August 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] alsa settings for sound device Intel cherry trail chipset
by fulgor fulgor 16 Aug '17

16 Aug '17

hello from europe I can provide logging informations http://www.alsa-project.org/db/?f=6fb68d0201f421b41fe65677fde2ec5b1c9d4429 Thanks you -- Lionel Van Winnendaele

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[alsa-devel] [PATCH v4] ASoC: samsung: i2s: Null pointer dereference on samsung_i2s_remove
by Anton Vasilyev 15 Aug '17

15 Aug '17

If (quirks & QUIRK_SEC_DAI == 0) then samsung_i2s_probe() doesn't allocate sec_dai and pri_dai->sec_dai remains Null, but samsung_i2s_remove() performs pri_dai->sec_dai dereference in any case. The patch removes useless reinitialization of sec_dai at samsung_i2s_remove(), because resources are under devm control. Found by Linux Driver Verification project (linuxtesting.org). Signed-off-by: Anton Vasilyev <vasilyev(a)ispras.ru> --- v2: Drop initialization of sec_dai by NULL at samsung_i2s_remove as Sylwester Nawrocki suggest. --- v3: Fix typo in the comment --- v4: Change comment based on Krzysztof review. --- sound/soc/samsung/i2s.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index af3ba4d..6f896e3 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1376,13 +1376,9 @@ static int samsung_i2s_probe(struct platform_device *pdev) static int samsung_i2s_remove(struct platform_device *pdev) { - struct i2s_dai *pri_dai, *sec_dai; + struct i2s_dai *pri_dai; pri_dai = dev_get_drvdata(&pdev->dev); - sec_dai = pri_dai->sec_dai; - - pri_dai->sec_dai = NULL; - sec_dai->pri_dai = NULL; pm_runtime_get_sync(&pdev->dev); pm_runtime_disable(&pdev->dev); -- 2.7.4

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[alsa-devel] Applied "ASoC: Freescale: Delete an error message for a failed memory allocation in three functions" to the asoc tree
by Mark Brown 15 Aug '17

15 Aug '17

The patch ASoC: Freescale: Delete an error message for a failed memory allocation in three functions has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 443be77e4b09d98c28196dca7132a1191ecf742b Mon Sep 17 00:00:00 2001 From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Fri, 11 Aug 2017 15:10:43 +0200 Subject: [PATCH] ASoC: Freescale: Delete an error message for a failed memory allocation in three functions Omit an extra message for a memory allocation failure in these functions. This issue was detected by using the Coccinelle software. Link: http://events.linuxfoundation.org/sites/events/files/slides/LCJ16-Refactor_… Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/fsl/fsl_asrc_dma.c | 4 +--- sound/soc/fsl/fsl_dma.c | 1 - sound/soc/fsl/fsl_ssi.c | 4 +--- 3 files changed, 2 insertions(+), 7 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index 282d841840b1..2baf19608bd0 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -279,10 +279,8 @@ static int fsl_asrc_dma_startup(struct snd_pcm_substream *substream) struct fsl_asrc_pair *pair; pair = kzalloc(sizeof(struct fsl_asrc_pair), GFP_KERNEL); - if (!pair) { - dev_err(dev, "failed to allocate pair\n"); + if (!pair) return -ENOMEM; - } pair->asrc_priv = asrc_priv; diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 979ed1b3ecc4..c17359648915 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -905,7 +905,6 @@ static int fsl_soc_dma_probe(struct platform_device *pdev) dma = kzalloc(sizeof(*dma) + strlen(np->full_name), GFP_KERNEL); if (!dma) { - dev_err(&pdev->dev, "could not allocate dma object\n"); of_node_put(ssi_np); return -ENOMEM; } diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 173cb8496641..64598d1183f8 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1432,10 +1432,8 @@ static int fsl_ssi_probe(struct platform_device *pdev) ssi_private = devm_kzalloc(&pdev->dev, sizeof(*ssi_private), GFP_KERNEL); - if (!ssi_private) { - dev_err(&pdev->dev, "could not allocate DAI object\n"); + if (!ssi_private) return -ENOMEM; - } ssi_private->soc = of_id->data; ssi_private->dev = &pdev->dev; -- 2.13.3

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[alsa-devel] [PATCH 0/5] constify snd_soc_dai_ops structures
by Julia Lawall 15 Aug '17

