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[PATCH] ASoC: twl4030: make read-only array ramp_base static const
by Colin Ian King 19 Oct '22

19 Oct '22

Don't populate the read-only array ramp_base on the stack but instead make it static const. Also makes the object code a little smaller. Signed-off-by: Colin Ian King <colin.i.king(a)gmail.com> --- sound/soc/codecs/twl4030.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index e48768233e20..9c50ac356c89 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -700,8 +700,10 @@ static void headset_ramp(struct snd_soc_component *component, int ramp) struct twl4030_priv *twl4030 = snd_soc_component_get_drvdata(component); struct twl4030_board_params *board_params = twl4030->board_params; /* Base values for ramp delay calculation: 2^19 - 2^26 */ - unsigned int ramp_base[] = {524288, 1048576, 2097152, 4194304, - 8388608, 16777216, 33554432, 67108864}; + static const unsigned int ramp_base[] = { + 524288, 1048576, 2097152, 4194304, + 8388608, 16777216, 33554432, 67108864 + }; unsigned int delay; hs_gain = twl4030_read(component, TWL4030_REG_HS_GAIN_SET); -- 2.37.3

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[PATCH 0/3] ASoC: Intel/SOF: simplify S3 resume flows
by Pierre-Louis Bossart 19 Oct '22

19 Oct '22

All Intel drivers for cAVS platforms contain a sequence for S3 resume which doesn't seem justified nor necessary. Forensic Git investigation in internal repositories did not provide any rationale for the implementation, and tests show no impact when those sequences are removed. This sequence was identified as problematic during a large HDaudio cleanup where all programming sequences were revisited before extensions are added. Pierre-Louis Bossart (3): ASoC: Intel: Skylake: simplify S3 resume flows ASoC: Intel: avs: simplify S3 resume flows ASoC: SOF: Intel: hda-dsp: simplify S3 resume flows sound/soc/intel/avs/core.c | 11 ----------- sound/soc/intel/skylake/skl.c | 9 --------- sound/soc/sof/intel/hda-dsp.c | 14 -------------- 3 files changed, 34 deletions(-) -- 2.34.1

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[PATCH] ASoC: SOF: ipc3: Log the tx message before sending it
by Peter Ujfalusi 19 Oct '22

19 Oct '22

It makes more sense to log the message before it is sent to the DSP. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-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/sof/ipc3.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/ipc3.c b/sound/soc/sof/ipc3.c index b28af3a48b70..1fef4dcc0936 100644 --- a/sound/soc/sof/ipc3.c +++ b/sound/soc/sof/ipc3.c @@ -329,6 +329,8 @@ static int ipc3_tx_msg_unlocked(struct snd_sof_ipc *ipc, struct snd_sof_dev *sdev = ipc->sdev; int ret; + ipc3_log_header(sdev->dev, "ipc tx", hdr->cmd); + ret = sof_ipc_send_msg(sdev, msg_data, msg_bytes, reply_bytes); if (ret) { @@ -338,8 +340,6 @@ static int ipc3_tx_msg_unlocked(struct snd_sof_ipc *ipc, return ret; } - ipc3_log_header(sdev->dev, "ipc tx", hdr->cmd); - /* now wait for completion */ return ipc3_wait_tx_done(ipc, reply_data); } -- 2.38.0

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[PATCH][next][V2] ASoC: codecs: jz4725b: Fix spelling mistake "Sourc" -> "Source", "Routee" -> "Route"
by Colin Ian King 19 Oct '22

19 Oct '22

There are two spelling mistakes in codec routing description. Fix it. Signed-off-by: Colin Ian King <colin.i.king(a)gmail.com> --- V2: Fix "Routee" -> "Route" too --- sound/soc/codecs/jz4725b.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/jz4725b.c b/sound/soc/codecs/jz4725b.c index 685ba1d3a644..64b14b1c74b9 100644 --- a/sound/soc/codecs/jz4725b.c +++ b/sound/soc/codecs/jz4725b.c @@ -359,7 +359,7 @@ static const struct snd_soc_dapm_route jz4725b_codec_dapm_routes[] = { {"Mixer to ADC", NULL, "Mixer"}, {"ADC Source Capture Route", "Mixer", "Mixer to ADC"}, - {"ADC Sourc Capture Routee", "Line In", "Line In"}, + {"ADC Source Capture Route", "Line In", "Line In"}, {"ADC Source Capture Route", "Mic 1", "Mic 1"}, {"ADC Source Capture Route", "Mic 2", "Mic 2"}, {"ADC", NULL, "ADC Source Capture Route"}, -- 2.37.3

