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[alsa-devel] [PATCH - hda 1/1] Add ultra dock support for Thinkpad X240.
by Lukas 30 Oct '14

30 Oct '14

Adding ultra doch support for Lenovo Thinkpad X240 (17aa:2214). Signed-off-by: Lukas Bossard <mr.bobukas(a)gmail.com> diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 34b7bdb..528bb2d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4803,6 +4803,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x220e, "Thinkpad T440p", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2210, "Thinkpad T540p", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad T440", ALC292_FIXUP_TPT440_DOCK), + SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad X240", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP), -- 2.1.0

2 1

[alsa-devel] [PATCH] ALSA: es1968: Replace timeval with ktime_t
by Tina Ruchandani 30 Oct '14

30 Oct '14

es1968_measure_clock uses struct timeval, which on 32-bit systems will overflow in 2038, leading to incorrect interpretation of time.This patch changes the function to use ktime_t instead of struct timeval, which implies: - no y2038: ktime_t uses a 64-bit datatype explicitly. - efficent subtraction: The earlier version computes the difference in usecs while dealing with secs and nsecs. It requires checks to see if the nsecs of stop is less than start. This patch uses a direct subtract of ktime_t and converts to usecs. - use of monotonic clock (ktime_get) over real time (do_gettimeofday), which simplifies timekeeping, as it does not have to deal with cases where stop_time is less than start_time. Signed-off-by: Tina Ruchandani <ruchandani.tina(a)gmail.com> Reviewed-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/pci/es1968.c | 15 ++++++--------- 1 file changed, 6 insertions(+), 9 deletions(-) diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index a9956a7..6039700 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -1710,7 +1710,8 @@ static void es1968_measure_clock(struct es1968 *chip) int i, apu; unsigned int pa, offset, t; struct esm_memory *memory; - struct timeval start_time, stop_time; + ktime_t start_time, stop_time; + ktime_t diff; if (chip->clock == 0) chip->clock = 48000; /* default clock value */ @@ -1761,12 +1762,12 @@ static void es1968_measure_clock(struct es1968 *chip) snd_es1968_bob_inc(chip, ESM_BOB_FREQ); __apu_set_register(chip, apu, 5, pa & 0xffff); snd_es1968_trigger_apu(chip, apu, ESM_APU_16BITLINEAR); - do_gettimeofday(&start_time); + start_time = ktime_get(); spin_unlock_irq(&chip->reg_lock); msleep(50); spin_lock_irq(&chip->reg_lock); offset = __apu_get_register(chip, apu, 5); - do_gettimeofday(&stop_time); + stop_time = ktime_get(); snd_es1968_trigger_apu(chip, apu, 0); /* stop */ snd_es1968_bob_dec(chip); chip->in_measurement = 0; @@ -1777,12 +1778,8 @@ static void es1968_measure_clock(struct es1968 *chip) offset &= 0xfffe; offset += chip->measure_count * (CLOCK_MEASURE_BUFSIZE/2); - t = stop_time.tv_sec - start_time.tv_sec; - t *= 1000000; - if (stop_time.tv_usec < start_time.tv_usec) - t -= start_time.tv_usec - stop_time.tv_usec; - else - t += stop_time.tv_usec - start_time.tv_usec; + diff = ktime_sub(stop_time, start_time); + t = ktime_to_us(diff); if (t == 0) { dev_err(chip->card->dev, "?? calculation error..\n"); } else { -- 2.1.0.rc2.206.gedb03e5

2 1

[alsa-devel] Shared volume control again
by Takashi Iwai 30 Oct '14

30 Oct '14

Hi, thinking of the problem about shared volume controls again, I stumbled on a case where your suggested solution doesn't work well. Namely, when a headphone volume is shared for two headphones (one on the laptop and one on the dock), how should it be named? One is "Headphone" and another is "Dock Headphone". That's a bit tricky. So, in the end, we need the mapping representation somehow more explicitly than the control name. Thoughts? Takashi

4 7

[alsa-devel] ASoC HDMI receiver driver
by Tim Harvey 30 Oct '14

30 Oct '14

Greetings, I'm working on an ASoC driver (an mfd driver for an HDMI receiver chip with an accompanying codec driver for the audio capture portion, and a cpu-dai driver for the specific SoC its hooked to). I can't seem to figure out what kind of hook I need in order to fill out the snd_pcm_substream fields specifying the rate/channels/sample-format of the receiving HDMI audio stream. In this case HDMI audio (and thus the codec) supports several rates but it doesn't make sense that the codec is told what rate to capture at as the rate is dictated by the current stream the receiver is being provided. I'm thinking there is some hook I'm missing that allows the codec to fill in the params dynamically when the capture device is opened. Any pointers to how I deal with this? Thanks, Tim

1 0

[alsa-devel] [PATCHv5 1/9] ASOC: wm8971: clean for checkpatch --strict
by Xavier Hsu 30 Oct '14

30 Oct '14

Before adding wm8973 support, clean up existing wm8971 style. 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 | 110 +++++++++++++++++++++++---------------------- sound/soc/codecs/wm8971.h | 30 ++++++------- 2 files changed, 72 insertions(+), 68 deletions(-) diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 39ddb9b..01c76a4 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -29,9 +29,9 @@ #include "wm8971.h" -#define WM8971_REG_COUNT 43 +#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 = snd_soc_dapm_to_codec(dapm); + 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); @@ -675,7 +680,6 @@ static int wm8971_probe(struct snd_soc_codec *codec) return ret; } - /* power down chip */ static int wm8971_remove(struct snd_soc_codec *codec) { @@ -729,8 +733,8 @@ 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; } diff --git a/sound/soc/codecs/wm8971.h b/sound/soc/codecs/wm8971.h index f31c38f..dc38e21 100644 --- a/sound/soc/codecs/wm8971.h +++ b/sound/soc/codecs/wm8971.h @@ -22,25 +22,25 @@ #define WM8971_ADCDAC 0x05 #define WM8971_IFACE 0x07 #define WM8971_SRATE 0x08 -#define WM8971_LDAC 0x0a -#define WM8971_RDAC 0x0b -#define WM8971_BASS 0x0c +#define WM8971_LDAC 0x0a +#define WM8971_RDAC 0x0b +#define WM8971_BASS 0x0c #define WM8971_TREBLE 0x0d #define WM8971_RESET 0x0f -#define WM8971_ALC1 0x11 -#define WM8971_ALC2 0x12 -#define WM8971_ALC3 0x13 +#define WM8971_ALC1 0x11 +#define WM8971_ALC2 0x12 +#define WM8971_ALC3 0x13 #define WM8971_NGATE 0x14 -#define WM8971_LADC 0x15 -#define WM8971_RADC 0x16 -#define WM8971_ADCTL1 0x17 -#define WM8971_ADCTL2 0x18 -#define WM8971_PWR1 0x19 -#define WM8971_PWR2 0x1a -#define WM8971_ADCTL3 0x1b +#define WM8971_LADC 0x15 +#define WM8971_RADC 0x16 +#define WM8971_ADCTL1 0x17 +#define WM8971_ADCTL2 0x18 +#define WM8971_PWR1 0x19 +#define WM8971_PWR2 0x1a +#define WM8971_ADCTL3 0x1b #define WM8971_ADCIN 0x1f -#define WM8971_LADCIN 0x20 -#define WM8971_RADCIN 0x21 +#define WM8971_LADCIN 0x20 +#define WM8971_RADCIN 0x21 #define WM8971_LOUTM1 0x22 #define WM8971_LOUTM2 0x23 #define WM8971_ROUTM1 0x24 -- 1.7.9.5

2 10

[alsa-devel] [PATCH v6] ASoC: dapm: add code to configure dai link parameters
by Nikesh Oswal 29 Oct '14

29 Oct '14

dai-link params for codec-codec links were fixed. The fixed link between codec and another chip which may be another codec, baseband, bluetooth codec etc may require run time configuaration changes. This change provides an optional alsa control to select one of the params from a list of params. Signed-off-by: Nikesh Oswal <nikesh(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 3 + include/sound/soc.h | 1 + sound/soc/soc-core.c | 6 +- sound/soc/soc-dapm.c | 143 ++++++++++++++++++++++++++++++++++++++++++++-- 4 files changed, 146 insertions(+), 7 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index aac04ff..20863f2 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -378,6 +378,7 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card); void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card); int snd_soc_dapm_new_pcm(struct snd_soc_card *card, const struct snd_soc_pcm_stream *params, + unsigned int num_params, struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink); @@ -532,6 +533,8 @@ struct snd_soc_dapm_widget { void *priv; /* widget specific data */ struct regulator *regulator; /* attached regulator */ const struct snd_soc_pcm_stream *params; /* params for dai links */ + unsigned int num_params; /* number of params for dai links */ + unsigned int params_select; /* currently selected param for dai link */ /* dapm control */ int reg; /* negative reg = no direct dapm */ diff --git a/include/sound/soc.h b/include/sound/soc.h index c83a334..a97848f 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -926,6 +926,7 @@ struct snd_soc_dai_link { int be_id; /* optional ID for machine driver BE identification */ const struct snd_soc_pcm_stream *params; + unsigned int num_params; unsigned int dai_fmt; /* format to set on init */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d074aa9..93850be 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1436,7 +1436,8 @@ static int soc_link_dai_widgets(struct snd_soc_card *card, capture_w = cpu_dai->capture_widget; if (play_w && capture_w) { ret = snd_soc_dapm_new_pcm(card, dai_link->params, - capture_w, play_w); + dai_link->num_params, capture_w, + play_w); if (ret != 0) { dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", play_w->name, capture_w->name, ret); @@ -1448,7 +1449,8 @@ static int soc_link_dai_widgets(struct snd_soc_card *card, capture_w = codec_dai->capture_widget; if (play_w && capture_w) { ret = snd_soc_dapm_new_pcm(card, dai_link->params, - capture_w, play_w); + dai_link->num_params, capture_w, + play_w); if (ret != 0) { dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", play_w->name, capture_w->name, ret); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 177bd86..5f7a342 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -750,6 +750,36 @@ static int dapm_new_pga(struct snd_soc_dapm_widget *w) return 0; } +/* create new dapm dai link control */ +static int dapm_new_dai_link(struct snd_soc_dapm_widget *w) +{ + int i, ret; + struct snd_kcontrol *kcontrol; + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_card *card = dapm->card->snd_card; + + /* skip control creation for links with 1 config */ + if (w->num_params == 1) + return 0; + + /* add kcontrol */ + for (i = 0; i < w->num_kcontrols; i++) { + kcontrol = snd_soc_cnew(&w->kcontrol_news[i], w, + w->name, NULL); + ret = snd_ctl_add(card, kcontrol); + if (ret < 0) { + dev_err(dapm->dev, + "ASoC: failed to add widget %s dapm kcontrol %s: %d\n", + w->name, w->kcontrol_news[i].name, ret); + return ret; + } + kcontrol->private_data = w; + w->kcontrols[i] = kcontrol; + } + + return 0; +} + /* reset 'walked' bit for each dapm path */ static void dapm_clear_walk_output(struct snd_soc_dapm_context *dapm, struct list_head *sink) @@ -2701,6 +2731,9 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) case snd_soc_dapm_out_drv: dapm_new_pga(w); break; + case snd_soc_dapm_dai_link: + dapm_new_dai_link(w); + break; default: break; } @@ -3185,6 +3218,9 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, source = source_p->source->priv; sink = sink_p->sink->priv; + /* Select the configuration set by alsa control */ + config += w->params_select; + /* Be a little careful as we don't want to overflow the mask array */ if (config->formats) { fmt = ffs(config->formats) - 1; @@ -3253,8 +3289,35 @@ out: return ret; } +static int snd_soc_dapm_dai_link_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = w->params_select; + + return 0; +} + +static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + + if (ucontrol->value.integer.value[0] == w->params_select) + return 0; + + if (ucontrol->value.integer.value[0] >= w->num_params) + return -EINVAL; + + w->params_select = ucontrol->value.integer.value[0]; + + return 0; +} + int snd_soc_dapm_new_pcm(struct snd_soc_card *card, const struct snd_soc_pcm_stream *params, + unsigned int num_params, struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { @@ -3262,14 +3325,48 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, struct snd_soc_dapm_widget *w; size_t len; char *link_name; - int ret; + int ret, count; + unsigned long private_value; + const char **w_param_text; + struct soc_enum w_param_enum[] = { + SOC_ENUM_SINGLE(0, 0, 0, NULL), + }; + struct snd_kcontrol_new kcontrol_dai_link[] = { + SOC_ENUM_EXT(NULL, w_param_enum[0], + snd_soc_dapm_dai_link_get, + snd_soc_dapm_dai_link_put), + }; + const struct snd_soc_pcm_stream *config = params; + + w_param_text = kcalloc(num_params, sizeof(char *), GFP_KERNEL); + if (!w_param_text) + return -ENOMEM; len = strlen(source->name) + strlen(sink->name) + 2; link_name = devm_kzalloc(card->dev, len, GFP_KERNEL); - if (!link_name) - return -ENOMEM; + if (!link_name) { + ret = -ENOMEM; + goto outfree_w_param; + } + snprintf(link_name, len, "%s-%s", source->name, sink->name); + for (count = 0 ; count < num_params; count++) { + if (!config->stream_name) + dev_warn(card->dapm.dev, + "ASoC: anonymous config %d for dai link %s\n", + count, link_name); + w_param_text[count] = kmemdup((void *)(config->stream_name), + strlen(config->stream_name) + 1, GFP_KERNEL); + if (!w_param_text[count]) { + ret = -ENOMEM; + goto outfree_link_name; + } + config++; + } + w_param_enum[0].items = num_params; + w_param_enum[0].texts = w_param_text; + memset(&template, 0, sizeof(template)); template.reg = SND_SOC_NOPM; template.id = snd_soc_dapm_dai_link; @@ -3277,6 +3374,25 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, template.event = snd_soc_dai_link_event; template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD; + template.num_kcontrols = 1; + private_value = + (unsigned long) kmemdup((void *)(kcontrol_dai_link[0].private_value), + sizeof(struct soc_enum), GFP_KERNEL); + if (!private_value) { + dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", + link_name); + ret = -ENOMEM; + goto outfree_link_name; + } + kcontrol_dai_link[0].private_value = private_value; + template.kcontrol_news = kmemdup(&kcontrol_dai_link[0], + sizeof(struct snd_kcontrol_new), GFP_KERNEL); + if (!template.kcontrol_news) { + dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", + link_name); + ret = -ENOMEM; + goto outfree_private_value; + } dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); @@ -3284,15 +3400,32 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, if (!w) { dev_err(card->dev, "ASoC: Failed to create %s widget\n", link_name); - return -ENOMEM; + ret = -ENOMEM; + goto outfree_kcontrol_news; } w->params = params; + w->num_params = num_params; ret = snd_soc_dapm_add_path(&card->dapm, source, w, NULL, NULL); if (ret) - return ret; + goto outfree_w; return snd_soc_dapm_add_path(&card->dapm, w, sink, NULL, NULL); + +outfree_w: + kfree(w); +outfree_kcontrol_news: + kfree(template.kcontrol_news); +outfree_private_value: + kfree((void *)private_value); +outfree_link_name: + kfree(link_name); +outfree_w_param: + for (count = 0 ; count < num_params; count++) + kfree(w_param_text[count]); + kfree(w_param_text); + + return ret; } int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, -- 1.7.9.5

2 3

[alsa-devel] [PATCH] ASoC: add xtensa xtfpga I2S interface and platform
by Max Filippov 29 Oct '14

29 Oct '14

XTFPGA boards providean audio subsystem that consists of TI CDCE706 clock synthesizer, I2S transmitter and TLV320AIC23 audio codec. I2S transmitter has MMIO-based interface that resembles that of the OpenCores I2S transmitter. I2S transmitter is always a master on I2S bus. There's no specialized audio DMA, sample data are transferred to I2S transmitter FIFO by CPU through memory-mapped queue interface. Signed-off-by: Max Filippov <jcmvbkbc(a)gmail.com> --- .../devicetree/bindings/sound/cdns,xtfpga-i2s.txt | 18 + MAINTAINERS | 1 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/xtensa/Kconfig | 14 + sound/soc/xtensa/Makefile | 1 + sound/soc/xtensa/xtfpga-i2s.c | 647 +++++++++++++++++++++ 7 files changed, 683 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.txt create mode 100644 sound/soc/xtensa/Kconfig create mode 100644 sound/soc/xtensa/Makefile create mode 100644 sound/soc/xtensa/xtfpga-i2s.c diff --git a/Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.txt b/Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.txt new file mode 100644 index 0000000..befd125 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.txt @@ -0,0 +1,18 @@ +Bindings for I2S controller built into xtfpga Xtensa bitstreams. + +Required properties: +- compatible: shall be "cdns,xtfpga-i2s". +- reg: memory region (address and length) with device registers. +- interrupts: interrupt for the device. +- clocks: phandle to the clk used as master clock. I2S bus clock + is derived from it. + +Examples: + + i2s0: xtfpga-i2s@0d080000 { + #sound-dai-cells = <0>; + compatible = "cdns,xtfpga-i2s"; + reg = <0x0d080000 0x40>; + interrupts = <2 1>; + clocks = <&cdce706 4>; + }; diff --git a/MAINTAINERS b/MAINTAINERS index acb4523..d7bfeec 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -10204,6 +10204,7 @@ M: Max Filippov <jcmvbkbc(a)gmail.com> L: linux-xtensa(a)linux-xtensa.org S: Maintained F: drivers/spi/spi-xtensa-xtfpga.c +F: sound/soc/xtensa/xtfpga-i2s.c YAM DRIVER FOR AX.25 M: Jean-Paul Roubelat <jpr(a)f6fbb.org> diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 0e96233..9ca12e7 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -56,6 +56,7 @@ source "sound/soc/spear/Kconfig" source "sound/soc/tegra/Kconfig" source "sound/soc/txx9/Kconfig" source "sound/soc/ux500/Kconfig" +source "sound/soc/xtensa/Kconfig" # Supported codecs source "sound/soc/codecs/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 534714a..99d7222 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -33,3 +33,4 @@ obj-$(CONFIG_SND_SOC) += spear/ obj-$(CONFIG_SND_SOC) += tegra/ obj-$(CONFIG_SND_SOC) += txx9/ obj-$(CONFIG_SND_SOC) += ux500/ +obj-$(CONFIG_SND_SOC) += xtensa/ diff --git a/sound/soc/xtensa/Kconfig b/sound/soc/xtensa/Kconfig new file mode 100644 index 0000000..98d6e0c --- /dev/null +++ b/sound/soc/xtensa/Kconfig @@ -0,0 +1,14 @@ +config SND_SOC_XTFPGA_I2S + tristate + select REGMAP_MMIO + +config SND_SOC_XTENSA_XTFPGA + tristate "SoC Audio for xtensa xtfpga" + depends on XTENSA_PLATFORM_XTFPGA + select SND_SOC_XTFPGA_I2S + select SND_SOC_TLV320AIC23_SPI + select SND_SIMPLE_CARD + help + Say Y or M if you want to add support for codecs attached to + xtfpga daughter board. You will also need to select the platform + to support below. diff --git a/sound/soc/xtensa/Makefile b/sound/soc/xtensa/Makefile new file mode 100644 index 0000000..97b8eb5 --- /dev/null +++ b/sound/soc/xtensa/Makefile @@ -0,0 +1 @@ +obj-$(CONFIG_SND_SOC_XTFPGA_I2S) += xtfpga-i2s.o diff --git a/sound/soc/xtensa/xtfpga-i2s.c b/sound/soc/xtensa/xtfpga-i2s.c new file mode 100644 index 0000000..393d5c3 --- /dev/null +++ b/sound/soc/xtensa/xtfpga-i2s.c @@ -0,0 +1,647 @@ +/* + * Xtfpga I2S controller driver + * + * Copyright (c) 2014 Cadence Design Systems Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/clk.h> +#include <linux/io.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/platform_device.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#define DRV_NAME "xtfpga-i2s" + +#define XTFPGA_I2S_VERSION 0x00 +#define XTFPGA_I2S_CONFIG 0x04 +#define XTFPGA_I2S_INT_MASK 0x08 +#define XTFPGA_I2S_INT_STATUS 0x0c +#define XTFPGA_I2S_CHAN0_DATA 0x10 +#define XTFPGA_I2S_CHAN1_DATA 0x14 +#define XTFPGA_I2S_CHAN2_DATA 0x18 +#define XTFPGA_I2S_CHAN3_DATA 0x1c + +#define XTFPGA_I2S_CONFIG_TX_ENABLE 0x1 +#define XTFPGA_I2S_CONFIG_INT_ENABLE 0x2 +#define XTFPGA_I2S_CONFIG_LEFT 0x4 +#define XTFPGA_I2S_CONFIG_RATIO_BASE 8 +#define XTFPGA_I2S_CONFIG_RATIO_MASK 0x0000ff00 +#define XTFPGA_I2S_CONFIG_RES_BASE 16 +#define XTFPGA_I2S_CONFIG_RES_MASK 0x003f0000 +#define XTFPGA_I2S_CONFIG_LEVEL_BASE 24 +#define XTFPGA_I2S_CONFIG_LEVEL_MASK 0x0f000000 +#define XTFPGA_I2S_CONFIG_CHANNEL_BASE 28 + +#define XTFPGA_I2S_INT_UNDERRUN 0x1 +#define XTFPGA_I2S_INT_LEVEL 0x2 +#define XTFPGA_I2S_INT_VALID 0x3 + +#define XTFPGA_I2S_FIFO_SIZE 8192 +#define XTFPGA_I2S_CONFIG_DEFAULT_LEVEL 3 + +/* + * I2S controller operation: + * + * Enabling TX: output 1 period of zeros (starting with left channel) + * and then queued data. + * + * Level status and interrupt: whenever FIFO level is below FIFO trigger, + * level status is 1 and an IRQ is asserted (if enabled). + * + * Underrun status and interrupt: whenever FIFO is empty, underrun status + * is 1 and an IRQ is asserted (if enabled). + */ +struct xtfpga_i2s { + struct device *dev; + struct clk *clk; + bool clk_enabled; + struct regmap *regmap; + void __iomem *regs; + + struct snd_pcm_substream *tx_substream; + unsigned tx_ptr; /* next frame index in the sample buffer */ + unsigned tx_fifo_sz; /* current fifo level estimate */ + unsigned tx_fifo_low; /* FIFO level when level interrupt occurs */ + unsigned tx_fifo_high; /* maximal FIFO level */ + struct tasklet_struct refill_fifo; /* FIFO refilling tasklet */ +}; + +static bool xtfpga_i2s_wr_reg(struct device *dev, unsigned int reg) +{ + return reg >= XTFPGA_I2S_CONFIG; +} + +static bool xtfpga_i2s_rd_reg(struct device *dev, unsigned int reg) +{ + return reg < XTFPGA_I2S_CHAN0_DATA; +} + +static bool xtfpga_i2s_volatile_reg(struct device *dev, unsigned int reg) +{ + return reg == XTFPGA_I2S_INT_STATUS; +} + +static const struct regmap_config xtfpga_i2s_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = XTFPGA_I2S_CHAN3_DATA, + .writeable_reg = xtfpga_i2s_wr_reg, + .readable_reg = xtfpga_i2s_rd_reg, + .volatile_reg = xtfpga_i2s_volatile_reg, + .cache_type = REGCACHE_FLAT, +}; + +static void xtfpga_i2s_push_tx(struct xtfpga_i2s *i2s) +{ + struct snd_pcm_substream *tx_substream; + struct snd_pcm_runtime *runtime; + + rcu_read_lock(); + tx_substream = rcu_dereference(i2s->tx_substream); + runtime = tx_substream->runtime; +#define xtfpga_i2s_tx_loop(i2s, runtime, channels, sample_bits) \ + do { \ + const u##sample_bits (*p)[channels] = \ + (void *)runtime->dma_area; \ + for (; i2s->tx_fifo_sz < i2s->tx_fifo_high; \ + i2s->tx_fifo_sz += 2) { \ + iowrite32(p[i2s->tx_ptr][0], \ + i2s->regs + XTFPGA_I2S_CHAN0_DATA); \ + iowrite32(p[i2s->tx_ptr][channels - 1], \ + i2s->regs + XTFPGA_I2S_CHAN0_DATA); \ + if (++i2s->tx_ptr >= runtime->buffer_size) \ + i2s->tx_ptr = 0; \ + } \ + } while (0) + + switch (runtime->sample_bits | (runtime->channels << 8)) { + case 0x110: + xtfpga_i2s_tx_loop(i2s, runtime, 1, 16); + break; + case 0x210: + xtfpga_i2s_tx_loop(i2s, runtime, 2, 16); + break; + case 0x120: + xtfpga_i2s_tx_loop(i2s, runtime, 1, 32); + break; + case 0x220: + xtfpga_i2s_tx_loop(i2s, runtime, 2, 32); + break; + } + rcu_read_unlock(); +} + +static void xtfpga_i2s_irq_update_config(struct xtfpga_i2s *i2s, + unsigned int_status) +{ + if (!(int_status & XTFPGA_I2S_INT_LEVEL)) + regmap_write(i2s->regmap, XTFPGA_I2S_INT_MASK, + XTFPGA_I2S_INT_VALID); + else if (!(int_status & XTFPGA_I2S_INT_UNDERRUN)) + regmap_write(i2s->regmap, XTFPGA_I2S_INT_MASK, + XTFPGA_I2S_INT_UNDERRUN); + + if (!(int_status & XTFPGA_I2S_INT_UNDERRUN)) + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_INT_ENABLE | + XTFPGA_I2S_CONFIG_TX_ENABLE, + XTFPGA_I2S_CONFIG_INT_ENABLE | + XTFPGA_I2S_CONFIG_TX_ENABLE); +} + +static void xtfpga_i2s_refill_fifo(struct xtfpga_i2s *i2s) +{ + struct snd_pcm_substream *tx_substream; + unsigned int_status; + unsigned i; + + rcu_read_lock(); + tx_substream = rcu_dereference(i2s->tx_substream); + if (tx_substream && snd_pcm_running(tx_substream)) { + for (i = 0; i < 2; ++i) { + xtfpga_i2s_push_tx(i2s); + regmap_write(i2s->regmap, XTFPGA_I2S_INT_STATUS, + XTFPGA_I2S_INT_VALID); + regmap_read(i2s->regmap, XTFPGA_I2S_INT_STATUS, + &int_status); + + if (!(int_status & XTFPGA_I2S_INT_LEVEL)) + break; + i2s->tx_fifo_sz = i2s->tx_fifo_low; + } + } else { + regmap_read(i2s->regmap, XTFPGA_I2S_INT_STATUS, + &int_status); + } + rcu_read_unlock(); + xtfpga_i2s_irq_update_config(i2s, int_status); +} + +static void xtfpga_i2s_refill_fifo_tasklet(unsigned long p) +{ + xtfpga_i2s_refill_fifo((struct xtfpga_i2s *)p); +} + +static irqreturn_t xtfpga_i2s_interrupt(int irq, void *dev_id) +{ + struct xtfpga_i2s *i2s = dev_id; + struct snd_pcm_substream *tx_substream; + int tx_active; + unsigned int_status; + + regmap_read(i2s->regmap, XTFPGA_I2S_INT_STATUS, &int_status); + if (!(int_status & XTFPGA_I2S_INT_VALID)) + return IRQ_NONE; + + if (int_status & XTFPGA_I2S_INT_UNDERRUN) { + i2s->tx_fifo_sz = 0; + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_INT_ENABLE | + XTFPGA_I2S_CONFIG_TX_ENABLE, 0); + } else { + i2s->tx_fifo_sz = i2s->tx_fifo_low; + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_INT_ENABLE, 0); + } + + rcu_read_lock(); + tx_substream = rcu_dereference(i2s->tx_substream); + tx_active = tx_substream && snd_pcm_running(tx_substream); + if (tx_active) { + snd_pcm_period_elapsed(tx_substream); + if (int_status & XTFPGA_I2S_INT_UNDERRUN) + dev_dbg_ratelimited(i2s->dev, "%s: underrun\n", + __func__); + tx_substream = rcu_dereference(i2s->tx_substream); + tx_active = tx_substream && snd_pcm_running(tx_substream); + } + rcu_read_unlock(); + + if (tx_active) { + if (i2s->tx_fifo_high < 256) + xtfpga_i2s_refill_fifo(i2s); + else + tasklet_hi_schedule(&i2s->refill_fifo); + } else { + xtfpga_i2s_irq_update_config(i2s, int_status); + } + + return IRQ_HANDLED; +} + +static int xtfpga_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct xtfpga_i2s *i2s = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_set_dma_data(dai, substream, i2s); + return 0; +} + +static int xtfpga_i2s_start(struct xtfpga_i2s *i2s, + struct snd_pcm_substream *substream) +{ + i2s->tx_ptr = 0; + rcu_assign_pointer(i2s->tx_substream, substream); + + xtfpga_i2s_push_tx(i2s); + regmap_write(i2s->regmap, XTFPGA_I2S_INT_STATUS, XTFPGA_I2S_INT_VALID); + regmap_write(i2s->regmap, XTFPGA_I2S_INT_MASK, XTFPGA_I2S_INT_VALID); + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_INT_ENABLE | + XTFPGA_I2S_CONFIG_TX_ENABLE, + XTFPGA_I2S_CONFIG_INT_ENABLE | + XTFPGA_I2S_CONFIG_TX_ENABLE); + return 0; +} + +static int xtfpga_i2s_stop(struct xtfpga_i2s *i2s, + struct snd_pcm_substream *substream) +{ + rcu_assign_pointer(i2s->tx_substream, NULL); + return 0; +} + +static int xtfpga_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + int ret; + struct xtfpga_i2s *i2s = snd_soc_dai_get_drvdata(dai); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ret = xtfpga_i2s_start(i2s, substream); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ret = xtfpga_i2s_stop(i2s, substream); + break; + + default: + ret = -EINVAL; + break; + } + return ret; +} + +static int xtfpga_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct xtfpga_i2s *i2s = snd_soc_dai_get_drvdata(dai); + unsigned srate = params_rate(params); + unsigned channels = params_channels(params); + unsigned period_size = params_period_size(params); + unsigned sample_size = snd_pcm_format_width(params_format(params)); + unsigned freq, ratio, level; + int err; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + case SNDRV_PCM_FORMAT_S32_LE: + break; + + default: + return -EINVAL; + } + + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_RES_MASK, + sample_size << XTFPGA_I2S_CONFIG_RES_BASE); + + if (i2s->clk_enabled) { + clk_disable_unprepare(i2s->clk); + i2s->clk_enabled = false; + } + freq = 256 * srate; + err = clk_set_rate(i2s->clk, freq); + if (err < 0) + return err; + + err = clk_prepare_enable(i2s->clk); + if (err < 0) + return err; + i2s->clk_enabled = true; + + ratio = (freq - srate * sample_size * 8) / + (srate * sample_size * 4); + + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_RATIO_MASK, + ratio << XTFPGA_I2S_CONFIG_RATIO_BASE); + + i2s->tx_fifo_low = XTFPGA_I2S_FIFO_SIZE / 2; + for (level = 1; + /* period_size * 2: FIFO always gets 2 samples per frame */ + i2s->tx_fifo_low / 2 >= period_size * 2; + ++level) + i2s->tx_fifo_low /= 2; + + i2s->tx_fifo_high = 2 * i2s->tx_fifo_low; + i2s->tx_fifo_sz = 0; + i2s->tx_ptr = 0; + + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_LEVEL_MASK, + level << XTFPGA_I2S_CONFIG_LEVEL_BASE); + + dev_dbg(i2s->dev, + "%s srate: %u, channels: %u, sample_size: %u, period_size: %u\n", + __func__, srate, channels, sample_size, period_size); + dev_dbg(i2s->dev, "%s freq: %u, ratio: %u, level: %u\n", + __func__, freq, ratio, level); + + return 0; +} + +static int xtfpga_i2s_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct xtfpga_i2s *i2s = snd_soc_dai_get_drvdata(dai); + + if (i2s->clk_enabled) { + clk_disable_unprepare(i2s->clk); + i2s->clk_enabled = false; + } + return 0; +} + +static int xtfpga_i2s_set_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) + return -EINVAL; + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) + return -EINVAL; + if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_I2S) + return -EINVAL; + + return 0; +} + +/* PCM */ + +static const struct snd_pcm_hardware xtfpga_pcm_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BLOCK_TRANSFER, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 1, + .channels_max = 2, + .period_bytes_min = 2, + .period_bytes_max = XTFPGA_I2S_FIFO_SIZE / 2 * 8, + .periods_min = 2, + .periods_max = XTFPGA_I2S_FIFO_SIZE * 8 / 2, + .buffer_bytes_max = XTFPGA_I2S_FIFO_SIZE * 8, + .fifo_size = 16, +}; + +static int xtfpga_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + void *p; + + snd_soc_set_runtime_hwparams(substream, &xtfpga_pcm_hardware); + p = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + runtime->private_data = p; + + return 0; +} + +static int xtfpga_pcm_close(struct snd_pcm_substream *substream) +{ + synchronize_rcu(); + return 0; +} + +static int xtfpga_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + int ret; + + ret = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + return ret; +} + +static int xtfpga_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + int ret; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ret = 0; + break; + + default: + ret = -EINVAL; + break; + } + return ret; +} + +static snd_pcm_uframes_t xtfpga_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct xtfpga_i2s *i2s = runtime->private_data; + snd_pcm_uframes_t pos = i2s->tx_ptr; + + return pos < runtime->buffer_size ? pos : 0; +} + +static int xtfpga_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_card *card = rtd->card->snd_card; + size_t size = xtfpga_pcm_hardware.buffer_bytes_max; + + return snd_pcm_lib_preallocate_pages_for_all(rtd->pcm, + SNDRV_DMA_TYPE_DEV, + card->dev, size, size); +} + +static void xtfpga_pcm_free(struct snd_pcm *pcm) +{ + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static struct snd_pcm_ops xtfpga_pcm_ops = { + .open = xtfpga_pcm_open, + .close = xtfpga_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = xtfpga_pcm_hw_params, + .trigger = xtfpga_pcm_trigger, + .pointer = xtfpga_pcm_pointer, +}; + +struct snd_soc_platform_driver xtfpga_soc_platform = { + .pcm_new = xtfpga_pcm_new, + .pcm_free = xtfpga_pcm_free, + .ops = &xtfpga_pcm_ops, +}; + +static const struct snd_soc_component_driver xtfpga_i2s_component = { + .name = DRV_NAME, +}; + +static const struct snd_soc_dai_ops xtfpga_i2s_dai_ops = { + .startup = xtfpga_i2s_startup, + .trigger = xtfpga_i2s_trigger, + .hw_params = xtfpga_i2s_hw_params, + .hw_free = xtfpga_i2s_hw_free, + .set_fmt = xtfpga_i2s_set_fmt, +}; + +static struct snd_soc_dai_driver xtfpga_i2s_dai[] = { + { + .name = "xtfpga-i2s", + .id = 0, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &xtfpga_i2s_dai_ops, + }, +}; + +static int xtfpga_i2s_probe(struct platform_device *pdev) +{ + struct xtfpga_i2s *i2s; + struct resource *mem, *irq; + int err; + + i2s = devm_kzalloc(&pdev->dev, sizeof(*i2s), GFP_KERNEL); + if (!i2s) { + err = -ENOMEM; + goto err; + } + dev_set_drvdata(&pdev->dev, i2s); + i2s->dev = &pdev->dev; + dev_dbg(&pdev->dev, "dev: %p, i2s: %p\n", &pdev->dev, i2s); + + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!mem) { + dev_err(&pdev->dev, "No memory resource\n"); + err = -ENODEV; + goto err; + } + i2s->regs = devm_ioremap_resource(&pdev->dev, mem); + if (IS_ERR(i2s->regs)) { + err = PTR_ERR(i2s->regs); + goto err; + } + + i2s->regmap = devm_regmap_init_mmio(&pdev->dev, i2s->regs, + &xtfpga_i2s_regmap_config); + if (IS_ERR(i2s->regmap)) { + dev_err(&pdev->dev, "regmap init failed\n"); + err = PTR_ERR(i2s->regmap); + goto err; + } + + i2s->clk = devm_clk_get(&pdev->dev, NULL); + if (IS_ERR(i2s->clk)) { + dev_err(&pdev->dev, "couldn't get clock\n"); + err = PTR_ERR(i2s->clk); + goto err; + } + + irq = platform_get_resource(pdev, IORESOURCE_IRQ, 0); + if (!irq) { + dev_err(&pdev->dev, "No IRQ resource\n"); + err = -ENODEV; + goto err; + } + err = devm_request_irq(&pdev->dev, irq->start, xtfpga_i2s_interrupt, + IRQF_SHARED, pdev->name, i2s); + if (err < 0) { + dev_err(&pdev->dev, "request_irq failed\n"); + goto err; + } + tasklet_init(&i2s->refill_fifo, xtfpga_i2s_refill_fifo_tasklet, + (unsigned long)i2s); + + regmap_write(i2s->regmap, XTFPGA_I2S_CONFIG, + (0x1 << XTFPGA_I2S_CONFIG_CHANNEL_BASE)); + regmap_write(i2s->regmap, XTFPGA_I2S_INT_STATUS, XTFPGA_I2S_INT_VALID); + regmap_write(i2s->regmap, XTFPGA_I2S_INT_MASK, XTFPGA_I2S_INT_UNDERRUN); + + err = snd_soc_register_platform(&pdev->dev, &xtfpga_soc_platform); + if (err < 0) { + dev_err(&pdev->dev, "couldn't register platform\n"); + goto err; + } + err = devm_snd_soc_register_component(&pdev->dev, + &xtfpga_i2s_component, + xtfpga_i2s_dai, + ARRAY_SIZE(xtfpga_i2s_dai)); + if (err < 0) + dev_err(&pdev->dev, "couldn't register component\n"); +err: + if (err) + dev_err(&pdev->dev, "%s: err = %d\n", __func__, err); + return err; +} + +static int xtfpga_i2s_remove(struct platform_device *pdev) +{ + struct xtfpga_i2s *i2s = dev_get_drvdata(&pdev->dev); + + if (i2s) { + tasklet_kill(&i2s->refill_fifo); + if (i2s->regmap && !IS_ERR(i2s->regmap)) { + regmap_write(i2s->regmap, XTFPGA_I2S_CONFIG, 0); + regmap_write(i2s->regmap, XTFPGA_I2S_INT_MASK, 0); + regmap_write(i2s->regmap, XTFPGA_I2S_INT_STATUS, + XTFPGA_I2S_INT_VALID); + } + if (i2s->clk_enabled) + clk_disable_unprepare(i2s->clk); + } + + return 0; +} + +#ifdef CONFIG_OF +static const struct of_device_id xtfpga_i2s_of_match[] = { + { .compatible = "cdns,xtfpga-i2s", }, + {}, +}; +MODULE_DEVICE_TABLE(of, xtfpga_i2s_of_match); +#endif + +static struct platform_driver xtfpga_i2s_driver = { + .probe = xtfpga_i2s_probe, + .remove = xtfpga_i2s_remove, + .driver = { + .name = "xtfpga-i2s", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(xtfpga_i2s_of_match), + }, +}; + +module_platform_driver(xtfpga_i2s_driver); + +MODULE_AUTHOR("Max Filippov <jcmvbkbc(a)gmail.com>"); +MODULE_DESCRIPTION("xtfpga I2S controller driver"); +MODULE_LICENSE("GPL v2"); -- 1.8.1.4

4 18

[alsa-devel] [PATCH] ASoC: Fix snd_soc_find_dai() matching component by name
by Lars-Peter Clausen 29 Oct '14

29 Oct '14

Commit 14621c7e5e72 ("ASoC: Consolidate CPU and CODEC DAI lookup") consolidated the lookup of CPU DAIs and CODEC DAIs into a single function. When matching a component by name for CODEC DAIs the code previous to the patch compared the name in the DAI link table with component->name. For CPU DAIs the code compared to dev_name(component->dev). The newly introduced function ended up using the later as well. For most components dev_name(component->dev) and component->name are the same. The main notable exception are I2C devices where the driver name and the device name are concatenated to form the component name. By using dev_name(component->dev) instead of component->name the patch broke the matching of I2C CODECs by name. This patch restores the original behavior by using component->name instead of dev_name(component->dev). This will be safe even for CPU DAIs since for CPU DAIs both are the same. Fixes: 14621c7e5e72 ("ASoC: Consolidate CPU and CODEC DAI lookup") Reported-by: Dmitry Eremin-Solenikov <dbaryshkov(a)gmail.com> Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 62fbb63..b900f0d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -884,7 +884,7 @@ static struct snd_soc_dai *snd_soc_find_dai( list_for_each_entry(component, &component_list, list) { if (dlc->of_node && component->dev->of_node != dlc->of_node) continue; - if (dlc->name && strcmp(dev_name(component->dev), dlc->name)) + if (dlc->name && strcmp(component->name, dlc->name)) continue; list_for_each_entry(dai, &component->dai_list, list) { if (dlc->dai_name && strcmp(dai->name, dlc->dai_name)) -- 1.8.0

2 1

[alsa-devel] [PATCH 1/3] ASoC: Intel: Add PM support to the HSW/BDW DSP core.
by Liam Girdwood 29 Oct '14

29 Oct '14

Add support for PM wake, sleep and stall calls to the core HSW/BDW driver. This includes reworking the reset and boot code and adding new calls for setting D3/D0 state. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/intel/sst-haswell-dsp.c | 158 ++++++++++++++++++++++++++++---------- 1 file changed, 119 insertions(+), 39 deletions(-) diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c index 5058dc8..86aea34 100644 --- a/sound/soc/intel/sst-haswell-dsp.c +++ b/sound/soc/intel/sst-haswell-dsp.c @@ -247,8 +247,67 @@ static irqreturn_t hsw_irq(int irq, void *context) return ret; } -static void hsw_boot(struct sst_dsp *sst) +static void hsw_set_dsp_D3(struct sst_dsp *sst) +{ + u32 val; + + /* switch off audio PLL, DRAM & IRAM blocks */ + val = readl(sst->addr.pci_cfg + SST_VDRTCTL0); + val |= SST_VDRTCL0_APLLSE_MASK | SST_VDRTCL0_DSRAMPGE_MASK | + SST_VDRTCL0_ISRAMPGE_MASK; + writel(val, sst->addr.pci_cfg + SST_VDRTCTL0); + + /* Set D3 state */ + val = readl(sst->addr.pci_cfg + SST_PMCS); + val |= SST_PMCS_PS_MASK; + writel(val, sst->addr.pci_cfg + SST_PMCS); +} + +static void hsw_reset(struct sst_dsp *sst) +{ + /* put DSP into reset and stall */ + sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, + SST_CSR_RST | SST_CSR_STALL, + SST_CSR_RST | SST_CSR_STALL); + + /* keep in reset for 10ms */ + mdelay(10); + + /* take DSP out of reset and keep stalled for FW loading */ + sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, + SST_CSR_RST | SST_CSR_STALL, SST_CSR_STALL); +} + +static int hsw_set_dsp_D0(struct sst_dsp *sst) { + int tries = 10; + u32 reg; + + /* Set D0 state */ + reg = readl(sst->addr.pci_cfg + SST_PMCS); + reg &= ~SST_PMCS_PS_MASK; + writel(reg, sst->addr.pci_cfg + SST_PMCS); + + /* check that ADSP shim is enabled */ + while (tries--) { + reg = readl(sst->addr.pci_cfg + SST_PMCS) & SST_PMCS_PS_MASK; + if (reg == 0) + goto finish; + + msleep(1); + } + + return -ENODEV; + +finish: + hsw_reset(sst); + + /* switch on audio PLL, DRAM & IRAM blocks */ + reg = readl(sst->addr.pci_cfg + SST_VDRTCTL0); + reg &= ~(SST_VDRTCL0_APLLSE_MASK | SST_VDRTCL0_DSRAMPGE_MASK | + SST_VDRTCL0_ISRAMPGE_MASK); + writel(reg, sst->addr.pci_cfg + SST_VDRTCTL0); + /* select SSP1 19.2MHz base clock, SSP clock 0, turn off Low Power Clock */ sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, SST_CSR_S1IOCS | SST_CSR_SBCS1 | SST_CSR_LPCS, 0x0); @@ -267,30 +326,73 @@ static void hsw_boot(struct sst_dsp *sst) sst_dsp_shim_update_bits_unlocked(sst, SST_CSR2, SST_CSR2_SDFD_SSP1, SST_CSR2_SDFD_SSP1); - /* enable DMA engine 0,1 all channels to access host memory */ - sst_dsp_shim_update_bits_unlocked(sst, SST_HMDC, - SST_HMDC_HDDA1(0xff) | SST_HMDC_HDDA0(0xff), - SST_HMDC_HDDA1(0xff) | SST_HMDC_HDDA0(0xff)); + /* set on-demond mode on engine 0,1 for all channels */ + sst_dsp_shim_update_bits(sst, SST_HMDC, + SST_HMDC_HDDA_E0_ALLCH | SST_HMDC_HDDA_E1_ALLCH, + SST_HMDC_HDDA_E0_ALLCH | SST_HMDC_HDDA_E1_ALLCH); + + /* Enable Interrupt from both sides */ + sst_dsp_shim_update_bits(sst, SST_IMRX, (SST_IMRX_BUSY | SST_IMRX_DONE), + 0x0); + sst_dsp_shim_update_bits(sst, SST_IMRD, (SST_IMRD_DONE | SST_IMRD_BUSY | + SST_IMRD_SSP0 | SST_IMRD_DMAC), 0x0); + + /* clear IPC registers */ + sst_dsp_shim_write(sst, SST_IPCX, 0x0); + sst_dsp_shim_write(sst, SST_IPCD, 0x0); + sst_dsp_shim_write(sst, 0x80, 0x6); + sst_dsp_shim_write(sst, 0xe0, 0x300a); /* disable all clock gating */ writel(0x0, sst->addr.pci_cfg + SST_VDRTCTL2); + return 0; +} + +static void hsw_boot(struct sst_dsp *sst) +{ + /* set oportunistic mode on engine 0,1 for all channels */ + sst_dsp_shim_update_bits(sst, SST_HMDC, + SST_HMDC_HDDA_E0_ALLCH | SST_HMDC_HDDA_E1_ALLCH, 0); + /* set DSP to RUN */ sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, SST_CSR_STALL, 0x0); } -static void hsw_reset(struct sst_dsp *sst) +static void hsw_stall(struct sst_dsp *sst) { + /* stall DSP */ + sst_dsp_shim_update_bits(sst, SST_CSR, + SST_CSR_24MHZ_LPCS | SST_CSR_STALL, + SST_CSR_STALL | SST_CSR_24MHZ_LPCS); +} + +static void hsw_sleep(struct sst_dsp *sst) +{ + dev_dbg(sst->dev, "HSW_PM dsp runtime suspend\n"); + /* put DSP into reset and stall */ - sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, - SST_CSR_RST | SST_CSR_STALL, SST_CSR_RST | SST_CSR_STALL); + sst_dsp_shim_update_bits(sst, SST_CSR, + SST_CSR_24MHZ_LPCS | SST_CSR_RST | SST_CSR_STALL, + SST_CSR_RST | SST_CSR_STALL | SST_CSR_24MHZ_LPCS); - /* keep in reset for 10ms */ - mdelay(10); + hsw_set_dsp_D3(sst); + dev_dbg(sst->dev, "HSW_PM dsp runtime suspend exit\n"); +} - /* take DSP out of reset and keep stalled for FW loading */ - sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, - SST_CSR_RST | SST_CSR_STALL, SST_CSR_STALL); +static int hsw_wake(struct sst_dsp *sst) +{ + int ret; + + dev_dbg(sst->dev, "HSW_PM dsp runtime resume\n"); + + ret = hsw_set_dsp_D0(sst); + if (ret < 0) + return ret; + + dev_dbg(sst->dev, "HSW_PM dsp runtime resume exit\n"); + + return 0; } struct sst_adsp_memregion { @@ -431,27 +533,6 @@ static struct sst_block_ops sst_hsw_ops = { .disable = hsw_block_disable, }; -static int hsw_enable_shim(struct sst_dsp *sst) -{ - int tries = 10; - u32 reg; - - /* enable shim */ - reg = readl(sst->addr.pci_cfg + SST_SHIM_PM_REG); - writel(reg & ~0x3, sst->addr.pci_cfg + SST_SHIM_PM_REG); - - /* check that ADSP shim is enabled */ - while (tries--) { - reg = sst_dsp_shim_read_unlocked(sst, SST_CSR); - if (reg != 0xffffffff) - return 0; - - msleep(1); - } - - return -ENODEV; -} - static int hsw_init(struct sst_dsp *sst, struct sst_pdata *pdata) { const struct sst_adsp_memregion *region; @@ -490,7 +571,7 @@ static int hsw_init(struct sst_dsp *sst, struct sst_pdata *pdata) } /* enable the DSP SHIM */ - ret = hsw_enable_shim(sst); + ret = hsw_set_dsp_D0(sst); if (ret < 0) { dev_err(dev, "error: failed to set DSP D0 and reset SHIM\n"); return ret; @@ -500,10 +581,6 @@ static int hsw_init(struct sst_dsp *sst, struct sst_pdata *pdata) if (ret) return ret; - /* Enable Interrupt from both sides */ - sst_dsp_shim_update_bits_unlocked(sst, SST_IMRX, 0x3, 0x0); - sst_dsp_shim_update_bits_unlocked(sst, SST_IMRD, - (0x3 | 0x1 << 16 | 0x3 << 21), 0x0); /* register DSP memory blocks - ideally we should get this from ACPI */ for (i = 0; i < region_count; i++) { @@ -535,6 +612,9 @@ static void hsw_free(struct sst_dsp *sst) struct sst_ops haswell_ops = { .reset = hsw_reset, .boot = hsw_boot, + .stall = hsw_stall, + .wake = hsw_wake, + .sleep = hsw_sleep, .write = sst_shim32_write, .read = sst_shim32_read, .write64 = sst_shim32_write64, -- 2.1.0

2 3

[alsa-devel] [PATCH] ALSA: hda - Add workaround for CMI8888 snoop behavior
by Takashi Iwai 29 Oct '14

29 Oct '14

CMI8888 shows the stuttering playback when the snooping is disabled on the audio buffer. Meanwhile, we've got reports that CORB/RIRB doesn't work in the snooped mode. So, as a compromise, disable the snoop only for CORB/RIRB and enable the snoop for the stream buffers. The resultant patch became a bit ugly, unfortunately, but we still can live with it. Reported-and-tested-by: Geoffrey McRae <geoff(a)spacevs.com> Cc: <stable(a)vger.kernel.org> # 3.17+ Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index cfcca4c30d4d..9ab1e631cb32 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -374,6 +374,8 @@ static void __mark_pages_wc(struct azx *chip, struct snd_dma_buffer *dmab, bool #ifdef CONFIG_SND_DMA_SGBUF if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_SG) { struct snd_sg_buf *sgbuf = dmab->private_data; + if (chip->driver_type == AZX_DRIVER_CMEDIA) + return; /* deal with only CORB/RIRB buffers */ if (on) set_pages_array_wc(sgbuf->page_table, sgbuf->pages); else @@ -1769,7 +1771,7 @@ static void pcm_mmap_prepare(struct snd_pcm_substream *substream, #ifdef CONFIG_X86 struct azx_pcm *apcm = snd_pcm_substream_chip(substream); struct azx *chip = apcm->chip; - if (!azx_snoop(chip)) + if (!azx_snoop(chip) && chip->driver_type != AZX_DRIVER_CMEDIA) area->vm_page_prot = pgprot_writecombine(area->vm_page_prot); #endif } -- 2.1.2

1 0

[alsa-devel] [PATCH] ASoC: Intel: dw_pdata can be static
by Liam Girdwood 29 Oct '14

29 Oct '14

From: kbuild test robot <fengguang.wu(a)intel.com> sound/soc/intel/sst-firmware.c:172:29: sparse: symbol 'dw_pdata' was not declared. Should it be static? Signed-off-by: Fengguang Wu <fengguang.wu(a)intel.com> --- sound/soc/intel/sst-firmware.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index 692a6ae..35788ad 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -169,7 +169,7 @@ err: return ret; } -struct dw_dma_platform_data dw_pdata = { +static struct dw_dma_platform_data dw_pdata = { .is_private = 1, .chan_allocation_order = CHAN_ALLOCATION_ASCENDING, .chan_priority = CHAN_PRIORITY_ASCENDING, -- 2.1.0

2 2

[alsa-devel] [PATCH] ALSA: hdspm: remove unused variable
by Sudip Mukherjee 29 Oct '14

29 Oct '14

removed the unused variables. These variables were only being assigned some value, but the values were never being used. it has been build tested after removing the variables. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- sound/pci/rme9652/hdspm.c | 23 +++-------------------- 1 file changed, 3 insertions(+), 20 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 7f7277b..e09348c1 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -1257,14 +1257,13 @@ static int hdspm_rate_multiplier(struct hdspm *hdspm, int rate) /* check for external sample rate, returns the sample rate in Hz*/ static int hdspm_external_sample_rate(struct hdspm *hdspm) { - unsigned int status, status2, timecode; + unsigned int status, status2; int syncref, rate = 0, rate_bits; switch (hdspm->io_type) { case AES32: status2 = hdspm_read(hdspm, HDSPM_statusRegister2); status = hdspm_read(hdspm, HDSPM_statusRegister); - timecode = hdspm_read(hdspm, HDSPM_timecodeRegister); syncref = hdspm_autosync_ref(hdspm); switch (syncref) { @@ -4862,18 +4861,15 @@ snd_hdspm_proc_read_madi(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct hdspm *hdspm = entry->private_data; - unsigned int status, status2, control, freq; + unsigned int status, status2; char *pref_sync_ref; char *autosync_ref; char *system_clock_mode; - char *insel; int x, x2; status = hdspm_read(hdspm, HDSPM_statusRegister); status2 = hdspm_read(hdspm, HDSPM_statusRegister2); - control = hdspm->control_register; - freq = hdspm_read(hdspm, HDSPM_timecodeRegister); snd_iprintf(buffer, "%s (Card #%d) Rev.%x Status2first3bits: %x\n", hdspm->card_name, hdspm->card->number + 1, @@ -4936,17 +4932,6 @@ snd_hdspm_proc_read_madi(struct snd_info_entry *entry, snd_iprintf(buffer, "Line out: %s\n", (hdspm->control_register & HDSPM_LineOut) ? "on " : "off"); - switch (hdspm->control_register & HDSPM_InputMask) { - case HDSPM_InputOptical: - insel = "Optical"; - break; - case HDSPM_InputCoaxial: - insel = "Coaxial"; - break; - default: - insel = "Unknown"; - } - snd_iprintf(buffer, "ClearTrackMarker = %s, Transmit in %s Channel Mode, " "Auto Input %s\n", @@ -5191,15 +5176,13 @@ snd_hdspm_proc_read_raydat(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct hdspm *hdspm = entry->private_data; - unsigned int status1, status2, status3, control, i; + unsigned int status1, status2, status3, i; unsigned int lock, sync; status1 = hdspm_read(hdspm, HDSPM_RD_STATUS_1); /* s1 */ status2 = hdspm_read(hdspm, HDSPM_RD_STATUS_2); /* freq */ status3 = hdspm_read(hdspm, HDSPM_RD_STATUS_3); /* s2 */ - control = hdspm->control_register; - snd_iprintf(buffer, "STATUS1: 0x%08x\n", status1); snd_iprintf(buffer, "STATUS2: 0x%08x\n", status2); snd_iprintf(buffer, "STATUS3: 0x%08x\n", status3); -- 1.8.1.2

2 1

[alsa-devel] [PATCH 1/2] ALSA: hda - Correct kerneldoc comments
by Takashi Iwai 29 Oct '14

29 Oct '14

Complete the missing parameters and fix anything wrong there. Just comment changes. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_auto_parser.c | 15 +++++- sound/pci/hda/hda_codec.c | 114 ++++++++++++++++++++++++++++++++++++++-- sound/pci/hda/hda_eld.c | 2 +- sound/pci/hda/hda_jack.c | 21 ++++++++ 4 files changed, 146 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index fcc5e478c9a1..7388958b01af 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -464,8 +464,12 @@ EXPORT_SYMBOL_GPL(snd_hda_get_input_pin_attr); /** * hda_get_input_pin_label - Give a label for the given input pin + * @codec: the HDA codec + * @item: ping config item to refer + * @pin: the pin NID + * @check_location: flag to add the jack location prefix * - * When check_location is true, the function checks the pin location + * When @check_location is true, the function checks the pin location * for mic and line-in pins, and set an appropriate prefix like "Front", * "Rear", "Internal". */ @@ -550,6 +554,9 @@ static int check_mic_location_need(struct hda_codec *codec, /** * hda_get_autocfg_input_label - Get a label for the given input + * @codec: the HDA codec + * @cfg: the parsed pin configuration + * @input: the input index number * * Get a label for the given input pin defined by the autocfg item. * Unlike hda_get_input_pin_label(), this function checks all inputs @@ -677,6 +684,12 @@ static int fill_audio_out_name(struct hda_codec *codec, hda_nid_t nid, /** * snd_hda_get_pin_label - Get a label for the given I/O pin + * @codec: the HDA codec + * @nid: pin NID + * @cfg: the parsed pin configuration + * @label: the string buffer to store + * @maxlen: the max length of string buffer (including termination) + * @indexp: the pointer to return the index number (for multiple ctls) * * Get a label for the given pin. This function works for both input and * output pins. When @cfg is given as non-NULL, the function tries to get diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 0025bf4c2f44..e15254204c72 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -416,7 +416,6 @@ static int read_and_add_raw_conns(struct hda_codec *codec, hda_nid_t nid) * snd_hda_get_conn_list - get connection list * @codec: the HDA codec * @nid: NID to parse - * @len: number of connection list entries * @listp: the pointer to store NID list * * Parses the connection list of the given widget and stores the pointer @@ -2004,6 +2003,7 @@ EXPORT_SYMBOL_GPL(query_amp_caps); * @codec: the HD-audio codec * @nid: the NID to query * @dir: either #HDA_INPUT or #HDA_OUTPUT + * @bits: bit mask to check the result * * Check whether the widget has the given amp capability for the direction. */ @@ -2023,7 +2023,7 @@ EXPORT_SYMBOL_GPL(snd_hda_check_amp_caps); * snd_hda_override_amp_caps - Override the AMP capabilities * @codec: the CODEC to clean up * @nid: the NID to clean up - * @direction: either #HDA_INPUT or #HDA_OUTPUT + * @dir: either #HDA_INPUT or #HDA_OUTPUT * @caps: the capability bits to set * * Override the cached AMP caps bits value by the given one. @@ -2320,6 +2320,8 @@ static u32 get_amp_max_value(struct hda_codec *codec, hda_nid_t nid, int dir, /** * snd_hda_mixer_amp_volume_info - Info callback for a standard AMP mixer + * @kcontrol: referred ctl element + * @uinfo: pointer to get/store the data * * The control element is supposed to have the private_value field * set up via HDA_COMPOSE_AMP_VAL*() or related macros. @@ -2381,6 +2383,8 @@ update_amp_value(struct hda_codec *codec, hda_nid_t nid, /** * snd_hda_mixer_amp_volume_get - Get callback for a standard AMP mixer volume + * @kcontrol: ctl element + * @ucontrol: pointer to get/store the data * * The control element is supposed to have the private_value field * set up via HDA_COMPOSE_AMP_VAL*() or related macros. @@ -2406,6 +2410,8 @@ EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_volume_get); /** * snd_hda_mixer_amp_volume_put - Put callback for a standard AMP mixer volume + * @kcontrol: ctl element + * @ucontrol: pointer to get/store the data * * The control element is supposed to have the private_value field * set up via HDA_COMPOSE_AMP_VAL*() or related macros. @@ -2436,6 +2442,10 @@ EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_volume_put); /** * snd_hda_mixer_amp_volume_put - TLV callback for a standard AMP mixer volume + * @kcontrol: ctl element + * @op_flag: operation flag + * @size: byte size of input TLV + * @_tlv: TLV data * * The control element is supposed to have the private_value field * set up via HDA_COMPOSE_AMP_VAL*() or related macros. @@ -3012,6 +3022,8 @@ EXPORT_SYMBOL_GPL(snd_hda_sync_vmaster_hook); /** * snd_hda_mixer_amp_switch_info - Info callback for a standard AMP mixer switch + * @kcontrol: referred ctl element + * @uinfo: pointer to get/store the data * * The control element is supposed to have the private_value field * set up via HDA_COMPOSE_AMP_VAL*() or related macros. @@ -3031,6 +3043,8 @@ EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_switch_info); /** * snd_hda_mixer_amp_switch_get - Get callback for a standard AMP mixer switch + * @kcontrol: ctl element + * @ucontrol: pointer to get/store the data * * The control element is supposed to have the private_value field * set up via HDA_COMPOSE_AMP_VAL*() or related macros. @@ -3057,6 +3071,8 @@ EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_switch_get); /** * snd_hda_mixer_amp_switch_put - Put callback for a standard AMP mixer switch + * @kcontrol: ctl element + * @ucontrol: pointer to get/store the data * * The control element is supposed to have the private_value field * set up via HDA_COMPOSE_AMP_VAL*() or related macros. @@ -3100,6 +3116,8 @@ EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_switch_put); /** * snd_hda_mixer_bind_switch_get - Get callback for a bound volume control + * @kcontrol: ctl element + * @ucontrol: pointer to get/store the data * * The control element is supposed to have the private_value field * set up via HDA_BIND_MUTE*() macros. @@ -3123,6 +3141,8 @@ EXPORT_SYMBOL_GPL(snd_hda_mixer_bind_switch_get); /** * snd_hda_mixer_bind_switch_put - Put callback for a bound volume control + * @kcontrol: ctl element + * @ucontrol: pointer to get/store the data * * The control element is supposed to have the private_value field * set up via HDA_BIND_MUTE*() macros. @@ -3153,6 +3173,8 @@ EXPORT_SYMBOL_GPL(snd_hda_mixer_bind_switch_put); /** * snd_hda_mixer_bind_ctls_info - Info callback for a generic bound control + * @kcontrol: referred ctl element + * @uinfo: pointer to get/store the data * * The control element is supposed to have the private_value field * set up via HDA_BIND_VOL() or HDA_BIND_SW() macros. @@ -3176,6 +3198,8 @@ EXPORT_SYMBOL_GPL(snd_hda_mixer_bind_ctls_info); /** * snd_hda_mixer_bind_ctls_get - Get callback for a generic bound control + * @kcontrol: ctl element + * @ucontrol: pointer to get/store the data * * The control element is supposed to have the private_value field * set up via HDA_BIND_VOL() or HDA_BIND_SW() macros. @@ -3199,6 +3223,8 @@ EXPORT_SYMBOL_GPL(snd_hda_mixer_bind_ctls_get); /** * snd_hda_mixer_bind_ctls_put - Put callback for a generic bound control + * @kcontrol: ctl element + * @ucontrol: pointer to get/store the data * * The control element is supposed to have the private_value field * set up via HDA_BIND_VOL() or HDA_BIND_SW() macros. @@ -3228,6 +3254,10 @@ EXPORT_SYMBOL_GPL(snd_hda_mixer_bind_ctls_put); /** * snd_hda_mixer_bind_tlv - TLV callback for a generic bound control + * @kcontrol: ctl element + * @op_flag: operation flag + * @size: byte size of input TLV + * @tlv: TLV data * * The control element is supposed to have the private_value field * set up via HDA_BIND_VOL() macro. @@ -4305,6 +4335,7 @@ static struct hda_rate_tbl rate_bits[] = { * @channels: the number of channels * @format: the PCM format (SNDRV_PCM_FORMAT_XXX) * @maxbps: the max. bps + * @spdif_ctls: HD-audio SPDIF status bits (0 if irrelevant) * * Calculate the format bitset from the given rate, channels and th PCM format. * @@ -4980,6 +5011,7 @@ static void __snd_hda_power_down(struct hda_codec *codec) * snd_hda_power_save - Power-up/down/sync the codec * @codec: HD-audio codec * @delta: the counter delta to change + * @d3wait: sync for D3 transition complete * * Change the power-up counter via @delta, and power up or down the hardware * appropriately. For the power-down, queue to the delayed action. @@ -5055,6 +5087,10 @@ EXPORT_SYMBOL_GPL(snd_hda_check_amp_list_power); /** * snd_hda_ch_mode_info - Info callback helper for the channel mode enum + * @codec: the HDA codec + * @uinfo: pointer to get/store the data + * @chmode: channel mode array + * @num_chmodes: channel mode array size */ int snd_hda_ch_mode_info(struct hda_codec *codec, struct snd_ctl_elem_info *uinfo, @@ -5074,6 +5110,11 @@ EXPORT_SYMBOL_GPL(snd_hda_ch_mode_info); /** * snd_hda_ch_mode_get - Get callback helper for the channel mode enum + * @codec: the HDA codec + * @ucontrol: pointer to get/store the data + * @chmode: channel mode array + * @num_chmodes: channel mode array size + * @max_channels: max number of channels */ int snd_hda_ch_mode_get(struct hda_codec *codec, struct snd_ctl_elem_value *ucontrol, @@ -5095,6 +5136,11 @@ EXPORT_SYMBOL_GPL(snd_hda_ch_mode_get); /** * snd_hda_ch_mode_put - Put callback helper for the channel mode enum + * @codec: the HDA codec + * @ucontrol: pointer to get/store the data + * @chmode: channel mode array + * @num_chmodes: channel mode array size + * @max_channelsp: pointer to store the max channels */ int snd_hda_ch_mode_put(struct hda_codec *codec, struct snd_ctl_elem_value *ucontrol, @@ -5123,6 +5169,8 @@ EXPORT_SYMBOL_GPL(snd_hda_ch_mode_put); /** * snd_hda_input_mux_info_info - Info callback helper for the input-mux enum + * @imux: imux helper object + * @uinfo: pointer to get/store the data */ int snd_hda_input_mux_info(const struct hda_input_mux *imux, struct snd_ctl_elem_info *uinfo) @@ -5144,6 +5192,11 @@ EXPORT_SYMBOL_GPL(snd_hda_input_mux_info); /** * snd_hda_input_mux_info_put - Put callback helper for the input-mux enum + * @codec: the HDA codec + * @imux: imux helper object + * @ucontrol: pointer to get/store the data + * @nid: input mux NID + * @cur_val: pointer to get/store the current imux value */ int snd_hda_input_mux_put(struct hda_codec *codec, const struct hda_input_mux *imux, @@ -5168,7 +5221,13 @@ int snd_hda_input_mux_put(struct hda_codec *codec, EXPORT_SYMBOL_GPL(snd_hda_input_mux_put); -/* +/** + * snd_hda_enum_helper_info - Helper for simple enum ctls + * @kcontrol: ctl element + * @uinfo: pointer to get/store the data + * @num_items: number of enum items + * @texts: enum item string array + * * process kcontrol info callback of a simple string enum array * when @num_items is 0 or @texts is NULL, assume a boolean enum array */ @@ -5257,6 +5316,8 @@ EXPORT_SYMBOL_GPL(snd_hda_bus_reboot_notify); /** * snd_hda_multi_out_dig_open - open the digital out in the exclusive mode + * @codec: the HDA codec + * @mout: hda_multi_out object */ int snd_hda_multi_out_dig_open(struct hda_codec *codec, struct hda_multi_out *mout) @@ -5273,6 +5334,11 @@ EXPORT_SYMBOL_GPL(snd_hda_multi_out_dig_open); /** * snd_hda_multi_out_dig_prepare - prepare the digital out stream + * @codec: the HDA codec + * @mout: hda_multi_out object + * @stream_tag: stream tag to assign + * @format: format id to assign + * @substream: PCM substream to assign */ int snd_hda_multi_out_dig_prepare(struct hda_codec *codec, struct hda_multi_out *mout, @@ -5289,6 +5355,8 @@ EXPORT_SYMBOL_GPL(snd_hda_multi_out_dig_prepare); /** * snd_hda_multi_out_dig_cleanup - clean-up the digital out stream + * @codec: the HDA codec + * @mout: hda_multi_out object */ int snd_hda_multi_out_dig_cleanup(struct hda_codec *codec, struct hda_multi_out *mout) @@ -5302,6 +5370,8 @@ EXPORT_SYMBOL_GPL(snd_hda_multi_out_dig_cleanup); /** * snd_hda_multi_out_dig_close - release the digital out stream + * @codec: the HDA codec + * @mout: hda_multi_out object */ int snd_hda_multi_out_dig_close(struct hda_codec *codec, struct hda_multi_out *mout) @@ -5315,6 +5385,10 @@ EXPORT_SYMBOL_GPL(snd_hda_multi_out_dig_close); /** * snd_hda_multi_out_analog_open - open analog outputs + * @codec: the HDA codec + * @mout: hda_multi_out object + * @substream: PCM substream to assign + * @hinfo: PCM information to assign * * Open analog outputs and set up the hw-constraints. * If the digital outputs can be opened as slave, open the digital @@ -5365,6 +5439,11 @@ EXPORT_SYMBOL_GPL(snd_hda_multi_out_analog_open); /** * snd_hda_multi_out_analog_prepare - Preapre the analog outputs. + * @codec: the HDA codec + * @mout: hda_multi_out object + * @stream_tag: stream tag to assign + * @format: format id to assign + * @substream: PCM substream to assign * * Set up the i/o for analog out. * When the digital out is available, copy the front out to digital out, too. @@ -5442,6 +5521,8 @@ EXPORT_SYMBOL_GPL(snd_hda_multi_out_analog_prepare); /** * snd_hda_multi_out_analog_cleanup - clean up the setting for analog out + * @codec: the HDA codec + * @mout: hda_multi_out object */ int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec, struct hda_multi_out *mout) @@ -5473,6 +5554,8 @@ EXPORT_SYMBOL_GPL(snd_hda_multi_out_analog_cleanup); /** * snd_hda_get_default_vref - Get the default (mic) VREF pin bits + * @codec: the HDA codec + * @pin: referred pin NID * * Guess the suitable VREF pin bits to be set as the pin-control value. * Note: the function doesn't set the AC_PINCTL_IN_EN bit. @@ -5498,7 +5581,12 @@ unsigned int snd_hda_get_default_vref(struct hda_codec *codec, hda_nid_t pin) } EXPORT_SYMBOL_GPL(snd_hda_get_default_vref); -/* correct the pin ctl value for matching with the pin cap */ +/** + * snd_hda_correct_pin_ctl - correct the pin ctl value for matching with the pin cap + * @codec: the HDA codec + * @pin: referred pin NID + * @val: pin ctl value to audit + */ unsigned int snd_hda_correct_pin_ctl(struct hda_codec *codec, hda_nid_t pin, unsigned int val) { @@ -5549,6 +5637,19 @@ unsigned int snd_hda_correct_pin_ctl(struct hda_codec *codec, } EXPORT_SYMBOL_GPL(snd_hda_correct_pin_ctl); +/** + * _snd_hda_pin_ctl - Helper to set pin ctl value + * @codec: the HDA codec + * @pin: referred pin NID + * @val: pin control value to set + * @cached: access over codec pinctl cache or direct write + * + * This function is a helper to set a pin ctl value more safely. + * It corrects the pin ctl value via snd_hda_correct_pin_ctl(), stores the + * value in pin target array via snd_hda_codec_set_pin_target(), then + * actually writes the value via either snd_hda_codec_update_cache() or + * snd_hda_codec_write() depending on @cached flag. + */ int _snd_hda_set_pin_ctl(struct hda_codec *codec, hda_nid_t pin, unsigned int val, bool cached) { @@ -5565,6 +5666,11 @@ EXPORT_SYMBOL_GPL(_snd_hda_set_pin_ctl); /** * snd_hda_add_imux_item - Add an item to input_mux + * @codec: the HDA codec + * @imux: imux helper object + * @label: the name of imux item to assign + * @index: index number of imux item to assign + * @type_idx: pointer to store the resultant label index * * When the same label is used already in the existing items, the number * suffix is appended to the label. This label index number is stored diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index e1cd34d9011d..0e6d7534f491 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -371,7 +371,7 @@ error: return ret; } -/** +/* * SNDRV_PCM_RATE_* and AC_PAR_PCM values don't match, print correct rates with * hdmi-specific routine. */ diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index f56765ae73a7..b2d81ab22fb0 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -57,6 +57,8 @@ static u32 read_pin_sense(struct hda_codec *codec, hda_nid_t nid) /** * snd_hda_jack_tbl_get - query the jack-table entry for the given NID + * @codec: the HDA codec + * @nid: pin NID to refer to */ struct hda_jack_tbl * snd_hda_jack_tbl_get(struct hda_codec *codec, hda_nid_t nid) @@ -75,6 +77,8 @@ EXPORT_SYMBOL_GPL(snd_hda_jack_tbl_get); /** * snd_hda_jack_tbl_get_from_tag - query the jack-table entry for the given tag + * @codec: the HDA codec + * @tag: tag value to refer to */ struct hda_jack_tbl * snd_hda_jack_tbl_get_from_tag(struct hda_codec *codec, unsigned char tag) @@ -93,6 +97,8 @@ EXPORT_SYMBOL_GPL(snd_hda_jack_tbl_get_from_tag); /** * snd_hda_jack_tbl_new - create a jack-table entry for the given NID + * @codec: the HDA codec + * @nid: pin NID to assign */ static struct hda_jack_tbl * snd_hda_jack_tbl_new(struct hda_codec *codec, hda_nid_t nid) @@ -162,6 +168,7 @@ static void jack_detect_update(struct hda_codec *codec, /** * snd_hda_set_dirty_all - Mark all the cached as dirty + * @codec: the HDA codec * * This function sets the dirty flag to all entries of jack table. * It's called from the resume path in hda_codec.c. @@ -218,6 +225,9 @@ EXPORT_SYMBOL_GPL(snd_hda_jack_detect_state); /** * snd_hda_jack_detect_enable - enable the jack-detection + * @codec: the HDA codec + * @nid: pin NID to enable + * @func: callback function to register * * In the case of error, the return value will be a pointer embedded with * errno. Check and handle the return value appropriately with standard @@ -266,6 +276,9 @@ EXPORT_SYMBOL_GPL(snd_hda_jack_detect_enable); /** * snd_hda_jack_set_gating_jack - Set gating jack. + * @codec: the HDA codec + * @gated_nid: gated pin NID + * @gating_nid: gating pin NID * * Indicates the gated jack is only valid when the gating jack is plugged. */ @@ -287,6 +300,7 @@ EXPORT_SYMBOL_GPL(snd_hda_jack_set_gating_jack); /** * snd_hda_jack_report_sync - sync the states of all jacks and report if changed + * @codec: the HDA codec */ void snd_hda_jack_report_sync(struct hda_codec *codec) { @@ -349,6 +363,11 @@ static void hda_free_jack_priv(struct snd_jack *jack) /** * snd_hda_jack_add_kctl - Add a kctl for the given pin + * @codec: the HDA codec + * @nid: pin NID to assign + * @name: string name for the jack + * @idx: index number for the jack + * @phantom_jack: flag to deal as a phantom jack * * This assigns a jack-detection kctl to the given pin. The kcontrol * will have the given name and index. @@ -456,6 +475,8 @@ static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid, /** * snd_hda_jack_add_kctls - Add kctls for all pins included in the given pincfg + * @codec: the HDA codec + * @cfg: pin config table to parse */ int snd_hda_jack_add_kctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) -- 2.1.2

1 1

[alsa-devel] [PATCH 00/11] Add AXD Audio Processing IP driver
by Qais Yousef 29 Oct '14

29 Oct '14

AXD Audio Processing IP performs audio decoding, encoding, mixing, equalisation, synchronisation and playback. This series adds the driver and DT binding document. I tried to split it into 'logical' units as there are many files and some of them are big (necessarily I believe). If it is still difficult to review as is I'd be happy to take on any suggestion to make things easier. I added it to drivers/char as this is the best location I could think of at the moment. It doesn't fit in alsa subsystem but I Cced them to confirm. Also I'm not sure if new drivers should go to staging first then get promoted or they can be accepted directly. This is based on Linus' 3.18-rc2 Cc: Arnd Bergmann <arnd(a)arndb.de> Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: <devicetree(a)vger.kernel.org> Cc: <alsa-devel(a)alsa-project.org> Qais Yousef (11): MAINTANERS: Add AXD as a supported driver dt: bindings: add AXD Audio Processing IP binding document drivers: char: add AXD Audio Processing IP driver drivers: char: axd: add fw binary header manipulation files drivers: char: axd: add buffers manipulation files drivers: char: axd: add basic files for sending/receiving axd cmds drivers: char: axd: add cmd interfce helper functions drivers: char: axd: add low level AXD platform setup files drivers: char: axd: add sysfs setup files drivers: char: axd: add ts interface file drivers: char: axd: add Kconfig and Makefile Documentation/devicetree/bindings/axd.txt | 39 + MAINTAINERS | 6 + drivers/char/Kconfig | 2 + drivers/char/Makefile | 2 + drivers/char/axd/Kconfig | 19 + drivers/char/axd/Makefile | 20 + drivers/char/axd/axd_api.h | 641 ++++++ drivers/char/axd/axd_buffers.c | 245 +++ drivers/char/axd/axd_buffers.h | 70 + drivers/char/axd/axd_cmds.c | 98 + drivers/char/axd/axd_cmds.h | 531 +++++ drivers/char/axd/axd_cmds_codec_internal.c | 58 + drivers/char/axd/axd_cmds_codec_internal.h | 28 + drivers/char/axd/axd_cmds_config.c | 1194 ++++++++++ drivers/char/axd/axd_cmds_decoder_config.c | 1806 ++++++++++++++++ drivers/char/axd/axd_cmds_encoder_config.c | 519 +++++ drivers/char/axd/axd_cmds_info.c | 1409 ++++++++++++ drivers/char/axd/axd_cmds_internal.c | 3237 ++++++++++++++++++++++++++++ drivers/char/axd/axd_cmds_internal.h | 306 +++ drivers/char/axd/axd_cmds_pipes.c | 1368 ++++++++++++ drivers/char/axd/axd_hdr.c | 69 + drivers/char/axd/axd_hdr.h | 20 + drivers/char/axd/axd_module.c | 1064 +++++++++ drivers/char/axd/axd_module.h | 99 + drivers/char/axd/axd_platform.h | 31 + drivers/char/axd/axd_platform_mips.c | 270 +++ drivers/char/axd/axd_sysfs.h | 37 + drivers/char/axd/axd_sysfs_ctrl.c | 539 +++++ drivers/char/axd/axd_sysfs_input.c | 406 ++++ drivers/char/axd/axd_sysfs_output.c | 1373 ++++++++++++ drivers/char/axd/axd_ts_driver.h | 24 + include/linux/axd.h | 32 + 32 files changed, 15562 insertions(+) create mode 100644 Documentation/devicetree/bindings/axd.txt create mode 100644 drivers/char/axd/Kconfig create mode 100644 drivers/char/axd/Makefile create mode 100644 drivers/char/axd/axd_api.h create mode 100644 drivers/char/axd/axd_buffers.c create mode 100644 drivers/char/axd/axd_buffers.h create mode 100644 drivers/char/axd/axd_cmds.c create mode 100644 drivers/char/axd/axd_cmds.h create mode 100644 drivers/char/axd/axd_cmds_codec_internal.c create mode 100644 drivers/char/axd/axd_cmds_codec_internal.h create mode 100644 drivers/char/axd/axd_cmds_config.c create mode 100644 drivers/char/axd/axd_cmds_decoder_config.c create mode 100644 drivers/char/axd/axd_cmds_encoder_config.c create mode 100644 drivers/char/axd/axd_cmds_info.c create mode 100644 drivers/char/axd/axd_cmds_internal.c create mode 100644 drivers/char/axd/axd_cmds_internal.h create mode 100644 drivers/char/axd/axd_cmds_pipes.c create mode 100644 drivers/char/axd/axd_hdr.c create mode 100644 drivers/char/axd/axd_hdr.h create mode 100644 drivers/char/axd/axd_module.c create mode 100644 drivers/char/axd/axd_module.h create mode 100644 drivers/char/axd/axd_platform.h create mode 100644 drivers/char/axd/axd_platform_mips.c create mode 100644 drivers/char/axd/axd_sysfs.h create mode 100644 drivers/char/axd/axd_sysfs_ctrl.c create mode 100644 drivers/char/axd/axd_sysfs_input.c create mode 100644 drivers/char/axd/axd_sysfs_output.c create mode 100644 drivers/char/axd/axd_ts_driver.h create mode 100644 include/linux/axd.h -- 2.1.0

6 37

[alsa-devel] [PATCH 1/2] ALSA: hda - Fix probing and stuttering on CMI8888 HD-audio controller
by Takashi Iwai 29 Oct '14

29 Oct '14

ASUS Phoebus with CMI8888 HD-audio chip (PCI id 13f6:5011) doesn't work with HD-audio driver as is because of some weird nature. For making DMA properly working, we need to disable MSI. The position report buffer doesn't work, thus we need to force reading LPIB instead. And yet, the codec CORB/RIRB communication gives errors unless we disable the snooping (caching). In this patch, all these workarounds are added as a quirk for the device. The HD-audio *codec* chip needs yet another workaround, but it'll be provided in the succeeding patch. Reported-and-tested-by: Vincent Lejeune <vljn(a)ovi.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 5db1948699d8..aa302fb03fc5 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -265,6 +265,7 @@ enum { AZX_DRIVER_TERA, AZX_DRIVER_CTX, AZX_DRIVER_CTHDA, + AZX_DRIVER_CMEDIA, AZX_DRIVER_GENERIC, AZX_NUM_DRIVERS, /* keep this as last entry */ }; @@ -330,6 +331,7 @@ static char *driver_short_names[] = { [AZX_DRIVER_TERA] = "HDA Teradici", [AZX_DRIVER_CTX] = "HDA Creative", [AZX_DRIVER_CTHDA] = "HDA Creative", + [AZX_DRIVER_CMEDIA] = "HDA C-Media", [AZX_DRIVER_GENERIC] = "HD-Audio Generic", }; @@ -1373,6 +1375,7 @@ static void azx_check_snoop_available(struct azx *chip) snoop = false; break; case AZX_DRIVER_CTHDA: + case AZX_DRIVER_CMEDIA: snoop = false; break; } @@ -2154,6 +2157,10 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_CTX | AZX_DCAPS_CTX_WORKAROUND | AZX_DCAPS_RIRB_PRE_DELAY | AZX_DCAPS_POSFIX_LPIB }, #endif + /* CM8888 */ + { PCI_DEVICE(0x13f6, 0x5011), + .driver_data = AZX_DRIVER_CMEDIA | + AZX_DCAPS_NO_MSI | AZX_DCAPS_POSFIX_LPIB }, /* Vortex86MX */ { PCI_DEVICE(0x17f3, 0x3010), .driver_data = AZX_DRIVER_GENERIC }, /* VMware HDAudio */ -- 2.0.3

2 6

[alsa-devel] [PATCH v03 0/4] ASoC: davinci-mcasp: Correct start/stop sequences
by Peter Ujfalusi 29 Oct '14

29 Oct '14

Hi, For some reason these patches did not made it to mainline kernel. V2 was sent on 28.04.2014: http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/074871.html and got applied the same day: http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/074878.html I just noticed that they are missing. Changes since v2: - rebased on next Changes since v1: - Correction for the stop sequence is added - start code cleanup patch added The startup sequence for TX and RX was not correct, not following the programming sequence from the TRM. This could cause initial channel swap when starting TX. When stopping the stream the AFIFO should be stopped as the last step to avoid unpredictable behavior. The change has been tested on AM335x and AM437x but it should be valid for all devices. Regards, Peter --- Peter Ujfalusi (4): ASoC: davinci-mcasp: Correct TX start sequence ASoC: davinci-mcasp: Correct RX start sequence ASoC: davinci-mcasp: When stopping TX/RX stop the AFIFO as the last step ASoC: davinci-mcasp: Move the AFIFO related code under start_tx/rx functions sound/soc/davinci/davinci-mcasp.c | 95 ++++++++++++++++++--------------------- sound/soc/davinci/davinci-mcasp.h | 6 +++ 2 files changed, 49 insertions(+), 52 deletions(-) -- 2.1.2

2 5

[alsa-devel] Kerneldoc warnings in ASoC source codes
by Takashi Iwai 29 Oct '14

29 Oct '14

Hi, I tried to process kerneldoc to sound/soc/soc-*.c and include/sound/soc.h, and hit quite lots of warnings. Many of them are trivial, typos, but some are unclear. I Cc'ed relevant people: could you guys fix these properly for 3.19 kernel? FYI, you can get the errors, e.g. scripts/kernel-doc sound/soc/soc-core.c > /dev/null thanks, Takashi Warning(.//include/sound/soc.h:595): No description found for parameter 'list' Warning(.//include/sound/soc.h:613): No description found for parameter 'list' Warning(.//include/sound/soc.h:648): No description found for parameter 'gpiod_dev' Warning(.//include/sound/soc.h:648): No description found for parameter 'debounce_time' Warning(.//include/sound/soc.h:648): No description found for parameter 'jack' Warning(.//include/sound/soc.h:648): No description found for parameter 'work' Warning(.//include/sound/soc.h:648): No description found for parameter 'desc' Warning(.//include/sound/soc.h:648): No description found for parameter 'data' Warning(.//include/sound/soc.h:648): Excess struct/union/enum/typedef member 'debouce_time' description in 'snd_soc_jack_gpio' Warning(.//sound/soc/soc-core.c:2122): No description found for parameter 'ops' Warning(.//sound/soc/soc-core.c:2122): No description found for parameter 'pdev' Warning(.//sound/soc/soc-core.c:2696): No description found for parameter 'ucontrol' Warning(.//sound/soc/soc-core.c:2696): Excess function parameter 'uinfo' description in 'snd_soc_put_volsw_sx' Warning(.//sound/soc/soc-core.c:3466): No description found for parameter 'ratio' Warning(.//sound/soc/soc-core.c:4088): No description found for parameter 'dev' Warning(.//sound/soc/soc-core.c:4126): No description found for parameter 'platform_drv' Warning(.//sound/soc/soc-core.c:4126): Excess function parameter 'platform_driver' description in 'snd_soc_add_platform' Warning(.//sound/soc/soc-core.c:4165): No description found for parameter 'dev' Warning(.//sound/soc/soc-core.c:4165): No description found for parameter 'platform_drv' Warning(.//sound/soc/soc-core.c:4165): Excess function parameter 'platform' description in 'snd_soc_register_platform' Warning(.//sound/soc/soc-core.c:4221): No description found for parameter 'dev' Warning(.//sound/soc/soc-core.c:4221): Excess function parameter 'platform' description in 'snd_soc_unregister_platform' Warning(.//sound/soc/soc-core.c:4315): No description found for parameter 'dev' Warning(.//sound/soc/soc-core.c:4315): No description found for parameter 'codec_drv' Warning(.//sound/soc/soc-core.c:4315): No description found for parameter 'dai_drv' Warning(.//sound/soc/soc-core.c:4315): No description found for parameter 'num_dai' Warning(.//sound/soc/soc-core.c:4315): Excess function parameter 'codec' description in 'snd_soc_register_codec' Warning(.//sound/soc/soc-core.c:4412): No description found for parameter 'dev' Warning(.//sound/soc/soc-core.c:4412): Excess function parameter 'codec' description in 'snd_soc_unregister_codec' Warning(.//sound/soc/soc-devres.c:70): No description found for parameter 'platform_drv' Warning(.//sound/soc/soc-devres.c:70): Excess function parameter 'platform' description in 'devm_snd_soc_register_platform' Warning(.//sound/soc/soc-dapm.c:2690): No description found for parameter 'card' Warning(.//sound/soc/soc-dapm.c:2690): Excess function parameter 'dapm' description in 'snd_soc_dapm_new_widgets' Warning(.//sound/soc/soc-jack.c:126): No description found for parameter 'zones' Warning(.//sound/soc/soc-jack.c:126): Excess function parameter 'zone' description in 'snd_soc_jack_add_zones'

1 0

[alsa-devel] [PATCH] ASoC: rsnd: remove unsupported PAUSE flag
by Kuninori Morimoto 29 Oct '14

29 Oct '14

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> R-Car sound doesn't support PAUSE. Remove SNDRV_PCM_INFO_PAUSE flags from snd_pcm_hardware info Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 1922ec5..7004219 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -886,8 +886,7 @@ static int rsnd_dai_probe(struct platform_device *pdev, static struct snd_pcm_hardware rsnd_pcm_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE, + SNDRV_PCM_INFO_MMAP_VALID, .buffer_bytes_max = 64 * 1024, .period_bytes_min = 32, .period_bytes_max = 8192, -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: fsi: remove unsupported PAUSE flag
by Kuninori Morimoto 29 Oct '14

29 Oct '14

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> FSI doesn't support PAUSE. Remove SNDRV_PCM_INFO_PAUSE flags from snd_pcm_hardware info Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/fsi.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 66fddec..88e5df4 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1711,8 +1711,7 @@ static const struct snd_soc_dai_ops fsi_dai_ops = { static struct snd_pcm_hardware fsi_pcm_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE, + SNDRV_PCM_INFO_MMAP_VALID, .buffer_bytes_max = 64 * 1024, .period_bytes_min = 32, .period_bytes_max = 8192, -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: rt5645:Mark RT5645_TDM_CTRL_3 as readable
by yang.a.fang＠intel.com 29 Oct '14

29 Oct '14

From: "Fang, Yang A" <yang.a.fang(a)intel.com> amixer query fails due to it is not readable reigster Signed-off-by: Fang, Yang A <yang.a.fang(a)intel.com> --- sound/soc/codecs/rt5645.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 3fb83bf..d16331e 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -139,6 +139,7 @@ static const struct reg_default rt5645_reg[] = { { 0x76, 0x000a }, { 0x77, 0x0c00 }, { 0x78, 0x0000 }, + { 0x79, 0x0123 }, { 0x80, 0x0000 }, { 0x81, 0x0000 }, { 0x82, 0x0000 }, @@ -334,6 +335,7 @@ static bool rt5645_readable_register(struct device *dev, unsigned int reg) case RT5645_DMIC_CTRL2: case RT5645_TDM_CTRL_1: case RT5645_TDM_CTRL_2: + case RT5645_TDM_CTRL_3: case RT5645_GLB_CLK: case RT5645_PLL_CTRL1: case RT5645_PLL_CTRL2: -- 1.7.9.5

3 2

[alsa-devel] [PATCH] ASoC: Intel: Add dependency on DesignWare DMA controller
by Mark Brown 29 Oct '14

29 Oct '14

We have calls into the controller so we need to ensure it is being built. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 2a3af880868b..ae7f87221a3c 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -20,6 +20,7 @@ config SND_SOC_INTEL_SST tristate "ASoC support for Intel(R) Smart Sound Technology" select SND_SOC_INTEL_SST_ACPI if ACPI depends on (X86 || COMPILE_TEST) + depends on DW_DMAC_CORE help This adds support for Intel(R) Smart Sound Technology (SST). Say Y if you have such a device -- 2.1.1

1 0

[alsa-devel] - snd-usb-audio - SPDIF clock source mixer control for Digidesign Mbox
by Damien Zammit 29 Oct '14

29 Oct '14

Hi all, As some of you may remember, the Digidesign Mbox has a multiple endpoint interface and does not follow the standard USB audio spec. I am pretty sure that the first interface 0 is for firmware upgrades only, and the second interface has 2 endpoints one for playback and one for capture. I have heard that the SPDIF output is always on, but there is a special command to change the clock source to external sync with SPDIF. I think I have found the correct USB command for changing the clock source of the device to SPDIF external sync. I tried to use a pair of the devices in JACK using a pcm_multi configuration and syncing one of them to the other with a RCA cable, but unfortunately I get lots of XRUNs in JACK and I am wondering if there is a reason for this? Can someone who knows about the pcm_multi stuff shed some light on why I might be getting thousands of XRUNs when I use two identical USB soundcards that are (hopefully) synced to each other? I have written a non-standard mixer control for this so alsamixer can select the external clock source. Basically, the command to set SPDIF sync seems to include a sample rate command, but the sample rate is set to zero for SPDIF sync. Could JACK be resetting the sample rate and overriding my command perhaps? How would I override the ALSA driver code so that the sample rate is not reset by JACK to 48000 instead of 0 but only when the mixer control sets the flag? Damien

1 0

[alsa-devel] [PATCH] ASoC: rockchip-i2s: fix infinite loop in rockchip_snd_rxctrl
by Jianqun 29 Oct '14

29 Oct '14

We can get into an infinite loop if the I2S_CLR register fails to clear due to a missing break statement, so add that. Signed-off-by: Jianqun <jay.xu(a)rock-chips.com> --- sound/soc/rockchip/rockchip_i2s.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index f373e37..c74ba37 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -154,8 +154,10 @@ static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) while (val) { regmap_read(i2s->regmap, I2S_CLR, &val); retry--; - if (!retry) + if (!retry) { dev_warn(i2s->dev, "fail to clear\n"); + break; + } } } } -- 1.9.1

2 1

[alsa-devel] [PATCH 1/3] ASoC: sgtl5000: Use the preferred form for passing a size of a struct
by Fabio Estevam 29 Oct '14

29 Oct '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> According to Documentation/CodingStyle - Chapter 14: "The preferred form for passing a size of a struct is the following: p = kmalloc(sizeof(*p), ...); The alternative form where struct name is spelled out hurts readability and introduces an opportunity for a bug when the pointer variable type is changed but the corresponding sizeof that is passed to a memory allocator is not." So do it as recommeded. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/sgtl5000.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 6bb77d7..3f8c05b 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1419,8 +1419,7 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, int ret, reg, rev; unsigned int mclk; - sgtl5000 = devm_kzalloc(&client->dev, sizeof(struct sgtl5000_priv), - GFP_KERNEL); + sgtl5000 = devm_kzalloc(&client->dev, sizeof(*sgtl5000), GFP_KERNEL); if (!sgtl5000) return -ENOMEM; -- 1.9.1

3 4

[alsa-devel] [PATCH - hda 1/1] Add ultra dock support for Thinkpad X240.
by Lukas Bossard 29 Oct '14

29 Oct '14

Adding ultra doch support for Lenovo Thinkpad X240 (17aa:2214). In the same process also rename ALC292_FIXUP_TPT440_DOCK to a more generic ALC292_FIXUP_LENOVO_ULTRA_DOCK Signed-off-by: Lukas Bossard <mr.bobukas(a)gmail.com> diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 34b7bdb..693f083 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4212,7 +4212,7 @@ enum { ALC255_FIXUP_HEADSET_MODE, ALC255_FIXUP_HEADSET_MODE_NO_HP_MIC, ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, - ALC292_FIXUP_TPT440_DOCK, + ALC292_FIXUP_LENOVO_ULTRA_DOCK, ALC283_FIXUP_BXBT2807_MIC, ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED, ALC282_FIXUP_ASPIRE_V5_PINS, @@ -4640,7 +4640,7 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE }, - [ALC292_FIXUP_TPT440_DOCK] = { + [ALC292_FIXUP_LENOVO_ULTRA_DOCK] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { { 0x16, 0x21211010 }, /* dock headphone */ @@ -4799,16 +4799,17 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK), - SND_PCI_QUIRK(0x17aa, 0x220c, "Thinkpad T440s", ALC292_FIXUP_TPT440_DOCK), - SND_PCI_QUIRK(0x17aa, 0x220e, "Thinkpad T440p", ALC292_FIXUP_TPT440_DOCK), - SND_PCI_QUIRK(0x17aa, 0x2210, "Thinkpad T540p", ALC292_FIXUP_TPT440_DOCK), - SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad T440", ALC292_FIXUP_TPT440_DOCK), + SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad X240", ALC292_FIXUP_LENOVO_ULTRA_DOCK), + SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad T440", ALC292_FIXUP_LENOVO_ULTRA_DOCK), + SND_PCI_QUIRK(0x17aa, 0x220c, "Thinkpad T440s", ALC292_FIXUP_LENOVO_ULTRA_DOCK), + SND_PCI_QUIRK(0x17aa, 0x220e, "Thinkpad T440p", ALC292_FIXUP_LENOVO_ULTRA_DOCK), + SND_PCI_QUIRK(0x17aa, 0x2210, "Thinkpad T540p", ALC292_FIXUP_LENOVO_ULTRA_DOCK), SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP), SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC283_FIXUP_INT_MIC), - SND_PCI_QUIRK(0x17aa, 0x501e, "Thinkpad L440", ALC292_FIXUP_TPT440_DOCK), + SND_PCI_QUIRK(0x17aa, 0x501e, "Thinkpad L440", ALC292_FIXUP_LENOVO_ULTRA_DOCK), SND_PCI_QUIRK(0x17aa, 0x5026, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), @@ -4887,7 +4888,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC269_FIXUP_DELL2_MIC_NO_PRESENCE, .name = "dell-headset-dock"}, {.id = ALC283_FIXUP_CHROME_BOOK, .name = "alc283-dac-wcaps"}, {.id = ALC283_FIXUP_SENSE_COMBO_JACK, .name = "alc283-sense-combo"}, - {.id = ALC292_FIXUP_TPT440_DOCK, .name = "tpt440-dock"}, + {.id = ALC292_FIXUP_LENOVO_ULTRA_DOCK, .name = "lenovo-ultra-dock"}, {} }; -- 2.1.0

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[alsa-devel] [PATCH 1/2] ALSA: doc: Add missing headers and compress stuff to alsa-driver-api.tmpl
by Takashi Iwai 29 Oct '14

29 Oct '14

Some header files have kereldoc comments but are not referred properly. Let's add them. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Documentation/DocBook/alsa-driver-api.tmpl | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/Documentation/DocBook/alsa-driver-api.tmpl b/Documentation/DocBook/alsa-driver-api.tmpl index 0230a96f0564..13f8b241f042 100644 --- a/Documentation/DocBook/alsa-driver-api.tmpl +++ b/Documentation/DocBook/alsa-driver-api.tmpl @@ -57,6 +57,7 @@ !Esound/core/pcm.c !Esound/core/pcm_lib.c !Esound/core/pcm_native.c +!include/sound/pcm.h </sect1> <sect1><title>PCM Format Helpers</title> !Esound/core/pcm_misc.c @@ -64,6 +65,10 @@ <sect1><title>PCM Memory Management</title> !Esound/core/pcm_memory.c </sect1> + <sect1><title>PCM DMA Engine API</title> +!Esound/core/pcm_dmaengine.c +!Iinclude/sound/dmaengine_pcm.h + </sect1> </chapter> <chapter><title>Control/Mixer API</title> <sect1><title>General Control Interface</title> @@ -91,12 +96,19 @@ !Esound/core/info.c </sect1> </chapter> + <chapter><title>Compress Offload</title> + <sect1><title>Compress Offload API</title> +!Esound/core/compress_offload.c +!Iinclude/sound/compress_driver.h + </sect1> + </chapter> <chapter><title>Miscellaneous Functions</title> <sect1><title>Hardware-Dependent Devices API</title> !Esound/core/hwdep.c </sect1> <sect1><title>Jack Abstraction Layer API</title> !Esound/core/jack.c +!Iinclude/sound/jack.h </sect1> <sect1><title>ISA DMA Helpers</title> !Esound/core/isadma.c -- 2.1.2

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[alsa-devel] [PATCH 0/3] documentation fixes
by Takashi Iwai 29 Oct '14

29 Oct '14

Hi, here is a few more minor fixes for kerneldoc comments. Per request from Vinod, uapid/sound/compress_*.h are included, too. [PATCH 1/3] ALSA: doc: Fix enum snd_jack_types comments [PATCH 2/3] ALSA: doc: Fix uapi/sound/compress_offload.h kerneldoc [PATCH 3/3] ALSA: doc: Include uapi/sound/compress_*.h, too Takashi

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[alsa-devel] [PATCH] 6fire: Convert byte_rev_table uses to bitrev8
by Joe Perches 29 Oct '14

29 Oct '14

Use the inline function instead of directly indexing the array. This allows some architectures with hardware instructions for bit reversals to eliminate the array. Signed-off-by: Joe Perches <joe(a)perches.com> --- On Sun, 2014-10-26 at 23:46 -0700, Joe Perches wrote: > On Mon, 2014-10-27 at 14:37 +0800, Wang, Yalin wrote: > > this change add CONFIG_HAVE_ARCH_BITREVERSE config option, > > so that we can use arm/arm64 rbit instruction to do bitrev operation > > by hardware. [] > > diff --git a/include/linux/bitrev.h b/include/linux/bitrev.h > > index 7ffe03f..ef5b2bb 100644 > > --- a/include/linux/bitrev.h > > +++ b/include/linux/bitrev.h > > @@ -3,6 +3,14 @@ > > > > #include <linux/types.h> > > > > +#ifdef CONFIG_HAVE_ARCH_BITREVERSE > > +#include <asm/bitrev.h> > > + > > +#define bitrev32 __arch_bitrev32 > > +#define bitrev16 __arch_bitrev16 > > +#define bitrev8 __arch_bitrev8 > > + > > +#else > > extern u8 const byte_rev_table[256]; sound/usb/6fire/firmware.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c index 3b02e54..62c25e7 100644 --- a/sound/usb/6fire/firmware.c +++ b/sound/usb/6fire/firmware.c @@ -316,7 +316,7 @@ static int usb6fire_fw_fpga_upload( while (c != end) { for (i = 0; c != end && i < FPGA_BUFSIZE; i++, c++) - buffer[i] = byte_rev_table[(u8) *c]; + buffer[i] = bitrev8((u8)*c); ret = usb6fire_fw_fpga_write(device, buffer, i); if (ret < 0) {

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[alsa-devel] [PATCH v2] ASoC: rt5677: Add TDM channel mapping function
by Oder Chiou 28 Oct '14

