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

Re: [alsa-devel] [pulseaudio-discuss] [PATCH] alsa-mixer: recognize Dock headphone jack
by Raymond Yau 02 Sep '14

02 Sep '14

> > > > > > Recognize the Dock headphone jack in the same way the normal & front > > > headphone jacks are detected. > > > --- > > > src/modules/alsa/mixer/paths/analog-output-headphones.conf | 8 ++++++++ > > > src/modules/alsa/mixer/paths/analog-output-speaker.conf | 4 ++++ > > > 2 files changed, 12 insertions(+) > > > > > > diff --git a/src/modules/alsa/mixer/paths/analog-output-headphones.conf > > b/src/modules/alsa/mixer/paths/analog-output-headphones.conf > > > index 89d794f..199b2c5 100644 > > > --- a/src/modules/alsa/mixer/paths/analog-output-headphones.conf > > > +++ b/src/modules/alsa/mixer/paths/analog-output-headphones.conf > > > @@ -25,6 +25,14 @@ description-key = analog-output-headphones > > > [Properties] > > > device.icon_name = audio-headphones > > > > > > +[Jack Dock Headphone] > > > +required-any = any > > > > As you assume dock headphone and headphone share the same headphone volume > > control > > The do on my machine indeed. For reference i've uploaded my alsa-info > into: >http://www.alsa-project.org/db/?f=5a1c208208ff33a2231d2d03d5bea5ec452dc24d > > > Do you mean the current alsa driver support surround 40 for notebook with > > dock headphone is incorrect ? > https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… You will get auto mute lineout+speaker if the pin fixup is dock line out instead of dock headphone but pulseaudio don't like headphone and line out share pcm playback volume control and no headphone playback switch https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… For notebook with dual headphone jacks which support multi channel and headphone Jack won't mute the other headphone Jack As you expect headphone can mute dock headphone and speaker

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[alsa-devel] [PATCHv3 1/9] Clean WM8971 through checkpatch
by Xavier Hsu 02 Sep '14

