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Re: [alsa-devel] [PATCH] ASoC: sta32x: Move resource allocation and release to the corresponding callback functions
by Johannes Stezenbach 18 Aug '11

18 Aug '11

On Thu, Aug 18, 2011 at 03:31:04PM +0800, Axel Lin wrote: > This patch includes below small fixes: > > 1. Move sta32x_set_bias_level() from sta32x_i2c_remove() to sta32x_remove(). > 2. Remove a redundant regulator_bulk_free() call in sta32x_i2c_remove(), > as we will call regulator_bulk_free() in sta32x_remove(). > 3. Remove unneeded snd_soc_codec_set_drvdata(codec, NULL) in sta32x_i2c_remove. > The i2c core will set the clientdata to NULL. > > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> > --- > hi, > I don't have this hardware handy. I appreciate if someone can test this patch. > > Axel Your patch looks good, but I also don't have the hardware currently. Maybe Sven or Daniel could test? But even though it's untested: Acked-by: Johannes Stezenbach <js(a)sig21.net> Thanks Johannes > sound/soc/codecs/sta32x.c | 13 ++----------- > 1 files changed, 2 insertions(+), 11 deletions(-) > > diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c > index 3d155f5..5c7def3 100644 > --- a/sound/soc/codecs/sta32x.c > +++ b/sound/soc/codecs/sta32x.c > @@ -812,6 +812,7 @@ static int sta32x_remove(struct snd_soc_codec *codec) > { > struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); > > + sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF); > regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); > regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); > > @@ -871,18 +872,8 @@ static __devinit int sta32x_i2c_probe(struct i2c_client *i2c, > static __devexit int sta32x_i2c_remove(struct i2c_client *client) > { > struct sta32x_priv *sta32x = i2c_get_clientdata(client); > - struct snd_soc_codec *codec = sta32x->codec; > - > - if (codec) > - sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF); > - > - regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); > - > - if (codec) { > - snd_soc_unregister_codec(&client->dev); > - snd_soc_codec_set_drvdata(codec, NULL); > - } > > + snd_soc_unregister_codec(&client->dev); > kfree(sta32x); > return 0; > } > -- > 1.7.4.1 > > > >

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Re: [alsa-devel] Correct way to set NONAUDIO flag
by Clemens Ladisch 17 Aug '11

17 Aug '11

Pierre-Louis Bossart wrote: > If you try to set the bits with ALSA controls in PulseAudio, somehow the > settings are overwritten when the iec958 device is opened and initialized > with default values. That's how the iec958 device is intended to work. The AES0..3 parameters of the iec958 device are the _only_ way to set these flags. If you don't set them, they get a static default value; see <http://git.alsa-project.org/?p=alsa-lib.git;a=blob;hb=HEAD;f=src/conf/pcm/i…>. If PA wants to set these flags, it must do so when opening the device; any other mechanism will not work. Regards, Clemens

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Re: [alsa-devel] Correct way to set NONAUDIO flag
by Arun Raghavan 17 Aug '11

17 Aug '11

On Wed, 2011-08-17 at 10:31 -0500, Pierre-Louis Bossart wrote: > > > > I'm trying to figure out the right way to set the NONAUDIO flag on > > an > > > > iec958 device. > > > > > > Use the device name "iec958:AES0=6". > > > > Thanks. As I understand it, this forces us to set all the flags instead > > of just the one that we're interested in. Is this the only way? > > I banged my head against the wall on this one for eAC3 support. For some > strange reason my receiver works fine with AC3 without touching the C bits, > but for eAC3 setting the NONAUDIO bit is required. > If you try to set the bits with ALSA controls in PulseAudio, somehow the > settings are overwritten when the iec958 device is opened and initialized > with default values. Clemens is right, setting the AES0=6 in the hdmi > profile proved to be the only solution. It'd be nice to find a solution in > which the iec958 device preserves the AES values set beforehand. That way we > could set the NONAUDIO bit only in passthrough mode. Well, given this is the best way to do it, my plan is to just close (suspend) and reopen (unsuspend) and tack on the the right flags to the device name when we're in passthrough mode. -- Arun

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[alsa-devel] Correct way to set NONAUDIO flag
by Arun Raghavan 17 Aug '11

17 Aug '11

Hello, I'm trying to figure out the right way to set the NONAUDIO flag on an iec958 device. The idea is that when we get a new stream in PulseAudio that signals a non-PCM format, we'd like to set the NONAUDIO flag to let receivers know that this isn't PCM data. I tried doing what iecset does in the PA alsa-sink code but this fails. As far as I can tell, this happens because: (a) if I set the flag before snd_pcm_open(), the AES0 value from iec958.conf gets used, and (b) if I set the flag after snd_pcm_open(), changing the value is no longer permitted. Any help/pointers would be appreciated. Thanks, Arun

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[PATCH 1/1] sgtl5000: fix record unwork issue
by Dong Aisheng 17 Aug '11

17 Aug '11

Signed-off-by: Dong Aisheng <b29396(a)freescale.com> --- sound/soc/codecs/sgtl5000.c | 32 ++++++++++++++++++-------------- 1 files changed, 18 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 5a0d8e4..ef52890 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -198,13 +198,13 @@ static int small_pop_event(struct snd_soc_dapm_widget= *w, { switch (event) { case SND_SOC_DAPM_PRE_PMU: - snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, - SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP); +// snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, +// SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP); break; =20 case SND_SOC_DAPM_PRE_PMD: - snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, - SGTL5000_VAG_POWERUP, 0); +// snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, +// SGTL5000_VAG_POWERUP, 0); msleep(400); break; default: @@ -1153,11 +1153,13 @@ static int sgtl5000_set_power_regs(struct snd_soc_c= odec *codec) SGTL5000_VDDC_CHRGPMP_POWERUP, 0); =20 /* VDDC use VDDIO rail */ - lreg_ctrl |=3D SGTL5000_VDDC_ASSN_OVRD; - lreg_ctrl |=3D SGTL5000_VDDC_MAN_ASSN_VDDIO << - SGTL5000_VDDC_MAN_ASSN_SHIFT; +// lreg_ctrl |=3D SGTL5000_VDDC_ASSN_OVRD; +// lreg_ctrl |=3D SGTL5000_VDDC_MAN_ASSN_VDDIO << +// SGTL5000_VDDC_MAN_ASSN_SHIFT; } =20 +// ana_pwr |=3D SGTL5000_VDDC_CHRGPMP_POWERUP | SGTL5000_VAG_POWERUP | SGT= L5000_HP_POWERUP; + ana_pwr |=3D 0x5afb; snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL, lreg_ctrl); =20 snd_soc_write(codec, SGTL5000_CHIP_ANA_POWER, ana_pwr); @@ -1194,9 +1196,9 @@ static int sgtl5000_set_power_regs(struct snd_soc_cod= ec *codec) else vag =3D (vag - SGTL5000_ANA_GND_BASE) / SGTL5000_ANA_GND_STP; =20 - snd_soc_update_bits(codec, SGTL5000_CHIP_REF_CTRL, - vag << SGTL5000_ANA_GND_SHIFT, - vag << SGTL5000_ANA_GND_SHIFT); +// snd_soc_update_bits(codec, SGTL5000_CHIP_REF_CTRL, +// vag << SGTL5000_ANA_GND_SHIFT, +/// vag << SGTL5000_ANA_GND_SHIFT); =20 /* set line out VAG to vddio / 2, in range (0.8v, 1.675v) */ vag =3D vddio / 2; @@ -1208,14 +1210,15 @@ static int sgtl5000_set_power_regs(struct snd_soc_c= odec *codec) else vag =3D (vag - SGTL5000_LINE_OUT_GND_BASE) / SGTL5000_LINE_OUT_GND_STP; - +#if 1 snd_soc_update_bits(codec, SGTL5000_CHIP_LINE_OUT_CTRL, - vag << SGTL5000_LINE_OUT_GND_SHIFT | +// vag << SGTL5000_LINE_OUT_GND_SHIFT | SGTL5000_LINE_OUT_CURRENT_360u << SGTL5000_LINE_OUT_CURRENT_SHIFT, - vag << SGTL5000_LINE_OUT_GND_SHIFT | +// vag << SGTL5000_LINE_OUT_GND_SHIFT | SGTL5000_LINE_OUT_CURRENT_360u << SGTL5000_LINE_OUT_CURRENT_SHIFT); +#endif =20 return 0; } @@ -1378,7 +1381,8 @@ static int sgtl5000_probe(struct snd_soc_codec *codec= ) =20 snd_soc_write(codec, SGTL5000_CHIP_ANA_CTRL, SGTL5000_HP_ZCD_EN | - SGTL5000_ADC_ZCD_EN); + SGTL5000_ADC_ZCD_EN=20 + | SGTL5000_LINE_OUT_MUTE); =20 snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 0); =20 --=20 1.7.0.4

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[alsa-devel] How to pair Wine with ALSA? part1: intro & underruns (long)
by Joerg-Cyril.Hoehle＠t-systems.com 17 Aug '11

17 Aug '11

Dear knowledgeable ALSA developers, Wine currently undergoes a rewrite of its audio systems. That's a welcome point in time to reflect on what it needs from ALSA and how it should use the ALSA API. What combination of hw & sw_params makes sense? How to feed ALSA with samples? I've identified 3 uses cases and several other topics to help thinking about the issues: UC1 Intermittent Mouse Clicks UC2 Video UC3 Background Music T1 underrun T2 buffer and period size/time T3 blocking or not T4 mmap Due to size, I'll cover T2-T4 in a later e-mail. UC1 Intermittent Mouse Clicks An app may use the winmm API as follows: 1. open the device 2. from time to time, e.g. on mouse clicks, send wave data. The requirements are: R1 Nothing (or silence) is to be played when the app submits no data. R2 Once the app submits data, it should play ASAP. That is the app's view on it. It seems to translate to silence_threshold=0 and silence_size=boundary, doesn't it? Alternatively, Wine's internal periodic audio worker could as well send silence samples when it receives nothing from the app. If so, I fear additional complexity with snd_pcm_rewind to meet R2, e.g. in case where the app submits samples shortly after wine decided to play some silence. UC2 Video I've not much to say here because I basically ignore how apps synchronize audio with video. GetPosition returns "the stream position of the sample that is currently playing through the speakers". This begs for snd_pcm_delay, not avail_update. An app may select a particular device for output (e.g. 5:1 may not be available with the default mixer). R3 Wine must offer access to several sound cards if available. R4 Try and find hw/sw_params that work with the "default" device as well as individual cards (hw:x or perhaps plughw:x). mmdevapi has a notion of exclusive and shared modes. I expect shared mode to be asked with the "default" device (for mixing to work), and exclusive with others (e.g. 5:1 sound). OTOH, apps may also ask for the default device in exclusive mode, which is said to be granted if no other app is playing sound (much like I can grab hw:0 only when dmix has nothing to do). UC3 Background Music The app ought to send a continuous stream of samples, perhaps mixing sources itself (e.g. what games let the dsound API do). R5 No samples are ever skipped. If late, play late. See R1/R2. I've observed that "dmix"+"pulse" and the "hw:x" devices approach underruns radically differently. In case of a 10s underrun, the former silently skip over the next 10s worth of samples(!) -- if the app does not submit those *fast* to catch up, you'll hear no more sound! -- while the latter immediately write them to their HW buffer. IOW, dmix and pulse streaming semantics do not match winmm nor mmdevapi's semantics. To counter that, I'm considering using: /* call avail_update prior to every write: */ if (snd_pcm_avail_update() > buffer_size) snd_pcm_reset() /* skip over late samples */ /* should be equivalent to snd_pcm_forward(avail) */ snd_pcm_write() However, there's still a short underrun between reset() and write(). Topic T1 underrun I argue that the MS APIs winmm and mmdevapi have almost no notion of underrun during playback. You derive it indirectly at best ("buffer is empty", "all headers returned", "speaker position = amount of samples sent"). R5 GetPosition only counts submitted samples. Silence played during underruns must not count and is not reported to the app. This seems unlike snd_pcm_delay. I've observed snd_pcm_delay grow towards minus infinity during underruns. How to nevertheless make use of the ALSA API to compute that? I'm thinking about err = snd_pcm_avail_delay(&avail,&delay) if (err<0) ...? if (avail > buffer_size) /* underrun => everything was played */ position = sum_of_submitted_samples; else position = some_function_of(delay, submitted_samples) What do you recommend? Thank you for your help and for reading up to this point, Jörg Höhle

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Re: [alsa-devel] snd soc spi read/write
by Scott Jiang 17 Aug '11

17 Aug '11

> > Why are you replying off-list? You should implement 16x8 SPI reads in > the core stuff I'd expect, then disable cache for the CODEC. > Sorry, I just forgot to CC. I wonder if rbtree compress type can support this kind of sparse register? And should the patch aginst 3.0 or 3.2?

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Re: [alsa-devel] [PATCH] ASoC: Remove unreachable code in au1xac97c_drvprobe and au1xi2s_drvprobe
by Mark Brown 17 Aug '11

17 Aug '11

On Wed, Aug 17, 2011 at 03:14:33PM +0800, Axel Lin wrote: > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> Applied, thanks.

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[alsa-devel] [PATCH] ASoC: Fix check for symmetric rate enforcement
by Sascha Hauer 17 Aug '11

17 Aug '11

The ASoC core tries to not enforce symmetric rates when two streams open simultaneously. It does so by checking rtd->rate being zero. This works exactly once after booting because it is not set to zero again when the streams close. Fix this by clearing rtd->rate when no active stream is left. Signed-off-by: Sascha Hauer <s.hauer(a)pengutronix.de> --- sound/soc/soc-core.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d75043e..c25649a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -746,6 +746,9 @@ static int soc_codec_close(struct snd_pcm_substream *substream) codec_dai->active--; codec->active--; + if (!cpu_dai->active && !codec_dai->active) + rtd->rate = 0; + /* Muting the DAC suppresses artifacts caused during digital * shutdown, for example from stopping clocks. */ -- 1.7.5.4 -- Pengutronix e.K. | | Industrial Linux Solutions | http://www.pengutronix.de/ | Peiner Str. 6-8, 31137 Hildesheim, Germany | Phone: +49-5121-206917-0 | Amtsgericht Hildesheim, HRA 2686 | Fax: +49-5121-206917-5555 |

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[alsa-devel] [PATCH] ASoC: fsl: fix build warning in fsl_dma
by Timur Tabi 17 Aug '11

17 Aug '11

The previous patch to fsl_dma.c ("fix initialization of DMA buffers") left behind an unused local variable that causes a build warning. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/fsl/fsl_dma.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 732208c..0efc04a 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -297,7 +297,6 @@ static irqreturn_t fsl_dma_isr(int irq, void *dev_id) static int fsl_dma_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; - struct snd_soc_dai *dai = rtd->cpu_dai; struct snd_pcm *pcm = rtd->pcm; static u64 fsl_dma_dmamask = DMA_BIT_MASK(36); int ret; -- 1.7.4.4

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Re: [alsa-devel] USB Audio questions
by Torsten Schenk 17 Aug '11

17 Aug '11

Hi Pierre, I also spent some time learning about USB and audio while developing the sound/usb/6fire driver. I made the following discoveries: >What I had in mind is a case where the ring buffer is divided in separate >periods, which would each correspond to an integer number of URBs/packets. >This way there is _never_ a case of wrap-around and no need for memcpys. >It's probably less generic than the current driver but a whole lot >simpler/efficient. I was also not very happy to implement the double buffering with memcpy's. But from the experience with the 6fire device I can say this: - The device/driver sends per urb a constant number of packets (as you already also discovered). The number of samples per packet differ however (+- 1 sample per packet). I accuse the asynchronousity between the urb interval and the sound card's internal clock for this fact. The device expects the driver to send back an urb with exactly the same structure. The device produced crackling noise if the samples per packet were not equal. I cannot say to what extend this is true (i.e. if you could wait some packets until catching up the difference without producing crackles) but due to this fact it's at least not a simple task to send urbs exactly corresponding to the period limit. - Another thing came into my mind while reading your post, but I'm not sure if that would apply in praxis: If sending an urb is scheduled with the alsa buffer, it might be possible that the buffer is overwritten before the urb is sent, leading to complete audio-rubbish. - And: at least the 6fire requires a header for each packet, which couldn't be placed directly in the alsa buffer. But perhaps this doesn't apply to most cards. Greets, Torsten

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[alsa-devel] WM8727 & AT91SAM9G45: not enough memory
by conn intel 17 Aug '11

17 Aug '11

Dear Friends, I am using WM8727 codec with AT91SAM9G45 processor. I have implemented machine code as http://pastebin.com/axadqGhd . After insterting the module I got following output: # insmod sam9g45_wm8727.ko atmel-pcm:preallocate_dma_buffer: area=ffadf000, addr=77c40000, size=32768 asoc: WM8727 <-> atmel-ssc0 mapping ok I am getting following error when I tried to play the wav file using command "aplay -D default sample_16.wav ": atmel_ssc_startup: SSC_SR=0x192 Playing WAVE 'sample_16.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo atmel_ssc_hw_params: RCMR=00010721 RFMR=0000008f TCMR=00010702 TFMR=0000008f atmel_ssc_dai: Starting clock atmel_ssc_dai,hw_params: SSC initialized atmel-pcm: hw_params: DMA for SSC0 PCM out initialized (dma_bytes=32768, period_size=8192) transmit enabled SSC_SR=0x000104c3 aplayatmel_ssc_shutdown: transmit disabled SSC_SR=0x000000c3 :atmel_ssc_dau: Stopping clock set_params:1167: not enough memory Does anyone throw some light where and how to solve the issue? Kindly share if one needs further information. Have a Good Day, Thank you, Ankur.

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[alsa-devel] [PATCH] ASoC: Fix backport of WM8994 thermal warning
by Mark Brown 17 Aug '11

17 Aug '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> --- sound/soc/codecs/wm8994.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 9412491..e537267 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3141,9 +3141,9 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994_request_irq(codec->control_data, WM8994_IRQ_FIFOS_ERR, wm8994_fifo_error, "FIFO error", codec); - wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_TEMP_WARN, + wm8994_request_irq(wm8994->control_data, WM8994_IRQ_TEMP_WARN, wm8994_temp_warn, "Thermal warning", codec); - wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_TEMP_SHUT, + wm8994_request_irq(wm8994->control_data, WM8994_IRQ_TEMP_SHUT, wm8994_temp_shut, "Thermal shutdown", codec); ret = wm8994_request_irq(codec->control_data, WM8994_IRQ_DCS_DONE, -- 1.7.5.4

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[alsa-devel] Playback PCM drain and card fifos
by Eliot Blennerhassett 17 Aug '11

17 Aug '11

Greetings, I am seeing playback audio get truncated at the end of playback. The amount lost is about what is in the on-card fifo. (asihpi driver) For example in a specific test this is about 100ms of audio, though in theory could go much higher if I let the card fifo fill completely. The current driver calls snd_pcm_period_elapsed when the card has read from the alsa ringbuffer. I.e. it is notifying that the buffer is available for more data to be written. As soon as the driver notifies that the final data has been read by the card, the trigger stop is called, which in turn stops the card playback immediately. However, the data just read has not been played back yet so the audio is truncated. During playback the driver sets runtime->delay = on_card_bytes I'm trying to work out what I can do in the driver (or in alsa mid layer if necessary), to make this work better? regards -- Eliot Blennerhassett AudioScience Inc.

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[alsa-devel] [PATCH] [v2] ASoC: claim the IRQ when the fsl_ssi device is probed, not opened
by Timur Tabi 17 Aug '11

17 Aug '11

The PowerPC Freescale SSI driver is claiming the IRQ when the IRQ when the device is opened, which means that the /proc/interrupts entry for the SSI exists only during playback or capture. This also meant that the user won't know that the IRQ number is wrong until he tries to use the device. Instead, we should claim the IRQ when the device is probed. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 61 ++++++++++++++++++++++++++++------------------ 1 files changed, 37 insertions(+), 24 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index d48afea..06ac2b9 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -289,16 +289,6 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, */ if (!ssi_private->playback && !ssi_private->capture) { struct ccsr_ssi __iomem *ssi = ssi_private->ssi; - int ret; - - /* The 'name' should not have any slashes in it. */ - ret = request_irq(ssi_private->irq, fsl_ssi_isr, 0, - ssi_private->name, ssi_private); - if (ret < 0) { - dev_err(substream->pcm->card->dev, - "could not claim irq %u\n", ssi_private->irq); - return ret; - } /* * Section 16.5 of the MPC8610 reference manual says that the @@ -522,15 +512,12 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream, ssi_private->second_stream = NULL; /* - * If this is the last active substream, disable the SSI and release - * the IRQ. + * If this is the last active substream, disable the SSI. */ if (!ssi_private->playback && !ssi_private->capture) { struct ccsr_ssi __iomem *ssi = ssi_private->ssi; clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); - - free_irq(ssi_private->irq, ssi_private); } } @@ -675,17 +662,30 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) ret = of_address_to_resource(np, 0, &res); if (ret) { dev_err(&pdev->dev, "could not determine device resources\n"); - kfree(ssi_private); - return ret; + goto error_kmalloc; } ssi_private->ssi = of_iomap(np, 0); if (!ssi_private->ssi) { dev_err(&pdev->dev, "could not map device resources\n"); - kfree(ssi_private); - return -ENOMEM; + ret = -ENOMEM; + goto error_kmalloc; } ssi_private->ssi_phys = res.start; + ssi_private->irq = irq_of_parse_and_map(np, 0); + if (ssi_private->irq == NO_IRQ) { + dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); + ret = -ENXIO; + goto error_iomap; + } + + /* The 'name' should not have any slashes in it. */ + ret = request_irq(ssi_private->irq, fsl_ssi_isr, 0, ssi_private->name, + ssi_private); + if (ret < 0) { + dev_err(&pdev->dev, "could not claim irq %u\n", ssi_private->irq); + goto error_irqmap; + } /* Are the RX and the TX clocks locked? */ if (of_find_property(np, "fsl,ssi-asynchronous", NULL)) @@ -711,7 +711,7 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) if (ret) { dev_err(&pdev->dev, "could not create sysfs %s file\n", ssi_private->dev_attr.attr.name); - goto error; + goto error_irq; } /* Register with ASoC */ @@ -720,7 +720,7 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) ret = snd_soc_register_dai(&pdev->dev, &ssi_private->cpu_dai_drv); if (ret) { dev_err(&pdev->dev, "failed to register DAI: %d\n", ret); - goto error; + goto error_dev; } /* Trigger the machine driver's probe function. The platform driver @@ -741,18 +741,28 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) if (IS_ERR(ssi_private->pdev)) { ret = PTR_ERR(ssi_private->pdev); dev_err(&pdev->dev, "failed to register platform: %d\n", ret); - goto error; + goto error_dai; } return 0; -error: +error_dai: snd_soc_unregister_dai(&pdev->dev); + +error_dev: dev_set_drvdata(&pdev->dev, NULL); - if (dev_attr) - device_remove_file(&pdev->dev, dev_attr); + device_remove_file(&pdev->dev, dev_attr); + +error_irq: + free_irq(ssi_private->irq, ssi_private); + +error_irqmap: irq_dispose_mapping(ssi_private->irq); + +error_iomap: iounmap(ssi_private->ssi); + +error_kmalloc: kfree(ssi_private); return ret; @@ -766,6 +776,9 @@ static int fsl_ssi_remove(struct platform_device *pdev) snd_soc_unregister_dai(&pdev->dev); device_remove_file(&pdev->dev, &ssi_private->dev_attr); + free_irq(ssi_private->irq, ssi_private); + irq_dispose_mapping(ssi_private->irq); + kfree(ssi_private); dev_set_drvdata(&pdev->dev, NULL); -- 1.7.4.4

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[alsa-devel] [PATCH] ASoC: claim the IRQ when the fsl_ssi device is probed, not opened
by Timur Tabi 17 Aug '11

17 Aug '11

The PowerPC Freescale SSI driver is claiming the IRQ when the IRQ when the device is opened, which means that the /proc/interrupts entry for the SSI exists only during playback or capture. This also meant that the user won't know that the IRQ number is wrong until he tries to use the device. Instead, we should claim the IRQ when the device is probed. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- arch/powerpc/configs/mpc85xx_defconfig | 1 + arch/powerpc/configs/mpc85xx_smp_defconfig | 1 + 2 files changed, 2 insertions(+), 0 deletions(-) diff --git a/arch/powerpc/configs/mpc85xx_defconfig b/arch/powerpc/configs/mpc85xx_defconfig index fcd85d2..a3467bf 100644 --- a/arch/powerpc/configs/mpc85xx_defconfig +++ b/arch/powerpc/configs/mpc85xx_defconfig @@ -139,6 +139,7 @@ CONFIG_SND=y CONFIG_SND_INTEL8X0=y # CONFIG_SND_PPC is not set # CONFIG_SND_USB is not set +CONFIG_SND_SOC=y CONFIG_HID_A4TECH=y CONFIG_HID_APPLE=y CONFIG_HID_BELKIN=y diff --git a/arch/powerpc/configs/mpc85xx_smp_defconfig b/arch/powerpc/configs/mpc85xx_smp_defconfig index 908c941..9693f6e 100644 --- a/arch/powerpc/configs/mpc85xx_smp_defconfig +++ b/arch/powerpc/configs/mpc85xx_smp_defconfig @@ -140,6 +140,7 @@ CONFIG_SND=y CONFIG_SND_INTEL8X0=y # CONFIG_SND_PPC is not set # CONFIG_SND_USB is not set +CONFIG_SND_SOC=y CONFIG_HID_A4TECH=y CONFIG_HID_APPLE=y CONFIG_HID_BELKIN=y -- 1.7.4.4

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Re: [alsa-devel] USB Audio questions
by Torsten Schenk 17 Aug '11

17 Aug '11

> Thanks for sharing your experience. > > > - The device/driver sends per urb a constant number of packets (as you > > already also discovered). The number of samples per packet differ > > however (+- 1 sample per packet). I accuse the asynchronousity between > > the urb interval and the sound card's internal clock for this fact. The > > device expects the driver to send back an urb with exactly the same > > structure. The device produced crackling noise if the samples per > > packet were not equal. I cannot say to what extend this is true (i.e. > > if you could wait some packets until catching up the difference without > > producing crackles) but due to this fact it's at least not a simple > > task to send urbs exactly corresponding to the period limit. > > I am still unclear on what makes the number of samples per packet vary, I > didn't see this in the code. I understand that it could vary for 44.1 but > for 48kHz (my case) I'd think it should be constant. The variation is not directly in the code. I remember that the caiaq driver enlightened me since this behaviour is commented there. The variation comes from the device and these steps lead to this variation: 1. the driver sends n urbs containing m packets with size 0 (i.e. no audio) to the device 2. the device sends back each of the n urbs one-by-one, filled with the captured audio; the urbs will contain m packets, each packet having an individual number of samples s_p ("sample" being defined here as: 1 sample contains one value for each channel) 3. for each urb returned from the device, the driver sends another urb containing again m packets, but this time with s_p samples per packet (corresponding to the packet-dependent s_p received from the device) I indeed found out that when using 48kHz, I first thought that the number of samples per packet were identical every time. When using this heuristic, crackling noise occured in the sound output and I looked at the number of samples per packet. They varied not very often but sometimes - around once per 2 seconds. My explanation for this problem is, that the sound card has its own clock, that will not oscillate at exactly an integer multiple as the computer's/usb's internal clock. Even temperature will change the frequency a little bit, meaning that 48kHz is not always exactly 48kHz. I would have to start speculating about which clock drives the usb urb scheduling, so this question should be left for the more experienced USB programmers. > > > - Another thing came into my mind while reading your post, but I'm not > > sure if that would apply in praxis: If sending an urb is scheduled with > > the alsa buffer, it might be possible that the buffer is overwritten > > before the urb is sent, leading to complete audio-rubbish. > > Indeed there could be some issues with pulseaudio rewinds, but the fact is > that rewinds are broken anyway when the data is copied... I gave this another thought and came to the conclusion: if you'd call snd_pcm_period_elapsed in the output urb retire callback, the theoretical problem wouldn't occur. Torsten

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[alsa-devel] [HELP WANTED] ALSA Driver developer
by Gabriel Beddingfield 16 Aug '11

16 Aug '11

All: A recruiter approached me looking for an ALSA driver developer to enhance an existing driver. This is a short-term, off-site gig. Some details below. Please contact Mr. DeAmelio directly. Thanks, Gabriel ---------- Forwarded message ---------- From: David DeAmelio <david_deamelio(a)oxfordcorp.com> Date: Tue, Aug 16, 2011 at 2:29 PM Subject: Job Description - Oxford International To: gabrbedd(a)gmail.com Hey Gabriel, Here’s the description….Thanks. Location: OFFSITE Start Date: ASAP Duration: 1-2 Weeks Manager needs consultant to come onsite for 1st few days. Client will pay for flight there and back. Must: Linux ALSA Device Drivers—Alsa- advanced linux sound architecture Blackfin Plus: D2D: Consultant will be enhancing an audio driver to support a voice quality recorder __________________________________ David De Amelio Team Lead Software/Hardware Consulting Services Oxford International a division of Oxford Global Resources 100 Cummings Center Suite 206N Beverly, MA 01915 877.258.9982 x 7608 Office 978.922.7547 FAX david_deamelio(a)oxfordcorp.com www.oxfordcorp.com Oxford Europe The Right Talent. Right Now.® This e-mail, and any attachments thereto, is intended only for the addressee(s) named herein and may contain privileged and/or confidential information. If you are not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail, and any attachments thereto, is strictly prohibited. If you have received this e-mail in error, immediately notify the sender by return e-mail and permanently delete the original, including any attachments thereto, and any copy and printout thereof.

