March 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] aplay/arecord: option to treat any xrun as fatal
by Ben Gardiner 19 Mar '12

19 Mar '12

Signed-off-by: Ben Gardiner <bengardiner(a)nanometrics.ca> --- aplay/aplay.1 | 4 ++++ aplay/aplay.c | 17 +++++++++++++++-- 2 files changed, 19 insertions(+), 2 deletions(-) diff --git a/aplay/aplay.1 b/aplay/aplay.1 index 0195322..bf9b53e 100644 --- a/aplay/aplay.1 +++ b/aplay/aplay.1 @@ -191,6 +191,10 @@ lists capabilities of the selected device such as supported formats, sampling rates, numbers of channels, period and buffer bytes/sizes/times. For raw device hw:X this option basically lists hardware capabilities of the soundcard. +.TP +\fI\-\-fatal\-errors\fP +Disables recovery attempts when errors (e.g. xrun) are encountered; the +aplay process instead aborts immediately. .SH SIGNALS When recording, SIGINT, SIGTERM and SIGABRT will close the output diff --git a/aplay/aplay.c b/aplay/aplay.c index 0633cbd..1ce34c4 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -115,6 +115,7 @@ static int stop_delay = 0; static int monotonic = 0; static int interactive = 0; static int can_pause = 0; +static int fatal_errors = 0; static int verbose = 0; static int vumeter = VUMETER_NONE; static int buffer_pos = 0; @@ -225,7 +226,8 @@ _("Usage: %s [OPTION]... [FILE]...\n" " for this many seconds\n" " --process-id-file write the process ID here\n" " --use-strftime apply the strftime facility to the output file name\n" -" --dump-hw-params dump hw_params of the device\n") +" --dump-hw-params dump hw_params of the device\n" +" --fatal-errors treat all errors as fatal\n") , command); printf(_("Recognized sample formats are:")); for (k = 0; k < SND_PCM_FORMAT_LAST; ++k) { @@ -419,7 +421,8 @@ enum { OPT_MAX_FILE_TIME, OPT_PROCESS_ID_FILE, OPT_USE_STRFTIME, - OPT_DUMP_HWPARAMS + OPT_DUMP_HWPARAMS, + OPT_FATAL_ERRORS, }; int main(int argc, char *argv[]) @@ -465,6 +468,7 @@ int main(int argc, char *argv[]) {"use-strftime", 0, 0, OPT_USE_STRFTIME}, {"interactive", 0, 0, 'i'}, {"dump-hw-params", 0, 0, OPT_DUMP_HWPARAMS}, + {"fatal-errors", 0, 0, OPT_FATAL_ERRORS}, {0, 0, 0, 0} }; char *pcm_name = "default"; @@ -669,6 +673,9 @@ int main(int argc, char *argv[]) case OPT_DUMP_HWPARAMS: dump_hw_params = 1; break; + case OPT_FATAL_ERRORS: + fatal_errors = 1; + break; default: fprintf(stderr, _("Try `%s --help' for more information.\n"), command); return 1; @@ -1350,6 +1357,12 @@ static void xrun(void) prg_exit(EXIT_FAILURE); } if (snd_pcm_status_get_state(status) == SND_PCM_STATE_XRUN) { + if (fatal_errors) { + error(_("fatal %s: %s"), + stream == SND_PCM_STREAM_PLAYBACK ? _("underrun") : _("overrun"), + snd_strerror(res)); + prg_exit(EXIT_FAILURE); + } if (monotonic) { #ifdef HAVE_CLOCK_GETTIME struct timespec now, diff, tstamp; -- 1.7.4.1

2 1

[alsa-devel] [PATCH] ALSA: ASoC: pxa-ssp: atomically set stream active masks
by Daniel Mack 19 Mar '12