02 Sep '14

This patch improves WM8971. We clean the file through checkpatch. Any comments about improving the patch are welcome. Thanks. Signed-off-by: Xavier Hsu <xavier.hsu(a)linaro.org> Signed-off-by: Andy Green <andy.green(a)linaro.org> --- sound/soc/codecs/wm8971.c | 107 ++++++++++++++++++++++++--------------------- 1 file changed, 56 insertions(+), 51 deletions(-) diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 0499cd4..064278f 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -31,7 +31,7 @@ #define WM8971_REG_COUNT 43 -static struct workqueue_struct *wm8971_workq = NULL; +static struct workqueue_struct *wm8971_workq; /* codec private data */ struct wm8971_priv { @@ -92,25 +92,28 @@ static const struct reg_default wm8971_reg_defaults[] = { #define wm8971_reset(c) snd_soc_write(c, WM8971_RESET, 0) /* WM8971 Controls */ -static const char *wm8971_bass[] = { "Linear Control", "Adaptive Boost" }; -static const char *wm8971_bass_filter[] = { "130Hz @ 48kHz", - "200Hz @ 48kHz" }; -static const char *wm8971_treble[] = { "8kHz", "4kHz" }; -static const char *wm8971_alc_func[] = { "Off", "Right", "Left", "Stereo" }; -static const char *wm8971_ng_type[] = { "Constant PGA Gain", - "Mute ADC Output" }; -static const char *wm8971_deemp[] = { "None", "32kHz", "44.1kHz", "48kHz" }; -static const char *wm8971_mono_mux[] = {"Stereo", "Mono (Left)", - "Mono (Right)", "Digital Mono"}; -static const char *wm8971_dac_phase[] = { "Non Inverted", "Inverted" }; -static const char *wm8971_lline_mux[] = {"Line", "NC", "NC", "PGA", - "Differential"}; -static const char *wm8971_rline_mux[] = {"Line", "Mic", "NC", "PGA", - "Differential"}; -static const char *wm8971_lpga_sel[] = {"Line", "NC", "NC", "Differential"}; -static const char *wm8971_rpga_sel[] = {"Line", "Mic", "NC", "Differential"}; -static const char *wm8971_adcpol[] = {"Normal", "L Invert", "R Invert", - "L + R Invert"}; +static const char const *wm8971_bass[] = {"Linear Control", "Adaptive Boost"}; +static const char const *wm8971_bass_filter[] = {"130Hz @ 48kHz", + "200Hz @ 48kHz"}; +static const char const *wm8971_treble[] = {"8kHz", "4kHz"}; +static const char const *wm8971_alc_func[] = {"Off", "Right", + "Left", "Stereo"}; +static const char const *wm8971_ng_type[] = {"Constant PGA Gain", + "Mute ADC Output"}; +static const char const *wm8971_deemp[] = {"None", "32kHz", "44.1kHz", "48kHz"}; +static const char const *wm8971_mono_mux[] = {"Stereo", "Mono (Left)", + "Mono (Right)", "Digital Mono"}; +static const char const *wm8971_dac_phase[] = {"Non Inverted", "Inverted"}; +static const char const *wm8971_lline_mux[] = {"Line", "NC", "NC", + "PGA", "Differential"}; +static const char const *wm8971_rline_mux[] = {"Line", "Mic", "NC", + "PGA", "Differential"}; +static const char const *wm8971_lpga_sel[] = {"Line", "NC", "NC", + "Differential"}; +static const char const *wm8971_rpga_sel[] = {"Line", "Mic", "NC", + "Differential"}; +static const char const *wm8971_adcpol[] = {"Normal", "L Invert", + "R Invert", "L + R Invert"}; static const struct soc_enum wm8971_enum[] = { SOC_ENUM_SINGLE(WM8971_BASS, 7, 2, wm8971_bass), /* 0 */ @@ -136,24 +139,24 @@ static const struct snd_kcontrol_new wm8971_snd_controls[] = { SOC_DOUBLE_R("Capture Switch", WM8971_LINVOL, WM8971_RINVOL, 7, 1, 1), SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8971_LOUT1V, - WM8971_ROUT1V, 7, 1, 0), + WM8971_ROUT1V, 7, 1, 0), SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8971_LOUT2V, - WM8971_ROUT2V, 7, 1, 0), + WM8971_ROUT2V, 7, 1, 0), SOC_SINGLE("Mono Playback ZC Switch", WM8971_MOUTV, 7, 1, 0), SOC_DOUBLE_R("PCM Volume", WM8971_LDAC, WM8971_RDAC, 0, 255, 0), SOC_DOUBLE_R("Bypass Left Playback Volume", WM8971_LOUTM1, - WM8971_LOUTM2, 4, 7, 1), + WM8971_LOUTM2, 4, 7, 1), SOC_DOUBLE_R("Bypass Right Playback Volume", WM8971_ROUTM1, - WM8971_ROUTM2, 4, 7, 1), + WM8971_ROUTM2, 4, 7, 1), SOC_DOUBLE_R("Bypass Mono Playback Volume", WM8971_MOUTM1, - WM8971_MOUTM2, 4, 7, 1), + WM8971_MOUTM2, 4, 7, 1), SOC_DOUBLE_R("Headphone Playback Volume", WM8971_LOUT1V, - WM8971_ROUT1V, 0, 127, 0), + WM8971_ROUT1V, 0, 127, 0), SOC_DOUBLE_R("Speaker Playback Volume", WM8971_LOUT2V, - WM8971_ROUT2V, 0, 127, 0), + WM8971_ROUT2V, 0, 127, 0), SOC_ENUM("Bass Boost", wm8971_enum[0]), SOC_ENUM("Bass Filter", wm8971_enum[1]), @@ -238,14 +241,14 @@ SOC_DAPM_ENUM("Route", wm8971_enum[13]); static const struct snd_soc_dapm_widget wm8971_dapm_widgets[] = { SND_SOC_DAPM_MIXER("Left