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

[alsa-devel] [PATCH AUTOSEL 4.19 475/671] ALSA: aoa: onyx: always initialize register read value
by Sasha Levin 16 Jan '20

16 Jan '20

From: Johannes Berg <johannes(a)sipsolutions.net> [ Upstream commit f474808acb3c4b30552d9c59b181244e0300d218 ] A lot of places in the driver use onyx_read_register() without checking the return value, and it's been working OK for ~10 years or so, so probably never fails ... Rather than trying to check the return value everywhere, which would be relatively intrusive, at least make sure we don't use an uninitialized value. Fixes: f3d9478b2ce4 ("[ALSA] snd-aoa: add snd-aoa") Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Johannes Berg <johannes(a)sipsolutions.net> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/aoa/codecs/onyx.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/aoa/codecs/onyx.c b/sound/aoa/codecs/onyx.c index d2d96ca082b7..6224fd3bbf7c 100644 --- a/sound/aoa/codecs/onyx.c +++ b/sound/aoa/codecs/onyx.c @@ -74,8 +74,10 @@ static int onyx_read_register(struct onyx *onyx, u8 reg, u8 *value) return 0; } v = i2c_smbus_read_byte_data(onyx->i2c, reg); - if (v < 0) + if (v < 0) { + *value = 0; return -1; + } *value = (u8)v; onyx->cache[ONYX_REG_CONTROL-FIRSTREGISTER] = *value; return 0; -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.19 424/671] ASoC: ti: davinci-mcasp: Fix slot mask settings when using multiple AXRs
by Sasha Levin 16 Jan '20

16 Jan '20

From: Peter Ujfalusi <peter.ujfalusi(a)ti.com> [ Upstream commit fd14f4436fd47d5418023c90e933e66d3645552e ] If multiple serializers are connected in the system and the number of channels will need to use more than one serializer the mask to enable the serializers were left to 0 if tdm_mask is provided Fixes: dd55ff8346a97 ("ASoC: davinci-mcasp: Add set_tdm_slots() support") Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/davinci/davinci-mcasp.c | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 6a8c279a4b20..14ab16e1369f 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -874,14 +874,13 @@ static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream, active_slots = hweight32(mcasp->tdm_mask[stream]); active_serializers = (channels + active_slots - 1) / active_slots; - if (active_serializers == 1) { + if (active_serializers == 1) active_slots = channels; - for (i = 0; i < total_slots; i++) { - if ((1 << i) & mcasp->tdm_mask[stream]) { - mask |= (1 << i); - if (--active_slots <= 0) - break; - } + for (i = 0; i < total_slots; i++) { + if ((1 << i) & mcasp->tdm_mask[stream]) { + mask |= (1 << i); + if (--active_slots <= 0) + break; } } } else { -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.19 411/671] ASoC: meson: axg-tdmout: right_j is not supported
by Sasha Levin 16 Jan '20

16 Jan '20

From: Jerome Brunet <jbrunet(a)baylibre.com> [ Upstream commit 7e0d7d0fbd06af0507611f85dba8daf24832abd9 ] Right justified format is actually not supported by the amlogic tdm output encoder. Fixes: c41c2a355b86 ("ASoC: meson: add tdm output driver") Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/meson/axg-tdmout.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/meson/axg-tdmout.c b/sound/soc/meson/axg-tdmout.c index f73368ee1088..d11acb3cc696 100644 --- a/sound/soc/meson/axg-tdmout.c +++ b/sound/soc/meson/axg-tdmout.c @@ -136,7 +136,6 @@ static int axg_tdmout_prepare(struct regmap *map, struct axg_tdm_stream *ts) break; case SND_SOC_DAIFMT_LEFT_J: - case SND_SOC_DAIFMT_RIGHT_J: case SND_SOC_DAIFMT_DSP_B: val |= TDMOUT_CTRL0_INIT_BITNUM(2); break; -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.19 410/671] ASoC: meson: axg-tdmin: right_j is not supported
by Sasha Levin 16 Jan '20

16 Jan '20

From: Jerome Brunet <jbrunet(a)baylibre.com> [ Upstream commit 47c317b786b6c1efc2cb3cdb894fd323422fe5ea ] Right justified format is actually not supported by the amlogic tdm input decoder. Fixes: 13a22e6a98f8 ("ASoC: meson: add tdm input driver") Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/meson/axg-tdmin.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/meson/axg-tdmin.c b/sound/soc/meson/axg-tdmin.c index bbac44c81688..37207bbebb2a 100644 --- a/sound/soc/meson/axg-tdmin.c +++ b/sound/soc/meson/axg-tdmin.c @@ -119,7 +119,6 @@ static int axg_tdmin_prepare(struct regmap *map, struct axg_tdm_stream *ts) break; case SND_SOC_DAIFMT_LEFT_J: - case SND_SOC_DAIFMT_RIGHT_J: case SND_SOC_DAIFMT_DSP_B: val = TDMIN_CTRL_IN_BIT_SKEW(2); break; -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.19 344/671] ALSA: aica: Fix a long-time build breakage
by Sasha Levin 16 Jan '20

