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

[PATCH] soundwire: export sdw_nwrite_no_pm and sdw_nread_no_pm functions
by Simon Trimmer 14 Sep '21

14 Sep '21

The patch "soundwire: export sdw_write/read_no_pm functions" exposed the single byte no_pm versions of the IO functions that can be used without touching PM, export the multi byte no_pm versions for the same reason. Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> --- drivers/soundwire/bus.c | 8 ++++---- include/linux/soundwire/sdw.h | 2 ++ 2 files changed, 6 insertions(+), 4 deletions(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index 1b115734a8f6..859de302dee2 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -379,8 +379,7 @@ int sdw_fill_msg(struct sdw_msg *msg, struct sdw_slave *slave, * all clients need to use the pm versions */ -static int -sdw_nread_no_pm(struct sdw_slave *slave, u32 addr, size_t count, u8 *val) +int sdw_nread_no_pm(struct sdw_slave *slave, u32 addr, size_t count, u8 *val) { struct sdw_msg msg; int ret; @@ -395,9 +394,9 @@ sdw_nread_no_pm(struct sdw_slave *slave, u32 addr, size_t count, u8 *val) ret = 0; return ret; } +EXPORT_SYMBOL(sdw_nread_no_pm); -static int -sdw_nwrite_no_pm(struct sdw_slave *slave, u32 addr, size_t count, const u8 *val) +int sdw_nwrite_no_pm(struct sdw_slave *slave, u32 addr, size_t count, const u8 *val) { struct sdw_msg msg; int ret; @@ -412,6 +411,7 @@ sdw_nwrite_no_pm(struct sdw_slave *slave, u32 addr, size_t count, const u8 *val) ret = 0; return ret; } +EXPORT_SYMBOL(sdw_nwrite_no_pm); int sdw_write_no_pm(struct sdw_slave *slave, u32 addr, u8 value) { diff --git a/include/linux/soundwire/sdw.h b/include/linux/soundwire/sdw.h index 76ce3f3ac0f2..2a5395f0dcf1 100644 --- a/include/linux/soundwire/sdw.h +++ b/include/linux/soundwire/sdw.h @@ -1042,7 +1042,9 @@ int sdw_write(struct sdw_slave *slave, u32 addr, u8 value); int sdw_write_no_pm(struct sdw_slave *slave, u32 addr, u8 value); int sdw_read_no_pm(struct sdw_slave *slave, u32 addr); int sdw_nread(struct sdw_slave *slave, u32 addr, size_t count, u8 *val); +int sdw_nread_no_pm(struct sdw_slave *slave, u32 addr, size_t count, u8 *val); int sdw_nwrite(struct sdw_slave *slave, u32 addr, size_t count, const u8 *val); +int sdw_nwrite_no_pm(struct sdw_slave *slave, u32 addr, size_t count, const u8 *val); int sdw_update(struct sdw_slave *slave, u32 addr, u8 mask, u8 val); int sdw_update_no_pm(struct sdw_slave *slave, u32 addr, u8 mask, u8 val); -- 2.33.0

4 3

[PATCH AUTOSEL 5.13 07/19] ASoC: audio-graph: respawn Platform Support
by Sasha Levin 14 Sep '21

