNK_EN 0x2033 +#define MAX98390_R2039_AMP_DSP_CFG 0x2039 +#define MAX98390_R203A_AMP_EN 0x203a +#define MAX98390_TONE_GEN_DC_CFG 0x203b +#define MAX98390_SPK_SRC_SEL 0x203c +#define MAX98390_R203D_SPK_GAIN 0x203d +#define MAX98390_SSM_CFG 0x203e +#define MAX98390_MEAS_EN 0x203f +#define MAX98390_MEAS_DSP_CFG 0x2040 +#define MAX98390_BOOST_CTRL0 0x2041 +#define MAX98390_BOOST_CTRL3 0x2042 +#define MAX98390_BOOST_CTRL1 0x2043 +#define MAX98390_MEAS_ADC_CFG 0x2044 +#define MAX98390_MEAS_ADC_BASE_MSB 0x2045 +#define MAX98390_MEAS_ADC_BASE_LSB 0x2046 +#define MAX98390_ADC_CH0_DIVIDE 0x2047 +#define MAX98390_ADC_CH1_DIVIDE 0x2048 +#define MAX98390_ADC_CH2_DIVIDE 0x2049 +#define MAX98390_ADC_CH0_FILT_CFG 0x204a +#define MAX98390_ADC_CH1_FILT_CFG 0x204b +#define MAX98390_ADC_CH2_FILT_CFG 0x204c +#define MAX98390_MEAS_ADC_CH0_READ 0x204d +#define MAX98390_MEAS_ADC_CH1_READ 0x204e +#define MAX98390_MEAS_ADC_CH2_READ 0x204f +#define MAX98390_PWR_GATE_CTL 0x2050 +#define MAX98390_PWR_GATE_STATUS 0x2051 +#define MAX98390_VBAT_LOW_STATUS 0x2052 +#define MAX98390_PVDD_LOW_STATUS 0x2053 +#define MAX98390_BROWNOUT_STATUS 0x2054 +#define MAX98390_BROWNOUT_EN 0x2055 +#define MAX98390_BROWNOUT_INFINITE_HOLD 0x2056 +#define MAX98390_BROWNOUT_INFINITE_HOLD_CLR 0x2057 +#define MAX98390_BROWNOUT_LVL_HOLD 0x2058 +#define MAX98390_BROWNOUT_LVL1_THRESH 0x2059 +#define MAX98390_BROWNOUT_LVL2_THRESH 0x205a +#define MAX98390_BROWNOUT_LVL3_THRESH 0x205b +#define MAX98390_BROWNOUT_LVL4_THRESH 0x205c +#define MAX98390_BROWNOUT_THRESH_HYSTERYSIS 0x205d +#define MAX98390_BROWNOUT_AMP_LIMITER_ATK_REL 0x205e +#define MAX98390_BROWNOUT_AMP_GAIN_ATK_REL 0x205f +#define MAX98390_BROWNOUT_AMP1_CLIP_MODE 0x2060 +#define MAX98390_BROWNOUT_LVL1_CUR_LIMIT 0x2061 +#define MAX98390_BROWNOUT_LVL1_AMP1_CTRL1 0x2062 +#define MAX98390_BROWNOUT_LVL1_AMP1_CTRL2 0x2063 +#define MAX98390_BROWNOUT_LVL1_AMP1_CTRL3 0x2064 +#define MAX98390_BROWNOUT_LVL2_CUR_LIMIT 0x2065 +#define MAX98390_BROWNOUT_LVL2_AMP1_CTRL1 0x2066 +#define MAX98390_BROWNOUT_LVL2_AMP1_CTRL2 0x2067 +#define MAX98390_BROWNOUT_LVL2_AMP1_CTRL3 0x2068 +#define MAX98390_BROWNOUT_LVL3_CUR_LIMIT 0x2069 +#define MAX98390_BROWNOUT_LVL3_AMP1_CTRL1 0x206a +#define MAX98390_BROWNOUT_LVL3_AMP1_CTRL2 0x206b +#define MAX98390_BROWNOUT_LVL3_AMP1_CTRL3 0x206c +#define MAX98390_BROWNOUT_LVL4_CUR_LIMIT 0x206d +#define MAX98390_BROWNOUT_LVL4_AMP1_CTRL1 0x206e +#define MAX98390_BROWNOUT_LVL4_AMP1_CTRL2 0x206f +#define MAX98390_BROWNOUT_LVL4_AMP1_CTRL3 0x2070 +#define MAX98390_BROWNOUT_LOWEST_STATUS 0x2071 +#define MAX98390_BROWNOUT_ILIM_HLD 0x2072 +#define MAX98390_BROWNOUT_LIM_HLD 0x2073 +#define MAX98390_BROWNOUT_CLIP_HLD 0x2074 +#define MAX98390_BROWNOUT_GAIN_HLD 0x2075 +#define MAX98390_ENV_TRACK_VOUT_HEADROOM 0x2076 +#define MAX98390_ENV_TRACK_BOOST_VOUT_DELAY 0x2077 +#define MAX98390_ENV_TRACK_REL_RATE 0x2078 +#define MAX98390_ENV_TRACK_HOLD_RATE 0x2079 +#define MAX98390_ENV_TRACK_CTRL 0x207a +#define MAX98390_ENV_TRACK_BOOST_VOUT_READ 0x207b +#define MAX98390_BOOST_BYPASS1 0x207c +#define MAX98390_BOOST_BYPASS2 0x207d +#define MAX98390_BOOST_BYPASS3 0x207e +#define MAX98390_FET_SCALING1 0x207f +#define MAX98390_FET_SCALING2 0x2080 +#define MAX98390_FET_SCALING3 0x2081 +#define MAX98390_FET_SCALING4 0x2082 +#define MAX98390_SPK_SPEEDUP 0x2084 + +#define DSM_STBASS_HPF_B0_BYTE0 0x2101 +#define DSM_STBASS_HPF_B0_BYTE1 0x2102 +#define DSM_STBASS_HPF_B0_BYTE2 0x2103 +#define DSM_STBASS_HPF_B1_BYTE0 0x2105 +#define DSM_STBASS_HPF_B1_BYTE1 0x2106 +#define DSM_STBASS_HPF_B1_BYTE2 0x2107 +#define DSM_STBASS_HPF_B2_BYTE0 0x2109 +#define DSM_STBASS_HPF_B2_BYTE1 0x210a +#define DSM_STBASS_HPF_B2_BYTE2 0x210b +#define DSM_STBASS_HPF_A1_BYTE0 0x210d +#define DSM_STBASS_HPF_A1_BYTE1 0x210e +#define DSM_STBASS_HPF_A1_BYTE2 0x210f +#define DSM_STBASS_HPF_A2_BYTE0 0x2111 +#define DSM_STBASS_HPF_A2_BYTE1 0x2112 +#define DSM_STBASS_HPF_A2_BYTE2 0x2113 +#define DSM_STBASS_LPF_B0_BYTE0 0x2115 +#define DSM_STBASS_LPF_B0_BYTE1 0x2116 +#define DSM_STBASS_LPF_B0_BYTE2 0x2117 +#define DSM_STBASS_LPF_B1_BYTE0 0x2119 +#define DSM_STBASS_LPF_B1_BYTE1 0x211a +#define DSM_STBASS_LPF_B1_BYTE2 0x211b +#define DSM_STBASS_LPF_B2_BYTE0 0x211d +#define DSM_STBASS_LPF_B2_BYTE1 0x211e +#define DSM_STBASS_LPF_B2_BYTE2 0x211f +#define DSM_STBASS_LPF_A1_BYTE0 0x2121 +#define DSM_STBASS_LPF_A1_BYTE1 0x2122 +#define DSM_STBASS_LPF_A1_BYTE2 0x2123 +#define DSM_STBASS_LPF_A2_BYTE0 0x2125 +#define DSM_STBASS_LPF_A2_BYTE1 0x2126 +#define DSM_STBASS_LPF_A2_BYTE2 0x2127 +#define DSM_EQ_BQ1_B0_BYTE0 0x2129 +#define DSM_EQ_BQ1_B0_BYTE1 0x212a +#define DSM_EQ_BQ1_B0_BYTE2 0x212b +#define DSM_EQ_BQ1_B1_BYTE0 0x212d +#define DSM_EQ_BQ1_B1_BYTE1 0x212e +#define DSM_EQ_BQ1_B1_BYTE2 0x212f +#define DSM_EQ_BQ1_B2_BYTE0 0x2131 +#define DSM_EQ_BQ1_B2_BYTE1 0x2132 +#define DSM_EQ_BQ1_B2_BYTE2 0x2133 +#define DSM_EQ_BQ1_A1_BYTE0 0x2135 +#define DSM_EQ_BQ1_A1_BYTE1 0x2136 +#define DSM_EQ_BQ1_A1_BYTE2 0x2137 +#define DSM_EQ_BQ1_A2_BYTE0 0x2139 +#define DSM_EQ_BQ1_A2_BYTE1 0x213a +#define DSM_EQ_BQ1_A2_BYTE2 0x213b +#define DSM_EQ_BQ2_B0_BYTE0 0x213d +#define DSM_EQ_BQ2_B0_BYTE1 0x213e +#define DSM_EQ_BQ2_B0_BYTE2 0x213f +#define DSM_EQ_BQ2_B1_BYTE0 0x2141 +#define DSM_EQ_BQ2_B1_BYTE1 0x2142 +#define DSM_EQ_BQ2_B1_BYTE2 0x2143 +#define DSM_EQ_BQ2_B2_BYTE0 0x2145 +#define DSM_EQ_BQ2_B2_BYTE1 0x2146 +#define DSM_EQ_BQ2_B2_BYTE2 0x2147 +#define DSM_EQ_BQ2_A1_BYTE0 0x2149 +#define DSM_EQ_BQ2_A1_BYTE1 0x214a +#define DSM_EQ_BQ2_A1_BYTE2 0x214b +#define DSM_EQ_BQ2_A2_BYTE0 0x214d +#define DSM_EQ_BQ2_A2_BYTE1 0x214e +#define DSM_EQ_BQ2_A2_BYTE2 0x214f +#define DSM_EQ_BQ3_B0_BYTE0 0x2151 +#define DSM_EQ_BQ3_B0_BYTE1 0x2152 +#define DSM_EQ_BQ3_B0_BYTE2 0x2153 +#define DSM_EQ_BQ3_B1_BYTE0 0x2155 +#define DSM_EQ_BQ3_B1_BYTE1 0x2156 +#define DSM_EQ_BQ3_B1_BYTE2 0x2157 +#define DSM_EQ_BQ3_B2_BYTE0 0x2159 +#define DSM_EQ_BQ3_B2_BYTE1 0x215a +#define DSM_EQ_BQ3_B2_BYTE2 0x215b +#define DSM_EQ_BQ3_A1_BYTE0 0x215d +#define DSM_EQ_BQ3_A1_BYTE1 0x215e +#define DSM_EQ_BQ3_A1_BYTE2 0x215f +#define DSM_EQ_BQ3_A2_BYTE0 0x2161 +#define DSM_EQ_BQ3_A2_BYTE1 0x2162 +#define DSM_EQ_BQ3_A2_BYTE2 0x2163 +#define DSM_EQ_BQ4_B0_BYTE0 0x2165 +#define DSM_EQ_BQ4_B0_BYTE1 0x2166 +#define DSM_EQ_BQ4_B0_BYTE2 0x2167 +#define DSM_EQ_BQ4_B1_BYTE0 0x2169 +#define DSM_EQ_BQ4_B1_BYTE1 0x216a +#define DSM_EQ_BQ4_B1_BYTE2 0x216b +#define DSM_EQ_BQ4_B2_BYTE0 0x216d +#define DSM_EQ_BQ4_B2_BYTE1 0x216e +#define DSM_EQ_BQ4_B2_BYTE2 0x216f +#define DSM_EQ_BQ4_A1_BYTE0 0x2171 +#define DSM_EQ_BQ4_A1_BYTE1 0x2172 +#define DSM_EQ_BQ4_A1_BYTE2 0x2173 +#define DSM_EQ_BQ4_A2_BYTE0 0x2175 +#define DSM_EQ_BQ4_A2_BYTE1 0x2176 +#define DSM_EQ_BQ4_A2_BYTE2 0x2177 +#define DSM_EQ_BQ5_B0_BYTE0 0x2179 +#define DSM_EQ_BQ5_B0_BYTE1 0x217a +#define DSM_EQ_BQ5_B0_BYTE2 0x217b +#define DSM_EQ_BQ5_B1_BYTE0 0x217d +#define DSM_EQ_BQ5_B1_BYTE1 0x217e +#define DSM_EQ_BQ5_B1_BYTE2 0x217f +#define DSM_EQ_BQ5_B2_BYTE0 0x2181 +#define DSM_EQ_BQ5_B2_BYTE1 0x2182 +#define DSM_EQ_BQ5_B2_BYTE2 0x2183 +#define DSM_EQ_BQ5_A1_BYTE0 0x2185 +#define DSM_EQ_BQ5_A1_BYTE1 0x2186 +#define DSM_EQ_BQ5_A1_BYTE2 0x2187 +#define DSM_EQ_BQ5_A2_BYTE0 0x2189 +#define DSM_EQ_BQ5_A2_BYTE1 0x218a +#define DSM_EQ_BQ5_A2_BYTE2 0x218b +#define DSM_EQ_BQ6_B0_BYTE0 0x218d +#define DSM_EQ_BQ6_B0_BYTE1 0x218e +#define DSM_EQ_BQ6_B0_BYTE2 0x218f +#define DSM_EQ_BQ6_B1_BYTE0 0x2191 +#define DSM_EQ_BQ6_B1_BYTE1 0x2192 +#define DSM_EQ_BQ6_B1_BYTE2 0x2193 +#define DSM_EQ_BQ6_B2_BYTE0 0x2195 +#define DSM_EQ_BQ6_B2_BYTE1 0x2196 +#define DSM_EQ_BQ6_B2_BYTE2 0x2197 +#define DSM_EQ_BQ6_A1_BYTE0 0x2199 +#define DSM_EQ_BQ6_A1_BYTE1 0x219a +#define DSM_EQ_BQ6_A1_BYTE2 0x219b +#define DSM_EQ_BQ6_A2_BYTE0 0x219d +#define DSM_EQ_BQ6_A2_BYTE1 0x219e +#define DSM_EQ_BQ6_A2_BYTE2 0x219f +#define DSM_EQ_BQ7_B0_BYTE0 0x21a1 +#define DSM_EQ_BQ7_B0_BYTE1 0x21a2 +#define DSM_EQ_BQ7_B0_BYTE2 0x21a3 +#define DSM_EQ_BQ7_B1_BYTE0 0x21a5 +#define DSM_EQ_BQ7_B1_BYTE1 0x21a6 +#define DSM_EQ_BQ7_B1_BYTE2 0x21a7 +#define DSM_EQ_BQ7_B2_BYTE0 0x21a9 +#define DSM_EQ_BQ7_B2_BYTE1 0x21aa +#define DSM_EQ_BQ7_B2_BYTE2 0x21ab +#define DSM_EQ_BQ7_A1_BYTE0 0x21ad +#define DSM_EQ_BQ7_A1_BYTE1 0x21ae +#define DSM_EQ_BQ7_A1_BYTE2 0x21af +#define DSM_EQ_BQ7_A2_BYTE0 0x21b1 +#define DSM_EQ_BQ7_A2_BYTE1 0x21b2 +#define DSM_EQ_BQ7_A2_BYTE2 0x21b3 +#define DSM_EQ_BQ8_B0_BYTE0 0x21b5 +#define DSM_EQ_BQ8_B0_BYTE1 0x21b6 +#define DSM_EQ_BQ8_B0_BYTE2 0x21b7 +#define DSM_EQ_BQ8_B1_BYTE0 0x21b9 +#define DSM_EQ_BQ8_B1_BYTE1 0x21ba +#define DSM_EQ_BQ8_B1_BYTE2 0x21bb +#define DSM_EQ_BQ8_B2_BYTE0 0x21bd +#define DSM_EQ_BQ8_B2_BYTE1 0x21be +#define DSM_EQ_BQ8_B2_BYTE2 0x21bf +#define DSM_EQ_BQ8_A1_BYTE0 0x21c1 +#define DSM_EQ_BQ8_A1_BYTE1 0x21c2 +#define DSM_EQ_BQ8_A1_BYTE2 0x21c3 +#define DSM_EQ_BQ8_A2_BYTE0 0x21c5 +#define DSM_EQ_BQ8_A2_BYTE1 0x21c6 +#define DSM_EQ_BQ8_A2_BYTE2 0x21c7 +#define DSM_LFX_BQ_B0_BYTE0 0x21c9 +#define DSM_LFX_BQ_B0_BYTE1 0x21ca +#define DSM_LFX_BQ_B0_BYTE2 0x21cb +#define DSM_LFX_BQ_B1_BYTE0 0x21cd +#define DSM_LFX_BQ_B1_BYTE1 0x21ce +#define DSM_LFX_BQ_B1_BYTE2 0x21cf +#define DSM_LFX_BQ_B2_BYTE0 0x21d1 +#define DSM_LFX_BQ_B2_BYTE1 0x21d2 +#define DSM_LFX_BQ_B2_BYTE2 0x21d3 +#define DSM_LFX_BQ_A1_BYTE0 0x21d5 +#define DSM_LFX_BQ_A1_BYTE1 0x21d6 +#define DSM_LFX_BQ_A1_BYTE2 0x21d7 +#define DSM_LFX_BQ_A2_BYTE0 0x21d9 +#define DSM_LFX_BQ_A2_BYTE1 0x21da +#define DSM_LFX_BQ_A2_BYTE2 0x21db +#define DSM_PPR_HPF_B0_BYTE0 0x21dd +#define DSM_PPR_HPF_B0_BYTE1 0x21de +#define DSM_PPR_HPF_B0_BYTE2 0x21df +#define DSM_PPR_HPF_B1_BYTE0 0x21e1 +#define DSM_PPR_HPF_B1_BYTE1 0x21e2 +#define DSM_PPR_HPF_B1_BYTE2 0x21e3 +#define DSM_PPR_HPF_B2_BYTE0 0x21e5 +#define DSM_PPR_HPF_B2_BYTE1 0x21e6 +#define DSM_PPR_HPF_B2_BYTE2 0x21e7 +#define DSM_PPR_HPF_A1_BYTE0 0x21e9 +#define DSM_PPR_HPF_A1_BYTE1 0x21ea +#define DSM_PPR_HPF_A1_BYTE2 0x21eb +#define DSM_PPR_HPF_A2_BYTE0 0x21ed +#define DSM_PPR_HPF_A2_BYTE1 0x21ee +#define DSM_PPR_HPF_A2_BYTE2 0x21ef +#define DSM_PPR_LPF_B0_BYTE0 0x21f1 +#define DSM_PPR_LPF_B0_BYTE1 0x21f2 +#define DSM_PPR_LPF_B0_BYTE2 0x21f3 +#define DSM_PPR_LPF_B1_BYTE0 0x21f5 +#define DSM_PPR_LPF_B1_BYTE1 0x21f6 +#define DSM_PPR_LPF_B1_BYTE2 0x21f7 +#define DSM_PPR_LPF_B2_BYTE0 0x21f9 +#define DSM_PPR_LPF_B2_BYTE1 0x21fa +#define DSM_PPR_LPF_B2_BYTE2 0x21fb +#define DSM_PPR_LPF_A1_BYTE0 0x21fd +#define DSM_PPR_LPF_A1_BYTE1 0x21fe +#define DSM_PPR_LPF_A1_BYTE2 0x21ff +#define DSM_PPR_LPF_A2_BYTE0 0x2201 +#define DSM_PPR_LPF_A2_BYTE1 0x2202 +#define DSM_PPR_LPF_A2_BYTE2 0x2203 +#define DSM_SPL_BQ_B0_BYTE0 0x2205 +#define DSM_SPL_BQ_B0_BYTE1 0x2206 +#define DSM_SPL_BQ_B0_BYTE2 0x2207 +#define DSM_SPL_BQ_B1_BYTE0 0x2209 +#define DSM_SPL_BQ_B1_BYTE1 0x220a +#define DSM_SPL_BQ_B1_BYTE2 0x220b +#define DSM_SPL_BQ_B2_BYTE0 0x220d +#define DSM_SPL_BQ_B2_BYTE1 0x220e +#define DSM_SPL_BQ_B2_BYTE2 0x220f +#define DSM_SPL_BQ_A1_BYTE0 0x2211 +#define DSM_SPL_BQ_A1_BYTE1 0x2212 +#define DSM_SPL_BQ_A1_BYTE2 0x2213 +#define DSM_SPL_BQ_A2_BYTE0 0x2215 +#define DSM_SPL_BQ_A2_BYTE1 0x2216 +#define DSM_SPL_BQ_A2_BYTE2 0x2217 +#define DSM_EXCUR_BQ_B0_BYTE0 0x2219 +#define DSM_EXCUR_BQ_B0_BYTE1 0x221a +#define DSM_EXCUR_BQ_B0_BYTE2 0x221b +#define DSM_EXCUR_BQ_B1_BYTE0 0x221d +#define DSM_EXCUR_BQ_B1_BYTE1 0x221e +#define DSM_EXCUR_BQ_B1_BYTE2 0x221f +#define DSM_EXCUR_BQ_B2_BYTE0 0x2221 +#define DSM_EXCUR_BQ_B2_BYTE1 0x2222 +#define DSM_EXCUR_BQ_B2_BYTE2 0x2223 +#define DSM_EXCUR_BQ_A1_BYTE0 0x2225 +#define DSM_EXCUR_BQ_A1_BYTE1 0x2226 +#define DSM_EXCUR_BQ_A1_BYTE2 0x2227 +#define DSM_EXCUR_BQ_A2_BYTE0 0x2229 +#define DSM_EXCUR_BQ_A2_BYTE1 0x222a +#define DSM_EXCUR_BQ_A2_BYTE2 0x222b +#define DSM_EXCPROT_HPF1_B0_BYTE0 0x222d +#define DSM_EXCPROT_HPF1_B0_BYTE1 0x222e +#define DSM_EXCPROT_HPF1_B0_BYTE2 0x222f +#define DSM_EXCPROT_HPF1_B1_BYTE0 0x2231 +#define DSM_EXCPROT_HPF1_B1_BYTE1 0x2232 +#define DSM_EXCPROT_HPF1_B1_BYTE2 0x2233 +#define DSM_EXCPROT_HPF1_B2_BYTE0 0x2235 +#define DSM_EXCPROT_HPF1_B2_BYTE1 0x2236 +#define DSM_EXCPROT_HPF1_B2_BYTE2 0x2237 +#define DSM_EXCPROT_HPF1_A1_BYTE0 0x2239 +#define DSM_EXCPROT_HPF1_A1_BYTE1 0x223a +#define DSM_EXCPROT_HPF1_A1_BYTE2 0x223b +#define DSM_EXCPROT_HPF1_A2_BYTE0 0x223d +#define DSM_EXCPROT_HPF1_A2_BYTE1 0x223e +#define DSM_EXCPROT_HPF1_A2_BYTE2 0x223f +#define DSM_EXCPROT_HPF2_B0_BYTE0 0x2241 +#define DSM_EXCPROT_HPF2_B0_BYTE1 0x2242 +#define DSM_EXCPROT_HPF2_B0_BYTE2 0x2243 +#define DSM_EXCPROT_HPF2_B1_BYTE0 0x2245 +#define DSM_EXCPROT_HPF2_B1_BYTE1 0x2246 +#define DSM_EXCPROT_HPF2_B1_BYTE2 0x2247 +#define DSM_EXCPROT_HPF2_B2_BYTE0 0x2249 +#define DSM_EXCPROT_HPF2_B2_BYTE1 0x224a +#define DSM_EXCPROT_HPF2_B2_BYTE2 0x224b +#define DSM_EXCPROT_HPF2_A1_BYTE0 0x224d +#define DSM_EXCPROT_HPF2_A1_BYTE1 0x224e +#define DSM_EXCPROT_HPF2_A1_BYTE2 0x224f +#define DSM_EXCPROT_HPF2_A2_BYTE0 0x2251 +#define DSM_EXCPROT_HPF2_A2_BYTE1 0x2252 +#define DSM_EXCPROT_HPF2_A2_BYTE2 0x2253 +#define DSM_EXCPROT_HPF3_B0_BYTE0 0x2255 +#define DSM_EXCPROT_HPF3_B0_BYTE1 0x2256 +#define DSM_EXCPROT_HPF3_B0_BYTE2 0x2257 +#define DSM_EXCPROT_HPF3_B1_BYTE0 0x2259 +#define DSM_EXCPROT_HPF3_B1_BYTE1 0x225a +#define DSM_EXCPROT_HPF3_B1_BYTE2 0x225b +#define DSM_EXCPROT_HPF3_B2_BYTE0 0x225d +#define DSM_EXCPROT_HPF3_B2_BYTE1 0x225e +#define DSM_EXCPROT_HPF3_B2_BYTE2 0x225f +#define DSM_EXCPROT_HPF3_A1_BYTE0 0x2261 +#define DSM_EXCPROT_HPF3_A1_BYTE1 0x2262 +#define DSM_EXCPROT_HPF3_A1_BYTE2 0x2263 +#define DSM_EXCPROT_HPF3_A2_BYTE0 0x2265 +#define DSM_EXCPROT_HPF3_A2_BYTE1 0x2266 +#define DSM_EXCPROT_HPF3_A2_BYTE2 0x2267 +#define DSM_EXCPROT_HPF4_B0_BYTE0 0x2269 +#define DSM_EXCPROT_HPF4_B0_BYTE1 0x226a +#define DSM_EXCPROT_HPF4_B0_BYTE2 0x226b +#define DSM_EXCPROT_HPF4_B1_BYTE0 0x226d +#define DSM_EXCPROT_HPF4_B1_BYTE1 0x226e +#define DSM_EXCPROT_HPF4_B1_BYTE2 0x226f +#define DSM_EXCPROT_HPF4_B2_BYTE0 0x2271 +#define DSM_EXCPROT_HPF4_B2_BYTE1 0x2272 +#define DSM_EXCPROT_HPF4_B2_BYTE2 0x2273 +#define DSM_EXCPROT_HPF4_A1_BYTE0 0x2275 +#define DSM_EXCPROT_HPF4_A1_BYTE1 0x2276 +#define DSM_EXCPROT_HPF4_A1_BYTE2 0x2277 +#define DSM_EXCPROT_HPF4_A2_BYTE0 0x2279 +#define DSM_EXCPROT_HPF4_A2_BYTE1 0x227a +#define DSM_EXCPROT_HPF4_A2_BYTE2 0x227b +#define DSM_EXCPROT_HPF5_B0_BYTE0 0x227d +#define