September 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 1/2] ASoC: Initial WM8996 headphone impedance measurement support
by Mark Brown 19 Sep '11

19 Sep '11

The WM8996 can measure the impedance of accessories connected to the headphone output. Implement initial support for this, measuring the left channel impedance when an accessory is detected and using this to distinguish between a line load and a headphone load. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8996.c | 137 ++++++++++++++++++++++++++++++++++++-------- 1 files changed, 112 insertions(+), 25 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 8e9d7f7..a6168c5 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2350,12 +2350,94 @@ int wm8996_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, /* Enable interrupts and we're off */ snd_soc_update_bits(codec, WM8996_INTERRUPT_STATUS_2_MASK, - WM8996_IM_MICD_EINT, 0); + WM8996_IM_MICD_EINT | WM8996_HP_DONE_EINT, 0); return 0; } EXPORT_SYMBOL_GPL(wm8996_detect); +static void wm8996_hpdet_irq(struct snd_soc_codec *codec) +{ + struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); + int val, reg, report; + + /* Assume headphone in error conditions; we need to report + * something or we stall our state machine. + */ + report = SND_JACK_HEADPHONE; + + reg = snd_soc_read(codec, WM8996_HEADPHONE_DETECT_2); + if (reg < 0) { + dev_err(codec->dev, "Failed to read HPDET status\n"); + goto out; + } + + if (!(reg & WM8996_HP_DONE)) { + dev_err(codec->dev, "Got HPDET IRQ but HPDET is busy\n"); + goto out; + } + + val = reg & WM8996_HP_LVL_MASK; + + dev_dbg(codec->dev, "HPDET measured %d ohms\n", val); + + /* If we've got high enough impedence then report as line, + * otherwise assume headphone. + */ + if (val >= 126) + report = SND_JACK_LINEOUT; + else + report = SND_JACK_HEADPHONE; + +out: + if (wm8996->jack_mic) + report |= SND_JACK_MICROPHONE; + + snd_soc_jack_report(wm8996->jack, report, + SND_JACK_LINEOUT | SND_JACK_HEADSET); + + wm8996->detecting = false; + + /* If the output isn't running re-clamp it */ + if (!(snd_soc_read(codec, WM8996_POWER_MANAGEMENT_1) & + (WM8996_HPOUT1L_ENA | WM8996_HPOUT1R_RMV_SHORT))) + snd_soc_update_bits(codec, WM8996_ANALOGUE_HP_1, + WM8996_HPOUT1L_RMV_SHORT | + WM8996_HPOUT1R_RMV_SHORT, 0); + + /* Go back to looking at the microphone */ + snd_soc_update_bits(codec, WM8996_ACCESSORY_DETECT_MODE_1, + WM8996_JD_MODE_MASK, 0); + snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, WM8996_MICD_ENA, + WM8996_MICD_ENA); + + snd_soc_dapm_disable_pin(&codec->dapm, "Bandgap"); + snd_soc_dapm_sync(&codec->dapm); +} + +static void wm8996_hpdet_start(struct snd_soc_codec *codec) +{ + /* Unclamp the output, we can't measure while we're shorting it */ + snd_soc_update_bits(codec, WM8996_ANALOGUE_HP_1, + WM8996_HPOUT1L_RMV_SHORT | + WM8996_HPOUT1R_RMV_SHORT, + WM8996_HPOUT1L_RMV_SHORT | + WM8996_HPOUT1R_RMV_SHORT); + + /* We need bandgap for HPDET */ + snd_soc_dapm_force_enable_pin(&codec->dapm, "Bandgap"); + snd_soc_dapm_sync(&codec->dapm); + + /* Go into headphone detect