August 2024 - Alsa-devel - mailman.alsa-project.org

[PATCH] ASoC: dt-bindings: qcom,lpass-wsa-macro: correct clocks on SM8250
by srinivas.kandagatla＠linaro.org 16 Aug '24

16 Aug '24

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> we seems to have ended up with duplicate clocks for frame-sync on sm8250, it has both va and fsgen which are exactly same things. Remove the redundant va clock and make it align with other SoCs. Codec driver does not even handle va clock, so remove this from the bindings and examples to avoid any confusion. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- .../bindings/sound/qcom,lpass-wsa-macro.yaml | 22 ++----------------- 1 file changed, 2 insertions(+), 20 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml index 06b5f7be3608..6f5644a89feb 100644 --- a/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml @@ -64,6 +64,7 @@ allOf: compatible: enum: - qcom,sc7280-lpass-wsa-macro + - qcom,sm8250-lpass-wsa-macro - qcom,sm8450-lpass-wsa-macro - qcom,sc8280xp-lpass-wsa-macro then: @@ -79,24 +80,6 @@ allOf: - const: dcodec - const: fsgen - - if: - properties: - compatible: - enum: - - qcom,sm8250-lpass-wsa-macro - then: - properties: - clocks: - minItems: 6 - clock-names: - items: - - const: mclk - - const: npl - - const: macro - - const: dcodec - - const: va - - const: fsgen - - if: properties: compatible: @@ -130,8 +113,7 @@ examples: <&audiocc 0>, <&q6afecc LPASS_HW_MACRO_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>, <&q6afecc LPASS_HW_DCODEC_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>, - <&aoncc LPASS_CDC_VA_MCLK>, <&vamacro>; - clock-names = "mclk", "npl", "macro", "dcodec", "va", "fsgen"; + clock-names = "mclk", "npl", "macro", "dcodec", "fsgen"; clock-output-names = "mclk"; }; -- 2.25.1

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[PATCH v1] ASoc: tas2781: Rename dai_driver name to unify the name between TAS2563 and TAS2781
by Shenghao Ding 16 Aug '24

16 Aug '24

Rename dai_driver name to unify the name between TAS2563 and TAS2781, remove unnecessary line feed, strscpy replace scnprintf. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - Use strscpy to replace scnprintf. - rename dai_driver name from "tas2781_codec" to "tasdev_codec", in case of misunderstanding whether tas2781 or tas2563. - Remove unnecessary line feed for tasdevice_dsp_create_ctrls --- sound/soc/codecs/tas2781-i2c.c | 14 ++++++-------- 1 file changed, 6 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 8a57c045afdd..6631254d4985 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -353,7 +353,7 @@ static int tasdevice_create_control(struct tasdevice_priv *tas_priv) ret = -ENOMEM; goto out; } - scnprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "Speaker Profile Id"); + strscpy(name, "Speaker Profile Id", SNDRV_CTL_ELEM_ID_NAME_MAXLEN); prof_ctrls[mix_index].name = name; prof_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; prof_ctrls[mix_index].info = tasdevice_info_profile; @@ -424,8 +424,7 @@ static int tasdevice_configuration_put( return ret; } -static int tasdevice_dsp_create_ctrls( - struct tasdevice_priv *tas_priv) +static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) { struct snd_kcontrol_new *dsp_ctrls; char *prog_name, *conf_name; @@ -453,8 +452,8 @@ static int tasdevice_dsp_create_ctrls( goto out; } - scnprintf(prog_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, - "Speaker Program Id"); + strscpy(prog_name, "Speaker Program Id", + SNDRV_CTL_ELEM_ID_NAME_MAXLEN); dsp_ctrls[mix_index].name = prog_name; dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; dsp_ctrls[mix_index].info = tasdevice_info_programs; @@ -462,8 +461,7 @@ static int tasdevice_dsp_create_ctrls( dsp_ctrls[mix_index].put = tasdevice_program_put; mix_index++; - scnprintf(conf_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, - "Speaker Config Id"); + strscpy(conf_name, "Speaker Config Id", SNDRV_CTL_ELEM_ID_NAME_MAXLEN); dsp_ctrls[mix_index].name = conf_name; dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; dsp_ctrls[mix_index].info = tasdevice_info_configurations; @@ -673,7 +671,7 @@ static const struct snd_soc_dai_ops tasdevice_dai_ops = { static struct snd_soc_dai_driver tasdevice_dai_driver[] = { { - .name = "tas2781_codec", + .name = "tasdev_codec", .id = 0, .playback = { .stream_name = "Playback", -- 2.34.1

