October 2009 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 06/34] don't use __devexit_p to wrap hal2_remove
by Uwe Kleine-König 02 Oct '09

02 Oct '09

The function hal2_remove is defined using __exit, so don't use __devexit_p but __exit_p to wrap it. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org --- sound/mips/hal2.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/mips/hal2.c b/sound/mips/hal2.c index c52691c..72523a7 100644 --- a/sound/mips/hal2.c +++ b/sound/mips/hal2.c @@ -926,7 +926,7 @@ static int __exit hal2_remove(struct platform_device *pdev) static struct platform_driver hal2_driver = { .probe = hal2_probe, - .remove = __devexit_p(hal2_remove), + .remove = __exit_p(hal2_remove), .driver = { .name = "sgihal2", .owner = THIS_MODULE, -- 1.6.4.3

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[alsa-devel] [PATCH] Don't check if period size is equal to buffer size
by Lubomir Rintel 02 Oct '09

02 Oct '09

At least some drivers seem to allow a single period and those which don't would correct us anyway. Moreover, Pulseaudio with tsched sets this frequently. --- aplay/aplay.c | 5 ----- 1 files changed, 0 insertions(+), 5 deletions(-) diff --git a/aplay/aplay.c b/aplay/aplay.c index c7c82a1..4fa36cb 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -1044,11 +1044,6 @@ static void set_params(void) } snd_pcm_hw_params_get_period_size(params, &chunk_size, 0); snd_pcm_hw_params_get_buffer_size(params, &buffer_size); - if (chunk_size == buffer_size) { - error(_("Can't use period equal to buffer size (%lu == %lu)"), - chunk_size, buffer_size); - exit(EXIT_FAILURE); - } snd_pcm_sw_params_current(handle, swparams); if (avail_min < 0) n = chunk_size; -- 1.6.2.5

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[alsa-devel] [PATCH] sound: Make keywest_driver static
by Jean Delvare 02 Oct '09

02 Oct '09

From: Jean Delvare <khali(a)linux-fr.org> Subject: sound: Make keywest_driver static I can't see any reason for struct i2c_driver keywest_driver to not be static. Signed-off-by: Jean Delvare <khali(a)linux-fr.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> --- sound/ppc/keywest.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- linux-2.6.32-rc1.orig/sound/ppc/keywest.c 2009-10-01 09:45:18.000000000 +0200 +++ linux-2.6.32-rc1/sound/ppc/keywest.c 2009-10-01 18:05:06.000000000 +0200 @@ -93,7 +93,7 @@ static const struct i2c_device_id keywes { } }; -struct i2c_driver keywest_driver = { +static struct i2c_driver keywest_driver = { .driver = { .name = "PMac Keywest Audio", }, -- Jean Delvare

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[alsa-devel] [PATCH] sscape: remove MIDI instances counting with limit ULONG_MAX
by Krzysztof Helt 02 Oct '09

