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

[PATCH] ASoC: dt-bindings: q6asm: Add Q6ASM_DAI_{TX_RX, TX, RX} defines
by Stephan Gerhold 27 Jul '20

27 Jul '20

Right now the direction of a DAI has to be specified as a literal number in the device tree, e.g.: dai@0 { reg = <0>; direction = <2>; }; but this does not make it immediately clear that this is a playback/RX-only DAI. Actually, q6asm-dai.c has useful defines for this. Move them to the dt-bindings header to allow using them in the dts(i) files. The example above then becomes: dai@0 { reg = <0>; direction = <Q6ASM_DAI_RX>; }; which is immediately recognizable as playback/RX-only DAI. Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- Documentation/devicetree/bindings/sound/qcom,q6asm.txt | 9 +++++---- include/dt-bindings/sound/qcom,q6asm.h | 4 ++++ sound/soc/qcom/qdsp6/q6asm-dai.c | 3 --- 3 files changed, 9 insertions(+), 7 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,q6asm.txt b/Documentation/devicetree/bindings/sound/qcom,q6asm.txt index 6b9a88d0ea3f..8c4883becae9 100644 --- a/Documentation/devicetree/bindings/sound/qcom,q6asm.txt +++ b/Documentation/devicetree/bindings/sound/qcom,q6asm.txt @@ -39,9 +39,9 @@ configuration of each dai. Must contain the following properties. Usage: Required for Compress offload dais Value type: <u32> Definition: Specifies the direction of the dai stream - 0 for both tx and rx - 1 for only tx (Capture/Encode) - 2 for only rx (Playback/Decode) + Q6ASM_DAI_TX_RX (0) for both tx and rx + Q6ASM_DAI_TX (1) for only tx (Capture/Encode) + Q6ASM_DAI_RX (2) for only rx (Playback/Decode) - is-compress-dai: Usage: Required for Compress offload dais @@ -50,6 +50,7 @@ configuration of each dai. Must contain the following properties. = EXAMPLE +#include <dt-bindings/sound/qcom,q6asm.h> apr-service@7 { compatible = "qcom,q6asm"; @@ -62,7 +63,7 @@ apr-service@7 { dai@0 { reg = <0>; - direction = <2>; + direction = <Q6ASM_DAI_RX>; is-compress-dai; }; }; diff --git a/include/dt-bindings/sound/qcom,q6asm.h b/include/dt-bindings/sound/qcom,q6asm.h index 1eb77d87c2e8..f59d74f14395 100644 --- a/include/dt-bindings/sound/qcom,q6asm.h +++ b/include/dt-bindings/sound/qcom,q6asm.h @@ -19,4 +19,8 @@ #define MSM_FRONTEND_DAI_MULTIMEDIA15 14 #define MSM_FRONTEND_DAI_MULTIMEDIA16 15 +#define Q6ASM_DAI_TX_RX 0 +#define Q6ASM_DAI_TX 1 +#define Q6ASM_DAI_RX 2 + #endif /* __DT_BINDINGS_Q6_ASM_H__ */ diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index a2acb7564eb8..9b7b218f2a20 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -37,9 +37,6 @@ #define COMPR_PLAYBACK_MAX_FRAGMENT_SIZE (128 * 1024) #define COMPR_PLAYBACK_MIN_NUM_FRAGMENTS (4) #define COMPR_PLAYBACK_MAX_NUM_FRAGMENTS (16 * 4) -#define Q6ASM_DAI_TX_RX 0 -#define Q6ASM_DAI_TX 1 -#define Q6ASM_DAI_RX 2 #define ALAC_CH_LAYOUT_MONO ((101 << 16) | 1) #define ALAC_CH_LAYOUT_STEREO ((101 << 16) | 2) -- 2.27.0

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[PATCH] ASoC: AMD: Restore PME_EN state at Power On
by Akshu Agrawal 27 Jul '20

