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

[alsa-devel] [patch 1/6] sound azt3328: cosmetics, minor updates
by andi＠lisas.de 27 Dec '10

27 Dec '10

ChangeLog: - correct samples to be POSIX shell compatible - add logging of jiffies value in _pointer() - several comments - cleanup Signed-off-by: Andreas Mohr <andi(a)lisas.de> Index: linux-2.6/sound/pci/azt3328.c =================================================================== --- linux-2.6.orig/sound/pci/azt3328.c 2010-12-27 14:28:57.722851002 -0500 +++ linux-2.6/sound/pci/azt3328.c 2010-12-27 14:29:00.449517668 -0500 @@ -1,6 +1,6 @@ /* * azt3328.c - driver for Aztech AZF3328 based soundcards (e.g. PCI168). - * Copyright (C) 2002, 2005 - 2009 by Andreas Mohr <andi AT lisas.de> + * Copyright (C) 2002, 2005 - 2010 by Andreas Mohr <andi AT lisas.de> * * Framework borrowed from Bart Hartgers's als4000.c. * Driver developed on PCI168 AP(W) version (PCI rev. 10, subsystem ID 1801), @@ -201,14 +201,15 @@ MODULE_SUPPORTED_DEVICE("{{Aztech,AZF332 /* === Debug settings === Further diagnostic functionality than the settings below - does not need to be provided, since one can easily write a bash script + does not need to be provided, since one can easily write a POSIX shell script to dump the card's I/O ports (those listed in lspci -v -v): - function dump() + dump() { local descr=$1; local addr=$2; local count=$3 echo "${descr}: ${count} @ ${addr}:" - dd if=/dev/port skip=$[${addr}] count=${count} bs=1 2>/dev/null| hexdump -C + dd if=/dev/port skip=`printf %d ${addr}` count=${count} bs=1 \ + 2>/dev/null| hexdump -C } and then use something like "dump joy200 0x200 8", "dump mpu388 0x388 4", "dump joy 0xb400 8", @@ -216,14 +217,14 @@ MODULE_SUPPORTED_DEVICE("{{Aztech,AZF332 possibly within a "while true; do ... sleep 1; done" loop. Tweaking ports could be done using VALSTRING="`printf "%02x" $value`" - printf "\x""$VALSTRING"|dd of=/dev/port seek=$[${addr}] bs=1 2>/dev/null + printf "\x""$VALSTRING"|dd of=/dev/port seek=`printf %d ${addr}` bs=1 \ + 2>/dev/null */ #define DEBUG_MISC 0 #define DEBUG_CALLS 0 #define DEBUG_MIXER 0 #define DEBUG_CODEC 0 -#define DEBUG_IO 0 #define DEBUG_TIMER 0 #define DEBUG_GAME 0 #define DEBUG_PM 0 @@ -299,6 +300,7 @@ struct snd_azf3328_codec_data { }; enum snd_azf3328_codec_type { + /* warning: fixed indices (also used for bitmask checks!) */ AZF_CODEC_PLAYBACK = 0, AZF_CODEC_CAPTURE = 1, AZF_CODEC_I2S_OUT = 2, @@ -362,6 +364,9 @@ MODULE_DEVICE_TABLE(pci, snd_azf3328_ids static int snd_azf3328_io_reg_setb(unsigned reg, u8 mask, bool do_set) { + /* Well, strictly spoken, the inb/outb sequence isn't atomic + and would need locking. However we currently don't care + since it potentially complicates matters. */ u8 prev = inb(reg), new; new = (do_set) ? (prev|mask) : (prev & ~mask); @@ -1004,7 +1009,8 @@ snd_azf3328_codec_setfmt(struct snd_azf3 * (FIXME: yes, it works, but what exactly am I doing here?? :) * FIXME: does this have some side effects for full-duplex * or other dramatic side effects? */ - if (codec_type == AZF_CODEC_PLAYBACK) /* only do it for playback */ + /* do it for non-capture codecs only */ + if (codec_type == AZF_CODEC_PLAYBACK) snd_azf3328_codec_outw(codec, IDX_IO_CODEC_DMA_FLAGS, snd_azf3328_codec_inw(codec, IDX_IO_CODEC_DMA_FLAGS) | DMA_RUN_SOMETHING1 | @@ -1368,8 +1374,8 @@ snd_azf3328_codec_pointer(struct snd_pcm /* calculate offset */ result -= bufptr; frmres = bytes_to_frames( substream->runtime, result); - snd_azf3328_dbgcodec("%s @ 0x%8lx, frames %8ld\n", - codec->name, result, frmres); + snd_azf3328_dbgcodec("%08li %s @ 0x%8lx, frames %8ld\n", + jiffies, codec->name, result, frmres); return frmres; } @@ -1532,7 +1538,7 @@ snd_azf3328_gameport_cooked_read(struct } } - /* trigger next axes sampling, to be evaluated the next time we + /* trigger next sampling of axes, to be evaluated the next time we * enter this function */ /* for some very, very strange reason we cannot enable @@ -1966,7 +1972,7 @@ snd_azf3328_timer_start(struct snd_timer snd_azf3328_dbgtimer("delay was too low (%d)!\n", delay); delay = 49; /* minimum time is 49 ticks */ } - snd_azf3328_dbgtimer("setting timer countdown value %d, add COUNTDOWN|IRQ\n", delay); + snd_azf3328_dbgtimer("setting timer countdown value %d\n", delay); delay |= TIMER_COUNTDOWN_ENABLE | TIMER_IRQ_ENABLE; spin_lock_irqsave(&chip->reg_lock, flags); snd_azf3328_ctrl_outl(chip, IDX_IO_TIMER_VALUE, delay); @@ -2257,7 +2263,7 @@ snd_azf3328_create(struct snd_card *card struct snd_azf3328_codec_data *codec = &chip->codecs[codec_type]; - /* shutdown codecs to save power */ + /* shutdown codecs to reduce power / noise */ /* have ...ctrl_codec_activity() act properly */ codec->running = 1; snd_azf3328_ctrl_codec_activity(chip, codec_type, 0); @@ -2419,6 +2425,7 @@ snd_azf3328_suspend(struct pci_dev *pci, snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + /* same pcm object for playback/capture */ snd_pcm_suspend_all(chip->pcm[AZF_CODEC_PLAYBACK]); snd_pcm_suspend_all(chip->pcm[AZF_CODEC_I2S_OUT]); --