16 Jan '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 534420c6ff87d3052540f1fd346e0adcff440819 ] The build of aica sound driver has been broken since the timer API conversion and some code rewrite. This patch fixes the breakage by using the common substream field, as well as a bit cleaning up wrt the timer handling in the code. Fixes: d522bb6a105f ("ALSA: sh: aica: Convert timers to use timer_setup()") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/sh/aica.c | 14 ++++---------- 1 file changed, 4 insertions(+), 10 deletions(-) diff --git a/sound/sh/aica.c b/sound/sh/aica.c index 2b26311405a4..ad3f71358486 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -303,7 +303,7 @@ static void aica_period_elapsed(struct timer_list *t) { struct snd_card_aica *dreamcastcard = from_timer(dreamcastcard, t, timer); - struct snd_pcm_substream *substream = dreamcastcard->timer_substream; + struct snd_pcm_substream *substream = dreamcastcard->substream; /*timer function - so cannot sleep */ int play_period; struct snd_pcm_runtime *runtime; @@ -335,13 +335,6 @@ static void spu_begin_dma(struct snd_pcm_substream *substream) dreamcastcard = substream->pcm->private_data; /*get the queue to do the work */ schedule_work(&(dreamcastcard->spu_dma_work)); - /* Timer may already be running */ - if (unlikely(dreamcastcard->timer_substream)) { - mod_timer(&dreamcastcard->timer, jiffies + 4); - return; - } - timer_setup(&dreamcastcard->timer, aica_period_elapsed, 0); - dreamcastcard->timer_substream = substream; mod_timer(&dreamcastcard->timer, jiffies + 4); } @@ -379,8 +372,8 @@ static int snd_aicapcm_pcm_close(struct snd_pcm_substream { struct snd_card_aica *dreamcastcard = substream->pcm->private_data; flush_work(&(dreamcastcard->spu_dma_work)); - if (dreamcastcard->timer_substream) - del_timer(&dreamcastcard->timer); + del_timer(&dreamcastcard->timer); + dreamcastcard->substream = NULL; kfree(dreamcastcard->channel); spu_disable(); return 0; @@ -615,6 +608,7 @@ static int snd_aica_probe(struct platform_device *devptr) "Yamaha AICA Super Intelligent Sound Processor for SEGA Dreamcast"); /* Prepare to use the queue */ INIT_WORK(&(dreamcastcard->spu_dma_work), run_spu_dma); + timer_setup(&dreamcastcard->timer, aica_period_elapsed, 0); /* Load the PCM 'chip' */ err = snd_aicapcmchip(dreamcastcard, 0); if (unlikely(err < 0)) -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.19 324/671] ASoC: fix valid stream condition
by Sasha Levin 16 Jan '20

16 Jan '20

From: Jerome Brunet <jbrunet(a)baylibre.com> [ Upstream commit 6a7c59c6d9f3b280e81d7a04bbe4e55e90152dce ] A stream may specify a rate range using 'rate_min' and 'rate_max', so a stream may be valid and not specify any rates. However, as stream cannot be valid and not have any channel. Let's use this condition instead to determine if a stream is valid or not. Fixes: cde79035c6cf ("ASoC: Handle multiple codecs with split playback / capture") Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-pcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 551bfc581fc1..53fefa7c982f 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -42,8 +42,8 @@ static bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream) else codec_stream = &dai->driver->capture; - /* If the codec specifies any rate at all, it supports the stream. */ - return codec_stream->rates; + /* If the codec specifies any channels at all, it supports the stream */ + return codec_stream->channels_min; } /** -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.19 309/671] ALSA: usb-audio: Handle the error from snd_usb_mixer_apply_create_quirk()
by Sasha Levin 16 Jan '20

16 Jan '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 328e9f6973be2ee67862cb17bf6c0c5c5918cd72 ] The error from snd_usb_mixer_apply_create_quirk() is ignored in the current usb-audio driver code, which will continue the probing even after the error. Let's take it more serious. Fixes: 7b1eda223deb ("ALSA: usb-mixer: factor out quirks") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/mixer.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index d7778f2bcbf8..6ac6a0980124 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -3480,7 +3480,9 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, if (err < 0) goto _error; - snd_usb_mixer_apply_create_quirk(mixer); + err = snd_usb_mixer_apply_create_quirk(mixer); + if (err < 0) + goto _error; err = snd_device_new(chip->card, SNDRV_DEV_CODEC, mixer, &dev_ops); if (err < 0) -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.19 206/671] ASoC: qcom: Fix of-node refcount unbalance in apq8016_sbc_parse_of()
by Sasha Levin 16 Jan '20