28 Oct '14

It is for channel to slot mapping, and it is not only for 8 channels mapping, but also in 2, 4 and 6 channels mapping. If we want to use the 2 channels in the stereo2 adc path, we need to set the item "2/1/3/4" or "2/3/1/4". It also adds for stereo channel swap. It can map the sterero channels "L/R" to "R/L", "L/L" or R/R. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 239 +++++++++++++++++++++++++++++++++++++++++++--- sound/soc/codecs/rt5677.h | 38 +++++++- 2 files changed, 262 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index d17d079..5a75552 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -1834,6 +1834,93 @@ static SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5677_if4_adc_mux = SOC_DAPM_ENUM("IF4 ADC Source", rt5677_if4_adc_enum); +/* TDM IF1/2 ADC Data Selection */ /* MX-3B MX-40 [7:6][5:4][3:2][1:0] */ +static const char * const rt5677_if12_adc_swap_src[] = { + "L/R", "R/L", "L/L", "R/R" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5677_if1_adc1_swap_enum, RT5677_TDM1_CTRL1, + RT5677_IF1_ADC1_SWAP_SFT, rt5677_if12_adc_swap_src); + +static const struct snd_kcontrol_new rt5677_if1_adc1_swap_mux = + SOC_DAPM_ENUM("IF1 ADC1 Swap Source", rt5677_if1_adc1_swap_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if1_adc2_swap_enum, RT5677_TDM1_CTRL1, + RT5677_IF1_ADC2_SWAP_SFT, rt5677_if12_adc_swap_src); + +static const struct snd_kcontrol_new rt5677_if1_adc2_swap_mux = + SOC_DAPM_ENUM("IF1 ADC2 Swap Source", rt5677_if1_adc2_swap_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if1_adc3_swap_enum, RT5677_TDM1_CTRL1, + RT5677_IF1_ADC3_SWAP_SFT, rt5677_if12_adc_swap_src); + +static const struct snd_kcontrol_new rt5677_if1_adc3_swap_mux = + SOC_DAPM_ENUM("IF1 ADC3 Swap Source", rt5677_if1_adc3_swap_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if1_adc4_swap_enum, RT5677_TDM1_CTRL1, + RT5677_IF1_ADC4_SWAP_SFT, rt5677_if12_adc_swap_src); + +static const struct snd_kcontrol_new rt5677_if1_adc4_swap_mux = + SOC_DAPM_ENUM("IF1 ADC4 Swap Source", rt5677_if1_adc4_swap_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if2_adc1_swap_enum, RT5677_TDM2_CTRL1, + RT5677_IF1_ADC2_SWAP_SFT, rt5677_if12_adc_swap_src); + +static const struct snd_kcontrol_new rt5677_if2_adc1_swap_mux = + SOC_DAPM_ENUM("IF1 ADC2 Swap Source", rt5677_if2_adc1_swap_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if2_adc2_swap_enum, RT5677_TDM2_CTRL1, + RT5677_IF2_ADC2_SWAP_SFT, rt5677_if12_adc_swap_src); + +static const struct snd_kcontrol_new rt5677_if2_adc2_swap_mux = + SOC_DAPM_ENUM("IF2 ADC2 Swap Source", rt5677_if2_adc2_swap_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if2_adc3_swap_enum, RT5677_TDM2_CTRL1, + RT5677_IF2_ADC3_SWAP_SFT, rt5677_if12_adc_swap_src); + +static const struct snd_kcontrol_new rt5677_if2_adc3_swap_mux = + SOC_DAPM_ENUM("IF2 ADC3 Swap Source", rt5677_if2_adc3_swap_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if2_adc4_swap_enum, RT5677_TDM2_CTRL1, + RT5677_IF2_ADC4_SWAP_SFT, rt5677_if12_adc_swap_src); + +static const struct snd_kcontrol_new rt5677_if2_adc4_swap_mux = + SOC_DAPM_ENUM("IF2 ADC4 Swap Source", rt5677_if2_adc4_swap_enum); + +/* TDM IF1 ADC Data Selection */ /* MX-3C [2:0] */ +static const char * const rt5677_if1_adc_tdm_swap_src[] = { + "1/2/3/4", "2/1/3/4", "2/3/1/4", "4/1/2/3", "1/3/2/4", "1/4/2/3", + "3/1/2/4", "3/4/1/2" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5677_if1_adc_tdm_swap_enum, RT5677_TDM1_CTRL2, + RT5677_IF1_ADC_CTRL_SFT, rt5677_if1_adc_tdm_swap_src); + +static const struct snd_kcontrol_new rt5677_if1_adc_tdm_swap_mux = + SOC_DAPM_ENUM("IF1 ADC TDM Swap Source", rt5677_if1_adc_tdm_swap_enum); + +/* TDM IF2 ADC Data Selection */ /* MX-41[2:0] */ +static const char * const rt5677_if2_adc_tdm_swap_src[] = { + "1/2/3/4", "2/1/3/4", "3/1/2/4", "4/1/2/3", "1/3/2/4", "1/4/2/3", + "2/3/1/4", "3/4/1/2" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5677_if2_adc_tdm_swap_enum, RT5677_TDM2_CTRL2, + RT5677_IF2_ADC_CTRL_SFT, rt5677_if2_adc_tdm_swap_src); + +static const struct snd_kcontrol_new rt5677_if2_adc_tdm_swap_mux = + SOC_DAPM_ENUM("IF2 ADC TDM Swap Source", rt5677_if2_adc_tdm_swap_enum); + static int rt5677_bst1_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -1945,6 +2032,52 @@ static int rt5677_set_micbias1_event(struct snd_soc_dapm_widget *w, return 0; } +static int rt5677_if1_adc_tdm_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + unsigned int value; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + regmap_read(rt5677->regmap, RT5677_TDM1_CTRL2, &value); + if (value & RT5677_IF1_ADC_CTRL_MASK) + regmap_update_bits(rt5677->regmap, RT5677_TDM1_CTRL1, + RT5677_IF1_ADC_MODE_MASK, + RT5677_IF1_ADC_MODE_TDM); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5677_if2_adc_tdm_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + unsigned int value; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + regmap_read(rt5677->regmap, RT5677_TDM2_CTRL2, &value); + if (value & RT5677_IF2_ADC_CTRL_MASK) + regmap_update_bits(rt5677->regmap, RT5677_TDM2_CTRL1, + RT5677_IF2_ADC_MODE_MASK, + RT5677_IF2_ADC_MODE_TDM); + break; + + default: + return 0; + } + + return 0; +} + static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("PLL1", RT5677_PWR_ANLG2, RT5677_PWR_PLL1_BIT, 0, rt5677_set_pll1_event, SND_SOC_DAPM_POST_PMU), @@ -2104,10 +2237,8 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { SND_SOC_DAPM_PGA("Stereo4 ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("Sto2 ADC LR MIX", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("Mono ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_PGA("IF1_ADC1", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_PGA("IF1_ADC2", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_PGA("IF1_ADC3", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_PGA("IF1_ADC4", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 ADC", SND_SOC_NOPM, 0, 0, NULL, 0), /* DSP */ SND_SOC_DAPM_MUX("IB9 Mux", SND_SOC_NOPM, 0, 0, @@ -2230,6 +2361,17 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { &rt5677_if1_adc3_mux), SND_SOC_DAPM_MUX("IF1 ADC4 Mux", SND_SOC_NOPM, 0, 0, &rt5677_if1_adc4_mux), + SND_SOC_DAPM_MUX("IF1 ADC1 Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if1_adc1_swap_mux), + SND_SOC_DAPM_MUX("IF1 ADC2 Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if1_adc2_swap_mux), + SND_SOC_DAPM_MUX("IF1 ADC3 Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if1_adc3_swap_mux), + SND_SOC_DAPM_MUX("IF1 ADC4 Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if1_adc4_swap_mux), + SND_SOC_DAPM_MUX_E("IF1 ADC TDM Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if1_adc_tdm_swap_mux, rt5677_if1_adc_tdm_event, + SND_SOC_DAPM_PRE_PMU), SND_SOC_DAPM_MUX("IF2 ADC1 Mux", SND_SOC_NOPM, 0, 0, &rt5677_if2_adc1_mux), SND_SOC_DAPM_MUX("IF2 ADC2 Mux", SND_SOC_NOPM, 0, 0, @@ -2238,6 +2380,17 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { &rt5677_if2_adc3_mux), SND_SOC_DAPM_MUX("IF2 ADC4 Mux", SND_SOC_NOPM, 0, 0, &rt5677_if2_adc4_mux), + SND_SOC_DAPM_MUX("IF2 ADC1 Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if2_adc1_swap_mux), + SND_SOC_DAPM_MUX("IF2 ADC2 Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if2_adc2_swap_mux), + SND_SOC_DAPM_MUX("IF2 ADC3 Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if2_adc3_swap_mux), + SND_SOC_DAPM_MUX("IF2 ADC4 Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if2_adc4_swap_mux), + SND_SOC_DAPM_MUX_E("IF2 ADC TDM Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if2_adc_tdm_swap_mux, rt5677_if2_adc_tdm_event, + SND_SOC_DAPM_PRE_PMU), SND_SOC_DAPM_MUX("IF3 ADC Mux", SND_SOC_NOPM, 0, 0, &rt5677_if3_adc_mux), SND_SOC_DAPM_MUX("IF4 ADC Mux", SND_SOC_NOPM, 0, 0, @@ -2621,11 +2774,42 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "IF1 ADC4 Mux", "OB67", "OB67" }, { "IF1 ADC4 Mux", "OB01", "OB01 Bypass Mux" }, + { "IF1 ADC1 Swap Mux", "L/R", "IF1 ADC1 Mux" }, + { "IF1 ADC1 Swap Mux", "R/L", "IF1 ADC1 Mux" }, + { "IF1 ADC1 Swap Mux", "L/L", "IF1 ADC1 Mux" }, + { "IF1 ADC1 Swap Mux", "R/R", "IF1 ADC1 Mux" }, + + { "IF1 ADC2 Swap Mux", "L/R", "IF1 ADC2 Mux" }, + { "IF1 ADC2 Swap Mux", "R/L", "IF1 ADC2 Mux" }, + { "IF1 ADC2 Swap Mux", "L/L", "IF1 ADC2 Mux" }, + { "IF1 ADC2 Swap Mux", "R/R", "IF1 ADC2 Mux" }, + + { "IF1 ADC3 Swap Mux", "L/R", "IF1 ADC3 Mux" }, + { "IF1 ADC3 Swap Mux", "R/L", "IF1 ADC3 Mux" }, + { "IF1 ADC3 Swap Mux", "L/L", "IF1 ADC3 Mux" }, + { "IF1 ADC3 Swap Mux", "R/R", "IF1 ADC3 Mux" }, + + { "IF1 ADC4 Swap Mux", "L/R", "IF1 ADC4 Mux" }, + { "IF1 ADC4 Swap Mux", "R/L", "IF1 ADC4 Mux" }, + { "IF1 ADC4 Swap Mux", "L/L", "IF1 ADC4 Mux" }, + { "IF1 ADC4 Swap Mux", "R/R", "IF1 ADC4 Mux" }, + + { "IF1 ADC", NULL, "IF1 ADC1 Swap Mux" }, + { "IF1 ADC", NULL, "IF1 ADC2 Swap Mux" }, + { "IF1 ADC", NULL, "IF1 ADC3 Swap Mux" }, + { "IF1 ADC", NULL, "IF1 ADC4 Swap Mux" }, + + { "IF1 ADC TDM Swap Mux", "1/2/3/4", "IF1 ADC" }, + { "IF1 ADC TDM Swap Mux", "2/1/3/4", "IF1 ADC" }, + { "IF1 ADC TDM Swap Mux", "2/3/1/4", "IF1 ADC" }, + { "IF1 ADC TDM Swap Mux", "4/1/2/3", "IF1 ADC" }, + { "IF1 ADC TDM Swap Mux", "1/3/2/4", "IF1 ADC" }, + { "IF1 ADC TDM Swap Mux", "1/4/2/3", "IF1 ADC" }, + { "IF1 ADC TDM Swap Mux", "3/1/2/4", "IF1 ADC" }, + { "IF1 ADC TDM Swap Mux", "3/4/1/2", "IF1 ADC" }, + { "AIF1TX", NULL, "I2S1" }, - { "AIF1TX", NULL, "IF1 ADC1 Mux" }, - { "AIF1TX", NULL, "IF1 ADC2 Mux" }, - { "AIF1TX", NULL, "IF1 ADC3 Mux" }, - { "AIF1TX", NULL, "IF1 ADC4 Mux" }, + { "AIF1TX", NULL, "IF1 ADC TDM Swap Mux" }, { "IF2 ADC1 Mux", "STO1 ADC MIX", "Stereo1 ADC MIX" }, { "IF2 ADC1 Mux", "OB01", "OB01 Bypass Mux" }, @@ -2642,11 +2826,42 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "IF2 ADC4 Mux", "OB67", "OB67" }, { "IF2 ADC4 Mux", "OB01", "OB01 Bypass Mux" }, + { "IF2 ADC1 Swap Mux", "L/R", "IF2 ADC1 Mux" }, + { "IF2 ADC1 Swap Mux", "R/L", "IF2 ADC1 Mux" }, + { "IF2 ADC1 Swap Mux", "L/L", "IF2 ADC1 Mux" }, + { "IF2 ADC1 Swap Mux", "R/R", "IF2 ADC1 Mux" }, + + { "IF2 ADC2 Swap Mux", "L/R", "IF2 ADC2 Mux" }, + { "IF2 ADC2 Swap Mux", "R/L", "IF2 ADC2 Mux" }, + { "IF2 ADC2 Swap Mux", "L/L", "IF2 ADC2 Mux" }, + { "IF2 ADC2 Swap Mux", "R/R", "IF2 ADC2 Mux" }, + + { "IF2 ADC3 Swap Mux", "L/R", "IF2 ADC3 Mux" }, + { "IF2 ADC3 Swap Mux", "R/L", "IF2 ADC3 Mux" }, + { "IF2 ADC3 Swap Mux", "L/L", "IF2 ADC3 Mux" }, + { "IF2 ADC3 Swap Mux", "R/R", "IF2 ADC3 Mux" }, + + { "IF2 ADC4 Swap Mux", "L/R", "IF2 ADC4 Mux" }, + { "IF2 ADC4 Swap Mux", "R/L", "IF2 ADC4 Mux" }, + { "IF2 ADC4 Swap Mux", "L/L", "IF2 ADC4 Mux" }, + { "IF2 ADC4 Swap Mux", "R/R", "IF2 ADC4 Mux" }, + + { "IF2 ADC", NULL, "IF2 ADC1 Swap Mux" }, + { "IF2 ADC", NULL, "IF2 ADC2 Swap Mux" }, + { "IF2 ADC", NULL, "IF2 ADC3 Swap Mux" }, + { "IF2 ADC", NULL, "IF2 ADC4 Swap Mux" }, + + { "IF2 ADC TDM Swap Mux", "1/2/3/4", "IF2 ADC" }, + { "IF2 ADC TDM Swap Mux", "2/1/3/4", "IF2 ADC" }, + { "IF2 ADC TDM Swap Mux", "3/1/2/4", "IF2 ADC" }, + { "IF2 ADC TDM Swap Mux", "4/1/2/3", "IF2 ADC" }, + { "IF2 ADC TDM Swap Mux", "1/3/2/4", "IF2 ADC" }, + { "IF2 ADC TDM Swap Mux", "1/4/2/3", "IF2 ADC" }, + { "IF2 ADC TDM Swap Mux", "2/3/1/4", "IF2 ADC" }, + { "IF2 ADC TDM Swap Mux", "3/4/1/2", "IF2 ADC" }, + { "AIF2TX", NULL, "I2S2" }, - { "AIF2TX", NULL, "IF2 ADC1 Mux" }, - { "AIF2TX", NULL, "IF2 ADC2 Mux" }, - { "AIF2TX", NULL, "IF2 ADC3 Mux" }, - { "AIF2TX", NULL, "IF2 ADC4 Mux" }, + { "AIF2TX", NULL, "IF2 ADC TDM Swap Mux" }, { "IF3 ADC Mux", "STO1 ADC MIX", "Stereo1 ADC MIX" }, { "IF3 ADC Mux", "STO2 ADC MIX", "Stereo2 ADC MIX" }, diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index 20efa4a..a0ae32f 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -799,7 +799,21 @@ #define RT5677_PDM2_I2C_EXE (0x1 << 1) #define RT5677_PDM2_I2C_BUSY (0x1 << 0) -/* MX3C TDM1 control 1 (0x3c) */ +/* MX3B TDM1 control 1 (0x3b) */ +#define RT5677_IF1_ADC_MODE_MASK (0x1 << 12) +#define RT5677_IF1_ADC_MODE_SFT 12 +#define RT5677_IF1_ADC_MODE_I2S (0x0 << 12) +#define RT5677_IF1_ADC_MODE_TDM (0x1 << 12) +#define RT5677_IF1_ADC1_SWAP_MASK (0x3 << 6) +#define RT5677_IF1_ADC1_SWAP_SFT 6 +#define RT5677_IF1_ADC2_SWAP_MASK (0x3 << 4) +#define RT5677_IF1_ADC2_SWAP_SFT 4 +#define RT5677_IF1_ADC3_SWAP_MASK (0x3 << 2) +#define RT5677_IF1_ADC3_SWAP_SFT 2 +#define RT5677_IF1_ADC4_SWAP_MASK (0x3 << 0) +#define RT5677_IF1_ADC4_SWAP_SFT 0 + +/* MX3C TDM1 control 2 (0x3c) */ #define RT5677_IF1_ADC4_MASK (0x3 << 10) #define RT5677_IF1_ADC4_SFT 10 #define RT5677_IF1_ADC3_MASK (0x3 << 8) @@ -808,8 +822,24 @@ #define RT5677_IF1_ADC2_SFT 6 #define RT5677_IF1_ADC1_MASK (0x3 << 4) #define RT5677_IF1_ADC1_SFT 4 - -/* MX41 TDM2 control 1 (0x41) */ +#define RT5677_IF1_ADC_CTRL_MASK (0x7 << 0) +#define RT5677_IF1_ADC_CTRL_SFT 0 + +/* MX40 TDM2 control 1 (0x40) */ +#define RT5677_IF2_ADC_MODE_MASK (0x1 << 12) +#define RT5677_IF2_ADC_MODE_SFT 12 +#define RT5677_IF2_ADC_MODE_I2S (0x0 << 12) +#define RT5677_IF2_ADC_MODE_TDM (0x1 << 12) +#define RT5677_IF2_ADC1_SWAP_MASK (0x3 << 6) +#define RT5677_IF2_ADC1_SWAP_SFT 6 +#define RT5677_IF2_ADC2_SWAP_MASK (0x3 << 4) +#define RT5677_IF2_ADC2_SWAP_SFT 4 +#define RT5677_IF2_ADC3_SWAP_MASK (0x3 << 2) +#define RT5677_IF2_ADC3_SWAP_SFT 2 +#define RT5677_IF2_ADC4_SWAP_MASK (0x3 << 0) +#define RT5677_IF2_ADC4_SWAP_SFT 0 + +/* MX41 TDM2 control 2 (0x41) */ #define RT5677_IF2_ADC4_MASK (0x3 << 10) #define RT5677_IF2_ADC4_SFT 10 #define RT5677_IF2_ADC3_MASK (0x3 << 8) @@ -818,6 +848,8 @@ #define RT5677_IF2_ADC2_SFT 6 #define RT5677_IF2_ADC1_MASK (0x3 << 4) #define RT5677_IF2_ADC1_SFT 4 +#define RT5677_IF2_ADC_CTRL_MASK (0x7 << 0) +#define RT5677_IF2_ADC_CTRL_SFT 0 /* Digital Microphone Control 1 (0x50) */ #define RT5677_DMIC_1_EN_MASK (0x1 << 15) -- 1.9.1

2 1

[alsa-devel] [PATCH v2] ASoC: Intel: Update HSW/BDW SST driver to new FW architecture
by Liam Girdwood 28 Oct '14

28 Oct '14

The following patch series updates the current upstream driver to use the new more feature rich ADSP FW architecture. The new FW arch allows for multiple instantiation of objects in dynamic memory locations. In summary :- o Modules can be created multiple times. o Modules persistent data can be allocated in any free blocks o Global scratch data can be allocated into any free blocks. o DMA can be used to load the module data. The driver now has to track and manage the new multiple module resource allocations and pass this data to FW on stream initialisations. Changes since V1 :- o Rebased on top of Vinods latest changes to remove a merge conflict. o Added mutex unlock in block error path. The series is sent as 7 patches for ease of review but should be applied as a single patch to preserve git bisect build. The following changes since commit 9a80e8f597f3bde0e1d4a4abb021d475520005a5: ASoC: Intel: mrfld: Define sst_res_info for acpi (2014-10-27 18:02:38 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc.git intel/new-fw-arch for you to fetch changes up to 86ce414b582c79fa2e5a5f19e046eaee5bfa84e1: ASoC: Intel: Update HSW/BDW SST driver to new FW architecture (2014-10-28 17:05:46 +0000) ---------------------------------------------------------------- Liam Girdwood (1): ASoC: Intel: Update HSW/BDW SST driver to new FW architecture sound/soc/intel/sst-baytrail-dsp.c | 24 +- sound/soc/intel/sst-dsp-priv.h | 133 ++++++--- sound/soc/intel/sst-dsp.c | 8 + sound/soc/intel/sst-dsp.h | 7 + sound/soc/intel/sst-firmware.c | 929 +++++++++++++++++++++++++++++++++++++++++++++++++------------ sound/soc/intel/sst-haswell-dsp.c | 59 ++-- sound/soc/intel/sst-haswell-ipc.c | 112 ++++---- sound/soc/intel/sst-haswell-ipc.h | 11 +- sound/soc/intel/sst-haswell-pcm.c | 83 ++++-- 9 files changed, 1025 insertions(+), 341 deletions(-)

2 1

[alsa-devel] [PATCH] ASoC: core: Split ops out of soc-core.c
by Mark Brown 28 Oct '14

28 Oct '14

The main ASoC source file is getting quite large and the standard ops don't really have anything to do with the rest of the file so split them out into a separate file. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/Makefile | 2 +- sound/soc/soc-core.c | 1006 ------------------------------------------------ sound/soc/soc-ops.c | 1039 ++++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 1040 insertions(+), 1007 deletions(-) create mode 100644 sound/soc/soc-ops.c diff --git a/sound/soc/Makefile b/sound/soc/Makefile index d88edfced8c4..2a043804a2bc 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,5 +1,5 @@ snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o -snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o soc-devres.o +snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o soc-devres.o soc-ops.o ifneq ($(CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM),) snd-soc-core-objs += soc-generic-dmaengine-pcm.o diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 96ecdc30eb60..a2b51edf6d83 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2334,1012 +2334,6 @@ int snd_soc_add_dai_controls(struct snd_soc_dai *dai, EXPORT_SYMBOL_GPL(snd_soc_add_dai_controls); /** - * snd_soc_info_enum_double - enumerated double mixer info callback - * @kcontrol: mixer control - * @uinfo: control element information - * - * Callback to provide information about a double enumerated - * mixer control. - * - * Returns 0 for success. - */ -int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - - return snd_ctl_enum_info(uinfo, e->shift_l == e->shift_r ? 1 : 2, - e->items, e->texts); -} -EXPORT_SYMBOL_GPL(snd_soc_info_enum_double); - -/** - * snd_soc_get_enum_double - enumerated double mixer get callback - * @kcontrol: mixer control - * @ucontrol: control element information - * - * Callback to get the value of a double enumerated mixer. - * - * Returns 0 for success. - */ -int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned int val, item; - unsigned int reg_val; - int ret; - - ret = snd_soc_component_read(component, e->reg, &reg_val); - if (ret) - return ret; - val = (reg_val >> e->shift_l) & e->mask; - item = snd_soc_enum_val_to_item(e, val); - ucontrol->value.enumerated.item[0] = item; - if (e->shift_l != e->shift_r) { - val = (reg_val >> e->shift_l) & e->mask; - item = snd_soc_enum_val_to_item(e, val); - ucontrol->value.enumerated.item[1] = item; - } - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_get_enum_double); - -/** - * snd_soc_put_enum_double - enumerated double mixer put callback - * @kcontrol: mixer control - * @ucontrol: control element information - * - * Callback to set the value of a double enumerated mixer. - * - * Returns 0 for success. - */ -int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned int *item = ucontrol->value.enumerated.item; - unsigned int val; - unsigned int mask; - - if (item[0] >= e->items) - return -EINVAL; - val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l; - mask = e->mask << e->shift_l; - if (e->shift_l != e->shift_r) { - if (item[1] >= e->items) - return -EINVAL; - val |= snd_soc_enum_item_to_val(e, item[1]) << e->shift_r; - mask |= e->mask << e->shift_r; - } - - return snd_soc_component_update_bits(component, e->reg, mask, val); -} -EXPORT_SYMBOL_GPL(snd_soc_put_enum_double); - -/** - * snd_soc_read_signed - Read a codec register and interprete as signed value - * @component: component - * @reg: Register to read - * @mask: Mask to use after shifting the register value - * @shift: Right shift of register value - * @sign_bit: Bit that describes if a number is negative or not. - * @signed_val: Pointer to where the read value should be stored - * - * This functions reads a codec register. The register value is shifted right - * by 'shift' bits and masked with the given 'mask'. Afterwards it translates - * the given registervalue into a signed integer if sign_bit is non-zero. - * - * Returns 0 on sucess, otherwise an error value - */ -static int snd_soc_read_signed(struct snd_soc_component *component, - unsigned int reg, unsigned int mask, unsigned int shift, - unsigned int sign_bit, int *signed_val) -{ - int ret; - unsigned int val; - - ret = snd_soc_component_read(component, reg, &val); - if (ret < 0) - return ret; - - val = (val >> shift) & mask; - - if (!sign_bit) { - *signed_val = val; - return 0; - } - - /* non-negative number */ - if (!(val & BIT(sign_bit))) { - *signed_val = val; - return 0; - } - - ret = val; - - /* - * The register most probably does not contain a full-sized int. - * Instead we have an arbitrary number of bits in a signed - * representation which has to be translated into a full-sized int. - * This is done by filling up all bits above the sign-bit. - */ - ret |= ~((int)(BIT(sign_bit) - 1)); - - *signed_val = ret; - - return 0; -} - -/** - * snd_soc_info_volsw - single mixer info callback - * @kcontrol: mixer control - * @uinfo: control element information - * - * Callback to provide information about a single mixer control, or a double - * mixer control that spans 2 registers. - * - * Returns 0 for success. - */ -int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - int platform_max; - - if (!mc->platform_max) - mc->platform_max = mc->max; - platform_max = mc->platform_max; - - if (platform_max == 1 && !strstr(kcontrol->id.name, " Volume")) - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - else - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - - uinfo->count = snd_soc_volsw_is_stereo(mc) ? 2 : 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = platform_max - mc->min; - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_info_volsw); - -/** - * snd_soc_get_volsw - single mixer get callback - * @kcontrol: mixer control - * @ucontrol: control element information - * - * Callback to get the value of a single mixer control, or a double mixer - * control that spans 2 registers. - * - * Returns 0 for success. - */ -int snd_soc_get_volsw(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - unsigned int reg = mc->reg; - unsigned int reg2 = mc->rreg; - unsigned int shift = mc->shift; - unsigned int rshift = mc->rshift; - int max = mc->max; - int min = mc->min; - int sign_bit = mc->sign_bit; - unsigned int mask = (1 << fls(max)) - 1; - unsigned int invert = mc->invert; - int val; - int ret; - - if (sign_bit) - mask = BIT(sign_bit + 1) - 1; - - ret = snd_soc_read_signed(component, reg, mask, shift, sign_bit, &val); - if (ret) - return ret; - - ucontrol->value.integer.value[0] = val - min; - if (invert) - ucontrol->value.integer.value[0] = - max - ucontrol->value.integer.value[0]; - - if (snd_soc_volsw_is_stereo(mc)) { - if (reg == reg2) - ret = snd_soc_read_signed(component, reg, mask, rshift, - sign_bit, &val); - else - ret = snd_soc_read_signed(component, reg2, mask, shift, - sign_bit, &val); - if (ret) - return ret; - - ucontrol->value.integer.value[1] = val - min; - if (invert) - ucontrol->value.integer.value[1] = - max - ucontrol->value.integer.value[1]; - } - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_get_volsw); - -/** - * snd_soc_put_volsw - single mixer put callback - * @kcontrol: mixer control - * @ucontrol: control element information - * - * Callback to set the value of a single mixer control, or a double mixer - * control that spans 2 registers. - * - * Returns 0 for success. - */ -int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - unsigned int reg = mc->reg; - unsigned int reg2 = mc->rreg; - unsigned int shift = mc->shift; - unsigned int rshift = mc->rshift; - int max = mc->max; - int min = mc->min; - unsigned int sign_bit = mc->sign_bit; - unsigned int mask = (1 << fls(max)) - 1; - unsigned int invert = mc->invert; - int err; - bool type_2r = false; - unsigned int val2 = 0; - unsigned int val, val_mask; - - if (sign_bit) - mask = BIT(sign_bit + 1) - 1; - - val = ((ucontrol->value.integer.value[0] + min) & mask); - if (invert) - val = max - val; - val_mask = mask << shift; - val = val << shift; - if (snd_soc_volsw_is_stereo(mc)) { - val2 = ((ucontrol->value.integer.value[1] + min) & mask); - if (invert) - val2 = max - val2; - if (reg == reg2) { - val_mask |= mask << rshift; - val |= val2 << rshift; - } else { - val2 = val2 << shift; - type_2r = true; - } - } - err = snd_soc_component_update_bits(component, reg, val_mask, val); - if (err < 0) - return err; - - if (type_2r) - err = snd_soc_component_update_bits(component, reg2, val_mask, - val2); - - return err; -} -EXPORT_SYMBOL_GPL(snd_soc_put_volsw); - -/** - * snd_soc_get_volsw_sx - single mixer get callback - * @kcontrol: mixer control - * @ucontrol: control element information - * - * Callback to get the value of a single mixer control, or a double mixer - * control that spans 2 registers. - * - * Returns 0 for success. - */ -int snd_soc_get_volsw_sx(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - unsigned int reg = mc->reg; - unsigned int reg2 = mc->rreg; - unsigned int shift = mc->shift; - unsigned int rshift = mc->rshift; - int max = mc->max; - int min = mc->min; - int mask = (1 << (fls(min + max) - 1)) - 1; - unsigned int val; - int ret; - - ret = snd_soc_component_read(component, reg, &val); - if (ret < 0) - return ret; - - ucontrol->value.integer.value[0] = ((val >> shift) - min) & mask; - - if (snd_soc_volsw_is_stereo(mc)) { - ret = snd_soc_component_read(component, reg2, &val); - if (ret < 0) - return ret; - - val = ((val >> rshift) - min) & mask; - ucontrol->value.integer.value[1] = val; - } - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_get_volsw_sx); - -/** - * snd_soc_put_volsw_sx - double mixer set callback - * @kcontrol: mixer control - * @uinfo: control element information - * - * Callback to set the value of a double mixer control that spans 2 registers. - * - * Returns 0 for success. - */ -int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - - unsigned int reg = mc->reg; - unsigned int reg2 = mc->rreg; - unsigned int shift = mc->shift; - unsigned int rshift = mc->rshift; - int max = mc->max; - int min = mc->min; - int mask = (1 << (fls(min + max) - 1)) - 1; - int err = 0; - unsigned int val, val_mask, val2 = 0; - - val_mask = mask << shift; - val = (ucontrol->value.integer.value[0] + min) & mask; - val = val << shift; - - err = snd_soc_component_update_bits(component, reg, val_mask, val); - if (err < 0) - return err; - - if (snd_soc_volsw_is_stereo(mc)) { - val_mask = mask << rshift; - val2 = (ucontrol->value.integer.value[1] + min) & mask; - val2 = val2 << rshift; - - err = snd_soc_component_update_bits(component, reg2, val_mask, - val2); - } - return err; -} -EXPORT_SYMBOL_GPL(snd_soc_put_volsw_sx); - -/** - * snd_soc_info_volsw_s8 - signed mixer info callback - * @kcontrol: mixer control - * @uinfo: control element information - * - * Callback to provide information about a signed mixer control. - * - * Returns 0 for success. - */ -int snd_soc_info_volsw_s8(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - int platform_max; - int min = mc->min; - - if (!mc->platform_max) - mc->platform_max = mc->max; - platform_max = mc->platform_max; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 2; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = platform_max - min; - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_info_volsw_s8); - -/** - * snd_soc_get_volsw_s8 - signed mixer get callback - * @kcontrol: mixer control - * @ucontrol: control element information - * - * Callback to get the value of a signed mixer control. - * - * Returns 0 for success. - */ -int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - unsigned int reg = mc->reg; - unsigned int val; - int min = mc->min; - int ret; - - ret = snd_soc_component_read(component, reg, &val); - if (ret) - return ret; - - ucontrol->value.integer.value[0] = - ((signed char)(val & 0xff))-min; - ucontrol->value.integer.value[1] = - ((signed char)((val >> 8) & 0xff))-min; - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_get_volsw_s8); - -/** - * snd_soc_put_volsw_sgn - signed mixer put callback - * @kcontrol: mixer control - * @ucontrol: control element information - * - * Callback to set the value of a signed mixer control. - * - * Returns 0 for success. - */ -int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - unsigned int reg = mc->reg; - int min = mc->min; - unsigned int val; - - val = (ucontrol->value.integer.value[0]+min) & 0xff; - val |= ((ucontrol->value.integer.value[1]+min) & 0xff) << 8; - - return snd_soc_component_update_bits(component, reg, 0xffff, val); -} -EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8); - -/** - * snd_soc_info_volsw_range - single mixer info callback with range. - * @kcontrol: mixer control - * @uinfo: control element information - * - * Callback to provide information, within a range, about a single - * mixer control. - * - * returns 0 for success. - */ -int snd_soc_info_volsw_range(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - int platform_max; - int min = mc->min; - - if (!mc->platform_max) - mc->platform_max = mc->max; - platform_max = mc->platform_max; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = snd_soc_volsw_is_stereo(mc) ? 2 : 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = platform_max - min; - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_info_volsw_range); - -/** - * snd_soc_put_volsw_range - single mixer put value callback with range. - * @kcontrol: mixer control - * @ucontrol: control element information - * - * Callback to set the value, within a range, for a single mixer control. - * - * Returns 0 for success. - */ -int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - unsigned int reg = mc->reg; - unsigned int rreg = mc->rreg; - unsigned int shift = mc->shift; - int min = mc->min; - int max = mc->max; - unsigned int mask = (1 << fls(max)) - 1; - unsigned int invert = mc->invert; - unsigned int val, val_mask; - int ret; - - if (invert) - val = (max - ucontrol->value.integer.value[0]) & mask; - else - val = ((ucontrol->value.integer.value[0] + min) & mask); - val_mask = mask << shift; - val = val << shift; - - ret = snd_soc_component_update_bits(component, reg, val_mask, val); - if (ret < 0) - return ret; - - if (snd_soc_volsw_is_stereo(mc)) { - if (invert) - val = (max - ucontrol->value.integer.value[1]) & mask; - else - val = ((ucontrol->value.integer.value[1] + min) & mask); - val_mask = mask << shift; - val = val << shift; - - ret = snd_soc_component_update_bits(component, rreg, val_mask, - val); - } - - return ret; -} -EXPORT_SYMBOL_GPL(snd_soc_put_volsw_range); - -/** - * snd_soc_get_volsw_range - single mixer get callback with range - * @kcontrol: mixer control - * @ucontrol: control element information - * - * Callback to get the value, within a range, of a single mixer control. - * - * Returns 0 for success. - */ -int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - unsigned int reg = mc->reg; - unsigned int rreg = mc->rreg; - unsigned int shift = mc->shift; - int min = mc->min; - int max = mc->max; - unsigned int mask = (1 << fls(max)) - 1; - unsigned int invert = mc->invert; - unsigned int val; - int ret; - - ret = snd_soc_component_read(component, reg, &val); - if (ret) - return ret; - - ucontrol->value.integer.value[0] = (val >> shift) & mask; - if (invert) - ucontrol->value.integer.value[0] = - max - ucontrol->value.integer.value[0]; - else - ucontrol->value.integer.value[0] = - ucontrol->value.integer.value[0] - min; - - if (snd_soc_volsw_is_stereo(mc)) { - ret = snd_soc_component_read(component, rreg, &val); - if (ret) - return ret; - - ucontrol->value.integer.value[1] = (val >> shift) & mask; - if (invert) - ucontrol->value.integer.value[1] = - max - ucontrol->value.integer.value[1]; - else - ucontrol->value.integer.value[1] = - ucontrol->value.integer.value[1] - min; - } - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_get_volsw_range); - -/** - * snd_soc_limit_volume - Set new limit to an existing volume control. - * - * @codec: where to look for the control - * @name: Name of the control - * @max: new maximum limit - * - * Return 0 for success, else error. - */ -int snd_soc_limit_volume(struct snd_soc_codec *codec, - const char *name, int max) -{ - struct snd_card *card = codec->component.card->snd_card; - struct snd_kcontrol *kctl; - struct soc_mixer_control *mc; - int found = 0; - int ret = -EINVAL; - - /* Sanity check for name and max */ - if (unlikely(!name || max <= 0)) - return -EINVAL; - - list_for_each_entry(kctl, &card->controls, list) { - if (!strncmp(kctl->id.name, name, sizeof(kctl->id.name))) { - found = 1; - break; - } - } - if (found) { - mc = (struct soc_mixer_control *)kctl->private_value; - if (max <= mc->max) { - mc->platform_max = max; - ret = 0; - } - } - return ret; -} -EXPORT_SYMBOL_GPL(snd_soc_limit_volume); - -int snd_soc_bytes_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct soc_bytes *params = (void *)kcontrol->private_value; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; - uinfo->count = params->num_regs * component->val_bytes; - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_bytes_info); - -int snd_soc_bytes_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct soc_bytes *params = (void *)kcontrol->private_value; - int ret; - - if (component->regmap) - ret = regmap_raw_read(component->regmap, params->base, - ucontrol->value.bytes.data, - params->num_regs * component->val_bytes); - else - ret = -EINVAL; - - /* Hide any masked bytes to ensure consistent data reporting */ - if (ret == 0 && params->mask) { - switch (component->val_bytes) { - case 1: - ucontrol->value.bytes.data[0] &= ~params->mask; - break; - case 2: - ((u16 *)(&ucontrol->value.bytes.data))[0] - &= cpu_to_be16(~params->mask); - break; - case 4: - ((u32 *)(&ucontrol->value.bytes.data))[0] - &= cpu_to_be32(~params->mask); - break; - default: - return -EINVAL; - } - } - - return ret; -} -EXPORT_SYMBOL_GPL(snd_soc_bytes_get); - -int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct soc_bytes *params = (void *)kcontrol->private_value; - int ret, len; - unsigned int val, mask; - void *data; - - if (!component->regmap || !params->num_regs) - return -EINVAL; - - len = params->num_regs * component->val_bytes; - - data = kmemdup(ucontrol->value.bytes.data, len, GFP_KERNEL | GFP_DMA); - if (!data) - return -ENOMEM; - - /* - * If we've got a mask then we need to preserve the register - * bits. We shouldn't modify the incoming data so take a - * copy. - */ - if (params->mask) { - ret = regmap_read(component->regmap, params->base, &val); - if (ret != 0) - goto out; - - val &= params->mask; - - switch (component->val_bytes) { - case 1: - ((u8 *)data)[0] &= ~params->mask; - ((u8 *)data)[0] |= val; - break; - case 2: - mask = ~params->mask; - ret = regmap_parse_val(component->regmap, - &mask, &mask); - if (ret != 0) - goto out; - - ((u16 *)data)[0] &= mask; - - ret = regmap_parse_val(component->regmap, - &val, &val); - if (ret != 0) - goto out; - - ((u16 *)data)[0] |= val; - break; - case 4: - mask = ~params->mask; - ret = regmap_parse_val(component->regmap, - &mask, &mask); - if (ret != 0) - goto out; - - ((u32 *)data)[0] &= mask; - - ret = regmap_parse_val(component->regmap, - &val, &val); - if (ret != 0) - goto out; - - ((u32 *)data)[0] |= val; - break; - default: - ret = -EINVAL; - goto out; - } - } - - ret = regmap_raw_write(component->regmap, params->base, - data, len); - -out: - kfree(data); - - return ret; -} -EXPORT_SYMBOL_GPL(snd_soc_bytes_put); - -int snd_soc_bytes_info_ext(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *ucontrol) -{ - struct soc_bytes_ext *params = (void *)kcontrol->private_value; - - ucontrol->type = SNDRV_CTL_ELEM_TYPE_BYTES; - ucontrol->count = params->max; - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_bytes_info_ext); - -int snd_soc_bytes_tlv_callback(struct snd_kcontrol *kcontrol, int op_flag, - unsigned int size, unsigned int __user *tlv) -{ - struct soc_bytes_ext *params = (void *)kcontrol->private_value; - unsigned int count = size < params->max ? size : params->max; - int ret = -ENXIO; - - switch (op_flag) { - case SNDRV_CTL_TLV_OP_READ: - if (params->get) - ret = params->get(tlv, count); - break; - case SNDRV_CTL_TLV_OP_WRITE: - if (params->put) - ret = params->put(tlv, count); - break; - } - return ret; -} -EXPORT_SYMBOL_GPL(snd_soc_bytes_tlv_callback); - -/** - * snd_soc_info_xr_sx - signed multi register info callback - * @kcontrol: mreg control - * @uinfo: control element information - * - * Callback to provide information of a control that can - * span multiple codec registers which together - * forms a single signed value in a MSB/LSB manner. - * - * Returns 0 for success. - */ -int snd_soc_info_xr_sx(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct soc_mreg_control *mc = - (struct soc_mreg_control *)kcontrol->private_value; - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 1; - uinfo->value.integer.min = mc->min; - uinfo->value.integer.max = mc->max; - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_info_xr_sx); - -/** - * snd_soc_get_xr_sx - signed multi register get callback - * @kcontrol: mreg control - * @ucontrol: control element information - * - * Callback to get the value of a control that can span - * multiple codec registers which together forms a single - * signed value in a MSB/LSB manner. The control supports - * specifying total no of bits used to allow for bitfields - * across the multiple codec registers. - * - * Returns 0 for success. - */ -int snd_soc_get_xr_sx(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct soc_mreg_control *mc = - (struct soc_mreg_control *)kcontrol->private_value; - unsigned int regbase = mc->regbase; - unsigned int regcount = mc->regcount; - unsigned int regwshift = component->val_bytes * BITS_PER_BYTE; - unsigned int regwmask = (1<<regwshift)-1; - unsigned int invert = mc->invert; - unsigned long mask = (1UL<<mc->nbits)-1; - long min = mc->min; - long max = mc->max; - long val = 0; - unsigned int regval; - unsigned int i; - int ret; - - for (i = 0; i < regcount; i++) { - ret = snd_soc_component_read(component, regbase+i, &regval); - if (ret) - return ret; - val |= (regval & regwmask) << (regwshift*(regcount-i-1)); - } - val &= mask; - if (min < 0 && val > max) - val |= ~mask; - if (invert) - val = max - val; - ucontrol->value.integer.value[0] = val; - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_get_xr_sx); - -/** - * snd_soc_put_xr_sx - signed multi register get callback - * @kcontrol: mreg control - * @ucontrol: control element information - * - * Callback to set the value of a control that can span - * multiple codec registers which together forms a single - * signed value in a MSB/LSB manner. The control supports - * specifying total no of bits used to allow for bitfields - * across the multiple codec registers. - * - * Returns 0 for success. - */ -int snd_soc_put_xr_sx(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct soc_mreg_control *mc = - (struct soc_mreg_control *)kcontrol->private_value; - unsigned int regbase = mc->regbase; - unsigned int regcount = mc->regcount; - unsigned int regwshift = component->val_bytes * BITS_PER_BYTE; - unsigned int regwmask = (1<<regwshift)-1; - unsigned int invert = mc->invert; - unsigned long mask = (1UL<<mc->nbits)-1; - long max = mc->max; - long val = ucontrol->value.integer.value[0]; - unsigned int i, regval, regmask; - int err; - - if (invert) - val = max - val; - val &= mask; - for (i = 0; i < regcount; i++) { - regval = (val >> (regwshift*(regcount-i-1))) & regwmask; - regmask = (mask >> (regwshift*(regcount-i-1))) & regwmask; - err = snd_soc_component_update_bits(component, regbase+i, - regmask, regval); - if (err < 0) - return err; - } - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_put_xr_sx); - -/** - * snd_soc_get_strobe - strobe get callback - * @kcontrol: mixer control - * @ucontrol: control element information - * - * Callback get the value of a strobe mixer control. - * - * Returns 0 for success. - */ -int snd_soc_get_strobe(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - unsigned int reg = mc->reg; - unsigned int shift = mc->shift; - unsigned int mask = 1 << shift; - unsigned int invert = mc->invert != 0; - unsigned int val; - int ret; - - ret = snd_soc_component_read(component, reg, &val); - if (ret) - return ret; - - val &= mask; - - if (shift != 0 && val != 0) - val = val >> shift; - ucontrol->value.enumerated.item[0] = val ^ invert; - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_get_strobe); - -/** - * snd_soc_put_strobe - strobe put callback - * @kcontrol: mixer control - * @ucontrol: control element information - * - * Callback strobe a register bit to high then low (or the inverse) - * in one pass of a single mixer enum control. - * - * Returns 1 for success. - */ -int snd_soc_put_strobe(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - unsigned int reg = mc->reg; - unsigned int shift = mc->shift; - unsigned int mask = 1 << shift; - unsigned int invert = mc->invert != 0; - unsigned int strobe = ucontrol->value.enumerated.item[0] != 0; - unsigned int val1 = (strobe ^ invert) ? mask : 0; - unsigned int val2 = (strobe ^ invert) ? 0 : mask; - int err; - - err = snd_soc_component_update_bits(component, reg, mask, val1); - if (err < 0) - return err; - - return snd_soc_component_update_bits(component, reg, mask, val2); -} -EXPORT_SYMBOL_GPL(snd_soc_put_strobe); - -/** * snd_soc_dai_set_sysclk - configure DAI system or master clock. * @dai: DAI * @clk_id: DAI specific clock ID diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c new file mode 100644 index 000000000000..c326452a8f8e --- /dev/null +++ b/sound/soc/soc-ops.c @@ -0,0 +1,1039 @@ +/* + * soc-ops.c -- Generic ASoC operations + * + * Copyright 2005 Wolfson Microelectronics PLC. + * Copyright 2005 Openedhand Ltd. + * Copyright (C) 2010 Slimlogic Ltd. + * Copyright (C) 2010 Texas Instruments Inc. + * + * Author: Liam Girdwood <lrg(a)slimlogic.co.uk> + * with code, comments and ideas from :- + * Richard Purdie <richard(a)openedhand.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/bitops.h> +#include <linux/ctype.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dpcm.h> +#include <sound/initval.h> + +/** + * snd_soc_info_enum_double - enumerated double mixer info callback + * @kcontrol: mixer control + * @uinfo: control element information + * + * Callback to provide information about a double enumerated + * mixer control. + * + * Returns 0 for success. + */ +int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + + return snd_ctl_enum_info(uinfo, e->shift_l == e->shift_r ? 1 : 2, + e->items, e->texts); +} +EXPORT_SYMBOL_GPL(snd_soc_info_enum_double); + +/** + * snd_soc_get_enum_double - enumerated double mixer get callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to get the value of a double enumerated mixer. + * + * Returns 0 for success. + */ +int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int val, item; + unsigned int reg_val; + int ret; + + ret = snd_soc_component_read(component, e->reg, &reg_val); + if (ret) + return ret; + val = (reg_val >> e->shift_l) & e->mask; + item = snd_soc_enum_val_to_item(e, val); + ucontrol->value.enumerated.item[0] = item; + if (e->shift_l != e->shift_r) { + val = (reg_val >> e->shift_l) & e->mask; + item = snd_soc_enum_val_to_item(e, val); + ucontrol->value.enumerated.item[1] = item; + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_get_enum_double); + +/** + * snd_soc_put_enum_double - enumerated double mixer put callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to set the value of a double enumerated mixer. + * + * Returns 0 for success. + */ +int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int *item = ucontrol->value.enumerated.item; + unsigned int val; + unsigned int mask; + + if (item[0] >= e->items) + return -EINVAL; + val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l; + mask = e->mask << e->shift_l; + if (e->shift_l != e->shift_r) { + if (item[1] >= e->items) + return -EINVAL; + val |= snd_soc_enum_item_to_val(e, item[1]) << e->shift_r; + mask |= e->mask << e->shift_r; + } + + return snd_soc_component_update_bits(component, e->reg, mask, val); +} +EXPORT_SYMBOL_GPL(snd_soc_put_enum_double); + +/** + * snd_soc_read_signed - Read a codec register and interprete as signed value + * @component: component + * @reg: Register to read + * @mask: Mask to use after shifting the register value + * @shift: Right shift of register value + * @sign_bit: Bit that describes if a number is negative or not. + * @signed_val: Pointer to where the read value should be stored + * + * This functions reads a codec register. The register value is shifted right + * by 'shift' bits and masked with the given 'mask'. Afterwards it translates + * the given registervalue into a signed integer if sign_bit is non-zero. + * + * Returns 0 on sucess, otherwise an error value + */ +static int snd_soc_read_signed(struct snd_soc_component *component, + unsigned int reg, unsigned int mask, unsigned int shift, + unsigned int sign_bit, int *signed_val) +{ + int ret; + unsigned int val; + + ret = snd_soc_component_read(component, reg, &val); + if (ret < 0) + return ret; + + val = (val >> shift) & mask; + + if (!sign_bit) { + *signed_val = val; + return 0; + } + + /* non-negative number */ + if (!(val & BIT(sign_bit))) { + *signed_val = val; + return 0; + } + + ret = val; + + /* + * The register most probably does not contain a full-sized int. + * Instead we have an arbitrary number of bits in a signed + * representation which has to be translated into a full-sized int. + * This is done by filling up all bits above the sign-bit. + */ + ret |= ~((int)(BIT(sign_bit) - 1)); + + *signed_val = ret; + + return 0; +} + +/** + * snd_soc_info_volsw - single mixer info callback + * @kcontrol: mixer control + * @uinfo: control element information + * + * Callback to provide information about a single mixer control, or a double + * mixer control that spans 2 registers. + * + * Returns 0 for success. + */ +int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int platform_max; + + if (!mc->platform_max) + mc->platform_max = mc->max; + platform_max = mc->platform_max; + + if (platform_max == 1 && !strstr(kcontrol->id.name, " Volume")) + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + else + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + + uinfo->count = snd_soc_volsw_is_stereo(mc) ? 2 : 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = platform_max - mc->min; + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_info_volsw); + +/** + * snd_soc_get_volsw - single mixer get callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to get the value of a single mixer control, or a double mixer + * control that spans 2 registers. + * + * Returns 0 for success. + */ +int snd_soc_get_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; + unsigned int shift = mc->shift; + unsigned int rshift = mc->rshift; + int max = mc->max; + int min = mc->min; + int sign_bit = mc->sign_bit; + unsigned int mask = (1 << fls(max)) - 1; + unsigned int invert = mc->invert; + int val; + int ret; + + if (sign_bit) + mask = BIT(sign_bit + 1) - 1; + + ret = snd_soc_read_signed(component, reg, mask, shift, sign_bit, &val); + if (ret) + return ret; + + ucontrol->value.integer.value[0] = val - min; + if (invert) + ucontrol->value.integer.value[0] = + max - ucontrol->value.integer.value[0]; + + if (snd_soc_volsw_is_stereo(mc)) { + if (reg == reg2) + ret = snd_soc_read_signed(component, reg, mask, rshift, + sign_bit, &val); + else + ret = snd_soc_read_signed(component, reg2, mask, shift, + sign_bit, &val); + if (ret) + return ret; + + ucontrol->value.integer.value[1] = val - min; + if (invert) + ucontrol->value.integer.value[1] = + max - ucontrol->value.integer.value[1]; + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_get_volsw); + +/** + * snd_soc_put_volsw - single mixer put callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to set the value of a single mixer control, or a double mixer + * control that spans 2 registers. + * + * Returns 0 for success. + */ +int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; + unsigned int shift = mc->shift; + unsigned int rshift = mc->rshift; + int max = mc->max; + int min = mc->min; + unsigned int sign_bit = mc->sign_bit; + unsigned int mask = (1 << fls(max)) - 1; + unsigned int invert = mc->invert; + int err; + bool type_2r = false; + unsigned int val2 = 0; + unsigned int val, val_mask; + + if (sign_bit) + mask = BIT(sign_bit + 1) - 1; + + val = ((ucontrol->value.integer.value[0] + min) & mask); + if (invert) + val = max - val; + val_mask = mask << shift; + val = val << shift; + if (snd_soc_volsw_is_stereo(mc)) { + val2 = ((ucontrol->value.integer.value[1] + min) & mask); + if (invert) + val2 = max - val2; + if (reg == reg2) { + val_mask |= mask << rshift; + val |= val2 << rshift; + } else { + val2 = val2 << shift; + type_2r = true; + } + } + err = snd_soc_component_update_bits(component, reg, val_mask, val); + if (err < 0) + return err; + + if (type_2r) + err = snd_soc_component_update_bits(component, reg2, val_mask, + val2); + + return err; +} +EXPORT_SYMBOL_GPL(snd_soc_put_volsw); + +/** + * snd_soc_get_volsw_sx - single mixer get callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to get the value of a single mixer control, or a double mixer + * control that spans 2 registers. + * + * Returns 0 for success. + */ +int snd_soc_get_volsw_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; + unsigned int shift = mc->shift; + unsigned int rshift = mc->rshift; + int max = mc->max; + int min = mc->min; + int mask = (1 << (fls(min + max) - 1)) - 1; + unsigned int val; + int ret; + + ret = snd_soc_component_read(component, reg, &val); + if (ret < 0) + return ret; + + ucontrol->value.integer.value[0] = ((val >> shift) - min) & mask; + + if (snd_soc_volsw_is_stereo(mc)) { + ret = snd_soc_component_read(component, reg2, &val); + if (ret < 0) + return ret; + + val = ((val >> rshift) - min) & mask; + ucontrol->value.integer.value[1] = val; + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_get_volsw_sx); + +/** + * snd_soc_put_volsw_sx - double mixer set callback + * @kcontrol: mixer control + * @uinfo: control element information + * + * Callback to set the value of a double mixer control that spans 2 registers. + * + * Returns 0 for success. + */ +int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; + unsigned int shift = mc->shift; + unsigned int rshift = mc->rshift; + int max = mc->max; + int min = mc->min; + int mask = (1 << (fls(min + max) - 1)) - 1; + int err = 0; + unsigned int val, val_mask, val2 = 0; + + val_mask = mask << shift; + val = (ucontrol->value.integer.value[0] + min) & mask; + val = val << shift; + + err = snd_soc_component_update_bits(component, reg, val_mask, val); + if (err < 0) + return err; + + if (snd_soc_volsw_is_stereo(mc)) { + val_mask = mask << rshift; + val2 = (ucontrol->value.integer.value[1] + min) & mask; + val2 = val2 << rshift; + + err = snd_soc_component_update_bits(component, reg2, val_mask, + val2); + } + return err; +} +EXPORT_SYMBOL_GPL(snd_soc_put_volsw_sx); + +/** + * snd_soc_info_volsw_s8 - signed mixer info callback + * @kcontrol: mixer control + * @uinfo: control element information + * + * Callback to provide information about a signed mixer control. + * + * Returns 0 for success. + */ +int snd_soc_info_volsw_s8(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int platform_max; + int min = mc->min; + + if (!mc->platform_max) + mc->platform_max = mc->max; + platform_max = mc->platform_max; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = platform_max - min; + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_info_volsw_s8); + +/** + * snd_soc_get_volsw_s8 - signed mixer get callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to get the value of a signed mixer control. + * + * Returns 0 for success. + */ +int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + unsigned int reg = mc->reg; + unsigned int val; + int min = mc->min; + int ret; + + ret = snd_soc_component_read(component, reg, &val); + if (ret) + return ret; + + ucontrol->value.integer.value[0] = + ((signed char)(val & 0xff))-min; + ucontrol->value.integer.value[1] = + ((signed char)((val >> 8) & 0xff))-min; + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_get_volsw_s8); + +/** + * snd_soc_put_volsw_sgn - signed mixer put callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to set the value of a signed mixer control. + * + * Returns 0 for success. + */ +int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + unsigned int reg = mc->reg; + int min = mc->min; + unsigned int val; + + val = (ucontrol->value.integer.value[0]+min) & 0xff; + val |= ((ucontrol->value.integer.value[1]+min) & 0xff) << 8; + + return snd_soc_component_update_bits(component, reg, 0xffff, val); +} +EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8); + +/** + * snd_soc_info_volsw_range - single mixer info callback with range. + * @kcontrol: mixer control + * @uinfo: control element information + * + * Callback to provide information, within a range, about a single + * mixer control. + * + * returns 0 for success. + */ +int snd_soc_info_volsw_range(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int platform_max; + int min = mc->min; + + if (!mc->platform_max) + mc->platform_max = mc->max; + platform_max = mc->platform_max; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = snd_soc_volsw_is_stereo(mc) ? 2 : 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = platform_max - min; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_info_volsw_range); + +/** + * snd_soc_put_volsw_range - single mixer put value callback with range. + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to set the value, within a range, for a single mixer control. + * + * Returns 0 for success. + */ +int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + unsigned int reg = mc->reg; + unsigned int rreg = mc->rreg; + unsigned int shift = mc->shift; + int min = mc->min; + int max = mc->max; + unsigned int mask = (1 << fls(max)) - 1; + unsigned int invert = mc->invert; + unsigned int val, val_mask; + int ret; + + if (invert) + val = (max - ucontrol->value.integer.value[0]) & mask; + else + val = ((ucontrol->value.integer.value[0] + min) & mask); + val_mask = mask << shift; + val = val << shift; + + ret = snd_soc_component_update_bits(component, reg, val_mask, val); + if (ret < 0) + return ret; + + if (snd_soc_volsw_is_stereo(mc)) { + if (invert) + val = (max - ucontrol->value.integer.value[1]) & mask; + else + val = ((ucontrol->value.integer.value[1] + min) & mask); + val_mask = mask << shift; + val = val << shift; + + ret = snd_soc_component_update_bits(component, rreg, val_mask, + val); + } + + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_put_volsw_range); + +/** + * snd_soc_get_volsw_range - single mixer get callback with range + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to get the value, within a range, of a single mixer control. + * + * Returns 0 for success. + */ +int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int reg = mc->reg; + unsigned int rreg = mc->rreg; + unsigned int shift = mc->shift; + int min = mc->min; + int max = mc->max; + unsigned int mask = (1 << fls(max)) - 1; + unsigned int invert = mc->invert; + unsigned int val; + int ret; + + ret = snd_soc_component_read(component, reg, &val); + if (ret) + return ret; + + ucontrol->value.integer.value[0] = (val >> shift) & mask; + if (invert) + ucontrol->value.integer.value[0] = + max - ucontrol->value.integer.value[0]; + else + ucontrol->value.integer.value[0] = + ucontrol->value.integer.value[0] - min; + + if (snd_soc_volsw_is_stereo(mc)) { + ret = snd_soc_component_read(component, rreg, &val); + if (ret) + return ret; + + ucontrol->value.integer.value[1] = (val >> shift) & mask; + if (invert) + ucontrol->value.integer.value[1] = + max - ucontrol->value.integer.value[1]; + else + ucontrol->value.integer.value[1] = + ucontrol->value.integer.value[1] - min; + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_get_volsw_range); + +/** + * snd_soc_limit_volume - Set new limit to an existing volume control. + * + * @codec: where to look for the control + * @name: Name of the control + * @max: new maximum limit + * + * Return 0 for success, else error. + */ +int snd_soc_limit_volume(struct snd_soc_codec *codec, + const char *name, int max) +{ + struct snd_card *card = codec->component.card->snd_card; + struct snd_kcontrol *kctl; + struct soc_mixer_control *mc; + int found = 0; + int ret = -EINVAL; + + /* Sanity check for name and max */ + if (unlikely(!name || max <= 0)) + return -EINVAL; + + list_for_each_entry(kctl, &card->controls, list) { + if (!strncmp(kctl->id.name, name, sizeof(kctl->id.name))) { + found = 1; + break; + } + } + if (found) { + mc = (struct soc_mixer_control *)kctl->private_value; + if (max <= mc->max) { + mc->platform_max = max; + ret = 0; + } + } + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_limit_volume); + +int snd_soc_bytes_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_bytes *params = (void *)kcontrol->private_value; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + uinfo->count = params->num_regs * component->val_bytes; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_bytes_info); + +int snd_soc_bytes_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_bytes *params = (void *)kcontrol->private_value; + int ret; + + if (component->regmap) + ret = regmap_raw_read(component->regmap, params->base, + ucontrol->value.bytes.data, + params->num_regs * component->val_bytes); + else + ret = -EINVAL; + + /* Hide any masked bytes to ensure consistent data reporting */ + if (ret == 0 && params->mask) { + switch (component->val_bytes) { + case 1: + ucontrol->value.bytes.data[0] &= ~params->mask; + break; + case 2: + ((u16 *)(&ucontrol->value.bytes.data))[0] + &= cpu_to_be16(~params->mask); + break; + case 4: + ((u32 *)(&ucontrol->value.bytes.data))[0] + &= cpu_to_be32(~params->mask); + break; + default: + return -EINVAL; + } + } + + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_bytes_get); + +int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_bytes *params = (void *)kcontrol->private_value; + int ret, len; + unsigned int val, mask; + void *data; + + if (!component->regmap || !params->num_regs) + return -EINVAL; + + len = params->num_regs * component->val_bytes; + + data = kmemdup(ucontrol->value.bytes.data, len, GFP_KERNEL | GFP_DMA); + if (!data) + return -ENOMEM; + + /* + * If we've got a mask then we need to preserve the register + * bits. We shouldn't modify the incoming data so take a + * copy. + */ + if (params->mask) { + ret = regmap_read(component->regmap, params->base, &val); + if (ret != 0) + goto out; + + val &= params->mask; + + switch (component->val_bytes) { + case 1: + ((u8 *)data)[0] &= ~params->mask; + ((u8 *)data)[0] |= val; + break; + case 2: + mask = ~params->mask; + ret = regmap_parse_val(component->regmap, + &mask, &mask); + if (ret != 0) + goto out; + + ((u16 *)data)[0] &= mask; + + ret = regmap_parse_val(component->regmap, + &val, &val); + if (ret != 0) + goto out; + + ((u16 *)data)[0] |= val; + break; + case 4: + mask = ~params->mask; + ret = regmap_parse_val(component->regmap, + &mask, &mask); + if (ret != 0) + goto out; + + ((u32 *)data)[0] &= mask; + + ret = regmap_parse_val(component->regmap, + &val, &val); + if (ret != 0) + goto out; + + ((u32 *)data)[0] |= val; + break; + default: + ret = -EINVAL; + goto out; + } + } + + ret = regmap_raw_write(component->regmap, params->base, + data, len); + +out: + kfree(data); + + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_bytes_put); + +int snd_soc_bytes_info_ext(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *ucontrol) +{ + struct soc_bytes_ext *params = (void *)kcontrol->private_value; + + ucontrol->type = SNDRV_CTL_ELEM_TYPE_BYTES; + ucontrol->count = params->max; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_bytes_info_ext); + +int snd_soc_bytes_tlv_callback(struct snd_kcontrol *kcontrol, int op_flag, + unsigned int size, unsigned int __user *tlv) +{ + struct soc_bytes_ext *params = (void *)kcontrol->private_value; + unsigned int count = size < params->max ? size : params->max; + int ret = -ENXIO; + + switch (op_flag) { + case SNDRV_CTL_TLV_OP_READ: + if (params->get) + ret = params->get(tlv, count); + break; + case SNDRV_CTL_TLV_OP_WRITE: + if (params->put) + ret = params->put(tlv, count); + break; + } + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_bytes_tlv_callback); + +/** + * snd_soc_info_xr_sx - signed multi register info callback + * @kcontrol: mreg control + * @uinfo: control element information + * + * Callback to provide information of a control that can + * span multiple codec registers which together + * forms a single signed value in a MSB/LSB manner. + * + * Returns 0 for success. + */ +int snd_soc_info_xr_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct soc_mreg_control *mc = + (struct soc_mreg_control *)kcontrol->private_value; + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = mc->min; + uinfo->value.integer.max = mc->max; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_info_xr_sx); + +/** + * snd_soc_get_xr_sx - signed multi register get callback + * @kcontrol: mreg control + * @ucontrol: control element information + * + * Callback to get the value of a control that can span + * multiple codec registers which together forms a single + * signed value in a MSB/LSB manner. The control supports + * specifying total no of bits used to allow for bitfields + * across the multiple codec registers. + * + * Returns 0 for success. + */ +int snd_soc_get_xr_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_mreg_control *mc = + (struct soc_mreg_control *)kcontrol->private_value; + unsigned int regbase = mc->regbase; + unsigned int regcount = mc->regcount; + unsigned int regwshift = component->val_bytes * BITS_PER_BYTE; + unsigned int regwmask = (1<<regwshift)-1; + unsigned int invert = mc->invert; + unsigned long mask = (1UL<<mc->nbits)-1; + long min = mc->min; + long max = mc->max; + long val = 0; + unsigned int regval; + unsigned int i; + int ret; + + for (i = 0; i < regcount; i++) { + ret = snd_soc_component_read(component, regbase+i, &regval); + if (ret) + return ret; + val |= (regval & regwmask) << (regwshift*(regcount-i-1)); + } + val &= mask; + if (min < 0 && val > max) + val |= ~mask; + if (invert) + val = max - val; + ucontrol->value.integer.value[0] = val; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_get_xr_sx); + +/** + * snd_soc_put_xr_sx - signed multi register get callback + * @kcontrol: mreg control + * @ucontrol: control element information + * + * Callback to set the value of a control that can span + * multiple codec registers which together forms a single + * signed value in a MSB/LSB manner. The control supports + * specifying total no of bits used to allow for bitfields + * across the multiple codec registers. + * + * Returns 0 for success. + */ +int snd_soc_put_xr_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_mreg_control *mc = + (struct soc_mreg_control *)kcontrol->private_value; + unsigned int regbase = mc->regbase; + unsigned int regcount = mc->regcount; + unsigned int regwshift = component->val_bytes * BITS_PER_BYTE; + unsigned int regwmask = (1<<regwshift)-1; + unsigned int invert = mc->invert; + unsigned long mask = (1UL<<mc->nbits)-1; + long max = mc->max; + long val = ucontrol->value.integer.value[0]; + unsigned int i, regval, regmask; + int err; + + if (invert) + val = max - val; + val &= mask; + for (i = 0; i < regcount; i++) { + regval = (val >> (regwshift*(regcount-i-1))) & regwmask; + regmask = (mask >> (regwshift*(regcount-i-1))) & regwmask; + err = snd_soc_component_update_bits(component, regbase+i, + regmask, regval); + if (err < 0) + return err; + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_put_xr_sx); + +/** + * snd_soc_get_strobe - strobe get callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback get the value of a strobe mixer control. + * + * Returns 0 for success. + */ +int snd_soc_get_strobe(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + unsigned int mask = 1 << shift; + unsigned int invert = mc->invert != 0; + unsigned int val; + int ret; + + ret = snd_soc_component_read(component, reg, &val); + if (ret) + return ret; + + val &= mask; + + if (shift != 0 && val != 0) + val = val >> shift; + ucontrol->value.enumerated.item[0] = val ^ invert; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_get_strobe); + +/** + * snd_soc_put_strobe - strobe put callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback strobe a register bit to high then low (or the inverse) + * in one pass of a single mixer enum control. + * + * Returns 1 for success. + */ +int snd_soc_put_strobe(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + unsigned int mask = 1 << shift; + unsigned int invert = mc->invert != 0; + unsigned int strobe = ucontrol->value.enumerated.item[0] != 0; + unsigned int val1 = (strobe ^ invert) ? mask : 0; + unsigned int val2 = (strobe ^ invert) ? 0 : mask; + int err; + + err = snd_soc_component_update_bits(component, reg, mask, val1); + if (err < 0) + return err; + + return snd_soc_component_update_bits(component, reg, mask, val2); +} +EXPORT_SYMBOL_GPL(snd_soc_put_strobe); -- 2.1.1

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[alsa-devel] [PATCH] ASoC: soc-pcm: always cancel any pending delayed stream shutdown
by Hector Palacios 28 Oct '14

28 Oct '14

After playback a shutdown is scheduled to happen within 5 seconds. If another playback takes place before the scheduled work times out, the work is cancelled, but if a recording takes place, the work will shutdown in the middle of the recording, causing problems. To reproduce the issue chain the following commands: arecord -M --duration 5 -c 2 /tmp/test.wav; aplay /tmp/test.wav; sleep 2; arecord -M --duration 5 -c 2 /tmp/test.wav; aplay /tmp/test.wav; The fist command records 5 seconds of audio. The second command plays the recorded audio. At the end of the playback a work will be scheduled to shutdown in 5 seconds, but only 2 seconds after the first playback, we run another recording. When we have recorded 3 seconds, the shutdown will occur which can cause incorrect behavior and bad recorded data. As a consequence, the last playback reproduces correct audio for the first 3 seconds and incorrect audio after that. Signed-off-by: Hector Palacios <hector.palacios(a)digi.com> --- sound/soc/soc-pcm.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 642c86240752..d19a926872e8 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -760,8 +760,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) } /* cancel any delayed stream shutdown that is pending */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && - rtd->pop_wait) { + if (rtd->pop_wait) { rtd->pop_wait = 0; cancel_delayed_work(&rtd->delayed_work); }

4 5

[alsa-devel] [PATCH v2 1/7] ASoC: Intel: Make ADSP memory block allocation more generic
by Liam Girdwood 28 Oct '14

28 Oct '14

Current block allocation is tied to block type and requestor type. Make the allocation more generic by removing the struct module parameter and adding a generic block allocator structure. Also pass in the list that the blocks have to be added too in order to remove dependence on block requestor type. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/intel/sst-baytrail-dsp.c | 24 ++- sound/soc/intel/sst-dsp-priv.h | 83 +++++----- sound/soc/intel/sst-firmware.c | 317 +++++++++++++++++++++++++------------ sound/soc/intel/sst-haswell-dsp.c | 41 ++--- 4 files changed, 288 insertions(+), 177 deletions(-) diff --git a/sound/soc/intel/sst-baytrail-dsp.c b/sound/soc/intel/sst-baytrail-dsp.c index fc58876..5a9e567 100644 --- a/sound/soc/intel/sst-baytrail-dsp.c +++ b/sound/soc/intel/sst-baytrail-dsp.c @@ -67,17 +67,12 @@ static int sst_byt_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, { struct dma_block_info *block; struct sst_module *mod; - struct sst_module_data block_data; struct sst_module_template template; int count; memset(&template, 0, sizeof(template)); template.id = module->type; template.entry = module->entry_point; - template.p.type = SST_MEM_DRAM; - template.p.data_type = SST_DATA_P; - template.s.type = SST_MEM_DRAM; - template.s.data_type = SST_DATA_S; mod = sst_module_new(fw, &template, NULL); if (mod == NULL) @@ -94,19 +89,19 @@ static int sst_byt_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, switch (block->type) { case SST_BYT_IRAM: - block_data.offset = block->ram_offset + + mod->offset = block->ram_offset + dsp->addr.iram_offset; - block_data.type = SST_MEM_IRAM; + mod->type = SST_MEM_IRAM; break; case SST_BYT_DRAM: - block_data.offset = block->ram_offset + + mod->offset = block->ram_offset + dsp->addr.dram_offset; - block_data.type = SST_MEM_DRAM; + mod->type = SST_MEM_DRAM; break; case SST_BYT_CACHE: - block_data.offset = block->ram_offset + + mod->offset = block->ram_offset + (dsp->addr.fw_ext - dsp->addr.lpe); - block_data.type = SST_MEM_CACHE; + mod->type = SST_MEM_CACHE; break; default: dev_err(dsp->dev, "wrong ram type 0x%x in block0x%x\n", @@ -114,11 +109,10 @@ static int sst_byt_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, return -EINVAL; } - block_data.size = block->size; - block_data.data_type = SST_DATA_M; - block_data.data = (void *)block + sizeof(*block); + mod->size = block->size; + mod->data = (void *)block + sizeof(*block); - sst_module_insert_fixed_block(mod, &block_data); + sst_module_alloc_blocks(mod); block = (void *)block + sizeof(*block) + block->size; } diff --git a/sound/soc/intel/sst-dsp-priv.h b/sound/soc/intel/sst-dsp-priv.h index ffb308b..f58a049 100644 --- a/sound/soc/intel/sst-dsp-priv.h +++ b/sound/soc/intel/sst-dsp-priv.h @@ -84,15 +84,6 @@ struct sst_mailbox { }; /* - * Audio DSP Firmware data types. - */ -enum sst_data_type { - SST_DATA_M = 0, /* module block data */ - SST_DATA_P = 1, /* peristant data (text, data) */ - SST_DATA_S = 2, /* scratch data (usually buffers) */ -}; - -/* * Audio DSP memory block types. */ enum sst_mem_type { @@ -125,23 +116,6 @@ struct sst_fw { }; /* - * Audio DSP Generic Module data. - * - * This is used to dsecribe any sections of persistent (text and data) and - * scratch (buffers) of module data in ADSP memory space. - */ -struct sst_module_data { - - enum sst_mem_type type; /* destination memory type */ - enum sst_data_type data_type; /* type of module data */ - - u32 size; /* size in bytes */ - int32_t offset; /* offset in FW file */ - u32 data_offset; /* offset in ADSP memory space */ - void *data; /* module data */ -}; - -/* * Audio DSP Generic Module Template. * * Used to define and register a new FW module. This data is extracted from @@ -150,15 +124,42 @@ struct sst_module_data { struct sst_module_template { u32 id; u32 entry; /* entry point */ - struct sst_module_data s; /* scratch data */ - struct sst_module_data p; /* peristant data */ + u32 scratch_size; + u32 persistent_size; +}; + +/* + * Block Allocator - Used to allocate blocks of DSP memory. + */ +struct sst_block_allocator { + u32 id; + u32 offset; + int size; + enum sst_mem_type type; +}; + +/* + * Runtime Module Instance - A module object can be instanciated multiple + * times within the DSP FW. + */ +struct sst_module_runtime { + struct sst_dsp *dsp; + int id; + struct sst_module *module; /* parent module we belong too */ + + u32 persistent_offset; /* private memory size */ + void *private; + + struct list_head list; + struct list_head block_list; /* list of blocks used */ }; /* * Audio DSP Generic Module. * * Each Firmware file can consist of 1..N modules. A module can span multiple - * ADSP memory blocks. The simplest FW will be a file with 1 module. + * ADSP memory blocks. The simplest FW will be a file with 1 module. A module + * can be instanciated multiple times in the DSP. */ struct sst_module { struct sst_dsp *dsp; @@ -167,10 +168,13 @@ struct sst_module { /* module configuration */ u32 id; u32 entry; /* module entry point */ - u32 offset; /* module offset in firmware file */ + s32 offset; /* module offset in firmware file */ u32 size; /* module size */ - struct sst_module_data s; /* scratch data */ - struct sst_module_data p; /* peristant data */ + u32 scratch_size; /* global scratch memory required */ + u32 persistent_size; /* private memory required */ + enum sst_mem_type type; /* destination memory type */ + u32 data_offset; /* offset in ADSP memory space */ + void *data; /* module data */ /* runtime */ u32 usage_count; /* can be unloaded if count == 0 */ @@ -180,6 +184,7 @@ struct sst_module { struct list_head block_list; /* Module list of blocks in use */ struct list_head list; /* DSP list of modules */ struct list_head list_fw; /* FW list of modules */ + struct list_head runtime_list; /* list of runtime module objects*/ }; /* @@ -208,7 +213,6 @@ struct sst_mem_block { struct sst_block_ops *ops; /* block operations, if any */ /* block status */ - enum sst_data_type data_type; /* data type held in this block */ u32 bytes_used; /* bytes in use by modules */ void *private; /* generic core does not touch this */ int users; /* number of modules using this block */ @@ -290,19 +294,20 @@ void sst_fw_unload(struct sst_fw *sst_fw); /* Create/Free firmware modules */ struct sst_module *sst_module_new(struct sst_fw *sst_fw, struct sst_module_template *template, void *private); -void sst_module_free(struct sst_module *sst_module); -int sst_module_insert(struct sst_module *sst_module); -int sst_module_remove(struct sst_module *sst_module); -int sst_module_insert_fixed_block(struct sst_module *module, - struct sst_module_data *data); +void sst_module_free(struct sst_module *module); struct sst_module *sst_module_get_from_id(struct sst_dsp *dsp, u32 id); /* allocate/free pesistent/scratch memory regions managed by drv */ struct sst_module *sst_mem_block_alloc_scratch(struct sst_dsp *dsp); void sst_mem_block_free_scratch(struct sst_dsp *dsp, struct sst_module *scratch); -int sst_block_module_remove(struct sst_module *module); +int sst_module_alloc_blocks(struct sst_module *module); +int sst_module_free_blocks(struct sst_module *module); +/* generic block allocation */ +int sst_alloc_blocks(struct sst_dsp *dsp, struct sst_block_allocator *ba, + struct list_head *block_list); +int sst_free_blocks(struct sst_dsp *dsp, struct list_head *block_list); /* Register the DSPs memory blocks - would be nice to read from ACPI */ struct sst_mem_block *sst_mem_block_register(struct sst_dsp *dsp, u32 offset, u32 size, enum sst_mem_type type, struct sst_block_ops *ops, u32 index, diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index cf3d199..ca90278 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -30,7 +30,7 @@ #include "sst-dsp.h" #include "sst-dsp-priv.h" -static void block_module_remove(struct sst_module *module); +#define SST_HSW_BLOCK_ANY 0xffffffff static inline void sst_memcpy32(volatile void __iomem *dest, void *src, u32 bytes) { @@ -38,6 +38,65 @@ static inline void sst_memcpy32(volatile void __iomem *dest, void *src, u32 byte __iowrite32_copy((void *)dest, src, bytes/4); } + +/* remove module from memory - callers hold locks */ +static void block_list_remove(struct sst_dsp *dsp, + struct list_head *block_list) +{ + struct sst_mem_block *block, *tmp; + int err; + + /* disable each block */ + list_for_each_entry(block, block_list, module_list) { + + if (block->ops && block->ops->disable) { + err = block->ops->disable(block); + if (err < 0) + dev_err(dsp->dev, + "error: cant disable block %d:%d\n", + block->type, block->index); + } + } + + /* mark each block as free */ + list_for_each_entry_safe(block, tmp, block_list, module_list) { + list_del(&block->module_list); + list_move(&block->list, &dsp->free_block_list); + dev_dbg(dsp->dev, "block freed %d:%d at offset 0x%x\n", + block->type, block->index, block->offset); + } +} + +/* prepare the memory block to receive data from host - callers hold locks */ +static int block_list_prepare(struct sst_dsp *dsp, + struct list_head *block_list) +{ + struct sst_mem_block *block; + int ret = 0; + + /* enable each block so that's it'e ready for data */ + list_for_each_entry(block, block_list, module_list) { + + if (block->ops && block->ops->enable) { + ret = block->ops->enable(block); + if (ret < 0) { + dev_err(dsp->dev, + "error: cant disable block %d:%d\n", + block->type, block->index); + goto err; + } + } + } + return ret; + +err: + list_for_each_entry(block, block_list, module_list) { + if (block->ops && block->ops->disable) + block->ops->disable(block); + } + return ret; +} + /* create new generic firmware object */ struct sst_fw *sst_fw_new(struct sst_dsp *dsp, const struct firmware *fw, void *private) @@ -199,73 +258,122 @@ void sst_module_free(struct sst_module *sst_module) } EXPORT_SYMBOL_GPL(sst_module_free); -static struct sst_mem_block *find_block(struct sst_dsp *dsp, int type, - u32 offset) +struct sst_module_runtime *sst_module_runtime_new(struct sst_module *module, + int id, void *private) +{ + struct sst_dsp *dsp = module->dsp; + struct sst_module_runtime *runtime; + + runtime = kzalloc(sizeof(*runtime), GFP_KERNEL); + if (runtime == NULL) + return NULL; + + runtime->id = id; + runtime->dsp = dsp; + runtime->module = module; + INIT_LIST_HEAD(&runtime->block_list); + + mutex_lock(&dsp->mutex); + list_add(&runtime->list, &module->runtime_list); + mutex_unlock(&dsp->mutex); + + return runtime; +} +EXPORT_SYMBOL_GPL(sst_module_runtime_new); + +void sst_module_runtime_free(struct sst_module_runtime *runtime) +{ + struct sst_dsp *dsp = runtime->dsp; + + mutex_lock(&dsp->mutex); + list_del(&runtime->list); + mutex_unlock(&dsp->mutex); + + kfree(runtime); +} +EXPORT_SYMBOL_GPL(sst_module_runtime_free); + +static struct sst_mem_block *find_block(struct sst_dsp *dsp, + struct sst_block_allocator *ba) { struct sst_mem_block *block; list_for_each_entry(block, &dsp->free_block_list, list) { - if (block->type == type && block->offset == offset) + if (block->type == ba->type && block->offset == ba->offset) return block; } return NULL; } -static int block_alloc_contiguous(struct sst_module *module, - struct sst_module_data *data, u32 offset, int size) +/* Block allocator must be on block boundary */ +static int block_alloc_contiguous(struct sst_dsp *dsp, + struct sst_block_allocator *ba, struct list_head *block_list) { struct list_head tmp = LIST_HEAD_INIT(tmp); - struct sst_dsp *dsp = module->dsp; struct sst_mem_block *block; + u32 block_start = SST_HSW_BLOCK_ANY; + int size = ba->size, offset = ba->offset; + + while (ba->size > 0) { - while (size > 0) { - block = find_block(dsp, data->type, offset); + block = find_block(dsp, ba); if (!block) { list_splice(&tmp, &dsp->free_block_list); + + ba->size = size; + ba->offset = offset; return -ENOMEM; } list_move_tail(&block->list, &tmp); - offset += block->size; - size -= block->size; + ba->offset += block->size; + ba->size -= block->size; } + ba->size = size; + ba->offset = offset; + + list_for_each_entry(block, &tmp, list) { + + if (block->offset < block_start) + block_start = block->offset; - list_for_each_entry(block, &tmp, list) - list_add(&block->module_list, &module->block_list); + list_add(&block->module_list, block_list); + + dev_dbg(dsp->dev, "block allocated %d:%d at offset 0x%x\n", + block->type, block->index, block->offset); + } list_splice(&tmp, &dsp->used_block_list); return 0; } -/* allocate free DSP blocks for module data - callers hold locks */ -static int block_alloc(struct sst_module *module, - struct sst_module_data *data) +/* allocate first free DSP blocks for data - callers hold locks */ +static int block_alloc(struct sst_dsp *dsp, struct sst_block_allocator *ba, + struct list_head *block_list) { - struct sst_dsp *dsp = module->dsp; struct sst_mem_block *block, *tmp; int ret = 0; - if (data->size == 0) + if (ba->size == 0) return 0; /* find first free whole blocks that can hold module */ list_for_each_entry_safe(block, tmp, &dsp->free_block_list, list) { /* ignore blocks with wrong type */ - if (block->type != data->type) + if (block->type != ba->type) continue; - if (data->size > block->size) + if (ba->size > block->size) continue; - data->offset = block->offset; - block->data_type = data->data_type; - block->bytes_used = data->size % block->size; - list_add(&block->module_list, &module->block_list); + ba->offset = block->offset; + block->bytes_used = ba->size % block->size; + list_add(&block->module_list, block_list); list_move(&block->list, &dsp->used_block_list); - dev_dbg(dsp->dev, " *module %d added block %d:%d\n", - module->id, block->type, block->index); + dev_dbg(dsp->dev, "block allocated %d:%d at offset 0x%x\n", + block->type, block->index, block->offset); return 0; } @@ -273,15 +381,19 @@ static int block_alloc(struct sst_module *module, list_for_each_entry_safe(block, tmp, &dsp->free_block_list, list) { /* ignore blocks with wrong type */ - if (block->type != data->type) + if (block->type != ba->type) continue; /* do we span > 1 blocks */ - if (data->size > block->size) { - ret = block_alloc_contiguous(module, data, - block->offset, data->size); + if (ba->size > block->size) { + + /* align ba to block boundary */ + ba->offset = block->offset; + + ret = block_alloc_contiguous(dsp, ba, block_list); if (ret == 0) return ret; + } } @@ -289,93 +401,74 @@ static int block_alloc(struct sst_module *module, return -ENOMEM; } -/* remove module from memory - callers hold locks */ -static void block_module_remove(struct sst_module *module) +int sst_alloc_blocks(struct sst_dsp *dsp, struct sst_block_allocator *ba, + struct list_head *block_list) { - struct sst_mem_block *block, *tmp; - struct sst_dsp *dsp = module->dsp; - int err; + int ret; - /* disable each block */ - list_for_each_entry(block, &module->block_list, module_list) { + dev_dbg(dsp->dev, "block request 0x%x bytes at offset 0x%x type %d\n", + ba->size, ba->offset, ba->type); - if (block->ops && block->ops->disable) { - err = block->ops->disable(block); - if (err < 0) - dev_err(dsp->dev, - "error: cant disable block %d:%d\n", - block->type, block->index); - } - } + mutex_lock(&dsp->mutex); - /* mark each block as free */ - list_for_each_entry_safe(block, tmp, &module->block_list, module_list) { - list_del(&block->module_list); - list_move(&block->list, &dsp->free_block_list); + ret = block_alloc(dsp, ba, block_list); + if (ret < 0) { + dev_err(dsp->dev, "error: can't alloc blocks %d\n", ret); + goto out; } -} - -/* prepare the memory block to receive data from host - callers hold locks */ -static int block_module_prepare(struct sst_module *module) -{ - struct sst_mem_block *block; - int ret = 0; - /* enable each block so that's it'e ready for module P/S data */ - list_for_each_entry(block, &module->block_list, module_list) { + /* prepare DSP blocks for module usage */ + ret = block_list_prepare(dsp, block_list); + if (ret < 0) + dev_err(dsp->dev, "error: prepare failed\n"); - if (block->ops && block->ops->enable) { - ret = block->ops->enable(block); - if (ret < 0) { - dev_err(module->dsp->dev, - "error: cant disable block %d:%d\n", - block->type, block->index); - goto err; - } - } - } +out: + mutex_unlock(&dsp->mutex); return ret; +} +EXPORT_SYMBOL_GPL(sst_alloc_blocks); -err: - list_for_each_entry(block, &module->block_list, module_list) { - if (block->ops && block->ops->disable) - block->ops->disable(block); - } - return ret; +int sst_free_blocks(struct sst_dsp *dsp, struct list_head *block_list) +{ + mutex_lock(&dsp->mutex); + block_list_remove(dsp, block_list); + mutex_unlock(&dsp->mutex); + return 0; } +EXPORT_SYMBOL_GPL(sst_free_blocks); /* allocate memory blocks for static module addresses - callers hold locks */ -static int block_alloc_fixed(struct sst_module *module, - struct sst_module_data *data) +static int block_alloc_fixed(struct sst_dsp *dsp, struct sst_block_allocator *ba, + struct list_head *block_list) { - struct sst_dsp *dsp = module->dsp; struct sst_mem_block *block, *tmp; - u32 end = data->offset + data->size, block_end; + u32 end = ba->offset + ba->size, block_end; int err; /* only IRAM/DRAM blocks are managed */ - if (data->type != SST_MEM_IRAM && data->type != SST_MEM_DRAM) + if (ba->type != SST_MEM_IRAM && ba->type != SST_MEM_DRAM) return 0; /* are blocks already attached to this module */ - list_for_each_entry_safe(block, tmp, &module->block_list, module_list) { + list_for_each_entry_safe(block, tmp, block_list, module_list) { - /* force compacting mem blocks of the same data_type */ - if (block->data_type != data->data_type) + /* ignore blocks with wrong type */ + if (block->type != ba->type) continue; block_end = block->offset + block->size; /* find block that holds section */ - if (data->offset >= block->offset && end < block_end) + if (ba->offset >= block->offset && end <= block_end) return 0; /* does block span more than 1 section */ - if (data->offset >= block->offset && data->offset < block_end) { + if (ba->offset >= block->offset && ba->offset < block_end) { - err = block_alloc_contiguous(module, data, - block->offset + block->size, - data->size - block->size); + /* align ba to block boundary */ + ba->size -= block_end - ba->offset; + ba->offset = block_end; + err = block_alloc_contiguous(dsp, ba, block_list); if (err < 0) return -ENOMEM; @@ -388,53 +481,67 @@ static int block_alloc_fixed(struct sst_module *module, list_for_each_entry_safe(block, tmp, &dsp->free_block_list, list) { block_end = block->offset + block->size; + /* ignore blocks with wrong type */ + if (block->type != ba->type) + continue; + /* find block that holds section */ - if (data->offset >= block->offset && end < block_end) { + if (ba->offset >= block->offset && end <= block_end) { /* add block */ - block->data_type = data->data_type; list_move(&block->list, &dsp->used_block_list); - list_add(&block->module_list, &module->block_list); + list_add(&block->module_list, block_list); + dev_dbg(dsp->dev, "block allocated %d:%d at offset 0x%x\n", + block->type, block->index, block->offset); return 0; } /* does block span more than 1 section */ - if (data->offset >= block->offset && data->offset < block_end) { + if (ba->offset >= block->offset && ba->offset < block_end) { + + /* align ba to block boundary */ + ba->offset = block->offset; - err = block_alloc_contiguous(module, data, - block->offset, data->size); + err = block_alloc_contiguous(dsp, ba, block_list); if (err < 0) return -ENOMEM; return 0; } - } return -ENOMEM; } /* Load fixed module data into DSP memory blocks */ -int sst_module_insert_fixed_block(struct sst_module *module, - struct sst_module_data *data) +int sst_module_alloc_blocks(struct sst_module *module) { struct sst_dsp *dsp = module->dsp; + struct sst_fw *sst_fw = module->sst_fw; + struct sst_block_allocator ba; int ret; + ba.size = module->size; + ba.type = module->type; + ba.offset = module->offset; + + dev_dbg(dsp->dev, "block request 0x%x bytes at offset 0x%x type %d\n", + ba.size, ba.offset, ba.type); + mutex_lock(&dsp->mutex); /* alloc blocks that includes this section */ - ret = block_alloc_fixed(module, data); + ret = block_alloc_fixed(dsp, &ba, &module->block_list); if (ret < 0) { dev_err(dsp->dev, "error: no free blocks for section at offset 0x%x size 0x%x\n", - data->offset, data->size); + module->offset, module->size); mutex_unlock(&dsp->mutex); return -ENOMEM; } /* prepare DSP blocks for module copy */ - ret = block_module_prepare(module); + ret = block_list_prepare(dsp, &module->block_list); if (ret < 0) { dev_err(dsp->dev, "error: fw module prepare failed\n"); goto err; @@ -447,19 +554,23 @@ int sst_module_insert_fixed_block(struct sst_module *module, return ret; err: - block_module_remove(module); + block_list_remove(dsp, &module->block_list); mutex_unlock(&dsp->mutex); return ret; } -EXPORT_SYMBOL_GPL(sst_module_insert_fixed_block); +EXPORT_SYMBOL_GPL(sst_module_alloc_blocks); /* Unload entire module from DSP memory */ -int sst_block_module_remove(struct sst_module *module) +int sst_module_free_blocks(struct sst_module *module) { struct sst_dsp *dsp = module->dsp; mutex_lock(&dsp->mutex); - block_module_remove(module); + block_list_remove(dsp, &module->block_list); + mutex_unlock(&dsp->mutex); + return 0; +} +EXPORT_SYMBOL_GPL(sst_module_free_blocks); mutex_unlock(&dsp->mutex); return 0; } diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c index 4b6c163..22cc697 100644 --- a/sound/soc/intel/sst-haswell-dsp.c +++ b/sound/soc/intel/sst-haswell-dsp.c @@ -86,9 +86,8 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, { struct dma_block_info *block; struct sst_module *mod; - struct sst_module_data block_data; struct sst_module_template template; - int count; + int count, ret; void __iomem *ram; /* TODO: allowed module types need to be configurable */ @@ -109,13 +108,9 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, memset(&template, 0, sizeof(template)); template.id = module->type; - template.entry = module->entry_point; - template.p.size = module->info.persistent_size; - template.p.type = SST_MEM_DRAM; - template.p.data_type = SST_DATA_P; - template.s.size = module->info.scratch_size; - template.s.type = SST_MEM_DRAM; - template.s.data_type = SST_DATA_S; + template.entry = module->entry_point - 4; + template.persistent_size = module->info.persistent_size; + template.scratch_size = module->info.scratch_size; mod = sst_module_new(fw, &template, NULL); if (mod == NULL) @@ -135,14 +130,14 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, switch (block->type) { case SST_HSW_IRAM: ram = dsp->addr.lpe; - block_data.offset = + mod->offset = block->ram_offset + dsp->addr.iram_offset; - block_data.type = SST_MEM_IRAM; + mod->type = SST_MEM_IRAM; break; case SST_HSW_DRAM: ram = dsp->addr.lpe; - block_data.offset = block->ram_offset; - block_data.type = SST_MEM_DRAM; + mod->offset = block->ram_offset; + mod->type = SST_MEM_DRAM; break; default: dev_err(dsp->dev, "error: bad type 0x%x for block 0x%x\n", @@ -151,20 +146,26 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, return -EINVAL; } - block_data.size = block->size; - block_data.data_type = SST_DATA_M; - block_data.data = (void *)block + sizeof(*block); - block_data.data_offset = block_data.data - fw->dma_buf; + mod->size = block->size; + mod->data = (void *)block + sizeof(*block); + mod->data_offset = mod->data - fw->dma_buf; - dev_dbg(dsp->dev, "copy firmware block %d type 0x%x " + dev_dbg(dsp->dev, "module block %d type 0x%x " "size 0x%x ==> ram %p offset 0x%x\n", - count, block->type, block->size, ram, + count, mod->type, block->size, ram, block->ram_offset); - sst_module_insert_fixed_block(mod, &block_data); + ret = sst_module_alloc_blocks(mod); + if (ret < 0) { + dev_err(dsp->dev, "error: could not allocate blocks for module %d\n", + count); + sst_module_free(mod); + return ret; + } block = (void *)block + sizeof(*block) + block->size; } + return 0; } -- 1.9.1

1 6

[alsa-devel] [PATCH] ALSA: compress: fix documentation errors
by Vinod Koul 28 Oct '14

28 Oct '14

Some structure documentation was not right so fix it now Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- include/sound/compress_driver.h | 5 +++-- 1 files changed, 3 insertions(+), 2 deletions(-) diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index ae6c3b8..396e8f7 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -42,12 +42,11 @@ struct snd_compr_ops; * @buffer_size: size of the above buffer * @fragment_size: size of buffer fragment in bytes * @fragments: number of such fragments - * @hw_pointer: offset of last location in buffer where DSP copied data - * @app_pointer: offset of last location in buffer where app wrote data * @total_bytes_available: cumulative number of bytes made available in * the ring buffer * @total_bytes_transferred: cumulative bytes transferred by offload DSP * @sleep: poll sleep + * @private_data: driver private data pointer */ struct snd_compr_runtime { snd_pcm_state_t state; @@ -94,6 +93,8 @@ struct snd_compr_stream { * This can be called in during stream creation only to set codec params * and the stream properties * @get_params: retrieve the codec parameters, mandatory + * @set_metadata: Set the metadata values for a stream + * @get_metadata: retreives the requested metadata values from stream * @trigger: Trigger operations like start, pause, resume, drain, stop. * This callback is mandatory * @pointer: Retrieve current h/w pointer information. Mandatory -- 1.7.0.4

2 1

[alsa-devel] [PATCH] ASoC: fsl_asrc: Add reg_defaults for regmap to fix kernel dump
by Nicolin Chen 28 Oct '14

28 Oct '14

Kernel dump (WARN_ON) ocurred during system boot-up inside regmap_write(): ------------[ cut here ]------------ WARNING: CPU: 0 PID: 47 at kernel/locking/lockdep.c:2744 lockdep_trace_alloc+0xe8/0x108() DEBUG_LOCKS_WARN_ON(irqs_disabled_flags(flags)) Modules linked in: CPU: 0 PID: 47 Comm: kworker/u2:2 Not tainted 3.18.0-rc1-10245-gb75d289-dirty #56 Workqueue: deferwq deferred_probe_work_func Backtrace: [<80012294>] (dump_backtrace) from [<80012578>] (show_stack+0x18/0x1c) r6:8097c73c r5:8097c73c r4:00000000 r3:be33ba80 [<80012560>] (show_stack) from [<806aac48>] (dump_stack+0x8c/0xa4) [<806aabbc>] (dump_stack) from [<8002a694>] (warn_slowpath_common+0x70/0x94) r6:80062838 r5:00000009 r4:bd827b30 r3:be33ba80 [<8002a624>] (warn_slowpath_common) from [<8002a6f0>] (warn_slowpath_fmt+0x38/0x40) r8:00000004 r7:00000001 r6:000080d0 r5:60000193 r4:bd826010 [<8002a6bc>] (warn_slowpath_fmt) from [<80062838>] (lockdep_trace_alloc+0xe8/0x108) r3:80831590 r2:8082e160 [<80062750>] (lockdep_trace_alloc) from [<800ea5dc>] (kmem_cache_alloc+0x28/0x134) r5:000080d0 r4:be001f00 [<800ea5b4>] (kmem_cache_alloc) from [<8038d72c>] (regcache_rbtree_write+0x15c/0x648) r10:00000000 r9:0000001c r8:00000004 r7:00000001 r6:00000000 r5:bd819a00 r4:00000000 r3:811aea88 [<8038d5d0>] (regcache_rbtree_write) from [<8038c4d8>] (regcache_write+0x5c/0x64) r10:be3f9f88 r9:00000000 r8:00000004 r7:00000001 r6:00000000 r5:00000001 r4:bd819a00 [<8038c47c>] (regcache_write) from [<8038b0dc>] (_regmap_raw_write+0x134/0x5f4) r6:be3f9f84 r5:00000001 r4:bd819a00 r3:00000001 [<8038afa8>] (_regmap_raw_write) from [<8038b610>] (_regmap_bus_raw_write+0x74/0x94) r10:00000000 r9:00000001 r8:be3fb080 r7:bd819a00 r6:00000001 r5:00000000 r4:bd819a00 [<8038b59c>] (_regmap_bus_raw_write) from [<8038a8b4>] (_regmap_write+0x60/0x9c) r6:00000001 r5:00000000 r4:bd819a00 r3:8038b59c [<8038a854>] (_regmap_write) from [<8038ba24>] (regmap_write+0x48/0x68) r7:bd81ad80 r6:00000001 r5:00000000 r4:bd819a00 [<8038b9dc>] (regmap_write) from [<80528f30>] (fsl_asrc_dai_probe+0x34/0x104) r6:bd888628 r5:be3fb080 r4:be3b4410 r3:be3b442c ------------[ dump end ]------------ ============================================================================= 2741 /* 2742 * Oi! Can't be having __GFP_FS allocations with IRQs disabled. 2743 */ 2744 if (DEBUG_LOCKS_WARN_ON(irqs_disabled_flags(flags))) 2745 return; ============================================================================= By looking at 2744 line, we can get that it's because regcache_rbtree_write() would call kmalloc() with GFP flag if it couldn't find an existing block to insert nodes while this kmalloc() call is inside a spin_lock_irq_save pair, i.e. IRQs disabled. Even though this may be a bug that should be fixed, I still try to send this patch as a quick fix (work around) since it does no harm to assign default values of every registers when using regcache. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- sound/soc/fsl/fsl_asrc.c | 26 ++++++++++++++++++++++++++ 1 file changed, 26 insertions(+) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index f2d8652..817d304 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -684,12 +684,38 @@ static bool fsl_asrc_writeable_reg(struct device *dev, unsigned int reg) } } +static struct reg_default fsl_asrc_reg[] = { + { REG_ASRCTR, 0x0000 }, { REG_ASRIER, 0x0000 }, + { REG_ASRCNCR, 0x0000 }, { REG_ASRCFG, 0x0000 }, + { REG_ASRCSR, 0x0000 }, { REG_ASRCDR1, 0x0000 }, + { REG_ASRCDR2, 0x0000 }, { REG_ASRSTR, 0x0000 }, + { REG_ASRRA, 0x0000 }, { REG_ASRRB, 0x0000 }, + { REG_ASRRC, 0x0000 }, { REG_ASRPM1, 0x0000 }, + { REG_ASRPM2, 0x0000 }, { REG_ASRPM3, 0x0000 }, + { REG_ASRPM4, 0x0000 }, { REG_ASRPM5, 0x0000 }, + { REG_ASRTFR1, 0x0000 }, { REG_ASRCCR, 0x0000 }, + { REG_ASRDIA, 0x0000 }, { REG_ASRDOA, 0x0000 }, + { REG_ASRDIB, 0x0000 }, { REG_ASRDOB, 0x0000 }, + { REG_ASRDIC, 0x0000 }, { REG_ASRDOC, 0x0000 }, + { REG_ASRIDRHA, 0x0000 }, { REG_ASRIDRLA, 0x0000 }, + { REG_ASRIDRHB, 0x0000 }, { REG_ASRIDRLB, 0x0000 }, + { REG_ASRIDRHC, 0x0000 }, { REG_ASRIDRLC, 0x0000 }, + { REG_ASR76K, 0x0A47 }, { REG_ASR56K, 0x0DF3 }, + { REG_ASRMCRA, 0x0000 }, { REG_ASRFSTA, 0x0000 }, + { REG_ASRMCRB, 0x0000 }, { REG_ASRFSTB, 0x0000 }, + { REG_ASRMCRC, 0x0000 }, { REG_ASRFSTC, 0x0000 }, + { REG_ASRMCR1A, 0x0000 }, { REG_ASRMCR1B, 0x0000 }, + { REG_ASRMCR1C, 0x0000 }, +}; + static const struct regmap_config fsl_asrc_regmap_config = { .reg_bits = 32, .reg_stride = 4, .val_bits = 32, .max_register = REG_ASRMCR1C, + .reg_defaults = fsl_asrc_reg, + .num_reg_defaults = ARRAY_SIZE(fsl_asrc_reg), .readable_reg = fsl_asrc_readable_reg, .volatile_reg = fsl_asrc_volatile_reg, .writeable_reg = fsl_asrc_writeable_reg, -- 1.9.1

2 4

[alsa-devel] [PATCH][resend] ASoC: simple-card / rsnd resend
by Kuninori Morimoto 28 Oct '14

28 Oct '14

Hi Mark These are resend patches which are still not applied, and, no response. The original patch-set had sent in... Subject: [PATCH 0/3] ASoC: simple-card: cleanup simple-card daifmt Date: Tue, 09 Sep 2014 21:36:43 -0700 (PDT) Subject: [PATCH 0/8] ASoC: rsnd: add Volume Ramp support Date: Tue, 21 Oct 2014 18:13:17 -0700 (PDT) Best regards --- Kuninori Morimoto

2 8

[alsa-devel] [PATCH v2] sound: pci: pcxhr: convert timeval to ktime_t
by Aya Mahfouz 28 Oct '14

28 Oct '14

This patch is concerned with migrating the time variables in the pcxhr module found in the sound driver. The changes are concerend with the y2038 problem where timeval will overflow in the year 2038. ktime_t was used instead of timeval to get the wall time. The difference is displayed now in nanoseconds instead of microseconds. Signed-off-by: Aya Mahfouz <mahfouz.saif.elyazal(a)gmail.com> Reviewed-by: Arnd Bergmann <arnd(a)arndb.de> --- v2: changed variables names to avoid redeclaration conflicts. sound/pci/pcxhr/pcxhr.c | 10 ++++++---- sound/pci/pcxhr/pcxhr_core.c | 10 ++++++---- 2 files changed, 12 insertions(+), 8 deletions(-) diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index b854fc5..7c33c97 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -711,8 +711,9 @@ static void pcxhr_start_linked_stream(struct pcxhr_mgr *mgr) int playback_mask = 0; #ifdef CONFIG_SND_DEBUG_VERBOSE - struct timeval my_tv1, my_tv2; - do_gettimeofday(&my_tv1); + ktime_t start_time, stop_time, diff_time; + + start_time = ktime_get(); #endif mutex_lock(&mgr->setup_mutex); @@ -823,9 +824,10 @@ static void pcxhr_start_linked_stream(struct pcxhr_mgr *mgr) mutex_unlock(&mgr->setup_mutex); #ifdef CONFIG_SND_DEBUG_VERBOSE - do_gettimeofday(&my_tv2); + stop_time = ktime_get(); + diff_time = ktime_sub(stop_time, start_time); dev_dbg(&mgr->pci->dev, "***TRIGGER START*** TIME = %ld (err = %x)\n", - (long)(my_tv2.tv_usec - my_tv1.tv_usec), err); + (long)(ktime_to_ns(diff_time)), err); #endif } diff --git a/sound/pci/pcxhr/pcxhr_core.c b/sound/pci/pcxhr/pcxhr_core.c index a584acb..181f772 100644 --- a/sound/pci/pcxhr/pcxhr_core.c +++ b/sound/pci/pcxhr/pcxhr_core.c @@ -910,8 +910,9 @@ int pcxhr_set_pipe_state(struct pcxhr_mgr *mgr, int playback_mask, int audio_mask; #ifdef CONFIG_SND_DEBUG_VERBOSE - struct timeval my_tv1, my_tv2; - do_gettimeofday(&my_tv1); + ktime_t start_time, stop_time, diff_time; + + start_time = ktime_get(); #endif audio_mask = (playback_mask | (capture_mask << PCXHR_PIPE_STATE_CAPTURE_OFFSET)); @@ -960,9 +961,10 @@ int pcxhr_set_pipe_state(struct pcxhr_mgr *mgr, int playback_mask, return err; } #ifdef CONFIG_SND_DEBUG_VERBOSE - do_gettimeofday(&my_tv2); + stop_time = ktime_get(); + diff_time = ktime_sub(stop_time, start_time); dev_dbg(&mgr->pci->dev, "***SET PIPE STATE*** TIME = %ld (err = %x)\n", - (long)(my_tv2.tv_usec - my_tv1.tv_usec), err); + (long)(ktime_to_ns(diff_time)), err); #endif return 0; } -- 1.9.3

2 1

[alsa-devel] [PATCH] sound: pci: pcxhr: convert timeval to ktime_t
by Aya Mahfouz 28 Oct '14

28 Oct '14

This patch is concerned with migrating the time variables in the pcxhr module found in the sound driver. The changes are concerend with the y2038 problem where timeval will overflow in the year 2038. ktime_t was used instead of timeval to get the wall time. The difference is displayed now in nanoseconds instead of microseconds. Signed-off-by: Aya Mahfouz <mahfouz.saif.elyazal(a)gmail.com> Reviewed-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/pci/pcxhr/pcxhr.c | 10 ++++++---- sound/pci/pcxhr/pcxhr_core.c | 10 ++++++---- 2 files changed, 12 insertions(+), 8 deletions(-) diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index b854fc5..3f4607c 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -711,8 +711,9 @@ static void pcxhr_start_linked_stream(struct pcxhr_mgr *mgr) int playback_mask = 0; #ifdef CONFIG_SND_DEBUG_VERBOSE - struct timeval my_tv1, my_tv2; - do_gettimeofday(&my_tv1); + ktime_t start, stop, diff; + + start = ktime_get(); #endif mutex_lock(&mgr->setup_mutex); @@ -823,9 +824,10 @@ static void pcxhr_start_linked_stream(struct pcxhr_mgr *mgr) mutex_unlock(&mgr->setup_mutex); #ifdef CONFIG_SND_DEBUG_VERBOSE - do_gettimeofday(&my_tv2); + stop = ktime_get(); + diff = ktime_sub(stop, start); dev_dbg(&mgr->pci->dev, "***TRIGGER START*** TIME = %ld (err = %x)\n", - (long)(my_tv2.tv_usec - my_tv1.tv_usec), err); + (long)(ktime_to_ns(diff)), err); #endif } diff --git a/sound/pci/pcxhr/pcxhr_core.c b/sound/pci/pcxhr/pcxhr_core.c index a584acb..be3c003 100644 --- a/sound/pci/pcxhr/pcxhr_core.c +++ b/sound/pci/pcxhr/pcxhr_core.c @@ -910,8 +910,9 @@ int pcxhr_set_pipe_state(struct pcxhr_mgr *mgr, int playback_mask, int audio_mask; #ifdef CONFIG_SND_DEBUG_VERBOSE - struct timeval my_tv1, my_tv2; - do_gettimeofday(&my_tv1); + ktime_t start, stop, diff; + + start = ktime_get(); #endif audio_mask = (playback_mask | (capture_mask << PCXHR_PIPE_STATE_CAPTURE_OFFSET)); @@ -960,9 +961,10 @@ int pcxhr_set_pipe_state(struct pcxhr_mgr *mgr, int playback_mask, return err; } #ifdef CONFIG_SND_DEBUG_VERBOSE - do_gettimeofday(&my_tv2); + stop = ktime_get(); + diff = ktime_sub(stop, start); dev_dbg(&mgr->pci->dev, "***SET PIPE STATE*** TIME = %ld (err = %x)\n", - (long)(my_tv2.tv_usec - my_tv1.tv_usec), err); + (long)(ktime_to_ns(diff)), err); #endif return 0; } -- 1.9.3

2 3

[alsa-devel] [PATCH] ALSA: pcm: Zero-clear reserved fields of PCM status ioctl in compat mode
by Takashi Iwai 28 Oct '14

28 Oct '14

In compat mode, we copy each field of snd_pcm_status struct but don't touch the reserved fields, and this leaves uninitialized values there. Meanwhile the native ioctl does zero-clear the whole structure, so we should follow the same rule in compat mode, too. Reported-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_compat.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c index 102e8fd1d450..2d957ba63557 100644 --- a/sound/core/pcm_compat.c +++ b/sound/core/pcm_compat.c @@ -210,6 +210,8 @@ static int snd_pcm_status_user_compat(struct snd_pcm_substream *substream, if (err < 0) return err; + if (clear_user(src, sizeof(*src))) + return -EFAULT; if (put_user(status.state, &src->state) || compat_put_timespec(&status.trigger_tstamp, &src->trigger_tstamp) || compat_put_timespec(&status.tstamp, &src->tstamp) || -- 2.1.2

1 0

[alsa-devel] [PATCH] ASoC: es8328-i2c: Fix i2c_device_id name field in es8328_id
by Axel Lin 28 Oct '14

28 Oct '14

The convention for i2c_device_id name does not need to have company prefix. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/es8328-i2c.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8328-i2c.c b/sound/soc/codecs/es8328-i2c.c index aae410d..2d05b5d 100644 --- a/sound/soc/codecs/es8328-i2c.c +++ b/sound/soc/codecs/es8328-i2c.c @@ -19,7 +19,7 @@ #include "es8328.h" static const struct i2c_device_id es8328_id[] = { - { "everest,es8328", 0 }, + { "es8328", 0 }, { } }; MODULE_DEVICE_TABLE(i2c, es8328_id); -- 1.9.1

2 1

[alsa-devel] What are subdevices?
by dE 28 Oct '14

28 Oct '14

Hi! I tried this question in the alsa-users mailing list but it doesn't appear to be much active. So I ask here. I'm facing problems with my system, so decided to learn ALSA to see what's the problem. My first question -- what's a device and a sub device? A device seems like the output streams the sound card supports. Like HDMI, DP, analog etc... So what's a subdevice? Is this related to multichannel? Does it designate physical ports on the card? Thanks.