02 Oct '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> There is no sense to limit open MIDI connections with limit as high as ULONG_MAX. Also, remove redundant "sscape:" prefix in messages printed by snd_printk and convert more messages to use the snd_printk. Correct few old and misleading comments as well. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- sound/isa/sscape.c | 85 +++++++++++++--------------------------------------- 1 files changed, 21 insertions(+), 64 deletions(-) diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 1ce465c..ece2bb9 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -149,9 +149,8 @@ struct soundscape { /* * The MIDI device won't work until we've loaded - * its firmware via a hardware-dependent device IOCTL + * its firmware. */ - spinlock_t fwlock; unsigned long midi_usage; unsigned char midi_vol; }; @@ -164,11 +163,6 @@ static inline struct soundscape *get_card_soundscape(struct snd_card *c) return (struct soundscape *) (c->private_data); } -static inline struct soundscape *get_mpu401_soundscape(struct snd_mpu401 * mpu) -{ - return (struct soundscape *) (mpu->private_data); -} - /* * Allocates some kernel memory that we can use for DMA. * I think this means that the memory has to map to @@ -179,7 +173,9 @@ static struct snd_dma_buffer *get_dmabuf(struct snd_dma_buffer *buf, unsigned lo if (buf) { if (snd_dma_alloc_pages_fallback(SNDRV_DMA_TYPE_DEV, snd_dma_isa_data(), size, buf) < 0) { - snd_printk(KERN_ERR "sscape: Failed to allocate %lu bytes for DMA\n", size); + snd_printk(KERN_ERR + "Failed to allocate %lu bytes for DMA\n", + size); return NULL; } } @@ -797,37 +793,13 @@ static int mpu401_open(struct snd_mpu401 * mpu) int err; if (!verify_mpu401(mpu)) { - snd_printk(KERN_ERR "sscape: MIDI disabled, please load firmware\n"); + snd_printk(KERN_ERR "MIDI disabled, please load firmware\n"); err = -ENODEV; - } else { - register struct soundscape *sscape = get_mpu401_soundscape(mpu); - unsigned long flags; - - spin_lock_irqsave(&sscape->fwlock, flags); - - if (sscape->midi_usage == ULONG_MAX) { - err = -EBUSY; - } else { - ++(sscape->midi_usage); - err = 0; - } - - spin_unlock_irqrestore(&sscape->fwlock, flags); } return err; } -static void mpu401_close(struct snd_mpu401 * mpu) -{ - register struct soundscape *sscape = get_mpu401_soundscape(mpu); - unsigned long flags; - - spin_lock_irqsave(&sscape->fwlock, flags); - --(sscape->midi_usage); - spin_unlock_irqrestore(&sscape->fwlock, flags); -} - /* * Initialse an MPU-401 subdevice for MIDI support on the SoundScape. */ @@ -845,8 +817,6 @@ static int __devinit create_mpu401(struct snd_card *card, int devnum, unsigned l struct snd_mpu401 *mpu = (struct snd_mpu401 *) rawmidi->private_data; mpu->open_input = mpu401_open; mpu->open_output = mpu401_open; - mpu->close_input = mpu401_close; - mpu->close_output = mpu401_close; mpu->private_data = sscape; sscape->mpu = mpu; @@ -908,21 +878,20 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, err = snd_wss_pcm(chip, 0, &pcm); if (err < 0) { - snd_printk(KERN_ERR "sscape: No PCM device " - "for AD1845 chip\n"); + snd_printk(KERN_ERR "No PCM device for AD1845 chip\n"); goto _error; } err = snd_wss_mixer(chip); if (err < 0) { - snd_printk(KERN_ERR "sscape: No mixer device " + snd_printk(KERN_ERR "No mixer device " "for AD1845 chip\n"); goto _error; } if (chip->hardware != WSS_HW_AD1848) { err = snd_wss_timer(chip, 0, NULL); if (err < 0) { - snd_printk(KERN_ERR "sscape: No timer device " + snd_printk(KERN_ERR "No timer device " "for AD1845 chip\n"); goto _error; } @@ -932,7 +901,7 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, err = snd_ctl_add(card, snd_ctl_new1(&midi_mixer_ctl, chip)); if (err < 0) { - snd_printk(KERN_ERR "sscape: Could not create " + snd_printk(KERN_ERR "Could not create " "MIDI mixer control\n"); goto _error; } @@ -968,15 +937,14 @@ static int __devinit create_sscape(int dev, struct snd_card *card) */ io_res = request_region(port[dev], 8, "SoundScape"); if (!io_res) { - snd_printk(KERN_ERR - "sscape: can't grab port 