February 2010 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] 2.6.30-rc8 skipping regression also affects cmipci
by Frédéric Brière 07 Mar '10

07 Mar '10

Hi everybody, I was only recently able to upgrade from 2.6.26 to 2.6.32 (long story). In so doing, I noticed infrequent occurrences of audio skipping, which I managed to bisect down to c87d973. It would appear that this commit also affects cmipci (CMI8738), albeit not as badly as it did for intel8x0. (I guess I'm the only person left on Earth still using a stand-alone PCI sound card, which is why this was never reported. <g>) Here's a sample log output after turning xrun debugging on: [ 171.555860] PCM: hw_ptr skipping! (pos=6341, delta=16383, period=1024, jdelta=0/85) [ 181.114680] PCM: hw_ptr skipping! (pos=12529, delta=16370, period=1024, jdelta=0/85) [ 181.114770] PCM: hw_ptr skipping! (pos=12534, delta=16375, period=1024, jdelta=0/85) [ 1713.774316] PCM: hw_ptr skipping! (pos=9456, delta=16369, period=1024, jdelta=0/85) [ 1713.774346] PCM: hw_ptr skipping! (pos=9458, delta=16371, period=1024, jdelta=0/85) [ 1713.774409] PCM: hw_ptr skipping! (pos=9461, delta=16374, period=1024, jdelta=0/85) [ 1713.774613] PCM: hw_ptr skipping! (pos=9470, delta=16383, period=1024, jdelta=0/85) [ 1831.092000] PCM: hw_ptr skipping! (pos=1712, delta=16373, period=1024, jdelta=0/85) [ 1953.347453] PCM: hw_ptr skipping! (pos=3723, delta=16316, period=1024, jdelta=0/84) [ 1953.347465] PCM: hw_ptr skipping! (pos=3724, delta=16317, period=1024, jdelta=0/84) [ 1953.347485] PCM: hw_ptr skipping! (pos=3725, delta=16318, period=1024, jdelta=0/84) You'll notice that this occurs much less frequently than it did for intel8x0. However, I left MPlayer on pause for the night, and at some point the kernel started spewing several syslog messages every couple of seconds: [25332.211286] PCM: hw_ptr skipping! [Q] (pos=4, delta=-1073740800, period=1024, jdelta=5/44744/0) [25332.232637] PCM: hw_ptr skipping! [Q] (pos=1029, delta=-1073739771, period=1024, jdelta=6/44749/0) [25332.253957] PCM: hw_ptr skipping! [Q] (pos=2053, delta=-1073738747, period=1024, jdelta=5/44755/0) [25332.275292] PCM: hw_ptr skipping! [Q] (pos=3077, delta=-1073737723, period=1024, jdelta=5/44760/0) [25332.296616] PCM: hw_ptr skipping! [Q] (pos=4101, delta=-1073736699, period=1024, jdelta=6/44765/0) [25332.317932] PCM: hw_ptr skipping! [Q] (pos=5125, delta=-1073735675, period=1024, jdelta=5/44771/0) [25332.339269] PCM: hw_ptr skipping! [Q] (pos=6149, delta=-1073734651, period=1024, jdelta=5/44776/0) [25332.360600] PCM: hw_ptr skipping! [Q] (pos=7173, delta=-1073733627, period=1024, jdelta=6/44781/0) [25332.381942] PCM: hw_ptr skipping! [Q] (pos=8197, delta=-1073732603, period=1024, jdelta=5/44787/0) [25332.403261] PCM: hw_ptr skipping! [Q] (pos=9221, delta=-1073731579, period=1024, jdelta=5/44792/0) [25337.244901] PCM: hw_ptr skipping! [Q] (pos=12293, delta=-1073728507, period=1024, jdelta=6/44808/0) [25337.266223] PCM: hw_ptr skipping! [Q] (pos=13317, delta=-1073727483, period=1024, jdelta=5/44813/0) This kept going on until I quit MPlayer (which, as I said, was not actually playing anything all this time). Not that skipped silence is much of an issue, but I thought this might add useful information. Let me know if I can be of any help in resolving this. -- If you want to travel around the world and be invited to speak at a lot of different places, just write a Unix operating system. -- Linus Torvalds

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[alsa-devel] snd-powermac @ G4 DA (PowerMac3, 4) - headphone detection doesn't work
by Douglas Mencken 04 Mar '10