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Re: [alsa-devel] USB Audio questions
by Clemens Ladisch 16 Aug '11

16 Aug '11

Pierre-Louis Bossart wrote: > What is this USB frame counter and what would to enable its use, at > least for playback where there is scope for lots of power > optimizations. USB timing is defined by frames, of which there are about 1000 per second. The audio driver schedules one packet to be sent/received for each frame. Multiple packets are grouped in a URB (USB request buffer); the controller raises an interrupt when the last packet of the URB has been completed. By reading the current frame number, the driver could deduce how many packets (and therefore, how many samples) have been processed by the controller. > > Why more efficient? You cannot avoid double-buffering when a USB packet > > wraps around the end of the ALSA ring buffer. > > What I had in mind is a case where the ring buffer is divided in separate > periods, which would each correspond to an integer number of URBs/packets. This works only when the driver has control over how many samples actually are transmitted in each USB frame. The playback streams of many USB audio devices work this way, but quite a few others, and capture streams, don't. > It's probably less generic than the current driver but a whole lot > simpler/efficient. The code wouldn't be simpler because we'd still need the other algorithm. This would be just an optimization. Regards, Clemens

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Re: [alsa-devel] USB Audio questions
by Clemens Ladisch 16 Aug '11

16 Aug '11

Pierre-Louis Bossart wrote: > - Is there any good reason why the max number of packets per urb defaults to > 8? Not really. It is an attempt to compromise between interrupt frequency and the latency resulting from packet queueing. > - Increasing the number of packets/urbs solves my power issue but not the > synchronization issue. If I reduce the number of urbs to reduce the > interrupt rate, then the accuracy of the hw_pointer is decreased big time > and it becomes difficult to synchronize with video. The .pointer routine > only keeps track of retired urbs, it seems there's no way to know how many > samples were played at a given time? Only the URB completion callback tells us about finished packets. For playback, we know how many bytes we had submitted, so we could derive the current position from the USB frame counter, but for capture, we don't know the precise byte count until the HCD has looked at the descriptors. > Isn't there a direct way to query the usb core to know how many bytes > have been transmitted and make the hw_ptr/delay information independent > of the urb buffer size? No, this information isn't returned earlier than the completion callback. It would be nice to be able to tell the HCD to collect information about all currently completed USB frames, regardless of whether the entire URB has finished or whether its interrupt flag is set, but this would require changes in all host controller drivers. (I'm contemplating something like this for FireWire, but there we have only one controller driver.) > - Is there any reason why the driver relies on vmalloc and memcpy? It seems > like using physically-contiguous memory would be more efficient, as done in > the CAIQ driver? Why more efficient? You cannot avoid double-buffering when a USB packet wraps around the end of the ALSA ring buffer. > - The period information is not updated when the urbs are allocated. In my > case the period specified was 25ms, but with all the approximations the > actual period (number of urbs*buffer_size_per_urb) was 24ms. Please note that URBs are usually not completely filled. For playback, the driver makes URBs shorter so that they end as nearly as possible on or after a period boundary; for capture, we cannot predict sample counts, so the driver just uses one URB per ms. Regards, Clemens

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Re: [alsa-devel] USB Audio questions
by Pierre-Louis Bossart 16 Aug '11

16 Aug '11

> For USB 3.0 host controllers, the HCD doesn't get the information about > which frame exactly the transfer went out in. This is because all the > scheduling is done in hardware, not software. The xHCI driver can only > record the frame that was current when the interrupt for the URB > happened. > > Pierre, I'm not sure you can get the precise measurements you want out > of a USB 3.0 host controller to do the audio/video sync. You're > welcome > to suggest something to Steve McGowan to add to the xHCI spec, and you > should probably double check with him that what I'm saying is correct. Thanks Sarah for the heads-up. I was only looking at existing USB Audio solutions for now, looks like there's some work to be done for USB3 as well. We do need precise timing information for all audio drivers. Probably a good topic for LPC discussions... -Pierre

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[alsa-devel] [PATCH] ASoC: Run Speyside WM8962 at 512fs
by Mark Brown 16 Aug '11

16 Aug '11

Ensure we have access to all the advanced DSP functinality offered by the WM8962 by running the system clock at 512fs. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/speyside_wm8962.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/speyside_wm8962.c b/sound/soc/samsung/speyside_wm8962.c index 0b9eb5f..753e1c2 100644 --- a/sound/soc/samsung/speyside_wm8962.c +++ b/sound/soc/samsung/speyside_wm8962.c @@ -28,13 +28,13 @@ static int speyside_wm8962_set_bias_level(struct snd_soc_card *card, if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, WM8962_FLL_MCLK, 32768, - 44100 * 256); + 44100 * 512); if (ret < 0) pr_err("Failed to start FLL: %d\n", ret); ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_FLL, - 44100 * 256, + 44100 * 512, SND_SOC_CLOCK_IN); if (ret < 0) { pr_err("Failed to set SYSCLK: %d\n", ret); -- 1.7.5.4

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[alsa-devel] [PATCH] ASoC: Clear completions from late WM8996 FLL lock IRQs
by Mark Brown 16 Aug '11

16 Aug '11

In case we have a pending completion, for example due to a problem with the input clock which got corrected after we timed out. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8996.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index ab8e9d1..a44a0e5 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2106,6 +2106,9 @@ static int wm8996_set_fll(struct snd_soc_codec *codec, int fll_id, int source, snd_soc_write(codec, WM8996_FLL_EFS_1, fll_div.lambda); + /* Clear any pending completions (eg, from failed startups) */ + try_wait_for_completion(&wm8996->fll_lock); + snd_soc_update_bits(codec, WM8996_FLL_CONTROL_1, WM8996_FLL_ENA, WM8996_FLL_ENA); -- 1.7.5.4

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[alsa-devel] How to pair Wine with ALSA? part2: buffer&period size / blocking / mmap (long)
by Joerg-Cyril.Hoehle＠t-systems.com 16 Aug '11

16 Aug '11

Dear ALSA developers, [Please refer to part1:intro & underruns for the introductory text] http://mailman.alsa-project.org/pipermail/alsa-devel/2011-August/042746.html Topic T2 period and buffer size and time Wine apps using the old winmm API cannot tell at waveOutOpen time "give me a large buffer" or "give me fast reaction to explosions". R6 A good compromise is needed for when Wine opens ALSA on behalf of winmm. Years later, the mmdevapi introduced with Vista provides "period" and duration parameters. It also implements sort of a dmix device: it appears to mix at a fixed rate (48000 or 44100 samples/sec, user settable) and to mix data in packets of 10ms. Incidentally, that's why MS claims 10ms latency. For compatibility, Wine should match that rate -- at least when accessing the "default" device. Does that really translate to set_period_time? I doubt it. I wonder why Wine up to now insists on particular ALSA buffer and period sizes. I tend to consider it's none of its business. It might make sense for Wine to use a periodic timer that calls snd_pcm_write (see T3 below). *That* timer should be set to mmdevapi's 10ms. Doesn't that translate to set_period_time_*max*(10ms)? However if ALSA wants/requires 50ms, why not? Furthermore, if the app wants to use large buffers, why insist on a tiny period on the ALSA side? (MS appears to stick to 10ms packets regardless of the app's requested buffer size). I expect that setting Wine's timer period to at least ALSA's allows it to actually find room for new data each turn. So here's what makes sense to me: if (shared_mode && device=="default") set_period_time_near(10ms); else if (exclusive_mode) /* mmdevapi:Initialize receives period from the app */ set_period_time_near(clamp(3ms, app's wish in mmdevapi:Intialize, 100ms)); /* 100ms is arbitrary, why not 1s? */ snd_pcm_hw_params() wine_set_timer_rate(clamp(3ms, snd_pcm_get_period_time(), 0.5s)) Topic T2.b Duration / buffer size mmdevapi's Initialize method receives a duration parameter as a hint towards either small latency or large buffering. One would think that it makes perfect sense to forward that to snd_pcm_set_buffer_time. *However*, mmdevapi also requires to hand out a pointer to a buffer that large (GetBuffer). Thus Wine must maintain a buffer that large (possibly even two of them, for reasons not relevant here). Now should ALSA really keep yet another buffer that large? Isn't that precisely why people have gripes with PA's 2s buffer? I'm wondering whether Wine should solely rely on its periodic timer to regularly submit data and e.g. ask ALSA to use a buffer 3 times the period? 30ms (3x10ms) seems to play a role in MS systems. Dmix seems to prefer 4 to 6 times period size. Unfortunately, snd_pcm_get_period_time may be known only after invoking snd_pcm_hw_params(), so I can't express: set_period_time_near(10ms); set_buffer_time_near(3 x actual_period); Prefer set_buffer_time_near(3 x period_above); or simply not call set_buffer at all? Finally, there's that snd_pcm_sw_params_set_avail_min whose purpose I cannot figure out. Should Wine call snd_pcm_sw_params_set_avail_min(1); or snd_pcm_sw_params_set_avail_min(0); or not at all? Topic T3 blocking or not Wine has traditionally used ALSA in non-blocking mode, which ALSA people recommended against (still?). Now suppose every write is preceded with avail_update, for reasons I gave in part1: underruns. Remember my example from part 1, slightly refined: if (snd_pcm_avail_update(&avail) > buffer_size) snd_pcm_reset() /* skip over late samples */ /* should be equivalent to snd_pcm_forward(avail) */ written = snd_pcm_write(min(avail,frames)); My understanding of the semantics is that write(<avail) will not block, thus Wine could as well dispose of SND_PCM_NONBLOCK. In the NONBLOCK case, this could be simplified to written = snd_pcm_write(frames); since I discovered that ALSA can write a little more than what avail returns, and NONBLOCK implies it'll not wait in an attempt to write a 2s data buffer, but return written < frames instead. Actually, I've a slightly more elaborate sequence in mind. I've read that snd_pcm_open() may be delayed because of networking issues, which I want to avoid in the audio thread. Therefore I'm considering using: snd_pcm_open(SND_PCM_NONBLOCK); ... setup hw&sw_params snd_pcm_prepare() snd_pcm_nonblock(0); /* or should it be called before prepare? */ I.e. allow blocking while playing, which does not actually happen thanks to avail_update() and a push model based on periodic timers to feed data with pcm_write(). Note that so far, I've not questioned whether a timer-based push model actually makes sense for Wine with ALSA... Topic T4 mmap After I re-read the mmdevapi documentation, I believe that mmdevapi's GetBuffer/ReleaseBuffer rendering protocol may be compatible with ALSA's snd_pcm_mmap_begin+commit after all. It all depends on whether ALSA grants requests for sizes as large as buffer_size, with buffer_size up to 2 seconds * samples/sec. Hence, as an optimization, one could imagine a driver using mmap, like Winealsa used to have prior to the 2011 rewrite. Yet it always had the fallback without mmap, so let's concentrate on the non-mmap case initially and get that right. Thank you for your help and for reading up to this point, Jörg Höhle

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Re: [alsa-devel] USB Audio questions
by David Henningsson 16 Aug '11

16 Aug '11

On 08/15/2011 10:31 PM, Pierre-Louis Bossart wrote: > - Increasing the number of packets/urbs solves my power issue but not the > synchronization issue. If I reduce the number of urbs to reduce the > interrupt rate, then the accuracy of the hw_pointer is decreased big time > and it becomes difficult to synchronize with video. I think the same thing is a problem for quite a few other devices as well - I wonder if we need some kind of "pointer granularity" variable to be exported through the ALSA API? PulseAudio could use that to determine whether or not to enable timer-based scheduling. And in these cases, maybe a call to hw_pointer could return hw_pointer and time, and then PulseAudio etc could use that for extrapolation (or the extrapolation could be done in alsa-lib). -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

2 1

[alsa-devel] [PATCH] ASoC: Implement WM8994 thermal warning and shutdown interrupt support
by Mark Brown 16 Aug '11

16 Aug '11

ALSA doesn't really have good mechanisms for dealing with these so we just log them - the hardware already has automatic shutdown support. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 26 ++++++++++++++++++++++++++ 1 files changed, 26 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index a0d6274..9412491 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3045,6 +3045,24 @@ static irqreturn_t wm8994_fifo_error(int irq, void *data) return IRQ_HANDLED; } +static irqreturn_t wm8994_temp_warn(int irq, void *data) +{ + struct snd_soc_codec *codec = data; + + dev_err(codec->dev, "Thermal warning\n"); + + return IRQ_HANDLED; +} + +static irqreturn_t wm8994_temp_shut(int irq, void *data) +{ + struct snd_soc_codec *codec = data; + + dev_crit(codec->dev, "Thermal shutdown\n"); + + return IRQ_HANDLED; +} + static int wm8994_codec_probe(struct snd_soc_codec *codec) { struct wm8994 *control; @@ -3123,6 +3141,10 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994_request_irq(codec->control_data, WM8994_IRQ_FIFOS_ERR, wm8994_fifo_error, "FIFO error", codec); + wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_TEMP_WARN, + wm8994_temp_warn, "Thermal warning", codec); + wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_TEMP_SHUT, + wm8994_temp_shut, "Thermal shutdown", codec); ret = wm8994_request_irq(codec->control_data, WM8994_IRQ_DCS_DONE, wm_hubs_dcs_done, "DC servo done", @@ -3387,6 +3409,8 @@ err_irq: wm8994_free_irq(codec->control_data, WM8994_IRQ_DCS_DONE, &wm8994->hubs); wm8994_free_irq(codec->control_data, WM8994_IRQ_FIFOS_ERR, codec); + wm8994_free_irq(codec->control_data, WM8994_IRQ_TEMP_SHUT, codec); + wm8994_free_irq(codec->control_data, WM8994_IRQ_TEMP_WARN, codec); err: kfree(wm8994); return ret; @@ -3409,6 +3433,8 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) wm8994_free_irq(codec->control_data, WM8994_IRQ_DCS_DONE, &wm8994->hubs); wm8994_free_irq(codec->control_data, WM8994_IRQ_FIFOS_ERR, codec); + wm8994_free_irq(codec->control_data, WM8994_IRQ_TEMP_SHUT, codec); + wm8994_free_irq(codec->control_data, WM8994_IRQ_TEMP_WARN, codec); switch (control->type) { case WM8994: -- 1.7.5.4

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[alsa-devel] [PATCH] ASoC: Add WM8958 noise gate support
by Mark Brown 16 Aug '11

16 Aug '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/linux/mfd/wm8994/registers.h | 45 ++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8994-tables.c | 12 ++++---- sound/soc/codecs/wm8994.c | 38 ++++++++++++++++++++++++++++ 3 files changed, 89 insertions(+), 6 deletions(-) diff --git a/include/linux/mfd/wm8994/registers.h b/include/linux/mfd/wm8994/registers.h index 54dbaa9..d7be463 100644 --- a/include/linux/mfd/wm8994/registers.h +++ b/include/linux/mfd/wm8994/registers.h @@ -138,6 +138,8 @@ #define WM8994_AIF1_DAC1_FILTERS_2 0x421 #define WM8994_AIF1_DAC2_FILTERS_1 0x422 #define WM8994_AIF1_DAC2_FILTERS_2 0x423 +#define WM8958_AIF1_DAC1_NOISE_GATE 0x430 +#define WM8958_AIF1_DAC2_NOISE_GATE 0x431 #define WM8994_AIF1_DRC1_1 0x440 #define WM8994_AIF1_DRC1_2 0x441 #define WM8994_AIF1_DRC1_3 0x442 @@ -195,6 +197,7 @@ #define WM8994_AIF2_ADC_FILTERS 0x510 #define WM8994_AIF2_DAC_FILTERS_1 0x520 #define WM8994_AIF2_DAC_FILTERS_2 0x521 +#define WM8958_AIF2_DAC_NOISE_GATE 0x530 #define WM8994_AIF2_DRC_1 0x540 #define WM8994_AIF2_DRC_2 0x541 #define WM8994_AIF2_DRC_3 0x542 @@ -3040,6 +3043,34 @@ #define WM8994_AIF1DAC2_3D_ENA_WIDTH 1 /* AIF1DAC2_3D_ENA */ /* + * R1072 (0x430) - AIF1 DAC1 Noise Gate + */ +#define WM8958_AIF1DAC1_NG_HLD_MASK 0x0060 /* AIF1DAC1_NG_HLD - [6:5] */ +#define WM8958_AIF1DAC1_NG_HLD_SHIFT 5 /* AIF1DAC1_NG_HLD - [6:5] */ +#define WM8958_AIF1DAC1_NG_HLD_WIDTH 2 /* AIF1DAC1_NG_HLD - [6:5] */ +#define WM8958_AIF1DAC1_NG_THR_MASK 0x000E /* AIF1DAC1_NG_THR - [3:1] */ +#define WM8958_AIF1DAC1_NG_THR_SHIFT 1 /* AIF1DAC1_NG_THR - [3:1] */ +#define WM8958_AIF1DAC1_NG_THR_WIDTH 3 /* AIF1DAC1_NG_THR - [3:1] */ +#define WM8958_AIF1DAC1_NG_ENA 0x0001 /* AIF1DAC1_NG_ENA */ +#define WM8958_AIF1DAC1_NG_ENA_MASK 0x0001 /* AIF1DAC1_NG_ENA */ +#define WM8958_AIF1DAC1_NG_ENA_SHIFT 0 /* AIF1DAC1_NG_ENA */ +#define WM8958_AIF1DAC1_NG_ENA_WIDTH 1 /* AIF1DAC1_NG_ENA */ + +/* + * R1073 (0x431) - AIF1 DAC2 Noise Gate + */ +#define WM8958_AIF1DAC2_NG_HLD_MASK 0x0060 /* AIF1DAC2_NG_HLD - [6:5] */ +#define WM8958_AIF1DAC2_NG_HLD_SHIFT 5 /* AIF1DAC2_NG_HLD - [6:5] */ +#define WM8958_AIF1DAC2_NG_HLD_WIDTH 2 /* AIF1DAC2_NG_HLD - [6:5] */ +#define WM8958_AIF1DAC2_NG_THR_MASK 0x000E /* AIF1DAC2_NG_THR - [3:1] */ +#define WM8958_AIF1DAC2_NG_THR_SHIFT 1 /* AIF1DAC2_NG_THR - [3:1] */ +#define WM8958_AIF1DAC2_NG_THR_WIDTH 3 /* AIF1DAC2_NG_THR - [3:1] */ +#define WM8958_AIF1DAC2_NG_ENA 0x0001 /* AIF1DAC2_NG_ENA */ +#define WM8958_AIF1DAC2_NG_ENA_MASK 0x0001 /* AIF1DAC2_NG_ENA */ +#define WM8958_AIF1DAC2_NG_ENA_SHIFT 0 /* AIF1DAC2_NG_ENA */ +#define WM8958_AIF1DAC2_NG_ENA_WIDTH 1 /* AIF1DAC2_NG_ENA */ + +/* * R1088 (0x440) - AIF1 DRC1 (1) */ #define WM8994_AIF1DRC1_SIG_DET_RMS_MASK 0xF800 /* AIF1DRC1_SIG_DET_RMS - [15:11] */ @@ -3651,6 +3682,20 @@ #define WM8994_AIF2DAC_3D_ENA_WIDTH 1 /* AIF2DAC_3D_ENA */ /* + * R1328 (0x530) - AIF2 DAC Noise Gate + */ +#define WM8958_AIF2DAC_NG_HLD_MASK 0x0060 /* AIF2DAC_NG_HLD - [6:5] */ +#define WM8958_AIF2DAC_NG_HLD_SHIFT 5 /* AIF2DAC_NG_HLD - [6:5] */ +#define WM8958_AIF2DAC_NG_HLD_WIDTH 2 /* AIF2DAC_NG_HLD - [6:5] */ +#define WM8958_AIF2DAC_NG_THR_MASK 0x000E /* AIF2DAC_NG_THR - [3:1] */ +#define WM8958_AIF2DAC_NG_THR_SHIFT 1 /* AIF2DAC_NG_THR - [3:1] */ +#define WM8958_AIF2DAC_NG_THR_WIDTH 3 /* AIF2DAC_NG_THR - [3:1] */ +#define WM8958_AIF2DAC_NG_ENA 0x0001 /* AIF2DAC_NG_ENA */ +#define WM8958_AIF2DAC_NG_ENA_MASK 0x0001 /* AIF2DAC_NG_ENA */ +#define WM8958_AIF2DAC_NG_ENA_SHIFT 0 /* AIF2DAC_NG_ENA */ +#define WM8958_AIF2DAC_NG_ENA_WIDTH 1 /* AIF2DAC_NG_ENA */ + +/* * R1344 (0x540) - AIF2 DRC (1) */ #define WM8994_AIF2DRC_SIG_DET_RMS_MASK 0xF800 /* AIF2DRC_SIG_DET_RMS - [15:11] */ diff --git a/sound/soc/codecs/wm8994-tables.c b/sound/soc/codecs/wm8994-tables.c index 428d4c5..43e6603 100644 --- a/sound/soc/codecs/wm8994-tables.c +++ b/sound/soc/codecs/wm8994-tables.c @@ -1073,8 +1073,8 @@ const struct wm8994_access_mask wm8994_access_masks[WM8994_CACHE_SIZE] = { { 0x0000, 0x0000 }, /* R1069 */ { 0x0000, 0x0000 }, /* R1070 */ { 0x0000, 0x0000 }, /* R1071 */ - { 0x0000, 0x0000 }, /* R1072 */ - { 0x0000, 0x0000 }, /* R1073 */ + { 0x006F, 0x006F }, /* R1072 - AIF1 DAC1 Noise Gate */ + { 0x006F, 0x006F }, /* R1073 - AIF1 DAC2 Noise Gate */ { 0x0000, 0x0000 }, /* R1074 */ { 0x0000, 0x0000 }, /* R1075 */ { 0x0000, 0x0000 }, /* R1076 */ @@ -1329,7 +1329,7 @@ const struct wm8994_access_mask wm8994_access_masks[WM8994_CACHE_SIZE] = { { 0x0000, 0x0000 }, /* R1325 */ { 0x0000, 0x0000 }, /* R1326 */ { 0x0000, 0x0000 }, /* R1327 */ - { 0x0000, 0x0000 }, /* R1328 */ + { 0x006F, 0x006F }, /* R1328 - AIF2 DAC Noise Gate */ { 0x0000, 0x0000 }, /* R1329 */ { 0x0000, 0x0000 }, /* R1330 */ { 0x0000, 0x0000 }, /* R1331 */ @@ -2646,8 +2646,8 @@ const u16 wm8994_reg_defaults[WM8994_CACHE_SIZE] = { 0x0000, /* R1069 */ 0x0000, /* R1070 */ 0x0000, /* R1071 */ - 0x0000, /* R1072 */ - 0x0000, /* R1073 */ + 0x0068, /* R1072 - AIF1 DAC1 Noise Gate */ + 0x0068, /* R1073 - AIF1 DAC2 Noise Gate */ 0x0000, /* R1074 */ 0x0000, /* R1075 */ 0x0000, /* R1076 */ @@ -2902,7 +2902,7 @@ const u16 wm8994_reg_defaults[WM8994_CACHE_SIZE] = { 0x0000, /* R1325 */ 0x0000, /* R1326 */ 0x0000, /* R1327 */ - 0x0000, /* R1328 */ + 0x0068, /* R1328 - AIF2 DAC Noise Gate */ 0x0000, /* R1329 */ 0x0000, /* R1330 */ 0x0000, /* R1331 */ diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index f328369..c45b634 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -299,6 +299,7 @@ static const DECLARE_TLV_DB_SCALE(st_tlv, -3600, 300, 0); static const DECLARE_TLV_DB_SCALE(wm8994_3d_tlv, -1600, 183, 0); static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); static const DECLARE_TLV_DB_SCALE(mixin_boost_tlv, 0, 900, 0); +static const DECLARE_TLV_DB_SCALE(ng_tlv, -10200, 600, 0); #define WM8994_DRC_SWITCH(xname, reg, shift) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ @@ -678,8 +679,45 @@ SOC_SINGLE_TLV("AIF2 EQ5 Volume", WM8994_AIF2_EQ_GAINS_2, 6, 31, 0, eq_tlv), }; +static const char *wm8958_ng_text[] = { + "30ms", "125ms", "250ms", "500ms", +}; + +static const struct soc_enum wm8958_aif1dac1_ng_hold = + SOC_ENUM_SINGLE(WM8958_AIF1_DAC1_NOISE_GATE, + WM8958_AIF1DAC1_NG_THR_SHIFT, 4, wm8958_ng_text); + +static const struct soc_enum wm8958_aif1dac2_ng_hold = + SOC_ENUM_SINGLE(WM8958_AIF1_DAC2_NOISE_GATE, + WM8958_AIF1DAC2_NG_THR_SHIFT, 4, wm8958_ng_text); + +static const struct soc_enum wm8958_aif2dac_ng_hold = + SOC_ENUM_SINGLE(WM8958_AIF2_DAC_NOISE_GATE, + WM8958_AIF2DAC_NG_THR_SHIFT, 4, wm8958_ng_text); + static const struct snd_kcontrol_new wm8958_snd_controls[] = { SOC_SINGLE_TLV("AIF3 Boost Volume", WM8958_AIF3_CONTROL_2, 10, 3, 0, aif_tlv), + +SOC_SINGLE("AIF1DAC1 Noise Gate Switch", WM8958_AIF1_DAC1_NOISE_GATE, + WM8958_AIF1DAC1_NG_ENA_SHIFT, 1, 0), +SOC_ENUM("AIF1DAC1 Noise Gate Hold Time", wm8958_aif1dac1_ng_hold), +SOC_SINGLE_TLV("AIF1DAC1 Noise Gate Threshold Volume", + WM8958_AIF1_DAC1_NOISE_GATE, WM8958_AIF1DAC1_NG_THR_SHIFT, + 7, 1, ng_tlv), + +SOC_SINGLE("AIF1DAC2 Noise Gate Switch", WM8958_AIF1_DAC2_NOISE_GATE, + WM8958_AIF1DAC2_NG_ENA_SHIFT, 1, 0), +SOC_ENUM("AIF1DAC2 Noise Gate Hold Time", wm8958_aif1dac2_ng_hold), +SOC_SINGLE_TLV("AIF1DAC2 Noise Gate Threshold Volume", + WM8958_AIF1_DAC2_NOISE_GATE, WM8958_AIF1DAC2_NG_THR_SHIFT, + 7, 1, ng_tlv), + +SOC_SINGLE("AIF2DAC Noise Gate Switch", WM8958_AIF2_DAC_NOISE_GATE, + WM8958_AIF2DAC_NG_ENA_SHIFT, 1, 0), +SOC_ENUM("AIF2DAC Noise Gate Hold Time", wm8958_aif2dac_ng_hold), +SOC_SINGLE_TLV("AIF2DAC Noise Gate Threshold Volume", + WM8958_AIF2_DAC_NOISE_GATE, WM8958_AIF2DAC_NG_THR_SHIFT, + 7, 1, ng_tlv), }; static const struct snd_kcontrol_new wm1811_snd_controls[] = { -- 1.7.5.4

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Re: [alsa-devel] USB Audio questions
by Daniel Mack 16 Aug '11

16 Aug '11

Hi, On Mon, Aug 15, 2011 at 10:31 PM, Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> wrote: > I am trying to prototype an application where the key requirement is > playback with minimal power on USB audio accessories while maintaining tight > synchronization with video. With the default ALSA USB code, I am getting > ~130 interrupts/s due to audio, a fairly high CPU load and a very poor sync, > exactly the opposite of my goals... > After a careful analysis of the code I found a couple of issues where the > help of experts would be welcome: > > - Is there any good reason why the max number of packets per urb defaults to > 8? This end-ups being the root cause for the large number of interrupts. My > headset supports up to 192 bytes/packet, that means 1536 bytes/urb and 125 > interrupts/s. Using more packets/urb results in a linear decrease of the > interrupt rate. You cannot use more than 8 subframes per ISO urb, that's what how USB is designed. What exactly did you do in you test? Did you increase MAX_URBS? > - Increasing the number of packets/urbs solves my power issue but not the > synchronization issue. If I reduce the number of urbs to reduce the > interrupt rate, then the accuracy of the hw_pointer is decreased big time > and it becomes difficult to synchronize with video. The .pointer routine > only keeps track of retired urbs, it seems there's no way to know how many > samples were played at a given time? It also makes pulseaudio behave > somewhat strangely since the driver only provides an approximate position. > In the CAIQ driver code, the .pointer routine seems to rely on a direct > register read, a more efficient way to do things. Isn't there a direct way > to query the usb core to know how many bytes have been transmitted and make > the hw_ptr/delay information independent of the urb buffer size? Just to be sure - you're talking about the driver in sound/usb/caiaq, right? After a short comparison, I'd say both the caiaq driver and the generic driver do more or less the same in regard to the .pointer value, as for the output stream, they both increase their internal counter when the packet is prepared, shortly before it is submitted to the host controller. Increasing the number of URBs. Admittedly, this approach is questionable, as a larger number of prepared output buffers will actually cause the client believe that more data has already been written already. You could try moving the hwptr_done increase statement into the retire callback (which is the first place we get to know about packets that have been sent to the device)? > - Is there any reason why the driver relies on vmalloc and memcpy? It seems > like using physically-contiguous memory would be more efficient, as done in > the CAIQ driver? I hacked something to use prealloc_pages/malloc_pages and > things seem to work fine. Can you show a patch? Daniel

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Re: [alsa-devel] hda-intel and runtime power-management
by Tatulea, Dragos 16 Aug '11

16 Aug '11

On Mon, Aug 8, 2011 at 9:11 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > At Sun, 7 Aug 2011 20:55:52 +0300, > Tatulea, Dragos wrote: > > > >> >> I'm in the process of enabling pm runtime pm for the hda_intel > driver, > > >> >> not being sure if this is a sane task :). Noticed that > > >> >> CONFIG_SND_HDA_POWER_SAVE disables codecs and chip when they are > not > > >> >> in use, but the power gain is not noticeable on my hw. On the other > > >> >> hand, a full suspend (azx_suspend) does save a few hundred mW's. > > >> >> I am aware that this is hw specific, but I would like to take > > >> >> advantage of the full suspend. Suspend/resume latency is below > jiffy > > >> >> resolution. > > >> > > > >> > Hm, it's strange. Basically the current power-save does almost > > >> > equivalent with the normal suspend/resume, i.e. the codec is > > >> > suspended and azx_stop_chip() is called. Make sure that the > > >> > power-saving is really kicked in during your test. > > >> > > > >> Indeed, I checked and a codec is stuck in a power transition for some > reason. > > > > > > You should check who accessing the device files via fuser, etc. > > > > Yes, there's a non-muted pcm around there that prevents the codec from sleeping. I have a patch that checks for all substream states and puts the chip to sleep when they are all paused/suspended. Is this a bad idea? It could be that resuming the chip takes too long, depending on hw... > > >> Besides that, it would be nicer to have the currently standard runtime > > >> pm on codec bus instead of doing the same thing with > > >> CONFIG_SND_HDA_POWER_SAVE, right? > > > > > > Ideally, yes, but practically it's difficult, so far, because the > > > driver can manage multiple codecs and we need the individual codec > > > power-up/down. With the current runtime pm scheme, it's difficult to > > > implement like this, as it's no simple suspend/resume of the driver > > > itself. > > > > > The codec bus can contain the generic runtime pm part which will call > > runtime_suspend/_resume for individual codecs, depending on usage. > > Each codec can register it's own dev_pm_ops. > > Yes, but this would need relatively big changes in the structure > of the hd-audio driver, e.g. assigning struct device per codec > instance, etc. > > Still looking into it, maybe there's a less invasive way of doing this. Thanks, Dragos

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Re: [alsa-devel] HDAudio period/buffer sizes
by Takashi Iwai 15 Aug '11

15 Aug '11

At Tue, 26 Jul 2011 14:23:40 -0500, Pierre-Louis Bossart wrote: > > that the buffer itself was already aligned to 128 bytes from the > > very beginning as it's allocated via dma_alloc_coherent(). > > The fact that the commit fixes the real problem means that the buffer > > size must be aligned, too, at least for some hardware. > > I checked internally and the reply was "There is no requirement the buffer > must be 128 byte multiples. As long as it is integer number of samples, we > are good." > > What's the best way to deal with it? Add a driver option? A compilation > option? I'd say a driver option since this must be dynamically adjustable. Care to make a patch? For 3.2 (after 3.1 merge window), we can disable 128-bytes limit as default again for testing. If someone hits the problem, he can switch easily and we can investigate more at this time. thanks, Takashi

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Re: [alsa-devel] [PATCH] ASoC: sta32x: shortcut the for loop to get ir and mcs
by Mark Brown 15 Aug '11

15 Aug '11

On Sun, Aug 14, 2011 at 11:31:04AM +0800, Axel Lin wrote: > There is exactly one match or no match at all during the for loop iteration, > thus we can break from the for loop once a match is found. > > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> Applied, thanks.