2 5

[alsa-devel] [PATCH V3 1/2] ARM: pxa: spitz: register spitz-audio device
by Dmitry Eremin-Solenikov 28 Oct '14

28 Oct '14

Register spitz-audio device to be used by ASoC driver to bind ASoC platform driver. Currently old 'soc-audio' approach is used, which needs to be replaced with proper device. Signed-off-by: Dmitry Eremin-Solenikov <dbaryshkov(a)gmail.com> --- arch/arm/mach-pxa/spitz.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/arch/arm/mach-pxa/spitz.c b/arch/arm/mach-pxa/spitz.c index 840c3a4..962a7f3 100644 --- a/arch/arm/mach-pxa/spitz.c +++ b/arch/arm/mach-pxa/spitz.c @@ -924,6 +924,14 @@ static inline void spitz_i2c_init(void) {} #endif /****************************************************************************** + * Audio devices + ******************************************************************************/ +static inline void spitz_audio_init(void) +{ + platform_device_register_simple("spitz-audio", -1, NULL, 0); +} + +/****************************************************************************** * Machine init ******************************************************************************/ static void spitz_poweroff(void) @@ -970,6 +978,7 @@ static void __init spitz_init(void) spitz_nor_init(); spitz_nand_init(); spitz_i2c_init(); + spitz_audio_init(); } static void __init spitz_fixup(struct tag *tags, char **cmdline) -- 2.1.1

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[alsa-devel] [PATCH] ASoC: Use generic control handlers for S8 control
by Lars-Peter Clausen 28 Oct '14

28 Oct '14

Commit f227b88f0fce ("ASoC: core: Add signed register volume control logic") added support for signed control to the generic volsw control handler. This makes it possible to use them for the S8 control as well, rather than having to use a custom control handler implementation. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc.h | 19 +++++------- sound/soc/soc-core.c | 87 ---------------------------------------------------- 2 files changed, 8 insertions(+), 98 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 7ba7130..ad47e96 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -36,6 +36,11 @@ {.reg = xreg, .rreg = xreg, .shift = shift_left, \ .rshift = shift_right, .max = xmax, .platform_max = xmax, \ .invert = xinvert, .autodisable = xautodisable}) +#define SOC_DOUBLE_S_VALUE(xreg, shift_left, shift_right, xmin, xmax, xsign_bit, xinvert, xautodisable) \ + ((unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .rreg = xreg, .shift = shift_left, \ + .rshift = shift_right, .min = xmin, .max = xmax, .platform_max = xmax, \ + .sign_bit = xsign_bit, .invert = xinvert, .autodisable = xautodisable}) #define SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert, xautodisable) \ SOC_DOUBLE_VALUE(xreg, xshift, xshift, xmax, xinvert, xautodisable) #define SOC_SINGLE_VALUE_EXT(xreg, xmax, xinvert) \ @@ -171,11 +176,9 @@ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ SNDRV_CTL_ELEM_ACCESS_READWRITE, \ .tlv.p = (tlv_array), \ - .info = snd_soc_info_volsw_s8, .get = snd_soc_get_volsw_s8, \ - .put = snd_soc_put_volsw_s8, \ - .private_value = (unsigned long)&(struct soc_mixer_control) \ - {.reg = xreg, .min = xmin, .max = xmax, \ - .platform_max = xmax} } + .info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\ + .put = snd_soc_put_volsw, \ + .private_value = SOC_DOUBLE_S_VALUE(xreg, 0, 8, xmin, xmax, 7, 0, 0) } #define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xitems, xtexts) \ { .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ .items = xitems, .texts = xtexts, \ @@ -545,12 +548,6 @@ int snd_soc_get_volsw_sx(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); -int snd_soc_info_volsw_s8(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo); -int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol); -int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol); int snd_soc_info_volsw_range(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 96ecdc3..47c378a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2721,93 +2721,6 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, EXPORT_SYMBOL_GPL(snd_soc_put_volsw_sx); /** - * snd_soc_info_volsw_s8 - signed mixer info callback - * @kcontrol: mixer control - * @uinfo: control element information - * - * Callback to provide information about a signed mixer control. - * - * Returns 0 for success. - */ -int snd_soc_info_volsw_s8(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - int platform_max; - int min = mc->min; - - if (!mc->platform_max) - mc->platform_max = mc->max; - platform_max = mc->platform_max; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 2; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = platform_max - min; - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_info_volsw_s8); - -/** - * snd_soc_get_volsw_s8 - signed mixer get callback - * @kcontrol: mixer control - * @ucontrol: control element information - * - * Callback to get the value of a signed mixer control. - * - * Returns 0 for success. - */ -int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - unsigned int reg = mc->reg; - unsigned int val; - int min = mc->min; - int ret; - - ret = snd_soc_component_read(component, reg, &val); - if (ret) - return ret; - - ucontrol->value.integer.value[0] = - ((signed char)(val & 0xff))-min; - ucontrol->value.integer.value[1] = - ((signed char)((val >> 8) & 0xff))-min; - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_get_volsw_s8); - -/** - * snd_soc_put_volsw_sgn - signed mixer put callback - * @kcontrol: mixer control - * @ucontrol: control element information - * - * Callback to set the value of a signed mixer control. - * - * Returns 0 for success. - */ -int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - unsigned int reg = mc->reg; - int min = mc->min; - unsigned int val; - - val = (ucontrol->value.integer.value[0]+min) & 0xff; - val |= ((ucontrol->value.integer.value[1]+min) & 0xff) << 8; - - return snd_soc_component_update_bits(component, reg, 0xffff, val); -} -EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8); - -/** * snd_soc_info_volsw_range - single mixer info callback with range. * @kcontrol: mixer control * @uinfo: control element information -- 1.8.0

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[alsa-devel] [PATCH v2 0/8] ASoC: dapm: Cleanups and speed-ups
by Lars-Peter Clausen 28 Oct '14

28 Oct '14

Only as few minor cleanup changes since v1. No change in functionallity. Original cover letter from v1: This series contains a few cleanups and speed-ups for the DAPM implementation. It starts with removing some dead or outdated code and optimizing some lesser used operations. The first larger change introduced by this series is the categorization of widgets into sinks, sources, supplies and normal widgets. This is all the information the main DAPM algorithm needs and using these categories makes the code simpler and faster and hopefully easier to understand. The second major change is to use more aggressive caching on the connected input and output paths of a widget. It takes advantage of the fact that typically only as subsection of the DAPM graph changes per operation. This allows to drastically reduce the number of neighbor and path checks for some DAPM operations. The change is most effective when the system is idle and when changing the graph routing for a widget with multiple inputs. This e.g. happens a lot when switching the audio profile. It still has quite good improvements for other operations as well and typically leads to at least a 50% reduction of path and neighbor checks. Some before and after DAPM stats. All testing was done with the ADAU1761. Before is asoc/for-next, After~1 is asoc/for-next + this series except the last patch, After is asoc/for-next + this series. Changing a single control on a 5 input mixer while the system is idle: Power Path Neighbour Before: 2 25 32 After~1: 2 12 30 After: 2 1 2 Cumulative numbers for switching the audio profile which changes 7 controls while the system is idle: Power Path Neighbour Before: 16 141 188 After~1: 16 80 170 After: 16 7 23 Cumulative numbers for switching the audio profile which changes 7 controls while playback is active: Power Path Neighbour Before: 52 207 297 After~1: 51 123 273 After: 51 29 109 Starting (or stopping) the playback stream: Power Path Neighbour Before: 35 80 125 After~1: 34 34 117 After: 34 17 69 - Lars Lars-Peter Clausen (8): ASoC: dapm: Only mark paths dirty when the connection status changed ASoC: dapm: Do not add un-muxed paths to MUX control ASoC: dapm: Do not pretend to support controls for non mixer/mux widgets ASoC: dapm: Introduce toplevel widget categories ASoC: dapm: Add a flag to mark paths connected to supply widgets ASoC: dapm: Mark endpoints instead of IO widgets dirty during suspend/resume ASoC: dapm: Add a few supply widget sanity checks ASoC: dapm: Use more aggressive caching include/sound/soc-dapm.h | 8 +- sound/soc/soc-core.c | 8 +- sound/soc/soc-dapm.c | 576 ++++++++++++++++++++++++++++------------------- 3 files changed, 353 insertions(+), 239 deletions(-) -- 1.8.0

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[alsa-devel] [PATCH 00/12] ASoC: dapm: Cleanups and speed-ups
by Lars-Peter Clausen 28 Oct '14

28 Oct '14

Hi, This series contains a few cleanups and speed-ups for the DAPM implementation. It starts with removing some dead or outdated code and optimizing some lesser used operations. The first larger change introduced by this series is the categorization of widgets into sinks, sources, supplies and normal widgets. This is all the information the main DAPM algorithm needs and using these categories makes the code simpler and faster and hopefully easier to understand. The second major change is to use more aggressive caching on the connected input and output paths of a widget. It takes advantage of the fact that typically only as subsection of the DAPM graph changes per operation. This allows to drastically reduce the number of neighbor and path checks for some DAPM operations. The change is most effective when the system is idle and when changing the graph routing for a widget with multiple inputs. This e.g. happens a lot when switching the audio profile. It still has quite good improvements for other operations as well and typically leads to at least a 50% reduction of path and neighbor checks. Some before and after DAPM stats. All testing was done with the ADAU1761. Before is asoc/for-next, After~1 is asoc/for-next + this series except the last patch, After is asoc/for-next + this series. Changing a single control on a 5 input mixer while the system is idle: Power Path Neighbour Before: 2 25 32 After~1: 2 12 30 After: 2 1 2 Cumulative numbers for switching the audio profile which changes 7 controls while the system is idle: Power Path Neighbour Before: 16 141 188 After~1: 16 80 170 After: 16 7 23 Cumulative numbers for switching the audio profile which changes 7 controls while playback is active: Power Path Neighbour Before: 52 207 297 After~1: 51 123 273 After: 51 29 109 Starting (or stopping) the playback stream: Power Path Neighbour Before: 35 80 125 After~1: 34 34 117 After: 34 17 69 - Lars Lars-Peter Clausen (12): ASoC: dapm: Reduce number of checked paths in dapm_widget_in_card_paths() ASoC: dapm: Remove always true path source/sink checks ASoC: dapm: Only mark paths dirty when the connection status changed ASoC: dapm: Do not add un-muxed paths to MUX control ASoC: dapm: Do not pretend to support controls for non mixer/mux widgets ASoC: dapm: Remove special DAI widget power check functions ASoC: dapm: Remove path 'walked' flag ASoC: dapm: Introduce toplevel widget categories ASoC: dapm: Add a flag to mark paths connected to supply widgets ASoC: dapm: Mark endpoints instead of IO widgets dirty during suspend/resume ASoC: dapm: Add a few supply widget sanity checks ASoC: dapm: Use more aggressive caching include/sound/soc-dapm.h | 9 +- sound/soc/soc-core.c | 8 +- sound/soc/soc-dapm.c | 755 +++++++++++++++++++++++++---------------------- 3 files changed, 409 insertions(+), 363 deletions(-) -- 1.8.0

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[alsa-devel] 2014 ALSA micro-conference meeting notes - 10/14 Dussedorf
by Patrick Lai 27 Oct '14

27 Oct '14

Attendees: Takashi Iwai, Mark Brown, Vinod Koul, Patrick Lai, Lars-Peter Clausen, Eric Laurent, David Henningsson, Tanu Kaskinen, Arun Raghavan, Paul Handrigan, Daniel Mack, Dylan Reid, Charles Keepax, Nariman Poushin, Peter Meerwald, Alexander Patrakov, Hui Wang, Tim Cussins, Nick Stoughton Topic: ASoC componentization Motivation - hardware is becoming more complex as CPU and CODEC functionality are overlapping. Codec now has DSP. DSP on CPU need DAPM and IO. Current ASoC APIs are based on clearly distinguished component type, platform, CPU, CODEC. It makes difficult to support complex configuration such as CPU -> CODEC -> CODEC or DSP with DAC, ADC, SRC Points discussed - Componentization is coming on V 3.18. - Lars-Peter is cleaning up suspend/resume logic in ASoC framework as conflicting with individual driver suspend/resume. Goal is to move toward bus suspend/resume - Grand scheme is to treat hardware module as component instead of clearly distinguish component type(platform/cpu/codec). Russel King introduced component framework recently - one master with many slave components. ASoC could adopt component framework. Sound card would be the master while individual drivers register slave components. Moving to use component framework would help unbinding of drivers cleanly as well. - Discuss possible phasing out/refactor DPCM, DAI-LINK and add support to DAPM for PCM configuration propagation. Topic: Dynamic FW framework Motivation: It is very common for SoC/CODEC vendors putting DSP onto their hardware. Topology/functionality can change based on DSP image being loaded. Dynamic FW framework is designed to address the need to run-time create DSP topology graph without hard coding into the driver - For DSP with large memory case, swapping DSP image at run-time might not be a concern so firmware download can happen once after boot up but CODEC DSP tend to be memory constraint. Run-time swapping requiring deregister of existing controls and register with new one. ALSA framework does not have good support for this requirement currently. - DFW tool allows each solution vendor to develop its own plug-in and can convert static description to DFW binary format. Through this way, solution vendor can specify its own data and associate with ASoC component such as DAPM widget. Topic: Mapping channels through DAI links in ASoC Motivation: Currently, it is difficult to track how each channel get propagated from one component to another i.e 8 channel streams from CPU to CODECs with stereo channels in TDM interface - Consensus that there is value to track the channel mapping - Possible solution is to create widget per channel of PCM object Topic: ALSA test - Discussed test ALSA core framework on virtual platform and patch submission should go through testing. For embedded system, the benefit may not be significant as bug tend to show up during use case transition - David demonstrated HDA CODEC test. Perhaps, this test can be leveraged for ALSA core test framework - There is no ALSA compliance test right now. There is small tool called alsa-time-test which could potentially be part of ALSA test suite. Another way to validate conformance is to run PulseAudio as PulseAudio exercises a lot of alsa-lib API - Vinod suggested a test framework Liam has been developing, to compare the sign wave form with the captured one. Contribution is welcome and can be merged to any public repo. In addition, for *any* other test programs or metrics, contribution is welcomed. Topic: ALSA documentation - For mixer control related query, Documentation/sound/alsa/ControlNames.txt in kernel tree is a good resource to read - Each kernel function can have a Javadoc style comment. This can be formated through DocBook directory in kernel tree as PDF or HTML (or anything else). So, from now on, the javadoc style comment is mandatory for each patch including exported symbols. - the root DocBook template file is missing for ASoC. Presentation given by Mark in ELC few years back could be a good supplement for someone to get a quick intro to develop drivers to plug into ASoC framework Topic: ALSA control names - Even though in embedded world, SoC architecture from different vendors are not so consistent enough to come up with common name. There is still value to come up with standard name for different types of controls such as Volume, Ramp up/down time (in unit of DB per ms or sample count) - Media controller discussion came up again as it can be solution to help user-space recognize mixer control for the topology - General consensus is that each vendor should have control definition document to describe what each control do - As part of user case transition, user-space often execute mute as well. Instead of ASoC framework handles digital mute as part of startup/shutdown sequence, let user-space have the control through mixer control command. - Suggest to have volume, mute control per PCM object which takes the same approach as channel MAP API by exposing mixer control with name "<pcm device id> volume" Topic: ALSA code removal - Can OSS be removed? Anyone is still using OSS? Perhaps, OSS emulation can be done in user-space - Add warning to notify people to contact usage of APIs to prevent deprecation - Discuss removal of old driver but does not look safe to remove - Suggestion to remove ADPCM support in ALSA core framework - Removal in alsa-library: aserver, shm, and share plugins were discussed. The decision seems to be putting them under a ./configure switch now and remove later Topic: DSP effect library/more DSP support - For DSP effect, Intel goes through mixer control approach which controls are defined in the dynamic firmware. It is not clear how Android effect would be associate stream to corresponding mixer controls for the DSP effect - Discuss couple issues when dealing with DSP architecture. These problems surfaces on PulseAudio 1. DSP DMA data as soon as it can put into its internal memory. Hence, hw_ptr does not reflect actual consumption point in DSP. This causes rewinding logic not function properly. It would be ideal if ALSA could allow driver expose granularity of hardware pointer 2. DSP continuous latency can vary. ALSA should have mechanism to allow driver expose latency information to user-space Topic: consolidate jack report mechanism Currently, there are K-control, input event, extcon mechanism to report headset jack events. Intel would like to see k-control as final solution in order to align with DSP notification through k-control Discussed to consolidate into one mechanism. One proposal from Mark is making a common in-kernel framework which all driver would use this jack framework, the framework would in turn expose this jack through different user-space interfaces depending on kernel configuration Topic: Convergence of single configuration file format UCM vs mixer_path Single format would make vendors who supports both UCM and mixer_path systems life easier. PulseAudio extract use cases list from UCM while mixer_path does not have such support yet. Action item: Arun to look into the convergence effort Topic: scheduled trigger start of PCM - Nick to submit snd_pcm_start_at() patch. The change is software based solution but margin of error is acceptable for the use case in mind - Brought up the use case which time stamp is sent along with every buffer so DSP knows to drop when the buffer is given pass the deadline. Topic: ACPI Intel is moving system-level device configuration file toward ACPI. Device specific data(_DSD) section defined in ACPI 5.1 specification allows device vendor to pass device specific data. (i.e CODEC device data). Topic: synchronization of HDMI/USB hot-plug and configuration events - Instead of waiting for USB hotplug event, Android should wait for udev event to notify new USB sound card is created - Discuss making HDMI a separate sound card. However, HDMI could simply be another routing path in a sound card for embedded architecture. - Suggest to return error when reading EDID info is HDMI is not ready Topic: Sharing hardware component between ALSA framework and other drivers - When hot word is detected, go through DAPM to enable mic path. Topic: Rewindng in ALSA PCM plugins 1. Discussion of the method to calculate the amount of samples that should not be rewound over after the hardware pointer. Result: the audience demonstrated general understanding of the issue. Lars suggested having two hardware pointers, because there are two boundaries that are unsafe to cross when overwriting samples: one from the left (normal write path), and one from the right (rewind path). No action items. 2. Discussion of the criteria when to enable timer-based scheduling. Suggestion: do not enumerate badness [*] when deciding whether to enable tsched in PulseAudio, because the bad things tend to go unreported by the drivers and people treat these as PulseAudio bugs. Move to a whitelist-based approach. The suggestion was rejected. [*] BLOCK_TRANSFER (unimplemented), BATCH (implemented), huge FIFO (unimplemented), completely unrewindable (unimplemented, but a workaround for ioplug exists), what else? 3. Suggestion to revive the snd_pcm_hw_params_can_rewind() API function. Approved by Takashi. 4. Discussion whether the snd_pcm_rewindable() function is useful at all. The uselessness comes from the need to call snd_pcm_avail_update() beforehand, and the position of Clemens Ladisch that the result of snd_pcm_rewindable() must not take pointer granularity and other reasons why non-rewindable samples exist into account. That way, the caller can just deduce the result from the snd_pcm_avail_update() return value. David Henningsson does not like this situation. Mark Brown suggests reporting the DMA transfer size, with the possibility for architectures to override the returned value with something larger such as the cache line size. 5. Status report on rewind support in PCM plugins. Summary: at least 10 plugins out of 26 have known problems and should have rewind support either fixed (which is not always possible) or disabled. Thanks Patrick -- Patrick Lai Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum,a Linux Foundation Collaborative Project

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[alsa-devel] [PATCH 0/3] ASoC: intel: add support for future acpi platforms
by Vinod Koul 27 Oct '14

27 Oct '14

We have some SoCs which are ACPI based. Current driver supports Merrifield type PCI SoC. SO make some minor changes required for ACPI support in subsequent series Subhransu S. Prusty (3): ASoC: Intel: mrfld: Replace pci_id with unique device id ASoC: Intel: mrfld - Define ipc_info structure ASoC: Intel: mrfld: Define sst_res_info for acpi arch/x86/include/asm/platform_sst_audio.h | 28 ++++++++++++++++++++++++++++ sound/soc/intel/sst/sst.c | 10 +++++----- sound/soc/intel/sst/sst.h | 5 +++-- sound/soc/intel/sst/sst_pvt.c | 4 ++-- 4 files changed, 38 insertions(+), 9 deletions(-)

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[alsa-devel] [PATCH] ALSA: hda - Pass printf argument directly to request_module()
by Takashi Iwai 27 Oct '14

27 Oct '14

request_module() handles the printf style arguments, so we don't have to render strings in the caller side. Not only it reduces the unnecessary temporary string buffer, it's even safer from the security POV. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 259fbeaa37bd..0025bf4c2f44 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -966,14 +966,12 @@ find_codec_preset(struct hda_codec *codec) mutex_unlock(&preset_mutex); if (mod_requested < HDA_MODREQ_MAX_COUNT) { - char name[32]; if (!mod_requested) - snprintf(name, sizeof(name), "snd-hda-codec-id:%08x", - codec->vendor_id); + request_module("snd-hda-codec-id:%08x", + codec->vendor_id); else - snprintf(name, sizeof(name), "snd-hda-codec-id:%04x*", - (codec->vendor_id >> 16) & 0xffff); - request_module(name); + request_module("snd-hda-codec-id:%04x*", + (codec->vendor_id >> 16) & 0xffff); mod_requested++; goto again; } -- 2.1.2

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[alsa-devel] Uninitialized id returned by saffirepro_both_clk_src_get.
by Christian Vogel 27 Oct '14

27 Oct '14

Hi, there's a possibility to get a Oops caused by an uninitialized value in snd_bebob_stream_check_internal_clock for a SaffirePro running on the internal clock. [ 88.100531] BUG: unable to handle kernel paging request at 8a3c85fc [ 88.103808] IP: [<e8553aa0>] snd_bebob_stream_check_internal_clock+0x66/0x11e [snd_bebob] ...which is dereferencing of clk_spec->labels[id] in... sound/firewire/bebob/bebob_stream.c : /* 1.The device has its own operation to switch source of clock */ if (clk_spec) { err = clk_spec->get(bebob, &id); if (err < 0) dev_err(&bebob->unit->device, "fail to get clock source: %d\n", err); --> else if (strncmp(clk_spec->labels[id], SND_BEBOB_CLOCK_INTERNAL, strlen(SND_BEBOB_CLOCK_INTERNAL)) == 0) *internal = true; goto end; } id is uninitialized, and will not be set by clk_spec->get (which is saffirepro_both_clk_src_get(), even if it returns ok(0). Attached patch tries to clean up the logic in saffirepro_both_clk_src_get() and also adds a safety check to snd_bebob_stream_check_internal_clock(). Thanks for Takashi Sakamoto to whom I sent the patch initially and who suggested some cleanup to my code, reviewed the patch and suggested I send it to alsa-dev. Chris

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[alsa-devel] [PATCH] pcm: Fix snd_pcm_avail_update description
by David Henningsson 27 Oct '14

27 Oct '14

While reading alsa-lib code, I discovered that snd_pcm_avail_update does require a user-kernel context switch: snd_pcm_avail_update() -> snd_pcm_hw_avail_update() -> sync_ptr() -> sync_ptr1() -> ioctl() The documentation says it does not, so let's fix the documentation. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- src/pcm/pcm.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 4a7be6c..e57e362 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -403,10 +403,12 @@ The function #snd_pcm_avail_update() updates the current available count of samples for writing (playback) or filled samples for reading (capture). This call is mandatory for updating actual r/w pointer. Using standalone, it is a light method to obtain current stream position, -because it does not require the user <-> kernel context switch, but the value -is less accurate, because ring buffer pointers are updated in kernel drivers +because while it does require a user <-> kernel context switch, the value is +just copied from the kernel, and not updated by synchronously asking the hardware +about its current position. The ring buffer pointers are updated in kernel drivers only when an interrupt occurs. If you want to get accurate stream state, -use functions #snd_pcm_avail(), #snd_pcm_delay() or #snd_pcm_avail_delay(). +use functions #snd_pcm_avail(), #snd_pcm_delay(), #snd_pcm_avail_delay() or +#snd_pcm_status(). </p> <p> The function #snd_pcm_avail() reads the current hardware pointer -- 1.9.1

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[alsa-devel] [GIT PULL] ASoC fixes for v3.18
by Mark Brown 27 Oct '14

27 Oct '14

The following changes since commit cac7f2429872d3733dc3f9915857b1691da2eb2f: Linux 3.18-rc2 (2014-10-26 16:48:41 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.18-rc2 for you to fetch changes up to c0d018bd5b1aabad59dffdec568e189359d93a14: Merge remote-tracking branches 'asoc/fix/adau1761', 'asoc/fix/fsl', 'asoc/fix/intel', 'asoc/fix/s6000' and 'asoc/fix/sgtl5000' into asoc-linus (2014-10-27 11:17:41 +0000) ---------------------------------------------------------------- ASoC: Fixes for v3.18 A few small driver fixes for v3.18 plus the removal of the s6000 support since the relevant chip is no longer supported in mainline. ---------------------------------------------------------------- Daniel Glöckner (1): ASoC: s6000: remove driver Daniel Mack (1): ASoC: fsl: use strncpy() to prevent copying of over-long names Fabio Estevam (1): ASoC: sgtl500: Document the required supplies Lars-Peter Clausen (1): ASoC: adau1761: Fix input PGA volume Liam Girdwood (1): ASoC: Intel: HSW/BDW only support S16 and S24 formats. Mark Brown (1): Merge remote-tracking branches 'asoc/fix/adau1761', 'asoc/fix/fsl', 'asoc/fix/intel', 'asoc/fix/s6000' and 'asoc/fix/sgtl5000' into asoc-linus .../devicetree/bindings/sound/sgtl5000.txt | 10 + sound/soc/Kconfig | 1 - sound/soc/Makefile | 1 - sound/soc/codecs/adau1761.c | 4 + sound/soc/fsl/fsl_asrc.c | 2 +- sound/soc/fsl/fsl_esai.c | 2 +- sound/soc/intel/sst-haswell-pcm.c | 4 +- sound/soc/s6000/Kconfig | 26 - sound/soc/s6000/Makefile | 11 - sound/soc/s6000/s6000-i2s.c | 617 --------------------- sound/soc/s6000/s6000-i2s.h | 23 - sound/soc/s6000/s6000-pcm.c | 521 ----------------- sound/soc/s6000/s6000-pcm.h | 33 -- sound/soc/s6000/s6105-ipcam.c | 221 -------- 14 files changed, 17 insertions(+), 1459 deletions(-) delete mode 100644 sound/soc/s6000/Kconfig delete mode 100644 sound/soc/s6000/Makefile delete mode 100644 sound/soc/s6000/s6000-i2s.c delete mode 100644 sound/soc/s6000/s6000-i2s.h delete mode 100644 sound/soc/s6000/s6000-pcm.c delete mode 100644 sound/soc/s6000/s6000-pcm.h delete mode 100644 sound/soc/s6000/s6105-ipcam.c

2 1

Re: [alsa-devel] [PATCH v2 8/8] ASoC: dapm: Use more aggressive caching
by noman pouigt 27 Oct '14

27 Oct '14

>E.g. imagine the following example: > > A --> B ----. > v > M --> N --> Z <-- S <-- R > | > v > X> > >Widget Z has multiple input paths, if any change was made that cause Z to be >marked as dirty the power state of Z has to be re-computed. This requires to >know the number of inputs and outputs of Z, which requires to know the >number of inputs and outputs of all widgets on all paths from or to Z. >Previously this meant re-computing all inputs and outputs of all the path >going into or out of Z. With this patch in place only paths that actually >have changed need to be re-computed. so suppose "B" was the widget which caused "Z" to be dirtied then with your patch only you will calculate A->B->N->Z ? Is my understanding right? A->B->C | v G->P v | D->E->F What will happen in the above diagram if P gets dirtied?

2 1

[alsa-devel] [PATCH v2] ASoC: Intel: Add Cherrytrail & Braswell machine driver cht_bsw_rt5672
by mengdong.lin＠intel.com 27 Oct '14

27 Oct '14

From: Mengdong Lin <mengdong.lin(a)intel.com> Add machine driver for two Intel Cherryview-based platforms, Cherrytrail and Braswell, with RT5672 codec. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 2a3af88..7479ce0 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -16,6 +16,9 @@ config SND_SST_MFLD_PLATFORM config SND_SST_IPC tristate +config SND_SST_MACHINE + tristate + config SND_SOC_INTEL_SST tristate "ASoC support for Intel(R) Smart Sound Technology" select SND_SOC_INTEL_SST_ACPI if ACPI @@ -76,3 +79,20 @@ config SND_SOC_INTEL_BROADWELL_MACH Ultrabook platforms. Say Y if you have such a device If unsure select "N". + +config SND_SOC_INTEL_CHT_BSW_RT5672_MACH + tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5672 codec" + depends on X86_INTEL_LPSS + select SND_SOC_RT5670 + select SND_SST_MFLD_PLATFORM + select SND_SOC_INTEL_SST + select SND_SST_IPC + select SND_SST_MACHINE + default n + + help + This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell + platforms with RT5672 audio codec. + Say Y if you have such a device + If unsure select "N". + diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index 9ab43be..4069d3f 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -31,6 +31,7 @@ obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o +obj-$(CONFIG_SND_SST_MACHINE) += board/ # DSP driver obj-$(CONFIG_SND_SST_IPC) += sst/ diff --git a/sound/soc/intel/board/Makefile b/sound/soc/intel/board/Makefile new file mode 100644 index 0000000..9ecc227 --- /dev/null +++ b/sound/soc/intel/board/Makefile @@ -0,0 +1,2 @@ +snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o +obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH) += snd-soc-sst-cht-bsw-rt5672.o diff --git a/sound/soc/intel/board/cht_bsw_rt5672.c b/sound/soc/intel/board/cht_bsw_rt5672.c new file mode 100644 index 0000000..789d1a0 --- /dev/null +++ b/sound/soc/intel/board/cht_bsw_rt5672.c @@ -0,0 +1,246 @@ +/* + * cht_bsw_rt5672.c - ASoc Machine driver for Intel Cherryview-based platforms + * Cherrytrail and Braswell, with RT5672 codec. + * + * Copyright (C) 2014 Intel Corp + * Author: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> + * Mengdong Lin <mengdong.lin(a)intel.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include "../../codecs/rt5670.h" +#include "../sst-atom-controls.h" + +#define CHT_PLAT_CLK_3_HZ 19200000 + +static const struct snd_soc_dapm_widget cht_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + +static const struct snd_soc_dapm_route cht_audio_map[] = { + {"IN1P", NULL, "Headset Mic"}, + {"IN1N", NULL, "Headset Mic"}, + {"DMIC L1", NULL, "Int Mic"}, + {"DMIC R1", NULL, "Int Mic"}, + {"Headphone", NULL, "HPOL"}, + {"Headphone", NULL, "HPOR"}, + {"Ext Spk", NULL, "SPOLP"}, + {"Ext Spk", NULL, "SPOLN"}, + {"Ext Spk", NULL, "SPORP"}, + {"Ext Spk", NULL, "SPORN"}, + {"AIF1 Playback", NULL, "ssp2 Tx"}, + {"ssp2 Tx", NULL, "codec_out0"}, + {"ssp2 Tx", NULL, "codec_out1"}, + {"codec_in0", NULL, "ssp2 Rx"}, + {"codec_in1", NULL, "ssp2 Rx"}, + {"ssp2 Rx", NULL, "AIF1 Capture"}, +}; + +static const struct snd_kcontrol_new cht_mc_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Int Mic"), + SOC_DAPM_PIN_SWITCH("Ext Spk"), +}; + +static int cht_aif1_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + if (strncmp(codec_dai->name, "rt5670-aif1", 11)) + return 0; + + ret = snd_soc_dai_set_pll(codec_dai, 0, RT5670_PLL1_S_MCLK, + CHT_PLAT_CLK_3_HZ, params_rate(params) * 512); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec pll: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, RT5670_SCLK_S_PLL1, + params_rate(params) * 512, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec sysclk: %d\n", ret); + return ret; + } + return 0; +} + +static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) +{ + int ret; + struct snd_soc_dai *codec_dai = runtime->codec_dai; + + /* 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) { + dev_err(runtime->dev, "can't set codec TDM slot %d\n", ret); + return ret; + } + + return 0; +} + +static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + /* The DSP will covert the FE rate to 48k, stereo, 24bits */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP2 to 24-bit */ + snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT - + SNDRV_PCM_HW_PARAM_FIRST_MASK], + SNDRV_PCM_FORMAT_S24_LE); + return 0; +} + +static unsigned int rates_48000[] = { + 48000, +}; + +static struct snd_pcm_hw_constraint_list constraints_48000 = { + .count = ARRAY_SIZE(rates_48000), + .list = rates_48000, +}; + +static int cht_aif1_startup(struct snd_pcm_substream *substream) +{ + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_48000); +} + +static struct snd_soc_ops cht_aif1_ops = { + .startup = cht_aif1_startup, +}; + +static struct snd_soc_ops cht_be_ssp2_ops = { + .hw_params = cht_aif1_hw_params, +}; + +static struct snd_soc_dai_link cht_dailink[] = { + /* Front End DAI links */ + [MERR_DPCM_AUDIO] = { + .name = "Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "media-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + .ignore_suspend = 1, + .dynamic = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &cht_aif1_ops, + }, + [MERR_DPCM_COMPR] = { + .name = "Compressed Port", + .stream_name = "Compress", + .cpu_dai_name = "compress-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + }, + + /* Back End DAI links */ + { + /* SSP2 - Codec */ + .name = "SSP2-Codec", + .be_id = 1, + .cpu_dai_name = "ssp2-port", + .platform_name = "sst-mfld-platform", + .no_pcm = 1, + .codec_dai_name = "rt5670-aif1", + .codec_name = "i2c-10EC5670:00", + .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF + | SND_SOC_DAIFMT_CBS_CFS, + .init = cht_codec_init, + .be_hw_params_fixup = cht_codec_fixup, + .ignore_suspend = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &cht_be_ssp2_ops, + }, +}; + +/* SoC card */ +static struct snd_soc_card snd_soc_card_cht = { + .name = "cherrytrailcraudio", + .dai_link = cht_dailink, + .num_links = ARRAY_SIZE(cht_dailink), + .dapm_widgets = cht_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cht_dapm_widgets), + .dapm_routes = cht_audio_map, + .num_dapm_routes = ARRAY_SIZE(cht_audio_map), + .controls = cht_mc_controls, + .num_controls = ARRAY_SIZE(cht_mc_controls), +}; + +static int snd_cht_mc_probe(struct platform_device *pdev) +{ + int ret_val = 0; + + /* register the soc card */ + snd_soc_card_cht.dev = &pdev->dev; + ret_val = snd_soc_register_card(&snd_soc_card_cht); + if (ret_val) { + dev_err(&pdev->dev, + "snd_soc_register_card failed %d\n", ret_val); + return ret_val; + } + platform_set_drvdata(pdev, &snd_soc_card_cht); + return ret_val; +} + +static int snd_cht_mc_remove(struct platform_device *pdev) +{ + struct snd_soc_card *soc_card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(soc_card); + platform_set_drvdata(pdev, NULL); + return 0; +} + +static struct platform_driver snd_cht_mc_driver = { + .driver = { + .owner = THIS_MODULE, + .name = "cht-bsw-rt5672", + .pm = &snd_soc_pm_ops, + }, + .probe = snd_cht_mc_probe, + .remove = snd_cht_mc_remove, +}; + +module_platform_driver(snd_cht_mc_driver); + +MODULE_DESCRIPTION("ASoC Intel(R) Baytrail CR Machine driver"); +MODULE_AUTHOR("Subhransu S. Prusty, Mengdong Lin"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:cht-bsw-rt5672"); -- 1.9.1

1 0

[alsa-devel] Customizing simple audio card?
by Max Filippov 27 Oct '14

27 Oct '14

Hello, I'm developing ALSA driver for sound hardware on Xtensa FPGA boards. The hardware is very simple, we have AIC23 codec connected to I2S bus and I2S bus master that pumps data pushed by CPU into its FIFO. Both are connected to the same clock synthesizer output. There are no other devices that need to be supported now. Due to its simplicity the setup could happily use the simple sound card driver. But to support both 44100 and 48000 frequency ranges I need to change master clock, setting it either to 11289600 or to 12288000 Hz. I can do it in the I2S driver, but I see no way to inform audio codec of that change. I used the following hack to debug everything with the simple audio card: diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index cef7776..abef5ed 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -43,6 +44,22 @@ static int asoc_simple_card_hw_params(struct snd_pcm_substream *substream, SND_SOC_CLOCK_IN); } + switch (params_rate(params)) { + case 48000: + case 32000: + case 16000: + case 8000: + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 12288000, + SND_SOC_CLOCK_IN); + break; + case 44100: + case 22050: + case 11025: + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 11289600, + SND_SOC_CLOCK_IN); + break; + + } return ret; } but that's of course not a solution. So my question is is there any way I can subclass or inherit from the simple audio card driver to add such functionality to it? -- Thanks. -- Max

2 2

[alsa-devel] [PATCH 1/7] ASoC: Intel: Make ADSP memory block allocation more generic
by Liam Girdwood 27 Oct '14

27 Oct '14

Current block allocation is tied to block type and requestor type. Make the allocation more generic by removing the struct module parameter and adding a generic block allocator structure. Also pass in the list that the blocks have to be added too in order to remove dependence on block requestor type. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/intel/sst-baytrail-dsp.c | 24 ++- sound/soc/intel/sst-dsp-priv.h | 83 +++++----- sound/soc/intel/sst-firmware.c | 317 +++++++++++++++++++++++++------------ sound/soc/intel/sst-haswell-dsp.c | 41 ++--- 4 files changed, 288 insertions(+), 177 deletions(-) diff --git a/sound/soc/intel/sst-baytrail-dsp.c b/sound/soc/intel/sst-baytrail-dsp.c index fc58876..5a9e567 100644 --- a/sound/soc/intel/sst-baytrail-dsp.c +++ b/sound/soc/intel/sst-baytrail-dsp.c @@ -67,17 +67,12 @@ static int sst_byt_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, { struct dma_block_info *block; struct sst_module *mod; - struct sst_module_data block_data; struct sst_module_template template; int count; memset(&template, 0, sizeof(template)); template.id = module->type; template.entry = module->entry_point; - template.p.type = SST_MEM_DRAM; - template.p.data_type = SST_DATA_P; - template.s.type = SST_MEM_DRAM; - template.s.data_type = SST_DATA_S; mod = sst_module_new(fw, &template, NULL); if (mod == NULL) @@ -94,19 +89,19 @@ static int sst_byt_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, switch (block->type) { case SST_BYT_IRAM: - block_data.offset = block->ram_offset + + mod->offset = block->ram_offset + dsp->addr.iram_offset; - block_data.type = SST_MEM_IRAM; + mod->type = SST_MEM_IRAM; break; case SST_BYT_DRAM: - block_data.offset = block->ram_offset + + mod->offset = block->ram_offset + dsp->addr.dram_offset; - block_data.type = SST_MEM_DRAM; + mod->type = SST_MEM_DRAM; break; case SST_BYT_CACHE: - block_data.offset = block->ram_offset + + mod->offset = block->ram_offset + (dsp->addr.fw_ext - dsp->addr.lpe); - block_data.type = SST_MEM_CACHE; + mod->type = SST_MEM_CACHE; break; default: dev_err(dsp->dev, "wrong ram type 0x%x in block0x%x\n", @@ -114,11 +109,10 @@ static int sst_byt_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, return -EINVAL; } - block_data.size = block->size; - block_data.data_type = SST_DATA_M; - block_data.data = (void *)block + sizeof(*block); + mod->size = block->size; + mod->data = (void *)block + sizeof(*block); - sst_module_insert_fixed_block(mod, &block_data); + sst_module_alloc_blocks(mod); block = (void *)block + sizeof(*block) + block->size; } diff --git a/sound/soc/intel/sst-dsp-priv.h b/sound/soc/intel/sst-dsp-priv.h index ffb308b..f58a049 100644 --- a/sound/soc/intel/sst-dsp-priv.h +++ b/sound/soc/intel/sst-dsp-priv.h @@ -84,15 +84,6 @@ struct sst_mailbox { }; /* - * Audio DSP Firmware data types. - */ -enum sst_data_type { - SST_DATA_M = 0, /* module block data */ - SST_DATA_P = 1, /* peristant data (text, data) */ - SST_DATA_S = 2, /* scratch data (usually buffers) */ -}; - -/* * Audio DSP memory block types. */ enum sst_mem_type { @@ -125,23 +116,6 @@ struct sst_fw { }; /* - * Audio DSP Generic Module data. - * - * This is used to dsecribe any sections of persistent (text and data) and - * scratch (buffers) of module data in ADSP memory space. - */ -struct sst_module_data { - - enum sst_mem_type type; /* destination memory type */ - enum sst_data_type data_type; /* type of module data */ - - u32 size; /* size in bytes */ - int32_t offset; /* offset in FW file */ - u32 data_offset; /* offset in ADSP memory space */ - void *data; /* module data */ -}; - -/* * Audio DSP Generic Module Template. * * Used to define and register a new FW module. This data is extracted from @@ -150,15 +124,42 @@ struct sst_module_data { struct sst_module_template { u32 id; u32 entry; /* entry point */ - struct sst_module_data s; /* scratch data */ - struct sst_module_data p; /* peristant data */ + u32 scratch_size; + u32 persistent_size; +}; + +/* + * Block Allocator - Used to allocate blocks of DSP memory. + */ +struct sst_block_allocator { + u32 id; + u32 offset; + int size; + enum sst_mem_type type; +}; + +/* + * Runtime Module Instance - A module object can be instanciated multiple + * times within the DSP FW. + */ +struct sst_module_runtime { + struct sst_dsp *dsp; + int id; + struct sst_module *module; /* parent module we belong too */ + + u32 persistent_offset; /* private memory size */ + void *private; + + struct list_head list; + struct list_head block_list; /* list of blocks used */ }; /* * Audio DSP Generic Module. * * Each Firmware file can consist of 1..N modules. A module can span multiple - * ADSP memory blocks. The simplest FW will be a file with 1 module. + * ADSP memory blocks. The simplest FW will be a file with 1 module. A module + * can be instanciated multiple times in the DSP. */ struct sst_module { struct sst_dsp *dsp; @@ -167,10 +168,13 @@ struct sst_module { /* module configuration */ u32 id; u32 entry; /* module entry point */ - u32 offset; /* module offset in firmware file */ + s32 offset; /* module offset in firmware file */ u32 size; /* module size */ - struct sst_module_data s; /* scratch data */ - struct sst_module_data p; /* peristant data */ + u32 scratch_size; /* global scratch memory required */ + u32 persistent_size; /* private memory required */ + enum sst_mem_type type; /* destination memory type */ + u32 data_offset; /* offset in ADSP memory space */ + void *data; /* module data */ /* runtime */ u32 usage_count; /* can be unloaded if count == 0 */ @@ -180,6 +184,7 @@ struct sst_module { struct list_head block_list; /* Module list of blocks in use */ struct list_head list; /* DSP list of modules */ struct list_head list_fw; /* FW list of modules */ + struct list_head runtime_list; /* list of runtime module objects*/ }; /* @@ -208,7 +213,6 @@ struct sst_mem_block { struct sst_block_ops *ops; /* block operations, if any */ /* block status */ - enum sst_data_type data_type; /* data type held in this block */ u32 bytes_used; /* bytes in use by modules */ void *private; /* generic core does not touch this */ int users; /* number of modules using this block */ @@ -290,19 +294,20 @@ void sst_fw_unload(struct sst_fw *sst_fw); /* Create/Free firmware modules */ struct sst_module *sst_module_new(struct sst_fw *sst_fw, struct sst_module_template *template, void *private); -void sst_module_free(struct sst_module *sst_module); -int sst_module_insert(struct sst_module *sst_module); -int sst_module_remove(struct sst_module *sst_module); -int sst_module_insert_fixed_block(struct sst_module *module, - struct sst_module_data *data); +void sst_module_free(struct sst_module *module); struct sst_module *sst_module_get_from_id(struct sst_dsp *dsp, u32 id); /* allocate/free pesistent/scratch memory regions managed by drv */ struct sst_module *sst_mem_block_alloc_scratch(struct sst_dsp *dsp); void sst_mem_block_free_scratch(struct sst_dsp *dsp, struct sst_module *scratch); -int sst_block_module_remove(struct sst_module *module); +int sst_module_alloc_blocks(struct sst_module *module); +int sst_module_free_blocks(struct sst_module *module); +/* generic block allocation */ +int sst_alloc_blocks(struct sst_dsp *dsp, struct sst_block_allocator *ba, + struct list_head *block_list); +int sst_free_blocks(struct sst_dsp *dsp, struct list_head *block_list); /* Register the DSPs memory blocks - would be nice to read from ACPI */ struct sst_mem_block *sst_mem_block_register(struct sst_dsp *dsp, u32 offset, u32 size, enum sst_mem_type type, struct sst_block_ops *ops, u32 index, diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index 3bb43da..f2f1d5f 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -30,7 +30,7 @@ #include "sst-dsp.h" #include "sst-dsp-priv.h" -static void block_module_remove(struct sst_module *module); +#define SST_HSW_BLOCK_ANY 0xffffffff static void sst_memcpy32(volatile void __iomem *dest, void *src, u32 bytes) { @@ -41,6 +41,65 @@ static void sst_memcpy32(volatile void __iomem *dest, void *src, u32 bytes) memcpy_toio(dest + i, src + i, 4); } + +/* remove module from memory - callers hold locks */ +static void block_list_remove(struct sst_dsp *dsp, + struct list_head *block_list) +{ + struct sst_mem_block *block, *tmp; + int err; + + /* disable each block */ + list_for_each_entry(block, block_list, module_list) { + + if (block->ops && block->ops->disable) { + err = block->ops->disable(block); + if (err < 0) + dev_err(dsp->dev, + "error: cant disable block %d:%d\n", + block->type, block->index); + } + } + + /* mark each block as free */ + list_for_each_entry_safe(block, tmp, block_list, module_list) { + list_del(&block->module_list); + list_move(&block->list, &dsp->free_block_list); + dev_dbg(dsp->dev, "block freed %d:%d at offset 0x%x\n", + block->type, block->index, block->offset); + } +} + +/* prepare the memory block to receive data from host - callers hold locks */ +static int block_list_prepare(struct sst_dsp *dsp, + struct list_head *block_list) +{ + struct sst_mem_block *block; + int ret = 0; + + /* enable each block so that's it'e ready for data */ + list_for_each_entry(block, block_list, module_list) { + + if (block->ops && block->ops->enable) { + ret = block->ops->enable(block); + if (ret < 0) { + dev_err(dsp->dev, + "error: cant disable block %d:%d\n", + block->type, block->index); + goto err; + } + } + } + return ret; + +err: + list_for_each_entry(block, block_list, module_list) { + if (block->ops && block->ops->disable) + block->ops->disable(block); + } + return ret; +} + /* create new generic firmware object */ struct sst_fw *sst_fw_new(struct sst_dsp *dsp, const struct firmware *fw, void *private) @@ -202,73 +261,122 @@ void sst_module_free(struct sst_module *sst_module) } EXPORT_SYMBOL_GPL(sst_module_free); -static struct sst_mem_block *find_block(struct sst_dsp *dsp, int type, - u32 offset) +struct sst_module_runtime *sst_module_runtime_new(struct sst_module *module, + int id, void *private) +{ + struct sst_dsp *dsp = module->dsp; + struct sst_module_runtime *runtime; + + runtime = kzalloc(sizeof(*runtime), GFP_KERNEL); + if (runtime == NULL) + return NULL; + + runtime->id = id; + runtime->dsp = dsp; + runtime->module = module; + INIT_LIST_HEAD(&runtime->block_list); + + mutex_lock(&dsp->mutex); + list_add(&runtime->list, &module->runtime_list); + mutex_unlock(&dsp->mutex); + + return runtime; +} +EXPORT_SYMBOL_GPL(sst_module_runtime_new); + +void sst_module_runtime_free(struct sst_module_runtime *runtime) +{ + struct sst_dsp *dsp = runtime->dsp; + + mutex_lock(&dsp->mutex); + list_del(&runtime->list); + mutex_unlock(&dsp->mutex); + + kfree(runtime); +} +EXPORT_SYMBOL_GPL(sst_module_runtime_free); + +static struct sst_mem_block *find_block(struct sst_dsp *dsp, + struct sst_block_allocator *ba) { struct sst_mem_block *block; list_for_each_entry(block, &dsp->free_block_list, list) { - if (block->type == type && block->offset == offset) + if (block->type == ba->type && block->offset == ba->offset) return block; } return NULL; } -static int block_alloc_contiguous(struct sst_module *module, - struct sst_module_data *data, u32 offset, int size) +/* Block allocator must be on block boundary */ +static int block_alloc_contiguous(struct sst_dsp *dsp, + struct sst_block_allocator *ba, struct list_head *block_list) { struct list_head tmp = LIST_HEAD_INIT(tmp); - struct sst_dsp *dsp = module->dsp; struct sst_mem_block *block; + u32 block_start = SST_HSW_BLOCK_ANY; + int size = ba->size, offset = ba->offset; + + while (ba->size > 0) { - while (size > 0) { - block = find_block(dsp, data->type, offset); + block = find_block(dsp, ba); if (!block) { list_splice(&tmp, &dsp->free_block_list); + + ba->size = size; + ba->offset = offset; return -ENOMEM; } list_move_tail(&block->list, &tmp); - offset += block->size; - size -= block->size; + ba->offset += block->size; + ba->size -= block->size; } + ba->size = size; + ba->offset = offset; + + list_for_each_entry(block, &tmp, list) { + + if (block->offset < block_start) + block_start = block->offset; - list_for_each_entry(block, &tmp, list) - list_add(&block->module_list, &module->block_list); + list_add(&block->module_list, block_list); + + dev_dbg(dsp->dev, "block allocated %d:%d at offset 0x%x\n", + block->type, block->index, block->offset); + } list_splice(&tmp, &dsp->used_block_list); return 0; } -/* allocate free DSP blocks for module data - callers hold locks */ -static int block_alloc(struct sst_module *module, - struct sst_module_data *data) +/* allocate first free DSP blocks for data - callers hold locks */ +static int block_alloc(struct sst_dsp *dsp, struct sst_block_allocator *ba, + struct list_head *block_list) { - struct sst_dsp *dsp = module->dsp; struct sst_mem_block *block, *tmp; int ret = 0; - if (data->size == 0) + if (ba->size == 0) return 0; /* find first free whole blocks that can hold module */ list_for_each_entry_safe(block, tmp, &dsp->free_block_list, list) { /* ignore blocks with wrong type */ - if (block->type != data->type) + if (block->type != ba->type) continue; - if (data->size > block->size) + if (ba->size > block->size) continue; - data->offset = block->offset; - block->data_type = data->data_type; - block->bytes_used = data->size % block->size; - list_add(&block->module_list, &module->block_list); + ba->offset = block->offset; + block->bytes_used = ba->size % block->size; + list_add(&block->module_list, block_list); list_move(&block->list, &dsp->used_block_list); - dev_dbg(dsp->dev, " *module %d added block %d:%d\n", - module->id, block->type, block->index); + dev_dbg(dsp->dev, "block allocated %d:%d at offset 0x%x\n", + block->type, block->index, block->offset); return 0; } @@ -276,15 +384,19 @@ static int block_alloc(struct sst_module *module, list_for_each_entry_safe(block, tmp, &dsp->free_block_list, list) { /* ignore blocks with wrong type */ - if (block->type != data->type) + if (block->type != ba->type) continue; /* do we span > 1 blocks */ - if (data->size > block->size) { - ret = block_alloc_contiguous(module, data, - block->offset, data->size); + if (ba->size > block->size) { + + /* align ba to block boundary */ + ba->offset = block->offset; + + ret = block_alloc_contiguous(dsp, ba, block_list); if (ret == 0) return ret; + } } @@ -292,93 +404,74 @@ static int block_alloc(struct sst_module *module, return -ENOMEM; } -/* remove module from memory - callers hold locks */ -static void block_module_remove(struct sst_module *module) +int sst_alloc_blocks(struct sst_dsp *dsp, struct sst_block_allocator *ba, + struct list_head *block_list) { - struct sst_mem_block *block, *tmp; - struct sst_dsp *dsp = module->dsp; - int err; + int ret; - /* disable each block */ - list_for_each_entry(block, &module->block_list, module_list) { + dev_dbg(dsp->dev, "block request 0x%x bytes at offset 0x%x type %d\n", + ba->size, ba->offset, ba->type); - if (block->ops && block->ops->disable) { - err = block->ops->disable(block); - if (err < 0) - dev_err(dsp->dev, - "error: cant disable block %d:%d\n", - block->type, block->index); - } - } + mutex_lock(&dsp->mutex); - /* mark each block as free */ - list_for_each_entry_safe(block, tmp, &module->block_list, module_list) { - list_del(&block->module_list); - list_move(&block->list, &dsp->free_block_list); + ret = block_alloc(dsp, ba, block_list); + if (ret < 0) { + dev_err(dsp->dev, "error: can't alloc blocks %d\n", ret); + goto out; } -} - -/* prepare the memory block to receive data from host - callers hold locks */ -static int block_module_prepare(struct sst_module *module) -{ - struct sst_mem_block *block; - int ret = 0; - /* enable each block so that's it'e ready for module P/S data */ - list_for_each_entry(block, &module->block_list, module_list) { + /* prepare DSP blocks for module usage */ + ret = block_list_prepare(dsp, block_list); + if (ret < 0) + dev_err(dsp->dev, "error: prepare failed\n"); - if (block->ops && block->ops->enable) { - ret = block->ops->enable(block); - if (ret < 0) { - dev_err(module->dsp->dev, - "error: cant disable block %d:%d\n", - block->type, block->index); - goto err; - } - } - } +out: + mutex_unlock(&dsp->mutex); return ret; +} +EXPORT_SYMBOL_GPL(sst_alloc_blocks); -err: - list_for_each_entry(block, &module->block_list, module_list) { - if (block->ops && block->ops->disable) - block->ops->disable(block); - } - return ret; +int sst_free_blocks(struct sst_dsp *dsp, struct list_head *block_list) +{ + mutex_lock(&dsp->mutex); + block_list_remove(dsp, block_list); + mutex_unlock(&dsp->mutex); + return 0; } +EXPORT_SYMBOL_GPL(sst_free_blocks); /* allocate memory blocks for static module addresses - callers hold locks */ -static int block_alloc_fixed(struct sst_module *module, - struct sst_module_data *data) +static int block_alloc_fixed(struct sst_dsp *dsp, struct sst_block_allocator *ba, + struct list_head *block_list) { - struct sst_dsp *dsp = module->dsp; struct sst_mem_block *block, *tmp; - u32 end = data->offset + data->size, block_end; + u32 end = ba->offset + ba->size, block_end; int err; /* only IRAM/DRAM blocks are managed */ - if (data->type != SST_MEM_IRAM && data->type != SST_MEM_DRAM) + if (ba->type != SST_MEM_IRAM && ba->type != SST_MEM_DRAM) return 0; /* are blocks already attached to this module */ - list_for_each_entry_safe(block, tmp, &module->block_list, module_list) { + list_for_each_entry_safe(block, tmp, block_list, module_list) { - /* force compacting mem blocks of the same data_type */ - if (block->data_type != data->data_type) + /* ignore blocks with wrong type */ + if (block->type != ba->type) continue; block_end = block->offset + block->size; /* find block that holds section */ - if (data->offset >= block->offset && end < block_end) + if (ba->offset >= block->offset && end <= block_end) return 0; /* does block span more than 1 section */ - if (data->offset >= block->offset && data->offset < block_end) { + if (ba->offset >= block->offset && ba->offset < block_end) { - err = block_alloc_contiguous(module, data, - block->offset + block->size, - data->size - block->size); + /* align ba to block boundary */ + ba->size -= block_end - ba->offset; + ba->offset = block_end; + err = block_alloc_contiguous(dsp, ba, block_list); if (err < 0) return -ENOMEM; @@ -391,53 +484,67 @@ static int block_alloc_fixed(struct sst_module *module, list_for_each_entry_safe(block, tmp, &dsp->free_block_list, list) { block_end = block->offset + block->size; + /* ignore blocks with wrong type */ + if (block->type != ba->type) + continue; + /* find block that holds section */ - if (data->offset >= block->offset && end < block_end) { + if (ba->offset >= block->offset && end <= block_end) { /* add block */ - block->data_type = data->data_type; list_move(&block->list, &dsp->used_block_list); - list_add(&block->module_list, &module->block_list); + list_add(&block->module_list, block_list); + dev_dbg(dsp->dev, "block allocated %d:%d at offset 0x%x\n", + block->type, block->index, block->offset); return 0; } /* does block span more than 1 section */ - if (data->offset >= block->offset && data->offset < block_end) { + if (ba->offset >= block->offset && ba->offset < block_end) { + + /* align ba to block boundary */ + ba->offset = block->offset; - err = block_alloc_contiguous(module, data, - block->offset, data->size); + err = block_alloc_contiguous(dsp, ba, block_list); if (err < 0) return -ENOMEM; return 0; } - } return -ENOMEM; } /* Load fixed module data into DSP memory blocks */ -int sst_module_insert_fixed_block(struct sst_module *module, - struct sst_module_data *data) +int sst_module_alloc_blocks(struct sst_module *module) { struct sst_dsp *dsp = module->dsp; + struct sst_fw *sst_fw = module->sst_fw; + struct sst_block_allocator ba; int ret; + ba.size = module->size; + ba.type = module->type; + ba.offset = module->offset; + + dev_dbg(dsp->dev, "block request 0x%x bytes at offset 0x%x type %d\n", + ba.size, ba.offset, ba.type); + mutex_lock(&dsp->mutex); /* alloc blocks that includes this section */ - ret = block_alloc_fixed(module, data); + ret = block_alloc_fixed(dsp, &ba, &module->block_list); if (ret < 0) { dev_err(dsp->dev, "error: no free blocks for section at offset 0x%x size 0x%x\n", - data->offset, data->size); + module->offset, module->size); mutex_unlock(&dsp->mutex); return -ENOMEM; } /* prepare DSP blocks for module copy */ - ret = block_module_prepare(module); + ret = block_list_prepare(dsp, &module->block_list); if (ret < 0) { dev_err(dsp->dev, "error: fw module prepare failed\n"); goto err; @@ -450,19 +557,23 @@ int sst_module_insert_fixed_block(struct sst_module *module, return ret; err: - block_module_remove(module); + block_list_remove(dsp, &module->block_list); mutex_unlock(&dsp->mutex); return ret; } -EXPORT_SYMBOL_GPL(sst_module_insert_fixed_block); +EXPORT_SYMBOL_GPL(sst_module_alloc_blocks); /* Unload entire module from DSP memory */ -int sst_block_module_remove(struct sst_module *module) +int sst_module_free_blocks(struct sst_module *module) { struct sst_dsp *dsp = module->dsp; mutex_lock(&dsp->mutex); - block_module_remove(module); + block_list_remove(dsp, &module->block_list); + mutex_unlock(&dsp->mutex); + return 0; +} +EXPORT_SYMBOL_GPL(sst_module_free_blocks); mutex_unlock(&dsp->mutex); return 0; } diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c index 4b6c163..22cc697 100644 --- a/sound/soc/intel/sst-haswell-dsp.c +++ b/sound/soc/intel/sst-haswell-dsp.c @@ -86,9 +86,8 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, { struct dma_block_info *block; struct sst_module *mod; - struct sst_module_data block_data; struct sst_module_template template; - int count; + int count, ret; void __iomem *ram; /* TODO: allowed module types need to be configurable */ @@ -109,13 +108,9 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, memset(&template, 0, sizeof(template)); template.id = module->type; - template.entry = module->entry_point; - template.p.size = module->info.persistent_size; - template.p.type = SST_MEM_DRAM; - template.p.data_type = SST_DATA_P; - template.s.size = module->info.scratch_size; - template.s.type = SST_MEM_DRAM; - template.s.data_type = SST_DATA_S; + template.entry = module->entry_point - 4; + template.persistent_size = module->info.persistent_size; + template.scratch_size = module->info.scratch_size; mod = sst_module_new(fw, &template, NULL); if (mod == NULL) @@ -135,14 +130,14 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, switch (block->type) { case SST_HSW_IRAM: ram = dsp->addr.lpe; - block_data.offset = + mod->offset = block->ram_offset + dsp->addr.iram_offset; - block_data.type = SST_MEM_IRAM; + mod->type = SST_MEM_IRAM; break; case SST_HSW_DRAM: ram = dsp->addr.lpe; - block_data.offset = block->ram_offset; - block_data.type = SST_MEM_DRAM; + mod->offset = block->ram_offset; + mod->type = SST_MEM_DRAM; break; default: dev_err(dsp->dev, "error: bad type 0x%x for block 0x%x\n", @@ -151,20 +146,26 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, return -EINVAL; } - block_data.size = block->size; - block_data.data_type = SST_DATA_M; - block_data.data = (void *)block + sizeof(*block); - block_data.data_offset = block_data.data - fw->dma_buf; + mod->size = block->size; + mod->data = (void *)block + sizeof(*block); + mod->data_offset = mod->data - fw->dma_buf; - dev_dbg(dsp->dev, "copy firmware block %d type 0x%x " + dev_dbg(dsp->dev, "module block %d type 0x%x " "size 0x%x ==> ram %p offset 0x%x\n", - count, block->type, block->size, ram, + count, mod->type, block->size, ram, block->ram_offset); - sst_module_insert_fixed_block(mod, &block_data); + ret = sst_module_alloc_blocks(mod); + if (ret < 0) { + dev_err(dsp->dev, "error: could not allocate blocks for module %d\n", + count); + sst_module_free(mod); + return ret; + } block = (void *)block + sizeof(*block) + block->size; } + return 0; } -- 1.9.1

3 11

[alsa-devel] New SSID for Dell Headset mode
by Kailang 27 Oct '14

27 Oct '14

Hi Takashi, Dell has new machine. That SSID was 0x069f. It is the lite version of 0x0626. The audio layout was the same. Attach patch was add the SSID for the fixup table. Many Thanks. BR, Kailang

2 1

[alsa-devel] [PATCH 00/11] treewide: mask then shift defects and style updates
by Joe Perches 27 Oct '14

27 Oct '14

logical mask has lower precedence than shift but should be done before the shift so parentheses are generally required. And when masking with a fixed value after a shift, normal kernel style has the shift on the left, then the shift on the right so convert a few non-conforming uses. Joe Perches (11): block: nvme-scsi: Fix probable mask then right shift defects radeon: evergreen: Fix probable mask then right shift defects aiptek: Fix probable mask then right shift defects dvb-net: Fix probable mask then right shift defects cx25840/cx18: Use standard ordering of mask and shift wm8350-core: Fix probable mask then right shift defect iwlwifi: dvm: Fix probable mask then right shift defect ssb: driver_chip_comon_pmu: Fix probable mask then right shift defect tty: ipwireless: Fix probable mask then right shift defects hwa-hc: Fix probable mask then right shift defect sound: ad1889: Fix probable mask then right shift defects drivers/block/nvme-scsi.c | 12 ++++++------ drivers/gpu/drm/radeon/evergreen.c | 3 ++- drivers/input/tablet/aiptek.c | 6 +++--- drivers/media/dvb-core/dvb_net.c | 4 +++- drivers/media/i2c/cx25840/cx25840-core.c | 12 ++++++------ drivers/media/pci/cx18/cx18-av-core.c | 16 ++++++++-------- drivers/mfd/wm8350-core.c | 2 +- drivers/net/wireless/iwlwifi/dvm/lib.c | 4 ++-- drivers/ssb/driver_chipcommon_pmu.c | 4 ++-- drivers/tty/ipwireless/hardware.c | 12 ++++++------ drivers/usb/host/hwa-hc.c | 2 +- sound/pci/ad1889.c | 8 ++++---- 12 files changed, 44 insertions(+), 41 deletions(-) -- 2.1.2

2 2

[alsa-devel] [PATCH][fix] ALSA: bebob: fix wrong decoding of clock information for Terratec PHASE 88 Rack FW
by Takashi Sakamoto 27 Oct '14

27 Oct '14

Terratec PHASE 88 rack fw has two registers for source of clock, one is for internal/external, and another is for wordclock/spdif for external. When clock source is internal, information in another register has no meaning. Thus it must be ignored, but current implementation decodes it. This causes over-indexing reference to labels. Reported-by: András Murányi <muranyia(a)gmail.com> Tested-by: András Murányi <muranyia(a)gmail.com> Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/bebob/bebob_terratec.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/firewire/bebob/bebob_terratec.c b/sound/firewire/bebob/bebob_terratec.c index 0e4c0bf..9940611 100644 --- a/sound/firewire/bebob/bebob_terratec.c +++ b/sound/firewire/bebob/bebob_terratec.c @@ -24,7 +24,12 @@ phase88_rack_clk_src_get(struct snd_bebob *bebob, unsigned int *id) if (err < 0) goto end; - *id = (enable_ext & 0x01) | ((enable_word & 0x01) << 1); + if (enable_ext == 0) + *id = 0; + else if (enable_word == 0) + *id = 1; + else + *id = 2; end: return err; } -- 1.9.1

3 2

Re: [alsa-devel] ALSA - AD1893
by Raymond Yau 27 Oct '14

27 Oct '14

> I am working on Chromium and trying to get a machine with AD1983 to > function correctly. > > I'm on 3.14 kernel. My issue is that I need a way for CRAS (no pulse) > to tell HDA driver to switch capture from Front mic to rear Line/Mic. > The way alsamixer does it, but I want to add that functionality to Cras. > Could you point my to which IOCTL to call or alsa API to look at? You have to post the outpout of alsa-info.sh It depend on the pin default of the rear jack is line in or mic https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/Documenta… - line_in_auto_switch (bool): enable/disable the line-in auto-switch feature; default false - add_stereo_mix_input (bool): add the stereo mix (analog-loopback mix) to the input mux if available - add_jack_modes (bool): add "xxx Jack Mode" enum controls to each I/O jack for allowing to change the headphone amp and mic bias VREF capabilities https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… Stereo mix was already added to those desktop using VIA or ADI codec when auto-mic is not enabled Stereo mix also has input from DAC, line in and mic node 0x14 Record Selector can also select input from mono mixer node 0xf (mono mix) For those HDA codec with one ADC and most AC97 codecs amixer -c0 cset "Capture Source" "Line" Any reason to use codec specific ioctl since it won't work on other hda codec

1 0

[alsa-devel] [RFC][PATCH 00/15 v4] OXFW driver, a succesor to firewire-speakers
by Takashi Sakamoto 26 Oct '14

26 Oct '14

This patchset updates a part of my previous series: [alsa-devel] [RFC v3] [PATCH 00/52] Enhancement for support of firewire devices http://mailman.alsa-project.org/pipermail/alsa-devel/2014-January/071820.ht… This patchset is based on this series. (This series is under reviewing.) [alsa-devel] [PATCH 00/49 v4] Enhancement for support of Firewire devices http://mailman.alsa-project.org/pipermail/alsa-devel/2014-April/075841.html If you try this patchset, please apply the series in advance and apply this patchset. Or just use backport codes in my github repository: https://github.com/takaswie/snd-firewire-improve Or if you use Ubuntu 14.04, DKMS package is available from David's PPA. (Special thanks to David Henningsson.) https://launchpad.net/~diwic/+archive/snd-firewire-improve If you have Griffin FireWave, I would strongly request to test. Please see 'An issue for FireWave' in this message. == Background Currently, in ALSA, 'firewire-speakers' driver supports OXFW970/971 chipset which Oxford Semiconductor produced. But this driver has some restrictions: - Don't support MIDI substreams - Don't support substreams for PCM/MIDI capturing There are several devices which are based on OXFW970/971. Especially, devices which Behringer and Mackie produced support playback/capture PCM samples/MIDI messages. So firewire-speakers needs re-work to support these devices. This patchset renames 'firewire-speakers' to 'oxfw'. Maintainers already agreed to rename firewire-speakers to suitable name, in my first RFC in Jan 2014. == Supported devices Currently these devices are supported: * Griffin Firewave * LaCie Firewire Speakers * Behringer F-Control Audio 202 * Mackie(Loud) Onyx-i series (former models) * Mackie(Loud) Onyx Satellite * Mackie(Loud) Tapco Link.Firewire These devices can be supported but its IDs are unknown: * Mackie(Loud), d.2 pro * Mackie(Loud), d.4 pro * Mackie(Loud), U.420 * Mackie(Loud), U.420d When you have unknown devices, please inform me output of this command: $ hexdump -C /sys/bus/firewire/devices/fwX/config_rom (X is node id for your device. Tipically '1'.) I tested this driver with Behringer F-Control Audio 202. == Features - Capturing/Playbacking PCM samples/MIDI messages at supported sampling rates via ALSA PCM/MIDI interfaces - Retrieving firewire node information via ALSA hwdep interface - Locking/Unlocking kernel streaming via ALSA hwdep interface - Receiving notification at starting/stopping kernel streaming via ALSA hwdep interface == Solutions for reported regressions Two regressions were reported for my previous series: 1.Some names in card structure is trimmed. http://mailman.alsa-project.org/pipermail/alsa-devel/2014-February/073128.h… 2.The number of PCM channels are not detected correctly. http://mailman.alsa-project.org/pipermail/alsa-devel/2014-February/073145.h… The cause of issue 1 is that the device has too long names in its config rom. To solve this issue, for the driver, this patchset uses the same way which firewire-speakers uses, propper alias in device-specific structure. The cause of issue 2 is that the device has several format entries for the same sampling rate: http://mailman.alsa-project.org/pipermail/alsa-devel/2014-February/073561.h… To solve this issue, this patchset includes improved mechanism to detect stream formation. Additionally, this patchset adds new device entry for Mackie(Loud) Tapco Link.Firewire, refering to FFADO bug ticket 236: http://subversion.ffado.org/ticket/236 == An issue for FireWave I have an issue for Griffin Firewave. This device can handle AV/C Stream Format Information command and driver can get to know stream formations via the command. With Stefan's help, responses from FireWave against this command are clear. http://mailman.alsa-project.org/pipermail/alsa-devel/2014-February/073561.h… According to these responses, supported formations are: Input Stream to device: Rate PCM MIDI 32000 6 0 44100 6 0 48000 6 0 96000 6 0 32000 2 0 44100 2 0 48000 2 0 0 0 0 0 0 0 0 0 0 (I also expect to get this output from /proc/asound/FireWave/firewire/formation. I note it's order is not important.) But current 'firewire-speakers' has PCM constraints below: - 6ch at 32/44.1/48/96 kHz - 2ch at 48/96 kHz http://mailman.alsa-project.org/pipermail/alsa-devel/2014-February/073562.h… So this OXFW driver may lost a support for 2ch/96.0kHz for FireWave and gain 2ch/32.0kHz and 2ch/44.1kHz. I don't know exactly the reason of these gaps. So I strongly hope testers to check these two items: - FireWave really can handle 2ch/96kHz with firewire-speakers driver. - FireWave really can handle 2ch/32.0kHz and 2ch/44.1kHz with oxfw driver. I think a test with aplay is easy to execute: $ aplay -D hw:FireWave -f S32_LE -c 2 -r 96000 /dev/urandom $ aplay -D hw:FireWave -f S32_LE -c 2 -r 48000 /dev/urandom $ aplay -D hw:FireWave -f S32_LE -c 2 -r 44100 /dev/urandom $ aplay -D hw:FireWave -f S32_LE -c 2 -r 32000 /dev/urandom If 2ch/96kHz is apparently available, I'll change codes as a quirk. FYI, when I tested unavailable number of PCM channels with a hack, FCA202 playback PCM substreams but becomes freezed after stopping streaming. $ cat /var/log/syslog ... kernel: [10098.670390] snd_oxfw fw1.0: transaction failed: timeout ... I believe the other OXFW970/971 based devices have a similar behaviour. ================ Takashi Sakamoto (15): speakers: Rename to oxfw and rename some members oxfw: Move to its own directory oxfw: Split stream functionality to a new file and add a header file oxfw: Split PCM functionality to a new file oxfw: Split control functionality to a new file oxfw: Change the way to name card oxfw: Change the way to make PCM rules/constraints oxfw: Add proc interface for debugging purpose oxfw: Change the way to start stream oxfw: Add support for AV/C stream format command to get supported stream formation oxfw: Add a quirk for Griffin FireWave oxfw: Add support for Behringer/Mackie devices oxfw: Add support AMDTP in-stream and PCM capture oxfw: Add support for capture/playback MIDI messages oxfw: Add hwdep interface include/uapi/sound/asound.h | 3 +- include/uapi/sound/firewire.h | 3 +- sound/firewire/Kconfig | 18 +- sound/firewire/Makefile | 3 +- sound/firewire/oxfw/Makefile | 3 + sound/firewire/oxfw/oxfw.c | 284 +++++++++++++ sound/firewire/oxfw/oxfw.h | 141 +++++++ sound/firewire/oxfw/oxfw_command.c | 113 ++++++ sound/firewire/oxfw/oxfw_control.c | 283 +++++++++++++ sound/firewire/oxfw/oxfw_hwdep.c | 190 +++++++++ sound/firewire/oxfw/oxfw_midi.c | 168 ++++++++ sound/firewire/oxfw/oxfw_pcm.c | 367 +++++++++++++++++ sound/firewire/oxfw/oxfw_proc.c | 83 ++++ sound/firewire/oxfw/oxfw_stream.c | 658 ++++++++++++++++++++++++++++++ sound/firewire/speakers.c | 792 ------------------------------------- 15 files changed, 2308 insertions(+), 801 deletions(-) create mode 100644 sound/firewire/oxfw/Makefile create mode 100644 sound/firewire/oxfw/oxfw.c create mode 100644 sound/firewire/oxfw/oxfw.h create mode 100644 sound/firewire/oxfw/oxfw_command.c create mode 100644 sound/firewire/oxfw/oxfw_control.c create mode 100644 sound/firewire/oxfw/oxfw_hwdep.c create mode 100644 sound/firewire/oxfw/oxfw_midi.c create mode 100644 sound/firewire/oxfw/oxfw_pcm.c create mode 100644 sound/firewire/oxfw/oxfw_proc.c create mode 100644 sound/firewire/oxfw/oxfw_stream.c delete mode 100644 sound/firewire/speakers.c -- 1.8.3.2

3 34

[alsa-devel] Request for comments for SND_SOC_IMX_WM8731
by Jonathan Bennett 26 Oct '14

26 Oct '14

This patch is a continuation of my work on supporting the Utilite devices on the vanilla kernel. I've modified the code slightly to conform to established practice. Namely instead of using src-port and ext-port, we use mux-int-port and mux-ext-port. This driver has support for putting the codec in either master mode or slave mode. So far I've only had success with using slave mode. For trying to submit for kernel inclusion, should I strip out the code for codec master mode? As a kernel dev noob, any other comments are welcome. Thanks, Jonathan Bennett diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 081e406..09c923f 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -229,6 +229,17 @@ config SND_SOC_EUKREA_TLV320 Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface +config SND_SOC_IMX_WM8731 + tristate "SoC Audio support for i.MX boards with wm8731 + depends on OF && I2C + select SND_SOC_WM8731 + select SND_SOC_IMX_PCM_DMA + select SND_SOC_IMX_AUDMUX + select SND_SOC_FSL_SSI + select SND_SOC_FSL_UTILS + help + Soc Audio support for i.MX with wm8731 + config SND_SOC_IMX_WM8962 tristate "SoC Audio support for i.MX boards with wm8962" depends on OF && I2C && INPUT diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index d28dc25..4292fac 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -54,6 +54,7 @@ snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o snd-soc-imx-es8328-objs := imx-es8328.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o +snd-soc-imx-wm8731-objs := imx-wm8731.o snd-soc-imx-wm8962-objs := imx-wm8962.o snd-soc-imx-spdif-objs := imx-spdif.o snd-soc-imx-mc13783-objs := imx-mc13783.o @@ -65,5 +66,6 @@ obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o obj-$(CONFIG_SND_SOC_IMX_ES8328) += snd-soc-imx-es8328.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o +obj-$(CONFIG_SND_SOC_IMX_WM8731) += snd-soc-imx-wm8731.o obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o diff --git a/sound/soc/fsl/imx-wm8731.c b/sound/soc/fsl/imx-wm8731.c new file mode 100644 index 0000000..8ce4e9f --- /dev/null +++ b/sound/soc/fsl/imx-wm8731.c @@ -0,0 +1,687 @@ +/* + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Based on imx-sgtl5000.c + * Copyright (C) 2012 Freescale Semiconductor, Inc. + * Copyright (C) 2012 Linaro Ltd. + * + * The code contained herein is licensed under the GNU General Public + * License. You may obtain a copy of the GNU General Public License + * Version 2 or later at the following locations: + * + * http://www.opensource.org/licenses/gpl-license.html + * http://www.gnu.org/copyleft/gpl.html + */ + +#include <linux/module.h> +#include <linux/of_platform.h> +#include <linux/i2c.h> +#include <linux/of_gpio.h> +#include <linux/slab.h> +#include <linux/gpio.h> +#include <linux/clk.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <sound/pcm_params.h> +#include <sound/soc-dapm.h> +#include <linux/pinctrl/consumer.h> + +#include "../codecs/wm8731.h" +#include "imx-audmux.h" +#include "imx-ssi.h" + +#define DAI_NAME_SIZE 32 +#define WM8731_MCLK_FREQ (24000000 / 2) + +struct imx_wm8731_data { + struct snd_soc_dai_link dai; + struct snd_soc_card card; + char codec_dai_name[DAI_NAME_SIZE]; + char platform_name[DAI_NAME_SIZE]; + struct i2c_client *codec_dev; + /* audio_clocking_data */ + struct clk *pll; + struct clk *clock_root; + long sysclk; + long current_rate; + /* apis */ + int (*clock_enable)(int enable,struct imx_wm8731_data *data); +}; + +static int imx_wm8731_init(struct snd_soc_pcm_runtime *rtd); +static int imx_hifi_hw_params_slv_mode(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params); +static void imx_hifi_shutdown(struct snd_pcm_substream *substream); + +struct imx_priv { + struct platform_device *pdev; + struct imx_wm8731_data *data; +}; + +static struct imx_priv card_priv; + +static struct snd_soc_ops imx_hifi_ops = { + .shutdown = imx_hifi_shutdown, +}; + +/* imx card dapm widgets */ +static const struct snd_soc_dapm_widget imx_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_LINE("Line Jack", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), +}; + +/* imx machine connections to the codec pins */ +static const struct snd_soc_dapm_route audio_map[] = { + { "Headphone Jack", NULL, "LHPOUT" }, + { "Headphone Jack", NULL, "RHPOUT" }, + + { "Ext Spk", NULL, "LOUT" }, + { "Ext Spk", NULL, "ROUT" }, + + { "LLINEIN", NULL, "Line Jack" }, + { "RLINEIN", NULL, "Line Jack" }, + + { "MICIN", NULL, "Mic Bias" }, + { "Mic Bias", NULL, "Mic Jack"}, +}; + +static int wm8731_slv_mode_init(struct imx_wm8731_data *data) +{ + struct clk *new_parent; + struct clk *ssi_clk; + struct i2c_client *codec_dev = data->codec_dev; + + new_parent = devm_clk_get(&codec_dev->dev, "pll4"); + if (IS_ERR(new_parent)) { + pr_err("Could not get \"pll4\" clock \n"); + return PTR_ERR(new_parent); + } + + ssi_clk = devm_clk_get(&codec_dev->dev, "imx-ssi.1"); + if (IS_ERR(ssi_clk)) { + pr_err("Could not get \"imx-ssi.1\" clock \n"); + return PTR_ERR(ssi_clk); + } + + clk_set_parent(ssi_clk, new_parent); + + data->pll = new_parent; + data->clock_root = ssi_clk; + data->current_rate = 0; + + data->sysclk = 0; + + return 0; +} + +static int wm8731_slv_mode_clock_enable(int enable, struct imx_wm8731_data *data) +{ + long pll_rate; + long rate_req; + long rate_avail; + + if (!enable) + return 0; + + if (data->sysclk == data->current_rate) + return 0; + + switch (data->sysclk) { + case 11289600: + pll_rate = 632217600; + break; + + case 12288000: + pll_rate = 688128000; + break; + + default: + return -EINVAL; + } + + rate_req = pll_rate; + rate_avail = clk_round_rate(data->pll, rate_req); + clk_set_rate(data->pll, rate_avail); + + rate_req = data->sysclk; + rate_avail = clk_round_rate(data->clock_root, + rate_req); + clk_set_rate(data->clock_root, rate_avail); + + pr_info("%s: \"imx-ssi.1\" rate = %ld (= %ld)\n", + __func__, rate_avail, rate_req); + + data->current_rate = data->sysclk; + + return 0; +} + +static int imx_hifi_startup_slv_mode(struct snd_pcm_substream *substream) +{ + /* + * As SSI's sys clock rate depends on sampling rate, + * the clock enabling code is moved to imx_hifi_hw_params(). + */ + return 0; +} + +static int wm8731_mst_mode_init(struct imx_wm8731_data *data) +{ + long rate; + struct clk *new_parent; + struct clk *ssi_clk; + struct i2c_client *codec_dev = data->codec_dev; + + new_parent = devm_clk_get(&codec_dev->dev, "cko2"); + if (IS_ERR(new_parent)) { + pr_err("Could not get \"cko2\" clock \n"); + return PTR_ERR(new_parent); + } + + ssi_clk = devm_clk_get(&codec_dev->dev, "cko"); + if (IS_ERR(ssi_clk)) { + pr_err("Could not get \"cko\" clock \n"); + return PTR_ERR(ssi_clk); + } + + rate = clk_round_rate(new_parent, WM8731_MCLK_FREQ); + clk_set_rate(new_parent, rate); + + clk_set_parent(ssi_clk, new_parent); + + rate = clk_round_rate(ssi_clk, WM8731_MCLK_FREQ); + clk_set_rate(ssi_clk, rate); + + pr_info("%s: \"CLKO\" rate = %ld (= %d)\n", + __func__, rate, WM8731_MCLK_FREQ); + + data->pll = new_parent; + data->clock_root = ssi_clk; + data->sysclk = rate; + + return 0; +} + +static int wm8731_mst_mode_clock_enable(int enable, struct imx_wm8731_data *data) +{ + struct clk *clko = data->clock_root; + + if (enable) + clk_enable(clko); + else + clk_disable(clko); + + return 0; +} + +static int imx_hifi_startup_mst_mode(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_card *card = codec_dai->card; + struct imx_wm8731_data *data = snd_soc_card_get_drvdata(card); + + if (!codec_dai->active) + data->clock_enable(1,data); + + return 0; +} + + +static int imx_hifi_hw_params_slv_mode(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_card *card = codec_dai->card; + struct imx_wm8731_data *data = snd_soc_card_get_drvdata(card); + + u32 dai_format; + snd_pcm_format_t sample_format; + unsigned int channels; + unsigned int tx_mask, rx_mask; + unsigned int sampling_rate; + unsigned int div_2, div_psr, div_pm; + int ret; + + sampling_rate = params_rate(params); + sample_format = params_format(params); + + channels = params_channels(params); + printk("%s:%s sampling rate = %u channels = %u \n", __FUNCTION__, + (substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? "Playback" : "Capture"), + sampling_rate, channels); + + /* set CPU DAI configuration */ + switch (sampling_rate) { + case 8000: + case 32000: + case 48000: + case 96000: + data->sysclk = 12288000; + break; + + case 44100: + case 88200: + data->sysclk = 11289600; + break; + + default: + return -EINVAL; + } + + wm8731_slv_mode_clock_enable(1,data); + + dai_format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBS_CFS; + + ret = snd_soc_dai_set_fmt(cpu_dai, dai_format); + if (ret < 0) + return ret; + + /* set i.MX active slot mask */ + /* S[TR]CCR:DC */ + tx_mask = ~((1 << channels) - 1); + rx_mask = tx_mask; + snd_soc_dai_set_tdm_slot(cpu_dai, tx_mask, rx_mask, 2, 32); + + /* + * SSI sysclk divider: + * div_2: /1 or /2 + * div_psr: /1 or /8 + * div_pm: /1 .. /256 + */ + div_2 = 0; + div_psr = 0; + switch (sampling_rate) { + case 8000: + // 1x1x12 + div_pm = 11; + break; + case 32000: + // 1x1x3 + div_pm = 2; + break; + case 48000: + // 1x1x2 + div_pm = 1; + break; + case 96000: + // 1x1x1 + div_pm = 0; + break; + case 44100: + // 1x1x2 + div_pm = 1; + break; + case 88200: + // 1x1x1 + div_pm = 0; + break; + default: + return -EINVAL; + } + + /* sync mode: a single clock controls both playback and capture */ + snd_soc_dai_set_clkdiv(cpu_dai, IMX_SSI_TX_DIV_2, (div_2 ? SSI_STCCR_DIV2 : 0)); + snd_soc_dai_set_clkdiv(cpu_dai, IMX_SSI_TX_DIV_PSR, (div_psr ? SSI_STCCR_PSR : 0)); + snd_soc_dai_set_clkdiv(cpu_dai, IMX_SSI_TX_DIV_PM, div_pm); + + /* set codec DAI configuration */ + dai_format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS; + + ret = snd_soc_dai_set_fmt(codec_dai, dai_format); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, + WM8731_SYSCLK_MCLK, + data->sysclk, + SND_SOC_CLOCK_IN); + + if (ret < 0) { + pr_err("Failed to set codec master clock to %u: %d \n", + data->sysclk, ret); + return ret; + } + + return 0; +} + +static int imx_hifi_hw_params_mst_mode(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_card *card = codec_dai->card; + struct imx_wm8731_data *data = snd_soc_card_get_drvdata(card); + u32 dai_format; + unsigned int channels; + unsigned int tx_mask, rx_mask; + unsigned int sampling_rate; + int ret; + + + sampling_rate = params_rate(params); + channels = params_channels(params); + pr_debug("%s:%s sampling rate = %u channels = %u \n", __FUNCTION__, + (substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? "Playback" : "Capture"), + sampling_rate, channels); + + /* set cpu DAI configuration */ + dai_format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBM_CFM; + + ret = snd_soc_dai_set_fmt(cpu_dai, dai_format); + if (ret < 0) + return ret; + + /* set i.MX active slot mask */ + /* S[TR]CCR:DC */ + tx_mask = ~((1 << channels) - 1); + rx_mask = tx_mask; + snd_soc_dai_set_tdm_slot(cpu_dai, tx_mask, rx_mask, 2, 32); + + /* set codec DAI configuration */ + dai_format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM; + + ret = snd_soc_dai_set_fmt(codec_dai, dai_format); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, + WM8731_SYSCLK_MCLK, + data->sysclk, + SND_SOC_CLOCK_IN); + + if (ret < 0) { + pr_err("Failed to set codec master clock to %u: %d \n", + data->sysclk, ret); + return ret; + } + + return 0; +} + +static void imx_hifi_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_card *card = codec_dai->card; + struct imx_wm8731_data *data = snd_soc_card_get_drvdata(card); + + if (!codec_dai->active) + data->clock_enable(0,data); + + return; +} + +static int imx_wm8731_init(struct snd_soc_pcm_runtime *rtd) +{ + int ret = 0; + struct snd_soc_codec *codec = rtd->codec; + + /* Add imx specific widgets */ + ret = snd_soc_dapm_new_controls(&codec->dapm, imx_dapm_widgets, + ARRAY_SIZE(imx_dapm_widgets)); + if (ret) + goto out_retcode; + + /* Set up imx specific audio path audio_map */ + ret = snd_soc_dapm_add_routes(&codec->dapm, audio_map, ARRAY_SIZE(audio_map)); + if (ret) + goto out_retcode; + + ret = snd_soc_dapm_enable_pin(&codec->dapm, "Headphone Jack"); + if (ret) + goto out_retcode; + + ret = snd_soc_dapm_nc_pin(&codec->dapm, "Ext Spk"); + if (ret) + goto out_retcode; + +out_retcode: + + if (ret) + pr_err("%s: failed with error code: %d \n", __FUNCTION__, ret); + else + pr_info("%s: success \n", __FUNCTION__); + + return ret; +} + +/** + * Configure AUDMUX interconnection between + * _slave (CPU side) and _master (codec size) + * + * When SSI operates in master mode, 5-wire interconnect with + * audio codec is required: + * TXC - BCLK + * TXD - DAC data + * RXD - ADC data + * TXFS - {DAC|ADC}LRC, i.e. word clock + * RXC - MCLK, i.e. oversampling clock + * Audmux is operated in asynchronous mode to enable 6-wire + * interface (as opposed to 4-wire interface in sync mode). + */ +static int imx_audmux_config_slv_mode(int _slave, int _master) +{ + unsigned int ptcr, pdcr; + int slave = _slave - 1; + int master = _master - 1; + + ptcr = IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TFSDIR | + IMX_AUDMUX_V2_PTCR_TFSEL(slave) | + IMX_AUDMUX_V2_PTCR_RCLKDIR | + IMX_AUDMUX_V2_PTCR_RCSEL(slave | 0x8) | + IMX_AUDMUX_V2_PTCR_TCLKDIR | + IMX_AUDMUX_V2_PTCR_TCSEL(slave); + + pdcr = IMX_AUDMUX_V2_PDCR_RXDSEL(slave); + imx_audmux_v2_configure_port(master, ptcr, pdcr); + ptcr = ptcr & ~IMX_AUDMUX_V2_PTCR_SYN; + imx_audmux_v2_configure_port(master, ptcr, pdcr); + + ptcr = IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_RCLKDIR | + IMX_AUDMUX_V2_PTCR_RCSEL(master | 0x8) | + IMX_AUDMUX_V2_PTCR_TCLKDIR | + IMX_AUDMUX_V2_PTCR_TCSEL(master); + + pdcr = IMX_AUDMUX_V2_PDCR_RXDSEL(master); + imx_audmux_v2_configure_port(slave, ptcr, pdcr); + ptcr = ptcr & ~IMX_AUDMUX_V2_PTCR_SYN; + imx_audmux_v2_configure_port(slave, ptcr, pdcr); + + return 0; +} + +static int imx_audmux_config_mst_mode(int _slave, int _master) +{ + unsigned int ptcr, pdcr; + int slave = _slave - 1; + int master = _master - 1; + + ptcr = IMX_AUDMUX_V2_PTCR_SYN; + ptcr |= IMX_AUDMUX_V2_PTCR_TFSDIR | + IMX_AUDMUX_V2_PTCR_TFSEL(master) | + IMX_AUDMUX_V2_PTCR_TCLKDIR | + IMX_AUDMUX_V2_PTCR_TCSEL(master); + pdcr = IMX_AUDMUX_V2_PDCR_RXDSEL(master); + imx_audmux_v2_configure_port(slave, ptcr, pdcr); + + ptcr = IMX_AUDMUX_V2_PTCR_SYN; + pdcr = IMX_AUDMUX_V2_PDCR_RXDSEL(slave); + imx_audmux_v2_configure_port(master, ptcr, pdcr); + + return 0; +} + +static int imx_wm8731_probe(struct platform_device *pdev) +{ + struct device_node *ssi_np, *codec_np; + struct platform_device *ssi_pdev; + struct imx_priv *priv = &card_priv; + struct i2c_client *codec_dev; + struct imx_wm8731_data *data; + unsigned int src_port, ext_port; + unsigned int ssi_mode; + const char *ssi_mode_str; + + int ret; + + priv->pdev = pdev; + + ssi_np = of_parse_phandle(pdev->dev.of_node, "ssi-controller", 0); + codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0); + if (!ssi_np || !codec_np) { + dev_err(&pdev->dev, "phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + ssi_pdev = of_find_device_by_node(ssi_np); + if (!ssi_pdev) { + dev_err(&pdev->dev, "failed to find SSI platform device\n"); + ret = -EINVAL; + goto fail; + } + + codec_dev = of_find_i2c_device_by_node(codec_np); + if (!codec_dev || !codec_dev->dev.driver) { + dev_err(&pdev->dev, "failed to find codec platform device\n"); + ret = -EINVAL; + goto fail; + } + + data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); + if (!data) { + ret = -ENOMEM; + goto fail; + } + + card_priv.data = data; + + data->codec_dev = codec_dev; + + data->dai.name = "HiFi"; + data->dai.stream_name = "HiFi"; + data->dai.codec_dai_name = "wm8731-hifi"; + data->dai.codec_of_node = codec_np; + data->dai.cpu_dai_name = dev_name(&ssi_pdev->dev); + data->dai.platform_of_node = ssi_np; + data->dai.ops = &imx_hifi_ops; + data->dai.init = &imx_wm8731_init; + + ret = of_property_read_u32(pdev->dev.of_node, "mux-int-port", &src_port); + if (ret) { + dev_err(&pdev->dev, "failed to get \"mux-int-port\" value\n"); + ret = -EINVAL; + goto fail; + } + + ret = of_property_read_u32(pdev->dev.of_node, "mux-ext-port", &ext_port); + if (ret) { + dev_err(&pdev->dev, "failed to get \"mux-ext-port\" value\n"); + ret = -EINVAL; + goto fail; + } + + ret = of_property_read_string(ssi_np, "fsl,mode", &ssi_mode_str); + if (ret) { + dev_err(&pdev->dev, "failed to get \"fsl,mode\" value\n"); + ret = -EINVAL; + goto fail; + } + + ssi_mode = strcmp(ssi_mode_str, "i2s-master"); + + if (ssi_mode) { + /* Master Mode */ + imx_audmux_config_mst_mode(src_port, ext_port); + wm8731_mst_mode_init(data); + data->clock_enable = wm8731_mst_mode_clock_enable; + imx_hifi_ops.hw_params = imx_hifi_hw_params_mst_mode; + imx_hifi_ops.startup = imx_hifi_startup_mst_mode; + } else { + /* Slave Mode */ + imx_audmux_config_slv_mode(src_port, ext_port); + wm8731_slv_mode_init(data); + data->clock_enable = wm8731_slv_mode_clock_enable; + imx_hifi_ops.hw_params = imx_hifi_hw_params_slv_mode; + imx_hifi_ops.startup = imx_hifi_startup_slv_mode; + } + + data->card.dev = &pdev->dev; + ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) + goto fail; + + ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); + if (ret) + goto fail; + + data->card.num_links = 1; + data->card.dai_link = &data->dai; + + data->card.dapm_widgets = imx_dapm_widgets; + data->card.num_dapm_widgets = ARRAY_SIZE(imx_dapm_widgets); + + platform_set_drvdata(pdev, &data->card); + snd_soc_card_set_drvdata(&data->card, data); + + ret = devm_snd_soc_register_card(&pdev->dev, &data->card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + goto fail; + } +/* of_node_put(ssi_np); + of_node_put(codec_np); */ + return 0; +fail: + + if (ssi_np) + of_node_put(ssi_np); + + if (codec_np) + of_node_put(codec_np); + + return ret; +} + +static int imx_wm8731_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + snd_soc_unregister_card(card); + + return 0; +} + +static const struct of_device_id imx_wm8731_dt_ids[] = { + { .compatible = "fsl,imx-audio-wm8731", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_wm8731_dt_ids); + +static struct platform_driver imx_wm8731_driver = { + .driver = { + .name = "imx-wm8731", + .owner = THIS_MODULE, + .of_match_table = imx_wm8731_dt_ids, + }, + .probe = imx_wm8731_probe, + .remove = imx_wm8731_remove, +}; +module_platform_driver(imx_wm8731_driver); + +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("Freescale i.MX WM8731 ASoC machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:imx-wm8731");

4 10

[alsa-devel] [PATCH] ASoC: rt5645:Mark RT5645_TDM_CTRL_3 as readable
by yang.a.fang＠intel.com 26 Oct '14

26 Oct '14

From: "Fang, Yang A" <yang.a.fang(a)intel.com> amixer query fails due to it is not readable reigster Signed-off-by: Fang, Yang A <yang.a.fang(a)intel.com> --- sound/soc/codecs/rt5645.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 3fb83bf..d16331e 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -139,6 +139,7 @@ static const struct reg_default rt5645_reg[] = { { 0x76, 0x000a }, { 0x77, 0x0c00 }, { 0x78, 0x0000 }, + { 0x79, 0x0123 }, { 0x80, 0x0000 }, { 0x81, 0x0000 }, { 0x82, 0x0000 }, @@ -334,6 +335,7 @@ static bool rt5645_readable_register(struct device *dev, unsigned int reg) case RT5645_DMIC_CTRL2: case RT5645_TDM_CTRL_1: case RT5645_TDM_CTRL_2: + case RT5645_TDM_CTRL_3: case RT5645_GLB_CLK: case RT5645_PLL_CTRL1: case RT5645_PLL_CTRL2: -- 1.7.9.5

1 0

[alsa-devel] [PATCH 0/2] Disable the ThinkPad HW mute/level control when possible
by Andy Lutomirski 25 Oct '14

25 Oct '14

This series is being resurrected after several years, and it's quite different from before, so I'm restarting the version number at 1 :) ThinkPad volume and mute controls are a mess. For whatever reason, ThinkPads have mute buttons that send KEY_MUTE *and* control an invisible-to-ALSA mute switch. Some of them have volume controls that interact with this switch as well. This is a perennial source of problems. On most ThinkPads, if you press mute and then unmute using GUI controls, you have no sound, because userspace and ALSA state gets out of sync with the hardware switch. There's a separate "sound card" that exposes the hardware switch, but userspace code generally doesn't understand that. There are already a few _OSI(Linux) overrides to turn all the hardware buttons into regular buttons. Rather than quirking ACPI everywhere, just teach thinkpad-acpi to program the buttons for full software control and to disable hardware controls. That allows us to remove the ACPI quirks and have normal mute controls. This approach should be much simpler than adding even more kludgey ALSA integration for questionable gain. I haven't tested on my X220 (which has a mute light in addition to a mute button), since that laptop's SATA controller died last week. It should work, though, and if someone else could test this on a ThinkPad with a mute light, that would be great. Andy Lutomirski (2): thinkpad-acpi: Try to use full software mute control acpi: Remove _OSI(Linux) for ThinkPads drivers/acpi/blacklist.c | 54 ------------------ drivers/platform/x86/thinkpad_acpi.c | 107 +++++++++++++++++++++++++++++++---- 2 files changed, 97 insertions(+), 64 deletions(-) -- 1.9.3

2 10

[alsa-devel] Recording with asus xonar essence st card
by Vasily Postnicov 25 Oct '14

25 Oct '14

Hello. This is not related to alsa, but I decided to write this message here, because as far as I know alsa has the best open source driver for asus xonar series. I am using DragonFlyBSD and a bit modified driver originally written for FreeBSD by Alexander Polakov https://github.com/shamazmazum/xonar-dragonfly It lacks support for recording. I am trying to add this functionality from alsa/OSS drivers, but I have some unanswered questions. There are they here: 1) How do I initialize my soundcard for recording properly? I think I must enable CS5381 ADC using GPIO registers, but what are OXYGEN_REC_ROUTING, OXYGEN_ADC_MONITOR and OXYGEN_A_MONITOR_ROUTING for? Does MONITOR mean the same thing as in the user manual? If ROUTING registers are just switches for source of a signal to be recorded, is it enough to use OXYGEN_REC_B_ROUTE_I2S_ADC_2 for ST card to record from line-in/mic jack? 2) How do I control volume of sound while recording? Have Xonar Essence cards any controls for this? I cannot find it out looking at oxygen_mixer.c