15 Aug '17

These snd_soc_dai_ops structures are only stored in the ops field of a snd_soc_dai_driver structure, which is const. Thus, the snd_soc_dai_ops structures can be const too. Done with the help of Coccinelle. --- sound/soc/blackfin/bf6xx-i2s.c | 2 +- sound/soc/codecs/es8316.c | 2 +- sound/soc/codecs/zx_aud96p22.c | 2 +- sound/soc/rockchip/rockchip_pdm.c | 2 +- sound/soc/spear/spdif_in.c | 2 +- 5 files changed, 5 insertions(+), 5 deletions(-)

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[alsa-devel] [PATCH] ASoC: spear: spdif_in: constify snd_soc_dai_ops structure
by Gustavo A. R. Silva 15 Aug '17

15 Aug '17

This structure is only stored in the ops field of a snd_soc_dai_driver structure. That field is declared const, so snd_soc_dai_ops structures that have this property can be declared as const also. Signed-off-by: Gustavo A. R. Silva <garsilva(a)embeddedor.com> --- sound/soc/spear/spdif_in.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/spear/spdif_in.c b/sound/soc/spear/spdif_in.c index 7f32527..d9368e1 100644 --- a/sound/soc/spear/spdif_in.c +++ b/sound/soc/spear/spdif_in.c @@ -151,7 +151,7 @@ static int spdif_in_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } -static struct snd_soc_dai_ops spdif_in_dai_ops = { +static const struct snd_soc_dai_ops spdif_in_dai_ops = { .shutdown = spdif_in_shutdown, .trigger = spdif_in_trigger, .hw_params = spdif_in_hw_params, -- 2.5.0

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[alsa-devel] [PATCH] arizona: anc: Correct setting of FCx_MIC_MODE_SEL
by Charles Keepax 15 Aug '17

15 Aug '17

From: Sapthagiri Baratam <sapthagiri.baratam(a)cirrus.com> The mask rather than the shift is accidentally passed to the SOC_ENUM_SINGLE macro which results in the wrong bits being set for this control, change to using the shift to correct this. Signed-off-by: Sapthagiri Baratam <sapthagiri.baratam(a)cirrus.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/arizona.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 0a734d910850..a1149f6a8450 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -710,7 +710,7 @@ const struct soc_enum arizona_anc_input_src[] = { ARRAY_SIZE(arizona_anc_input_src_text), arizona_anc_input_src_text), SOC_ENUM_SINGLE(ARIZONA_FCL_ADC_REFORMATTER_CONTROL, - ARIZONA_FCL_MIC_MODE_SEL, + ARIZONA_FCL_MIC_MODE_SEL_SHIFT, ARRAY_SIZE(arizona_anc_channel_src_text), arizona_anc_channel_src_text), SOC_ENUM_SINGLE(ARIZONA_ANC_SRC, @@ -718,7 +718,7 @@ const struct soc_enum arizona_anc_input_src[] = { ARRAY_SIZE(arizona_anc_input_src_text), arizona_anc_input_src_text), SOC_ENUM_SINGLE(ARIZONA_FCR_ADC_REFORMATTER_CONTROL, - ARIZONA_FCR_MIC_MODE_SEL, + ARIZONA_FCR_MIC_MODE_SEL_SHIFT, ARRAY_SIZE(arizona_anc_channel_src_text), arizona_anc_channel_src_text), }; -- 2.11.0

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[alsa-devel] [PATCH 3/3] ALSA: firewire-motu: constify snd_rawmidi_ops structures
by Julia Lawall 15 Aug '17

15 Aug '17

These snd_rawmidi_ops structures are only passed as the third argument of snd_rawmidi_set_ops. This argument is const, so the snd_rawmidi_ops structures can be const too. Done with the help of Coccinelle. Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> --- sound/firewire/motu/motu-midi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/firewire/motu/motu-midi.c b/sound/firewire/motu/motu-midi.c index e3acfcc..e55cab6 100644 --- a/sound/firewire/motu/motu-midi.c +++ b/sound/firewire/motu/motu-midi.c @@ -128,12 +128,12 @@ static void set_midi_substream_names(struct snd_motu *motu, int snd_motu_create_midi_devices(struct snd_motu *motu) { - static struct snd_rawmidi_ops capture_ops = { + static const struct snd_rawmidi_ops capture_ops = { .open = midi_capture_open, .close = midi_capture_close, .trigger = midi_capture_trigger, }; - static struct snd_rawmidi_ops playback_ops = { + static const struct snd_rawmidi_ops playback_ops = { .open = midi_playback_open, .close = midi_playback_close, .trigger = midi_playback_trigger,

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[alsa-devel] [PATCH 0/3] constify snd_rawmidi_ops structures
by Julia Lawall 15 Aug '17

