June 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] *** ASoC: core: refine of node id parse of cpu/codec dai ***
by Qiao Zhou 20 Jun '14

20 Jun '14

Hi Mark, Liam This patch is to add another check besides cpu_dai/codec_dai name during dai_link bind. currently if the cpu_dai/codec_dai name match corresponding dai_link cpu/codec name, then a match is found. in this patch, it also checks whether cpu/codec dai id match dailink cpu_dai_id/codec_dai_id. Still check name first. 1. if it doesn't match, it will keep checking whether cpu_id/codec_id match corresponding dai_link cpu_dai_id/codec_dai_id. if the ids are equal, then a match is found. 2. if it does match, then a match is already found. no need to further check. By this way, it's convenient to maintain dai_link in DT files, and machine driver can parse related node point to get dai id info. Then there is no need to parse cpu_dai/codec_dai name for each dai. Like below: In DT files: sound { compatible = "sound card name xx"; dai_link_1:dai_link_1 { compatible = "xx-dailink-1"; cpu-dai = <&cpu-dai-node CPU_DAI_ID_1>; codec-dai = <&codec-dai-node CODEC_DAI_ID_1>; ...... } dai_link_2:dai_link_2 { compatible = "xx-dailink-2"; cpu-dai = <&cpu-dai-node CPU_DAI_ID_2>; codec-dai = <&codec-dai-node CODEC_DAI_ID_2>; ...... } ...... } In machine driver: static xx_sound_card_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; i = 0; for_each_child_of_node(np, dailink_np) { of_parse_phandle_with_args(dailink_np, "cpu-dai", "#dailink-cells", 0, &out_args); dailink[i].cpu_of_node = out_args.np; dailink[i].cpu_dai_id = out_args.args[0]; dailink[i].platform_of_node = out_args.np; of_parse_phandle_with_args(dailink_np, "codec-dai", "#dailink-cells", 0, &out_args); dailink[i].codec_of_node = out_args.np; dailink[i].codec_dai_id = out_args.args[0]; ...... i++ } ...... ret = snd_soc_register_card(card); ...... } If some drivers don't have DT support, then we have no choice but to add correct cpu/codec(dai) name in DT and parse name info in machine driver. Could you help share your opinions? Thanks in advance. Qiao Zhou (1): ASoC: core: refine of node id parse of cpu/codec dai include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 32 +++++++++++++++++++++++--------- 2 files changed, 25 insertions(+), 9 deletions(-) -- 1.7.9.5

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[alsa-devel] [PATCH] ALSA: hda - Fix usage of "model" module parameter
by David Henningsson 19 Jun '14

19 Jun '14

A recent refactoring broke the possibility to manually specify model name as a module parameter. This patch restores the desired functionality. Reported-by: Kent Baxley <kent.baxley(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_auto_parser.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index b684c6e..dabe419 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -898,6 +898,7 @@ void snd_hda_pick_fixup(struct hda_codec *codec, if (!strcmp(codec->modelname, models->name)) { codec->fixup_id = models->id; codec->fixup_name = models->name; + codec->fixup_list = fixlist; codec->fixup_forced = 1; return; } -- 1.9.1

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[alsa-devel] [PATCH v4 0/3] Add ES8328 audio codec
by Sean Cross 19 Jun '14