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[PATCH] ASoC: SOF: amd: Fix for acp init sequence
by Vijendar Mukunda 16 Aug '24

16 Aug '24

When ACP is not powered on by default, acp power on sequence explicitly invoked by programming pgfsm control mask. The existing implementation checks the same PGFSM status mask and programs the same PGFSM control mask in all ACP variants which breaks acp power on sequence for ACP6.0 and ACP6.3 variants. So to fix this issue, update ACP pgfsm control mask and status mask based on acp descriptor rev field, which will vary based on acp variant. Fixes: 846aef1d7cc0 ("ASoC: SOF: amd: Add Renoir ACP HW support") Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/sof/amd/acp.c | 19 +++++++++++++++++-- sound/soc/sof/amd/acp.h | 7 +++++-- 2 files changed, 22 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index 35e56d22930f..0f6115c8b005 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -433,6 +433,7 @@ static int acp_power_on(struct snd_sof_dev *sdev) const struct sof_amd_acp_desc *desc = get_chip_info(sdev->pdata); unsigned int base = desc->pgfsm_base; unsigned int val; + unsigned int acp_pgfsm_status_mask, acp_pgfsm_cntl_mask; int ret; val = snd_sof_dsp_read(sdev, ACP_DSP_BAR, base + PGFSM_STATUS_OFFSET); @@ -440,9 +441,23 @@ static int acp_power_on(struct snd_sof_dev *sdev) if (val == ACP_POWERED_ON) return 0; - if (val & ACP_PGFSM_STATUS_MASK) + switch (desc->rev) { + case 3: + case 5: + acp_pgfsm_status_mask = ACP3X_PGFSM_STATUS_MASK; + acp_pgfsm_cntl_mask = ACP3X_PGFSM_CNTL_POWER_ON_MASK; + break; + case 6: + acp_pgfsm_status_mask = ACP6X_PGFSM_STATUS_MASK; + acp_pgfsm_cntl_mask = ACP6X_PGFSM_CNTL_POWER_ON_MASK; + break; + default: + return -EINVAL; + } + + if (val & acp_pgfsm_status_mask) snd_sof_dsp_write(sdev, ACP_DSP_BAR, base + PGFSM_CONTROL_OFFSET, - ACP_PGFSM_CNTL_POWER_ON_MASK); + acp_pgfsm_cntl_mask); ret = snd_sof_dsp_read_poll_timeout(sdev, ACP_DSP_BAR, base + PGFSM_STATUS_OFFSET, val, !val, ACP_REG_POLL_INTERVAL, ACP_REG_POLL_TIMEOUT_US); diff --git a/sound/soc/sof/amd/acp.h b/sound/soc/sof/amd/acp.h index f6f0fcfeb691..321c40cc6d50 100644 --- a/sound/soc/sof/amd/acp.h +++ b/sound/soc/sof/amd/acp.h @@ -25,8 +25,11 @@ #define ACP_REG_POLL_TIMEOUT_US 2000 #define ACP_DMA_COMPLETE_TIMEOUT_US 5000 -#define ACP_PGFSM_CNTL_POWER_ON_MASK 0x01 -#define ACP_PGFSM_STATUS_MASK 0x03 +#define ACP3X_PGFSM_CNTL_POWER_ON_MASK 0x01 +#define ACP3X_PGFSM_STATUS_MASK 0x03 +#define ACP6X_PGFSM_CNTL_POWER_ON_MASK 0x07 +#define ACP6X_PGFSM_STATUS_MASK 0x0F + #define ACP_POWERED_ON 0x00 #define ACP_ASSERT_RESET 0x01 #define ACP_RELEASE_RESET 0x00 -- 2.34.1