04 Mar '10

I can hear sound with internal speakers only if I plug headphones jack, without plugged jack, there's no sound at all; to listen thru headphones, I need to half-insert that jack and this leads to distortions; with h/phones jack plugged fully, no sound in phones, but sound in internal speaker. With the same half-plug, in OS X I head sound both via h/phones and internal speaker. Driver: snd-powermac. Audio card: Tumbler @ i2ca @ KeyLargo i2c (I can provide the full dump of OpenFirmware device tree). As <gnubien> at #alsa @ irc.freenode.net suggested, I ran `alsamixer -c 0` and MM-ed 'Auto Mute'. After that, I can fully plug h/phones jack and enjoy full sounds w/o distortions but still can't get any sound when it's unplugged. The result of running ./alsa-info.sh: Your ALSA information is located at http://www.alsa-project.org/db/?f=daac9e06cbe772222b90d7fba14c1a49a65b980c

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[alsa-devel] [RFC] Cleanup/refactor the usbaudio driver
by Daniel Mack 04 Mar '10

04 Mar '10

I've spent some time cleaning up the usbaudio driver. I felt that the existing implementation was quite difficult to follow because it contained code for all sorts of things mixed up in one big file. So I split the driver into pieces and tried to be as less intrusive as I can, meaning that I didn't touch much real code but rather copied the functions as they were and factored them out. Of course, non-static functions were properly prefixed to avoid name space pollution. I also moved almost all quirks to a seperate file and introduced two more generic hooks to handle them, so the excecption add less to the actual code of the driver. Unfortunately it wasn't easily possible to put that big change into smaller pieces, and so 3/3 became a huge chunk. After all, I feel that it is easier to read now, but I don't know whether you agree. So take this as RFC for now :) Daniel

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[alsa-devel] [PATCH] ALSA: ASoC: fix ak4104 register array access
by Daniel Mack 03 Mar '10

03 Mar '10

Don't touch the variable 'reg' to construct the value for the actual SPI transport. This variable is again used to access the driver's register cache, and so random memory is overwritten. Compute the value in-place instead. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: stable(a)kernel.org --- sound/soc/codecs/ak4104.c | 6 ++---- 1 files changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index b9ef7e4..b68d99f 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -90,12 +90,10 @@ static int ak4104_spi_write(struct snd_soc_codec *codec, unsigned int reg, if (reg >= codec->reg_cache_size) return -EINVAL; - reg &= AK4104_REG_MASK; - reg |= AK4104_WRITE; - /* only write to the hardware if value has changed */ if (cache[reg] != value) { - u8 tmp[2] = { reg, value }; + u8 tmp[2] = { (reg & AK4104_REG_MASK) | AK4104_WRITE, value }; + if (spi_write(spi, tmp, sizeof(tmp))) { dev_err(&spi->dev, "SPI write failed\n"); return -EIO; -- 1.6.3.3

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[alsa-devel] snd_pcm_drop causing lockup with DMix
by John Lindgren 03 Mar '10

03 Mar '10

Hi, A couple of weeks ago I reported a rather serious bug #4914 that can cause a multi-threaded using ALSA to lock up so that it can only be ended with SIGKILL. I maintain the ALSA output part of Audacious, and I need to be able to play audio in one thread of the program and then signal that thread to stop from another. The way to do this in ALSA, if I understand correctly, is to interrupt the thread that is sending audio with a call to snd_pcm_drop. But using DMix makes the first thread hang up in a snd_pcm_wait call with a timeout of -1. Is anyone working on this bug? Is there any more help I can provide? John Lindgren

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[alsa-devel] No CD mixer on Intel G35 (Asus P5E-V(M))
by Mario 'BitKoenig' Holbe 02 Mar '10

02 Mar '10

Hello, on my Asus P5E-V HDMI and my Asus P5E-VM HDMI (both Intel G35) alsamixer shows no CD mixer. Both boards have a CD audio connector and the CD-drive connected to it. holbe@darkside:/home/holbe% uname -a Linux darkside 2.6.32-3-686-bigmem #1 SMP Thu Feb 25 06:54:30 UTC 2010 i686 GNU/Linux holbe@darkside:/home/holbe% cat /proc/asound/cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xfe5f4000 irq 22 1 [Bt878 ]: Bt87x - Brooktree Bt878 Brooktree Bt878 at 0xfdfff000, irq 16 alsamixer says: Card: HDA Intel Chip: Silicon Image SiI1392 HDMI Same with 2.6.33. On my P5Q-EM (Intel G45) the CD mixer appears in alsamixer. I have no idea how to dig deeper in this, if you need more information, please just tell me. Thanks for your work & regards Mario -- Do I contradict myself? Very well, then I contradict myself, I am large, I contain multitudes. -- Walt Whitman

