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

[alsa-devel] [PATCH v5 0/9] ASoC: rockchip: Parse dai links from dts
by Jeffy Chen 22 Aug '17

22 Aug '17

Currently we are using a fixed list of dai links in the driver. This serial of patches would let the driver parse dai links from dts, so that we can make some of them optional for future boards. Tested on my chromebook bob(with cros 4.4 kernel), it still works after disabled rt5514 codecs in the dts. Changes in v5: Allow searching for dai driver name in snd_soc_find_dai, instead of hacking non-zero device id to avoid legay dai naming. Changes in v4: Use non-zero drv id to avoid legacy dai naming instead of switching to snd_soc_register_codec. Changes in v3: Use compatible to match audio codecs -- Suggested-by Matthias Kaehlcke <mka(a)chromium.org> Changes in v2: Let rockchip,codec-names be a required property, because we plan to add more supported codecs to the fixed dai link list in the driver. Jeffy Chen (9): ASoC: rockchip: Remove obsolete dmic-delay ASoC: soc-core: Allow searching dai driver name in snd_soc_find_dai ASoC: rockchip: Use codec of_node and dai_name for rt5514 dsp arm64: dts: rockchip: Add rt5514 dsp for Gru arm64: dts: rockchip: Update rt5514 devices' compatible for Gru ASoC: rockchip: Parse dai links from dts ASoC: rockchip: Add support for DP codec ASoC: rockchip: Add support for DMIC codec dt-bindings: ASoC: rockchip: Update description of rockchip,codec .../bindings/sound/rockchip,rk3399-gru-sound.txt | 9 +- arch/arm64/boot/dts/rockchip/rk3399-gru.dtsi | 7 +- sound/soc/rockchip/Kconfig | 2 + sound/soc/rockchip/rk3399_gru_sound.c | 263 ++++++++++++++------- sound/soc/soc-core.c | 5 +- 5 files changed, 185 insertions(+), 101 deletions(-) -- 2.11.0

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[alsa-devel] [PATCH] ALSA: firewire: add const qualifier to identifiers for read-only symbols
by Takashi Sakamoto 22 Aug '17