DSM_EXCPROT_HPF5_B0_BYTE1 0x227e +#define DSM_EXCPROT_HPF5_B0_BYTE2 0x227f +#define DSM_EXCPROT_HPF5_B1_BYTE0 0x2281 +#define DSM_EXCPROT_HPF5_B1_BYTE1 0x2282 +#define DSM_EXCPROT_HPF5_B1_BYTE2 0x2283 +#define DSM_EXCPROT_HPF5_B2_BYTE0 0x2285 +#define DSM_EXCPROT_HPF5_B2_BYTE1 0x2286 +#define DSM_EXCPROT_HPF5_B2_BYTE2 0x2287 +#define DSM_EXCPROT_HPF5_A1_BYTE0 0x2289 +#define DSM_EXCPROT_HPF5_A1_BYTE1 0x228a +#define DSM_EXCPROT_HPF5_A1_BYTE2 0x228b +#define DSM_EXCPROT_HPF5_A2_BYTE0 0x228d +#define DSM_EXCPROT_HPF5_A2_BYTE1 0x228e +#define DSM_EXCPROT_HPF5_A2_BYTE2 0x228f +#define DSM_DEBUZZ_BPF_B0_BYTE0 0x2291 +#define DSM_DEBUZZ_BPF_B0_BYTE1 0x2292 +#define DSM_DEBUZZ_BPF_B0_BYTE2 0x2293 +#define DSM_DEBUZZ_BPF_B1_BYTE0 0x2295 +#define DSM_DEBUZZ_BPF_B1_BYTE1 0x2296 +#define DSM_DEBUZZ_BPF_B1_BYTE2 0x2297 +#define DSM_DEBUZZ_BPF_B2_BYTE0 0x2299 +#define DSM_DEBUZZ_BPF_B2_BYTE1 0x229a +#define DSM_DEBUZZ_BPF_B2_BYTE2 0x229b +#define DSM_DEBUZZ_BPF_A1_BYTE0 0x229d +#define DSM_DEBUZZ_BPF_A1_BYTE1 0x229e +#define DSM_DEBUZZ_BPF_A1_BYTE2 0x229f +#define DSM_DEBUZZ_BPF_A2_BYTE0 0x22a1 +#define DSM_DEBUZZ_BPF_A2_BYTE1 0x22a2 +#define DSM_DEBUZZ_BPF_A2_BYTE2 0x22a3 +#define DSM_DEBUZZ_PORT_B0_BYTE0 0x22a5 +#define DSM_DEBUZZ_PORT_B0_BYTE1 0x22a6 +#define DSM_DEBUZZ_PORT_B0_BYTE2 0x22a7 +#define DSM_DEBUZZ_PORT_B1_BYTE0 0x22a9 +#define DSM_DEBUZZ_PORT_B1_BYTE1 0x22aa +#define DSM_DEBUZZ_PORT_B1_BYTE2 0x22ab +#define DSM_DEBUZZ_PORT_B2_BYTE0 0x22ad +#define DSM_DEBUZZ_PORT_B2_BYTE1 0x22ae +#define DSM_DEBUZZ_PORT_B2_BYTE2 0x22af +#define DSM_DEBUZZ_PORT_A1_BYTE0 0x22b1 +#define DSM_DEBUZZ_PORT_A1_BYTE1 0x22b2 +#define DSM_DEBUZZ_PORT_A1_BYTE2 0x22b3 +#define DSM_DEBUZZ_PORT_A2_BYTE0 0x22b5 +#define DSM_DEBUZZ_PORT_A2_BYTE1 0x22b6 +#define DSM_DEBUZZ_PORT_A2_BYTE2 0x22b7 +#define DSM_DEBUZZ_NOTCH_B0_BYTE0 0x22b9 +#define DSM_DEBUZZ_NOTCH_B0_BYTE1 0x22ba +#define DSM_DEBUZZ_NOTCH_B0_BYTE2 0x22bb +#define DSM_DEBUZZ_NOTCH_B1_BYTE0 0x22bd +#define DSM_DEBUZZ_NOTCH_B1_BYTE1 0x22be +#define DSM_DEBUZZ_NOTCH_B1_BYTE2 0x22bf +#define DSM_DEBUZZ_NOTCH_B2_BYTE0 0x22c1 +#define DSM_DEBUZZ_NOTCH_B2_BYTE1 0x22c2 +#define DSM_DEBUZZ_NOTCH_B2_BYTE2 0x22c3 +#define DSM_DEBUZZ_NOTCH_A1_BYTE0 0x22c5 +#define DSM_DEBUZZ_NOTCH_A1_BYTE1 0x22c6 +#define DSM_DEBUZZ_NOTCH_A1_BYTE2 0x22c7 +#define DSM_DEBUZZ_NOTCH_A2_BYTE0 0x22c9 +#define DSM_DEBUZZ_NOTCH_A2_BYTE1 0x22ca +#define DSM_DEBUZZ_NOTCH_A2_BYTE2 0x22cb +#define DSM_THERMAL_BQ_B0_BYTE0 0x22cd +#define DSM_THERMAL_BQ_B0_BYTE1 0x22ce +#define DSM_THERMAL_BQ_B0_BYTE2 0x22cf +#define DSM_THERMAL_BQ_B1_BYTE0 0x22d1 +#define DSM_THERMAL_BQ_B1_BYTE1 0x22d2 +#define DSM_THERMAL_BQ_B1_BYTE2 0x22d3 +#define DSM_THERMAL_BQ_B2_BYTE0 0x22d5 +#define DSM_THERMAL_BQ_B2_BYTE1 0x22d6 +#define DSM_THERMAL_BQ_B2_BYTE2 0x22d7 +#define DSM_THERMAL_BQ_A1_BYTE0 0x22d9 +#define DSM_THERMAL_BQ_A1_BYTE1 0x22da +#define DSM_THERMAL_BQ_A1_BYTE2 0x22db +#define DSM_THERMAL_BQ_A2_BYTE0 0x22dd +#define DSM_THERMAL_BQ_A2_BYTE1 0x22de +#define DSM_THERMAL_BQ_A2_BYTE2 0x22df +#define DSM_WBDRC_FILT1_B0_BYTE0 0x22e1 +#define DSM_WBDRC_FILT1_B0_BYTE1 0x22e2 +#define DSM_WBDRC_FILT1_B0_BYTE2 0x22e3 +#define DSM_WBDRC_FILT1_B1_BYTE0 0x22e5 +#define DSM_WBDRC_FILT1_B1_BYTE1 0x22e6 +#define DSM_WBDRC_FILT1_B1_BYTE2 0x22e7 +#define DSM_WBDRC_FILT1_B2_BYTE0 0x22e9 +#define DSM_WBDRC_FILT1_B2_BYTE1 0x22ea +#define DSM_WBDRC_FILT1_B2_BYTE2 0x22eb +#define DSM_WBDRC_FILT1_A1_BYTE0 0x22ed +#define DSM_WBDRC_FILT1_A1_BYTE1 0x22ee +#define DSM_WBDRC_FILT1_A1_BYTE2 0x22ef +#define DSM_WBDRC_FILT1_A2_BYTE0 0x22f1 +#define DSM_WBDRC_FILT1_A2_BYTE1 0x22f2 +#define DSM_WBDRC_FILT1_A2_BYTE2 0x22f3 +#define DSM_WBDRC_FILT2_B0_BYTE0 0x22f5 +#define DSM_WBDRC_FILT2_B0_BYTE1 0x22f6 +#define DSM_WBDRC_FILT2_B0_BYTE2 0x22f7 +#define DSM_WBDRC_FILT2_B1_BYTE0 0x22f9 +#define DSM_WBDRC_FILT2_B1_BYTE1 0x22fa +#define DSM_WBDRC_FILT2_B1_BYTE2 0x22fb +#define DSM_WBDRC_FILT2_B2_BYTE0 0x22fd +#define DSM_WBDRC_FILT2_B2_BYTE1 0x22fe +#define DSM_WBDRC_FILT2_B2_BYTE2 0x22ff +#define DSM_WBDRC_FILT2_A1_BYTE0 0x2301 +#define DSM_WBDRC_FILT2_A1_BYTE1 0x2302 +#define DSM_WBDRC_FILT2_A1_BYTE2 0x2303 +#define DSM_WBDRC_FILT2_A2_BYTE0 0x2305 +#define DSM_WBDRC_FILT2_A2_BYTE1 0x2306 +#define DSM_WBDRC_FILT2_A2_BYTE2 0x2307 +#define DSM_PPR_RELEASE_TIME_BYTE0 0x2309 +#define DSM_PPR_RELEASE_TIME_BYTE1 0x230a +#define DSM_PPR_RELEASE_TIME_BYTE2 0x230b +#define DSM_PPR_ATTACK_TIME_BYTE0 0x230d +#define DSM_PPR_ATTACK_TIME_BYTE1 0x230e +#define DSM_PPR_ATTACK_TIME_BYTE2 0x230f +#define DSM_DEBUZZER_RELEASE_TIME_BYTE0 0x2311 +#define DSM_DEBUZZER_RELEASE_TIME_BYTE1 0x2312 +#define DSM_DEBUZZER_RELEASE_TIME_BYTE2 0x2313 +#define DSM_DEBUZZER_ATTACK_TIME_BYTE0 0x2315 +#define