October 2008 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH3/4] hda: add snd_hda_get_jack* functions
by Matthew Ranostay 25 Oct '08

25 Oct '08

This patch adds snd_hda_get_jack* functions for reporting jack location, device, and connectivity type. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 9a8adc6..eaa8b56 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -107,6 +107,52 @@ static void hda_keep_power_on(struct hda_codec *codec); static inline void hda_keep_power_on(struct hda_codec *codec) {} #endif +const char *snd_hda_get_jack_location(u32 cfg) +{ + static char *bases[7] = { + "N/A", "Rear", "Front", "Left", "Right", "Top", "Bottom", + }; + static unsigned char specials_idx[] = { + 0x07, 0x08, + 0x17, 0x18, 0x19, + 0x37, 0x38 + }; + static char *specials[] = { + "Rear Panel", "Drive Bar", + "Riser", "HDMI", "ATAPI", + "Mobile-In", "Mobile-Out" + }; + int i; + cfg = (cfg & AC_DEFCFG_LOCATION) >> AC_DEFCFG_LOCATION_SHIFT; + if ((cfg & 0x0f) < 7) + return bases[cfg & 0x0f]; + for (i = 0; i < ARRAY_SIZE(specials_idx); i++) { + if (cfg == specials_idx[i]) + return specials[i]; + } + return "UNKNOWN"; +} + +const char *snd_hda_get_jack_connectivity(u32 cfg) +{ + static char *jack_locations[4] = { "Ext", "Int", "Sep", "Oth" }; + + return jack_locations[(cfg >> (AC_DEFCFG_LOCATION_SHIFT + 4)) & 3]; +} + +const char *snd_hda_get_jack_type(u32 cfg) +{ + static char *jack_types[16] = { + "Line Out", "Speaker", "HP Out", "CD", + "SPDIF Out", "Digital Out", "Modem Line", "Modem Hand", + "Line In", "Aux", "Mic", "Telephony", + "SPDIF In", "Digitial In", "Reserved", "Other" + }; + + return jack_types[(cfg & AC_DEFCFG_DEVICE) + >> AC_DEFCFG_DEVICE_SHIFT]; +} + /** * snd_hda_codec_read - send a command and get the response * @codec: the HDA codec diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index a77ba22..07ddd3d 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -859,6 +859,13 @@ int snd_hda_suspend(struct hda_bus *bus, pm_message_t state); int snd_hda_resume(struct hda_bus *bus); #endif +/* + * get widget information + */ +const char *snd_hda_get_jack_connectivity(u32 cfg); +const char *snd_hda_get_jack_type(u32 cfg); +const char *snd_hda_get_jack_location(u32 cfg); + /* * power saving */ diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 64ab19f..b36d4d0 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -145,32 +145,6 @@ static void print_pcm_caps(struct snd_info_buffer *buffer, print_pcm_formats(buffer, stream); } -static const char *get_jack_location(u32 cfg) -{ - static char *bases[7] = { - "N/A", "Rear", "Front", "Left", "Right", "Top", "Bottom", - }; - static unsigned char specials_idx[] = { - 0x07, 0x08, - 0x17, 0x18, 0x19, - 0x37, 0x38 - }; - static char *specials[] = { - "Rear Panel", "Drive Bar", - "Riser", "HDMI", "ATAPI", - "Mobile-In", "Mobile-Out" - }; - int i; - cfg = (cfg & AC_DEFCFG_LOCATION) >> AC_DEFCFG_LOCATION_SHIFT; - if ((cfg & 0x0f) < 7) - return bases[cfg & 0x0f]; - for (i = 0; i < ARRAY_SIZE(specials_idx); i++) { - if (cfg == specials_idx[i]) - return specials[i]; - } - return "UNKNOWN"; -} - static const char *get_jack_connection(u32 cfg) { static char *names[16] = { @@ -206,13 +180,6 @@ static void print_pin_caps(struct snd_info_buffer *buffer, int *supports_vref) { static char *jack_conns[4] = { "Jack", "N/A", "Fixed", "Both" }; - static char *jack_types[16] = { - "Line Out", "Speaker", "HP Out", "CD", - "SPDIF Out", "Digital Out", "Modem Line", "Modem Hand", - "Line In", "Aux", "Mic", "Telephony", - "SPDIF In", "Digitial In", "Reserved", "Other" - }; - static char *jack_locations[4] = { "Ext", "Int", "Sep", "Oth" }; unsigned int caps, val; caps = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); @@ -274,9 +241,9 @@ static void print_pin_caps(struct snd_info_buffer *buffer, caps = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); snd_iprintf(buffer, " Pin Default 0x%08x: [%s] %s at %s %s\n", caps, jack_conns[(caps & AC_DEFCFG_PORT_CONN) >> AC_DEFCFG_PORT_CONN_SHIFT], - jack_types[(caps & AC_DEFCFG_DEVICE) >> AC_DEFCFG_DEVICE_SHIFT], - jack_locations[(caps >> (AC_DEFCFG_LOCATION_SHIFT + 4)) & 3], - get_jack_location(caps)); + snd_hda_get_jack_type(caps), + snd_hda_get_jack_connectivity(caps), + snd_hda_get_jack_location(caps)); snd_iprintf(buffer, " Conn = %s, Color = %s\n", get_jack_connection(caps), get_jack_color(caps));