Mixer", SND_SOC_NOPM, 0, 0, - &wm8971_left_mixer_controls[0], - ARRAY_SIZE(wm8971_left_mixer_controls)), + &wm8971_left_mixer_controls[0], + ARRAY_SIZE(wm8971_left_mixer_controls)), SND_SOC_DAPM_MIXER("Right Mixer", SND_SOC_NOPM, 0, 0, - &wm8971_right_mixer_controls[0], - ARRAY_SIZE(wm8971_right_mixer_controls)), + &wm8971_right_mixer_controls[0], + ARRAY_SIZE(wm8971_right_mixer_controls)), SND_SOC_DAPM_MIXER("Mono Mixer", WM8971_PWR2, 2, 0, - &wm8971_mono_mixer_controls[0], - ARRAY_SIZE(wm8971_mono_mixer_controls)), + &wm8971_mono_mixer_controls[0], + ARRAY_SIZE(wm8971_mono_mixer_controls)), SND_SOC_DAPM_PGA("Right Out 2", WM8971_PWR2, 3, 0, NULL, 0), SND_SOC_DAPM_PGA("Left Out 2", WM8971_PWR2, 4, 0, NULL, 0), @@ -260,18 +263,18 @@ static const struct snd_soc_dapm_widget wm8971_dapm_widgets[] = { SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8971_PWR1, 3, 0), SND_SOC_DAPM_MUX("Left PGA Mux", WM8971_PWR1, 5, 0, - &wm8971_left_pga_controls), + &wm8971_left_pga_controls), SND_SOC_DAPM_MUX("Right PGA Mux", WM8971_PWR1, 4, 0, - &wm8971_right_pga_controls), + &wm8971_right_pga_controls), SND_SOC_DAPM_MUX("Left Line Mux", SND_SOC_NOPM, 0, 0, - &wm8971_left_line_controls), + &wm8971_left_line_controls), SND_SOC_DAPM_MUX("Right Line Mux", SND_SOC_NOPM, 0, 0, - &wm8971_right_line_controls), + &wm8971_right_line_controls), SND_SOC_DAPM_MUX("Left ADC Mux", SND_SOC_NOPM, 0, 0, - &wm8971_monomux_controls), + &wm8971_monomux_controls), SND_SOC_DAPM_MUX("Right ADC Mux", SND_SOC_NOPM, 0, 0, - &wm8971_monomux_controls), + &wm8971_monomux_controls), SND_SOC_DAPM_OUTPUT("LOUT1"), SND_SOC_DAPM_OUTPUT("ROUT1"), @@ -431,7 +434,7 @@ static int get_coeff(int mclk, int rate) } static int wm8971_set_dai_sysclk(struct snd_soc_dai *codec_dai, - int clk_id, unsigned int freq, int dir) + int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; struct wm8971_priv *wm8971 = snd_soc_codec_get_drvdata(codec); @@ -449,7 +452,7 @@ static int wm8971_set_dai_sysclk(struct snd_soc_dai *codec_dai, } static int wm8971_set_dai_fmt(struct snd_soc_dai *codec_dai, - unsigned int fmt) + unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; u16 iface = 0; @@ -507,8 +510,8 @@ static int wm8971_set_dai_fmt(struct snd_soc_dai *codec_dai, } static int wm8971_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; struct wm8971_priv *wm8971 = snd_soc_codec_get_drvdata(codec); @@ -553,7 +556,7 @@ static int wm8971_mute(struct snd_soc_dai *dai, int mute) } static int wm8971_set_bias_level(struct snd_soc_codec *codec, - enum snd_soc_bias_level level) + enum snd_soc_bias_level level) { u16 pwr_reg = snd_soc_read(codec, WM8971_PWR1) & 0xfe3e; @@ -580,11 +583,12 @@ static int wm8971_set_bias_level(struct snd_soc_codec *codec, } #define WM8971_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ - SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \ - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) #define WM8971_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ - SNDRV_PCM_FMTBIT_S24_LE) + SNDRV_PCM_FMTBIT_S24_LE) static const struct snd_soc_dai_ops wm8971_dai_ops = { .hw_params = wm8971_pcm_hw_params, @@ -616,6 +620,7 @@ static void wm8971_work(struct work_struct *work) container_of(work, struct snd_soc_dapm_context, delayed_work.work); struct snd_soc_codec *codec = dapm->codec; + wm8971_set_bias_level(codec, codec->dapm.bias_level); } @@ -637,7 +642,7 @@ static int wm8971_resume(struct snd_soc_codec *codec) snd_soc_write(codec, WM8971_PWR1, reg | 0x01c0); codec->dapm.bias_level = SND_SOC_BIAS_ON; queue_delayed_work(wm8971_workq, &codec->dapm.delayed_work, - msecs_to_jiffies(1000)); + msecs_to_jiffies(1000)); } return 0; @@ -660,7 +665,7 @@ static int wm8971_probe(struct snd_soc_codec *codec) snd_soc_write(codec, WM8971_PWR1, reg | 0x01c0); codec->dapm.bias_level = SND_SOC_BIAS_STANDBY; queue_delayed_work(wm8971_workq, &codec->dapm.delayed_work, - msecs_to_jiffies(1000)); + msecs_to_jiffies(1000)); /* set the update bits */ snd_soc_update_bits(codec, WM8971_LDAC, 0x0100, 0x0100); @@ -729,8 +734,8 @@ static int wm8971_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, wm8971); - ret = snd_soc_register_codec(&i2c->dev, - &soc_codec_dev_wm8971, &wm8971_dai, 1); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8971, + &wm8971_dai, 1); return ret; } -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: ab8500-codec: Revert back to regmap
by Lars-Peter Clausen 01 Sep '14