19 Jun '14

This patchset adds support for the Everest Semi ES8328 audio codec. It also adds support for using the es8328 on IMX boards. We write a machine driver rather than using simple-card because the machine driver needs to support regulators for the codec itself and speaker amps, have headphone support, and must supply an unusual frequency to the codec chip. Changes since v3: - Add OF bindings for everest,es8328 - Add back in patch to add everest as a vendor prefix - Give clocks more generic names - Document required clocks in DT bindings document Changes since v2: - Change deemph to a bool, and update based on sample rate - Replace value enums with regular enums - Break SPI and I2C drivers into their own modules - Rename everest,es8328 to es8328 as the module wasn't getting detected. Because of this, the patch to vendor-prefixes.txt has been dropped. Changes since v1: - Rename HP/Speaker to OUT1/OUT2 - Use DAPM widgets for startup/shutdown - Add regulator support to machine driver Sean Cross (3): devicetree: bindings: Add Everest Semicodunctor ASoC: add es8328 codec driver ASoC: fsl: add imx-es8328 machine driver Documentation/devicetree/bindings/sound/es8328.txt | 18 + .../devicetree/bindings/sound/imx-audio-es8328.txt | 68 +++ .../devicetree/bindings/vendor-prefixes.txt | 1 + sound/soc/codecs/Kconfig | 13 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/es8328-i2c.c | 62 ++ sound/soc/codecs/es8328-spi.c | 51 ++ sound/soc/codecs/es8328.c | 625 +++++++++++++++++++++ sound/soc/codecs/es8328.h | 313 +++++++++++ sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-es8328.c | 359 ++++++++++++ 12 files changed, 1532 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/es8328.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-es8328.txt create mode 100644 sound/soc/codecs/es8328-i2c.c create mode 100644 sound/soc/codecs/es8328-spi.c create mode 100644 sound/soc/codecs/es8328.c create mode 100644 sound/soc/codecs/es8328.h create mode 100644 sound/soc/fsl/imx-es8328.c -- 2.0.0

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[alsa-devel] [PATCH] ASoC: max98090: Remove needless defines and line feeds
by Jarkko Nikula 19 Jun '14

19 Jun '14

Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- sound/soc/codecs/max98090.c | 13 ------------- 1 file changed, 13 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index d97f1ce7ff7d..6ea0fdd42eb5 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -26,10 +26,6 @@ #include <sound/max98090.h> #include "max98090.h" -#define DEBUG -#define EXTMIC_METHOD -#define EXTMIC_METHOD_TEST - /* Allows for sparsely populated register maps */ static struct reg_default max98090_reg[] = { { 0x00, 0x00 }, /* 00 Software Reset */ @@ -820,7 +816,6 @@ static int max98090_micinput_event(struct snd_soc_dapm_widget *w, else val = (val & M98090_MIC_PA2EN_MASK) >> M98090_MIC_PA2EN_SHIFT; - if (val >= 1) { if (w->reg == M98090_REG_MIC1_INPUT_LEVEL) { max98090->pa1en = val - 1; /* Update for volatile */ @@ -1140,7 +1135,6 @@ static const struct snd_kcontrol_new max98090_mixhprsel_mux = SOC_DAPM_ENUM("MIXHPRSEL Mux", mixhprsel_mux_enum); static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = { - SND_SOC_DAPM_INPUT("MIC1"), SND_SOC_DAPM_INPUT("MIC2"), SND_SOC_DAPM_INPUT("DMICL"), @@ -1304,7 +1298,6 @@ static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = { }; static const struct snd_soc_dapm_widget max98091_dapm_widgets[] = { - SND_SOC_DAPM_INPUT("DMIC3"), SND_SOC_DAPM_INPUT("DMIC4"), @@ -1315,7 +1308,6 @@ static const struct snd_soc_dapm_widget max98091_dapm_widgets[] = { }; static const struct snd_soc_dapm_route max98090_dapm_routes[] = { - {"MIC1 Input", NULL, "MIC1"}, {"MIC2 Input", NULL, "MIC2"}, @@ -1493,17 +1485,14 @@ static const struct snd_soc_dapm_route max98090_dapm_routes[] = { {"SPKR", NULL, "SPK Right Out"}, {"RCVL", NULL, "RCV Left Out"}, {"RCVR", NULL, "RCV Right Out"}, - }; static const struct snd_soc_dapm_route max98091_dapm_routes[] = { - /* DMIC inputs */ {"DMIC3", NULL, "DMIC3_ENA"}, {"DMIC4", NULL, "DMIC4_ENA"}, {"DMIC3", NULL, "AHPF"}, {"DMIC4", NULL, "AHPF"}, - }; static int max98090_add_widgets(struct snd_soc_codec *codec) @@ -1531,7 +1520,6 @@ static int max98090_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(dapm, max98091_dapm_routes, ARRAY_SIZE(max98091_dapm_routes)); - } return 0; @@ -2221,7 +2209,6 @@ static void max98090_handle_pdata(struct snd_soc_codec *codec) dev_err(codec->dev, "No platform data\n"); return; } - } static int max98090_probe(struct snd_soc_codec *codec) -- 2.0.0

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[alsa-devel] [PATCH] ASoC: max98090: Fix missing free_irq
by Jarkko Nikula 19 Jun '14

