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

[PATCH AUTOSEL 5.10 04/17] ASoC: tlv320aic31xx: fix reversed bclk/wclk master bits
by Sasha Levin 27 Jul '21

27 Jul '21

From: Kyle Russell <bkylerussell(a)gmail.com> [ Upstream commit 9cf76a72af6ab81030dea6481b1d7bdd814fbdaf ] These are backwards from Table 7-71 of the TLV320AIC3100 spec [1]. This was broken in 12eb4d66ba2e when BCLK_MASTER and WCLK_MASTER were converted from 0x08 and 0x04 to BIT(2) and BIT(3), respectively. -#define AIC31XX_BCLK_MASTER 0x08 -#define AIC31XX_WCLK_MASTER 0x04 +#define AIC31XX_BCLK_MASTER BIT(2) +#define AIC31XX_WCLK_MASTER BIT(3) Probably just a typo since the defines were not listed in bit order. [1] https://www.ti.com/lit/gpn/tlv320aic3100 Signed-off-by: Kyle Russell <bkylerussell(a)gmail.com> Link: https://lore.kernel.org/r/20210622010941.241386-1-bkylerussell@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/tlv320aic31xx.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h index 81952984613d..2513922a0292 100644 --- a/sound/soc/codecs/tlv320aic31xx.h +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -151,8 +151,8 @@ struct aic31xx_pdata { #define AIC31XX_WORD_LEN_24BITS 0x02 #define AIC31XX_WORD_LEN_32BITS 0x03 #define AIC31XX_IFACE1_MASTER_MASK GENMASK(3, 2) -#define AIC31XX_BCLK_MASTER BIT(2) -#define AIC31XX_WCLK_MASTER BIT(3) +#define AIC31XX_BCLK_MASTER BIT(3) +#define AIC31XX_WCLK_MASTER BIT(2) /* AIC31XX_DATA_OFFSET */ #define AIC31XX_DATA_OFFSET_MASK GENMASK(7, 0) -- 2.30.2

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[PATCH AUTOSEL 5.13 14/21] ASoC: ti: j721e-evm: Check for not initialized parent_clk_id
by Sasha Levin 27 Jul '21

27 Jul '21

From: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> [ Upstream commit 82d28b67f780910f816fe1cfb0f676fc38c4cbb3 ] During probe the parent_clk_id is set to -1 which should not be used to array index within hsdiv_rates[]. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> Link: https://lore.kernel.org/r/20210717122820.1467-3-peter.ujfalusi@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/ti/j721e-evm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/ti/j721e-evm.c b/sound/soc/ti/j721e-evm.c index 017c4ad11ca6..265bbc5a2f96 100644 --- a/sound/soc/ti/j721e-evm.c +++ b/sound/soc/ti/j721e-evm.c @@ -197,7 +197,7 @@ static int j721e_configure_refclk(struct j721e_priv *priv, return ret; } - if (priv->hsdiv_rates[domain->parent_clk_id] != scki) { + if (domain->parent_clk_id == -1 || priv->hsdiv_rates[domain->parent_clk_id] != scki) { dev_dbg(priv->dev, "%s configuration for %u Hz: %s, %dxFS (SCKI: %u Hz)\n", audio_domain == J721E_AUDIO_DOMAIN_CPB ? "CPB" : "IVI", -- 2.30.2

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[PATCH AUTOSEL 5.13 13/21] ASoC: ti: j721e-evm: Fix unbalanced domain activity tracking during startup
by Sasha Levin 27 Jul '21

27 Jul '21

From: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> [ Upstream commit 78d2a05ef22e7b5863b01e073dd6a06b3979bb00 ] In case of an error within j721e_audio_startup() the domain->active must be decremented to avoid unbalanced counter. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> Link: https://lore.kernel.org/r/20210717122820.1467-2-peter.ujfalusi@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/ti/j721e-evm.c | 16 +++++++++++----- 1 file changed, 11 insertions(+), 5 deletions(-) diff --git a/sound/soc/ti/j721e-evm.c b/sound/soc/ti/j721e-evm.c index a7c0484d44ec..017c4ad11ca6 100644 --- a/sound/soc/ti/j721e-evm.c +++ b/sound/soc/ti/j721e-evm.c @@ -278,23 +278,29 @@ static int j721e_audio_startup(struct snd_pcm_substream *substream) j721e_rule_rate, &priv->rate_range, SNDRV_PCM_HW_PARAM_RATE, -1); - mutex_unlock(&priv->mutex); if (ret) - return ret; + goto out; /* Reset TDM slots to 32 */ ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 32); if (ret && ret != -ENOTSUPP) - return ret; + goto out; for_each_rtd_codec_dais(rtd, i, codec_dai) { ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x3, 0x3, 2, 32); if (ret && ret != -ENOTSUPP) - return ret; + goto out; } - return 0; + if (ret == -ENOTSUPP) + ret = 0; +out: + if (ret) + domain->active--; + mutex_unlock(&priv->mutex); + + return ret; } static int j721e_audio_hw_params(struct snd_pcm_substream *substream, -- 2.30.2