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[alsa-devel] ASoC updates for 2.6.38
by Mark Brown 27 Dec '10

27 Dec '10

The following changes since commit d08935711b542c3fe5c51abf290e172536d4cf09: Merge branch 'for-2.6.38' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc (2010-12-23 13:39:59 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.38 Sorry, just realised that I've forgotten the (trivial) merge up of your warning fix for soc-dapm.c but it'll take kernel.org longer than I have to mirror out the merge - it's done locally and will be in the next pull request. Mark Brown (4): ASoC: Remove incorrect WM8903 erratum workaround ASoC: Remove WM8995 write sequencer bitfield definitions ASoC: Fix typo in x86 workaround ASoC: Fix double comment start sound/soc/codecs/wm8903.c | 20 +- sound/soc/codecs/wm8903.h | 21 - sound/soc/codecs/wm8962.c | 2 +- sound/soc/codecs/wm8995.h | 4480 --------------------------------------------- 4 files changed, 7 insertions(+), 4516 deletions(-)

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[alsa-devel] ASoC updates for 2.6.38
by Mark Brown 27 Dec '10

27 Dec '10

The following changes since commit d08935711b542c3fe5c51abf290e172536d4cf09: Merge branch 'for-2.6.38' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc (2010-12-23 13:39:59 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.38 Mark Brown (4): ASoC: Remove incorrect WM8903 erratum workaround ASoC: Remove WM8995 write sequencer bitfield definitions ASoC: Fix typo in x86 workaround ASoC: Fix double comment start sound/soc/codecs/wm8903.c | 20 +- sound/soc/codecs/wm8903.h | 21 - sound/soc/codecs/wm8962.c | 2 +- sound/soc/codecs/wm8995.h | 4480 --------------------------------------------- 4 files changed, 7 insertions(+), 4516 deletions(-)

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[alsa-devel] HDA: Sound mutes when combination is below -48 dB
by David Henningsson 27 Dec '10

27 Dec '10

I'm trying to nail down a strange behaviour. I have one "Master" and one "Speaker" (or "Headphone") volume control, each ranging from -48 dB to 0 dB. Whenever the sum of "Master" and "Speaker" is below -48 dB, the output mutes. Codec is attached. "Master" is a "Virtual master" control, and both "Master" and "Speaker" control amp-out on nid 0x13. Nid 0x13 does not have amp-out caps, but the AFG specifies a range of -96 dB to 0 dB in 0.75 dB steps. There is some strange things here. First, in patch_sigmatel:create_controls_idx, there is this comment: /* if dB scale is over -64dB, and finer enough, * let's reduce it to half */ Hmm? Removing half of the volume slider? Second, assuming we actually buy this decision, wouldn't it make sense for people to be able to combine Master and Speaker to get the full -96 to 0 dB range? Instead, "Master" is just working over the range defined by "Speaker". Third, there is an interesting condition in hda_codec.c:update_amp_value: if (val > 0) val += ofs; Now this is in practice turning all volume controls which have ofs (e g those being "reduced to half") to a minimum-is-mute error: I can set the amp value to either 0 (-96 dB) or the 65 - 127 range (~ -48 dB - 0 dB). So it isn't really mute, it's -96 dB signal, but I can't hear that anyway. So; I think I understand the problem, but I'm not sure of where I should try to patch it, to make the best solution. Finally, is there right to exist for both "virtual masters" and the "PCM" softvol talked about a while ago? If I were to decide, I'd personally like them both removed (the additional complexity isn't motivated IMO), but wouldn't it at least make sense to name the virtual master "PCM", to avoid the PCM softvol? -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

4 5

[alsa-devel] New sound device question
by Laurent Epinat 26 Dec '10

26 Dec '10

Hello All I'm a newbie in this group and the sound systems. I try to developpe a usb sound driver for a Alesis Multix 8 USB2.0 http://www.alesis.com/multimix8usb20 The output interface is allready fonctional, but the Input not :-( Because there is 2 end point for it. The first one is on Isochronous mode and the other is on Bulk mode I don't know how to declare the second endpoint in the quirks_table. Where can I get a good documentation or help about the alsa USB sound structure. The only informations I have about the devices, are dump from windows thanks -- Salutations __ /----------------\ |~~\______/====\__ | ( Laurent Epinat )---- \LE______======= )-+ \----------------/ ----|/-- | mailto:l.epinat@free.fr (O) http://lepinat.free.fr