0x%lx\n", port[dev]); + snd_printk(KERN_ERR "can't grab port 0x%lx\n", port[dev]); return -EBUSY; } wss_res = NULL; if (sscape->type == SSCAPE_VIVO) { wss_res = request_region(wss_port[dev], 4, "SoundScape"); if (!wss_res) { - snd_printk(KERN_ERR "sscape: can't grab port 0x%lx\n", + snd_printk(KERN_ERR "can't grab port 0x%lx\n", wss_port[dev]); err = -EBUSY; goto _release_region; @@ -988,18 +956,18 @@ static int __devinit create_sscape(int dev, struct snd_card *card) */ err = request_dma(dma[dev], "SoundScape"); if (err < 0) { - snd_printk(KERN_ERR "sscape: can't grab DMA %d\n", dma[dev]); + snd_printk(KERN_ERR "can't grab DMA %d\n", dma[dev]); goto _release_region; } spin_lock_init(&sscape->lock); - spin_lock_init(&sscape->fwlock); sscape->io_res = io_res; sscape->wss_res = wss_res; sscape->io_base = port[dev]; if (!detect_sscape(sscape, wss_port[dev])) { - printk(KERN_ERR "sscape: hardware not detected at 0x%x\n", sscape->io_base); + printk(KERN_ERR "sscape: hardware not detected at 0x%x\n", + sscape->io_base); err = -ENODEV; goto _release_dma; } @@ -1030,13 +998,13 @@ static int __devinit create_sscape(int dev, struct snd_card *card) */ irq_cfg = get_irq_config(sscape->type, irq[dev]); if (irq_cfg == INVALID_IRQ) { - snd_printk(KERN_ERR "sscape: Invalid IRQ %d\n", irq[dev]); + snd_printk(KERN_ERR "Invalid IRQ %d\n", irq[dev]); return -ENXIO; } mpu_irq_cfg = get_irq_config(sscape->type, mpu_irq[dev]); if (mpu_irq_cfg == INVALID_IRQ) { - printk(KERN_ERR "sscape: Invalid IRQ %d\n", mpu_irq[dev]); + snd_printk(KERN_ERR "Invalid IRQ %d\n", mpu_irq[dev]); return -ENXIO; } @@ -1073,7 +1041,7 @@ static int __devinit create_sscape(int dev, struct snd_card *card) err = create_ad1845(card, wss_port[dev], irq[dev], dma[dev], dma2[dev]); if (err < 0) { - printk(KERN_ERR "sscape: No AD1845 device at 0x%lx, IRQ %d\n", + snd_printk(KERN_ERR "No AD1845 device at 0x%lx, IRQ %d\n", wss_port[dev], irq[dev]); goto _release_dma; } @@ -1094,7 +1062,7 @@ static int __devinit create_sscape(int dev, struct snd_card *card) err = create_mpu401(card, MIDI_DEVNUM, port[dev], mpu_irq[dev]); if (err < 0) { - printk(KERN_ERR "sscape: Failed to create " + snd_printk(KERN_ERR "Failed to create " "MPU-401 device at 0x%lx\n", port[dev]); goto _release_dma; @@ -1191,7 +1159,7 @@ static int __devinit snd_sscape_probe(struct device *pdev, unsigned int dev) goto _release_card; if ((ret = snd_card_register(card)) < 0) { - printk(KERN_ERR "sscape: Failed to register sound card\n"); + snd_printk(KERN_ERR "Failed to register sound card\n"); goto _release_card; } dev_set_drvdata(pdev, card); @@ -1250,18 +1218,7 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, * We have found a candidate ISA PnP card. Now we * have to check that it has the devices that we * expect it to have. - * - * We will NOT try and autoconfigure all of the resources - * needed and then activate the card as we are assuming that - * has already been done at boot-time using /proc/isapnp. - * We shall simply try to give each active card the resources - * that it wants. This is a sensible strategy for a modular - * system where unused modules are unloaded regularly. - * - * This strategy is utterly useless if we compile the driver - * into the kernel, of course. */ - // printk(KERN_INFO "sscape: %s\n", card->name); /* * Check that we still have room for another sound card ... @@ -1272,7 +1229,7 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, if (!pnp_is_active(dev)) { if (pnp_activate_dev(dev) < 0) { - printk(KERN_INFO "sscape: device is inactive\n"); + snd_printk(KERN_INFO "device is inactive\n"); return -EBUSY; } } @@ -1317,7 +1274,7 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, goto _release_card; if ((ret = snd_card_register(card)) < 0) { - printk(KERN_ERR "sscape: Failed to register sound card\n"); + snd_printk(KERN_ERR "Failed to register sound card\n"); goto _release_card; } -- 1.6.0.3 --------------------------------------------------------------- 2012 rok wcale nie musi by� tragiczny! Kliknij >>> http://link.interia.pl/f238b