19 Mar '12

PXA's SSP engine fails to take its current channel phase into account when enabling a stream while the engine is already running. This results in randomly swapped left/right channels on either the record or the playback side, depending on which one was enabled first. The following patch fixes this by factoring out the bit field modifications in question to a separate function that pauses the engine temporarily, modifies the bits and kicks it off again afterwards. Appearantly, a transition of SSCR0_SSE syncs both directions properly. The patch has been rolled out to quite a number of devices over the last weeks and seems to fix the issue reliably. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Reported-and-tested-by: Sven Neumann <s.neumann(a)raumfeld.com> Cc: Eric Miao <eric.y.miao(a)gmail.com> Cc: Haojian Zhuang <haojian.zhuang(a)marvell.com> Cc: Liam Girdwood <lrg(a)ti.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Johannes Stezenbach <js(a)sig21.net> Cc: stable(a)kernel.org --- sound/soc/pxa/pxa-ssp.c | 61 +++++++++++++++++++++++++++------------------- 1 files changed, 36 insertions(+), 25 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index a16df0f..fd04ce1 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -668,6 +668,38 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, return 0; } +static void pxa_ssp_set_running_bit(struct snd_pcm_substream *substream, + struct ssp_device *ssp, int value) +{ + uint32_t sscr0 = pxa_ssp_read_reg(ssp, SSCR0); + uint32_t sscr1 = pxa_ssp_read_reg(ssp, SSCR1); + uint32_t sspsp = pxa_ssp_read_reg(ssp, SSPSP); + uint32_t sssr = pxa_ssp_read_reg(ssp, SSSR); + + if (value && (sscr0 & SSCR0_SSE)) + pxa_ssp_write_reg(ssp, SSCR0, sscr0 & ~SSCR0_SSE); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (value) + sscr1 |= SSCR1_TSRE; + else + sscr1 &= ~SSCR1_TSRE; + } else { + if (value) + sscr1 |= SSCR1_RSRE; + else + sscr1 &= ~SSCR1_RSRE; + } + + pxa_ssp_write_reg(ssp, SSCR1, sscr1); + + if (value) { + pxa_ssp_write_reg(ssp, SSSR, sssr); + pxa_ssp_write_reg(ssp, SSPSP, sspsp); + pxa_ssp_write_reg(ssp, SSCR0, sscr0 | SSCR0_SSE); + } +} + static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *cpu_dai) { @@ -681,42 +713,21 @@ static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd, pxa_ssp_enable(ssp); break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - val = pxa_ssp_read_reg(ssp, SSCR1); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - val |= SSCR1_TSRE; - else - val |= SSCR1_RSRE; - pxa_ssp_write_reg(ssp, SSCR1, val); + pxa_ssp_set_running_bit(substream, ssp, 1); val = pxa_ssp_read_reg(ssp, SSSR); pxa_ssp_write_reg(ssp, SSSR, val); break; case SNDRV_PCM_TRIGGER_START: - val = pxa_ssp_read_reg(ssp, SSCR1); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - val |= SSCR1_TSRE; - else - val |= SSCR1_RSRE; - pxa_ssp_write_reg(ssp, SSCR1, val); - pxa_ssp_enable(ssp); + pxa_ssp_set_running_bit(substream, ssp, 1); break; case SNDRV_PCM_TRIGGER_STOP: - val = pxa_ssp_read_reg(ssp, SSCR1); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - val &= ~SSCR1_TSRE; - else - val &= ~SSCR1_RSRE; - pxa_ssp_write_reg(ssp, SSCR1, val); + pxa_ssp_set_running_bit(substream, ssp, 0); break; case SNDRV_PCM_TRIGGER_SUSPEND: pxa_ssp_disable(ssp); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - val = pxa_ssp_read_reg(ssp, SSCR1); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - val &= ~SSCR1_TSRE; - else - val &= ~SSCR1_RSRE; - pxa_ssp_write_reg(ssp, SSCR1, val); + pxa_ssp_set_running_bit(substream, ssp, 0); break; default: -- 1.7.7.6