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Re: [alsa-devel] [PATCH] ASoC: nuc900-pcm: remove unused variable 'dai'
by Mark Brown 15 Aug '11

15 Aug '11

On Mon, Aug 15, 2011 at 04:38:34PM +0800, Axel Lin wrote: > Remove unused variable 'dai' to eliminate below warning. Applied, thanks.

1 0

Re: [alsa-devel] [PATCH] ASoC: soc-pcm: Remove unused global mutex
by Mark Brown 15 Aug '11

15 Aug '11

On Sun, Aug 14, 2011 at 08:00:12AM +0800, Axel Lin wrote: > Since commit b8c0dab9bf3373010e857a8d3f1b594c60a348dd > "ASoC: core - PCM mutex per rtd", > the global pcm_mutex is not being used any more. Applied, thanks.

1 0

Re: [alsa-devel] [PATCH] ASoC: soc-cache: Remove unneeded codec_drv pointer variable in snd_soc_lzo_get_blksize
by Mark Brown 15 Aug '11

15 Aug '11

On Sat, Aug 13, 2011 at 09:27:16PM +0800, Axel Lin wrote: > Since commit aea170a099793abcd0e6de46b947458073204241 > "ASoC: soc-cache: Add reg_size as a member to snd_soc_codec", > the codec_drv pointer variable is not used in snd_soc_lzo_get_blksize. Applied, thanks.

1 0

Re: [alsa-devel] [PATCH] ASoC: soc-cache: Remove unneeded codec_drv pointer variable in snd_soc_flat_cache_init
by Mark Brown 15 Aug '11

15 Aug '11

On Sat, Aug 13, 2011 at 09:12:01PM +0800, Axel Lin wrote: > Since commit d779fce5d79525d66269c8f6e430e1515d697f3d > "ASoC: soc-cache: Ensure flat compression uses a copy of the defaults cache", > the codec_drv pointer variable is not used any more. Applied, thanks.

1 0

Re: [alsa-devel] [PATCH] ASoC: s6000-pcm: remove unused variable 'dai'
by Mark Brown 15 Aug '11

15 Aug '11

On Sat, Aug 13, 2011 at 11:32:56AM +0800, Axel Lin wrote: > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> Applied, thanks.

1 0

[alsa-devel] [PATCH v2 1/5] ASoC: ad193x: fix registers definition
by Scott Jiang 15 Aug '11

15 Aug '11

fix dac word len mask and adc tdm fmt shift value Signed-off-by: Scott Jiang <scott.jiang.linux(a)gmail.com> --- sound/soc/codecs/ad193x.h | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h index 9747b54..c1029d2 100644 --- a/sound/soc/codecs/ad193x.h +++ b/sound/soc/codecs/ad193x.h @@ -34,7 +34,7 @@ #define AD193X_DAC_LEFT_HIGH (1 << 3) #define AD193X_DAC_BCLK_INV (1 << 7) #define AD193X_DAC_CTRL2 0x804 -#define AD193X_DAC_WORD_LEN_MASK 0xC +#define AD193X_DAC_WORD_LEN_MASK 0x18 #define AD193X_DAC_MASTER_MUTE 1 #define AD193X_DAC_CHNL_MUTE 0x805 #define AD193X_DACL1_MUTE 0 @@ -63,7 +63,7 @@ #define AD193X_ADC_CTRL1 0x80f #define AD193X_ADC_SERFMT_MASK 0x60 #define AD193X_ADC_SERFMT_STEREO (0 << 5) -#define AD193X_ADC_SERFMT_TDM (1 << 2) +#define AD193X_ADC_SERFMT_TDM (1 << 5) #define AD193X_ADC_SERFMT_AUX (2 << 5) #define AD193X_ADC_WORD_LEN_MASK 0x3 #define AD193X_ADC_CTRL2 0x810 -- 1.7.0.4

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Re: [alsa-devel] [PATCH] ASoC: soc-jack: Fix checking return value of request_any_context_irq
by Mark Brown 15 Aug '11

15 Aug '11

On Sat, Aug 13, 2011 at 07:15:01PM +0800, Axel Lin wrote: > request_any_context_irq() returns a negative value on failure. > On success, it returns either IRQC_IS_HARDIRQ or IRQC_IS_NESTED. Applied, thanks.

1 0

[alsa-devel] [PATCH 0/4] RME RayDAT/AIO - enable period size of 32 samples
by Adrian Knoth 15 Aug '11

15 Aug '11

Hi! Here is a series of patches that enable period sizes of 32 samples on RME RayDAT and AIO cards. It also adds 8192 samples per period for MADI and AES cards, this mode was present in the original driver but got lost during the merge. Cheers Adrian Knoth (4): ALSA: hdspm - Allow for 8192 period size on RME MADI and AES cards ALSA: hdspm - Set period_bytes_min to 32 * 4 for new RME cards ALSA: hdspm - Reorder period sizes according to their bit representation ALSA: hdspm - Enable 32 samples/period on RME RayDAT/AIO sound/pci/rme9652/hdspm.c | 50 ++++++++++++++++++++++++++++++++++++-------- 1 files changed, 41 insertions(+), 9 deletions(-) -- 1.7.5.4

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Re: [alsa-devel] [PATCH] ASoC: sta32x: shortcut the for loop to get ir and mcs
by Johannes Stezenbach 15 Aug '11

15 Aug '11

On Sun, Aug 14, 2011 at 11:31:04AM +0800, Axel Lin wrote: > There is exactly one match or no match at all during the for loop iteration, > thus we can break from the for loop once a match is found. > > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> Looks good. Acked-by: Johannes Stezenbach <js(a)sig21.net> Thanks, Johannes > --- > sound/soc/codecs/sta32x.c | 8 ++++++-- > 1 files changed, 6 insertions(+), 2 deletions(-) > > diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c > index fbd7eb9..3d155f5 100644 > --- a/sound/soc/codecs/sta32x.c > +++ b/sound/soc/codecs/sta32x.c > @@ -524,13 +524,17 @@ static int sta32x_hw_params(struct snd_pcm_substream *substream, > rate = params_rate(params); > pr_debug("rate: %u\n", rate); > for (i = 0; i < ARRAY_SIZE(interpolation_ratios); i++) > - if (interpolation_ratios[i].fs == rate) > + if (interpolation_ratios[i].fs == rate) { > ir = interpolation_ratios[i].ir; > + break; > + } > if (ir < 0) > return -EINVAL; > for (i = 0; mclk_ratios[ir][i].ratio; i++) > - if (mclk_ratios[ir][i].ratio * rate == sta32x->mclk) > + if (mclk_ratios[ir][i].ratio * rate == sta32x->mclk) { > mcs = mclk_ratios[ir][i].mcs; > + break; > + } > if (mcs < 0) > return -EINVAL; > > -- > 1.7.4.1 > > > >

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[alsa-devel] Questions about the alsa-driver-unstable tarballs
by David Henningsson 15 Aug '11

15 Aug '11

Hi Takashi, 1) there seems to be a build failure with current alsa-driver-unstable-snapshot [1] : ...as alsa-kernel/soc/Makefile references a mxs/ directory, but this directory is not present in the soc/ directory. 2) it seems like the .bz2 version of alsa-driver-snapshot has disappeared, is this intentional? 3) Could you explain the difference between alsa-driver-snapshot and alsa-driver-unstable-snapshot, and how the alsa-kernel.git, alsa-driver.git and sound-2.6.git come together to form the two different tarballs? Thanks! -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic [1] http://ftp.kernel.org/pub/linux/kernel/people/tiwai/alsa/alsa-driver/alsa-d…

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[alsa-devel] [PATCH] ALSA: ac97: Add HP Compaq dc5100 SFF(PT003AW) to Headphone Jack Sense whitelist
by Dan Chen 15 Aug '11

15 Aug '11

Hi, Attached please find a patch against sound-2.6 master HEAD fixing inaudible audio in the bug reporter's system by unmuting 'Headphone Jack Sense'. To make it permanent, his PCI SSID is added to the whitelist. Please consider applying. Cheers, -Dan

2 1

[alsa-devel] [PATCH] ASoC: Disable pulls on WM8994 AIF2 when starting it
by Mark Brown 15 Aug '11

15 Aug '11

Pull control is availalbe for WM8994 AIF2, generally disabled as part of the GPIO configuration in order to save power after system startup. As on newer devices in the series there is no GPIO functionality on these pins this will happen less naturally so have the driver disable the pulls as the AIF is probed. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 16 ++++++++++++++++ 1 files changed, 16 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index b63c4c8..f328369 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2498,6 +2498,21 @@ static int wm8994_set_tristate(struct snd_soc_dai *codec_dai, int tristate) return snd_soc_update_bits(codec, reg, mask, val); } +static int wm8994_aif2_probe(struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + + /* Disable the pulls on the AIF if we're using it to save power. */ + snd_soc_update_bits(codec, WM8994_GPIO_3, + WM8994_GPN_PU | WM8994_GPN_PD, 0); + snd_soc_update_bits(codec, WM8994_GPIO_4, + WM8994_GPN_PU | WM8994_GPN_PD, 0); + snd_soc_update_bits(codec, WM8994_GPIO_5, + WM8994_GPN_PU | WM8994_GPN_PD, 0); + + return 0; +} + #define WM8994_RATES SNDRV_PCM_RATE_8000_96000 #define WM8994_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ @@ -2565,6 +2580,7 @@ static struct snd_soc_dai_driver wm8994_dai[] = { .rates = WM8994_RATES, .formats = WM8994_FORMATS, }, + .probe = wm8994_aif2_probe, .ops = &wm8994_aif2_dai_ops, }, { -- 1.7.5.4

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[alsa-devel] Investigating ALC662 3stack noise
by Miguel Freitas 15 Aug '11

15 Aug '11

Hi developers, I'm trying to fix an annoying noise i get from my D945GCLF2D Intel Motherboard, but only from Surround/clfe outputs, not from Front jack. Since i'm running out of ideas i thought people here might be able to suggest some pointers. The problem is: whenever I change from 2ch to 6ch Channel Mode in mixer I get the background noise in my speakers. In 2ch mode there is no noise but that doesn't help at all since I want that pin as output. I fail to admit Intel or Realtek would do such a horrible engineering job (Intel is happy to say in their datasheet the motherboard has 95dB S/N ratio). It's more like we did connect something wrong is our driver, or failed to change some required settings, i don't know. What I've tried so far: 1) I removed the cable to front case connectors guessing noise might be coming from there. Nothing. 2) The motherboard/ALC662 assembly is obviously 3stack kind. I double-checked all settings in patch_realtek.c against ALC662 datasheet. Changing NID 0x1A from input to output, unmuting and connecting it to idx 0x01 (surround) seems like the right thing to me, afterall, I do get sound as expected from this port. 3) To eliminate the possibility that the noise was coming from the ALC662 itself (but from a different port) I changed NID 0x0d (surround) to mute idx 01, which is the connection from NID 0x0b (that big mixer with Mic, Line, Beeper, CD etc). The leaves just DAC PCM-2 connected and the noise is still there. 4) Muting/lowering volume for Surround (DAC PCM-2 / NID 0x03) also has no effect on noise. 5) Thinking that some external amplifier on motherboard might need to be disabled I've played with EAPD and GPIOs 0,1. I can clearly see at least GPIO 1 is routed on the motherboard, although no idea if it is just ends in a testpoint pad or that could be a via to some internal layer. Different settings of EAPD/GPIOs pins made no difference to noise. So that is pretty much the point i'm now. I'm starting to consider soldering another jack to unused Port-A (surround) pins... but that would be kind of rude, don't you think so? :-) Any ideas are very welcomed... regards, Miguel

1 0

[alsa-devel] [PATCH] ALSA: snd_usb_caiaq: track submitted output urbs
by Daniel Mack 14 Aug '11

14 Aug '11

The snd_usb_caiaq driver currently assumes that output urbs are serviced in time and doesn't track when and whether they are given back by the USB core. That usually works fine, but due to temporary limitations of the XHCI stack, we faced that urbs were submitted more than once with this approach. As it's no good practice to fire and forget urbs anyway, this patch introduces a proper bit mask to track which requests have been submitted and given back. That alone however doesn't make the driver work in case the host controller is broken and doesn't give back urbs at all, and the output stream will stop once all pre-allocated output urbs are consumed. But it does prevent crashes of the controller stack in such cases. See http://bugzilla.kernel.org/show_bug.cgi?id=40702 for more details. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Reported-and-tested-by: Matej Laitl <matej(a)laitl.cz> Cc: Sarah Sharp <sarah.a.sharp(a)linux.intel.com> Cc: stable(a)kernel.org --- sound/usb/caiaq/audio.c | 31 +++++++++++++++++++++++++++---- sound/usb/caiaq/device.h | 1 + 2 files changed, 28 insertions(+), 4 deletions(-) diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index aa52b3e..2cf87f5 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -139,8 +139,12 @@ static void stream_stop(struct snd_usb_caiaqdev *dev) for (i = 0; i < N_URBS; i++) { usb_kill_urb(dev->data_urbs_in[i]); - usb_kill_urb(dev->data_urbs_out[i]); + + if (test_bit(i, &dev->outurb_active_mask)) + usb_kill_urb(dev->data_urbs_out[i]); } + + dev->outurb_active_mask = 0; } static int snd_usb_caiaq_substream_open(struct snd_pcm_substream *substream) @@ -612,8 +616,8 @@ static void read_completed(struct urb *urb) { struct snd_usb_caiaq_cb_info *info = urb->context; struct snd_usb_caiaqdev *dev; - struct urb *out; - int frame, len, send_it = 0, outframe = 0; + struct urb *out = NULL; + int i, frame, len, send_it = 0, outframe = 0; size_t offset = 0; if (urb->status || !info) @@ -624,7 +628,17 @@ static void read_completed(struct urb *urb) if (!dev->streaming) return; - out = dev->data_urbs_out[info->index]; + /* find an unused output urb that is unused */ + for (i = 0; i < N_URBS; i++) + if (test_and_set_bit(i, &dev->outurb_active_mask) == 0) { + out = dev->data_urbs_out[i]; + break; + } + + if (!out) { + log("Unable to find an output urb to use\n"); + goto requeue; + } /* read the recently received packet and send back one which has * the same layout */ @@ -655,8 +669,12 @@ static void read_completed(struct urb *urb) out->number_of_packets = outframe; out->transfer_flags = URB_ISO_ASAP; usb_submit_urb(out, GFP_ATOMIC); + } else { + struct snd_usb_caiaq_cb_info *oinfo = out->context; + clear_bit(oinfo->index, &dev->outurb_active_mask); } +requeue: /* re-submit inbound urb */ for (frame = 0; frame < FRAMES_PER_URB; frame++) { urb->iso_frame_desc[frame].offset = BYTES_PER_FRAME * frame; @@ -678,6 +696,8 @@ static void write_completed(struct urb *urb) dev->output_running = 1; wake_up(&dev->prepare_wait_queue); } + + clear_bit(info->index, &dev->outurb_active_mask); } static struct urb **alloc_urbs(struct snd_usb_caiaqdev *dev, int dir, int *ret) @@ -829,6 +849,9 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) if (!dev->data_cb_info) return -ENOMEM; + dev->outurb_active_mask = 0; + BUILD_BUG_ON(N_URBS > (sizeof(dev->outurb_active_mask) * 8)); + for (i = 0; i < N_URBS; i++) { dev->data_cb_info[i].dev = dev; dev->data_cb_info[i].index = i; diff --git a/sound/usb/caiaq/device.h b/sound/usb/caiaq/device.h index b2b3101..3f9c633 100644 --- a/sound/usb/caiaq/device.h +++ b/sound/usb/caiaq/device.h @@ -96,6 +96,7 @@ struct snd_usb_caiaqdev { int input_panic, output_panic, warned; char *audio_in_buf, *audio_out_buf; unsigned int samplerates, bpp; + unsigned long outurb_active_mask; struct snd_pcm_substream *sub_playback[MAX_STREAMS]; struct snd_pcm_substream *sub_capture[MAX_STREAMS]; -- 1.7.6

2 1

[alsa-devel] No sound with HDA ATI HDMI since kernel 3.0
by Yannick Palanque 14 Aug '11

14 Aug '11

Hello, First, I'm sorry for my english is awkward and I'm not sure if I'm at the right place. I use Linux amd64 and have two sound "cards": Motherboard: 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xfe9f8000 irq 47 Graphic card: 1 [HDMI ]: HDA-Intel - HDA ATI HDMI HDA ATI HDMI at 0xfeaec000 irq 48 (lspci ouput and alsa-info.txt attached) I have no problem with 2.6.39.3. With kernel 3.0 and 3.0.1, I have no sound with HDMI output. VLC 1.1.3 gives me no sound (I don't see any error message). MPlayer SVN-r31918 (with correct -ao alsa:device=hw=1.3 option, it works with 2.6.39.3) gives me no sound and I see the audio position in seconds on status line going approximatively 6 times faster than normal (in fact, video is 6 times faster, too). I attached dmesg output for kernels 2.6.39.3 and 3.0.1. For both kernels, I have the line hda-intel: IRQ timing workaround is activated for card #1. Suggest a bigger bdl_pos_adj. I tried modifying parameters bdl_pos_adj and position_fix, because I read that some people tried this, when I searched for this problem, but only 2.6.39.3 worked, with default setup (see attached tests.txt file). I also tried to replace the files in sound/pci/hda/* from 2.6.39.3 (I read that about Debian bug 635362 in this very list) but it doesn't work better. I'm only a modest user and I don't know what to do best, but I would be delighted if I could help to solve what I believe to be a bug on a specific hardware. Cordially, -- Yannick Palanque

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Re: [alsa-devel] Please review rt5631 driver on alsa 1.0.24.
by Mark Brown 14 Aug '11

14 Aug '11

On Fri, Aug 12, 2011 at 05:31:40PM +0800, johnnyhsu wrote: > Dear Mr. Brown and Mr. Girdwood, > > This patch is made against sound-2.6.git for-next. > Because the git send-email can't work in my PC, I send the patch by Outlook. > Hope it won't make error. No, this doesn't work at all - Outlook mangles your patch so it can't be applied and text like this at the top of the mail means it's not in the format the tools expect. There's quite a few problems, overall my main comment is the same as the last time - please look at how other drivers do things and try to fit in with the general style. There's a lot of code in here which doesn't look like normal ASoC code with no hint as to why. > +#define RT5631_VERSION "0.01 alsa 1.0.24" The kernel has a perfectly good versioning system, remove this. > +static const u16 rt5631_reg[0x80]; > +static int timesofbclk = 32; No global variables. > +static void snd_soc_write_index(struct snd_soc_codec *codec, > + unsigned int reg, unsigned int value) All your snd_soc_ functions should have some driver specific name, unless they really are generic in which case they should be added to the subsystem code rather than kept in your driver. This one doesn't appear to be generic. > +{ > + snd_soc_write(codec, RT5631_INDEX_ADD, reg); > + snd_soc_write(codec, RT5631_INDEX_DATA, value); > + > + return; > +} The return statement is pointless here. > +static void snd_soc_update_bits_index(struct snd_soc_codec *codec, > + unsigned int reg, unsigned int mask, unsigned int value) > +{ > + unsigned int reg_val; > + > + if (!mask) > + return; > + > + if (mask != 0xffff) { > + reg_val = snd_soc_read_index(codec, reg); > + reg_val &= ~mask; > + reg_val |= (value & mask); > + snd_soc_write_index(codec, reg, reg_val); > + } else { > + snd_soc_write_index(codec, reg, value); > + } > + > + return; > +} Looking at this I'd expect regular snd_soc_update_bits() would work just fine for you. > +/* > + * speaker channel volume 0dB by default > + * Headphone channel volume 0dB by default > + * AXO1/AXO2 channel volume 0dB by default > + * Mic1/Mic2 boost 40dB by default > + * Speaker AMP ratio gain is 1.44X > + * enable HP zero cross > + * change Mic1 & mic2 to differential mode > + */ As I said in reply to your original mail: | This sort of stuff is not suitable for mainline either, the driver | should just use chip defaults for audio paths and leave the use case up | to the application layer. | Please compare your driver with the other CODEC drivers and make sure it | is following a simmilar style to them. > +static const char *rt5631_mic_boost[] = {"Bypass", "+20db", "+24db", > "+30db", > + "+35db", "+40db", "+44db", "+50db", "+52db"}; This should be TLV dB information. > +static const char *rt5631_hpl_source_sel[] = {"LEFT HPVOL", "LEFT DAC"}; > +static const char *rt5631_hpr_source_sel[] = {"RIGHT HPVOL", "RIGHT DAC"}; > +static const char *rt5631_eq_sel[] = {"NORMAL", "CLUB", "DANCE", "LIVE", > "POP", > + "ROCK", "OPPO", "TREBLE", "BASS"}; The general style is not all caps. > +static const struct soc_enum rt5631_enum[] = { Don't use arrays of enums, they're hard to read and error prone. > +static int spk_event(struct snd_soc_dapm_widget *w, > + struct snd_kcontrol *kcontrol, int event) > +{ > + struct snd_soc_codec *codec = w->codec; > + static int spkl_out_enable, spkr_out_enable; > + > + switch (event) { > + case SND_SOC_DAPM_POST_PMU: > + if (!spkl_out_enable && !strcmp(w->name, "SPKL Amp")) { DAPM will take care of tracking if the widget is already enabled or disabled. > + snd_soc_update_bits(codec, RT5631_PWR_MANAG_ADD1, > + PWR_CLASS_D, PWR_CLASS_D); I suspect you want a supply widget. > +static void hp_depop_mode2_onebit(struct snd_soc_codec *codec, int enable) > +{ > + unsigned int soft_vol, hp_zc; This function is rather abstruse in name and code terms, it should be clarified. > +static void hp_mute_unmute_depop_onebit(struct snd_soc_codec *codec, int Similarly here and with the other depop functions, they could all could use rather more blank lines and comments. > + if (enable) { > + snd_soc_write_index(codec, RT5631_SPK_INTL_CTRL, 0x303e); > + snd_soc_update_bits(codec, RT5631_PWR_MANAG_ADD3, > + PWR_CHARGE_PUMP | PWR_HP_L_AMP | PWR_HP_R_AMP, > + PWR_CHARGE_PUMP | PWR_HP_L_AMP | PWR_HP_R_AMP); I rather suspect there should be some supply widgets in here. > + case SND_SOC_DAPM_POST_PMU: > + /* > + * If microphone is stereo, need not copy ADC channel > + * If mic1 is used, copy ADC left to right > + * If mic2 is used, copy ADC right to left > + */ Expose the routes via DAPM and leave the control up to the user. > +static int auxo1_event(struct snd_soc_dapm_widget *w, > + struct snd_kcontrol *kcontrol, int event) > +{ > + struct snd_soc_codec *codec = w->codec; > + static bool aux1_en; > + > + switch (event) { > + case SND_SOC_DAPM_PRE_PMD: > + if (aux1_en) { > + snd_soc_update_bits(codec, RT5631_MONO_AXO_1_2_VOL, > + RT_L_MUTE, RT_L_MUTE); > + aux1_en = false; Again, DAPM will keep track of power for you. I'm not clear why you're managing mutes here rather than letting the user control them. > +/** > + * config_common_power - control all common power of codec system > + * @pmu: power up or not > + */ > +static int config_common_power(struct snd_soc_codec *codec, bool pmu) This should just be inline in set_bias_level(). Though... > + if (pmu) { > + snd_soc_update_bits(codec, RT5631_PWR_MANAG_ADD1, > + PWR_MAIN_I2S_EN | PWR_DAC_REF, > + PWR_MAIN_I2S_EN | PWR_DAC_REF); > + mux_val = snd_soc_read(codec, RT5631_SPK_MONO_HP_OUT_CTRL); > + if (!(mux_val & HP_L_MUX_SEL_DAC_L)) > + snd_soc_update_bits(codec, RT5631_PWR_MANAG_ADD1, > + PWR_DAC_L_TO_MIXER, PWR_DAC_L_TO_MIXER); > + if (!(mux_val & HP_R_MUX_SEL_DAC_R)) > + snd_soc_update_bits(codec, RT5631_PWR_MANAG_ADD1, > + PWR_DAC_R_TO_MIXER, PWR_DAC_R_TO_MIXER); > + if (rt5631->pll_used_flag) > + snd_soc_update_bits(codec, RT5631_PWR_MANAG_ADD2, > + PWR_PLL, PWR_PLL); ..this all looks rather like it should be managed via DAPM. > +static int adc_event(struct snd_soc_dapm_widget *w, > + struct snd_kcontrol *kcontrol, int event) > +{ > + struct snd_soc_codec *codec = w->codec; > + static bool pmu; This isn't going to work if you've got more than one device in the system. > + > + switch (event) { > + case SND_SOC_DAPM_POST_PMD: > + if (pmu) { > + config_common_power(codec, false); > + pmu = false; > + } Again, everything is refcounted for you. Though I'm not entirely clear what this is supposed to do it looks rather like you may want to use a supply widget. In general please try to work with the subsystem rather than layering stuff on top of it - it makes the code more complicated and harder to follow. > +static int rt5631_add_widgets(struct snd_soc_codec *codec) > +{ > + struct snd_soc_dapm_context *dapm = &codec->dapm; > + > + snd_soc_dapm_new_controls(dapm, rt5631_dapm_widgets, > + ARRAY_SIZE(rt5631_dapm_widgets)); > + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); Just point to the tables from the driver, no need to do this manually. > +static const struct pll_div codec_slave_pll_div[] = { > + {256000, 2048000, 0x46f0}, > + {256000, 4096000, 0x3ea0}, > + {352800, 5644800, 0x3ea0}, > + {512000, 8192000, 0x3ea0}, Try to keep the indentation consistent within the table. > +static ssize_t rt5631_index_reg_show(struct device *dev, > + struct device_attribute *attr, char *buf) > +{ > + #define IDX_REG_FMT "%02x: %04x\n" > + #define IDX_REG_LEN 9 Ick, don't indent #defines and don't put them in the middle of functions. > + cnt += sprintf(buf, "RT5631 index register\n"); > + for (i = 0; i < 0x55; i++) { > + if (cnt + IDX_REG_LEN >= PAGE_SIZE - 1) > + break; > + val = snd_soc_read_index(codec, i); > + if (!val) > + continue; > + cnt += sprintf(buf + cnt, IDX_REG_FMT, i, val); > + } Use the subsystem, please. > +EXPORT_SYMBOL_GPL(rt5631_dai); No, don't do this. There's no need for anyone to reference the DAI. > + case SND_SOC_BIAS_OFF: > + snd_soc_update_bits(codec, RT5631_SPK_OUT_VOL, > + RT_L_MUTE | RT_R_MUTE, RT_L_MUTE | RT_R_MUTE); > + snd_soc_update_bits(codec, RT5631_HP_OUT_VOL, > + RT_L_MUTE | RT_R_MUTE, RT_L_MUTE | RT_R_MUTE); This looks very suspicous, it'll mess with whatever the user has configured for the mutes. This applies to quite a few places in the driver > + snd_soc_add_controls(codec, rt5631_snd_controls, > + ARRAY_SIZE(rt5631_snd_controls)); Again, use the tables. > + pr_info("RT5631 initial ok!\n"); The subsystem does a perfectly good job of announcing things, and if you do need to do prints they should use dev_ prints. > +static int rt5631_resume(struct snd_soc_codec *codec) > +{ > + struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec); > + > + snd_soc_update_bits(codec, RT5631_PWR_MANAG_ADD3, > + PWR_VREF | PWR_MAIN_BIAS, PWR_VREF | PWR_MAIN_BIAS); > + schedule_timeout_uninterruptible(msecs_to_jiffies(110)); > + snd_soc_update_bits(codec, RT5631_PWR_MANAG_ADD3, > + PWR_FAST_VREF_CTRL, PWR_FAST_VREF_CTRL); > + rt5631_reg_init(codec); > + > + /* power off ClassD auto Recovery */ > + if (rt5631->codec_version) > + snd_soc_update_bits(codec, RT5631_INT_ST_IRQ_CTRL_2, > + 0x2000, 0x2000); > + else > + snd_soc_update_bits(codec, RT5631_INT_ST_IRQ_CTRL_2, > + 0x2000, 0); This looks like it should share code with the probe. > +/* > + * detect short current for mic1 > + */ > +int rt5631_ext_mic_detect(struct snd_soc_codec *codec) > +{ > + int det; > + > + snd_soc_update_bits(codec, RT5631_MIC_CTRL_2, > + MICBIAS1_S_C_DET_MASK, MICBIAS1_S_C_DET_ENA); > + det = snd_soc_read(codec, RT5631_INT_ST_IRQ_CTRL_2) & 0x0001; > + snd_soc_update_bits(codec, RT5631_INT_ST_IRQ_CTRL_2, 0x0001, > 0x00001); > + > + return det; > +} > +EXPORT_SYMBOL_GPL(rt5631_ext_mic_detect); It looks like you want to write something that slots into soc-jack. > + pr_info("RT5631 Audio Codec %s\n", RT5631_VERSION); This is needlessly chatty. > +struct i2c_driver rt5631_i2c_driver = { > + .driver = { > + .name = "rt5631-codec", > + .owner = THIS_MODULE, No need for a -codec, the device only does one thing. > +/* global definition */ > +#define RT_L_MUTE (0x1 << 15) > +#define RT_R_MUTE (0x1 << 7) Most of the constants need namespacing.