DSM_DEBUZZER_ATTACK_TIME_BYTE1 0x2316 +#define DSM_DEBUZZER_ATTACK_TIME_BYTE2 0x2317 + +#define DSMIG_WB_DRC_RELEASE_TIME_1 0x2380 +#define DSMIG_WB_DRC_RELEASE_TIME_2 0x2381 +#define DSMIG_WB_DRC_ATTACK_TIME_1 0x2382 +#define DSMIG_WB_DRC_ATTACK_TIME_2 0x2383 +#define DSMIG_WB_DRC_COMPRESSION_RATIO 0x2384 +#define DSMIG_WB_DRC_COMPRESSION_THRESHOLD 0x2385 +#define DSMIG_WB_DRC_MAKEUPGAIN 0x2386 +#define DSMIG_WB_DRC_NOISE_GATE_THRESHOLD 0x2387 +#define DSMIG_WBDRC_HPF_ENABLE 0x2388 +#define DSMIG_WB_DRC_TEST_SMOOTHER_OUT_EN 0x2389 +#define DSMIG_PPR_THRESHOLD 0x238b +#define DSM_STEREO_BASS_CHANNEL_SELECT 0x238d +#define DSM_TPROT_THRESHOLD_BYTE0 0x238e +#define DSM_TPROT_THRESHOLD_BYTE1 0x238f +#define DSM_TPROT_ROOM_TEMPERATURE_BYTE0 0x2390 +#define DSM_TPROT_ROOM_TEMPERATURE_BYTE1 0x2391 +#define DSM_TPROT_RECIP_RDC_ROOM_BYTE0 0x2392 +#define DSM_TPROT_RECIP_RDC_ROOM_BYTE1 0x2393 +#define DSM_TPROT_RECIP_RDC_ROOM_BYTE2 0x2394 +#define DSM_TPROT_RECIP_TCONST_BYTE0 0x2395 +#define DSM_TPROT_RECIP_TCONST_BYTE1 0x2396 +#define DSM_TPROT_RECIP_TCONST_BYTE2 0x2397 +#define DSM_THERMAL_ATTENUATION_SETTINGS 0x2398 +#define DSM_THERMAL_PILOT_TONE_ATTENUATION 0x2399 +#define DSM_TPROT_PG_TEMP_THRESH_BYTE0 0x239a +#define DSM_TPROT_PG_TEMP_THRESH_BYTE1 0x239b + +#define THERMAL_RDC_RD_BACK_BYTE1 0x239c +#define THERMAL_RDC_RD_BACK_BYTE0 0x239d +#define THERMAL_COILTEMP_RD_BACK_BYTE1 0x239e +#define THERMAL_COILTEMP_RD_BACK_BYTE0 0x239f + +#define DSMIG_DEBUZZER_THRESHOLD 0x23b5 +#define DSMIG_DEBUZZER_ALPHA_COEF_TEST_ONLY 0x23b6 +#define DSM_VOL_ENA 0x23b9 +#define DSM_VOL_CTRL 0x23ba + +#define DSMIG_EN 0x23e0 +#define MAX98390_R23E1_DSP_GLOBAL_EN 0x23e1 + +#define DSM_THERMAL_GAIN 0x23f0 +#define DSM_PPR_GAIN 0x23f1 +#define DSM_DBZ_GAIN 0x23f2 +#define DSM_WBDRC_GAIN 0x23f3 + +#define MAX98390_R23FF_GLOBAL_EN 0x23FF +#define MAX98390_R24FF_REV_ID 0x24FF + +/* MAX98390_R2021_PCM_RX_SRC_1 */ +#define MAX98390_PCM_RX_CH_SRC_SHIFT (0) +#define MAX98390_PCM_RX_CH_SRC_BASS_SHIFT (4) + +/* MAX98390_R2022_PCM_TX_SRC_1 */ +#define MAX98390_PCM_TX_CH_SRC_A_V_SHIFT (0) +#define MAX98390_PCM_TX_CH_SRC_A_I_SHIFT (4) + +/* MAX98390_R2024_PCM_DATA_FMT_CFG */ +#define MAX98390_PCM_MODE_CFG_FORMAT_MASK (0x7 << 3) +#define MAX98390_PCM_MODE_CFG_FORMAT_SHIFT (3) +#define MAX98390_PCM_TX_CH_INTERLEAVE_MASK (0x1 << 2) +#define MAX98390_PCM_FORMAT_I2S (0x0 << 0) +#define MAX98390_PCM_FORMAT_LJ (0x1 << 0) +#define MAX98390_PCM_FORMAT_TDM_MODE0 (0x3 << 0) +#define MAX98390_PCM_FORMAT_TDM_MODE1 (0x4 << 0) +#define MAX98390_PCM_FORMAT_TDM_MODE2 (0x5 << 0) +#define MAX98390_PCM_MODE_CFG_CHANSZ_MASK (0x3 << 6) +#define MAX98390_PCM_MODE_CFG_CHANSZ_16 (0x1 << 6) +#define MAX98390_PCM_MODE_CFG_CHANSZ_24 (0x2 << 6) +#define MAX98390_PCM_MODE_CFG_CHANSZ_32 (0x3 << 6) + +/* MAX98390_R2039_AMP_DSP_CFG */ +#define MAX98390_AMP_DSP_CFG_RMP_UP_SHIFT (4) +#define MAX98390_AMP_DSP_CFG_RMP_DN_SHIFT (5) + +/* MAX98390_R203A_AMP_EN */ +#define MAX98390_R203A_AMP_EN_SHIFT (0) + +/* MAX98390_PCM_MASTER_MODE */ +#define MAX98390_PCM_MASTER_MODE_MASK (0x3 << 0) +#define MAX98390_PCM_MASTER_MODE_SLAVE (0x0 << 0) +#define MAX98390_PCM_MASTER_MODE_MASTER (0x3 << 0) + +#define MAX98390_PCM_MASTER_MODE_MCLK_MASK (0xF << 2) +#define MAX98390_PCM_MASTER_MODE_MCLK_RATE_SHIFT (2) + +/* PCM_CLK_SETUP */ +#define MAX98390_PCM_MODE_CFG_PCM_BCLKEDGE (0x1 << 2) +#define MAX98390_PCM_CLK_SETUP_BSEL_MASK (0xF << 0) + +/* PCM_SR_SETUP */ +#define MAX98390_PCM_SR_SET1_SR_MASK (0xF << 0) +#define MAX98390_PCM_SR_SET1_SR_8000 (0x0 << 0) +#define MAX98390_PCM_SR_SET1_SR_11025 (0x1 << 0) +#define MAX98390_PCM_SR_SET1_SR_12000 (0x2 << 0) +#define MAX98390_PCM_SR_SET1_SR_16000 (0x3 << 0) +#define MAX98390_PCM_SR_SET1_SR_22050 (0x4 << 0) +#define MAX98390_PCM_SR_SET1_SR_24000 (0x5 << 0) +#define MAX98390_PCM_SR_SET1_SR_32000 (0x6 << 0) +#define MAX98390_PCM_SR_SET1_SR_44100 (0x7 << 0) +#define MAX98390_PCM_SR_SET1_SR_48000 (0x8 << 0) + +/* PCM_TO_SPK_MONO_MIX_1 */ +#define MAX98390_PCM_TO_SPK_MONOMIX_CFG_MASK (0x3 << 6) +#define MAX98390_PCM_TO_SPK_MONOMIX_CFG_SHIFT (6) +#define MAX98390_PCM_TO_SPK_CH0_SRC_MASK (0xF << 0) +#define MAX98390_PCM_TO_SPK_CH1_SRC_MASK (0xF << 4) + +/* MAX98390_BOOST_CTRL3 */ +#define MAX98390_BOOST_CLK_PHASE_CFG_SHIFT (2) + +/* SOFT_RESET */ +#define MAX98390_SOFT_RESET_MASK (0x1 << 0) + +#define MAX98390_GLOBAL_EN_MASK (0x1 << 0) +#define MAX98390_AMP_EN_MASK (0x1 << 0) + +/* DSM register offset */ +#define MAX98390_DSM_PAYLOAD_OFFSET 16 +#define MAX98390_DSM_PAYLOAD_OFFSET_2 495 + +struct max98390_priv { + struct regmap *regmap; + unsigned int sysclk; + unsigned int master; + unsigned int tdm_mode; + unsigned int ref_rdc_value; + unsigned int ambient_temp_value; +}; +#endif -- 2.17.1