2 1

[alsa-devel] static /dev configuration
by Bob Tracy 19 Mar '12

19 Mar '12

Does ALSA now require dynamic devices (udev) for certain soundcards (specifically, cards handled by the snd-hda-intel driver)? My problem is that I moved a Slackware installation to new hardware, and the old devices under /dev/snd I originally created back in 2003 don't seem to work correctly with snd-hda-intel. "lspci" shows my new system having two audio devices as follows: 00:1b.0 Audio device: Intel Corporation Cougar Point High Definition Audio Controller (rev 04) 01:00.1 Audio device: ATI Technologies Inc Device aa60 (The ATI entry corresponds to a DisplayPort output on a Radeon graphics card that isn't presently of interest to me: REDWOOD firmware.) I had an opportunity to boot a modern Linux distro with udev support that accesses the Intel audio device just fine, so I took note of the device entries under /dev/snd and replicated those on the Slackware setup as follows (ignore the "root" group and 666 permissions rather than the more correct "audio" group and 660 permissions): crw-rw-rw- 1 root root 116, 9 Mar 18 00:24 controlC0 crw-rw-rw- 1 root root 116, 12 Mar 18 00:26 controlC1 crw-rw-rw- 1 root root 116, 8 Mar 18 00:27 hwC0D2 crw-rw-rw- 1 root root 116, 7 Jun 18 2003 hwC0D3 crw-rw-rw- 1 root root 116, 11 Mar 18 00:30 hwC1D0 crw-rw-rw- 1 root root 116, 6 Mar 18 00:34 pcmC0D0c crw-rw-rw- 1 root root 116, 5 Mar 18 20:44 pcmC0D0p crw-rw-rw- 1 root root 116, 4 Mar 18 00:36 pcmC0D1p crw-rw-rw- 1 root root 116, 3 Mar 18 00:39 pcmC0D2c crw-rw-rw- 1 root root 116, 2 Mar 18 00:44 pcmC0D3p crw-rw-rw- 1 root root 116, 10 Mar 18 00:42 pcmC1D3p crw-rw-rw- 1 root root 116, 1 Jun 18 2003 seq crw-rw-rw- 1 root root 116, 33 Jun 18 2003 timer Not all that surprisingly, the above entries work: "alsamixer" and "aplay" do what I expect and seem to work correctly. Note that the minor device numbers are NOT the same as what the "snddevices" script in "alsa-driver-1.0.25" creates (and that script creates pretty much exactly what I had in my static /dev tree back in 2003). According to "snddevices", the "controlCn" devices should have minor numbers 0, 32, 64, etc. The "hwC0Dn" minors are 4, 5, etc., "hwC1Dn" minors are 36, 37, etc., "hwC2Dn" minors are 68, 69, etc. Note that "hwC0D3" is at minor 7 where it should be under the old minor device numbering scheme, but "hwC0D2" is at minor 8 where 6 was expected. The various "pcm*" devices were expected to be as follows: crw-rw-rw- 1 root root 116, 24 Jun 18 2003 pcmC0D0c crw-rw-rw- 1 root root 116, 16 Mar 3 20:50 pcmC0D0p crw-rw-rw- 1 root root 116, 25 Jun 18 2003 pcmC0D1c crw-rw-rw- 1 root root 116, 17 Jun 18 2003 pcmC0D1p crw-rw-rw- 1 root root 116, 26 Jun 18 2003 pcmC0D2c crw-rw-rw- 1 root root 116, 18 Jun 18 2003 pcmC0D2p crw-rw-rw- 1 root root 116, 27 Jun 18 2003 pcmC0D3c crw-rw-rw- 1 root root 116, 19 Jun 18 2003 pcmC0D3p crw-rw-rw- 1 root root 116, 56 Jun 18 2003 pcmC1D0c crw-rw-rw- 1 root root 116, 48 Jun 18 2003 pcmC1D0p crw-rw-rw- 1 root root 116, 57 Jun 18 2003 pcmC1D1c crw-rw-rw- 1 root root 116, 49 Jun 18 2003 pcmC1D1p crw-rw-rw- 1 root root 116, 58 Jun 18 2003 pcmC1D2c crw-rw-rw- 1 root root 116, 50 Jun 18 2003 pcmC1D2p crw-rw-rw- 1 root root 116, 59 Jun 18 2003 pcmC1D3c crw-rw-rw- 1 root root 116, 51 Jun 18 2003 pcmC1D3p I'm thinking the only reason I'm getting away with using any kind of static device scheme is that my sound devices are getting detected and initialized in a consistent order each time I boot, and the /dev/snd entries I copied from the udev-enabled system just happen to agree with what "udev" would consistently create if I had "udev" installed. So, just to restate the original question for clarity, is "udev" required for certain soundcards? Thanks! --Bob

2 2

[alsa-devel] [PATCH] ASoC: Add MICBIAS routing information for WM8904
by Hong Xu 19 Mar '12

19 Mar '12

Add MICBIAS routing entries for WM8904. Tested on ATMEL's AT91SAM9N12-EK with a electret condenser microphone connected to IN1L. Signed-off-by: Hong Xu <hong.xu(a)atmel.com> --- sound/soc/codecs/wm8904.c | 8 ++++++++ 1 files changed, 8 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 285ef87..2158f30 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1318,6 +1318,14 @@ static const struct snd_soc_dapm_route core_intercon[] = { }; static const struct snd_soc_dapm_route adc_intercon[] = { + { "MICBIAS", NULL, "IN1L" }, + { "MICBIAS", NULL, "IN1R" }, + + { "Left Capture Mux", NULL, "MICBIAS" }, + { "Right Capture Mux", NULL, "MICBIAS" }, + { "Left Capture Inverting Mux", NULL, "MICBIAS" }, + { "Right Capture Inverting Mux", NULL, "MICBIAS" }, + { "Left Capture Mux", "IN1L", "IN1L" }, { "Left Capture Mux", "IN2L", "IN2L" }, { "Left Capture Mux", "IN3L", "IN3L" }, -- 1.7.6