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[alsa-devel] [PATCH2/4] hda: dynamic jack id
by Matthew Ranostay 25 Oct '08

25 Oct '08

This patch duplicates the jack->id pointer with kstrdup() to prevent scoping issues from calling autoprobing functions from the HDA section. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/sound/core/jack.c b/sound/core/jack.c index 4978db1..284432f 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -34,6 +34,7 @@ static int snd_jack_dev_free(struct snd_device *device) else input_free_device(jack->input_dev); + kfree(jack->id); kfree(jack); return 0; @@ -87,7 +88,7 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, if (jack == NULL) return -ENOMEM; - jack->id = id; + jack->id = kstrdup(id, GFP_KERNEL); jack->input_dev = input_allocate_device(); if (jack->input_dev == NULL) {

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Re: [alsa-devel] Problem setting mixer switch with amixer/alsactl.
by Mark Brown 24 Oct '08

24 Oct '08

On Thu, Oct 23, 2008 at 11:12:34PM +0100, Oliver Ford wrote: > Am I right in understanding that setting all the controls in amixer > doesn't actually enable the control in the codec chip at that time but > that the powering on/off is done by soc-dapm.c when the the DAC is > actually enabled when playing some audio? This appears to be what is > happening when it does work (looking at the log). I hadn't realised this. Sort of. The power control happens whenever any power-related control changes and whenever audio is started or stopped. > I've put some debugging in pxa2xx-ac97-write and in dapm_power_widgets() > and the log is attached (stripped of non-audio stuff). I'll have a look at these on Monday.

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Re: [alsa-devel] Nvidia hdmi problems
by Takashi Iwai 24 Oct '08

24 Oct '08

At Wed, 22 Oct 2008 06:43:06 +0200, Nir Tzachar wrote: > > On Wed, Oct 22, 2008 at 4:42 AM, Wei Ni <wni(a)nvidia.com> wrote: > > Hi, Takshi > > We have no official plan to support this nforce 630i HDMI audio in linux > > now, > > and we didn't do any test on it yet. So we can't provide the official > > patch for this addition. > > > > I think if someone interest in it and do some test, he can generate > > patch and submit it. > > As I am interested, and willing o do the tests, I would be happy to > know which tests are needed. If you can play both audio streams and non-audio streams over the device by some applications, it's basically confirmed to work :) Takashi

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[alsa-devel] Audio on omap3+twl4030 codec
by naveen krishna ch 24 Oct '08

24 Oct '08

Hi all, I am working on omap3430 and twl4030 combination for audio. I am wokring on 2.6.26 whre the support was not there for omap3 mcbsp. so i ported some stuff from the 2.6.27 latest git tree and now audio is working on omap3 evm. with the very same kernel when i try with my custom board. I am unable to hear audoi. I get audio when i boot out of MMC (x-load and u-boot in mmc and fiel system is nfs). But, when i usse nand(x-load and u-boot in nand and file system is on nfs) i dont get audio out meams aplay and arecord seems working but i dont hear any audio. I probe and checked the Bit clock and Word clock of the I2S which seems are comming when i try a aplay and arecord.. I am sure that hardware has no problem. Can any one tell me what might be the issue and where i have to start debugging the issue. If this is not the right palce to put my question can any one tell me the correct mailing list. -- Thanks, NeyO

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Re: [alsa-devel] Problem setting mixer switch with amixer/alsactl.
by Mark Brown 24 Oct '08