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[PATCH][next] ASoC: codecs: jz4725b: Fix spelling mistake "Sourc" -> "Source"
by Colin Ian King 19 Oct '22

19 Oct '22

There is a spelling mistake in codec routing description. Fix it. Signed-off-by: Colin Ian King <colin.i.king(a)gmail.com> --- sound/soc/codecs/jz4725b.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/jz4725b.c b/sound/soc/codecs/jz4725b.c index 685ba1d3a644..64b14b1c74b9 100644 --- a/sound/soc/codecs/jz4725b.c +++ b/sound/soc/codecs/jz4725b.c @@ -359,7 +359,7 @@ static const struct snd_soc_dapm_route jz4725b_codec_dapm_routes[] = { {"Mixer to ADC", NULL, "Mixer"}, {"ADC Source Capture Route", "Mixer", "Mixer to ADC"}, - {"ADC Sourc Capture Routee", "Line In", "Line In"}, + {"ADC Source Capture Routee", "Line In", "Line In"}, {"ADC Source Capture Route", "Mic 1", "Mic 1"}, {"ADC Source Capture Route", "Mic 2", "Mic 2"}, {"ADC", NULL, "ADC Source Capture Route"}, -- 2.37.3

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[PATCH] ASoC: sun50i-dmic: avoid unused variable warning for sun50i_dmic_of_match
by Ban Tao 19 Oct '22

19 Oct '22

In configurations with CONFIG_OF=n, we get a harmless build warning: sound/soc/sunxi/sun50i-dmic.c:268:34: warning: unused variable 'sun50i_dmic_of_match' [-Wunused-const-variable] Signed-off-by: Ban Tao <fengzheng923(a)gmail.com> Reported-by: kernel test robot <lkp(a)intel.com> --- sound/soc/sunxi/sun50i-dmic.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sunxi/sun50i-dmic.c b/sound/soc/sunxi/sun50i-dmic.c index 86cff5a..62509cb 100644 --- a/sound/soc/sunxi/sun50i-dmic.c +++ b/sound/soc/sunxi/sun50i-dmic.c @@ -391,7 +391,7 @@ static const struct dev_pm_ops sun50i_dmic_pm = { static struct platform_driver sun50i_dmic_driver = { .driver = { .name = "sun50i-dmic", - .of_match_table = of_match_ptr(sun50i_dmic_of_match), + .of_match_table = sun50i_dmic_of_match, .pm = &sun50i_dmic_pm, }, .probe = sun50i_dmic_probe, -- 2.7.4

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[PATCH v1] ASoC: simple-card: Fix up checks for HW param fixups
by Aidan MacDonald 19 Oct '22