2 1

[alsa-devel] [PATCH] ASoC: rt5645:Mark RT5645_TDM_CTRL_3 as readable
by yang.a.fang＠intel.com 25 Oct '14

25 Oct '14

From: "Fang, Yang A" <yang.a.fang(a)intel.com> amixer query fails due to it is not readable reigster Signed-off-by: Fang, Yang A <yang.a.fang(a)intel.com> --- sound/soc/codecs/rt5645.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 3fb83bf..d16331e 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -139,6 +139,7 @@ static const struct reg_default rt5645_reg[] = { { 0x76, 0x000a }, { 0x77, 0x0c00 }, { 0x78, 0x0000 }, + { 0x79, 0x0123 }, { 0x80, 0x0000 }, { 0x81, 0x0000 }, { 0x82, 0x0000 }, @@ -334,6 +335,7 @@ static bool rt5645_readable_register(struct device *dev, unsigned int reg) case RT5645_DMIC_CTRL2: case RT5645_TDM_CTRL_1: case RT5645_TDM_CTRL_2: + case RT5645_TDM_CTRL_3: case RT5645_GLB_CLK: case RT5645_PLL_CTRL1: case RT5645_PLL_CTRL2: -- 1.7.9.5

1 0

[alsa-devel] [PATCH] ALSA: pcm: Avoid mmap warnings on x86
by Takashi Iwai 24 Oct '14

24 Oct '14

On x86, using dma_mmap_coherent() for the pages allocated via dma_alloc_coherent() results in a warning like: aplay:32536 map pfn RAM range req uncached-minus for [mem 0x21d500000-0x21d51ffff], got write-back Until the issue is addressed in the core side, take back to the old good way in PCM code only for x86. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_native.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index a915ed07aee1..2f7ad10ee7c3 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3324,6 +3324,7 @@ int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream, area->vm_end - area->vm_start, area->vm_page_prot); } #endif /* CONFIG_GENERIC_ALLOCATOR */ +#ifndef CONFIG_X86 /* for avoiding warnings arch/x86/mm/pat.c */ if (!substream->ops->page && substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) return dma_mmap_coherent(substream->dma_buffer.dev.dev, @@ -3331,6 +3332,7 @@ int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream, substream->runtime->dma_area, substream->runtime->dma_addr, area->vm_end - area->vm_start); +#endif /* CONFIG_X86 */ /* mmap with fault handler */ area->vm_ops = &snd_pcm_vm_ops_data_fault; return 0; -- 2.1.2

1 0

Re: [alsa-devel] Default value change for restore default in initial state
by Takashi Iwai 24 Oct '14

24 Oct '14

At Fri, 24 Oct 2014 08:35:12 +0000, Kailang wrote: > > Hi Takashi, > > We have some changed for restore default for ALC282 and ALC283. Could you give a bit more information, what these updates actually fix? thanks, Takashi

1 0

[alsa-devel] [PATCH] ALSA: hda - Add missing terminating entry to SND_HDA_PIN_QUIRK macro
by David Henningsson 24 Oct '14

24 Oct '14

Without this terminating entry, the pin matching would continue across random memory until a zero or a non-matching entry was found. The result being that in some cases, the pin quirk would not be applied correctly. Cc: stable(a)vger.kernel.org Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_local.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 7eb44e7..62658f2 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -419,7 +419,7 @@ struct snd_hda_pin_quirk { .subvendor = _subvendor,\ .name = _name,\ .value = _value,\ - .pins = (const struct hda_pintbl[]) { _pins } \ + .pins = (const struct hda_pintbl[]) { _pins, {0, 0}} \ } #else @@ -427,7 +427,7 @@ struct snd_hda_pin_quirk { { .codec = _codec,\ .subvendor = _subvendor,\ .value = _value,\ - .pins = (const struct hda_pintbl[]) { _pins } \ + .pins = (const struct hda_pintbl[]) { _pins, {0, 0}} \ } #endif -- 1.9.1

2 1

[alsa-devel] [PATCH] hda-emu: Add snd_ctl_enum_info
by David Henningsson 24 Oct '14

24 Oct '14

Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- include/sound/control.h | 2 ++ snd-control.c | 18 ++++++++++++++++++ 2 files changed, 20 insertions(+) diff --git a/include/sound/control.h b/include/sound/control.h index 47b92c5..eaaada4 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -101,6 +101,8 @@ int snd_ctl_boolean_mono_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); int snd_ctl_boolean_stereo_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); +int snd_ctl_enum_info(struct snd_ctl_elem_info *info, unsigned int channels, + unsigned int items, const char *const names[]); /* * virtual master control diff --git a/snd-control.c b/snd-control.c index 3e3686f..a4a7a14 100644 --- a/snd-control.c +++ b/snd-control.c @@ -153,6 +153,24 @@ int snd_ctl_boolean_stereo_info(struct snd_kcontrol *kcontrol, } #endif +int snd_ctl_enum_info(struct snd_ctl_elem_info *info, unsigned int channels, + unsigned int items, const char *const names[]) +{ + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + info->count = channels; + info->value.enumerated.items = items; + if (!items) + return 0; + if (info->value.enumerated.item >= items) + info->value.enumerated.item = items - 1; + if (strlen(names[info->value.enumerated.item]) >= sizeof(info->value.enumerated.name)) + hda_log(HDA_LOG_WARN, "ALSA: too long item name '%s'\n", names[info->value.enumerated.item]); + strlcpy(info->value.enumerated.name, + names[info->value.enumerated.item], + sizeof(info->value.enumerated.name)); + return 0; +} + int snd_ctl_activate_id(struct snd_card *card, struct snd_ctl_elem_id *id, int active) { -- 1.9.1

2 1

[alsa-devel] ALSA HDMI output not detected
by samuelchemla＠orange.fr 23 Oct '14

23 Oct '14

Hi guys, I opened a bug about HDMI sound output not detected on an NVIDIA optimus laptop. I don't know if HDMI sound card detection is up to ALSA, the kernel or the video driver. Could anyone give me a clue so the bug can be routed to its correct destination? Link to the bug: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1377653 Regards, Sam

2 2

[alsa-devel] [PATCH] ASoC: jack: update calls to gpiod_get*()
by Alexandre Courbot 23 Oct '14

23 Oct '14

Add the new flags argument to calls of (devm_)gpiod_get*() and remove any direction setting code afterwards. Currently both forms (with or without the flags argument) are valid thanks to transitional macros in <linux/gpio/consumer.h>. These macros will be removed once all consumers are updated and the flags argument will become compulsary. Signed-off-by: Alexandre Courbot <acourbot(a)nvidia.com> --- sound/soc/soc-jack.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index ab47fea997a3..f921d0098518 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -309,7 +309,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, /* GPIO descriptor */ gpios[i].desc = gpiod_get_index(gpios[i].gpiod_dev, gpios[i].name, - gpios[i].idx); + gpios[i].idx, GPIOD_IN); if (IS_ERR(gpios[i].desc)) { ret = PTR_ERR(gpios[i].desc); dev_err(gpios[i].gpiod_dev, @@ -327,17 +327,14 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, goto undo; } - ret = gpio_request(gpios[i].gpio, gpios[i].name); + ret = gpio_request_one(gpios[i].gpio, GPIOF_IN, + gpios[i].name); if (ret) goto undo; gpios[i].desc = gpio_to_desc(gpios[i].gpio); } - ret = gpiod_direction_input(gpios[i].desc); - if (ret) - goto err; - INIT_DELAYED_WORK(&gpios[i].work, gpio_work); gpios[i].jack = jack; -- 2.1.2

2 1

Re: [alsa-devel] [pulseaudio-discuss] Problem with timing and USB scanning...
by Raymond Yau 23 Oct '14

23 Oct '14

> > Howdy! I have a situation where PulseAudio is "grabbing" a usb headset before > fully initialized, thus not all features are being recognized. > > More details here: > > https://askubuntu.com/questions/537166/usb%2daudio%2ddevice%2dnot%2dfully%2… You have to post output of alsa-info.sh and pactl list before and after the hot plug This look like priority issue if you have usb web cam mic and hotplug usb headset https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/u… Another issue is some usb devices only obtained dB scale at a later time after the control is created This may need some delay for pulseaudio to get dB scale You can add printk statement to find out how much time you need > > and a related, I think, bug here: > > https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/841433 > > Anyone have a pointer to a udev rule I could create (or a PulseAudio config I > could tweak) that would case a delay before the device is scanned by > PulseAudio? > > I'm pretty sure it's a timing issue, because: > > Works OK: > Cold boot (or reboot) > Resume from hibernate > Resume (or hotplug) -> killall/restart pulseaudio > > Fails to work: > Hotplug > Resume from sleep > > Any tips? >

2 3

[alsa-devel] Echo Audio Echo 2 USB Interface
by Tristan Chambers 23 Oct '14

23 Oct '14

I picked up an Echo Audio Echo 2 USB Interface. The product specs list iPad support through camera connection kit, so I assume it is class compliant. http://echoaudio.com/blogs/news/6454472-echo-digital-audio-introduces-the-e… However, on most systems when I plug the device in, it shows up in my sound settings panel under the Input tab, but not under the Output tab. Recording works. I can successfully record with arecord -vv -fdat -D "plughw:CARD=Audio,DEV=0" echo.wav. I also can increase the sample rate by changing the flags. I verified that the sample rate actually increases by recording a 50khz test tone into the input and examining the wave form in an editor. It's there! So this is very promising. (anyone know of a test to verify bit depth? I want to make sure it's actually recording at 24bits.) Playback however, is nothing but trouble. Like I said, the output device doesn't show up in the Output tab in my sound settings (gnome/pulse). However, when I list devices and available PCMs with aplay -l and aplay -L the device shows up on the lists. See specimens #1, #2, and #3. When I try to play though I get about a 1/4 second of audio and then nothing, and subsequent attempts to play sound output nothing. The device then must be unplugged and plugged back in. I'm playing with the command aplay -vv foo.wav --device="plughw:CARD=Audio,DEV=0". See output screen on specimen #4. I installed the latest kernel and had the exact same phenomenon. See specimen #5. To move forward here. What kind of debugging tools should I use to help figure out where the problem is? Any suggestions you have to get it working? To be sure that the device is in good working order I hooked it up to an OSX machine running 10.6.8. Everything works great with no additional drivers installed, or special configurations. Tanks for your help, Tristan --------------- #1 -------------------- tristan@ilya:~$ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC259 Analog [ALC259 Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Audio [Echo 2 Audio], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 tristan@ilya:~$ aplay -L default Playback/recording through the PulseAudio sound server sysdefault:CARD=PCH HDA Intel PCH, ALC259 Analog Default Audio Device front:CARD=PCH,DEV=0 HDA Intel PCH, ALC259 Analog Front speakers surround40:CARD=PCH,DEV=0 HDA Intel PCH, ALC259 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=PCH,DEV=0 HDA Intel PCH, ALC259 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=PCH,DEV=0 HDA Intel PCH, ALC259 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=PCH,DEV=0 HDA Intel PCH, ALC259 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=PCH,DEV=0 HDA Intel PCH, ALC259 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers hdmi:CARD=PCH,DEV=0 HDA Intel PCH, HDMI 0 HDMI Audio Output dmix:CARD=PCH,DEV=0 HDA Intel PCH, ALC259 Analog Direct sample mixing device dmix:CARD=PCH,DEV=3 HDA Intel PCH, HDMI 0 Direct sample mixing device dsnoop:CARD=PCH,DEV=0 HDA Intel PCH, ALC259 Analog Direct sample snooping device dsnoop:CARD=PCH,DEV=3 HDA Intel PCH, HDMI 0 Direct sample snooping device hw:CARD=PCH,DEV=0 HDA Intel PCH, ALC259 Analog Direct hardware device without any conversions hw:CARD=PCH,DEV=3 HDA Intel PCH, HDMI 0 Direct hardware device without any conversions plughw:CARD=PCH,DEV=0 HDA Intel PCH, ALC259 Analog Hardware device with all software conversions plughw:CARD=PCH,DEV=3 HDA Intel PCH, HDMI 0 Hardware device with all software conversions sysdefault:CARD=Audio Echo 2 Audio, USB Audio Default Audio Device front:CARD=Audio,DEV=0 Echo 2 Audio, USB Audio Front speakers surround40:CARD=Audio,DEV=0 Echo 2 Audio, USB Audio 4.0 Surround output to Front and Rear speakers surround41:CARD=Audio,DEV=0 Echo 2 Audio, USB Audio 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=Audio,DEV=0 Echo 2 Audio, USB Audio 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=Audio,DEV=0 Echo 2 Audio, USB Audio 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=Audio,DEV=0 Echo 2 Audio, USB Audio 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=Audio,DEV=0 Echo 2 Audio, USB Audio IEC958 (S/PDIF) Digital Audio Output dmix:CARD=Audio,DEV=0 Echo 2 Audio, USB Audio Direct sample mixing device dsnoop:CARD=Audio,DEV=0 Echo 2 Audio, USB Audio Direct sample snooping device hw:CARD=Audio,DEV=0 Echo 2 Audio, USB Audio Direct hardware device without any conversions plughw:CARD=Audio,DEV=0 Echo 2 Audio, USB Audio Hardware device with all software conversions --------------- #2 -------------------- tristan@ilya:~$ arecord -l **** List of CAPTURE Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC259 Analog [ALC259 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Audio [Echo 2 Audio], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 tristan@ilya:~$ arecord -L default Playback/recording through the PulseAudio sound server sysdefault:CARD=PCH HDA Intel PCH, ALC259 Analog Default Audio Device front:CARD=PCH,DEV=0 HDA Intel PCH, ALC259 Analog Front speakers surround40:CARD=PCH,DEV=0 HDA Intel PCH, ALC259 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=PCH,DEV=0 HDA Intel PCH, ALC259 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=PCH,DEV=0 HDA Intel PCH, ALC259 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=PCH,DEV=0 HDA Intel PCH, ALC259 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=PCH,DEV=0 HDA Intel PCH, ALC259 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers dmix:CARD=PCH,DEV=0 HDA Intel PCH, ALC259 Analog Direct sample mixing device dsnoop:CARD=PCH,DEV=0 HDA Intel PCH, ALC259 Analog Direct sample snooping device hw:CARD=PCH,DEV=0 HDA Intel PCH, ALC259 Analog Direct hardware device without any conversions plughw:CARD=PCH,DEV=0 HDA Intel PCH, ALC259 Analog Hardware device with all software conversions sysdefault:CARD=Audio Echo 2 Audio, USB Audio Default Audio Device front:CARD=Audio,DEV=0 Echo 2 Audio, USB Audio Front speakers surround40:CARD=Audio,DEV=0 Echo 2 Audio, USB Audio 4.0 Surround output to Front and Rear speakers surround41:CARD=Audio,DEV=0 Echo 2 Audio, USB Audio 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=Audio,DEV=0 Echo 2 Audio, USB Audio 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=Audio,DEV=0 Echo 2 Audio, USB Audio 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=Audio,DEV=0 Echo 2 Audio, USB Audio 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=Audio,DEV=0 Echo 2 Audio, USB Audio IEC958 (S/PDIF) Digital Audio Output dmix:CARD=Audio,DEV=0 Echo 2 Audio, USB Audio Direct sample mixing device dsnoop:CARD=Audio,DEV=0 Echo 2 Audio, USB Audio Direct sample snooping device hw:CARD=Audio,DEV=0 Echo 2 Audio, USB Audio Direct hardware device without any conversions plughw:CARD=Audio,DEV=0 Echo 2 Audio, USB Audio Hardware device with all software conversions ---------------- #3 ------------------- tristan@ilya:/proc/asound/card1$ cat stream0 ECHO Digital Audio Echo 2 Audio at usb-0000:00:14.0-1, high speed : USB Audio Playback: Status: Stop Interface 1 Altset 1 Format: S32_LE Channels: 4 Endpoint: 1 OUT (ASYNC) Rates: 44100, 48000, 88200, 96000, 176400, 192000 Data packet interval: 125 us Capture: Status: Stop Interface 2 Altset 1 Format: S32_LE Channels: 2 Endpoint: 2 IN (ASYNC) Rates: 44100, 48000, 88200, 96000, 176400, 192000 Data packet interval: 125 us ------------------- #4 ------------------ aplay -vv foo.wav --device="plughw:CARD=Audio,DEV=0" Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 4 rate : 48000 exact rate : 48000 (48000/1) msbits : 32 buffer_size : 24000 period_size : 6000 period_time : 125000 tstamp_mode : NONE period_step : 1 avail_min : 6000 period_event : 0 start_threshold : 24000 stop_threshold : 24000 silence_threshold: 0 silence_size : 0 boundary : 1572864000 appl_ptr : 0 hw_ptr : 0 ############+ | 23% ------------------- #5 ------------------ tristan@ilya:~$ uname -a Linux ilya 3.17.0-031700-lowlatency #201410060605 SMP PREEMPT Mon Oct 6 11:06:16 UTC 2014 i686 i686 i686 GNU/Linux tristan@ilya:~$ cat /proc/asound/version Advanced Linux Sound Architecture Driver Version k3.17.0-031700-lowlatency.

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[alsa-devel] [PATCH v3 03/59] dmaengine: Make the destination abbreviation coherent
by Maxime Ripard 22 Oct '14

22 Oct '14

The dmaengine header abbreviates destination as at least two different strings. Make a coherent use of a single one. Signed-off-by: Maxime Ripard <maxime.ripard(a)free-electrons.com> Acked-by: Mark Brown <broonie(a)kernel.org> Acked-by: Laurent Pinchart <laurent.pinchart(a)ideasonboard.com> Acked-by: Stephen Warren <swarren(a)wwwdotorg.org> --- drivers/dma/bcm2835-dma.c | 2 +- drivers/dma/edma.c | 2 +- drivers/dma/fsl-edma.c | 2 +- drivers/dma/nbpfaxi.c | 2 +- drivers/dma/omap-dma.c | 2 +- drivers/dma/pl330.c | 2 +- drivers/dma/sirf-dma.c | 2 +- include/linux/dmaengine.h | 8 ++++---- sound/soc/soc-generic-dmaengine-pcm.c | 2 +- 9 files changed, 12 insertions(+), 12 deletions(-) diff --git a/drivers/dma/bcm2835-dma.c b/drivers/dma/bcm2835-dma.c index 68007974961a..8a6a9c27fe83 100644 --- a/drivers/dma/bcm2835-dma.c +++ b/drivers/dma/bcm2835-dma.c @@ -571,7 +571,7 @@ static int bcm2835_dma_device_slave_caps(struct dma_chan *dchan, struct dma_slave_caps *caps) { caps->src_addr_widths = BIT(DMA_SLAVE_BUSWIDTH_4_BYTES); - caps->dstn_addr_widths = BIT(DMA_SLAVE_BUSWIDTH_4_BYTES); + caps->dst_addr_widths = BIT(DMA_SLAVE_BUSWIDTH_4_BYTES); caps->directions = BIT(DMA_DEV_TO_MEM) | BIT(DMA_MEM_TO_DEV); caps->cmd_pause = false; caps->cmd_terminate = true; diff --git a/drivers/dma/edma.c b/drivers/dma/edma.c index 123f578d6dd3..790054eef76c 100644 --- a/drivers/dma/edma.c +++ b/drivers/dma/edma.c @@ -998,7 +998,7 @@ static int edma_dma_device_slave_caps(struct dma_chan *dchan, struct dma_slave_caps *caps) { caps->src_addr_widths = EDMA_DMA_BUSWIDTHS; - caps->dstn_addr_widths = EDMA_DMA_BUSWIDTHS; + caps->dst_addr_widths = EDMA_DMA_BUSWIDTHS; caps->directions = BIT(DMA_DEV_TO_MEM) | BIT(DMA_MEM_TO_DEV); caps->cmd_pause = true; caps->cmd_terminate = true; diff --git a/drivers/dma/fsl-edma.c b/drivers/dma/fsl-edma.c index 3c5711d5fe97..13df85a16438 100644 --- a/drivers/dma/fsl-edma.c +++ b/drivers/dma/fsl-edma.c @@ -781,7 +781,7 @@ static int fsl_dma_device_slave_caps(struct dma_chan *dchan, struct dma_slave_caps *caps) { caps->src_addr_widths = FSL_EDMA_BUSWIDTHS; - caps->dstn_addr_widths = FSL_EDMA_BUSWIDTHS; + caps->dst_addr_widths = FSL_EDMA_BUSWIDTHS; caps->directions = BIT(DMA_DEV_TO_MEM) | BIT(DMA_MEM_TO_DEV); caps->cmd_pause = true; caps->cmd_terminate = true; diff --git a/drivers/dma/nbpfaxi.c b/drivers/dma/nbpfaxi.c index 5aeada56a442..b4a61ccbb443 100644 --- a/drivers/dma/nbpfaxi.c +++ b/drivers/dma/nbpfaxi.c @@ -1076,7 +1076,7 @@ static int nbpf_slave_caps(struct dma_chan *dchan, struct dma_slave_caps *caps) { caps->src_addr_widths = NBPF_DMA_BUSWIDTHS; - caps->dstn_addr_widths = NBPF_DMA_BUSWIDTHS; + caps->dst_addr_widths = NBPF_DMA_BUSWIDTHS; caps->directions = BIT(DMA_DEV_TO_MEM) | BIT(DMA_MEM_TO_DEV); caps->cmd_pause = false; caps->cmd_terminate = true; diff --git a/drivers/dma/omap-dma.c b/drivers/dma/omap-dma.c index bbea8243f9e8..8035b8c33cfd 100644 --- a/drivers/dma/omap-dma.c +++ b/drivers/dma/omap-dma.c @@ -1100,7 +1100,7 @@ static int omap_dma_device_slave_caps(struct dma_chan *dchan, struct dma_slave_caps *caps) { caps->src_addr_widths = OMAP_DMA_BUSWIDTHS; - caps->dstn_addr_widths = OMAP_DMA_BUSWIDTHS; + caps->dst_addr_widths = OMAP_DMA_BUSWIDTHS; caps->directions = BIT(DMA_DEV_TO_MEM) | BIT(DMA_MEM_TO_DEV); caps->cmd_pause = true; caps->cmd_terminate = true; diff --git a/drivers/dma/pl330.c b/drivers/dma/pl330.c index 4839bfa74a10..b563bfa6cc4d 100644 --- a/drivers/dma/pl330.c +++ b/drivers/dma/pl330.c @@ -2576,7 +2576,7 @@ static int pl330_dma_device_slave_caps(struct dma_chan *dchan, struct dma_slave_caps *caps) { caps->src_addr_widths = PL330_DMA_BUSWIDTHS; - caps->dstn_addr_widths = PL330_DMA_BUSWIDTHS; + caps->dst_addr_widths = PL330_DMA_BUSWIDTHS; caps->directions = BIT(DMA_DEV_TO_MEM) | BIT(DMA_MEM_TO_DEV); caps->cmd_pause = false; caps->cmd_terminate = true; diff --git a/drivers/dma/sirf-dma.c b/drivers/dma/sirf-dma.c index aac03ab10c54..149d7fddfa75 100644 --- a/drivers/dma/sirf-dma.c +++ b/drivers/dma/sirf-dma.c @@ -652,7 +652,7 @@ static int sirfsoc_dma_device_slave_caps(struct dma_chan *dchan, struct dma_slave_caps *caps) { caps->src_addr_widths = SIRFSOC_DMA_BUSWIDTHS; - caps->dstn_addr_widths = SIRFSOC_DMA_BUSWIDTHS; + caps->dst_addr_widths = SIRFSOC_DMA_BUSWIDTHS; caps->directions = BIT(DMA_DEV_TO_MEM) | BIT(DMA_MEM_TO_DEV); caps->cmd_pause = true; caps->cmd_terminate = true; diff --git a/include/linux/dmaengine.h b/include/linux/dmaengine.h index 653a1fd07ae8..4d912575fec0 100644 --- a/include/linux/dmaengine.h +++ b/include/linux/dmaengine.h @@ -387,7 +387,7 @@ enum dma_residue_granularity { /* struct dma_slave_caps - expose capabilities of a slave channel only * * @src_addr_widths: bit mask of src addr widths the channel supports - * @dstn_addr_widths: bit mask of dstn addr widths the channel supports + * @dst_addr_widths: bit mask of dstn addr widths the channel supports * @directions: bit mask of slave direction the channel supported * since the enum dma_transfer_direction is not defined as bits for each * type of direction, the dma controller should fill (1 << <TYPE>) and same @@ -398,7 +398,7 @@ enum dma_residue_granularity { */ struct dma_slave_caps { u32 src_addr_widths; - u32 dstn_addr_widths; + u32 dst_addr_widths; u32 directions; bool cmd_pause; bool cmd_terminate; @@ -638,10 +638,10 @@ struct dma_device { void (*device_free_chan_resources)(struct dma_chan *chan); struct dma_async_tx_descriptor *(*device_prep_dma_memcpy)( - struct dma_chan *chan, dma_addr_t dest, dma_addr_t src, + struct dma_chan *chan, dma_addr_t dst, dma_addr_t src, size_t len, unsigned long flags); struct dma_async_tx_descriptor *(*device_prep_dma_xor)( - struct dma_chan *chan, dma_addr_t dest, dma_addr_t *src, + struct dma_chan *chan, dma_addr_t dst, dma_addr_t *src, unsigned int src_cnt, size_t len, unsigned long flags); struct dma_async_tx_descriptor *(*device_prep_dma_xor_val)( struct dma_chan *chan, dma_addr_t *src, unsigned int src_cnt, diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index b329b84bc5af..851f7afcd5dc 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -151,7 +151,7 @@ static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substrea hw.info |= SNDRV_PCM_INFO_BATCH; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - addr_widths = dma_caps.dstn_addr_widths; + addr_widths = dma_caps.dst_addr_widths; else addr_widths = dma_caps.src_addr_widths; } -- 2.1.1

1 0

[alsa-devel] [PATCH 0/2] ASoC: two trivial fixes
by Daniel Mack 22 Oct '14

22 Oct '14

Just two random fixes spotted by Coverity. Daniel Mack (2): ASoC: soc-compress: consolidate two identical branches ASoC: fsl: use strncpy() to prevent copying of over-long names sound/soc/fsl/fsl_asrc.c | 2 +- sound/soc/fsl/fsl_esai.c | 2 +- sound/soc/soc-compress.c | 11 ++--------- 3 files changed, 4 insertions(+), 11 deletions(-) -- 2.1.0

4 5

[alsa-devel] [PATCH] ASoC: atmel_ssc_dai: Track playback and capture CMR dividers separately.
by Peter Rosin 22 Oct '14

22 Oct '14

>From 1e5621d7b9887c648d1a66238dc82d715c1e2cad Mon Sep 17 00:00:00 2001 From: Peter Rosin <peda(a)axentia.se> Date: Mon, 20 Oct 2014 14:38:04 +0200 Subject: [PATCH] ASoC: atmel_ssc_dai: Track playback and capture CMR dividers separately. The CMR divider register is shared by playback and capture. The SSC driver therefore tries to enforce rules so that the needed register content do not conflict during simultaneous playback/capture. However, the implementation also prevents changing the register content in a half-duplex scenario, which is needed when changing sample rates. Thus, keep track of the desired playback and capture clock dividers separately, and allow changing rates without closing the stream. Signed-off-by: Peter Rosin <peda(a)axentia.se> --- sound/soc/atmel/atmel_ssc_dai.c | 31 ++++++++++++++++++++++--------- sound/soc/atmel/atmel_ssc_dai.h | 10 ++++++---- 2 files changed, 28 insertions(+), 13 deletions(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index f403f39..fec14fb 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -277,7 +277,8 @@ static void atmel_ssc_shutdown(struct snd_pcm_substream *substream, /* Reset the SSC */ ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); /* Clear the SSC dividers */ - ssc_p->cmr_div = ssc_p->tcmr_period = ssc_p->rcmr_period = 0; + ssc_p->tcmr_div = ssc_p->rcmr_div = 0; + ssc_p->tcmr_period = ssc_p->rcmr_period = 0; } spin_unlock_irq(&ssc_p->lock); } @@ -304,17 +305,27 @@ static int atmel_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, struct atmel_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; switch (div_id) { - case ATMEL_SSC_CMR_DIV: + case ATMEL_SSC_TCMR_DIV: /* * The same master clock divider is used for both * transmit and receive, so if a value has already - * been set, it must match this value. + * been set for the other direction, it must match + * this value. */ - if (ssc_p->cmr_div == 0) - ssc_p->cmr_div = div; - else - if (div != ssc_p->cmr_div) - return -EBUSY; + if (ssc_p->rcmr_div == 0) + ssc_p->tcmr_div = div; + else if (div != ssc_p->rcmr_div) + return -EBUSY; + break; + + case ATMEL_SSC_RCMR_DIV: + /* + * See ATMEL_SSC_TCMR_DIV. + */ + if (ssc_p->tcmr_div == 0) + ssc_p->rcmr_div = div; + else if (div != ssc_p->tcmr_div) + return -EBUSY; break; case ATMEL_SSC_TCMR_PERIOD: @@ -345,6 +356,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, struct atmel_pcm_dma_params *dma_params; int dir, channels, bits; u32 tfmr, rfmr, tcmr, rcmr; + u16 cmr; int start_event; int ret; int fslen, fslen_ext; @@ -626,7 +638,8 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, } /* set SSC clock mode register */ - ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->cmr_div); + cmr = ssc_p->tcmr_div ? ssc_p->tcmr_div : ssc_p->rcmr_div; + ssc_writel(ssc_p->ssc->regs, CMR, cmr); /* set receive clock mode and format */ ssc_writel(ssc_p->ssc->regs, RCMR, rcmr); diff --git a/sound/soc/atmel/atmel_ssc_dai.h b/sound/soc/atmel/atmel_ssc_dai.h index b1f08d5..a25df7a 100644 --- a/sound/soc/atmel/atmel_ssc_dai.h +++ b/sound/soc/atmel/atmel_ssc_dai.h @@ -39,9 +39,10 @@ #define ATMEL_SYSCLK_MCK 0 /* SSC uses AT91 MCK as system clock */ /* SSC divider ids */ -#define ATMEL_SSC_CMR_DIV 0 /* MCK divider for BCLK */ -#define ATMEL_SSC_TCMR_PERIOD 1 /* BCLK divider for transmit FS */ -#define ATMEL_SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */ +#define ATMEL_SSC_TCMR_DIV 0 /* MCK divider for transmit BCLK */ +#define ATMEL_SSC_RCMR_DIV 1 /* MCK divider for receive BCLK */ +#define ATMEL_SSC_TCMR_PERIOD 2 /* BCLK divider for transmit FS */ +#define ATMEL_SSC_RCMR_PERIOD 3 /* BCLK divider for receive FS */ /* * SSC direction masks */ @@ -110,7 +111,8 @@ struct atmel_ssc_info { unsigned short dir_mask; /* 0=unused, 1=playback, 2=capture */ unsigned short initialized; /* true if SSC has been initialized */ unsigned short daifmt; - unsigned short cmr_div; + unsigned short tcmr_div; + unsigned short rcmr_div; unsigned short tcmr_period; unsigned short rcmr_period; struct atmel_pcm_dma_params *dma_params[2]; -- 1.7.10.4

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[alsa-devel] [PATCH] ASoC: rt5677: add GPIO IRQ support
by Anatol Pomozov 22 Oct '14

22 Oct '14

This allows to enable Mic Jack detection feature Author: Oder Chiou <oder_chiou(a)realtek.com> Modified-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- Documentation/devicetree/bindings/sound/rt5677.txt | 10 ++ include/sound/rt5677.h | 7 ++ sound/soc/codecs/rt5677.c | 134 +++++++++++++++++++++ sound/soc/codecs/rt5677.h | 49 ++++++++ 4 files changed, 200 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/rt5677.txt b/Documentation/devicetree/bindings/sound/rt5677.txt index f82f0e9..740ff77 100644 --- a/Documentation/devicetree/bindings/sound/rt5677.txt +++ b/Documentation/devicetree/bindings/sound/rt5677.txt @@ -33,6 +33,15 @@ Optional properties: 1 - pull down 2 - pull up +- realtek,jd1-gpio + Configures GPIO Mic Jack detection 1. + Select 0 ~ 3 as OFF, GPIO1, GPIO2 and GPIO3 respectively. + +- realtek,jd2-gpio +- realtek,jd3-gpio + Configures GPIO Mic Jack detection 2 and 3. + Select 0 ~ 3 as OFF, GPIO4, GPIO5 and GPIO6 respectively. + Pins on the device (for linking into audio routes): * IN1P @@ -63,4 +72,5 @@ rt5677 { <&gpio TEGRA_GPIO(V, 3) GPIO_ACTIVE_HIGH>; realtek,in1-differential = "true"; realtek,gpio-config = /bits/ 8 <0 0 0 0 0 2>; /* pull up GPIO6 */ + realtek,jd2-gpio = <3>; /* Enables Jack detection for GPIO6 */ }; diff --git a/include/sound/rt5677.h b/include/sound/rt5677.h index a56b429..d9eb7d8 100644 --- a/include/sound/rt5677.h +++ b/include/sound/rt5677.h @@ -30,6 +30,13 @@ struct rt5677_platform_data { /* configures GPIO, 0 - floating, 1 - pulldown, 2 - pullup */ u8 gpio_config[6]; + + /* jd1 can select 0 ~ 3 as OFF, GPIO1, GPIO2 and GPIO3 respectively */ + unsigned int jd1_gpio; + /* jd2 and jd3 can select 0 ~ 3 as + OFF, GPIO4, GPIO5 and GPIO6 respectively */ + unsigned int jd2_gpio; + unsigned int jd3_gpio; }; #endif diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index f6df276..1f5897f 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -3614,6 +3614,46 @@ static void rt5677_gpio_config(struct rt5677_priv *rt5677, unsigned offset, } } +static int rt5677_to_irq(struct gpio_chip *chip, unsigned offset) +{ + struct rt5677_priv *rt5677 = gpio_to_rt5677(chip); + struct regmap_irq_chip_data *data = rt5677->irq_data; + int irq; + + if (offset >= RT5677_GPIO1 && offset <= RT5677_GPIO3) { + if ((rt5677->pdata.jd1_gpio == 1 && offset == RT5677_GPIO1) || + (rt5677->pdata.jd1_gpio == 2 && + offset == RT5677_GPIO2) || + (rt5677->pdata.jd1_gpio == 3 && + offset == RT5677_GPIO3)) { + irq = RT5677_IRQ_JD1; + } else { + return -ENXIO; + } + } + + if (offset >= RT5677_GPIO4 && offset <= RT5677_GPIO6) { + if ((rt5677->pdata.jd2_gpio == 1 && offset == RT5677_GPIO4) || + (rt5677->pdata.jd2_gpio == 2 && + offset == RT5677_GPIO5) || + (rt5677->pdata.jd2_gpio == 3 && + offset == RT5677_GPIO6)) { + irq = RT5677_IRQ_JD2; + } else if ((rt5677->pdata.jd3_gpio == 1 && + offset == RT5677_GPIO4) || + (rt5677->pdata.jd3_gpio == 2 && + offset == RT5677_GPIO5) || + (rt5677->pdata.jd3_gpio == 3 && + offset == RT5677_GPIO6)) { + irq = RT5677_IRQ_JD3; + } else { + return -ENXIO; + } + } + + return regmap_irq_get_virq(data, irq); +} + static struct gpio_chip rt5677_template_chip = { .label = "rt5677", .owner = THIS_MODULE, @@ -3621,6 +3661,7 @@ static struct gpio_chip rt5677_template_chip = { .set = rt5677_gpio_set, .direction_input = rt5677_gpio_direction_in, .get = rt5677_gpio_get, + .to_irq = rt5677_to_irq, .can_sleep = 1, }; @@ -3688,6 +3729,31 @@ static int rt5677_probe(struct snd_soc_codec *codec) for (i = 0; i < RT5677_GPIO_NUM; i++) rt5677_gpio_config(rt5677, i, rt5677->pdata.gpio_config[i]); + if (rt5677->irq_data) { + regmap_update_bits(rt5677->regmap, RT5677_GPIO_CTRL1, 0x8000, + 0x8000); + regmap_update_bits(rt5677->regmap, RT5677_DIG_MISC, 0x0018, + 0x0008); + + if (rt5677->pdata.jd1_gpio) + regmap_update_bits(rt5677->regmap, RT5677_JD_CTRL1, + RT5677_SEL_GPIO_JD1_MASK, + rt5677->pdata.jd1_gpio << + RT5677_SEL_GPIO_JD1_SFT); + + if (rt5677->pdata.jd2_gpio) + regmap_update_bits(rt5677->regmap, RT5677_JD_CTRL1, + RT5677_SEL_GPIO_JD2_MASK, + rt5677->pdata.jd2_gpio << + RT5677_SEL_GPIO_JD2_SFT); + + if (rt5677->pdata.jd3_gpio) + regmap_update_bits(rt5677->regmap, RT5677_JD_CTRL1, + RT5677_SEL_GPIO_JD3_MASK, + rt5677->pdata.jd3_gpio << + RT5677_SEL_GPIO_JD3_SFT); + } + mutex_init(&rt5677->dsp_cmd_lock); return 0; @@ -3918,9 +3984,74 @@ static int rt5677_parse_dt(struct rt5677_priv *rt5677, struct device_node *np) of_property_read_u8_array(np, "realtek,gpio-config", rt5677->pdata.gpio_config, RT5677_GPIO_NUM); + of_property_read_u32(np, "realtek,jd1-gpio", &rt5677->pdata.jd1_gpio); + of_property_read_u32(np, "realtek,jd2-gpio", &rt5677->pdata.jd2_gpio); + of_property_read_u32(np, "realtek,jd3-gpio", &rt5677->pdata.jd3_gpio); + return 0; } +static struct regmap_irq rt5677_irqs[] = { + [RT5677_IRQ_JD1] = { + .reg_offset = 0, + .mask = RT5677_EN_IRQ_GPIO_JD1, + }, + [RT5677_IRQ_JD2] = { + .reg_offset = 0, + .mask = RT5677_EN_IRQ_GPIO_JD2, + }, + [RT5677_IRQ_JD3] = { + .reg_offset = 0, + .mask = RT5677_EN_IRQ_GPIO_JD3, + }, +}; + +static struct regmap_irq_chip rt5677_irq_chip = { + .name = "rt5677", + .irqs = rt5677_irqs, + .num_irqs = ARRAY_SIZE(rt5677_irqs), + + .num_regs = 1, + .status_base = RT5677_IRQ_CTRL1, + .mask_base = RT5677_IRQ_CTRL1, + .mask_invert = 1, +}; + +int rt5677_irq_init(struct i2c_client *i2c) +{ + int ret; + struct rt5677_priv *rt5677 = i2c_get_clientdata(i2c); + + if (!rt5677->pdata.jd1_gpio && + !rt5677->pdata.jd2_gpio && + !rt5677->pdata.jd3_gpio) + return 0; + + if (!i2c->irq) { + dev_err(&i2c->dev, "No interrupt specified\n"); + return -EINVAL; + } + + ret = regmap_add_irq_chip(rt5677->regmap, i2c->irq, + IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, 0, + &rt5677_irq_chip, &rt5677->irq_data); + + if (ret != 0) { + dev_err(&i2c->dev, "Failed to register IRQ chip: %d\n", ret); + return ret; + } + + return 0; +} + +void rt5677_irq_exit(struct i2c_client *i2c) +{ + struct rt5677_priv *rt5677 = i2c_get_clientdata(i2c); + + if (rt5677->irq_data) + regmap_del_irq_chip(i2c->irq, rt5677->irq_data); +} + static int rt5677_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -4018,6 +4149,7 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, } rt5677_init_gpio(i2c); + rt5677_irq_init(i2c); return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5677, rt5677_dai, ARRAY_SIZE(rt5677_dai)); @@ -4025,6 +4157,8 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, static int rt5677_i2c_remove(struct i2c_client *i2c) { + rt5677_irq_exit(i2c); + snd_soc_unregister_codec(&i2c->dev); rt5677_free_gpio(i2c); diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index 20efa4a..d2c743c 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -1368,6 +1368,48 @@ #define RT5677_SEL_SRC_IB01 (0x1 << 0) #define RT5677_SEL_SRC_IB01_SFT 0 +/* Jack Detect Control 1 (0xb5) */ +#define RT5677_SEL_GPIO_JD1_MASK (0x3 << 14) +#define RT5677_SEL_GPIO_JD1_SFT 14 +#define RT5677_SEL_GPIO_JD2_MASK (0x3 << 12) +#define RT5677_SEL_GPIO_JD2_SFT 12 +#define RT5677_SEL_GPIO_JD3_MASK (0x3 << 10) +#define RT5677_SEL_GPIO_JD3_SFT 10 + +/* IRQ Control 1 (0xbd) */ +#define RT5677_STA_GPIO_JD1 (0x1 << 15) +#define RT5677_STA_GPIO_JD1_SFT 15 +#define RT5677_EN_IRQ_GPIO_JD1 (0x1 << 14) +#define RT5677_EN_IRQ_GPIO_JD1_SFT 14 +#define RT5677_EN_GPIO_JD1_STICKY (0x1 << 13) +#define RT5677_EN_GPIO_JD1_STICKY_SFT 13 +#define RT5677_INV_GPIO_JD1 (0x1 << 12) +#define RT5677_INV_GPIO_JD1_SFT 12 +#define RT5677_STA_GPIO_JD2 (0x1 << 11) +#define RT5677_STA_GPIO_JD2_SFT 11 +#define RT5677_EN_IRQ_GPIO_JD2 (0x1 << 10) +#define RT5677_EN_IRQ_GPIO_JD2_SFT 10 +#define RT5677_EN_GPIO_JD2_STICKY (0x1 << 9) +#define RT5677_EN_GPIO_JD2_STICKY_SFT 9 +#define RT5677_INV_GPIO_JD2 (0x1 << 8) +#define RT5677_INV_GPIO_JD2_SFT 8 +#define RT5677_STA_MICBIAS1_OVCD (0x1 << 7) +#define RT5677_STA_MICBIAS1_OVCD_SFT 7 +#define RT5677_EN_IRQ_MICBIAS1_OVCD (0x1 << 6) +#define RT5677_EN_IRQ_MICBIAS1_OVCD_SFT 6 +#define RT5677_EN_MICBIAS1_OVCD_STICKY (0x1 << 5) +#define RT5677_EN_MICBIAS1_OVCD_STICKY_SFT 5 +#define RT5677_INV_MICBIAS1_OVCD (0x1 << 4) +#define RT5677_INV_MICBIAS1_OVCD_SFT 4 +#define RT5677_STA_GPIO_JD3 (0x1 << 3) +#define RT5677_STA_GPIO_JD3_SFT 3 +#define RT5677_EN_IRQ_GPIO_JD3 (0x1 << 2) +#define RT5677_EN_IRQ_GPIO_JD3_SFT 2 +#define RT5677_EN_GPIO_JD3_STICKY (0x1 << 1) +#define RT5677_EN_GPIO_JD3_STICKY_SFT 1 +#define RT5677_INV_GPIO_JD3 (0x1 << 0) +#define RT5677_INV_GPIO_JD3_SFT 0 + /* GPIO status (0xbf) */ #define RT5677_GPIO6_STATUS_MASK (0x1 << 5) #define RT5677_GPIO6_STATUS_SFT 5 @@ -1545,6 +1587,12 @@ enum { RT5677_GPIO_NUM, }; +enum { + RT5677_IRQ_JD1, + RT5677_IRQ_JD2, + RT5677_IRQ_JD3, +}; + struct rt5677_priv { struct snd_soc_codec *codec; struct rt5677_platform_data pdata; @@ -1565,6 +1613,7 @@ struct rt5677_priv { struct gpio_chip gpio_chip; #endif bool dsp_vad_en; + struct regmap_irq_chip_data *irq_data; }; #endif /* __RT5677_H__ */ -- 2.1.0.rc2.206.gedb03e5

3 5

[alsa-devel] [PATCH 01/12] ASoC: ak4535: Cleanup manual bias level transitions
by Lars-Peter Clausen 22 Oct '14

22 Oct '14

Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/ak4535.c | 21 ++------------------- 1 file changed, 2 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 30e2978..eced46d 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -373,36 +373,19 @@ static struct snd_soc_dai_driver ak4535_dai = { .ops = &ak4535_dai_ops, }; -static int ak4535_suspend(struct snd_soc_codec *codec) -{ - ak4535_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static int ak4535_resume(struct snd_soc_codec *codec) { snd_soc_cache_sync(codec); - ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; } static int ak4535_probe(struct snd_soc_codec *codec) { - /* power on device */ - ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - snd_soc_add_codec_controls(codec, ak4535_snd_controls, ARRAY_SIZE(ak4535_snd_controls)); return 0; } -/* power down chip */ -static int ak4535_remove(struct snd_soc_codec *codec) -{ - ak4535_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static const struct regmap_config ak4535_regmap = { .reg_bits = 8, .val_bits = 8, @@ -417,10 +400,10 @@ static const struct regmap_config ak4535_regmap = { static struct snd_soc_codec_driver soc_codec_dev_ak4535 = { .probe = ak4535_probe, - .remove = ak4535_remove, - .suspend = ak4535_suspend, .resume = ak4535_resume, .set_bias_level = ak4535_set_bias_level, + .suspend_bias_off = true, + .dapm_widgets = ak4535_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(ak4535_dapm_widgets), .dapm_routes = ak4535_audio_map, -- 1.8.0

2 12

[alsa-devel] [PATCH next 1/2] ALSA: pcm: Remove arch-dependent mmap kludges
by Takashi Iwai 22 Oct '14

22 Oct '14

Since we have consistently dma_mmap_coherent() for all architectures, the current ifdef and arch-specific codes in pcm core can be cleaned up gracefully. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_native.c | 30 ------------------------------ 1 file changed, 30 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 815396d8427f..aa6754da7929 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -35,9 +35,6 @@ #include <sound/timer.h> #include <sound/minors.h> #include <asm/io.h> -#if defined(CONFIG_MIPS) && defined(CONFIG_DMA_NONCOHERENT) -#include <dma-coherence.h> -#endif /* * Compatibility @@ -3251,20 +3248,6 @@ static inline struct page * snd_pcm_default_page_ops(struct snd_pcm_substream *substream, unsigned long ofs) { void *vaddr = substream->runtime->dma_area + ofs; -#if defined(CONFIG_MIPS) && defined(CONFIG_DMA_NONCOHERENT) - if (substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) - return virt_to_page(CAC_ADDR(vaddr)); -#endif -#if defined(CONFIG_PPC32) && defined(CONFIG_NOT_COHERENT_CACHE) - if (substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) { - dma_addr_t addr = substream->runtime->dma_addr + ofs; - addr -= get_dma_offset(substream->dma_buffer.dev.dev); - /* assume dma_handle set via pfn_to_phys() in - * mm/dma-noncoherent.c - */ - return pfn_to_page(addr >> PAGE_SHIFT); - } -#endif return virt_to_page(vaddr); } @@ -3309,13 +3292,6 @@ static const struct vm_operations_struct snd_pcm_vm_ops_data_fault = { .fault = snd_pcm_mmap_data_fault, }; -#ifndef ARCH_HAS_DMA_MMAP_COHERENT -/* This should be defined / handled globally! */ -#if defined(CONFIG_ARM) || defined(CONFIG_ARM64) -#define ARCH_HAS_DMA_MMAP_COHERENT -#endif -#endif - /* * mmap the DMA buffer on RAM */ @@ -3331,7 +3307,6 @@ int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream, area->vm_end - area->vm_start, area->vm_page_prot); } #endif /* CONFIG_GENERIC_ALLOCATOR */ -#ifdef ARCH_HAS_DMA_MMAP_COHERENT if (!substream->ops->page && substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) return dma_mmap_coherent(substream->dma_buffer.dev.dev, @@ -3339,11 +3314,6 @@ int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream, substream->runtime->dma_area, substream->runtime->dma_addr, area->vm_end - area->vm_start); -#elif defined(CONFIG_MIPS) && defined(CONFIG_DMA_NONCOHERENT) - if (substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV && - !plat_device_is_coherent(substream->dma_buffer.dev.dev)) - area->vm_page_prot = pgprot_noncached(area->vm_page_prot); -#endif /* ARCH_HAS_DMA_MMAP_COHERENT */ /* mmap with fault handler */ area->vm_ops = &snd_pcm_vm_ops_data_fault; return 0; -- 2.1.2

1 1

[alsa-devel] [PATCH 0/2] ASoC: spread usage of snd_ctl_elem_info()
by Takashi Iwai 22 Oct '14

22 Oct '14

Hi, just like my previous patch series, this convers the simple open-code for enum info callbacks with the existing snd_ctl_enum_info() helper. They can be applied individually, as far as I know, but I think it'd fit better to the same branch for other similar patches. Takashi

3 5

[alsa-devel] [PATCH V2 1/2] ARM: pxa: spitz: register spitz-audio device
by Dmitry Eremin-Solenikov 22 Oct '14

22 Oct '14

Register spitz-audio device to be used by ASoC driver. [V2: Removed ifdefs around spitz_audio_init]. Signed-off-by: Dmitry Eremin-Solenikov <dbaryshkov(a)gmail.com> --- arch/arm/mach-pxa/spitz.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/arch/arm/mach-pxa/spitz.c b/arch/arm/mach-pxa/spitz.c index 840c3a4..af1dea3 100644 --- a/arch/arm/mach-pxa/spitz.c +++ b/arch/arm/mach-pxa/spitz.c @@ -924,6 +924,19 @@ static inline void spitz_i2c_init(void) {} #endif /****************************************************************************** + * Audio devices + ******************************************************************************/ +static struct platform_device spitz_audio_device = { + .name = "spitz-audio", + .id = -1, +}; + +static inline void spitz_audio_init(void) +{ + platform_device_register(&spitz_audio_device); +} + +/****************************************************************************** * Machine init ******************************************************************************/ static void spitz_poweroff(void) @@ -970,6 +983,7 @@ static void __init spitz_init(void) spitz_nor_init(); spitz_nand_init(); spitz_i2c_init(); + spitz_audio_init(); } static void __init spitz_fixup(struct tag *tags, char **cmdline) -- 2.1.1

2 3

[alsa-devel] [PATCH] ASoC: Intel: Fix block is enabled multiple times issue
by Jie Yang 22 Oct '14

22 Oct '14

During FW parsing and loading, block_list_prepare() may be called for each raw data block copying and this may made the hsw_block_enable() called mutiple times, which increase block->users many times. The result of this is hsw_block_disable() can't power gated the related block when trying to free the blocks during suspend, and the power gating status also confused. Here check the block user status, only calling enable() for those blocks who has no user yet. Remember that this works correctlly on current case, where there are enough SRAM memory so different module won't share a memory block. For further usage, we may need restructure the struct sst_mem_block to save the module list who is using it. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/sst-firmware.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index f2f1d5f..db60398 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -80,7 +80,7 @@ static int block_list_prepare(struct sst_dsp *dsp, /* enable each block so that's it'e ready for data */ list_for_each_entry(block, block_list, module_list) { - if (block->ops && block->ops->enable) { + if (block->ops && block->ops->enable && !block->users) { ret = block->ops->enable(block); if (ret < 0) { dev_err(dsp->dev, -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: adau1761: Fix input PGA volume
by Lars-Peter Clausen 22 Oct '14

22 Oct '14

For the input PGA to work correctly the ALC clock needs to be active. Otherwise volume changes are not applied. Fixes: dab464b60b2 ("ASoC: Add ADAU1361/ADAU1761 audio CODEC support") Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Cc: stable(a)vger.kernel.org --- sound/soc/codecs/adau1761.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c index 5518ebd..91f6028 100644 --- a/sound/soc/codecs/adau1761.c +++ b/sound/soc/codecs/adau1761.c @@ -405,6 +405,7 @@ static const struct snd_soc_dapm_widget adau1761_dapm_widgets[] = { 2, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("Slew Clock", ADAU1761_CLK_ENABLE0, 6, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ALC Clock", ADAU1761_CLK_ENABLE0, 5, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("Digital Clock 0", 1, ADAU1761_CLK_ENABLE1, 0, 0, NULL, 0), @@ -436,6 +437,9 @@ static const struct snd_soc_dapm_route adau1761_dapm_routes[] = { { "Right Playback Mixer", NULL, "Slew Clock" }, { "Left Playback Mixer", NULL, "Slew Clock" }, + { "Left Input Mixer", NULL, "ALC Clock" }, + { "Right Input Mixer", NULL, "ALC Clock" }, + { "Digital Clock 0", NULL, "SYSCLK" }, { "Digital Clock 1", NULL, "SYSCLK" }, }; -- 1.8.0

2 1

[alsa-devel] [PATCH 0/5] Add Phytec pbab01 audio support
by Dmitry Lavnikevich 22 Oct '14

22 Oct '14

This patchset implements audio support on Phytec's phyFLEX-i.MX6 boards. This includes creating imx-tlv320aic3x machine driver, fixing tlv320aic3x driver playback speed issue and enabling audio for pbab01 itself.

10 44

[alsa-devel] [PATCH v2 0/2] ASoC: simple-card: Add multi-CODEC support
by Jean-Francois Moine 22 Oct '14

22 Oct '14

This patch set adds multi-CODEC support to the simple card. v2: accept list of CODECs in 'sound-dais' (Benoit Cousson) Jean-Francois Moine (2): ASoC: core: add multi-codec support in DT ASoC: simple-card: add multi-CODECs in DT .../devicetree/bindings/sound/simple-card.txt | 28 +++-- include/sound/soc.h | 3 + sound/soc/generic/simple-card.c | 114 +++++++++++++-------- sound/soc/soc-core.c | 75 +++++++++++--- 4 files changed, 149 insertions(+), 71 deletions(-) -- 2.1.1

1 2

[alsa-devel] Alsa-lib 32-bit on 64-bit machine
by Alan Horstmann 22 Oct '14

22 Oct '14

Hi, I've tried to find info or examples of this on the web, but to no avail. What is the recommended way to build Alsa-lib on a 64-bit machine to create the additional alternative 32-bit libraries for compatibility with 32-bit applications? The original distro (in this case a Suse Enterprise) has them, but standard './configure make install' only builds the 64-bit variants. I only succeeded by explicitly adding CFLAGS, and wondered if it should work using configure options, or another way? I'm also not sure exactly which flags should be set with that method (m32, march=i686 others?). Does anyone know how it is done in the Suse builds? Thanks, Alan

2 3

[alsa-devel] [RFC PATCH 1/2] ALSA: hda - Fix "PCM" name being used on one DAC when there are two DACs
by David Henningsson 22 Oct '14

22 Oct '14

In the scenario where there is one "Line Out", one "Speaker" and one "Headphone", and there are only two DACs, two outputs will share a DAC. Currently any mixer on such a DAC will get the "PCM" name, which is misleading. Instead use "Headphone+LO" or "Speaker+LO" to better specify what the volume actually controls. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_generic.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 64220c0..1d742ea 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -1038,6 +1038,19 @@ static const char *get_line_out_pfx(struct hda_codec *codec, int ch, break; *index = ch; return "Headphone"; + case AUTO_PIN_LINE_OUT: + /* This deals with the case where we have two DACs and + * one LO, one HP and one Speaker */ + if (!ch && cfg->speaker_outs && cfg->hp_outs) { + bool hp_lo_shared = !path_has_mixer(codec, spec->hp_paths[0], ctl_type); + bool spk_lo_shared = !path_has_mixer(codec, spec->speaker_paths[0], ctl_type); + if (hp_lo_shared && spk_lo_shared) + return spec->vmaster_mute.hook ? "PCM" : "Master"; + if (hp_lo_shared) + return "Headphone+LO"; + if (spk_lo_shared) + return "Speaker+LO"; + } } /* for a single channel output, we don't have to name the channel */ -- 1.9.1

4 28

[alsa-devel] [PATCH] ASoC: simple-card: Add mic and hp detect gpios.
by Dylan Reid 22 Oct '14

22 Oct '14

Allow Headphone and Microphone jack detect gpios to be specified in device tree. This will allow a few systems including rk3288_max98090 to use simple-card instead of having their own board file. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- .../devicetree/bindings/sound/simple-card.txt | 4 ++ sound/soc/generic/simple-card.c | 73 ++++++++++++++++++++++ 2 files changed, 77 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt index c2e9841..72d94b7 100644 --- a/Documentation/devicetree/bindings/sound/simple-card.txt +++ b/Documentation/devicetree/bindings/sound/simple-card.txt @@ -17,6 +17,10 @@ Optional properties: source. - simple-audio-card,mclk-fs : Multiplication factor between stream rate and codec mclk. +- simple-audio-card,hp_det_gpio : Reference to GPIO that signals when + headphones are attached. +- simple-audio-card,mic_det_gpio : Reference to GPIO that signals when + a microphone is attached. Optional subnodes: diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 709ce67..fcb431f 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -10,10 +10,13 @@ */ #include <linux/clk.h> #include <linux/device.h> +#include <linux/gpio.h> #include <linux/module.h> #include <linux/of.h> +#include <linux/of_gpio.h> #include <linux/platform_device.h> #include <linux/string.h> +#include <sound/jack.h> #include <sound/simple_card.h> #include <sound/soc-dai.h> #include <sound/soc.h> @@ -25,6 +28,8 @@ struct simple_card_data { struct asoc_simple_dai codec_dai; } *dai_props; unsigned int mclk_fs; + int gpio_hp_det; + int gpio_mic_det; struct snd_soc_dai_link dai_link[]; /* dynamically allocated */ }; @@ -54,6 +59,32 @@ static struct snd_soc_ops asoc_simple_card_ops = { .hw_params = asoc_simple_card_hw_params, }; +static struct snd_soc_jack simple_card_hp_jack; +static struct snd_soc_jack_pin simple_card_hp_jack_pins[] = { + { + .pin = "Headphones", + .mask = SND_JACK_HEADPHONE, + }, +}; +static struct snd_soc_jack_gpio simple_card_hp_jack_gpio = { + .name = "Headphone detection", + .report = SND_JACK_HEADPHONE, + .debounce_time = 150, +}; + +static struct snd_soc_jack simple_card_mic_jack; +static struct snd_soc_jack_pin simple_card_mic_jack_pins[] = { + { + .pin = "Mic Jack", + .mask = SND_JACK_MICROPHONE, + }, +}; +static struct snd_soc_jack_gpio simple_card_mic_jack_gpio = { + .name = "Mic detection", + .report = SND_JACK_MICROPHONE, + .debounce_time = 150, +}; + static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, struct asoc_simple_dai *set) { @@ -109,6 +140,28 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) if (ret < 0) return ret; + if (gpio_is_valid(priv->gpio_hp_det)) { + snd_soc_jack_new(codec->codec, "Headphones", SND_JACK_HEADPHONE, + &simple_card_hp_jack); + snd_soc_jack_add_pins(&simple_card_hp_jack, + ARRAY_SIZE(simple_card_hp_jack_pins), + simple_card_hp_jack_pins); + + simple_card_hp_jack_gpio.gpio = priv->gpio_hp_det; + snd_soc_jack_add_gpios(&simple_card_hp_jack, 1, + &simple_card_hp_jack_gpio); + } + + if (gpio_is_valid(priv->gpio_mic_det)) { + snd_soc_jack_new(codec->codec, "Mic Jack", SND_JACK_MICROPHONE, + &simple_card_mic_jack); + snd_soc_jack_add_pins(&simple_card_mic_jack, + ARRAY_SIZE(simple_card_mic_jack_pins), + simple_card_mic_jack_pins); + simple_card_mic_jack_gpio.gpio = priv->gpio_mic_det; + snd_soc_jack_add_gpios(&simple_card_mic_jack, 1, + &simple_card_mic_jack_gpio); + } return 0; } @@ -383,6 +436,16 @@ static int asoc_simple_card_parse_of(struct device_node *node, return ret; } + priv->gpio_hp_det = of_get_named_gpio(node, + "simple-audio-card,hp-det-gpio", 0); + if (priv->gpio_hp_det == -EPROBE_DEFER) + return -EPROBE_DEFER; + + priv->gpio_mic_det = of_get_named_gpio(node, + "simple-audio-card,mic-det-gpio", 0); + if (priv->gpio_mic_det == -EPROBE_DEFER) + return -EPROBE_DEFER; + if (!priv->snd_card.name) priv->snd_card.name = priv->snd_card.dai_link->name; @@ -502,6 +565,16 @@ err: static int asoc_simple_card_remove(struct platform_device *pdev) { + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct simple_card_data *priv = snd_soc_card_get_drvdata(card); + + if (gpio_is_valid(priv->gpio_hp_det)) + snd_soc_jack_free_gpios(&simple_card_hp_jack, 1, + &simple_card_hp_jack_gpio); + if (gpio_is_valid(priv->gpio_mic_det)) + snd_soc_jack_free_gpios(&simple_card_mic_jack, 1, + &simple_card_mic_jack_gpio); + return asoc_simple_card_unref(pdev); } -- 2.1.0.rc2.206.gedb03e5

7 11

[alsa-devel] [PATCH v3 0/5] ASoC: intel: fix to sst driver series
by Vinod Koul 22 Oct '14

22 Oct '14

This series fixes comments observed while applying the sst series last week. changes in v3: - fix the makefile artifact not required changes in v2: - typo fixes - make wrapper inline - add newline on err log Vinod Koul (5): ASoC: intel: use __iowrite32_copy for 32 bit copy ASoC: intel: log an error on double free ASoC: intel: fix the kernldoc comment ASoC: intel: explain why block not found isn't error always ASoC: intel: use __iowrite32_copy for 32 bit copy sound/soc/intel/sst-firmware.c | 9 +++------ sound/soc/intel/sst/sst_drv_interface.c | 4 +++- sound/soc/intel/sst/sst_ipc.c | 15 +++++++++++++++ sound/soc/intel/sst/sst_loader.c | 12 +++++------- 4 files changed, 26 insertions(+), 14 deletions(-)

2 6

[alsa-devel] [PATCH 1/2] MAINTAINERS: add atmel ssc driver maintainer entry
by Bo Shen 21 Oct '14

21 Oct '14

Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- MAINTAINERS | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/MAINTAINERS b/MAINTAINERS index 3705430..c004e6f 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -1703,6 +1703,13 @@ M: Nicolas Ferre <nicolas.ferre(a)atmel.com> S: Supported F: drivers/spi/spi-atmel.* +ATMEL SSC DRIVER +M: Bo Shen <voice.shen(a)atmel.com> +L: linux-arm-kernel(a)lists.infradead.org (moderated for non-subscribers) +S: Supported +F: drivers/misc/atmel-ssc.c +F: include/linux/atmel-ssc.h + ATMEL Timer Counter (TC) AND CLOCKSOURCE DRIVERS M: Nicolas Ferre <nicolas.ferre(a)atmel.com> L: linux-arm-kernel(a)lists.infradead.org (moderated for non-subscribers) -- 2.1.0.24.g4109c28

4 6

[alsa-devel] [PATCH] ASoC: rt5677: fix rt5677 spi driver build
by Ben Zhang 21 Oct '14

21 Oct '14

Create a separate module for rt5677 spi driver. Without this patch, the build fails due to multiple defs of 'init_module' and 'cleanup_module'. module_spi_driver() defines its own module, so it can't be part of the rt5677 module. Signed-off-by: Ben Zhang <benzh(a)chromium.org> --- sound/soc/codecs/Kconfig | 4 ++++ sound/soc/codecs/Makefile | 4 +++- sound/soc/codecs/rt5677-spi.c | 2 ++ 3 files changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2c7482e..6f21a76 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -504,6 +504,10 @@ config SND_SOC_RT5670 config SND_SOC_RT5677 tristate +config SND_SOC_RT5677_SPI + tristate + default SND_SOC_RT5677 + #Freescale sgtl5000 codec config SND_SOC_SGTL5000 tristate "Freescale SGTL5000 CODEC" diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 4435f9f..3e57edc 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -79,7 +79,8 @@ snd-soc-rt5640-objs := rt5640.o snd-soc-rt5645-objs := rt5645.o snd-soc-rt5651-objs := rt5651.o snd-soc-rt5670-objs := rt5670.o -snd-soc-rt5677-objs := rt5677.o rt5677-spi.o +snd-soc-rt5677-objs := rt5677.o +snd-soc-rt5677-spi-objs := rt5677-spi.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o snd-soc-alc5632-objs := alc5632.o @@ -256,6 +257,7 @@ obj-$(CONFIG_SND_SOC_RT5645) += snd-soc-rt5645.o obj-$(CONFIG_SND_SOC_RT5651) += snd-soc-rt5651.o obj-$(CONFIG_SND_SOC_RT5670) += snd-soc-rt5670.o obj-$(CONFIG_SND_SOC_RT5677) += snd-soc-rt5677.o +obj-$(CONFIG_SND_SOC_RT5677_SPI) += snd-soc-rt5677-spi.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o obj-$(CONFIG_SND_SOC_SIGMADSP_I2C) += snd-soc-sigmadsp-i2c.o diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c index 11c38f3..ef6348c 100644 --- a/sound/soc/codecs/rt5677-spi.c +++ b/sound/soc/codecs/rt5677-spi.c @@ -52,6 +52,7 @@ int rt5677_spi_write(u8 *txbuf, size_t len) return status; } +EXPORT_SYMBOL_GPL(rt5677_spi_write); /** * rt5677_spi_burst_write - Write data to SPI by rt5677 dsp memory address. @@ -107,6 +108,7 @@ int rt5677_spi_burst_write(u32 addr, const struct firmware *fw) return 0; } +EXPORT_SYMBOL_GPL(rt5677_spi_burst_write); static int rt5677_spi_probe(struct spi_device *spi) { -- 2.1.0.rc2.206.gedb03e5

2 1

[alsa-devel] [PATCH] ASoC: dapm: Remove redundant cast
by Rasmus Villemoes 21 Oct '14

21 Oct '14

Both path->name and e->texts[i] have type const char*, so the cast is slightly confusing and certainly unnecessary. Signed-off-by: Rasmus Villemoes <linux(a)rasmusvillemoes.dk> --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c61cb9c..39f992b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -496,7 +496,7 @@ static int dapm_connect_mux(struct snd_soc_dapm_context *dapm, list_add(&path->list, &dapm->card->paths); list_add(&path->list_sink, &dest->sources); list_add(&path->list_source, &src->sinks); - path->name = (char*)e->texts[i]; + path->name = e->texts[i]; if (i == item) path->connect = 1; else -- 2.0.4

2 1

[alsa-devel] [PATCH] ASoC: tegra: Read and use the GPIO flags of the headphone detect
by Alban Bedel 21 Oct '14

21 Oct '14

The headphone detect was hardcoded to low-active, use the flags from DT to allow high-active as well. Signed-off-by: Alban Bedel <alban.bedel(a)avionic-design.de> --- sound/soc/tegra/tegra_rt5640.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/tegra/tegra_rt5640.c b/sound/soc/tegra/tegra_rt5640.c index a689883..4ebe387 100644 --- a/sound/soc/tegra/tegra_rt5640.c +++ b/sound/soc/tegra/tegra_rt5640.c @@ -44,6 +44,7 @@ struct tegra_rt5640 { struct tegra_asoc_utils_data util_data; int gpio_hp_det; + enum of_gpio_flags gpio_hp_det_flags; }; static int tegra_rt5640_asoc_hw_params(struct snd_pcm_substream *substream, @@ -119,6 +120,8 @@ static int tegra_rt5640_asoc_init(struct snd_soc_pcm_runtime *rtd) if (gpio_is_valid(machine->gpio_hp_det)) { tegra_rt5640_hp_jack_gpio.gpio = machine->gpio_hp_det; + tegra_rt5640_hp_jack_gpio.invert = + !!(machine->gpio_hp_det_flags & OF_GPIO_ACTIVE_LOW); snd_soc_jack_add_gpios(&tegra_rt5640_hp_jack, 1, &tegra_rt5640_hp_jack_gpio); @@ -180,7 +183,8 @@ static int tegra_rt5640_probe(struct platform_device *pdev) platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, machine); - machine->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0); + machine->gpio_hp_det = of_get_named_gpio_flags( + np, "nvidia,hp-det-gpios", 0, &machine->gpio_hp_det_flags); if (machine->gpio_hp_det == -EPROBE_DEFER) return -EPROBE_DEFER; -- 2.1.1

3 2

[alsa-devel] [PATCH 1/2] ARM: pxa: spitz: register spitz-audio device
by Dmitry Eremin-Solenikov 21 Oct '14

21 Oct '14

Register spitz-audio device to be used by ASoC driver. Signed-off-by: Dmitry Eremin-Solenikov <dbaryshkov(a)gmail.com> --- arch/arm/mach-pxa/spitz.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/arch/arm/mach-pxa/spitz.c b/arch/arm/mach-pxa/spitz.c index 840c3a4..a9f971f 100644 --- a/arch/arm/mach-pxa/spitz.c +++ b/arch/arm/mach-pxa/spitz.c @@ -924,6 +924,23 @@ static inline void spitz_i2c_init(void) {} #endif /****************************************************************************** + * Audio devices + ******************************************************************************/ +#if defined(CONFIG_SND_PXA2XX_SOC_SPITZ) || defined(SND_PXA2XX_SOC_SPITZ_MODULE) +static struct platform_device spitz_audio_device = { + .name = "spitz-audio", + .id = -1, +}; + +static inline void spitz_audio_init(void) +{ + platform_device_register(&spitz_audio_device); +} +#else +static inline void spitz_audio_init(void) {} +#endif + +/****************************************************************************** * Machine init ******************************************************************************/ static void spitz_poweroff(void) @@ -970,6 +987,7 @@ static void __init spitz_init(void) spitz_nor_init(); spitz_nand_init(); spitz_i2c_init(); + spitz_audio_init(); } static void __init spitz_fixup(struct tag *tags, char **cmdline) -- 2.1.1

2 3

[alsa-devel] 3.18.0-rc1 pulseaudio possible recursive locking
by Arthur Marsh 21 Oct '14

21 Oct '14

Hi, I have been seeing this a few times lately, and noticed that it was still present Is this the right place to report it or should I be reporting a bug against pulseaudio? [ 182.273991] ============================================= [ 182.273996] [ INFO: possible recursive locking detected ] [ 182.274003] 3.18.0-rc1 #1297 Not tainted [ 182.274008] --------------------------------------------- [ 182.274013] pulseaudio/5183 is trying to acquire lock: [ 182.274018] (&(&substream->self_group.lock)->rlock/1){......}, at: [<f8432284>] snd_pcm_action_group+0x96/0x215 [snd_pcm] [ 182.274063] but task is already holding lock: [ 182.274069] (&(&substream->self_group.lock)->rlock/1){......}, at: [<f8432284>] snd_pcm_action_group+0x96/0x215 [snd_pcm] [ 182.274092] other info that might help us debug this: [ 182.274098] Possible unsafe locking scenario: [ 182.274104] CPU0 [ 182.274108] ---- [ 182.274112] lock(&(&substream->self_group.lock)->rlock/1); [ 182.274122] lock(&(&substream->self_group.lock)->rlock/1); [ 182.274132] *** DEADLOCK *** [ 182.274138] May be due to missing lock nesting notation [ 182.274146] 4 locks held by pulseaudio/5183: [ 182.274151] #0: (snd_pcm_link_rwlock){......}, at: [<f8432fe3>] snd_pcm_stream_lock+0x1b/0x43 [snd_pcm] [ 182.274177] #1: (&(&substream->self_group.lock)->rlock){......}, at: [<f8432fee>] snd_pcm_stream_lock+0x26/0x43 [snd_pcm] [ 182.274202] #2: (&(&substream->group->lock)->rlock){......}, at: [<f8432cf4>] snd_pcm_action+0x34/0x10b [snd_pcm] [ 182.274227] #3: (&(&substream->self_group.lock)->rlock/1){......}, at: [<f8432284>] snd_pcm_action_group+0x96/0x215 [snd_pcm] [ 182.274254] stack backtrace: [ 182.274263] CPU: 1 PID: 5183 Comm: pulseaudio Not tainted 3.18.0-rc1 #1297 [ 182.274269] Hardware name: System Manufacturer System Name/P4S800, BIOS ASUS P4S800 ACPI BIOS Revision 1011 Beta 001 08/30/2005 [ 182.274275] c1bfc4e0 00000000 e55c9d50 c145d969 c1bfc4e0 e55c9dd4 c1082b7b c15b7f9a [ 182.274295] e55b630c 0000143f 00000000 e55b6010 e55b6070 00000000 5402a095 00000004 [ 182.274315] c17a4034 c1bfc4e0 e55b6508 e55b6520 00000000 00000008 00000000 e55b6508 [ 182.274335] Call Trace: [ 182.274353] [<c145d969>] dump_stack+0x4b/0x75 [ 182.274366] [<c1082b7b>] __lock_acquire+0x187c/0x19b3 [ 182.274378] [<c10834ad>] lock_acquire+0xa1/0x126 [ 182.274394] [<f8432284>] ? snd_pcm_action_group+0x96/0x215 [snd_pcm] [ 182.274404] [<c1463a9d>] _raw_spin_lock_nested+0x41/0x4e [ 182.274420] [<f8432284>] ? snd_pcm_action_group+0x96/0x215 [snd_pcm] [ 182.274436] [<f8432284>] snd_pcm_action_group+0x96/0x215 [snd_pcm] [ 182.274445] [<c146393b>] ? _raw_spin_trylock+0x6b/0x80 [ 182.274462] [<f8432d0f>] snd_pcm_action+0x4f/0x10b [snd_pcm] [ 182.274479] [<f843332a>] snd_pcm_drop+0x4a/0x7d [snd_pcm] [ 182.274496] [<f84351d9>] snd_pcm_common_ioctl1+0x7b0/0xcc4 [snd_pcm] [ 182.274514] [<f84357ce>] snd_pcm_playback_ioctl1+0xe1/0x3a8 [snd_pcm] [ 182.274532] [<f8435ab7>] snd_pcm_playback_ioctl+0x22/0x35 [snd_pcm] [ 182.274549] [<f8435a95>] ? snd_pcm_playback_ioctl1+0x3a8/0x3a8 [snd_pcm] [ 182.274561] [<c1171e82>] do_vfs_ioctl+0x306/0x576 [ 182.274573] [<c117d3ca>] ? mntput_no_expire+0x61/0x1e7 [ 182.274583] [<c117d377>] ? mntput_no_expire+0xe/0x1e7 [ 182.274594] [<c117d56b>] ? mntput+0x1b/0x29 [ 182.274603] [<c1161438>] ? __fput+0x144/0x1aa [ 182.274614] [<c1172147>] SyS_ioctl+0x55/0x75 [ 182.274624] [<c14646e0>] sysenter_do_call+0x12/0x12 Regards, Arthur.

2 3

[alsa-devel] [PATCH] ALSA: ctxfi: remove unused variable
by Sudip Mukherjee 21 Oct '14

21 Oct '14

As of now the pointer to struct dai is not being used anywhere in the function. So it is safe to remove the variable. If we are ever doing anything with the container_of(daio, struct dai, daio), then at that time we can again add the variable. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- sound/pci/ctxfi/ctatc.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index 4546590..632e843 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -1145,7 +1145,6 @@ static int atc_release_resources(struct ct_atc *atc) int i; struct daio_mgr *daio_mgr = NULL; struct dao *dao = NULL; - struct dai *dai = NULL; struct daio *daio = NULL; struct sum_mgr *sum_mgr = NULL; struct src_mgr *src_mgr = NULL; @@ -1172,9 +1171,6 @@ static int atc_release_resources(struct ct_atc *atc) dao = container_of(daio, struct dao, daio); dao->ops->clear_left_input(dao); dao->ops->clear_right_input(dao); - } else { - dai = container_of(daio, struct dai, daio); - /* some thing to do for dai ... */ } daio_mgr->put_daio(daio_mgr, daio); } -- 1.8.1.2

2 1

[alsa-devel] [PATCH 00/43] Spread usage of snd_ctl_elem_info()
by Takashi Iwai 21 Oct '14

21 Oct '14

Hi, this is a series of patches to just convert the plain info callback for enum ctl elements to snd_ctl_elem_info(). Also, it includes the extension of snd_ctl_elem_info(), for catching the unexpected string cut-off and handling the zero items. Takashi

1 43

[alsa-devel] bug in alsa-utils 1.0.28 - arecord
by Jordan Kagan 21 Oct '14

21 Oct '14

Hello, I encountered an issue with using the arecord binary in this most recent version. When using the '-d' duration flag, the program will continue to open files AFTER the duration has expired. The issue occurs when the capture() while loop hits the break due to a value returned by pcm_read(). When the while loop is exited, the program hits the do-while loop and continues to open files indefinitely. If rcount is returned from pcm_read() instead of "result", the problem does not occur. Thanks, Jordan

2 1

Re: [alsa-devel] [PATCH v2] ASoC: simple-card: add asoc_simple_card_parse_daifmt()
by Mark Brown 21 Oct '14

21 Oct '14

On Tue, Sep 23, 2014 at 07:29:34PM -0700, Kuninori Morimoto wrote: > > Hi Mark > > Should I re-send this patch ? Don't top post... I've been at Linaro Connect so haven't had a chance to look at it.

2 3

[alsa-devel] [PATCH] ALSA: Update control names documentation
by David Henningsson 21 Oct '14

21 Oct '14

This document was not really up-to-date. Add recent additions to this standard - based on what the HDA driver currently does, which is some kind of a de facto standard. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- Documentation/sound/alsa/ControlNames.txt | 32 ++++++++++++++++++++++++++----- 1 file changed, 27 insertions(+), 5 deletions(-) diff --git a/Documentation/sound/alsa/ControlNames.txt b/Documentation/sound/alsa/ControlNames.txt index fea65bb..79a6127 100644 --- a/Documentation/sound/alsa/ControlNames.txt +++ b/Documentation/sound/alsa/ControlNames.txt @@ -1,6 +1,6 @@ This document describes standard names of mixer controls. -Syntax: SOURCE [DIRECTION] FUNCTION +Syntax: [LOCATION] SOURCE [CHANNEL] [DIRECTION] FUNCTION DIRECTION: <nothing> (both directions) @@ -14,12 +14,29 @@ FUNCTION: Volume Route (route control, hardware specific) +CHANNEL: + <nothing> (channel independent, or applies to all channels) + Front + Surround (rear left/right in 4.0/5.1 surround) + CLFE + Center + LFE + Side (side left/right for 7.1 surround) + +LOCATION: (physical location of source) + Front + Rear + Dock (docking station) + Internal + SOURCE: Master Master Mono Hardware Master Speaker (internal speaker) + Bass Speaker (internal LFE speaker) Headphone + Line Out Beep (beep generator) Phone Phone Input @@ -27,14 +44,14 @@ SOURCE: Synth FM Mic - Line + Headset Mic (mic part of combined headset jack - 4-pin headphone + mic) + Headphone Mic (mic part of either/or - 3-pin headphone or mic) + Line (input only, use "Line Out" for output) CD Video Zoom Video Aux PCM - PCM Front - PCM Rear PCM Pan Loopback Analog Loopback (D/A -> A/D loopback) @@ -47,8 +64,13 @@ SOURCE: Music I2S IEC958 + HDMI + SPDIF (output only) + SPDIF In + Digital In + HDMI/DP (either HDMI or DisplayPort) -Exceptions: +Exceptions (deprecated): [Digital] Capture Source [Digital] Capture Switch (aka input gain switch) [Digital] Capture Volume (aka input gain volume) -- 1.9.1

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[alsa-devel] Advanced Gtk+ Sequencer 0.4.2
by Joël Krähemann 20 Oct '14

20 Oct '14

Advanced Gtk+ Sequencer 0.4.2 is out Check download at: http://sourceforge.net/projects/ags/files/ags-0_4_2.tar.bz2/download or API Reference by issuing: http://ags.sourceforge.net/api/ags/index.html Advanced GTK+ Sequencer is intended to use for music composition. Use its piano roll, synth, matrix editor, drum machine, soundfont2 player, mixer or the output panel.

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[alsa-devel] [PATCH v4 0/7] ASoC: Intel: sst - add the merrifield IPC driver
by Vinod Koul 20 Oct '14

20 Oct '14

This path series adds the Merrifield IPC driver which is used to load and manage the DSP along with sending and receiving IPCs for PCM and compressed audio. v4 updates: - fix the comment style issues - fix for id allocation handling - modularize the code to remove few repated patterns - comments from previos rev (hopefully didnt miss anything) Vinod Koul (7): ASoC: Intel: mrfld - add the dsp sst driver ASoC: Intel: mrfld - Add DSP load and management ASoC: Intel: sst - add pcm ops handling ASoC: Intel: sst: Add IPC handling ASoC: Intel: sst: add stream operations ASoC: Intel: sst: Add some helper functions ASoC: Intel: sst: Add makefile and kconfig changes arch/x86/include/asm/platform_sst_audio.h | 34 ++ sound/soc/intel/Kconfig | 4 + sound/soc/intel/Makefile | 3 + sound/soc/intel/sst/Makefile | 3 + sound/soc/intel/sst/sst.c | 434 +++++++++++++++++++++++ sound/soc/intel/sst/sst.h | 539 +++++++++++++++++++++++++++++ sound/soc/intel/sst/sst_drv_interface.c | 403 +++++++++++++++++++++ sound/soc/intel/sst/sst_ipc.c | 358 +++++++++++++++++++ sound/soc/intel/sst/sst_loader.c | 461 ++++++++++++++++++++++++ sound/soc/intel/sst/sst_pvt.c | 446 ++++++++++++++++++++++++ sound/soc/intel/sst/sst_stream.c | 437 +++++++++++++++++++++++ 11 files changed, 3122 insertions(+), 0 deletions(-) create mode 100644 sound/soc/intel/sst/Makefile create mode 100644 sound/soc/intel/sst/sst.c create mode 100644 sound/soc/intel/sst/sst.h create mode 100644 sound/soc/intel/sst/sst_drv_interface.c create mode 100644 sound/soc/intel/sst/sst_ipc.c create mode 100644 sound/soc/intel/sst/sst_loader.c create mode 100644 sound/soc/intel/sst/sst_pvt.c create mode 100644 sound/soc/intel/sst/sst_stream.c

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[alsa-devel] [PATCH 1/2] ALSA: snd-usb: drop unused varible assigments
by Daniel Mack 20 Oct '14

20 Oct '14

Don't assign 'len' in cases where we don't make use of the returned value. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- sound/usb/mixer.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 2e4a9db..63a8adb 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1290,9 +1290,8 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, kctl->id.name, sizeof(kctl->id.name), 1); if (!len) - len = snprintf(kctl->id.name, - sizeof(kctl->id.name), - "Feature %d", unitid); + snprintf(kctl->id.name, sizeof(kctl->id.name), + "Feature %d", unitid); } if (!mapped_name) @@ -1305,9 +1304,9 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, */ if (!mapped_name && !(state->oterm.type >> 16)) { if ((state->oterm.type & 0xff00) == 0x0100) - len = append_ctl_name(kctl, " Capture"); + append_ctl_name(kctl, " Capture"); else - len = append_ctl_name(kctl, " Playback"); + append_ctl_name(kctl, " Playback"); } append_ctl_name(kctl, control == UAC_FU_MUTE ? " Switch" : " Volume"); -- 2.1.0

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[alsa-devel] [PATCH 0/7] ASoC: Intel: Update HSW/BDW SST driver to new FW architecture
by Liam Girdwood 20 Oct '14

20 Oct '14

The following patch series updates the current upstream driver to use the new more feature rich ADSP FW architecture. The new FW arch allows for multiple instantiation of objects in dynamic memory locations. In summary :- o Modules can be created multiple times. o Modules persistent data can be allocated in any free blocks o Global scratch data can be allocated into any free blocks. o DMA can be used to load the module data. The driver now has to track and manage the new multiple module resource allocations and pass this data to FW on stream initialisations. Thanks Liam

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[alsa-devel] [RFC PATCH] Inverted internal mic
by rodney byne 20 Oct '14

20 Oct '14

Hello to David Henningsson and associates, I have been trawling the internet to find out why the internal mic on my Acer Aspire One AOA-110 netbook is permanently inhibited by Alsa. This little pc runs Linux Peppermint 5 OS, which contains PulseAudio ESD over-riding the Alsa sound architecture. Thinking initially this was a "programming" fault, I have read as much of the history up to the patch need as I can understand and so appreciate this action on your part was deliberate - but must comment nonetheless that this matter is disconcerting from a casual user perspective. In short, the owner just wants everything on his machine to work normally and so I would really like to use the internal mic function for what it was intended. Personally I have often wondered why PA was ever introduced, as Alsa & AlsaMixer are perfectly capable of working on their own, as testament to early releases of Linux Mint. My opinion is PA just makes the sound system too complicated. I call it guilding the lilly - so use the principle K.I.S.S My question therefore is, can the patch be easily lifted at will to re-enable the internal microphone? Thanks and best regards, from Rodney.

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[alsa-devel] [PATCH v2 0/5] ASoC: intel: fix to sst driver series
by Vinod Koul 20 Oct '14

20 Oct '14

This series fixes comments observed while applying the sst series last week. canges in v2: - typo fixes - make wrapper inline - add newline on err log Vinod Koul (5): ASoC: intel: use __iowrite32_copy for 32 bit copy ASoC: intel: log an error on double free ASoC: intel: fix the kernldoc comment ASoC: intel: explain why block not found isn't error always ASoC: intel: use __iowrite32_copy for 32 bit copy sound/soc/intel/Kconfig | 2 +- sound/soc/intel/sst-firmware.c | 9 +++------ sound/soc/intel/sst/sst_drv_interface.c | 4 +++- sound/soc/intel/sst/sst_ipc.c | 15 +++++++++++++++ sound/soc/intel/sst/sst_loader.c | 12 +++++------- 5 files changed, 27 insertions(+), 15 deletions(-)

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[alsa-devel] [PATCH 0/5] ASoC: intel: fix to sst driver series
by Vinod Koul 20 Oct '14

20 Oct '14

Hi, This series fixes comments observed while applying the sst series last week. Vinod Koul (5): ASoC: intel: use __iowrite32_copy for 32 bit copy ASoC: intel: log an error on double free ASoC: intel: fix the kernldoc comment ASoC: intel: explain why block not found isn't error always ASoC: intel: use __iowrite32_copy for 32 bit copy sound/soc/intel/sst-firmware.c | 7 ++----- sound/soc/intel/sst/sst_drv_interface.c | 4 +++- sound/soc/intel/sst/sst_ipc.c | 15 +++++++++++++++ sound/soc/intel/sst/sst_loader.c | 9 +++------ 4 files changed, 23 insertions(+), 12 deletions(-)

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[alsa-devel] [PATCH] ASoC: s6000: remove driver
by Daniel Glöckner 20 Oct '14