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[alsa-devel] [PATCH 22/66] sound/oss/v_midi.h: Checkpatch cleanup
by Andrea Gelmini 02 Mar '10

02 Mar '10

sound/oss/v_midi.h:5: ERROR: code indent should use tabs where possible sound/oss/v_midi.h:7: ERROR: trailing whitespace Signed-off-by: Andrea Gelmini <andrea.gelmini(a)gelma.net> --- sound/oss/v_midi.h | 5 ++--- 1 files changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/oss/v_midi.h b/sound/oss/v_midi.h index 1b86cb4..08e2185 100644 --- a/sound/oss/v_midi.h +++ b/sound/oss/v_midi.h @@ -2,9 +2,9 @@ typedef struct vmidi_devc { int dev; /* State variables */ - int opened; + int opened; spinlock_t lock; - + /* MIDI fields */ int my_mididev; int pair_mididev; @@ -12,4 +12,3 @@ typedef struct vmidi_devc { int intr_active; void (*midi_input_intr) (int dev, unsigned char data); } vmidi_devc; - -- 1.7.0.90.g251a4

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[alsa-devel] [PATCH 1/1] sound/pci/hda/hda_codec.c: various coding style fixes
by Norberto Lopes 02 Mar '10

02 Mar '10

Signed-off-by: Norberto Lopes <nlopes.ml(a)gmail.com> --- sound/pci/hda/hda_codec.c | 69 ++++++++++++++++++++++++-------------------- 1 files changed, 38 insertions(+), 31 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index f98b47c..444b989 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -955,8 +955,9 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, * * Returns 0 if successful, or a negative error code. */ -int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr, - struct hda_codec **codecp) +int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, + unsigned int codec_addr, + struct hda_codec **codecp) { struct hda_codec *codec; char component[31]; @@ -1162,7 +1163,7 @@ EXPORT_SYMBOL_HDA(snd_hda_codec_cleanup_stream); */ /* FIXME: more better hash key? */ -#define HDA_HASH_KEY(nid,dir,idx) (u32)((nid) + ((idx) << 16) + ((dir) << 24)) +#define HDA_HASH_KEY(nid, dir, idx) (u32)((nid) + ((idx) << 16) + ((dir) << 24)) #define HDA_HASH_PINCAP_KEY(nid) (u32)((nid) + (0x02 << 24)) #define HDA_HASH_PARPCM_KEY(nid) (u32)((nid) + (0x03 << 24)) #define HDA_HASH_PARSTR_KEY(nid) (u32)((nid) + (0x04 << 24)) @@ -1332,7 +1333,8 @@ u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid) if (!codec->no_trigger_sense) { pincap = snd_hda_query_pin_caps(codec, nid); if (pincap & AC_PINCAP_TRIG_REQ) /* need trigger? */ - snd_hda_codec_read(codec, nid, 0, AC_VERB_SET_PIN_SENSE, 0); + snd_hda_codec_read(codec, nid, 0, + AC_VERB_SET_PIN_SENSE, 0); } return snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_SENSE, 0); @@ -1348,8 +1350,8 @@ EXPORT_SYMBOL_HDA(snd_hda_pin_sense); */ int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid) { - u32 sense = snd_hda_pin_sense(codec, nid); - return !!(sense & AC_PINSENSE_PRESENCE); + u32 sense = snd_hda_pin_sense(codec, nid); + return !!(sense & AC_PINSENSE_PRESENCE); } EXPORT_SYMBOL_HDA(snd_hda_jack_detect); @@ -1891,7 +1893,7 @@ int snd_hda_add_vmaster(struct hda_codec *codec, char *name, err = snd_hda_ctl_add(codec, 0, kctl); if (err < 0) return err; - + for (s = slaves; *s; s++) { struct snd_kcontrol *sctl; int i = 0; @@ -2378,27 +2380,27 @@ static struct snd_kcontrol_new dig_mixes[] = { { .access = SNDRV_CTL_ELEM_ACCESS_READ, .