left mode */ + snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, WM8996_MICD_ENA, 0); + snd_soc_update_bits(codec, WM8996_ACCESSORY_DETECT_MODE_1, + WM8996_JD_MODE_MASK, 1); + + /* Trigger a measurement */ + snd_soc_update_bits(codec, WM8996_HEADPHONE_DETECT_1, + WM8996_HP_POLL, WM8996_HP_POLL); +} + static void wm8996_micd(struct snd_soc_codec *codec) { struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); @@ -2376,28 +2458,36 @@ static void wm8996_micd(struct snd_soc_codec *codec) wm8996->jack_mic = false; wm8996->detecting = true; snd_soc_jack_report(wm8996->jack, 0, - SND_JACK_HEADSET | SND_JACK_BTN_0); + SND_JACK_LINEOUT | SND_JACK_HEADSET | + SND_JACK_BTN_0); + snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, WM8996_MICD_RATE_MASK, WM8996_MICD_RATE_MASK); return; } - /* If the measurement is very high we've got a microphone but - * do a little debounce to account for mechanical issues. + /* If the measurement is very high we've got a microphone, + * either we just detected one or if we already reported then + * we've got a button release event. */ if (val & 0x400) { - dev_dbg(codec->dev, "Microphone detected\n"); - snd_soc_jack_report(wm8996->jack, SND_JACK_HEADSET, - SND_JACK_HEADSET | SND_JACK_BTN_0); - wm8996->jack_mic = true; - wm8996->detecting = false; - - /* Increase poll rate to give better responsiveness - * for buttons */ - snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, - WM8996_MICD_RATE_MASK, - 5 << WM8996_MICD_RATE_SHIFT); + if (wm8996->detecting) { + dev_dbg(codec->dev, "Microphone detected\n"); + wm8996->jack_mic = true; + wm8996_hpdet_start(codec); + + /* Increase poll rate to give better responsiveness + * for buttons */ + snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, + WM8996_MICD_RATE_MASK, + 5 << WM8996_MICD_RATE_SHIFT); + } else { + dev_dbg(codec->dev, "Mic button up\n"); + snd_soc_jack_report(wm8996->jack, 0, SND_JACK_BTN_0); + } + + return; } /* If we detected a lower impedence during initial startup @@ -2429,15 +2519,11 @@ static void wm8996_micd(struct snd_soc_codec *codec) if (val & 0x3fc) { if (wm8996->jack_mic) { dev_dbg(codec->dev, "Mic button detected\n"); - snd_soc_jack_report(wm8996->jack, - SND_JACK_HEADSET | SND_JACK_BTN_0, - SND_JACK_HEADSET | SND_JACK_BTN_0); - } else { - dev_dbg(codec->dev, "Headphone detected\n"); - snd_soc_jack_report(wm8996->jack, - SND_JACK_HEADPHONE, - SND_JACK_HEADSET | + snd_soc_jack_report(wm8996->jack, SND_JACK_BTN_0, SND_JACK_BTN_0); + } else if (wm8996->detecting) { + dev_dbg(codec->dev, "Headphone detected\n"); + wm8996_hpdet_start(codec); /* Increase the detection rate a bit for * responsiveness. @@ -2445,8 +2531,6 @@ static void wm8996_micd(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, WM8996_MICD_RATE_MASK, 7 << WM8996_MICD_RATE_SHIFT); - - wm8996->detecting = false; } } } @@ -2486,6 +2570,9 @@ static irqreturn_t wm8996_irq(int irq, void *data) if (irq_val & WM8996_MICD_EINT) wm8996_micd(codec); + if (irq_val & WM8996_HP_DONE_EINT) + wm8996_hpdet_irq(codec); + return IRQ_HANDLED; } -- 1.7.6.3