24 Oct '08

On 23 Oct 2008, at 23:12, Oliver Ford <oliford(a)oliford.co.uk> wrote: > Oliver Ford wrote: > >>> name 'Speaker Playback Volume' >>> value 1 >>> } >>> > This is unrelated, I have just left it there from playing around. > > >> >>> One other thing that would be useful would be to put trace which >>> logs >>> the register writes in ac97_write() in the driver and then try >>> testing >>> both using alsamixer and then amixer to do the final enable of the >>> Speaker Mixer PCM Playback Switch. >>> > Well... > > In terms of the control's direct effect, setting it in alsamixer > gives: > pxa2xx_ac97_write: 0c = af08 > > setting it in amixer: > pxa2xx_ac97_write: 0c = af09 > > > > Am I right in understanding that setting all the controls in amixer > doesn't actually enable the control in the codec chip at that time > but that the powering on/off is done by soc-dapm.c when the the DAC > is actually enabled when playing some audio? This appears to be what > is happening when it does work (looking at the log). I hadn't > realised this. > > However, after running the offending amixer command this no longer > happens. It seems that the dapm_power_widgets() function determines > that the DAC is off and so doesn't power any widgets on at all. > > I've put some debugging in pxa2xx-ac97-write and in > dapm_power_widgets() and the log is attached (stripped of non-audio > stuff). > > > > Oliver > > Clean boot... > > > ** run speaker.sh setup without PCM->speaker setting... ** > pxa2xx_ac97_write: 000c = ef08 > pxa2xx_ac97_write: 001c = 0002 > pxa2xx_ac97_write: 001c = 1802 > pxa2xx_ac97_write: 001e = 0800 > pxa2xx_ac97_write: 0006 = 0081 > pxa2xx_ac97_write: 0002 = 0080 > > ** run alsamixer, set PCM->speaker on: ** > pxa2xx_ac97_write: 000c = af08 > > > # aplay /stuff/400Hz-wav.wav > +hpipaq214_startup() > hpipaq214-audio: Enabling iPAQ Speaker... > hpipaq214_startup, gpio96 -> on > -hpipaq214_startup() > > Playing WAVE '/stuff/400Hz-wav.wav' : Signed 16 bit Little Endian, > Rate 44100 Hz, Mono > pxa2xx_ac97_write: 002a = 0411 > pxa2xx_ac97_write: 002c = ac44 > DAPM: Path Widget 'DAC Inv Mux 2': off --> on > DAPM: Path Widget 'Out 4 Mux': off --> on > DAPM: Path Widget 'Left Speaker Out Mux': off --> on > DAPM: DAC Widget 'Right DAC': off --> on (w->active==1) > pxa2xx_ac97_write: 003c = 3abf > DAPM: DAC Widget 'Left DAC': off --> on (w->active==1) > pxa2xx_ac97_write: 003c = 3a3f > DAPM: Path Widget 'AC97 Mixer': off --> on > DAPM: Path Widget 'Speaker Mixer': off --> on > pxa2xx_ac97_write: 003c = 3a3d > DAPM: Path Widget 'Out 4': off --> on > pxa2xx_ac97_write: 003e = efff > DAPM: Path Widget 'Left Speaker': off --> on > pxa2xx_ac97_write: 003e = eeff > freq changed, set PCD to 5 (LCCR3 = 04700005, LCCR4 = 80000000) > DAPM: Path Widget 'iPAQ Speaker': off --> on > pxa2xx_ac97_write: 003c = 1a3d > hpipaq214_shutdown, gpio96 -> off > > ** aplay exits. ** > > ** ... few seconds delay ... ** > > DAPM: Path Widget 'iPAQ Speaker': on --> off > DAPM: Path Widget 'Out 4': on --> off > pxa2xx_ac97_write: 003e = feff > DAPM: Path Widget 'Left Speaker': on --> off > pxa2xx_ac97_write: 003e = ffff > DAPM: Path Widget 'AC97 Mixer': on --> off > DAPM: Path Widget 'Speaker Mixer': on --> off > pxa2xx_ac97_write: 003c = 1a3f > DAPM: Path Widget 'Right DAC': on --> off > pxa2xx_ac97_write: 003c = 1a7f > DAPM: Path Widget 'Left DAC': on --> off > pxa2xx_ac97_write: 003c = 1aff > DAPM: Path Widget 'DAC Inv Mux 2': on --> off > DAPM: Path Widget 'Out 4 Mux': on --> off > DAPM: Path Widget 'Left Speaker Out Mux': on --> off > pxa2xx_ac97_write: 003c = 1aff > pxa2xx_ac97_write: 0026 = 0000 > > > ** all quiet ** > > # > # > # amixer cset numid=62 1 > pxa2xx_ac97_write: 000c = af09 > numid=62,iface=MIXER,name='Speaker Mixer PCM Playback Swit' > ; type=BOOLEAN,access=rw---,values=1 > : values=on > > > > # aplay /stuff/400Hz-wav.wav > +hpipaq214_startup() > hpipaq214-audio: Enabling iPAQ Speaker... > hpipaq214_startup, gpio96 -> on > -hpipaq214_startup() > > Playing WAVE '/stuff/400Hz-wav.wav' : Signed 16 bit Little Endian, > Rate 44100 Hz, Mono > freq changed, set PCD to 5 (LCCR3 = 04700005, LCCR4 = 80000000) > pxa2xx_ac97_write: 002a = 0411 > pxa2xx_ac97_write: 002c = ac44 > DAPM: DAC Widget 'Right DAC': off --> off (w->active==1) > DAPM: DAC Widget 'Left DAC': off --> off (w->active==1) > pxa2xx_ac97_write: 003c = 1aff > hpipaq214_shutdown, gpio96 -> off > # > # > pxa2xx_ac97_write: 003c = 1aff > pxa2xx_ac97_write: 0026 = 0000