19 Jun '14

max98090.c doesn't free the threaded interrupt it requests. This causes an oops when doing "cat /proc/interrupts" after snd-soc-max98090.ko is unloaded. Fix this by requesting the interrupt by using devm_request_threaded_irq(). Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> Cc: Stable <stable(a)vger.kernel.org> # 3.10+ --- This issue has been in max98090.c since v3.8 commit 685e42154dcf ("ASoC: Replace max98090 Device Driver"). However this patch applies only after v3.10 which has 3d15aacbb802 ("ASoC: max98090: request IRQF_ONESHOT interrupt"). I'm not sure is there need to go before that since first upstream max98090 user was introduced in a v3.14 commit 7637af2e17f1 ("ASoC: tegra: add tegra+MAX98090 machine driver"). However I'm aware there is an out-of-tree 3.10 user. --- sound/soc/codecs/max98090.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index f5fccc7a8e89..d97f1ce7ff7d 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2284,7 +2284,7 @@ static int max98090_probe(struct snd_soc_codec *codec) /* Register for interrupts */ dev_dbg(codec->dev, "irq = %d\n", max98090->irq); - ret = request_threaded_irq(max98090->irq, NULL, + ret = devm_request_threaded_irq(codec->dev, max98090->irq, NULL, max98090_interrupt, IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "max98090_interrupt", codec); if (ret < 0) { -- 2.0.0

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[alsa-devel] [PATCH] ASoC: wm5100/wm8903/wm8996: Replace open-coded snd_soc_dapm_to_codec()
by Lars-Peter Clausen 19 Jun '14

19 Jun '14

We now have a generic helper function to cast from a DAPM context to a CODEC. Make use of it in the places which previously open-coded it. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/wm5100.c | 3 +-- sound/soc/codecs/wm8903.c | 3 +-- sound/soc/codecs/wm8996.c | 3 +-- 3 files changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 91a9ea2..7bb0d36 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -735,8 +735,7 @@ WM5100_MIXER_CONTROLS("LHPF4", WM5100_HPLP4MIX_INPUT_1_SOURCE), static void wm5100_seq_notifier(struct snd_soc_dapm_context *dapm, enum snd_soc_dapm_type event, int subseq) { - struct snd_soc_codec *codec = container_of(dapm, - struct snd_soc_codec, dapm); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); u16 val, expect, i; diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index b84940c..ec3250d 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -281,8 +281,7 @@ static int wm8903_dcs_event(struct snd_soc_dapm_widget *w, static void wm8903_seq_notifier(struct snd_soc_dapm_context *dapm, enum snd_soc_dapm_type event, int subseq) { - struct snd_soc_codec *codec = container_of(dapm, - struct snd_soc_codec, dapm); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); int dcs_mode = WM8903_DCS_MODE_WRITE_STOP; int i, val; diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 6926633..9304a91 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -690,8 +690,7 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, u16 mask) static void wm8996_seq_notifier(struct snd_soc_dapm_context *dapm, enum snd_soc_dapm_type event, int subseq) { - struct snd_soc_codec *codec = container_of(dapm, - struct snd_soc_codec, dapm); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); u16 val, mask; -- 1.8.0

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[alsa-devel] [PATCH 1/2] tinycompress: update compress headers for sample rate field
by Vinod Koul 19 Jun '14

19 Jun '14

commit "929559b: ALSA: compress: add num_sample_rates in snd_codec_desc" adds array for passing sample rate instead of bit map. Update the local header as well Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- include/sound/compress_params.h | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/include/sound/compress_params.h b/include/sound/compress_params.h index deda425..378417a 100644 --- a/include/sound/compress_params.h +++ b/include/sound/compress_params.h @@ -23,6 +23,7 @@ #define MAX_NUM_CODECS 32 #define MAX_NUM_CODEC_DESCRIPTORS 32 #define MAX_NUM_BITRATES 32 +#define MAX_NUM_SAMPLE_RATES 32 #define SND_AUDIOCODEC_PCM ((__u32) 0x00000001) #define SND_AUDIOCODEC_MP3 ((__u32) 0x00000002) @@ -209,7 +210,8 @@ union snd_codec_options { struct snd_codec_desc { __u32 max_ch; - __u32 sample_rates; + __u32 sample_rates[MAX_NUM_SAMPLE_RATES]; + __u32 num_sample_rates; __u32 bit_rate[MAX_NUM_BITRATES]; __u32 num_bitrates; __u32 rate_control; -- 1.7.0.4