3 3

[alsa-devel] [PATCH] ASoC: Avoid direct register cache access when setting defaults
by Mark Brown 24 Dec '10

24 Dec '10

Directly accessing the register cache means that we can't use anything except a flat register cache so use snd_soc_update_bits(). Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8523.c | 6 +++--- sound/soc/codecs/wm8741.c | 13 ++++++++----- sound/soc/codecs/wm8904.c | 41 ++++++++++++++++++++++++++--------------- sound/soc/codecs/wm8955.c | 27 +++++++++++++++++++-------- sound/soc/codecs/wm8962.c | 30 ++++++++++++++++++++---------- sound/soc/codecs/wm8978.c | 2 +- sound/soc/codecs/wm9090.c | 41 ++++++++++++++++++++++++----------------- 7 files changed, 101 insertions(+), 59 deletions(-) diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 5eb2f50..9b8603f 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -414,7 +414,6 @@ static int wm8523_resume(struct snd_soc_codec *codec) static int wm8523_probe(struct snd_soc_codec *codec) { struct wm8523_priv *wm8523 = snd_soc_codec_get_drvdata(codec); - u16 *reg_cache = codec->reg_cache; int ret, i; codec->hw_write = (hw_write_t)i2c_master_send; @@ -471,8 +470,9 @@ static int wm8523_probe(struct snd_soc_codec *codec) } /* Change some default settings - latch VU and enable ZC */ - reg_cache[WM8523_DAC_GAINR] |= WM8523_DACR_VU; - reg_cache[WM8523_DAC_CTRL3] |= WM8523_ZC; + snd_soc_update_bits(codec, WM8523_DAC_GAINR, + WM8523_DACR_VU, WM8523_DACR_VU); + snd_soc_update_bits(codec, WM8523_DAC_CTRL3, WM8523_ZC, WM8523_ZC); wm8523_set_bias_level(codec, SND_SOC_BIAS_STANDBY); diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 494f2d3..25af901 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -421,7 +421,6 @@ static int wm8741_resume(struct snd_soc_codec *codec) static int wm8741_probe(struct snd_soc_codec *codec) { struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); - u16 *reg_cache = codec->reg_cache; int ret = 0; ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8741->control_type); @@ -437,10 +436,14 @@ static int wm8741_probe(struct snd_soc_codec *codec) } /* Change some default settings - latch VU */ - reg_cache[WM8741_DACLLSB_ATTENUATION] |= WM8741_UPDATELL; - reg_cache[WM8741_DACLMSB_ATTENUATION] |= WM8741_UPDATELM; - reg_cache[WM8741_DACRLSB_ATTENUATION] |= WM8741_UPDATERL; - reg_cache[WM8741_DACRLSB_ATTENUATION] |= WM8741_UPDATERM; + snd_soc_update_bits(codec, WM8741_DACLLSB_ATTENUATION, + WM8741_UPDATELL, WM8741_UPDATELL); + snd_soc_update_bits(codec, WM8741_DACLMSB_ATTENUATION, + WM8741_UPDATELM, WM8741_UPDATELM); + snd_soc_update_bits(codec, WM8741_DACRLSB_ATTENUATION, + WM8741_UPDATERL, WM8741_UPDATERL); + snd_soc_update_bits(codec, WM8741_DACRLSB_ATTENUATION, + WM8741_UPDATERM, WM8741_UPDATERM); snd_soc_add_controls(codec, wm8741_snd_controls, ARRAY_SIZE(wm8741_snd_controls)); diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 9de44a4..86c1cb8 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2436,19 +2436,28 @@ static int wm8904_probe(struct snd_soc_codec *codec) } /* Change some default settings - latch VU and enable ZC */ - reg_cache[WM8904_ADC_DIGITAL_VOLUME_LEFT] |= WM8904_ADC_VU; - reg_cache[WM8904_ADC_DIGITAL_VOLUME_RIGHT] |= WM8904_ADC_VU; - reg_cache[WM8904_DAC_DIGITAL_VOLUME_LEFT] |= WM8904_DAC_VU; - reg_cache[WM8904_DAC_DIGITAL_VOLUME_RIGHT] |= WM8904_DAC_VU; - reg_cache[WM8904_ANALOGUE_OUT1_LEFT] |= WM8904_HPOUT_VU | - WM8904_HPOUTLZC; - reg_cache[WM8904_ANALOGUE_OUT1_RIGHT] |= WM8904_HPOUT_VU | - WM8904_HPOUTRZC; - reg_cache[WM8904_ANALOGUE_OUT2_LEFT] |= WM8904_LINEOUT_VU | - WM8904_LINEOUTLZC; - reg_cache[WM8904_ANALOGUE_OUT2_RIGHT] |= WM8904_LINEOUT_VU | - WM8904_LINEOUTRZC; - reg_cache[WM8904_CLOCK_RATES_0] &= ~WM8904_SR_MODE; + snd_soc_update_bits(codec, WM8904_ADC_DIGITAL_VOLUME_LEFT, + WM8904_ADC_VU, WM8904_ADC_VU); + snd_soc_update_bits(codec, WM8904_ADC_DIGITAL_VOLUME_RIGHT, + WM8904_ADC_VU, WM8904_ADC_VU); + snd_soc_update_bits(codec, WM8904_DAC_DIGITAL_VOLUME_LEFT, + WM8904_DAC_VU, WM8904_DAC_VU); + snd_soc_update_bits(codec, WM8904_DAC_DIGITAL_VOLUME_RIGHT, + WM8904_DAC_VU, WM8904_DAC_VU); + snd_soc_update_bits(codec, WM8904_ANALOGUE_OUT1_LEFT, + WM8904_HPOUT_VU | WM8904_HPOUTLZC, + WM8904_HPOUT_VU | WM8904_HPOUTLZC); + snd_soc_update_bits(codec, WM8904_ANALOGUE_OUT1_RIGHT, + WM8904_HPOUT_VU | WM8904_HPOUTRZC, + WM8904_HPOUT_VU | WM8904_HPOUTRZC); + snd_soc_update_bits(codec, WM8904_ANALOGUE_OUT2_LEFT, + WM8904_LINEOUT_VU | WM8904_LINEOUTLZC, + WM8904_LINEOUT_VU | WM8904_LINEOUTLZC); + snd_soc_update_bits(codec, WM8904_ANALOGUE_OUT2_RIGHT, + WM8904_LINEOUT_VU | WM8904_LINEOUTRZC, + WM8904_LINEOUT_VU | WM8904_LINEOUTRZC); + snd_soc_update_bits(codec, WM8904_CLOCK_RATES_0, + WM8904_SR_MODE, 0); /* Apply configuration from the platform data. */ if (wm8904->pdata) { @@ -2469,10 +2478,12 @@ static int wm8904_probe(struct snd_soc_codec *codec) /* Set Class W by default - this will be managed by the Class * G widget at runtime where bypass paths are available. */ - reg_cache[WM8904_CLASS_W_0] |= WM8904_CP_DYN_PWR; + snd_soc_update_bits(codec, WM8904_CLASS_W_0, + WM8904_CP_DYN_PWR, WM8904_CP_DYN_PWR); /* Use normal bias source */ - reg_cache[WM8904_BIAS_CONTROL_0] &= ~WM8904_POBCTRL; + snd_soc_update_bits(codec, WM8904_BIAS_CONTROL_0, + WM8904_POBCTRL, 0); wm8904_set_bias_level(codec, SND_SOC_BIAS_STANDBY); diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index 70e9516..34c0e69 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -934,16 +934,27 @@ static int wm8955_probe(struct snd_soc_codec *codec) } /* Change some default settings - latch VU and enable ZC */ - reg_cache[WM8955_LEFT_DAC_VOLUME] |= WM8955_LDVU; - reg_cache[WM8955_RIGHT_DAC_VOLUME] |= WM8955_RDVU; - reg_cache[WM8955_LOUT1_VOLUME] |= WM8955_LO1VU | WM8955_LO1ZC; - reg_cache[WM8955_ROUT1_VOLUME] |= WM8955_RO1VU | WM8955_RO1ZC; - reg_cache[WM8955_LOUT2_VOLUME] |= WM8955_LO2VU | WM8955_LO2ZC; - reg_cache[WM8955_ROUT2_VOLUME] |= WM8955_RO2VU | WM8955_RO2ZC; - reg_cache[WM8955_MONOOUT_VOLUME] |= WM8955_MOZC; + snd_soc_update_bits(codec, WM8955_LEFT_DAC_VOLUME, + WM8955_LDVU, WM8955_LDVU); + snd_soc_update_bits(codec, WM8955_RIGHT_DAC_VOLUME, + WM8955_RDVU, WM8955_RDVU); + snd_soc_update_bits(codec, WM8955_LOUT1_VOLUME, + WM8955_LO1VU | WM8955_LO1ZC, + WM8955_LO1VU | WM8955_LO1ZC); + snd_soc_update_bits(codec, WM8955_ROUT1_VOLUME, + WM8955_RO1VU | WM8955_RO1ZC, + WM8955_RO1VU | WM8955_RO1ZC); + snd_soc_update_bits(codec, WM8955_LOUT2_VOLUME, + WM8955_LO2VU | WM8955_LO2ZC, + WM8955_LO2VU | WM8955_LO2ZC); + snd_soc_update_bits(codec, WM8955_ROUT2_VOLUME, + WM8955_RO2VU | WM8955_RO2ZC, + WM8955_RO2VU | WM8955_RO2ZC); + snd_soc_update_bits(codec, WM8955_MONOOUT_VOLUME, + WM8955_MOZC, WM8955_MOZC); /* Also enable adaptive bass boost by default */ - reg_cache[WM8955_BASS_CONTROL] |= WM8955_BB; + snd_soc_update_bits(codec, WM8955_BASS_CONTROL, WM8955_BB, WM8955_BB); /* Set platform data values */ if (pdata) { diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 82826f5..a598ff2 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3822,16 +3822,26 @@ static int wm8962_probe(struct snd_soc_codec *codec) } /* Latch volume update bits */ - reg_cache[WM8962_LEFT_INPUT_VOLUME] |= WM8962_IN_VU; - reg_cache[WM8962_RIGHT_INPUT_VOLUME] |= WM8962_IN_VU; - reg_cache[WM8962_LEFT_ADC_VOLUME] |= WM8962_ADC_VU; - reg_cache[WM8962_RIGHT_ADC_VOLUME] |= WM8962_ADC_VU; - reg_cache[WM8962_LEFT_DAC_VOLUME] |= WM8962_DAC_VU; - reg_cache[WM8962_RIGHT_DAC_VOLUME] |= WM8962_DAC_VU; - reg_cache[WM8962_SPKOUTL_VOLUME] |= WM8962_SPKOUT_VU; - reg_cache[WM8962_SPKOUTR_VOLUME] |= WM8962_SPKOUT_VU; - reg_cache[WM8962_HPOUTL_VOLUME] |= WM8962_HPOUT_VU; - reg_cache[WM8962_HPOUTR_VOLUME] |= WM8962_HPOUT_VU; + snd_soc_update_bits(codec, WM8962_LEFT_INPUT_VOLUME, + WM8962_IN_VU, WM8962_IN_VU); + snd_soc_update_bits(codec, WM8962_RIGHT_INPUT_VOLUME, + WM8962_IN_VU, WM8962_IN_VU); + snd_soc_update_bits(codec, WM8962_LEFT_ADC_VOLUME, + WM8962_ADC_VU, WM8962_ADC_VU); + snd_soc_update_bits(codec, WM8962_RIGHT_ADC_VOLUME, + WM8962_ADC_VU, WM8962_ADC_VU); + snd_soc_update_bits(codec, WM8962_LEFT_DAC_VOLUME, + WM8962_DAC_VU, WM8962_DAC_VU); + snd_soc_update_bits(codec, WM8962_RIGHT_DAC_VOLUME, + WM8962_DAC_VU, WM8962_DAC_VU); + snd_soc_update_bits(codec, WM8962_SPKOUTL_VOLUME, + WM8962_SPKOUT_VU, WM8962_SPKOUT_VU); + snd_soc_update_bits(codec, WM8962_SPKOUTR_VOLUME, + WM8962_SPKOUT_VU, WM8962_SPKOUT_VU); + snd_soc_update_bits(codec, WM8962_HPOUTL_VOLUME, + WM8962_HPOUT_VU, WM8962_HPOUT_VU); + snd_soc_update_bits(codec, WM8962_HPOUTR_VOLUME, + WM8962_HPOUT_VU, WM8962_HPOUT_VU); wm8962_add_widgets(codec); diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 4bbc344..30fb48e 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -965,7 +965,7 @@ static int wm8978_probe(struct snd_soc_codec *codec) * written. */ for (i = 0; i < ARRAY_SIZE(update_reg); i++) - ((u16 *)codec->reg_cache)[update_reg[i]] |= 0x100; + snd_soc_update_bits(codec, update_reg[i], 0x100, 0x100); /* Reset the codec */ ret = snd_soc_write(codec, WM8978_RESET, 0); diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index a788c42..3c5d325 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -551,7 +551,6 @@ static int wm9090_set_bias_level(struct snd_soc_codec *codec, static int wm9090_probe(struct snd_soc_codec *codec) { struct wm9090_priv *wm9090 = snd_soc_codec_get_drvdata(codec); - u16 *reg_cache = codec->reg_cache; int ret; codec->control_data = wm9090->control_data; @@ -576,22 +575,30 @@ static int wm9090_probe(struct snd_soc_codec *codec) /* Configure some defaults; they will be written out when we * bring the bias up. */ - reg_cache[WM9090_IN1_LINE_INPUT_A_VOLUME] |= WM9090_IN1_VU - | WM9090_IN1A_ZC; - reg_cache[WM9090_IN1_LINE_INPUT_B_VOLUME] |= WM9090_IN1_VU - | WM9090_IN1B_ZC; - reg_cache[WM9090_IN2_LINE_INPUT_A_VOLUME] |= WM9090_IN2_VU - | WM9090_IN2A_ZC; - reg_cache[WM9090_IN2_LINE_INPUT_B_VOLUME] |= WM9090_IN2_VU - | WM9090_IN2B_ZC; - reg_cache[WM9090_SPEAKER_VOLUME_LEFT] |= - WM9090_SPKOUT_VU | WM9090_SPKOUTL_ZC; - reg_cache[WM9090_LEFT_OUTPUT_VOLUME] |= - WM9090_HPOUT1_VU | WM9090_HPOUT1L_ZC; - reg_cache[WM9090_RIGHT_OUTPUT_VOLUME] |= - WM9090_HPOUT1_VU | WM9090_HPOUT1R_ZC; - - reg_cache[WM9090_CLOCKING_1] |= WM9090_TOCLK_ENA; + snd_soc_update_bits(codec, WM9090_IN1_LINE_INPUT_A_VOLUME, + WM9090_IN1_VU | WM9090_IN1A_ZC, + WM9090_IN1_VU | WM9090_IN1A_ZC); + snd_soc_update_bits(codec, WM9090_IN1_LINE_INPUT_B_VOLUME, + WM9090_IN1_VU | WM9090_IN1B_ZC, + WM9090_IN1_VU | WM9090_IN1B_ZC); + snd_soc_update_bits(codec, WM9090_IN2_LINE_INPUT_A_VOLUME, + WM9090_IN2_VU | WM9090_IN2A_ZC, + WM9090_IN2_VU | WM9090_IN2A_ZC); + snd_soc_update_bits(codec, WM9090_IN2_LINE_INPUT_B_VOLUME, + WM9090_IN2_VU | WM9090_IN2B_ZC, + WM9090_IN2_VU | WM9090_IN2B_ZC); + snd_soc_update_bits(codec, WM9090_SPEAKER_VOLUME_LEFT, + WM9090_SPKOUT_VU | WM9090_SPKOUTL_ZC, + WM9090_SPKOUT_VU | WM9090_SPKOUTL_ZC); + snd_soc_update_bits(codec, WM9090_LEFT_OUTPUT_VOLUME, + WM9090_HPOUT1_VU | WM9090_HPOUT1L_ZC, + WM9090_HPOUT1_VU | WM9090_HPOUT1L_ZC); + snd_soc_update_bits(codec, WM9090_RIGHT_OUTPUT_VOLUME, + WM9090_HPOUT1_VU | WM9090_HPOUT1R_ZC, + WM9090_HPOUT1_VU | WM9090_HPOUT1R_ZC); + + snd_soc_update_bits(codec, WM9090_CLOCKING_1, + WM9090_TOCLK_ENA, WM9090_TOCLK_ENA); wm9090_set_bias_level(codec, SND_SOC_BIAS_STANDBY); -- 1.7.2.3