3 2

Re: [alsa-devel] ASoC:Question rate constraint between the dais
by Omair Mohammed Abdullah 18 Mar '12

18 Mar '12

>> I need to make sure that all simultaneous playback and capture (across >> the all dai_links) needs to be of same frequency family. >> If the sampling frequency of a new request is different from ongoing >> playback/capture, then it needs to be rejected. > >> How do I handle it in machine driver? > > Why do you need to do this in the machine driver? Such a constraint sounds like it's coming from one of the chips in the system rather than from the machine itself so it'd seem sensible for the chip itself to impose the constraint so that other machines don't need to replicate the code. > > See wm8988 for an example of adding sample rate constraints, obviously you can change the criteria dynamically. You are right to say that it is not a machine constraint but a codec constraint. The problem is that if a DAI (say the "SN95031 Headset DAI") is opened with a 44.1 khz rate, then another DAI (say "SN95031 Voice") is opened with a 48 khz rate, then we need to reject the second open because the codec cannot support both rates at the same time. Is there any way to do this with constraint_lists? The way it is done in the wm8988 driver does not apply here because there it is a single DAI.

4 7

[alsa-devel] [PATCH 1/1] a52: set time_base to avoid crash in avcodec
by Alex Hornung 18 Mar '12

18 Mar '12

* Not setting the avctx->time_base properly results in a crash in avcodec_encode_audio() in more recent versions of it. * Fix it by setting the time_base according to the sampling rate. Signed-off-by: Alex Hornung <alex(a)alexhornung.com> --- a52/pcm_a52.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/a52/pcm_a52.c b/a52/pcm_a52.c index 00c7c59..560071d 100644 --- a/a52/pcm_a52.c +++ b/a52/pcm_a52.c @@ -436,6 +436,8 @@ static int a52_prepare(snd_pcm_ioplug_t *io) rec->avctx->bit_rate = rec->bitrate * 1000; rec->avctx->sample_rate = io->rate; rec->avctx->channels = io->channels; + rec->avctx->time_base.num = 1; + rec->avctx->time_base.den = io->rate; #if LIBAVCODEC_VERSION_MAJOR > 52 || (LIBAVCODEC_VERSION_MAJOR == 52 && LIBAVCODEC_VERSION_MINOR >= 95) rec->avctx->sample_fmt = AV_SAMPLE_FMT_S16; #else -- 1.7.9.1

1 0

[alsa-devel] [PATCH v2] ASoC: max98095: add jack detection
by Rhyland Klein 17 Mar '12