3 4

Missing hotplug support
by GitHub issues - opened 31 May '20

31 May '20

alsa-project/alsa-lib issue #55 was opened from wm4: Hotplug support is mandatory in any modern system. But in ALSA there is no way to be notified by audio device changes. I'm aware that as a hack, one could listen to udev notifications. But this is not appropriate, because: - udev is a different API, which makes this is inelegant and messy - it requires a dependency on udev, which may not be a workable solution even if the host OS is Linux - there may be devices that do not have anything to do with udev (for example network audio interfaces implemented in userspace) - changes in udev may not be reflected in ALSA at the same time (creating the need to avoid race conditions with timeouts or whatever) - it's ALSA's responsibility to provide this; I don't know any OS where you have to go to a completely different API for this Issue URL : https://github.com/alsa-project/alsa-lib/issues/55 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

No way to get notified of playback progress
by GitHub issues - opened 31 May '20

31 May '20

alsa-project/alsa-lib issue #54 was opened from wm4: See subject. There is a way to run user code in a signal handler with snd_async_add_pcm_handler(). But that seems extremely questionable, is for sure not supported everywhere, and is probably full of subtle synchronization bugs. There is snd_pcm_poll_descriptors() and related, but that seems to signal only whether at least 1 period is available, nothing else. Judging from the docs. That seems like a rather big missing thing - I probably overlooked something. Otherwise this is a feature request for an asynchronous notification on period boundaries/underruns/device disconnection, that does not run in a UNIX signal handler. Maybe something relatively similar to that poll() mechanism. Issue URL : https://github.com/alsa-project/alsa-lib/issues/54 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v2 00/19] ASoC: add soc-card
by Kuninori Morimoto 30 May '20

30 May '20

Hi Mark Current ALSA SoC is handling snd_soc_card related operation, but it is implmemented directly without using function/macro, and at random place. This v2 patch-set creates new snd_soc_card_xxx() functions which handles snd_soc_card related operation, and implmement these at new soc-card.c. v1 -> v2 - careed scripts/checkpatch.pl except for MAINTAINERS file update - added detail log at git-log, and/or comment at file. - cares "if (card)" under snd_soc_card_xxx() Link: https://lore.kernel.org/r/87h7w3339l.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (19): 1 ASoC: soc.h: convert bool to bit field for snd_soc_card 2 ASoC: add soc-card.c 3 ASoC: soc-card: move snd_soc_card_get_kcontrol() to soc-card 4 ASoC: soc-card: move snd_soc_card_jack_new() to soc-card 5 ASoC: soc-card: move snd_soc_card_set/get_drvdata() to soc-card 6 ASoC: soc-card: move snd_soc_card_get_codec_dai() to soc-card 7 ASoC: soc-card: move snd_soc_card_subclass to soc-card 8 ASoC: soc-card: add snd_soc_card_suspend_pre() 9 ASoC: soc-card: add snd_soc_card_suspend_post() 10 ASoC: soc-card: add snd_soc_card_resume_pre() 11 ASoC: soc-card: add snd_soc_card_resume_post() 12 ASoC: soc-card: add probed bit field to snd_soc_card 13 ASoC: soc-card: add snd_soc_card_probe() 14 ASoC: soc-card: add snd_soc_card_late_probe() 15 ASoC: soc-card: add snd_soc_card_remove() 16 ASoC: soc-card: add snd_soc_card_set_bias_level() 17 ASoC: soc-card: add snd_soc_card_set_bias_level_post() 18 ASoC: soc-card: add snd_soc_card_add_dai_link() 19 ASoC: soc-card: add snd_soc_card_remove_dai_link() include/sound/soc-card.h | 69 ++++++++++++ include/sound/soc.h | 53 ++------- sound/soc/Makefile | 2 +- sound/soc/soc-card.c | 225 +++++++++++++++++++++++++++++++++++++++ sound/soc/soc-core.c | 71 ++++-------- sound/soc/soc-dapm.c | 6 +- sound/soc/soc-jack.c | 38 ------- 7 files changed, 325 insertions(+), 139 deletions(-) create mode 100644 include/sound/soc-card.h create mode 100644 sound/soc/soc-card.c -- 2.17.1

3 21

[PATCH] ASoC: SOF: Intel: byt: fix unused-function warnings
by Arnd Bergmann 30 May '20

30 May '20

Several functions in this driver are only referenced for baytrail-class configurations and building configurations with only merrifield enabled causes a warning: sound/soc/sof/intel/byt.c:447:12: error: unused function 'byt_suspend' [-Werror,-Wunused-function] static int byt_suspend(struct snd_sof_dev *sdev, u32 target_state) ^ sound/soc/sof/intel/byt.c:454:12: error: unused function 'byt_resume' [-Werror,-Wunused-function] static int byt_resume(struct snd_sof_dev *sdev) ^ sound/soc/sof/intel/byt.c:464:12: error: unused function 'byt_remove' [-Werror,-Wunused-function] static int byt_remove(struct snd_sof_dev *sdev) Move these into a corresponding #ifdef section. Fixes: c691f0c6e267 ("ASoC: SOF: Intel: BYT: add .remove op") Fixes: ddcccd543f5d ("ASoC: SOF: Intel: byt: Add PM callbacks") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/sof/intel/byt.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sof/intel/byt.c b/sound/soc/sof/intel/byt.c index e6ba8382b1de..6ae62a8a9daf 100644 --- a/sound/soc/sof/intel/byt.c +++ b/sound/soc/sof/intel/byt.c @@ -432,6 +432,7 @@ static void byt_set_mach_params(const struct snd_soc_acpi_mach *mach, mach_params->platform = dev_name(dev); } +#if IS_ENABLED(CONFIG_SND_SOC_SOF_BAYTRAIL) static void byt_reset_dsp_disable_int(struct snd_sof_dev *sdev) { /* Disable Interrupt from both sides */ @@ -467,6 +468,7 @@ static int byt_remove(struct snd_sof_dev *sdev) return 0; } +#endif /* Baytrail DAIs */ static struct snd_soc_dai_driver byt_dai[] = { -- 2.26.2

2 1

[PATCH 0/2] RZ/G1H enable sound support
by Lad Prabhakar 29 May '20

29 May '20

Hi All, This patch series adds support for sound in R8A7742 SoC DT. Cheers, Prabhakar Lad Prabhakar (2): dt-bindings: ASoC: renesas,rsnd: Add r8a7742 support ARM: dts: r8a7742: Add audio support .../devicetree/bindings/sound/renesas,rsnd.txt | 1 + arch/arm/boot/dts/r8a7742.dtsi | 284 +++++++++++++++++++++ 2 files changed, 285 insertions(+) -- 2.7.4

5 8

[PATCH v2 0/3] Kconfig updates for DMIC and SOF HDMI support
by Pierre-Louis Bossart 29 May '20

29 May '20

This series provides the following updates to the Intel machine driver Kconfig: 1. The first patch adds the explicit dependency on GPIOLIB when SND_SOC_DMIC is selected. 2. SND_SOC_SOF_HDA_AUDIO_CODEC is required for using the legacy HDA codec driver for HDMI support in SOF. The last 3 three patches make the required changes to account for this. changes since v1: first patch for DMIC was merged already rebase for HDMI on top of Arnd's RT5682 changes. Libin Yang (3): ASoC: intel: add depends on SND_SOC_SOF_HDA_AUDIO_CODEC for common hdmi ASoC: sof-sdw: remove CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC condition ASoC: sof_pcm512x: remove CONFIG_SND_HDA_CODEC_HDMI condition sound/soc/intel/boards/Kconfig | 20 ++++++++++---------- sound/soc/intel/boards/sof_pcm512x.c | 9 --------- sound/soc/intel/boards/sof_sdw.c | 8 -------- sound/soc/intel/boards/sof_sdw_hdmi.c | 7 ------- 4 files changed, 10 insertions(+), 34 deletions(-) base-commit: 89d73ccab20a684d8446cea4d8ac6a2608c8d390 -- 2.20.1