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[PATCH AUTOSEL 5.13 10/21] ASoC: rt5682: Fix the issue of garbled recording after powerd_dbus_suspend
by Sasha Levin 27 Jul '21

27 Jul '21

From: Oder Chiou <oder_chiou(a)realtek.com> [ Upstream commit 6a503e1c455316fd0bfd8188c0a62cce7c5525ca ] While using the DMIC recording, the garbled data will be captured by the DMIC. It is caused by the critical power of PLL closed in the jack detect function. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> Link: https://lore.kernel.org/r/20210716085853.20170-1-oder_chiou@realtek.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5682.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index e4c91571abae..abcd6f483788 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -973,10 +973,14 @@ 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_dapm_get_pin_status(dapm, "PLL1") && + !snd_soc_dapm_get_pin_status(dapm, "PLL2B")) snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, RT5682_PWR_MB, 0); - if (!snd_soc_dapm_get_pin_status(dapm, "Vref2")) + if (!snd_soc_dapm_get_pin_status(dapm, "Vref2") && + !snd_soc_dapm_get_pin_status(dapm, "PLL1") && + !snd_soc_dapm_get_pin_status(dapm, "PLL2B")) snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, RT5682_PWR_VREF2, 0); snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3, -- 2.30.2

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[PATCH AUTOSEL 5.13 04/21] ASoC: tlv320aic31xx: fix reversed bclk/wclk master bits
by Sasha Levin 27 Jul '21

27 Jul '21

From: Kyle Russell <bkylerussell(a)gmail.com> [ Upstream commit 9cf76a72af6ab81030dea6481b1d7bdd814fbdaf ] These are backwards from Table 7-71 of the TLV320AIC3100 spec [1]. This was broken in 12eb4d66ba2e when BCLK_MASTER and WCLK_MASTER were converted from 0x08 and 0x04 to BIT(2) and BIT(3), respectively. -#define AIC31XX_BCLK_MASTER 0x08 -#define AIC31XX_WCLK_MASTER 0x04 +#define AIC31XX_BCLK_MASTER BIT(2) +#define AIC31XX_WCLK_MASTER BIT(3) Probably just a typo since the defines were not listed in bit order. [1] https://www.ti.com/lit/gpn/tlv320aic3100 Signed-off-by: Kyle Russell <bkylerussell(a)gmail.com> Link: https://lore.kernel.org/r/20210622010941.241386-1-bkylerussell@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/tlv320aic31xx.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h index 81952984613d..2513922a0292 100644 --- a/sound/soc/codecs/tlv320aic31xx.h +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -151,8 +151,8 @@ struct aic31xx_pdata { #define AIC31XX_WORD_LEN_24BITS 0x02 #define AIC31XX_WORD_LEN_32BITS 0x03 #define AIC31XX_IFACE1_MASTER_MASK GENMASK(3, 2) -#define AIC31XX_BCLK_MASTER BIT(2) -#define AIC31XX_WCLK_MASTER BIT(3) +#define AIC31XX_BCLK_MASTER BIT(3) +#define AIC31XX_WCLK_MASTER BIT(2) /* AIC31XX_DATA_OFFSET */ #define AIC31XX_DATA_OFFSET_MASK GENMASK(7, 0) -- 2.30.2

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[PATCH 01/54] ASoC: dt-bindings: Add WM8978 Binding
by Maxime Ripard 27 Jul '21

27 Jul '21

Even though we had the wm8978 driver for some time and a number of boards using it already, we never had a binding for it. Let's add it based on what the driver expects and the boards are providing. Cc: alsa-devel(a)alsa-project.org Cc: devicetree(a)vger.kernel.org Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Signed-off-by: Maxime Ripard <maxime(a)cerno.tech> --- .../devicetree/bindings/sound/wlf,wm8978.yaml | 58 +++++++++++++++++++ 1 file changed, 58 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/wlf,wm8978.yaml diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8978.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8978.yaml new file mode 100644 index 000000000000..6761380261e3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8978.yaml @@ -0,0 +1,58 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,wm8978.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Wolfson WM8978 Codec Device Tree Bindings + +maintainers: + - Mark Brown <broonie(a)kernel.org> + +properties: + '#sound-dai-cells': + const: 0 + + compatible: + const: wlf,wm8978 + + reg: + maxItems: 1 + + spi-max-frequency: + maximum: 526000 + +required: + - '#sound-dai-cells' + - compatible + - reg + +additionalProperties: false + +examples: + - | + spi { + #address-cells = <1>; + #size-cells = <0>; + + codec@0 { + #sound-dai-cells = <0>; + compatible = "wlf,wm8978"; + reg = <0>; + spi-max-frequency = <500000>; + }; + }; + + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@0 { + #sound-dai-cells = <0>; + compatible = "wlf,wm8978"; + reg = <0>; + }; + }; + +... -- 2.31.1