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[alsa-devel] [PATCH] aoa i2sbus: don't overwrite module parameter
by Johannes Berg 23 Oct '08

23 Oct '08

We shouldn't modify a global variable here. Signed-off-by: Johannes Berg <johannes(a)sipsolutions.net> --- sound/aoa/soundbus/i2sbus/i2sbus-core.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) --- everything.orig/sound/aoa/soundbus/i2sbus/i2sbus-core.c 2008-10-23 13:55:14.000000000 +0200 +++ everything/sound/aoa/soundbus/i2sbus/i2sbus-core.c 2008-10-23 13:55:33.000000000 +0200 @@ -159,7 +159,7 @@ static int i2sbus_add_dev(struct macio_d struct i2sbus_dev *dev; struct device_node *child = NULL, *sound = NULL; struct resource *r; - int i, layout = 0, rlen; + int i, layout = 0, rlen, ok = force; static const char *rnames[] = { "i2sbus: %s (control)", "i2sbus: %s (tx)", "i2sbus: %s (rx)" }; @@ -192,7 +192,7 @@ static int i2sbus_add_dev(struct macio_d layout = *layout_id; snprintf(dev->sound.modalias, 32, "sound-layout-%d", layout); - force = 1; + ok = 1; } } /* for the time being, until we can handle non-layout-id @@ -201,7 +201,7 @@ static int i2sbus_add_dev(struct macio_d * When there are two i2s busses and only one has a layout-id, * then this depends on the order, but that isn't important * either as the second one in that case is just a modem. */ - if (!force) { + if (!ok) { kfree(dev); return -ENODEV; }

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[alsa-devel] [PATCH] ASoC: tlv320aic3x: Fix DSP DAI format and signal polarities matching
by Jarkko Nikula 23 Oct '08

23 Oct '08

- Codec doesn't support to configure bit clock and frame sync polarities - Codec doesn't support DSP_A format but DSP_B with inverted bit clock polarity - Match also other formats with their signal polarities Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> --- sound/soc/codecs/tlv320aic3x.c | 16 ++++++++++------ 1 files changed, 10 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 05336ed..cff276e 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -863,17 +863,21 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - /* interface format */ - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_I2S: + /* + * match both interface format and signal polarities since they + * are fixed + */ + switch (fmt & (SND_SOC_DAIFMT_FORMAT_MASK | + SND_SOC_DAIFMT_INV_MASK)) { + case (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF): break; - case SND_SOC_DAIFMT_DSP_A: + case (SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF): iface_breg |= (0x01 << 6); break; - case SND_SOC_DAIFMT_RIGHT_J: + case (SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_NB_NF): iface_breg |= (0x02 << 6); break; - case SND_SOC_DAIFMT_LEFT_J: + case (SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF): iface_breg |= (0x03 << 6); break; default: -- 1.5.6.5

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[alsa-devel] [PATCH 0/4] ASoC: Blackfin updates
by Bryan Wu 23 Oct '08

23 Oct '08

Hi Mark and Takashi: We updates the patchset: - split out AD1980 change to a seperated patch - coding style change Thanks -Bryan

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[alsa-devel] Select source support in alsa soc
by naveen krishna ch 23 Oct '08

23 Oct '08

I am in a process of making a asoc driver for costum board. with OMAP3430 and twl4030 combination.. I have a driver now at a palce where i can play back with stereo headset and record through line in cable. I need some understanding of how can i give support for selecting output and input sources. I have understood the way it is done in ALSA but i feel the sane should not be used in ALSA SOC frame work. I would be thankful if someone can help me solve this issue. Thanks, Neyo

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