17 Mar '12

This change adds the logic to support using the jack detect mechanism built in to the codec to detect both when a jack was inserted and what type of jack is present. This change also supports the use of an external mechanism for headphone detection. If this mechanism exists, when the max98095_jack_detect function is called, the hp_jack is simply passed NULL. This change supports both simple headphones, powered headphones, microphones and headsets with both headphones and a mic. v2: - Removed useless function for irq_handler - removed enable/disable irq - Added pdata entry for jack detections slew rate - get irq directly from dev Signed-off-by: Rhyland Klein <rklein(a)nvidia.com> --- include/sound/max98095.h | 12 +++ sound/soc/codecs/max98095.c | 158 ++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/max98095.h | 22 ++++++ 3 files changed, 191 insertions(+), 1 deletions(-) diff --git a/include/sound/max98095.h b/include/sound/max98095.h index 7513a42..65b1c90 100644 --- a/include/sound/max98095.h +++ b/include/sound/max98095.h @@ -49,6 +49,18 @@ struct max98095_pdata { */ unsigned int digmic_left_mode:1; unsigned int digmic_right_mode:1; + + /* Pin5 is the mechanical method of sensing jack insertion + * but it is something that might not be supported. + * 0 = PIN5 not supported + * 1 = PIN5 supported + */ + int jack_detect_pin5en:1; + + /* Slew amount for jack detection. Calculated as 4 * (delay + 1). + * Default delay is 24 to get a time of 100ms. + */ + unsigned int jack_detect_delay; }; #endif diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 0bb511a..0752840 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -24,6 +24,7 @@ #include <linux/slab.h> #include <asm/div64.h> #include <sound/max98095.h> +#include <sound/jack.h> #include "max98095.h" enum max98095_type { @@ -51,6 +52,8 @@ struct max98095_priv { u8 lin_state; unsigned int mic1pre; unsigned int mic2pre; + struct snd_soc_jack *headphone_jack; + struct snd_soc_jack *mic_jack; }; static const u8 max98095_reg_def[M98095_REG_CNT] = { @@ -2173,9 +2176,125 @@ static void max98095_handle_pdata(struct snd_soc_codec *codec) max98095_handle_bq_pdata(codec); } +static irqreturn_t max98095_report_jack(int irq, void *data) +{ + struct snd_soc_codec *codec = data; + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + unsigned int value; + int hp_report = 0; + int mic_report = 0; + + /* Read the Jack Status Register */ + value = snd_soc_read(codec, M98095_007_JACK_AUTO_STS); + + /* If ddone is not set, then detection isn't finished yet */ + if ((value & M98095_DDONE) == 0) + return IRQ_NONE; + + /* if hp, check its bit, and if set, clear it */ + if ((value & M98095_HP_IN || value & M98095_LO_IN) && + max98095->headphone_jack) + hp_report |= SND_JACK_HEADPHONE; + + /* if mic, check its bit, and if set, clear it */ + if ((value & M98095_MIC_IN) && max98095->mic_jack) + mic_report |= SND_JACK_MICROPHONE; + + if (max98095->headphone_jack == max98095->mic_jack) { + snd_soc_jack_report(max98095->headphone_jack, + hp_report | mic_report, + SND_JACK_HEADSET); + } else { + if (max98095->headphone_jack) + snd_soc_jack_report(max98095->headphone_jack, + hp_report, SND_JACK_HEADPHONE); + if (max98095->mic_jack) + snd_soc_jack_report(max98095->mic_jack, + mic_report, SND_JACK_MICROPHONE); + } + + return IRQ_HANDLED; +} + +int max98095_jack_detect_enable(struct snd_soc_codec *codec) +{ + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + int detect_enable = M98095_JDEN; + unsigned int slew = M98095_DEFAULT_SLEW_DELAY; + + if (max98095->pdata->jack_detect_pin5en) + detect_enable |= M98095_PIN5EN; + + if (max98095->jack_detect_delay) + slew = max98095->jack_detect_delay; + + ret = snd_soc_write(codec, M98095_08E_JACK_DC_SLEW, slew); + if (ret < 0) { + dev_err(codec->dev, "Failed to cfg auto detect %d\n", ret); + return ret; + } + + /* configure auto detection to be enabled */ + ret = snd_soc_write(codec, M98095_089_JACK_DET_AUTO, detect_enable); + if (ret < 0) { + dev_err(codec->dev, "Failed to cfg auto detect %d\n", ret); + return ret; + } + + return ret; +} + +int max98095_jack_detect_disable(struct snd_soc_codec *codec) +{ + int ret = 0; + + /* configure auto detection to be disabled */ + ret = snd_soc_write(codec, M98095_089_JACK_DET_AUTO, 0x0); + if (ret < 0) { + dev_err(codec->dev, "Failed to cfg auto detect %d\n", ret); + return ret; + } + + return ret; +} + +int max98095_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack) +{ + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + struct i2c_client *client = to_i2c_client(codec->dev); + int ret = 0; + + max98095->headphone_jack = hp_jack; + max98095->mic_jack = mic_jack; + + /* only progress if we have at least 1 jack pointer */ + if (!hp_jack && !mic_jack) + return -EINVAL; + + max98095_jack_detect_enable(codec); + + /* enable interrupts for headphone jack detection */ + ret = snd_soc_update_bits(codec, M98095_013_JACK_INT_EN, + M98095_IDDONE, M98095_IDDONE); + if (ret < 0) { + dev_err(codec->dev, "Failed to cfg jack irqs %d\n", ret); + return ret; + } + + max98095_report_jack(client->irq, codec); + return 0; +} + #ifdef CONFIG_PM static int max98095_suspend(struct snd_soc_codec *codec) { + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + + if (max98095->headphone_jack || max98095->mic_jack) + max98095_jack_detect_disable(codec); + max98095_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; @@ -2183,8 +2302,16 @@ static int max98095_suspend(struct snd_soc_codec *codec) static int max98095_resume(struct snd_soc_codec *codec) { + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + struct i2c_client *client = to_i2c_client(codec->dev); + max98095_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + if (max98095->headphone_jack || max98095->mic_jack) { + max98095_jack_detect_enable(codec); + max98095_report_jack(client->irq, codec); + } + return 0; } #else @@ -2227,6 +2354,7 @@ static int max98095_probe(struct snd_soc_codec *codec) { struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); struct max98095_cdata *cdata; + struct i2c_client *client; int ret = 0; ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); @@ -2238,6 +2366,8 @@ static int max98095_probe(struct snd_soc_codec *codec) /* reset the codec, the DSP core, and disable all interrupts */ max98095_reset(codec); + client = to_i2c_client(codec->dev); + /* initialize private data */ max98095->sysclk = (unsigned)-1; @@ -2266,11 +2396,23 @@ static int max98095_probe(struct snd_soc_codec *codec) max98095->mic1pre = 0; max98095->mic2pre = 0; + if (client->irq) { + /* register an audio interrupt */ + ret = request_threaded_irq(client->irq, NULL, + max98095_report_jack, + IRQF_TRIGGER_FALLING | IRQF_TRIGGER_RISING, + "max98095", codec); + if (ret) { + dev_err(codec->dev, "Failed to request IRQ: %d\n", ret); + goto err_access; + } + } + ret = snd_soc_read(codec, M98095_0FF_REV_ID); if (ret < 0) { dev_err(codec->dev, "Failure reading hardware revision: %d\n", ret); - goto err_access; + goto err_irq; } dev_info(codec->dev, "Hardware revision: %c\n", ret - 0x40 + 'A'); @@ -2306,14 +2448,28 @@ static int max98095_probe(struct snd_soc_codec *codec) max98095_add_widgets(codec); + return 0; + +err_irq: + if (client->irq) + free_irq(client->irq, codec); err_access: return ret; } static int max98095_remove(struct snd_soc_codec *codec) { + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + struct i2c_client *client = to_i2c_client(codec->dev); + max98095_set_bias_level(codec, SND_SOC_BIAS_OFF); + if (max98095->headphone_jack || max98095->mic_jack) + max98095_jack_detect_disable(codec); + + if (client->irq) + free_irq(client->irq, codec); + return 0; } diff --git a/sound/soc/codecs/max98095.h b/sound/soc/codecs/max98095.h index 891584a..2ebbe4e 100644 --- a/sound/soc/codecs/max98095.h +++ b/sound/soc/codecs/max98095.h @@ -175,11 +175,23 @@ /* MAX98095 Registers Bit Fields */ +/* M98095_007_JACK_AUTO_STS */ + #define M98095_MIC_IN (1<<3) + #define M98095_LO_IN (1<<5) + #define M98095_HP_IN (1<<6) + #define M98095_DDONE (1<<7) + /* M98095_00F_HOST_CFG */ #define M98095_SEG (1<<0) #define M98095_XTEN (1<<1) #define M98095_MDLLEN (1<<2) +/* M98095_013_JACK_INT_EN */ + #define M98095_IMIC_IN (1<<3) + #define M98095_ILO_IN (1<<5) + #define M98095_IHP_IN (1<<6) + #define M98095_IDDONE (1<<7) + /* M98095_027_DAI1_CLKMODE, M98095_031_DAI2_CLKMODE, M98095_03B_DAI3_CLKMODE */ #define M98095_CLKMODE_MASK 0xFF @@ -255,6 +267,10 @@ #define M98095_EQ2EN (1<<1) #define M98095_EQ1EN (1<<0) +/* M98095_089_JACK_DET_AUTO */ + #define M98095_PIN5EN (1<<2) + #define M98095_JDEN (1<<7) + /* M98095_090_PWR_EN_IN */ #define M98095_INEN (1<<7) #define M98095_MB2EN (1<<3) @@ -296,4 +312,10 @@ #define M98095_174_DAI1_BQ_BASE 0x74 #define M98095_17E_DAI2_BQ_BASE 0x7E +/* Default Delay used in Slew Rate Calculation for Jack detection */ +#define M98095_DEFAULT_SLEW_DELAY 0x18 + +extern int max98095_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack); + #endif -- 1.7.0.4