22 Aug '17

Drivers in ALSA firewire stack still includes some symbols which can be moved to a section for read-only symbols. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/bebob/bebob.h | 2 +- sound/firewire/bebob/bebob_focusrite.c | 6 +++--- sound/firewire/bebob/bebob_maudio.c | 2 +- sound/firewire/bebob/bebob_terratec.c | 2 +- sound/firewire/bebob/bebob_yamaha_terratec.c | 2 +- sound/firewire/dice/dice.c | 2 +- sound/firewire/fireface/ff-protocol-ff400.c | 2 +- sound/firewire/fireface/ff.c | 2 +- sound/firewire/fireface/ff.h | 4 ++-- sound/firewire/fireworks/fireworks_proc.c | 2 +- sound/firewire/motu/motu.c | 6 +++--- sound/firewire/oxfw/oxfw-scs1x.c | 11 +++++------ sound/firewire/tascam/tascam.c | 2 +- 13 files changed, 22 insertions(+), 23 deletions(-) diff --git a/sound/firewire/bebob/bebob.h b/sound/firewire/bebob/bebob.h index 17678d6ab5a2..df1b1e94c43c 100644 --- a/sound/firewire/bebob/bebob.h +++ b/sound/firewire/bebob/bebob.h @@ -58,7 +58,7 @@ enum snd_bebob_clock_type { struct snd_bebob_clock_spec { unsigned int num; const char *const *labels; - enum snd_bebob_clock_type *types; + const enum snd_bebob_clock_type *types; int (*get)(struct snd_bebob *bebob, unsigned int *id); }; struct snd_bebob_rate_spec { diff --git a/sound/firewire/bebob/bebob_focusrite.c b/sound/firewire/bebob/bebob_focusrite.c index f11090057949..52b8b61ecddd 100644 --- a/sound/firewire/bebob/bebob_focusrite.c +++ b/sound/firewire/bebob/bebob_focusrite.c @@ -103,12 +103,12 @@ saffire_write_quad(struct snd_bebob *bebob, u64 offset, u32 value) &data, sizeof(__be32), 0); } -static enum snd_bebob_clock_type saffirepro_10_clk_src_types[] = { +static const enum snd_bebob_clock_type saffirepro_10_clk_src_types[] = { SND_BEBOB_CLOCK_TYPE_INTERNAL, SND_BEBOB_CLOCK_TYPE_EXTERNAL, /* S/PDIF */ SND_BEBOB_CLOCK_TYPE_EXTERNAL, /* Word Clock */ }; -static enum snd_bebob_clock_type saffirepro_26_clk_src_types[] = { +static const enum snd_bebob_clock_type saffirepro_26_clk_src_types[] = { SND_BEBOB_CLOCK_TYPE_INTERNAL, SND_BEBOB_CLOCK_TYPE_EXTERNAL, /* S/PDIF */ SND_BEBOB_CLOCK_TYPE_EXTERNAL, /* ADAT1 */ @@ -201,7 +201,7 @@ saffirepro_both_clk_src_get(struct snd_bebob *bebob, unsigned int *id) } const struct snd_bebob_spec saffire_le_spec; -static enum snd_bebob_clock_type saffire_both_clk_src_types[] = { +static const enum snd_bebob_clock_type saffire_both_clk_src_types[] = { SND_BEBOB_CLOCK_TYPE_INTERNAL, SND_BEBOB_CLOCK_TYPE_EXTERNAL, }; diff --git a/sound/firewire/bebob/bebob_maudio.c b/sound/firewire/bebob/bebob_maudio.c index d10208f92edf..bd55620c6a47 100644 --- a/sound/firewire/bebob/bebob_maudio.c +++ b/sound/firewire/bebob/bebob_maudio.c @@ -340,7 +340,7 @@ static int special_set_rate(struct snd_bebob *bebob, unsigned int rate) } /* Clock source control for special firmware */ -static enum snd_bebob_clock_type special_clk_types[] = { +static const enum snd_bebob_clock_type special_clk_types[] = { SND_BEBOB_CLOCK_TYPE_INTERNAL, /* With digital mute */ SND_BEBOB_CLOCK_TYPE_EXTERNAL, /* SPDIF/ADAT */ SND_BEBOB_CLOCK_TYPE_EXTERNAL, /* Word Clock */ diff --git a/sound/firewire/bebob/bebob_terratec.c b/sound/firewire/bebob/bebob_terratec.c index 2fdaf93e7a8d..9770c2127a7a 100644 --- a/sound/firewire/bebob/bebob_terratec.c +++ b/sound/firewire/bebob/bebob_terratec.c @@ -8,7 +8,7 @@ #include "./bebob.h" -static enum snd_bebob_clock_type phase88_rack_clk_src_types[] = { +static const enum snd_bebob_clock_type phase88_rack_clk_src_types[] = { SND_BEBOB_CLOCK_TYPE_INTERNAL, SND_BEBOB_CLOCK_TYPE_EXTERNAL, /* S/PDIF */ SND_BEBOB_CLOCK_TYPE_EXTERNAL, /* Word Clock */ diff --git a/sound/firewire/bebob/bebob_yamaha_terratec.c b/sound/firewire/bebob/bebob_yamaha_terratec.c index a6be3e7138e0..8bd78fef3516 100644 --- a/sound/firewire/bebob/bebob_yamaha_terratec.c +++ b/sound/firewire/bebob/bebob_yamaha_terratec.c @@ -31,7 +31,7 @@ * Yamaha GO 44 and GO 46. Yamaha and Terratec had cooperated for these models. */ -static enum snd_bebob_clock_type clk_src_types[] = { +static const enum snd_bebob_clock_type clk_src_types[] = { SND_BEBOB_CLOCK_TYPE_INTERNAL, SND_BEBOB_CLOCK_TYPE_EXTERNAL, /* S/PDIF */ }; diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index 25e9f77275c4..4ddb4cdd054b 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -26,7 +26,7 @@ MODULE_LICENSE("GPL v2"); */ static bool force_two_pcm_support(struct fw_unit *unit) { - const char *const models[] = { + static const char *const models[] = { /* TC Electronic models. */ "StudioKonnekt48", /* Focusrite models. */ diff --git a/sound/firewire/fireface/ff-protocol-ff400.c b/sound/firewire/fireface/ff-protocol-ff400.c