1 0

[alsa-devel] [PATCH] ASoC: Correct revision display for WM1250-EV1 module
by Mark Brown 14 Aug '11

14 Aug '11

The hardware documentation uses revision numbers starting at 1. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm1250-ev1.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c index a98a3ff1..4523c4c 100644 --- a/sound/soc/codecs/wm1250-ev1.c +++ b/sound/soc/codecs/wm1250-ev1.c @@ -74,7 +74,7 @@ static int __devinit wm1250_ev1_probe(struct i2c_client *i2c, return -ENODEV; } - dev_info(&i2c->dev, "revision %d\n", rev); + dev_info(&i2c->dev, "revision %d\n", rev + 1); return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm1250_ev1, &wm1250_ev1_dai, 1); -- 1.7.5.4

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[alsa-devel] [PATCH 1/4] ASoC: DAPM: Allow multiple mixer sources to be routed via the same switch
by Lars-Peter Clausen 14 Aug '11

14 Aug '11

Currently it is only possible to route one source per switch into a mixer. This patch modifies the code, so that it is possible to route multiple sources into a mixer via the same switch. One use-case for this is routing a stereo channel pair into a mono-mixer via the same switch. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-dapm.c | 6 +++++- 1 files changed, 5 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c265311..170c4ff 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -443,6 +443,11 @@ static int dapm_new_mixer(struct snd_soc_dapm_widget *w) if (path->name != (char *)w->kcontrol_news[i].name) continue; + if (w->kcontrols[i]) { + path->kcontrol = w->kcontrols[i]; + continue; + } + wlistsize = sizeof(struct snd_soc_dapm_widget_list) + sizeof(struct snd_soc_dapm_widget *), wlist = kzalloc(wlistsize, GFP_KERNEL); @@ -1556,7 +1561,6 @@ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, /* found, now check type */ found = 1; path->connect = connect; - break; } if (found) -- 1.7.2.5

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[alsa-devel] [PATCH 1/3] soc/soc-io.c: Use regmap_init_{i2c, spi} only if it's available
by Vasily Khoruzhick 14 Aug '11

14 Aug '11

Otherwise we get linker error like: ERROR: "regmap_init_spi" [sound/soc/snd-soc-core.ko] undefined! Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> --- sound/soc/soc-io.c | 6 ++++-- 1 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index e471ed6..c1565e4 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -122,16 +122,18 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, config.val_bits = data_bits; switch (control) { +#ifdef CONFIG_REGMAP_I2C case SND_SOC_I2C: codec->control_data = regmap_init_i2c(to_i2c_client(codec->dev), &config); break; - +#endif +#ifdef CONFIG_REGMAP_SPI case SND_SOC_SPI: codec->control_data = regmap_init_spi(to_spi_device(codec->dev), &config); break; - +#endif case SND_SOC_REGMAP: /* Device has made its own regmap arrangements */ break; -- 1.7.5.rc3

3 5

Re: [alsa-devel] [PATCH] ASoC: soc-core: Fix checking return value of debugfs_create_dir
by Mark Brown 14 Aug '11

14 Aug '11

On Sun, Aug 14, 2011 at 11:23:13AM +0800, Axel Lin wrote: > We have CONFIG_DEBUG_FS guard around the place calling debugfs_create_dir, > thus debugfs_create_dir returns NULL on failure. On the other hand the existing code doesn't actually have any problem, the IS_ERR() won't give false positives, so the changelog isn't really accurate and if anything we'll introduce a bug if someone cut'n'pastes the code elsewhere.

1 0

Re: [alsa-devel] [linux-next] ASoC: sound/soc/samsung/i2s.c build failure
by Mark Brown 13 Aug '11

13 Aug '11

On Sat, Aug 13, 2011 at 12:00:43PM +0800, Axel Lin wrote: > CC sound/soc/samsung/i2s.o > sound/soc/samsung/i2s.c:24:18: error: idma.h: No such file or directory > sound/soc/samsung/i2s.c: In function 'samsung_i2s_dai_probe': > sound/soc/samsung/i2s.c:883: error: implicit declaration of function 'idma_reg_addr_init' > make[3]: *** [sound/soc/samsung/i2s.o] Error 1 > make[2]: *** [sound/soc/samsung] Error 2 > make[1]: *** [sound/soc] Error 2 > make: *** [sound] Error 2 > It is caused by commit a0c27ab2421c47dc7c53f797fffcc0d17cdb122c > ASoC: Revert "ASoC: SAMSUNG: Add I2S0 internal dma driver" > To fix the build error, I think we also need to revert > commit 61100f405de5c16a0866de7843ed442090436e6a > "ASoC: SAMSUNG: Modify I2S driver to support idma". This is already fixed by cherry picking back the original commit, it depended on some stuff from Kukjin's tree.

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Re: [alsa-devel] [PATCH] ASoC: wm8750: Fix missing braces around initializer warning
by Mark Brown 13 Aug '11

13 Aug '11

On Sat, Aug 13, 2011 at 11:45:14AM +0800, Axel Lin wrote: > Fix below warning: > CC sound/soc/codecs/wm8750.o > sound/soc/codecs/wm8750.c:784: warning: missing braces around initializer > sound/soc/codecs/wm8750.c:784: warning: (near initialization for 'wm8750_spi_ids[2].name') Takashi already made this change.

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[alsa-devel] [PATCH] ASoC: Support !CONFIG_REGMAP builds
by Mark Brown 13 Aug '11

13 Aug '11

Since we changed regmap to be selected and register per bus rather than via the core only we can't rely on it being enabled by the ASoC core. Support compiling it out. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Reported-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/soc-io.c | 11 ++++++++++- 1 files changed, 10 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index be5aac3..3f56d52 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -18,6 +18,7 @@ #include <trace/events/asoc.h> +#ifdef CONFIG_REGMAP static int hw_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { @@ -150,4 +151,12 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, return 0; } EXPORT_SYMBOL_GPL(snd_soc_codec_set_cache_io); - +#else +int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, + int addr_bits, int data_bits, + enum snd_soc_control_type control) +{ + return -ENOTSUPP; +} +EXPORT_SYMBOL_GPL(snd_soc_codec_set_cache_io); +#endif -- 1.7.5.4

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Re: [alsa-devel] [linux-next] ASoC: build failure with make eseries_pxa_defconfig
by Mark Brown 13 Aug '11

13 Aug '11

On Sat, Aug 13, 2011 at 10:12:01AM +0800, Axel Lin wrote: > In the case of build with "make eseries_pxa_defconfig;make", > both i2c and spi support are not configured. > Thus CONFIG_REGMAP is default to disable. Meh, right. This is all due to Grant insisting on having direct registration functions for each bus, previously we could just select regmap.

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[alsa-devel] ASoC updates for 3.2
by Mark Brown 12 Aug '11

12 Aug '11

The following changes since commit 7ec41ee5ad5f716f67041c0d49014d0becb5332c: ASoC: omap: Update e-mail address of Jarkko Nikula (2011-08-12 11:45:10 +0900) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-3.2 Axel Lin (2): ASoC: soc-io: Add CONFIG_REGMAP_I2C/CONFIG_REGMAP_SPI guards for regmap_init_i2c/regmap_init_spi ASoC: sgtl5000: fix module device table type for sgtl5000_dt_ids Dong Aisheng (5): ASoC: mxs: add mxs-pcm driver ASoC: mxs: add mxs-saif driver ASoC: mxs: add mxs-sgtl5000 machine driver ASoC: mxs: add asoc configuration files ASoC: sgtl5000: add one missed cache reg Liam Girdwood (2): ASoC: dapm - Add DAPM stream completion event. ASoC: dapm - change stream event dbg to vdgb Manuel Lauss (6): ASoC: Alchemy AC97C/I2SC audio support ASoC: Add a DB1x00 AC97 machine driver ALSA: deprecate MIPS AU1X00 AC97 driver ASoC: au1x: remove automatic DMA device registration from PSC drivers ASoC: au1x: update db1200 machine to the new way of things ASoC: au1x: use substream stream info directly Mark Brown (26): ASoC: Add device tree binding for WM8731 ASoC: Trivial formatting fix in soc-core.c ASoC: Implement new DC servo readback mode for late WM8994 revisions ASoC: Support separate left and right channel dcs_codes values ASoC: Parse board ID/revision information from WM1250-EV1 board ASoC: Remove -codec from WM8523 driver name ASoC: Remove redundant -codec from WM8580 driver name Merge branch 'for-3.1' into for-3.2 ASoC: Use new register map API for ASoC generic physical I/O ASoC: Add regmap as a control type Merge branch 'regmap-asoc' into for-3.2 ASoC: Remove unneeded -codec from WM8753 driver name Merge branch 'for-3.1' into for-3.2 ASoC: Update SMDKs for WM8580 -codec removal ASoC: Remove some more redundant -codecs from driver names Merge branch 'for-3.1' into for-3.2 ASoC: Refactor WM8741 regulator handling into CODEC generic code ASoC: Add SPI support for WM8741 ASoC: Fix warning in WM1250-EV1 driver Merge branch 'for-3.1' into for-3.2 ASoC: Specify register defaults for WM8958 MICBIAS1 and MICBIAS2 Merge branch 'for-3.1' into for-3.2 Merge branch 'for-3.1' into for-3.2 Merge branch 'for-3.1' into for-3.2 ASoC: Allow userspace control of Speyside headphone output ASoC: Add bitfield definitions for WM8958 MICBIAS registers Shawn Guo (1): ASoC: sgtl5000: add device tree probe support Stephen Warren (1): ASoC: soc-io: Fix CONFIG_REGMAP_I2C/SPI guards to support regmap modules .../bindings/sound/soc/codecs/fsl-sgtl5000.txt | 11 + Documentation/devicetree/bindings/sound/wm8731.txt | 18 + arch/mips/alchemy/devboards/db1200/platform.c | 16 + arch/mips/alchemy/devboards/db1x00/platform.c | 48 ++ include/linux/mfd/wm8994/registers.h | 39 ++ include/sound/soc-dapm.h | 2 + include/sound/soc.h | 9 + sound/mips/Kconfig | 5 +- sound/soc/Kconfig | 3 + sound/soc/Makefile | 1 + sound/soc/au1x/Kconfig | 28 + sound/soc/au1x/Makefile | 10 + sound/soc/au1x/ac97c.c | 365 +++++++++++ sound/soc/au1x/db1000.c | 75 +++ sound/soc/au1x/db1200.c | 64 ++- sound/soc/au1x/dbdma2.c | 91 +--- sound/soc/au1x/dma.c | 377 +++++++++++ sound/soc/au1x/i2sc.c | 347 ++++++++++ sound/soc/au1x/psc-ac97.c | 48 +- sound/soc/au1x/psc-i2s.c | 42 +- sound/soc/au1x/psc.h | 16 +- sound/soc/codecs/sgtl5000.c | 8 + sound/soc/codecs/wm1250-ev1.c | 20 +- sound/soc/codecs/wm8523.c | 2 +- sound/soc/codecs/wm8580.c | 2 +- sound/soc/codecs/wm8711.c | 4 +- sound/soc/codecs/wm8728.c | 4 +- sound/soc/codecs/wm8731.c | 10 + sound/soc/codecs/wm8741.c | 126 +++- sound/soc/codecs/wm8753.c | 4 +- sound/soc/codecs/wm8993.c | 3 +- sound/soc/codecs/wm8994-tables.c | 4 +- sound/soc/codecs/wm8994.c | 6 +- sound/soc/codecs/wm_hubs.c | 32 +- sound/soc/codecs/wm_hubs.h | 3 +- sound/soc/mxs/Kconfig | 20 + sound/soc/mxs/Makefile | 10 + sound/soc/mxs/mxs-pcm.c | 359 +++++++++++ sound/soc/mxs/mxs-pcm.h | 43 ++ sound/soc/mxs/mxs-saif.c | 677 ++++++++++++++++++++ sound/soc/mxs/mxs-saif.h | 130 ++++ sound/soc/mxs/mxs-sgtl5000.c | 165 +++++ sound/soc/samsung/smdk_wm8580.c | 6 +- sound/soc/samsung/smdk_wm8580pcm.c | 4 +- sound/soc/samsung/speyside.c | 5 +- sound/soc/soc-core.c | 4 +- sound/soc/soc-dapm.c | 6 +- sound/soc/soc-io.c | 323 ++-------- 48 files changed, 3101 insertions(+), 494 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/soc/codecs/fsl-sgtl5000.txt create mode 100644 Documentation/devicetree/bindings/sound/wm8731.txt create mode 100644 sound/soc/au1x/ac97c.c create mode 100644 sound/soc/au1x/db1000.c create mode 100644 sound/soc/au1x/dma.c create mode 100644 sound/soc/au1x/i2sc.c create mode 100644 sound/soc/mxs/Kconfig create mode 100644 sound/soc/mxs/Makefile create mode 100644 sound/soc/mxs/mxs-pcm.c create mode 100644 sound/soc/mxs/mxs-pcm.h create mode 100644 sound/soc/mxs/mxs-saif.c create mode 100644 sound/soc/mxs/mxs-saif.h create mode 100644 sound/soc/mxs/mxs-sgtl5000.c

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[alsa-devel] ASoC updates for 3.1
by Mark Brown 12 Aug '11

12 Aug '11

The following changes since commit 8c285645ab3b05942124020b5f0b89d3b539823a: Merge branch 'wm8996-rename' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into fix/asoc (2011-08-08 10:45:31 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-3.1 So more changes ended up cooking in -next for the wm8750 than had been anticipated - it should've been a simple SPI driver ID addition but it got messed up for the modular x86_64 builds that I have difficultly in doing while travelling. The change from Sangbeom is another change percolating through after the merge window that I'd forgotten about until Kukjin reminded me. Jarkko Nikula (1): ASoC: omap: Update e-mail address of Jarkko Nikula Mark Brown (8): Merge branch 'for-3.1' of git://git.kernel.org/.../broonie/sound-2.6.git into for-3.1 Merge branch 'fix/asoc' of git://git.kernel.org/.../tiwai/sound-2.6 into for-3.1 ASoC: Fix binding of WM8750 on Jive ASoC: Fix SPI driver binding for WM8987 ASoC: Fix warning in Speyside WM8962 ASoC: Fix typo in wm8750 spi_ids ASoC: Add missing break in WM8994 probe ASoC: Terminate WM8750 SPI device ID table Sangbeom Kim (1): ASoC: SAMSUNG: Add I2S0 internal dma driver Stephen Warren (3): ASoC: Tegra: tegra_pcm_deallocate_dma_buffer: Don't OOPS ASoC: Tegra: wm8903 machine driver: Allow re-insertion of module ASoC: WM8903: Free IRQ on device removal MAINTAINERS | 2 +- include/sound/tlv320aic3x.h | 2 +- sound/soc/codecs/wm8750.c | 8 + sound/soc/codecs/wm8903.c | 5 + sound/soc/codecs/wm8994.c | 1 + sound/soc/omap/n810.c | 4 +- sound/soc/omap/omap-mcbsp.c | 4 +- sound/soc/omap/omap-mcbsp.h | 2 +- sound/soc/omap/omap-pcm.c | 4 +- sound/soc/omap/omap-pcm.h | 2 +- sound/soc/omap/rx51.c | 2 +- sound/soc/samsung/Makefile | 2 + sound/soc/samsung/idma.c | 453 +++++++++++++++++++++++++++++++++++ sound/soc/samsung/idma.h | 26 ++ sound/soc/samsung/jive_wm8750.c | 2 +- sound/soc/samsung/speyside_wm8962.c | 2 +- sound/soc/tegra/tegra_pcm.c | 9 +- sound/soc/tegra/tegra_wm8903.c | 15 +- 18 files changed, 526 insertions(+), 19 deletions(-) create mode 100644 sound/soc/samsung/idma.c create mode 100644 sound/soc/samsung/idma.h

2 1

[alsa-devel] MP3 files with ALSA
by Oscar Salvador 12 Aug '11

12 Aug '11

Hi, how are you. I started to read the alsa documentation a few days ago, because i want to see how works alsa internally, and because i want to write an audio application (just for fun). Well, with .wav files i don't have any problem, but i have a doubt with mp3 files. I know that i need to decode mp3 into wav, for example, for playing it with alsa, but my doubt is if alsa decode this internally, or if is my job doing it. In the last case, maybe you can give me a web that talking about this. Thanks a lot, Regards

2 1

Re: [alsa-devel] Allocating buffers for USB transfers (again)
by Daniel Mack 12 Aug '11

12 Aug '11

On Fri, Aug 12, 2011 at 6:46 AM, Sarah Sharp <sarah.a.sharp(a)linux.intel.com> wrote: > On Fri, Aug 12, 2011 at 03:28:42AM +0200, Daniel Mack wrote: >> On Fri, Aug 12, 2011 at 1:50 AM, Matěj Laitl <matej(a)laitl.cz> wrote: >> > On 12. 8. 2011 Daniel Mack wrote: >> >> > This patch actually makes the sound playback _worse_. Now I get strange >> >> > squawks where previously at least first seconds of a song sounded >> >> > normally. >> >> > >> >> > However, I no longer get "zero length buffer submitted" or that "... >> >> > Weird." debug messages, only several megabytes of: (should I post these >> >> > somewhere?) >> >> > >> >> > xhci_hcd 0000:05:00.0: Giveback URB ffff880114cec000, len = 880, >> >> > expected = 1000, status = -115 >> >> > xhci_hcd 0000:05:00.0: Giveback URB ffff880114740000, len = 352, >> >> > expected = 1000, status = -115 >> >> > xhci_hcd 0000:05:00.0: underrun event on endpoint >> >> >> >> Might be the hardware doesn't like this. As I said, the patch is blindly >> >> written and I couldn't test it. Did you test this on a EHCI port as well? >> > >> > Good point, this patch distorts the playback also on the EHCI port. (the >> > playback seems to be 10 times slower than it should be) >> >> And it is only this *last* patch that causes it, yes? >> >> If that is the case, we need empty subframes in the middle of urbs, >> and we need to figure a way how to teach the XHCI driver to cope with >> that, too. > > By empty subframes, you mean zero-length frame buffers in the middle of > the URB, correct? Yep. > Andiry and Alan had some suggestions for how to fix > that, so I'll try to cook up a patch tomorrow. Then we'll have to > "unpeel" the debug patches and make sure the system is still usable. Good :) And I'll make a patch to make double-submission impossible in a nice way. Daniel

1 0

[alsa-devel] Template Based Coefficients Setting for Filters
by Ashish Chavan 12 Aug '11

12 Aug '11

Hi, I am working on updating ASoC codec driver for Dialog's DA7210 codec (sound/soc/codec/da7210.c). This update would be a major functionality update in nature. Currently, I am implementing support for GP (general purpose) filters of this codec and need some opinions/comments. Brief info about GP filters: The additional general purpose (GP) filter engine offers up to eight 2nd order biquad filter stages with freely configurable 16 bit coefficients and supports flexible digital audio routing and mixing arrangements. This engine implements transfer function to derive Hz which uses five coefficients, a1, a1, a2, b1 and b2. The coefficients are specified as 16 bit twos-complement numbers. There are total eight blocks which can be interconnected as per requirement. Total number of registers to configure = (5 * 2 * 8) = 80 The GP filters are mainly designed to be used for following modes, (1) Stereo Playback (2) Stereo Capture (3) Capture with monitoring (4) Spatializer mode (5) Left-Sidetone filtering (6) Right-Sidetone filtering (7) Left-Right Sidetone filtering Each of this mode requires different set of coefficients to be programmed in to those 80 registers. My idea of implementation: I believe that it would be quite non practical(because of number of controls required to do that) to provide individual control to set each coefficient independently. What I am thinking is a virtual control named something like, "GP Filter Mode Selection". It would offer above seven options to user. On selection of any option, driver will populate that set of 80 registers with predefined coefficients. It would be like selecting one out of seven predefined templates. (Apart from the coefficients part, some interconnections between these blocks and IOs need to be done to achieve desired mode.) Is this a good/valid way to handle this? Is there any alternative? Thanks, -- Ashish

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[alsa-devel] [PATCH] ASoC: Add device tree binding for WM8753
by Mark Brown 12 Aug '11

12 Aug '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- Documentation/devicetree/bindings/sound/wm8753.txt | 18 ++++++++++++++++++ sound/soc/codecs/wm8753.c | 9 +++++++++ 2 files changed, 27 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/wm8753.txt diff --git a/Documentation/devicetree/bindings/sound/wm8753.txt b/Documentation/devicetree/bindings/sound/wm8753.txt new file mode 100644 index 0000000..e65277a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8753.txt @@ -0,0 +1,18 @@ +WM8753 audio CODEC + +This device supports both I2C and SPI (configured with pin strapping +on the board). + +Required properties: + + - compatible : "wlf,wm8753" + + - reg : the I2C address of the device for I2C, the chip select + number for SPI. + +Example: + +codec: wm8737@1a { + compatible = "wlf,wm8753"; + reg = <0x1a>; +}; diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index a702550..fe04a10 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -38,6 +38,7 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> +#include <linux/of_device.h> #include <linux/platform_device.h> #include <linux/spi/spi.h> #include <linux/slab.h> @@ -1490,6 +1491,12 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8753 = { .reg_cache_default = wm8753_reg, }; +static const struct of_device_id wm8753_of_match[] = { + { .compatible = "wlf,wm8753", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8753_of_match); + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8753_spi_probe(struct spi_device *spi) { @@ -1521,6 +1528,7 @@ static struct spi_driver wm8753_spi_driver = { .driver = { .name = "wm8753", .owner = THIS_MODULE, + .of_match_table = wm8753_of_match, }, .probe = wm8753_spi_probe, .remove = __devexit_p(wm8753_spi_remove), @@ -1565,6 +1573,7 @@ static struct i2c_driver wm8753_i2c_driver = { .driver = { .name = "wm8753", .owner = THIS_MODULE, + .of_match_table = wm8753_of_match, }, .probe = wm8753_i2c_probe, .remove = __devexit_p(wm8753_i2c_remove), -- 1.7.5.4

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Re: [alsa-devel] [uclinux-dist-devel] [PATCH 3.0]ASoC: ad193x: add spi_hw_read, fix sysclk and register definition
by Greg KH 12 Aug '11

12 Aug '11

On Thu, Aug 11, 2011 at 04:52:18AM -0400, Zhang, Sonic wrote: > > > >-----Original Message----- > >From: uclinux-dist-devel-bounces(a)blackfin.uclinux.org > >[mailto:uclinux-dist-devel-bounces@blackfin.uclinux.org] On > >Behalf Of Scott Jiang > >Sent: Thursday, August 11, 2011 4:27 PM > >To: Greg Kroah-Hartman > >Cc: uclinux-dist-devel(a)blackfin.uclinux.org; > >alsa-devel(a)alsa-project.org; stable(a)kernel.org > >Subject: [uclinux-dist-devel] [alsa-devel][PATCH 3.0]ASoC: > >ad193x: add spi_hw_read, fix sysclk and register definition > > > >asoc cache layer can't support this kind of spi registers, so > >bypass cache and read regiters directly > > > > This patch is against stable kernel 3.0 only, the ASOC framework in kernel 3.1 and later are different. Then I need some acks from all of the relevant maintainers before I can apply it to the stable kernel tree. Is there a corrisponding patch in Linus's tree that does the same thing already? greg k-h

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[alsa-devel] [PATCH 1/2] regmap: Specify a module license
by Stephen Warren 12 Aug '11

12 Aug '11

CONFIG_REGMAP_I2C/SPI are set to m when selected by a tristate config option that's set to m. The regmap modules don't specify a license, so fail to link to regmap_init at load time, since that is EXPORT_SYMBOL_GPL. Fix this by specifying a license for the regmap modules. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- drivers/base/regmap/regmap-i2c.c | 1 + drivers/base/regmap/regmap-spi.c | 2 ++ 2 files changed, 3 insertions(+), 0 deletions(-) diff --git a/drivers/base/regmap/regmap-i2c.c b/drivers/base/regmap/regmap-i2c.c index e6ce82d..e7d916d 100644 --- a/drivers/base/regmap/regmap-i2c.c +++ b/drivers/base/regmap/regmap-i2c.c @@ -112,3 +112,4 @@ struct regmap *regmap_init_i2c(struct i2c_client *i2c, } EXPORT_SYMBOL_GPL(regmap_init_i2c); +MODULE_LICENSE("GPL"); diff --git a/drivers/base/regmap/regmap-spi.c b/drivers/base/regmap/regmap-spi.c index 07633bd..cc0f116 100644 --- a/drivers/base/regmap/regmap-spi.c +++ b/drivers/base/regmap/regmap-spi.c @@ -69,3 +69,5 @@ struct regmap *regmap_init_spi(struct spi_device *spi, return regmap_init(&spi->dev, &regmap_spi, config); } EXPORT_SYMBOL_GPL(regmap_init_spi); + +MODULE_LICENSE("GPL"); -- 1.7.0.4

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[alsa-devel] [PATCH] ASoC: omap: Update e-mail address of Jarkko Nikula
by Jarkko Nikula 12 Aug '11

12 Aug '11

My gmail account got disabled and I'm not going to reopen it. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> --- Be careful about your age in google+ registration... --- MAINTAINERS | 2 +- include/sound/tlv320aic3x.h | 2 +- sound/soc/omap/n810.c | 4 ++-- sound/soc/omap/omap-mcbsp.c | 4 ++-- sound/soc/omap/omap-mcbsp.h | 2 +- sound/soc/omap/omap-pcm.c | 4 ++-- sound/soc/omap/omap-pcm.h | 2 +- sound/soc/omap/rx51.c | 2 +- 8 files changed, 11 insertions(+), 11 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 1f8267f..3e6b34c 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -4604,7 +4604,7 @@ F: arch/arm/mach-omap2/clockdomain2xxx_3xxx.c F: arch/arm/mach-omap2/clockdomain44xx.c OMAP AUDIO SUPPORT -M: Jarkko Nikula <jhnikula(a)gmail.com> +M: Jarkko Nikula <jarkko.nikula(a)bitmer.com> L: alsa-devel(a)alsa-project.org (subscribers-only) L: linux-omap(a)vger.kernel.org S: Maintained diff --git a/include/sound/tlv320aic3x.h b/include/sound/tlv320aic3x.h index 99e0308..ffd9bc7 100644 --- a/include/sound/tlv320aic3x.h +++ b/include/sound/tlv320aic3x.h @@ -1,7 +1,7 @@ /* * Platform data for Texas Instruments TLV320AIC3x codec * - * Author: Jarkko Nikula <jhnikula(a)gmail.com> + * Author: Jarkko Nikula <jarkko.nikula(a)bitmer.com> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 83d213b..62e292f 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -3,7 +3,7 @@ * * Copyright (C) 2008 Nokia Corporation * - * Contact: Jarkko Nikula <jhnikula(a)gmail.com> + * Contact: Jarkko Nikula <jarkko.nikula(a)bitmer.com> * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License @@ -402,6 +402,6 @@ static void __exit n810_soc_exit(void) module_init(n810_soc_init); module_exit(n810_soc_exit); -MODULE_AUTHOR("Jarkko Nikula <jhnikula(a)gmail.com>"); +MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula(a)bitmer.com>"); MODULE_DESCRIPTION("ALSA SoC Nokia N810"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 07b7723..ebcc2d4 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -3,7 +3,7 @@ * * Copyright (C) 2008 Nokia Corporation * - * Contact: Jarkko Nikula <jhnikula(a)gmail.com> + * Contact: Jarkko Nikula <jarkko.nikula(a)bitmer.com> * Peter Ujfalusi <peter.ujfalusi(a)ti.com> * * This program is free software; you can redistribute it and/or @@ -780,6 +780,6 @@ static void __exit snd_omap_mcbsp_exit(void) } module_exit(snd_omap_mcbsp_exit); -MODULE_AUTHOR("Jarkko Nikula <jhnikula(a)gmail.com>"); +MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula(a)bitmer.com>"); MODULE_DESCRIPTION("OMAP I2S SoC Interface"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h index 9a7dedd..65cde9d 100644 --- a/sound/soc/omap/omap-mcbsp.h +++ b/sound/soc/omap/omap-mcbsp.h @@ -3,7 +3,7 @@ * * Copyright (C) 2008 Nokia Corporation * - * Contact: Jarkko Nikula <jhnikula(a)gmail.com> + * Contact: Jarkko Nikula <jarkko.nikula(a)bitmer.com> * Peter Ujfalusi <peter.ujfalusi(a)ti.com> * * This program is free software; you can redistribute it and/or diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index b2f5751..9b5c88a 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -3,7 +3,7 @@ * * Copyright (C) 2008 Nokia Corporation * - * Contact: Jarkko Nikula <jhnikula(a)gmail.com> + * Contact: Jarkko Nikula <jarkko.nikula(a)bitmer.com> * Peter Ujfalusi <peter.ujfalusi(a)ti.com> * * This program is free software; you can redistribute it and/or @@ -436,6 +436,6 @@ static void __exit snd_omap_pcm_exit(void) } module_exit(snd_omap_pcm_exit); -MODULE_AUTHOR("Jarkko Nikula <jhnikula(a)gmail.com>"); +MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula(a)bitmer.com>"); MODULE_DESCRIPTION("OMAP PCM DMA module"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap-pcm.h b/sound/soc/omap/omap-pcm.h index a0ed1db..f95fe30 100644 --- a/sound/soc/omap/omap-pcm.h +++ b/sound/soc/omap/omap-pcm.h @@ -3,7 +3,7 @@ * * Copyright (C) 2008 Nokia Corporation * - * Contact: Jarkko Nikula <jhnikula(a)gmail.com> + * Contact: Jarkko Nikula <jarkko.nikula(a)bitmer.com> * Peter Ujfalusi <peter.ujfalusi(a)ti.com> * * This program is free software; you can redistribute it and/or diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 0aae998..893300a 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -5,7 +5,7 @@ * * Contact: Peter Ujfalusi <peter.ujfalusi(a)ti.com> * Eduardo Valentin <eduardo.valentin(a)nokia.com> - * Jarkko Nikula <jhnikula(a)gmail.com> + * Jarkko Nikula <jarkko.nikula(a)bitmer.com> * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License -- 1.7.2.5

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Re: [alsa-devel] [uclinux-dist-devel] [PATCH 3.0]ASoC: ad193x: add spi_hw_read, fix sysclk and register definition
by Barry Song 12 Aug '11

12 Aug '11

>>Subject: [uclinux-dist-devel] [alsa-devel][PATCH 3.0]ASoC: >>ad193x: add spi_hw_read, fix sysclk and register definition >> >>asoc cache layer can't support this kind of spi registers, so >>bypass cache and read regiters directly >> > > This patch is against stable kernel 3.0 only, the ASOC framework in kernel 3.1 and later are different. this can't be one patch, but be n patches since it is doing n different work. it is really difficult to understand what the patch is really doing. > > Sonic > >>Signed-off-by: Scott Jiang <scott.jiang.linux(a)gmail.com> -barry

1 0

Re: [alsa-devel] [PATCH] ASoC: sgtl5000: fix module device table type for sgtl5000_dt_ids
by Mark Brown 12 Aug '11

12 Aug '11

On Thu, Aug 11, 2011 at 10:19:16PM +0800, Axel Lin wrote: > The module device table for of_device_id should use "of" type. Applied, thanks.