14 Sep '21

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> [ Upstream commit 5f939f49771002f347039edf984aca42f30fc31a ] commit 63f2f9cceb09f8 ("ASoC: audio-graph: remove Platform support") removed Platform support from audio-graph, because it doesn't have "plat" support on DT (simple-card has). But, Platform support is needed if user is using snd_dmaengine_pcm_register() which adds generic DMA as Platform. And this Platform dev is using CPU dev. Without this patch, at least STM32MP15 audio sound card is no more functional (v5.13 or later). This patch respawn Platform Support on audio-graph again. Reported-by: Olivier MOYSAN <olivier.moysan(a)foss.st.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Olivier MOYSAN <olivier.moysan(a)foss.st.com> Link: https://lore.kernel.org/r/878s0jzrpf.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/generic/audio-graph-card.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 5e71382467e8..546f6fd0609e 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -285,6 +285,7 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, if (li->cpu) { struct snd_soc_card *card = simple_priv_to_card(priv); struct snd_soc_dai_link_component *cpus = asoc_link_to_cpu(dai_link, 0); + struct snd_soc_dai_link_component *platforms = asoc_link_to_platform(dai_link, 0); int is_single_links = 0; /* Codec is dummy */ @@ -313,6 +314,7 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, dai_link->no_pcm = 1; asoc_simple_canonicalize_cpu(cpus, is_single_links); + asoc_simple_canonicalize_platform(platforms, cpus); } else { struct snd_soc_codec_conf *cconf = simple_props_to_codec_conf(dai_props, 0); struct snd_soc_dai_link_component *codecs = asoc_link_to_codec(dai_link, 0); @@ -366,6 +368,7 @@ static int graph_dai_link_of(struct asoc_simple_priv *priv, struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); struct snd_soc_dai_link_component *cpus = asoc_link_to_cpu(dai_link, 0); struct snd_soc_dai_link_component *codecs = asoc_link_to_codec(dai_link, 0); + struct snd_soc_dai_link_component *platforms = asoc_link_to_platform(dai_link, 0); char dai_name[64]; int ret, is_single_links = 0; @@ -383,6 +386,7 @@ static int graph_dai_link_of(struct asoc_simple_priv *priv, "%s-%s", cpus->dai_name, codecs->dai_name); asoc_simple_canonicalize_cpu(cpus, is_single_links); + asoc_simple_canonicalize_platform(platforms, cpus); ret = graph_link_init(priv, cpu_ep, codec_ep, li, dai_name); if (ret < 0) @@ -608,6 +612,7 @@ static int graph_count_noml(struct asoc_simple_priv *priv, li->num[li->link].cpus = 1; li->num[li->link].codecs = 1; + li->num[li->link].platforms = 1; li->link += 1; /* 1xCPU-Codec */ @@ -630,6 +635,7 @@ static int graph_count_dpcm(struct asoc_simple_priv *priv, if (li->cpu) { li->num[li->link].cpus = 1; + li->num[li->link].platforms = 1; li->link++; /* 1xCPU-dummy */ } else { -- 2.30.2

1 0

[PATCH AUTOSEL 5.14 09/25] ASoC: audio-graph: respawn Platform Support
by Sasha Levin 14 Sep '21

14 Sep '21

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> [ Upstream commit 5f939f49771002f347039edf984aca42f30fc31a ] commit 63f2f9cceb09f8 ("ASoC: audio-graph: remove Platform support") removed Platform support from audio-graph, because it doesn't have "plat" support on DT (simple-card has). But, Platform support is needed if user is using snd_dmaengine_pcm_register() which adds generic DMA as Platform. And this Platform dev is using CPU dev. Without this patch, at least STM32MP15 audio sound card is no more functional (v5.13 or later). This patch respawn Platform Support on audio-graph again. Reported-by: Olivier MOYSAN <olivier.moysan(a)foss.st.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Olivier MOYSAN <olivier.moysan(a)foss.st.com> Link: https://lore.kernel.org/r/878s0jzrpf.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/generic/audio-graph-card.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 5e71382467e8..546f6fd0609e 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -285,6 +285,7 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, if (li->cpu) { struct snd_soc_card *card = simple_priv_to_card(priv); struct snd_soc_dai_link_component *cpus = asoc_link_to_cpu(dai_link, 0); + struct snd_soc_dai_link_component *platforms = asoc_link_to_platform(dai_link, 0); int is_single_links = 0; /* Codec is dummy */ @@ -313,6 +314,7 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, dai_link->no_pcm = 1; asoc_simple_canonicalize_cpu(cpus, is_single_links); + asoc_simple_canonicalize_platform(platforms, cpus); } else { struct snd_soc_codec_conf *cconf = simple_props_to_codec_conf(dai_props, 0); struct snd_soc_dai_link_component *codecs = asoc_link_to_codec(dai_link, 0); @@ -366,6 +368,7 @@ static int graph_dai_link_of(struct asoc_simple_priv *priv, struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); struct snd_soc_dai_link_component *cpus = asoc_link_to_cpu(dai_link, 0); struct snd_soc_dai_link_component *codecs = asoc_link_to_codec(dai_link, 0); + struct snd_soc_dai_link_component *platforms = asoc_link_to_platform(dai_link, 0); char dai_name[64]; int ret, is_single_links = 0; @@ -383,6 +386,7 @@ static int graph_dai_link_of(struct asoc_simple_priv *priv, "%s-%s", cpus->dai_name, codecs->dai_name); asoc_simple_canonicalize_cpu(cpus, is_single_links); + asoc_simple_canonicalize_platform(platforms, cpus); ret = graph_link_init(priv, cpu_ep, codec_ep, li, dai_name); if (ret < 0) @@ -608,6 +612,7 @@ static int graph_count_noml(struct asoc_simple_priv *priv, li->num[li->link].cpus = 1; li->num[li->link].codecs = 1; + li->num[li->link].platforms = 1; li->link += 1; /* 1xCPU-Codec */ @@ -630,6 +635,7 @@ static int graph_count_dpcm(struct asoc_simple_priv *priv, if (li->cpu) { li->num[li->link].cpus = 1; + li->num[li->link].platforms = 1; li->link++; /* 1xCPU-dummy */ } else { -- 2.30.2