01 Sep '14

Commit ff795d614bfa ("ASoC: ab8500: Convert register I/O to regmap") initially converted the ab8500 CODEC driver to use regmap rather than legacy ASoC IO. This was reverted though in commit 63e6d43bf80d ("ASoC: ab8500: Revert to using custom I/O functions") since the inital conversion was not working properly. This was presumebly because the SOC_SINGLE_XR_SX controls, which are used by this driver, did not properly support regmap at that point. This has since been fixed in commit 6137a5ca326d ("ASoC: Prepare SOC_SINGLE_XR_SX controls for regmap"). So revert back to regmap again. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- I don't have the hardware, so I haven't been able to verify that it actually works now, but I think there is a very good chance. Lee, do you have access to a Snowball board and can give this a test? --- sound/soc/codecs/ab8500-codec.c | 62 +++++++++++++++++++---------------------- 1 file changed, 28 insertions(+), 34 deletions(-) diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index 62cf231..fd43827 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -125,6 +125,8 @@ struct ab8500_codec_drvdata_dbg { /* Private data for AB8500 device-driver */ struct ab8500_codec_drvdata { + struct regmap *regmap; + /* Sidetone */ long *sid_fir_values; enum sid_state sid_status; @@ -165,49 +167,35 @@ static inline const char *amic_type_str(enum amic_type type) */ /* Read a register from the audio-bank of AB8500 */ -static unsigned int ab8500_codec_read_reg(struct snd_soc_codec *codec, - unsigned int reg) +static int ab8500_codec_read_reg(void *context, unsigned int reg, + unsigned int *value) { + struct device *dev = context; int status; - unsigned int value = 0; u8 value8; - status = abx500_get_register_interruptible(codec->dev, AB8500_AUDIO, - reg, &value8); - if (status < 0) { - dev_err(codec->dev, - "%s: ERROR: Register (0x%02x:0x%02x) read failed (%d).\n", - __func__, (u8)AB8500_AUDIO, (u8)reg, status); - } else { - dev_dbg(codec->dev, - "%s: Read 0x%02x from register 0x%02x:0x%02x\n", - __func__, value8, (u8)AB8500_AUDIO, (u8)reg); - value = (unsigned int)value8; - } + status = abx500_get_register_interruptible(dev, AB8500_AUDIO, + reg, &value8); + *value = (unsigned int)value8; - return value; + return status; } /* Write to a register in the audio-bank of AB8500 */ -static int ab8500_codec_write_reg(struct snd_soc_codec *codec, - unsigned int reg, unsigned int value) +static int ab8500_codec_write_reg(void *context, unsigned int reg, + unsigned int value) { - int status; - - status = abx500_set_register_interruptible(codec->dev, AB8500_AUDIO, - reg, value); - if (status < 0) - dev_err(codec->dev, - "%s: ERROR: Register (%02x:%02x) write failed (%d).\n", - __func__, (u8)AB8500_AUDIO, (u8)reg, status); - else - dev_dbg(codec->dev, - "%s: Wrote 0x%02x into register %02x:%02x\n", - __func__, (u8)value, (u8)AB8500_AUDIO, (u8)reg); + struct device *dev = context; - return status; + return abx500_set_register_interruptible(dev, AB8500_AUDIO, + reg, value); } +static const struct regmap_config ab8500_codec_regmap = { + .reg_read = ab8500_codec_read_reg, + .reg_write = ab8500_codec_write_reg, +}; + /* * Controls - DAPM */ @@ -2564,9 +2552,6 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec) static struct snd_soc_codec_driver ab8500_codec_driver = { .probe = ab8500_codec_probe, - .read = ab8500_codec_read_reg, - .write = ab8500_codec_write_reg, - .reg_word_size = sizeof(u8), .controls = ab8500_ctrls, .num_controls = ARRAY_SIZE(ab8500_ctrls), .dapm_widgets = ab8500_dapm_widgets, @@ -2591,6 +2576,15 @@ static int ab8500_codec_driver_probe(struct platform_device *pdev) drvdata->anc_status = ANC_UNCONFIGURED; dev_set_drvdata(&pdev->dev, drvdata); + drvdata->regmap = devm_regmap_init(&pdev->dev, NULL, &pdev->dev, + &ab8500_codec_regmap); + if (IS_ERR(drvdata->regmap)) { + status = PTR_ERR(drvdata->regmap); + dev_err(&pdev->dev, "%s: Failed to allocate regmap: %d\n", + __func__, status); + return status; + } + dev_dbg(&pdev->dev, "%s: Register codec.\n", __func__); status = snd_soc_register_codec(&pdev->dev, &ab8500_codec_driver, ab8500_codec_dai, -- 1.8.0