2 2

[alsa-devel] [PATCH 4/4] src/pcm/pcm_multi.c: add missing free
by Julia Lawall 18 Sep '11

18 Sep '11

From: Julia Lawall <julia(a)diku.dk> Something that is allocated using calloc is not freed on an error path. Signed-off-by: Julia Lawall <julia(a)diku.dk> Signed-off-by: Suman Saha <sumsaha(a)gmail.com> --- Compile-tested only. src/pcm/pcm_multi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/src/pcm/pcm_multi.c b/src/pcm/pcm_multi.c index 68f2d68..6b39c7a 100644 --- a/src/pcm/pcm_multi.c +++ b/src/pcm/pcm_multi.c @@ -886,6 +886,8 @@ int snd_pcm_multi_open(snd_pcm_t **pcmp, const char *name, err = snd_pcm_new(&pcm, SND_PCM_TYPE_MULTI, name, stream, multi->slaves[0].pcm->mode); if (err < 0) { + free(multi->slaves); + free(multi->channels); free(multi); return err; }

1 0

[alsa-devel] [PATCH 3/4] src/pcm/pcm_ladspa.c: add missing free
by Julia Lawall 18 Sep '11

18 Sep '11

From: Julia Lawall <julia(a)diku.dk> Something that is allocated using calloc is not freed on some error paths. Signed-off-by: Julia Lawall <julia(a)diku.dk> Signed-off-by: Suman Saha <sumsaha(a)gmail.com> --- Compile-tested only. src/pcm/pcm_ladspa.c | 16 ++++++++++++---- 1 file changed, 12 insertions(+), 4 deletions(-) diff --git a/src/pcm/pcm_ladspa.c b/src/pcm/pcm_ladspa.c index c413c10..84ebaa5 100644 --- a/src/pcm/pcm_ladspa.c +++ b/src/pcm/pcm_ladspa.c @@ -750,8 +750,10 @@ static int snd_pcm_ladspa_allocate_memory(snd_pcm_t *pcm, snd_pcm_ladspa_t *lads if (instance->input.data == NULL || instance->input.m_data == NULL || instance->output.data == NULL || - instance->output.m_data == NULL) + instance->output.m_data == NULL) { + free(pchannels); return -ENOMEM; + } for (idx = 0; idx < instance->input.channels.size; idx++) { chn = instance->output.channels.array[idx]; if (pchannels[chn] == NULL && chn < ichannels) { @@ -761,8 +763,10 @@ static int snd_pcm_ladspa_allocate_memory(snd_pcm_t *pcm, snd_pcm_ladspa_t *lads instance->input.data[idx] = pchannels[chn]; if (instance->input.data[idx] == NULL) { instance->input.data[idx] = snd_pcm_ladspa_allocate_zero(ladspa, 0); - if (instance->input.data[idx] == NULL) + if (instance->input.data[idx] == NULL) { + free(pchannels); return -ENOMEM; + } } } for (idx = 0; idx < instance->output.channels.size; idx++) { @@ -770,8 +774,10 @@ static int snd_pcm_ladspa_allocate_memory(snd_pcm_t *pcm, snd_pcm_ladspa_t *lads /* FIXME/OPTIMIZE: check if we can remove double alloc */ /* if LADSPA plugin has no broken inplace */ instance->output.data[idx] = malloc(sizeof(LADSPA_Data) * ladspa->allocated); - if (instance->output.data[idx] == NULL) + if (instance->output.data[idx] == NULL) { + free(pchannels); return -ENOMEM; + } pchannels[chn] = instance->output.m_data[idx] = instance->output.data[idx]; } } @@ -793,8 +799,10 @@ static int snd_pcm_ladspa_allocate_memory(snd_pcm_t *pcm, snd_pcm_ladspa_t *lads instance->output.data[idx] = NULL; } else { instance->output.data[idx] = snd_pcm_ladspa_allocate_zero(ladspa, 1); - if (instance->output.data[idx] == NULL) + if (instance->output.data[idx] == NULL) { + free(pchannels); return -ENOMEM; + } } } }

1 0

[alsa-devel] [PATCH 1/1] sound/usb: Added support for Roland UM-ONE midi-usb interface.
by Daniele Guerrieri 18 Sep '11