20 Oct '14

The s6000 Xtensa support is removed from the kernel. There are no other chips known to use this I2S controller. Signed-off-by: Daniel Glöckner <dg(a)emlix.com> --- sound/soc/Kconfig | 1 - sound/soc/Makefile | 1 - sound/soc/s6000/Kconfig | 26 -- sound/soc/s6000/Makefile | 11 - sound/soc/s6000/s6000-i2s.c | 617 ------------------------------------------ sound/soc/s6000/s6000-i2s.h | 23 -- sound/soc/s6000/s6000-pcm.c | 521 ----------------------------------- sound/soc/s6000/s6000-pcm.h | 33 --- sound/soc/s6000/s6105-ipcam.c | 221 --------------- 9 files changed, 1454 deletions(-) delete mode 100644 sound/soc/s6000/Kconfig delete mode 100644 sound/soc/s6000/Makefile delete mode 100644 sound/soc/s6000/s6000-i2s.c delete mode 100644 sound/soc/s6000/s6000-i2s.h delete mode 100644 sound/soc/s6000/s6000-pcm.c delete mode 100644 sound/soc/s6000/s6000-pcm.h delete mode 100644 sound/soc/s6000/s6105-ipcam.c diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 0e96233..7d5d644 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -49,7 +49,6 @@ source "sound/soc/mxs/Kconfig" source "sound/soc/pxa/Kconfig" source "sound/soc/rockchip/Kconfig" source "sound/soc/samsung/Kconfig" -source "sound/soc/s6000/Kconfig" source "sound/soc/sh/Kconfig" source "sound/soc/sirf/Kconfig" source "sound/soc/spear/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 534714a..d88edfc 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -26,7 +26,6 @@ obj-$(CONFIG_SND_SOC) += kirkwood/ obj-$(CONFIG_SND_SOC) += pxa/ obj-$(CONFIG_SND_SOC) += rockchip/ obj-$(CONFIG_SND_SOC) += samsung/ -obj-$(CONFIG_SND_SOC) += s6000/ obj-$(CONFIG_SND_SOC) += sh/ obj-$(CONFIG_SND_SOC) += sirf/ obj-$(CONFIG_SND_SOC) += spear/ diff --git a/sound/soc/s6000/Kconfig b/sound/soc/s6000/Kconfig deleted file mode 100644 index f244a25..0000000 --- a/sound/soc/s6000/Kconfig +++ /dev/null @@ -1,26 +0,0 @@ -config SND_S6000_SOC - tristate "SoC Audio for the Stretch s6000 family" - depends on XTENSA_VARIANT_S6000 || COMPILE_TEST - depends on HAS_IOMEM - select SND_S6000_SOC_PCM if XTENSA_VARIANT_S6000 - help - Say Y or M if you want to add support for codecs attached to - s6000 family chips. You will also need to select the platform - to support below. - -config SND_S6000_SOC_PCM - tristate - -config SND_S6000_SOC_I2S - tristate - -config SND_S6000_SOC_S6IPCAM - bool "SoC Audio support for Stretch 6105 IP Camera" - depends on SND_S6000_SOC=y - depends on I2C=y - depends on XTENSA_PLATFORM_S6105 || COMPILE_TEST - select SND_S6000_SOC_I2S - select SND_SOC_TLV320AIC3X - help - Say Y if you want to add support for SoC audio on the - Stretch s6105 IP Camera Reference Design. diff --git a/sound/soc/s6000/Makefile b/sound/soc/s6000/Makefile deleted file mode 100644 index 0f0ae2a..0000000 --- a/sound/soc/s6000/Makefile +++ /dev/null @@ -1,11 +0,0 @@ -# s6000 Platform Support -snd-soc-s6000-objs := s6000-pcm.o -snd-soc-s6000-i2s-objs := s6000-i2s.o - -obj-$(CONFIG_SND_S6000_SOC_PCM) += snd-soc-s6000.o -obj-$(CONFIG_SND_S6000_SOC_I2S) += snd-soc-s6000-i2s.o - -# s6105 Machine Support -snd-soc-s6ipcam-objs := s6105-ipcam.o - -obj-$(CONFIG_SND_S6000_SOC_S6IPCAM) += snd-soc-s6ipcam.o diff --git a/sound/soc/s6000/s6000-i2s.c b/sound/soc/s6000/s6000-i2s.c deleted file mode 100644 index 1c8d011..0000000 --- a/sound/soc/s6000/s6000-i2s.c +++ /dev/null @@ -1,617 +0,0 @@ -/* - * ALSA SoC I2S Audio Layer for the Stretch S6000 family - * - * Author: Daniel Gloeckner, <dg(a)emlix.com> - * Copyright: (C) 2009 emlix GmbH <info(a)emlix.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#include <linux/init.h> -#include <linux/module.h> -#include <linux/device.h> -#include <linux/delay.h> -#include <linux/clk.h> -#include <linux/interrupt.h> -#include <linux/io.h> -#include <linux/slab.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/initval.h> -#include <sound/soc.h> - -#include "s6000-i2s.h" -#include "s6000-pcm.h" - -struct s6000_i2s_dev { - dma_addr_t sifbase; - u8 __iomem *scbbase; - unsigned int wide; - unsigned int channel_in; - unsigned int channel_out; - unsigned int lines_in; - unsigned int lines_out; - struct s6000_pcm_dma_params dma_params; -}; - -#define S6_I2S_INTERRUPT_STATUS 0x00 -#define S6_I2S_INT_OVERRUN 1 -#define S6_I2S_INT_UNDERRUN 2 -#define S6_I2S_INT_ALIGNMENT 4 -#define S6_I2S_INTERRUPT_ENABLE 0x04 -#define S6_I2S_INTERRUPT_RAW 0x08 -#define S6_I2S_INTERRUPT_CLEAR 0x0C -#define S6_I2S_INTERRUPT_SET 0x10 -#define S6_I2S_MODE 0x20 -#define S6_I2S_DUAL 0 -#define S6_I2S_WIDE 1 -#define S6_I2S_TX_DEFAULT 0x24 -#define S6_I2S_DATA_CFG(c) (0x40 + 0x10 * (c)) -#define S6_I2S_IN 0 -#define S6_I2S_OUT 1 -#define S6_I2S_UNUSED 2 -#define S6_I2S_INTERFACE_CFG(c) (0x44 + 0x10 * (c)) -#define S6_I2S_DIV_MASK 0x001fff -#define S6_I2S_16BIT 0x000000 -#define S6_I2S_20BIT 0x002000 -#define S6_I2S_24BIT 0x004000 -#define S6_I2S_32BIT 0x006000 -#define S6_I2S_BITS_MASK 0x006000 -#define S6_I2S_MEM_16BIT 0x000000 -#define S6_I2S_MEM_32BIT 0x008000 -#define S6_I2S_MEM_MASK 0x008000 -#define S6_I2S_CHANNELS_SHIFT 16 -#define S6_I2S_CHANNELS_MASK 0x030000 -#define S6_I2S_SCK_IN 0x000000 -#define S6_I2S_SCK_OUT 0x040000 -#define S6_I2S_SCK_DIR 0x040000 -#define S6_I2S_WS_IN 0x000000 -#define S6_I2S_WS_OUT 0x080000 -#define S6_I2S_WS_DIR 0x080000 -#define S6_I2S_LEFT_FIRST 0x000000 -#define S6_I2S_RIGHT_FIRST 0x100000 -#define S6_I2S_FIRST 0x100000 -#define S6_I2S_CUR_SCK 0x200000 -#define S6_I2S_CUR_WS 0x400000 -#define S6_I2S_ENABLE(c) (0x48 + 0x10 * (c)) -#define S6_I2S_DISABLE_IF 0x02 -#define S6_I2S_ENABLE_IF 0x03 -#define S6_I2S_IS_BUSY 0x04 -#define S6_I2S_DMA_ACTIVE 0x08 -#define S6_I2S_IS_ENABLED 0x10 - -#define S6_I2S_NUM_LINES 4 - -#define S6_I2S_SIF_PORT0 0x0000000 -#define S6_I2S_SIF_PORT1 0x0000080 /* docs say 0x0000010 */ - -static inline void s6_i2s_write_reg(struct s6000_i2s_dev *dev, int reg, u32 val) -{ - writel(val, dev->scbbase + reg); -} - -static inline u32 s6_i2s_read_reg(struct s6000_i2s_dev *dev, int reg) -{ - return readl(dev->scbbase + reg); -} - -static inline void s6_i2s_mod_reg(struct s6000_i2s_dev *dev, int reg, - u32 mask, u32 val) -{ - val ^= s6_i2s_read_reg(dev, reg) & ~mask; - s6_i2s_write_reg(dev, reg, val); -} - -static void s6000_i2s_start_channel(struct s6000_i2s_dev *dev, int channel) -{ - int i, j, cur, prev; - - /* - * Wait for WCLK to toggle 5 times before enabling the channel - * s6000 Family Datasheet 3.6.4: - * "At least two cycles of WS must occur between commands - * to disable or enable the interface" - */ - j = 0; - prev = ~S6_I2S_CUR_WS; - for (i = 1000000; --i && j < 6; ) { - cur = s6_i2s_read_reg(dev, S6_I2S_INTERFACE_CFG(channel)) - & S6_I2S_CUR_WS; - if (prev != cur) { - prev = cur; - j++; - } - } - if (j < 6) - printk(KERN_WARNING "s6000-i2s: timeout waiting for WCLK\n"); - - s6_i2s_write_reg(dev, S6_I2S_ENABLE(channel), S6_I2S_ENABLE_IF); -} - -static void s6000_i2s_stop_channel(struct s6000_i2s_dev *dev, int channel) -{ - s6_i2s_write_reg(dev, S6_I2S_ENABLE(channel), S6_I2S_DISABLE_IF); -} - -static void s6000_i2s_start(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s6000_i2s_dev *dev = snd_soc_dai_get_drvdata(rtd->cpu_dai); - int channel; - - channel = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? - dev->channel_out : dev->channel_in; - - s6000_i2s_start_channel(dev, channel); -} - -static void s6000_i2s_stop(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s6000_i2s_dev *dev = snd_soc_dai_get_drvdata(rtd->cpu_dai); - int channel; - - channel = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? - dev->channel_out : dev->channel_in; - - s6000_i2s_stop_channel(dev, channel); -} - -static int s6000_i2s_trigger(struct snd_pcm_substream *substream, int cmd, - int after) -{ - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if ((substream->stream == SNDRV_PCM_STREAM_CAPTURE) ^ !after) - s6000_i2s_start(substream); - break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (!after) - s6000_i2s_stop(substream); - } - return 0; -} - -static unsigned int s6000_i2s_int_sources(struct s6000_i2s_dev *dev) -{ - unsigned int pending; - pending = s6_i2s_read_reg(dev, S6_I2S_INTERRUPT_RAW); - pending &= S6_I2S_INT_ALIGNMENT | - S6_I2S_INT_UNDERRUN | - S6_I2S_INT_OVERRUN; - s6_i2s_write_reg(dev, S6_I2S_INTERRUPT_CLEAR, pending); - - return pending; -} - -static unsigned int s6000_i2s_check_xrun(struct snd_soc_dai *cpu_dai) -{ - struct s6000_i2s_dev *dev = snd_soc_dai_get_drvdata(cpu_dai); - unsigned int errors; - unsigned int ret; - - errors = s6000_i2s_int_sources(dev); - if (likely(!errors)) - return 0; - - ret = 0; - if (errors & S6_I2S_INT_ALIGNMENT) - printk(KERN_ERR "s6000-i2s: WCLK misaligned\n"); - if (errors & S6_I2S_INT_UNDERRUN) - ret |= 1 << SNDRV_PCM_STREAM_PLAYBACK; - if (errors & S6_I2S_INT_OVERRUN) - ret |= 1 << SNDRV_PCM_STREAM_CAPTURE; - return ret; -} - -static void s6000_i2s_wait_disabled(struct s6000_i2s_dev *dev) -{ - int channel; - int n = 50; - for (channel = 0; channel < 2; channel++) { - while (--n >= 0) { - int v = s6_i2s_read_reg(dev, S6_I2S_ENABLE(channel)); - if ((v & S6_I2S_IS_ENABLED) - || !(v & (S6_I2S_DMA_ACTIVE | S6_I2S_IS_BUSY))) - break; - udelay(20); - } - } - if (n < 0) - printk(KERN_WARNING "s6000-i2s: timeout disabling interfaces"); -} - -static int s6000_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, - unsigned int fmt) -{ - struct s6000_i2s_dev *dev = snd_soc_dai_get_drvdata(cpu_dai); - u32 w; - - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - w = S6_I2S_SCK_IN | S6_I2S_WS_IN; - break; - case SND_SOC_DAIFMT_CBS_CFM: - w = S6_I2S_SCK_OUT | S6_I2S_WS_IN; - break; - case SND_SOC_DAIFMT_CBM_CFS: - w = S6_I2S_SCK_IN | S6_I2S_WS_OUT; - break; - case SND_SOC_DAIFMT_CBS_CFS: - w = S6_I2S_SCK_OUT | S6_I2S_WS_OUT; - break; - default: - return -EINVAL; - } - - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_NB_NF: - w |= S6_I2S_LEFT_FIRST; - break; - case SND_SOC_DAIFMT_NB_IF: - w |= S6_I2S_RIGHT_FIRST; - break; - default: - return -EINVAL; - } - - s6_i2s_mod_reg(dev, S6_I2S_INTERFACE_CFG(0), - S6_I2S_FIRST | S6_I2S_WS_DIR | S6_I2S_SCK_DIR, w); - s6_i2s_mod_reg(dev, S6_I2S_INTERFACE_CFG(1), - S6_I2S_FIRST | S6_I2S_WS_DIR | S6_I2S_SCK_DIR, w); - - return 0; -} - -static int s6000_i2s_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) -{ - struct s6000_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); - - if (!div || (div & 1) || div > (S6_I2S_DIV_MASK + 1) * 2) - return -EINVAL; - - s6_i2s_mod_reg(dev, S6_I2S_INTERFACE_CFG(div_id), - S6_I2S_DIV_MASK, div / 2 - 1); - return 0; -} - -static int s6000_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct s6000_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); - int interf; - u32 w = 0; - - if (dev->wide) - interf = 0; - else { - w |= (((params_channels(params) - 2) / 2) - << S6_I2S_CHANNELS_SHIFT) & S6_I2S_CHANNELS_MASK; - interf = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - ? dev->channel_out : dev->channel_in; - } - - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: - w |= S6_I2S_16BIT | S6_I2S_MEM_16BIT; - break; - case SNDRV_PCM_FORMAT_S32_LE: - w |= S6_I2S_32BIT | S6_I2S_MEM_32BIT; - break; - default: - printk(KERN_WARNING "s6000-i2s: unsupported PCM format %x\n", - params_format(params)); - return -EINVAL; - } - - if (s6_i2s_read_reg(dev, S6_I2S_INTERFACE_CFG(interf)) - & S6_I2S_IS_ENABLED) { - printk(KERN_ERR "s6000-i2s: interface already enabled\n"); - return -EBUSY; - } - - s6_i2s_mod_reg(dev, S6_I2S_INTERFACE_CFG(interf), - S6_I2S_CHANNELS_MASK|S6_I2S_MEM_MASK|S6_I2S_BITS_MASK, - w); - - return 0; -} - -static int s6000_i2s_dai_probe(struct snd_soc_dai *dai) -{ - struct s6000_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); - struct s6000_snd_platform_data *pdata = dai->dev->platform_data; - - if (!pdata) - return -EINVAL; - - dai->capture_dma_data = &dev->dma_params; - dai->playback_dma_data = &dev->dma_params; - - dev->wide = pdata->wide; - dev->channel_in = pdata->channel_in; - dev->channel_out = pdata->channel_out; - dev->lines_in = pdata->lines_in; - dev->lines_out = pdata->lines_out; - - s6_i2s_write_reg(dev, S6_I2S_MODE, - dev->wide ? S6_I2S_WIDE : S6_I2S_DUAL); - - if (dev->wide) { - int i; - - if (dev->lines_in + dev->lines_out > S6_I2S_NUM_LINES) - return -EINVAL; - - dev->channel_in = 0; - dev->channel_out = 1; - dai->driver->capture.channels_min = 2 * dev->lines_in; - dai->driver->capture.channels_max = dai->driver->capture.channels_min; - dai->driver->playback.channels_min = 2 * dev->lines_out; - dai->driver->playback.channels_max = dai->driver->playback.channels_min; - - for (i = 0; i < dev->lines_out; i++) - s6_i2s_write_reg(dev, S6_I2S_DATA_CFG(i), S6_I2S_OUT); - - for (; i < S6_I2S_NUM_LINES - dev->lines_in; i++) - s6_i2s_write_reg(dev, S6_I2S_DATA_CFG(i), - S6_I2S_UNUSED); - - for (; i < S6_I2S_NUM_LINES; i++) - s6_i2s_write_reg(dev, S6_I2S_DATA_CFG(i), S6_I2S_IN); - } else { - unsigned int cfg[2] = {S6_I2S_UNUSED, S6_I2S_UNUSED}; - - if (dev->lines_in > 1 || dev->lines_out > 1) - return -EINVAL; - - dai->driver->capture.channels_min = 2 * dev->lines_in; - dai->driver->capture.channels_max = 8 * dev->lines_in; - dai->driver->playback.channels_min = 2 * dev->lines_out; - dai->driver->playback.channels_max = 8 * dev->lines_out; - - if (dev->lines_in) - cfg[dev->channel_in] = S6_I2S_IN; - if (dev->lines_out) - cfg[dev->channel_out] = S6_I2S_OUT; - - s6_i2s_write_reg(dev, S6_I2S_DATA_CFG(0), cfg[0]); - s6_i2s_write_reg(dev, S6_I2S_DATA_CFG(1), cfg[1]); - } - - if (dev->lines_out) { - if (dev->lines_in) { - if (!dev->dma_params.dma_out) - return -ENODEV; - } else { - dev->dma_params.dma_out = dev->dma_params.dma_in; - dev->dma_params.dma_in = 0; - } - } - dev->dma_params.sif_in = dev->sifbase + (dev->channel_in ? - S6_I2S_SIF_PORT1 : S6_I2S_SIF_PORT0); - dev->dma_params.sif_out = dev->sifbase + (dev->channel_out ? - S6_I2S_SIF_PORT1 : S6_I2S_SIF_PORT0); - dev->dma_params.same_rate = pdata->same_rate | pdata->wide; - return 0; -} - -#define S6000_I2S_RATES SNDRV_PCM_RATE_CONTINUOUS -#define S6000_I2S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE) - -static const struct snd_soc_dai_ops s6000_i2s_dai_ops = { - .set_fmt = s6000_i2s_set_dai_fmt, - .set_clkdiv = s6000_i2s_set_clkdiv, - .hw_params = s6000_i2s_hw_params, -}; - -static struct snd_soc_dai_driver s6000_i2s_dai = { - .probe = s6000_i2s_dai_probe, - .playback = { - .channels_min = 2, - .channels_max = 8, - .formats = S6000_I2S_FORMATS, - .rates = S6000_I2S_RATES, - .rate_min = 0, - .rate_max = 1562500, - }, - .capture = { - .channels_min = 2, - .channels_max = 8, - .formats = S6000_I2S_FORMATS, - .rates = S6000_I2S_RATES, - .rate_min = 0, - .rate_max = 1562500, - }, - .ops = &s6000_i2s_dai_ops, -}; - -static const struct snd_soc_component_driver s6000_i2s_component = { - .name = "s6000-i2s", -}; - -static int s6000_i2s_probe(struct platform_device *pdev) -{ - struct s6000_i2s_dev *dev; - struct resource *scbmem, *sifmem, *region, *dma1, *dma2; - u8 __iomem *mmio; - int ret; - - scbmem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!scbmem) { - dev_err(&pdev->dev, "no mem resource?\n"); - ret = -ENODEV; - goto err_release_none; - } - - region = request_mem_region(scbmem->start, resource_size(scbmem), - pdev->name); - if (!region) { - dev_err(&pdev->dev, "I2S SCB region already claimed\n"); - ret = -EBUSY; - goto err_release_none; - } - - mmio = ioremap(scbmem->start, resource_size(scbmem)); - if (!mmio) { - dev_err(&pdev->dev, "can't ioremap SCB region\n"); - ret = -ENOMEM; - goto err_release_scb; - } - - sifmem = platform_get_resource(pdev, IORESOURCE_MEM, 1); - if (!sifmem) { - dev_err(&pdev->dev, "no second mem resource?\n"); - ret = -ENODEV; - goto err_release_map; - } - - region = request_mem_region(sifmem->start, resource_size(sifmem), - pdev->name); - if (!region) { - dev_err(&pdev->dev, "I2S SIF region already claimed\n"); - ret = -EBUSY; - goto err_release_map; - } - - dma1 = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!dma1) { - dev_err(&pdev->dev, "no dma resource?\n"); - ret = -ENODEV; - goto err_release_sif; - } - - region = request_mem_region(dma1->start, resource_size(dma1), - pdev->name); - if (!region) { - dev_err(&pdev->dev, "I2S DMA region already claimed\n"); - ret = -EBUSY; - goto err_release_sif; - } - - dma2 = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (dma2) { - region = request_mem_region(dma2->start, resource_size(dma2), - pdev->name); - if (!region) { - dev_err(&pdev->dev, - "I2S DMA region already claimed\n"); - ret = -EBUSY; - goto err_release_dma1; - } - } - - dev = kzalloc(sizeof(struct s6000_i2s_dev), GFP_KERNEL); - if (!dev) { - ret = -ENOMEM; - goto err_release_dma2; - } - dev_set_drvdata(&pdev->dev, dev); - - dev->sifbase = sifmem->start; - dev->scbbase = mmio; - - s6_i2s_write_reg(dev, S6_I2S_INTERRUPT_ENABLE, 0); - s6_i2s_write_reg(dev, S6_I2S_INTERRUPT_CLEAR, - S6_I2S_INT_ALIGNMENT | - S6_I2S_INT_UNDERRUN | - S6_I2S_INT_OVERRUN); - - s6000_i2s_stop_channel(dev, 0); - s6000_i2s_stop_channel(dev, 1); - s6000_i2s_wait_disabled(dev); - - dev->dma_params.check_xrun = s6000_i2s_check_xrun; - dev->dma_params.trigger = s6000_i2s_trigger; - dev->dma_params.dma_in = dma1->start; - dev->dma_params.dma_out = dma2 ? dma2->start : 0; - dev->dma_params.irq = platform_get_irq(pdev, 0); - if (dev->dma_params.irq < 0) { - dev_err(&pdev->dev, "no irq resource?\n"); - ret = -ENODEV; - goto err_release_dev; - } - - s6_i2s_write_reg(dev, S6_I2S_INTERRUPT_ENABLE, - S6_I2S_INT_ALIGNMENT | - S6_I2S_INT_UNDERRUN | - S6_I2S_INT_OVERRUN); - - ret = snd_soc_register_component(&pdev->dev, &s6000_i2s_component, - &s6000_i2s_dai, 1); - if (ret) - goto err_release_dev; - - return 0; - -err_release_dev: - kfree(dev); -err_release_dma2: - if (dma2) - release_mem_region(dma2->start, resource_size(dma2)); -err_release_dma1: - release_mem_region(dma1->start, resource_size(dma1)); -err_release_sif: - release_mem_region(sifmem->start, resource_size(sifmem)); -err_release_map: - iounmap(mmio); -err_release_scb: - release_mem_region(scbmem->start, resource_size(scbmem)); -err_release_none: - return ret; -} - -static int s6000_i2s_remove(struct platform_device *pdev) -{ - struct s6000_i2s_dev *dev = dev_get_drvdata(&pdev->dev); - struct resource *region; - void __iomem *mmio = dev->scbbase; - - snd_soc_unregister_component(&pdev->dev); - - s6000_i2s_stop_channel(dev, 0); - s6000_i2s_stop_channel(dev, 1); - - s6_i2s_write_reg(dev, S6_I2S_INTERRUPT_ENABLE, 0); - kfree(dev); - - region = platform_get_resource(pdev, IORESOURCE_DMA, 0); - release_mem_region(region->start, resource_size(region)); - - region = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (region) - release_mem_region(region->start, resource_size(region)); - - region = platform_get_resource(pdev, IORESOURCE_MEM, 0); - release_mem_region(region->start, resource_size(region)); - - iounmap(mmio); - region = platform_get_resource(pdev, IORESOURCE_IO, 0); - release_mem_region(region->start, resource_size(region)); - - return 0; -} - -static struct platform_driver s6000_i2s_driver = { - .probe = s6000_i2s_probe, - .remove = s6000_i2s_remove, - .driver = { - .name = "s6000-i2s", - .owner = THIS_MODULE, - }, -}; - -module_platform_driver(s6000_i2s_driver); - -MODULE_AUTHOR("Daniel Gloeckner"); -MODULE_DESCRIPTION("Stretch s6000 family I2S SoC Interface"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s6000/s6000-i2s.h b/sound/soc/s6000/s6000-i2s.h deleted file mode 100644 index 86aa192..0000000 --- a/sound/soc/s6000/s6000-i2s.h +++ /dev/null @@ -1,23 +0,0 @@ -/* - * ALSA SoC I2S Audio Layer for the Stretch s6000 family - * - * Author: Daniel Gloeckner, <dg(a)emlix.com> - * Copyright: (C) 2009 emlix GmbH <info(a)emlix.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef _S6000_I2S_H -#define _S6000_I2S_H - -struct s6000_snd_platform_data { - int lines_in; - int lines_out; - int channel_in; - int channel_out; - int wide; - int same_rate; -}; -#endif diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c deleted file mode 100644 index fb8461e..0000000 --- a/sound/soc/s6000/s6000-pcm.c +++ /dev/null @@ -1,521 +0,0 @@ -/* - * ALSA PCM interface for the Stetch s6000 family - * - * Author: Daniel Gloeckner, <dg(a)emlix.com> - * Copyright: (C) 2009 emlix GmbH <info(a)emlix.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#include <linux/module.h> -#include <linux/init.h> -#include <linux/platform_device.h> -#include <linux/slab.h> -#include <linux/dma-mapping.h> -#include <linux/interrupt.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> - -#include <asm/dma.h> -#include <variant/dmac.h> - -#include "s6000-pcm.h" - -#define S6_PCM_PREALLOCATE_SIZE (96 * 1024) -#define S6_PCM_PREALLOCATE_MAX (2048 * 1024) - -static struct snd_pcm_hardware s6000_pcm_hardware = { - .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_JOINT_DUPLEX), - .buffer_bytes_max = 0x7ffffff0, - .period_bytes_min = 16, - .period_bytes_max = 0xfffff0, - .periods_min = 2, - .periods_max = 1024, /* no limit */ - .fifo_size = 0, -}; - -struct s6000_runtime_data { - spinlock_t lock; - int period; /* current DMA period */ -}; - -static void s6000_pcm_enqueue_dma(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct s6000_runtime_data *prtd = runtime->private_data; - struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; - struct s6000_pcm_dma_params *par; - int channel; - unsigned int period_size; - unsigned int dma_offset; - dma_addr_t dma_pos; - dma_addr_t src, dst; - - par = snd_soc_dai_get_dma_data(soc_runtime->cpu_dai, substream); - - period_size = snd_pcm_lib_period_bytes(substream); - dma_offset = prtd->period * period_size; - dma_pos = runtime->dma_addr + dma_offset; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - src = dma_pos; - dst = par->sif_out; - channel = par->dma_out; - } else { - src = par->sif_in; - dst = dma_pos; - channel = par->dma_in; - } - - if (!s6dmac_channel_enabled(DMA_MASK_DMAC(channel), - DMA_INDEX_CHNL(channel))) - return; - - if (s6dmac_fifo_full(DMA_MASK_DMAC(channel), DMA_INDEX_CHNL(channel))) { - printk(KERN_ERR "s6000-pcm: fifo full\n"); - return; - } - - if (WARN_ON(period_size & 15)) - return; - s6dmac_put_fifo(DMA_MASK_DMAC(channel), DMA_INDEX_CHNL(channel), - src, dst, period_size); - - prtd->period++; - if (unlikely(prtd->period >= runtime->periods)) - prtd->period = 0; -} - -static irqreturn_t s6000_pcm_irq(int irq, void *data) -{ - struct snd_pcm *pcm = data; - struct snd_soc_pcm_runtime *runtime = pcm->private_data; - struct s6000_runtime_data *prtd; - unsigned int has_xrun; - int i, ret = IRQ_NONE; - - for (i = 0; i < 2; ++i) { - struct snd_pcm_substream *substream = pcm->streams[i].substream; - struct s6000_pcm_dma_params *params = - snd_soc_dai_get_dma_data(runtime->cpu_dai, substream); - u32 channel; - unsigned int pending; - - if (substream == SNDRV_PCM_STREAM_PLAYBACK) - channel = params->dma_out; - else - channel = params->dma_in; - - has_xrun = params->check_xrun(runtime->cpu_dai); - - if (!channel) - continue; - - if (unlikely(has_xrun & (1 << i)) && - substream->runtime && - snd_pcm_running(substream)) { - dev_dbg(pcm->dev, "xrun\n"); - snd_pcm_stream_lock(substream); - snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock(substream); - ret = IRQ_HANDLED; - } - - pending = s6dmac_int_sources(DMA_MASK_DMAC(channel), - DMA_INDEX_CHNL(channel)); - - if (pending & 1) { - ret = IRQ_HANDLED; - if (likely(substream->runtime && - snd_pcm_running(substream))) { - snd_pcm_period_elapsed(substream); - dev_dbg(pcm->dev, "period elapsed %x %x\n", - s6dmac_cur_src(DMA_MASK_DMAC(channel), - DMA_INDEX_CHNL(channel)), - s6dmac_cur_dst(DMA_MASK_DMAC(channel), - DMA_INDEX_CHNL(channel))); - prtd = substream->runtime->private_data; - spin_lock(&prtd->lock); - s6000_pcm_enqueue_dma(substream); - spin_unlock(&prtd->lock); - } - } - - if (unlikely(pending & ~7)) { - if (pending & (1 << 3)) - printk(KERN_WARNING - "s6000-pcm: DMA %x Underflow\n", - channel); - if (pending & (1 << 4)) - printk(KERN_WARNING - "s6000-pcm: DMA %x Overflow\n", - channel); - if (pending & 0x1e0) - printk(KERN_WARNING - "s6000-pcm: DMA %x Master Error " - "(mask %x)\n", - channel, pending >> 5); - - } - } - - return ret; -} - -static int s6000_pcm_start(struct snd_pcm_substream *substream) -{ - struct s6000_runtime_data *prtd = substream->runtime->private_data; - struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; - struct s6000_pcm_dma_params *par; - unsigned long flags; - int srcinc; - u32 dma; - - par = snd_soc_dai_get_dma_data(soc_runtime->cpu_dai, substream); - - spin_lock_irqsave(&prtd->lock, flags); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - srcinc = 1; - dma = par->dma_out; - } else { - srcinc = 0; - dma = par->dma_in; - } - s6dmac_enable_chan(DMA_MASK_DMAC(dma), DMA_INDEX_CHNL(dma), - 1 /* priority 1 (0 is max) */, - 0 /* peripheral requests w/o xfer length mode */, - srcinc /* source address increment */, - srcinc^1 /* destination address increment */, - 0 /* chunksize 0 (skip impossible on this dma) */, - 0 /* source skip after chunk (impossible) */, - 0 /* destination skip after chunk (impossible) */, - 4 /* 16 byte burst size */, - -1 /* don't conserve bandwidth */, - 0 /* low watermark irq descriptor threshold */, - 0 /* disable hardware timestamps */, - 1 /* enable channel */); - - s6000_pcm_enqueue_dma(substream); - s6000_pcm_enqueue_dma(substream); - - spin_unlock_irqrestore(&prtd->lock, flags); - - return 0; -} - -static int s6000_pcm_stop(struct snd_pcm_substream *substream) -{ - struct s6000_runtime_data *prtd = substream->runtime->private_data; - struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; - struct s6000_pcm_dma_params *par; - unsigned long flags; - u32 channel; - - par = snd_soc_dai_get_dma_data(soc_runtime->cpu_dai, substream); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - channel = par->dma_out; - else - channel = par->dma_in; - - s6dmac_set_terminal_count(DMA_MASK_DMAC(channel), - DMA_INDEX_CHNL(channel), 0); - - spin_lock_irqsave(&prtd->lock, flags); - - s6dmac_disable_chan(DMA_MASK_DMAC(channel), DMA_INDEX_CHNL(channel)); - - spin_unlock_irqrestore(&prtd->lock, flags); - - return 0; -} - -static int s6000_pcm_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; - struct s6000_pcm_dma_params *par; - int ret; - - par = snd_soc_dai_get_dma_data(soc_runtime->cpu_dai, substream); - - ret = par->trigger(substream, cmd, 0); - if (ret < 0) - return ret; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - ret = s6000_pcm_start(substream); - break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - ret = s6000_pcm_stop(substream); - break; - default: - ret = -EINVAL; - } - if (ret < 0) - return ret; - - return par->trigger(substream, cmd, 1); -} - -static int s6000_pcm_prepare(struct snd_pcm_substream *substream) -{ - struct s6000_runtime_data *prtd = substream->runtime->private_data; - - prtd->period = 0; - - return 0; -} - -static snd_pcm_uframes_t s6000_pcm_pointer(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; - struct s6000_pcm_dma_params *par; - struct snd_pcm_runtime *runtime = substream->runtime; - struct s6000_runtime_data *prtd = runtime->private_data; - unsigned long flags; - unsigned int offset; - dma_addr_t count; - - par = snd_soc_dai_get_dma_data(soc_runtime->cpu_dai, substream); - - spin_lock_irqsave(&prtd->lock, flags); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - count = s6dmac_cur_src(DMA_MASK_DMAC(par->dma_out), - DMA_INDEX_CHNL(par->dma_out)); - else - count = s6dmac_cur_dst(DMA_MASK_DMAC(par->dma_in), - DMA_INDEX_CHNL(par->dma_in)); - - count -= runtime->dma_addr; - - spin_unlock_irqrestore(&prtd->lock, flags); - - offset = bytes_to_frames(runtime, count); - if (unlikely(offset >= runtime->buffer_size)) - offset = 0; - - return offset; -} - -static int s6000_pcm_open(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; - struct s6000_pcm_dma_params *par; - struct snd_pcm_runtime *runtime = substream->runtime; - struct s6000_runtime_data *prtd; - int ret; - - par = snd_soc_dai_get_dma_data(soc_runtime->cpu_dai, substream); - snd_soc_set_runtime_hwparams(substream, &s6000_pcm_hardware); - - ret = snd_pcm_hw_constraint_step(runtime, 0, - SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 16); - if (ret < 0) - return ret; - ret = snd_pcm_hw_constraint_step(runtime, 0, - SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 16); - if (ret < 0) - return ret; - ret = snd_pcm_hw_constraint_integer(runtime, - SNDRV_PCM_HW_PARAM_PERIODS); - if (ret < 0) - return ret; - - if (par->same_rate) { - int rate; - spin_lock(&par->lock); /* needed? */ - rate = par->rate; - spin_unlock(&par->lock); - if (rate != -1) { - ret = snd_pcm_hw_constraint_minmax(runtime, - SNDRV_PCM_HW_PARAM_RATE, - rate, rate); - if (ret < 0) - return ret; - } - } - - prtd = kzalloc(sizeof(struct s6000_runtime_data), GFP_KERNEL); - if (prtd == NULL) - return -ENOMEM; - - spin_lock_init(&prtd->lock); - - runtime->private_data = prtd; - - return 0; -} - -static int s6000_pcm_close(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct s6000_runtime_data *prtd = runtime->private_data; - - kfree(prtd); - - return 0; -} - -static int s6000_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params) -{ - struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; - struct s6000_pcm_dma_params *par; - int ret; - ret = snd_pcm_lib_malloc_pages(substream, - params_buffer_bytes(hw_params)); - if (ret < 0) { - printk(KERN_WARNING "s6000-pcm: allocation of memory failed\n"); - return ret; - } - - par = snd_soc_dai_get_dma_data(soc_runtime->cpu_dai, substream); - - if (par->same_rate) { - spin_lock(&par->lock); - if (par->rate == -1 || - !(par->in_use & ~(1 << substream->stream))) { - par->rate = params_rate(hw_params); - par->in_use |= 1 << substream->stream; - } else if (params_rate(hw_params) != par->rate) { - snd_pcm_lib_free_pages(substream); - par->in_use &= ~(1 << substream->stream); - ret = -EBUSY; - } - spin_unlock(&par->lock); - } - return ret; -} - -static int s6000_pcm_hw_free(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; - struct s6000_pcm_dma_params *par = - snd_soc_dai_get_dma_data(soc_runtime->cpu_dai, substream); - - spin_lock(&par->lock); - par->in_use &= ~(1 << substream->stream); - if (!par->in_use) - par->rate = -1; - spin_unlock(&par->lock); - - return snd_pcm_lib_free_pages(substream); -} - -static struct snd_pcm_ops s6000_pcm_ops = { - .open = s6000_pcm_open, - .close = s6000_pcm_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = s6000_pcm_hw_params, - .hw_free = s6000_pcm_hw_free, - .trigger = s6000_pcm_trigger, - .prepare = s6000_pcm_prepare, - .pointer = s6000_pcm_pointer, -}; - -static void s6000_pcm_free(struct snd_pcm *pcm) -{ - struct snd_soc_pcm_runtime *runtime = pcm->private_data; - struct s6000_pcm_dma_params *params = - snd_soc_dai_get_dma_data(runtime->cpu_dai, - pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream); - - free_irq(params->irq, pcm); - snd_pcm_lib_preallocate_free_for_all(pcm); -} - -static int s6000_pcm_new(struct snd_soc_pcm_runtime *runtime) -{ - struct snd_card *card = runtime->card->snd_card; - struct snd_pcm *pcm = runtime->pcm; - struct s6000_pcm_dma_params *params; - int res; - - params = snd_soc_dai_get_dma_data(runtime->cpu_dai, - pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream); - - res = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); - if (res) - return res; - - if (params->dma_in) { - s6dmac_disable_chan(DMA_MASK_DMAC(params->dma_in), - DMA_INDEX_CHNL(params->dma_in)); - s6dmac_int_sources(DMA_MASK_DMAC(params->dma_in), - DMA_INDEX_CHNL(params->dma_in)); - } - - if (params->dma_out) { - s6dmac_disable_chan(DMA_MASK_DMAC(params->dma_out), - DMA_INDEX_CHNL(params->dma_out)); - s6dmac_int_sources(DMA_MASK_DMAC(params->dma_out), - DMA_INDEX_CHNL(params->dma_out)); - } - - res = request_irq(params->irq, s6000_pcm_irq, IRQF_SHARED, - "s6000-audio", pcm); - if (res) { - printk(KERN_ERR "s6000-pcm couldn't get IRQ\n"); - return res; - } - - res = snd_pcm_lib_preallocate_pages_for_all(pcm, - SNDRV_DMA_TYPE_DEV, - card->dev, - S6_PCM_PREALLOCATE_SIZE, - S6_PCM_PREALLOCATE_MAX); - if (res) - printk(KERN_WARNING "s6000-pcm: preallocation failed\n"); - - spin_lock_init(&params->lock); - params->in_use = 0; - params->rate = -1; - return 0; -} - -static struct snd_soc_platform_driver s6000_soc_platform = { - .ops = &s6000_pcm_ops, - .pcm_new = s6000_pcm_new, - .pcm_free = s6000_pcm_free, -}; - -static int s6000_soc_platform_probe(struct platform_device *pdev) -{ - return snd_soc_register_platform(&pdev->dev, &s6000_soc_platform); -} - -static int s6000_soc_platform_remove(struct platform_device *pdev) -{ - snd_soc_unregister_platform(&pdev->dev); - return 0; -} - -static struct platform_driver s6000_pcm_driver = { - .driver = { - .name = "s6000-pcm-audio", - .owner = THIS_MODULE, - }, - - .probe = s6000_soc_platform_probe, - .remove = s6000_soc_platform_remove, -}; - -module_platform_driver(s6000_pcm_driver); - -MODULE_AUTHOR("Daniel Gloeckner"); -MODULE_DESCRIPTION("Stretch s6000 family PCM DMA module"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s6000/s6000-pcm.h b/sound/soc/s6000/s6000-pcm.h deleted file mode 100644 index 09d9b88..0000000 --- a/sound/soc/s6000/s6000-pcm.h +++ /dev/null @@ -1,33 +0,0 @@ -/* - * ALSA PCM interface for the Stretch s6000 family - * - * Author: Daniel Gloeckner, <dg(a)emlix.com> - * Copyright: (C) 2009 emlix GmbH <info(a)emlix.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef _S6000_PCM_H -#define _S6000_PCM_H - -struct snd_soc_dai; -struct snd_pcm_substream; - -struct s6000_pcm_dma_params { - unsigned int (*check_xrun)(struct snd_soc_dai *cpu_dai); - int (*trigger)(struct snd_pcm_substream *substream, int cmd, int after); - dma_addr_t sif_in; - dma_addr_t sif_out; - u32 dma_in; - u32 dma_out; - int irq; - int same_rate; - - spinlock_t lock; - int in_use; - int rate; -}; - -#endif diff --git a/sound/soc/s6000/s6105-ipcam.c b/sound/soc/s6000/s6105-ipcam.c deleted file mode 100644 index 3510c01..0000000 --- a/sound/soc/s6000/s6105-ipcam.c +++ /dev/null @@ -1,221 +0,0 @@ -/* - * ASoC driver for Stretch s6105 IP camera platform - * - * Author: Daniel Gloeckner, <dg(a)emlix.com> - * Copyright: (C) 2009 emlix GmbH <info(a)emlix.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/timer.h> -#include <linux/interrupt.h> -#include <linux/platform_device.h> -#include <linux/i2c.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> - -#include "s6000-pcm.h" -#include "s6000-i2s.h" - -#define S6105_CAM_CODEC_CLOCK 12288000 - -static int s6105_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int ret = 0; - - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBM_CFM | - SND_SOC_DAIFMT_NB_NF); - if (ret < 0) - return ret; - - /* set the codec system clock */ - ret = snd_soc_dai_set_sysclk(codec_dai, 0, S6105_CAM_CODEC_CLOCK, - SND_SOC_CLOCK_OUT); - if (ret < 0) - return ret; - - return 0; -} - -static struct snd_soc_ops s6105_ops = { - .hw_params = s6105_hw_params, -}; - -/* s6105 machine dapm widgets */ -static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { - SND_SOC_DAPM_LINE("Audio Out Differential", NULL), - SND_SOC_DAPM_LINE("Audio Out Stereo", NULL), - SND_SOC_DAPM_LINE("Audio In", NULL), -}; - -/* s6105 machine audio_mapnections to the codec pins */ -static const struct snd_soc_dapm_route audio_map[] = { - /* Audio Out connected to HPLOUT, HPLCOM, HPROUT */ - {"Audio Out Differential", NULL, "HPLOUT"}, - {"Audio Out Differential", NULL, "HPLCOM"}, - {"Audio Out Stereo", NULL, "HPLOUT"}, - {"Audio Out Stereo", NULL, "HPROUT"}, - - /* Audio In connected to LINE1L, LINE1R */ - {"LINE1L", NULL, "Audio In"}, - {"LINE1R", NULL, "Audio In"}, -}; - -static int output_type_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 2; - if (uinfo->value.enumerated.item) { - uinfo->value.enumerated.item = 1; - strcpy(uinfo->value.enumerated.name, "HPLOUT/HPROUT"); - } else { - strcpy(uinfo->value.enumerated.name, "HPLOUT/HPLCOM"); - } - return 0; -} - -static int output_type_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.enumerated.item[0] = kcontrol->private_value; - return 0; -} - -static int output_type_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_card *card = kcontrol->private_data; - struct snd_soc_dapm_context *dapm = &card->dapm; - unsigned int val = (ucontrol->value.enumerated.item[0] != 0); - char *differential = "Audio Out Differential"; - char *stereo = "Audio Out Stereo"; - - if (kcontrol->private_value == val) - return 0; - kcontrol->private_value = val; - snd_soc_dapm_disable_pin(dapm, val ? differential : stereo); - snd_soc_dapm_sync(dapm); - snd_soc_dapm_enable_pin(dapm, val ? stereo : differential); - snd_soc_dapm_sync(dapm); - - return 1; -} - -static const struct snd_kcontrol_new audio_out_mux = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Output Mux", - .index = 0, - .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, - .info = output_type_info, - .get = output_type_get, - .put = output_type_put, - .private_value = 1 /* default to stereo */ -}; - -/* Logic for a aic3x as connected on the s6105 ip camera ref design */ -static int s6105_aic3x_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_card *card = rtd->card; - - /* must correspond to audio_out_mux.private_value initializer */ - snd_soc_dapm_disable_pin(&card->dapm, "Audio Out Differential"); - - snd_ctl_add(card->snd_card, snd_ctl_new1(&audio_out_mux, card)); - - return 0; -} - -/* s6105 digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link s6105_dai = { - .name = "TLV320AIC31", - .stream_name = "AIC31", - .cpu_dai_name = "s6000-i2s", - .codec_dai_name = "tlv320aic3x-hifi", - .platform_name = "s6000-pcm-audio", - .codec_name = "tlv320aic3x-codec.0-001a", - .init = s6105_aic3x_init, - .ops = &s6105_ops, -}; - -/* s6105 audio machine driver */ -static struct snd_soc_card snd_soc_card_s6105 = { - .name = "Stretch IP Camera", - .owner = THIS_MODULE, - .dai_link = &s6105_dai, - .num_links = 1, - - .dapm_widgets = aic3x_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(aic3x_dapm_widgets), - .dapm_routes = audio_map, - .num_dapm_routes = ARRAY_SIZE(audio_map), - .fully_routed = true, -}; - -static struct s6000_snd_platform_data s6105_snd_data __initdata = { - .wide = 0, - .channel_in = 0, - .channel_out = 1, - .lines_in = 1, - .lines_out = 1, - .same_rate = 1, -}; - -static struct platform_device *s6105_snd_device; - -/* temporary i2c device creation until this can be moved into the machine - * support file. -*/ -static struct i2c_board_info i2c_device[] = { - { I2C_BOARD_INFO("tlv320aic33", 0x18), } -}; - -static int __init s6105_init(void) -{ - int ret; - - i2c_register_board_info(0, i2c_device, ARRAY_SIZE(i2c_device)); - - s6105_snd_device = platform_device_alloc("soc-audio", -1); - if (!s6105_snd_device) - return -ENOMEM; - - platform_set_drvdata(s6105_snd_device, &snd_soc_card_s6105); - platform_device_add_data(s6105_snd_device, &s6105_snd_data, - sizeof(s6105_snd_data)); - - ret = platform_device_add(s6105_snd_device); - if (ret) - platform_device_put(s6105_snd_device); - - return ret; -} - -static void __exit s6105_exit(void) -{ - platform_device_unregister(s6105_snd_device); -} - -module_init(s6105_init); -module_exit(s6105_exit); - -MODULE_AUTHOR("Daniel Gloeckner"); -MODULE_DESCRIPTION("Stretch s6105 IP camera ASoC driver"); -MODULE_LICENSE("GPL"); -- 2.1.2

2 1

[alsa-devel] [PATCH] ASoC: rt5677: add build dependency to spi
by Anatol Pomozov 20 Oct '14

20 Oct '14

Since 9cb715a9d4c the codec has a hardcoded dependency to spi. Add this dependency to Kconfig. It fixes buildbot compilation failure: sound/built-in.o: In function `spi_write': >> rt5677-spi.c:(.text+0x8265f): undefined reference to `spi_sync' sound/built-in.o: In function `rt5677_spi_driver_init': >> rt5677-spi.c:(.init.text+0x17db): undefined reference to `spi_register_driver' Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 8838838e..1ff565e 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -81,7 +81,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_RT5645 if I2C select SND_SOC_RT5651 if I2C select SND_SOC_RT5670 if I2C - select SND_SOC_RT5677 if I2C + select SND_SOC_RT5677 if I2C && SPI_MASTER select SND_SOC_SGTL5000 if I2C select SND_SOC_SI476X if MFD_SI476X_CORE select SND_SOC_SIRF_AUDIO_CODEC -- 2.1.0.rc2.206.gedb03e5

3 2

[alsa-devel] [PATCH] ALSA: hda - Fix inverted LED gpio setup for Lenovo Ideapad
by Takashi Iwai 20 Oct '14

20 Oct '14

We implemented in a wrong way for mute LED on Lenovo Ideapad; the bit must be flipped. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=16373 Cc: <stable(a)vger.kernel.org> # 3.15+ Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4c35490cf119..34b7bdb510c7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5613,9 +5613,9 @@ static void alc662_led_gpio1_mute_hook(void *private_data, int enabled) unsigned int oldval = spec->gpio_led; if (enabled) - spec->gpio_led &= ~0x01; - else spec->gpio_led |= 0x01; + else + spec->gpio_led &= ~0x01; if (spec->gpio_led != oldval) snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, spec->gpio_led); -- 2.1.2

1 0

[alsa-devel] Wolfson codec, problem on mixer levels on kernel 3.17.0
by Angelo Dureghello 20 Oct '14

20 Oct '14

Hi, moving from kernel 3.5.1 to 3.17, amixer cannot read certain mixer controls, giving Input/output error: [root@test ~]# amixer cget numid=12 numid=12,iface=MIXER,name='Capture Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=255,step=0 amixer: Control default element read error: Input/output error So i had to disable those controls in the codec code: ... //SOC_DOUBLE_R_TLV("Capture Volume", WM8758_LEFT_ADC_DIGITAL_VOL, // WM8758_RIGHT_ADC_DIGITAL_VOL, 0, 255, 0, adc_tlv), ... //SOC_SINGLE("DAC Playback Limiter Switch", WM8758_DAC_LIMITER_1, 8, 1, 0), //SOC_SINGLE("DAC Playback Limiter Decay", WM8758_DAC_LIMITER_1, 4, 10, 0), //SOC_SINGLE("DAC Playback Limiter Attack", WM8758_DAC_LIMITER_1, 0, 11, 0), ... //SOC_ENUM("EQ4 Bandwith", eq4_bw), //SOC_ENUM("EQ4 Cutoff", eq4_cutoff), //SOC_SINGLE_TLV("EQ4 Volume", WM8758_EQ4_PEAK_3, 0, 24, 1, eq_tlv), ... Is it possible sonething is changed in these macroes, so that the amixer cannot access the controls anymore ? Every help is appreciated, many thanks angelo -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

2 1

[alsa-devel] [PATCH] ALSA: hda - hdmi: Fix missing ELD change event on plug/unplug
by Anssi Hannula 20 Oct '14

20 Oct '14

The ELD ALSA control change event is sent by hdmi_present_sense() when eld_changed is true. Currently, it is only true when the ELD buffer contents have been modified. However, the user-visible ELD controls also change to a zero-length value and back when eld_valid is unset/set, and no event is currently sent in such cases (such as when unplugging or replugging a sink). Fix the code to always set eld_changed if eld_valid value is changed, and therefore to always send the change event when the user-visible value changes. Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> Cc: David Henningsson <david.henningsson(a)canonical.com> Cc: <stable(a)vger.kernel.org> # 3.9+ --- sound/pci/hda/patch_hdmi.c | 15 +++++++++------ 1 file changed, 9 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 39862e98551c..9dc9cf8c90e9 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1583,19 +1583,22 @@ static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) } } - if (pin_eld->eld_valid && !eld->eld_valid) { - update_eld = true; + if (pin_eld->eld_valid != eld->eld_valid) eld_changed = true; - } + + if (pin_eld->eld_valid && !eld->eld_valid) + update_eld = true; + if (update_eld) { bool old_eld_valid = pin_eld->eld_valid; pin_eld->eld_valid = eld->eld_valid; - eld_changed = pin_eld->eld_size != eld->eld_size || + if (pin_eld->eld_size != eld->eld_size || memcmp(pin_eld->eld_buffer, eld->eld_buffer, - eld->eld_size) != 0; - if (eld_changed) + eld->eld_size) != 0) { memcpy(pin_eld->eld_buffer, eld->eld_buffer, eld->eld_size); + eld_changed = true; + } pin_eld->eld_size = eld->eld_size; pin_eld->info = eld->info; -- 1.8.4.5

2 1

[alsa-devel] [PATCH 1/1] snd-usb-audio: Add support for Steinberg UR22 USB interface
by Vlad Catoi 20 Oct '14

20 Oct '14

Adding support for Steinberg UR22 USB interface via quirks table patch See Ubuntu bug report: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1317244 Also see threads: http://linux-audio.4202.n7.nabble.com/Support-for-Steinberg-UR22-Yamaha-USB… http://www.steinberg.net/forums/viewtopic.php?t=62290 Tested by at least 4 people judging by the threads. Did not test MIDI interface, but audio output and capture both are functional. Built 3.17 kernel with this driver on Ubuntu 14.04 & tested with mpg123 Patch applied to 3.13 Ubuntu kernel works well enough for daily use. Signed-off-by: Vlad Catoi <vladcatoi(a)gmail.com> --- sound/usb/quirks-table.h | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 223c47b..c657752 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -385,6 +385,36 @@ YAMAHA_DEVICE(0x105d, NULL), } }, { + USB_DEVICE(0x0499, 0x1509), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "Yamaha", */ + /* .product_name = "Steinberg UR22", */ + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 3, + .type = QUIRK_MIDI_YAMAHA + }, + { + .ifnum = 4, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = -1 + } + } + } +}, +{ USB_DEVICE(0x0499, 0x150a), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { /* .vendor_name = "Yamaha", */ -- 1.9.1

3 2

[alsa-devel] [PATCH 1/1] ALSA: ALC283 codec - Avoid pop noise on headphones during suspend/resume
by Harsha Priya 20 Oct '14

20 Oct '14

This patch sets the headphones mode to default before suspending which helps avoid the pop noise on headphones Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 69d1236..e839e7a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2884,6 +2884,9 @@ static void alc283_shutup(struct hda_codec *codec) alc_write_coef_idx(codec, 0x43, 0x9004); + /*depop hp during suspend*/ + alc_write_coef_idx(codec, 0x06, 0x2100); + snd_hda_codec_write(codec, hp_pin, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); -- 1.9.1

4 6

[alsa-devel] Direct Stream Digital (DSD/DST, One Bit Audio) HDMI pass-trough
by Andrej Falout 19 Oct '14

19 Oct '14

Hello, I am collecting requirements for implementation a DSD HDMI pass-trough, according to HDMI specs version 1.2 and 1.3a I am wondering if anyone previously worked on this, or has any interest in this functionality. Any advice or pointers are gratefully received. Here is my list of the core facts so far: 1. hda_eld.c in current kernel git ignores raw DSD (One Bit Audio) labelled AUDIO_CODING_TYPE_SACD ("SACD" because originally DSD formats where only used commercially in SACDs) and compressed DSD (DST) is labelled AUDIO_CODING_TYPE_DST (1) 2. It is my understanding that apart from "special handling" denoted in "Table 47 CEA Short Audio Descriptor for Audio Format Codes 9 to 13", (eg setting the format code, etc) this should be a reasonably straightforward stream pass-trough (2) 3. Typical use scenario for DSD over HDMI: an DVD/BluRay player that supports SACD (Super Audio CD) outputs a DSD stream over HDMI to a DSD capable receiver. For instance, my Oppo and Sony BluRay players both do that, and all my receivers (Onkyo, Pioneer, Marantz, Yamaha, NAD) have DSD DACs and I can play both stereo and surround music this way, without converting DSD to PCM. There is currently about 30 million DSD capable receivers on the planet, conservatively speaking (Just Onkyo makes 1 million a year, and makes DSD capable ones since 2003) 4. Wanted scenario: play DSD files using DSD enabled players (Linux: MPD, mplayer, VLC, Squeezeboxserver, etc / Windows: FooBar, JRiver, etc) to an HDMI audio card/device on a PC motherboard or add-on video card with HDMI, connected via HDMI cable to a DSD enabled receiver/DAC 5. Note that as far as I know, no proprietary drivers on Windows (ATI/AMD, Nvidia, Intel) support this; It will be the first time a device that is not a SACD player outputs DSD over HDMI, on any OS. I am assuming that this has nothing to do with hardware limitations of the HDMI chips used for video outputs (I could not find any reference for such assumption in any HDMI chips I examined specs for). Therefore, I assume that implementing this has no hardware prerequisites. 6. DSD format general info: http://en.wikipedia.org/wiki/Direct_Stream_Digital http://dsd-guide.com/ http://dsd-guide.com/what-dsd-audio-simple-explanation Thanks, Andrej Falout References: (1) https://git.kernel.org/cgit/linux/kernel/git/torvalds/linux.git/tree/sound/… - Same source file in Alsa GIT tree: http://git.alsa-project.org/?p=alsa-kernel.git;a=blob;f=sound/pci/hda/hda_e… (2) CEA Standard PDF, page 66, table 47. HDMI specs version 1.3a, pages 72-73 for DSD(One Bit Audio) and DST audio packet header and body formats <%2B385%2891%291520436>

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[alsa-devel] [PATCH] ASoC: rt5677: Add TDM channel mapping function
by Oder Chiou 18 Oct '14

18 Oct '14

It is for channel to slot mapping, and it is not only for 8 channels mapping, but also in 2, 4 and 6 channels mapping. If we want to use the 2 channels in the stereo2 adc path, we need to set the item "2/1/3/4" or "2/3/1/4". It also adds for stereo channel swap. It can map the sterero channels "L/R" to "R/L", "L/L" or R/R. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 241 +++++++++++++++++++++++++++++++++++++++++++--- sound/soc/codecs/rt5677.h | 38 +++++++- 2 files changed, 264 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index f6df276..8be899d 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -1834,6 +1834,87 @@ static SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5677_if4_adc_mux = SOC_DAPM_ENUM("IF4 ADC Source", rt5677_if4_adc_enum); +/* TDM IF1/2 ADC Data Selection */ /* MX-3B MX-40 [7:6][5:4][3:2][1:0] */ +static const char * const rt5677_if12_adc_swap_src[] = { + "L/R", "R/L", "L/L", "R/R" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5677_if1_adc1_swap_enum, RT5677_TDM1_CTRL1, + RT5677_IF1_ADC1_SWAP_SFT, rt5677_if12_adc_swap_src); + +static const struct snd_kcontrol_new rt5677_if1_adc1_swap_mux = + SOC_DAPM_ENUM("IF1 ADC1 Swap Source", rt5677_if1_adc1_swap_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5677_if1_adc2_swap_enum, RT5677_TDM1_CTRL1, + RT5677_IF1_ADC2_SWAP_SFT, rt5677_if12_adc_swap_src); + +static const struct snd_kcontrol_new rt5677_if1_adc2_swap_mux = + SOC_DAPM_ENUM("IF1 ADC2 Swap Source", rt5677_if1_adc2_swap_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5677_if1_adc3_swap_enum, RT5677_TDM1_CTRL1, + RT5677_IF1_ADC3_SWAP_SFT, rt5677_if12_adc_swap_src); + +static const struct snd_kcontrol_new rt5677_if1_adc3_swap_mux = + SOC_DAPM_ENUM("IF1 ADC3 Swap Source", rt5677_if1_adc3_swap_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5677_if1_adc4_swap_enum, RT5677_TDM1_CTRL1, + RT5677_IF1_ADC4_SWAP_SFT, rt5677_if12_adc_swap_src); + +static const struct snd_kcontrol_new rt5677_if1_adc4_swap_mux = + SOC_DAPM_ENUM("IF1 ADC4 Swap Source", rt5677_if1_adc4_swap_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5677_if2_adc1_swap_enum, RT5677_TDM2_CTRL1, + RT5677_IF1_ADC2_SWAP_SFT, rt5677_if12_adc_swap_src); + +static const struct snd_kcontrol_new rt5677_if2_adc1_swap_mux = + SOC_DAPM_ENUM("IF1 ADC2 Swap Source", rt5677_if2_adc1_swap_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5677_if2_adc2_swap_enum, RT5677_TDM2_CTRL1, + RT5677_IF2_ADC2_SWAP_SFT, rt5677_if12_adc_swap_src); + +static const struct snd_kcontrol_new rt5677_if2_adc2_swap_mux = + SOC_DAPM_ENUM("IF2 ADC2 Swap Source", rt5677_if2_adc2_swap_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5677_if2_adc3_swap_enum, RT5677_TDM2_CTRL1, + RT5677_IF2_ADC3_SWAP_SFT, rt5677_if12_adc_swap_src); + +static const struct snd_kcontrol_new rt5677_if2_adc3_swap_mux = + SOC_DAPM_ENUM("IF2 ADC3 Swap Source", rt5677_if2_adc3_swap_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5677_if2_adc4_swap_enum, RT5677_TDM2_CTRL1, + RT5677_IF2_ADC4_SWAP_SFT, rt5677_if12_adc_swap_src); + +static const struct snd_kcontrol_new rt5677_if2_adc4_swap_mux = + SOC_DAPM_ENUM("IF2 ADC4 Swap Source", rt5677_if2_adc4_swap_enum); + +/* TDM IF1/2 ADC Data Selection */ /* MX-3C MX-41 [2:0] */ +static const char * const rt5677_if12_adc_tdm_swap_src[] = { + "1/2/3/4", "2/1/3/4", "2/3/1/4", "4/1/2/3", "1/3/2/4", "1/4/2/3", + "3/1/2/4", "3/4/1/2" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5677_if1_adc_tdm_swap_enum, RT5677_TDM1_CTRL2, + RT5677_IF1_ADC_CTRL_SFT, rt5677_if12_adc_tdm_swap_src); + +static const struct snd_kcontrol_new rt5677_if1_adc_tdm_swap_mux = + SOC_DAPM_ENUM("IF1 ADC TDM Swap Source", rt5677_if1_adc_tdm_swap_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5677_if2_adc_tdm_swap_enum, RT5677_TDM2_CTRL2, + RT5677_IF2_ADC_CTRL_SFT, rt5677_if12_adc_tdm_swap_src); + +static const struct snd_kcontrol_new rt5677_if2_adc_tdm_swap_mux = + SOC_DAPM_ENUM("IF2 ADC TDM Swap Source", rt5677_if2_adc_tdm_swap_enum); + static int rt5677_bst1_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -1945,6 +2026,60 @@ static int rt5677_set_micbias1_event(struct snd_soc_dapm_widget *w, return 0; } +static int rt5677_if1_adc_tdm_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + unsigned int value; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + regmap_read(rt5677->regmap, RT5677_TDM1_CTRL2, &value); + if (value & RT5677_IF1_ADC_CTRL_MASK) + regmap_update_bits(rt5677->regmap, RT5677_TDM1_CTRL1, + RT5677_IF1_ADC_MODE_MASK, + RT5677_IF1_ADC_MODE_TDM); + else + regmap_update_bits(rt5677->regmap, RT5677_TDM1_CTRL1, + RT5677_IF1_ADC_MODE_MASK, + RT5677_IF1_ADC_MODE_I2S); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5677_if2_adc_tdm_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + unsigned int value; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + regmap_read(rt5677->regmap, RT5677_TDM2_CTRL2, &value); + if (value & RT5677_IF2_ADC_CTRL_MASK) + regmap_update_bits(rt5677->regmap, RT5677_TDM2_CTRL1, + RT5677_IF2_ADC_MODE_MASK, + RT5677_IF2_ADC_MODE_TDM); + else + regmap_update_bits(rt5677->regmap, RT5677_TDM2_CTRL1, + RT5677_IF2_ADC_MODE_MASK, + RT5677_IF2_ADC_MODE_I2S); + break; + + default: + return 0; + } + + return 0; +} + static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("PLL1", RT5677_PWR_ANLG2, RT5677_PWR_PLL1_BIT, 0, rt5677_set_pll1_event, SND_SOC_DAPM_POST_PMU), @@ -2104,10 +2239,8 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { SND_SOC_DAPM_PGA("Stereo4 ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("Sto2 ADC LR MIX", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("Mono ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_PGA("IF1_ADC1", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_PGA("IF1_ADC2", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_PGA("IF1_ADC3", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_PGA("IF1_ADC4", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 ADC", SND_SOC_NOPM, 0, 0, NULL, 0), /* DSP */ SND_SOC_DAPM_MUX("IB9 Mux", SND_SOC_NOPM, 0, 0, @@ -2230,6 +2363,17 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { &rt5677_if1_adc3_mux), SND_SOC_DAPM_MUX("IF1 ADC4 Mux", SND_SOC_NOPM, 0, 0, &rt5677_if1_adc4_mux), + SND_SOC_DAPM_MUX("IF1 ADC1 Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if1_adc1_swap_mux), + SND_SOC_DAPM_MUX("IF1 ADC2 Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if1_adc2_swap_mux), + SND_SOC_DAPM_MUX("IF1 ADC3 Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if1_adc3_swap_mux), + SND_SOC_DAPM_MUX("IF1 ADC4 Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if1_adc4_swap_mux), + SND_SOC_DAPM_MUX_E("IF1 ADC TDM Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if1_adc_tdm_swap_mux, rt5677_if1_adc_tdm_event, + SND_SOC_DAPM_PRE_PMU), SND_SOC_DAPM_MUX("IF2 ADC1 Mux", SND_SOC_NOPM, 0, 0, &rt5677_if2_adc1_mux), SND_SOC_DAPM_MUX("IF2 ADC2 Mux", SND_SOC_NOPM, 0, 0, @@ -2238,6 +2382,17 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { &rt5677_if2_adc3_mux), SND_SOC_DAPM_MUX("IF2 ADC4 Mux", SND_SOC_NOPM, 0, 0, &rt5677_if2_adc4_mux), + SND_SOC_DAPM_MUX("IF2 ADC1 Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if2_adc1_swap_mux), + SND_SOC_DAPM_MUX("IF2 ADC2 Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if2_adc2_swap_mux), + SND_SOC_DAPM_MUX("IF2 ADC3 Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if2_adc3_swap_mux), + SND_SOC_DAPM_MUX("IF2 ADC4 Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if2_adc4_swap_mux), + SND_SOC_DAPM_MUX_E("IF2 ADC TDM Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if2_adc_tdm_swap_mux, rt5677_if2_adc_tdm_event, + SND_SOC_DAPM_PRE_PMU), SND_SOC_DAPM_MUX("IF3 ADC Mux", SND_SOC_NOPM, 0, 0, &rt5677_if3_adc_mux), SND_SOC_DAPM_MUX("IF4 ADC Mux", SND_SOC_NOPM, 0, 0, @@ -2621,11 +2776,42 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "IF1 ADC4 Mux", "OB67", "OB67" }, { "IF1 ADC4 Mux", "OB01", "OB01 Bypass Mux" }, + { "IF1 ADC1 Swap Mux", "L/R", "IF1 ADC1 Mux" }, + { "IF1 ADC1 Swap Mux", "R/L", "IF1 ADC1 Mux" }, + { "IF1 ADC1 Swap Mux", "L/L", "IF1 ADC1 Mux" }, + { "IF1 ADC1 Swap Mux", "R/R", "IF1 ADC1 Mux" }, + + { "IF1 ADC2 Swap Mux", "L/R", "IF1 ADC2 Mux" }, + { "IF1 ADC2 Swap Mux", "R/L", "IF1 ADC2 Mux" }, + { "IF1 ADC2 Swap Mux", "L/L", "IF1 ADC2 Mux" }, + { "IF1 ADC2 Swap Mux", "R/R", "IF1 ADC2 Mux" }, + + { "IF1 ADC3 Swap Mux", "L/R", "IF1 ADC3 Mux" }, + { "IF1 ADC3 Swap Mux", "R/L", "IF1 ADC3 Mux" }, + { "IF1 ADC3 Swap Mux", "L/L", "IF1 ADC3 Mux" }, + { "IF1 ADC3 Swap Mux", "R/R", "IF1 ADC3 Mux" }, + + { "IF1 ADC4 Swap Mux", "L/R", "IF1 ADC4 Mux" }, + { "IF1 ADC4 Swap Mux", "R/L", "IF1 ADC4 Mux" }, + { "IF1 ADC4 Swap Mux", "L/L", "IF1 ADC4 Mux" }, + { "IF1 ADC4 Swap Mux", "R/R", "IF1 ADC4 Mux" }, + + { "IF1 ADC", NULL, "IF1 ADC1 Swap Mux" }, + { "IF1 ADC", NULL, "IF1 ADC2 Swap Mux" }, + { "IF1 ADC", NULL, "IF1 ADC3 Swap Mux" }, + { "IF1 ADC", NULL, "IF1 ADC4 Swap Mux" }, + + { "IF1 ADC TDM Swap Mux", "1/2/3/4", "IF1 ADC" }, + { "IF1 ADC TDM Swap Mux", "2/1/3/4", "IF1 ADC" }, + { "IF1 ADC TDM Swap Mux", "2/3/1/4", "IF1 ADC" }, + { "IF1 ADC TDM Swap Mux", "4/1/2/3", "IF1 ADC" }, + { "IF1 ADC TDM Swap Mux", "1/3/2/4", "IF1 ADC" }, + { "IF1 ADC TDM Swap Mux", "1/4/2/3", "IF1 ADC" }, + { "IF1 ADC TDM Swap Mux", "3/1/2/4", "IF1 ADC" }, + { "IF1 ADC TDM Swap Mux", "3/4/1/2", "IF1 ADC" }, + { "AIF1TX", NULL, "I2S1" }, - { "AIF1TX", NULL, "IF1 ADC1 Mux" }, - { "AIF1TX", NULL, "IF1 ADC2 Mux" }, - { "AIF1TX", NULL, "IF1 ADC3 Mux" }, - { "AIF1TX", NULL, "IF1 ADC4 Mux" }, + { "AIF1TX", NULL, "IF1 ADC TDM Swap Mux" }, { "IF2 ADC1 Mux", "STO1 ADC MIX", "Stereo1 ADC MIX" }, { "IF2 ADC1 Mux", "OB01", "OB01 Bypass Mux" }, @@ -2642,11 +2828,42 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "IF2 ADC4 Mux", "OB67", "OB67" }, { "IF2 ADC4 Mux", "OB01", "OB01 Bypass Mux" }, + { "IF2 ADC1 Swap Mux", "L/R", "IF2 ADC1 Mux" }, + { "IF2 ADC1 Swap Mux", "R/L", "IF2 ADC1 Mux" }, + { "IF2 ADC1 Swap Mux", "L/L", "IF2 ADC1 Mux" }, + { "IF2 ADC1 Swap Mux", "R/R", "IF2 ADC1 Mux" }, + + { "IF2 ADC2 Swap Mux", "L/R", "IF2 ADC2 Mux" }, + { "IF2 ADC2 Swap Mux", "R/L", "IF2 ADC2 Mux" }, + { "IF2 ADC2 Swap Mux", "L/L", "IF2 ADC2 Mux" }, + { "IF2 ADC2 Swap Mux", "R/R", "IF2 ADC2 Mux" }, + + { "IF2 ADC3 Swap Mux", "L/R", "IF2 ADC3 Mux" }, + { "IF2 ADC3 Swap Mux", "R/L", "IF2 ADC3 Mux" }, + { "IF2 ADC3 Swap Mux", "L/L", "IF2 ADC3 Mux" }, + { "IF2 ADC3 Swap Mux", "R/R", "IF2 ADC3 Mux" }, + + { "IF2 ADC4 Swap Mux", "L/R", "IF2 ADC4 Mux" }, + { "IF2 ADC4 Swap Mux", "R/L", "IF2 ADC4 Mux" }, + { "IF2 ADC4 Swap Mux", "L/L", "IF2 ADC4 Mux" }, + { "IF2 ADC4 Swap Mux", "R/R", "IF2 ADC4 Mux" }, + + { "IF2 ADC", NULL, "IF2 ADC1 Swap Mux" }, + { "IF2 ADC", NULL, "IF2 ADC2 Swap Mux" }, + { "IF2 ADC", NULL, "IF2 ADC3 Swap Mux" }, + { "IF2 ADC", NULL, "IF2 ADC4 Swap Mux" }, + + { "IF2 ADC TDM Swap Mux", "1/2/3/4", "IF2 ADC" }, + { "IF2 ADC TDM Swap Mux", "2/1/3/4", "IF2 ADC" }, + { "IF2 ADC TDM Swap Mux", "2/3/1/4", "IF2 ADC" }, + { "IF2 ADC TDM Swap Mux", "4/1/2/3", "IF2 ADC" }, + { "IF2 ADC TDM Swap Mux", "1/3/2/4", "IF2 ADC" }, + { "IF2 ADC TDM Swap Mux", "1/4/2/3", "IF2 ADC" }, + { "IF2 ADC TDM Swap Mux", "3/1/2/4", "IF2 ADC" }, + { "IF2 ADC TDM Swap Mux", "3/4/1/2", "IF2 ADC" }, + { "AIF2TX", NULL, "I2S2" }, - { "AIF2TX", NULL, "IF2 ADC1 Mux" }, - { "AIF2TX", NULL, "IF2 ADC2 Mux" }, - { "AIF2TX", NULL, "IF2 ADC3 Mux" }, - { "AIF2TX", NULL, "IF2 ADC4 Mux" }, + { "AIF2TX", NULL, "IF2 ADC TDM Swap Mux" }, { "IF3 ADC Mux", "STO1 ADC MIX", "Stereo1 ADC MIX" }, { "IF3 ADC Mux", "STO2 ADC MIX", "Stereo2 ADC MIX" }, diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index 20efa4a..a0ae32f 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -799,7 +799,21 @@ #define RT5677_PDM2_I2C_EXE (0x1 << 1) #define RT5677_PDM2_I2C_BUSY (0x1 << 0) -/* MX3C TDM1 control 1 (0x3c) */ +/* MX3B TDM1 control 1 (0x3b) */ +#define RT5677_IF1_ADC_MODE_MASK (0x1 << 12) +#define RT5677_IF1_ADC_MODE_SFT 12 +#define RT5677_IF1_ADC_MODE_I2S (0x0 << 12) +#define RT5677_IF1_ADC_MODE_TDM (0x1 << 12) +#define RT5677_IF1_ADC1_SWAP_MASK (0x3 << 6) +#define RT5677_IF1_ADC1_SWAP_SFT 6 +#define RT5677_IF1_ADC2_SWAP_MASK (0x3 << 4) +#define RT5677_IF1_ADC2_SWAP_SFT 4 +#define RT5677_IF1_ADC3_SWAP_MASK (0x3 << 2) +#define RT5677_IF1_ADC3_SWAP_SFT 2 +#define RT5677_IF1_ADC4_SWAP_MASK (0x3 << 0) +#define RT5677_IF1_ADC4_SWAP_SFT 0 + +/* MX3C TDM1 control 2 (0x3c) */ #define RT5677_IF1_ADC4_MASK (0x3 << 10) #define RT5677_IF1_ADC4_SFT 10 #define RT5677_IF1_ADC3_MASK (0x3 << 8) @@ -808,8 +822,24 @@ #define RT5677_IF1_ADC2_SFT 6 #define RT5677_IF1_ADC1_MASK (0x3 << 4) #define RT5677_IF1_ADC1_SFT 4 - -/* MX41 TDM2 control 1 (0x41) */ +#define RT5677_IF1_ADC_CTRL_MASK (0x7 << 0) +#define RT5677_IF1_ADC_CTRL_SFT 0 + +/* MX40 TDM2 control 1 (0x40) */ +#define RT5677_IF2_ADC_MODE_MASK (0x1 << 12) +#define RT5677_IF2_ADC_MODE_SFT 12 +#define RT5677_IF2_ADC_MODE_I2S (0x0 << 12) +#define RT5677_IF2_ADC_MODE_TDM (0x1 << 12) +#define RT5677_IF2_ADC1_SWAP_MASK (0x3 << 6) +#define RT5677_IF2_ADC1_SWAP_SFT 6 +#define RT5677_IF2_ADC2_SWAP_MASK (0x3 << 4) +#define RT5677_IF2_ADC2_SWAP_SFT 4 +#define RT5677_IF2_ADC3_SWAP_MASK (0x3 << 2) +#define RT5677_IF2_ADC3_SWAP_SFT 2 +#define RT5677_IF2_ADC4_SWAP_MASK (0x3 << 0) +#define RT5677_IF2_ADC4_SWAP_SFT 0 + +/* MX41 TDM2 control 2 (0x41) */ #define RT5677_IF2_ADC4_MASK (0x3 << 10) #define RT5677_IF2_ADC4_SFT 10 #define RT5677_IF2_ADC3_MASK (0x3 << 8) @@ -818,6 +848,8 @@ #define RT5677_IF2_ADC2_SFT 6 #define RT5677_IF2_ADC1_MASK (0x3 << 4) #define RT5677_IF2_ADC1_SFT 4 +#define RT5677_IF2_ADC_CTRL_MASK (0x7 << 0) +#define RT5677_IF2_ADC_CTRL_SFT 0 /* Digital Microphone Control 1 (0x50) */ #define RT5677_DMIC_1_EN_MASK (0x1 << 15) -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH] ASoC: arm64: use the same dma mmap codepath both for arm and arm64
by Anatol Pomozov 18 Oct '14

18 Oct '14

This avoids following kernel crash when try to playback on arm64 [ 107.497203] [<ffffffc00046b310>] snd_pcm_mmap_data_fault+0x90/0xd4 [ 107.503405] [<ffffffc0001541ac>] __do_fault+0xb0/0x498 [ 107.508565] [<ffffffc0001576a0>] handle_mm_fault+0x224/0x7b0 [ 107.514246] [<ffffffc000092640>] do_page_fault+0x11c/0x310 [ 107.519738] [<ffffffc000081100>] do_mem_abort+0x38/0x98 Tested: backported to 3.14 and tried to playback on arm64 machine Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- sound/core/pcm_native.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 8cd2f93..a95356f 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3193,7 +3193,7 @@ static const struct vm_operations_struct snd_pcm_vm_ops_data_fault = { #ifndef ARCH_HAS_DMA_MMAP_COHERENT /* This should be defined / handled globally! */ -#ifdef CONFIG_ARM +#if defined(CONFIG_ARM) || defined(CONFIG_ARM64) #define ARCH_HAS_DMA_MMAP_COHERENT #endif #endif -- 2.1.0.rc2.206.gedb03e5

2 1

[alsa-devel] [PATCH] ASoC: Intel: Add jack detection for Broadwell
by Jie Yang 17 Oct '14

17 Oct '14

Add jack dectection and event reporting for Broadwell. It use combo jack on BDW platform, which including Mic Jack pin and Headphone jack pin. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/broadwell.c | 49 ++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 46 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/broadwell.c b/sound/soc/intel/broadwell.c index 0e550f1..52cb764 100644 --- a/sound/soc/intel/broadwell.c +++ b/sound/soc/intel/broadwell.c @@ -19,6 +19,7 @@ #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> +#include <sound/jack.h> #include <sound/pcm_params.h> #include "sst-dsp.h" @@ -26,8 +27,26 @@ #include "../codecs/rt286.h" +static struct snd_soc_jack broadwell_headset; +/* Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin broadwell_headset_pins[] = { + { + .pin = "Mic Jack", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, +}; + +static const struct snd_kcontrol_new broadwell_controls[] = { + SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Headphone Jack"), +}; + static const struct snd_soc_dapm_widget broadwell_widgets[] = { - SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_SPK("Speaker", NULL), SND_SOC_DAPM_MIC("Mic Jack", NULL), SND_SOC_DAPM_MIC("DMIC1", NULL), @@ -42,7 +61,7 @@ static const struct snd_soc_dapm_route broadwell_rt286_map[] = { {"Speaker", NULL, "SPOL"}, /* HP jack connectors - unknown if we have jack deteck */ - {"Headphones", NULL, "HPO Pin"}, + {"Headphone Jack", NULL, "HPO Pin"}, /* other jacks */ {"MIC1", NULL, "Mic Jack"}, @@ -57,6 +76,27 @@ static const struct snd_soc_dapm_route broadwell_rt286_map[] = { {"AIF1 Playback", NULL, "SSP0 CODEC OUT"}, }; +static int broadwell_rt286_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int ret = 0; + ret = snd_soc_jack_new(codec, "Headset", + SND_JACK_HEADSET | SND_JACK_BTN_0, &broadwell_headset); + + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(&broadwell_headset, + ARRAY_SIZE(broadwell_headset_pins), + broadwell_headset_pins); + if (ret) + return ret; + + rt286_mic_detect(codec, &broadwell_headset); + return 0; +} + + static int broadwell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { @@ -116,7 +156,7 @@ static int broadwell_rtd_init(struct snd_soc_pcm_runtime *rtd) } /* always connected - check HP for jack detect */ - snd_soc_dapm_enable_pin(dapm, "Headphones"); + snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); snd_soc_dapm_enable_pin(dapm, "Speaker"); snd_soc_dapm_enable_pin(dapm, "Mic Jack"); snd_soc_dapm_enable_pin(dapm, "Line Jack"); @@ -196,6 +236,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { .no_pcm = 1, .codec_name = "i2c-INT343A:00", .codec_dai_name = "rt286-aif1", + .init = broadwell_rt286_codec_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, .ignore_suspend = 1, @@ -213,6 +254,8 @@ static struct snd_soc_card broadwell_rt286 = { .owner = THIS_MODULE, .dai_link = broadwell_rt286_dais, .num_links = ARRAY_SIZE(broadwell_rt286_dais), + .controls = broadwell_controls, + .num_controls = ARRAY_SIZE(broadwell_controls), .dapm_widgets = broadwell_widgets, .num_dapm_widgets = ARRAY_SIZE(broadwell_widgets), .dapm_routes = broadwell_rt286_map, -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rt5677: fix warnings when kernel compiled without GPIOLIB support
by Anatol Pomozov 17 Oct '14

17 Oct '14

Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- sound/soc/codecs/rt5677.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 3b63772..f6df276 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -3649,6 +3649,11 @@ static void rt5677_free_gpio(struct i2c_client *i2c) dev_err(&i2c->dev, "Failed to remove GPIOs: %d\n", ret); } #else +static void rt5677_gpio_config(struct rt5677_priv *rt5677, unsigned offset, + int value) +{ +} + static void rt5677_init_gpio(struct i2c_client *i2c) { } -- 2.1.0.rc2.206.gedb03e5

2 2

[alsa-devel] [PATCH 0/2] ASoC: Intel: Add missing build dependencies
by Ricardo Neri 17 Oct '14

17 Oct '14

The Broadwell and Haswell machines utilize the RT286 and RT5640 codecs, respectively. To communicate with the codecs, the machine utilizes the Designware I2C platform adapter. Hence, such adapter should be present in order to probe the codec drivers correctly. This short patch series fixes the missing dependency. Thanks and BR, Ricardo Ricardo Neri (2): ASoC: Intel: Add new dependency for Broadwell machine ASoC: Intel: Add new dependency for Haswell machine sound/soc/intel/Kconfig | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: atmel_ssc_dai: Fix spelling in pr_debug.
by Peter Rosin 17 Oct '14

17 Oct '14

>From 64e0a6c2fe731eb24c81c7c56a3cacdda3d51610 Mon Sep 17 00:00:00 2001 From: Peter Rosin <peda(a)axentia.se> Date: Thu, 16 Oct 2014 17:22:05 +0200 Subject: [PATCH] ASoC: atmel_ssc_dai: Fix spelling in pr_debug. Signed-off-by: Peter Rosin <peda(a)axentia.se> --- sound/soc/atmel/atmel_ssc_dai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index f403f39..775a2ba 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -267,7 +267,7 @@ static void atmel_ssc_shutdown(struct snd_pcm_substream *substream, if (!ssc_p->dir_mask) { if (ssc_p->initialized) { /* Shutdown the SSC clock. */ - pr_debug("atmel_ssc_dau: Stopping clock\n"); + pr_debug("atmel_ssc_dai: Stopping clock\n"); clk_disable(ssc_p->ssc->clk); free_irq(ssc_p->ssc->irq, ssc_p); -- 1.7.10.4

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[alsa-devel] Some thoughts on ALSA support for VCO audio clocks
by Tim Cussins 17 Oct '14

17 Oct '14

Hi all, I thought I'd brain-dump my thoughts on VCO support. It's not as coherent as I'd like it to be, but it's important I get it out there as the mini summit is only a few days away. Linn use-case ------------- We make network media players, and support synchronised playback where a master device sends audio to several slave devices for simultaneous rendering. On a given network there may be several masters, each coordinating several slaves. These masters might be recovering their clocks from one of many different types of external source: SPDIF or internet radio, for example. By keeping the audio clocks synchronised using an appropriate network protocol, we can achieve simultaneous playback across many devices, which enable simultaneous start, and no drift. Network devices can be dynamically configured as master/slave, and therefore must dynamically subscribe/unsubscribe to/from any master as appropriate. Incidentally, we use our own UPnP services to make this happen. Our hardware ------------ Our SoC of choice has an event timer, a notion of an event clock. This allows signals from peripherals to be timestamped in hardware, and for signals from the event module to be raised at peripherals using compare registers. Think of it as a capture/compare module with multiple capture channels and compare registers for almost all peripherals. To get synchronised start, We intend to drive the event timer using a VCO, and trigger our I2S module (with preloaded output fifo) using the appropriate compare register in the event module. Handwaving ---------- There are two use cases that are of interest here: VCO and PTP. The pure VCO model implies that the PCM driver *owns* the VCO. The PTP model implies that the PCM is a slave to a PTP clock, governed elsewhere. As our SoC event clock is system-wide, other drivers/processes may be timestamping using this clock. Therefore this proposal seeks to address the PTP model (PCM doesn't own VCO). The Linux kernel provides a kernel framework for supporting VCO-based clocks, the PCH framework, which exposes such clocks to userspace with a common API. This API isn't PTP-specific however, and would allow for other synchronisation mechanisms. The kernel refers to these as 'ptp' clocks, so I'll use PTP from here on in: This doesn't imply IEEE-1588 however. Takashi added support for the selection of timestamp type at runtime. These types currently correspond to posix clock types, and don't allow for other time-sources such as PTP clocks. I imagine that PCM modules should be able to advertise their supported timestamp types: In our case, our driver would additionally advertise some unique PTP clock. It would be ideal if we could uniquely identify a PTP clock using a posix clockid_t, the value of which could be meaningfully shared amongst userspace processes, for example over D-Bus. However it's not immediately clear that you can establish such a value. So for now let's design as though we can't. Unique clocks from userspace might be represented as some tuple of a _clock protocol_ (e.g. POSIX, PTP), and a _clock id_ that represents a specific clock (e.g. CLOCK_MONOTONIC, CLOCK_REALTIME, "/dev/ptp0"). I'll admit that a path to a device node is not ideal, but you get the idea: It's a conversation starter. Maybe major/minor number would be better. Either way, kernel-side ALSA must be able to timestamp using the information. Takashi's work presumes that the clock protocol is posix: I would advocate for allowing other protocols. No surprise there :) Timestamp Clock Enumeration --------------------------- PCM devices could be probed for supported timestamps. Some enumeration mechanism, akin to snd_pcm_query_chmaps_from_hw(), could return an array of snd_pcm_clock_descriptor_t*, as in: typedef struct _snd_pcm_clock_descriptor { /* enum indicating type of chewy center */ snd_pcm_clock_protocol_t protocol; union { /* Essentially a clockid_t */ struct snd_pcm_clock_posix *posix_data; /* Perhaps "/dev/ptp0", or a major/minor number. */ struct snd_pcm_clock_ptp *ptp_data; } id; } snd_pcm_clock_descriptor_t; A revised version of Takashi's snd_pcm_sw_params_set_tstamp_type() would take a snd_pcm_clock_descriptor_t*. This allows for existing posix clock types, PTP clock types, and allows further clock types without an API (ABI?) change. Summary ------- This document focuses on using a VCO as the output clock for a PCM device, and hints at userspace API amendments that would support such a configuration. It mentions: - Allowing the timestamp type to support other clock protocols - PCM devices can be probed for a list of supported clock protocol/id tuples. - These tuples are sufficient to uniquely identify clocks from userspace. - Proposal scales to multiple PTP clocks, with no implied relationship to the system clock. It doesn't attempt to tackle configurations where the VCO is owned by the PCM hardware: In this case, I would advocate for using the PTP interface anyway, rather than adding an ALSA API for controlling the VCO. Any comments and feedback would be very welcome. Cheers, Tim

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[alsa-devel] [TINYCOMPRESS][PATCH] crec: Fix error with opening file using O_CREAT
by Richard Fitzgerald 17 Oct '14

17 Oct '14

From: Ammar Zahid Ali Syed <ammar.zhd(a)googlemail.com> The code was not compiling when mode was not passed to open() while using O_CREAT offset. Passed mode to the open() alongwith O_CREAT to specify the attributes of file and fix compile issue. Signed-off-by: Ammar Zahid Ali Syed <ammar.zhd(a)googlemail.com> --- crec.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/crec.c b/crec.c index 31703c5..e3f189f 100644 --- a/crec.c +++ b/crec.c @@ -66,6 +66,8 @@ #include <stdbool.h> #include <getopt.h> #include <sys/time.h> +#include <sys/types.h> +#include <sys/stat.h> #define __force #define __bitwise #define __user @@ -256,7 +258,7 @@ void capture_samples(char *name, unsigned int card, unsigned int device, if (verbose) printf("%s: entry, reading %u bytes\n", __func__, length); - file = open(name, O_RDWR | O_CREAT); + file = open(name, O_RDWR | O_CREAT, S_IRUSR | S_IWUSR | S_IRGRP | S_IWGRP); if (file == -1) { fprintf(stderr, "Unable to open file '%s'\n", name); exit(EXIT_FAILURE); -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ASoC: Intel: mrfld: fix semicolon.cocci warnings
by Fengguang Wu 17 Oct '14

17 Oct '14

sound/soc/intel/sst-atom-controls.c:249:2-3: Unneeded semicolon sound/soc/intel/sst-atom-controls.c:289:2-3: Unneeded semicolon Removes unneeded semicolon. Generated by: scripts/coccinelle/misc/semicolon.cocci CC: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Fengguang Wu <fengguang.wu(a)intel.com> --- sst-atom-controls.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) --- a/sound/soc/intel/sst-atom-controls.c +++ b/sound/soc/intel/sst-atom-controls.c @@ -246,7 +246,7 @@ static int sst_gain_get(struct snd_kcont dev_err(component->dev, "Invalid Input- gain type:%d\n", mc->type); return -EINVAL; - }; + } return 0; } @@ -286,7 +286,7 @@ static int sst_gain_put(struct snd_kcont dev_err(cmpnt->dev, "Invalid Input- gain type:%d\n", mc->type); return -EINVAL; - }; + } if (mc->w && mc->w->power) ret = sst_send_gain_cmd(drv, gv, mc->task_id,

3 2

[alsa-devel] [PATCH v2 0/3] ALSA: seq: bind seq driver automatically
by Takashi Iwai 17 Oct '14

17 Oct '14

Hi, this is a patchset split from the previous patch for fixing the autoloading issue of sequencer stuff. The end result is the exactly same changes. [PATCH 1/3] ALSA: seq: Use atomic ops for autoload refcount [PATCH 2/3] ALSA: seq: bind seq driver automatically [PATCH 3/3] Subject: ALSA: seq: Remove autoload locks in driver Takashi

1 3

[alsa-devel] [PATCH v2 1/2] ALSA: au88x0: added reference of vortex_t
by Sudip Mukherjee 17 Oct '14

17 Oct '14

added a pointer of the vortex in the following functions : vortex_alsafmt_aspfmt vortex_Vort3D_InitializeSource a3dsrc_ZeroStateA3D so that we can have a reference of the vortex in the function. this reference of the vortex will actually be used in a later patch to convert the pr_* macro to dev_*. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- no change from v1. sound/pci/au88x0/au88x0.h | 4 ++-- sound/pci/au88x0/au88x0_a3d.c | 6 +++--- sound/pci/au88x0/au88x0_core.c | 5 +++-- sound/pci/au88x0/au88x0_pcm.c | 2 +- 4 files changed, 9 insertions(+), 8 deletions(-) diff --git a/sound/pci/au88x0/au88x0.h b/sound/pci/au88x0/au88x0.h index 466a5c8..3a8fefe 100644 --- a/sound/pci/au88x0/au88x0.h +++ b/sound/pci/au88x0/au88x0.h @@ -243,7 +243,7 @@ static int vortex_core_init(vortex_t * card); static int vortex_core_shutdown(vortex_t * card); static void vortex_enable_int(vortex_t * card); static irqreturn_t vortex_interrupt(int irq, void *dev_id); -static int vortex_alsafmt_aspfmt(int alsafmt); +static int vortex_alsafmt_aspfmt(int alsafmt, vortex_t *v); /* Connection stuff. */ static void vortex_connect_default(vortex_t * vortex, int en); @@ -278,7 +278,7 @@ static void vortex_mix_setvolumebyte(vortex_t * vortex, unsigned char mix, static void vortex_Vort3D_enable(vortex_t * v); static void vortex_Vort3D_disable(vortex_t * v); static void vortex_Vort3D_connect(vortex_t * vortex, int en); -static void vortex_Vort3D_InitializeSource(a3dsrc_t * a, int en); +static void vortex_Vort3D_InitializeSource(a3dsrc_t *a, int en, vortex_t *v); #endif /* Driver stuff. */ diff --git a/sound/pci/au88x0/au88x0_a3d.c b/sound/pci/au88x0/au88x0_a3d.c index 30f760e..bc9cda3a 100644 --- a/sound/pci/au88x0/au88x0_a3d.c +++ b/sound/pci/au88x0/au88x0_a3d.c @@ -484,7 +484,7 @@ static void a3dsrc_ZeroState(a3dsrc_t * a) } /* Reset entire A3D engine */ -static void a3dsrc_ZeroStateA3D(a3dsrc_t * a) +static void a3dsrc_ZeroStateA3D(a3dsrc_t *a, vortex_t *v) { int i, var, var2; @@ -601,7 +601,7 @@ static void vortex_Vort3D_enable(vortex_t *v) Vort3DRend_Initialize(v, XT_HEADPHONE); for (i = 0; i < NR_A3D; i++) { vortex_A3dSourceHw_Initialize(v, i % 4, i >> 2); - a3dsrc_ZeroStateA3D(&(v->a3d[0])); + a3dsrc_ZeroStateA3D(&v->a3d[0], v); } /* Register ALSA controls */ vortex_a3d_register_controls(v); @@ -676,7 +676,7 @@ static void vortex_Vort3D_connect(vortex_t * v, int en) } /* Initialize one single A3D source. */ -static void vortex_Vort3D_InitializeSource(a3dsrc_t * a, int en) +static void vortex_Vort3D_InitializeSource(a3dsrc_t *a, int en, vortex_t *v) { if (a->vortex == NULL) { pr_warn diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c index 72e8128..00e2096 100644 --- a/sound/pci/au88x0/au88x0_core.c +++ b/sound/pci/au88x0/au88x0_core.c @@ -2177,7 +2177,8 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir, return -EBUSY; } /* (De)Initialize A3D hardware source. */ - vortex_Vort3D_InitializeSource(&(vortex->a3d[a3d]), en); + vortex_Vort3D_InitializeSource(&vortex->a3d[a3d], en, + vortex); } /* Make SPDIF out exclusive to "spdif" device when in use. */ if ((stream->type == VORTEX_PCM_SPDIF) && (en)) { @@ -2765,7 +2766,7 @@ static int vortex_core_shutdown(vortex_t * vortex) /* Alsa support. */ -static int vortex_alsafmt_aspfmt(int alsafmt) +static int vortex_alsafmt_aspfmt(int alsafmt, vortex_t *v) { int fmt; diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c index 5adc6b9..bdde182 100644 --- a/sound/pci/au88x0/au88x0_pcm.c +++ b/sound/pci/au88x0/au88x0_pcm.c @@ -332,7 +332,7 @@ static int snd_vortex_pcm_prepare(struct snd_pcm_substream *substream) dir = 1; else dir = 0; - fmt = vortex_alsafmt_aspfmt(runtime->format); + fmt = vortex_alsafmt_aspfmt(runtime->format, chip); spin_lock_irq(&chip->lock); if (VORTEX_PCM_TYPE(substream->pcm) != VORTEX_PCM_WT) { vortex_adbdma_setmode(chip, dma, 1, dir, fmt, -- 1.8.1.2

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[alsa-devel] [PATCH v2 0/2] Disable the ThinkPad HW mute/level control when possible
by Andy Lutomirski 16 Oct '14

16 Oct '14

This series is being resurrected after several years, and it's quite different from before, so I'm restarting the version number at 1 :) ThinkPad volume and mute controls are a mess. For whatever reason, ThinkPads have mute buttons that send KEY_MUTE *and* control an invisible-to-ALSA mute switch. Some of them have volume controls that interact with this switch as well. This is a perennial source of problems. On most ThinkPads, if you press mute and then unmute using GUI controls, you have no sound, because userspace and ALSA state gets out of sync with the hardware switch. There's a separate "sound card" that exposes the hardware switch, but userspace code generally doesn't understand that. There are already a few _OSI(Linux) overrides to turn all the hardware buttons into regular buttons. Rather than quirking ACPI everywhere, just teach thinkpad-acpi to program the buttons for full software control and to disable hardware controls. That allows us to remove the ACPI quirks and have normal mute controls. This approach should be much simpler than adding even more kludgey ALSA integration for questionable gain. I haven't tested on my X220 (which has a mute light in addition to a mute button), since that laptop's SATA controller died last week. It should work, though, and if someone else could test this on a ThinkPad with a mute light, that would be great. The only issue I know of with this version is that unloading and reloading the module to change the software_mute setting doesn't work quite right. I can fix that if it matters. Changes from v1: - Simplified the code a bit. - Improved suspend/hibernate behavior. Andy Lutomirski (2): thinkpad-acpi: Try to use full software mute control acpi: Remove _OSI(Linux) for ThinkPads drivers/acpi/blacklist.c | 54 -------------------- drivers/platform/x86/thinkpad_acpi.c | 98 ++++++++++++++++++++++++++++++++---- 2 files changed, 88 insertions(+), 64 deletions(-) -- 1.9.3

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[alsa-devel] Cards that support ADPCM in hardware
by Alexander E. Patrakov 16 Oct '14

16 Oct '14

Hello. Due to the "remove support for old hardware" topic that is going to be discussed at the Linux Plumbers Conference, let me see if there is actually any supported hardware that supports ADPCM. A quick grep for ADPCM in the sound directory yields this: ./isa/wss/wss_lib.c: ISA, Windows sound system ./isa/sb/sb16_{main,csp}.c: ISA, classic SB16 card ./oss/ad1848.c: Another ISA card, OSS-only ./sh/aica.c: for SEGA Dreamcast console ./sparc/cs4231.c: I don't know about this Is any of the hardware listed above still used (or usable) with the current kernels? -- Alexander E. Patrakov

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[alsa-devel] [PATCH] ASoC: rt5677: fix warnings when kernel compiled without GPIOLIB support
by Anatol Pomozov 16 Oct '14

16 Oct '14

Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- sound/soc/codecs/rt5677.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index d1fc9a1..2f0d19f 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -3686,6 +3686,11 @@ static void rt5677_free_gpio(struct i2c_client *i2c) gpiochip_remove(&rt5677->gpio_chip); } #else +static void rt5677_gpio_config(struct rt5677_priv *rt5677, unsigned offset, + int value) +{ +} + static void rt5677_init_gpio(struct i2c_client *i2c) { } -- 2.1.0.rc2.206.gedb03e5

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[alsa-devel] [PATCH v2 01/53] dmaengine: Make the destination abbreviation coherent
by Maxime Ripard 16 Oct '14

16 Oct '14

The dmaengine header abbreviates destination as at least two different strings. Make a coherent use of a single one. Signed-off-by: Maxime Ripard <maxime.ripard(a)free-electrons.com> --- drivers/dma/bcm2835-dma.c | 2 +- drivers/dma/edma.c | 2 +- drivers/dma/fsl-edma.c | 2 +- drivers/dma/nbpfaxi.c | 2 +- drivers/dma/omap-dma.c | 2 +- drivers/dma/pl330.c | 2 +- drivers/dma/sirf-dma.c | 2 +- include/linux/dmaengine.h | 12 ++++++------ sound/soc/soc-generic-dmaengine-pcm.c | 2 +- 9 files changed, 14 insertions(+), 14 deletions(-) diff --git a/drivers/dma/bcm2835-dma.c b/drivers/dma/bcm2835-dma.c index 68007974961a..8a6a9c27fe83 100644 --- a/drivers/dma/bcm2835-dma.c +++ b/drivers/dma/bcm2835-dma.c @@ -571,7 +571,7 @@ static int bcm2835_dma_device_slave_caps(struct dma_chan *dchan, struct dma_slave_caps *caps) { caps->src_addr_widths = BIT(DMA_SLAVE_BUSWIDTH_4_BYTES); - caps->dstn_addr_widths = BIT(DMA_SLAVE_BUSWIDTH_4_BYTES); + caps->dst_addr_widths = BIT(DMA_SLAVE_BUSWIDTH_4_BYTES); caps->directions = BIT(DMA_DEV_TO_MEM) | BIT(DMA_MEM_TO_DEV); caps->cmd_pause = false; caps->cmd_terminate = true; diff --git a/drivers/dma/edma.c b/drivers/dma/edma.c index 7b65633f495e..b3d641881d9d 100644 --- a/drivers/dma/edma.c +++ b/drivers/dma/edma.c @@ -998,7 +998,7 @@ static int edma_dma_device_slave_caps(struct dma_chan *dchan, struct dma_slave_caps *caps) { caps->src_addr_widths = EDMA_DMA_BUSWIDTHS; - caps->dstn_addr_widths = EDMA_DMA_BUSWIDTHS; + caps->dst_addr_widths = EDMA_DMA_BUSWIDTHS; caps->directions = BIT(DMA_DEV_TO_MEM) | BIT(DMA_MEM_TO_DEV); caps->cmd_pause = true; caps->cmd_terminate = true; diff --git a/drivers/dma/fsl-edma.c b/drivers/dma/fsl-edma.c index 3c5711d5fe97..13df85a16438 100644 --- a/drivers/dma/fsl-edma.c +++ b/drivers/dma/fsl-edma.c @@ -781,7 +781,7 @@ static int fsl_dma_device_slave_caps(struct dma_chan *dchan, struct dma_slave_caps *caps) { caps->src_addr_widths = FSL_EDMA_BUSWIDTHS; - caps->dstn_addr_widths = FSL_EDMA_BUSWIDTHS; + caps->dst_addr_widths = FSL_EDMA_BUSWIDTHS; caps->directions = BIT(DMA_DEV_TO_MEM) | BIT(DMA_MEM_TO_DEV); caps->cmd_pause = true; caps->cmd_terminate = true; diff --git a/drivers/dma/nbpfaxi.c b/drivers/dma/nbpfaxi.c index 5aeada56a442..b4a61ccbb443 100644 --- a/drivers/dma/nbpfaxi.c +++ b/drivers/dma/nbpfaxi.c @@ -1076,7 +1076,7 @@ static int nbpf_slave_caps(struct dma_chan *dchan, struct dma_slave_caps *caps) { caps->src_addr_widths = NBPF_DMA_BUSWIDTHS; - caps->dstn_addr_widths = NBPF_DMA_BUSWIDTHS; + caps->dst_addr_widths = NBPF_DMA_BUSWIDTHS; caps->directions = BIT(DMA_DEV_TO_MEM) | BIT(DMA_MEM_TO_DEV); caps->cmd_pause = false; caps->cmd_terminate = true; diff --git a/drivers/dma/omap-dma.c b/drivers/dma/omap-dma.c index 4cf7d9a950d7..51dee6636a37 100644 --- a/drivers/dma/omap-dma.c +++ b/drivers/dma/omap-dma.c @@ -1095,7 +1095,7 @@ static int omap_dma_device_slave_caps(struct dma_chan *dchan, struct dma_slave_caps *caps) { caps->src_addr_widths = OMAP_DMA_BUSWIDTHS; - caps->dstn_addr_widths = OMAP_DMA_BUSWIDTHS; + caps->dst_addr_widths = OMAP_DMA_BUSWIDTHS; caps->directions = BIT(DMA_DEV_TO_MEM) | BIT(DMA_MEM_TO_DEV); caps->cmd_pause = true; caps->cmd_terminate = true; diff --git a/drivers/dma/pl330.c b/drivers/dma/pl330.c index d5149aacd2fe..8904891ed5d5 100644 --- a/drivers/dma/pl330.c +++ b/drivers/dma/pl330.c @@ -2583,7 +2583,7 @@ static int pl330_dma_device_slave_caps(struct dma_chan *dchan, struct dma_slave_caps *caps) { caps->src_addr_widths = PL330_DMA_BUSWIDTHS; - caps->dstn_addr_widths = PL330_DMA_BUSWIDTHS; + caps->dst_addr_widths = PL330_DMA_BUSWIDTHS; caps->directions = BIT(DMA_DEV_TO_MEM) | BIT(DMA_MEM_TO_DEV); caps->cmd_pause = false; caps->cmd_terminate = true; diff --git a/drivers/dma/sirf-dma.c b/drivers/dma/sirf-dma.c index aac03ab10c54..149d7fddfa75 100644 --- a/drivers/dma/sirf-dma.c +++ b/drivers/dma/sirf-dma.c @@ -652,7 +652,7 @@ static int sirfsoc_dma_device_slave_caps(struct dma_chan *dchan, struct dma_slave_caps *caps) { caps->src_addr_widths = SIRFSOC_DMA_BUSWIDTHS; - caps->dstn_addr_widths = SIRFSOC_DMA_BUSWIDTHS; + caps->dst_addr_widths = SIRFSOC_DMA_BUSWIDTHS; caps->directions = BIT(DMA_DEV_TO_MEM) | BIT(DMA_MEM_TO_DEV); caps->cmd_pause = true; caps->cmd_terminate = true; diff --git a/include/linux/dmaengine.h b/include/linux/dmaengine.h index 1f9e642c66ad..f4b71ce1a0a6 100644 --- a/include/linux/dmaengine.h +++ b/include/linux/dmaengine.h @@ -388,7 +388,7 @@ enum dma_residue_granularity { /* struct dma_slave_caps - expose capabilities of a slave channel only * * @src_addr_widths: bit mask of src addr widths the channel supports - * @dstn_addr_widths: bit mask of dstn addr widths the channel supports + * @dst_addr_widths: bit mask of dstn addr widths the channel supports * @directions: bit mask of slave direction the channel supported * since the enum dma_transfer_direction is not defined as bits for each * type of direction, the dma controller should fill (1 << <TYPE>) and same @@ -399,7 +399,7 @@ enum dma_residue_granularity { */ struct dma_slave_caps { u32 src_addr_widths; - u32 dstn_addr_widths; + u32 dst_addr_widths; u32 directions; bool cmd_pause; bool cmd_terminate; @@ -639,10 +639,10 @@ struct dma_device { void (*device_free_chan_resources)(struct dma_chan *chan); struct dma_async_tx_descriptor *(*device_prep_dma_memcpy)( - struct dma_chan *chan, dma_addr_t dest, dma_addr_t src, + struct dma_chan *chan, dma_addr_t dst, dma_addr_t src, size_t len, unsigned long flags); struct dma_async_tx_descriptor *(*device_prep_dma_xor)( - struct dma_chan *chan, dma_addr_t dest, dma_addr_t *src, + struct dma_chan *chan, dma_addr_t dst, dma_addr_t *src, unsigned int src_cnt, size_t len, unsigned long flags); struct dma_async_tx_descriptor *(*device_prep_dma_xor_val)( struct dma_chan *chan, dma_addr_t *src, unsigned int src_cnt, @@ -935,11 +935,11 @@ async_dma_find_channel(enum dma_transaction_type type) #endif /* CONFIG_ASYNC_TX_DMA */ dma_cookie_t dma_async_memcpy_buf_to_buf(struct dma_chan *chan, - void *dest, void *src, size_t len); + void *dst, void *src, size_t len); dma_cookie_t dma_async_memcpy_buf_to_pg(struct dma_chan *chan, struct page *page, unsigned int offset, void *kdata, size_t len); dma_cookie_t dma_async_memcpy_pg_to_pg(struct dma_chan *chan, - struct page *dest_pg, unsigned int dest_off, struct page *src_pg, + struct page *dst_pg, unsigned int dst_off, struct page *src_pg, unsigned int src_off, size_t len); void dma_async_tx_descriptor_init(struct dma_async_tx_descriptor *tx, struct dma_chan *chan); diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 6307f85e871b..f4661be7de1d 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -151,7 +151,7 @@ static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substrea hw.info |= SNDRV_PCM_INFO_BATCH; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - addr_widths = dma_caps.dstn_addr_widths; + addr_widths = dma_caps.dst_addr_widths; else addr_widths = dma_caps.src_addr_widths; } -- 2.1.1

4 3

[alsa-devel] [PATCH 1/2] ASoC: rt5677: Add TDM channel select function
by Oder Chiou 16 Oct '14

16 Oct '14

Add TDM channel select function. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 75 +++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5677.h | 16 ++++++++-- 2 files changed, 89 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 9f07146..52cbb56 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -795,6 +795,14 @@ static unsigned int bst_tlv[] = { 8, 8, TLV_DB_SCALE_ITEM(5200, 0, 0), }; +static const char * const rt5677_adc_tdm_mode[] = { + "1/2/3/4", "2/1/3/4", "2/3/1/4", "4/1/2/3", "1/3/2/4", "1/4/2/3", + "3/1/2/4", "3/4/1/2" +}; + +static SOC_ENUM_SINGLE_DECL(rt5677_adc_tdm_mode_enum, + SND_SOC_NOPM, 0, rt5677_adc_tdm_mode); + static int rt5677_dsp_vad_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -820,6 +828,68 @@ static int rt5677_dsp_vad_put(struct snd_kcontrol *kcontrol, return 0; } +static int rt5677_if1_adc_tdm_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + unsigned int value; + + regmap_read(rt5677->regmap, RT5677_TDM1_CTRL2, &value); + + ucontrol->value.integer.value[0] = value & RT5677_IF1_ADC_CTRL_MASK; + + return 0; +} + +static int rt5677_if1_adc_tdm_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + + unsigned int value = ucontrol->value.integer.value[0]; + + regmap_update_bits(rt5677->regmap, RT5677_TDM1_CTRL2, + RT5677_IF1_ADC_CTRL_MASK, value); + + regmap_update_bits(rt5677->regmap, RT5677_TDM1_CTRL1, + RT5677_IF1_ADC_MODE_MASK, (!!value) << RT5677_IF1_ADC_MODE_SFT); + + return 0; +} + +static int rt5677_if2_adc_tdm_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + unsigned int value; + + regmap_read(rt5677->regmap, RT5677_TDM2_CTRL2, &value); + + ucontrol->value.integer.value[0] = value & RT5677_IF2_ADC_CTRL_MASK; + + return 0; +} + +static int rt5677_if2_adc_tdm_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + + unsigned int value = ucontrol->value.integer.value[0]; + + regmap_update_bits(rt5677->regmap, RT5677_TDM2_CTRL2, + RT5677_IF2_ADC_CTRL_MASK, value); + + regmap_update_bits(rt5677->regmap, RT5677_TDM2_CTRL1, + RT5677_IF2_ADC_MODE_MASK, (!!value) << RT5677_IF2_ADC_MODE_SFT); + + return 0; +} + static const struct snd_kcontrol_new rt5677_snd_controls[] = { /* OUTPUT Control */ SOC_SINGLE("OUT1 Playback Switch", RT5677_LOUT1, @@ -894,6 +964,11 @@ static const struct snd_kcontrol_new rt5677_snd_controls[] = { SOC_SINGLE_EXT("DSP VAD Switch", SND_SOC_NOPM, 0, 1, 0, rt5677_dsp_vad_get, rt5677_dsp_vad_put), + + SOC_ENUM_EXT("ADC1 TDM Channel Switch", rt5677_adc_tdm_mode_enum, + rt5677_if1_adc_tdm_get, rt5677_if1_adc_tdm_put), + SOC_ENUM_EXT("ADC2 TDM Channel Switch", rt5677_adc_tdm_mode_enum, + rt5677_if2_adc_tdm_get, rt5677_if2_adc_tdm_put), }; /** diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index 20efa4a..20223eb 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -799,7 +799,11 @@ #define RT5677_PDM2_I2C_EXE (0x1 << 1) #define RT5677_PDM2_I2C_BUSY (0x1 << 0) -/* MX3C TDM1 control 1 (0x3c) */ +/* MX3B TDM1 control 1 (0x3b) */ +#define RT5677_IF1_ADC_MODE_MASK (0x1 << 12) +#define RT5677_IF1_ADC_MODE_SFT 12 + +/* MX3C TDM1 control 2 (0x3c) */ #define RT5677_IF1_ADC4_MASK (0x3 << 10) #define RT5677_IF1_ADC4_SFT 10 #define RT5677_IF1_ADC3_MASK (0x3 << 8) @@ -808,8 +812,14 @@ #define RT5677_IF1_ADC2_SFT 6 #define RT5677_IF1_ADC1_MASK (0x3 << 4) #define RT5677_IF1_ADC1_SFT 4 +#define RT5677_IF1_ADC_CTRL_MASK (0x7 << 0) +#define RT5677_IF1_ADC_CTRL_SFT 0 + +/* MX40 TDM2 control 1 (0x40) */ +#define RT5677_IF2_ADC_MODE_MASK (0x1 << 12) +#define RT5677_IF2_ADC_MODE_SFT 12 -/* MX41 TDM2 control 1 (0x41) */ +/* MX41 TDM2 control 2 (0x41) */ #define RT5677_IF2_ADC4_MASK (0x3 << 10) #define RT5677_IF2_ADC4_SFT 10 #define RT5677_IF2_ADC3_MASK (0x3 << 8) @@ -818,6 +828,8 @@ #define RT5677_IF2_ADC2_SFT 6 #define RT5677_IF2_ADC1_MASK (0x3 << 4) #define RT5677_IF2_ADC1_SFT 4 +#define RT5677_IF2_ADC_CTRL_MASK (0x7 << 0) +#define RT5677_IF2_ADC_CTRL_SFT 0 /* Digital Microphone Control 1 (0x50) */ #define RT5677_DMIC_1_EN_MASK (0x1 << 15) -- 1.8.1.1.439.g50a6b54

4 5

[alsa-devel] [PATCH] ASoC: Intel: HSW/BDW only support S16 and S24 formats.
by Liam Girdwood 16 Oct '14

16 Oct '14

Fix driver with correct formats. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/intel/sst-haswell-pcm.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 33fc5c3..4df867c 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -691,9 +691,7 @@ static int hsw_pcm_new(struct snd_soc_pcm_runtime *rtd) } #define HSW_FORMATS \ - (SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S24_LE | \ - SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S16_LE |\ - SNDRV_PCM_FMTBIT_S8) + (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE) static struct snd_soc_dai_driver hsw_dais[] = { { -- 1.9.1

2 7

[alsa-devel] [RFC] ALSA: provide a simple mechanism to start playback at a given time
by Nick Stoughton 16 Oct '14

16 Oct '14

Initial implementation / Request For Comment. Given an absolute time based on a given clock (CLOCK_MONOTONIC, CLOCK_MONOTONIC_RAW, CLOCK_REALTIME etc), setup a high resolution timer to cause playback to be triggered at that time. --- include/linux/timekeeping.h | 3 ++ include/uapi/sound/asound.h | 6 +++ kernel/time/timekeeping.c | 16 ++++++++ sound/core/pcm_native.c | 99 +++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 124 insertions(+) diff --git a/include/linux/timekeeping.h b/include/linux/timekeeping.h index 1caa6b0..74eb6b2 100644 --- a/include/linux/timekeeping.h +++ b/include/linux/timekeeping.h @@ -32,6 +32,9 @@ extern void ktime_get_ts64(struct timespec64 *ts); extern int __getnstimeofday64(struct timespec64 *tv); extern void getnstimeofday64(struct timespec64 *tv); +/* Get the offset between monotonic and monotonic raw clocks */ +extern ktime_t ktime_get_raw_offset(void); + #if BITS_PER_LONG == 64 static inline int __getnstimeofday(struct timespec *ts) { diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 6ee5867..c7aa88c 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -462,6 +462,11 @@ struct snd_xfern { snd_pcm_uframes_t frames; }; +struct snd_clock_time { + clockid_t clock; + struct timespec time; +}; + enum { SNDRV_PCM_TSTAMP_TYPE_GETTIMEOFDAY = 0, /* gettimeofday equivalent */ SNDRV_PCM_TSTAMP_TYPE_MONOTONIC, /* posix_clock_monotonic equivalent */ @@ -547,6 +552,7 @@ enum { #define SNDRV_PCM_IOCTL_READN_FRAMES _IOR('A', 0x53, struct snd_xfern) #define SNDRV_PCM_IOCTL_LINK _IOW('A', 0x60, int) #define SNDRV_PCM_IOCTL_UNLINK _IO('A', 0x61) +#define SNDRV_PCM_IOCTL_START_AT _IOR('A', 0x62, struct snd_clock_time) /***************************************************************************** * * diff --git a/kernel/time/timekeeping.c b/kernel/time/timekeeping.c index ec1791f..3d5d0bc 100644 --- a/kernel/time/timekeeping.c +++ b/kernel/time/timekeeping.c @@ -789,6 +789,22 @@ error: /* even if we error out, we forwarded the time, so call update */ } EXPORT_SYMBOL(timekeeping_inject_offset); +/** + * ktime_get_raw_offset - get the offset in ktime format between + * the monotonic_raw clock and the monotonic clock + */ +ktime_t ktime_get_raw_offset(void) +{ + struct timespec rawtime = {0,0}; + struct timespec now = {0,0}; + struct timespec delta; + + ktime_get_ts(&now); + getrawmonotonic(&rawtime); + delta = timespec_sub(now, rawtime); + return timespec_to_ktime(delta); +} +EXPORT_SYMBOL(ktime_get_raw_offset); /** * timekeeping_get_tai_offset - Returns current TAI offset from UTC diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 85fe1a2..1121127 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -27,6 +27,7 @@ #include <linux/pm_qos.h> #include <linux/aio.h> #include <linux/dma-mapping.h> +#include <linux/time.h> #include <sound/core.h> #include <sound/control.h> #include <sound/info.h> @@ -38,6 +39,9 @@ #if defined(CONFIG_MIPS) && defined(CONFIG_DMA_NONCOHERENT) #include <dma-coherence.h> #endif +#if defined(CONFIG_HIGH_RES_TIMERS) +#include <linux/hrtimer.h> +#endif /* * Compatibility @@ -1016,6 +1020,99 @@ static struct action_ops snd_pcm_action_start = { .post_action = snd_pcm_post_start }; +#ifdef CONFIG_HIGH_RES_TIMERS +/* + * hrtimer interface + */ + +struct hrtimer_pcm { + struct hrtimer timer; + struct snd_pcm_substream *substream; +}; + +/* + * called from a hard irq context - no need for locks. + * only problem is that the caller might have gone away and closed the substream + * before the timer expires. + */ +enum hrtimer_restart snd_pcm_do_start_time(struct hrtimer *timer) +{ + struct hrtimer_pcm *pcm_timer; + struct snd_pcm_substream *substream; + int ret; + + pcm_timer = container_of(timer, struct hrtimer_pcm, timer); + substream = pcm_timer->substream; + + if (substream->runtime) { + ret = snd_pcm_do_start(substream, SNDRV_PCM_STATE_RUNNING); + if (ret == 0) { + snd_pcm_post_start(substream, SNDRV_PCM_STATE_RUNNING); + } + } + kfree(pcm_timer); + + return HRTIMER_NORESTART; +} + +int snd_pcm_start_at(struct snd_pcm_substream *substream, + struct snd_clock_time __user *_start_time) +{ + struct hrtimer_pcm *pcm_timer; + struct snd_clock_time start_time; + int clock; + ktime_t start_kt; + ktime_t raw_offset; + int ret; + + if (copy_from_user(&start_time, _start_time, sizeof(start_time))) + return -EFAULT; + + if (!timespec_valid(&start_time.time)) + return -EINVAL; + + start_kt = timespec_to_ktime(start_time.time); + clock = start_time.clock; + if (start_time.clock == CLOCK_MONOTONIC_RAW) { + raw_offset = ktime_get_raw_offset(); + + /* convert to clock monotonic */ + start_kt = ktime_add(raw_offset, start_kt); + clock = CLOCK_MONOTONIC; + } + + /* if not playback substream or less than 100 ms give up */ + if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK || + (ktime_compare(start_kt, ktime_add_ns(ktime_get(), 100000000)) <= 0)) { + return snd_pcm_action_lock_irq(&snd_pcm_action_start, + substream, SNDRV_PCM_STATE_RUNNING); + } + + ret = snd_pcm_pre_start(substream, SNDRV_PCM_STATE_PREPARED); + if (ret != 0) { + return ret; + } + + pcm_timer = kmalloc(sizeof(*pcm_timer), GFP_KERNEL); + hrtimer_init(&pcm_timer->timer, clock, HRTIMER_MODE_ABS); + + pcm_timer->timer.function = snd_pcm_do_start_time; + pcm_timer->substream = substream; + + hrtimer_start(&pcm_timer->timer, start_kt, HRTIMER_MODE_ABS); + + return 0; +} +#else +/* without high res time, interface is identical to snd_pcm_start() */ +int snd_pcm_start_at(struct snd_pcm_substream *substream, + struct snd_clock_time __user *_start_time) +{ + return snd_pcm_action_lock_irq(&snd_pcm_action_start, + substream, SNDRV_PCM_STATE_RUNNING); +} +#endif + /** * snd_pcm_start - start all linked streams * @substream: the PCM substream instance @@ -2698,6 +2795,8 @@ static int snd_pcm_common_ioctl1(struct file *file, return snd_pcm_reset(substream); case SNDRV_PCM_IOCTL_START: return snd_pcm_action_lock_irq(&snd_pcm_action_start, substream, SNDRV_PCM_STATE_RUNNING); + case SNDRV_PCM_IOCTL_START_AT: + return snd_pcm_start_at(substream, arg); case SNDRV_PCM_IOCTL_LINK: return snd_pcm_link(substream, (int)(unsigned long) arg); case SNDRV_PCM_IOCTL_UNLINK: -- 1.9.3