19 Oct '22

The "convert-xxx" properties only have an effect for DPCM DAI links. A DAI link is only created as DPCM if the device tree requires it; part of this involves checking for the use of "convert-xxx" properties. When the convert-sample-format property was added, the checks got out of sync. A DAI link that specified only convert-sample-format but did not pass any of the other DPCM checks would not go into DPCM mode and the convert-sample-format property would be silently ignored. Fix this by adding a function to do the "convert-xxx" property checks, instead of open-coding it in simple-card and audio-graph-card. And add "convert-sample-format" to the check function so that DAI links using it will be initialized correctly. Fixes: 047a05366f4b ("ASoC: simple-card-utils: Fixup DAI sample format") Signed-off-by: Aidan MacDonald <aidanmacdonald.0x0(a)gmail.com> --- include/sound/simple_card_utils.h | 1 + sound/soc/generic/audio-graph-card.c | 2 +- sound/soc/generic/simple-card-utils.c | 15 +++++++++++++++ sound/soc/generic/simple-card.c | 3 +-- 4 files changed, 18 insertions(+), 3 deletions(-) diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index a0b827f0c2f6..cfed741e0181 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -177,6 +177,7 @@ void asoc_simple_convert_fixup(struct asoc_simple_data *data, struct snd_pcm_hw_params *params); void asoc_simple_parse_convert(struct device_node *np, char *prefix, struct asoc_simple_data *data); +bool asoc_simple_convert_is_required(const struct asoc_simple_data *data); int asoc_simple_parse_routing(struct snd_soc_card *card, char *prefix); diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index b327372f2e4a..7804625b131e 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -417,7 +417,7 @@ static inline bool parse_as_dpcm_link(struct asoc_simple_priv *priv, * or has convert-xxx property */ if ((of_get_child_count(codec_port) > 1) || - (adata->convert_rate || adata->convert_channels)) + asoc_simple_convert_is_required(adata)) return true; return false; diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index bef16833c487..7a708be24656 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -85,6 +85,21 @@ void asoc_simple_parse_convert(struct device_node *np, } EXPORT_SYMBOL_GPL(asoc_simple_parse_convert); +/** + * asoc_simple_convert_is_required() - Query if HW param conversion was requested + * @data: Link data. + * + * Returns true if any HW param conversion was requested for this DAI link with + * any "convert-xxx" properties. + */ +bool asoc_simple_convert_is_required(const struct asoc_simple_data *data) +{ + return data->convert_rate || + data->convert_channels || + data->convert_sample_format; +} +EXPORT_SYMBOL_GPL(asoc_simple_convert_is_required); + int asoc_simple_parse_daifmt(struct device *dev, struct device_node *node, struct device_node *codec, diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 78419e18717d..675991cdee87 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -393,8 +393,7 @@ static int __simple_for_each_link(struct asoc_simple_priv *priv, * or has convert-xxx property */ if (dpcm_selectable && - (num > 2 || - adata.convert_rate || adata.convert_channels)) { + (num > 2 || asoc_simple_convert_is_required(&adata))) { /* * np * |1(CPU)|0(Codec) li->cpu -- 2.38.0

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sof-es8336.conf
by GitHub issues - edited 18 Oct '22

18 Oct '22

alsa-project/alsa-ucm-conf issue #231 was edited from junocomp: The only way for the `Headphone Mixer Volume` to be enabled at startup the following commands needs to be added. `sset Headphone 3` When adding this to `sof-essx8336.conf` and reload alsa I get the following error ``` sudo alsaucm reload ALSA lib parser.c:1194:(parse_sequence) error: sequence command 'sset' is ignored ALSA lib parser.c:1045:(parse_sequence) error: string type is expected for sequence command ALSA lib parser.c:2289:(parse_controls_boot) Unable to parse BootSequence ALSA lib main.c:1541:(snd_use_case_mgr_open) error: failed to import hw:0 use case configuration -22 alsaucm: error failed to open sound card hw:0: Invalid argument ``` When running `amixer -c 0 sset Headphone 3` manually it works fine Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/231 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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sof-es8336.conf
by GitHub issues - opened 18 Oct '22

18 Oct '22

alsa-project/alsa-ucm-conf issue #231 was opened from junocomp: The only way for the `Headphone Mixer Volume` to be enabled at startup the following commands needs to be added. `sset Headphone 3 ` When adding this to `sof-essx8336.conf` and reload alsa I get the following error ``` sudo alsaucm reload ALSA lib parser.c:1194:(parse_sequence) error: sequence command 'sset' is ignored ALSA lib parser.c:1045:(parse_sequence) error: string type is expected for sequence command ALSA lib parser.c:2289:(parse_controls_boot) Unable to parse BootSequence ALSA lib main.c:1541:(snd_use_case_mgr_open) error: failed to import hw:0 use case configuration -22 alsaucm: error failed to open sound card hw:0: Invalid argument ``` When running `amixer -c 0 sset Headphone 3` manually it works fine Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/231 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH v1] ALSA: hda/realtek: Add quirk for ASUS Zenbook using CS35L41
by Stefan Binding 18 Oct '22

18 Oct '22

This Asus Zenbook laptop use Realtek HDA codec combined with 2xCS35L41 Amplifiers using SPI with External Boost. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> --- 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 7c177426bf303..79acd2a2caf20 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9395,6 +9395,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1d42, "ASUS Zephyrus G14 2022", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1d4e, "ASUS TM420", ALC256_FIXUP_ASUS_HPE), + SND_PCI_QUIRK(0x1043, 0x1e02, "ASUS UX3402", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1e11, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA502), SND_PCI_QUIRK(0x1043, 0x1e51, "ASUS Zephyrus M15", ALC294_FIXUP_ASUS_GU502_PINS), SND_PCI_QUIRK(0x1043, 0x1e5e, "ASUS ROG Strix G513", ALC294_FIXUP_ASUS_G513_PINS), -- 2.34.1

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