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[alsa-devel] [PATCHv3] ASoC: simple-card: Fix bug of wrong decrement DT node's refcount
by Xiubo Li 01 Sep '14

01 Sep '14

DAI links's cpu_of_node's and codec_of_node's refcounts shouldn't be decremented immediately at the end of the probe() fucntion. Because we will still use them before the audio card is removed. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> Cc: Jean-Francois Moine <moinejf(a)free.fr> --- Change in v3: - Follow Jean-Francois Moine's advice. - Remove the following patch: "ASoC: simple-card: Fix bug of forgetting decrement DT node's refcount" sound/soc/generic/simple-card.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index fd8b045..986d2c7 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -481,12 +481,19 @@ static int asoc_simple_card_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(&priv->snd_card, priv); ret = devm_snd_soc_register_card(&pdev->dev, &priv->snd_card); + if (ret >= 0) + return ret; err: asoc_simple_card_unref(pdev); return ret; } +static int asoc_simple_card_remove(struct platform_device *pdev) +{ + return asoc_simple_card_unref(pdev); +} + static const struct of_device_id asoc_simple_of_match[] = { { .compatible = "simple-audio-card", }, {}, @@ -500,6 +507,7 @@ static struct platform_driver asoc_simple_card = { .of_match_table = asoc_simple_of_match, }, .probe = asoc_simple_card_probe, + .remove = asoc_simple_card_remove, }; module_platform_driver(asoc_simple_card); -- 1.8.4

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[alsa-devel] [PATCH v4 0/2] ASoC: tda998x: add a codec to the HDMI transmitter
by Jean-Francois Moine 01 Sep '14

01 Sep '14

The NXP TDA998x HDMI transmitter may transmit audio to the HDMI link from 2 different sources, I2S and S/PDIF. This patch set adds a generic HDMI CODEC module which is used by the TDA998x driver to update the audio constraints from the display characteristics (EDID) and by the audio subsystem to connect the chosen audio source to the HDMI link. v4: - remove all the TDA998x specific stuff from the CODEC - move the EDID scan from the CODEC to the TDA998x - move the CODEC to sound/soc (Mark Brown) - update the audio_sample_rate from the EDID (Andrew Jackson) v3: fix bad rate (Andrew Jackson) v2: check double stream start (Mark Brown) Jean-Francois Moine (2): ASoC:codecs: Add a generic HDMI audio CODEC drm/i2c:tda998x: Use the HDMI2 audio CODEC Documentation/devicetree/bindings/sound/hdmi2.txt | 32 ++++ drivers/gpu/drm/i2c/Kconfig | 1 + drivers/gpu/drm/i2c/tda998x_drv.c | 142 ++++++++++++++- include/sound/hdmi2.h | 24 +++ sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hdmi2.c | 204 ++++++++++++++++++++++ 7 files changed, 400 insertions(+), 8 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/hdmi2.txt create mode 100644 include/sound/hdmi2.h create mode 100644 sound/soc/codecs/hdmi2.c -- 2.1.0

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[alsa-devel] [PATCHv2] ASoC: fsl-sai: using 'lsb-first' property instead of 'big-endian-data'.
by Xiubo Li 01 Sep '14