3 2

[alsa-devel] [RFC] pcm: provide a simple mechanism to start playback at a given time
by Nick Stoughton 16 Oct '14

16 Oct '14

Initial implementation / Request For Comment. Given an absolute time based on a given clock (CLOCK_MONOTONIC, CLOCK_MONOTONIC_RAW, CLOCK_REALTIME etc), setup a high resolution timer to cause playback to be triggered at that time. --- include/global.h | 10 +++++++++- include/pcm.h | 2 ++ include/sound/asound.h | 7 +++++++ src/pcm/pcm.c | 22 +++++++++++++++++++++- src/pcm/pcm_hw.c | 15 +++++++++++++++ src/pcm/pcm_local.h | 1 + 6 files changed, 55 insertions(+), 2 deletions(-) diff --git a/include/global.h b/include/global.h index 16a26dc..f0fb661 100644 --- a/include/global.h +++ b/include/global.h @@ -144,6 +144,8 @@ struct timespec { time_t tv_sec; /* seconds */ long tv_nsec; /* nanoseconds */ }; + +typedef int clockid_t; #endif #endif @@ -151,7 +153,13 @@ struct timespec { typedef struct timeval snd_timestamp_t; /** Hi-res timestamp */ typedef struct timespec snd_htimestamp_t; - +/** clock type */ +typedef clockid_t snd_clock_t; +/** a combined clock and time */ +typedef struct { + snd_clock_t clock; + snd_htimestamp_t time; +} snd_clock_time_t; /** \} */ #ifdef __cplusplus diff --git a/include/pcm.h b/include/pcm.h index db88ad5..dfaf69f 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -474,6 +474,8 @@ int snd_pcm_prepare(snd_pcm_t *pcm); int snd_pcm_reset(snd_pcm_t *pcm); int snd_pcm_status(snd_pcm_t *pcm, snd_pcm_status_t *status); int snd_pcm_start(snd_pcm_t *pcm); +int snd_pcm_start_at(snd_pcm_t *pcm, snd_clock_time_t *start_time); +#define SND_PCM_START_AT(pcm, start_time) snd_pcm_start_at(pcm, start_time) int snd_pcm_drop(snd_pcm_t *pcm); int snd_pcm_drain(snd_pcm_t *pcm); int snd_pcm_pause(snd_pcm_t *pcm, int enable); diff --git a/include/sound/asound.h b/include/sound/asound.h index 6ee5867..92647ce 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -434,6 +434,11 @@ struct snd_pcm_mmap_control { snd_pcm_uframes_t avail_min; /* RW: min available frames for wakeup */ }; +struct snd_clock_time { + clockid_t clock; + struct timespec time; +}; + #define SNDRV_PCM_SYNC_PTR_HWSYNC (1<<0) /* execute hwsync */ #define SNDRV_PCM_SYNC_PTR_APPL (1<<1) /* get appl_ptr from driver (r/w op) */ #define SNDRV_PCM_SYNC_PTR_AVAIL_MIN (1<<2) /* get avail_min from driver */ @@ -547,6 +552,8 @@ enum { #define SNDRV_PCM_IOCTL_READN_FRAMES _IOR('A', 0x53, struct snd_xfern) #define SNDRV_PCM_IOCTL_LINK _IOW('A', 0x60, int) #define SNDRV_PCM_IOCTL_UNLINK _IO('A', 0x61) +#define SNDRV_PCM_IOCTL_START_AT _IOR('A', 0x62, struct snd_clock_time) + /***************************************************************************** * * diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 4a7be6c..7051a22 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -1084,6 +1084,25 @@ int snd_pcm_start(snd_pcm_t *pcm) } /** + * \brief Start a PCM at a specified point in the future + * \param pcm PCM handle + * \param start_time time to start - based on specified clock + * \return 0 on success otherwise a negative error code + */ +int snd_pcm_start_at(snd_pcm_t *pcm, snd_clock_time_t *start_at) +{ + assert(pcm); + assert(start_at); + if (CHECK_SANITY(! pcm->setup)) { + SNDMSG("PCM not set up"); + return -EIO; + } + if (pcm->fast_ops->start_at) + return pcm->fast_ops->start_at(pcm->fast_op_arg, start_at); + return -EINVAL; +} + +/** * \brief Stop a PCM dropping pending frames * \param pcm PCM handle * \return 0 on success otherwise a negative error code @@ -2444,8 +2463,9 @@ int snd_pcm_wait_nocheck(snd_pcm_t *pcm, int timeout) continue; return -errno; } - if (! err_poll) + if (! err_poll) { break; + } err = snd_pcm_poll_descriptors_revents(pcm, pfd, npfds, &revents); if (err < 0) return err; diff --git a/src/pcm/pcm_hw.c b/src/pcm/pcm_hw.c index c34b766..9888253 100644 --- a/src/pcm/pcm_hw.c +++ b/src/pcm/pcm_hw.c @@ -620,6 +620,20 @@ static int snd_pcm_hw_start(snd_pcm_t *pcm) return 0; } +static int snd_pcm_hw_start_at(snd_pcm_t *pcm, snd_clock_time_t *start_time) +{ + snd_pcm_hw_t *hw = pcm->private_data; + int err; + + sync_ptr(hw, 0); + if (ioctl(hw->fd, SNDRV_PCM_IOCTL_START_AT, start_time) < 0) { + err = -errno; + SYSMSG("SNDRV_PCM_IOCTL_START_AT failed (%i)", err); + return err; + } + return 0; +} + static int snd_pcm_hw_drop(snd_pcm_t *pcm) { snd_pcm_hw_t *hw = pcm->private_data; @@ -1336,6 +1350,7 @@ static const snd_pcm_fast_ops_t snd_pcm_hw_fast_ops = { .prepare = snd_pcm_hw_prepare, .reset = snd_pcm_hw_reset, .start = snd_pcm_hw_start, + .start_at = snd_pcm_hw_start_at, .drop = snd_pcm_hw_drop, .drain = snd_pcm_hw_drain, .pause = snd_pcm_hw_pause, diff --git a/src/pcm/pcm_local.h b/src/pcm/pcm_local.h index 394505f..c3e58af 100644 --- a/src/pcm/pcm_local.h +++ b/src/pcm/pcm_local.h @@ -154,6 +154,7 @@ typedef struct { int (*prepare)(snd_pcm_t *pcm); int (*reset)(snd_pcm_t *pcm); int (*start)(snd_pcm_t *pcm); + int (*start_at)(snd_pcm_t *pcm, snd_clock_time_t *start_time); int (*drop)(snd_pcm_t *pcm); int (*drain)(snd_pcm_t *pcm); int (*pause)(snd_pcm_t *pcm, int enable); -- 1.9.3

2 1

[alsa-devel] [PATCH] ASoC: tegra: Add platform driver for rt5677 audio codec
by Anatol Pomozov 16 Oct '14

16 Oct '14

The driver supports NVIDIA Tegra Ryu board Signed-off-by: Anatol Pomozov <anatol(a)google.com> --- .../bindings/sound/nvidia,tegra-audio-rt5677.txt | 73 ++++ sound/soc/tegra/Kconfig | 11 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra_rt5677.c | 383 +++++++++++++++++++++ 4 files changed, 469 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.txt create mode 100644 sound/soc/tegra/tegra_rt5677.c diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.txt new file mode 100644 index 0000000..0e483c5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.txt @@ -0,0 +1,73 @@ +NVIDIA Tegra audio complex, with RT5677 CODEC + +Required properties: +- compatible : "nvidia,tegra-audio-rt5677" +- clocks : Must contain an entry for each entry in clock-names. + See ../clocks/clock-bindings.txt for details. +- clock-names : Must include the following entries: + - pll_a + - pll_a_out0 + - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk) +- nvidia,model : The user-visible name of this sound complex. +- nvidia,audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the RT5677's pins (as documented in its binding), and the jacks + on the board: + + * Headphone + * Speaker + * Headset Mic + * Internal Mic 1 + * Internal Mic 2 + +- nvidia,i2s-controller : The phandle of the Tegra I2S controller that's + connected to the CODEC. +- nvidia,audio-codec : The phandle of the RT5677 audio codec. This binding + assumes that AIF1 on the CODEC is connected to Tegra. +- nvidia,speaker-codec: The phandle of the board specific speaker amplifier. + This binding assumes that AIF2 on the audio CODEC is connected to this + amplifier. + +Optional properties: +- nvidia,hp-det-gpios : The GPIO that detects headphones are plugged in +- nvidia,hp-en-gpios : The GPIO that enables headphone amplifier +- nvidia,mic-present-gpios: The GPIO that mic jack is plugged in +- nvidia,dmic-clk-en-gpios : The GPIO that gates DMIC clock signal + +Example: + +sound { + compatible = "nvidia,tegra-audio-rt5677-ryu", + "nvidia,tegra-audio-rt5677"; + nvidia,model = "NVIDIA Tegra Ryu"; + + nvidia,audio-routing = + /* rt5677 codec */ + "Headphone", "LOUT2", + "Headphone", "LOUT1", + "Headset Mic", "MICBIAS1", + "IN1P", "Headset Mic", + "IN1N", "Headset Mic", + "DMIC L1", "Internal Mic 1", + "DMIC R1", "Internal Mic 1", + "DMIC L2", "Internal Mic 2", + "DMIC R2", "Internal Mic 2", + /* speaker amp */ + "DAC", "AIF2TX", + "Speaker", "OUT"; + + nvidia,i2s-controller = <&tegra_i2s1>; + nvidia,audio-codec = <&rt5677>; + nvidia,speaker-codec = <&ssm4567>; + + nvidia,hp-det-gpios = <&gpio TEGRA_GPIO(R, 7) GPIO_ACTIVE_HIGH>; + nvidia,mic-present-gpios = <&rt5677 5 GPIO_ACTIVE_LOW>; + nvidia,hp-en-gpios = <&rt5677 1 GPIO_ACTIVE_HIGH>; + nvidia,dmic-clk-en-gpios = <&rt5677 2 GPIO_ACTIVE_HIGH>; + + clocks = <&tegra_car TEGRA124_CLK_PLL_A>, + <&tegra_car TEGRA124_CLK_PLL_A_OUT0>, + <&tegra_car TEGRA124_CLK_EXTERN1>; + clock-names = "pll_a", "pll_a_out0", "mclk"; +}; diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 31198cf7..c30611e 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -128,3 +128,14 @@ config SND_SOC_TEGRA_MAX98090 help Say Y or M here if you want to add support for SoC audio on Tegra boards using the MAX98090 codec, such as Venice2. + +config SND_SOC_TEGRA_RT5677 + tristate "SoC Audio support for Tegra boards using a RT5677 codec" + depends on SND_SOC_TEGRA && I2C && GPIOLIB + select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC + select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC + select SND_SOC_RT5677 + select SND_SOC_SSM4567 + help + Say Y or M here if you want to add support for SoC audio on Tegra + boards using the RT5677 codec, such as Ryu. diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile index 5ae588c..9171655 100644 --- a/sound/soc/tegra/Makefile +++ b/sound/soc/tegra/Makefile @@ -19,6 +19,7 @@ obj-$(CONFIG_SND_SOC_TEGRA30_I2S) += snd-soc-tegra30-i2s.o # Tegra machine Support snd-soc-tegra-rt5640-objs := tegra_rt5640.o +snd-soc-tegra-rt5677-objs := tegra_rt5677.o snd-soc-tegra-wm8753-objs := tegra_wm8753.o snd-soc-tegra-wm8903-objs := tegra_wm8903.o snd-soc-tegra-wm9712-objs := tegra_wm9712.o @@ -27,6 +28,7 @@ snd-soc-tegra-alc5632-objs := tegra_alc5632.o snd-soc-tegra-max98090-objs := tegra_max98090.o obj-$(CONFIG_SND_SOC_TEGRA_RT5640) += snd-soc-tegra-rt5640.o +obj-$(CONFIG_SND_SOC_TEGRA_RT5677) += snd-soc-tegra-rt5677.o obj-$(CONFIG_SND_SOC_TEGRA_WM8753) += snd-soc-tegra-wm8753.o obj-$(CONFIG_SND_SOC_TEGRA_WM8903) += snd-soc-tegra-wm8903.o obj-$(CONFIG_SND_SOC_TEGRA_WM9712) += snd-soc-tegra-wm9712.o diff --git a/sound/soc/tegra/tegra_rt5677.c b/sound/soc/tegra/tegra_rt5677.c new file mode 100644 index 0000000..ce15876 --- /dev/null +++ b/sound/soc/tegra/tegra_rt5677.c @@ -0,0 +1,383 @@ +/* +* tegra_rt5677.c - Tegra machine ASoC driver for boards using RT5677 codec. + * + * Copyright (c) 2014, The Chromium OS Authors. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>. + * + * Based on code copyright/by: + * + * Copyright (C) 2010-2012 - NVIDIA, Inc. + * Copyright (C) 2011 The AC100 Kernel Team <ac100(a)lists.lauchpad.net> + * (c) 2009, 2010 Nvidia Graphics Pvt. Ltd. + * Copyright 2007 Wolfson Microelectronics PLC. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> + +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "../codecs/rt5677.h" + +#include "tegra_asoc_utils.h" + +#define DRV_NAME "tegra-snd-rt5677" + +struct tegra_rt5677 { + struct tegra_asoc_utils_data util_data; + int gpio_hp_det; + int gpio_hp_en; + int gpio_mic_present; + int gpio_dmic_clk_en; +}; + +static struct snd_soc_pcm_stream tegra_rt5677_spk_params = { + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .channels_min = 2, + .channels_max = 2, +}; + +static int tegra_rt5677_asoc_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_card *card = rtd->card; + struct tegra_rt5677 *machine = snd_soc_card_get_drvdata(card); + int srate, mclk, err; + + srate = params_rate(params); + mclk = 256 * srate; + + err = tegra_asoc_utils_set_rate(&machine->util_data, srate, mclk); + if (err < 0) { + dev_err(card->dev, "Can't configure clocks\n"); + return err; + } + + err = snd_soc_dai_set_sysclk(codec_dai, RT5677_SCLK_S_MCLK, mclk, + SND_SOC_CLOCK_IN); + if (err < 0) { + dev_err(card->dev, "codec_dai clock not set\n"); + return err; + } + + /* both dai interfaces use the same bit clock, + * set rate for both interfaces the same, otherwise driver might not + * be able to configure clock divider. + */ + tegra_rt5677_spk_params.rate_min = srate; + tegra_rt5677_spk_params.rate_max = srate; + + return 0; +} + +static int tegra_rt5677_event_hp(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct tegra_rt5677 *machine = snd_soc_card_get_drvdata(card); + + if (!gpio_is_valid(machine->gpio_hp_en)) + return 0; + + gpio_set_value_cansleep(machine->gpio_hp_en, + SND_SOC_DAPM_EVENT_ON(event)); + + return 0; +} + +static struct snd_soc_ops tegra_rt5677_ops = { + .hw_params = tegra_rt5677_asoc_hw_params, +}; + +static struct snd_soc_jack tegra_rt5677_hp_jack; + +static struct snd_soc_jack_pin tegra_rt5677_hp_jack_pins = { + .pin = "Headphone", + .mask = SND_JACK_HEADPHONE, +}; +static struct snd_soc_jack_gpio tegra_rt5677_hp_jack_gpio = { + .name = "Headphone detection", + .report = SND_JACK_HEADPHONE, + .debounce_time = 150, +}; + +static struct snd_soc_jack tegra_rt5677_mic_jack; + +static struct snd_soc_jack_pin tegra_rt5677_mic_jack_pins = { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, +}; + +static struct snd_soc_jack_gpio tegra_rt5677_mic_jack_gpio = { + .name = "Headset Mic detection", + .report = SND_JACK_MICROPHONE, + .debounce_time = 150, + .invert = 1 +}; + +static const struct snd_soc_dapm_widget tegra_rt5677_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_HP("Headphone", tegra_rt5677_event_hp), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Internal Mic 1", NULL), + SND_SOC_DAPM_MIC("Internal Mic 2", NULL), +}; + +static const struct snd_kcontrol_new tegra_rt5677_controls[] = { + SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Internal Mic 1"), + SOC_DAPM_PIN_SWITCH("Internal Mic 2"), +}; + +static int tegra_rt5677_asoc_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = codec_dai->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + struct tegra_rt5677 *machine = snd_soc_card_get_drvdata(rtd->card); + + snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, + &tegra_rt5677_hp_jack); + snd_soc_jack_add_pins(&tegra_rt5677_hp_jack, 1, + &tegra_rt5677_hp_jack_pins); + + if (gpio_is_valid(machine->gpio_hp_det)) { + tegra_rt5677_hp_jack_gpio.gpio = machine->gpio_hp_det; + snd_soc_jack_add_gpios(&tegra_rt5677_hp_jack, 1, + &tegra_rt5677_hp_jack_gpio); + } + + + snd_soc_jack_new(codec, "Mic Jack", SND_JACK_MICROPHONE, + &tegra_rt5677_mic_jack); + snd_soc_jack_add_pins(&tegra_rt5677_mic_jack, 1, + &tegra_rt5677_mic_jack_pins); + + if (gpio_is_valid(machine->gpio_mic_present)) { + tegra_rt5677_mic_jack_gpio.gpio = machine->gpio_mic_present; + snd_soc_jack_add_gpios(&tegra_rt5677_mic_jack, 1, + &tegra_rt5677_mic_jack_gpio); + } + + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS1"); + + return 0; +} + +static int tegra_rt5677_card_remove(struct snd_soc_card *card) +{ + struct tegra_rt5677 *machine = snd_soc_card_get_drvdata(card); + + if (gpio_is_valid(machine->gpio_hp_det)) { + snd_soc_jack_free_gpios(&tegra_rt5677_hp_jack, 1, + &tegra_rt5677_hp_jack_gpio); + } + + if (gpio_is_valid(machine->gpio_mic_present)) { + snd_soc_jack_free_gpios(&tegra_rt5677_mic_jack, 1, + &tegra_rt5677_mic_jack_gpio); + } + + return 0; +} + +static struct snd_soc_dai_link tegra_rt5677_dai[] = { + { + .name = "RT5677", + .stream_name = "RT5677 PCM", + .codec_dai_name = "rt5677-aif1", + .init = tegra_rt5677_asoc_init, + .ops = &tegra_rt5677_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + }, { + .name = "SSM4567", + .stream_name = "Speaker", + .cpu_dai_name = "rt5677-aif2", + .codec_dai_name = "ssm4567-hifi", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .params = &tegra_rt5677_spk_params, + .playback_only = true, + } +}; + +static struct snd_soc_card snd_soc_tegra_rt5677 = { + .name = "tegra-rt5677", + .owner = THIS_MODULE, + .remove = tegra_rt5677_card_remove, + .dai_link = tegra_rt5677_dai, + .num_links = ARRAY_SIZE(tegra_rt5677_dai), + .controls = tegra_rt5677_controls, + .num_controls = ARRAY_SIZE(tegra_rt5677_controls), + .dapm_widgets = tegra_rt5677_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tegra_rt5677_dapm_widgets), + .fully_routed = true, +}; + +static int tegra_rt5677_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct snd_soc_card *card = &snd_soc_tegra_rt5677; + struct tegra_rt5677 *machine; + int ret; + struct snd_soc_dai_link *codec_dai = &tegra_rt5677_dai[0]; + struct snd_soc_dai_link *speaker_dai = &tegra_rt5677_dai[1]; + + machine = devm_kzalloc(&pdev->dev, + sizeof(struct tegra_rt5677), GFP_KERNEL); + if (!machine) + return -ENOMEM; + + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + snd_soc_card_set_drvdata(card, machine); + + machine->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0); + if (machine->gpio_hp_det == -EPROBE_DEFER) + return -EPROBE_DEFER; + + machine->gpio_mic_present = of_get_named_gpio(np, + "nvidia,mic-present-gpios", 0); + if (machine->gpio_mic_present == -EPROBE_DEFER) + return -EPROBE_DEFER; + + machine->gpio_hp_en = of_get_named_gpio(np, "nvidia,hp-en-gpios", 0); + if (machine->gpio_hp_en == -EPROBE_DEFER) + return -EPROBE_DEFER; + if (gpio_is_valid(machine->gpio_hp_en)) { + ret = devm_gpio_request_one(&pdev->dev, machine->gpio_hp_en, + GPIOF_OUT_INIT_LOW, "hp_en"); + if (ret) { + dev_err(card->dev, "cannot get hp_en gpio\n"); + return ret; + } + } + + machine->gpio_dmic_clk_en = of_get_named_gpio(np, + "nvidia,dmic-clk-en-gpios", 0); + if (machine->gpio_dmic_clk_en == -EPROBE_DEFER) + return -EPROBE_DEFER; + if (gpio_is_valid(machine->gpio_dmic_clk_en)) { + ret = devm_gpio_request_one(&pdev->dev, + machine->gpio_dmic_clk_en, + GPIOF_OUT_INIT_HIGH, "dmic_clk_en"); + if (ret) { + dev_err(card->dev, "cannot get dmic_clk_en gpio\n"); + return ret; + } + } + + ret = snd_soc_of_parse_card_name(card, "nvidia,model"); + if (ret) + goto err; + + ret = snd_soc_of_parse_audio_routing(card, "nvidia,audio-routing"); + if (ret) + goto err; + + codec_dai->codec_of_node = of_parse_phandle(np, + "nvidia,audio-codec", 0); + if (!codec_dai->codec_of_node) { + dev_err(&pdev->dev, + "Property 'nvidia,audio-codec' missing or invalid\n"); + ret = -EINVAL; + goto err; + } + + codec_dai->cpu_of_node = of_parse_phandle(np, + "nvidia,i2s-controller", 0); + if (!codec_dai->cpu_of_node) { + dev_err(&pdev->dev, + "Property 'nvidia,i2s-controller' missing or invalid\n"); + ret = -EINVAL; + goto err; + } + codec_dai->platform_of_node = codec_dai->cpu_of_node; + + speaker_dai->codec_of_node = of_parse_phandle(np, + "nvidia,speaker-codec", 0); + if (!speaker_dai->codec_of_node) { + dev_err(&pdev->dev, + "Property 'nvidia,speaker-codec' missing or invalid\n"); + ret = -EINVAL; + goto err; + } + speaker_dai->cpu_of_node = codec_dai->codec_of_node; + + ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev); + if (ret) + goto err; + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", + ret); + goto err_fini_utils; + } + + return 0; + +err_fini_utils: + tegra_asoc_utils_fini(&machine->util_data); +err: + return ret; +} + +static int tegra_rt5677_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct tegra_rt5677 *machine = snd_soc_card_get_drvdata(card); + + snd_soc_unregister_card(card); + + tegra_asoc_utils_fini(&machine->util_data); + + return 0; +} + +static const struct of_device_id tegra_rt5677_of_match[] = { + { .compatible = "nvidia,tegra-audio-rt5677", }, + {}, +}; + +static struct platform_driver tegra_rt5677_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + .of_match_table = tegra_rt5677_of_match, + }, + .probe = tegra_rt5677_probe, + .remove = tegra_rt5677_remove, +}; +module_platform_driver(tegra_rt5677_driver); + +MODULE_AUTHOR("Anatol Pomozov <anatol(a)google.com>"); +MODULE_DESCRIPTION("Tegra+RT5677 machine ASoC driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, tegra_rt5677_of_match); -- 2.1.0.rc2.206.gedb03e5

3 3

[alsa-devel] [PATCH] ASoC: intel: turn off COMPILE_TEST for medfield
by Vinod Koul 16 Oct '14

16 Oct '14

Since medfield machine uses SCU_IPC which is not availble for all archs, so compile test fails on these Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 726f7d8..f5b4a9c 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -1,6 +1,6 @@ config SND_MFLD_MACHINE tristate "SOC Machine Audio driver for Intel Medfield MID platform" - depends on INTEL_SCU_IPC || COMPILE_TEST + depends on INTEL_SCU_IPC select SND_SOC_SN95031 select SND_SST_MFLD_PLATFORM help -- 1.7.0.4

2 1

[alsa-devel] [PATCH v10 0/5] Add mrfld DSP topology and widgets
by Vinod Koul 16 Oct '14

16 Oct '14

This patch series adds DPCM and DAPM widgets to handle DSP topology in the SST platform driver for merrifield. changes in v10: - add SST_GAIN_NUM_CONTROLS for use in gain cell size calculation changes in v9: - fix some corner conditions and error checks changes in v8: - fix the locking and comments blobs - fix the send_ssp_slot to send IPC on none - fix controls to have "Volume" and "Switch" wherever neccessary - add mute for CPU dai in asoc core patch Vinod Koul (5): ASoC: Intel: mrfld: add the gain controls ASoC: Intel: mrfld: add DSP core controls ASoC: Intel: mrfld: add the DSP DAPM widgets ASoC: Intel: mfld-pcm: add FE and BE ops ASoC: Intel: mrfld: add the DSP mixers sound/soc/intel/Kconfig | 2 +- sound/soc/intel/sst-atom-controls.c | 1208 ++++++++++++++++++++++++++++++- sound/soc/intel/sst-atom-controls.h | 428 +++++++++++ sound/soc/intel/sst-mfld-platform-pcm.c | 153 ++++- sound/soc/intel/sst-mfld-platform.h | 4 + 5 files changed, 1764 insertions(+), 31 deletions(-)

2 6

[alsa-devel] [PATCH] ASoC: rt5677: Print more information if setting DAI clock failed
by Anatol Pomozov 16 Oct '14

16 Oct '14

Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- sound/soc/codecs/rt5677.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 2f0d19f..a8d9eab 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -3104,7 +3104,8 @@ static int rt5677_hw_params(struct snd_pcm_substream *substream, rt5677->lrck[dai->id] = params_rate(params); pre_div = rl6231_get_clk_info(rt5677->sysclk, rt5677->lrck[dai->id]); if (pre_div < 0) { - dev_err(codec->dev, "Unsupported clock setting\n"); + dev_err(codec->dev, "Unsupported clock setting: sysclk=%dHz lrck=%dHz\n", + rt5677->sysclk, rt5677->lrck[dai->id]); return -EINVAL; } frame_size = snd_soc_params_to_frame_size(params); -- 2.1.0.rc2.206.gedb03e5

2 1

[alsa-devel] [PATCH v2] ASoC: rt5677: Support DSP function for VAD application
by Oder Chiou 15 Oct '14

15 Oct '14

The ALC5677 has a programmable DSP, and there is a SPI that is dadicated for DSP firmware loading. Therefore, the patch includes a SPI driver for writing the DSP firmware. The VAD(Voice Activaty Detection) has be implemented and use the DSP to recognize the key phase. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/Makefile | 2 +- sound/soc/codecs/rt5677-spi.c | 129 +++++++++++++++++++ sound/soc/codecs/rt5677-spi.h | 21 +++ sound/soc/codecs/rt5677.c | 288 +++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/rt5677.h | 6 + 5 files changed, 441 insertions(+), 5 deletions(-) create mode 100644 sound/soc/codecs/rt5677-spi.c create mode 100644 sound/soc/codecs/rt5677-spi.h diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 5dce451..4435f9f 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -79,7 +79,7 @@ snd-soc-rt5640-objs := rt5640.o snd-soc-rt5645-objs := rt5645.o snd-soc-rt5651-objs := rt5651.o snd-soc-rt5670-objs := rt5670.o -snd-soc-rt5677-objs := rt5677.o +snd-soc-rt5677-objs := rt5677.o rt5677-spi.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o snd-soc-alc5632-objs := alc5632.o diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c new file mode 100644 index 0000000..805db12 --- /dev/null +++ b/sound/soc/codecs/rt5677-spi.c @@ -0,0 +1,129 @@ +/* + * rt5677-spi.c -- RT5677 ALSA SoC audio codec driver + * + * Copyright 2013 Realtek Semiconductor Corp. + * Author: Oder Chiou <oder_chiou(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/input.h> +#include <linux/spi/spi.h> +#include <linux/device.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/interrupt.h> +#include <linux/irq.h> +#include <linux/slab.h> +#include <linux/gpio.h> +#include <linux/sched.h> +#include <linux/kthread.h> +#include <linux/uaccess.h> +#include <linux/miscdevice.h> +#include <linux/regulator/consumer.h> +#include <linux/pm_qos.h> +#include <linux/sysfs.h> +#include <linux/clk.h> +#include <linux/firmware.h> + +#include "rt5677-spi.h" + +static struct spi_device *g_spi; + +/** + * rt5677_spi_write - Write data to SPI. + * @txbuf: Data Buffer for writing. + * @len: Data length. + * + * + * Returns true for success. + */ +int rt5677_spi_write(u8 *txbuf, size_t len) +{ + int status; + + status = spi_write(g_spi, txbuf, len); + + if (status) + dev_err(&g_spi->dev, "rt5677_spi_write error %d\n", status); + + return status; +} + +/** + * rt5677_spi_burst_write - Write data to SPI by rt5677 dsp memory address. + * @addr: Start address. + * @txbuf: Data Buffer for writng. + * @len: Data length, it must be a multiple of 8. + * + * + * Returns true for success. + */ +int rt5677_spi_burst_write(u32 addr, const struct firmware *fw) +{ + u8 spi_cmd = RT5677_SPI_CMD_BURST_WRITE; + u8 *write_buf; + unsigned int i, end, offset = 0; + + write_buf = kmalloc(RT5677_SPI_BUF_LEN + 6, GFP_KERNEL); + + if (write_buf == NULL) + return -ENOMEM; + + while (offset < fw->size) { + if (offset + RT5677_SPI_BUF_LEN <= fw->size) + end = RT5677_SPI_BUF_LEN; + else + end = fw->size % RT5677_SPI_BUF_LEN; + + write_buf[0] = spi_cmd; + write_buf[1] = ((addr + offset) & 0xff000000) >> 24; + write_buf[2] = ((addr + offset) & 0x00ff0000) >> 16; + write_buf[3] = ((addr + offset) & 0x0000ff00) >> 8; + write_buf[4] = ((addr + offset) & 0x000000ff) >> 0; + + for (i = 0; i < end; i += 8) { + write_buf[i + 12] = fw->data[offset + i + 0]; + write_buf[i + 11] = fw->data[offset + i + 1]; + write_buf[i + 10] = fw->data[offset + i + 2]; + write_buf[i + 9] = fw->data[offset + i + 3]; + write_buf[i + 8] = fw->data[offset + i + 4]; + write_buf[i + 7] = fw->data[offset + i + 5]; + write_buf[i + 6] = fw->data[offset + i + 6]; + write_buf[i + 5] = fw->data[offset + i + 7]; + } + + write_buf[end + 5] = spi_cmd; + + rt5677_spi_write(write_buf, end + 6); + + offset += RT5677_SPI_BUF_LEN; + } + + kfree(write_buf); + + return 0; +} + +static int rt5677_spi_probe(struct spi_device *spi) +{ + g_spi = spi; + return 0; +} + +static struct spi_driver rt5677_spi_driver = { + .driver = { + .name = "rt5677_spidev", + .bus = &spi_bus_type, + .owner = THIS_MODULE, + }, + .probe = rt5677_spi_probe, +}; +module_spi_driver(rt5677_spi_driver); + +MODULE_DESCRIPTION("ASoC RT5677 SPI driver"); +MODULE_AUTHOR("Oder Chiou <oder_chiou(a)realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt5677-spi.h b/sound/soc/codecs/rt5677-spi.h new file mode 100644 index 0000000..7528bfd --- /dev/null +++ b/sound/soc/codecs/rt5677-spi.h @@ -0,0 +1,21 @@ +/* + * rt5677-spi.h -- RT5677 ALSA SoC audio codec driver + * + * Copyright 2013 Realtek Semiconductor Corp. + * Author: Oder Chiou <oder_chiou(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __RT5671_SPI_H__ +#define __RT5671_SPI_H__ + +#define RT5677_SPI_BUF_LEN 240 +#define RT5677_SPI_CMD_BURST_WRITE 0x05 + +int rt5677_spi_write(u8 *txbuf, size_t len); +int rt5677_spi_burst_write(u32 addr, const struct firmware *fw); + +#endif /* __RT5677_SPI_H__ */ diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 16aa4d9..b5f6273 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -20,6 +20,7 @@ #include <linux/i2c.h> #include <linux/platform_device.h> #include <linux/spi/spi.h> +#include <linux/firmware.h> #include <linux/gpio.h> #include <sound/core.h> #include <sound/pcm.h> @@ -31,6 +32,7 @@ #include "rl6231.h" #include "rt5677.h" +#include "rt5677-spi.h" #define RT5677_DEVICE_ID 0x6327 @@ -537,6 +539,243 @@ static bool rt5677_readable_register(struct device *dev, unsigned int reg) } } +/** + * rt5677_dsp_mode_i2c_write_addr - Write value to address on DSP mode. + * @codec: SoC audio codec device. + * @addr: Address index. + * @value: Address data. + * + * + * Returns 0 for success or negative error code. + */ +static int rt5677_dsp_mode_i2c_write_addr(struct snd_soc_codec *codec, + unsigned int addr, unsigned int value, unsigned int opcode) +{ + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + int ret; + + mutex_lock(&rt5677->dsp_cmd_lock); + + ret = regmap_write(rt5677->regmap, RT5677_DSP_I2C_ADDR_MSB, addr >> 16); + if (ret < 0) { + dev_err(codec->dev, "Failed to set addr msb value: %d\n", ret); + goto err; + } + + ret = regmap_write(rt5677->regmap, RT5677_DSP_I2C_ADDR_LSB, + addr & 0xffff); + if (ret < 0) { + dev_err(codec->dev, "Failed to set addr lsb value: %d\n", ret); + goto err; + } + + ret = regmap_write(rt5677->regmap, RT5677_DSP_I2C_DATA_MSB, + value >> 16); + if (ret < 0) { + dev_err(codec->dev, "Failed to set data msb value: %d\n", ret); + goto err; + } + + ret = regmap_write(rt5677->regmap, RT5677_DSP_I2C_DATA_LSB, + value & 0xffff); + if (ret < 0) { + dev_err(codec->dev, "Failed to set data lsb value: %d\n", ret); + goto err; + } + + ret = regmap_write(rt5677->regmap, RT5677_DSP_I2C_OP_CODE, opcode); + if (ret < 0) { + dev_err(codec->dev, "Failed to set op code value: %d\n", ret); + goto err; + } + +err: + mutex_unlock(&rt5677->dsp_cmd_lock); + + return ret; +} + +/** + * rt5677_dsp_mode_i2c_read_addr - Read value from address on DSP mode. + * @codec: SoC audio codec device. + * @addr: Address index. + * @value: Address data. + * + * Returns 0 for success or negative error code. + */ +static int rt5677_dsp_mode_i2c_read_addr( + struct snd_soc_codec *codec, unsigned int addr, unsigned int *value) +{ + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + int ret; + unsigned int msb, lsb; + + mutex_lock(&rt5677->dsp_cmd_lock); + + ret = regmap_write(rt5677->regmap, RT5677_DSP_I2C_ADDR_MSB, addr >> 16); + if (ret < 0) { + dev_err(codec->dev, "Failed to set addr msb value: %d\n", ret); + goto err; + } + + ret = regmap_write(rt5677->regmap, RT5677_DSP_I2C_ADDR_LSB, + addr & 0xffff); + if (ret < 0) { + dev_err(codec->dev, "Failed to set addr lsb value: %d\n", ret); + goto err; + } + + ret = regmap_write(rt5677->regmap, RT5677_DSP_I2C_OP_CODE , 0x0002); + if (ret < 0) { + dev_err(codec->dev, "Failed to set op code value: %d\n", ret); + goto err; + } + + regmap_read(rt5677->regmap, RT5677_DSP_I2C_DATA_MSB, &msb); + regmap_read(rt5677->regmap, RT5677_DSP_I2C_DATA_LSB, &lsb); + *value = (msb << 16) | lsb; + +err: + mutex_unlock(&rt5677->dsp_cmd_lock); + + return ret; +} + +/** + * rt5677_dsp_mode_i2c_write - Write register on DSP mode. + * @codec: SoC audio codec device. + * @reg: Register index. + * @value: Register data. + * + * + * Returns 0 for success or negative error code. + */ +static int rt5677_dsp_mode_i2c_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + return rt5677_dsp_mode_i2c_write_addr(codec, 0x18020000 + reg * 2, + value, 0x0001); +} + +/** + * rt5677_dsp_mode_i2c_read - Read register on DSP mode. + * @codec: SoC audio codec device. + * @reg: Register index. + * + * + * Returns Register value. + */ +static unsigned int rt5677_dsp_mode_i2c_read( + struct snd_soc_codec *codec, unsigned int reg) +{ + unsigned int value = 0; + + rt5677_dsp_mode_i2c_read_addr(codec, 0x18020000 + reg * 2, &value); + + return value; +} + +/** + * rt5677_dsp_mode_i2c_update_bits - update register on DSP mode. + * @codec: audio codec + * @reg: register index. + * @mask: register mask + * @value: new value + * + * + * Returns 1 for change, 0 for no change, or negative error code. + */ +static int rt5677_dsp_mode_i2c_update_bits(struct snd_soc_codec *codec, + unsigned int reg, unsigned int mask, unsigned int value) +{ + unsigned int old, new; + int change, ret; + + ret = rt5677_dsp_mode_i2c_read(codec, reg); + if (ret < 0) { + dev_err(codec->dev, "Failed to read reg: %d\n", ret); + goto err; + } + + old = ret; + new = (old & ~mask) | (value & mask); + change = old != new; + if (change) { + ret = rt5677_dsp_mode_i2c_write(codec, reg, new); + if (ret < 0) { + dev_err(codec->dev, + "Failed to write reg: %d\n", ret); + goto err; + } + } + return change; + +err: + return ret; +} + +static int rt5677_set_dsp_vad(struct snd_soc_codec *codec, bool on) +{ + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + static bool activity; + int ret; + + if (on && !activity) { + activity = true; + + regcache_cache_only(rt5677->regmap, false); + regcache_cache_bypass(rt5677->regmap, true); + + regmap_update_bits(rt5677->regmap, RT5677_DIG_MISC, 0x1, 0x1); + regmap_update_bits(rt5677->regmap, + RT5677_PR_BASE + RT5677_BIAS_CUR4, 0x0f00, 0x0f00); + regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG1, + RT5677_LDO1_SEL_MASK, 0x0); + regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG2, + RT5677_PWR_LDO1, RT5677_PWR_LDO1); + regmap_write(rt5677->regmap, RT5677_GLB_CLK2, 0x0080); + regmap_write(rt5677->regmap, RT5677_PWR_DSP2, 0x07ff); + regmap_write(rt5677->regmap, RT5677_PWR_DSP1, 0x07ff); + + ret = request_firmware(&rt5677->fw1, RT5677_FIRMWARE1, + codec->dev); + if (ret == 0) { + rt5677_spi_burst_write(0x50000000, rt5677->fw1); + release_firmware(rt5677->fw1); + } + + ret = request_firmware(&rt5677->fw2, RT5677_FIRMWARE2, + codec->dev); + if (ret == 0) { + rt5677_spi_burst_write(0x60000000, rt5677->fw2); + release_firmware(rt5677->fw2); + } + + rt5677_dsp_mode_i2c_update_bits(codec, RT5677_PWR_DSP1, 0x1, + 0x0); + + regcache_cache_bypass(rt5677->regmap, false); + regcache_cache_only(rt5677->regmap, true); + } else if (!on && activity) { + activity = false; + + regcache_cache_only(rt5677->regmap, false); + regcache_cache_bypass(rt5677->regmap, true); + + rt5677_dsp_mode_i2c_update_bits(codec, RT5677_PWR_DSP1, 0x1, + 0x1); + rt5677_dsp_mode_i2c_write(codec, RT5677_PWR_DSP1, 0x0001); + + regmap_write(rt5677->regmap, RT5677_RESET, 0x10ec); + + regcache_cache_bypass(rt5677->regmap, false); + regcache_mark_dirty(rt5677->regmap); + regcache_sync(rt5677->regmap); + } + + return 0; +} + static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0); static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); @@ -556,6 +795,31 @@ static unsigned int bst_tlv[] = { 8, 8, TLV_DB_SCALE_ITEM(5200, 0, 0), }; +static int rt5677_dsp_vad_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = rt5677->dsp_vad_en; + + return 0; +} + +static int rt5677_dsp_vad_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + + rt5677->dsp_vad_en = !!ucontrol->value.integer.value[0]; + + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + rt5677_set_dsp_vad(codec, rt5677->dsp_vad_en); + + return 0; +} + static const struct snd_kcontrol_new rt5677_snd_controls[] = { /* OUTPUT Control */ SOC_SINGLE("OUT1 Playback Switch", RT5677_LOUT1, @@ -627,6 +891,9 @@ static const struct snd_kcontrol_new rt5677_snd_controls[] = { SOC_DOUBLE_TLV("Mono ADC Boost Volume", RT5677_ADC_BST_CTRL2, RT5677_MONO_ADC_L_BST_SFT, RT5677_MONO_ADC_R_BST_SFT, 3, 0, adc_bst_tlv), + + SOC_SINGLE_EXT("DSP VAD Switch", SND_SOC_NOPM, 0, 1, 0, + rt5677_dsp_vad_get, rt5677_dsp_vad_put), }; /** @@ -3181,6 +3448,8 @@ static int rt5677_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) { + rt5677_set_dsp_vad(codec, false); + regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG1, RT5677_LDO1_SEL_MASK | RT5677_LDO2_SEL_MASK, 0x0055); @@ -3214,6 +3483,9 @@ static int rt5677_set_bias_level(struct snd_soc_codec *codec, regmap_write(rt5677->regmap, RT5677_PWR_ANLG2, 0x0000); regmap_update_bits(rt5677->regmap, RT5677_PR_BASE + RT5677_BIAS_CUR4, 0x0f00, 0x0000); + + if (rt5677->dsp_vad_en) + rt5677_set_dsp_vad(codec, true); break; default: @@ -3371,6 +3643,8 @@ static int rt5677_probe(struct snd_soc_codec *codec) regmap_write(rt5677->regmap, RT5677_DIG_MISC, 0x0020); regmap_write(rt5677->regmap, RT5677_PWR_DSP2, 0x0c00); + mutex_init(&rt5677->dsp_cmd_lock); + return 0; } @@ -3390,8 +3664,11 @@ static int rt5677_suspend(struct snd_soc_codec *codec) { struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); - regcache_cache_only(rt5677->regmap, true); - regcache_mark_dirty(rt5677->regmap); + if (!rt5677->dsp_vad_en) { + regcache_cache_only(rt5677->regmap, true); + regcache_mark_dirty(rt5677->regmap); + } + if (gpio_is_valid(rt5677->pow_ldo2)) gpio_set_value_cansleep(rt5677->pow_ldo2, 0); @@ -3406,8 +3683,11 @@ static int rt5677_resume(struct snd_soc_codec *codec) gpio_set_value_cansleep(rt5677->pow_ldo2, 1); msleep(10); } - regcache_cache_only(rt5677->regmap, false); - regcache_sync(rt5677->regmap); + + if (!rt5677->dsp_vad_en) { + regcache_cache_only(rt5677->regmap, false); + regcache_sync(rt5677->regmap); + } return 0; } diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index d4eb6d5..a7ea27a 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -1506,6 +1506,9 @@ #define RT5677_GPIO5_FUNC_GPIO (0x0 << 9) #define RT5677_GPIO5_FUNC_DMIC (0x1 << 9) +#define RT5677_FIRMWARE1 "rt5677_dsp_fw1.bin" +#define RT5677_FIRMWARE2 "rt5677_dsp_fw2.bin" + /* System Clock Source */ enum { RT5677_SCLK_S_MCLK, @@ -1545,6 +1548,8 @@ struct rt5677_priv { struct snd_soc_codec *codec; struct rt5677_platform_data pdata; struct regmap *regmap; + const struct firmware *fw1, *fw2; + struct mutex dsp_cmd_lock; int sysclk; int sysclk_src; @@ -1558,6 +1563,7 @@ struct rt5677_priv { #ifdef CONFIG_GPIOLIB struct gpio_chip gpio_chip; #endif + bool dsp_vad_en; }; #endif /* __RT5677_H__ */ -- 1.8.1.1.439.g50a6b54

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[alsa-devel] davinci_soc_platform_register and davinci-pcm-audio question
by Angelo Dureghello 15 Oct '14

15 Oct '14

Dear all, i am referring to this patch: [PATCH V2 1/6] ASoC: davinci: davinci-pcm does not need to be a plaform_driver i am working on a board, based on am1808(da850) cpu and codec wolfson wm8758 (created based on wm8985.c) using mcbsp with a custom soc machine is actually broken, This moving from kernel 3.5.1 to 3.17. Alsa card is loaded correctly, but i can hear no sound: omap_rtc omap_rtc: setting system clock to 2014-10-15 15:46:57 UTC (1413388017) ALSA device list: #0: barix-ipam390 UBIFS: background thread "ubifs_bgt0_0" started, PID 1071 UBIFS: recovery needed In the patch, seems "davinci-pcm-audio" is no more a driver. And i should use "davinci_soc_platform_register" to have the pcm helper available. But i don't understand in where i have to call this function. Every help is appreciated. Reagrds angelo -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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[alsa-devel] davinci_soc_platform_register and davinci-pcm-audio question
by Angelo Dureghello 15 Oct '14

15 Oct '14

Dear all, i am referring to this patch: http://lists.infradead.org/pipermail/linux-arm-kernel/2012-August/116088.ht… My board, based on am1808(da850) cpu and codec wolfson wm8758 (created based on wm8985.c) using mcbsp with a custom soc machine is actually broken, This moving from kernel 3.5.1 to 3.17. Alsa card is loaded correctly, but i can hear no sound: omap_rtc omap_rtc: setting system clock to 2014-10-15 15:46:57 UTC (1413388017) ALSA device list: #0: barix-ipam390 UBIFS: background thread "ubifs_bgt0_0" started, PID 1071 UBIFS: recovery needed In the patch, seems "davinci-pcm-audio" is no more a driver. And i should use "davinci_soc_platform_register" to have the pcm helper available. But i don't understand in where i have to call this function. Every help is appreciated. Reagrds angelo -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

1 0

[alsa-devel] [PATCH 00/12] dmaengine: remove users of device_control
by Vinod Koul 15 Oct '14

15 Oct '14

The recent discussion [1] on the API have resulted in moving away from device_control ioctl method to proper channel APIs. There are still few users on the device_control which should use the wrappers existing rather than access device_control. This will aid us in deprecating and removing device_control, possibly after the merge window. These can be merged thru respective subsystem tree or dmaengine tree. Either way please just let me know. Feng's kbuild has tested these as well [2] [1]: http://www.spinics.net/lists/dmaengine/msg02212.html [2]: http://git.infradead.org/users/vkoul/slave-dma.git/shortlog/refs/heads/topi… Vinod Koul (12): pata_arasan_cf: use dmaengine_terminate_all() API dmaengine: coh901318: use dmaengine_terminate_all() API [media] V4L2: mx3_camer: use dmaengine_pause() API mtd: fsmc_nand: use dmaengine_terminate_all() API mtd: sh_flctl: use dmaengine_terminate_all() API net: ks8842: use dmaengine_terminate_all() API spi/atmel: use dmaengine_terminate_all() API spi/spi-dw-mid.c: use dmaengine_slave_config() API serial: sh-sci: use dmaengine_terminate_all() API usb: musb: ux500_dma: use dmaengine_xxx() APIs ASoC: txx9: use dmaengine_terminate_all() API video: mx3fb: use dmaengine_terminate_all() API drivers/ata/pata_arasan_cf.c | 5 ++--- drivers/dma/coh901318.c | 2 +- drivers/media/platform/soc_camera/mx3_camera.c | 6 ++---- drivers/mtd/nand/fsmc_nand.c | 2 +- drivers/mtd/nand/sh_flctl.c | 2 +- drivers/net/ethernet/micrel/ks8842.c | 6 ++---- drivers/spi/spi-atmel.c | 6 ++---- drivers/spi/spi-dw-mid.c | 6 ++---- drivers/tty/serial/sh-sci.c | 2 +- drivers/usb/musb/ux500_dma.c | 7 ++----- drivers/video/fbdev/mx3fb.c | 3 +-- sound/soc/txx9/txx9aclc.c | 7 +++---- 12 files changed, 20 insertions(+), 34 deletions(-)

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[alsa-devel] [PATCH v9 0/6] Add mrfld DSP topology and widgets
by Vinod Koul 15 Oct '14

15 Oct '14

This patch series adds DPCM and DAPM widgets to handle DSP topology in the SST platform driver for merrifield. changes in v9: - fix some corner conditions and error checks changes in v8: - fix the locking and comments blobs - fix the send_ssp_slot to send IPC on none - fix controls to have "Volume" and "Switch" wherever neccessary - add mute for CPU dai in asoc core patch Ramesh Babu (1): ASoC: core: Call mute for cpu dais as well Vinod Koul (5): ASoC: Intel: mrfld: add the gain controls ASoC: Intel: mrfld: add DSP core controls ASoC: Intel: mrfld: add the DSP DAPM widgets ASoC: Intel: mfld-pcm: add FE and BE ops ASoC: Intel: mrfld: add the DSP mixers sound/soc/intel/sst-atom-controls.c | 1208 ++++++++++++++++++++++++++++++- sound/soc/intel/sst-atom-controls.h | 428 +++++++++++ sound/soc/intel/sst-mfld-platform-pcm.c | 153 ++++- sound/soc/intel/sst-mfld-platform.h | 4 + sound/soc/soc-pcm.c | 3 + 5 files changed, 1766 insertions(+), 30 deletions(-)

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[alsa-devel] [RFC] provide a simple mechanism to start playback at a given time
by Nick Stoughton 15 Oct '14

15 Oct '14

As discussed at today's meeting, please find 2 patches (one to the 3.17 kernel, the other to alsa-lib) following this that implement a simple interface to trigger playback at a given point in the future. The API provides an absolute time, measured against a given clock, at which to trigger DMA. If the point in the future appears to be less than 100ms away, we fall back to the standard snd_pcm_start() behavior. This is because this interface is designed to support networked devices playing back in sync, and network latencies are likely to be over 100ms. This value should probably be tunable ... Also, we simply set up a timer and commence DMA when it expires. If the devices to be synchronized have identical hardware, then there is no problem. A future enhancement is to estimate the latency between the DMA trigger and the codec driver (FIFO sizes etc) so that we can take that into account. Any comments gratefully accepted! *______________________________* *Nick Stoughton* *Aether Things Inc * *San Francisco* +1 (510) 388 1413

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Re: [alsa-devel] [PATCH v6 09/18] ASoC: omap-hdmi-audio: Add platfrom device for OMAP HDMI audio support (Jyri Sarha)
by Andrej Falout 15 Oct '14

15 Oct '14

Hi Jyri, Is the audio support for OMAP HDMI supposed to support DSD audio? Eg, on page 6 of http://elinux.org/images/8/8e/HDMI_ELC_mythripk.pdf "Super Audio CD (DSD) support" starting from HDMI version 1.2 Thanks, Andrej Falout

2 3

[alsa-devel] [PATCH v6 00/18] Rework OMAP4+ HDMI audio support
by Jyri Sarha 15 Oct '14

15 Oct '14

The patches are based on: git://git.kernel.org/pub/scm/linux/kernel/git/tomba/linux.git for-next The base, the patches, and couple of additional not-to-be-merged omap2plus_defconfig patches can be found here: https://github.com/jsarha/linux.git It would make the most sense to get these in trough fbdev tree. So it would be nice to get acked-bys (if the patches are Ok) for ASoC side changes from appropriate maintainers. Changes since v5: Still lot of changes, most of them were suggested by Tomi Valkeinen. The most significant changes: - Fix HDMI unplugging crashes when playing audio etc. - Don't use simple-card but implement bare minimum ASoC card under pdev - Register cpu-dai under OMAPDSS HDMI device and get dai drvdata trough card drvdata via snd_pcm_substream - Some more cleaning up Changes since v4: - Turn ASoC parts into a platform device driver - Some cleaning up - reorder the patches to keep bisect working Changes since v3: - Move ASoC parts into library module under sound/soc/omap - Come up with API for the library - Some cleaning up The patch set fixes OMAP4+ HDMI audio. The structure of the implementation looks a bit different than before. Instead of creating a driver specific API for a separate ASoC component driver to connect to, these patches implement a platform device driver under sound/soc/omap for ASoC side implementation. This driver should only be registered from OMAPDSS HDMI code. The ASoC side driver implements cpu-dai component using the callbacks provided by OMAPDSS and registers the component under DSS HDMI device. Omap-pcm is registered for platform component, dummy hdmi-audio-codec is registered for codec component, and ASoC driver implements also a bare minimum card driver and registeres it under itself. Big part of the HDMI audio code is still unchanged and there is a need for a cleanup there. Also there is still probably something wrong with speaker mapping of multi-channel streams. I will get back to cleaning up these issues later. Best regards, Jyri OMAPDSS: hdmi.h: Add HDMI_AUDIO_LAYOUT_6CH enum value OMAPDSS: hdmi: Remove most of OMAP[45]_DSS_HDMI_AUDIO ifdefs OMAPDSS: hdmi4_core: Remove unused hdmi4_audio_get_dma_port() OMAPDSS: hdmi_wp: Add function for getting audio dma address OMAPDSS: hdmi: Make hdmi structure public OMAPDSS: hdmi: Add pdev pointer for audio_pdev in HDMI DRV data ASoC: codecs/hdmi: Mark the maximum significant bits to HDMI codec ASoC: codecs/hdmi: HDMI codec doesn't benefit from pmdown delay ASoC: omap-hdmi-audio: Add platfrom device for OMAP HDMI audio support OMAPDSS: Kconfig: Remove HDMI audio booleans from Kconfig OMAPDSS: hdmi: Make hdmi_mode_has_audio() more user friedly OMAPDSS: hdmi.h: Add members to hdmi drvdata for audio implementation OMAPDSS: hdmi4: Remove callbacks for the old ASoC DAI driver OMAPDSS: hdmi4: Register ASoC platfrom device for omap hdmi audio OMAPDSS: hdmi5: Remove callbacks for the old ASoC DAI driver OMAPDSS: hdmi5: Register ASoC platfrom device for omap hdmi audio ASoC: omap: Remove obsolete HDMI audio code and Kconfig options OMAPDSS: Remove all references to obsolete HDMI audio callbacks .../fbdev/omap2/displays-new/connector-hdmi.c | 99 ----- .../fbdev/omap2/displays-new/encoder-tpd12s015.c | 56 --- drivers/video/fbdev/omap2/dss/Kconfig | 7 - drivers/video/fbdev/omap2/dss/hdmi.h | 37 +- drivers/video/fbdev/omap2/dss/hdmi4.c | 268 ++++++------- drivers/video/fbdev/omap2/dss/hdmi4_core.c | 14 - drivers/video/fbdev/omap2/dss/hdmi4_core.h | 4 - drivers/video/fbdev/omap2/dss/hdmi5.c | 257 ++++++------ drivers/video/fbdev/omap2/dss/hdmi5_core.c | 6 - drivers/video/fbdev/omap2/dss/hdmi5_core.h | 2 - drivers/video/fbdev/omap2/dss/hdmi_common.c | 2 - drivers/video/fbdev/omap2/dss/hdmi_wp.c | 8 +- include/sound/omap-hdmi-audio.h | 43 ++ include/video/omapdss.h | 31 -- sound/soc/codecs/hdmi.c | 2 + sound/soc/omap/Kconfig | 21 +- sound/soc/omap/Makefile | 6 +- sound/soc/omap/omap-hdmi-audio.c | 420 ++++++++++++++++++++ sound/soc/omap/omap-hdmi-card.c | 87 ---- sound/soc/omap/omap-hdmi.c | 364 ----------------- sound/soc/omap/omap-hdmi.h | 38 -- 21 files changed, 772 insertions(+), 1000 deletions(-) create mode 100644 include/sound/omap-hdmi-audio.h create mode 100644 sound/soc/omap/omap-hdmi-audio.c delete mode 100644 sound/soc/omap/omap-hdmi-card.c delete mode 100644 sound/soc/omap/omap-hdmi.c delete mode 100644 sound/soc/omap/omap-hdmi.h -- 1.7.9.5

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[alsa-devel] [PATCH v3 1/2] ALSA: au88x0: added reference of vortex_t
by Sudip Mukherjee 15 Oct '14

15 Oct '14

added a pointer of the vortex in the following functions : vortex_alsafmt_aspfmt vortex_Vort3D_InitializeSource a3dsrc_ZeroStateA3D so that we can have a reference of the vortex in the function. this reference of the vortex will actually be used in a later patch to convert the pr_* macro to dev_*. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- no change from v1. no change from v2. sound/pci/au88x0/au88x0.h | 4 ++-- sound/pci/au88x0/au88x0_a3d.c | 6 +++--- sound/pci/au88x0/au88x0_core.c | 5 +++-- sound/pci/au88x0/au88x0_pcm.c | 2 +- 4 files changed, 9 insertions(+), 8 deletions(-) diff --git a/sound/pci/au88x0/au88x0.h b/sound/pci/au88x0/au88x0.h index 466a5c8..3a8fefe 100644 --- a/sound/pci/au88x0/au88x0.h +++ b/sound/pci/au88x0/au88x0.h @@ -243,7 +243,7 @@ static int vortex_core_init(vortex_t * card); static int vortex_core_shutdown(vortex_t * card); static void vortex_enable_int(vortex_t * card); static irqreturn_t vortex_interrupt(int irq, void *dev_id); -static int vortex_alsafmt_aspfmt(int alsafmt); +static int vortex_alsafmt_aspfmt(int alsafmt, vortex_t *v); /* Connection stuff. */ static void vortex_connect_default(vortex_t * vortex, int en); @@ -278,7 +278,7 @@ static void vortex_mix_setvolumebyte(vortex_t * vortex, unsigned char mix, static void vortex_Vort3D_enable(vortex_t * v); static void vortex_Vort3D_disable(vortex_t * v); static void vortex_Vort3D_connect(vortex_t * vortex, int en); -static void vortex_Vort3D_InitializeSource(a3dsrc_t * a, int en); +static void vortex_Vort3D_InitializeSource(a3dsrc_t *a, int en, vortex_t *v); #endif /* Driver stuff. */ diff --git a/sound/pci/au88x0/au88x0_a3d.c b/sound/pci/au88x0/au88x0_a3d.c index 30f760e..bc9cda3a 100644 --- a/sound/pci/au88x0/au88x0_a3d.c +++ b/sound/pci/au88x0/au88x0_a3d.c @@ -484,7 +484,7 @@ static void a3dsrc_ZeroState(a3dsrc_t * a) } /* Reset entire A3D engine */ -static void a3dsrc_ZeroStateA3D(a3dsrc_t * a) +static void a3dsrc_ZeroStateA3D(a3dsrc_t *a, vortex_t *v) { int i, var, var2; @@ -601,7 +601,7 @@ static void vortex_Vort3D_enable(vortex_t *v) Vort3DRend_Initialize(v, XT_HEADPHONE); for (i = 0; i < NR_A3D; i++) { vortex_A3dSourceHw_Initialize(v, i % 4, i >> 2); - a3dsrc_ZeroStateA3D(&(v->a3d[0])); + a3dsrc_ZeroStateA3D(&v->a3d[0], v); } /* Register ALSA controls */ vortex_a3d_register_controls(v); @@ -676,7 +676,7 @@ static void vortex_Vort3D_connect(vortex_t * v, int en) } /* Initialize one single A3D source. */ -static void vortex_Vort3D_InitializeSource(a3dsrc_t * a, int en) +static void vortex_Vort3D_InitializeSource(a3dsrc_t *a, int en, vortex_t *v) { if (a->vortex == NULL) { pr_warn diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c index 72e8128..00e2096 100644 --- a/sound/pci/au88x0/au88x0_core.c +++ b/sound/pci/au88x0/au88x0_core.c @@ -2177,7 +2177,8 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir, return -EBUSY; } /* (De)Initialize A3D hardware source. */ - vortex_Vort3D_InitializeSource(&(vortex->a3d[a3d]), en); + vortex_Vort3D_InitializeSource(&vortex->a3d[a3d], en, + vortex); } /* Make SPDIF out exclusive to "spdif" device when in use. */ if ((stream->type == VORTEX_PCM_SPDIF) && (en)) { @@ -2765,7 +2766,7 @@ static int vortex_core_shutdown(vortex_t * vortex) /* Alsa support. */ -static int vortex_alsafmt_aspfmt(int alsafmt) +static int vortex_alsafmt_aspfmt(int alsafmt, vortex_t *v) { int fmt; diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c index 5adc6b9..bdde182 100644 --- a/sound/pci/au88x0/au88x0_pcm.c +++ b/sound/pci/au88x0/au88x0_pcm.c @@ -332,7 +332,7 @@ static int snd_vortex_pcm_prepare(struct snd_pcm_substream *substream) dir = 1; else dir = 0; - fmt = vortex_alsafmt_aspfmt(runtime->format); + fmt = vortex_alsafmt_aspfmt(runtime->format, chip); spin_lock_irq(&chip->lock); if (VORTEX_PCM_TYPE(substream->pcm) != VORTEX_PCM_WT) { vortex_adbdma_setmode(chip, dma, 1, dir, fmt, -- 1.8.1.2

2 2

[alsa-devel] [PATCH 3/4] hda_intel: Add Device IDs for Intel Sunrise Point PCH
by James Ralston 15 Oct '14

15 Oct '14

This patch adds the HD Audio Device IDs for the Intel Sunrise Point PCH. Signed-off-by: James Ralston <james.d.ralston(a)intel.com> --- sound/pci/hda/hda_intel.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index aa302fb..8856604 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -218,6 +218,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6}," "{Intel, LPT}," "{Intel, LPT_LP}," "{Intel, WPT_LP}," + "{Intel, SPT}," "{Intel, HPT}," "{Intel, PBG}," "{Intel, SCH}," @@ -1984,6 +1985,9 @@ static const struct pci_device_id azx_ids[] = { /* 9 Series */ { PCI_DEVICE(0x8086, 0x8ca0), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, + /* Sunrise Point */ + { PCI_DEVICE(0x8086, 0xa170), + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, /* Wellsburg */ { PCI_DEVICE(0x8086, 0x8d20), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, -- 1.8.3.1

2 1

[alsa-devel] [PATCH] ASoC: sgtl500.txt: Document the required supplies
by Fabio Estevam 15 Oct '14

15 Oct '14

sgtl5000 has two required supplies: VDDA and VDDIO and one optional supply: VDDD, so document this properly. Not passing VDDA and VDDIO prevents the driver to probe successfully. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Documentation/devicetree/bindings/sound/sgtl5000.txt | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/sgtl5000.txt b/Documentation/devicetree/bindings/sound/sgtl5000.txt index 955df60..d556dcb 100644 --- a/Documentation/devicetree/bindings/sound/sgtl5000.txt +++ b/Documentation/devicetree/bindings/sound/sgtl5000.txt @@ -7,10 +7,20 @@ Required properties: - clocks : the clock provider of SYS_MCLK +- VDDA-supply : the regulator provider of VDDA + +- VDDIO-supply: the regulator provider of VDDIO + +Optional properties: + +- VDDD-supply : the regulator provider of VDDD + Example: codec: sgtl5000@0a { compatible = "fsl,sgtl5000"; reg = <0x0a>; clocks = <&clks 150>; + VDDA-supply = <&reg_3p3v>; + VDDIO-supply = <&reg_3p3v>; }; -- 1.9.1

2 1

[alsa-devel] disconnecting bluetooth device
by Thomas Green 14 Oct '14

14 Oct '14

I am using gstreamer and bluez-alsa to connect to a Bluetooth headset. I successfully connect to the device and can play audio from it. The issue comes when the device disconnects. Gstreamer lists these alsa errors: 0:00:25.312585875 24605 0x42430800 WARN alsa audio/pcm_bluetooth.c:247:playback_hw_thread: alsalib error: poll fd 1 revents 24 0:00:25.313233896 24605 0xd04f0 WARN alsa audio/pcm_bluetooth.c:835:bluetooth_playback_poll_revents: alsalib error: read error: Resource temporarily unavailable (11): Resource temporarily unavailable (lots and lots of these until I terminate the application) I've looked at the code for pcm_bluetooth.c, headset.c and pcm.c, and from what I can tell it looks like what is happening is that the playback_hw_thread detects that the device has disconnected and gets a message back with POLLERR | POLLHUP. In turn, when the bluetooth_playback_poll_revents is called, it can't read any further input. But the io->state should be getting set to SND_PCM_STATE_DISCONNECTED. From what I can see, I am guessing that what is happening is that above that is the bluetooth_playback_poll_revents keeps getting called without the io->state being checked to make sure whether or not the state is disconnected. I haven't dug too deeply into the call stack higher than that, so, I'm not sure. Is this what is happening, or is there something that I can do myself to mitigate this? The default alsa installation is 1.0.25, but I've also built libasound 1.0.28, and I still get the same behavior. Thanks... Tom

1 0

[alsa-devel] [PATCH 1/2] ALSA: pcm: Fix referred substream in snd_pcm_action_group() unlock loop
by Takashi Iwai 14 Oct '14

14 Oct '14

In the unlock loop of snd_pcm_action_group(), the object "s" is used as the check of nonatomic PCM, but it should be rather "s1", which is the iterator of the loop. This supposedly causes a kernel panic when the substreams in operatino are linked. Fixes: 257f8cce5d40 ('ALSA: pcm: Allow nonatomic trigger operations') Reported-and-tested-by: Arthur Marsh <arthur.marsh(a)internode.on.net> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_native.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 85fe1a216225..bfe1cf6b492f 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -818,7 +818,7 @@ static int snd_pcm_action_group(struct action_ops *ops, /* unlock streams */ snd_pcm_group_for_each_entry(s1, substream) { if (s1 != substream) { - if (s->pcm->nonatomic) + if (s1->pcm->nonatomic) mutex_unlock(&s1->self_group.mutex); else spin_unlock(&s1->self_group.lock); -- 2.1.2

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[alsa-devel] [PATCH] ALSA: provide a simple mechanism to start playback at a given time
by Nick Stoughton 14 Oct '14

14 Oct '14

Given an absolute time based on the clock used by the substream setup a high resolution timer to cause playback to be triggered at that time. Signed-off-by: Nick Stoughton <nstoughton(a)aether.com> --- include/uapi/sound/asound.h | 6 ++ sound/core/pcm_native.c | 142 ++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 148 insertions(+) diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 6ee5867..92d5856 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -462,6 +462,11 @@ struct snd_xfern { snd_pcm_uframes_t frames; }; +struct snd_clock_time { + clockid_t clock; + struct timespec time; +}; + enum { SNDRV_PCM_TSTAMP_TYPE_GETTIMEOFDAY = 0, /* gettimeofday equivalent */ SNDRV_PCM_TSTAMP_TYPE_MONOTONIC, /* posix_clock_monotonic equivalent */ @@ -547,6 +552,7 @@ enum { #define SNDRV_PCM_IOCTL_READN_FRAMES _IOR('A', 0x53, struct snd_xfern) #define SNDRV_PCM_IOCTL_LINK _IOW('A', 0x60, int) #define SNDRV_PCM_IOCTL_UNLINK _IO('A', 0x61) +#define SNDRV_PCM_IOCTL_START_AT _IOW('A', 0x62, struct snd_clock_time) /***************************************************************************** * * diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 85fe1a2..f16ac3c 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -27,6 +27,7 @@ #include <linux/pm_qos.h> #include <linux/aio.h> #include <linux/dma-mapping.h> +#include <linux/time.h> #include <sound/core.h> #include <sound/control.h> #include <sound/info.h> @@ -38,6 +39,9 @@ #if defined(CONFIG_MIPS) && defined(CONFIG_DMA_NONCOHERENT) #include <dma-coherence.h> #endif +#if defined(CONFIG_HIGH_RES_TIMERS) +#include <linux/hrtimer.h> +#endif /* * Compatibility @@ -1016,6 +1020,142 @@ static struct action_ops snd_pcm_action_start = { .post_action = snd_pcm_post_start }; +#ifdef CONFIG_HIGH_RES_TIMERS +/** + * ktime_get_raw_offset - get the offset in ktime format between + * the monotonic_raw clock and the monotonic clock + */ +static ktime_t ktime_get_raw_offset(void) +{ + struct timespec rawtime = {0,0}; + struct timespec now = {0,0}; + struct timespec delta; + + ktime_get_ts(&now); + getrawmonotonic(&rawtime); + delta = timespec_sub(now, rawtime); + return timespec_to_ktime(delta); +} + +/* + * hrtimer interface + */ + +struct hrtimer_pcm { + struct hrtimer timer; + struct snd_pcm_substream *substream; +}; + +/* + * called from a hard irq context - no need for locks. + */ +enum hrtimer_restart snd_pcm_do_start_time(struct hrtimer *timer) +{ + struct hrtimer_pcm *pcm_timer; + struct snd_pcm_substream *substream; + struct snd_pcm_runtime *runtime; + int ret; + + pcm_timer = container_of(timer, struct hrtimer_pcm, timer); + substream = pcm_timer->substream; + runtime = substream->runtime; + + /* Check that the stream is still open and in the right state. */ + if (runtime && runtime->status->state == SNDRV_PCM_STATE_PREPARED) { + + ret = snd_pcm_do_start(substream, SNDRV_PCM_STATE_RUNNING); + if (ret == 0) { + snd_pcm_post_start(substream, SNDRV_PCM_STATE_RUNNING); + } + } + kfree(pcm_timer); + + return HRTIMER_NORESTART; +} + +int snd_pcm_start_at(struct snd_pcm_substream *substream, + struct timespec __user *_start_time) +{ + struct hrtimer_pcm *pcm_timer; + struct timespec start_time; + int clock; + ktime_t start_kt; + ktime_t raw_offset; + int ret; + + if (copy_from_user(&start_time, _start_time, sizeof(start_time))) + return -EFAULT; + + if (!timespec_valid(&start_time)) + return -EINVAL; + + start_kt = timespec_to_ktime(start_time); + switch (substream->runtime->tstamp_type) { + case SNDRV_PCM_TSTAMP_TYPE_GETTIMEOFDAY: + clock = CLOCK_REALTIME; + break; + case SNDRV_PCM_TSTAMP_TYPE_MONOTONIC: + clock = CLOCK_MONOTONIC; + break; + case SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW: + clock = CLOCK_MONOTONIC_RAW; + break; + default: + return -EINVAL; + } + switch(clock) { + case CLOCK_MONOTONIC_RAW: + /* + * convert to clock monotonic: + * the raw clock and the monotonic clock run at different + * rates. Currently there is no timer based on the + * monotonic_raw clock but for our purposes, we assume that + * the divergence between the raw and the monotonic clock + * over the life of the timer will be minimal. + * Typically the difference is less than 20PPM. + * TODO: implement high res timer based on CLOCK_MONOTONIC_RAW + */ + raw_offset = ktime_get_raw_offset(); + start_kt = ktime_add(raw_offset, start_kt); + clock = CLOCK_MONOTONIC; + /* and fall through ... */ + case CLOCK_MONOTONIC: + if (ktime_compare(start_kt, ktime_get()) <= 0) + return -ETIME; + break; + case CLOCK_REALTIME: + if (ktime_compare(start_kt, + ktime_get_with_offset(TK_OFFS_REAL)) <= 0) + return -EINVAL; + } + + /* Fail if not playback substream */ + if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + return -EBADFD; + + ret = snd_pcm_pre_start(substream, SNDRV_PCM_STATE_PREPARED); + if (ret != 0) + return ret; + + pcm_timer = kmalloc(sizeof(*pcm_timer), GFP_KERNEL); + hrtimer_init(&pcm_timer->timer, clock, HRTIMER_MODE_ABS); + + pcm_timer->timer.function = snd_pcm_do_start_time; + pcm_timer->substream = substream; + + hrtimer_start(&pcm_timer->timer, start_kt, HRTIMER_MODE_ABS); + + return 0; +} +#else +/* without high res time, interface is identical to snd_pcm_start() */ +int snd_pcm_start_at(struct snd_pcm_substream *substream, + struct snd_clock_time __user *_start_time) +{ + return -ENOSYS; +} +#endif + /** * snd_pcm_start - start all linked streams * @substream: the PCM substream instance @@ -2698,6 +2838,8 @@ static int snd_pcm_common_ioctl1(struct file *file, return snd_pcm_reset(substream); case SNDRV_PCM_IOCTL_START: return snd_pcm_action_lock_irq(&snd_pcm_action_start, substream, SNDRV_PCM_STATE_RUNNING); + case SNDRV_PCM_IOCTL_START_AT: + return snd_pcm_start_at(substream, arg); case SNDRV_PCM_IOCTL_LINK: return snd_pcm_link(substream, (int)(unsigned long) arg); case SNDRV_PCM_IOCTL_UNLINK: -- 1.9.3

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[alsa-devel] [PATCH v2 0/2] sgtl5000: Add controls for MicBias in DT
by Jean-Michel Hautbois 14 Oct '14

14 Oct '14

This serie adds two simple controls on resistor and voltage for MicBias. Jean-Michel Hautbois (2): audio: sgtl5000: Add MicBias resistor support in DT audio: sgtl5000: Add MicBias voltage support .../devicetree/bindings/sound/sgtl5000.txt | 13 +++++ sound/soc/codecs/sgtl5000.c | 68 ++++++++++++++++++++-- 2 files changed, 77 insertions(+), 4 deletions(-) -- 2.1.1

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[alsa-devel] lock-up when loading desktop
by Arthur Marsh 14 Oct '14

14 Oct '14

I have been experiencing a lock-up situation on a dual-core P4 machine since some time after kernel 3.17.0 was released. After the lock-up, it takes a couple of reboots into a known good kernel (3.17.0) to successfully boot as the corruption seems to hit the hard disks. After doing a git-bisect I received the following result: git bisect bad 257f8cce5d40b811d229ed71602882baa0012808 is the first bad commit commit 257f8cce5d40b811d229ed71602882baa0012808 Author: Takashi Iwai <tiwai(a)suse.de> Date: Fri Aug 29 15:32:29 2014 +0200 ALSA: pcm: Allow nonatomic trigger operations Currently, many PCM operations are performed in a critical section protected by spinlock, typically the trigger and pointer callbacks are assumed to be atomic. This is basically because some trigger action (e.g. PCM stop after drain or xrun) is done in the interrupt handler. If a driver runs in a threaded irq, however, this doesn't have to be atomic. And many devices want to handle trigger in a non-atomic context due to lengthy communications. This patch tries all PCM calls operational in non-atomic context. What it does is very simple: replaces the substream spinlock with the corresponding substream mutex when pcm->nonatomic flag is set. The driver that wants to use the non-atomic PCM ops just needs to set the flag and keep the rest as is. (Of course, it must not handle any PCM ops in irq context.) Note that the code doesn't check whether it's atomic-safe or not, but trust in 100% that the driver sets pcm->nonatomic correctly. One possible problem is the case where linked PCM substreams have inconsistent nonatomic states. For avoiding this, snd_pcm_link() returns an error if one tries to link an inconsistent PCM substream. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> :040000 040000 e395bf17236b9d109745444ae818b2ecdc21f206 e002045c29bc96fe0a99c81db9c905db04e87e03 M include :040000 040000 44044ea9f3c2aacbd488524c060554256c2b2ceb 77d1a1e452b9321876f9e1a8f6926f11814a9cd9 M sound git bisect log git bisect start # bad: [ca321885b0511a85e2d1cd40caafedbeb18f4af6] Merge git://git.kernel.org/pub/scm/linux/kernel/git/davem/net git bisect bad ca321885b0511a85e2d1cd40caafedbeb18f4af6 # good: [bfe01a5ba2490f299e1d2d5508cbbbadd897bbe9] Linux 3.17 git bisect good bfe01a5ba2490f299e1d2d5508cbbbadd897bbe9 # good: [f86dc4b04dd5292cae3708c16ca6e46dbb5c95fa] Merge tag 'defconfig-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/arm/arm-soc git bisect good f86dc4b04dd5292cae3708c16ca6e46dbb5c95fa # good: [5e5f6dc10546f5c03bc572e3ba3089af30c66e2d] arm64: mm: enable HAVE_RCU_TABLE_FREE logic git bisect good 5e5f6dc10546f5c03bc572e3ba3089af30c66e2d # good: [4d9708ea5e5a45973df7cf965805fdfb185dd5bf] Merge tag 'media/v3.18-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-media git bisect good 4d9708ea5e5a45973df7cf965805fdfb185dd5bf # bad: [e98d6e7f7625ed60c7bc1d39aeb2375ed3918fd5] Merge tag 'devicetree-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/glikely/linux git bisect bad e98d6e7f7625ed60c7bc1d39aeb2375ed3918fd5 # good: [bdf20b4291eaa3b327398b8dd330065ad8e6d3ce] Merge remote-tracking branches 'asoc/fix/88pm860x', 'asoc/fix/fsl', 'asoc/fix/imx', 'asoc/fix/mc13783', 'asoc/fix/rockchip' and 'asoc/fix/simple' into asoc-linus git bisect good bdf20b4291eaa3b327398b8dd330065ad8e6d3ce # good: [3db3525196a992da628fb210776d73ec4bb59460] mmc: sdhci-acpi: Get UID directly from acpi_device git bisect good 3db3525196a992da628fb210776d73ec4bb59460 # bad: [3d0fdc86e4b500dfcfbf2f68039d2d6853536c2e] ALSA: ctxfi: added reference of snd_card git bisect bad 3d0fdc86e4b500dfcfbf2f68039d2d6853536c2e # bad: [7fd4394dfe1db02ba904dfa1048f718cbca822d1] Merge branch 'topic/pcm-nonatomic' into for-next git bisect bad 7fd4394dfe1db02ba904dfa1048f718cbca822d1 # good: [c77900e63abd9e2bdf385ba846a22858a0ed50a7] ALSA: hda/realtek - move DELL2_MIC_NO_PRESENCE quirk for alc255 git bisect good c77900e63abd9e2bdf385ba846a22858a0ed50a7 # good: [d89c6c0c91af0344b52dd21ca48dd29821fee677] ALSA: hda - Add TLV_DB_SCALE_MUTE bit for relevant controls git bisect good d89c6c0c91af0344b52dd21ca48dd29821fee677 # good: [dd38dc1a9bf780b619ab93b3d7a5e90ebad441f5] ALSA: virtuoso: add one more headphone impedance setting git bisect good dd38dc1a9bf780b619ab93b3d7a5e90ebad441f5 # bad: [7af142f752116e86adbe2073f2922d8265a77709] ALSA: pcm: Uninline snd_pcm_stream_lock() and _unlock() git bisect bad 7af142f752116e86adbe2073f2922d8265a77709 # bad: [257f8cce5d40b811d229ed71602882baa0012808] ALSA: pcm: Allow nonatomic trigger operations git bisect bad 257f8cce5d40b811d229ed71602882baa0012808 # first bad commit: [257f8cce5d40b811d229ed71602882baa0012808] ALSA: pcm: Allow nonatomic trigger operations The problem still exists with the Linus git head as of earlier today, but only seems to get triggered when loading the desktop (which is actually an ancient KDE 3.51 with a library recompiled to work with newer kernels). The soundcard on the machine is: 00:0a.0 Multimedia audio controller: Creative Labs SB Audigy (rev 04) Subsystem: Creative Labs SB Audigy 2 ZS (SB0350) Flags: bus master, medium devsel, latency 32, IRQ 18 I/O ports at 8400 [size=64] Capabilities: <access denied> Kernel driver in use: snd_emu10k1 00:0a.1 Input device controller: Creative Labs SB Audigy Game Port (rev 04) Subsystem: Creative Labs SB Audigy Game Port Flags: bus master, medium devsel, latency 32 I/O ports at 8000 [size=8] Capabilities: <access denied> Kernel driver in use: Emu10k1_gameport 00:0a.2 FireWire (IEEE 1394): Creative Labs SB Audigy FireWire Port (rev 04) (prog-if 10 [OHCI]) Subsystem: Creative Labs SB Audigy FireWire Port Flags: bus master, medium devsel, latency 32, IRQ 19 Memory at bc800000 (32-bit, non-prefetchable) [size=2K] Memory at bc000000 (32-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: firewire_ohci I have not looked at the "first bad patch", only built against it. I am happy to supply further build and machine details, and run extra tests to help identify the problem.

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[alsa-devel] [PATCH 0/2] sgtl5000: Add controls for MicBias in DT
by Jean-Michel Hautbois 14 Oct '14

14 Oct '14

This serie adds two simple controls on resistor and voltage for MicBias. Jean-Michel Hautbois (2): audio: sgtl5000: Add MicBias resistor support in DT audio: sgtl5000: Add MicBias voltage support .../devicetree/bindings/sound/sgtl5000.txt | 13 +++++ sound/soc/codecs/sgtl5000.c | 68 ++++++++++++++++++++-- 2 files changed, 77 insertions(+), 4 deletions(-) -- 2.1.1

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[alsa-devel] Max volume is too low compared to Windows
by Mateus Rodrigues Costa 13 Oct '14

13 Oct '14

I have been getting this for a while on this laptop: the 100% volume on Linux is too low compared to what I can get on Windows. This is a Dell Inspiron 14R (5420, mid 2012) with a Intel HD Audio card. alsa-info.sh output: upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.4.63 !!################################ !!Script ran on: Mon Oct 13 17:29:37 UTC 2014 !!Linux Distribution !!------------------ Arch Linux \r (\l) NAME="Arch Linux" ID=arch PRETTY_NAME="Arch Linux" HOME_URL="https://www.archlinux.org/" SUPPORT_URL=" https://bbs.archlinux.org/" BUG_REPORT_URL="https://bugs.archlinux.org/" !!DMI Information !!--------------- Manufacturer: Dell Inc. Product Name: Inspiron 5420 Product Version: Firmware Version: A19 !!Kernel Information !!------------------ Kernel release: 3.16.4-1-ARCH Operating System: GNU/Linux Architecture: x86_64 Processor: unknown SMP Enabled: Yes !!ALSA Version !!------------ Driver version: k3.16.4-1-ARCH Library version: 1.0.28 Utilities version: 1.0.28 !!Loaded ALSA modules !!------------------- snd_hda_intel !!Sound Servers on this system !!---------------------------- Pulseaudio: Installed - Yes (/usr/bin/pulseaudio) Running - Yes Jack: Installed - Yes (/usr/bin/jackd) Running - No !!Soundcards recognised by ALSA !!----------------------------- 0 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xf1a10000 irq 45 !!PCI Soundcards installed in the system !!-------------------------------------- 00:1b.0 Audio device: Intel Corporation 7 Series/C210 Series Chipset Family High Definition Audio Controller (rev 04) !!Advanced information - PCI Vendor/Device/Subsystem ID's !!------------------------------------------------------- 00:1b.0 0403: 8086:1e20 (rev 04) Subsystem: 1028:055e !!Loaded sound module options !!--------------------------- !!Module: snd_hda_intel align_buffer_size : -1 bdl_pos_adj : 1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 beep_mode : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_msi : -1 id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 jackpoll_ms : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 model : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) patch : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) position_fix : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 power_save : 0 power_save_controller : N probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 probe_only : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 single_cmd : N snoop : Y !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: Conexant CX20590 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x14f1506e Subsystem Id: 0x1028055e Revision Id: 0x100003 No Modem Function Group found Default PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D1 D2 D3 D3cold CLKSTOP EPSS Power: setting=D0, actual=D0 GPIO: io=4, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x10 [Audio Output] wcaps 0xc1d: Stereo Amp-Out R/L Control: name="Headphone Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="CX20590 Analog", type="Audio", device=0 Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-Out vals: [0x80 0x80] Converter: stream=8, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x11 [Audio Output] wcaps 0xc1d: Stereo Amp-Out R/L Control: name="Speaker Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Speaker Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-Out vals: [0x4a 0x4a] Converter: stream=8, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x12 [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x13 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Control: name="Beep Playback Volume", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="Beep Playback Switch", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x07, nsteps=0x07, stepsize=0x0f, mute=0 Amp-Out vals: [0x07] Node 0x14 [Audio Input] wcaps 0x100d1b: Stereo Amp-In R/L Control: name="Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Capture Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Device: name="CX20590 Analog", type="Audio", device=0 Amp-In caps: ofs=0x4a, nsteps=0x50, stepsize=0x03, mute=1 Amp-In vals: [0x50 0x50] [0x80 0x80] [0x50 0x50] [0x80 0x80] Converter: stream=4, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 4 0x17 0x18 0x23* 0x24 Node 0x15 [Audio Input] wcaps 0x100d1b: Stereo Amp-In R/L Amp-In caps: ofs=0x4a, nsteps=0x50, stepsize=0x03, mute=1 Amp-In vals: [0x4a 0x4a] [0x4a 0x4a] [0x4a 0x4a] [0x4a 0x4a] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 4 0x17* 0x18 0x23 0x24 Node 0x16 [Audio Input] wcaps 0x100d1b: Stereo Amp-In R/L Amp-In caps: ofs=0x4a, nsteps=0x50, stepsize=0x03, mute=1 Amp-In vals: [0x4a 0x4a] [0x4a 0x4a] [0x4a 0x4a] [0x4a 0x4a] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 4 0x17* 0x18 0x23 0x24 Node 0x17 [Audio Selector] wcaps 0x30050d: Stereo Amp-Out Control: name="Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-Out vals: [0x00 0x00] Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 4 0x1a* 0x1b 0x1d 0x1e Node 0x18 [Audio Selector] wcaps 0x30050d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-Out vals: [0x00 0x00] Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 4 0x1a* 0x1b 0x1d 0x1e Node 0x19 [Pin Complex] wcaps 0x400581: Stereo Control: name="Headphone Jack", index=0, device=0 Pincap 0x0000001c: OUT HP Detect Pin Default 0x04211040: [Jack] HP Out at Ext Right Conn = 1/8, Color = Black DefAssociation = 0x4, Sequence = 0x0 Pin-ctls: 0xc0: OUT HP Unsolicited: tag=01, enabled=1 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x1a [Pin Complex] wcaps 0x400481: Stereo Control: name="Mic Jack", index=0, device=0 Pincap 0x00001324: IN Detect Vref caps: HIZ 50 80 Pin Default 0x04a11030: [Jack] Mic at Ext Right Conn = 1/8, Color = Black DefAssociation = 0x3, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=02, enabled=1 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x1b [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00011334: IN OUT EAPD Detect Vref caps: HIZ 50 80 EAPD 0x2: EAPD Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x1c [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x1d [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00010034: IN OUT EAPD Detect EAPD 0x2: EAPD Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x1e [Pin Complex] wcaps 0x400481: Stereo Pincap 0x00000024: IN Detect Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x1f [Pin Complex] wcaps 0x400501: Stereo Control: name="Speaker Phantom Jack", index=0, device=0 Pincap 0x00000010: OUT Pin Default 0x92170110: [Fixed] Speaker at Int Front Conn = Analog, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 2 0x10 0x11* Node 0x20 [Pin Complex] wcaps 0x400781: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 1 0x12 Node 0x21 [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x22 [Pin Complex] wcaps 0x400781: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 1 0x21 Node 0x23 [Pin Complex] wcaps 0x40040b: Stereo Amp-In Control: name="Internal Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Internal Mic Phantom Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x04, stepsize=0x2f, mute=0 Amp-In vals: [0x02 0x02] Pincap 0x00000020: IN Pin Default 0x95a60150: [Fixed] Mic at Int Top Conn = Digital, Color = Unknown DefAssociation = 0x5, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x24 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Amp-In caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 2 0x10 0x11 Node 0x25 [Vendor Defined Widget] wcaps 0xf00000: Mono Codec: Intel PantherPoint HDMI Address: 3 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x80862806 Subsystem Id: 0x80860101 Revision Id: 0x100000 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D3 CLKSTOP EPSS Power: setting=D0, actual=D0, Clock-stop-OK GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 Node 0x02 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Node 0x03 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Node 0x04 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Node 0x05 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Control: name="HDMI/DP,pcm=3 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=0, device=0 Control: name="IEC958 Playback Pro Mask", index=0, device=0 Control: name="IEC958 Playback Default", index=0, device=0 Control: name="IEC958 Playback Switch", index=0, device=0 Control: name="ELD", index=0, device=3 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x09000094: OUT Detect HBR HDMI DP Pin Default 0x18560010: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=01, enabled=1 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x02 Node 0x06 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x80] Pincap 0x09000094: OUT Detect HBR HDMI DP Pin Default 0x58560020: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x03 Node 0x07 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x80] Pincap 0x09000094: OUT Detect HBR HDMI DP Pin Default 0x58560030: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x3, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x04 Node 0x08 [Vendor Defined Widget] wcaps 0xf00000: Mono --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw----+ 1 root audio 116, 2 Oct 13 10:00 /dev/snd/controlC0 crw-rw----+ 1 root audio 116, 6 Oct 13 10:00 /dev/snd/hwC0D0 crw-rw----+ 1 root audio 116, 7 Oct 13 10:00 /dev/snd/hwC0D3 crw-rw----+ 1 root audio 116, 4 Oct 13 11:46 /dev/snd/pcmC0D0c crw-rw----+ 1 root audio 116, 3 Oct 13 14:29 /dev/snd/pcmC0D0p crw-rw----+ 1 root audio 116, 5 Oct 13 10:01 /dev/snd/pcmC0D3p crw-rw----+ 1 root audio 116, 1 Oct 13 09:59 /dev/snd/seq crw-rw----+ 1 root audio 116, 33 Oct 13 10:00 /dev/snd/timer /dev/snd/by-path: total 0 drwxr-xr-x 2 root root 60 Oct 13 10:00 . drwxr-xr-x 3 root root 220 Oct 13 10:00 .. lrwxrwxrwx 1 root root 12 Oct 13 10:00 pci-0000:00:1b.0 -> ../controlC0 !!ALSA configuration files !!------------------------ !!System wide config file (/etc/asound.conf) # Use PulseAudio by default pcm.!default { type pulse fallback "sysdefault" hint { show on description "Default ALSA Output (currently PulseAudio Sound Server)" } } ctl.!default { type pulse fallback "sysdefault" } # vim:set ft=alsaconf: !!Aplay/Arecord output !!-------------------- APLAY **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: CX20590 Analog [CX20590 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: CX20590 Analog [CX20590 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [PCH] Card hw:0 'PCH'/'HDA Intel PCH at 0xf1a10000 irq 45' Mixer name : 'Intel PantherPoint HDMI' Components : 'HDA:14f1506e,1028055e,00100003 HDA:80862806,80860101,00100000' Controls : 28 Simple ctrls : 11 Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 74 Mono: Playback 74 [100%] [0.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 74 Mono: Front Left: Playback 0 [0%] [-74.00dB] [off] Front Right: Playback 0 [0%] [-74.00dB] [off] Simple mixer control 'Speaker',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 74 Mono: Front Left: Playback 74 [100%] [0.00dB] [on] Front Right: Playback 74 [100%] [0.00dB] [on] Simple mixer control 'PCM',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [0.00dB] Front Right: Playback 255 [100%] [0.00dB] Simple mixer control 'Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 4 Front Left: 0 [0%] [0.00dB] Front Right: 0 [0%] [0.00dB] Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] Simple mixer control 'Beep',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 7 Mono: Playback 7 [100%] [0.00dB] [off] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 80 Front Left: Capture 80 [100%] [6.00dB] [on] Front Right: Capture 80 [100%] [6.00dB] [on] Simple mixer control 'Auto-Mute Mode',0 Capabilities: enum Items: 'Disabled' 'Enabled' Item0: 'Enabled' Simple mixer control 'Digital',0 Capabilities: cvolume Capture channels: Front Left - Front Right Limits: Capture 0 - 120 Front Left: Capture 120 [100%] [30.00dB] Front Right: Capture 120 [100%] [30.00dB] Simple mixer control 'Internal Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 4 Front Left: 2 [50%] [24.00dB] Front Right: 2 [50%] [24.00dB] !!Alsactl output !!-------------- --startcollapse-- state.PCH { control.1 { iface MIXER name 'Headphone Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 74' dbmin -7400 dbmax 0 dbvalue.0 -7400 dbvalue.1 -7400 } } control.2 { iface MIXER name 'Headphone Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.3 { iface MIXER name 'Speaker Playback Volume' value.0 74 value.1 74 comment { access 'read write' type INTEGER count 2 range '0 - 74' dbmin -7400 dbmax 0 dbvalue.0 0 dbvalue.1 0 } } control.4 { iface MIXER name 'Speaker Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.5 { iface MIXER name 'Auto-Mute Mode' value Enabled comment { access 'read write' type ENUMERATED count 1 item.0 Disabled item.1 Enabled } } control.6 { iface MIXER name 'Capture Volume' value.0 80 value.1 80 comment { access 'read write' type INTEGER count 2 range '0 - 80' dbmin -7400 dbmax 600 dbvalue.0 600 dbvalue.1 600 } } control.7 { iface MIXER name 'Capture Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.8 { iface MIXER name 'Internal Mic Boost Volume' value.0 2 value.1 2 comment { access 'read write' type INTEGER count 2 range '0 - 4' dbmin 0 dbmax 4800 dbvalue.0 2400 dbvalue.1 2400 } } control.9 { iface MIXER name 'Mic Boost Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 4' dbmin 0 dbmax 4000 dbvalue.0 0 dbvalue.1 0 } } control.10 { iface MIXER name 'Master Playback Volume' value 74 comment { access 'read write' type INTEGER count 1 range '0 - 74' dbmin -7400 dbmax 0 dbvalue.0 0 } } control.11 { iface MIXER name 'Master Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.12 { iface CARD name 'Internal Mic Phantom Jack' value true comment { access read type BOOLEAN count 1 } } control.13 { iface CARD name 'Mic Jack' value false comment { access read type BOOLEAN count 1 } } control.14 { iface CARD name 'Headphone Jack' value false comment { access read type BOOLEAN count 1 } } control.15 { iface CARD name 'Speaker Phantom Jack' value true comment { access read type BOOLEAN count 1 } } control.16 { iface MIXER name 'Beep Playback Volume' value 7 comment { access 'read write' type INTEGER count 1 range '0 - 7' dbmin -2800 dbmax 0 dbvalue.0 0 } } control.17 { iface MIXER name 'Beep Playback Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.18 { iface PCM name 'Playback Channel Map' value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } control.19 { iface PCM name 'Capture Channel Map' value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } control.20 { iface CARD name 'HDMI/DP,pcm=3 Jack' value false comment { access read type BOOLEAN count 1 } } control.21 { iface MIXER name 'IEC958 Playback Con Mask' value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.22 { iface MIXER name 'IEC958 Playback Pro Mask' value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.23 { iface MIXER name 'IEC958 Playback Default' value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.24 { iface MIXER name 'IEC958 Playback Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.25 { iface PCM device 3 name ELD value '' comment { access 'read volatile' type BYTES count 0 } } control.26 { iface PCM device 3 name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 value.4 0 value.5 0 value.6 0 value.7 0 comment { access 'read write' type INTEGER count 8 range '0 - 36' } } control.27 { iface MIXER name 'PCM Playback Volume' value.0 255 value.1 255 comment { access 'read write user' type INTEGER count 2 range '0 - 255' tlv '0000000100000008ffffec1400000014' dbmin -5100 dbmax 0 dbvalue.0 0 dbvalue.1 0 } } control.28 { iface MIXER name 'Digital Capture Volume' value.0 120 value.1 120 comment { access 'read write user' type INTEGER count 2 range '0 - 120' tlv '0000000100000008fffff44800000032' dbmin -3000 dbmax 3000 dbvalue.0 3000 dbvalue.1 3000 } } } --endcollapse-- !!All Loaded Modules !!------------------ Module cpufreq_stats rfcomm ctr ccm fuse bbswitch bnep ecb rtsx_usb_ms memstick uvcvideo ath3k videobuf2_vmalloc videobuf2_memops videobuf2_core btusb bluetooth v4l2_common videodev 6lowpan_iphc media ext4 crc16 mbcache jbd2 nvram arc4 ath9k ath9k_common ath9k_hw intel_rapl ath x86_pkg_temp_thermal intel_powerclamp crct10dif_pclmul crc32_pclmul joydev ghash_clmulni_intel aesni_intel mac80211 aes_x86_64 lrw gf128mul cfg80211 mousedev glue_helper ablk_helper cryptd dell_laptop iTCO_wdt alx dell_wmi iTCO_vendor_support sparse_keymap serio_raw psmouse rfkill dcdbas pcspkr thermal microcode snd_hda_codec_hdmi snd_hda_codec_conexant snd_hda_codec_generic snd_hda_intel snd_hda_controller snd_hda_codec mdio snd_hwdep snd_pcm snd_timer snd wmi mei_me mei lpc_ich evdev shpchp i2c_i801 processor mac_hid soundcore battery ac usbip_host usbip_core coretemp msr kvm_intel kvm i8k hwmon btrfs xor raid6_pq sd_mod sr_mod crc_t10dif cdrom crct10dif_common rtsx_usb_sdmmc led_class mmc_core rtsx_usb hid_generic usbhid hid atkbd libps2 crc32c_intel ahci libahci libata scsi_mod i915 xhci_hcd ehci_pci intel_gtt i2c_algo_bit ehci_hcd drm_kms_helper usbcore drm usb_common i2c_core i8042 serio video button !!Sysfs Files !!----------- /sys/class/sound/hwC0D0/init_pin_configs: 0x19 0x04211040 0x1a 0x04a11030 0x1b 0x400001f0 0x1c 0x400001f0 0x1d 0x400001f0 0x1e 0x400001f0 0x1f 0x92170110 0x20 0x400001f0 0x22 0x400001f0 0x23 0x95a60150 /sys/class/sound/hwC0D0/driver_pin_configs: /sys/class/sound/hwC0D0/user_pin_configs: /sys/class/sound/hwC0D0/init_verbs: /sys/class/sound/hwC0D0/hints: /sys/class/sound/hwC0D3/init_pin_configs: 0x05 0x18560010 0x06 0x58560020 0x07 0x58560030 /sys/class/sound/hwC0D3/driver_pin_configs: /sys/class/sound/hwC0D3/user_pin_configs: /sys/class/sound/hwC0D3/init_verbs: /sys/class/sound/hwC0D3/hints: !!ALSA/HDA dmesg !!-------------- [ 13.954329] wmi: Mapper loaded [ 14.228325] snd_hda_intel 0000:00:1b.0: irq 45 for MSI/MSI-X [ 14.240346] sound hdaudioC0D0: CX20590: BIOS auto-probing. [ 14.240798] sound hdaudioC0D0: autoconfig: line_outs=1 (0x1f/0x0/0x0/0x0/0x0) type:speaker [ 14.240801] sound hdaudioC0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 14.240803] sound hdaudioC0D0: hp_outs=1 (0x19/0x0/0x0/0x0/0x0) [ 14.240805] sound hdaudioC0D0: mono: mono_out=0x0 [ 14.240807] sound hdaudioC0D0: inputs: [ 14.240809] sound hdaudioC0D0: Internal Mic=0x23 [ 14.240812] sound hdaudioC0D0: Mic=0x1a [ 14.241749] sound hdaudioC0D0: Enable sync_write for stable communication [ 14.249169] input: HDA Digital PCBeep as /devices/pci0000:00/0000:00:1b.0/sound/card0/hdaudioC0D0/input9 [ 14.249439] input: HDA Intel PCH Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input10 [ 14.249502] input: HDA Intel PCH Headphone as /devices/pci0000:00/0000:00:1b.0/sound/card0/input11 [ 14.249564] input: HDA Intel PCH HDMI/DP,pcm=3 as /devices/pci0000:00/0000:00:1b.0/sound/card0/input12 [ 14.260752] microcode: CPU0 sig=0x306a9, pf=0x10, revision=0x15 -- Mateus Rodrigues Costa

1 0

[alsa-devel] Audio Mini-Summit info updates
by Takashi Iwai 13 Oct '14

13 Oct '14

Hi, some information updates about Audio Mini-Summit: * Venue we'll have the room 110 for our meeting in Dusseldorf Congress Center. The LPC web page shows the map: http://www.linuxplumbersconf.org/2014/directions-and-transportation/ * Schedule The room will be open from 8:30AM, so I think we can start at 9:00AM, if there is no objection. The registration for LPC is open at 8:00AM. Dylan has a talk for ELCE at 12:15PM:; should we put the UCM discussion afternoon or better done in morning before? * Pre/post-meeting events I added new fields in the attendee list for dinners on Monday and/or Tuesday. Please update by yourselves. (URL see below) The restaurants aren't decided yet at all, I'm looking for it. Are German restaurants interesting in general? It's usually not the best choice for Vegetarian, though. I have been at Dusseldorf only once, so have little idea. If you have a recommendation or a suggestion, it'd be appreciated! *** For those still haven't signed up or missed the announcement: we'll have the Audio mini-summit at Dusseldorf, Oct. 14, along with Linux Plumbers Conference and other Linux Foundation meetings. For joining, just add your name to the attendee list below, which is a Google doc everyone can edit: http://goo.gl/VbXLPW The proposed topics are seen in: http://goo.gl/JSasQ9 Note that you'd need a pass for LPC or other LF conferences to enter the building. It's the only restriction. Takashi

6 12

[alsa-devel] How to select input from muxer in front of SSI on i.MX6
by Jean-Michel Hautbois 13 Oct '14

13 Oct '14

Hi, I am now working on making my audio path working completely. And I get some difficulties to write a DT for my needs. I join a picture which is not the reality but a simplified version of the board. I have a FPGA between my codecs and my i.MX6. This FPGA is responsible of lots of things, and in particular of muxing signals. I can program it in order to get the audio signal I want on the SSI I want (so sgtl5000 can ben on SSI1 or SSI2 or both). This path is not a single codec, but a multiple codec one. So I can have on SSI1 LineIn from sgtl5000 or audio from hdmi (adv7611). I don't know how to write it in DT, regarding audmux for instance, and the sound{} node... And I don't know how to drive all these peripherals (from arecord, I wish I have hw:0,0 is LineIn, hw:0;1 is HDMI, etc.). Any help would be appreciated :). Thanks, JM

3 7

[alsa-devel] [PATCH v2] ASoC: rt5645: Add the workqueue of the jack detect function for the debouncing
by Oder Chiou 13 Oct '14

13 Oct '14

Add the workqueue of the jack detect function for the debouncing. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5645.c | 15 ++++++++++++++- sound/soc/codecs/rt5645.h | 1 + 2 files changed, 15 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index cfe1eff..16ee4f3 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2166,11 +2166,20 @@ int rt5645_set_jack_detect(struct snd_soc_codec *codec, } EXPORT_SYMBOL_GPL(rt5645_set_jack_detect); +static void rt5645_jack_detect_work(struct work_struct *work) +{ + struct rt5645_priv *rt5645 = + container_of(work, struct rt5645_priv, jack_detect_work.work); + + rt5645_jack_detect(rt5645->codec, rt5645->jack); +} + static irqreturn_t rt5645_irq(int irq, void *data) { struct rt5645_priv *rt5645 = data; - rt5645_jack_detect(rt5645->codec, rt5645->jack); + queue_delayed_work(system_power_efficient_wq, + &rt5645->jack_detect_work, msecs_to_jiffies(250)); return IRQ_HANDLED; } @@ -2436,6 +2445,8 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, dev_err(&i2c->dev, "Fail gpio_direction hp_det_gpio\n"); } + INIT_DELAYED_WORK(&rt5645->jack_detect_work, rt5645_jack_detect_work); + return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5645, rt5645_dai, ARRAY_SIZE(rt5645_dai)); } @@ -2447,6 +2458,8 @@ static int rt5645_i2c_remove(struct i2c_client *i2c) if (i2c->irq) free_irq(i2c->irq, rt5645); + cancel_delayed_work_sync(&rt5645->jack_detect_work); + if (gpio_is_valid(rt5645->pdata.hp_det_gpio)) gpio_free(rt5645->pdata.hp_det_gpio); diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index 50c62c5..5ec2520 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -2168,6 +2168,7 @@ struct rt5645_priv { struct regmap *regmap; struct i2c_client *i2c; struct snd_soc_jack *jack; + struct delayed_work jack_detect_work; int sysclk; int sysclk_src; -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [WIP PATCH] Scarlett mixer interface for 6i6, 18i6, 18i8 and 18i20.
by David Henningsson 13 Oct '14