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[alsa-devel] Patches for alsactl init db
by Dan Chen 02 Oct '09

02 Oct '09

Hi, Attached are two patches against alsa-utils.git HEAD that create an intel8x0 init db file for ICH4. This is necessary from bug reports in Debian and Ubuntu that state that External Amplifier must be muted to have audible (analog) playback by default on the Sony VAIO VGN-T350P. Thanks, Dan

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[alsa-devel] Help! Playing multiple audio stream using alsa libraries
by Arjun Sambamoorthy 02 Oct '09

02 Oct '09

Hello all, I am trying to play two audio streams using alsa libraries at the same time, but I am not able to do so. I could play a single audio stream though. I get bi-directional G711 MU Law RTP streams and I am using the sample playback capture code from the tutorial section for playing it. I am using the Non blocking and interleaved mode for playing the streams. The alsa library plays both the audio streams for a few seconds but it is incomprehensible and after few seconds I get the error "write to audio interface failed (broken pipe)". I also tried opening two pcm handles for each stream to the same device ("default"), but the second handle is unusable gives the same error "write to audio interface failed (broken pipe)". I have the same problem playing two audio streams to the /dev/dsp device. I am pretty sure I am not doing it the right way. Any help is highly appreciated or pointer to links that explains how to play multiple streams using alsa. Thanks Arjunen

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[alsa-devel] [PATCH 1/3] ASoC: DaVinci: i2s, reduce underruns by combining into 1 element
by Troy Kisky 01 Oct '09