index fcec6de80eeb..12aa15df435d 100644 --- a/sound/firewire/fireface/ff-protocol-ff400.c +++ b/sound/firewire/fireface/ff-protocol-ff400.c @@ -356,7 +356,7 @@ static void ff400_dump_clock_config(struct snd_ff *ff, snd_iprintf(buffer, "Sync to clock source: %s\n", src); } -struct snd_ff_protocol snd_ff_protocol_ff400 = { +const struct snd_ff_protocol snd_ff_protocol_ff400 = { .get_clock = ff400_get_clock, .begin_session = ff400_begin_session, .finish_session = ff400_finish_session, diff --git a/sound/firewire/fireface/ff.c b/sound/firewire/fireface/ff.c index eee7c8eac7a6..4974bc7980e9 100644 --- a/sound/firewire/fireface/ff.c +++ b/sound/firewire/fireface/ff.c @@ -157,7 +157,7 @@ static void snd_ff_remove(struct fw_unit *unit) } } -static struct snd_ff_spec spec_ff400 = { +static const struct snd_ff_spec spec_ff400 = { .name = "Fireface400", .pcm_capture_channels = {18, 14, 10}, .pcm_playback_channels = {18, 14, 10}, diff --git a/sound/firewire/fireface/ff.h b/sound/firewire/fireface/ff.h index 3cb812a50030..64df44beb950 100644 --- a/sound/firewire/fireface/ff.h +++ b/sound/firewire/fireface/ff.h @@ -47,7 +47,7 @@ struct snd_ff_spec { unsigned int midi_in_ports; unsigned int midi_out_ports; - struct snd_ff_protocol *protocol; + const struct snd_ff_protocol *protocol; }; struct snd_ff { @@ -112,7 +112,7 @@ struct snd_ff_protocol { u64 midi_rx_port_1_reg; }; -extern struct snd_ff_protocol snd_ff_protocol_ff400; +extern const struct snd_ff_protocol snd_ff_protocol_ff400; int snd_ff_transaction_register(struct snd_ff *ff); int snd_ff_transaction_reregister(struct snd_ff *ff); diff --git a/sound/firewire/fireworks/fireworks_proc.c b/sound/firewire/fireworks/fireworks_proc.c index beb0a0ffee57..9c21f31b8b21 100644 --- a/sound/firewire/fireworks/fireworks_proc.c +++ b/sound/firewire/fireworks/fireworks_proc.c @@ -12,7 +12,7 @@ static inline const char* get_phys_name(struct snd_efw_phys_grp *grp, bool input) { - const char *const ch_type[] = { + static const char *const ch_type[] = { "Analog", "S/PDIF", "ADAT", "S/PDIF or ADAT", "Mirroring", "Headphones", "I2S", "Guitar", "Pirzo Guitar", "Guitar String", }; diff --git a/sound/firewire/motu/motu.c b/sound/firewire/motu/motu.c index 2bbb764822fc..0d6b526105ab 100644 --- a/sound/firewire/motu/motu.c +++ b/sound/firewire/motu/motu.c @@ -194,7 +194,7 @@ static void motu_bus_update(struct fw_unit *unit) snd_motu_transaction_reregister(motu); } -static struct snd_motu_spec motu_828mk2 = { +static const struct snd_motu_spec motu_828mk2 = { .name = "828mk2", .protocol = &snd_motu_protocol_v2, .flags = SND_MOTU_SPEC_SUPPORT_CLOCK_X2 | @@ -208,7 +208,7 @@ static struct snd_motu_spec motu_828mk2 = { .analog_out_ports = 8, }; -static struct snd_motu_spec motu_828mk3 = { +static const struct snd_motu_spec motu_828mk3 = { .name = "828mk3", .protocol = &snd_motu_protocol_v3, .flags = SND_MOTU_SPEC_SUPPORT_CLOCK_X2 | @@ -225,7 +225,7 @@ static struct snd_motu_spec motu_828mk3 = { .analog_out_ports = 8, }; -static struct snd_motu_spec motu_audio_express = { +static const struct snd_motu_spec motu_audio_express = { .name = "AudioExpress", .protocol = &snd_motu_protocol_v3, .flags = SND_MOTU_SPEC_SUPPORT_CLOCK_X2 | diff --git a/sound/firewire/oxfw/oxfw-scs1x.c b/sound/firewire/oxfw/oxfw-scs1x.c index 93209ebd9121..02d595665898 100644 --- a/sound/firewire/oxfw/oxfw-scs1x.c +++ b/sound/firewire/oxfw/oxfw-scs1x.c @@ -297,12 +297,6 @@ static void midi_capture_trigger(struct snd_rawmidi_substream *stream, int up) } } -static const struct snd_rawmidi_ops midi_capture_ops = { - .open = midi_capture_open, - .close = midi_capture_close, - .trigger = midi_capture_trigger, -}; - static int midi_playback_open(struct snd_rawmidi_substream *stream) { return 0; @@ -363,6 +357,11 @@ void snd_oxfw_scs1x_update(struct snd_oxfw *oxfw) int snd_oxfw_scs1x_add(struct snd_oxfw *oxfw) { + static const struct snd_rawmidi_ops midi_capture_ops = { + .open = midi_capture_open, + .close = midi_capture_close, + .trigger = midi_capture_trigger, + }; static const struct snd_rawmidi_ops midi_playback_ops = { .open = midi_playback_open, .close = midi_playback_close, diff --git a/sound/firewire/tascam/tascam.c b/sound/firewire/tascam/tascam.c index 9dc93a7eb9da..44ad41fb7374 100644 --- a/sound/firewire/tascam/tascam.c +++ b/sound/firewire/tascam/tascam.c @@ -12,7 +12,7 @@ MODULE_DESCRIPTION("TASCAM FireWire series Driver"); MODULE_AUTHOR("Takashi Sakamoto <o-takashi(a)sakamocchi.jp>"); MODULE_LICENSE("GPL v2"); -static struct snd_tscm_spec model_specs[] = { +static const struct snd_tscm_spec model_specs[] = { { .name = "FW-1884", .has_adat = true, -- 2.11.0