27 Jul '20

PME_EN state needs to restored to the value set by fmw. For the devices which are not using I2S wake event which gets enabled by PME_EN bit, keeping PME_EN enabled burns considerable amount of power as it blocks low power state. For the devices using I2S wake event, PME_EN gets enabled in fmw and the state should be maintained after ACP Power On. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/raven/pci-acp3x.c | 21 +++++++++++++-------- 1 file changed, 13 insertions(+), 8 deletions(-) diff --git a/sound/soc/amd/raven/pci-acp3x.c b/sound/soc/amd/raven/pci-acp3x.c index ebf4388b6262..31b797c8bfe6 100644 --- a/sound/soc/amd/raven/pci-acp3x.c +++ b/sound/soc/amd/raven/pci-acp3x.c @@ -19,10 +19,12 @@ struct acp3x_dev_data { bool acp3x_audio_mode; struct resource *res; struct platform_device *pdev[ACP3x_DEVS]; + u32 pme_en; }; -static int acp3x_power_on(void __iomem *acp3x_base) +static int acp3x_power_on(struct acp3x_dev_data *adata) { + void __iomem *acp3x_base = adata->acp3x_base; u32 val; int timeout; @@ -39,10 +41,10 @@ static int acp3x_power_on(void __iomem *acp3x_base) while (++timeout < 500) { val = rv_readl(acp3x_base + mmACP_PGFSM_STATUS); if (!val) { - /* Set PME_EN as after ACP power On, - * PME_EN gets cleared + /* ACP power On clears PME_EN. + * Restore the value to its prior state */ - rv_writel(0x1, acp3x_base + mmACP_PME_EN); + rv_writel(adata->pme_en, acp3x_base + mmACP_PME_EN); return 0; } udelay(1); @@ -74,12 +76,13 @@ static int acp3x_reset(void __iomem *acp3x_base) return -ETIMEDOUT; } -static int acp3x_init(void __iomem *acp3x_base) +static int acp3x_init(struct acp3x_dev_data *adata) { + void __iomem *acp3x_base = adata->acp3x_base; int ret; /* power on */ - ret = acp3x_power_on(acp3x_base); + ret = acp3x_power_on(adata); if (ret) { pr_err("ACP3x power on failed\n"); return ret; @@ -151,7 +154,9 @@ static int snd_acp3x_probe(struct pci_dev *pci, } pci_set_master(pci); pci_set_drvdata(pci, adata); - ret = acp3x_init(adata->acp3x_base); + /* Save ACP_PME_EN state */ + adata->pme_en = rv_readl(adata->acp3x_base + mmACP_PME_EN); + ret = acp3x_init(adata); if (ret) goto disable_msi; @@ -274,7 +279,7 @@ static int snd_acp3x_resume(struct device *dev) struct acp3x_dev_data *adata; adata = dev_get_drvdata(dev); - ret = acp3x_init(adata->acp3x_base); + ret = acp3x_init(adata); if (ret) { dev_err(dev, "ACP init failed\n"); return ret; -- 2.20.1

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[PATCH] ALSA: hda/realtek: Fix add a "ultra_low_power" function for intel reference board (alc256)
by Takashi Iwai 27 Jul '20

27 Jul '20

From: PeiSen Hou <pshou(a)realtek.com> Intel requires to enable power saving mode for intel reference board (alc256) Signed-off-by: PeiSen Hou <pshou(a)realtek.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3f2512942daf..0f640d99a396 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7590,7 +7590,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x10cf, 0x1629, "Lifebook U7x7", ALC255_FIXUP_LIFEBOOK_U7x7_HEADSET_MIC), SND_PCI_QUIRK(0x10cf, 0x1845, "Lifebook U904", ALC269_FIXUP_LIFEBOOK_EXTMIC), SND_PCI_QUIRK(0x10ec, 0x10f2, "Intel Reference board", ALC700_FIXUP_INTEL_REFERENCE), - SND_PCI_QUIRK(0x10ec, 0x1230, "Intel Reference board", ALC225_FIXUP_HEADSET_JACK), + SND_PCI_QUIRK(0x10ec, 0x1230, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), 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), -- 2.16.4

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[PATCH] alsaucm: Fix ending with quotes commands
by Cheng Yueh 27 Jul '20

27 Jul '20

If we use 'alsaucm -n -b -' with 'get "CapturePCM/Internal Mic"' then the alsaucm will report error and stop immediately. The reason is that the parse_line in usecase.c appends an empty argument if a command ends with quotes. This change adds a patch to fix the parse_line function in usecase.c. Signed-off-by: Cheng Yueh <cyueh(a)chromium.org> --- alsaucm/usecase.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/alsaucm/usecase.c b/alsaucm/usecase.c index 1ed91ea..d39a159 100644 --- a/alsaucm/usecase.c +++ b/alsaucm/usecase.c @@ -130,6 +130,8 @@ static int parse_line(struct context *context, char *line) *line == '\n')) line++; c = *line; + if (c == '\0') + return 0; if (c == '\"' || c == '\'') { start = ++line; while (*line && *line != c) -- 2.26.2

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snd_soc_component_driver substream ops
by Daniel Baluta 27 Jul '20

27 Jul '20

Hi Morimoto-san, Looking at snd_soc_component_driver I see there are some operations like: open, close, hw_params, hw_free. (1) Now, snd_soc_component_driver has snd_compress_ops. Do you think it is worth it to group operations from (1) in a similar structure maybe snd_<xyz>_ops. Daniel.