01 Oct '09

Allow the left and right 16 bit samples to be shifted out as 1 32 bit sample. Signed-off-by: Troy Kisky <troy.kisky(a)boundarydevices.com> -- This applies to Kevin's temp/asoc branch --- arch/arm/mach-davinci/include/mach/asp.h | 6 ++ sound/soc/davinci/davinci-i2s.c | 74 ++++++++++++++++++++++-------- 2 files changed, 61 insertions(+), 19 deletions(-) diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index 18e4ce3..a3d2aa1 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -51,6 +51,12 @@ struct snd_platform_data { u32 rx_dma_offset; enum dma_event_q eventq_no; /* event queue number */ unsigned int codec_fmt; + /* + * Allowing this is more efficient and eliminates left and right swaps + * caused by underruns, but will swap the left and right channels + * when compared to previous behavior. + */ + unsigned disable_channel_combine:1; /* McASP specific fields */ int tdm_slots; diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 12a6c54..081b2d4 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -97,6 +97,23 @@ enum { DAVINCI_MCBSP_WORD_32, }; +static const unsigned char data_type[SNDRV_PCM_FORMAT_S32_LE + 1] = { + [SNDRV_PCM_FORMAT_S8] = 1, + [SNDRV_PCM_FORMAT_S16_LE] = 2, + [SNDRV_PCM_FORMAT_S32_LE] = 4, +}; + +static const unsigned char asp_word_length[SNDRV_PCM_FORMAT_S32_LE + 1] = { + [SNDRV_PCM_FORMAT_S8] = DAVINCI_MCBSP_WORD_8, + [SNDRV_PCM_FORMAT_S16_LE] = DAVINCI_MCBSP_WORD_16, + [SNDRV_PCM_FORMAT_S32_LE] = DAVINCI_MCBSP_WORD_32, +}; + +static const unsigned char double_fmt[SNDRV_PCM_FORMAT_S32_LE + 1] = { + [SNDRV_PCM_FORMAT_S8] = SNDRV_PCM_FORMAT_S16_LE, + [SNDRV_PCM_FORMAT_S16_LE] = SNDRV_PCM_FORMAT_S32_LE, +}; + static struct davinci_pcm_dma_params davinci_i2s_pcm_out = { .name = "I2S PCM Stereo out", }; @@ -113,6 +130,27 @@ struct davinci_mcbsp_dev { u32 pcr; struct clk *clk; struct davinci_pcm_dma_params *dma_params[2]; + /* + * Combining both channels into 1 element will at least double the + * amount of time between servicing the dma channel, increase + * effiency, and reduce the chance of overrun/underrun. But, + * it will result in the left & right channels being swapped. + * + * If relabeling the left and right channels is not possible, + * you may want to let the codec know to swap them back. + * + * It may allow x10 the amount of time to service dma requests, + * if the codec is master and is using an unnecessarily fast bit clock + * (ie. tlvaic23b), independent of the sample rate. So, having an + * entire frame at once means it can be serviced at the sample rate + * instead of the bit clock rate. + * + * In the now unlikely case that an underrun still + * occurs, both the left and right samples will be repeated + * so that no pops are heard, and the left and right channels + * won't end up being swapped because of the underrun. + */ + unsigned disable_channel_combine:1; }; static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev, @@ -359,6 +397,8 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, int mcbsp_word_length; unsigned int rcr, xcr, srgr; u32 spcr; + snd_pcm_format_t fmt; + unsigned element_cnt = 1; /* general line settings */ spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); @@ -388,27 +428,22 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, xcr |= DAVINCI_MCBSP_XCR_XDATDLY(1); } /* Determine xfer data type */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: - dma_params->data_type = 1; - mcbsp_word_length = DAVINCI_MCBSP_WORD_8; - break; - case SNDRV_PCM_FORMAT_S16_LE: - dma_params->data_type = 2; - mcbsp_word_length = DAVINCI_MCBSP_WORD_16; - break; - case SNDRV_PCM_FORMAT_S32_LE: - dma_params->data_type = 4; - mcbsp_word_length = DAVINCI_MCBSP_WORD_32; - break; - default: + fmt = params_format(params); + if ((fmt > SNDRV_PCM_FORMAT_S32_LE) || !data_type[fmt]) { printk(KERN_WARNING "davinci-i2s: unsupported PCM format\n"); return -EINVAL; } - - dma_params->acnt = dma_params->data_type; - rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(1); - xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(1); + if (params_channels(params) == 2) { + element_cnt = 2; + if (double_fmt[fmt] && !dev->disable_channel_combine) { + element_cnt = 1; + fmt = double_fmt[fmt]; + } + } + dma_params->acnt = dma_params->data_type = data_type[fmt]; + mcbsp_word_length = asp_word_length[fmt]; + rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(element_cnt - 1); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(element_cnt - 1); rcr |= DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) | DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length); @@ -524,7 +559,8 @@ static int davinci_i2s_probe(struct platform_device *pdev) ret = -ENOMEM; goto err_release_region; } - + if (pdata) + dev->disable_channel_combine = pdata->disable_channel_combine; dev->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(dev->clk)) { ret = -ENODEV; -- 1.5.6.3

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[alsa-devel] A few fix patches for ASoC
by Takashi Iwai 01 Oct '09

01 Oct '09

Mark, I applied a few fix patches to fix/asoc and topic/asoc branches for build errors I found with cross-build tests in this morning. Check below and sync to your tree. commit bb26276744a80d066681836f4d49c70010b129d6 Author: Takashi Iwai <tiwai(a)suse.de> Date: Thu Oct 1 07:39:45 2009 +0200 ASoC: Fix build errors of wm8711.c with SPI commit c877c25170e2655d519b29e91d6c91d5d1a72a6f Author: Takashi Iwai <tiwai(a)suse.de> Date: Thu Oct 1 08:33:47 2009 +0200 ASoC: Fix dependency of CONFIG_SND_PXA2XX_SOC_IMOTE2 commit 140318aaa924ce9664ff59366993228cf1547f1d Author: Takashi Iwai <tiwai(a)suse.de> Date: Thu Oct 1 08:40:32 2009 +0200 ASoC: Fix snd_soc_dai_set_pll() calls in neo1973_*.c