15 Aug '17

These snd_rawmidi_ops structures are only passed as the third argument of snd_rawmidi_set_ops. This argument is const, so the snd_rawmidi_ops structures can be const too. Done with the help of Coccinelle. --- drivers/hid/hid-prodikeys.c | 2 +- drivers/usb/gadget/function/f_midi.c | 4 ++-- sound/firewire/motu/motu-midi.c | 4 ++-- 3 files changed, 5 insertions(+), 5 deletions(-)

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[alsa-devel] Headphone and headset jack control naming
by Tanu Kaskinen 15 Aug '17

15 Aug '17

A user reported various audio problems with the Alienware 14 laptop in the PulseAudio bugzilla. One of the problems is that the jack control names that alsa reports are incorrect. However, I'm not sure what the correct names would be, so I'd like to get that clarified (I will then file a proper bug report in the kernel bugzilla). The laptop has three physical jacks: headphone, headset and microphone. In general, headset jacks have been so far represented by two jack controls: "Headphone" + "Headset Mic" (or is "Headphone" + "Mic" more common? I think I've seen both being used). However, in this case there's also a headphone-only jack. Should we invent a new name for headset outputs, so that the headphone and headset outputs can be distinguished, or should there be two "Headphone" controls, or should one of the controls (which one?) use the "Headphone2" name that is already used by some hardware? FWIW, currently the output jack control names on Alienware 14 are "Headphone Front" for the headset jack and "Headphone Surround" for the headphone jack, which I assume we all agree is incorrect. Furthermore, there's no "Headset Mic" jack control, or any other control corresponding to the headset mic. "Mic" control exists, but it corresponds to the separate microphone jack. -- Tanu https://www.patreon.com/tanuk

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[alsa-devel] [PATCH v3 00/11] ASoC: Add I2S support for Allwinner H3 SoCs
by codekipper＠gmail.com 15 Aug '17

15 Aug '17

From: Marcus Cooper <codekipper(a)gmail.com> Hi All, please find attached a series of patches to bring i2s support to the Allwinner H3 SoC. This has been tested with the following setups: A20 Olimex EVB connected to a pcm5102 Orange Pi 2 connected to a uda1380 Orange Pi 2 hdmi audio playback Pine 64 connected to the audio DAC board To get i2s working some additional patches are required which will be delivered later. For now they have been pushed here https://github.com/codekipper/linux-sunxi/commits/sunxi-audio-h3 I don't own a A33 device which uses the i2s block for the audio codec so if someone could test against that it would be much appreciated. I'm also wondering if there is a preferred way of setting the lrclk size in the dts?..currently it is set to the sample width but for example the pcm5102a wants it to be 32 bits whatever the sample rate. Thanks in advance, CK --- v4 changes compared to v3 are: - moved clkdiv variant adjustment out of function - used PTR_ERR_OR_ZERO for checks - tidy up of extra lines and lines over 80 chars. - reduced names of polarity, wss and sr reg fields. - added reviewed-by to commit messages - added comments for functionality that hasn't been implemented yet. v3 changes compared to v2 are: - initial changes to prepare driver for newer SoCs has been broken down into smaller patches - reduce use of regmap fields to where just needed. - clkdiv expansion will be delivered later. - defines for H3 variant segregated. - fixed regmap config issue with SUN8I_I2S_FIFO_TX_REG. v2 changes compared to v1 are: - massive refactoring to remove duplicate code making use of regmap_fields. - extending the clock divisors. - removed code that should be delivered when we support 20/24bits --- Marcus Cooper (11): ASoC: sun4i-i2s: Add clkdiv offsets to quirks ASoC: sun4i-i2s: Add regmap config to quirks ASoC: sun4i-i2s: Add TX FIFO offset to quirks ASoC: sun4i-i2s: Add regmap fields for channels ASoC: sun4i-i2s: Add regfields for word size select and sample resolution ASoC: sun4i-i2s: bclk and lrclk polarity tidyup ASoC: sun4i-i2s: Add mclk enable regmap field ASoC: sun4i-i2s: Add regmap field to set DAI format ASoC: sun4i-i2s: Check for slave select bit ASoC: sun4i-i2s: Update global enable with bitmask ASoC: sun4i-i2s: Add support for H3 .../devicetree/bindings/sound/sun4i-i2s.txt | 2 + sound/soc/sunxi/sun4i-i2s.c | 444 ++++++++++++++++++--- 2 files changed, 391 insertions(+), 55 deletions(-) -- 2.14.1

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