2 4

[PATCH] ASoC: SOF: Intel: hda: fix generic hda codec support
by Kai Vehmanen 29 May '20

29 May '20

Add support for using generic codec driver with SOF. Generic driver is used if: - snd_sof_intel_hda_common.hda_model="generic" is set, or - fallback if no other codec driver is found The implementation is aligned with snd-hda-intel driver, and fixes audio support for systems like Acer Swift 3 SF314-57G, on which this issue was originally reported. BugLink: https://github.com/thesofproject/linux/issues/1807 BugLink: https://bugs.launchpad.net/bugs/1877757 Reviewed-by: Hui Wang <hui.wang(a)canonical.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/sof/intel/hda-codec.c | 51 +++++++++++++++++++++++++++++---- 1 file changed, 45 insertions(+), 6 deletions(-) diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index c3299be119a1..2c5c451fa19d 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -24,19 +24,44 @@ #define IDISP_VID_INTEL 0x80860000 /* load the legacy HDA codec driver */ -static int hda_codec_load_module(struct hda_codec *codec) +static int request_codec_module(struct hda_codec *codec) { #ifdef MODULE char alias[MODULE_NAME_LEN]; - const char *module = alias; + const char *mod = NULL; - snd_hdac_codec_modalias(&codec->core, alias, sizeof(alias)); - dev_dbg(&codec->core.dev, "loading codec module: %s\n", module); - request_module(module); + switch (codec->probe_id) { + case HDA_CODEC_ID_GENERIC: +#if IS_MODULE(CONFIG_SND_HDA_GENERIC) + mod = "snd-hda-codec-generic"; #endif + break; + default: + snd_hdac_codec_modalias(&codec->core, alias, sizeof(alias)); + mod = alias; + break; + } + + if (mod) { + dev_dbg(&codec->core.dev, "loading codec module: %s\n", mod); + request_module(mod); + } +#endif /* MODULE */ return device_attach(hda_codec_dev(codec)); } +static int hda_codec_load_module(struct hda_codec *codec) +{ + int ret = request_codec_module(codec); + + if (ret <= 0) { + codec->probe_id = HDA_CODEC_ID_GENERIC; + ret = request_codec_module(codec); + } + + return ret; +} + /* enable controller wake up event for all codecs with jack connectors */ void hda_codec_jack_wake_enable(struct snd_sof_dev *sdev) { @@ -78,6 +103,13 @@ void hda_codec_jack_check(struct snd_sof_dev *sdev) {} EXPORT_SYMBOL_NS(hda_codec_jack_wake_enable, SND_SOC_SOF_HDA_AUDIO_CODEC); EXPORT_SYMBOL_NS(hda_codec_jack_check, SND_SOC_SOF_HDA_AUDIO_CODEC); +#if IS_ENABLED(CONFIG_SND_HDA_GENERIC) +#define is_generic_config(bus) \ + ((bus)->modelname && !strcmp((bus)->modelname, "generic")) +#else +#define is_generic_config(x) 0 +#endif + /* probe individual codec */ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, bool hda_codec_use_common_hdmi) @@ -87,6 +119,7 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, #endif struct hda_bus *hbus = sof_to_hbus(sdev); struct hdac_device *hdev; + struct hda_codec *codec; u32 hda_cmd = (address << 28) | (AC_NODE_ROOT << 20) | (AC_VERB_PARAMETERS << 8) | AC_PAR_VENDOR_ID; u32 resp = -1; @@ -108,6 +141,7 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, hda_priv->codec.bus = hbus; hdev = &hda_priv->codec.core; + codec = &hda_priv->codec; ret = snd_hdac_ext_bus_device_init(&hbus->core, address, hdev); if (ret < 0) @@ -122,6 +156,11 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, hda_priv->need_display_power = true; } + if (is_generic_config(hbus)) + codec->probe_id = HDA_CODEC_ID_GENERIC; + else + codec->probe_id = 0; + /* * if common HDMI codec driver is not used, codec load * is skipped here and hdac_hdmi is used instead @@ -129,7 +168,7 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, if (hda_codec_use_common_hdmi || (resp & 0xFFFF0000) != IDISP_VID_INTEL) { hdev->type = HDA_DEV_LEGACY; - ret = hda_codec_load_module(&hda_priv->codec); + ret = hda_codec_load_module(codec); /* * handle ret==0 (no driver bound) as an error, but pass * other return codes without modification -- 2.26.2

2 1

[PATCH] ASoC: mediatek: mt6358: support DMIC one-wire mode
by Jiaxin Yu 29 May '20

29 May '20

Supports DMIC one-wire mode. Adds a mixer control to enable and disable. Signed-off-by: Jiaxin Yu <jiaxin.yu(a)mediatek.com> Reviewed-by: Tzung-Bi Shih <tzungbi(a)google.com> --- sound/soc/codecs/mt6358.c | 33 ++++++++++++++++++++++++++++++++- 1 file changed, 32 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/mt6358.c b/sound/soc/codecs/mt6358.c index 1b830ea..ca7628d 100644 --- a/sound/soc/codecs/mt6358.c +++ b/sound/soc/codecs/mt6358.c @@ -95,6 +95,8 @@ struct mt6358_priv { struct regulator *avdd_reg; int wov_enabled; + + int dmic_one_wire_mode; }; int mt6358_set_mtkaif_protocol(struct snd_soc_component *cmpnt, @@ -566,6 +568,28 @@ static int mt6358_put_wov(struct snd_kcontrol *kcontrol, return 0; } +static int mt6358_dmic_one_wire_mode_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct mt6358_priv *priv = snd_soc_component_get_drvdata(cmpnt); + + ucontrol->value.integer.value[0] = priv->dmic_one_wire_mode; + + return 0; +} + +static int mt6358_dmic_one_wire_mode_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct mt6358_priv *priv = snd_soc_component_get_drvdata(cmpnt); + + priv->dmic_one_wire_mode = ucontrol->value.integer.value[0]; + + return 0; +} + static const DECLARE_TLV_DB_SCALE(playback_tlv, -1000, 100, 0); static const DECLARE_TLV_DB_SCALE(pga_tlv, 0, 600, 0); @@ -588,6 +612,10 @@ static int mt6358_put_wov(struct snd_kcontrol *kcontrol, SOC_SINGLE_BOOL_EXT("Wake-on-Voice Phase2 Switch", 0, mt6358_get_wov, mt6358_put_wov), + + SOC_SINGLE_BOOL_EXT("Dmic One Wire Mode", 0, + mt6358_dmic_one_wire_mode_get, + mt6358_dmic_one_wire_mode_set), }; /* MUX */ @@ -1740,7 +1768,10 @@ static int mt6358_amic_enable(struct mt6358_priv *priv) mt6358_mtkaif_tx_enable(priv); /* UL dmic setting off */ - regmap_write(priv->regmap, MT6358_AFE_UL_SRC_CON0_H, 0x0000); + if (priv->dmic_one_wire_mode) + regmap_write(priv->regmap, MT6358_AFE_UL_SRC_CON0_H, 0x0400); + else + regmap_write(priv->regmap, MT6358_AFE_UL_SRC_CON0_H, 0x0080); /* UL turn on */ regmap_write(priv->regmap, MT6358_AFE_UL_SRC_CON0_L, 0x0001); -- 1.8.1.1.dirty

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[PATCH] ALSA: Fixing usage of plain int instead of NULL
by Thomas Ebeling 29 May '20

29 May '20

As reported by kbuild test robot, mixer quirks for RME Babyface Pro used plain integer instead of NULL. Signed-off-by: Thomas Ebeling <penguins(a)bollie.de> Reported-by: kbuild test robot <lkp(a)intel.com> --- sound/usb/mixer_quirks.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index aad2683ff793..b6bcf2f92383 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -2255,7 +2255,7 @@ static int snd_bbfpro_ctl_update(struct usb_mixer_interface *mixer, u8 reg, err = snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), usb_req, USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_DEVICE, - usb_val, usb_idx, 0, 0); + usb_val, usb_idx, NULL, 0); snd_usb_unlock_shutdown(chip); return err; @@ -2394,7 +2394,7 @@ static int snd_bbfpro_vol_update(struct usb_mixer_interface *mixer, u16 index, SND_BBFPRO_USBREQ_MIXER, USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_DEVICE, - usb_val, usb_idx, 0, 0); + usb_val, usb_idx, NULL, 0); snd_usb_unlock_shutdown(chip); return err; -- 2.26.2

2 1

[PATCH] ASoC: img-i2s-out: Fix runtime PM imbalance on error
by Dinghao Liu 29 May '20