18 Sep '11

Hi all, this is my first post to this list. Recently i bought a roland UM-ONE midi-usb interface which i thought worked well with alsa, but it didn't. So i got from lsusb id/vid and added them to sound/usb/quirks-table.h; it seems to work well now with snd-usb-audio, tested on 2 machines. Below there is the patch output for my changes. As a master branch i cloned git://github.com/torvalds/linux.git ----output of git format-patch ----- Roland UM-ONE midi usb interface differs from Roland UM-1. Signed-off-by: Daniele Guerrieri <d.guerrieri(a)gmail.com> --- sound/usb/quirks-table.h | 14 ++++++++++++++ 1 files changed, 14 insertions(+), 0 deletions(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index a42e3ef..f041b2e 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1192,6 +1192,20 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { + /* Added support for Roland UM-ONE which differs from UM-1 */ + USB_DEVICE(0x0582, 0x012a), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "ROLAND", */ + /* .product_name = "UM-ONE", */ + .ifnum = 0, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0001, + .in_cables = 0x0003 + } + } +}, +{ /* has ID 0x006b when not in "Advanced Driver" mode */ USB_DEVICE_VENDOR_SPEC(0x0582, 0x006a), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { -- 1.7.6

2 2

[PATCH 2/2] saa7134: introduce "std" module parameter to force video std
by Stas Sergeev 18 Sep '11

18 Sep '11

--- drivers/media/video/saa7134/saa7134-video.c | 39 ++++++++++++++++++++++++-- drivers/media/video/saa7134/saa7134.h | 1 + 2 files changed, 37 insertions(+), 3 deletions(-) diff --git a/drivers/media/video/saa7134/saa7134-video.c b/drivers/media/video/saa7134/saa7134-video.c index 776ba2d..04ac7de 100644 --- a/drivers/media/video/saa7134/saa7134-video.c +++ b/drivers/media/video/saa7134/saa7134-video.c @@ -40,12 +40,15 @@ static unsigned int noninterlaced; /* 0 */ static unsigned int gbufsize = 720*576*4; static unsigned int gbufsize_max = 720*576*4; static char secam[] = "--"; +static char std[16] = "--"; module_param(video_debug, int, 0644); MODULE_PARM_DESC(video_debug,"enable debug messages [video]"); module_param(gbuffers, int, 0444); MODULE_PARM_DESC(gbuffers,"number of capture buffers, range 2-32"); module_param(noninterlaced, int, 0644); MODULE_PARM_DESC(noninterlaced,"capture non interlaced video"); +module_param_string(std, std, sizeof(std), 0644); +MODULE_PARM_DESC(secam, "force TV standard, either PAL, SECAM or NTSC"); module_param_string(secam, secam, sizeof(secam), 0644); MODULE_PARM_DESC(secam, "force SECAM variant, either DK,L or Lc"); @@ -1847,14 +1850,20 @@ int saa7134_s_std_internal(struct saa7134_dev *dev, struct saa7134_fh *fh, v4l2_ return -EBUSY; } - for (i = 0; i < TVNORMS; i++) + for (i = 0; i < TVNORMS; i++) { + if (!tvnorms[i].enabled) + continue; if (*id == tvnorms[i].id) break; + } if (i == TVNORMS) - for (i = 0; i < TVNORMS; i++) + for (i = 0; i < TVNORMS; i++) { + if (!tvnorms[i].enabled) + continue; if (*id & tvnorms[i].id) break; + } if (i == TVNORMS) return -EINVAL; @@ -1871,6 +1880,8 @@ int saa7134_s_std_internal(struct saa7134_dev *dev, struct saa7134_fh *fh, v4l2_ fixup = V4L2_STD_SECAM; } for (i = 0; i < TVNORMS; i++) { + if (!tvnorms[i].enabled) + continue; if (fixup == tvnorms[i].id) break; } @@ -2579,8 +2590,30 @@ int saa7134_videoport_init(struct saa7134_dev *dev) int saa7134_video_init2(struct saa7134_dev *dev) { + int i, idx = -1; + for (i = 0; i < TVNORMS; i++) { + int enable = (std[0] == '-' || strncasecmp(std, + tvnorms[i].name, strlen(std)) == 0); + + if (secam[0] != '-') { + char buf[16] = "SECAM"; + if (strncasecmp(buf, tvnorms[i].name, + strlen(buf)) == 0) { + strcat(buf, "-"); + strcat(buf, secam); + if (strncasecmp(buf, tvnorms[i].name, + strlen(buf)) != 0) + enable = 0; + } + } + tvnorms[i].enabled = enable; + if (enable && idx == -1) + idx = i; + } + if (idx == -1) + return -EINVAL; /* init video hw */ - set_tvnorm(dev,&tvnorms[0]); + set_tvnorm(dev,&tvnorms[idx]); video_mux(dev,0); saa7134_tvaudio_setmute(dev); saa7134_tvaudio_setvolume(dev,dev->ctl_volume); diff --git a/drivers/media/video/saa7134/saa7134.h b/drivers/media/video/saa7134/saa7134.h index 28eb103..a0eddce 100644 --- a/drivers/media/video/saa7134/saa7134.h +++ b/drivers/media/video/saa7134/saa7134.h @@ -77,6 +77,7 @@ enum saa7134_video_out { struct saa7134_tvnorm { char *name; v4l2_std_id id; + int enabled; /* video decoder */ unsigned int sync_control; -- 1.7.6 --------------080905040006020600030205--