1 0

[PATCH] ALSA: hda/realtek: Quirks to enable speaker output for Lenovo Legion 7i 15IMHG05, Yoga 7i 14ITL5/15ITL5, and 13s Gen2 laptops.
by Cameron Berkenpas 14 Sep '21

14 Sep '21

This patch initializes and enables speaker output on the Lenovo Legion 7i 15IMHG05, Yoga 7i 14ITL5/15ITL5, and 13s Gen2 series of laptops using the HDA verb sequence specific to each model. Speaker automute is disabled for the Lenovo Legion 7i 15IMHG05 to avoid breaking speaker output on resume and when devices are unplugged from its headphone jack. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208555 Signed-off-by: Cameron Berkenpas <cam(a)neo-zeon.de> --- sound/pci/hda/patch_realtek.c | 129 ++++++++++++++++++++++++++++++++++ 1 file changed, 129 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8b7a389b6aed..7e94107e8e59 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6429,6 +6429,20 @@ static void alc_fixup_thinkpad_acpi(struct hda_codec *codec, hda_fixup_thinkpad_acpi(codec, fix, action); } +/* Fixup for Lenovo Legion 15IMHg05 speaker output on headset removal. */ +static void alc287_fixup_legion_15imhg05_speakers(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + struct alc_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_INIT: + spec->gen.automute_speaker = 0; + break; + } +} + /* for alc295_fixup_hp_top_speakers */ #include "hp_x360_helper.c" @@ -6646,6 +6660,10 @@ enum { ALC623_FIXUP_LENOVO_THINKSTATION_P340, ALC255_FIXUP_ACER_HEADPHONE_AND_MIC, ALC236_FIXUP_HP_LIMIT_INT_MIC_BOOST, + ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS, + ALC287_FIXUP_LEGION_15IMHG05_AUTOMUTE, + ALC287_FIXUP_YOGA7_14ITL_SPEAKERS, + ALC287_FIXUP_13S_GEN2_SPEAKERS }; static const struct hda_fixup alc269_fixups[] = { @@ -8236,6 +8254,113 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF, }, + [ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS] = { + .type = HDA_FIXUP_VERBS, + //.v.verbs = legion_15imhg05_coefs, + .v.verbs = (const struct hda_verb[]) { + // set left speaker Legion 7i. + {0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x41 }, + + {0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xc }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x1a }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + {0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x2 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + // set right speaker Legion 7i. + {0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x42 }, + + {0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xc }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x2a }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + {0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x2 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + {} + }, + .chained = true, + .chain_id = ALC287_FIXUP_LEGION_15IMHG05_AUTOMUTE, + }, + [ALC287_FIXUP_LEGION_15IMHG05_AUTOMUTE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc287_fixup_legion_15imhg05_speakers, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE, + }, + [ALC287_FIXUP_YOGA7_14ITL_SPEAKERS] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + // set left speaker Yoga 7i. + {0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x41 }, + + {0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xc }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x1a }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + {0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x2 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + // set right speaker Yoga 7i. + {0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x46 }, + + {0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xc }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x2a }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + {0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x2 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + {} + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE, + }, + [ALC287_FIXUP_13S_GEN2_SPEAKERS] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + {0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x41 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x2 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + {0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x42 }, + {0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x2 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + {} + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -8630,6 +8755,10 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3818, "Lenovo C940", ALC298_FIXUP_LENOVO_SPK_VOLUME), SND_PCI_QUIRK(0x17aa, 0x3827, "Ideapad S740", ALC285_FIXUP_IDEAPAD_S740_COEF), SND_PCI_QUIRK(0x17aa, 0x3843, "Yoga 9i", ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP), + SND_PCI_QUIRK(0x17aa, 0x3813, "Legion 7i 15IMHG05", ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS), + SND_PCI_QUIRK(0x17aa, 0x3852, "Lenovo Yoga 7 14ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), + SND_PCI_QUIRK(0x17aa, 0x3853, "Lenovo Yoga 7 15ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), + SND_PCI_QUIRK(0x17aa, 0x3819, "Lenovo 13s Gen2 ITL", ALC287_FIXUP_13S_GEN2_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI), -- 2.30.2