1 0

[alsa-devel] [PATCH 3.1]ASoC: ad193x: add spi_hw_read, fix sysclk and register definition
by Scott Jiang 12 Aug '11

12 Aug '11

asoc cache layer can't support this kind of spi registers, so bypass cache and read regiters directly Signed-off-by: Scott Jiang <scott.jiang.linux(a)gmail.com> --- sound/soc/blackfin/bf5xx-ad193x.c | 2 +- sound/soc/codecs/ad193x.c | 8 ++++++-- sound/soc/codecs/ad193x.h | 5 +++-- sound/soc/soc-io.c | 23 +++++++++++++++++++++++ 4 files changed, 33 insertions(+), 5 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ad193x.c b/sound/soc/blackfin/bf5xx-ad193x.c index d6651c0..a118a0f 100644 --- a/sound/soc/blackfin/bf5xx-ad193x.c +++ b/sound/soc/blackfin/bf5xx-ad193x.c @@ -56,7 +56,7 @@ static int bf5xx_ad193x_hw_params(struct snd_pcm_substream *substream, switch (params_rate(params)) { case 48000: - clk = 12288000; + clk = 24576000; break; } diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 2374ca5..1447cc8 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -307,7 +307,8 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream, snd_soc_write(codec, AD193X_PLL_CLK_CTRL0, reg); reg = snd_soc_read(codec, AD193X_DAC_CTRL2); - reg = (reg & (~AD193X_DAC_WORD_LEN_MASK)) | word_len; + reg = (reg & (~AD193X_DAC_WORD_LEN_MASK)) + | (word_len << AD193X_DAC_WORD_LEN_SHFT); snd_soc_write(codec, AD193X_DAC_CTRL2, reg); reg = snd_soc_read(codec, AD193X_ADC_CTRL1); @@ -361,7 +362,10 @@ static int ad193x_probe(struct snd_soc_codec *codec) dev_err(codec->dev, "failed to set cache I/O: %d\n", ret); return ret; } - +#if defined(CONFIG_SPI_MASTER) + /* asoc cache layer can't support this kind of spi registers now */ + codec->cache_bypass = 1; +#endif /* default setting for ad193x */ /* unmute dac channels */ diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h index 9747b54..cccc2e8 100644 --- a/sound/soc/codecs/ad193x.h +++ b/sound/soc/codecs/ad193x.h @@ -34,7 +34,8 @@ #define AD193X_DAC_LEFT_HIGH (1 << 3) #define AD193X_DAC_BCLK_INV (1 << 7) #define AD193X_DAC_CTRL2 0x804 -#define AD193X_DAC_WORD_LEN_MASK 0xC +#define AD193X_DAC_WORD_LEN_SHFT 3 +#define AD193X_DAC_WORD_LEN_MASK 0x18 #define AD193X_DAC_MASTER_MUTE 1 #define AD193X_DAC_CHNL_MUTE 0x805 #define AD193X_DACL1_MUTE 0 @@ -63,7 +64,7 @@ #define AD193X_ADC_CTRL1 0x80f #define AD193X_ADC_SERFMT_MASK 0x60 #define AD193X_ADC_SERFMT_STEREO (0 << 5) -#define AD193X_ADC_SERFMT_TDM (1 << 2) +#define AD193X_ADC_SERFMT_TDM (1 << 5) #define AD193X_ADC_SERFMT_AUX (2 << 5) #define AD193X_ADC_WORD_LEN_MASK 0x3 #define AD193X_ADC_CTRL2 0x810 diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index cca490c..a62f7dd 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -205,6 +205,25 @@ static unsigned int snd_soc_16_8_read_i2c(struct snd_soc_codec *codec, #define snd_soc_16_8_read_i2c NULL #endif +#if defined(CONFIG_SPI_MASTER) +static unsigned int snd_soc_16_8_read_spi(struct snd_soc_codec *codec, + unsigned int r) +{ + struct spi_device *spi = codec->control_data; + + const u16 reg = cpu_to_be16(r | 0x100); + u8 data; + int ret; + + ret = spi_write_then_read(spi, &reg, 2, &data, 1); + if (ret < 0) + return 0; + return data; +} +#else +#define snd_soc_16_8_read_spi NULL +#endif + static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { @@ -295,6 +314,7 @@ static struct { int (*write)(struct snd_soc_codec *codec, unsigned int, unsigned int); unsigned int (*read)(struct snd_soc_codec *, unsigned int); unsigned int (*i2c_read)(struct snd_soc_codec *, unsigned int); + unsigned int (*spi_read)(struct snd_soc_codec *, unsigned int); } io_types[] = { { .addr_bits = 4, .data_bits = 12, @@ -318,6 +338,7 @@ static struct { .addr_bits = 16, .data_bits = 8, .write = snd_soc_16_8_write, .i2c_read = snd_soc_16_8_read_i2c, + .spi_read = snd_soc_16_8_read_spi, }, { .addr_bits = 16, .data_bits = 16, @@ -383,6 +404,8 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, #ifdef CONFIG_SPI_MASTER codec->hw_write = do_spi_write; #endif + if (io_types[i].spi_read) + codec->hw_read = io_types[i].spi_read; codec->control_data = container_of(codec->dev, struct spi_device, -- 1.7.0.4

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Re: [alsa-devel] Allocating buffers for USB transfers (again)
by Daniel Mack 12 Aug '11

12 Aug '11

On Fri, Aug 12, 2011 at 1:50 AM, Matěj Laitl <matej(a)laitl.cz> wrote: > On 12. 8. 2011 Daniel Mack wrote: >> > This patch actually makes the sound playback _worse_. Now I get strange >> > squawks where previously at least first seconds of a song sounded >> > normally. >> > >> > However, I no longer get "zero length buffer submitted" or that "... >> > Weird." debug messages, only several megabytes of: (should I post these >> > somewhere?) >> > >> > xhci_hcd 0000:05:00.0: Giveback URB ffff880114cec000, len = 880, >> > expected = 1000, status = -115 >> > xhci_hcd 0000:05:00.0: Giveback URB ffff880114740000, len = 352, >> > expected = 1000, status = -115 >> > xhci_hcd 0000:05:00.0: underrun event on endpoint >> >> Might be the hardware doesn't like this. As I said, the patch is blindly >> written and I couldn't test it. Did you test this on a EHCI port as well? > > Good point, this patch distorts the playback also on the EHCI port. (the > playback seems to be 10 times slower than it should be) And it is only this *last* patch that causes it, yes? If that is the case, we need empty subframes in the middle of urbs, and we need to figure a way how to teach the XHCI driver to cope with that, too. Daniel

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[alsa-devel] [PATCH] ASoC: Revert "ASoC: SAMSUNG: Add I2S0 internal dma driver"
by Mark Brown 12 Aug '11

12 Aug '11

This reverts commit d7c3e9525ac8e898f1156a1f3a7c5038f6560186 as it does not currently build due to missing dependencies in the Samsung tree. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/Makefile | 2 - sound/soc/samsung/idma.c | 453 -------------------------------------------- sound/soc/samsung/idma.h | 26 --- 3 files changed, 0 insertions(+), 481 deletions(-) delete mode 100644 sound/soc/samsung/idma.c delete mode 100644 sound/soc/samsung/idma.h diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 8509d3c..9eb3b12 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -1,6 +1,5 @@ # S3c24XX Platform Support snd-soc-s3c24xx-objs := dma.o -snd-soc-idma-objs := idma.o snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o snd-soc-ac97-objs := ac97.o @@ -17,7 +16,6 @@ obj-$(CONFIG_SND_S3C_I2SV2_SOC) += snd-soc-s3c-i2s-v2.o obj-$(CONFIG_SND_SAMSUNG_SPDIF) += snd-soc-samsung-spdif.o obj-$(CONFIG_SND_SAMSUNG_PCM) += snd-soc-pcm.o obj-$(CONFIG_SND_SAMSUNG_I2S) += snd-soc-i2s.o -obj-$(CONFIG_SND_SAMSUNG_I2S) += snd-soc-idma.o # S3C24XX Machine Support snd-soc-jive-wm8750-objs := jive_wm8750.o diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c deleted file mode 100644 index ebde074..0000000 --- a/sound/soc/samsung/idma.c +++ /dev/null @@ -1,453 +0,0 @@ -/* - * sound/soc/samsung/idma.c - * - * Copyright (c) 2011 Samsung Electronics Co., Ltd. - * http://www.samsung.com - * - * I2S0's Internal DMA driver - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ -#include <linux/interrupt.h> -#include <linux/platform_device.h> -#include <linux/dma-mapping.h> -#include <linux/slab.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> - -#include "i2s.h" -#include "idma.h" -#include "dma.h" -#include "i2s-regs.h" - -#define ST_RUNNING (1<<0) -#define ST_OPENED (1<<1) - -static const struct snd_pcm_hardware idma_hardware = { - .info = SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME, - .formats = SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_U16_LE | - SNDRV_PCM_FMTBIT_S24_LE | - SNDRV_PCM_FMTBIT_U24_LE | - SNDRV_PCM_FMTBIT_U8 | - SNDRV_PCM_FMTBIT_S8, - .channels_min = 2, - .channels_max = 2, - .buffer_bytes_max = MAX_IDMA_BUFFER, - .period_bytes_min = 128, - .period_bytes_max = MAX_IDMA_PERIOD, - .periods_min = 1, - .periods_max = 2, -}; - -struct idma_ctrl { - spinlock_t lock; - int state; - dma_addr_t start; - dma_addr_t pos; - dma_addr_t end; - dma_addr_t period; - dma_addr_t periodsz; - void *token; - void (*cb)(void *dt, int bytes_xfer); -}; - -static struct idma_info { - spinlock_t lock; - void __iomem *regs; - dma_addr_t lp_tx_addr; -} idma; - -static void idma_getpos(dma_addr_t *src) -{ - *src = idma.lp_tx_addr + - (readl(idma.regs + I2STRNCNT) & 0xffffff) * 4; -} - -static int idma_enqueue(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct idma_ctrl *prtd = substream->runtime->private_data; - u32 val; - - spin_lock(&prtd->lock); - prtd->token = (void *) substream; - spin_unlock(&prtd->lock); - - /* Internal DMA Level0 Interrupt Address */ - val = idma.lp_tx_addr + prtd->periodsz; - writel(val, idma.regs + I2SLVL0ADDR); - - /* Start address0 of I2S internal DMA operation. */ - val = idma.lp_tx_addr; - writel(val, idma.regs + I2SSTR0); - - /* - * Transfer block size for I2S internal DMA. - * Should decide transfer size before start dma operation - */ - val = readl(idma.regs + I2SSIZE); - val &= ~(I2SSIZE_TRNMSK << I2SSIZE_SHIFT); - val |= (((runtime->dma_bytes >> 2) & - I2SSIZE_TRNMSK) << I2SSIZE_SHIFT); - writel(val, idma.regs + I2SSIZE); - - val = readl(idma.regs + I2SAHB); - val |= AHB_INTENLVL0; - writel(val, idma.regs + I2SAHB); - - return 0; -} - -static void idma_setcallbk(struct snd_pcm_substream *substream, - void (*cb)(void *, int)) -{ - struct idma_ctrl *prtd = substream->runtime->private_data; - - spin_lock(&prtd->lock); - prtd->cb = cb; - spin_unlock(&prtd->lock); -} - -static void idma_control(int op) -{ - u32 val = readl(idma.regs + I2SAHB); - - spin_lock(&idma.lock); - - switch (op) { - case LPAM_DMA_START: - val |= (AHB_INTENLVL0 | AHB_DMAEN); - break; - case LPAM_DMA_STOP: - val &= ~(AHB_INTENLVL0 | AHB_DMAEN); - break; - default: - spin_unlock(&idma.lock); - return; - } - - writel(val, idma.regs + I2SAHB); - spin_unlock(&idma.lock); -} - -static void idma_done(void *id, int bytes_xfer) -{ - struct snd_pcm_substream *substream = id; - struct idma_ctrl *prtd = substream->runtime->private_data; - - if (prtd && (prtd->state & ST_RUNNING)) - snd_pcm_period_elapsed(substream); -} - -static int idma_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct idma_ctrl *prtd = substream->runtime->private_data; - u32 mod = readl(idma.regs + I2SMOD); - u32 ahb = readl(idma.regs + I2SAHB); - - ahb |= (AHB_DMARLD | AHB_INTMASK); - mod |= MOD_TXS_IDMA; - writel(ahb, idma.regs + I2SAHB); - writel(mod, idma.regs + I2SMOD); - - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - runtime->dma_bytes = params_buffer_bytes(params); - - prtd->start = prtd->pos = runtime->dma_addr; - prtd->period = params_periods(params); - prtd->periodsz = params_period_bytes(params); - prtd->end = runtime->dma_addr + runtime->dma_bytes; - - idma_setcallbk(substream, idma_done); - - return 0; -} - -static int idma_hw_free(struct snd_pcm_substream *substream) -{ - snd_pcm_set_runtime_buffer(substream, NULL); - - return 0; -} - -static int idma_prepare(struct snd_pcm_substream *substream) -{ - struct idma_ctrl *prtd = substream->runtime->private_data; - - prtd->pos = prtd->start; - - /* flush the DMA channel */ - idma_control(LPAM_DMA_STOP); - idma_enqueue(substream); - - return 0; -} - -static int idma_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct idma_ctrl *prtd = substream->runtime->private_data; - int ret = 0; - - spin_lock(&prtd->lock); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - prtd->state |= ST_RUNNING; - idma_control(LPAM_DMA_START); - break; - - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - prtd->state &= ~ST_RUNNING; - idma_control(LPAM_DMA_STOP); - break; - - default: - ret = -EINVAL; - break; - } - - spin_unlock(&prtd->lock); - - return ret; -} - -static snd_pcm_uframes_t - idma_pointer(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct idma_ctrl *prtd = runtime->private_data; - dma_addr_t src; - unsigned long res; - - spin_lock(&prtd->lock); - - idma_getpos(&src); - res = src - prtd->start; - - spin_unlock(&prtd->lock); - - return bytes_to_frames(substream->runtime, res); -} - -static int idma_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - unsigned long size, offset; - int ret; - - /* From snd_pcm_lib_mmap_iomem */ - vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot); - vma->vm_flags |= VM_IO; - size = vma->vm_end - vma->vm_start; - offset = vma->vm_pgoff << PAGE_SHIFT; - ret = io_remap_pfn_range(vma, vma->vm_start, - (runtime->dma_addr + offset) >> PAGE_SHIFT, - size, vma->vm_page_prot); - - return ret; -} - -static irqreturn_t iis_irq(int irqno, void *dev_id) -{ - struct idma_ctrl *prtd = (struct idma_ctrl *)dev_id; - u32 iiscon, iisahb, val, addr; - - iisahb = readl(idma.regs + I2SAHB); - iiscon = readl(idma.regs + I2SCON); - - val = (iisahb & AHB_LVL0INT) ? AHB_CLRLVL0INT : 0; - - if (val) { - iisahb |= val; - writel(iisahb, idma.regs + I2SAHB); - - addr = readl(idma.regs + I2SLVL0ADDR) - idma.lp_tx_addr; - addr += prtd->periodsz; - addr %= (prtd->end - prtd->start); - addr += idma.lp_tx_addr; - - writel(addr, idma.regs + I2SLVL0ADDR); - - if (prtd->cb) - prtd->cb(prtd->token, prtd->period); - } - - return IRQ_HANDLED; -} - -static int idma_open(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct idma_ctrl *prtd; - int ret; - - snd_soc_set_runtime_hwparams(substream, &idma_hardware); - - prtd = kzalloc(sizeof(struct idma_ctrl), GFP_KERNEL); - if (prtd == NULL) - return -ENOMEM; - - ret = request_irq(IRQ_I2S0, iis_irq, 0, "i2s", prtd); - if (ret < 0) { - pr_err("fail to claim i2s irq , ret = %d\n", ret); - kfree(prtd); - return ret; - } - - spin_lock_init(&prtd->lock); - - runtime->private_data = prtd; - - return 0; -} - -static int idma_close(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct idma_ctrl *prtd = runtime->private_data; - - free_irq(IRQ_I2S0, prtd); - - if (!prtd) - pr_err("idma_close called with prtd == NULL\n"); - - kfree(prtd); - - return 0; -} - -static struct snd_pcm_ops idma_ops = { - .open = idma_open, - .close = idma_close, - .ioctl = snd_pcm_lib_ioctl, - .trigger = idma_trigger, - .pointer = idma_pointer, - .mmap = idma_mmap, - .hw_params = idma_hw_params, - .hw_free = idma_hw_free, - .prepare = idma_prepare, -}; - -static void idma_free(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - - substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; - if (!substream) - return; - - buf = &substream->dma_buffer; - if (!buf->area) - return; - - iounmap(buf->area); - - buf->area = NULL; - buf->addr = 0; -} - -static int preallocate_idma_buffer(struct snd_pcm *pcm, int stream) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - - /* Assign PCM buffer pointers */ - buf->dev.type = SNDRV_DMA_TYPE_CONTINUOUS; - buf->addr = idma.lp_tx_addr; - buf->bytes = idma_hardware.buffer_bytes_max; - buf->area = (unsigned char *)ioremap(buf->addr, buf->bytes); - - return 0; -} - -static u64 idma_mask = DMA_BIT_MASK(32); - -static int idma_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_card *card = rtd->card->snd_card; - struct snd_soc_dai *dai = rtd->cpu_dai; - struct snd_pcm *pcm = rtd->pcm; - int ret = 0; - - if (!card->dev->dma_mask) - card->dev->dma_mask = &idma_mask; - if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = DMA_BIT_MASK(32); - - if (dai->driver->playback.channels_min) - ret = preallocate_idma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); - - return ret; -} - -void idma_reg_addr_init(void *regs, dma_addr_t addr) -{ - spin_lock_init(&idma.lock); - idma.regs = regs; - idma.lp_tx_addr = addr; -} - -struct snd_soc_platform_driver asoc_idma_platform = { - .ops = &idma_ops, - .pcm_new = idma_new, - .pcm_free = idma_free, -}; - -static int __devinit asoc_idma_platform_probe(struct platform_device *pdev) -{ - return snd_soc_register_platform(&pdev->dev, &asoc_idma_platform); -} - -static int __devexit asoc_idma_platform_remove(struct platform_device *pdev) -{ - snd_soc_unregister_platform(&pdev->dev); - return 0; -} - -static struct platform_driver asoc_idma_driver = { - .driver = { - .name = "samsung-idma", - .owner = THIS_MODULE, - }, - - .probe = asoc_idma_platform_probe, - .remove = __devexit_p(asoc_idma_platform_remove), -}; - -static int __init asoc_idma_init(void) -{ - return platform_driver_register(&asoc_idma_driver); -} -module_init(asoc_idma_init); - -static void __exit asoc_idma_exit(void) -{ - platform_driver_unregister(&asoc_idma_driver); -} -module_exit(asoc_idma_exit); - -MODULE_AUTHOR("Jaswinder Singh, <jassisinghbrar(a)gmail.com>"); -MODULE_DESCRIPTION("Samsung ASoC IDMA Driver"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/idma.h b/sound/soc/samsung/idma.h deleted file mode 100644 index 4827321..0000000 --- a/sound/soc/samsung/idma.h +++ /dev/null @@ -1,26 +0,0 @@ -/* - * sound/soc/samsung/idma.h - * - * Copyright (c) 2011 Samsung Electronics Co., Ltd - * http://www.samsung.com - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - */ - -#ifndef __SND_SOC_SAMSUNG_IDMA_H_ -#define __SND_SOC_SAMSUNG_IDMA_H_ - -extern void idma_reg_addr_init(void *regs, dma_addr_t addr); - -/* dma_state */ -#define LPAM_DMA_STOP 0 -#define LPAM_DMA_START 1 - -#define MAX_IDMA_PERIOD (128 * 1024) -#define MAX_IDMA_BUFFER (160 * 1024) - -#endif /* __SND_SOC_SAMSUNG_IDMA_H_ */ -- 1.7.5.4

5 7

Re: [alsa-devel] linux-next: Tree for Aug 11 (sound/soc + regmap)
by Randy Dunlap 12 Aug '11

12 Aug '11

On Thu, 11 Aug 2011 16:20:58 +1000 Stephen Rothwell wrote: > Hi all, When CONFIG_SPI is not enabled (so REGMAP_SPI is not enabled): (on i386 and x86_64) sound/built-in.o: In function `snd_soc_codec_set_cache_io': (.text+0x85635): undefined reference to `regmap_init_spi' --- ~Randy *** Remember to use Documentation/SubmitChecklist when testing your code ***

2 1

Re: [alsa-devel] Allocating buffers for USB transfers (again)
by Daniel Mack 12 Aug '11

12 Aug '11

On 08/12/2011 01:29 AM, Matěj Laitl wrote: > On 11. 8. 2011 Daniel Mack wrote: >> On Thu, Aug 11, 2011 at 8:05 PM, Sarah Sharp >>> Because according to Matej, he applied that patch, plus my patch to >>> reject zero-length buffers[1], and he saw debugging that indicated he >>> *did* see zero-length buffers. Is there any chance your driver might >>> submit a zero-length buffer in the middle of the isochronous URB >>> transfer array? >> >> Hmm, judging from the code, this can only ever happen if we receive an >> inbound iso frame which has a valid status and an actual_length of >> zero. Also, it was not neccessary to catch this case for EHCI. >> >> Maetj, does this patch make any difference? > > This patch actually makes the sound playback _worse_. Now I get strange > squawks where previously at least first seconds of a song sounded normally. > > However, I no longer get "zero length buffer submitted" or that "... Weird." > debug messages, only several megabytes of: (should I post these somewhere?) > > xhci_hcd 0000:05:00.0: Giveback URB ffff880114cec000, len = 880, expected = > 1000, status = -115 > xhci_hcd 0000:05:00.0: Giveback URB ffff880114740000, len = 352, expected = > 1000, status = -115 > xhci_hcd 0000:05:00.0: underrun event on endpoint Might be the hardware doesn't like this. As I said, the patch is blindly written and I couldn't test it. Did you test this on a EHCI port as well? Daniel

1 0

Re: [alsa-devel] Allocating buffers for USB transfers (again)
by Daniel Mack 12 Aug '11

12 Aug '11

On Thu, Aug 11, 2011 at 8:05 PM, Sarah Sharp <sarah.a.sharp(a)linux.intel.com> wrote: > On Thu, Aug 11, 2011 at 07:27:15PM +0200, Daniel Mack wrote: >> On Thu, Aug 11, 2011 at 6:45 PM, Sarah Sharp >> <sarah.a.sharp(a)linux.intel.com> wrote: >> > On Thu, Aug 11, 2011 at 02:57:41AM +0200, Daniel Mack wrote: >> >> On Thu, Aug 11, 2011 at 1:15 AM, Sarah Sharp >> >> <sarah.a.sharp(a)linux.intel.com> wrote: >> >> > On Wed, Aug 10, 2011 at 05:33:02PM +0200, Daniel Mack wrote: >> >> >> On 08/10/2011 04:32 PM, Alan Stern wrote: >> >> >> >Looking at the driver's current code, it appears that your patch >> >> >> >does not fix the bug properly. Using discontiguous regions in the >> >> >> >transfer buffer is perfectly okay. The real problem is later on, >> >> >> >where you do: >> >> >> > >> >> >> >if (send_it) { out->number_of_packets = FRAMES_PER_URB; >> >> >> > >> >> >> >This should be >> >> >> > >> >> >> >out->number_of_packets = outframe; >> >> >> > >> >> >> >The way it is now, the USB stack will try to use data from all the >> >> >> >frame descriptors, and the last few will be stale because the loop >> >> >> >doesn't set them. >> >> >> >> >> >> That's actually true, even though it doesn't seem to cause any trouble. >> >> >> I tested everything here of course, and the output URBs return back from >> >> >> the USB stack with their length fields zeroed out, which then >> >> >> causes the stack to send packets with zero-length fields at the end. >> >> > >> >> > Actually, it causes system hangs when the driver is loaded on a device >> >> > attached to a USB 3.0 port, as Alan Stern pointed out: >> >> > >> >> > https://bugzilla.kernel.org/show_bug.cgi?id=40702 >> >> >> >> Yes, I've noticed this. >> >> >> >> > Please don't submit zero-length transfers. The xHCI driver just isn't >> >> > able to handle it. Arguably, it probably should have just rejected your >> >> > URB when it found a zero length buffer, so I'll probably be submitting a >> >> > patch to fix that. >> >> >> >> According to the spec, sending zero-length frames should be fine, no? >> >> Is there any particular reason why XCHI can't handle this while EHCI >> >> can? And does my patch fix the driver for XHCI? >> > >> > Ok, yes, you're correct that the xHCI spec allows the transfer length to >> > be set to zero. In the case where the frame buffer is zero-length, is >> > the buffer pointer still valid? It's not clear from the spec whether it >> > needs to be. >> >> Well, the buffer pointer is set in the URB, not in its individual iso >> subframes which just denotes them via the offset field. So yes, it is >> valid in my case. But it doesn't matter anymore, as the code which >> does that is now gone :) > > Do you mean it was removed with this patch: > > http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=commitdiff;… > > Because according to Matej, he applied that patch, plus my patch to > reject zero-length buffers[1], and he saw debugging that indicated he > *did* see zero-length buffers. Is there any chance your driver might > submit a zero-length buffer in the middle of the isochronous URB > transfer array? Hmm, judging from the code, this can only ever happen if we receive an inbound iso frame which has a valid status and an actual_length of zero. Also, it was not neccessary to catch this case for EHCI. Maetj, does this patch make any difference? Daniel diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index aa52b3e..b48adb9 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -633,6 +634,10 @@ static void read_completed(struct urb *urb) continue; len = urb->iso_frame_desc[outframe].actual_length; + + if (len == 0) + continue; + out->iso_frame_desc[outframe].length = len; out->iso_frame_desc[outframe].actual_length = 0; out->iso_frame_desc[outframe].offset = offset;

1 0

Re: [alsa-devel] Allocating buffers for USB transfers (again)
by Daniel Mack 11 Aug '11