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[alsa-devel] [PATCH 0/9] ASoC: codecs: Fix register cache incoherencies
by Lars-Peter Clausen 24 Dec '10

24 Dec '10

The multi-component patch(commit f0fba2ad1) introduced a generic register cache framework. Most of the drivers where converted to use this new framework. But there are also some drivers now which use a mixture of their own register cache and the generic register cache. Thus these end up with two from each other incoherent register caches, which can lead to undefined behaviour. This patch series tries to fix these drivers by converting them to always use the generic register cache framework instead of their own. There are also quite a few drivers left, which do not use their own register cache, but still have a field for it in their private device structure. The first patch of this series addresses those drivers and removes the unused register cache arrays from their private device structs. Please note, these patches, with the exception of wm8753, were only compile tested. - Lars Lars-Peter Clausen (9): ASoC: codecs: Remove unused reg_cache fields from device structs ASoC: codecs: max98088: Fix register cache incoherency ASoC: codecs: wm8523: Fix register cache incoherency ASoC: codecs: wm8741: Fix register cache incoherency ASoC: codecs: wm8904: Fix register cache incoherency ASoC: codecs: wm8955: Fix register cache incoherency ASoC: codecs: wm8962: Fix register cache incoherency ASoC: codecs: wm9090: Fix register cache incoherency ASoC: codecs: wm8753: Fix register cache incoherency sound/soc/codecs/88pm860x-codec.c | 1 - sound/soc/codecs/ad193x.c | 1 - sound/soc/codecs/ak4671.c | 1 - sound/soc/codecs/cs4270.c | 1 - sound/soc/codecs/cs42l51.c | 1 - sound/soc/codecs/cx20442.c | 1 - sound/soc/codecs/max98088.c | 13 +- sound/soc/codecs/tlv320aic26.c | 4 +- sound/soc/codecs/uda1380.c | 1 - sound/soc/codecs/wm8523.c | 9 +- sound/soc/codecs/wm8580.c | 1 - sound/soc/codecs/wm8711.c | 1 - sound/soc/codecs/wm8731.c | 1 - sound/soc/codecs/wm8741.c | 10 +- sound/soc/codecs/wm8750.c | 1 - sound/soc/codecs/wm8753.c | 267 ++++++++++++++----------------------- sound/soc/codecs/wm8900.c | 1 - sound/soc/codecs/wm8903.c | 2 - sound/soc/codecs/wm8904.c | 37 +++--- sound/soc/codecs/wm8940.c | 1 - sound/soc/codecs/wm8955.c | 30 ++-- sound/soc/codecs/wm8960.c | 1 - sound/soc/codecs/wm8961.c | 1 - sound/soc/codecs/wm8962.c | 45 +++---- sound/soc/codecs/wm8974.c | 1 - sound/soc/codecs/wm8978.c | 1 - sound/soc/codecs/wm8988.c | 1 - sound/soc/codecs/wm8993.c | 1 - sound/soc/codecs/wm9081.c | 1 - sound/soc/codecs/wm9090.c | 18 ++-- 30 files changed, 179 insertions(+), 276 deletions(-) -- 1.7.2.3

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[alsa-devel] asoc SND_SOC_DAPM_AIF_IN question
by Koul, Vinod 24 Dec '10