1 0

[PATCH 1/2] saa7134: fix automute
by Stas Sergeev 18 Sep '11

18 Sep '11

--- drivers/media/video/saa7134/saa7134-tvaudio.c | 7 +++++-- 1 files changed, 5 insertions(+), 2 deletions(-) diff --git a/drivers/media/video/saa7134/saa7134-tvaudio.c b/drivers/media/video/saa7134/saa7134-tvaudio.c index 57e646b..62a6287 100644 --- a/drivers/media/video/saa7134/saa7134-tvaudio.c +++ b/drivers/media/video/saa7134/saa7134-tvaudio.c @@ -332,7 +332,7 @@ static int tvaudio_checkcarrier(struct saa7134_dev *dev, struct mainscan *scan) { __s32 left,right,value; - if (audio_debug > 1) { + if (audio_debug > 1 && (dev->tvnorm->id & scan->std)) { int i; dprintk("debug %d:",scan->carr); for (i = -150; i <= 150; i += 30) { @@ -546,6 +546,7 @@ static int tvaudio_thread(void *data) dev->tvnorm->name, carrier/1000, carrier%1000, max1, max2); dev->last_carrier = carrier; + dev->automute = !(dev->thread.scan1 > 1); } else if (0 != dev->last_carrier) { /* no carrier -- try last detected one as fallback */ @@ -553,6 +554,7 @@ static int tvaudio_thread(void *data) dprintk("audio carrier scan failed, " "using %d.%03d MHz [last detected]\n", carrier/1000, carrier%1000); + dev->automute = 1; } else { /* no carrier + no fallback -- use default */ @@ -560,9 +562,9 @@ static int tvaudio_thread(void *data) dprintk("audio carrier scan failed, " "using %d.%03d MHz [default]\n", carrier/1000, carrier%1000); + dev->automute = 1; } tvaudio_setcarrier(dev,carrier,carrier); - dev->automute = 0; saa_andorb(SAA7134_STEREO_DAC_OUTPUT_SELECT, 0x30, 0x00); saa7134_tvaudio_setmute(dev); /* find the exact tv audio norm */ @@ -1020,6 +1022,7 @@ int saa7134_tvaudio_init2(struct saa7134_dev *dev) } dev->thread.thread = NULL; + dev->thread.scan1 = dev->thread.scan2 = 0; if (my_thread) { saa7134_tvaudio_init(dev); /* start tvaudio thread */ -- 1.7.6 --------------080905040006020600030205 Content-Type: text/x-patch; name="0002-saa7134-introduce-std-module-parameter-to-force-vide.patch" Content-Transfer-Encoding: 7bit Content-Disposition: attachment; filename*0="0002-saa7134-introduce-std-module-parameter-to-force-vide.pa"; filename*1="tch"