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[alsa-devel] GPIO issue with RT5670
by Pierre-Louis Bossart 22 Aug '17

22 Aug '17

the commit f10e4bf6632b5be11cea875b66ba959833a69258 gpio: acpi: Even more tighten up ACPI GPIO lookups now generates the following issue [ 9.694204] rt5670 i2c-10EC5672:00: ASoC: Cannot get gpio at index 0: -2 [ 9.694293] rt5670 i2c-10EC5672:00: Adding jack GPIO failed This is the 3rd occurrence of an audio issue with this commit [1][2], what is the recommended fix here since this codec is not only used in ACPI-based devices and do we need to check for more issues in audio drivers? Thanks -Pierre [1] https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git/commit/?h… [2] https://bugzilla.kernel.org/show_bug.cgi?id=115531

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[alsa-devel] [PATCH 0/8] ASoC: jack: Manage gpios via devres
by Takashi Iwai 22 Aug '17

22 Aug '17

Hi, while debugging the recent GPIO regression with (still not-merged) cx2072x codec driver, I noticed the driver misses the snd_soc_jack_free_gpios() call. This patch set is rather making it redundant, change the ASoC jack code to manage gpios via devres. It follows the cleanups of superfluous snd_soc_jack_free_gpios() in each driver. This resulted in lots of reductions of remove callbacks. Takashi === Takashi Iwai (8): ASoC: jack: Manage gpios via devres ASoC: fsl: Remove superfluous snd_soc_jack_free_gpios() call ASoC: simple-card: Remove superfluous snd_soc_jack_free_gpios() call ASoC: intel: Remove superfluous snd_soc_jack_free_gpios() call ASoC: omap: Remove superfluous snd_soc_jack_free_gpios() call ASoC: pxa: Remove superfluous snd_soc_jack_free_gpios() call ASoC: samsung: Remove superfluous snd_soc_jack_free_gpios() call ASoC: tegra: Remove superfluous snd_soc_jack_free_gpios() call sound/soc/fsl/imx-es8328.c | 3 -- sound/soc/generic/simple-card.c | 10 ------- sound/soc/intel/boards/byt-max98090.c | 12 -------- sound/soc/omap/ams-delta.c | 10 ------- sound/soc/omap/omap-twl4030.c | 13 --------- sound/soc/omap/rx51.c | 9 ------ sound/soc/pxa/hx4700.c | 8 ------ sound/soc/samsung/h1940_uda1380.c | 9 ------ sound/soc/samsung/rx1950_uda1380.c | 10 ------- sound/soc/samsung/smartq_wm8987.c | 9 ------ sound/soc/soc-jack.c | 52 +++++++++++++++++++++++++++-------- sound/soc/tegra/tegra_alc5632.c | 13 --------- sound/soc/tegra/tegra_max98090.c | 18 ------------ sound/soc/tegra/tegra_rt5640.c | 13 --------- sound/soc/tegra/tegra_rt5677.c | 18 ------------ sound/soc/tegra/tegra_wm8903.c | 5 ---- 16 files changed, 41 insertions(+), 171 deletions(-) -- 2.14.0