24 Dec '10

Hi, In the codec driver I am writing I added 2 paths in map, one for Headset and second for Speaker. Both of these paths have common PCM port enable button so I added that using SND_SOC_DAPM_AIF_IN So my codec map looks like: /*headset map*/ { "HSDAC Left", NULL, "PCM_In"}, { "HSDAC Right", NULL, "PCM_In"}, { "PCM_In", NULL, "Headset Rail"}, { "Headset Rail", NULL, "Audio Rail"}, /*speaker map*/ { "IHFDAC Left", NULL, "PCM_In"}, { "IHFDAC Right", NULL, "PCM_In"}, { "PCM_In", NULL, "Speaker Rail"}, { "Speaker Rail", NULL, "Audio Rail"}, The debugfs tell me this seems right [root@localhost dapm]# cat PCM_In PCM_In: Off in 0 out 5 - R636(0x27c) bit 0 stream HeadSet Speaker inactive in static Speaker Rail in static Headset Rail out static IHFDAC Right out static IHFDAC Left out static HSDAC Right out static HSDAC Left But when the stream is started now all the widgets get powered on. The debugfs entry says [root@localhost dapm]# cat PCM_In PCM_In: On in 1 out 5 - R636(0x27c) bit 0 stream HeadSet Speaker active in static Speaker Rail in static Headset Rail out static IHFDAC Right out static IHFDAC Left out static HSDAC Right out static HSDAC Left How is this detecting input as 1 (is this due to AIF_IN and thus turning on all the widgets in map? (turns on IHFDAC for HeadSet stream as well :( What could I be doing wrong here? I want to add this PCM button for all streams (4 eventually) and this be enabled whenever anyone of them is started. ~Vinod

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[alsa-devel] [PATCH] ASoC: SAMSUNG: Cleanup dependency of ASOC_SAMSUNG
by Kukjin Kim 24 Dec '10

24 Dec '10

This patch adds cleanup dependency of ASOC_SAMSUNG in each Samsung's audio config. Cc: Jassi Brar <jassi.brar(a)samsung.com> Signed-off-by: Kukjin Kim <kgene.kim(a)samsung.com> --- sound/soc/samsung/Kconfig | 46 ++++++++++++++++++++++++-------------------- 1 files changed, 25 insertions(+), 21 deletions(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 11ff4ad..91931e1 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -34,9 +34,16 @@ config SND_SAMSUNG_SPDIF config SND_SAMSUNG_I2S tristate +config ASOC_SAMSUNG_SIMTEC + tristate + help + Internal node for common S3C24XX/Simtec suppor + +if ASOC_SAMSUNG + config ASOC_SAMSUNG_NEO1973_WM8753 tristate "SoC I2S Audio support for NEO1973 - WM8753" - depends on ASOC_SAMSUNG && MACH_NEO1973_GTA01 + depends on MACH_NEO1973_GTA01 select SND_S3C24XX_I2S select SND_SOC_WM8753 help @@ -45,7 +52,7 @@ config ASOC_SAMSUNG_NEO1973_WM8753 config ASOC_SAMSUNG_NEO1973_GTA02_WM8753 tristate "Audio support for the Openmoko Neo FreeRunner (GTA02)" - depends on ASOC_SAMSUNG && MACH_NEO1973_GTA02 + depends on MACH_NEO1973_GTA02 select SND_S3C24XX_I2S select SND_SOC_WM8753 help @@ -54,7 +61,7 @@ config ASOC_SAMSUNG_NEO1973_GTA02_WM8753 config ASOC_SAMSUNG_JIVE_WM8750 tristate "SoC I2S Audio support for Jive" - depends on ASOC_SAMSUNG && MACH_JIVE + depends on MACH_JIVE select SND_SOC_WM8750 select SND_S3C2412_SOC_I2S help @@ -62,7 +69,7 @@ config ASOC_SAMSUNG_JIVE_WM8750 config ASOC_SAMSUNG_SMDK_WM8580 tristate "SoC I2S Audio support for WM8580 on SMDK" - depends on ASOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDK6440 || MACH_SMDK6450 || MACH_SMDK6442 || MACH_SMDKV210 || MACH_SMDKC110) + depends on (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDK6440 || MACH_SMDK6450 || MACH_SMDK6442 || MACH_SMDKV210 || MACH_SMDKC110) select SND_SOC_WM8580 select SND_SAMSUNG_I2S help @@ -70,7 +77,7 @@ config ASOC_SAMSUNG_SMDK_WM8580 config ASOC_SAMSUNG_SMDK_WM8994 tristate "SoC I2S Audio support for WM8994 on SMDK" - depends on ASOC_SAMSUNG && (MACH_SMDKV310 || MACH_SMDKC210) + depends on (MACH_SMDKV310 || MACH_SMDKC210) select SND_SOC_WM8994 select SND_SAMSUNG_I2S help @@ -78,7 +85,7 @@ config ASOC_SAMSUNG_SMDK_WM8994 config ASOC_SAMSUNG_SMDK2443_WM9710 tristate "SoC AC97 Audio support for SMDK2443 - WM9710" - depends on ASOC_SAMSUNG && MACH_SMDK2443 + depends on MACH_SMDK2443 select S3C2410_DMA select AC97_BUS select SND_SOC_AC97_CODEC @@ -89,7 +96,7 @@ config ASOC_SAMSUNG_SMDK2443_WM9710 config ASOC_SAMSUNG_LN2440SBC_ALC650 tristate "SoC AC97 Audio support for LN2440SBC - ALC650" - depends on ASOC_SAMSUNG && ARCH_S3C2410 + depends on ARCH_S3C2410 select S3C2410_DMA select AC97_BUS select SND_SOC_AC97_CODEC @@ -100,33 +107,28 @@ config ASOC_SAMSUNG_LN2440SBC_ALC650 config ASOC_SAMSUNG_S3C24XX_UDA134X tristate "SoC I2S Audio support UDA134X wired to a S3C24XX" - depends on ASOC_SAMSUNG && ARCH_S3C2410 + depends on ARCH_S3C2410 select SND_S3C24XX_I2S select SND_SOC_L3 select SND_SOC_UDA134X -config ASOC_SAMSUNG_SIMTEC - tristate - help - Internal node for common S3C24XX/Simtec suppor - config ASOC_SAMSUNG_SIMTEC_TLV320AIC23 tristate "SoC I2S Audio support for TLV320AIC23 on Simtec boards" - depends on ASOC_SAMSUNG && ARCH_S3C2410 + depends on ARCH_S3C2410 select SND_S3C24XX_I2S select SND_SOC_TLV320AIC23 select ASOC_SAMSUNG_SIMTEC config ASOC_SAMSUNG_SIMTEC_HERMES tristate "SoC I2S Audio support for Simtec Hermes board" - depends on ASOC_SAMSUNG && ARCH_S3C2410 + depends on ARCH_S3C2410 select SND_S3C24XX_I2S select SND_SOC_TLV320AIC3X select ASOC_SAMSUNG_SIMTEC config ASOC_SAMSUNG_H1940_UDA1380 tristate "Audio support for the HP iPAQ H1940" - depends on ASOC_SAMSUNG && ARCH_H1940 + depends on ARCH_H1940 select SND_S3C24XX_I2S select SND_SOC_UDA1380 help @@ -134,7 +136,7 @@ config ASOC_SAMSUNG_H1940_UDA1380 config ASOC_SAMSUNG_RX1950_UDA1380 tristate "Audio support for the HP iPAQ RX1950" - depends on ASOC_SAMSUNG && MACH_RX1950 + depends on MACH_RX1950 select SND_S3C24XX_I2S select SND_SOC_UDA1380 help @@ -142,7 +144,7 @@ config ASOC_SAMSUNG_RX1950_UDA1380 config ASOC_SAMSUNG_SMDK_WM9713 tristate "SoC AC97 Audio support for SMDK with WM9713" - depends on ASOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110 || MACH_SMDKV310 || MACH_SMDKC210) + depends on (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110 || MACH_SMDKV310 || MACH_SMDKC210) select SND_SOC_WM9713 select SND_SAMSUNG_AC97 help @@ -150,13 +152,13 @@ config ASOC_SAMSUNG_SMDK_WM9713 config ASOC_SMARTQ tristate "SoC I2S Audio support for SmartQ board" - depends on ASOC_SAMSUNG && MACH_SMARTQ + depends on MACH_SMARTQ select SND_SAMSUNG_I2S select SND_SOC_WM8750 config ASOC_GONI_AQUILA_WM8994 tristate "SoC I2S Audio support for AQUILA/GONI - WM8994" - depends on ASOC_SAMSUNG && (MACH_GONI || MACH_AQUILA) + depends on (MACH_GONI || MACH_AQUILA) select SND_SAMSUNG_I2S select SND_SOC_WM8994 help @@ -165,7 +167,9 @@ config ASOC_GONI_AQUILA_WM8994 config ASOC_SAMSUNG_SMDK_SPDIF tristate "SoC S/PDIF Audio support for SMDK" - depends on ASOC_SAMSUNG && (MACH_SMDKC100 || MACH_SMDKC110 || MACH_SMDKV210) + depends on (MACH_SMDKC100 || MACH_SMDKC110 || MACH_SMDKV210) select SND_SAMSUNG_SPDIF help Say Y if you want to add support for SoC S/PDIF audio on the SMDK. + +endif -- 1.6.2.5

