[alsa-devel] [PATCH AUTOSEL 4.19 309/671] ALSA: usb-audio: Handle the error from snd_usb_mixer_apply_create_quirk()

[alsa-devel] [PATCH AUTOSEL 4.19 309/671] ALSA: usb-audio: Handle the error from snd_usb_mixer_apply_create_quirk()

[Sasha Levin]

From: Takashi Iwai <tiwai@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@suse.de>
Signed-off-by: Sasha Levin <sashal@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 4.19 324/671] ASoC: fix valid stream condition

[Sasha Levin]

From: Jerome Brunet <jbrunet@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@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@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 4.19 344/671] ALSA: aica: Fix a long-time build breakage

[Sasha Levin]

From: Takashi Iwai <tiwai@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@suse.de>
Signed-off-by: Sasha Levin <sashal@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 4.19 410/671] ASoC: meson: axg-tdmin: right_j is not supported

[Sasha Levin]

From: Jerome Brunet <jbrunet@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@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@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 4.19 411/671] ASoC: meson: axg-tdmout: right_j is not supported

[Sasha Levin]

From: Jerome Brunet <jbrunet@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@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@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 4.19 424/671] ASoC: ti: davinci-mcasp: Fix slot mask settings when using multiple AXRs

[Sasha Levin]

From: Peter Ujfalusi <peter.ujfalusi@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@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@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 4.19 475/671] ALSA: aoa: onyx: always initialize register read value

[Sasha Levin]

From: Johannes Berg <johannes@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@canb.auug.org.au>
Signed-off-by: Johannes Berg <johannes@sipsolutions.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@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 4.19 490/671] ASoC: es8328: Fix copy-paste error in es8328_right_line_controls

[Sasha Levin]

From: YueHaibing <yuehaibing@huawei.com>

[ Upstream commit 630742c296341a8cfe00dfd941392025ba8dd4e8 ]

It seems 'es8328_rline_enum' should be used
in es8328_right_line_controls

Fixes: 567e4f98922c ("ASoC: add es8328 codec driver")
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190815092300.68712-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/codecs/es8328.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c
index e9fc2fd97d2f..3afa163f7652 100644
--- a/sound/soc/codecs/es8328.c
+++ b/sound/soc/codecs/es8328.c
@@ -231,7 +231,7 @@ static const struct soc_enum es8328_rline_enum =
 			      ARRAY_SIZE(es8328_line_texts),
 			      es8328_line_texts);
 static const struct snd_kcontrol_new es8328_right_line_controls =
-	SOC_DAPM_ENUM("Route", es8328_lline_enum);
+	SOC_DAPM_ENUM("Route", es8328_rline_enum);
 
 /* Left Mixer */
 static const struct snd_kcontrol_new es8328_left_mixer_controls[] = {
-- 
2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.19 491/671] ASoC: cs4349: Use PM ops 'cs4349_runtime_pm'

[Sasha Levin]

From: YueHaibing <yuehaibing@huawei.com>

[ Upstream commit 9b4275c415acca6264a3d7f1182589959c93d530 ]

sound/soc/codecs/cs4349.c:358:32: warning:
 cs4349_runtime_pm defined but not used [-Wunused-const-variable=]

cs4349_runtime_pm ops already defined, it seems
we should enable it.

Reported-by: Hulk Robot <hulkci@huawei.com>
Fixes: e40da86 ("ASoC: cs4349: Add support for Cirrus Logic CS4349")
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190815090157.70036-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/codecs/cs4349.c | 1 +
 1 file changed, 1 insertion(+)

diff --git a/sound/soc/codecs/cs4349.c b/sound/soc/codecs/cs4349.c
index bee0e343723f..f08d7a296c0c 100644
--- a/sound/soc/codecs/cs4349.c
+++ b/sound/soc/codecs/cs4349.c
@@ -381,6 +381,7 @@ static struct i2c_driver cs4349_i2c_driver = {
 	.driver = {
 		.name		= "cs4349",
 		.of_match_table	= cs4349_of_match,
+		.pm = &cs4349_runtime_pm,
 	},
 	.id_table	= cs4349_i2c_id,
 	.probe		= cs4349_i2c_probe,
-- 
2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.19 492/671] ASoC: wm8737: Fix copy-paste error in wm8737_snd_controls

[Sasha Levin]

From: YueHaibing <yuehaibing@huawei.com>

[ Upstream commit 554b75bde64bcad9662530726d1483f7ef012069 ]

sound/soc/codecs/wm8737.c:112:29: warning:
 high_3d defined but not used [-Wunused-const-variable=]

'high_3d' should be used for 3D High Cut-off.

Reported-by: Hulk Robot <hulkci@huawei.com>
Fixes: 2a9ae13a2641 ("ASoC: Add initial WM8737 driver")
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20190815091920.64480-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/codecs/wm8737.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c
index e9ae821e7609..568b77692f5f 100644
--- a/sound/soc/codecs/wm8737.c
+++ b/sound/soc/codecs/wm8737.c
@@ -170,7 +170,7 @@ SOC_DOUBLE("Polarity Invert Switch", WM8737_ADC_CONTROL, 5, 6, 1, 0),
 SOC_SINGLE("3D Switch", WM8737_3D_ENHANCE, 0, 1, 0),
 SOC_SINGLE("3D Depth", WM8737_3D_ENHANCE, 1, 15, 0),
 SOC_ENUM("3D Low Cut-off", low_3d),
-SOC_ENUM("3D High Cut-off", low_3d),
+SOC_ENUM("3D High Cut-off", high_3d),
 SOC_SINGLE_TLV("3D ADC Volume", WM8737_3D_ENHANCE, 7, 1, 1, adc_tlv),
 
 SOC_SINGLE("Noise Gate Switch", WM8737_NOISE_GATE, 0, 1, 0),
-- 
2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.19 500/671] ASoC: sun4i-i2s: RX and TX counter registers are swapped

[Sasha Levin]

From: Maxime Ripard <maxime.ripard@bootlin.com>

[ Upstream commit cf2c0e1ce9544df42170fb921f12da82dc0cc8d6 ]

The RX and TX counters registers offset have been swapped, fix that.

Fixes: fa7c0d13cb26 ("ASoC: sunxi: Add Allwinner A10 Digital Audio driver")
Signed-off-by: Maxime Ripard <maxime.ripard@bootlin.com>
Link: https://lore.kernel.org/r/8b26477560ad5fd8f69e037b167c5e61de5c26a3.1566242458.git-series.maxime.ripard@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/sunxi/sun4i-i2s.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index 18cf8404d27c..f248e563986c 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -80,8 +80,8 @@
 #define SUN4I_I2S_CLK_DIV_MCLK_MASK		GENMASK(3, 0)
 #define SUN4I_I2S_CLK_DIV_MCLK(mclk)			((mclk) << 0)
 
-#define SUN4I_I2S_RX_CNT_REG		0x28
-#define SUN4I_I2S_TX_CNT_REG		0x2c
+#define SUN4I_I2S_TX_CNT_REG		0x28
+#define SUN4I_I2S_RX_CNT_REG		0x2c
 
 #define SUN4I_I2S_TX_CHAN_SEL_REG	0x30
 #define SUN4I_I2S_CHAN_SEL(num_chan)		(((num_chan) - 1) << 0)
-- 
2.20.1

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