11 Aug '11

On Thu, Aug 11, 2011 at 6:45 PM, Sarah Sharp <sarah.a.sharp(a)linux.intel.com> wrote: > On Thu, Aug 11, 2011 at 02:57:41AM +0200, Daniel Mack wrote: >> On Thu, Aug 11, 2011 at 1:15 AM, Sarah Sharp >> <sarah.a.sharp(a)linux.intel.com> wrote: >> > On Wed, Aug 10, 2011 at 05:33:02PM +0200, Daniel Mack wrote: >> >> On 08/10/2011 04:32 PM, Alan Stern wrote: >> >> >Looking at the driver's current code, it appears that your patch >> >> >does not fix the bug properly. Using discontiguous regions in the >> >> >transfer buffer is perfectly okay. The real problem is later on, >> >> >where you do: >> >> > >> >> >if (send_it) { out->number_of_packets = FRAMES_PER_URB; >> >> > >> >> >This should be >> >> > >> >> >out->number_of_packets = outframe; >> >> > >> >> >The way it is now, the USB stack will try to use data from all the >> >> >frame descriptors, and the last few will be stale because the loop >> >> >doesn't set them. >> >> >> >> That's actually true, even though it doesn't seem to cause any trouble. >> >> I tested everything here of course, and the output URBs return back from >> >> the USB stack with their length fields zeroed out, which then >> >> causes the stack to send packets with zero-length fields at the end. >> > >> > Actually, it causes system hangs when the driver is loaded on a device >> > attached to a USB 3.0 port, as Alan Stern pointed out: >> > >> > https://bugzilla.kernel.org/show_bug.cgi?id=40702 >> >> Yes, I've noticed this. >> >> > Please don't submit zero-length transfers. The xHCI driver just isn't >> > able to handle it. Arguably, it probably should have just rejected your >> > URB when it found a zero length buffer, so I'll probably be submitting a >> > patch to fix that. >> >> According to the spec, sending zero-length frames should be fine, no? >> Is there any particular reason why XCHI can't handle this while EHCI >> can? And does my patch fix the driver for XHCI? > > Ok, yes, you're correct that the xHCI spec allows the transfer length to > be set to zero. In the case where the frame buffer is zero-length, is > the buffer pointer still valid? It's not clear from the spec whether it > needs to be. Well, the buffer pointer is set in the URB, not in its individual iso subframes which just denotes them via the offset field. So yes, it is valid in my case. But it doesn't matter anymore, as the code which does that is now gone :) Daniel

1 0

Re: [alsa-devel] [PATCH] ASoC: sgtl5000: fix module device table type for sgtl5000_dt_ids
by Liam Girdwood 11 Aug '11

11 Aug '11

On 11/08/11 15:19, Axel Lin wrote: > The module device table for of_device_id should use "of" type. > > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> > --- > sound/soc/codecs/sgtl5000.c | 2 +- > 1 files changed, 1 insertions(+), 1 deletions(-) > > diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c > index 666fae6..91130fb 100644 > --- a/sound/soc/codecs/sgtl5000.c > +++ b/sound/soc/codecs/sgtl5000.c > @@ -1441,7 +1441,7 @@ static const struct of_device_id sgtl5000_dt_ids[] = { > { .compatible = "fsl,sgtl5000", }, > { /* sentinel */ } > }; > -MODULE_DEVICE_TABLE(i2c, sgtl5000_dt_ids); > +MODULE_DEVICE_TABLE(of, sgtl5000_dt_ids); > > static struct i2c_driver sgtl5000_i2c_driver = { > .driver = { Acked-by: Liam Girdwood <lrg(a)ti.com>

1 0

[alsa-devel] [PATCH] ASoC: Allow userspace control of Speyside headphone output
by Mark Brown 11 Aug '11

11 Aug '11

In order to facilitate the widest range of use cases (especially things like speakerphone) allow the headphone output to be enabled and disabled by the application layer. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/speyside.c | 5 +---- 1 files changed, 1 insertions(+), 4 deletions(-) diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index 3b63995..67572c9 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -125,10 +125,6 @@ static struct snd_soc_jack_pin speyside_headset_pins[] = { .pin = "Headset Mic", .mask = SND_JACK_MICROPHONE, }, - { - .pin = "Headphone", - .mask = SND_JACK_HEADPHONE, - }, }; /* Default the headphone selection to active high */ @@ -256,6 +252,7 @@ static const struct snd_kcontrol_new controls[] = { SOC_DAPM_PIN_SWITCH("Main AMIC"), SOC_DAPM_PIN_SWITCH("WM1250 Input"), SOC_DAPM_PIN_SWITCH("WM1250 Output"), + SOC_DAPM_PIN_SWITCH("Headphone"), }; static struct snd_soc_dapm_widget widgets[] = { -- 1.7.5.4

2 1

[alsa-devel] [PATCH 1/2] ASoC: Add VMID widget for wm_hubs devices
by Mark Brown 11 Aug '11

11 Aug '11

Currently this does not actually do anything, it is being introduced in order to facilitate additional power optimisations for current generation devices. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8993.c | 4 ++++ sound/soc/codecs/wm8994.c | 3 +++ sound/soc/codecs/wm_hubs.c | 16 ++++++++++++++++ 3 files changed, 23 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index f014e56..eec8e14 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -847,6 +847,7 @@ SND_SOC_DAPM_SUPPLY("CLK_SYS", WM8993_BUS_CONTROL_1, 1, 0, clk_sys_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_SUPPLY("TOCLK", WM8993_CLOCKING_1, 14, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("CLK_DSP", WM8993_CLOCKING_3, 0, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("VMID", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_ADC("ADCL", NULL, WM8993_POWER_MANAGEMENT_2, 1, 0), SND_SOC_DAPM_ADC("ADCR", NULL, WM8993_POWER_MANAGEMENT_2, 0, 0), @@ -880,6 +881,9 @@ SND_SOC_DAPM_PGA("Direct Voice", SND_SOC_NOPM, 0, 0, NULL, 0), }; static const struct snd_soc_dapm_route routes[] = { + { "MICBIAS1", NULL, "VMID" }, + { "MICBIAS2", NULL, "VMID" }, + { "ADCL", NULL, "CLK_SYS" }, { "ADCL", NULL, "CLK_DSP" }, { "ADCR", NULL, "CLK_SYS" }, diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index b70901d..2fb3807 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1171,6 +1171,7 @@ SND_SOC_DAPM_INPUT("Clock"), SND_SOC_DAPM_SUPPLY_S("MICBIAS Supply", 1, SND_SOC_NOPM, 0, 0, micbias_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_SUPPLY("VMID", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("CLK_SYS", SND_SOC_NOPM, 0, 0, clk_sys_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), @@ -1488,6 +1489,8 @@ static const struct snd_soc_dapm_route wm8994_revd_intercon[] = { static const struct snd_soc_dapm_route wm8994_intercon[] = { { "AIF2DACL", NULL, "AIF2DAC Mux" }, { "AIF2DACR", NULL, "AIF2DAC Mux" }, + { "MICBIAS1", NULL, "VMID" }, + { "MICBIAS2", NULL, "VMID" }, }; static const struct snd_soc_dapm_route wm8958_intercon[] = { diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 017522e..ca8ce03 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -711,6 +711,11 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "IN1L PGA", "IN1LP Switch", "IN1LP" }, { "IN1L PGA", "IN1LN Switch", "IN1LN" }, + { "IN1L PGA", NULL, "VMID" }, + { "IN1R PGA", NULL, "VMID" }, + { "IN2L PGA", NULL, "VMID" }, + { "IN2R PGA", NULL, "VMID" }, + { "IN1R PGA", "IN1RP Switch", "IN1RP" }, { "IN1R PGA", "IN1RN Switch", "IN1RN" }, @@ -728,12 +733,14 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "MIXINL", NULL, "Direct Voice" }, { "MIXINL", NULL, "IN1LP" }, { "MIXINL", NULL, "Left Output Mixer" }, + { "MIXINL", NULL, "VMID" }, { "MIXINR", "IN1R Switch", "IN1R PGA" }, { "MIXINR", "IN2R Switch", "IN2R PGA" }, { "MIXINR", NULL, "Direct Voice" }, { "MIXINR", NULL, "IN1RP" }, { "MIXINR", NULL, "Right Output Mixer" }, + { "MIXINR", NULL, "VMID" }, { "ADCL", NULL, "MIXINL" }, { "ADCR", NULL, "MIXINR" }, @@ -764,6 +771,7 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "Earpiece Mixer", "Left Output Switch", "Left Output PGA" }, { "Earpiece Mixer", "Right Output Switch", "Right Output PGA" }, + { "Earpiece Driver", NULL, "VMID" }, { "Earpiece Driver", NULL, "Earpiece Mixer" }, { "HPOUT2N", NULL, "Earpiece Driver" }, { "HPOUT2P", NULL, "Earpiece Driver" }, @@ -786,9 +794,11 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "SPKR Boost", "SPKR Switch", "SPKR" }, { "SPKR Boost", "SPKL Switch", "SPKL" }, + { "SPKL Driver", NULL, "VMID" }, { "SPKL Driver", NULL, "SPKL Boost" }, { "SPKL Driver", NULL, "CLK_SYS" }, + { "SPKR Driver", NULL, "VMID" }, { "SPKR Driver", NULL, "SPKR Boost" }, { "SPKR Driver", NULL, "CLK_SYS" }, @@ -802,12 +812,18 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "Headphone PGA", NULL, "Left Headphone Mux" }, { "Headphone PGA", NULL, "Right Headphone Mux" }, + { "Headphone PGA", NULL, "VMID" }, { "Headphone PGA", NULL, "CLK_SYS" }, { "Headphone PGA", NULL, "Headphone Supply" }, { "HPOUT1L", NULL, "Headphone PGA" }, { "HPOUT1R", NULL, "Headphone PGA" }, + { "LINEOUT1N Driver", NULL, "VMID" }, + { "LINEOUT1P Driver", NULL, "VMID" }, + { "LINEOUT2N Driver", NULL, "VMID" }, + { "LINEOUT2P Driver", NULL, "VMID" }, + { "LINEOUT1N", NULL, "LINEOUT1N Driver" }, { "LINEOUT1P", NULL, "LINEOUT1P Driver" }, { "LINEOUT2N", NULL, "LINEOUT2N Driver" }, -- 1.7.5.4

2 2

[alsa-devel] [PATCH] ASoC: Add bitfield definitions for WM8958 MICBIAS registers
by Mark Brown 11 Aug '11

11 Aug '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/linux/mfd/wm8994/registers.h | 38 ++++++++++++++++++++++++++++++++++ 1 files changed, 38 insertions(+), 0 deletions(-) diff --git a/include/linux/mfd/wm8994/registers.h b/include/linux/mfd/wm8994/registers.h index 6152914..64bf91e 100644 --- a/include/linux/mfd/wm8994/registers.h +++ b/include/linux/mfd/wm8994/registers.h @@ -1922,6 +1922,44 @@ #define WM8994_LDO2_DISCH_WIDTH 1 /* LDO2_DISCH */ /* + * R61 (0x3D) - MICBIAS1 + */ +#define WM8958_MICB1_RATE 0x0020 /* MICB1_RATE */ +#define WM8958_MICB1_RATE_MASK 0x0020 /* MICB1_RATE */ +#define WM8958_MICB1_RATE_SHIFT 5 /* MICB1_RATE */ +#define WM8958_MICB1_RATE_WIDTH 1 /* MICB1_RATE */ +#define WM8958_MICB1_MODE 0x0010 /* MICB1_MODE */ +#define WM8958_MICB1_MODE_MASK 0x0010 /* MICB1_MODE */ +#define WM8958_MICB1_MODE_SHIFT 4 /* MICB1_MODE */ +#define WM8958_MICB1_MODE_WIDTH 1 /* MICB1_MODE */ +#define WM8958_MICB1_LVL_MASK 0x000E /* MICB1_LVL - [3:1] */ +#define WM8958_MICB1_LVL_SHIFT 1 /* MICB1_LVL - [3:1] */ +#define WM8958_MICB1_LVL_WIDTH 3 /* MICB1_LVL - [3:1] */ +#define WM8958_MICB1_DISCH 0x0001 /* MICB1_DISCH */ +#define WM8958_MICB1_DISCH_MASK 0x0001 /* MICB1_DISCH */ +#define WM8958_MICB1_DISCH_SHIFT 0 /* MICB1_DISCH */ +#define WM8958_MICB1_DISCH_WIDTH 1 /* MICB1_DISCH */ + +/* + * R62 (0x3E) - MICBIAS2 + */ +#define WM8958_MICB2_RATE 0x0020 /* MICB2_RATE */ +#define WM8958_MICB2_RATE_MASK 0x0020 /* MICB2_RATE */ +#define WM8958_MICB2_RATE_SHIFT 5 /* MICB2_RATE */ +#define WM8958_MICB2_RATE_WIDTH 1 /* MICB2_RATE */ +#define WM8958_MICB2_MODE 0x0010 /* MICB2_MODE */ +#define WM8958_MICB2_MODE_MASK 0x0010 /* MICB2_MODE */ +#define WM8958_MICB2_MODE_SHIFT 4 /* MICB2_MODE */ +#define WM8958_MICB2_MODE_WIDTH 1 /* MICB2_MODE */ +#define WM8958_MICB2_LVL_MASK 0x000E /* MICB2_LVL - [3:1] */ +#define WM8958_MICB2_LVL_SHIFT 1 /* MICB2_LVL - [3:1] */ +#define WM8958_MICB2_LVL_WIDTH 3 /* MICB2_LVL - [3:1] */ +#define WM8958_MICB2_DISCH 0x0001 /* MICB2_DISCH */ +#define WM8958_MICB2_DISCH_MASK 0x0001 /* MICB2_DISCH */ +#define WM8958_MICB2_DISCH_SHIFT 0 /* MICB2_DISCH */ +#define WM8958_MICB2_DISCH_WIDTH 1 /* MICB2_DISCH */ + +/* * R76 (0x4C) - Charge Pump (1) */ #define WM8994_CP_ENA 0x8000 /* CP_ENA */ -- 1.7.5.4

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[alsa-devel] [PATCH 3.0]ASoC: ad193x: add spi_hw_read, fix sysclk and register definition
by Scott Jiang 11 Aug '11

11 Aug '11

asoc cache layer can't support this kind of spi registers, so bypass cache and read regiters directly Signed-off-by: Scott Jiang <scott.jiang.linux(a)gmail.com> --- sound/soc/blackfin/bf5xx-ad193x.c | 2 +- sound/soc/codecs/ad193x.c | 8 ++++++-- sound/soc/codecs/ad193x.h | 5 +++-- sound/soc/soc-cache.c | 23 +++++++++++++++++++++++ 4 files changed, 33 insertions(+), 5 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ad193x.c b/sound/soc/blackfin/bf5xx-ad193x.c index d40cf68..417200b 100644 --- a/sound/soc/blackfin/bf5xx-ad193x.c +++ b/sound/soc/blackfin/bf5xx-ad193x.c @@ -56,7 +56,7 @@ static int bf5xx_ad193x_hw_params(struct snd_pcm_substream *substream, switch (params_rate(params)) { case 48000: - clk = 12288000; + clk = 24576000; break; } diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 2374ca5..1447cc8 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -307,7 +307,8 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream, snd_soc_write(codec, AD193X_PLL_CLK_CTRL0, reg); reg = snd_soc_read(codec, AD193X_DAC_CTRL2); - reg = (reg & (~AD193X_DAC_WORD_LEN_MASK)) | word_len; + reg = (reg & (~AD193X_DAC_WORD_LEN_MASK)) + | (word_len << AD193X_DAC_WORD_LEN_SHFT); snd_soc_write(codec, AD193X_DAC_CTRL2, reg); reg = snd_soc_read(codec, AD193X_ADC_CTRL1); @@ -361,7 +362,10 @@ static int ad193x_probe(struct snd_soc_codec *codec) dev_err(codec->dev, "failed to set cache I/O: %d\n", ret); return ret; } - +#if defined(CONFIG_SPI_MASTER) + /* asoc cache layer can't support this kind of spi registers now */ + codec->cache_bypass = 1; +#endif /* default setting for ad193x */ /* unmute dac channels */ diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h index 9747b54..cccc2e8 100644 --- a/sound/soc/codecs/ad193x.h +++ b/sound/soc/codecs/ad193x.h @@ -34,7 +34,8 @@ #define AD193X_DAC_LEFT_HIGH (1 << 3) #define AD193X_DAC_BCLK_INV (1 << 7) #define AD193X_DAC_CTRL2 0x804 -#define AD193X_DAC_WORD_LEN_MASK 0xC +#define AD193X_DAC_WORD_LEN_SHFT 3 +#define AD193X_DAC_WORD_LEN_MASK 0x18 #define AD193X_DAC_MASTER_MUTE 1 #define AD193X_DAC_CHNL_MUTE 0x805 #define AD193X_DACL1_MUTE 0 @@ -63,7 +64,7 @@ #define AD193X_ADC_CTRL1 0x80f #define AD193X_ADC_SERFMT_MASK 0x60 #define AD193X_ADC_SERFMT_STEREO (0 << 5) -#define AD193X_ADC_SERFMT_TDM (1 << 2) +#define AD193X_ADC_SERFMT_TDM (1 << 5) #define AD193X_ADC_SERFMT_AUX (2 << 5) #define AD193X_ADC_WORD_LEN_MASK 0x3 #define AD193X_ADC_CTRL2 0x810 diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 039b953..d9e9682 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -233,6 +233,25 @@ static unsigned int snd_soc_16_8_read_i2c(struct snd_soc_codec *codec, #define snd_soc_16_8_read_i2c NULL #endif +#if defined(CONFIG_SPI_MASTER) +static unsigned int snd_soc_16_8_read_spi(struct snd_soc_codec *codec, + unsigned int r) +{ + struct spi_device *spi = codec->control_data; + + const u16 reg = cpu_to_be16(r | 0x100); + u8 data; + int ret; + + ret = spi_write_then_read(spi, &reg, 2, &data, 1); + if (ret < 0) + return 0; + return data; +} +#else +#define snd_soc_16_8_read_spi NULL +#endif + static unsigned int snd_soc_16_8_read(struct snd_soc_codec *codec, unsigned int reg) { @@ -337,6 +356,7 @@ static struct { int (*write)(struct snd_soc_codec *codec, unsigned int, unsigned int); unsigned int (*read)(struct snd_soc_codec *, unsigned int); unsigned int (*i2c_read)(struct snd_soc_codec *, unsigned int); + unsigned int (*spi_read)(struct snd_soc_codec *, unsigned int); } io_types[] = { { .addr_bits = 4, .data_bits = 12, @@ -360,6 +380,7 @@ static struct { .addr_bits = 16, .data_bits = 8, .write = snd_soc_16_8_write, .read = snd_soc_16_8_read, .i2c_read = snd_soc_16_8_read_i2c, + .spi_read = snd_soc_16_8_read_spi, }, { .addr_bits = 16, .data_bits = 16, @@ -425,6 +446,8 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, #ifdef CONFIG_SPI_MASTER codec->hw_write = do_spi_write; #endif + if (io_types[i].spi_read) + codec->hw_read = io_types[i].spi_read; codec->control_data = container_of(codec->dev, struct spi_device, -- 1.7.0.4

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[alsa-devel] [PATCH] ASoC: Terminate WM8750 SPI device ID table
by Mark Brown 11 Aug '11

11 Aug '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> --- Again already applied. sound/soc/codecs/wm8750.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index f0ccbef..5f62300 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -789,6 +789,7 @@ static int __devexit wm8750_spi_remove(struct spi_device *spi) static const struct spi_device_id wm8750_spi_ids[] = { { "wm8750", 0 }, { "wm8987", 0 }, + { 0, 0 }, }; MODULE_DEVICE_TABLE(spi, wm8750_spi_id); -- 1.7.5.4

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Re: [alsa-devel] Allocating buffers for USB transfers (again)
by Daniel Mack 11 Aug '11

11 Aug '11

On Thu, Aug 11, 2011 at 1:15 AM, Sarah Sharp <sarah.a.sharp(a)linux.intel.com> wrote: > On Wed, Aug 10, 2011 at 05:33:02PM +0200, Daniel Mack wrote: >> On 08/10/2011 04:32 PM, Alan Stern wrote: >> >Looking at the driver's current code, it appears that your patch >> >does not fix the bug properly. Using discontiguous regions in the >> >transfer buffer is perfectly okay. The real problem is later on, >> >where you do: >> > >> >if (send_it) { out->number_of_packets = FRAMES_PER_URB; >> > >> >This should be >> > >> >out->number_of_packets = outframe; >> > >> >The way it is now, the USB stack will try to use data from all the >> >frame descriptors, and the last few will be stale because the loop >> >doesn't set them. >> >> That's actually true, even though it doesn't seem to cause any trouble. >> I tested everything here of course, and the output URBs return back from >> the USB stack with their length fields zeroed out, which then >> causes the stack to send packets with zero-length fields at the end. > > Actually, it causes system hangs when the driver is loaded on a device > attached to a USB 3.0 port, as Alan Stern pointed out: > > https://bugzilla.kernel.org/show_bug.cgi?id=40702 Yes, I've noticed this. > Please don't submit zero-length transfers. The xHCI driver just isn't > able to handle it. Arguably, it probably should have just rejected your > URB when it found a zero length buffer, so I'll probably be submitting a > patch to fix that. According to the spec, sending zero-length frames should be fine, no? Is there any particular reason why XCHI can't handle this while EHCI can? And does my patch fix the driver for XHCI? Daniel

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[alsa-devel] [PATCH] ASoC: Add missing break in WM8994 probe
by Mark Brown 11 Aug '11

11 Aug '11

This error would have no effect on current silicon revisions, the fall through case has the same behaviour. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 1a9ef7f..e1fb844 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3042,6 +3042,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994->hubs.dcs_readback_mode = 2; break; } + break; case WM8958: wm8994->hubs.dcs_readback_mode = 1; -- 1.7.5.4

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Re: [alsa-devel] Allocating buffers for USB transfers (again)
by Daniel Mack 10 Aug '11

10 Aug '11

On 08/10/2011 04:32 PM, Alan Stern wrote: > Looking at the driver's current code, it appears that your patch > does not fix the bug properly. Using discontiguous regions in the > transfer buffer is perfectly okay. The real problem is later on, > where you do: > > if (send_it) { out->number_of_packets = FRAMES_PER_URB; > > This should be > > out->number_of_packets = outframe; > > The way it is now, the USB stack will try to use data from all the > frame descriptors, and the last few will be stale because the loop > doesn't set them. That's actually true, even though it doesn't seem to cause any trouble. I tested everything here of course, and the output URBs return back from the USB stack with their length fields zeroed out, which then causes the stack to send packets with zero-length fields at the end. However, doing it right doesn't harm either. Thanks for spotting. Takashi, can you apply this version? Thanks, Daniel

2 1

Re: [alsa-devel] [PATCH] ASoC: soc-io: Add CONFIG_REGMAP_I2C/CONFIG_REGMAP_SPI guards for regmap_init_i2c/regmap_init_spi
by Mark Brown 10 Aug '11

10 Aug '11

On Wed, Aug 10, 2011 at 04:24:12PM +0800, Axel Lin wrote: > In the case of "make da8xx_omapl_defconfig;make", the SPI support is disabled. > Thus calling regmap_init_spi in soc-io.c has below build error. Applied, thanks.

1 0

[alsa-devel] [PATCH] sound/pci/azt3328.c: adjust error handling code to include debugging code
by Julia Lawall 10 Aug '11

10 Aug '11

From: Julia Lawall <julia(a)diku.dk> snd_azf3328_dbgcallenter is called at the very beginning of the function, so it could be useful to call snd_azf3328_dbgcallleave at all exit points. Signed-off-by: Julia Lawall <julia(a)diku.dk> --- sound/pci/azt3328.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index e4d76a2..579fc0d 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -2625,16 +2625,19 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) int err; snd_azf3328_dbgcallenter(); - if (dev >= SNDRV_CARDS) - return -ENODEV; + if (dev >= SNDRV_CARDS) { + err = -ENODEV; + goto out; + } if (!enable[dev]) { dev++; - return -ENOENT; + err = -ENOENT; + goto out; } err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); if (err < 0) - return err; + goto out; strcpy(card->driver, "AZF3328"); strcpy(card->shortname, "Aztech AZF3328 (PCI168)");

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[alsa-devel] Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers.
by ashwanik 10 Aug '11

10 Aug '11

Hi All, I am using USB mic on omap4 board(panda). When I am trying to use "gst-launch-0.10 pulsesrc ! pulsesink" pipeline, getting 1-2 sec audio delay and following error messages: E: alsa-source.c: ALSA woke us up to read new data from the device, but there was actually nothing to read! E: alsa-source.c: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. E: alsa-source.c: We were woken up with POLLIN set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. I am attaching the complete log. please find pointers. Regards, Ashwani

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[alsa-devel] [PATCH] ASoC: Specify register defaults for WM8958 MICBIAS1 and MICBIAS2
by Mark Brown 10 Aug '11

10 Aug '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994-tables.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8994-tables.c b/sound/soc/codecs/wm8994-tables.c index a87adbd..13e5a01 100644 --- a/sound/soc/codecs/wm8994-tables.c +++ b/sound/soc/codecs/wm8994-tables.c @@ -1635,8 +1635,8 @@ const u16 wm8994_reg_defaults[WM8994_CACHE_SIZE] = { 0x0000, /* R58 - MICBIAS */ 0x000D, /* R59 - LDO 1 */ 0x0003, /* R60 - LDO 2 */ - 0x0000, /* R61 */ - 0x0000, /* R62 */ + 0x0039, /* R61 - MICBIAS1 */ + 0x0039, /* R62 - MICBIAS2 */ 0x0000, /* R63 */ 0x0000, /* R64 */ 0x0000, /* R65 */ -- 1.7.5.4

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Re: [alsa-devel] Allocating buffers for USB transfers (again)
by Daniel Mack 10 Aug '11

10 Aug '11

On Thu, Jul 7, 2011 at 5:16 PM, Florian Mickler <florian(a)mickler.org> wrote: > [adding Robert Hancock] > see here for the whole thread: https://lkml.org/lkml/2011/7/7/150 > > 2011/7/7 Daniel Mack <zonque(a)gmail.com>: >> On Thu, Jul 7, 2011 at 2:33 PM, Oliver Neukum <oliver(a)neukum.org> wrote: >> >>> PS: Do you still see this if you enable 64bit DMA for EHCI? >> >> The problem is that I personally don't see that issue at all. I even >> installed 4GB of RAM to my development machine last year to be able to >> reproduce this, but I can't, even when the memory allocator is under >> heavy load. The only people who see this effect are Pedro Ribeiro and >> William Light (both in Cc:), and both have been very helpful in trying >> patches and reporting back in detail. Which instructions could we >> probably give to these people to finally hunt this issue down? >> >> Daniel >> >> > > Did someone affected already try the suggestions from Robert : > http://lkml.org/lkml/2010/4/10/66 > > That is checking if it really is a 64bit vs 32bit issue by booting > with mem=4096M ? So it seems we can finally close this issue. I got myself a new laptop recently, and was actually happy to see that this machine showed the bug as well. So I started investigating on what's going on and after hours of debugging in the wrong area, I stumbled over a bug in my own code. The patch attached fixes the problem for me, and William (one of the very few people who saw it, too) also reported green light. However, it remains a puzzle to me where the acutal root cause is for this is, and I would much like to know. Let me briefly explain what I'm seeing. The driver operates in a mode so that it copies the layout of its inbound stream to the output stream, thus guaranteeing the data rate of outbound data is equal to that of the input stream. For that, I walk the isochronous frames of each received urb and prepare an urb for sending that has the same iso packet structure in terms of lengths and offsets, jumping over and ignoring frames that are actually invalid (status != 0). So here was the problem: if there were frames with any different status field, the driver would calculate the wrong offset and the playback stream will have artefacts. So, a classic bug you might think, and the fix is trivial. However, what I really don't understand is why this wasn't observed any earlier by any of the many users, and why iit makes a difference which type of memory I use for this. Recall that my original patch that allocates DMA coherent memory for the transfer buffers also fixes the problem just fine. Or put it that way: when allocating "normal" memory using kmalloc(), the stream appears to have "holes" in the iso packet structure, while with DMA coherent memory, the layout is different. And all this only happens on fairly new 64bit-enabled machines. Can anyone explain this? Many thanks to everyone involved in this - and especially to William and Pedro who spent a lot of time in trying my patches and sending me debug information. Daniel

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[alsa-devel] [PATCH] ALSA: usb-audio - Add quirk for BOSS Micro BR-80
by Kazutomo Yoshii 10 Aug '11

10 Aug '11

Signed-off-by: Kazutomo Yoshii <kazutomo.yoshii(a)gmail.com> --- sound/usb/quirks-table.h | 34 ++++++++++++++++++++++++++++++++++ 1 files changed, 34 insertions(+), 0 deletions(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 4d4f865..a42e3ef 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1707,6 +1707,40 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + USB_DEVICE(0x0582, 0x0130), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "BOSS", */ + /* .product_name = "MICRO BR-80", */ + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 3, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0001, + .in_cables = 0x0001 + } + }, + { + .ifnum = -1 + } + } + } +}, /* Guillemot devices */ { -- 1.7.6

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[alsa-devel] [PATCH] ASoC: Fix typo in wm8750 spi_ids
by Mark Brown 10 Aug '11

10 Aug '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> --- I've already applied this. sound/soc/codecs/wm8750.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index f0ccbef..0fa742f 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -790,7 +790,7 @@ static const struct spi_device_id wm8750_spi_ids[] = { { "wm8750", 0 }, { "wm8987", 0 }, }; -MODULE_DEVICE_TABLE(spi, wm8750_spi_id); +MODULE_DEVICE_TABLE(spi, wm8750_spi_ids); static struct spi_driver wm8750_spi_driver = { .driver = { -- 1.7.5.4

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[alsa-devel] asoc: where to store board specific info?
by Scott Jiang 09 Aug '11

09 Aug '11

Hi Mark, I wonder where to store board specific info? For example, some customer boards have a reset pin to control codec. If I store it in machine driver, I must modify it according to the boards. I can store it in arch/board file and pass it to codec driver, but how about the external system clock? Regards, Scott

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Re: [alsa-devel] [PATCH] ASoc: new rt5631 driver from Realtek
by Mark Brown 09 Aug '11

09 Aug '11

On Tue, Aug 09, 2011 at 06:10:16PM +0800, johnnyhsu wrote: > We have prepared our driver and attach them in the mail. > The driver is made according to alsa architecture and patches by diff. Please follow the standard driver submission process documented in Documentation/SubmittingPatches. At the very least you need to submit patches in the format documented there, not attachments and not just directly attached source files as you have done. There are many examples of the appropriate mail format on the Linux mailing lists. git format-patch and git send-email will generally do the right thing, they're often the easiest way to get things right. > The driver is approved and worked on alsa 1.0.24 As also covered in SubmittingPatches you need to submit against the current development version of the standard Linux kernel. For ASoC this is: git://git.kernel.org/pub/scm/linux/kernel/broonie/sound-2.6.git for-next Looking briefly at the code I also see: > static struct snd_soc_codec *rt5631_codec; This is not suitable for mainline, please register your device using the standard kernel mechanisms. > module_param(timesofbclk, int, S_IRUSR | S_IWUSR | S_IRGRP | S_IROTH); > MODULE_PARM_DESC(timeofbclk, "relationship between bclk and fs"); This looks like something that should be configured using the machine driver. > static inline int rt5631_write(struct snd_soc_codec *codec, > unsigned int reg, unsigned int val) > { > return snd_soc_write(codec, reg, val); > } > static inline unsigned int rt5631_read(struct snd_soc_codec *codec, > unsigned int reg) > { > return snd_soc_read(codec, reg); > } Just use the relevant functions directly. > > static int rt5631_write_mask(struct snd_soc_codec *codec, > unsigned int reg, unsigned int value, unsigned int mask) This is snd_soc_update_bits(). > /* > * speaker channel volume select SPKMIXER, 0DB by default > * Headphone channel volume select OUTMIXER,0DB by default > * AXO1/AXO2 channel volume select OUTMIXER,0DB by default > * Record Mixer source from Mic1/Mic2 by default > * Mic1/Mic2 boost 40dB by default > * DAC_L-->OutMixer_L by default > * DAC_R-->OutMixer_R by default > * DAC-->SpeakerMixer > * Speaker volume-->SPOMixer(L-->L,R-->R) > * Speaker AMP ratio gain is 1.44X > * HP from OutMixer,speaker out from SpeakerOut Mixer > * enable HP zero cross > * change Mic1 & mic2 to differential mode > */ This sort of stuff is not suitable for mainline either, the driver should just use chip defaults for audio paths and leave the use case up to the application layer. Please compare your driver with the other CODEC drivers and make sure it is following a simmilar style to them.