2 5

kernel threads preventing wakeup of thread using snd_pcm_readi?
by Bert Schiettecatte 13 Sep '21

13 Sep '21

Hi all I am working on a realtime audio application on the rk3288 platform. I am using stable kernel 5.13.13. I'm using the open source tracy profiler to profile my application. My development PC receives trace data via ethernet from my platform. I have custom alsa drivers on my platform for my DACs and ADCs. My application has a thread in which I use snd_pcm_readi which blocks until it can read enough samples. I have another thread in which I write the samples to disk (sandisk extreme SD card). I use a ring-buffer so the code is lock-free. I've noticed that over a timespan of a few minutes I get a few clicks, which seem to come from the thread blocking on snd_pcm_readi not waking up in time. I am observing this in the traces I collect via tracy. Right before this happens, I can see kworker/kblockd being scheduled on one of the cores many times and rapidly, or ocasionally ksoftirqd, mmc_complete, or events_highpri. From the trace I collect, i have the impression nothing is being scheduled on the other cores, so I do not see a reason why my thread should not be woken up and scheduled. Why is the thread blocking on snd_pcm_readi not waking up in time, and is it possible that the kworker threads are preventing this, and why? Any suggestions on how to debug this further are greatly appreciated. Thanks Bert

1 0

[PATCH 0/2] ARM: dts: sun8i: r40: Add I2S nodes
by Jernej Skrabec 13 Sep '21

13 Sep '21

Allwinner R40 has 3 I2S controllers, compatible to those in H3. Patch 1 adds R40/H3 compatible pair to DT bindings. Patch 2 adds I2S nodes to R40 DT. Please take a look. Best regards, Jernej Jernej Skrabec (2): dt-bindings: sound: sun4i-i2s: add Allwinner R40 I2S compatible ARM: dts: sun8i: r40: Add I2S nodes .../sound/allwinner,sun4i-a10-i2s.yaml | 3 ++ arch/arm/boot/dts/sun8i-r40.dtsi | 39 +++++++++++++++++++ 2 files changed, 42 insertions(+) -- 2.33.0

3 6

[PATCH v2] ASoC: mediatek: common: handle NULL case in suspend/resume function
by Trevor Wu 13 Sep '21

13 Sep '21

When memory allocation for afe->reg_back_up fails, reg_back_up can't be used. Keep the suspend/resume flow but skip register backup when afe->reg_back_up is NULL, in case illegal memory access happens. Fixes: 283b612429a2 ("ASoC: mediatek: implement mediatek common structure") Signed-off-by: Trevor Wu <trevor.wu(a)mediatek.com> Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- Change since v1: add Reported-by tag --- sound/soc/mediatek/common/mtk-afe-fe-dai.c | 19 +++++++++++-------- 1 file changed, 11 insertions(+), 8 deletions(-) diff --git a/sound/soc/mediatek/common/mtk-afe-fe-dai.c b/sound/soc/mediatek/common/mtk-afe-fe-dai.c index baaa5881b1d4..e95c7c018e7d 100644 --- a/sound/soc/mediatek/common/mtk-afe-fe-dai.c +++ b/sound/soc/mediatek/common/mtk-afe-fe-dai.c @@ -334,9 +334,11 @@ int mtk_afe_suspend(struct snd_soc_component *component) devm_kcalloc(dev, afe->reg_back_up_list_num, sizeof(unsigned int), GFP_KERNEL); - for (i = 0; i < afe->reg_back_up_list_num; i++) - regmap_read(regmap, afe->reg_back_up_list[i], - &afe->reg_back_up[i]); + if (afe->reg_back_up) { + for (i = 0; i < afe->reg_back_up_list_num; i++) + regmap_read(regmap, afe->reg_back_up_list[i], + &afe->reg_back_up[i]); + } afe->suspended = true; afe->runtime_suspend(dev); @@ -356,12 +358,13 @@ int mtk_afe_resume(struct snd_soc_component *component) afe->runtime_resume(dev); - if (!afe->reg_back_up) + if (!afe->reg_back_up) { dev_dbg(dev, "%s no reg_backup\n", __func__); - - for (i = 0; i < afe->reg_back_up_list_num; i++) - mtk_regmap_write(regmap, afe->reg_back_up_list[i], - afe->reg_back_up[i]); + } else { + for (i = 0; i < afe->reg_back_up_list_num; i++) + mtk_regmap_write(regmap, afe->reg_back_up_list[i], + afe->reg_back_up[i]); + } afe->suspended = false; return 0; -- 2.18.0

