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

[alsa-devel] [PATCH AUTOSEL 4.9 181/251] ASoC: cs4349: Use PM ops 'cs4349_runtime_pm'
by Sasha Levin 16 Jan '20

16 Jan '20

From: YueHaibing <yuehaibing(a)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(a)huawei.com> Fixes: e40da86 ("ASoC: cs4349: Add support for Cirrus Logic CS4349") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> Link: https://lore.kernel.org/r/20190815090157.70036-1-yuehaibing@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)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 231ca935cdf3..c232c42ccead 100644 --- a/sound/soc/codecs/cs4349.c +++ b/sound/soc/codecs/cs4349.c @@ -380,6 +380,7 @@ 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 4.9 180/251] ASoC: es8328: Fix copy-paste error in es8328_right_line_controls
by Sasha Levin 16 Jan '20

16 Jan '20

From: YueHaibing <yuehaibing(a)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(a)huawei.com> Link: https://lore.kernel.org/r/20190815092300.68712-1-yuehaibing@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)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 37722194b107..6b22700842e2 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -234,7 +234,7 @@ 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 4.9 174/251] 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 a04edff8b729..ae50d59fb810 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 4.9 156/251] 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 624c209c9498..d1935c5c3602 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -882,14 +882,13 @@ 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 4.9 124/251] 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 d69559e45872..635b22fa1101 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -48,8 +48,8 @@ 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 4.9 122/251] 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 64fa1bbf0acb..54011f8543a7 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2626,7 +2626,9 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, (err = snd_usb_mixer_status_create(mixer)) < 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 4.9 082/251] 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 07f91e918b23..754742018515 100644 --- a/sound/soc/qcom/apq8016_sbc.c +++ b/sound/soc/qcom/apq8016_sbc.c @@ -114,13 +114,15 @@ 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; } link->codec_of_node = of_parse_phandle(codec, "sound-dai", 0); @@ -132,28 +134,37 @@ static struct apq8016_sbc_data *apq8016_sbc_parse_of(struct snd_soc_card *card) 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_name(codec, &link->codec_dai_name); if (ret) { 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 4.9 043/251] 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 8e525f7ac08d..3d99a8579c99 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -119,7 +119,8 @@ 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 4.9 006/251] ALSA: usb-audio: update quirk for B&W PX to remove microphone
by Sasha Levin 16 Jan '20

16 Jan '20

From: Nicolas Huaman <nicolas(a)herochao.de> [ Upstream commit c369c8db15d51fa175d2ba85928f79d16af6b562 ] A quirk in snd-usb-audio was added to automate setting sample rate to 4800k and remove the previously exposed nonfunctional microphone for the Bowers & Wilkins PX: commit 240a8af929c7c57dcde28682725b29cf8474e8e5 https://lore.kernel.org/patchwork/patch/919689/ However the headphones where updated shortly after that to remove the unintentional microphone functionality. I guess because of this the headphones now crash when connecting them via USB while the quirk is active. Dmesg: snd-usb-audio: probe of 2-3:1.0 failed with error -22 usb 2-3: 2:1: cannot get min/max values for control 2 (id 2) This patch removes the microfone and allows the headphones to connect and work out of the box. It is based on the current mainline kernel and successfully applied an tested on my machine (4.18.10.arch1-1). Fixes: 240a8af929c7 ("ALSA: usb-audio: Add a quirck for B&W PX headphones") Signed-off-by: Nicolas Huaman <nicolas(a)herochao.de> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/quirks-table.h | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index d32727c74a16..c892b4d1e733 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3293,19 +3293,14 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), .ifnum = 0, .type = QUIRK_AUDIO_STANDARD_MIXER, }, - /* Capture */ - { - .ifnum = 1, - .type = QUIRK_IGNORE_INTERFACE, - }, /* Playback */ { - .ifnum = 2, + .ifnum = 1, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = &(const struct audioformat) { .formats = SNDRV_PCM_FMTBIT_S16_LE, .channels = 2, - .iface = 2, + .iface = 1, .altsetting = 1, .altset_idx = 1, .attributes = UAC_EP_CS_ATTR_FILL_MAX | -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.9 003/251] ALSA: hda: fix unused variable warning
by Sasha Levin 16 Jan '20

16 Jan '20

From: Anders Roxell <anders.roxell(a)linaro.org> [ Upstream commit 5b03006d5c58ddd31caf542eef4d0269bcf265b3 ] When CONFIG_X86=n function azx_snoop doesn't use the variable chip it only returns true. sound/pci/hda/hda_intel.c: In function ‘dma_alloc_pages’: sound/pci/hda/hda_intel.c:2002:14: warning: unused variable ‘chip’ [-Wunused-variable] struct azx *chip = bus_to_azx(bus); ^~~~ Create a inline function of azx_snoop. Fixes: a41d122449be ("ALSA: hda - Embed bus into controller object") Signed-off-by: Anders Roxell <anders.roxell(a)linaro.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_controller.h | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index b83feecf1e40..8f9386998270 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -171,11 +171,10 @@ struct azx { #define azx_bus(chip) (&(chip)->bus.core) #define bus_to_azx(_bus) container_of(_bus, struct azx, bus.core) -#ifdef CONFIG_X86 -#define azx_snoop(chip) ((chip)->snoop) -#else -#define azx_snoop(chip) true -#endif +static inline bool azx_snoop(struct azx *chip) +{ + return !IS_ENABLED(CONFIG_X86) || chip->snoop; +} /* * macros for easy use -- 2.20.1

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