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[alsa-devel] LEFT or RIGHT Channel in alsa mixer
by Navink Singh 01 Oct '09

01 Oct '09

Hi Developers, I need to configure my audio device such as to use only LEFT or RIGHT Channel. I have carried out following steps to proceed with 1) Opened PCM playback Device 2) Configured its hardware and Software parameters 3) Open mixer 4) Attached Mixer to playback device 5) Registered Mixer 6) Loaded the mixer 7) Get the first element of mixer A) Now when i use following API, snd_mixer_selem_is_playback_mono(aElem); i found mixer is using MONO channel. But i am configuring my PCM Device for 2 channels I am using snd_mixer_selem_set_playback_switch() API to set right or Left channel to MUTE {Does not work} If i set all channels to Mute {Works fine} snd_mixer_selem_set_playback_switch_all (aElem, 0) Also i am able to set SND_MIXER_SCHN_MONO to mute {works fine} snd_mixer_selem_set_playback_switch(aElem, SND_MIXER_SCHN_MONO, 0) B) There is also an API in my MIXER driver to set the right and left balance, but i did not find any API in Application to do so {Is there any such API in Application}. Thanks in Advance, Navin Kumar Singh Tata Consultancy Services DLF Akruti Info Park, PN 28, Rajiv Gandhi Infotech Park Hinjewadi, PHASE II Pune - 411057,Maharashtra India Ph:- +91 20 66752458 Cell:- +919860227357 Mailto: navink.singh(a)tcs.com Website: http://www.tcs.com ____________________________________________ Experience certainty. IT Services Business Solutions Outsourcing ____________________________________________ =====-----=====-----===== Notice: The information contained in this e-mail message and/or attachments to it may contain confidential or privileged information. If you are not the intended recipient, any dissemination, use, review, distribution, printing or copying of the information contained in this e-mail message and/or attachments to it are strictly prohibited. If you have received this communication in error, please notify us by reply e-mail or telephone and immediately and permanently delete the message and any attachments. Thank you

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[alsa-devel] ASoC updates for 2.6.33
by Mark Brown 01 Oct '09

01 Oct '09

The following changes since commit 140318aaa924ce9664ff59366993228cf1547f1d: Takashi Iwai (1): ASoC: Fix snd_soc_dai_set_pll() calls in neo1973_*.c are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.33 Barry Song (1): ASoC: fix kconfig order of Blackfin drivers Chaithrika U S (1): ASoC: DaVinci: McASP FIFO related updates Lopez Cruz, Misael (1): ASoC: Add PDM DAI format definition Mark Brown (6): ASoC: Clean up WM8974 PLL configuration Merge branch 'upstream/wm8974' into for-2.6.33 ASoC: Factor out analogue platform data from WM8993 Merge remote branch 'takashi/fix/asoc' into for-2.6.32 Merge remote branch 'takashi/topic/asoc' into for-2.6.33 Merge branch 'for-2.6.32' into for-2.6.33 Peter Ujfalusi (1): ASoC: add support for multiple cards/codecs in debugfs Takashi Iwai (1): ASoC: Fix dependency of CONFIG_SND_PXA2XX_SOC_IMOTE2 include/sound/soc-dai.h | 1 + include/sound/soc.h | 1 + sound/soc/blackfin/Kconfig | 98 ++++++++++++++++++------------------ sound/soc/codecs/wm8974.c | 23 +++++---- sound/soc/codecs/wm8993.c | 36 +++---------- sound/soc/codecs/wm_hubs.c | 35 +++++++++++++ sound/soc/codecs/wm_hubs.h | 5 ++ sound/soc/davinci/davinci-i2s.c | 2 + sound/soc/davinci/davinci-mcasp.c | 17 +++---- sound/soc/davinci/davinci-pcm.c | 21 +++++++- sound/soc/davinci/davinci-pcm.h | 1 + sound/soc/pxa/Kconfig | 2 +- sound/soc/soc-core.c | 26 +++++++--- 13 files changed, 161 insertions(+), 107 deletions(-)

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