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Use PyBytes and not Unicode strings for IEC958 values
by GitHub pull_request - edited 16 Aug '24

16 Aug '24

alsa-project/alsa-python pull request #14 was edited from tormodvolden: Suggested fix for #13. - the first commit is not strictly related but also needed to avoid hctltest1.py crashing with my audio devices, where some have info.count = 0. Let me know if more information is needed. I don't know if this is expected or if a warning should be issued. - I went for PyBytes instead of PyByteArray - for setting values, if the provided PyBytes value is shorter than the value, it will be zero-padded. - if it is too long, it will be silently truncated - following the existing code style, no empty lines were inserted, but this could help readability Request URL : https://github.com/alsa-project/alsa-python/pull/14 Patch URL : https://github.com/alsa-project/alsa-python/pull/14.patch Repository URL: https://github.com/alsa-project/alsa-python

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Use PyBytes and not Unicode strings for IEC958 values
by GitHub pull_request - opened 16 Aug '24

16 Aug '24

alsa-project/alsa-python pull request #14 was opened from tormodvolden: Suggested fix for #13. - the first commit is not strictly related but also needed to avoid hctltest1.py crashing with my audio devices, where some have info.count = 0. Let me know if more information is needed. - I went for PyBytes instead of PyByteArray - for setting values, if the provided PyBytes value is shorter than the value, it will be zero-padded. - if it is too long, it will be silently truncated - following the existing code style, no empty lines were inserted, but this could help readability Request URL : https://github.com/alsa-project/alsa-python/pull/14 Patch URL : https://github.com/alsa-project/alsa-python/pull/14.patch Repository URL: https://github.com/alsa-project/alsa-python

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[PATCH] ASoC: codecs: lpass-va-macro: set the default codec version for sm8250
by srinivas.kandagatla＠linaro.org 16 Aug '24

16 Aug '24

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> sm8250 and sc7280 have lpass codec version 1.0, as these are very old platforms, they do not have a reliable way to get the codec version from core_id registers. Add the version info into of_data, so that it does not need to use core_id registers to get version number. Fixes: 378918d59181 ("ASoC: codecs: lpass-macro: add helpers to get codec version") Fixes: dbacef05898d ("ASoC: codec: lpass-rx-macro: prepare driver to accomdate new codec versions") Fixes: 727de4fbc546 ("ASoC: codecs: lpass-wsa-macro: Correct support for newer v2.5 version") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/codecs/lpass-va-macro.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c index 8454193ed22a..e95d1f29ef18 100644 --- a/sound/soc/codecs/lpass-va-macro.c +++ b/sound/soc/codecs/lpass-va-macro.c @@ -228,11 +228,13 @@ struct va_macro { struct va_macro_data { bool has_swr_master; bool has_npl_clk; + int version; }; static const struct va_macro_data sm8250_va_data = { .has_swr_master = false, .has_npl_clk = false, + .version = LPASS_CODEC_VERSION_1_0, }; static const struct va_macro_data sm8450_va_data = { @@ -1587,7 +1589,14 @@ static int va_macro_probe(struct platform_device *pdev) goto err_npl; } - va_macro_set_lpass_codec_version(va); + /** + * old version of codecs do not have a reliable way to determine the + * version from registers, get them from soc specific data + */ + if (data->version) + lpass_macro_set_codec_version(data->version); + else /* read version from register */ + va_macro_set_lpass_codec_version(va); if (va->has_swr_master) { /* Set default CLK div to 1 */ -- 2.25.1

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[PATCH v1] ASoC: codecs: ES8326: Adjust buttons
by zhangyi＠everest-semi.com 16 Aug '24

16 Aug '24

The content of this message was lost. It was probably cross-posted to multiple lists and previously handled on another list.