16 Jan '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 8d1667200850f8753c0265fa4bd25c9a6e5f94ce ] The apq8016 driver leaves the of-node refcount at aborting from the loop of for_each_child_of_node() in the error path. Not only the iterator node of for_each_child_of_node(), the children nodes referred from it for codec and cpu have to be properly unreferenced. Fixes: bdb052e81f62 ("ASoC: qcom: add apq8016 sound card support") Cc: Patrick Lai <plai(a)codeaurora.org> Cc: Banajit Goswami <bgoswami(a)codeaurora.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/qcom/apq8016_sbc.c | 21 ++++++++++++++++----- 1 file changed, 16 insertions(+), 5 deletions(-) diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c index 1dd23bba1bed..4b559932adc3 100644 --- a/sound/soc/qcom/apq8016_sbc.c +++ b/sound/soc/qcom/apq8016_sbc.c @@ -164,41 +164,52 @@ static struct apq8016_sbc_data *apq8016_sbc_parse_of(struct snd_soc_card *card) if (!cpu || !codec) { dev_err(dev, "Can't find cpu/codec DT node\n"); - return ERR_PTR(-EINVAL); + ret = -EINVAL; + goto error; } link->cpu_of_node = of_parse_phandle(cpu, "sound-dai", 0); if (!link->cpu_of_node) { dev_err(card->dev, "error getting cpu phandle\n"); - return ERR_PTR(-EINVAL); + ret = -EINVAL; + goto error; } ret = snd_soc_of_get_dai_name(cpu, &link->cpu_dai_name); if (ret) { dev_err(card->dev, "error getting cpu dai name\n"); - return ERR_PTR(ret); + goto error; } ret = snd_soc_of_get_dai_link_codecs(dev, codec, link); if (ret < 0) { dev_err(card->dev, "error getting codec dai name\n"); - return ERR_PTR(ret); + goto error; } link->platform_of_node = link->cpu_of_node; ret = of_property_read_string(np, "link-name", &link->name); if (ret) { dev_err(card->dev, "error getting codec dai_link name\n"); - return ERR_PTR(ret); + goto error; } link->stream_name = link->name; link->init = apq8016_sbc_dai_init; link++; + + of_node_put(cpu); + of_node_put(codec); } return data; + + error: + of_node_put(np); + of_node_put(cpu); + of_node_put(codec); + return ERR_PTR(ret); } static const struct snd_soc_dapm_widget apq8016_sbc_dapm_widgets[] = { -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.19 122/671] ASoC: imx-sgtl5000: put of nodes if finding codec fails
by Sasha Levin 16 Jan '20

16 Jan '20

From: Stefan Agner <stefan(a)agner.ch> [ Upstream commit d9866572486802bc598a3e8576a5231378d190de ] Make sure to properly put the of node in case finding the codec fails. Fixes: 81e8e4926167 ("ASoC: fsl: add sgtl5000 clock support for imx-sgtl5000") Signed-off-by: Stefan Agner <stefan(a)agner.ch> Reviewed-by: Daniel Baluta <daniel.baluta(a)nxp.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Reviewed-by: Fabio Estevam <festevam(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/imx-sgtl5000.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 9b9a7ec52905..4bd8da3a5f5b 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -112,7 +112,8 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) codec_dev = of_find_i2c_device_by_node(codec_np); if (!codec_dev) { dev_err(&pdev->dev, "failed to find codec platform device\n"); - return -EPROBE_DEFER; + ret = -EPROBE_DEFER; + goto fail; } data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.19 049/671] ASoC: sun8i-codec: add missing route for ADC
by Sasha Levin 16 Jan '20

16 Jan '20

From: Vasily Khoruzhick <anarsoul(a)gmail.com> [ Upstream commit 9ee325d029c4abb75716851ce38863845911d605 ] sun8i-codec misses a route from ADC to AIF1 Slot 0 ADC. Add it to the driver to avoid adding it to every dts. Fixes: eda85d1fee05d ("ASoC: sun8i-codec: Add ADC support for a33") Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sunxi/sun8i-codec.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/sunxi/sun8i-codec.c b/sound/soc/sunxi/sun8i-codec.c index bf615fa16dc8..a3db6a68dfe6 100644 --- a/sound/soc/sunxi/sun8i-codec.c +++ b/sound/soc/sunxi/sun8i-codec.c @@ -465,7 +465,11 @@ static const struct snd_soc_dapm_route sun8i_codec_dapm_routes[] = { { "Right Digital DAC Mixer", "AIF1 Slot 0 Digital DAC Playback Switch", "AIF1 Slot 0 Right"}, - /* ADC routes */ + /* ADC Routes */ + { "AIF1 Slot 0 Right ADC", NULL, "ADC" }, + { "AIF1 Slot 0 Left ADC", NULL, "ADC" }, + + /* ADC Mixer Routes */ { "Left Digital ADC Mixer", "AIF1 Data Digital ADC Capture Switch", "AIF1 Slot 0 Left ADC" }, { "Right Digital ADC Mixer", "AIF1 Data Digital ADC Capture Switch", -- 2.20.1

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