29 May '20

pm_runtime_get_sync() increments the runtime PM usage counter even the call returns an error code. Thus a pairing decrement is needed on the error handling path to keep the counter balanced. Signed-off-by: Dinghao Liu <dinghao.liu(a)zju.edu.cn> --- sound/soc/img/img-i2s-out.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/img/img-i2s-out.c b/sound/soc/img/img-i2s-out.c index db052ec17d5d..b56a18e7f3ac 100644 --- a/sound/soc/img/img-i2s-out.c +++ b/sound/soc/img/img-i2s-out.c @@ -347,8 +347,10 @@ static int img_i2s_out_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) chan_control_mask = IMG_I2S_OUT_CHAN_CTL_CLKT_MASK; ret = pm_runtime_get_sync(i2s->dev); - if (ret < 0) + if (ret < 0) { + pm_runtime_put_noidle(i2s->dev); return ret; + } img_i2s_out_disable(i2s); @@ -488,8 +490,10 @@ static int img_i2s_out_probe(struct platform_device *pdev) goto err_pm_disable; } ret = pm_runtime_get_sync(&pdev->dev); - if (ret < 0) + if (ret < 0) { + pm_runtime_put_noidle(&pdev->dev); goto err_suspend; + } reg = IMG_I2S_OUT_CTL_FRM_SIZE_MASK; img_i2s_out_writel(i2s, reg, IMG_I2S_OUT_CTL); -- 2.17.1

2 1

[PATCH] ASoC: SOF: Intel: Baytrail: fix 'defined but not used' warnings
by Pierre-Louis Bossart 29 May '20

29 May '20

With the allmodconfig option, CONFIG_SND_SOC_SOF_BAYTRAIL is disabled due to mutual exclusion with the legacy driver. This generates 'defined by not used' warnings. suspend/resume/remove are only supported for Baytrail for now, so move the code under the CONFIG_SND_SOC_SOF_BAYTRAIL checks. Fixes: ddcccd543f5d ("ASoC: SOF: Intel: byt: Add PM callbacks") Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/intel/byt.c | 72 +++++++++++++++++++-------------------- 1 file changed, 36 insertions(+), 36 deletions(-) diff --git a/sound/soc/sof/intel/byt.c b/sound/soc/sof/intel/byt.c index e6ba8382b1de..49f67f1b94e0 100644 --- a/sound/soc/sof/intel/byt.c +++ b/sound/soc/sof/intel/byt.c @@ -432,42 +432,6 @@ static void byt_set_mach_params(const struct snd_soc_acpi_mach *mach, mach_params->platform = dev_name(dev); } -static void byt_reset_dsp_disable_int(struct snd_sof_dev *sdev) -{ - /* Disable Interrupt from both sides */ - snd_sof_dsp_update_bits64(sdev, BYT_DSP_BAR, SHIM_IMRX, 0x3, 0x3); - snd_sof_dsp_update_bits64(sdev, BYT_DSP_BAR, SHIM_IMRD, 0x3, 0x3); - - /* Put DSP into reset, set reset vector */ - snd_sof_dsp_update_bits64(sdev, BYT_DSP_BAR, SHIM_CSR, - SHIM_BYT_CSR_RST | SHIM_BYT_CSR_VECTOR_SEL, - SHIM_BYT_CSR_RST | SHIM_BYT_CSR_VECTOR_SEL); -} - -static int byt_suspend(struct snd_sof_dev *sdev, u32 target_state) -{ - byt_reset_dsp_disable_int(sdev); - - return 0; -} - -static int byt_resume(struct snd_sof_dev *sdev) -{ - /* enable BUSY and disable DONE Interrupt by default */ - snd_sof_dsp_update_bits64(sdev, BYT_DSP_BAR, SHIM_IMRX, - SHIM_IMRX_BUSY | SHIM_IMRX_DONE, - SHIM_IMRX_DONE); - - return 0; -} - -static int byt_remove(struct snd_sof_dev *sdev) -{ - byt_reset_dsp_disable_int(sdev); - - return 0; -} - /* Baytrail DAIs */ static struct snd_soc_dai_driver byt_dai[] = { { @@ -695,6 +659,42 @@ EXPORT_SYMBOL_NS(tng_chip_info, SND_SOC_SOF_MERRIFIELD); #if IS_ENABLED(CONFIG_SND_SOC_SOF_BAYTRAIL) +static void byt_reset_dsp_disable_int(struct snd_sof_dev *sdev) +{ + /* Disable Interrupt from both sides */ + snd_sof_dsp_update_bits64(sdev, BYT_DSP_BAR, SHIM_IMRX, 0x3, 0x3); + snd_sof_dsp_update_bits64(sdev, BYT_DSP_BAR, SHIM_IMRD, 0x3, 0x3); + + /* Put DSP into reset, set reset vector */ + snd_sof_dsp_update_bits64(sdev, BYT_DSP_BAR, SHIM_CSR, + SHIM_BYT_CSR_RST | SHIM_BYT_CSR_VECTOR_SEL, + SHIM_BYT_CSR_RST | SHIM_BYT_CSR_VECTOR_SEL); +} + +static int byt_suspend(struct snd_sof_dev *sdev, u32 target_state) +{ + byt_reset_dsp_disable_int(sdev); + + return 0; +} + +static int byt_resume(struct snd_sof_dev *sdev) +{ + /* enable BUSY and disable DONE Interrupt by default */ + snd_sof_dsp_update_bits64(sdev, BYT_DSP_BAR, SHIM_IMRX, + SHIM_IMRX_BUSY | SHIM_IMRX_DONE, + SHIM_IMRX_DONE); + + return 0; +} + +static int byt_remove(struct snd_sof_dev *sdev) +{ + byt_reset_dsp_disable_int(sdev); + + return 0; +} + static const struct snd_sof_debugfs_map cht_debugfs[] = { {"dmac0", BYT_DSP_BAR, DMAC0_OFFSET, DMAC_SIZE, SOF_DEBUGFS_ACCESS_ALWAYS}, -- 2.20.1

2 1

[PATCH] dt-bindings: sound: tlv320adcx140: Fix dt-binding-check issue
by Dan Murphy 29 May '20

29 May '20

Fix dt-binding-check issue ti,gpi-config:0:0: 4 is greater than the maximum of 1 ti,gpi-config:0:1: 5 is greater than the maximum of 1 ti,gpi-config:0:2: 6 is greater than the maximum of 1 ti,gpi-config:0:3: 7 is greater than the maximum of 1 Reported-by: Rob Herring <robh(a)kernel.org> Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- Documentation/devicetree/bindings/sound/tlv320adcx140.yaml | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml index e8a69b1c7ca9..306ac3d006dc 100644 --- a/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml +++ b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml @@ -109,7 +109,7 @@ properties: - minItems: 1 maxItems: 4 items: - maximum: 1 + maximum: 7 default: [0, 0, 0, 0] required: -- 2.26.2

3 3

[PATCH] ASoC: fsl_asrc_dma: Fix dma_chan leak when config DMA channel failed
by Xiyu Yang 29 May '20

29 May '20

fsl_asrc_dma_hw_params() invokes dma_request_channel() or fsl_asrc_get_dma_channel(), which returns a reference of the specified dma_chan object to "pair->dma_chan[dir]" with increased refcnt. The reference counting issue happens in one exception handling path of fsl_asrc_dma_hw_params(). When config DMA channel failed for Back-End, the function forgets to decrease the refcnt increased by dma_request_channel() or fsl_asrc_get_dma_channel(), causing a refcnt leak. Fix this issue by calling dma_release_channel() when config DMA channel failed. Signed-off-by: Xiyu Yang <xiyuyang19(a)fudan.edu.cn> Signed-off-by: Xin Tan <tanxin.ctf(a)gmail.com> --- sound/soc/fsl/fsl_asrc_dma.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index e7178817d7a7..1ee10eafe3e6 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -252,6 +252,7 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, ret = dmaengine_slave_config(pair->dma_chan[dir], &config_be); if (ret) { dev_err(dev, "failed to config DMA channel for Back-End\n"); + dma_release_channel(pair->dma_chan[dir]); return ret; } -- 2.7.4

2 1

[PATCH v3 0/2] ASoC topology header parsing refinement
by Keyon Jie 29 May '20

29 May '20

This small series is to optimize the header logging during the topology parsing. This is verified work fine on both SOF and SST drivers. Change History: v3: - Remove using the separated soc_pass_load() function and merge it to the soc_tplg_load_header() body. - Add more Tested-by tags. v2: - Change the internal used array to be 'static' to fix the issue reported by: kbuild test robot <lkp(a)intel.com> - Add testing coverage including Intel SST driver also. v1: - Initial version. Keyon Jie (2): ASoC: topology: refine and log the header in the correct pass ASoC: topology: remove the redundant pass checks sound/soc/soc-topology.c | 96 ++++++++++++++++------------------------ 1 file changed, 38 insertions(+), 58 deletions(-) -- 2.25.1

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