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[PATCH] ALSA: aoa: Use helper function for_each_child_of_node()
by Zhang Zekun 16 Aug '24

16 Aug '24

for_each_child_of_node can help to iterate through the device_node, and we don't need to use while loop. No functional change with this conversion. Signed-off-by: Zhang Zekun <zhangzekun11(a)huawei.com> --- sound/aoa/soundbus/i2sbus/core.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index 5431d2c49421..ce84288168e4 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -335,7 +335,7 @@ static int i2sbus_add_dev(struct macio_dev *macio, static int i2sbus_probe(struct macio_dev* dev, const struct of_device_id *match) { - struct device_node *np = NULL; + struct device_node *np; int got = 0, err; struct i2sbus_control *control = NULL; @@ -347,7 +347,7 @@ static int i2sbus_probe(struct macio_dev* dev, const struct of_device_id *match) return -ENODEV; } - while ((np = of_get_next_child(dev->ofdev.dev.of_node, np))) { + for_each_child_of_node(dev->ofdev.dev.of_node, np) { if (of_device_is_compatible(np, "i2sbus") || of_device_is_compatible(np, "i2s-modem")) { got += i2sbus_add_dev(dev, control, np); -- 2.17.1

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[PATCH v2 0/6] ASoC: codecs: wsa88xx: add support for static port mapping.
by srinivas.kandagatla＠linaro.org 15 Aug '24

15 Aug '24

Existing way of allocating soundwire master ports on Qualcommm platforms is dynamic, and in linear order starting from 1 to MAX_PORTS. This will work as long as soundwire device ports are 1:1 mapped linearly. However on most Qcom SoCs like SM8550, SM8650, x1e80100, these are NOT mapped in that order. The result of this is that only one speaker among the pair of speakers is always silent, With recent changes for WSA codec to support codec versions and along with these patches we are able to get all speakers working on these SoCs. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- Changes in v2: - used dev_dbg instead of dev_info - Link to v1: https://lore.kernel.org/r/20240626-port-map-v1-0-bd8987d2b332@linaro.org --- Manikantan R (1): ASoC: dt-bindings: wsa883x: Document port mapping property Srinivas Kandagatla (5): ASoC: codecs: wsa883x: parse port-mapping information ASoC: dt-bindings: wsa8840: Document port mapping property ASoC: codecs: wsa884x: parse port-mapping information arm64: dts: x1e80100-crd: fix wsa soundwire port mapping arm64: dts: x1e80100-qcp: fix wsa soundwire port mapping Documentation/devicetree/bindings/sound/qcom,wsa883x.yaml | 8 ++++++++ Documentation/devicetree/bindings/sound/qcom,wsa8840.yaml | 8 ++++++++ arch/arm64/boot/dts/qcom/x1e80100-crd.dts | 4 ++++ arch/arm64/boot/dts/qcom/x1e80100-qcp.dts | 2 ++ sound/soc/codecs/wsa883x.c | 8 ++++++++ sound/soc/codecs/wsa884x.c | 8 ++++++++ 6 files changed, 38 insertions(+) --- base-commit: 9935be184a55dd84fc3275094f2df095491f6ea1 change-id: 20240626-port-map-ef50c3304d4a Best regards, -- Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org>

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[PATCH 0/6] soc: qcom: use scoped device node handling to simplify error paths
by Krzysztof Kozlowski 15 Aug '24

15 Aug '24

Use cleanup.h/scope on device nodes to get rid of of_node_put(). Code is a bit simpler. Best regards, Krzysztof --- Krzysztof Kozlowski (6): soc: qcom: apr: simplify with scoped for each OF child loop soc: qcom: aoss: simplify with scoped for each OF child loop soc: qcom: ice: use scoped device node handling to simplify error paths soc: qcom: ocmem: use scoped device node handling to simplify error paths soc: qcom: pbs: use scoped device node handling to simplify error paths soc: qcom: smp2p: use scoped device node handling to simplify error paths drivers/soc/qcom/apr.c | 5 +---- drivers/soc/qcom/ice.c | 14 +++++--------- drivers/soc/qcom/ocmem.c | 7 ++----- drivers/soc/qcom/qcom-pbs.c | 16 ++++++---------- drivers/soc/qcom/qcom_aoss.c | 8 +++----- drivers/soc/qcom/smp2p.c | 16 ++++------------ 6 files changed, 21 insertions(+), 45 deletions(-) --- base-commit: d54f33dafd9e93367ffaf95318ed371138a6c8c0 change-id: 20240813-b4-cleanup-h-of-node-put-other-7e9c7128abe4 Best regards, -- Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>

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