2 4

[PATCH] Revert "ALSA: hda: Drop workaround for a hang at shutdown again"
by Takashi Iwai 13 Sep '21

13 Sep '21

This reverts commit 8fc8e903156f42c66245838441d03607e9067381. It was expected that the fixes in HD-audio codec side would make the workaround redundant, but unfortunately it doesn't seem sufficing. Resurrect the workaround for now. Fixes: 8fc8e903156f ("ALSA: hda: Drop workaround for a hang at shutdown again") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=214045 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 3aa432d814a2..47777439961c 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -883,10 +883,11 @@ static unsigned int azx_get_pos_skl(struct azx *chip, struct azx_dev *azx_dev) return azx_get_pos_posbuf(chip, azx_dev); } -static void azx_shutdown_chip(struct azx *chip) +static void __azx_shutdown_chip(struct azx *chip, bool skip_link_reset) { azx_stop_chip(chip); - azx_enter_link_reset(chip); + if (!skip_link_reset) + azx_enter_link_reset(chip); azx_clear_irq_pending(chip); display_power(chip, false); } @@ -895,6 +896,11 @@ static void azx_shutdown_chip(struct azx *chip) static DEFINE_MUTEX(card_list_lock); static LIST_HEAD(card_list); +static void azx_shutdown_chip(struct azx *chip) +{ + __azx_shutdown_chip(chip, false); +} + static void azx_add_card_list(struct azx *chip) { struct hda_intel *hda = container_of(chip, struct hda_intel, chip); @@ -2357,7 +2363,7 @@ static void azx_shutdown(struct pci_dev *pci) return; chip = card->private_data; if (chip && chip->running) - azx_shutdown_chip(chip); + __azx_shutdown_chip(chip, true); } /* PCI IDs */ -- 2.26.2

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[PATCH] ASoC: pcm5102a: increase rate from 192k to 384k
by gearhead 13 Sep '21

13 Sep '21

the pcm5102a is capable of 384k, but the current code limits it to 192k. This commit extends to 384k Signed-off-by: gearhead <ys3al35l(a)gmail.com> --- sound/soc/codecs/pcm5102a.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/pcm5102a.c b/sound/soc/codecs/pcm5102a.c index b8cfc250612c..f39f98bbc97f 100644 --- a/sound/soc/codecs/pcm5102a.c +++ b/sound/soc/codecs/pcm5102a.c @@ -17,7 +17,7 @@ static struct snd_soc_dai_driver pcm5102a_dai = { .playback = { .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000, + .rates = SNDRV_PCM_RATE_8000_384000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE -- 2.33.0

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[PATCH] ASoC: mediatek: mt8195: Make use of the helper function devm_platform_ioremap_resource()
by Cai Huoqing 13 Sep '21

13 Sep '21

Use the devm_platform_ioremap_resource() helper instead of calling platform_get_resource() and devm_ioremap_resource() separately Signed-off-by: Cai Huoqing <caihuoqing(a)baidu.com> --- sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c b/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c index 6635c3f72ecc..8c697f41d51d 100644 --- a/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c +++ b/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c @@ -3057,7 +3057,6 @@ static int mt8195_afe_pcm_dev_probe(struct platform_device *pdev) { struct mtk_base_afe *afe; struct mt8195_afe_private *afe_priv; - struct resource *res; struct device *dev = &pdev->dev; int i, irq_id, ret; struct snd_soc_component *component; @@ -3078,8 +3077,7 @@ static int mt8195_afe_pcm_dev_probe(struct platform_device *pdev) afe_priv = afe->platform_priv; afe->dev = &pdev->dev; - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - afe->base_addr = devm_ioremap_resource(&pdev->dev, res); + afe->base_addr = devm_platform_ioremap_resource(pdev, 0); if (IS_ERR(afe->base_addr)) return PTR_ERR(afe->base_addr); -- 2.25.1

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