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[alsa-devel] TWL4030 Line In/Out Loopback issue
by Almohanad Fayez 09 Aug '11

09 Aug '11

I'verealized that there is a loopback dependency between the audioinput/output dependency with the 2.6.38 kernel running on a gumstix overo and also using the 2.6.35 kernel running on the Beaglenboard. You can replicate this issue by plugging some music into the Line injack and then fire up alsamixer. Then you would need to modify theDAC2 Dig level, the one on the right no the left. As you lower thevolume you'll hear your music input decreasing and as you increase itthe music level would increase. If you lower it enough you can captureaudio from the input jack without hearing it out of your speaker butthe bad thing is that also controls your speaker output so if you tryto play some sound out of gumstix/beagleboard you won't be able to do because youneed to increase your DAC2 Dig level in alsamixer ... it's a catch22!! I couldn't find any solutions readily. I am running a console-only Angstrom image on both boards. I've used the Beagleboard's line in/out capabilities with no issues using the 2.6.29 kernel in the past so is there an update in the TWL4030 ALSA driver that I should take into account? thanks al fayez

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[alsa-devel] [PATCH] ASoC: Fix SPI driver binding for WM8987
by Mark Brown 09 Aug '11

09 Aug '11

As we had no id_table only the driver name would be matched against meaning that WM8987 devices wouldn't be bound. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8750.c | 9 ++++++++- 1 files changed, 8 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 6cbf51b..2bb7bea 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -779,11 +779,18 @@ static int __devexit wm8750_spi_remove(struct spi_device *spi) return 0; } +static const struct spi_device_id wm8750_spi_ids[] = { + { "wm8750", 0 }, + { "wm8987", 0 }, +}; +MODULE_DEVICE_TABLE(spi, wm8750_spi_id); + static struct spi_driver wm8750_spi_driver = { .driver = { - .name = "wm8750-codec", + .name = "wm8750", .owner = THIS_MODULE, }, + .id_table = wm8750_spi_ids, .probe = wm8750_spi_probe, .remove = __devexit_p(wm8750_spi_remove), }; -- 1.7.5.4

2 6

[alsa-devel] [PATCH 1/3] ASoC: Tegra: tegra_pcm_deallocate_dma_buffer: Don't OOPS
by Stephen Warren 09 Aug '11

09 Aug '11

Not all PCM devices have all sub-streams. Specifically, the SPDIF driver only supports playback and hence has no capture substream. Check whether a substream exists before dereferencing it, when de-allocating DMA buffers in tegra_pcm_deallocate_dma_buffer. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/tegra/tegra_pcm.c | 9 +++++++-- 1 files changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index ff86e5e..c7cfd96 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -309,9 +309,14 @@ static int tegra_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) static void tegra_pcm_deallocate_dma_buffer(struct snd_pcm *pcm, int stream) { - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + + substream = pcm->streams[stream].substream; + if (!substream) + return; + buf = &substream->dma_buffer; if (!buf->area) return; -- 1.7.0.4

3 6

[alsa-devel] [PATCH] ASoC: Fix warning in WM1250-EV1 driver
by Mark Brown 09 Aug '11

09 Aug '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm1250-ev1.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c index bbcf9ec..a98a3ff1 100644 --- a/sound/soc/codecs/wm1250-ev1.c +++ b/sound/soc/codecs/wm1250-ev1.c @@ -58,12 +58,12 @@ static struct snd_soc_codec_driver soc_codec_dev_wm1250_ev1 = { static int __devinit wm1250_ev1_probe(struct i2c_client *i2c, const struct i2c_device_id *i2c_id) { - int ret, id, board, rev; + int id, board, rev; board = i2c_smbus_read_byte_data(i2c, 0); if (board < 0) { - dev_err(&i2c->dev, "Failed to read ID: %d\n", ret); - return ret; + dev_err(&i2c->dev, "Failed to read ID: %d\n", board); + return board; } id = (board & 0xfe) >> 2; -- 1.7.5.4

2 1

[alsa-devel] [PATCH 1/2] ASoC: Refactor WM8741 regulator handling into CODEC generic code
by Mark Brown 09 Aug '11

09 Aug '11

No meaningful runtime impact but is more in line with other CODECs and will support further work. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8741.c | 74 ++++++++++++++++++++++++++------------------ 1 files changed, 44 insertions(+), 30 deletions(-) diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 3def27c..d4bfffd 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -422,17 +422,35 @@ static int wm8741_probe(struct snd_soc_codec *codec) { struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); int ret = 0; + int i; + + for (i = 0; i < ARRAY_SIZE(wm8741->supplies); i++) + wm8741->supplies[i].supply = wm8741_supply_names[i]; + + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8741->supplies), + wm8741->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to request supplies: %d\n", ret); + goto err; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(wm8741->supplies), + wm8741->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + goto err_get; + } ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8741->control_type); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; + goto err_enable; } ret = wm8741_reset(codec); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset\n"); - return ret; + goto err_enable; } /* Change some default settings - latch VU */ @@ -451,10 +469,28 @@ static int wm8741_probe(struct snd_soc_codec *codec) dev_dbg(codec->dev, "Successful registration\n"); return ret; + +err_enable: + regulator_bulk_disable(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); +err_get: + regulator_bulk_free(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); +err: + return ret; +} + +static int wm8741_remove(struct snd_soc_codec *codec) +{ + struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); + + regulator_bulk_disable(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); + regulator_bulk_free(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); + + return 0; } static struct snd_soc_codec_driver soc_codec_dev_wm8741 = { .probe = wm8741_probe, + .remove = wm8741_remove, .resume = wm8741_resume, .reg_cache_size = ARRAY_SIZE(wm8741_reg_defaults), .reg_word_size = sizeof(u16), @@ -466,43 +502,22 @@ static int wm8741_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct wm8741_priv *wm8741; - int ret, i; + int ret; wm8741 = kzalloc(sizeof(struct wm8741_priv), GFP_KERNEL); if (wm8741 == NULL) return -ENOMEM; - for (i = 0; i < ARRAY_SIZE(wm8741->supplies); i++) - wm8741->supplies[i].supply = wm8741_supply_names[i]; - - ret = regulator_bulk_get(&i2c->dev, ARRAY_SIZE(wm8741->supplies), - wm8741->supplies); - if (ret != 0) { - dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret); - goto err; - } - - ret = regulator_bulk_enable(ARRAY_SIZE(wm8741->supplies), - wm8741->supplies); - if (ret != 0) { - dev_err(&i2c->dev, "Failed to enable supplies: %d\n", ret); - goto err_get; - } - i2c_set_clientdata(i2c, wm8741); wm8741->control_type = SND_SOC_I2C; - ret = snd_soc_register_codec(&i2c->dev, - &soc_codec_dev_wm8741, &wm8741_dai, 1); - if (ret < 0) - goto err_enable; - return ret; + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_wm8741, &wm8741_dai, 1); + if (ret != 0) + goto err; -err_enable: - regulator_bulk_disable(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); + return ret; -err_get: - regulator_bulk_free(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); err: kfree(wm8741); return ret; @@ -513,7 +528,6 @@ static int wm8741_i2c_remove(struct i2c_client *client) struct wm8741_priv *wm8741 = i2c_get_clientdata(client); snd_soc_unregister_codec(&client->dev); - regulator_bulk_free(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); kfree(i2c_get_clientdata(client)); return 0; } -- 1.7.5.4

2 4

[alsa-devel] [PATCH] Support for BOSS Micro BR-80
by Kazutomo Yoshii 09 Aug '11

09 Aug '11

Hi, [ I forget to add Signed-off-by. Sorry about re-posting ] This patch adds support for BOSS Micro BR-80, which allows us to use BR-80 as a USB Audio interface (both playback and capture) and external MIDI drum sound module(CH10). Thanks, Kaz Signed-off-by: Kazutomo Yoshii <kazutomo.yoshii(a)gmail.com> --- sound/usb/quirks-table.h | 34 ++++++++++++++++++++++++++++++++++ 1 files changed, 34 insertions(+), 0 deletions(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 4d4f865..a42e3ef 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1707,6 +1707,40 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + USB_DEVICE(0x0582, 0x0130), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "BOSS", */ + /* .product_name = "MICRO BR-80", */ + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 3, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0001, + .in_cables = 0x0001 + } + }, + { + .ifnum = -1 + } + } + } +}, /* Guillemot devices */ { -- 1.7.6

1 0

[alsa-devel] [PATCH] ASoC: Update SMDKs for WM8580 -codec removal
by Mark Brown 09 Aug '11

09 Aug '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/smdk_wm8580.c | 6 +++--- sound/soc/samsung/smdk_wm8580pcm.c | 4 ++-- 2 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/samsung/smdk_wm8580.c b/sound/soc/samsung/smdk_wm8580.c index 3d26f66..20deecf 100644 --- a/sound/soc/samsung/smdk_wm8580.c +++ b/sound/soc/samsung/smdk_wm8580.c @@ -210,7 +210,7 @@ static struct snd_soc_dai_link smdk_dai[] = { .cpu_dai_name = "samsung-i2s.0", .codec_dai_name = "wm8580-hifi-playback", .platform_name = "samsung-audio", - .codec_name = "wm8580-codec.0-001b", + .codec_name = "wm8580.0-001b", .init = smdk_wm8580_init_paifrx, .ops = &smdk_ops, }, @@ -220,7 +220,7 @@ static struct snd_soc_dai_link smdk_dai[] = { .cpu_dai_name = "samsung-i2s.0", .codec_dai_name = "wm8580-hifi-capture", .platform_name = "samsung-audio", - .codec_name = "wm8580-codec.0-001b", + .codec_name = "wm8580.0-001b", .init = smdk_wm8580_init_paiftx, .ops = &smdk_ops, }, @@ -230,7 +230,7 @@ static struct snd_soc_dai_link smdk_dai[] = { .cpu_dai_name = "samsung-i2s.x", .codec_dai_name = "wm8580-hifi-playback", .platform_name = "samsung-audio", - .codec_name = "wm8580-codec.0-001b", + .codec_name = "wm8580.0-001b", .init = smdk_wm8580_init_paifrx, .ops = &smdk_ops, }, diff --git a/sound/soc/samsung/smdk_wm8580pcm.c b/sound/soc/samsung/smdk_wm8580pcm.c index 0d12092..4b9c734 100644 --- a/sound/soc/samsung/smdk_wm8580pcm.c +++ b/sound/soc/samsung/smdk_wm8580pcm.c @@ -127,7 +127,7 @@ static struct snd_soc_dai_link smdk_dai[] = { .cpu_dai_name = "samsung-pcm.0", .codec_dai_name = "wm8580-hifi-playback", .platform_name = "samsung-audio", - .codec_name = "wm8580-codec.0-001b", + .codec_name = "wm8580.0-001b", .ops = &smdk_wm8580_pcm_ops, }, { .name = "WM8580 PAIF PCM TX", @@ -135,7 +135,7 @@ static struct snd_soc_dai_link smdk_dai[] = { .cpu_dai_name = "samsung-pcm.0", .codec_dai_name = "wm8580-hifi-capture", .platform_name = "samsung-audio", - .codec_name = "wm8580-codec.0-001b", + .codec_name = "wm8580.0-001b", .ops = &smdk_wm8580_pcm_ops, }, }; -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: Fix binding of WM8750 on Jive
by Mark Brown 09 Aug '11

09 Aug '11

The I2C address is misformatted and would never match. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/jive_wm8750.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/samsung/jive_wm8750.c b/sound/soc/samsung/jive_wm8750.c index 3b53ad5..14eb6ea 100644 --- a/sound/soc/samsung/jive_wm8750.c +++ b/sound/soc/samsung/jive_wm8750.c @@ -131,7 +131,7 @@ static struct snd_soc_dai_link jive_dai = { .cpu_dai_name = "s3c2412-i2s", .codec_dai_name = "wm8750-hifi", .platform_name = "samsung-audio", - .codec_name = "wm8750-codec.0-0x1a", + .codec_name = "wm8750-codec.0-001a", .init = jive_wm8750_init, .ops = &jive_ops, }; -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: Remove redundant -codec from WM8750
by Mark Brown 09 Aug '11

09 Aug '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8750.c | 2 +- sound/soc/pxa/spitz.c | 2 +- sound/soc/pxa/z2.c | 2 +- sound/soc/samsung/jive_wm8750.c | 2 +- sound/soc/samsung/smartq_wm8987.c | 2 +- 5 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index b7c96c6..f0ccbef 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -841,7 +841,7 @@ MODULE_DEVICE_TABLE(i2c, wm8750_i2c_id); static struct i2c_driver wm8750_i2c_driver = { .driver = { - .name = "wm8750-codec", + .name = "wm8750", .owner = THIS_MODULE, .of_match_table = wm8750_of_match, }, diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index b253d86..ce920e3 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -312,7 +312,7 @@ static struct snd_soc_dai_link spitz_dai = { .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "wm8750-hifi", .platform_name = "pxa-pcm-audio", - .codec_name = "wm8750-codec.0-001b", + .codec_name = "wm8750.0-001b", .init = spitz_wm8750_init, .ops = &spitz_ops, }; diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c index d69d9fc..4b81ffd 100644 --- a/sound/soc/pxa/z2.c +++ b/sound/soc/pxa/z2.c @@ -198,7 +198,7 @@ static struct snd_soc_dai_link z2_dai = { .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "wm8750-hifi", .platform_name = "pxa-pcm-audio", - .codec_name = "wm8750-codec.0-001b", + .codec_name = "wm8750.0-001b", .init = z2_wm8750_init, .ops = &z2_ops, }; diff --git a/sound/soc/samsung/jive_wm8750.c b/sound/soc/samsung/jive_wm8750.c index 14eb6ea..ed8f13a 100644 --- a/sound/soc/samsung/jive_wm8750.c +++ b/sound/soc/samsung/jive_wm8750.c @@ -131,7 +131,7 @@ static struct snd_soc_dai_link jive_dai = { .cpu_dai_name = "s3c2412-i2s", .codec_dai_name = "wm8750-hifi", .platform_name = "samsung-audio", - .codec_name = "wm8750-codec.0-001a", + .codec_name = "wm8750.0-001a", .init = jive_wm8750_init, .ops = &jive_ops, }; diff --git a/sound/soc/samsung/smartq_wm8987.c b/sound/soc/samsung/smartq_wm8987.c index 0a2c4f2..bbd1476 100644 --- a/sound/soc/samsung/smartq_wm8987.c +++ b/sound/soc/samsung/smartq_wm8987.c @@ -207,7 +207,7 @@ static struct snd_soc_dai_link smartq_dai[] = { .cpu_dai_name = "samsung-i2s.0", .codec_dai_name = "wm8750-hifi", .platform_name = "samsung-audio", - .codec_name = "wm8750-codec.0-0x1a", + .codec_name = "wm8750.0-0x1a", .init = smartq_wm8987_init, .ops = &smartq_hifi_ops, }, -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: Fix warning in Speyside WM8962
by Mark Brown 09 Aug '11

09 Aug '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/speyside_wm8962.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/samsung/speyside_wm8962.c b/sound/soc/samsung/speyside_wm8962.c index 8ac42bf..0b9eb5f 100644 --- a/sound/soc/samsung/speyside_wm8962.c +++ b/sound/soc/samsung/speyside_wm8962.c @@ -37,7 +37,7 @@ static int speyside_wm8962_set_bias_level(struct snd_soc_card *card, 44100 * 256, SND_SOC_CLOCK_IN); if (ret < 0) { - pr_err("Failed to set SYSCLK: %d\n"); + pr_err("Failed to set SYSCLK: %d\n", ret); return ret; } } -- 1.7.5.4

2 1

[alsa-devel] [PATCH 1/5] ASoC: Add device tree binding for WM8711
by Mark Brown 09 Aug '11

09 Aug '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- Documentation/devicetree/bindings/sound/wm8711.txt | 18 ++++++++++++++++++ sound/soc/codecs/wm8711.c | 9 +++++++++ 2 files changed, 27 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/wm8711.txt diff --git a/Documentation/devicetree/bindings/sound/wm8711.txt b/Documentation/devicetree/bindings/sound/wm8711.txt new file mode 100644 index 0000000..8ed9998 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8711.txt @@ -0,0 +1,18 @@ +WM8711 audio CODEC + +This device supports both I2C and SPI (configured with pin strapping +on the board). + +Required properties: + + - compatible : "wlf,wm8711" + + - reg : the I2C address of the device for I2C, the chip select + number for SPI. + +Example: + +codec: wm8711@1a { + compatible = "wlf,wm8711"; + reg = <0x1a>; +}; diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index e1db7e4..8457d3c 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -21,6 +21,7 @@ #include <linux/platform_device.h> #include <linux/spi/spi.h> #include <linux/slab.h> +#include <linux/of_device.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -414,6 +415,12 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8711 = { .num_dapm_routes = ARRAY_SIZE(wm8711_intercon), }; +static const struct of_device_id wm8711_of_match[] = { + { .compatible = "wlf,wm8711", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8711_of_match); + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8711_spi_probe(struct spi_device *spi) { @@ -445,6 +452,7 @@ static struct spi_driver wm8711_spi_driver = { .driver = { .name = "wm8711", .owner = THIS_MODULE, + .of_match_table = wm8711_of_match, }, .probe = wm8711_spi_probe, .remove = __devexit_p(wm8711_spi_remove), @@ -489,6 +497,7 @@ static struct i2c_driver wm8711_i2c_driver = { .driver = { .name = "wm8711", .owner = THIS_MODULE, + .of_match_table = wm8711_of_match, }, .probe = wm8711_i2c_probe, .remove = __devexit_p(wm8711_i2c_remove), -- 1.7.5.4

2 5

[alsa-devel] [PATCH] ASoC: Remove some more redundant -codecs from driver names
by Mark Brown 09 Aug '11

09 Aug '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8711.c | 4 ++-- sound/soc/codecs/wm8728.c | 4 ++-- sound/soc/codecs/wm8741.c | 2 +- 3 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index a537e4a..e1db7e4 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -443,7 +443,7 @@ static int __devexit wm8711_spi_remove(struct spi_device *spi) static struct spi_driver wm8711_spi_driver = { .driver = { - .name = "wm8711-codec", + .name = "wm8711", .owner = THIS_MODULE, }, .probe = wm8711_spi_probe, @@ -487,7 +487,7 @@ MODULE_DEVICE_TABLE(i2c, wm8711_i2c_id); static struct i2c_driver wm8711_i2c_driver = { .driver = { - .name = "wm8711-codec", + .name = "wm8711", .owner = THIS_MODULE, }, .probe = wm8711_i2c_probe, diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index 86d4718..c8564f7 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -298,7 +298,7 @@ static int __devexit wm8728_spi_remove(struct spi_device *spi) static struct spi_driver wm8728_spi_driver = { .driver = { - .name = "wm8728-codec", + .name = "wm8728", .owner = THIS_MODULE, }, .probe = wm8728_spi_probe, @@ -342,7 +342,7 @@ MODULE_DEVICE_TABLE(i2c, wm8728_i2c_id); static struct i2c_driver wm8728_i2c_driver = { .driver = { - .name = "wm8728-codec", + .name = "wm8728", .owner = THIS_MODULE, }, .probe = wm8728_i2c_probe, diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 1572a39..c1b6e91 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -541,7 +541,7 @@ MODULE_DEVICE_TABLE(i2c, wm8741_i2c_id); static struct i2c_driver wm8741_i2c_driver = { .driver = { - .name = "wm8741-codec", + .name = "wm8741", .owner = THIS_MODULE, }, .probe = wm8741_i2c_probe, -- 1.7.5.4

2 1

[alsa-devel] HDMI LPCM (7.1) with AMD/FGLRX
by Andrée 'Glaucous' 09 Aug '11

09 Aug '11

Hello. I'm wondering if there's any ongoing development on multichannel sound support for HDMI - LPCM/5.1/7.1, not bitstreamed audio such as DTS/AC3 (which works). I've tried both a AMD HD 4870 and AMD HD 6870, on both 2 channel and bitstreamed audio works with ALSA, but not LPCM. LPCM works perfectly (5.1, 7.1) on Windows. According to the mailing list there seem to have been some activity in this area (January or so), but no success. I'd gladly try out patches. My setup is: Arch Linux 64 bit fglrx 11.7 alsa 1.0.24 HD 6870 -> HDMI -> Denon AVR-1909 (Receiver) -> HDTV -- aplay -L hdmi:CARD=Generic,DEV=0 HD-Audio Generic, HDMI 0 HDMI Audio Output -- speaker-test -D hw:2,3 -c 6 speaker-test 1.0.24.2 Playback device is hw:2,3 Stream parameters are 48000Hz, S16_LE, 6 channels Using 16 octaves of pink noise Channels count (6) not available for playbacks: Invalid argument Setting of hwparams failed: Invalid argument -- speaker-test -D hw:2,3 -c 8 speaker-test 1.0.24.2 Playback device is hw:2,3 Stream parameters are 48000Hz, S16_LE, 8 channels Using 16 octaves of pink noise Channels count (8) not available for playbacks: Invalid argument Setting of hwparams failed: Invalid argument --

2 15

[alsa-devel] [PATCH] Support for BOSS Micro BR-80
by Kazutomo Yoshii 09 Aug '11

09 Aug '11

Hi, This patch adds support for BOSS Micro BR-80, which allows us to use BR-80 as a USB Audio interface (both playback and capture) and external MIDI drum sound module(CH10). Thanks, Kaz --- sound/usb/quirks-table.h | 34 ++++++++++++++++++++++++++++++++++ 1 files changed, 34 insertions(+), 0 deletions(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 4d4f865..a42e3ef 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1707,6 +1707,40 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + USB_DEVICE(0x0582, 0x0130), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "BOSS", */ + /* .product_name = "MICRO BR-80", */ + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 3, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0001, + .in_cables = 0x0001 + } + }, + { + .ifnum = -1 + } + } + } +}, /* Guillemot devices */ { -- 1.7.6

1 0

[alsa-devel] The sound skipping also exist in the alsa-driver-1.0.24. The sound card is CMI8738.
by 刘庆阳 09 Aug '11

09 Aug '11

The sound skipping also exist in the alsa-driver-1.0.24. The sound card is CMI8738. The alsa-driver-1.0.18a have no problem.But I can't build it on kernel 3.x.The error info:hwdep.c: In function ‘snd_hwdep_ioctl_old’:hwdep.c:322:2: warning: implicit declaration of function ‘unlock_kernel’hwdep.c:324:2: warning: implicit declaration of function ‘lock_kernel’hwdep.c: At top level:hwdep.c:348:2: error: unknown field ‘ioctl’ specified in initializerhwdep.c:348:2: warning: initialization from incompatible pointer typemake[1]: *** [hwdep.o] Error 1Help me, please.Thank you in advance.

1 0

[alsa-devel] switching pcm devices dynamically when using dmix
by Baek Chang 08 Aug '11

08 Aug '11

Hi, I'm wondering if the following is possible. If I have my asound.conf configured in such a way that pcm.media { } route to hw:0,0 throuh dmix. can I dynamically change the device to hw:0,1 or hw:1,0 if necessary? Thanks! -- -baeksanchang

3 5

[alsa-devel] [PATCH] ASoC: Remove unneeded -codec from WM8753 driver name
by Mark Brown 08 Aug '11

08 Aug '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8753.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index ffa2ffe..a702550 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1519,7 +1519,7 @@ static int __devexit wm8753_spi_remove(struct spi_device *spi) static struct spi_driver wm8753_spi_driver = { .driver = { - .name = "wm8753-codec", + .name = "wm8753", .owner = THIS_MODULE, }, .probe = wm8753_spi_probe, @@ -1563,7 +1563,7 @@ MODULE_DEVICE_TABLE(i2c, wm8753_i2c_id); static struct i2c_driver wm8753_i2c_driver = { .driver = { - .name = "wm8753-codec", + .name = "wm8753", .owner = THIS_MODULE, }, .probe = wm8753_i2c_probe, -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: Move WM8962 CLKREG_OVD earlier
by Mark Brown 08 Aug '11

08 Aug '11

When the clocking registers are not overriden some of the registers are not writable. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index a381104..e6d58b5 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2934,10 +2934,6 @@ static int wm8962_set_bias_level(struct snd_soc_codec *codec, WM8962_BIAS_ENA | 0x180); msleep(5); - - snd_soc_update_bits(codec, WM8962_CLOCKING2, - WM8962_CLKREG_OVD, - WM8962_CLKREG_OVD); } /* VMID 2*250k */ @@ -3875,6 +3871,10 @@ static int wm8962_probe(struct snd_soc_codec *codec) */ snd_soc_update_bits(codec, WM8962_CLOCKING2, WM8962_SYSCLK_ENA, 0); + /* Ensure we have soft control over all registers */ + snd_soc_update_bits(codec, WM8962_CLOCKING2, + WM8962_CLKREG_OVD, WM8962_CLKREG_OVD); + regulator_bulk_disable(ARRAY_SIZE(wm8962->supplies), wm8962->supplies); if (pdata) { -- 1.7.5.4

2 1

[alsa-devel] Rename of WM8996 for 3.1
by Mark Brown 08 Aug '11

08 Aug '11

This update to rename the WM8915 to WM8996 had some cross tree issues due to the use of the platform data for the device in the Cragganmore machine driver so got held off until things settled down during the merge window. I'm posting it separately to any other updates in case you're nervous about taking it. I can guarantee that only that one machine is affected. This is probably about the best time to do the update while minimizing the cross tree issues. The following changes since commit 322a8b034003c0d46d39af85bf24fee27b902f48: Linux 3.1-rc1 (2011-08-07 18:23:30 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git wm8996-rename Mark Brown (2): ASoC: Rename WM8915 to WM8996 ASoC: Acknowledge WM8996 interrupts before acting on them arch/arm/mach-s3c64xx/mach-crag6410.c | 18 +- include/sound/{wm8915.h => wm8996.h} | 28 +- sound/soc/codecs/Kconfig | 8 +- sound/soc/codecs/Makefile | 4 +- sound/soc/codecs/wm8915.c | 2995 -------------------------- sound/soc/codecs/wm8915.h | 3717 --------------------------------- sound/soc/codecs/wm8996.c | 2994 ++++++++++++++++++++++++++ sound/soc/codecs/wm8996.h | 3717 +++++++++++++++++++++++++++++++++ sound/soc/samsung/Kconfig | 2 +- sound/soc/samsung/speyside.c | 32 +- 10 files changed, 6757 insertions(+), 6758 deletions(-) rename include/sound/{wm8915.h => wm8996.h} (63%) delete mode 100644 sound/soc/codecs/wm8915.c delete mode 100644 sound/soc/codecs/wm8915.h create mode 100644 sound/soc/codecs/wm8996.c create mode 100644 sound/soc/codecs/wm8996.h