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[alsa-devel] [PATCH] ASoC: pxa2xx-i2s: Add clk_prepare/clk_unprepare calls
by Philipp Zabel 17 Mar '12

17 Mar '12

This patch adds clk_prepare/clk_unprepare calls to the pxa2xx-i2s driver by using the helper functions clk_prepare_enable and clk_disable_unprepare. Signed-off-by: Philipp Zabel <philipp.zabel(a)gmail.com> Cc: Haojian Zhuang <haojian.zhuang(a)marvell.com> Cc: Eric Miao <eric.y.miao(a)gmail.com> Cc: Liam Girdwood <lrg(a)ti.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/pxa/pxa2xx-i2s.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 609abd5..453b092 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -165,7 +165,7 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream, struct pxa2xx_pcm_dma_params *dma_data; BUG_ON(IS_ERR(clk_i2s)); - clk_enable(clk_i2s); + clk_prepare_enable(clk_i2s); clk_ena = 1; pxa_i2s_wait(); @@ -258,7 +258,7 @@ static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream, SACR0 &= ~SACR0_ENB; pxa_i2s_wait(); if (clk_ena) { - clk_disable(clk_i2s); + clk_disable_unprepare(clk_i2s); clk_ena = 0; } } -- 1.7.9.1

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[alsa-devel] [PATCH v3 00/11] ASoC: add imx-sgtl5000 machine driver working with fsl_ssi
by Shawn Guo 17 Mar '12

17 Mar '12

I believe I have addressed all the outstanding comments and requests reviewers put on previous versions. So, hopefully this will be the last iteration of the series. The current sound/for-next branch seems broken for me, and I have not tracking the problem down. But the series works good on SHA below. 3030763 (Merge tag 'asoc-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into topic/asoc) Timur, I did not pick up you Ack-by tag, because there are some changes since v2 you will need to take another look and have the testing, I guess. Regards, Shawn Shawn Guo (11): ASoC: core: missing set_fmt should not be complaint ASoC: fsl: separate SSI and DMA Kconfig options ASoC: imx: merge sound/soc/imx into sound/soc/fsl ASoC: fsl: rename imx-pcm Kconfig options and filename ASoC: fsl: create fsl_utils to accommodate the common functions ASoC: fsl: remove helper fsl_asoc_get_codec_dev_name ASoC: fsl: check property 'compatible' for the machine name ASoC: fsl: make fsl_ssi driver compilable on ARM/IMX ASoC: fsl: remove the fatal error checking on codec-handle ASoC: fsl: let fsl_ssi work with imx pcm and machine drivers ASoC: fsl: add imx-sgtl5000 machine driver .../bindings/sound/imx-audio-sgtl5000.txt | 24 +++ arch/powerpc/configs/86xx/mpc8610_hpcd_defconfig | 1 + arch/powerpc/configs/mpc85xx_defconfig | 1 + arch/powerpc/configs/mpc85xx_smp_defconfig | 1 + sound/soc/Kconfig | 1 - sound/soc/Makefile | 1 - sound/soc/fsl/Kconfig | 124 ++++++++++++- sound/soc/fsl/Makefile | 29 +++- sound/soc/{imx => fsl}/eukrea-tlv320.c | 2 +- sound/soc/fsl/fsl_ssi.c | 146 ++++++++++++---- sound/soc/fsl/fsl_utils.c | 91 ++++++++++ sound/soc/fsl/fsl_utils.h | 26 +++ sound/soc/{imx => fsl}/imx-audmux.c | 0 sound/soc/{imx => fsl}/imx-audmux.h | 0 .../{imx/imx-pcm-dma-mx2.c => fsl/imx-pcm-dma.c} | 0 sound/soc/{imx => fsl}/imx-pcm-fiq.c | 0 sound/soc/{imx => fsl}/imx-pcm.c | 0 sound/soc/{imx => fsl}/imx-pcm.h | 0 sound/soc/fsl/imx-sgtl5000.c | 177 ++++++++++++++++++ sound/soc/{imx => fsl}/imx-ssi.c | 2 +- sound/soc/{imx => fsl}/imx-ssi.h | 0 sound/soc/fsl/mpc8610_hpcd.c | 168 ++---------------- sound/soc/{imx => fsl}/mx27vis-aic32x4.c | 0 sound/soc/fsl/p1022_ds.c | 190 ++------------------ sound/soc/{imx => fsl}/phycore-ac97.c | 0 sound/soc/{imx => fsl}/wm1133-ev1.c | 0 sound/soc/imx/Kconfig | 79 -------- sound/soc/imx/Makefile | 22 --- sound/soc/soc-core.c | 11 +- 29 files changed, 616 insertions(+), 480 deletions(-)