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[alsa-devel] Enable HP machine to enable capture mute led and output mute led
by Kailang 19 Jun '14

19 Jun '14

Hi Takashi, Attach was HP mute led support patch. It use Line1 vref and Mic1 vref to control mute led for ALC282. Line1 vref to control output mute led. Mic1 vref to control input mute led. It use GPIO3 and Mic1 vref to control mute led for ALC280. GPIO3 to control output mute led. Mic1 vref to control input mute led. For the DMI string for output mute led application. I already request from HP to support it in next release. HP_Mute_LED_0_A => use Mic1 vref to control mute led. Is the string right? I have tested upper string for Mic1 mute led. BR, Kailang

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Re: [alsa-devel] [PATCH v11 2/8] ASoC: core: Generic ac97 link reset functions
by Markus Pargmann 18 Jun '14

18 Jun '14

Hi Silvio, On Wed, Jun 18, 2014 at 08:42:34PM +0200, S. Fricke wrote: > > Hi Markus, > > I'm fighting with your ac97 pinctrl configuration per dts. > > > [...] > > +ssi { > > + ... > > + > > + pinctrl-names = "default", "ac97-running", "ac97-reset", "ac97-warm-reset"; > > + pinctrl-0 = <&ac97link_running>; > > + pinctrl-1 = <&ac97link_running>; > > + pinctrl-2 = <&ac97link_reset>; > > + pinctrl-3 = <&ac97link_warm_reset>; > > + ac97-gpios = <&gpio3 20 0 &gpio3 22 0 &gpio3 28 0>; > > I see here the gpios for sync, sdata and reset. On my board I have two sdata > lines (sdata-in and sdata-out) which line should I specify here for this setting? Try to use sdata-out. This is the only data output line. sdata-in is the input to the SoC so we don't have to use this for an AC97 reset. Regards, Markus -- Pengutronix e.K. | | Industrial Linux Solutions | http://www.pengutronix.de/ | Peiner Str. 6-8, 31137 Hildesheim, Germany | Phone: +49-5121-206917-0 | Amtsgericht Hildesheim, HRA 2686 | Fax: +49-5121-206917-5555 |

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Re: [alsa-devel] [RFC PATCH] ASoC: tas2552: Support TI TAS2552 Amplifier
by Mark Brown 18 Jun '14

18 Jun '14

On Wed, Jun 18, 2014 at 12:45:46PM -0500, Dan Murphy wrote: > +static int tas2552_i2c_read(struct tas2552_data *tas_data, int reg) > +{ > + int val; > + > + if (WARN_ON(!tas_data->tas2552_client)) > + return -EINVAL; > + > + /* If powered off, return the cached value */ > + mutex_lock(&tas_data->mutex); > + if (tas_data->power_state) { > + val = i2c_smbus_read_byte_data(tas_data->tas2552_client, reg); Don't open code your I/O, use regmap like any other CODEC driver. This will remove a very large proportion of the code in the driver which replicates features of the regmap framework. > +static int tas2552_startup(struct snd_pcm_substream *substream, > + struct snd_soc_dai *dai) > +{ > + struct snd_soc_codec *codec = dai->codec; > + struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec); > + > + tas2552_sw_shutdown(tas2552, 1); > + tas2552_power(tas2552, 1); > + /* Turn on Class D amplifier */ > + tas2552_i2c_write(tas2552, TAS2552_CFG_2, 0x80); > + tas2552_sw_shutdown(tas2552, 0); I'd expect this to be done using DAPM. > +static const struct regmap_config tas2552_regmap_config = { > + .reg_bits = 8, > + .val_bits = 8, > + > + .max_register = TAS2552_MAX_REG, > + .reg_defaults = NULL, > + .num_reg_defaults = 0, No need to set things to zero or NULL in static variables, that's the default. > + dev = &client->dev; > + data = devm_kzalloc(&client->dev, sizeof(*data), GFP_KERNEL); > + if (data == NULL) { > + dev_err(dev, "Can not allocate memory\n"); No need to print an error, OOM messages are already quite noisy.

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