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,CON_MASK), + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, CON_MASK), .info = snd_hda_spdif_mask_info, .get = snd_hda_spdif_cmask_get, }, { .access = SNDRV_CTL_ELEM_ACCESS_READ, .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,PRO_MASK), + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, PRO_MASK), .info = snd_hda_spdif_mask_info, .get = snd_hda_spdif_pmask_get, }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT), + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT), .info = snd_hda_spdif_mask_info, .get = snd_hda_spdif_default_get, .put = snd_hda_spdif_default_put, }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,SWITCH), + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, SWITCH), .info = snd_hda_spdif_out_switch_info, .get = snd_hda_spdif_out_switch_get, .put = snd_hda_spdif_out_switch_put, @@ -2549,7 +2551,7 @@ static int snd_hda_spdif_in_status_get(struct snd_kcontrol *kcontrol, static struct snd_kcontrol_new dig_in_ctls[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("",CAPTURE,SWITCH), + .name = SNDRV_CTL_NAME_IEC958("", CAPTURE, SWITCH), .info = snd_hda_spdif_in_switch_info, .get = snd_hda_spdif_in_switch_get, .put = snd_hda_spdif_in_switch_put, @@ -2557,7 +2559,7 @@ static struct snd_kcontrol_new dig_in_ctls[] = { { .access = SNDRV_CTL_ELEM_ACCESS_READ, .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("",CAPTURE,DEFAULT), + .name = SNDRV_CTL_NAME_IEC958("", CAPTURE, DEFAULT), .info = snd_hda_spdif_mask_info, .get = snd_hda_spdif_in_status_get, }, @@ -2822,7 +2824,7 @@ int /*__devinit*/ snd_hda_build_controls(struct hda_bus *bus) int err = snd_hda_codec_build_controls(codec); if (err < 0) { printk(KERN_ERR "hda_codec: cannot build controls" - "for #%d (error %d)\n", codec->addr, err); + "for #%d (error %d)\n", codec->addr, err); err = snd_hda_codec_reset(codec); if (err < 0) { printk(KERN_ERR @@ -2918,8 +2920,12 @@ unsigned int snd_hda_calc_stream_format(unsigned int rate, val |= channels - 1; switch (snd_pcm_format_width(format)) { - case 8: val |= 0x00; break; - case 16: val |= 0x10; break; + case 8: + val |= 0x00; + break; + case 16: + val |= 0x10; + break; case 20: case 24: case 32: @@ -3235,7 +3241,8 @@ static int get_empty_pcm_device(struct hda_bus *bus, int type) if (!test_and_set_bit(audio_idx[type][i], bus->pcm_dev_bits)) return audio_idx[type][i]; - snd_printk(KERN_WARNING "Too many %s devices\n", snd_hda_pcm_type_name[type]); + snd_printk(KERN_WARNING "Too many %s devices\n", + snd_hda_pcm_type_name[type]); return -EAGAIN; } @@ -3273,7 +3280,7 @@ int snd_hda_codec_build_pcms(struct hda_codec *codec) err = codec->patch_ops.build_pcms(codec); if (err < 0) { printk(KERN_ERR "hda_codec: cannot build PCMs" - "for #%d (error %d)\n", codec->addr, err); + "for #%d (error %d)\n", codec->addr, err); err = snd_hda_codec_reset(codec); if (err < 0) { printk(KERN_ERR @@ -3403,8 +3410,8 @@ EXPORT_SYMBOL_HDA(snd_hda_check_board_config); /** * snd_hda_check_board_codec_sid_config - compare the current