13 Oct '14

Author: Tobias Hoffman <th55(a)gmx.de> Author: Robin Gareus <robin(a)gareus.org> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- So, this is how Tobias' and Robin's patch look now. I've merged it all to one big patch, both for my own simplicity and because I thought that made the most sense. I've also fixed most checkpatch stuff (apart from long lines warnings). It's not ready for merging yet, I assume. But it would be good with a hint w r t what the big issues are with this patch as you see it. And then we could see if I have some spare cycles to fix that up, or not. sound/usb/Makefile | 1 + sound/usb/mixer.c | 28 +- sound/usb/quirks-table.h | 51 -- sound/usb/scarlettmixer.c | 1374 +++++++++++++++++++++++++++++++++++++++++++++ sound/usb/scarlettmixer.h | 6 + 5 files changed, 1404 insertions(+), 56 deletions(-) create mode 100644 sound/usb/scarlettmixer.c create mode 100644 sound/usb/scarlettmixer.h diff --git a/sound/usb/Makefile b/sound/usb/Makefile index 2b92f0d..5d9008d 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -12,6 +12,7 @@ snd-usb-audio-objs := card.o \ pcm.o \ proc.o \ quirks.o \ + scarlettmixer.o \ stream.o snd-usbmidi-lib-objs := midi.o diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 2e4a9db..8360762 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -62,6 +62,7 @@ #include "helper.h" #include "mixer_quirks.h" #include "power.h" +#include "scarlettmixer.h" #define MAX_ID_ELEMS 256 @@ -2305,6 +2306,7 @@ static void snd_usb_mixer_interrupt_v2(struct usb_mixer_interface *mixer, __func__, channel); return; } +snd_printk(KERN_INFO "scarlett thobi mixer interrupt %x %x %x\n", attribute, value, index); /* TODO */ for (info = mixer->id_elems[unitid]; info; info = info->next_id_elem) { if (info->control != control) @@ -2462,11 +2464,27 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, break; } - if ((err = snd_usb_mixer_controls(mixer)) < 0 || - (err = snd_usb_mixer_status_create(mixer)) < 0) - goto _error; - - snd_usb_mixer_apply_create_quirk(mixer); + switch (chip->usb_id) { + case USB_ID(0x1235, 0x8012): /* Focusrite Scarlett 6i6 */ + case USB_ID(0x1235, 0x8002): /* Focusrite Scarlett 8i6 */ + case USB_ID(0x1235, 0x8004): /* Focusrite Scarlett 18i6 */ + case USB_ID(0x1235, 0x8014): /* Focusrite Scarlett 18i8 */ + case USB_ID(0x1235, 0x800c): /* Focusrite Scarlett 18i20 */ + /* don't even try to parse UAC2 descriptors */ + err = scarlett_mixer_controls(mixer); + if (err < 0) + goto _error; + break; + default: + err = snd_usb_mixer_controls(mixer); + if (err < 0) + goto _error; + err = snd_usb_mixer_status_create(mixer); + if (err < 0) + goto _error; + snd_usb_mixer_apply_create_quirk(mixer); + break; + } err = snd_device_new(chip->card, SNDRV_DEV_CODEC, mixer, &dev_ops); if (err < 0) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 223c47b..70ae637 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2637,57 +2637,6 @@ YAMAHA_DEVICE(0x7010, "UB99"), .type = QUIRK_MIDI_NOVATION } }, -{ - /* - * Focusrite Scarlett 18i6 - * - * Avoid mixer creation, which otherwise fails because some of - * the interface descriptor subtypes for interface 0 are - * unknown. That should be fixed or worked-around but this at - * least allows the device to be used successfully with a DAW - * and an external mixer. See comments below about other - * ignored interfaces. - */ - USB_DEVICE(0x1235, 0x8004), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "Focusrite", - .product_name = "Scarlett 18i6", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = & (const struct snd_usb_audio_quirk[]) { - { - /* InterfaceSubClass 1 (Control Device) */ - .ifnum = 0, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - /* InterfaceSubClass 1 (Control Device) */ - .ifnum = 3, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = 4, - .type = QUIRK_MIDI_STANDARD_INTERFACE - }, - { - /* InterfaceSubClass 1 (Device Firmware Update) */ - .ifnum = 5, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = -1 - } - } - } -}, /* Access Music devices */ { diff --git a/sound/usb/scarlettmixer.c b/sound/usb/scarlettmixer.c new file mode 100644 index 0000000..873d5d6 --- /dev/null +++ b/sound/usb/scarlettmixer.c @@ -0,0 +1,1374 @@ +/* + * (Tentative) Scarlett 18i6 Driver for ALSA + * + * Copyright (c) 2013 by Tobias Hoffmann + * Copyright (c) 2013 by Robin Gareus <robin(a)gareus.org> + * Copyright (c) 2002 by Takashi Iwai <tiwai(a)suse.de> + * + * Many codes borrowed from audio.c by + * Alan Cox (alan(a)lxorguk.ukuu.org.uk) + * Thomas Sailer (sailer(a)ife.ee.ethz.ch) + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + */ + +/* + * Rewritten and extended to support more models, e.g. Scarlett 18i8. + * TODO? reset_first/ channel init + * TODO... test meter? + */ + +/* + * Many features of Scarlett 18i6 do not lend themselves to be implemented + * as simple mixer-quirk -- or at least I don't see a way how to do that, yet. + * Hence the top parts of this file is a 1:1 copy of select static functions + * from mixer.c to implement the interface. + * Suggestions how to avoid this code duplication are very welcome. + * + * eventually this should either be integrated as quirk into mixer_quirks.c + * or become a standalone module. + * + * This source hardcodes the URBs for the Scarlett, + * Auto-detection via UAC2 is not feasible to properly discover the vast majority + * of features. It's related to both Linux/ALSA's UAC2 as well as Focusrite's + * implementation of it. Eventually quirks may be sufficient but right now + * it's a major headache to work arount these things. + * + * NB. Neither the OSX nor the win driver provided by Focusrite performs + * discovery, they seem to operate the same as this driver. + */ + +/* Mixer Interface for the Focusrite Scarlett 18i6 audio interface. + * + * The protocol was reverse engineered by looking at communication between + * Scarlett MixControl (v 1.2.128.0) and the Focusrite(R) Scarlett 18i6 (firmware + * v305) using wireshark and usbmon in January 2013. + * Extended in July 2013. + * + * this mixer gives complete access to all features of the device: + * - change Impedance of inputs (Line-in, Mic / Instrument, Hi-Z) + * - select clock source + * - dynamic input to mixer-matrix assignment + * - 18 x 6 mixer-matrix gain stages + * - bus routing & volume control + * - save setting to hardware + * - automatic re-initialization on connect if device was power-cycled + * - peak monitoring of all 3 buses (18 input, 6 DAW input, 6 route chanels) + * (changing the samplerate and buffersize is supported by the PCM interface) + * + * + * USB URB commands overview (bRequest = 0x01 = UAC2_CS_CUR) + * wIndex + * 0x01 Analog Input line/instrument impedance switch, wValue=0x0901 + channel, data=Line/Inst (2bytes) + * pad (-10dB) switch, wValue=0x0b01 + channel, data=Off/On (2bytes) + * ?? wValue=0x0803/04, ?? (2bytes) + * 0x0a Master Volume, wValue=0x0200+bus[0:all + only 1..4?] data(2bytes) + * Bus Mute/Unmute wValue=0x0100+bus[0:all + only 1..4?], data(2bytes) + * 0x28 Clock source, wValue=0x0100, data={1:int,2:spdif,3:adat} (1byte) + * 0x29 Set Sample-rate, wValue=0x0100, data=sample-rate(4bytes) + * 0x32 Mixer mux, wValue=0x0600 + mixer-channel, data=input-to-connect(2bytes) + * 0x33 Output mux, wValue=bus, data=input-to-connect(2bytes) + * 0x34 Capture mux, wValue=0...18, data=input-to-connect(2bytes) + * 0x3c Matrix Mixer gains, wValue=mixer-node data=gain(2bytes) + * ?? [sometimes](4bytes, e.g 0x000003be 0x000003bf ...03ff) + * + * USB reads: (i.e. actually issued by original software) + * 0x01 wValue=0x0901+channel (1byte!!), wValue=0x0b01+channed (1byte!!) + * 0x29 wValue=0x0100 sample-rate(4bytes) + * wValue=0x0200 ?? 1byte (only once) + * 0x2a wValue=0x0100 ?? 4bytes, sample-rate2 ?? + * + * USB reads with bRequest = 0x03 = UAC2_CS_MEM + * 0x3c wValue=0x0002 1byte: sync status (locked=1) + * wValue=0x0000 18*2byte: peak meter (inputs) + * wValue=0x0001 8(?)*2byte: peak meter (mix) + * wValue=0x0003 6*2byte: peak meter (pcm/daw) + * + * USB write with bRequest = 0x03 + * 0x3c Save settings to hardware: wValue=0x005a, data=0xa5 + * + * + * <ditaa> + * /--------------\ 18chn 6chn /--------------\ + * | Hardware in +--+-------\ /------+--+ ALSA PCM out | + * \--------------/ | | | | \--------------/ + * | | | | + * | v v | + * | +---------------+ | + * | \ Matrix Mux / | + * | +-----+-----+ | + * | | | + * | | 18chn | + * | v | + * | +-----------+ | + * | | Mixer | | + * | | Matrix | | + * | | | | + * | | 18x6 Gain | | + * | | stages | | + * | +-----+-----+ | + * | | | + * | | | + * | 18chn | 6chn | 6chn + * v v v + * ========================= + * +---------------+ +--—------------+ + * \ Output Mux / \ Capture Mux / + * +-----+-----+ +-----+-----+ + * | | + * | 6chn | + * v | + * +-------------+ | + * | Master Gain | | + * +------+------+ | + * | | + * | 6chn | 18chn + * | (3 stereo pairs) | + * /--------------\ | | /--------------\ + * | Hardware out |<--/ \-->| ALSA PCM in | + * \--------------/ \--------------/ + * </ditaa> + * + */ + +#include <linux/slab.h> +#include <linux/usb.h> +#include <linux/usb/audio-v2.h> + +#include <sound/core.h> +#include <sound/control.h> +#include <sound/tlv.h> + +#include "usbaudio.h" +#include "mixer.h" +#include "helper.h" +#include "power.h" + +#include "scarlettmixer.h" + +/* #define WITH_METER */ +/* #define WITH_LOGSCALEMETER */ + +#define LEVEL_BIAS 128 /* some gui mixers can't handle negative ctl values (alsamixergui, qasmixer, ...) */ + +#ifndef LEVEL_BIAS + #define LEVEL_BIAS 0 +#endif + +struct scarlett_enum_info { + int start, len; + const char **texts; +}; + +struct scarlett_device_info { + int matrix_in; + int matrix_out; + int input_len; + int output_len; + + int pcm_start; + int analog_start; + int spdif_start; + int adat_start; + int mix_start; + + struct scarlett_enum_info opt_master; + struct scarlett_enum_info opt_matrix; + + int (*controls_fn)(struct usb_mixer_interface *mixer, + const struct scarlett_device_info *info); + + int matrix_mux_init[]; +}; + +struct scarlett_mixer_elem_info { + struct usb_mixer_interface *mixer; + + /* URB command details */ + int wValue, index; + int val_len; + + int count; /* number of channels, using ++wValue */ + + const struct scarlett_enum_info *opt; + + int cached; + int cache_val[MAX_CHANNELS]; +}; + +static void scarlett_mixer_elem_free(struct snd_kcontrol *kctl) +{ + kfree(kctl->private_data); + kctl->private_data = NULL; +} + +/***************************** Low Level USB I/O *****************************/ + +/* stripped down/adapted from get_ctl_value_v2 */ +static int get_ctl_urb2(struct snd_usb_audio *chip, + int bRequest, int wValue, int index, + unsigned char *buf, int size) +{ + int ret, idx = 0; + + ret = snd_usb_autoresume(chip); + if (ret < 0 && ret != -ENODEV) { + ret = -EIO; + goto error; + } + + down_read(&chip->shutdown_rwsem); + if (chip->shutdown) { + ret = -ENODEV; + } else { + idx = snd_usb_ctrl_intf(chip) | (index << 8); + ret = snd_usb_ctl_msg(chip->dev, + usb_rcvctrlpipe(chip->dev, 0), + bRequest, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, + wValue, idx, buf, size); + } + up_read(&chip->shutdown_rwsem); + snd_usb_autosuspend(chip); + + if (ret < 0) { +error: + snd_printk(KERN_ERR "cannot get ctl value: req = %#x, wValue = %#x, wIndex = %#x, size = %d\n", + bRequest, wValue, idx, size); + return ret; + } +#if 0 /* rg debug XXX */ + snd_printk(KERN_ERR "req ctl value: req = %#x, rtype = %#x, wValue = %#x, wIndex = %#x, size = %d\n", + bRequest, (USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN), wValue, idx, size); +#endif + return 0; +} + +/* adopted from snd_usb_mixer_set_ctl_value */ +static int set_ctl_urb2(struct snd_usb_audio *chip, + int request, int wValue, int index, + unsigned char *buf, int val_len) +{ + int idx = 0, err, timeout = 10; + + err = snd_usb_autoresume(chip); + if (err < 0 && err != -ENODEV) + return -EIO; + down_read(&chip->shutdown_rwsem); + while (timeout-- > 0) { + if (chip->shutdown) + break; + idx = snd_usb_ctrl_intf(chip) | (index << 8); + if (snd_usb_ctl_msg(chip->dev, + usb_sndctrlpipe(chip->dev, 0), request, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT, + wValue, idx, buf, val_len) >= 0) { + err = 0; + goto out; + } + } + snd_printdd(KERN_ERR "cannot set ctl value: req = %#x, wValue = %#x, wIndex = %#x, len = %d, data = %#x/%#x\n", + request, wValue, idx, val_len, buf[0], buf[1]); + err = -EINVAL; + + out: + up_read(&chip->shutdown_rwsem); + snd_usb_autosuspend(chip); +#if 0 /* rg debug XXX */ + snd_printk(KERN_ERR "set ctl value: req = %#x, wValue = %#x, wIndex = %#x, len = %d, data = %#x/%#x\n", + request, wValue, idx, val_len, buf[0], buf[1]); +#endif + return err; +} + +/***************************** High Level USB *****************************/ + +static int set_ctl_value(struct scarlett_mixer_elem_info *elem, int channel, int value) +{ + struct snd_usb_audio *chip = elem->mixer->chip; + unsigned char buf[2]; + int err; + + if (elem->val_len == 2) { /* S16 */ + buf[0] = value & 0xff; + buf[1] = (value >> 8) & 0xff; + } else { /* U8 */ + buf[0] = value & 0xff; + } + + err = set_ctl_urb2(chip, UAC2_CS_CUR, elem->wValue + channel, elem->index, buf, elem->val_len); + if (err < 0) + return err; + + elem->cached |= 1 << channel; + elem->cache_val[channel] = value; + return 0; +} + +/* + TODO: can't read back any volume (master/mixer), only cache works [?] + [return 0xfe for enums???] +*/ +static int get_ctl_value(struct scarlett_mixer_elem_info *elem, int channel, int *value) +{ + struct snd_usb_audio *chip = elem->mixer->chip; + unsigned char buf[2] = {0, 0}; + int err, val_len; + + if (elem->cached & (1 << channel)) { + *value = elem->cache_val[channel]; + return 0; + } + + val_len = elem->val_len; + /* quirk: write 2bytes, but read 1byte */ + if ((elem->index == 0x01) || /* input impedance and input pad switch */ + ((elem->index == 0x0a) && (elem->wValue < 0x0200)) || /* bus mutes */ + (elem->index == 0x32) || (elem->index == 0x33)) { /* mux */ + val_len = 1; + } + + err = get_ctl_urb2(chip, UAC2_CS_CUR, elem->wValue + channel, elem->index, buf, val_len); + if (err < 0) { + snd_printd(KERN_ERR "cannot get current value for control %x ch %d: err = %d\n", + elem->wValue, channel, err); + return err; + } + + if (val_len == 2) { /* S16 */ + *value = buf[0] | ((unsigned int)buf[1] << 8); + if (*value >= 0x8000) + (*value) -= 0x10000; + } else { /* U8 */ + *value = buf[0]; + } + + elem->cached |= 1 << channel; + elem->cache_val[channel] = *value; + + return 0; +} + +/********************** Enum Strings *************************/ +static const char txtOff[] = "Off", + txtPcm1[] = "PCM 1", txtPcm2[] = "PCM 2", + txtPcm3[] = "PCM 3", txtPcm4[] = "PCM 4", + txtPcm5[] = "PCM 5", txtPcm6[] = "PCM 6", + txtPcm7[] = "PCM 7", txtPcm8[] = "PCM 8", + txtPcm9[] = "PCM 9", txtPcm10[] = "PCM 10", + txtPcm11[] = "PCM 11", txtPcm12[] = "PCM 12", + txtPcm13[] = "PCM 13", txtPcm14[] = "PCM 14", + txtPcm15[] = "PCM 15", txtPcm16[] = "PCM 16", + txtPcm17[] = "PCM 17", txtPcm18[] = "PCM 18", + txtPcm19[] = "PCM 19", txtPcm20[] = "PCM 20", + txtAnlg1[] = "Analog 1", txtAnlg2[] = "Analog 2", + txtAnlg3[] = "Analog 3", txtAnlg4[] = "Analog 4", + txtAnlg5[] = "Analog 5", txtAnlg6[] = "Analog 6", + txtAnlg7[] = "Analog 7", txtAnlg8[] = "Analog 8", + txtSpdif1[] = "SPDIF 1", txtSpdif2[] = "SPDIF 2", + txtAdat1[] = "ADAT 1", txtAdat2[] = "ADAT 2", + txtAdat3[] = "ADAT 3", txtAdat4[] = "ADAT 4", + txtAdat5[] = "ADAT 5", txtAdat6[] = "ADAT 6", + txtAdat7[] = "ADAT 7", txtAdat8[] = "ADAT 8", + txtMix1[] = "Mix A", txtMix2[] = "Mix B", + txtMix3[] = "Mix C", txtMix4[] = "Mix D", + txtMix5[] = "Mix E", txtMix6[] = "Mix F", + txtMix7[] = "Mix G", txtMix8[] = "Mix H"; + +static const struct scarlett_enum_info opt_pad = { + .start = 0, + .len = 2, + .texts = (const char *[]){ + txtOff, "-10dB" + } +}; + +static const struct scarlett_enum_info opt_impedance = { + .start = 0, + .len = 2, + .texts = (const char *[]){ + "Line", "Hi-Z" + } +}; + +static const struct scarlett_enum_info opt_clock = { + .start = 1, + .len = 3, + .texts = (const char *[]){ + "Internal", "SPDIF", "ADAT" + } +}; + +static const struct scarlett_enum_info opt_sync = { + .start = 0, + .len = 2, + .texts = (const char *[]){ + "No Lock", "Locked" + } +}; + +static const struct scarlett_enum_info opt_save = { + .start = 0, + .len = 2, + .texts = (const char *[]){ + "---", "Save" + } +}; + +#ifdef WITH_LOGSCALEMETER +/* approx ( 20.0 * log10(x) ) for 16bit + * map 0..65535 to range 0..194 = -97.0..0dB in .5dB steps */ +static int sig_to_db(const int sig16bit) +{ + int i; + const int dbtbl[148] = { + 13, 14, 15, 16, 16, 17, 18, 20, 21, 22, 23, 25, 26, 28, 29, 31, 33, 35, + 37, 39, 41, 44, 46, 49, 52, 55, 58, 62, 66, 69, 74, 78, 83, 87, 93, 98, + 104, 110, 117, 123, 131, 139, 147, 155, 165, 174, 185, 196, 207, 220, + 233, 246, 261, 276, 293, 310, 328, 348, 369, 390, 414, 438, 464, 491, + 521, 551, 584, 619, 655, 694, 735, 779, 825, 874, 926, 981, 1039, 1100, + 1165, 1234, 1308, 1385, 1467, 1554, 1646, 1744, 1847, 1957, 2072, 2195, + 2325, 2463, 2609, 2764, 2927, 3101, 3285, 3479, 3685, 3904, 4135, 4380, + 4640, 4915, 5206, 5514, 5841, 6187, 6554, 6942, 7353, 7789, 8250, 8739, + 9257, 9806, 10387, 11002, 11654, 12345, 13076, 13851, 14672, 15541, + 16462, 17437, 18471, 19565, 20724, 21952, 23253, 24631, 26090, 27636, + 29274, 31008, 32846, 34792, 36854, 39037, 41350, 43801, 46396, 49145, + 52057, 55142, 58409, 61870 + }; + + if (sig16bit < 13) { + switch (sig16bit) { + case 0: return 0; + case 1: return 1; /* -96.0dB */ + case 2: return 13; /* -90.5dB */ + case 3: return 20; /* -87.0dB */ + case 4: return 26; /* -84.0dB */ + case 5: return 29; /* -82.5dB */ + case 6: return 32; /* -81.0dB */ + case 7: return 35; /* -79.5dB */ + case 8: return 37; /* -78.5dB */ + case 9: return 39; /* -77.5dB */ + case 10: return 41; /* -76.5dB */ + case 11: return 43; /* -75.5dB */ + case 12: return 45; /* -74.5dB */ + } + } + for (i = 0; i < 148; ++i) { + if (sig16bit <= dbtbl[i]) + return 46+i; + } + return 194; +} +#endif + +static int scarlett_ctl_switch_info(struct snd_kcontrol *kctl, struct snd_ctl_elem_info *uinfo) +{ + struct scarlett_mixer_elem_info *elem = kctl->private_data; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = elem->count; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int scarlett_ctl_switch_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) +{ + struct scarlett_mixer_elem_info *elem = kctl->private_data; + int i, err, val; + + for (i = 0; i < elem->count; i++) { + err = get_ctl_value(elem, i, &val); + if (err < 0) + return err; + + val = !val; /* alsa uses 0: on, 1: off */ + ucontrol->value.integer.value[i] = val; + } + + return 0; +} + +static int scarlett_ctl_switch_put(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) +{ + struct scarlett_mixer_elem_info *elem = kctl->private_data; + int i, changed = 0; + int err, oval, val; + + for (i = 0; i < elem->count; i++) { + err = get_ctl_value(elem, i, &oval); + if (err < 0) + return err; + + val = ucontrol->value.integer.value[i]; + val = !val; + if (oval != val) { + err = set_ctl_value(elem, i, val); + if (err < 0) + return err; + + changed = 1; + } + } + + return changed; +} + +static int scarlett_ctl_info(struct snd_kcontrol *kctl, struct snd_ctl_elem_info *uinfo) +{ + struct scarlett_mixer_elem_info *elem = kctl->private_data; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = elem->count; + uinfo->value.integer.min = -128 + LEVEL_BIAS; + uinfo->value.integer.max = (int)kctl->private_value + LEVEL_BIAS; + uinfo->value.integer.step = 1; + return 0; +} + +static int scarlett_ctl_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) +{ + struct scarlett_mixer_elem_info *elem = kctl->private_data; + int i, err, val; + + for (i = 0; i < elem->count; i++) { + err = get_ctl_value(elem, i, &val); + if (err < 0) + return err; + + val = clamp(val / 256, -128, (int)kctl->private_value) + LEVEL_BIAS; + ucontrol->value.integer.value[i] = val; + } + + return 0; +} + +static int scarlett_ctl_put(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) +{ + struct scarlett_mixer_elem_info *elem = kctl->private_data; + int i, changed = 0; + int err, oval, val; + + for (i = 0; i < elem->count; i++) { + err = get_ctl_value(elem, i, &oval); + if (err < 0) + return err; + + val = ucontrol->value.integer.value[i] - LEVEL_BIAS; + val = val * 256; + if (oval != val) { + err = set_ctl_value(elem, i, val); + if (err < 0) + return err; + + changed = 1; + } + } + + return changed; +} + +static int scarlett_ctl_enum_info(struct snd_kcontrol *kctl, struct snd_ctl_elem_info *uinfo) +{ + struct scarlett_mixer_elem_info *elem = kctl->private_data; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = elem->count; + uinfo->value.enumerated.items = elem->opt->len; + if (uinfo->value.enumerated.item > uinfo->value.enumerated.items - 1) + uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; + strcpy(uinfo->value.enumerated.name, elem->opt->texts[uinfo->value.enumerated.item]); + return 0; +} + +static int scarlett_ctl_enum_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) +{ + struct scarlett_mixer_elem_info *elem = kctl->private_data; + int err, val; + + err = get_ctl_value(elem, 0, &val); + if (err < 0) + return err; + + /* snd_printk(KERN_WARNING "enum %s: %x %x\n", ucontrol->id.name, val, elem->opt->len); */ + if ((elem->opt->start == -1) && (val > elem->opt->len)) /* >= 0x20 ??? */ + val = 0; + else + val = clamp(val - elem->opt->start, 0, elem->opt->len-1); + + ucontrol->value.enumerated.item[0] = val; + + return 0; +} + +static int scarlett_ctl_enum_put(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) +{ + struct scarlett_mixer_elem_info *elem = kctl->private_data; + int changed = 0; + int err, oval, val; + + err = get_ctl_value(elem, 0, &oval); + if (err < 0) + return err; + + val = ucontrol->value.integer.value[0]; +#if 0 /* TODO? */ + if (val == -1) { + val = elem->enum->len + 1; /* only true for master, not for mixer [also master must be used] */ + /* ... or? > 0x20, 18i8: 0x22 */ + } else +#endif + val = val + elem->opt->start; + if (oval != val) { + err = set_ctl_value(elem, 0, val); + if (err < 0) + return err; + + changed = 1; + } + + return changed; +} + +static int scarlett_ctl_save_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.enumerated.item[0] = 0; + return 0; +} + +static int scarlett_ctl_save_put(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) +{ + struct scarlett_mixer_elem_info *elem = kctl->private_data; + int err; + + if (ucontrol->value.enumerated.item[0] > 0) { + char buf[1] = { 0xa5 }; + + err = set_ctl_urb2(elem->mixer->chip, UAC2_CS_MEM, 0x005a, 0x3c, buf, 1); + if (err < 0) + return err; + + snd_printk(KERN_INFO "Scarlett: Saved settings to hardware.\n"); + } + + return 0; /* (?) */ +} + +#ifdef WITH_METER +static int scarlett_ctl_meter_info(struct snd_kcontrol *kctl, struct snd_ctl_elem_info *uinfo) +{ + struct scarlett_mixer_elem_info *elem = kctl->private_data; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = elem->count; +#ifdef WITH_LOGSCALEMETER + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 194; +#else + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 255; /* 0xffff ? */ +#endif + uinfo->value.integer.step = 1; + return 0; +} +#endif + +static int scarlett_ctl_meter_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) +{ + struct scarlett_mixer_elem_info *elem = kctl->private_data; + unsigned char buf[2 * MAX_CHANNELS] = {0, }; + int err, val, i; + + err = get_ctl_urb2(elem->mixer->chip, UAC2_CS_MEM, elem->wValue, elem->index, buf, elem->val_len * elem->count); + if (err < 0) { + snd_printd(KERN_ERR "cannot get current value for mem %x: err = %d\n", + elem->wValue, err); + return err; + } + + if (elem->val_len == 1) { /* single U8 */ + ucontrol->value.enumerated.item[0] = clamp((int)buf[0], 0, 1); + } else { /* multiple S16 */ + for (i = 0; i < elem->count; i++) { + val = buf[2*i] | ((unsigned int)buf[2*i + 1] << 8); + if (val >= 0x8000) + val -= 0x10000; + +#ifdef WITH_LOGSCALEMETER + ucontrol->value.integer.value[i] = sig_to_db(val); +#else + ucontrol->value.integer.value[i] = clamp(val / 256, 0, 255); +#endif + } + } + + return 0; +} + +static struct snd_kcontrol_new usb_scarlett_ctl_switch = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = scarlett_ctl_switch_info, + .get = scarlett_ctl_switch_get, + .put = scarlett_ctl_switch_put, +}; + +static const DECLARE_TLV_DB_SCALE(db_scale_scarlett_gain, -12800, 100, 0); + +static struct snd_kcontrol_new usb_scarlett_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .name = "", + .info = scarlett_ctl_info, + .get = scarlett_ctl_get, + .put = scarlett_ctl_put, + .private_value = 6, /* max value */ + .tlv = { .p = db_scale_scarlett_gain } +}; + +static struct snd_kcontrol_new usb_scarlett_ctl_master = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .name = "", + .info = scarlett_ctl_info, + .get = scarlett_ctl_get, + .put = scarlett_ctl_put, + .private_value = 6, /* max value */ + .tlv = { .p = db_scale_scarlett_gain } +}; + +static struct snd_kcontrol_new usb_scarlett_ctl_enum = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = scarlett_ctl_enum_info, + .get = scarlett_ctl_enum_get, + .put = scarlett_ctl_enum_put, +}; + +static struct snd_kcontrol_new usb_scarlett_ctl_sync = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .name = "", + .info = scarlett_ctl_enum_info, + .get = scarlett_ctl_meter_get, +}; + +#ifdef WITH_METER +#ifdef WITH_LOGSCALEMETER +static const DECLARE_TLV_DB_SCALE(db_scale_scarlett_peak, -9700, 50, 0); +#endif + +static struct snd_kcontrol_new usb_scarlett_ctl_meter = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, +#ifdef WITH_LOGSCALEMETER + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .tlv = { .p = db_scale_scarlett_peak }, +#else + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, +#endif + .name = "", + .info = scarlett_ctl_meter_info, + .get = scarlett_ctl_meter_get, +}; +#endif + +static struct snd_kcontrol_new usb_scarlett_ctl_save = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = scarlett_ctl_enum_info, + .get = scarlett_ctl_save_get, + .get = scarlett_ctl_save_put, +}; + +static int add_new_ctl(struct usb_mixer_interface *mixer, + const struct snd_kcontrol_new *ncontrol, + int index, int offset, int num, + int val_len, int count, const char *name, + const struct scarlett_enum_info *opt, + struct scarlett_mixer_elem_info **elem_ret) +{ + struct snd_kcontrol *kctl; + struct scarlett_mixer_elem_info *elem; + int err; + + elem = kzalloc(sizeof(*elem), GFP_KERNEL); + if (!elem) + return -ENOMEM; + + elem->mixer = mixer; + elem->wValue = (offset << 8) | num; + elem->index = index; + elem->val_len = val_len; + elem->count = count; + elem->opt = opt; + + kctl = snd_ctl_new1(ncontrol, elem); + if (!kctl) { + snd_printk(KERN_ERR "cannot malloc kcontrol\n"); + kfree(elem); + return -ENOMEM; + } + kctl->private_free = scarlett_mixer_elem_free; + + snprintf(kctl->id.name, sizeof(kctl->id.name), "%s", name); + + err = snd_ctl_add(mixer->chip->card, kctl); + if (err < 0) + return err; + + if (elem_ret) + *elem_ret = elem; + + return 0; +} + +static int init_ctl(struct scarlett_mixer_elem_info *elem, int value) +{ + int err, channel; + + for (channel = 0; channel < elem->count; channel++) { + err = set_ctl_value(elem, channel, value); + if (err < 0) + return err; + } + + return 0; +} + +#define INIT(value) \ + do { \ + err = init_ctl(elem, value); \ + if (err < 0) \ + return err; \ + } while (0) + +#define CTL_SWITCH(cmd, off, no, count, name) \ + do { \ + err = add_new_ctl(mixer, &usb_scarlett_ctl_switch, cmd, off, no, 2, count, name, NULL, &elem); \ + if (err < 0) \ + return err; \ + } while (0) + +/* no multichannel enum, always count == 1 (at least for now) */ +#define CTL_ENUM(cmd, off, no, name, opt) \ + do { \ + err = add_new_ctl(mixer, &usb_scarlett_ctl_enum, cmd, off, no, 2, 1, name, opt, &elem); \ + if (err < 0) \ + return err; \ + } while (0) + +#define CTL_MIXER(cmd, off, no, count, name) \ + do { \ + err = add_new_ctl(mixer, &usb_scarlett_ctl, cmd, off, no, 2, count, name, NULL, &elem); \ + if (err < 0) \ + return err; \ + INIT(-32768); /* -128*256 */ \ + } while (0) + +#define CTL_MASTER(cmd, off, no, count, name) \ + do { \ + err = add_new_ctl(mixer, &usb_scarlett_ctl_master, cmd, off, no, 2, count, name, NULL, &elem); \ + if (err < 0) \ + return err; \ + INIT(0); \ + } while (0) + +#define CTL_PEAK(cmd, off, no, count, name) /* but UAC2_CS_MEM */ \ + do { \ + err = add_new_ctl(mixer, &usb_scarlett_ctl_meter, cmd, off, no, 2, count, name, NULL, NULL); \ + if (err < 0) \ + return err; \ + } while (0) + +static int add_output_ctls(struct usb_mixer_interface *mixer, + int index, const char *name, + const struct scarlett_device_info *info) +{ + int err; + char mx[45]; + struct scarlett_mixer_elem_info *elem; + + snprintf(mx, 45, "Master %d (%s) Playback Switch", index+1, name); /* mute */ + CTL_SWITCH(0x0a, 0x01, 2*index+1, 2, mx); + + snprintf(mx, 45, "Master %d (%s) Playback Volume", index+1, name); + CTL_MASTER(0x0a, 0x02, 2*index+1, 2, mx); + + snprintf(mx, 45, "Master %dL (%s) Source Playback Enum", index+1, name); + CTL_ENUM(0x33, 0x00, 2*index, mx, &info->opt_master); + INIT(info->mix_start); + + snprintf(mx, 45, "Master %dR (%s) Source Playback Enum", index+1, name); + CTL_ENUM(0x33, 0x00, 2*index+1, mx, &info->opt_master); + INIT(info->mix_start + 1); + + return 0; +} + +#define CTLS_OUTPUT(index, name) \ + do { \ + err = add_output_ctls(mixer, index, name, info); \ + if (err < 0) \ + return err; \ + while (0) + + +/********************** device-specific config *************************/ +static int scarlet_s6i6_controls(struct usb_mixer_interface *mixer, + const struct scarlett_device_info *info) +{ + struct scarlett_mixer_elem_info *elem; + int err; + + CTLS_OUTPUT(0, "Monitor"); + CTLS_OUTPUT(1, "Headphone 2"); + CTLS_OUTPUT(2, "SPDIF"); + + CTL_ENUM(0x01, 0x09, 1, "Input 1 Impedance Switch", &opt_impedance); + CTL_ENUM(0x01, 0x0b, 1, "Input 1 Pad Switch", &opt_pad); + + CTL_ENUM(0x01, 0x09, 2, "Input 2 Impedance Switch", &opt_impedance); + CTL_ENUM(0x01, 0x0b, 2, "Input 2 Pad Switch", &opt_pad); + + CTL_ENUM(0x01, 0x0b, 3, "Input 3 Pad Switch", &opt_pad); + CTL_ENUM(0x01, 0x0b, 4, "Input 4 Pad Switch", &opt_pad); + + return 0; +} + +static int scarlet_s8i6_controls(struct usb_mixer_interface *mixer, + const struct scarlett_device_info *info) +{ + struct scarlett_mixer_elem_info *elem; + int err; + + CTLS_OUTPUT(0, "Monitor"); + CTLS_OUTPUT(1, "Headphone"); + CTLS_OUTPUT(2, "SPDIF"); + + CTL_ENUM(0x01, 0x09, 1, "Input 1 Impedance Switch", &opt_impedance); + CTL_ENUM(0x01, 0x09, 2, "Input 2 Impedance Switch", &opt_impedance); + + CTL_ENUM(0x01, 0x0b, 3, "Input 3 Pad Switch", &opt_pad); + CTL_ENUM(0x01, 0x0b, 4, "Input 4 Pad Switch", &opt_pad); + + return 0; +} + +static int scarlet_s18i6_controls(struct usb_mixer_interface *mixer, + const struct scarlett_device_info *info) +{ + struct scarlett_mixer_elem_info *elem; + int err; + + CTLS_OUTPUT(0, "Monitor"); + CTLS_OUTPUT(1, "Headphone"); + CTLS_OUTPUT(2, "SPDIF"); + + CTL_ENUM(0x01, 0x09, 1, "Input 1 Impedance Switch", &opt_impedance); + CTL_ENUM(0x01, 0x09, 2, "Input 2 Impedance Switch", &opt_impedance); + + return 0; +} + +static int scarlet_s18i8_controls(struct usb_mixer_interface *mixer, + const struct scarlett_device_info *info) +{ + struct scarlett_mixer_elem_info *elem; + int err; + + CTLS_OUTPUT(0, "Monitor"); + CTLS_OUTPUT(1, "Headphone 1"); + CTLS_OUTPUT(2, "Headphone 2"); + CTLS_OUTPUT(3, "SPDIF"); + + CTL_ENUM(0x01, 0x09, 1, "Input 1 Impedance Switch", &opt_impedance); + CTL_ENUM(0x01, 0x0b, 1, "Input 1 Pad Switch", &opt_pad); + + CTL_ENUM(0x01, 0x09, 2, "Input 2 Impedance Switch", &opt_impedance); + CTL_ENUM(0x01, 0x0b, 2, "Input 2 Pad Switch", &opt_pad); + + CTL_ENUM(0x01, 0x0b, 3, "Input 3 Pad Switch", &opt_pad); + CTL_ENUM(0x01, 0x0b, 4, "Input 4 Pad Switch", &opt_pad); + + return 0; +} + +static int scarlet_s18i20_controls(struct usb_mixer_interface *mixer, + const struct scarlett_device_info *info) +{ +/* struct scarlett_mixer_elem_info *elem; */ + int err; + + CTLS_OUTPUT(0, "Monitor"); /* 1/2 */ + CTLS_OUTPUT(1, "Line 3/4"); + CTLS_OUTPUT(2, "Line 5/6"); + CTLS_OUTPUT(3, "Line 7/8"); /* = Headphone 1 */ + CTLS_OUTPUT(4, "Line 9/10"); /* = Headphone 2 */ + CTLS_OUTPUT(5, "SPDIF"); + CTLS_OUTPUT(6, "ADAT 1/2"); + CTLS_OUTPUT(7, "ADAT 3/4"); + CTLS_OUTPUT(8, "ADAT 5/6"); + CTLS_OUTPUT(9, "ADAT 7/8"); + +/* ? real hardware switches + CTL_ENUM (0x01, 0x09, 1, "Input 1 Impedance Switch", &opt_impedance); + CTL_ENUM (0x01, 0x0b, 1, "Input 1 Pad Switch", &opt_pad); + + CTL_ENUM (0x01, 0x09, 2, "Input 2 Impedance Switch", &opt_impedance); + CTL_ENUM (0x01, 0x0b, 2, "Input 2 Pad Switch", &opt_pad); + + CTL_ENUM (0x01, 0x0b, 3, "Input 3 Pad Switch", &opt_pad); + CTL_ENUM (0x01, 0x0b, 4, "Input 4 Pad Switch", &opt_pad); +*/ + + return 0; +} + +static const char * const s6i6_texts[] = { + txtOff, /* 'off' == 0xff */ + txtPcm1, txtPcm2, txtPcm3, txtPcm4, + txtPcm5, txtPcm6, txtPcm7, txtPcm8, + txtPcm9, txtPcm10, txtPcm11, txtPcm12, + txtAnlg1, txtAnlg2, txtAnlg3, txtAnlg4, + txtSpdif1, txtSpdif2, + txtMix1, txtMix2, txtMix3, txtMix4, + txtMix5, txtMix6, txtMix7, txtMix8 +}; + +/* untested... */ +static const struct scarlett_device_info s6i6_info = { + .matrix_in = 18, + .matrix_out = 8, + .input_len = 6, + .output_len = 6, + + .pcm_start = 0, + .analog_start = 12, + .spdif_start = 16, + .adat_start = 18, + .mix_start = 18, + + .opt_master = { + .start = -1, + .len = 27, + .texts = s6i6_texts + }, + + .opt_matrix = { + .start = -1, + .len = 19, + .texts = s6i6_texts + }, + + .controls_fn = scarlet_s6i6_controls, + .matrix_mux_init = { + 12, 13, 14, 15, /* Analog -> 1..4 */ + 16, 17, /* SPDIF -> 5,6 */ + 0, 1, 2, 3, 4, 5, 6, 7, /* PCM[1..12] -> 7..18 */ + 8, 9, 10, 11 + } +}; + +/* and 2 loop channels: Mix1, Mix2 */ +static const char * const s8i6_texts[] = { + txtOff, /* 'off' == 0xff */ + txtPcm1, txtPcm2, txtPcm3, txtPcm4, + txtPcm5, txtPcm6, txtPcm7, txtPcm8, + txtPcm9, txtPcm10, txtPcm11, txtPcm12, + txtAnlg1, txtAnlg2, txtAnlg3, txtAnlg4, + txtSpdif1, txtSpdif2, + txtMix1, txtMix2, txtMix3, txtMix4, + txtMix5, txtMix6 +}; + +/* untested... */ +static const struct scarlett_device_info s8i6_info = { + .matrix_in = 18, + .matrix_out = 6, + .input_len = 8, + .output_len = 6, + + .pcm_start = 0, + .analog_start = 12, + .spdif_start = 16, + .adat_start = 18, + .mix_start = 18, + + .opt_master = { + .start = -1, + .len = 25, + .texts = s8i6_texts + }, + + .opt_matrix = { + .start = -1, + .len = 19, + .texts = s8i6_texts + }, + + .controls_fn = scarlet_s8i6_controls, + .matrix_mux_init = { + 12, 13, 14, 15, /* Analog -> 1..4 */ + 16, 17, /* SPDIF -> 5,6 */ + 0, 1, 2, 3, 4, 5, 6, 7, /* PCM[1..12] -> 7..18 */ + 8, 9, 10, 11 + } +}; + +static const char * const s18i6_texts[] = { + txtOff, /* 'off' == 0xff */ + txtPcm1, txtPcm2, txtPcm3, txtPcm4, + txtPcm5, txtPcm6, + txtAnlg1, txtAnlg2, txtAnlg3, txtAnlg4, + txtAnlg5, txtAnlg6, txtAnlg7, txtAnlg8, + txtSpdif1, txtSpdif2, + txtAdat1, txtAdat2, txtAdat3, txtAdat4, + txtAdat5, txtAdat6, txtAdat7, txtAdat8, + txtMix1, txtMix2, txtMix3, txtMix4, + txtMix5, txtMix6 +}; + +static const struct scarlett_device_info s18i6_info = { + .matrix_in = 18, + .matrix_out = 6, + .input_len = 18, + .output_len = 6, + + .pcm_start = 0, + .analog_start = 6, + .spdif_start = 14, + .adat_start = 16, + .mix_start = 24, + + .opt_master = { + .start = -1, + .len = 31, + .texts = s18i6_texts + }, + + .opt_matrix = { + .start = -1, + .len = 25, + .texts = s18i6_texts + }, + + .controls_fn = scarlet_s18i6_controls, + .matrix_mux_init = { + 6, 7, 8, 9, 10, 11, 12, 13, /* Analog -> 1..8 */ + 16, 17, 18, 19, 20, 21, /* ADAT[1..6] -> 9..14 */ + 14, 15, /* SPDIF -> 15,16 */ + 0, 1 /* PCM[1,2] -> 17,18 */ + } +}; + +static const char * const s18i8_texts[] = { + txtOff, /* 'off' == 0xff (original software: 0x22) */ + txtPcm1, txtPcm2, txtPcm3, txtPcm4, + txtPcm5, txtPcm6, txtPcm7, txtPcm8, + txtAnlg1, txtAnlg2, txtAnlg3, txtAnlg4, + txtAnlg5, txtAnlg6, txtAnlg7, txtAnlg8, + txtSpdif1, txtSpdif2, + txtAdat1, txtAdat2, txtAdat3, txtAdat4, + txtAdat5, txtAdat6, txtAdat7, txtAdat8, + txtMix1, txtMix2, txtMix3, txtMix4, + txtMix5, txtMix6, txtMix7, txtMix8 +}; + +static const struct scarlett_device_info s18i8_info = { + .matrix_in = 18, + .matrix_out = 8, + .input_len = 18, + .output_len = 8, + + .pcm_start = 0, + .analog_start = 8, + .spdif_start = 16, + .adat_start = 18, + .mix_start = 26, + + .opt_master = { + .start = -1, + .len = 35, + .texts = s18i8_texts + }, + + .opt_matrix = { + .start = -1, + .len = 27, + .texts = s18i8_texts + }, + + .controls_fn = scarlet_s18i8_controls, + .matrix_mux_init = { + 8, 9, 10, 11, 12, 13, 14, 15, /* Analog -> 1..8 */ + 18, 19, 20, 21, 22, 23, /* ADAT[1..6] -> 9..14 */ + 16, 17, /* SPDIF -> 15,16 */ + 0, 1 /* PCM[1,2] -> 17,18 */ + } +}; + +static const char * const s18i20_texts[] = { + txtOff, /* 'off' == 0xff (original software: 0x22) */ + txtPcm1, txtPcm2, txtPcm3, txtPcm4, + txtPcm5, txtPcm6, txtPcm7, txtPcm8, + txtPcm9, txtPcm10, txtPcm11, txtPcm12, + txtPcm13, txtPcm14, txtPcm15, txtPcm16, + txtPcm17, txtPcm18, txtPcm19, txtPcm20, + txtAnlg1, txtAnlg2, txtAnlg3, txtAnlg4, + txtAnlg5, txtAnlg6, txtAnlg7, txtAnlg8, + txtSpdif1, txtSpdif2, + txtAdat1, txtAdat2, txtAdat3, txtAdat4, + txtAdat5, txtAdat6, txtAdat7, txtAdat8, + txtMix1, txtMix2, txtMix3, txtMix4, + txtMix5, txtMix6, txtMix7, txtMix8 +}; + +static const struct scarlett_device_info s18i20_info = { + .matrix_in = 18, + .matrix_out = 8, + .input_len = 18, + .output_len = 20, + + .pcm_start = 0, + .analog_start = 20, + .spdif_start = 28, + .adat_start = 30, + .mix_start = 38, + + .opt_master = { + .start = -1, + .len = 47, + .texts = s18i20_texts + }, + + .opt_matrix = { + .start = -1, + .len = 39, + .texts = s18i20_texts + }, + + .controls_fn = scarlet_s18i20_controls, + .matrix_mux_init = { + 20, 21, 22, 23, 24, 25, 26, 27, /* Analog -> 1..8 */ + 30, 31, 32, 33, 34, 35, /* ADAT[1..6] -> 9..14 */ + 28, 29, /* SPDIF -> 15,16 */ + 0, 1 /* PCM[1,2] -> 17,18 */ + } +}; + +/* +int scarlett_reset(struct usb_mixer_interface *mixer) +{ + TODO? save first-time init flag into device? + + unmute [master +] mixes (switches are currently not initialized) + [set(get!) impedance: 0x01, 0x09, 1..2] + [set(get!) 0x01, 0x08, 3..4] + [set(get!) pad: 0x01, 0x0b, 1..4] + + matrix inputs (currently in scarlett_mixer_controls) +} +*/ + +/* + * Create and initialize a mixer for the Focusrite(R) Scarlett + */ +int scarlett_mixer_controls(struct usb_mixer_interface *mixer) +{ + int err, i, o; + char mx[32]; + const struct scarlett_device_info *info; + struct scarlett_mixer_elem_info *elem; + + CTL_SWITCH(0x0a, 0x01, 0, 1, "Master Playback Switch"); + CTL_MASTER(0x0a, 0x02, 0, 1, "Master Playback Volume"); + + switch (mixer->chip->usb_id) { + case USB_ID(0x1235, 0x8012): + info = &s6i6_info; + break; + case USB_ID(0x1235, 0x8002): + info = &s8i6_info; + break; + case USB_ID(0x1235, 0x8004): + info = &s18i6_info; + break; + case USB_ID(0x1235, 0x8014): + info = &s18i8_info; + break; + case USB_ID(0x1235, 0x800c): + info = &s18i20_info; + break; + default: /* device not (yet) supported */ + return -EINVAL; + } + + err = (*info->controls_fn)(mixer, info); + if (err < 0) + return err; + + for (i = 0; i < info->matrix_in; i++) { + snprintf(mx, 32, "Matrix %02d Input Playback Route", i+1); + CTL_ENUM(0x32, 0x06, i, mx, &info->opt_matrix); + INIT(info->matrix_mux_init[i]); + + for (o = 0; o < info->matrix_out; o++) { + sprintf(mx, "Matrix %02d Mix %c Playback Volume", i+1, o+'A'); + CTL_MIXER(0x3c, 0x00, (i << 3) + (o & 0x07), 1, mx); + if (((o == 0) && (info->matrix_mux_init[i] == info->pcm_start)) || + ((o == 1) && (info->matrix_mux_init[i] == info->pcm_start + 1))) { + INIT(0); /* init hack: enable PCM 1 / 2 on Mix A / B */ + } + } + } + + for (i = 0; i < info->input_len; i++) { + snprintf(mx, 32, "Input Source %02d Capture Route", i+1); + CTL_ENUM(0x34, 0x00, i, mx, &info->opt_master); + INIT(info->analog_start + i); + } + + /* val_len == 1 needed here */ + err = add_new_ctl(mixer, &usb_scarlett_ctl_enum, 0x28, 0x01, 0, 1, + 1, "Sample Clock Source", &opt_clock, NULL); + if (err < 0) + return err; + + /* val_len == 1 and UAC2_CS_MEM */ + err = add_new_ctl(mixer, &usb_scarlett_ctl_sync, 0x3c, 0x00, 2, 1, + 1, "Sample Clock Sync Status", &opt_sync, NULL); + if (err < 0) + return err; + + /* val_len == 1 and UAC2_CS_MEM */ + err = add_new_ctl(mixer, &usb_scarlett_ctl_save, 0x3c, 0x00, 0x5a, 1, + 1, "Save To HW", &opt_save, NULL); + if (err < 0) + return err; + +#ifdef WITH_METER + CTL_PEAK(0x3c, 0x00, 0, info->input_len, "Input Meter"); + CTL_PEAK(0x3c, 0x00, 1, info->matrix_out, "Matrix Meter"); + CTL_PEAK(0x3c, 0x00, 3, info->output_len, "PCM Meter"); +#endif + + /* initialize sampling rate to 48000 */ + err = set_ctl_urb2(mixer->chip, UAC2_CS_CUR, 0x0100, 0x29, "\x80\xbb\x00\x00", 4); + if (err < 0) + return err; + +/* TODO(?) scarlett_reset(mixer); */ + + return 0; +} diff --git a/sound/usb/scarlettmixer.h b/sound/usb/scarlettmixer.h new file mode 100644 index 0000000..ebde17e --- /dev/null +++ b/sound/usb/scarlettmixer.h @@ -0,0 +1,6 @@ +#ifndef __USBSCARLETTMIXER_H +#define __USBSCARLETTMIXER_H + +int scarlett_mixer_controls(struct usb_mixer_interface *mixer); + +#endif /* __USBSCARLETTMIXER_H */ -- 1.9.1

6 15

[alsa-devel] [PATCH] ASoC: rt5677: Add option to configure gpio as floating/pullup/pulldown
by Anatol Pomozov 13 Oct '14

13 Oct '14

gpio_config is array of 6 elements that allows to set GPIO as floating, pullup, pulldown. Sponsored: Google ChromeOS Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- Documentation/devicetree/bindings/sound/rt5677.txt | 7 ++++ include/sound/rt5677.h | 3 ++ sound/soc/codecs/rt5677.c | 39 ++++++++++++++++++++++ 3 files changed, 49 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/rt5677.txt b/Documentation/devicetree/bindings/sound/rt5677.txt index 0701b83..f82f0e9 100644 --- a/Documentation/devicetree/bindings/sound/rt5677.txt +++ b/Documentation/devicetree/bindings/sound/rt5677.txt @@ -27,6 +27,12 @@ Optional properties: Boolean. Indicate MIC1/2 input and LOUT1/2/3 outputs are differential, rather than single-ended. +- realtek,gpio-config + Array of six 8bit elements that configures GPIO. + 0 - floating (reset value) + 1 - pull down + 2 - pull up + Pins on the device (for linking into audio routes): * IN1P @@ -56,4 +62,5 @@ rt5677 { realtek,pow-ldo2-gpio = <&gpio TEGRA_GPIO(V, 3) GPIO_ACTIVE_HIGH>; realtek,in1-differential = "true"; + realtek,gpio-config = /bits/ 8 <0 0 0 0 0 2>; /* pull up GPIO6 */ }; diff --git a/include/sound/rt5677.h b/include/sound/rt5677.h index 082670e..a56b429 100644 --- a/include/sound/rt5677.h +++ b/include/sound/rt5677.h @@ -27,6 +27,9 @@ struct rt5677_platform_data { bool lout3_diff; /* DMIC2 clock source selection */ enum rt5677_dmic2_clk dmic2_clk_pin; + + /* configures GPIO, 0 - floating, 1 - pulldown, 2 - pullup */ + u8 gpio_config[6]; }; #endif diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 16aa4d9..a454df3 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -3309,6 +3309,38 @@ static int rt5677_gpio_direction_in(struct gpio_chip *chip, unsigned offset) return 0; } +/** Configures the gpio as + * 0 - floating + * 1 - pull down + * 2 - pull up + */ +static void rt5677_gpio_config(struct rt5677_priv *rt5677, unsigned offset, + int value) +{ + int shift; + + switch (offset) { + case RT5677_GPIO1 ... RT5677_GPIO2: + shift = 2 * (1 - offset); + regmap_update_bits(rt5677->regmap, + RT5677_PR_BASE + RT5677_DIG_IN_PIN_ST_CTRL2, + 0x3 << shift, + (value & 0x3) << shift); + break; + + case RT5677_GPIO3 ... RT5677_GPIO6: + shift = 2 * (9 - offset); + regmap_update_bits(rt5677->regmap, + RT5677_PR_BASE + RT5677_DIG_IN_PIN_ST_CTRL3, + 0x3 << shift, + (value & 0x3) << shift); + break; + + default: + break; + } +} + static struct gpio_chip rt5677_template_chip = { .label = "rt5677", .owner = THIS_MODULE, @@ -3353,6 +3385,7 @@ static void rt5677_free_gpio(struct i2c_client *i2c) static int rt5677_probe(struct snd_soc_codec *codec) { struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + int i; rt5677->codec = codec; @@ -3371,6 +3404,9 @@ static int rt5677_probe(struct snd_soc_codec *codec) regmap_write(rt5677->regmap, RT5677_DIG_MISC, 0x0020); regmap_write(rt5677->regmap, RT5677_PWR_DSP2, 0x0c00); + for (i = 0; i < RT5677_GPIO_NUM; i++) + rt5677_gpio_config(rt5677, i, rt5677->pdata.gpio_config[i]); + return 0; } @@ -3590,6 +3626,9 @@ static int rt5677_parse_dt(struct rt5677_priv *rt5677, struct device_node *np) (rt5677->pow_ldo2 != -ENOENT)) return rt5677->pow_ldo2; + of_property_read_u8_array(np, "realtek,gpio-config", + rt5677->pdata.gpio_config, RT5677_GPIO_NUM); + return 0; } -- 2.1.2

2 1

[alsa-devel] [PATCH v2] audio: sgtl5000: Add MicBias resistor support in DT
by Jean-Michel Hautbois 13 Oct '14

13 Oct '14

Some systems may require a different resistor than the default one (4K). This adds a property in sgtl5000 codec. It keeps the default of 4K when nothing is specified so it does not break existing code. v2: modify the default case on DT parsing Signed-off-by: Jean-Michel Hautbois <jean-michel.hautbois(a)vodalys.com> --- .../devicetree/bindings/sound/sgtl5000.txt | 10 ++++ sound/soc/codecs/sgtl5000.c | 60 ++++++++++++++++++++-- 2 files changed, 66 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/sgtl5000.txt b/Documentation/devicetree/bindings/sound/sgtl5000.txt index 955df60..dd38f84 100644 --- a/Documentation/devicetree/bindings/sound/sgtl5000.txt +++ b/Documentation/devicetree/bindings/sound/sgtl5000.txt @@ -7,10 +7,20 @@ Required properties: - clocks : the clock provider of SYS_MCLK +- sgtl5000-micbias-resistor : the bias resistor to be used + 1 or SGTL5000_MICBIAS_2K - MICBIAS resistor is set to 2K + 2 or SGTL5000_MICBIAS_4K - MICBIAS resistor is set to 4K + 3 or SGTL5000_MICBIAS_8K - MICBIAS resistor is set to 8K + 0 or SGTL5000_MICBIAS_OFF - MICBIAS resistor is not used + If this node is not mentioned or if the value is unknown, then + micbias resistor is set to 4K. + + Example: codec: sgtl5000@0a { compatible = "fsl,sgtl5000"; reg = <0x0a>; clocks = <&clks 150>; + sgtl5000-micbias-resistor = <1>; }; diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 6bb77d7..a255754 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -121,6 +121,13 @@ struct ldo_regulator { bool enabled; }; +enum sgtl5000_micbias_resistor { + SGTL5000_MICBIAS_OFF = 0, + SGTL5000_MICBIAS_2K = 1, + SGTL5000_MICBIAS_4K = 2, + SGTL5000_MICBIAS_8K = 3, +}; + /* sgtl5000 private structure in codec */ struct sgtl5000_priv { int sysclk; /* sysclk rate */ @@ -131,6 +138,7 @@ struct sgtl5000_priv { struct regmap *regmap; struct clk *mclk; int revision; + enum sgtl5000_micbias_resistor micbias_resistor; }; /* @@ -145,12 +153,14 @@ struct sgtl5000_priv { static int mic_bias_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(w->codec); + switch (event) { case SND_SOC_DAPM_POST_PMU: - /* change mic bias resistor to 4Kohm */ + /* change mic bias resistor */ snd_soc_update_bits(w->codec, SGTL5000_CHIP_MIC_CTRL, - SGTL5000_BIAS_R_MASK, - SGTL5000_BIAS_R_4k << SGTL5000_BIAS_R_SHIFT); + SGTL5000_BIAS_R_MASK, + sgtl5000->micbias_resistor << SGTL5000_BIAS_R_SHIFT); break; case SND_SOC_DAPM_PRE_PMD: @@ -1326,7 +1336,16 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) SGTL5000_HP_ZCD_EN | SGTL5000_ADC_ZCD_EN); - snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 2); + switch (sgtl5000->micbias_resistor) { + case SGTL5000_MICBIAS_2K: + case SGTL5000_MICBIAS_4K: + case SGTL5000_MICBIAS_8K: + snd_soc_update_bits(codec, SGTL5000_CHIP_MIC_CTRL, + SGTL5000_BIAS_R_MASK, + sgtl5000->micbias_resistor << SGTL5000_BIAS_R_SHIFT); + default: + break; + } /* * disable DAP @@ -1418,6 +1437,8 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, struct sgtl5000_priv *sgtl5000; int ret, reg, rev; unsigned int mclk; + struct device_node *np = client->dev.of_node; + u32 value; sgtl5000 = devm_kzalloc(&client->dev, sizeof(struct sgtl5000_priv), GFP_KERNEL); @@ -1470,6 +1491,37 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, dev_info(&client->dev, "sgtl5000 revision 0x%x\n", rev); sgtl5000->revision = rev; + if (np) { + if (!of_property_read_u32(np, + "sgtl5000-micbias-resistor", &value)) { + switch (value) { + case 0: + sgtl5000->micbias_resistor = + SGTL5000_MICBIAS_OFF; + break; + case 1: + sgtl5000->micbias_resistor = + SGTL5000_MICBIAS_2K; + break; + case 2: + sgtl5000->micbias_resistor = + SGTL5000_MICBIAS_4K; + break; + case 3: + sgtl5000->micbias_resistor = + SGTL5000_MICBIAS_8K; + break; + default: + sgtl5000->micbias_resistor = + SGTL5000_MICBIAS_4K; + dev_err(&client->dev, + "Unsuitable MicBias resistor\n"); + } + } else { + sgtl5000->micbias_resistor = SGTL5000_MICBIAS_4K; + } + } + i2c_set_clientdata(client, sgtl5000); /* Ensure sgtl5000 will start with sane register values */ -- 2.1.1

3 3

[alsa-devel] Blackmagic card used in Linux, cause static white noise, but no noise when OSX 10.9.4 and Windows 8.1 is used.
by Shamun Toha Md 13 Oct '14

13 Oct '14

Dear Experts, I am using more then 3 years Blackmagic Intensity pro PCI-Express capture card with Linux, Mac OSX, Windows operating system. While using the card with Linux OS (Ubuntu 11, 12, 13, 14.04) and Blackmagic latest drivers i have constantly this white static noise when i use minijack microphones. But when i use external USB XLR microphone it get resolved. With same hardware when i use Windows 8.1 OS (not Linux), with same Blackmagic card, same microphone, same PC i have not a single white static noise which i noticed while using Linux. Same issue i simulated with OSX 10.9.4 and Windows 8.1, where i do not have white static noise, but Linux when used with Blackmagic intensity pro then there is this constant white static noise. I had this problem about 3 years now, and still no solution unless i move to Windows 8.1 or OSX. Please kindly advise. Thank you /Sha

1 0

[alsa-devel] [PATCH] bebob: Fix failure to detect source of clock for Terratec Phase 88
by Takashi Sakamoto 13 Oct '14

13 Oct '14

This patch fixes a failure to open PCM device with -ENOSYS in Terratec Phase 88. Terratec Phase 88 has two Selector Function Blocks of AVC Audio subunit to switch source of clock. One is to switch internal/external for the source and another is to switch word/spdif for the external clock. The IDs for these Selector Function Blocks are 9 and 8. But in current implementation they're 0 and 0. Reported-by: András Murányi <muranyia(a)gmail.com> Tested-by: András Murányi <muranyia(a)gmail.com> Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/bebob/bebob_terratec.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/firewire/bebob/bebob_terratec.c b/sound/firewire/bebob/bebob_terratec.c index eef8ea7..0e4c0bf 100644 --- a/sound/firewire/bebob/bebob_terratec.c +++ b/sound/firewire/bebob/bebob_terratec.c @@ -17,10 +17,10 @@ phase88_rack_clk_src_get(struct snd_bebob *bebob, unsigned int *id) unsigned int enable_ext, enable_word; int err; - err = avc_audio_get_selector(bebob->unit, 0, 0, &enable_ext); + err = avc_audio_get_selector(bebob->unit, 0, 9, &enable_ext); if (err < 0) goto end; - err = avc_audio_get_selector(bebob->unit, 0, 0, &enable_word); + err = avc_audio_get_selector(bebob->unit, 0, 8, &enable_word); if (err < 0) goto end; -- 1.9.1

2 5

[alsa-devel] [PATCH v3] ES938 support for ES18xx driver
by Ondrej Zary 13 Oct '14

13 Oct '14

Add support for ES938 3-D Audio Effects Processor found on some ES18xx (and possibly other) sound cards, doing bass/treble and 3D control. ES938 is controlled by MIDI SysEx commands sent through card's MPU401 port. The code opens/closes the MIDI device everytime a mixer control value is changed so userspace apps can still use the MIDI port. Changing the mixer controls will fail when the MIDI port is open by an application. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- Changes in v3: - constify buf argument to snd_rawmidi_kernel_write - remove double if from probe - spelling fix Changes in v2: - debug message when ES938 detected - reworked sysex messages to use structs - ktime instead of jiffies for timeout --- sound/isa/Kconfig | 4 + sound/isa/Makefile | 2 + sound/isa/es18xx.c | 13 +++- sound/isa/es938.c | 217 ++++++++++++++++++++++++++++++++++++++++++++++++++++ sound/isa/es938.h | 48 ++++++++++++ 5 files changed, 283 insertions(+), 1 deletion(-) create mode 100644 sound/isa/es938.c create mode 100644 sound/isa/es938.h diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 0216475..db0b678 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -17,6 +17,9 @@ config SND_SB16_DSP select SND_PCM select SND_SB_COMMON +config SND_ES938 + tristate + menuconfig SND_ISA bool "ISA sound devices" depends on ISA && ISA_DMA_API @@ -183,6 +186,7 @@ config SND_ES18XX select SND_OPL3_LIB select SND_MPU401_UART select SND_PCM + select SND_ES938 help Say Y here to include support for ESS AudioDrive ES18xx chips. diff --git a/sound/isa/Makefile b/sound/isa/Makefile index 9a15f14..d59e0bf 100644 --- a/sound/isa/Makefile +++ b/sound/isa/Makefile @@ -8,6 +8,7 @@ snd-als100-objs := als100.o snd-azt2320-objs := azt2320.o snd-cmi8328-objs := cmi8328.o snd-cmi8330-objs := cmi8330.o +snd-es938-objs := es938.o snd-es18xx-objs := es18xx.o snd-opl3sa2-objs := opl3sa2.o snd-sc6000-objs := sc6000.o @@ -19,6 +20,7 @@ obj-$(CONFIG_SND_ALS100) += snd-als100.o obj-$(CONFIG_SND_AZT2320) += snd-azt2320.o obj-$(CONFIG_SND_CMI8328) += snd-cmi8328.o obj-$(CONFIG_SND_CMI8330) += snd-cmi8330.o +obj-$(CONFIG_SND_ES938) += snd-es938.o obj-$(CONFIG_SND_ES18XX) += snd-es18xx.o obj-$(CONFIG_SND_OPL3SA2) += snd-opl3sa2.o obj-$(CONFIG_SND_SC6000) += snd-sc6000.o diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index 6faaac6..72dad69 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -96,6 +96,7 @@ #define SNDRV_LEGACY_FIND_FREE_IRQ #define SNDRV_LEGACY_FIND_FREE_DMA #include <sound/initval.h> +#include "es938.h" #define PFX "es18xx: " @@ -122,6 +123,7 @@ struct snd_es18xx { struct snd_pcm_substream *playback_b_substream; struct snd_rawmidi *rmidi; + struct snd_es938 es938; struct snd_kcontrol *hw_volume; struct snd_kcontrol *hw_switch; @@ -2167,7 +2169,16 @@ static int snd_audiodrive_probe(struct snd_card *card, int dev) return err; } - return snd_card_register(card); + err = snd_card_register(card); + if (err < 0) + return err; + + /* no error if this fails because ES938 is optional */ + if (mpu_port[dev] > 0 && mpu_port[dev] != SNDRV_AUTO_PORT && + snd_es938_init(&chip->es938, card, 0, 0) == 0) + snd_printk(KERN_DEBUG "found ES938 audio processor\n"); + + return 0; } static int snd_es18xx_isa_match(struct device *pdev, unsigned int dev) diff --git a/sound/isa/es938.c b/sound/isa/es938.c new file mode 100644 index 0000000..75d1fbe --- /dev/null +++ b/sound/isa/es938.c @@ -0,0 +1,217 @@ +/* + * Driver for ESS ES938 3-D Audio Effects Processor + * Copyright (c) 2014 Ondrej Zary + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include <linux/module.h> +#include <sound/asoundef.h> +#include <sound/control.h> +#include <sound/core.h> +#include <sound/rawmidi.h> +#include "es938.h" + +MODULE_AUTHOR("Ondrej Zary"); +MODULE_DESCRIPTION("ESS ES938"); +MODULE_LICENSE("GPL"); + +static int snd_es938_read_reg(struct snd_es938 *chip, u8 reg, u8 *out) +{ + int i = 0, res; + const struct snd_es938_sysex_reg req = { + .midi_cmd = MIDI_CMD_COMMON_SYSEX, + .id = ES938_ID, + .cmd = ES938_CMD_REG_R, + .reg = reg, + .midi_end = MIDI_CMD_COMMON_SYSEX_END, + }; + struct snd_es938_sysex_regval reply = { }; + u8 *p = (void *)&reply; + ktime_t end_time; + + snd_rawmidi_kernel_write(chip->rfile.output, + (const void *)&req, sizeof(req)); + + end_time = ktime_add_ms(ktime_get(), 100); + while (i < sizeof(reply)) { + res = snd_rawmidi_kernel_read(chip->rfile.input, p + i, + sizeof(reply) - i); + if (res > 0) + i += res; + if (ktime_after(ktime_get(), end_time)) + return -1; + } + + /* check if the reply is ours and has SYSEX_END at the end */ + if (memcmp(&reply, &req, sizeof(req) - 1) || + reply.midi_end != MIDI_CMD_COMMON_SYSEX_END) + return -1; + + if (out) + *out = reply.val; + + return 0; +} + +static void snd_es938_write_reg(struct snd_es938 *chip, u8 reg, u8 val) +{ + const struct snd_es938_sysex_regval req = { + .midi_cmd = MIDI_CMD_COMMON_SYSEX, + .id = ES938_ID, + .cmd = ES938_CMD_REG_W, + .reg = reg, + .val = val, + .midi_end = MIDI_CMD_COMMON_SYSEX_END, + }; + + snd_rawmidi_kernel_write(chip->rfile.output, + (const void *)&req, sizeof(req)); + chip->regs[reg] = val; +} + +#define ES938_MIXER(xname, xindex, reg, shift, mask) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ +.info = snd_es938_info_mixer, \ +.get = snd_es938_get_mixer, .put = snd_es938_put_mixer, \ +.private_value = reg | (shift << 8) | (mask << 16) } + +#define ES938_MIXER_TLV(xname, xindex, reg, shift, mask, xtlv) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ +.access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ +.info = snd_es938_info_mixer, \ +.get = snd_es938_get_mixer, .put = snd_es938_put_mixer, \ +.private_value = reg | (shift << 8) | (mask << 16), \ +.tlv = { .p = (xtlv) } } + +static const DECLARE_TLV_DB_SCALE(db_scale_tone, -900, 300, 0); + +static struct snd_kcontrol_new snd_es938_controls[] = { +ES938_MIXER_TLV("Tone Control - Bass", 0, ES938_REG_TONE, 0, 7, db_scale_tone), +ES938_MIXER_TLV("Tone Control - Treble", 0, ES938_REG_TONE, 4, 7, db_scale_tone), +ES938_MIXER("3D Control - Level", 0, ES938_REG_SPATIAL, 1, 63), +ES938_MIXER("3D Control - Switch", 0, ES938_REG_SPATIAL_EN, 0, 1), +}; + +int snd_es938_init(struct snd_es938 *chip, struct snd_card *card, int device, + int subdevice) +{ + int ret, i; + + ret = snd_rawmidi_kernel_open(card, device, subdevice, + SNDRV_RAWMIDI_LFLG_OPEN | SNDRV_RAWMIDI_LFLG_APPEND, + &chip->rfile); + if (ret < 0) { + snd_printk(KERN_WARNING "es938: unable to open MIDI device\n"); + return ret; + } + + /* try to read a register to detect chip presence */ + if (snd_es938_read_reg(chip, ES938_REG_MISC, NULL) < 0) { + ret = -ENODEV; + goto err; + } + + /* write default values (there's no reset) */ + snd_es938_write_reg(chip, ES938_REG_MISC, 0x49); + snd_es938_write_reg(chip, ES938_REG_TONE, 0x33); + /* reserved bit 0 must be set for 3D to work */ + snd_es938_write_reg(chip, ES938_REG_SPATIAL, 0x01); + /* datasheet specifies invalid value 0x00 as default */ + snd_es938_write_reg(chip, ES938_REG_SPATIAL_EN, 0x02); + snd_es938_write_reg(chip, ES938_REG_POWER, 0x0e); + + strlcat(card->mixername, " + ES938", sizeof(card->mixername)); + for (i = 0; i < ARRAY_SIZE(snd_es938_controls); i++) { + ret = snd_ctl_add(card, + snd_ctl_new1(&snd_es938_controls[i], chip)); + if (ret < 0) + goto err; + } + + chip->card = card; + chip->device = device; + chip->subdevice = subdevice; +err: + snd_rawmidi_kernel_release(&chip->rfile); + + return ret; +} +EXPORT_SYMBOL(snd_es938_init); + +int snd_es938_info_mixer(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + int mask = (kcontrol->private_value >> 16) & 0xff; + + uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : + SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = mask; + return 0; +} +EXPORT_SYMBOL(snd_es938_info_mixer); + +int snd_es938_get_mixer(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_es938 *chip = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int shift = (kcontrol->private_value >> 8) & 0xff; + int mask = (kcontrol->private_value >> 16) & 0xff; + u8 val = chip->regs[reg]; + + ucontrol->value.integer.value[0] = (val >> shift) & mask; + return 0; +} +EXPORT_SYMBOL(snd_es938_get_mixer); + +int snd_es938_put_mixer(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_es938 *chip = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int shift = (kcontrol->private_value >> 8) & 0xff; + int mask = (kcontrol->private_value >> 16) & 0xff; + u8 val = ucontrol->value.integer.value[0] & mask; + u8 oldval = chip->regs[reg]; + u8 newval; + int err; + + mask <<= shift; + val <<= shift; + + newval = (oldval & ~mask) | val; + if (newval == oldval) + return 0; + + /* this will fail if the MIDI port is used by an application */ + err = snd_rawmidi_kernel_open(chip->card, chip->device, + chip->subdevice, + SNDRV_RAWMIDI_LFLG_OPEN | SNDRV_RAWMIDI_LFLG_APPEND, + &chip->rfile); + if (err < 0) + return err; + + snd_es938_write_reg(chip, reg, newval); + + snd_rawmidi_kernel_release(&chip->rfile); + + return 1; +} +EXPORT_SYMBOL(snd_es938_put_mixer); diff --git a/sound/isa/es938.h b/sound/isa/es938.h new file mode 100644 index 0000000..08148ff --- /dev/null +++ b/sound/isa/es938.h @@ -0,0 +1,48 @@ +#include <sound/tlv.h> + +#define ES938_ID 0x7b + +#define ES938_CMD_REG_R 0x7e +#define ES938_CMD_REG_W 0x7f + +#define ES938_REG_MISC 0 +#define ES938_REG_TONE 1 +#define ES938_REG_SPATIAL 5 +#define ES938_REG_SPATIAL_EN 6 +#define ES938_REG_POWER 7 + +struct snd_es938 { + u8 regs[8]; + struct snd_card *card; + int device; + int subdevice; + struct snd_rawmidi_file rfile; +}; + +struct snd_es938_sysex_reg { + u8 midi_cmd; + u8 zeros[2]; + u8 id; + u8 cmd; + u8 reg; + u8 midi_end; +}; + +struct snd_es938_sysex_regval { + u8 midi_cmd; + u8 zeros[2]; + u8 id; + u8 cmd; + u8 reg; + u8 val; + u8 midi_end; +}; + +int snd_es938_init(struct snd_es938 *chip, struct snd_card *card, int device, + int subdevice); +int snd_es938_info_mixer(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo); +int snd_es938_get_mixer(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_es938_put_mixer(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); -- Ondrej Zary

5 10

[alsa-devel] [PATCH] audio: sgtl5000: Add MicBias resistor support in DT
by Jean-Michel Hautbois 13 Oct '14

13 Oct '14

Some systems may require a different resistor than the default one (4K). This adds a property in sgtl5000 codec. It keeps the default of 4K when nothing is specified so it does not break existing code. Signed-off-by: Jean-Michel Hautbois <jean-michel.hautbois(a)vodalys.com> --- .../devicetree/bindings/sound/sgtl5000.txt | 10 ++++ sound/soc/codecs/sgtl5000.c | 56 ++++++++++++++++++++-- 2 files changed, 62 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/sgtl5000.txt b/Documentation/devicetree/bindings/sound/sgtl5000.txt index 955df60..dd38f84 100644 --- a/Documentation/devicetree/bindings/sound/sgtl5000.txt +++ b/Documentation/devicetree/bindings/sound/sgtl5000.txt @@ -7,10 +7,20 @@ Required properties: - clocks : the clock provider of SYS_MCLK +- sgtl5000-micbias-resistor : the bias resistor to be used + 1 or SGTL5000_MICBIAS_2K - MICBIAS resistor is set to 2K + 2 or SGTL5000_MICBIAS_4K - MICBIAS resistor is set to 4K + 3 or SGTL5000_MICBIAS_8K - MICBIAS resistor is set to 8K + 0 or SGTL5000_MICBIAS_OFF - MICBIAS resistor is not used + If this node is not mentioned or if the value is unknown, then + micbias resistor is set to 4K. + + Example: codec: sgtl5000@0a { compatible = "fsl,sgtl5000"; reg = <0x0a>; clocks = <&clks 150>; + sgtl5000-micbias-resistor = <1>; }; diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 6bb77d7..f538957 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -121,6 +121,13 @@ struct ldo_regulator { bool enabled; }; +enum sgtl5000_micbias_resistor { + SGTL5000_MICBIAS_OFF = 0, + SGTL5000_MICBIAS_2K = 1, + SGTL5000_MICBIAS_4K = 2, + SGTL5000_MICBIAS_8K = 3, +}; + /* sgtl5000 private structure in codec */ struct sgtl5000_priv { int sysclk; /* sysclk rate */ @@ -131,6 +138,7 @@ struct sgtl5000_priv { struct regmap *regmap; struct clk *mclk; int revision; + enum sgtl5000_micbias_resistor micbias_resistor; }; /* @@ -145,12 +153,14 @@ struct sgtl5000_priv { static int mic_bias_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(w->codec); + switch (event) { case SND_SOC_DAPM_POST_PMU: - /* change mic bias resistor to 4Kohm */ + /* change mic bias resistor */ snd_soc_update_bits(w->codec, SGTL5000_CHIP_MIC_CTRL, - SGTL5000_BIAS_R_MASK, - SGTL5000_BIAS_R_4k << SGTL5000_BIAS_R_SHIFT); + SGTL5000_BIAS_R_MASK, + sgtl5000->micbias_resistor << SGTL5000_BIAS_R_SHIFT); break; case SND_SOC_DAPM_PRE_PMD: @@ -1326,7 +1336,16 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) SGTL5000_HP_ZCD_EN | SGTL5000_ADC_ZCD_EN); - snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 2); + switch (sgtl5000->micbias_resistor) { + case SGTL5000_MICBIAS_2K: + case SGTL5000_MICBIAS_4K: + case SGTL5000_MICBIAS_8K: + snd_soc_update_bits(codec, SGTL5000_CHIP_MIC_CTRL, + SGTL5000_BIAS_R_MASK, + sgtl5000->micbias_resistor << SGTL5000_BIAS_R_SHIFT); + default: + break; + } /* * disable DAP @@ -1418,6 +1437,8 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, struct sgtl5000_priv *sgtl5000; int ret, reg, rev; unsigned int mclk; + struct device_node *np = client->dev.of_node; + u32 value; sgtl5000 = devm_kzalloc(&client->dev, sizeof(struct sgtl5000_priv), GFP_KERNEL); @@ -1470,6 +1491,33 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, dev_info(&client->dev, "sgtl5000 revision 0x%x\n", rev); sgtl5000->revision = rev; + if (np) { + if (!of_property_read_u32(np, + "sgtl5000-micbias-resistor", &value)) { + switch (value) { + case 1: + sgtl5000->micbias_resistor = + SGTL5000_MICBIAS_2K; + break; + case 2: + sgtl5000->micbias_resistor = + SGTL5000_MICBIAS_4K; + break; + case 3: + sgtl5000->micbias_resistor = + SGTL5000_MICBIAS_8K; + break; + default: + sgtl5000->micbias_resistor = + SGTL5000_MICBIAS_OFF; + dev_err(&client->dev, + "Unsuitable MicBias resistor\n"); + } + } else { + sgtl5000->micbias_resistor = SGTL5000_MICBIAS_4K; + } + } + i2c_set_clientdata(client, sgtl5000); /* Ensure sgtl5000 will start with sane register values */ -- 2.1.1

2 4

[alsa-devel] [PATCH] snd-es18xx: Add GPO controls
by Ondrej Zary 12 Oct '14

12 Oct '14

Add GPO0 and GPO1 (General Purpose Outputs) controls to mixer. These can be used on some cards to control amplifier mute (seen in ES1868 datasheet) or additional onboard chips such as QX2130 QXpander processor. These GPOs are present on ES1868, ES1869, ES1887 and ES1888 chips. Tested on ES1868 with QX2130. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- sound/isa/es18xx.c | 49 +++++++++++++++++++++++++++++++++++++++---------- 1 file changed, 39 insertions(+), 10 deletions(-) diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index 6faaac6..8966ed2 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -156,6 +156,7 @@ struct snd_es18xx { #define ES18XX_I2S 0x0200 /* I2S mixer control */ #define ES18XX_MUTEREC 0x0400 /* Record source can be muted */ #define ES18XX_CONTROL 0x0800 /* Has control ports */ +#define ES18XX_GPO_2BIT 0x1000 /* GPO0,1 controlled by PM port */ /* Power Management */ #define ES18XX_PM 0x07 @@ -1136,11 +1137,14 @@ static int snd_es18xx_reg_read(struct snd_es18xx *chip, unsigned char reg) return snd_es18xx_read(chip, reg); } -#define ES18XX_SINGLE(xname, xindex, reg, shift, mask, invert) \ +#define ES18XX_SINGLE(xname, xindex, reg, shift, mask, flags) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ .info = snd_es18xx_info_single, \ .get = snd_es18xx_get_single, .put = snd_es18xx_put_single, \ - .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24) } + .private_value = reg | (shift << 8) | (mask << 16) | (flags << 24) } + +#define ES18XX_FL_INVERT (1 << 0) +#define ES18XX_FL_PMPORT (1 << 1) static int snd_es18xx_info_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -1159,10 +1163,14 @@ static int snd_es18xx_get_single(struct snd_kcontrol *kcontrol, struct snd_ctl_e int reg = kcontrol->private_value & 0xff; int shift = (kcontrol->private_value >> 8) & 0xff; int mask = (kcontrol->private_value >> 16) & 0xff; - int invert = (kcontrol->private_value >> 24) & 0xff; + int invert = (kcontrol->private_value >> 24) & ES18XX_FL_INVERT; + int pm_port = (kcontrol->private_value >> 24) & ES18XX_FL_PMPORT; int val; - - val = snd_es18xx_reg_read(chip, reg); + + if (pm_port) + val = inb(chip->port + ES18XX_PM); + else + val = snd_es18xx_reg_read(chip, reg); ucontrol->value.integer.value[0] = (val >> shift) & mask; if (invert) ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0]; @@ -1175,7 +1183,8 @@ static int snd_es18xx_put_single(struct snd_kcontrol *kcontrol, struct snd_ctl_e int reg = kcontrol->private_value & 0xff; int shift = (kcontrol->private_value >> 8) & 0xff; int mask = (kcontrol->private_value >> 16) & 0xff; - int invert = (kcontrol->private_value >> 24) & 0xff; + int invert = (kcontrol->private_value >> 24) & ES18XX_FL_INVERT; + int pm_port = (kcontrol->private_value >> 24) & ES18XX_FL_PMPORT; unsigned char val; val = (ucontrol->value.integer.value[0] & mask); @@ -1183,6 +1192,12 @@ static int snd_es18xx_put_single(struct snd_kcontrol *kcontrol, struct snd_ctl_e val = mask - val; mask <<= shift; val <<= shift; + if (pm_port) { + val = (inb(chip->port + ES18XX_PM) & ~mask) | val; + outb(val, chip->port + ES18XX_PM); + return 0; + } + return snd_es18xx_reg_bits(chip, reg, mask, val) != val; } @@ -1304,7 +1319,7 @@ static struct snd_kcontrol_new snd_es18xx_opt_speaker = ES18XX_SINGLE("Beep Playback Volume", 0, 0x3c, 0, 7, 0); static struct snd_kcontrol_new snd_es18xx_opt_1869[] = { -ES18XX_SINGLE("Capture Switch", 0, 0x1c, 4, 1, 1), +ES18XX_SINGLE("Capture Switch", 0, 0x1c, 4, 1, ES18XX_FL_INVERT), ES18XX_SINGLE("Video Playback Switch", 0, 0x7f, 0, 1, 0), ES18XX_DOUBLE("Mono Playback Volume", 0, 0x6d, 0x6d, 4, 0, 15, 0), ES18XX_DOUBLE("Mono Capture Volume", 0, 0x6f, 0x6f, 4, 0, 15, 0) @@ -1363,6 +1378,11 @@ static struct snd_kcontrol_new snd_es18xx_hw_volume_controls[] = { ES18XX_SINGLE("Hardware Master Volume Split", 0, 0x64, 7, 1, 0), }; +static struct snd_kcontrol_new snd_es18xx_opt_gpo_2bit[] = { +ES18XX_SINGLE("GPO0 Switch", 0, ES18XX_PM, 0, 1, ES18XX_FL_PMPORT), +ES18XX_SINGLE("GPO1 Switch", 0, ES18XX_PM, 1, 1, ES18XX_FL_PMPORT), +}; + static int snd_es18xx_config_read(struct snd_es18xx *chip, unsigned char reg) { int data; @@ -1629,10 +1649,10 @@ static int snd_es18xx_probe(struct snd_es18xx *chip, switch (chip->version) { case 0x1868: - chip->caps = ES18XX_DUPLEX_MONO | ES18XX_DUPLEX_SAME | ES18XX_CONTROL; + chip->caps = ES18XX_DUPLEX_MONO | ES18XX_DUPLEX_SAME | ES18XX_CONTROL | ES18XX_GPO_2BIT; break; case 0x1869: - chip->caps = ES18XX_PCM2 | ES18XX_SPATIALIZER | ES18XX_RECMIX | ES18XX_NEW_RATE | ES18XX_AUXB | ES18XX_MONO | ES18XX_MUTEREC | ES18XX_CONTROL | ES18XX_HWV; + chip->caps = ES18XX_PCM2 | ES18XX_SPATIALIZER | ES18XX_RECMIX | ES18XX_NEW_RATE | ES18XX_AUXB | ES18XX_MONO | ES18XX_MUTEREC | ES18XX_CONTROL | ES18XX_HWV | ES18XX_GPO_2BIT; break; case 0x1878: chip->caps = ES18XX_DUPLEX_MONO | ES18XX_DUPLEX_SAME | ES18XX_I2S | ES18XX_CONTROL; @@ -1642,7 +1662,7 @@ static int snd_es18xx_probe(struct snd_es18xx *chip, break; case 0x1887: case 0x1888: - chip->caps = ES18XX_PCM2 | ES18XX_RECMIX | ES18XX_AUXB | ES18XX_DUPLEX_SAME; + chip->caps = ES18XX_PCM2 | ES18XX_RECMIX | ES18XX_AUXB | ES18XX_DUPLEX_SAME | ES18XX_GPO_2BIT; break; default: snd_printk(KERN_ERR "[0x%lx] unsupported chip ES%x\n", @@ -1944,6 +1964,15 @@ static int snd_es18xx_mixer(struct snd_card *card) return err; } } + if (chip->caps & ES18XX_GPO_2BIT) { + for (idx = 0; idx < ARRAY_SIZE(snd_es18xx_opt_gpo_2bit); idx++) { + err = snd_ctl_add(card, + snd_ctl_new1(&snd_es18xx_opt_gpo_2bit[idx], + chip)); + if (err < 0) + return err; + } + } return 0; } -- Ondrej Zary

1 0

[alsa-devel] snd_pcm_hw_params_get_tick_time
by Shérab 12 Oct '14

12 Oct '14

Hello, I'm maintaing a TTS engine for french that use snd_pcm_hw_params_get_tick_time. Compiling the code produces a deprecation warning for this function. I tried to figure out how the code should be modified to not use this function but can't find a way. Any help would be warmly appreciated. Thanks, Shérab.

2 2

[alsa-devel] seq module loading issue
by Adam Goode 10 Oct '14

10 Oct '14

Hi, I've observed a scenario in ALSA that I would like some feedback on solving. On Linux, Chrome uses seq for WebMIDI. Eventually, there will be hotplug detection code in Chrome that looks like this: 1. Connect to announce port 2. Listen for any announcements for client/port start/end 3. Send MIDIConnectionEvent events to JavaScript clients ( http://webaudio.github.io/web-midi-api/#midiconnectionevent-interface) Here is the scenario where this fails: 1. Cold boot a machine with no MIDI devices plugged in 2. Start client, subscribe on the seq announce port 3. Insert USB MIDI device 4. snd_rawmidi module is loaded, triggered by insertion of MIDI device 5. snd_seq_midi is NOT loaded, but marked for loading later At this point, there is no client start event sent to the announce port. This is because of the deferred module loading, which won't trigger except for a handful of operations (snd_seq_client_info_get_client is an example). This is easily replicated with looking at aseqdump -p 0:1. My question is: how to solve this? There are a few options: 1. Periodically poll for new clients. This is fairly ugly, but will work to trigger module loading and won't result in latency once any device has been plugged in, since announce will again work. 2. Somehow trigger module loading if there are any clients listening on the announce port. This seems pretty tricky and requires keeping track of some new state. 3. Do away with deferred module loading in seq. This is pretty invasive and would increase the number of modules loaded, but would result in a lot of deleted code in the kernel and a clean fix. Number 3 is my preferred solution, but I wanted to see if these sorts of patches would be acceptable, since it is invasive and would result in more modules loaded. Thanks for any comments. Adam

3 9

[alsa-devel] ASoC machine driver: how to configure supported frame rates?
by Sergej Sawazki 10 Oct '14

10 Oct '14

Dear all, I'm writing an ALSA SoC machine driver. The codec on my board, as well as the cpu do support the following rates: SNDRV_PCM_RATE_8000_192000 However, the codec, as well as the cpu are clock slaves and the clocks are fixed. Hence, my board does only support the 96kHz sample rate. Can someone give me some advice on how to tell ALSA that my board does only support the rate SNDRV_PCM_RATE_96000? I don't want to modify the codec and the platform driver. Any help would be highly appreciated. Thanks! - Sergej

2 2

[alsa-devel] proposal: snd_pcm_start_at()
by Tim Cussins 10 Oct '14

10 Oct '14

Hi all, I'm Tim: I work at Linn Products Ltd - we make Network Music Players, amongst other things. As you might imagine, synchronised-start is important when multiple devices on the network are rendering the same audio. We'd be interested in contributing a small expansion of the alsa-lib API to support synchronised start. Assuming we can synchronise the audio clocks (I'm aware this is not trivial - It's not the topic of this post), we'd propose something like: int snd_pcm_start_at(snd_pcm_t* pcm, snd_htimestamp_t* tstamp); and playback would begin as close to tstamp as possible. If tstamp is in the past, it would should return an error. Recent work by Takashi Iwai enables client code to set the clock type of timestamps using snd_pcm_sw_params_set_tstamp_type(). This context could quite naturally extend to tstamp argument of snd_pcm_start_at(). Before I get stuck into working up the details under the hood, it'd be good to get some feedback/objections regarding this approach. Thanks in advance! Tim

5 11

[alsa-devel] No sound captured with SGTL5000 on i.MX6 in I²S master mode
by Jean-Michel Hautbois 10 Oct '14

10 Oct '14

Hi, Sorry if double posting, I have not received confirmation on subscription yesterday... So I try again :/. I have a custom i.MX6 board, and have a sgtl5000 connected on it. I would like to have it used in master mode (it is used only for audio capture) but I get a "pcm_read:2031: read error: Input/output error" when using arecord. I have checked my signals and I get something on I²S which seems to be good. I launch : $> arecord -f cd -Dhw:0,0 -v -V stereo Receive Frame Sync is 44.1KHz Receive Clock Signal is ~1MHz (I can be more precise if needed) Data Receive is present, only when arecord is launched. It stops when there is no capture asked. Both clocks are still present even when no arecord is running. This suggests, for me, that master mode is working. So, I don't know why I can't get any data to my arecord. Is this a left/right justified master mode ? Anything else ? I am wondering if I forgot something in my DT or if I need to do something else : sound { compatible = "fsl,imx6q-vbx3-sgtl5000", "fsl,imx-audio-sgtl5000"; model = "imx6q-vbx3-sgtl5000"; ssi-controller = <&ssi1>; audio-codec = <&codec>; audio-routing = "LINE_IN", "Line In Jack"; mux-int-port = <1>; mux-ext-port = <5>; }; &audmux { pinctrl-names = "default"; pinctrl-0 = <&pinctrl_audmux>; status = "okay"; }; &i2c2 { clock-frequency = <100000>; pinctrl-names = "default"; pinctrl-0 = <&pinctrl_i2c2>; status = "okay"; codec: sgtl5000@0a { compatible = "fsl,sgtl5000"; reg = <0x0a>; clocks = <&clks 201>; VDDA-supply = <&vgen6_reg>; VDDIO-supply = <&vgen6_reg>; sysclk = <24000000>; }; }; &iomuxc { pinctrl-names = "default"; pinctrl-0 = <&pinctrl_hog>; imx6qdl-vbx3 { pinctrl_audmux: audmuxgrp { fsl,pins = < MX6QDLPAD_DISP0_DAT13__AUD5_RXFS 0x130b0 MX6QDL_PAD_DISP0_DAT14__AUD5_RXC 0x130b0 MX6QDL_PAD_DISP0_DAT19__AUD5_RXD 0x130b0 >; }; }; &ssi1 { status = "okay"; codec-handle = <&codec>; fsl,mode = "i2s-master"; fsl,ssi-asynchronous; /* Not sure about it, but does not change anything */ }; Here is the dmesg part : [ 6.026803] sgtl5000 1-000a: sgtl5000 revision 0x11 [ 6.129380] imx-sgtl5000 sound: snd_soc_register_card failed (-517) [ 6.248849] platform sound: Driver imx-sgtl5000 requests probe deferral [ 6.311643] sgtl5000 1-000a: Using internal LDO instead of VDDD [ 6.393196] imx-sgtl5000 sound: sgtl5000 <-> 2028000.ssi mapping ok And the user space commands : $> arecord -l **** List of CAPTURE Hardware Devices **** card 0: imx6qvbx3sgtl50 [imx6q-vbx3-sgtl5000], device 0: HiFi sgtl5000-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 $> arecord -v -V stereo -fdat -D hw:0,0 somefile.wav Recording WAVE 'somefile.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo Hardware PCM card 0 'imx6q-vbx3-sgtl5000' device 0 subdevice 0 Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 16384 period_size : 4096 period_time : 85333 tstamp_mode : NONE period_step : 1 avail_min : 4096 period_event : 0 start_threshold : 1 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 1073741824 appl_ptr : 0 hw_ptr : 0 arecord: pcm_read:2031: read error: Input/output error The kernel is 3.17-rc5. Thanks for reading all this, and for any advice :). JM

4 35

[alsa-devel] Help Adjusting HDA HDMI settings
by bencoxdev＠gmail.com 10 Oct '14

10 Oct '14

Hi All, I need to change the of an ALSA driver on a proprietary arm linux box. Specifically the box in its standard configuration is hard coded to have 2 channel audio output only. The hardware on the box I’m assured by the manufacturer is capable of surround 5.1 output on it’s HDMI port. I therefore need to increase the ALSA driver’s capabilities to handle surround 5.1 output. I have the kernel source code for box. Could someone point me in the direction of what would need changing/adding in order to achieve this please. Thanks Ben

2 2

[alsa-devel] [PATCH 1/2] ALSA: au88x0: added reference of vortex_t
by Sudip Mukherjee 10 Oct '14

10 Oct '14

added a pointer of the vortex in the following functions : vortex_alsafmt_aspfmt vortex_Vort3D_InitializeSource a3dsrc_ZeroStateA3D so that we can have a reference of the vortex in the function. this reference of the vortex will actually be used in a later patch to convert the pr_* macro to dev_*. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- sound/pci/au88x0/au88x0.h | 4 ++-- sound/pci/au88x0/au88x0_a3d.c | 6 +++--- sound/pci/au88x0/au88x0_core.c | 5 +++-- sound/pci/au88x0/au88x0_pcm.c | 2 +- 4 files changed, 9 insertions(+), 8 deletions(-) diff --git a/sound/pci/au88x0/au88x0.h b/sound/pci/au88x0/au88x0.h index 466a5c8..3a8fefe 100644 --- a/sound/pci/au88x0/au88x0.h +++ b/sound/pci/au88x0/au88x0.h @@ -243,7 +243,7 @@ static int vortex_core_init(vortex_t * card); static int vortex_core_shutdown(vortex_t * card); static void vortex_enable_int(vortex_t * card); static irqreturn_t vortex_interrupt(int irq, void *dev_id); -static int vortex_alsafmt_aspfmt(int alsafmt); +static int vortex_alsafmt_aspfmt(int alsafmt, vortex_t *v); /* Connection stuff. */ static void vortex_connect_default(vortex_t * vortex, int en); @@ -278,7 +278,7 @@ static void vortex_mix_setvolumebyte(vortex_t * vortex, unsigned char mix, static void vortex_Vort3D_enable(vortex_t * v); static void vortex_Vort3D_disable(vortex_t * v); static void vortex_Vort3D_connect(vortex_t * vortex, int en); -static void vortex_Vort3D_InitializeSource(a3dsrc_t * a, int en); +static void vortex_Vort3D_InitializeSource(a3dsrc_t *a, int en, vortex_t *v); #endif /* Driver stuff. */ diff --git a/sound/pci/au88x0/au88x0_a3d.c b/sound/pci/au88x0/au88x0_a3d.c index 30f760e..bc9cda3a 100644 --- a/sound/pci/au88x0/au88x0_a3d.c +++ b/sound/pci/au88x0/au88x0_a3d.c @@ -484,7 +484,7 @@ static void a3dsrc_ZeroState(a3dsrc_t * a) } /* Reset entire A3D engine */ -static void a3dsrc_ZeroStateA3D(a3dsrc_t * a) +static void a3dsrc_ZeroStateA3D(a3dsrc_t *a, vortex_t *v) { int i, var, var2; @@ -601,7 +601,7 @@ static void vortex_Vort3D_enable(vortex_t *v) Vort3DRend_Initialize(v, XT_HEADPHONE); for (i = 0; i < NR_A3D; i++) { vortex_A3dSourceHw_Initialize(v, i % 4, i >> 2); - a3dsrc_ZeroStateA3D(&(v->a3d[0])); + a3dsrc_ZeroStateA3D(&v->a3d[0], v); } /* Register ALSA controls */ vortex_a3d_register_controls(v); @@ -676,7 +676,7 @@ static void vortex_Vort3D_connect(vortex_t * v, int en) } /* Initialize one single A3D source. */ -static void vortex_Vort3D_InitializeSource(a3dsrc_t * a, int en) +static void vortex_Vort3D_InitializeSource(a3dsrc_t *a, int en, vortex_t *v) { if (a->vortex == NULL) { pr_warn diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c index 72e8128..00e2096 100644 --- a/sound/pci/au88x0/au88x0_core.c +++ b/sound/pci/au88x0/au88x0_core.c @@ -2177,7 +2177,8 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir, return -EBUSY; } /* (De)Initialize A3D hardware source. */ - vortex_Vort3D_InitializeSource(&(vortex->a3d[a3d]), en); + vortex_Vort3D_InitializeSource(&vortex->a3d[a3d], en, + vortex); } /* Make SPDIF out exclusive to "spdif" device when in use. */ if ((stream->type == VORTEX_PCM_SPDIF) && (en)) { @@ -2765,7 +2766,7 @@ static int vortex_core_shutdown(vortex_t * vortex) /* Alsa support. */ -static int vortex_alsafmt_aspfmt(int alsafmt) +static int vortex_alsafmt_aspfmt(int alsafmt, vortex_t *v) { int fmt; diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c index 5adc6b9..bdde182 100644 --- a/sound/pci/au88x0/au88x0_pcm.c +++ b/sound/pci/au88x0/au88x0_pcm.c @@ -332,7 +332,7 @@ static int snd_vortex_pcm_prepare(struct snd_pcm_substream *substream) dir = 1; else dir = 0; - fmt = vortex_alsafmt_aspfmt(runtime->format); + fmt = vortex_alsafmt_aspfmt(runtime->format, chip); spin_lock_irq(&chip->lock); if (VORTEX_PCM_TYPE(substream->pcm) != VORTEX_PCM_WT) { vortex_adbdma_setmode(chip, dma, 1, dir, fmt, -- 1.8.1.2

3 6

[alsa-devel] [PATCH] ASoC: rt5677: make volume TLV closer to reality
by Dylan Reid 10 Oct '14

10 Oct '14

The volume blocks have an step of 0.375dB, but TLV uses 0.01dB for units. Approximate 0.375 as 0.37. It leaves the minimum value off by 1.15dB, but that is better than being off by ~500 as it was with 37.5 as the step. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- Just a suggestion. It seems to me to be the lesser of two evils. This probably won't be the last codec that has a higher resolution than 0.01dB, should we come up with a way to express that? --- sound/soc/codecs/rt5677.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 16aa4d9..f139d5a 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -538,9 +538,11 @@ static bool rt5677_readable_register(struct device *dev, unsigned int reg) } static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0); -static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); +/* Actual min=-65.625dB and step=0.375dB but only have 0.01 resolution. */ +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -6475, 37, 0); static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); -static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +/* Actual min=-17.625dB and step=0.375dB but only have 0.01 resolution. */ +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -1739, 37, 0); static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); static const DECLARE_TLV_DB_SCALE(st_vol_tlv, -4650, 150, 0); -- 2.1.2.330.g565301e

2 2

[alsa-devel] [PATCH 1/1] ALSA: ALC283 codec - Avoid pop noise on headphones during suspend/resume
by Harsha Priya 09 Oct '14

09 Oct '14

This patch sets the headphones mode to default before suspending which helps avoid the pop noise on headphones Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 69d1236..e839e7a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2884,6 +2884,9 @@ static void alc283_shutup(struct hda_codec *codec) alc_write_coef_idx(codec, 0x43, 0x9004); + /*depop hp during suspend*/ + alc_write_coef_idx(codec, 0x06, 0x2100); + snd_hda_codec_write(codec, hp_pin, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); -- 1.9.1

1 0

[alsa-devel] [GIT PULL] ASoC fixes for v3.18
by Mark Brown 09 Oct '14

09 Oct '14

The following changes since commit a5448c88b812390a3622e76d774e10c0da1fb970: Merge remote-tracking branches 'asoc/topic/txx9', 'asoc/topic/wm8978' and 'asoc/topic/wm8994' into asoc-next (2014-10-06 12:49:09 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.18-2 for you to fetch changes up to bdf20b4291eaa3b327398b8dd330065ad8e6d3ce: Merge remote-tracking branches 'asoc/fix/88pm860x', 'asoc/fix/fsl', 'asoc/fix/imx', 'asoc/fix/mc13783', 'asoc/fix/rockchip' and 'asoc/fix/simple' into asoc-linus (2014-10-08 16:44:50 +0100) ---------------------------------------------------------------- ASoC: Fixes for v3.18 A few small fixes, mostly small although a few really nasty ones like the interaction between GPIO 0 and simple-card. ---------------------------------------------------------------- Daniel Mack (4): ASoC: soc-dapm: fix use after free ASoC: core: fix use after free in snd_soc_remove_platform() ASoC: 88pm860x-codec: Fix possibly missing string termination ASoC: soc-pcm: fix sig_bits determination in soc_pcm_apply_msb() Geert Uytterhoeven (1): ASoC: simple-card: Initialize headphone and mic GPIO numbers Mark Brown (6): ASoC: mc13783: Ensure we only try to dereference valid of_nodes Merge tag 'asoc-v3.17-rc4' into asoc-linus Merge tag 'asoc-v3.18' into asoc-linus Merge remote-tracking branch 'asoc/fix/core' into asoc-linus Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branches 'asoc/fix/88pm860x', 'asoc/fix/fsl', 'asoc/fix/imx', 'asoc/fix/mc13783', 'asoc/fix/rockchip' and 'asoc/fix/simple' into asoc-linus Sonny Rao (1): ASoC: rockchip-i2s: fix infinite loop in rockchip_snd_txctrl Takashi Iwai (2): ASoC: imx-es8328: Fix missing return code in imx_es8328_probe() ASoC: imx-es8328: Fix of_node_put() call with uninitialized object sound/soc/codecs/88pm860x-codec.c | 2 +- sound/soc/codecs/mc13783.c | 16 ++++++++++------ sound/soc/fsl/imx-es8328.c | 3 ++- sound/soc/generic/simple-card.c | 3 +++ sound/soc/rockchip/rockchip_i2s.c | 4 +++- sound/soc/soc-core.c | 4 ++-- sound/soc/soc-dapm.c | 25 ++++++++++++++----------- sound/soc/soc-pcm.c | 2 +- 8 files changed, 36 insertions(+), 23 deletions(-)

2 1

[alsa-devel] [PATCH] ASoC: mc13783: Ensure we only try to dereference valid of_nodes
by Mark Brown 08 Oct '14