01 Sep '14

The 'big-endian-data' property is originally used to indicate whether the LSB firstly or MSB firstly will be transmitted to the CODEC or received from the CODEC, and there has nothing relation to the memory data. Generally, if the audio data in big endian format, which will be using the bytes reversion, Here this can only be used to bits reversion. So using the 'lsb-first' instead of 'big-endian-data' can make the code to be readable easier and more easy to understand what this property is used to do. This property used for configuring whether the LSB or the MSB is transmitted first for the fifo data. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> Cc: Nicolin Chen <nicoleotsuka(a)gmail.com> --- Documentation/devicetree/bindings/sound/fsl-sai.txt | 8 ++++---- sound/soc/fsl/fsl_sai.c | 6 +++--- sound/soc/fsl/fsl_sai.h | 2 +- 3 files changed, 8 insertions(+), 8 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt index 06a405e..4956b14 100644 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt @@ -20,9 +20,9 @@ Required properties: See ../pinctrl/pinctrl-bindings.txt for details of the property values. - big-endian: Boolean property, required if all the FTM_PWM registers are big-endian rather than little-endian. -- big-endian-data: If this property is absent, the little endian mode will - be in use as default, or the big endian mode will be in use for all the - fifo data. +- lsb-first: Configures whether the LSB or the MSB is transmitted first for + the fifo data. If this property is absent, the MSB is transmitted first as + default, or the LSB is transmitted first. - fsl,sai-synchronous-rx: This is a boolean property. If present, indicating that SAI will work in the synchronous mode (sync Tx with Rx) which means both the transimitter and receiver will send and receive data by following @@ -53,5 +53,5 @@ sai2: sai@40031000 { dmas = <&edma0 0 VF610_EDMA_MUXID0_SAI2_TX>, <&edma0 0 VF610_EDMA_MUXID0_SAI2_RX>; big-endian; - big-endian-data; + lsb-first; }; diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index a6eb784..7eeb1dd 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -175,7 +175,7 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, bool tx = fsl_dir == FSL_FMT_TRANSMITTER; u32 val_cr2 = 0, val_cr4 = 0; - if (!sai->big_endian_data) + if (!sai->is_lsb_first) val_cr4 |= FSL_SAI_CR4_MF; /* DAI mode */ @@ -304,7 +304,7 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, val_cr5 |= FSL_SAI_CR5_WNW(word_width); val_cr5 |= FSL_SAI_CR5_W0W(word_width); - if (sai->big_endian_data) + if (sai->is_lsb_first) val_cr5 |= FSL_SAI_CR5_FBT(0); else val_cr5 |= FSL_SAI_CR5_FBT(word_width - 1); @@ -573,7 +573,7 @@ static int fsl_sai_probe(struct platform_device *pdev) if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai")) sai->sai_on_imx = true; - sai->big_endian_data = of_property_read_bool(np, "big-endian-data"); + sai->is_lsb_first = of_property_read_bool(np, "lsb-first"); res = platform_get_resource(pdev, IORESOURCE_MEM, 0); base = devm_ioremap_resource(&pdev->dev, res); diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 2cded44..3466720 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -132,7 +132,7 @@ struct fsl_sai { struct clk *bus_clk; struct clk *mclk_clk[FSL_SAI_MCLK_MAX]; - bool big_endian_data; + bool is_lsb_first; bool is_dsp_mode; bool sai_on_imx; bool synchronous[2]; -- 1.8.5

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[alsa-devel] [PATCH 1/3 v4 RESEND] mfd: arizona: Export function to control subsystem DVFS
by Richard Fitzgerald 01 Sep '14