codec - subsystem ID with the - config table + subsystem ID with the + config table This is important for Gateway notebooks with SB450 HDA Audio where the vendor ID of the PCI device is: @@ -3542,7 +3549,7 @@ void snd_hda_update_power_acct(struct hda_codec *codec) * * Increment the power-up counter and power up the hardware really when * not turned on yet. - */ + */ void snd_hda_power_up(struct hda_codec *codec) { struct hda_bus *bus = codec->bus; @@ -3571,7 +3578,7 @@ EXPORT_SYMBOL_HDA(snd_hda_power_up); * * Decrement the power-up counter and schedules the power-off work if * the counter rearches to zero. - */ + */ void snd_hda_power_down(struct hda_codec *codec) { --codec->power_count; @@ -3597,7 +3604,7 @@ EXPORT_SYMBOL_HDA(snd_hda_power_down); * * This function is supposed to be set or called from the check_power_status * patch ops. - */ + */ int snd_hda_check_amp_list_power(struct hda_codec *codec, struct hda_loopback_check *check, hda_nid_t nid) @@ -3765,7 +3772,7 @@ static void setup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid, { /* turn off SPDIF once; otherwise the IEC958 bits won't be updated */ if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE)) - set_dig_out_convert(codec, nid, + set_dig_out_convert(codec, nid, codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff, -1); snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format); @@ -4024,13 +4031,13 @@ static int is_in_nid_list(hda_nid_t nid, hda_nid_t *list) /* * Sort an associated group of pins according to their sequence numbers. */ -static void sort_pins_by_sequence(hda_nid_t * pins, short * sequences, +static void sort_pins_by_sequence(hda_nid_t *pins, short *sequences, int num_pins) { int i, j; short seq; hda_nid_t nid; - + for (i = 0; i < num_pins; i++) { for (j = i + 1; j < num_pins; j++) { if (sequences[i] > sequences[j]) { @@ -4058,7 +4065,7 @@ static void sort_pins_by_sequence(hda_nid_t * pins, short * sequences, * is detected, one of speaker of HP pins is assigned as the primary * output, i.e. to line_out_pins[0]. So, line_outs is always positive * if any analog output exists. - * + * * The analog input pins are assigned to input_pins array. * The digital input/output pins are assigned to dig_in_pin and dig_out_pin, * respectively. @@ -4121,9 +4128,9 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, case AC_JACK_SPEAKER: seq = get_defcfg_sequence(def_conf); assoc = get_defcfg_association(def_conf); - if (! assoc) + if (!assoc) continue; - if (! assoc_speaker) + if (!assoc_speaker) assoc_speaker = assoc; else if (assoc_speaker != assoc) continue; @@ -4221,7 +4228,7 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, cfg->speaker_outs); sort_pins_by_sequence(cfg->hp_pins, sequences_hp, cfg->hp_outs); - + /* if we have only one mic, make it AUTO_PIN_MIC */ if (!cfg->input_pins[AUTO_PIN_MIC] && cfg->input_pins[AUTO_PIN_FRONT_MIC]) { @@ -4371,7 +4378,7 @@ EXPORT_SYMBOL_HDA(snd_hda_resume); /** * snd_array_new - get a new element from the given array * @array: the array object - * + * * Get a new element from the given array. If it exceeds the * pre-allocated array size, re-allocate the array. * -- 1.6.4