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[alsa-devel] ALSA alsaSink and NONINTERLEAVED vs INTERLEAVED
by Radivoje Jovanovic 24 Dec '10

24 Dec '10

Hello, I have a simple sound card with a stereo output. The driver works just fine when using aplay or gstreamer to play wave files. The problem occurs when I am trying to play videos through gstreamer. I use following gstremaer pipeline (just to test sound): gst-launch-0.10 filesrc location=MPEG4_MOVIE_NAME ! qtdemux name=t ! queue ! faad ! alsasink If I make my driver with following settings: .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME), gstreamer comes back with a following message: Pipeline is PREROLLING ... ERROR: from element /GstPipeline:pipeline0/GstAlsaSink:alsasink0: Could not get/set settings from/on resource. Additional debug info: gstalsasink.c(519): set_hwparams (): /GstPipeline:pipeline0/GstAlsaSink:alsasink0: Unable to set hw params for playback: Invalid argument ERROR: pipeline doesn't want to preroll. Setting pipeline to NULL ... If I make my driver with following settings: .info = (SNDRV_PCM_INFO_NONINTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME), gstreamer will play sound. Of course sound will not be correct since the data is interleaved. Any idea why is this happening? Here is the rest of the snd_pcm_hardware struct: .formats = SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_U8, .rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_CONTINUOUS, .rate_min = 8000, .rate_max = 48000, .channels_min = 1, .channels_max = 2, .period_bytes_min = 16384, .period_bytes_max = 16384*4, .periods_min = 4, .periods_max = 256, .buffer_bytes_max= 131072, Cheers Ogi

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