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[alsa-devel] [bug] alsa-tools-1.1.3 compilation fails with clang due to wrong inline declaration.
by moxian 22 Aug '17

22 Aug '17

I've tried compiling alsa-tools-1.3.0 with clang, and the compilation fails with the following: $ CC=clang CXX=clang++ make ./as10k1 <...snip...> clang -fno-strict-aliasing -O2 -Wall -pipe -g -o as10k1 as10k1.o parse.o assemble.o macro.o as10k1.o: In function `header': /home/moxian/work/alsa-tools/alsa-tools-1.1.3/as10k1/as10k1.c:524: undefined reference to `output_tram_line' /home/moxian/work/alsa-tools/alsa-tools-1.1.3/as10k1/as10k1.c:527: undefined reference to `output_tram_line' /home/moxian/work/alsa-tools/alsa-tools-1.1.3/as10k1/as10k1.c:539: undefined reference to `output_tram_line' /home/moxian/work/alsa-tools/alsa-tools-1.1.3/as10k1/as10k1.c:543: undefined reference to `output_tram_line' clang-3.9: error: linker command failed with exit code 1 (use -v to see invocation) make[2]: *** [Makefile:320: as10k1] Error 1 make[2]: Leaving directory '/home/moxian/work/alsa-tools/alsa-tools-1.1.3/as10k1' make[1]: *** [Makefile:354: all-recursive] Error 1 make[1]: Leaving directory '/home/moxian/work/alsa-tools/alsa-tools-1.1.3/as10k1' make: *** [Makefile:9: all] Error 1 Clang devs claim that it is not a bug in clang (as I thought originally), but rather invalid code, which happens to compile with gcc by accident. This is due to `output_tram_line` function being declared as `inline void output_tram_line`. The intent was probably to declare it as `static inline void output_tram_line`. Please see https://bugs.llvm.org/show_bug.cgi?id=33768 for more details. Adding `static` to the function definition makes everything compile (at least everything that compiles with gcc already).

2 1

[alsa-devel] [PATCH] ASoC: Intel: Headset button support in kabylake machine driver
by Harsha Priya 22 Aug '17

22 Aug '17

This patch adds headset button support for kabylake machine driver (kbl_rt5663_rt5514_max98927). Signed-off-by: Hsin-yu Chao <hychao(a)google.com> Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> --- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 441a397..b350aa7 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -18,6 +18,7 @@ * GNU General Public License for more details. */ +#include <linux/input.h> #include <linux/module.h> #include <linux/platform_device.h> #include <sound/core.h> @@ -174,6 +175,7 @@ static int kabylake_rt5663_codec_init(struct snd_soc_pcm_runtime *rtd) int ret; struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_jack *jack; /* * Headset buttons map to the google Reference headset. @@ -188,6 +190,12 @@ static int kabylake_rt5663_codec_init(struct snd_soc_pcm_runtime *rtd) return ret; } + jack = &ctx->kabylake_headset; + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + rt5663_set_jack_detect(codec, &ctx->kabylake_headset); ret = snd_soc_dapm_ignore_suspend(&rtd->card->dapm, "DMIC"); -- 1.9.1

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[alsa-devel] [PATCH] alsa-lib: topology: Fix private data for BEs
by Liam Girdwood 22 Aug '17

22 Aug '17

Private data was not being added to BEs. Fix this. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- src/topology/data.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/src/topology/data.c b/src/topology/data.c index 5b83b9f..f96ff9b 100644 --- a/src/topology/data.c +++ b/src/topology/data.c @@ -50,7 +50,9 @@ struct snd_soc_tplg_private *get_priv_data(struct tplg_elem *elem) case SND_TPLG_TYPE_DAI: priv = &elem->dai->priv; break; - + case SND_TPLG_TYPE_BE: + priv = &elem->link->priv; + break; default: SNDERR("error: '%s': no support for private data for type %d\n", elem->id, elem->type); -- 1.9.1

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Re: [alsa-devel] [PATCH] ASoC: dapm: Avoid creating kcontrol for params
by anish kumar 22 Aug '17