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Acer TravelMate 5735Z (Realtek ALC272X): Periodic crackling noises
by Paul Menzel 27 Jul '20

27 Jul '20

Dear Linux folks, On Acer TravelMate 5735Z with Debian Sid/unstable and Linux 5.7.6, there are recurring crackling noises on the internal speaker. I ran `alsa-info` from the Debian package, and the logs are uploaded [1]. It’d be awesome, if you took a look, and told me, what I can try to solve this. Kind regards, Paul [1]: https://alsa-project.org/db/?f=67bde144f2e2a4f06da89a8cc8abf93da8c91d5b

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No sound with Realtek ALC287
by GitHub issues - opened 27 Jul '20

27 Jul '20

alsa-project/alsa-lib issue #76 was opened from BLuFeNiX: I get no sound via the speakers of my Lenovo Legion 7i laptop, which alsamixer tells me is using a Realtek ALC287. Here is my alsa-info.sh output: http://alsa-project.org/db/?f=4272343a3590cc08f192f98113dedfc0418afe52 I made sure to disable Auto-Mute in alsamixer, and turn all volume levels to maximum. In all cases, I get no sounds from the speakers (running speaker-test, playing music, etc.). I *am* able to get sound via headphones and HDMI (though I believe HDMI is via a different sound card). I previously reported this here: https://bugzilla.kernel.org/show_bug.cgi?id=208555 but I received no response in almost 3 weeks, so I think maybe I have reported it to the wrong place. Please let me know if I can provide any assistance or additional information. Issue URL : https://github.com/alsa-project/alsa-lib/issues/76 Repository URL: https://github.com/alsa-project/alsa-lib

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[RESEND 3/3] ASoC: max98390: update dsm param bin max size
by Steve Lee 27 Jul '20

27 Jul '20

MAX98390_DSM_PARAM_MAX_SIZE is changed to support extended register update. Signed-off-by: Steve Lee <steves.lee(a)maximintegrated.com> --- sound/soc/codecs/max98390.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98390.h b/sound/soc/codecs/max98390.h index 5f444e7779b0..dff884f68e3e 100644 --- a/sound/soc/codecs/max98390.h +++ b/sound/soc/codecs/max98390.h @@ -650,7 +650,7 @@ /* DSM register offset */ #define MAX98390_DSM_PAYLOAD_OFFSET 16 -#define MAX98390_DSM_PARAM_MAX_SIZE 770 +#define MAX98390_DSM_PARAM_MAX_SIZE 1024 #define MAX98390_DSM_PARAM_MIN_SIZE 670 struct max98390_priv { -- 2.17.1

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[RESEND 2/3] ASoC: max98390: Update dsm init sequence and condition.
by Steve Lee 27 Jul '20

27 Jul '20

Modify dsm_init sequence and dsm param bin check condition. - Move dsm_init() to after amp init setting to make sure dsm init is last setting. - dsm param bin check condition changed for extended register setting. 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 44ffebac15ad..ff5cc9bbec29 100644 --- a/sound/soc/codecs/max98390.c +++ b/sound/soc/codecs/max98390.c @@ -790,7 +790,7 @@ static int max98390_dsm_init(struct snd_soc_component *component) 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 || + param_start_addr < MAX98390_IRQ_CTRL || fw->size < param_size + MAX98390_DSM_PAYLOAD_OFFSET) { dev_err(component->dev, "param fw is invalid.\n"); @@ -864,11 +864,11 @@ static int max98390_probe(struct snd_soc_component *component) regmap_write(max98390->regmap, MAX98390_SOFTWARE_RESET, 0x01); /* Sleep reset settle time */ msleep(20); - /* Update dsm bin param */ - max98390_dsm_init(component); /* Amp init setting */ max98390_init_regs(component); + /* Update dsm bin param */ + max98390_dsm_init(component); /* Dsm Setting */ if (max98390->ref_rdc_value) { -- 2.17.1

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[RESEND 1/3] ASoC: max98390: Fix dac event dapm mixer.
by Steve Lee 27 Jul '20

27 Jul '20

Global EN register guide to off before AMP_EN register when amp disable sequence. - remove AMP_EN control before max98390_dac_event call Signed-off-by: Steve Lee <steves.lee(a)maximintegrated.com> --- 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 3e8094241645..44ffebac15ad 100644 --- a/sound/soc/codecs/max98390.c +++ b/sound/soc/codecs/max98390.c @@ -678,7 +678,7 @@ static const struct snd_kcontrol_new max98390_dai_controls = 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_NOPM, 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), -- 2.17.1

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