08 Oct '14

Reported-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/mc13783.c | 16 ++++++++++------ 1 file changed, 10 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 388f90a597fa..71f775aad7c7 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -765,12 +765,18 @@ static int __init mc13783_codec_probe(struct platform_device *pdev) return -ENOSYS; ret = of_property_read_u32(np, "adc-port", &priv->adc_ssi_port); - if (ret) - goto out; + if (ret) { + of_node_put(np); + return ret; + } ret = of_property_read_u32(np, "dac-port", &priv->dac_ssi_port); - if (ret) - goto out; + if (ret) { + of_node_put(np); + return ret; + } + + of_node_put(np); } dev_set_drvdata(&pdev->dev, priv); @@ -783,8 +789,6 @@ static int __init mc13783_codec_probe(struct platform_device *pdev) ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_mc13783, mc13783_dai_async, ARRAY_SIZE(mc13783_dai_async)); -out: - of_node_put(np); return ret; } -- 2.1.1

2 3

[alsa-devel] 3.5 audio jack automatic detection
by manoj kumar 08 Oct '14

08 Oct '14

Hi All, I am new to ALSA development and I have a requirement where I need to have automatic detection of 3.5 audio jack and update in the user space GUI app. Is there any utility and sample code available where I can find about any plug and unplug of device in the jack of my machine. Regards Manoj

1 0

[alsa-devel] [PATCH 0/8] ASoC: fsl: Refine DT binding documents for more readability
by Nicolin Chen 08 Oct '14

08 Oct '14

This series of patches simply add indentations and blank lines to each DT binding documents of Freescale ASoC drivers so as to make them more clear and readable. @Mark Brown Not sure if they are okay to get applied through your tree. Otherwise, I might need to change these subjects and send them to DT maillist only. Thank you. Nicolin Chen (8): ASoC: fsl_esai: Add indentation for binding doc to increase readability ASoC: fsl_spdif: Add indentation for binding doc to increase readability ASoC: fsl_sai: Add indentation for binding doc to increase readability ASoC: eukrea-tlv320: Add indentation for binding doc to increase readability ASoC: imx-audmux: Add indentation for binding doc to increase readability ASoC: imx-sgtl5000: Add indentation for binding doc to increase readability ASoC: imx-spdif: Add indentation for binding doc to increase readability ASoC: imx-wm8962: Add indentation for binding doc to increase readability .../devicetree/bindings/sound/eukrea-tlv320.txt | 15 +++-- .../devicetree/bindings/sound/fsl,esai.txt | 44 ++++++++------- .../devicetree/bindings/sound/fsl,spdif.txt | 37 ++++++------ .../devicetree/bindings/sound/fsl-sai.txt | 66 ++++++++++++++-------- .../bindings/sound/imx-audio-sgtl5000.txt | 61 +++++++++++--------- .../devicetree/bindings/sound/imx-audio-spdif.txt | 22 ++++---- .../devicetree/bindings/sound/imx-audio-wm8962.txt | 45 ++++++++------- .../devicetree/bindings/sound/imx-audmux.txt | 22 +++++--- 8 files changed, 178 insertions(+), 134 deletions(-) -- 1.9.1

2 9

[alsa-devel] [PATCH] ASoC: rockchip-i2s: fix infinite loop in rockchip_snd_txctrl
by Sonny Rao 08 Oct '14

08 Oct '14

We can get into an infinite loop if the I2S_CLR register fails to clear due to a missing break statement, so add that. Signed-off-by: Sonny Rao <sonnyrao(a)chromium.org> --- sound/soc/rockchip/rockchip_i2s.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 033487c..f373e37 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -108,8 +108,10 @@ static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) while (val) { regmap_read(i2s->regmap, I2S_CLR, &val); retry--; - if (!retry) + if (!retry) { dev_warn(i2s->dev, "fail to clear\n"); + break; + } } } } -- 1.8.3.2

2 1

[alsa-devel] HDA: Missing Dock LineIn/Out with Lenovo L440 on Pro Dock
by Sascha Wilde 08 Oct '14

08 Oct '14

Hi *, using a Lenovo L440 Laptop on a Lenovo Pro Dock the combined line-in/out jack at the docking station can not be used for output (input untested). Output on the internal combined line-in/out jack works, as well as output on the internal speakers. Its intel-hda with Realtek ALC292. For more details see alsa-info.sh output below. Loading the driver with model=lenovo-dock does not help. I guess quirks for the pins leading to the external dock jack are missing. Please let me know if you need any further information. Any help highly appreciated. :) cheers sascha Output of alsa-info follows: ---------------------------------------------------------------------->8- upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.4.63 !!################################ !!Script ran on: Fri Jun 6 10:54:38 UTC 2014 !!Linux Distribution !!------------------ Arch Linux \r (\l) NAME="Arch Linux" ID=arch PRETTY_NAME="Arch Linux" HOME_URL="https://www.archlinux.org/" SUPPORT_URL="https://bbs.archlinux.org/" BUG_REPORT_URL="https://bugs.archlinux.org/" !!DMI Information !!--------------- Manufacturer: LENOVO Product Name: 20ATCTO1WW Product Version: ThinkPad L440 Firmware Version: J4ET62WW(1.62) !!Kernel Information !!------------------ Kernel release: 3.14.5-1-ARCH Operating System: GNU/Linux Architecture: x86_64 Processor: unknown SMP Enabled: Yes !!ALSA Version !!------------ Driver version: k3.14.5-1-ARCH Library version: 1.0.27.2 Utilities version: 1.0.27.2 !!Loaded ALSA modules !!------------------- snd_hda_intel snd_hda_intel snd_usb_audio thinkpad_acpi !!Sound Servers on this system !!---------------------------- Jack: Installed - Yes (/usr/bin/jackd) Running - No !!Soundcards recognised by ALSA !!----------------------------- 0 [HDMI ]: HDA-Intel - HDA Intel HDMI HDA Intel HDMI at 0xf2530000 irq 48 1 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xf2534000 irq 45 2 [Q9000 ]: USB-Audio - QuickCam Pro 9000 Logitech, Inc. QuickCam Pro 9000 at usb-0000:00:14.0-9.1, high speed 29 [ThinkPadEC ]: ThinkPad EC - ThinkPad Console Audio Control ThinkPad Console Audio Control at EC reg 0x30, fw unknown !!PCI Soundcards installed in the system !!-------------------------------------- 00:03.0 Audio device: Intel Corporation Xeon E3-1200 v3/4th Gen Core Processor HD Audio Controller (rev 06) 00:1b.0 Audio device: Intel Corporation 8 Series/C220 Series Chipset High Definition Audio Controller (rev 05) !!Advanced information - PCI Vendor/Device/Subsystem ID's !!------------------------------------------------------- 00:03.0 0403: 8086:0c0c (rev 06) Subsystem: 17aa:501e -- 00:1b.0 0403: 8086:8c20 (rev 05) Subsystem: 17aa:501e !!Modprobe options (Sound related) !!-------------------------------- snd_hda_intel: model=lenovo-dock !!Loaded sound module options !!--------------------------- !!Module: snd_hda_intel align_buffer_size : -1 bdl_pos_adj : 32,1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 beep_mode : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_msi : -1 id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 jackpoll_ms : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 model : lenovo-dock,(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) patch : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) position_fix : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 power_save : 0 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 probe_only : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 single_cmd : N snoop : Y !!Module: snd_hda_intel align_buffer_size : -1 bdl_pos_adj : 32,1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 beep_mode : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_msi : -1 id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 jackpoll_ms : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 model : lenovo-dock,(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) patch : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) position_fix : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 power_save : 0 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 probe_only : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 single_cmd : N snoop : Y !!Module: snd_usb_audio autoclock : Y device_setup : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) ignore_ctl_error : N index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 pid : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 vid : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 !!Module: thinkpad_acpi brightness_enable : 2 brightness_mode : 4 enable : Y experimental : 0 fan_control : N force_load : N id : ThinkPadEC index : -536870912 volume_capabilities : 0 volume_control : N volume_mode : 3 !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: Intel Haswell HDMI Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x80862807 Subsystem Id: 0x80860101 Revision Id: 0x100000 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D3 CLKSTOP EPSS Power: setting=D0, actual=D0, Clock-stop-OK GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 Node 0x02 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled KAE Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Node 0x03 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled KAE Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Node 0x04 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled KAE Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Node 0x05 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Control: name="HDMI/DP,pcm=3 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=0, device=0 Control: name="IEC958 Playback Pro Mask", index=0, device=0 Control: name="IEC958 Playback Default", index=0, device=0 Control: name="IEC958 Playback Switch", index=0, device=0 Control: name="ELD", index=0, device=3 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0b000094: OUT Detect HBR HDMI DP Pin Default 0x18560010: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=01, enabled=1 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Devices: 0 Connection: 0 In-driver Connection: 3 0x02 0x03 0x04 Node 0x06 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Control: name="HDMI/DP,pcm=7 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=1, device=0 Control: name="IEC958 Playback Pro Mask", index=1, device=0 Control: name="IEC958 Playback Default", index=1, device=0 Control: name="IEC958 Playback Switch", index=1, device=0 Control: name="ELD", index=0, device=7 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0b000094: OUT Detect HBR HDMI DP Pin Default 0x18560020: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=02, enabled=1 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Devices: 0 Connection: 0 In-driver Connection: 3 0x02 0x03 0x04 Node 0x07 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Control: name="HDMI/DP,pcm=8 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=2, device=0 Control: name="IEC958 Playback Pro Mask", index=2, device=0 Control: name="IEC958 Playback Default", index=2, device=0 Control: name="IEC958 Playback Switch", index=2, device=0 Control: name="ELD", index=0, device=8 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0b000094: OUT Detect HBR HDMI DP Pin Default 0x18560030: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x3, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=03, enabled=1 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Devices: 0 Connection: 0 In-driver Connection: 3 0x02 0x03 0x04 Node 0x08 [Vendor Defined Widget] wcaps 0xf00000: Mono Codec: Realtek ALC292 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0292 Subsystem Id: 0x17aa501e Revision Id: 0x100001 No Modem Function Group found Default PCM: rates [0x5f0]: 32000 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D1 D2 D3 CLKSTOP EPSS Power: setting=D0, actual=D0 GPIO: io=5, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[4]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0x41d: Stereo Amp-Out Control: name="Headphone Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="ALC292 Analog", type="Audio", device=0 Amp-Out caps: ofs=0x57, nsteps=0x57, stepsize=0x02, mute=0 Amp-Out vals: [0x3d 0x3d] Converter: stream=8, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x03 [Audio Output] wcaps 0x41d: Stereo Amp-Out Control: name="Speaker Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x57, nsteps=0x57, stepsize=0x02, mute=0 Amp-Out vals: [0x3d 0x3d] Converter: stream=8, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x04 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x05 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x06 [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x5f0]: 32000 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x08 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x17, nsteps=0x3f, stepsize=0x02, mute=1 Amp-In vals: [0x97 0x97] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x23 Node 0x09 [Audio Input] wcaps 0x10051b: Stereo Amp-In Control: name="Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Capture Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Device: name="ALC292 Analog", type="Audio", device=0 Amp-In caps: ofs=0x17, nsteps=0x3f, stepsize=0x02, mute=1 Amp-In vals: [0x27 0x27] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x22 Node 0x0a [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Control: name="Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Control: name="Beep Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=4, ofs=0 Control: name="Beep Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=4, ofs=0 Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 5 0x18 0x19 0x1a 0x1b 0x1d Node 0x0c [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Connection: 2 0x03 0x0b Node 0x0e [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0f [Audio Mixer] wcaps 0x20010a: Mono Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80] [0x00] Connection: 2 0x02 0x03 Node 0x10 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x12 [Pin Complex] wcaps 0x40040b: Stereo Amp-In Control: name="Internal Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Internal Mic Phantom Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 Amp-In vals: [0x02 0x02] Pincap 0x00000020: IN Pin Default 0x90a60130: [Fixed] Mic at Int N/A Conn = Digital, Color = Unknown DefAssociation = 0x3, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x13 [Pin Complex] wcaps 0x40040b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00000020: IN Pin Default 0x40000000: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0x0, Sequence = 0x0 Pin-ctls: 0x00: Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x14 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Control: name="Speaker Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Speaker Phantom Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00010010: OUT EAPD EAPD 0x2: EAPD Pin Default 0x90170110: [Fixed] Speaker at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x0c 0x0d* Node 0x15 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Headphone Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0001001c: OUT HP EAPD Detect EAPD 0x2: EAPD Pin Default 0x0321101f: [Jack] HP Out at Ext Left Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0xc0: OUT HP Unsolicited: tag=01, enabled=1 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x0c* 0x0d Node 0x16 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0001001c: OUT HP EAPD Detect EAPD 0x2: EAPD Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x0c* 0x0d Node 0x17 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x18 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00003734: IN OUT Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x0c* 0x0d Node 0x19 [Pin Complex] wcaps 0x40048b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00000024: IN Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x1a [Pin Complex] wcaps 0x40048b: Stereo Amp-In Control: name="Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Mic Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00003724: IN Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x03a11020: [Jack] Mic at Ext Left Conn = 1/8, Color = Black DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=02, enabled=1 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x1b [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00003734: IN OUT Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x0c* 0x0d Node 0x1c [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x1d [Pin Complex] wcaps 0x400400: Mono Pincap 0x00000020: IN Pin Default 0x40738105: [N/A] Modem Hand at Ext N/A Conn = ATAPI, Color = Purple DefAssociation = 0x0, Sequence = 0x5 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x1e [Pin Complex] wcaps 0x400781: Stereo Digital Pincap 0x00000014: OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x06 Node 0x1f [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=124 Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Audio Selector] wcaps 0x300101: Stereo Connection: 8 0x18 0x19 0x1a 0x1b 0x1d 0x0b 0x12* 0x13 Node 0x23 [Audio Selector] wcaps 0x300101: Stereo Connection: 6 0x18* 0x19 0x1a 0x1b 0x1d 0x0b Node 0x24 [Vendor Defined Widget] wcaps 0xf00000: Mono --endcollapse-- !!USB Mixer information !!--------------------- --startcollapse-- USB Mixer: usb_id=0x046d0990, ctrlif=2, ctlerr=0 Card: Logitech, Inc. QuickCam Pro 9000 at usb-0000:00:14.0-9.1, high speed Unit: 5 Control: name="Mic Capture Volume", index=0 Info: id=5, control=2, cmask=0x0, channels=1, type="S16" Volume: min=4608, max=7680, dBmin=1800, dBmax=3000 Unit: 5 Control: name="Mic Capture Switch", index=0 Info: id=5, control=1, cmask=0x0, channels=1, type="INV_BOOLEAN" Volume: min=0, max=1, dBmin=0, dBmax=0 --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw----+ 1 root audio 116, 11 Jun 5 09:35 /dev/snd/controlC0 crw-rw----+ 1 root audio 116, 6 Jun 5 09:35 /dev/snd/controlC1 crw-rw----+ 1 root audio 116, 13 Jun 6 12:38 /dev/snd/controlC2 crw-rw----+ 1 root audio 116, 2 Jun 5 09:35 /dev/snd/controlC29 crw-rw----+ 1 root audio 116, 10 Jun 5 09:35 /dev/snd/hwC0D0 crw-rw----+ 1 root audio 116, 5 Jun 5 09:35 /dev/snd/hwC1D0 crw-rw----+ 1 root audio 116, 9 Jun 5 09:35 /dev/snd/pcmC0D3p crw-rw----+ 1 root audio 116, 8 Jun 5 09:35 /dev/snd/pcmC0D7p crw-rw----+ 1 root audio 116, 7 Jun 5 09:35 /dev/snd/pcmC0D8p crw-rw----+ 1 root audio 116, 4 Jun 6 11:41 /dev/snd/pcmC1D0c crw-rw----+ 1 root audio 116, 3 Jun 6 12:38 /dev/snd/pcmC1D0p crw-rw----+ 1 root audio 116, 12 Jun 6 12:38 /dev/snd/pcmC2D0c crw-rw----+ 1 root audio 116, 1 Jun 5 17:36 /dev/snd/seq crw-rw----+ 1 root audio 116, 33 Jun 5 09:35 /dev/snd/timer /dev/snd/by-id: total 0 drwxr-xr-x 2 root root 60 Jun 6 12:38 . drwxr-xr-x 4 root root 360 Jun 6 12:38 .. lrwxrwxrwx 1 root root 12 Jun 6 12:38 usb-046d_0990_2648235F-02 -> ../controlC2 /dev/snd/by-path: total 0 drwxr-xr-x 2 root root 120 Jun 6 12:38 . drwxr-xr-x 4 root root 360 Jun 6 12:38 .. lrwxrwxrwx 1 root root 12 Jun 5 09:35 pci-0000:00:03.0 -> ../controlC0 lrwxrwxrwx 1 root root 12 Jun 6 12:38 pci-0000:00:14.0-usb-0:9.1:1.2 -> ../controlC2 lrwxrwxrwx 1 root root 12 Jun 5 09:35 pci-0000:00:1b.0 -> ../controlC1 lrwxrwxrwx 1 root root 13 Jun 5 09:35 platform-thinkpad_acpi -> ../controlC29 !!ALSA configuration files !!------------------------ !!System wide config file (/etc/asound.conf) pcm.!default { type plug slave.pcm { type asym playback.pcm "mixout" capture.pcm "mixin" } } pcm.mixout { type dmix ipc_key 1024 slave { pcm "hw:1,0" } bindings { 0 0 1 1 } } pcm.mixin { type dsnoop ipc_key 1234 slave { pcm "hw:1,0" } } ctl.!default { type hw card PCH } !!Aplay/Arecord output !!-------------------- APLAY **** List of PLAYBACK Hardware Devices **** card 0: HDMI [HDA Intel HDMI], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: HDMI [HDA Intel HDMI], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: HDMI [HDA Intel HDMI], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: PCH [HDA Intel PCH], device 0: ALC292 Analog [ALC292 Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 1: PCH [HDA Intel PCH], device 0: ALC292 Analog [ALC292 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: Q9000 [QuickCam Pro 9000], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [HDMI] Card hw:0 'HDMI'/'HDA Intel HDMI at 0xf2530000 irq 48' Mixer name : 'Intel Haswell HDMI' Components : 'HDA:80862807,80860101,00100000' Controls : 21 Simple ctrls : 3 Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'IEC958',1 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'IEC958',2 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] !!-------Mixer controls for card 1 [PCH] Card hw:1 'PCH'/'HDA Intel PCH at 0xf2534000 irq 45' Mixer name : 'Realtek ALC292' Components : 'HDA:10ec0292,17aa501e,00100001' Controls : 21 Simple ctrls : 9 Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 87 Mono: Playback 61 [70%] [-19.50dB] [on] Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 87 Mono: Front Left: Playback 87 [100%] [0.00dB] [on] Front Right: Playback 87 [100%] [0.00dB] [on] Simple mixer control 'Speaker',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 87 Mono: Front Left: Playback 87 [100%] [0.00dB] [on] Front Right: Playback 87 [100%] [0.00dB] [on] Simple mixer control 'Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] [0.00dB] Front Right: 0 [0%] [0.00dB] Simple mixer control 'Beep',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 63 Front Left: Capture 39 [62%] [12.00dB] [on] Front Right: Capture 39 [62%] [12.00dB] [on] Simple mixer control 'Auto-Mute Mode',0 Capabilities: enum Items: 'Disabled' 'Enabled' Item0: 'Enabled' Simple mixer control 'Internal Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 2 [67%] [24.00dB] Front Right: 2 [67%] [24.00dB] !!-------Mixer controls for card 2 [Q9000] Card hw:2 'Q9000'/'Logitech, Inc. QuickCam Pro 9000 at usb-0000:00:14.0-9.1, high speed' Mixer name : 'USB Mixer' Components : 'USB046d:0990' Controls : 3 Simple ctrls : 1 Simple mixer control 'Mic',0 Capabilities: cvolume cvolume-joined cswitch cswitch-joined Capture channels: Mono Limits: Capture 0 - 3072 Mono: Capture 1536 [50%] [24.00dB] [on] !!-------Mixer controls for card 29 [ThinkPadEC] Card hw:29 'ThinkPadEC'/'ThinkPad Console Audio Control at EC reg 0x30, fw unknown' Mixer name : 'ThinkPad EC (unknown)' Components : '' Controls : 1 Simple ctrls : 1 Simple mixer control 'Console',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] !!Alsactl output !!-------------- --startcollapse-- state.HDMI { control.1 { iface CARD name 'HDMI/DP,pcm=3 Jack' value false comment { access read type BOOLEAN count 1 } } control.2 { iface MIXER name 'IEC958 Playback Con Mask' value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.3 { iface MIXER name 'IEC958 Playback Pro Mask' value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.4 { iface MIXER name 'IEC958 Playback Default' value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.5 { iface MIXER name 'IEC958 Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.6 { iface PCM device 3 name ELD value '' comment { access 'read volatile' type BYTES count 0 } } control.7 { iface CARD name 'HDMI/DP,pcm=7 Jack' value false comment { access read type BOOLEAN count 1 } } control.8 { iface MIXER name 'IEC958 Playback Con Mask' index 1 value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.9 { iface MIXER name 'IEC958 Playback Pro Mask' index 1 value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.10 { iface MIXER name 'IEC958 Playback Default' index 1 value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.11 { iface MIXER name 'IEC958 Playback Switch' index 1 value true comment { access 'read write' type BOOLEAN count 1 } } control.12 { iface PCM device 7 name ELD value '' comment { access 'read volatile' type BYTES count 0 } } control.13 { iface CARD name 'HDMI/DP,pcm=8 Jack' value false comment { access read type BOOLEAN count 1 } } control.14 { iface MIXER name 'IEC958 Playback Con Mask' index 2 value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.15 { iface MIXER name 'IEC958 Playback Pro Mask' index 2 value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.16 { iface MIXER name 'IEC958 Playback Default' index 2 value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.17 { iface MIXER name 'IEC958 Playback Switch' index 2 value true comment { access 'read write' type BOOLEAN count 1 } } control.18 { iface PCM device 8 name ELD value '' comment { access 'read volatile' type BYTES count 0 } } control.19 { iface PCM device 3 name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 value.4 0 value.5 0 value.6 0 value.7 0 comment { access 'read write' type INTEGER count 8 range '0 - 36' } } control.20 { iface PCM device 7 name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 value.4 0 value.5 0 value.6 0 value.7 0 comment { access 'read write' type INTEGER count 8 range '0 - 36' } } control.21 { iface PCM device 8 name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 value.4 0 value.5 0 value.6 0 value.7 0 comment { access 'read write' type INTEGER count 8 range '0 - 36' } } } state.PCH { control.1 { iface MIXER name 'Headphone Playback Volume' value.0 87 value.1 87 comment { access 'read write' type INTEGER count 2 range '0 - 87' dbmin -6525 dbmax 0 dbvalue.0 0 dbvalue.1 0 } } control.2 { iface MIXER name 'Headphone Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.3 { iface MIXER name 'Speaker Playback Volume' value.0 87 value.1 87 comment { access 'read write' type INTEGER count 2 range '0 - 87' dbmin -6525 dbmax 0 dbvalue.0 0 dbvalue.1 0 } } control.4 { iface MIXER name 'Speaker Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.5 { iface MIXER name 'Mic Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -3450 dbvalue.1 -3450 } } control.6 { iface MIXER name 'Mic Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.7 { iface MIXER name 'Auto-Mute Mode' value Enabled comment { access 'read write' type ENUMERATED count 1 item.0 Disabled item.1 Enabled } } control.8 { iface MIXER name 'Capture Volume' value.0 39 value.1 39 comment { access 'read write' type INTEGER count 2 range '0 - 63' dbmin -1725 dbmax 3000 dbvalue.0 1200 dbvalue.1 1200 } } control.9 { iface MIXER name 'Capture Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.10 { iface MIXER name 'Mic Boost Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 3' dbmin 0 dbmax 3000 dbvalue.0 0 dbvalue.1 0 } } control.11 { iface MIXER name 'Internal Mic Boost Volume' value.0 2 value.1 2 comment { access 'read write' type INTEGER count 2 range '0 - 3' dbmin 0 dbmax 3600 dbvalue.0 2400 dbvalue.1 2400 } } control.12 { iface MIXER name 'Master Playback Volume' value 61 comment { access 'read write' type INTEGER count 1 range '0 - 87' dbmin -6525 dbmax 0 dbvalue.0 -1950 } } control.13 { iface MIXER name 'Master Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.14 { iface CARD name 'Mic Jack' value false comment { access read type BOOLEAN count 1 } } control.15 { iface CARD name 'Internal Mic Phantom Jack' value true comment { access read type BOOLEAN count 1 } } control.16 { iface CARD name 'Headphone Jack' value true comment { access read type BOOLEAN count 1 } } control.17 { iface CARD name 'Speaker Phantom Jack' value true comment { access read type BOOLEAN count 1 } } control.18 { iface MIXER name 'Beep Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -3450 dbvalue.1 -3450 } } control.19 { iface MIXER name 'Beep Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.20 { iface PCM name 'Playback Channel Map' value.0 3 value.1 4 comment { access read type INTEGER count 2 range '0 - 36' } } control.21 { iface PCM name 'Capture Channel Map' value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } } state.Q9000 { control.1 { iface PCM name 'Capture Channel Map' value 0 comment { access read type INTEGER count 1 range '0 - 36' } } control.2 { iface MIXER name 'Mic Capture Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.3 { iface MIXER name 'Mic Capture Volume' value 1536 comment { access 'read write' type INTEGER count 1 range '0 - 3072' dbmin 1800 dbmax 3000 dbvalue.0 2400 } } } state.ThinkPadEC { control.1 { iface MIXER name 'Console Playback Switch' value true comment { access read type BOOLEAN count 1 } } } --endcollapse-- !!All Loaded Modules !!------------------ Module rtsx_pci_sdmmc mmc_core ctr ccm uvcvideo videobuf2_vmalloc videobuf2_memops videobuf2_core videodev media snd_usb_audio snd_usbmidi_lib snd_rawmidi snd_seq_dummy snd_hrtimer snd_seq snd_seq_device usb_storage nls_iso8859_1 nls_cp437 vfat fat btusb bluetooth 6lowpan_iphc snd_hda_codec_hdmi joydev iTCO_wdt mousedev iTCO_vendor_support i915 arc4 iwlmvm coretemp x86_pkg_temp_thermal intel_powerclamp drm_kms_helper mac80211 kvm_intel drm iwlwifi cfg80211 snd_hda_codec_realtek snd_hda_codec_generic snd_hda_intel kvm microcode evdev thinkpad_acpi mac_hid nvram led_class intel_gtt hwmon serio_raw i2c_i801 i2c_algo_bit i2c_core psmouse pcspkr rtsx_pci_ms memstick tpm_tis tpm snd_hda_codec mei_me mei thermal rfkill e1000e shpchp wmi snd_hwdep snd_pcm snd_timer snd battery button soundcore video ptp lpc_ich ac pps_core processor vboxnetflt vboxnetadp pci_stub vboxpci vboxdrv ext4 crc16 mbcache jbd2 hid_generic usbhid hid dm_crypt dm_mod sr_mod sd_mod crc_t10dif cdrom crct10dif_pclmul crct10dif_common crc32_pclmul crc32c_intel ghash_clmulni_intel atkbd libps2 ahci aesni_intel libahci aes_x86_64 lrw gf128mul glue_helper ablk_helper cryptd libata ehci_pci ehci_hcd xhci_hcd scsi_mod rtsx_pci usbcore usb_common i8042 serio !!Sysfs Files !!----------- /sys/class/sound/hwC0D0/init_pin_configs: 0x05 0x18560010 0x06 0x18560010 0x07 0x18560010 /sys/class/sound/hwC0D0/driver_pin_configs: /sys/class/sound/hwC0D0/user_pin_configs: /sys/class/sound/hwC0D0/init_verbs: /sys/class/sound/hwC0D0/hints: /sys/class/sound/hwC1D0/init_pin_configs: 0x12 0x90a60130 0x13 0x40000000 0x14 0x90170110 0x15 0x0321101f 0x16 0x411111f0 0x18 0x411111f0 0x19 0x411111f0 0x1a 0x03a11020 0x1b 0x411111f0 0x1d 0x40738105 0x1e 0x411111f0 /sys/class/sound/hwC1D0/driver_pin_configs: /sys/class/sound/hwC1D0/user_pin_configs: /sys/class/sound/hwC1D0/init_verbs: /sys/class/sound/hwC1D0/hints: !!ALSA/HDA dmesg !!-------------- [ 24.883012] input: ThinkPad Extra Buttons as /devices/platform/thinkpad_acpi/input/input10 [ 24.920888] snd_hda_intel 0000:00:1b.0: irq 45 for MSI/MSI-X [ 24.945919] input: HDA Digital PCBeep as /devices/pci0000:00/0000:00:1b.0/input/input11 [ 24.952007] input: HDA Intel PCH Headphone as /devices/pci0000:00/0000:00:1b.0/sound/card1/input13 [ 24.952314] input: HDA Intel PCH Mic as /devices/pci0000:00/0000:00:1b.0/sound/card1/input12 [ 24.953189] cfg80211: Calling CRDA to update world regulatory domain -- [ 27.821545] [drm] Initialized i915 1.6.0 20080730 for 0000:00:02.0 on minor 0 [ 27.822590] snd_hda_intel 0000:00:03.0: irq 48 for MSI/MSI-X [ 27.870131] mousedev: PS/2 mouse device common for all mice -- [ 27.999737] Adding 8388604k swap on /dev/mapper/vg0-swap. Priority:-1 extents:1 across:8388604k FS [ 28.066542] input: HDA Intel HDMI HDMI/DP,pcm=8 as /devices/pci0000:00/0000:00:03.0/sound/card0/input18 [ 28.066845] input: HDA Intel HDMI HDMI/DP,pcm=7 as /devices/pci0000:00/0000:00:03.0/sound/card0/input17 [ 28.067107] input: HDA Intel HDMI HDMI/DP,pcm=3 as /devices/pci0000:00/0000:00:03.0/sound/card0/input16 [ 28.360819] [drm] Enabling RC6 states: RC6 on, RC6p off, RC6pp off -- [ 43.023277] sd 5:0:0:1: [sdc] Attached SCSI removable disk [ 3583.280102] hda-intel: IRQ timing workaround is activated for card #1. Suggest a bigger bdl_pos_adj. [ 3641.367542] warning: `VirtualBox' uses 32-bit capabilities (legacy support in use) -- [32892.031210] usb_audio: [5] FU [Mic Capture Volume] ch = 1, val = 4608/7680/1 [32892.031612] usbcore: registered new interface driver snd-usb-audio [32892.042989] uvcvideo: Found UVC 1.00 device <unnamed> (046d:0990) -- [33399.436941] usb usb2: root hub lost power or was reset [33399.437037] snd_hda_intel 0000:00:03.0: irq 41 for MSI/MSI-X [33399.437179] xhci_hcd 0000:00:14.0: irq 43 for MSI/MSI-X -- [33399.437386] e1000e 0000:00:19.0: irq 45 for MSI/MSI-X [33399.439612] snd_hda_intel 0000:00:1b.0: irq 48 for MSI/MSI-X [33399.439616] usb usb4: root hub lost power or was reset -- [33399.833408] eDP powered off while attempting aux channel communication. [33399.833512] Modules linked in: uvcvideo videobuf2_vmalloc videobuf2_memops videobuf2_core videodev media snd_usb_audio snd_usbmidi_lib snd_rawmidi snd_seq_dummy snd_hrtimer snd_seq snd_seq_device usb_storage nls_iso8859_1 nls_cp437 vfat fat btusb bluetooth 6lowpan_iphc snd_hda_codec_hdmi joydev iTCO_wdt mousedev iTCO_vendor_support i915 arc4 iwlmvm coretemp x86_pkg_temp_thermal intel_powerclamp drm_kms_helper mac80211 kvm_intel drm iwlwifi cfg80211 snd_hda_codec_realtek snd_hda_codec_generic snd_hda_intel kvm microcode evdev thinkpad_acpi mac_hid nvram led_class intel_gtt hwmon serio_raw i2c_i801 i2c_algo_bit i2c_core psmouse pcspkr rtsx_pci_ms memstick tpm_tis tpm snd_hda_codec mei_me mei thermal rfkill e1000e shpchp wmi snd_hwdep snd_pcm snd_timer snd battery button soundcore video ptp lpc_ich [33399.833609] ac pps_core processor vboxnetflt(O) vboxnetadp(O) pci_stub vboxpci(O) vboxdrv(O) ext4 crc16 mbcache jbd2 hid_generic usbhid hid dm_crypt dm_mod sr_mod sd_mod crc_t10dif cdrom crct10dif_pclmul crct10dif_common rtsx_pci_sdmmc crc32_pclmul crc32c_intel ghash_clmulni_intel mmc_core atkbd libps2 ahci aesni_intel libahci aes_x86_64 lrw gf128mul glue_helper ablk_helper cryptd libata ehci_pci ehci_hcd xhci_hcd scsi_mod rtsx_pci usbcore usb_common i8042 serio -- [33399.834171] eDP powered off while attempting aux channel communication. [33399.834261] Modules linked in: uvcvideo videobuf2_vmalloc videobuf2_memops videobuf2_core videodev media snd_usb_audio snd_usbmidi_lib snd_rawmidi snd_seq_dummy snd_hrtimer snd_seq snd_seq_device usb_storage nls_iso8859_1 nls_cp437 vfat fat btusb bluetooth 6lowpan_iphc snd_hda_codec_hdmi joydev iTCO_wdt mousedev iTCO_vendor_support i915 arc4 iwlmvm coretemp x86_pkg_temp_thermal intel_powerclamp drm_kms_helper mac80211 kvm_intel drm iwlwifi cfg80211 snd_hda_codec_realtek snd_hda_codec_generic snd_hda_intel kvm microcode evdev thinkpad_acpi mac_hid nvram led_class intel_gtt hwmon serio_raw i2c_i801 i2c_algo_bit i2c_core psmouse pcspkr rtsx_pci_ms memstick tpm_tis tpm snd_hda_codec mei_me mei thermal rfkill e1000e shpchp wmi snd_hwdep snd_pcm snd_timer snd battery button soundcore video ptp lpc_ich [33399.834316] ac pps_core processor vboxnetflt(O) vboxnetadp(O) pci_stub vboxpci(O) vboxdrv(O) ext4 crc16 mbcache jbd2 hid_generic usbhid hid dm_crypt dm_mod sr_mod sd_mod crc_t10dif cdrom crct10dif_pclmul crct10dif_common rtsx_pci_sdmmc crc32_pclmul crc32c_intel ghash_clmulni_intel mmc_core atkbd libps2 ahci aesni_intel libahci aes_x86_64 lrw gf128mul glue_helper ablk_helper cryptd libata ehci_pci ehci_hcd xhci_hcd scsi_mod rtsx_pci usbcore usb_common i8042 serio -- [33689.287384] PM: early resume of devices complete after 0.090 msecs [33689.287501] snd_hda_intel 0000:00:03.0: irq 41 for MSI/MSI-X [33689.287620] mei_me 0000:00:16.0: irq 44 for MSI/MSI-X -- [33689.287661] e1000e 0000:00:19.0: irq 45 for MSI/MSI-X [33689.288519] snd_hda_intel 0000:00:1b.0: irq 48 for MSI/MSI-X [33689.607497] ata1: SATA link up 1.5 Gbps (SStatus 113 SControl 300) -- [39741.199147] PM: early restore of devices complete after 0.277 msecs [39741.331890] snd_hda_intel 0000:00:03.0: irq 41 for MSI/MSI-X [39741.334458] usb usb1: root hub lost power or was reset -- [39741.335190] usb usb3: root hub lost power or was reset [39741.335279] snd_hda_intel 0000:00:1b.0: irq 48 for MSI/MSI-X [39741.335338] usb usb4: root hub lost power or was reset -- [45131.719540] usb usb2: root hub lost power or was reset [45131.719647] snd_hda_intel 0000:00:03.0: irq 41 for MSI/MSI-X [45131.719774] xhci_hcd 0000:00:14.0: irq 43 for MSI/MSI-X -- [45131.720005] usb usb4: root hub lost power or was reset [45131.720403] snd_hda_intel 0000:00:1b.0: irq 45 for MSI/MSI-X [45131.723605] e1000e 0000:00:19.0: irq 48 for MSI/MSI-X ---------------------------------------------------------------------->8- -- Sascha Wilde OpenPGP key: 4BB86568 http://www.intevation.de/~wilde/ http://www.intevation.de/ Intevation GmbH, Neuer Graben 17, 49074 Osnabrück; AG Osnabrück, HR B 18998 Geschäftsführer: Frank Koormann, Bernhard Reiter, Dr. Jan-Oliver Wagner

2 2

Re: [alsa-devel] ALSA: hda/realtek - Allow NULL bus->pci
by Dan Carpenter 08 Oct '14

08 Oct '14

Hello Takashi Iwai, The patch 5100cd07d43a: "ALSA: hda/realtek - Allow NULL bus->pci" from Feb 15, 2014, leads to the following static checker warning: sound/pci/hda/patch_realtek.c:5371 patch_alc269() warn: variable dereferenced before check 'codec->bus->pci' (see line 5364) sound/pci/hda/patch_realtek.c 5363 if (has_cdefine_beep(codec)) ^^^^^ We actually dereference "codec->bus->pci" inside the call to has_cdefine_beep(). 5364 spec->gen.beep_nid = 0x01; 5365 5366 switch (codec->vendor_id) { 5367 case 0x10ec0269: 5368 spec->codec_variant = ALC269_TYPE_ALC269VA; 5369 switch (alc_get_coef0(codec) & 0x00f0) { 5370 case 0x0010: 5371 if (codec->bus->pci && ^^^^^^^^^^^^^^^ This new check is too late. 5372 codec->bus->pci->subsystem_vendor == 0x1025 && 5373 spec->cdefine.platform_type == 1) 5374 err = alc_codec_rename(codec, "ALC271X"); 5375 spec->codec_variant = ALC269_TYPE_ALC269VB; 5376 break; There is another related warning: sound/pci/hda/patch_realtek.c:6270 patch_alc662() warn: variable dereferenced before check 'codec->bus->pci' (see line 6267) regards, dan carpenter

2 1

Re: [alsa-devel] Exposing the ThinkPad HW mute switch to ALSA?
by Takashi Iwai 08 Oct '14

08 Oct '14

At Tue, 23 Sep 2014 19:16:52 -0700, Andy Lutomirski wrote: > > ThinkPads have an extra mute switch that's controlled by the embedded > controller. There are some hardware buttons associated with it. > > We (thinkpad-acpi) can be notified about state changes in the mute > switch, we can see presses of the hardware buttons and program them to > send KEY_MUTE or not as appropriate, and we can (optionally) change > the HW mute state. > > However, I have no clue how to correctly expose the mute switch > through ALSA so that userspace will know what it controls. It > currently shows up as its own sound card, which is ridiculous. > > Can anyone give me any hints as to how to expose this bonus audio > control in a sensible manner? I know essentially nothing about ALSA. > > (Some older ThinkPads also have a hardware volume control with similar > behavior.) It's a bit difficult issue, since this needs the weak module dependency. And even worse, different sound drivers might be used for a single thinkpad_acpi. So, from the programming POV, the individual sound card instance is the easiest approach. Though, some HD-audio codec drivers already have a weak binding of thinkpad_acpi stuff for Mic-mute switch. I guess we can move the handling of the master mute switch to the sound driver as well. If doing so, each relevant sound driver needs to implement it. IIRC, snd-intel8x0 is used on old Thinkpads. Any others? Takashi

5 12

[alsa-devel] [PATCH 0/4] ASoC trivial fixes
by Takashi Iwai 08 Oct '14

08 Oct '14

Hi, this is a series of fixes for the bugs I pointed yesterday. Since no fix action was seen yet, I quickly put band-aid over them. thanks, Takashi

2 22

[alsa-devel] [PATCH 00/14] ALSA: Dice enhancement
by Takashi Sakamoto 07 Oct '14

07 Oct '14

This 14 patchset is to enable ALSA Dice driver to handle more models with some functionalities: - up to 192.0 kHz - PCM capture - MIDI playback/capture - Duplex streams with synchronization (with snd-firewire-lib) This driver doesn't support: - Synchronization between multiple devices on the same bus For this purpose, please join in user-space driver project (FFADO) - Any control functionality for internal DSP For this purpose, please use ffado-dbus-server/ffado-mixer, or write your own application Remarks: - When this driver is loaded, user-space driver fails to register notification address, thus cannot handle device for streaming. See: http://mailman.alsa-project.org/pipermail/alsa-devel/2014-September/081731.… Newly supported models (as long as I know): Supported devices: TC Electronic Desktop Konnekt 6 Impact Twin Konnekt 8/24D/Live Studio Konnekt 48 PreSonus FireStudio 26x26/Mobile/Project/Tube/LightPipe StudioLive 1602/1642/2442 FireStation Mackie Onyx Blackbird Onyx i-series (latter models) Focusrite Saffire Pro 14/24/24 DSP/26/40/56 M-Audio Profire 610/2626 Alessis IO 24/26 MultiMix 12/16 Weiss ADC2 Vesta Mineva AFI 1 TAG DAC 1 INT 202 DAC 202 Flex 5000 DnR Axus FireWire I/O card 16x16 Lexicon I-ONIX FW 810S Allen and Heath Zed R16 Midas Venice F32 Updates from my previous RFC: http://mailman.alsa-project.org/pipermail/alsa-devel/2014-September/081673.… - Improve bus reset handling, especially for registering notification address At bus reset, Dice devices initialize a register for notification address, then drivers need to register the address again. Just after powering on, devices generates any bus reset, then drivers often fail to detect stream formation because this operation requires to change sampling rate. My improvement is retries to register notification address when notification wait is timed out. - Remove experimental state - Fix minor bugs Takashi Sakamoto (14): ALSA: dice: Rename structure and its members ALSA: dice: Move file to its own directory ALSA: dice: Split transaction functionality into a file ALSA: dice: Split stream functionality into a file ALSA: dice: Split PCM functionality into a file ALSA: dice: Split hwdep functionality into a file ALSA: dice: Split proc functionality into a file ALSA: dice: Add new functions for restriction of PCM parameters ALSA: dice: Change the way to start stream ALSA: dice: Add support for duplex streams with synchronization ALSA: dice: Support for non SYT-Match sampling clock source mode ALSA: dice: Add support for PCM capture ALSA: dice: Add support for MIDI capture/playback ALSA: dice: remove experimental state sound/firewire/Kconfig | 7 +- sound/firewire/Makefile | 3 +- sound/firewire/dice.c | 1511 ---------------------------- sound/firewire/dice/Makefile | 3 + sound/firewire/dice/dice-hwdep.c | 190 ++++ sound/firewire/{ => dice}/dice-interface.h | 0 sound/firewire/dice/dice-midi.c | 176 ++++ sound/firewire/dice/dice-pcm.c | 404 ++++++++ sound/firewire/dice/dice-proc.c | 252 +++++ sound/firewire/dice/dice-stream.c | 424 ++++++++ sound/firewire/dice/dice-transaction.c | 380 +++++++ sound/firewire/dice/dice.c | 359 +++++++ sound/firewire/dice/dice.h | 189 ++++ 13 files changed, 2380 insertions(+), 1518 deletions(-) delete mode 100644 sound/firewire/dice.c create mode 100644 sound/firewire/dice/Makefile create mode 100644 sound/firewire/dice/dice-hwdep.c rename sound/firewire/{ => dice}/dice-interface.h (100%) create mode 100644 sound/firewire/dice/dice-midi.c create mode 100644 sound/firewire/dice/dice-pcm.c create mode 100644 sound/firewire/dice/dice-proc.c create mode 100644 sound/firewire/dice/dice-stream.c create mode 100644 sound/firewire/dice/dice-transaction.c create mode 100644 sound/firewire/dice/dice.c create mode 100644 sound/firewire/dice/dice.h -- 1.9.1

3 16

[alsa-devel] [PATCH 1/3] ASoC: rt5677: Include gpio driver header
by Ben Zhang 07 Oct '14

07 Oct '14

The header file is needed because struct gpio_chip is placed in struct rt5677_priv. Signed-off-by: Ben Zhang <benzh(a)chromium.org> --- sound/soc/codecs/rt5677.h | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index d4eb6d5..99fd023 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -13,6 +13,7 @@ #define __RT5677_H__ #include <sound/rt5677.h> +#include <linux/gpio/driver.h> /* Info */ #define RT5677_RESET 0x00 -- 2.1.0.rc2.206.gedb03e5

2 5

[alsa-devel] [PATCH 1/4] ASoC: mc13783: Delete unneeded test before of_node_put
by Fabio Estevam 07 Oct '14

07 Oct '14

of_node_put() supports NULL as its argument, so the initial test is not necessary. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/mc13783.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 2c1f1b0..388f90a 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -784,8 +784,7 @@ static int __init mc13783_codec_probe(struct platform_device *pdev) mc13783_dai_async, ARRAY_SIZE(mc13783_dai_async)); out: - if (np) - of_node_put(np); + of_node_put(np); return ret; } -- 1.9.1

3 8

[alsa-devel] [PATCH 1/2] ASoC: fsl: imx-wm8962: Delete unneeded test before of_node_put
by Fabio Estevam 07 Oct '14

07 Oct '14

of_node_put() supports NULL as its argument, so the initial test is not necessary. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/imx-wm8962.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 3a3d17c..48179ff 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -281,10 +281,8 @@ static int imx_wm8962_probe(struct platform_device *pdev) clk_fail: clk_disable_unprepare(data->codec_clk); fail: - if (ssi_np) - of_node_put(ssi_np); - if (codec_np) - of_node_put(codec_np); + of_node_put(ssi_np); + of_node_put(codec_np); return ret; } -- 1.9.1

3 3

[alsa-devel] [PATCH] ASoC: soc-pcm: fix sig_bits determination in soc_pcm_apply_msb()
by Daniel Mack 07 Oct '14

07 Oct '14

In the SNDRV_PCM_STREAM_CAPTURE branch in soc_pcm_apply_msb(), look at sig_bits of the capture stream, not the playback one. Spotted by coverity. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- sound/soc/soc-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 642c862..002311a 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -352,7 +352,7 @@ static void soc_pcm_apply_msb(struct snd_pcm_substream *substream) } else { for (i = 0; i < rtd->num_codecs; i++) { codec_dai = rtd->codec_dais[i]; - if (codec_dai->driver->playback.sig_bits == 0) { + if (codec_dai->driver->capture.sig_bits == 0) { bits = 0; break; } -- 2.1.0

2 1

[alsa-devel] [GIT PULL] ASoC updates for v3.18
by Mark Brown 07 Oct '14

07 Oct '14

The following changes since commit bfe01a5ba2490f299e1d2d5508cbbbadd897bbe9: Linux 3.17 (2014-10-05 12:23:04 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.18 for you to fetch changes up to a5448c88b812390a3622e76d774e10c0da1fb970: Merge remote-tracking branches 'asoc/topic/txx9', 'asoc/topic/wm8978' and 'asoc/topic/wm8994' into asoc-next (2014-10-06 12:49:09 +0100) ---------------------------------------------------------------- ASoC: Updates for v3.18 - More componentisation work from Lars-Peter, this time mainly cleaning up the suspend and bias level transition callbacks. - Real system support for the Intel drivers and a bunch of fixes and enhancements for the associated CODEC drivers, this is going to need a lot quirks over time due to the lack of any firmware description of the boards. - Jack detect support for simple card from Dylan Reid. - A bunch of small fixes and enhancements for the Freescale drivers. - New drivers for Analog Devices SSM4567, Cirrus Logic CS35L32, Everest Semiconductor ES8328 and Freescale cards using the ASRC in newer i.MX processors. ---------------------------------------------------------------- Anatol Pomozov (6): ASoC: rt5677: Add a configuration option for LDO2_POW pin ASoC: trace: Remove trailing new-lines in trace messages ASoC: rt5677: Add gpio-controller DTS documentation ASoC: ssm4567: Add driver for Analog Devices SSM4567 amplifier ASoC: rt5677: Add dts properties for input/output differential configuration ASoC: ssm4567: Remove duplicated else-if branch Arnd Bergmann (2): ASoC: simple-card: fix regression in clock rate lookup ASoC: davinci: vcif must be a module if SND_DAVINCI_SOC is Axel Lin (4): ASoC: cs35l32: Remove unneeded regulator_bulk_free call in cs35l32_i2c_remove ASoC: cs42l56: Remove unneeded regulator_bulk_free call in cs42l56_remove ASoC: cs35l32: Simplify implementation of cs35l32_codec_set_sysclk ASoC: Remove return value checking for gpiochip_remove() Bard Liao (2): ASoC: rt5677: Add DMIC2 clock selection ASoC: rt286: Add depends on I2C Brian Austin (6): ASoC: cs35l32: Add support for CS35L32 Boosted Amplifier ASoC: cs35l32: Add bindings for CS35L32 ASoC: cs35l32: fix compile warning for i2c_probe ASoC: cs35l32: use true/false returns for bool functions ASoC: cs42l52: use true/false returns for bool functions ASoC: cs42l56: use true/false returns for bool functions Dan Murphy (1): ASoC: tas2552: Add DAPM calls for amp and PLL Dmitry Lavnikevich (1): ASoC: tlv320aic3x: fix PLL D configuration Dylan Reid (3): ASoC: simple-card: Add mic and hp detect gpios. ASoC: tegra: add mic detect gpio to tegra_max98090 ASoC: simple-card: Fix detect gpio documentation. Fabian Frederick (1): ASoC: cs35l32: remove second linux/slab.h inclusion Fabio Estevam (6): ASoC: fsl_ssi: Remove unneeded 'i2s-slave' property ASoC: fsl ssi doc: Remove unused properties ASoC: fsl: Do not force codecs selection by SND_SOC_FSL_ASOC_CARD ASoC: sgtl5000: Improve the error message on slave mode setting ASoC: sgtl5000: Do a sanity check on SYS_MCLK ASoC: fsl_esai doc: Add "fsl,vf610-esai" as compatible string Fabio Falzoi (1): ASoC: fsl-ssi: Support for SND_SOC_DAIFMT_CBM_CFS Fengguang Wu (1): ASoC: Intel: byt-rt5640: fix coccinelle warnings Geert Uytterhoeven (2): regmap: Split regmap_get_endian() in two functions ASoC: Allow SND_SOC_WM8978 to be selected manually Howard Mitchell (1): ASoC: core: Fix volsw_range funcs so SOC_DOUBLE_R_RANGE_TLV works. Jarkko Nikula (16): ASoC: Remove unused cache_only from struct snd_soc_codec ASoC: max98090: Add recovery for PLL lock failure ASoC: max98090: Move interrupt request from codec probe to i2c probe ASoC: max98090: Remove structure member irq from private data ASoC: max98090: Remove structure member extmic_mux from private data ASoC: max98090: Remove unused version define ASoC: max98090: Remove unused byte access macros ASoC: max98090: Enable both DMIC channels also when using mono configuration ASoC: Intel: byt-max98090: Set card as fully routed ASoC: Intel: byt-rt5640: Remove IN2N pin from DAPM route table ASoC: Intel: byt-rt5640: Add quirk for Asus T100 ASoC: rt5640: Add function for enabling DMIC from ACPI probed machine ASoC: Intel: byt-rt5640: Enable DMIC interface for default DAPM route ASoC: Intel: byt-rt5640: Add quirk for Dell Venue 8 Pro tablet ASoC: Intel: byt-rt5640: Set card as fully routed ASoC: dapm: Fix NULL pointer dereference when registering card with widgets Javier Martinez Canillas (1): regmap: Fix DT endianess parsing logic Jianqun (3): ASoC: rockchip-i2s: enable "hclk" for rockchip I2S controller ASoC: rockchip-i2s: fix rockchip i2s defination more reasonable ASoC: rockchip-i2s: add dma data to snd_soc_dai Julia Lawall (1): ASoC: rsnd: delete unneeded test before of_node_put Jyri Sarha (1): ASoC: tlv320aic31xx: Choose PLL p divider automatically Kuninori Morimoto (7): ASoC: simple-card: use asoc_simple_xxx prefix ASoC: simple-card: remove dai_link->cpu_dai_name when DT ASoC: simple-card: dai_link->init should be cared when multi DAI ASoC: simple-card: use common for_each_child_of_node() for loop ASoC: simple-card: fixup cpu_dai_name clear case ASoC: simple-card: tidyup use priv in parameter ASoC: simple-card: tidyup get dai_link/dai_props from priv Lars-Peter Clausen (59): ASoC: dmic: Add to SND_SOC_ALL_CODECS ASoC: edma-pcm: Include edma-pcm.h ASoC: odrodix2_max98090: Make non exported symbols static ASoC: sh: Fix dma direction type ASoC: samsung idma: Add proper annotation for casting iomem pointers ASoC: ab8500-codec: Drop bank prefix from AB8500_GPIO_DIR4_REG register define ASoC: Move debugfs registration to the component level ASoC: Consolidate platform and CODEC probe/remove ASoC: Make rtd->codec optional ASoC: Add component level probe/remove support ASoC: Move AUX dev support to the component level ASoC: Pass component instead of DAPM context to AUX dev init callback ASoC: Move component->probed check into soc_{remove,probe}_component() ASoC: Cleanup DAI module reference counting ASoC: Consolidate CPU and CODEC DAI removal ASoC: Consolidate CPU and CODEC DAI lookup ASoC: Automatically initialize regmap for all components ASoC: Remove support for legacy snd_soc_platform IO ASoC: Replace list_empty(&card->codec_dev_list) with !card->instantiated ASoC: sh: Don't opencode DMAengine API calls ASoC: txx9: Don't opencode DMAengine API calls ASoC: ac97-codec: Remove ASoC level IO support ASoC: Restore idle_bias_off initialization ASoC: ab8500-codec: Revert back to regmap ASoC: adau1373: Remove unnecessary suspend/resume bias level changes ASoC: lm49453: Remove unnecessary suspend/resume bias level changes ASoC: tlv320aic3x: Remove unnecessary suspend/resume bias level changes ASoC: wm8804: Remove unnecessary suspend/resume bias level changes ASoC: wm8995: Remove unnecessary suspend/resume bias level changes ASoC: Set card->instantiated to false when removing the card ASoC: Shutdown DAPM contexts when removing a card ASoC: Add support for automatically going to BIAS_OFF on suspend ASoC: Always run default suspend/resume code ASoC: adau1373: Cleanup manual bias level transitions ASoC: adau17x1: Cleanup manual bias level transitions ASoC: adav80x: Cleanup manual bias level transitions ASoC: ssm2518: Cleanup manual bias level transitions ASoC: ssm2602: Cleanup manual bias level transitions ASoC: 88pm860x-codec: Cleanup manual bias level transitions ASoC: jz4740: Cleanup manual bias level transitions ASoC: ml26124: Cleanup manual bias level transitions ASoC: sgtl5000: Cleanup bias level transitions ASoC: es8328: Cleanup manual bias level transitions ASoC: Add snd_soc_component_{get,set}_drvdata() ASoC: sst-haswell-pcm: Alloc state struct in driver probe() ASoC: sst-haswell-pcm: Move controls and DAPM elements to component ASoC: Remove table based DAPM/control setup support from snd_soc_platform_driver ASoC: cs42l52: Cleanup manual bias level transitions ASoC: cs42l56: Cleanup manual bias level transitions ASoC: cs42l73: Cleanup manual bias level transitions ASoC: da732x: Remove unnecessary idle_bias_off initialization ASoC: da732x: Remove unused codec field form da732x_priv struct ASoC: da732x: Cleanup manual bias level transitions ASoC: Remove locking in snd_soc_{new,free}_ac97_codec() ASoC: Fix snd_soc_{new,free}_ac97_codec() locking removal ASoC: wm8741: Remove unused wm8741_suspend define ASoC: wm8{350,753,971}: Use snd_soc_dapm_to_codec() instead of dapm->codec ASoC: rx51: Use snd_soc_dapm_to_codec() instead of dapm->codec ASoC: Remove CODEC pointer from snd_soc_dapm_context Mark Brown (31): Merge branch 'topic/es8328' of git://git.kernel.org/.../broonie/sound into asoc-fsl Merge branch 'fix/fsl-esai' of git://git.kernel.org/.../broonie/sound into asoc-fsl-esai Merge tag 'dt-endian' of git://git.kernel.org/.../broonie/regmap into asoc-fsl Merge branch 'topic/fsl' of git://git.kernel.org/.../broonie/sound into asoc-fsl-esai Merge branch 'topic/fsl' of git://git.kernel.org/.../broonie/sound into asoc-fsl-sai Merge branch 'fix/tlv320aic31xx' of git://git.kernel.org/.../broonie/sound into asoc-tlv320aic31xx Merge branch 'topic/component' of git://git.kernel.org/.../broonie/sound into asoc-suspend Merge branch 'topic/suspend' of git://git.kernel.org/.../broonie/sound into asoc-es8328 Merge branch 'topic/component' of git://git.kernel.org/.../broonie/sound into asoc-intel Merge branch 'topic/cs42l56' of git://git.kernel.org/.../broonie/sound into asoc-suspend Merge remote-tracking branch 'asoc/fix/core' into asoc-linus Merge remote-tracking branch 'asoc/fix/pcm' into asoc-linus Merge remote-tracking branches 'asoc/fix/davinci', 'asoc/fix/max98090', 'asoc/fix/samsung' and 'asoc/fix/tlv320aic31xx' into asoc-linus Merge branch 'fix/max98090' of git://git.kernel.org/.../broonie/sound into asoc-intel Merge branch 'topic/rt5640' of git://git.kernel.org/.../broonie/sound into asoc-intel Merge tag 'v3.17-rc4' into asoc-simple Merge tag 'asoc-v3.17-rc4' into asoc-linus Merge remote-tracking branches 'asoc/fix/davinci' and 'asoc/fix/max98090' into asoc-linus Merge remote-tracking branch 'asoc/topic/component' into asoc-next Merge remote-tracking branch 'asoc/topic/core' into asoc-next Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/fsl-esai' into asoc-next Merge remote-tracking branches 'asoc/topic/ab8500', 'asoc/topic/ac97' and 'asoc/topic/cs35l32' into asoc-next Merge remote-tracking branches 'asoc/topic/cs4265', 'asoc/topic/cs42l52', 'asoc/topic/cs42l56' and 'asoc/topic/da732x' into asoc-next Merge remote-tracking branches 'asoc/topic/davinci', 'asoc/topic/dmic', 'asoc/topic/drivers', 'asoc/topic/es8328' and 'asoc/topic/fsl' into asoc-next Merge remote-tracking branches 'asoc/topic/fsl-easi', 'asoc/topic/fsl-sai', 'asoc/topic/fsl-ssi' and 'asoc/topic/intel' into asoc-next Merge remote-tracking branches 'asoc/topic/max98090', 'asoc/topic/rockchip', 'asoc/topic/rsnd' and 'asoc/topic/rt286' into asoc-next Merge remote-tracking branches 'asoc/topic/rt5645', 'asoc/topic/rt5677', 'asoc/topic/samsung', 'asoc/topic/sgtl5000' and 'asoc/topic/sh' into asoc-next Merge remote-tracking branches 'asoc/topic/simple', 'asoc/topic/sirf', 'asoc/topic/spdif', 'asoc/topic/ssm2602' and 'asoc/topic/ssm4567' into asoc-next Merge remote-tracking branches 'asoc/topic/suspend', 'asoc/topic/tas2552', 'asoc/topic/tegra', 'asoc/topic/tlv320aic31xx' and 'asoc/topic/tlv320aic3x' into asoc-next Merge remote-tracking branches 'asoc/topic/txx9', 'asoc/topic/wm8978' and 'asoc/topic/wm8994' into asoc-next Masanari Iida (1): ASoC: da732x: Remove unnecessary KERN_ERR in pr_err() Nicolin Chen (6): ASoC: fsl: Add Freescale Generic ASoC Sound Card with ASRC support ASoC: fsl_sai: Initialize with software reset ASoC: fsl_sai: Set SYNC bit of TCR2 to Asynchronous Mode ASoC: fsl_sai: Add asynchronous mode support ASoC: fsl_sai: Make Synchronous and Asynchronous modes exclusive ASoC: fsl_sai: Set SYNC bit of TCR2 to Asynchronous Mode Nikesh Oswal (1): ASoC: wm8994: Demux the microphone detection IRQ Oder Chiou (5): ASoC: rt5677: Revise the wrong name in the header file ASoC: rt5677: Add the GPIO function ASoC: rt5677: Add the TDM function ASoC: rt5677: Add sidetone function ASoC: rt5645: Add headset detect function Paul Handrigan (1): ASoC: cs4265: Add CHIP_ID as a readable register Peter Ujfalusi (4): ASoC: tlv320aic31xx: Correct interface register 2 variable name ASoC: tlv320aic31xx: Enable support for S24_LE format ASoC: davinci-mcasp: Convert the context save/restore to use array ASoC: davinvi-mcasp: Proper suspend/resume support while audio is active Rongjun Ying (1): ASoC: sirf: usp: Add bitclock inversion support Sean Cross (4): devicetree: bindings: Add Everest Semicodunctor ASoC: add es8328 codec driver ASoC: fsl: add imx-es8328 machine driver ASoC: fsl: Fix building of imx-es8328 on PPC Shengjiu Wang (6): ASoC: fsl_esai: refine esai for TDM support ASoC: fsl_esai: refine esai for TDM support ASoC: fsl-asoc-card: Fix build warning for maybe-uninitialized ASoC: fsl: fsl-asoc-card: Select SND_SOC_IMX_AUDMUX ASoC: fsl-asoc-card: move 'config SND_SOC_FSL_ASOC_CARD' to 'if SND_IMX_SOC' ASoC: fsl_ssi: refine ipg clock usage in this module Stefan Kristiansson (2): ASoC: ssm2602: add device tree bindings ASoC: ssm2602: add support for 11.025kHz and 22.5kHz sample rates Stephen Warren (1): regmap: of_regmap_get_endian() cleanup Subhransu S. Prusty (6): ASoC: Intel: mfld-pcm: Use function instead of ioctl ASoC: Intel: mfld-pcm: Change sst_ops prototypes to take dev parameter ASoC: mfld-compress: Use dedicated function instead of ioctl ASoC: mfld: pcm: Replace pr_ with dev_ ASoC: Intel: mrfld: Use snd_soc_dai_get_drvdata to derive drv data ASoC: Export dapm_kcontrol_get_value Sylwester Nawrocki (1): ASoC: Drop const from struct snd_soc_dai_link *of_node members Thierry Reding (2): ASoC: tas2552: Fix compilation warning for !PM_RUNTIME ASoC: rt286: Fix compilation warning for !PM Tomeu Vizoso (1): ASoC: fsl_spdif: Remove unused includes of linux/clk-private.h Vinod Koul (5): ASoC: Intel: Fix to use byte control interface ASoC: Intel: add mrfld DSP defines ASoC: Intel: mfld-pcm: don't call trigger ops to DSP for internal streams ASoC: Intel: mrfld: add bytes control for modules ASoC: Intel: mfld-pcm: add control for powering up/down dsp Wei Yongjun (1): ASoC: es8328: fix error return code in es8328_codec_probe() Xiubo Li (10): regmap: add DT endianness binding support. regmap: Add the DT binding documentation for endianness ASoC: simple-card: Fix the compile warning. ASoC: fsl-asrc: Convert to use regmap framework's endianness method. ASoC: fsl-esai: Convert to use regmap framework's endianness method. ASoC: fsl-spdif: Convert to use regmap framework's endianness method. ASoC: fsl-sai: Convert to use regmap framework's endianness method. ASoC: fsl-sai: using 'lsb-first' property instead of 'big-endian-data'. ASoC: simple-card: Merge single and muti DAI link(s) code. ASoC: simple-card: Adjust the comments of simple card. .../devicetree/bindings/regmap/regmap.txt | 47 ++ .../devicetree/bindings/sound/adi,ssm2602.txt | 19 + .../devicetree/bindings/sound/cs35l32.txt | 62 ++ Documentation/devicetree/bindings/sound/es8328.txt | 38 ++ .../devicetree/bindings/sound/fsl,esai.txt | 3 +- .../devicetree/bindings/sound/fsl,ssi.txt | 8 +- .../devicetree/bindings/sound/fsl-asoc-card.txt | 82 +++ .../devicetree/bindings/sound/fsl-sai.txt | 30 +- .../devicetree/bindings/sound/imx-audio-es8328.txt | 60 ++ .../bindings/sound/nvidia,tegra-audio-max98090.txt | 1 + Documentation/devicetree/bindings/sound/rt5677.txt | 59 ++ .../devicetree/bindings/sound/simple-card.txt | 4 + .../devicetree/bindings/sound/ssm4567.txt | 15 + .../devicetree/bindings/vendor-prefixes.txt | 1 + drivers/base/regmap/regmap-i2c.c | 2 + drivers/base/regmap/regmap-spi.c | 2 + drivers/base/regmap/regmap.c | 74 +- include/dt-bindings/sound/cs35l32.h | 26 + include/sound/rt5645.h | 3 + include/sound/rt5677.h | 13 +- include/sound/soc-dapm.h | 5 +- include/sound/soc.h | 101 +-- include/trace/events/asoc.h | 6 +- sound/soc/codecs/88pm860x-codec.c | 3 - sound/soc/codecs/Kconfig | 36 +- sound/soc/codecs/Makefile | 10 + sound/soc/codecs/ab8500-codec.c | 73 +- sound/soc/codecs/ac97.c | 15 - sound/soc/codecs/adau1373.c | 21 - sound/soc/codecs/adau1761.c | 2 +- sound/soc/codecs/adau1781.c | 2 +- sound/soc/codecs/adau17x1.c | 8 - sound/soc/codecs/adau17x1.h | 1 - sound/soc/codecs/adav80x.c | 23 +- sound/soc/codecs/cs35l32.c | 631 +++++++++++++++++ sound/soc/codecs/cs35l32.h | 93 +++ sound/soc/codecs/cs4265.c | 1 + sound/soc/codecs/cs42l52.c | 24 +- sound/soc/codecs/cs42l56.c | 27 +- sound/soc/codecs/cs42l73.c | 25 +- sound/soc/codecs/da732x.c | 27 +- sound/soc/codecs/es8328-i2c.c | 60 ++ sound/soc/codecs/es8328-spi.c | 49 ++ sound/soc/codecs/es8328.c | 756 +++++++++++++++++++++ sound/soc/codecs/es8328.h | 314 +++++++++ sound/soc/codecs/jz4740.c | 30 +- sound/soc/codecs/lm49453.c | 14 - sound/soc/codecs/max98090.c | 146 +++- sound/soc/codecs/max98090.h | 13 +- sound/soc/codecs/ml26124.c | 24 +- sound/soc/codecs/rt286.c | 2 + sound/soc/codecs/rt5640.c | 49 +- sound/soc/codecs/rt5640.h | 3 + sound/soc/codecs/rt5645.c | 99 +++ sound/soc/codecs/rt5645.h | 5 + sound/soc/codecs/rt5677.c | 327 ++++++++- sound/soc/codecs/rt5677.h | 171 ++++- sound/soc/codecs/sgtl5000.c | 42 +- sound/soc/codecs/ssm2518.c | 13 - sound/soc/codecs/ssm2602-i2c.c | 9 + sound/soc/codecs/ssm2602-spi.c | 7 + sound/soc/codecs/ssm2602.c | 39 +- sound/soc/codecs/ssm4567.c | 343 ++++++++++ sound/soc/codecs/tas2552.c | 70 +- sound/soc/codecs/tlv320aic31xx.c | 107 ++- sound/soc/codecs/tlv320aic31xx.h | 3 +- sound/soc/codecs/tlv320aic3x.c | 29 +- sound/soc/codecs/wm5100.c | 5 +- sound/soc/codecs/wm8350.c | 2 +- sound/soc/codecs/wm8741.c | 1 - sound/soc/codecs/wm8753.c | 2 +- sound/soc/codecs/wm8804.c | 19 - sound/soc/codecs/wm8903.c | 6 +- sound/soc/codecs/wm8962.c | 5 +- sound/soc/codecs/wm8971.c | 2 +- sound/soc/codecs/wm8994.c | 18 +- sound/soc/codecs/wm8995.c | 19 - sound/soc/codecs/wm8996.c | 6 +- sound/soc/davinci/Kconfig | 3 +- sound/soc/davinci/davinci-mcasp.c | 79 ++- sound/soc/davinci/edma-pcm.c | 2 + sound/soc/fsl/Kconfig | 26 + sound/soc/fsl/Makefile | 4 + sound/soc/fsl/fsl-asoc-card.c | 574 ++++++++++++++++ sound/soc/fsl/fsl_asrc.c | 6 +- sound/soc/fsl/fsl_esai.c | 19 +- sound/soc/fsl/fsl_esai.h | 8 +- sound/soc/fsl/fsl_sai.c | 58 +- sound/soc/fsl/fsl_sai.h | 8 +- sound/soc/fsl/fsl_spdif.c | 6 +- sound/soc/fsl/fsl_ssi.c | 89 ++- sound/soc/fsl/imx-es8328.c | 232 +++++++ sound/soc/generic/simple-card.c | 226 ++++-- sound/soc/intel/Makefile | 3 +- sound/soc/intel/byt-max98090.c | 1 + sound/soc/intel/byt-rt5640.c | 83 ++- sound/soc/intel/sst-atom-controls.c | 218 ++++++ sound/soc/intel/sst-atom-controls.h | 416 +++++++++++- sound/soc/intel/sst-haswell-pcm.c | 56 +- sound/soc/intel/sst-mfld-platform-compress.c | 38 +- sound/soc/intel/sst-mfld-platform-pcm.c | 106 +-- sound/soc/intel/sst-mfld-platform.h | 58 +- sound/soc/omap/rx51.c | 2 +- sound/soc/rockchip/Kconfig | 3 +- sound/soc/rockchip/Makefile | 2 +- sound/soc/rockchip/rockchip_i2s.c | 28 +- sound/soc/samsung/idma.c | 4 +- sound/soc/samsung/odroidx2_max98090.c | 4 +- sound/soc/samsung/speyside.c | 6 +- sound/soc/sh/fsi.c | 7 +- sound/soc/sh/rcar/core.c | 6 +- sound/soc/sh/siu_pcm.c | 4 +- sound/soc/sirf/sirf-usp.c | 24 +- sound/soc/soc-core.c | 669 ++++++++---------- sound/soc/soc-dapm.c | 26 +- sound/soc/soc-generic-dmaengine-pcm.c | 4 +- sound/soc/soc-io.c | 28 - sound/soc/tegra/tegra_max98090.c | 40 ++ sound/soc/txx9/txx9aclc.c | 14 +- 119 files changed, 6323 insertions(+), 1359 deletions(-) create mode 100644 Documentation/devicetree/bindings/regmap/regmap.txt create mode 100644 Documentation/devicetree/bindings/sound/adi,ssm2602.txt create mode 100644 Documentation/devicetree/bindings/sound/cs35l32.txt create mode 100644 Documentation/devicetree/bindings/sound/es8328.txt create mode 100644 Documentation/devicetree/bindings/sound/fsl-asoc-card.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-es8328.txt create mode 100644 Documentation/devicetree/bindings/sound/rt5677.txt create mode 100644 Documentation/devicetree/bindings/sound/ssm4567.txt create mode 100644 include/dt-bindings/sound/cs35l32.h create mode 100644 sound/soc/codecs/cs35l32.c create mode 100644 sound/soc/codecs/cs35l32.h 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/codecs/ssm4567.c create mode 100644 sound/soc/fsl/fsl-asoc-card.c create mode 100644 sound/soc/fsl/imx-es8328.c create mode 100644 sound/soc/intel/sst-atom-controls.c

4 12

[alsa-devel] [PATCH v3 0/3] ASoC/ARM: tegra: apalis/colibri t30: sgtl5000 audio
by Marcel Ziswiler 07 Oct '14

07 Oct '14

This series adds/integrates Freescale SGTL5000 analogue audio codec support. Changes in v3: - Revert to not using simple-audio-card being incompatible with tegra. - Rebased to for-next. Changes in v2: - Using simple-audio-card as suggested by Fabio. Marcel Ziswiler (3): ASoC: tegra: add tegra sgtl5000 machine driver ARM: tegra: apalis/colibri t30: integrate audio support ARM: tegra: enable sgtl5000 audio .../bindings/sound/nvidia,tegra-audio-sgtl5000.txt | 45 +++++ arch/arm/boot/dts/tegra30-apalis.dtsi | 49 +++++ arch/arm/boot/dts/tegra30-colibri.dtsi | 49 +++++ arch/arm/configs/tegra_defconfig | 1 + sound/soc/tegra/Kconfig | 10 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra_sgtl5000.c | 216 +++++++++++++++++++++ 7 files changed, 372 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-sgtl5000.txt create mode 100644 sound/soc/tegra/tegra_sgtl5000.c -- 1.9.3

2 4

[alsa-devel] [PATCH] ASoC: simple-card: Initialize headphone and mic GPIO numbers
by Geert Uytterhoeven 07 Oct '14

07 Oct '14

The uninitialized default of 0 for gpio_hp_det and gpio_mic_det doesn't play well with asm-generic's gpio_is_valid(): static inline bool gpio_is_valid(int number) { return number >= 0 && number < ARCH_NR_GPIOS; } Hence on r8a7740/armadillo-legacy: sh-mobile-hdmi sh-mobile-hdmi: SH Mobile HDMI Audio Codec sh-mobile-hdmi sh-mobile-hdmi: ASoC: DAPM unknown pin Headphones sh-mobile-hdmi sh-mobile-hdmi: ASoC: DAPM unknown pin Mic Jack After that the kernel log is spammed ca. 7 times per second with: sh-mobile-hdmi sh-mobile-hdmi: ASoC: DAPM unknown pin Headphones Initialize the GPIO numbers with a negative number (-ENOENT) to fix this. Fixes: 3fe240326cc395c6 ("ASoC: simple-card: Add mic and hp detect gpios.") Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- This is against sound-asoc/for-next sound/soc/generic/simple-card.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index fcb431fe20b4ae7b..d1b7293c133eaa45 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -501,6 +501,9 @@ static int asoc_simple_card_probe(struct platform_device *pdev) priv->snd_card.dai_link = dai_link; priv->snd_card.num_links = num_links; + priv->gpio_hp_det = -ENOENT; + priv->gpio_mic_det = -ENOENT; + /* Get room for the other properties */ priv->dai_props = devm_kzalloc(dev, sizeof(*priv->dai_props) * num_links, -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: sgtl5000: Cleanup the comments
by Fabio Estevam 07 Oct '14

07 Oct '14

Fix grammar and typos. Besides that, also fix the comment about the range of SYS_MCLK, which is from 8 to 27 MHz. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/sgtl5000.c | 27 ++++++++++++++------------- 1 file changed, 14 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 6bb77d7..10160e7 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -530,16 +530,16 @@ static int sgtl5000_set_dai_sysclk(struct snd_soc_dai *codec_dai, /* * set clock according to i2s frame clock, - * sgtl5000 provide 2 clock sources. - * 1. sys_mclk. sample freq can only configure to + * sgtl5000 provides 2 clock sources: + * 1. sys_mclk: sample freq can only be configured to * 1/256, 1/384, 1/512 of sys_mclk. - * 2. pll. can derive any audio clocks. + * 2. pll: can derive any audio clocks. * * clock setting rules: - * 1. in slave mode, only sys_mclk can use. - * 2. as constraint by sys_mclk, sample freq should - * set to 32k, 44.1k and above. - * 3. using sys_mclk prefer to pll to save power. + * 1. in slave mode, only sys_mclk can be used + * 2. as constraint by sys_mclk, sample freq should be set to 32 kHz, 44.1 kHz + * and above. + * 3. usage of sys_mclk is preferred over pll to save power. */ static int sgtl5000_set_clock(struct snd_soc_codec *codec, int frame_rate) { @@ -549,8 +549,8 @@ static int sgtl5000_set_clock(struct snd_soc_codec *codec, int frame_rate) /* * sample freq should be divided by frame clock, - * if frame clock lower than 44.1khz, sample feq should set to - * 32khz or 44.1khz. + * if frame clock is lower than 44.1 kHz, sample freq should be set to + * 32 kHz or 44.1 kHz. */ switch (frame_rate) { case 8000: @@ -603,7 +603,8 @@ static int sgtl5000_set_clock(struct snd_soc_codec *codec, int frame_rate) /* * calculate the divider of mclk/sample_freq, - * factor of freq =96k can only be 256, since mclk in range (12m,27m) + * factor of freq = 96 kHz can only be 256, since mclk is in the range + * of 8 MHz - 27 MHz */ switch (sgtl5000->sysclk / sys_fs) { case 256: @@ -619,7 +620,7 @@ static int sgtl5000_set_clock(struct snd_soc_codec *codec, int frame_rate) SGTL5000_MCLK_FREQ_SHIFT; break; default: - /* if mclk not satisify the divider, use pll */ + /* if mclk does not satisfy the divider, use pll */ if (sgtl5000->master) { clk_ctl |= SGTL5000_MCLK_FREQ_PLL << SGTL5000_MCLK_FREQ_SHIFT; @@ -795,7 +796,7 @@ static int ldo_regulator_enable(struct regulator_dev *dev) SGTL5000_LINEREG_D_POWERUP, SGTL5000_LINEREG_D_POWERUP); - /* when internal ldo enabled, simple digital power can be disabled */ + /* when internal ldo is enabled, simple digital power can be disabled */ snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_LINREG_SIMPLE_POWERUP, 0); @@ -1079,7 +1080,7 @@ static bool sgtl5000_readable(struct device *dev, unsigned int reg) /* * sgtl5000 has 3 internal power supplies: * 1. VAG, normally set to vdda/2 - * 2. chargepump, set to different value + * 2. charge pump, set to different value * according to voltage of vdda and vddio * 3. line out VAG, normally set to vddio/2 * -- 1.9.1

2 1

[alsa-devel] [PATCH/alsa-lib] conf/ucm: PAZ00: add Toshiba AC100/Dynabook AZ new config
by zombah 07 Oct '14

07 Oct '14

Recent mainline kernels threat Toshiba AC100 audio hardware as hw:PAZ00 vs old hw:tegraalc5632. This patch adds config files for new hw name and include them to makefiles. Signed-off-by: zombah <zombah(a)gmail.com> --- configure.ac | 1 + src/conf/ucm/Makefile.am | 2 +- src/conf/ucm/PAZ00/HiFi.conf | 45 +++++++++++++++++++++++++++++ src/conf/ucm/PAZ00/Makefile.am | 4 +++ src/conf/ucm/PAZ00/PAZ00.conf | 39 +++++++++++++++++++++++++ src/conf/ucm/PAZ00/Record.conf | 65 ++++++++++++++++++++++++++++++++++++++++++ 6 files changed, 155 insertions(+), 1 deletion(-) create mode 100644 src/conf/ucm/PAZ00/HiFi.conf create mode 100644 src/conf/ucm/PAZ00/Makefile.am create mode 100644 src/conf/ucm/PAZ00/PAZ00.conf create mode 100644 src/conf/ucm/PAZ00/Record.conf diff --git a/configure.ac b/configure.ac index b8353a0..734163c 100644 --- a/configure.ac +++ b/configure.ac @@ -653,6 +653,7 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ src/conf/ucm/PandaBoardES/Makefile \ src/conf/ucm/SDP4430/Makefile \ src/conf/ucm/tegraalc5632/Makefile \ + src/conf/ucm/PAZ00/Makefile \ modules/Makefile modules/mixer/Makefile modules/mixer/simple/Makefile \ alsalisp/Makefile aserver/Makefile \ test/Makefile test/lsb/Makefile \ diff --git a/src/conf/ucm/Makefile.am b/src/conf/ucm/Makefile.am index e089dde..bde89df 100644 --- a/src/conf/ucm/Makefile.am +++ b/src/conf/ucm/Makefile.am @@ -1 +1 @@ -SUBDIRS=DAISY-I2S PandaBoard PandaBoardES SDP4430 tegraalc5632 +SUBDIRS=DAISY-I2S PandaBoard PandaBoardES SDP4430 tegraalc5632 PAZ00 diff --git a/src/conf/ucm/PAZ00/HiFi.conf b/src/conf/ucm/PAZ00/HiFi.conf new file mode 100644 index 0000000..275516d --- /dev/null +++ b/src/conf/ucm/PAZ00/HiFi.conf @@ -0,0 +1,45 @@ +SectionVerb { + EnableSequence [ + cdev "hw:PAZ00" + + cset "name='Speaker Playback Switch' on" + cset "name='Int Spk Switch' on" + cset "name='Speaker Mix DAC2SPK Playback Switch' on" + ] + DisableSequence [ + cset "name='Speaker Playback Switch' off" + cset "name='Int Spk Switch' off" + cset "name='Speaker Mix DAC2SPK Playback Switch' off" + ] +} + +SectionDevice."Headset".0 { + Comment "3.5mm Headset" + + Value { + JackName "Paz00 Headset Jack" + } + + EnableSequence [ + cdev "hw:PAZ00" + + # Internal speaker + cset "name='Int Spk Switch' off" + + # Headphones + cset "name='Headphone Playback Switch' on" + cset "name='HPL Mix DACL2HP Playback Switch' on" + cset "name='HPR Mix DACR2HP Playback Switch' on" + ] + DisableSequence [ + cdev "hw:PAZ00" + + # Headphones + cset "name='Headphone Playback Switch' off" + cset "name='HPL Mix DACL2HP Playback Switch' off" + cset "name='HPR Mix DACR2HP Playback Switch' off" + + # Internal speaker + cset "name='Int Spk Switch' on" + ] +} diff --git a/src/conf/ucm/PAZ00/Makefile.am b/src/conf/ucm/PAZ00/Makefile.am new file mode 100644 index 0000000..1bf98a6 --- /dev/null +++ b/src/conf/ucm/PAZ00/Makefile.am @@ -0,0 +1,4 @@ +alsaconfigdir = @ALSA_CONFIG_DIR@ +ucmdir = $(alsaconfigdir)/ucm/PAZ00 +ucm_DATA = PAZ00.conf HiFi.conf Record.conf +EXTRA_DIST = $(ucm_DATA) diff --git a/src/conf/ucm/PAZ00/PAZ00.conf b/src/conf/ucm/PAZ00/PAZ00.conf new file mode 100644 index 0000000..8b9d904 --- /dev/null +++ b/src/conf/ucm/PAZ00/PAZ00.conf @@ -0,0 +1,39 @@ +Comment "PAZ00 alc5632" + +ValueDefaults { + PlaybackCTL "hw:0" + CaptureCTL "hw:0" +} + +SectionDefaults [ + cdev "hw:PAZ00" + + cset "name='Master Playback Switch' on" + cset "name='Master Playback Volume' 44,44" + cset "name='Headphone Playback Volume' 21,21" + cset "name='Speaker Playback Volume' 21,21" + cset "name='SpeakerOut Mux' 2" + cset "name='SpeakerOut N Mux' LN/-R" + cset "name='AB-D Amp Mux' 0" + cset "name='Left Headphone Mux' 1" + cset "name='Right Headphone Mux' 1" + cset "name='DMIC Boost Capture Volume' 4" + cset "name='Rec Capture Volume' 24" + + # Fix external mic record volumes + cset "name='Mic 1 Boost Volume' 0" + cset "name='Mic 2 Boost Volume' 0" + cset "name='Mic1 Playback Volume' 0" + cset "name='Mic2 Playback Volume' 0" + cset "name='Rec Capture Volume' 0" +] + +SectionUseCase."HiFi" { + File "HiFi.conf" + Comment "Music playback" +} + +SectionUseCase."Record" { + File "Record.conf" + Comment "Playback and capture" +} diff --git a/src/conf/ucm/PAZ00/Record.conf b/src/conf/ucm/PAZ00/Record.conf new file mode 100644 index 0000000..568e2fb --- /dev/null +++ b/src/conf/ucm/PAZ00/Record.conf @@ -0,0 +1,65 @@ +SectionVerb { + EnableSequence [ + cdev "hw:PAZ00" + + cset "name='Speaker Playback Switch' on" + cset "name='Int Spk Switch' on" + cset "name='Speaker Mix DAC2SPK Playback Switch' on" + + cset "name='DMICL Mix DMICL2ADC Capture Switch' on" + cset "name='DMIC En Capture Switch' on" + cset "name='DMIC PreFilter Capture Switch' on" + ] + DisableSequence [ + cset "name='Speaker Playback Switch' off" + cset "name='Int Spk Switch' off" + cset "name='Speaker Mix DAC2SPK Playback Switch' off" + + cset "name='DMICL Mix DMICL2ADC Capture Switch' off" + cset "name='DMIC En Capture Switch' off" + cset "name='DMIC PreFilter Capture Switch' off" + ] +} + +SectionDevice."Headset".0 { + Comment "3.5mm Headset" + + Value { + JackName "Paz00 Headset Jack" + } + + EnableSequence [ + cdev "hw:PAZ00" + + # Internal speaker + cset "name='Int Spk Switch' off" + + # Internal mic + cset "name='DMIC En Capture Switch' off" + + # Headphones + cset "name='Headphone Playback Switch' on" + cset "name='HPL Mix DACL2HP Playback Switch' on" + cset "name='HPR Mix DACR2HP Playback Switch' on" + + # External min + cset "name='Left Capture Mix MIC12REC_L Capture Switch' on" + ] + DisableSequence [ + cdev "hw:PAZ00" + + # Headphones + cset "name='Headphone Playback Switch' off" + cset "name='HPL Mix DACL2HP Playback Switch' off" + cset "name='HPR Mix DACR2HP Playback Switch' off" + + # External mic + cset "name='Left Capture Mix MIC12REC_L Capture Switch' on" + + # Internal speaker + cset "name='Int Spk Switch' on" + + # Internal mic + cset "name='DMIC En Capture Switch' on" + ] +} -- 2.0.1

2 1

[alsa-devel] [PATCH] ALSA: hda - Add dock support for Thinkpad T440 (17aa:2212)
by Takashi Iwai 07 Oct '14

07 Oct '14

There is another Thinkpad T440 with SSID 17aa:2212 that has a dock port. Reported-by: Siwei Luo <sluo(a)smartbeans.de> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 69d1236365e0..2cc2568af016 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4799,7 +4799,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x220c, "Thinkpad T440s", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x220e, "Thinkpad T440p", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2210, "Thinkpad T540p", ALC292_FIXUP_TPT440_DOCK), - SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad T440", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP), -- 2.1.2

1 0

[alsa-devel] [PATCH] ALSA: usb: caiaq: check for cdev->n_streams > 1
by Daniel Mack 07 Oct '14

07 Oct '14

Coverity spotted a possible DIV0 condition when cdev->n_streams is 0. Fix this by making sure the value is > 1 in snd_usb_caiaq_audio_init(). Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- sound/usb/caiaq/audio.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index 7103b09..2728447 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -816,6 +816,11 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *cdev) return -EINVAL; } + if (cdev->n_streams < 2) { + dev_err(dev, "bogus number of streams: %d\n", cdev->n_streams); + return -EINVAL; + } + ret = snd_pcm_new(cdev->chip.card, cdev->product_name, 0, cdev->n_audio_out, cdev->n_audio_in, &cdev->pcm); -- 2.1.0

2 1

[alsa-devel] [PATCH 0/3] ASoC: some trivial fixups
by Daniel Mack 07 Oct '14

07 Oct '14

Here are some trivial fixes for minor issues that were spotted by Coverity. Only compile-tested, but the issues are pretty obvious, and the fixes are simple. Applicable to the for-next branch of the asoc tree. Thanks, Daniel Daniel Mack (3): ASoC: core: fix use after free in snd_soc_remove_platform() ASoC: soc-dapm: fix use after free ASoC: 88pm860x-codec: Fix possibly missing string termination sound/soc/codecs/88pm860x-codec.c | 2 +- sound/soc/soc-core.c | 4 ++-- sound/soc/soc-dapm.c | 25 ++++++++++++++----------- 3 files changed, 17 insertions(+), 14 deletions(-) -- 2.1.0

2 6

[alsa-devel] [PATCH RESEND] ASoC: rt5670: Add dsp support
by Bard Liao 07 Oct '14

07 Oct '14

Add dsp part of rt5670 codec driver Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/Makefile | 2 +- sound/soc/codecs/rt5670-dsp.c | 383 ++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5670-dsp.h | 11 +- sound/soc/codecs/rt5670.c | 43 +++++ sound/soc/codecs/rt5670.h | 3 + 5 files changed, 435 insertions(+), 7 deletions(-) create mode 100644 sound/soc/codecs/rt5670-dsp.c diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 5dce451..19a4631 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -78,7 +78,7 @@ snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o snd-soc-rt5645-objs := rt5645.o snd-soc-rt5651-objs := rt5651.o -snd-soc-rt5670-objs := rt5670.o +snd-soc-rt5670-objs := rt5670.o rt5670-dsp.o snd-soc-rt5677-objs := rt5677.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o diff --git a/sound/soc/codecs/rt5670-dsp.c b/sound/soc/codecs/rt5670-dsp.c new file mode 100644 index 0000000..bbe039d --- /dev/null +++ b/sound/soc/codecs/rt5670-dsp.c @@ -0,0 +1,383 @@ +/* + * rt5670-dsp.c -- RT5670 ALSA SoC DSP driver + * + * Copyright 2014 Realtek Semiconductor Corp. + * Author: Bard Liao <bardliao(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/firmware.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include "rt5670.h" +#include "rt5670-dsp.h" + +#define DSP_CLK_RATE RT5670_DSP_CLK_96K + +static const struct firmware *rt5670_dsp_fw; + +/** + * rt5670_dsp_done - Wait until DSP is ready. + * @codec: SoC Audio Codec device. + * + * To check voice DSP status and confirm it's ready for next work. + * + * Returns 0 for success or negative error code. + */ +static int rt5670_dsp_done(struct snd_soc_codec *codec) +{ + unsigned int count = 0, dsp_val; + + dsp_val = snd_soc_read(codec, RT5670_DSP_CTRL1); + while (dsp_val & RT5670_DSP_BUSY_MASK) { + if (count > 10) + return -EBUSY; + dsp_val = snd_soc_read(codec, RT5670_DSP_CTRL1); + count++; + } + + return 0; +} + +/** + * rt5670_dsp_write - Write DSP register. + * @codec: SoC audio codec device. + * @param: DSP parameters. + * + * Modify voice DSP register for sound effect. The DSP can be controlled + * through DSP addr (0xe1), data (0xe2) and cmd (0xe0) + * registers. It has to wait until the DSP is ready. + * + * Returns 0 for success or negative error code. + */ +int rt5670_dsp_write(struct snd_soc_codec *codec, + unsigned int addr, unsigned int data) +{ + unsigned int dsp_val; + int ret; + + ret = snd_soc_write(codec, RT5670_DSP_CTRL2, addr); + if (ret < 0) { + dev_err(codec->dev, "Failed to write DSP addr reg: %d\n", ret); + goto err; + } + ret = snd_soc_write(codec, RT5670_DSP_CTRL3, data); + if (ret < 0) { + dev_err(codec->dev, "Failed to write DSP data reg: %d\n", ret); + goto err; + } + dsp_val = RT5670_DSP_I2C_AL_16 | RT5670_DSP_DL_2 | + RT5670_DSP_CMD_MW | DSP_CLK_RATE | RT5670_DSP_CMD_EN; + + ret = snd_soc_write(codec, RT5670_DSP_CTRL1, dsp_val); + if (ret < 0) { + dev_err(codec->dev, "Failed to write DSP cmd reg: %d\n", ret); + goto err; + } + ret = rt5670_dsp_done(codec); + if (ret < 0) { + dev_err(codec->dev, "DSP is busy: %d\n", ret); + goto err; + } + + return 0; + +err: + return ret; +} + +/** + * rt5670_dsp_read - Read DSP register. + * @codec: SoC audio codec device. + * @reg: DSP register index. + * + * Read DSP setting value from voice DSP. The DSP can be controlled + * through DSP addr (0xe1), data (0xe2) and cmd (0xe0) registers. Each + * command has to wait until the DSP is ready. + * + * Returns DSP register value or negative error code. + */ +unsigned int rt5670_dsp_read( + struct snd_soc_codec *codec, unsigned int reg) +{ + unsigned int value; + unsigned int dsp_val; + int ret = 0; + + ret = rt5670_dsp_done(codec); + if (ret < 0) { + dev_err(codec->dev, "DSP is busy: %d\n", ret); + goto err; + } + + ret = snd_soc_write(codec, RT5670_DSP_CTRL2, reg); + if (ret < 0) { + dev_err(codec->dev, "Failed to write DSP addr reg: %d\n", ret); + goto err; + } + dsp_val = RT5670_DSP_I2C_AL_16 | RT5670_DSP_DL_0 | RT5670_DSP_RW_MASK | + RT5670_DSP_CMD_MR | DSP_CLK_RATE | RT5670_DSP_CMD_EN; + + ret = snd_soc_write(codec, RT5670_DSP_CTRL1, dsp_val); + if (ret < 0) { + dev_err(codec->dev, "Failed to write DSP cmd reg: %d\n", ret); + goto err; + } + + ret = rt5670_dsp_done(codec); + if (ret < 0) { + dev_err(codec->dev, "DSP is busy: %d\n", ret); + goto err; + } + + ret = snd_soc_write(codec, RT5670_DSP_CTRL2, 0x26); + if (ret < 0) { + dev_err(codec->dev, "Failed to write DSP addr reg: %d\n", ret); + goto err; + } + dsp_val = RT5670_DSP_DL_1 | RT5670_DSP_CMD_RR | RT5670_DSP_RW_MASK | + DSP_CLK_RATE | RT5670_DSP_CMD_EN; + + ret = snd_soc_write(codec, RT5670_DSP_CTRL1, dsp_val); + if (ret < 0) { + dev_err(codec->dev, "Failed to write DSP cmd reg: %d\n", ret); + goto err; + } + + ret = rt5670_dsp_done(codec); + if (ret < 0) { + dev_err(codec->dev, "DSP is busy: %d\n", ret); + goto err; + } + + ret = snd_soc_write(codec, RT5670_DSP_CTRL2, 0x25); + if (ret < 0) { + dev_err(codec->dev, "Failed to write DSP addr reg: %d\n", ret); + goto err; + } + + dsp_val = RT5670_DSP_DL_1 | RT5670_DSP_CMD_RR | RT5670_DSP_RW_MASK | + DSP_CLK_RATE | RT5670_DSP_CMD_EN; + + ret = snd_soc_write(codec, RT5670_DSP_CTRL1, dsp_val); + if (ret < 0) { + dev_err(codec->dev, "Failed to write DSP cmd reg: %d\n", ret); + goto err; + } + + ret = rt5670_dsp_done(codec); + if (ret < 0) { + dev_err(codec->dev, "DSP is busy: %d\n", ret); + goto err; + } + + ret = snd_soc_read(codec, RT5670_DSP_CTRL5); + if (ret < 0) { + dev_err(codec->dev, "Failed to read DSP data reg: %d\n", ret); + goto err; + } + + value = ret; + return value; + +err: + return ret; +} + +static int rt5670_dsp_mode_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = rt5670->dsp_sw; + + return 0; +} + +static int rt5670_dsp_mode_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + if (rt5670->dsp_sw != ucontrol->value.integer.value[0]) + rt5670->dsp_sw = ucontrol->value.integer.value[0]; + + return 0; +} + +/* DSP SRC Control */ +static const char * const rt5670_src_rxdp_mode[] = { + "Normal", "Divided by 2", "Divided by 3" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_src_rxdp_enum, RT5670_DSP_PATH1, + RT5670_RXDP_SRC_SFT, rt5670_src_rxdp_mode); + +static const char * const rt5670_src_txdp_mode[] = { + "Normal", "Multiplied by 2", "Multiplied by 3" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_src_txdp_enum, RT5670_DSP_PATH1, + RT5670_TXDP_SRC_SFT, rt5670_src_txdp_mode); + +/* DSP Mode */ +static const char * const rt5670_dsp_mode[] = { + "Mode 1", "Mode 2", "Mode 3", "Mode 4", "Mode 5" +}; + +static const SOC_ENUM_SINGLE_DECL(rt5670_dsp_enum, 0, 0, + rt5670_dsp_mode); + +static const struct snd_kcontrol_new rt5670_dsp_snd_controls[] = { + SOC_ENUM("RxDP SRC Switch", rt5670_src_rxdp_enum), + SOC_ENUM("TxDP SRC Switch", rt5670_src_txdp_enum), + SOC_ENUM_EXT("DSP Function Switch", rt5670_dsp_enum, + rt5670_dsp_mode_get, rt5670_dsp_mode_put), +}; + +/** + * rt5670_dsp_set_mode - Set DSP mode parameters. + * + * @codec: SoC audio codec device. + * @mode: DSP mode. + * + * Set parameters of mode to DSP. + * + * Returns 0 for success or negative error code. + */ +static int rt5670_dsp_set_mode(struct snd_soc_codec *codec, int mode) +{ + int tab_num, n, ret = -EINVAL; + unsigned int pos = 0; + + if (rt5670_dsp_fw) { + n = rt5670_dsp_fw->data[0]; + + if (mode <= n) { + pos = rt5670_dsp_fw->data[mode * 3 + 2] | + rt5670_dsp_fw->data[mode * 3 + 1] << 8; + + tab_num = rt5670_dsp_fw->data[mode * 3 + 3]; + if (pos + tab_num * 5 > rt5670_dsp_fw->size) + return -EINVAL; + ret = rt5670_write_fw(codec, + rt5670_dsp_fw, pos, tab_num); + if (ret < 0) + dev_err(codec->dev, + "Fail to set mode %d parameters: %d\n", + mode, ret); + } + } + + return ret; +} + +static int rt5670_dsp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMD: + rt5670_dsp_write(codec, 0x22f9, 1); + break; + + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, RT5670_DIG_MISC, + RT5670_RST_DSP, RT5670_RST_DSP); + snd_soc_update_bits(codec, RT5670_DIG_MISC, + RT5670_RST_DSP, 0); + mdelay(10); + rt5670_dsp_set_mode(codec, rt5670->dsp_sw); + break; + + default: + return 0; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt5670_dsp_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY_S("Voice DSP", 1, SND_SOC_NOPM, + 0, 0, rt5670_dsp_event, + SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA("DSP Downstream", SND_SOC_NOPM, + 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DSP Upstream", SND_SOC_NOPM, + 0, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_route rt5670_dsp_dapm_routes[] = { + {"DSP Downstream", NULL, "Voice DSP"}, + {"DSP Downstream", NULL, "RxDP Mux"}, + {"DSP Upstream", NULL, "Voice DSP"}, + {"DSP Upstream", NULL, "TDM Data Mux"}, + {"DSP DL Mux", "DSP", "DSP Downstream"}, + {"DSP UL Mux", "DSP", "DSP Upstream"}, +}; + +static void rt5670_dsp_fw_loaded(const struct firmware *fw, void *context) +{ + if (fw) { + rt5670_dsp_fw = fw; + pr_debug("fw->size=%d\n", fw->size); + } + +} + +/** + * rt5670_dsp_probe - register DSP for rt5670 + * @codec: audio codec + * + * To register DSP function for rt5670. + * + * Returns 0 for success or negative error code. + */ +int rt5670_dsp_probe(struct snd_soc_codec *codec) +{ + if (codec == NULL) + return -EINVAL; + + snd_soc_update_bits(codec, RT5670_PWR_DIG2, + RT5670_PWR_I2S_DSP, RT5670_PWR_I2S_DSP); + + snd_soc_update_bits(codec, RT5670_DIG_MISC, RT5670_RST_DSP, + RT5670_RST_DSP); + snd_soc_update_bits(codec, RT5670_DIG_MISC, RT5670_RST_DSP, 0); + + mdelay(10); + + rt5670_dsp_set_mode(codec, 0); + /* power down DSP */ + mdelay(15); + rt5670_dsp_write(codec, 0x22f9, 1); + + snd_soc_update_bits(codec, RT5670_PWR_DIG2, + RT5670_PWR_I2S_DSP, 0); + + snd_soc_add_codec_controls(codec, rt5670_dsp_snd_controls, + ARRAY_SIZE(rt5670_dsp_snd_controls)); + snd_soc_dapm_new_controls(&codec->dapm, rt5670_dsp_dapm_widgets, + ARRAY_SIZE(rt5670_dsp_dapm_widgets)); + snd_soc_dapm_add_routes(&codec->dapm, rt5670_dsp_dapm_routes, + ARRAY_SIZE(rt5670_dsp_dapm_routes)); + + request_firmware_nowait(THIS_MODULE, FW_ACTION_HOTPLUG, + "rt567x_dsp.bin", codec->dev, GFP_KERNEL, + codec, rt5670_dsp_fw_loaded); + + return 0; +} diff --git a/sound/soc/codecs/rt5670-dsp.h b/sound/soc/codecs/rt5670-dsp.h index a34d0cd..7b789f3 100644 --- a/sound/soc/codecs/rt5670-dsp.h +++ b/sound/soc/codecs/rt5670-dsp.h @@ -43,12 +43,11 @@ #define RT5670_DSP_I2C_AL_16 (0x1 << 1) #define RT5670_DSP_CMD_EN (0x1) -struct rt5670_dsp_param { - u16 cmd_fmt; - u16 addr; - u16 data; - u8 cmd; -}; +int rt5670_dsp_probe(struct snd_soc_codec *codec); +int rt5670_dsp_write(struct snd_soc_codec *codec, + unsigned int addr, unsigned int data); +unsigned int rt5670_dsp_read( + struct snd_soc_codec *codec, unsigned int reg); #endif /* __RT5670_DSP_H__ */ diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index ba9d9b4..da4b689 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -16,6 +16,7 @@ #include <linux/pm.h> #include <linux/i2c.h> #include <linux/platform_device.h> +#include <linux/firmware.h> #include <linux/spi/spi.h> #include <sound/core.h> #include <sound/pcm.h> @@ -401,6 +402,44 @@ static bool rt5670_readable_register(struct device *dev, unsigned int reg) } } +int rt5670_write_fw(struct snd_soc_codec *codec, const struct firmware *fw, + unsigned int pos, unsigned int num) +{ + int i, ret; + + for (i = 0; i < num; i++) { + switch (fw->data[pos]) { + case 1: /*PR*/ + ret = snd_soc_write(codec, RT5670_PR_BASE + + ((fw->data[pos + 1] << 8) | fw->data[pos + 2]), + (fw->data[pos + 3] << 8) | fw->data[pos + 4]); + if (ret < 0) + return -1; + break; + case 2: /*dsp*/ + ret = rt5670_dsp_write(codec, + (fw->data[pos + 1] << 8) | fw->data[pos + 2], + (fw->data[pos + 3] << 8) | fw->data[pos + 4]); + if (ret < 0) + return -1; + break; + default: /*register*/ + ret = snd_soc_write(codec, + (fw->data[pos + 1] << 8) | fw->data[pos + 2], + (fw->data[pos + 3] << 8) | fw->data[pos + 4]); + if (ret < 0) + return -1; + break; + } + pos += 5; + if (fw->size < pos) + return 0; + } + + return 0; + +} + static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0); static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); @@ -2332,6 +2371,7 @@ static int rt5670_probe(struct snd_soc_codec *codec) struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); rt5670->codec = codec; + rt5670_dsp_probe(codec); return 0; } @@ -2623,6 +2663,9 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, } + /*Give sysclk a default value*/ + rt5670->sysclk = 24576000; + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5670, rt5670_dai, ARRAY_SIZE(rt5670_dai)); if (ret < 0) diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h index a0b5c85..70285c9 100644 --- a/sound/soc/codecs/rt5670.h +++ b/sound/soc/codecs/rt5670.h @@ -1997,4 +1997,7 @@ struct rt5670_priv { int jack_type; }; +int rt5670_write_fw(struct snd_soc_codec *codec, const struct firmware *fw, + unsigned int pos, unsigned int num); + #endif /* __RT5670_H__ */ -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH 1/3] ALSA: hda - Add inverted internal mic for Asus Aspire 4830T
by David Henningsson 07 Oct '14

07 Oct '14

Alsa-info at https://launchpadlibrarian.net/186697318/alsa-info.txt.37fYWkaJRc Reported-by: Tomas Nilsson <tomas.k.nilsson(a)gmail.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index d2eaf8b..71e4bad 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -752,6 +752,7 @@ 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_ASPIRE_DMIC), + SND_PCI_QUIRK(0x1025, 0x054f, "Acer Aspire 4830T", 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), -- 1.9.1

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