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Pulseaudio craches on start with SegFault error
by GitHub issues - opened 27 Jul '21

27 Jul '21

alsa-project/alsa-plugins issue #31 was opened from jmtrivial: Dear alsa-project developers, I am using a debian (testing) on my computer, together with deb-multimedia.org repository. As some other users with external USB audio devices, I recently discovered a segfault: https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=991337 I discussed withe Christian Marillat (maintainer of the deb-multimedia repository) and we tried a couple of alsa versions, but the bug is still present if I'm using a recent libasound2-plugins such as this one: https://www.deb-multimedia.org/tests/libasound2-plugins_1.2.2+git4219f4b-dm… It does not crash with libasound2-plugins 1.2.2-2 from debian testing. You will found attached the log file from pulseaudio -vvv. [pulseaudio.log](https://github.com/alsa-project/alsa-plugins/files/6884875/… Issue URL : https://github.com/alsa-project/alsa-plugins/issues/31 Repository URL: https://github.com/alsa-project/alsa-plugins

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[PATCH] ALSA: seq: Fix comments of wrong client number for MIDI Passthrough
by Takashi Iwai 27 Jul '21

27 Jul '21

MIDI Passthrough sequencer client is assigned always to the fixed number 14, while it's wrongly documented in the comments as if 62, which was an old number that was used during development. Fix all those numbers. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/seq_dummy.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/core/seq/seq_dummy.c b/sound/core/seq/seq_dummy.c index ac760b1e3d12..8c18d8c4177e 100644 --- a/sound/core/seq/seq_dummy.c +++ b/sound/core/seq/seq_dummy.c @@ -20,15 +20,15 @@ are redirected to output port immediately. The routing can be done via aconnect program in alsa-utils. - Each client has a static client number 62 (= SNDRV_SEQ_CLIENT_DUMMY). + Each client has a static client number 14 (= SNDRV_SEQ_CLIENT_DUMMY). If you want to auto-load this module, you may add the following alias in your /etc/conf.modules file. - alias snd-seq-client-62 snd-seq-dummy + alias snd-seq-client-14 snd-seq-dummy - The module is loaded on demand for client 62, or /proc/asound/seq/ + The module is loaded on demand for client 14, or /proc/asound/seq/ is accessed. If you don't need this module to be loaded, alias - snd-seq-client-62 as "off". This will help modprobe. + snd-seq-client-14 as "off". This will help modprobe. The number of ports to be created can be specified via the module parameter "ports". For example, to create four ports, add the -- 2.26.2

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[PATCH] ALSA: usb-audio: Add registration quirk for JBL Quantum 600
by Alexander Tsoy 27 Jul '21

27 Jul '21

Apparently JBL Quantum 600 has multiple hardware revisions. Apply registration quirk to another device id as well. Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index e7accd87e063..326d1b0ea5e6 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1899,6 +1899,7 @@ static const struct registration_quirk registration_quirks[] = { REG_QUIRK_ENTRY(0x0951, 0x16ea, 2), /* Kingston HyperX Cloud Flight S */ REG_QUIRK_ENTRY(0x0ecb, 0x1f46, 2), /* JBL Quantum 600 */ REG_QUIRK_ENTRY(0x0ecb, 0x2039, 2), /* JBL Quantum 400 */ + REG_QUIRK_ENTRY(0x0ecb, 0x203c, 2), /* JBL Quantum 600 */ REG_QUIRK_ENTRY(0x0ecb, 0x203e, 2), /* JBL Quantum 800 */ { 0 } /* terminator */ }; -- 2.31.1

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MIDI Through Port Number in ALSA Sequencer
by rv71＠onet.eu 27 Jul '21

27 Jul '21

Hello, ALSA Sequencer always creates MIDI Through Port with the id 14:0. However, I can't find where it is defined that this special port will have this particular id/number. I also don't see any definition in seq.h regarding this ( unlike the timer and announce special port, for example). Can you explain? Thanks Robert

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