1 0

[alsa-devel] [PATCH 0/7] ASoC: ep93xx: convert to use snd_soc_register_card()
by Mika Westerberg 18 Sep '11

18 Sep '11

Here is a series of patches which convert ep93xx ASoC drivers to use snd_soc_register_card(). This has the advantage that the machine drivers get loaded automatically by udev. Also this gets rid of machine_is_xxx() from the machine drivers. Tested on Sim.One. The edb93xx and snappercl15 changes are compile tested only. Mika Westerberg (7): ASoC: ep93xx-pcm: add MODULE_ALIAS ASoC: simone: convert to use snd_soc_register_card() ASoC: edb93xx: convert to use snd_soc_register_card() ASoC: snappercl15: convert to use snd_soc_register_card() ARM: ep93xx: simone: register audio platform device ARM: ep93xx: edb93xx: register audio platform device ARM: ep93xx: snappercl15: register audio platform device arch/arm/mach-ep93xx/edb93xx.c | 6 +++ arch/arm/mach-ep93xx/simone.c | 13 +++++++- arch/arm/mach-ep93xx/snappercl15.c | 13 +++++++- sound/soc/ep93xx/edb93xx.c | 60 ++++++++++++++++++--------------- sound/soc/ep93xx/ep93xx-pcm.c | 1 + sound/soc/ep93xx/simone.c | 64 ++++++++++++++++++++---------------- sound/soc/ep93xx/snappercl15.c | 53 ++++++++++++++++++++--------- 7 files changed, 136 insertions(+), 74 deletions(-) -- 1.7.5.4

6 19

[alsa-devel] hda- Disable Sticky PCM
by Raymond Yau 18 Sep '11

18 Sep '11

2011/9/15 Takashi Iwai <tiwai(a)suse.de>: > At Thu, 15 Sep 2011 22:23:12 +0800, > Raymond Yau wrote: >> >> BTW, there is some strange behaviour of the non-sticky pcm >> implementation on ad1988 >> >> Refer to commit 0e7adbe263f89ea2ef15b5af5e80a812b2a85025 >> >> Disable sticky PCM stream assignment for AD codecs >> >> why hda_codec_cleanup_stream is called twice and this also occur with >> capture and playback > > Isn't it just the power-save? The suspend calls > hda_cleanup_all_streams(). > > That is, the first call is the normal one, and it doesn't clean up the > stream actually because of stickiness. The second call is the real > cleanup at power-saving. > > > Takashi > by adding prinkt statement in really cleanup steam() function, really_cleanup_stream is called twice so why is second call if the stream is already really cleanup ALSA hda_intel.c:1755: azx_pcm_prepare: bufsize=0x10000, format=0x4011 ALSA hda_codec.c:1412: hda_codec_setup_stream: NID=0x4, stream=0x5, channel=0, format=0x4011 ALSA hda_codec.c:1412: hda_codec_setup_stream: NID=0x3, stream=0x5, channel=0, format=0x4011 ALSA hda_codec.c:1412: hda_codec_setup_stream: NID=0x6, stream=0x5, channel=0, format=0x4011 ALSA hda_codec.c:1412: hda_codec_setup_stream: NID=0x5, stream=0x5, channel=0, format=0x4011 ALSA hda_codec.c:1412: hda_codec_setup_stream: NID=0xa, stream=0x5, channel=0, format=0x4011 ALSA hda_codec.c:1475: hda_codec_cleanup_stream: NID=0x4 really cleanup steam ALSA hda_codec.c:1475: hda_codec_cleanup_stream: NID=0x6 really cleanup steam ALSA hda_codec.c:1475: hda_codec_cleanup_stream: NID=0x5 really cleanup steam ALSA hda_codec.c:1475: hda_codec_cleanup_stream: NID=0xa really cleanup steam ALSA hda_codec.c:1475: hda_codec_cleanup_stream: NID=0x3 really cleanup steam ALSA hda_codec.c:1475: hda_codec_cleanup_stream: NID=0x4 really cleanup steam ALSA hda_codec.c:1475: hda_codec_cleanup_stream: NID=0x6 really cleanup steam ALSA hda_codec.c:1475: hda_codec_cleanup_stream: NID=0x5 really cleanup steam ALSA hda_codec.c:1475: hda_codec_cleanup_stream: NID=0xa really cleanup steam ALSA hda_codec.c:1475: hda_codec_cleanup_stream: NID=0x3 really cleanup steam