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[alsa-devel] [PATCH] ASoC: fsl: p1022ds: tell the WM8776 codec driver that it's the master
by Timur Tabi 17 Mar '12

17 Mar '12

The WM8776 codec driver requires the machine driver to set one of the SND_SOC_DAIFMT_CBx_xxx values. The P1022DS machine driver should be setting SND_SOC_DAIFMT_CBM_CFM, but since that value was zero, no one noticed. Commit 75d9ac46 ("ASoC: Allow DAI formats to be specified in the dai_link"), however, changed the value of SND_SOC_DAIFMT_CBM_CFM from zero to a non-zero value, which means that it now needs to be specifically set by the machine driver. We also set SND_SOC_DAIFMT_NB_NF, for the same reason. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- This patch should be applied on top of Shawn's patchset "[PATCH v5] ASoC: add imx-sgtl5000 machine driver working with fsl_ssi", but it may also need to be back-ported to kernel 3.3. sound/soc/fsl/p1022_ds.c | 24 ++++++++++++++++-------- 1 file changed, 16 insertions(+), 8 deletions(-) diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c index 0f42f31..2022985 100644 --- a/sound/soc/fsl/p1022_ds.c +++ b/sound/soc/fsl/p1022_ds.c @@ -255,7 +255,8 @@ static int p1022_ds_probe(struct platform_device *pdev) } if (strcasecmp(sprop, "i2s-slave") == 0) { - mdata->dai_format = SND_SOC_DAIFMT_I2S; + mdata->dai_format = SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM; mdata->codec_clk_direction = SND_SOC_CLOCK_OUT; mdata->cpu_clk_direction = SND_SOC_CLOCK_IN; @@ -272,31 +273,38 @@ static int p1022_ds_probe(struct platform_device *pdev) } mdata->clk_frequency = be32_to_cpup(iprop); } else if (strcasecmp(sprop, "i2s-master") == 0) { - mdata->dai_format = SND_SOC_DAIFMT_I2S; + mdata->dai_format = SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS; mdata->codec_clk_direction = SND_SOC_CLOCK_IN; mdata->cpu_clk_direction = SND_SOC_CLOCK_OUT; } else if (strcasecmp(sprop, "lj-slave") == 0) { - mdata->dai_format = SND_SOC_DAIFMT_LEFT_J; + mdata->dai_format = SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBM_CFM; mdata->codec_clk_direction = SND_SOC_CLOCK_OUT; mdata->cpu_clk_direction = SND_SOC_CLOCK_IN; } else if (strcasecmp(sprop, "lj-master") == 0) { - mdata->dai_format = SND_SOC_DAIFMT_LEFT_J; + mdata->dai_format = SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBS_CFS; mdata->codec_clk_direction = SND_SOC_CLOCK_IN; mdata->cpu_clk_direction = SND_SOC_CLOCK_OUT; } else if (strcasecmp(sprop, "rj-slave") == 0) { - mdata->dai_format = SND_SOC_DAIFMT_RIGHT_J; + mdata->dai_format = SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_CBM_CFM; mdata->codec_clk_direction = SND_SOC_CLOCK_OUT; mdata->cpu_clk_direction = SND_SOC_CLOCK_IN; } else if (strcasecmp(sprop, "rj-master") == 0) { - mdata->dai_format = SND_SOC_DAIFMT_RIGHT_J; + mdata->dai_format = SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_CBS_CFS; mdata->codec_clk_direction = SND_SOC_CLOCK_IN; mdata->cpu_clk_direction = SND_SOC_CLOCK_OUT; } else if (strcasecmp(sprop, "ac97-slave") == 0) { - mdata->dai_format = SND_SOC_DAIFMT_AC97; + mdata->dai_format = SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_AC97 | SND_SOC_DAIFMT_CBM_CFM; mdata->codec_clk_direction = SND_SOC_CLOCK_OUT; mdata->cpu_clk_direction = SND_SOC_CLOCK_IN; } else if (strcasecmp(sprop, "ac97-master") == 0) { - mdata->dai_format = SND_SOC_DAIFMT_AC97; + mdata->dai_format = SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_AC97 | SND_SOC_DAIFMT_CBS_CFS; mdata->codec_clk_direction = SND_SOC_CLOCK_IN; mdata->cpu_clk_direction = SND_SOC_CLOCK_OUT; } else { -- 1.7.10.rc1.8.ga99c5.dirty

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