01 Sep '14

From: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> Moving this control from being a side-effect of the LDO1 regulator driver to a specific exported function. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- drivers/mfd/arizona-core.c | 89 ++++++++++++++++++++++++++++++++++++++ include/linux/mfd/arizona/core.h | 12 +++++ 2 files changed, 101 insertions(+), 0 deletions(-) diff --git a/drivers/mfd/arizona-core.c b/drivers/mfd/arizona-core.c index cfc191a..04cde4b 100644 --- a/drivers/mfd/arizona-core.c +++ b/drivers/mfd/arizona-core.c @@ -94,6 +94,94 @@ int arizona_clk32k_disable(struct arizona *arizona) } EXPORT_SYMBOL_GPL(arizona_clk32k_disable); +int arizona_dvfs_up(struct arizona *arizona, unsigned int flags) +{ + unsigned int new_flags; + int ret = 0; + + mutex_lock(&arizona->subsys_max_lock); + + new_flags = arizona->subsys_max_rq | flags; + + if (arizona->subsys_max_rq != new_flags) { + switch (arizona->type) { + case WM5102: + case WM8997: + ret = regulator_set_voltage(arizona->dcvdd, + 1800000, 1800000); + if (ret != 0) { + dev_err(arizona->dev, + "Failed to set DCVDD (DVFS up): %d\n", + ret); + goto err; + } + + ret = regmap_update_bits(arizona->regmap, + ARIZONA_DYNAMIC_FREQUENCY_SCALING_1, + ARIZONA_SUBSYS_MAX_FREQ, 1); + if (ret != 0) { + dev_err(arizona->dev, + "Failed to enable subsys max: %d\n", + ret); + regulator_set_voltage(arizona->dcvdd, + 1200000, 1800000); + goto err; + } + break; + + default: + break; + } + + arizona->subsys_max_rq = new_flags; + } +err: + mutex_unlock(&arizona->subsys_max_lock); + return ret; +} +EXPORT_SYMBOL_GPL(arizona_dvfs_up); + +int arizona_dvfs_down(struct arizona *arizona, unsigned int flags) +{ + int ret = 0; + + mutex_lock(&arizona->subsys_max_lock); + + if ((arizona->subsys_max_rq & flags) != flags) + dev_warn(arizona->dev, "Unbalanced DVFS down: %x\n", flags); + + arizona->subsys_max_rq &= ~flags; + + if (arizona->subsys_max_rq == 0) { + switch (arizona->type) { + case WM5102: + case WM8997: + ret = regmap_update_bits(arizona->regmap, + ARIZONA_DYNAMIC_FREQUENCY_SCALING_1, + ARIZONA_SUBSYS_MAX_FREQ, 0); + if (ret != 0) + dev_err(arizona->dev, + "Failed to disable subsys max: %d\n", + ret); + + ret = regulator_set_voltage(arizona->dcvdd, + 1200000, 1800000); + if (ret != 0) + dev_err(arizona->dev, + "Failed to set DCVDD (DVFS down): %d\n", + ret); + break; + + default: + break; + } + } + + mutex_unlock(&arizona->subsys_max_lock); + return ret; +} +EXPORT_SYMBOL_GPL(arizona_dvfs_down); + static irqreturn_t arizona_clkgen_err(int irq, void *data) { struct arizona *arizona = data; @@ -641,6 +729,7 @@ int arizona_dev_init(struct arizona *arizona) dev_set_drvdata(arizona->dev, arizona); mutex_init(&arizona->clk_lock); + mutex_init(&arizona->subsys_max_lock); if (dev_get_platdata(arizona->dev)) memcpy(&arizona->pdata, dev_get_platdata(arizona->dev), diff --git a/include/linux/mfd/arizona/core.h b/include/linux/mfd/arizona/core.h index a614b33..4aec6a3 100644 --- a/include/linux/mfd/arizona/core.h +++ b/include/linux/mfd/arizona/core.h @@ -109,6 +109,9 @@ struct arizona { struct mutex clk_lock; int clk32k_ref; + struct mutex subsys_max_lock; + unsigned int subsys_max_rq; + struct snd_soc_dapm_context *dapm; int tdm_width[ARIZONA_MAX_AIF]; @@ -118,8 +121,17 @@ struct arizona { uint8_t dac_comp_enabled; }; +#define ARIZONA_DVFS_SR1_RQ 0x00000001 +#define ARIZONA_DVFS_SR2_RQ 0x00000002 +#define ARIZONA_DVFS_SR3_RQ 0x00000004 +#define ARIZONA_DVFS_ASR1_RQ 0x00000010 +#define ARIZONA_DVFS_ASR2_RQ 0x00000020 +#define ARIZONA_DVFS_ADSP1_RQ 0x00010000 + int arizona_clk32k_enable(struct arizona *arizona); int arizona_clk32k_disable(struct arizona *arizona); +int arizona_dvfs_up(struct arizona *arizona, unsigned int mask); +int arizona_dvfs_down(struct arizona *arizona, unsigned int mask); int arizona_request_irq(struct arizona *arizona, int irq, char *name, irq_handler_t handler, void *data); -- 1.7.2.5