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Re: [alsa-devel] HD-Audio device unsuppored (Nvidia G2xx-series GPU implemented)
by Stefan Ringel 02 Mar '10

02 Mar '10

I forgot this attachment 1.patch and alsa-info.txt.

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[alsa-devel] [PATCHv4 7/7] ASoC: TWL6030: Detect power-up sequence completion
by Olaya, Margarita 02 Mar '10

02 Mar '10

From: Misael Lopez Cruz <x0052729(a)ti.com> When the codec is powered-up through external AUDPWRON line it starts its power-up sequence. The completion of the sequence is signaled through the audio interrupt, and then codec is operational. If NAUDINT irq is provided, CODEC driver starts a wait_for_completion just after AUDPWRON line transitions from low to high. It's signaled as complete when servicing READYINT interrupt. If AUDPWRON gpio line is provided but NAUDINT irq is not, then CODEC driver enables READYINT and polls on INTID register. If none of them are provided, then CODEC uses manual power sequences and disables all audio interrupts. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/codecs/twl6030.c | 61 +++++++++++++++++++++++++++++++++++++------ sound/soc/codecs/twl6030.h | 1 + 2 files changed, 53 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/twl6030.c b/sound/soc/codecs/twl6030.c index b8dd5ae..2847f1b 100644 --- a/sound/soc/codecs/twl6030.c +++ b/sound/soc/codecs/twl6030.c @@ -52,6 +52,7 @@ struct twl6030_data { int non_lp; unsigned int sysclk; struct snd_pcm_hw_constraint_list *sysclk_constraints; + struct completion ready; }; /* @@ -372,6 +373,7 @@ static int twl6030_power_mode_event(struct snd_soc_dapm_widget *w, static irqreturn_t twl6030_naudint_handler(int irq, void *data) { struct snd_soc_codec *codec = data; + struct twl6030_data *priv = codec->private_data; u8 intid; twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &intid, TWL6030_REG_INTID); @@ -391,7 +393,7 @@ static irqreturn_t twl6030_naudint_handler(int irq, void *data) dev_alert(codec->dev, "vib drivers over current detection\n"); break; case TWL6030_READYINT: - dev_alert(codec->dev, "codec is ready\n"); + complete(&priv->ready); break; default: dev_err(codec->dev, "unknown audio interrupt %d\n", intid); @@ -626,11 +628,45 @@ static int twl6030_add_widgets(struct snd_soc_codec *codec) return 0; } +static int twl6030_power_up_completion(struct snd_soc_codec *codec, + int naudint) +{ + struct twl6030_data *priv = codec->private_data; + int time_left; + u8 intid; + + if (naudint) { + /* wait for ready interrupt with 48 ms timeout */ + time_left = wait_for_completion_timeout(&priv->ready, + msecs_to_jiffies(48)); + } else { + /* retry 3 times only */ + for (time_left = 3; time_left > 0; time_left--) { + mdelay(16); + twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &intid, + TWL6030_REG_INTID); + if (intid & TWL6030_READYINT) + break; + } + } + + if (!time_left) { + dev_err(codec->dev, "timeout waiting for READYINT\n"); + return -ETIMEDOUT; + } + + priv->codec_powered = 1; + + return 0; +} + static int twl6030_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct twl6030_data *priv = codec->private_data; int audpwron = priv->audpwron; + int naudint = priv->naudint; + int ret; switch (level) { case SND_SOC_BIAS_ON: @@ -643,8 +679,10 @@ static int twl6030_set_bias_level(struct snd_soc_codec *codec, /* use AUDPWRON line */ gpio_set_value(audpwron, 1); - /* power-up sequence latency */ - mdelay(16); + /* wait for power-up completion */ + ret = twl6030_power_up_completion(codec, naudint); + if (ret) + return ret; /* sync registers updated during power-up sequence */ twl6030_read(codec, TWL6030_REG_NCPCTL); @@ -653,12 +691,11 @@ static int twl6030_set_bias_level(struct snd_soc_codec *codec, } else { /* use manual power-up sequence */ twl6030_power_up(codec); + priv->codec_powered = 1; } /* initialize vdd/vss registers with reg_cache */ twl6030_init_vdd_regs(codec); - - priv->codec_powered = 1; break; case SND_SOC_BIAS_OFF: if (!priv->codec_powered) @@ -1067,6 +1104,7 @@ static int __devinit twl6030_codec_probe(struct platform_device *pdev) mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); + init_completion(&priv->ready); if (gpio_is_valid(audpwron)) { ret = gpio_request(audpwron, "audpwron"); @@ -1089,10 +1127,15 @@ static int __devinit twl6030_codec_probe(struct platform_device *pdev) if (ret) goto gpio2_err; } else { - dev_warn(codec->dev, - "no naudint irq, audio interrupts disabled\n"); - twl6030_write_reg_cache(codec, TWL6030_REG_INTMR, - TWL6030_ALLINT_MSK); + if (gpio_is_valid(audpwron)) { + /* enable only codec ready interrupt */ + twl6030_write_reg_cache(codec, TWL6030_REG_INTMR, + ~TWL6030_READYMSK & TWL6030_ALLINT_MSK); + } else { + /* no interrupts at all */ + twl6030_write_reg_cache(codec, TWL6030_REG_INTMR, + TWL6030_ALLINT_MSK); + } } /* init vio registers */ diff --git a/sound/soc/codecs/twl6030.h b/sound/soc/codecs/twl6030.h index 90b8a44..d591be1 100644 --- a/sound/soc/codecs/twl6030.h +++ b/sound/soc/codecs/twl6030.h @@ -79,6 +79,7 @@ /* INTMR (0x04) fields */ +#define TWL6030_READYMSK 0x40 #define TWL6030_ALLINT_MSK 0x7B /* NCPCTL (0x05) fields */ -- 1.6.1.3

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