2 1

[alsa-devel] Noise on snd-hda-intel
by Sune Mølgaard 18 Sep '11

18 Sep '11

Plugging almost any headset into the front headset jacks on my machine results in noise in the headphones as well as in Skype conversations and/or recording. Said noise is, however not head in the regular speakers (except when playing back recordings, of course. lspci -vvnn gives 00:1b.0 Audio device [0403]: Intel Corporation 82801JI (ICH10 Family) HD Audio Controller [8086:3a3e] Subsystem: ASUSTeK Computer Inc. Device [1043:82ea] Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 256 bytes Interrupt: pin A routed to IRQ 70 Region 0: Memory at f7ff8000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel and aplay -l gives **** List of PLAYBACK Hardware Devices **** card 0: Device [USB Multimedia Audio Device], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: Intel [HDA Intel], device 0: AD198x Analog [AD198x Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 I have tried with no modules options as well as options snd_hda_intel model=6stack and options snd_hda_intel model=6stack-dig Best regards, Sune Mølgaard -- You buy Windows, and you put it on your machine-you can't do a whole lot with it. - Bob Muglia, Microsoft Senior Vice President

4 14

[alsa-devel] [PATCH] [v2] ASoC: support all possible sample rates in the WM8776 driver
by Timur Tabi 17 Sep '11

17 Sep '11

The .set_sysclk() function in ASoC codec drivers can be used to tell the driver what the frequency of master clock (mclk) is. Some codecs use a divider on this clock to determine the sample rate. The WM8776 is one such codec. So instead of hard-coding a list of specific sample rates supported, these codec drivers can specify SNDRV_PCM_RATE_CONTINUOUS instead, clamping the range to the absolute limits of the hardware. Then the .hw_params() function can use the mclk rate to determine whether any requested rate is actually supported. Although the WM8776 driver includes a .set_sysclk function, it was also hard-coding the list of supported sample rates. We change the hard-coded list to a range within the capabilities of the WM8776 itself. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/codecs/wm8776.c | 13 ++++++------- 1 files changed, 6 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index ad6f0fa..ca8a593 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -321,11 +321,6 @@ static int wm8776_set_bias_level(struct snd_soc_codec *codec, return 0; } -#define WM8776_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\ - SNDRV_PCM_RATE_96000) - - #define WM8776_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) @@ -350,7 +345,9 @@ static struct snd_soc_dai_driver wm8776_dai[] = { .stream_name = "Playback", .channels_min = 2, .channels_max = 2, - .rates = WM8776_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 32000, + .rate_max = 192000, .formats = WM8776_FORMATS, }, .ops = &wm8776_dac_ops, @@ -362,7 +359,9 @@ static struct snd_soc_dai_driver wm8776_dai[] = { .stream_name = "Capture", .channels_min = 2, .channels_max = 2, - .rates = WM8776_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 32000, + .rate_max = 96000, .formats = WM8776_FORMATS, }, .ops = &wm8776_adc_ops, -- 1.7.3.4

4 19