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Re: [alsa-devel] [v3 04/11] ASoC: Intel: sst: Add IPC handling
by Mark Brown 01 Sep '14

01 Sep '14

On Mon, Sep 01, 2014 at 05:47:53PM +0530, Subhransu S. Prusty wrote: > On Wed, Aug 27, 2014 at 09:37:37PM +0100, Mark Brown wrote: > > > + spin_unlock_bh(&ctx->block_lock); > > > + return 0; > > > + } > > > + } > > > + spin_unlock_bh(&ctx->block_lock); > > > + return -EINVAL; > > I'd expect much louder complaints if we try to free something that's not > > allocated - what happens if we end up reallocating something quickly and > > then double freeing? Better to complain if we hit such a code path. > "freed" is a block which is passed by the caller to be freed up. Will add a > comment. How would that address the problem? Obviously the caller is trying to free what they're passing in.

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[alsa-devel] [PATCH alsa-lib] pcm: fix snd_pcm_mmap_hw_avail() near the boundary
by Alexander E. Patrakov 01 Sep '14

01 Sep '14

This function returned incorrect results when hw.ptr was near the boundary and hw.appl_ptr was near zero. Here "incorrect" means "greater than the boundary". The result was incorrect, because it was used as a return value of various *_rewindable() functions and also as the delay for ioplug. Signed-off-by: Alexander E. Patrakov <patrakov(a)gmail.com> --- src/pcm/pcm_local.h | 8 +------- 1 file changed, 1 insertion(+), 7 deletions(-) diff --git a/src/pcm/pcm_local.h b/src/pcm/pcm_local.h index 80bbe59..74ebd60 100644 --- a/src/pcm/pcm_local.h +++ b/src/pcm/pcm_local.h @@ -461,13 +461,7 @@ static inline snd_pcm_sframes_t snd_pcm_mmap_capture_hw_avail(snd_pcm_t *pcm) static inline snd_pcm_sframes_t snd_pcm_mmap_hw_avail(snd_pcm_t *pcm) { - snd_pcm_sframes_t avail; - avail = *pcm->hw.ptr - *pcm->appl.ptr; - if (pcm->stream == SND_PCM_STREAM_PLAYBACK) - avail += pcm->buffer_size; - if (avail < 0) - avail += pcm->boundary; - return pcm->buffer_size - avail; + return pcm->buffer_size - snd_pcm_mmap_avail(pcm); } static inline const snd_pcm_channel_area_t *snd_pcm_mmap_areas(snd_pcm_t *pcm) -- 2.1.0

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[alsa-devel] [PATCH] ALSA: hda - Fix digital mic on Acer Aspire 3830TG
by Takashi Iwai 01 Sep '14

01 Sep '14

Acer Aspire 3830TG with CX20588 codec has a digital built-in mic that has the same problem like many others, the inverted signal in stereo. Apply the same fixup to this machine, too. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_conexant.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 6f2fa838b635..6e5d0cb4e3d7 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -217,6 +217,7 @@ enum { CXT_FIXUP_HEADPHONE_MIC_PIN, CXT_FIXUP_HEADPHONE_MIC, CXT_FIXUP_GPIO1, + CXT_FIXUP_ASPIRE_DMIC, CXT_FIXUP_THINKPAD_ACPI, CXT_FIXUP_OLPC_XO, CXT_FIXUP_CAP_MIX_AMP, @@ -664,6 +665,12 @@ static const struct hda_fixup cxt_fixups[] = { { } }, }, + [CXT_FIXUP_ASPIRE_DMIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = cxt_fixup_stereo_dmic, + .chained = true, + .chain_id = CXT_FIXUP_GPIO1, + }, [CXT_FIXUP_THINKPAD_ACPI] = { .type = HDA_FIXUP_FUNC, .v.func = hda_fixup_thinkpad_acpi, @@ -744,7 +751,7 @@ static const struct hda_model_fixup cxt5051_fixup_models[] = { static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x1025, 0x0543, "Acer Aspire One 522", CXT_FIXUP_STEREO_DMIC), - SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT_FIXUP_GPIO1), + SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT_FIXUP_ASPIRE_DMIC), SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN), SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT_FIXUP_OLPC_XO), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410), -- 2.1.0

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