22 Aug '17

> > When num_params is defined as one in case of > codec to codec dai link, there is no point in > allocating memory for kcontrol. > > Signed-off-by: anish kumar <yesanishhere(a)gmail.com> > --- > include/sound/soc-dapm.h | 1 - > sound/soc/soc-dapm.c | 147 ++++++++++++++++++++++++----------------------- >>This is a very big change and that commit log doesn't describe it in any >>detail or really motivate it. What exactly does this do and why? Currently in codec to codec dai link if there are multiple params defined then dapm can use created kcontrol to decide which param to apply at runtime. However, in case there is only single param configuration then there is no point in creating the kcontrol (this is currently happening) and also there is no point in allocating memory for kcontrol. In the snd_soc_dapm_new_pcm function, there is memory allocation happening for kcontrol which is later used or not used based on num_param. It is better to not allocate memory when there is only a single configuration. This change is to remedy that anomaly and also it remove unnecessary checks.

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[alsa-devel] [PATCH] ASoC: rt5670: Fix GPIO headset detection regression
by Takashi Iwai 22 Aug '17

22 Aug '17

RT5670 codec driver and its machine driver for Intel CHT assume the implicit GPIO mapping on the index 0 while BIOS on most devices don't provide it. The recent commit f10e4bf6632b ("gpio: acpi: Even more tighten up ACPI GPIO lookups") restricts such cases and it resulted in a regression where the headset jack setup fails like: rt5670 i2c-10EC5672:00: ASoC: Cannot get gpio at index 0: -2 rt5670 i2c-10EC5672:00: Adding jack GPIO failed For fixing this, we need to provide the GPIO mapping explicitly in the machine driver. Also this patch corrects the string to be passed to gpiolib to match with the pre-given mapping, too. Fixes: f10e4bf6632b ("gpio: acpi: Even more tighten up ACPI GPIO lookups") Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=115531 Reported-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Tested-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/codecs/rt5670.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 10 ++++++++++ 2 files changed, 11 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 0ec7985ed306..054b613cb0d0 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -567,7 +567,7 @@ int rt5670_set_jack_detect(struct snd_soc_codec *codec, rt5670->jack = jack; rt5670->hp_gpio.gpiod_dev = codec->dev; - rt5670->hp_gpio.name = "headphone detect"; + rt5670->hp_gpio.name = "headset"; rt5670->hp_gpio.report = SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2; rt5670->hp_gpio.debounce_time = 150; diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index bc2a52de06a3..f597d5582223 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -184,6 +184,13 @@ static int cht_aif1_hw_params(struct snd_pcm_substream *substream, return 0; } +static const struct acpi_gpio_params headset_gpios = { 0, 0, false }; + +static const struct acpi_gpio_mapping cht_rt5672_gpios[] = { + { "headset-gpios", &headset_gpios, 1 }, + {}, +}; + static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) { int ret; @@ -191,6 +198,9 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) struct snd_soc_codec *codec = codec_dai->codec; struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card); + if (devm_acpi_dev_add_driver_gpios(codec->dev, cht_rt5672_gpios)) + dev_warn(runtime->dev, "Unable to add GPIO mapping table\n"); + /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0xF, 4, 24); if (ret < 0) { -- 2.14.0

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[alsa-devel] [PATCH] ALSA: core: Fix unexpected error at replacing user TLV
by Takashi Iwai 22 Aug '17

22 Aug '17

When user tries to replace the user-defined control TLV, the kernel checks the change of its content via memcmp(). The problem is that the kernel passes the return value from memcmp() as is. memcmp() gives a non-zero negative value depending on the comparison result, and this shall be recognized as an error code. The patch covers that corner-case, return 1 properly for the changed TLV. Fixes: 8aa9b586e420 ("[ALSA] Control API - more robust TLV implementation") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/control.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/control.c b/sound/core/control.c index 3c6be1452e35..4525e127afd9 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1137,7 +1137,7 @@ static int snd_ctl_elem_user_tlv(struct snd_kcontrol *kcontrol, mutex_lock(&ue->card->user_ctl_lock); change = ue->tlv_data_size != size; if (!change) - change = memcmp(ue->tlv_data, new_data, size); + change = memcmp(ue->tlv_data, new_data, size) != 0; kfree(ue->tlv_data); ue->tlv_data = new_data; ue->tlv_data_size = size; -- 2.14.0

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