2 2

[alsa-devel] [PATCH] ALSA: asihpi - use kzalloc()
by Thomas Meyer 07 Aug '11

07 Aug '11

From: Thomas Meyer <thomas(a)m3y3r.de> Use kzalloc rather than kmalloc followed by memset with 0 This considers some simple cases that are common and easy to validate Note in particular that there are no ...s in the rule, so all of the matched code has to be contiguous The semantic patch that makes this output is available in scripts/coccinelle/api/alloc/kzalloc-simple.cocci. More information about semantic patching is available at http://coccinelle.lip6.fr/ Signed-off-by: Thomas Meyer <thomas(a)m3y3r.de> --- diff -u -p a/sound/pci/asihpi/hpicmn.c b/sound/pci/asihpi/hpicmn.c --- a/sound/pci/asihpi/hpicmn.c 2011-07-26 00:46:12.970154018 +0200 +++ b/sound/pci/asihpi/hpicmn.c 2011-08-01 20:42:34.859285236 +0200 @@ -631,13 +631,12 @@ struct hpi_control_cache *hpi_alloc_cont if (!p_cache) return NULL; - p_cache->p_info = - kmalloc(sizeof(*p_cache->p_info) * control_count, GFP_KERNEL); + p_cache->p_info = kzalloc(sizeof(*p_cache->p_info) * control_count, + GFP_KERNEL); if (!p_cache->p_info) { kfree(p_cache); return NULL; } - memset(p_cache->p_info, 0, sizeof(*p_cache->p_info) * control_count); p_cache->cache_size_in_bytes = size_in_bytes; p_cache->control_count = control_count; p_cache->p_cache = p_dsp_control_buffer;

2 1

[alsa-devel] [PATCH 1/1] ASoC: core: cache index fix
by Dong Aisheng 07 Aug '11

07 Aug '11

For codecs whose reg_cache_step is not 1, the original cache io code may fetch a wrong value since it does not check the reg_cache_step and remainly use the reg as index to fetch value from cache. This patch provides help functions for conversion between cache index and register index and the cache io functions will use them in right place to ensure to fetch a correct register value. Signed-off-by: Dong Aisheng <b29396(a)freescale.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lrg(a)ti.com> Cc: Sascha Hauer <s.hauer(a)pengutronix.de> Cc: Wolfram Sang <w.sang(a)pengutronix.de> --- include/sound/soc.h | 4 ++ sound/soc/soc-cache.c | 117 +++++++++++++++++++++++++++++++++++++------------ sound/soc/soc-core.c | 15 +++--- sound/soc/soc-io.c | 10 +++- 4 files changed, 107 insertions(+), 39 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index aa19f5a..b70789d 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -306,6 +306,10 @@ int snd_soc_cache_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value); int snd_soc_cache_read(struct snd_soc_codec *codec, unsigned int reg, unsigned int *value); +int snd_soc_cache_reg_to_idx(struct snd_soc_codec *codec, + unsigned int reg); +int snd_soc_cache_idx_to_reg(struct snd_soc_codec *codec, + unsigned int idx); int snd_soc_default_volatile_register(struct snd_soc_codec *codec, unsigned int reg); int snd_soc_default_readable_register(struct snd_soc_codec *codec, diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index d9f8ade..1a08bcf 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -66,6 +66,38 @@ static unsigned int snd_soc_get_cache_val(const void *base, unsigned int idx, return -1; } +int snd_soc_cache_reg_to_idx(struct snd_soc_codec *codec, + unsigned int reg) +{ + const struct snd_soc_codec_driver *codec_drv; + unsigned int idx = 0; + + codec_drv = codec->driver; + + if (codec_drv->reg_cache_step > 0) + idx = reg / codec_drv->reg_cache_step; + else + idx = reg; + + return idx; +} + +int snd_soc_cache_idx_to_reg(struct snd_soc_codec *codec, + unsigned int idx) +{ + const struct snd_soc_codec_driver *codec_drv; + unsigned int reg = 0; + + codec_drv = codec->driver; + + if (codec_drv->reg_cache_step > 0) + reg = idx * codec_drv->reg_cache_step; + else + reg = idx; + + return reg; +} + struct snd_soc_rbtree_node { struct rb_node node; /* the actual rbtree node holding this block */ unsigned int base_reg; /* base register handled by this block */ @@ -262,14 +294,17 @@ static int snd_soc_rbtree_cache_write(struct snd_soc_codec *codec, unsigned int pos; int i; int ret; + unsigned int idx; + + idx = snd_soc_cache_reg_to_idx(codec, reg); rbtree_ctx = codec->reg_cache; /* look up the required register in the cached rbnode */ rbnode = rbtree_ctx->cached_rbnode; if (rbnode) { snd_soc_rbtree_get_base_top_reg(rbnode, &base_reg, &top_reg); - if (reg >= base_reg && reg <= top_reg) { - reg_tmp = reg - base_reg; + if (idx >= base_reg && idx <= top_reg) { + reg_tmp = idx - base_reg; val = snd_soc_rbtree_get_register(rbnode, reg_tmp); if (val == value) return 0; @@ -280,9 +315,9 @@ static int snd_soc_rbtree_cache_write(struct snd_soc_codec *codec, /* if we can't locate it in the cached rbnode we'll have * to traverse the rbtree looking for it. */ - rbnode = snd_soc_rbtree_lookup(&rbtree_ctx->root, reg); + rbnode = snd_soc_rbtree_lookup(&rbtree_ctx->root, idx); if (rbnode) { - reg_tmp = reg - rbnode->base_reg; + reg_tmp = idx - rbnode->base_reg; val = snd_soc_rbtree_get_register(rbnode, reg_tmp); if (val == value) return 0; @@ -298,15 +333,15 @@ static int snd_soc_rbtree_cache_write(struct snd_soc_codec *codec, rbnode_tmp = rb_entry(node, struct snd_soc_rbtree_node, node); for (i = 0; i < rbnode_tmp->blklen; ++i) { reg_tmp = rbnode_tmp->base_reg + i; - if (abs(reg_tmp - reg) != 1) + if (abs(reg_tmp - idx) != 1) continue; /* decide where in the block to place our register */ - if (reg_tmp + 1 == reg) + if (reg_tmp + 1 == idx) pos = i + 1; else pos = i; ret = snd_soc_rbtree_insert_to_block(rbnode_tmp, pos, - reg, value); + idx, value); if (ret) return ret; rbtree_ctx->cached_rbnode = rbnode_tmp; @@ -322,7 +357,7 @@ static int snd_soc_rbtree_cache_write(struct snd_soc_codec *codec, if (!rbnode) return -ENOMEM; rbnode->blklen = 1; - rbnode->base_reg = reg; + rbnode->base_reg = idx; rbnode->word_size = codec->driver->reg_word_size; rbnode->block = kmalloc(rbnode->blklen * rbnode->word_size, GFP_KERNEL); @@ -345,14 +380,17 @@ static int snd_soc_rbtree_cache_read(struct snd_soc_codec *codec, struct snd_soc_rbtree_node *rbnode; unsigned int base_reg, top_reg; unsigned int reg_tmp; + unsigned int idx; + + idx = snd_soc_cache_reg_to_idx(codec, reg); rbtree_ctx = codec->reg_cache; /* look up the required register in the cached rbnode */ rbnode = rbtree_ctx->cached_rbnode; if (rbnode) { snd_soc_rbtree_get_base_top_reg(rbnode, &base_reg, &top_reg); - if (reg >= base_reg && reg <= top_reg) { - reg_tmp = reg - base_reg; + if (idx >= base_reg && reg <= top_reg) { + reg_tmp = idx - base_reg; *value = snd_soc_rbtree_get_register(rbnode, reg_tmp); return 0; } @@ -360,9 +398,9 @@ static int snd_soc_rbtree_cache_read(struct snd_soc_codec *codec, /* if we can't locate it in the cached rbnode we'll have * to traverse the rbtree looking for it. */ - rbnode = snd_soc_rbtree_lookup(&rbtree_ctx->root, reg); + rbnode = snd_soc_rbtree_lookup(&rbtree_ctx->root, idx); if (rbnode) { - reg_tmp = reg - rbnode->base_reg; + reg_tmp = idx - rbnode->base_reg; *value = snd_soc_rbtree_get_register(rbnode, reg_tmp); rbtree_ctx->cached_rbnode = rbnode; } else { @@ -408,6 +446,7 @@ static int snd_soc_rbtree_cache_init(struct snd_soc_codec *codec) unsigned int val; int i; int ret; + unsigned int reg; codec->reg_cache = kmalloc(sizeof *rbtree_ctx, GFP_KERNEL); if (!codec->reg_cache) @@ -426,7 +465,8 @@ static int snd_soc_rbtree_cache_init(struct snd_soc_codec *codec) word_size); if (!val) continue; - ret = snd_soc_rbtree_cache_write(codec, i, val); + reg = snd_soc_cache_idx_to_reg(codec, i); + ret = snd_soc_rbtree_cache_write(codec, reg, val); if (ret) goto err; } @@ -560,21 +600,23 @@ static int snd_soc_lzo_cache_sync(struct snd_soc_codec *codec) unsigned int val; int i; int ret; + unsigned int reg; lzo_blocks = codec->reg_cache; for_each_set_bit(i, lzo_blocks[0]->sync_bmp, lzo_blocks[0]->sync_bmp_nbits) { + reg = snd_soc_cache_idx_to_reg(codec, i); WARN_ON(codec->writable_register && - codec->writable_register(codec, i)); - ret = snd_soc_cache_read(codec, i, &val); + codec->writable_register(codec, reg)); + ret = snd_soc_cache_read(codec, reg, &val); if (ret) return ret; codec->cache_bypass = 1; - ret = snd_soc_write(codec, i, val); + ret = snd_soc_write(codec, reg, val); codec->cache_bypass = 0; if (ret) return ret; dev_dbg(codec->dev, "Synced register %#x, value = %#x\n", - i, val); + reg, val); } return 0; @@ -587,11 +629,14 @@ static int snd_soc_lzo_cache_write(struct snd_soc_codec *codec, int ret, blkindex, blkpos; size_t blksize, tmp_dst_len; void *tmp_dst; + unsigned int idx; + + idx = snd_soc_cache_reg_to_idx(codec, reg); /* index of the compressed lzo block */ - blkindex = snd_soc_lzo_get_blkindex(codec, reg); + blkindex = snd_soc_lzo_get_blkindex(codec, idx); /* register index within the decompressed block */ - blkpos = snd_soc_lzo_get_blkpos(codec, reg); + blkpos = snd_soc_lzo_get_blkpos(codec, idx); /* size of the compressed block */ blksize = snd_soc_lzo_get_blksize(codec); lzo_blocks = codec->reg_cache; @@ -632,7 +677,7 @@ static int snd_soc_lzo_cache_write(struct snd_soc_codec *codec, } /* set the bit so we know we have to sync this register */ - set_bit(reg, lzo_block->sync_bmp); + set_bit(idx, lzo_block->sync_bmp); kfree(tmp_dst); kfree(lzo_block->src); return 0; @@ -649,12 +694,15 @@ static int snd_soc_lzo_cache_read(struct snd_soc_codec *codec, int ret, blkindex, blkpos; size_t blksize, tmp_dst_len; void *tmp_dst; + unsigned int idx; + + idx = snd_soc_cache_reg_to_idx(codec, reg); *value = 0; /* index of the compressed lzo block */ - blkindex = snd_soc_lzo_get_blkindex(codec, reg); + blkindex = snd_soc_lzo_get_blkindex(codec, idx); /* register index within the decompressed block */ - blkpos = snd_soc_lzo_get_blkpos(codec, reg); + blkpos = snd_soc_lzo_get_blkpos(codec, idx); /* size of the compressed block */ blksize = snd_soc_lzo_get_blksize(codec); lzo_blocks = codec->reg_cache; @@ -820,23 +868,25 @@ static int snd_soc_flat_cache_sync(struct snd_soc_codec *codec) int ret; const struct snd_soc_codec_driver *codec_drv; unsigned int val; + unsigned int reg; codec_drv = codec->driver; for (i = 0; i < codec_drv->reg_cache_size; ++i) { + reg = snd_soc_cache_idx_to_reg(codec, i); WARN_ON(codec->writable_register && - codec->writable_register(codec, i)); - ret = snd_soc_cache_read(codec, i, &val); + codec->writable_register(codec, reg)); + ret = snd_soc_cache_read(codec, reg, &val); if (ret) return ret; if (codec->reg_def_copy) if (snd_soc_get_cache_val(codec->reg_def_copy, - i, codec_drv->reg_word_size) == val) + reg, codec_drv->reg_word_size) == val) continue; - ret = snd_soc_write(codec, i, val); + ret = snd_soc_write(codec, reg, val); if (ret) return ret; dev_dbg(codec->dev, "Synced register %#x, value = %#x\n", - i, val); + reg, val); } return 0; } @@ -844,15 +894,24 @@ static int snd_soc_flat_cache_sync(struct snd_soc_codec *codec) static int snd_soc_flat_cache_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { - snd_soc_set_cache_val(codec->reg_cache, reg, value, + unsigned int idx; + + idx = snd_soc_cache_reg_to_idx(codec, reg); + + snd_soc_set_cache_val(codec->reg_cache, idx, value, codec->driver->reg_word_size); + return 0; } static int snd_soc_flat_cache_read(struct snd_soc_codec *codec, unsigned int reg, unsigned int *value) { - *value = snd_soc_get_cache_val(codec->reg_cache, reg, + unsigned int idx; + + idx = snd_soc_cache_reg_to_idx(codec, reg); + + *value = snd_soc_get_cache_val(codec->reg_cache, idx, codec->driver->reg_word_size); return 0; } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 83ad8ca..a23971e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -125,11 +125,12 @@ static int format_register_str(struct snd_soc_codec *codec, static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf, size_t count, loff_t pos) { - int i, step = 1; + int i; int wordsize, regsize; int len; size_t total = 0; loff_t p = 0; + unsigned int reg; wordsize = min_bytes_needed(codec->driver->reg_cache_size) * 2; regsize = codec->driver->reg_word_size * 2; @@ -139,22 +140,20 @@ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf, if (!codec->driver->reg_cache_size) return 0; - if (codec->driver->reg_cache_step) - step = codec->driver->reg_cache_step; - - for (i = 0; i < codec->driver->reg_cache_size; i += step) { - if (codec->readable_register && !codec->readable_register(codec, i)) + for (i = 0; i <= codec->driver->reg_cache_size; i++) { + reg = snd_soc_cache_idx_to_reg(codec, i); + if (codec->readable_register && !codec->readable_register(codec, reg)) continue; if (codec->driver->display_register) { count += codec->driver->display_register(codec, buf + count, - PAGE_SIZE - count, i); + PAGE_SIZE - count, reg); } else { /* only support larger than PAGE_SIZE bytes debugfs * entries for the default case */ if (p >= pos) { if (total + len >= count - 1) break; - format_register_str(codec, i, buf + total, len); + format_register_str(codec, reg, buf + total, len); total += len; } p += len; diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index cca490c..e5c15d3 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -35,9 +35,12 @@ static int do_hw_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value, const void *data, int len) { int ret; + unsigned int idx; + + idx = snd_soc_cache_reg_to_idx(codec, reg); if (!snd_soc_codec_volatile_register(codec, reg) && - reg < codec->driver->reg_cache_size && + idx < codec->driver->reg_cache_size && !codec->cache_bypass) { ret = snd_soc_cache_write(codec, reg, value); if (ret < 0) @@ -62,8 +65,11 @@ static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg) { int ret; unsigned int val; + unsigned int idx; + + idx = snd_soc_cache_reg_to_idx(codec, reg); - if (reg >= codec->driver->reg_cache_size || + if (idx >= codec->driver->reg_cache_size || snd_soc_codec_volatile_register(codec, reg) || codec->cache_bypass) { if (codec->cache_only) -- 1.7.0.4

6 34

Re: [alsa-devel] [PATCH v3] pass ELD to HDMI/DP audio driver
by Wu Fengguang 06 Aug '11

06 Aug '11

On Fri, Aug 05, 2011 at 02:03:41AM +0800, Keith Packard wrote: > On Thu, 4 Aug 2011 17:40:24 +0800, Wu Fengguang <fengguang.wu(a)intel.com> wrote: > > > Right. I actually have this chunk. dmesg shows that in intel_hdmi_detect(), > > the drm_encoder object is there, however encoder->crtc is NULL at the > > time. > > Correct. encoder->crtc is set only when the output is active. Ah OK. > > You may wonder why the mode parameter is needed in intel_write_eld(). > > This is because the ELD field aud_synch_delay (ie. A/V sync delay) may > > have different values in progressive/interleaved display modes. > > Ok, so you can't write ELD data until the display is active, which > happens at mode_set time. > > Do you need to provide ELD when the display is inactive? Is this only to > enable audio output when the display is not on? In that case, we will Good questions! In general the audio functionalities should not depend on the display activeness. There are even audio-only HDMI devices. So I'll need to make intel_write_eld() work even without information about the current display mode. > need to essentially turn enough of the output on to allow for audio > data, picking a mode which provide sufficient bandwidth. That would be a good feature. For one thing, I find it annoying that the music playback fades out when the screen goes to power saving mode.. > > I tested Ivybridge yesterday and ironlake_write_eld seems to not work > > properly for Ivybridge. Need to double check its spec. > > That's unfortunate... But fixable :) Thanks, Fengguang

3 2

[alsa-devel] [PATCH 0/3] 3 Patches for better UAC2 handling
by Daniel Mack 06 Aug '11

06 Aug '11

Sorry, I've been distracted, but here are 3 patches that should be included. Thanks, Daniel Daniel Mack (2): ALSA: snd-usb: Accept UAC2 FORMAT_TYPE descriptors with bLength > 6 ALSA: snd-usb: operate on given mixer interface only Nicolai Krakowiak (1): ALSA: snd-usb: avoid dividing by zero on invalid input sound/usb/endpoint.c | 2 +- sound/usb/mixer.c | 25 +++++++++++++------------ sound/usb/mixer.h | 1 + 3 files changed, 15 insertions(+), 13 deletions(-) -- 1.7.5.4

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[alsa-devel] [PATCH] ALSA: snd-usb-caiaq: Fix keymap for RigKontrol3
by Daniel Mack 06 Aug '11

06 Aug '11

Signed-off-by: Daniel Mack <zonque(a)gmail.com> Reported-by: Renato <naretobh(a)gmail.com> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: stable(a)kernel.org --- sound/usb/caiaq/input.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/usb/caiaq/input.c b/sound/usb/caiaq/input.c index 4432ef7..a213813 100644 --- a/sound/usb/caiaq/input.c +++ b/sound/usb/caiaq/input.c @@ -30,7 +30,7 @@ static unsigned short keycode_ak1[] = { KEY_C, KEY_B, KEY_A }; static unsigned short keycode_rk2[] = { KEY_1, KEY_2, KEY_3, KEY_4, KEY_5, KEY_6, KEY_7 }; static unsigned short keycode_rk3[] = { KEY_1, KEY_2, KEY_3, KEY_4, - KEY_5, KEY_6, KEY_7, KEY_5, KEY_6 }; + KEY_5, KEY_6, KEY_7, KEY_8, KEY_9 }; static unsigned short keycode_kore[] = { KEY_FN_F1, /* "menu" */ -- 1.7.6

2 1

[alsa-devel] [PATCH] ALSA: snd-usb: Fix uninitialized variable usage
by Daniel Mack 06 Aug '11

06 Aug '11

Purely cosmetic, but fixes the following build warning. CC [M] sound/usb/quirks.o sound/usb/quirks.c: In function ‘snd_usb_apply_boot_quirk’: sound/usb/quirks.c:429:6: warning: ‘err’ may be used uninitialized in this function [-Wuninitialized] Signed-off-by: Daniel Mack <zonque(a)gmail.com> Cc: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/quirks.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 77762c9..81e07d8 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -426,7 +426,7 @@ static int snd_usb_cm106_boot_quirk(struct usb_device *dev) */ static int snd_usb_cm6206_boot_quirk(struct usb_device *dev) { - int err, reg; + int err = 0, reg; int val[] = {0x2004, 0x3000, 0xf800, 0x143f, 0x0000, 0x3000}; for (reg = 0; reg < ARRAY_SIZE(val); reg++) { -- 1.7.6

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Re: [alsa-devel] Sound BoF in LinuxCon Europe?
by Mark Brown 05 Aug '11

05 Aug '11

On Sat, Jul 09, 2011 at 09:43:32AM +0200, Takashi Iwai wrote: > Mark Brown wrote: > > Yes, I don't think it's good for covering at KS as it's far too > > detailed. If we're going to do something probably LinuxCon/ELC Europe > > make most sense. > Well, KS day 1 is basically an open-day for workshops, AFAIK. > (The days 2 and 3 are different.) > So, it can be suitable for covering subsystem(s) specific problems. > Though, I also think LinuxCon/ELC would be a better choice, too. If nothing else we can always arrange something in the hallway/beer tracks.

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Re: [alsa-devel] [PATCH - add Keith McMillen Stringport to quirks tableadd quirk for Keith McMillen StringPort
by Takashi Iwai 05 Aug '11

05 Aug '11

At Thu, 4 Aug 2011 12:25:56 -0700, msp(a)ucsd.edu wrote: > > From: Miller Puckette <msp(a)ucsd.edu> > > > Signed-off-by: Miller Puckette <msp(a)ucsd.edu> Applied. Thanks. Takashi > diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h > index dba0b7f..4d4f865 100644 > --- a/sound/usb/quirks-table.h > +++ b/sound/usb/quirks-table.h > @@ -2417,6 +2417,12 @@ YAMAHA_DEVICE(0x7010, "UB99"), > .idProduct = 0x1020, > }, > > +/* KeithMcMillen Stringport */ > +{ > + USB_DEVICE(0x1f38, 0x0001), > + .bInterfaceClass = USB_CLASS_AUDIO, > +}, > + > /* Miditech devices */ > { > USB_DEVICE(0x4752, 0x0011), > -- > 1.7.2.2 >

1 0

[PATCH] ALSA: snd-usb-caiaq: Correct offset fields of outbound iso_frame_desc
by Daniel Mack 05 Aug '11

05 Aug '11

This fixes faulty outbount packets in case the inbound packets received from the hardware are fragmented and contain bogus input iso frames. The bug has been there for ages, but for some strange reasons, it was only triggered by newer machines in 64bit mode. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Reported-and-tested-by: William Light <wrl(a)illest.net> Reported-by: Pedro Ribeiro <pedrib(a)gmail.com> Cc: stable(a)kernel.org --- sound/usb/caiaq/audio.c | 6 ++++-- 1 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index d0d493c..aa52b3e 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -614,6 +614,7 @@ static void read_completed(struct urb *urb) struct snd_usb_caiaqdev *dev; struct urb *out; int frame, len, send_it = 0, outframe = 0; + size_t offset = 0; if (urb->status || !info) return; @@ -634,7 +635,8 @@ static void read_completed(struct urb *urb) len = urb->iso_frame_desc[outframe].actual_length; out->iso_frame_desc[outframe].length = len; out->iso_frame_desc[outframe].actual_length = 0; - out->iso_frame_desc[outframe].offset = BYTES_PER_FRAME * frame; + out->iso_frame_desc[outframe].offset = offset; + offset += len; if (len > 0) { spin_lock(&dev->spinlock); @@ -650,7 +652,7 @@ static void read_completed(struct urb *urb) } if (send_it) { - out->number_of_packets = FRAMES_PER_URB; + out->number_of_packets = outframe; out->transfer_flags = URB_ISO_ASAP; usb_submit_urb(out, GFP_ATOMIC); } -- 1.7.6 --------------000700020004070602000804--

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[alsa-devel] Suspend/Resume failure of Alsa but OSS emulation still works
by Shane Volpe 05 Aug '11

05 Aug '11

My embedded system's audio is not resuming after the system resumes from suspend to RAM, below is a detailed discription of the problem and what I have done, to date, to identify the source of the problem. I have tested the following scenario with several audio applications such as Alsa aplay and ogg123: If I have an audio player (aplay or ogg123) using Alsa and I suspend my system to RAM when the system wakes up (resumes) there is no sound. If I kill the audio player and start it back up the sound returns so it seems the act of opening and closing Alsa fixes the failure. During the "no sound state" I also opened alsamixer and adjusted (successfully) several alsa settings including volume but the sound still did not resume until I restarted the player. I also configured one player to use the OSS (emulated by Alsa). When I used the emulated OSS audio driver (/dev/dsp) the sound did resume after the system woke up (resumed) from memory so everything WORKED! I then configured the player back to use the native Alsa driver and again there was no sound after resume. Has anyone seen this issue before or does anyone have additional tests or debugging ideas they can recommend to further narrow down the issue. I'm very comfortable with modifying and compiling the Kernel so please do not limit your suggestions to user-space tweaks. Embedded System Information: Processor: ARM/Xscale (PXA270) Audio Codec: AC97 (UCB1400) alsa lib version: 1.0.24.1 Linux Kernel Version: 2.6.39.2 Regards, Shane

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[alsa-devel] [PATCH] ASoC: OMAP4: mcpdm: Fix IRQ handling
by Peter Ujfalusi 05 Aug '11

05 Aug '11

Use the correct variable holding the IRQ reason flags to check the reason for the interrupt. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hello Liam, this patch as well with my previous series for McPDM can go for 3.1, since this is fixing the interrupt handler function. Regards, Peter sound/soc/omap/mcpdm.c | 14 ++++++-------- 1 files changed, 6 insertions(+), 8 deletions(-) diff --git a/sound/soc/omap/mcpdm.c b/sound/soc/omap/mcpdm.c index 1594fe0..9746a49 100644 --- a/sound/soc/omap/mcpdm.c +++ b/sound/soc/omap/mcpdm.c @@ -283,41 +283,39 @@ static irqreturn_t omap_mcpdm_irq_handler(int irq, void *dev_id) /* Acknowledge irq event */ omap_mcpdm_write(MCPDM_IRQSTATUS, irq_status); - if (irq & MCPDM_DN_IRQ_FULL) { + if (irq_status & MCPDM_DN_IRQ_FULL) { dev_err(mcpdm_irq->dev, "DN FIFO error %x\n", irq_status); omap_mcpdm_reset_playback(1); omap_mcpdm_playback_open(mcpdm_irq->downlink); omap_mcpdm_reset_playback(0); } - if (irq & MCPDM_DN_IRQ_EMPTY) { + if (irq_status & MCPDM_DN_IRQ_EMPTY) { dev_err(mcpdm_irq->dev, "DN FIFO error %x\n", irq_status); omap_mcpdm_reset_playback(1); omap_mcpdm_playback_open(mcpdm_irq->downlink); omap_mcpdm_reset_playback(0); } - if (irq & MCPDM_DN_IRQ) { + if (irq_status & MCPDM_DN_IRQ) dev_dbg(mcpdm_irq->dev, "DN write request\n"); - } - if (irq & MCPDM_UP_IRQ_FULL) { + if (irq_status & MCPDM_UP_IRQ_FULL) { dev_err(mcpdm_irq->dev, "UP FIFO error %x\n", irq_status); omap_mcpdm_reset_capture(1); omap_mcpdm_capture_open(mcpdm_irq->uplink); omap_mcpdm_reset_capture(0); } - if (irq & MCPDM_UP_IRQ_EMPTY) { + if (irq_status & MCPDM_UP_IRQ_EMPTY) { dev_err(mcpdm_irq->dev, "UP FIFO error %x\n", irq_status); omap_mcpdm_reset_capture(1); omap_mcpdm_capture_open(mcpdm_irq->uplink); omap_mcpdm_reset_capture(0); } - if (irq & MCPDM_UP_IRQ) { + if (irq_status & MCPDM_UP_IRQ) dev_dbg(mcpdm_irq->dev, "UP write request\n"); - } return IRQ_HANDLED; } -- 1.7.6

2 1

[alsa-devel] USB mic delay problem on OMAP4 PandaBoard
by ABDUL MOIZ 05 Aug '11

05 Aug '11

Hi, I am working on OMAP4 platform (Panda Board) using ubuntu 10.10(kernel: 2.6.35) and recording audio with USB mic and getting the following problem: When I am recording with the arecord utility as shown below: *$ arecord -Dplughw:1,0 -f cd foo.wav * audio is recorded with XRUN. So we have tested by changing buffer size and period size as shown below, *$ arecord -Dplughw:1,0 -f cd --buffer-size=65536 --period-size=128 foo.wav * and we are getting good recorded audio quality. I found from other posts that buffer-size of 65536 is very large and it should work with size of only 1024. same usb mic works without xruns on my desktop with buffer size 1024. The problem is that the application has to be told to use insanely large buffers in order to avoid xruns. I suspect the problem is either in the kernel (possibly just a misconfiguration) or in hardware. So Can anyone suggest, how to solve the above problem? Regards, Abdul Moiz

1 0

[alsa-devel] snd soc spi read/write
by Scott Jiang 05 Aug '11

05 Aug '11

Hi Mark, My version is linux 3.0, I found snd_soc_spi_read/write didn't work at 16-8 bit mode. The addr is 16bit, the function compares reg with reg_cache_size. This certainly causes failure. In the former version, reg &= 0xff seems right. So how to use snd_soc_16_8_write now? Regards, Scott

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Re: [alsa-devel] [PATCH 1/3] irq: If an IRQ is a GPIO, request and configure it
by Stephen Warren 05 Aug '11

05 Aug '11

Rob Herring wrote at Thursday, August 04, 2011 7:55 PM: > On 08/04/2011 06:00 PM, Stephen Warren wrote: > > Many IRQs are associated with GPIO pins. When the pin is used as an IRQ, > > it can't be used as anything else; it should be requested. Enhance the > > core interrupt code to call gpio_request() and gpio_direction_input() for > > any IRQ that is also a GPIO. This prevents duplication of these calls in > > each driver that uses an IRQ. ... > > diff --git a/kernel/irq/manage.c b/kernel/irq/manage.c ... > > @@ -978,6 +979,16 @@ __setup_irq(unsigned int irq, struct irq_desc *desc, struct irqaction *new) > > old = *old_ptr; > > } while (old); > > shared = 1; > > + } else { > > + gpio = irq_to_gpio(irq); > > If you read the documentation for gpio, it is not recommended to use > irq_to_gpio. There's only a handful of users. Part of the problem is it > is platform specific and the gpio core cannot convert irq to gpio > number It seems like that's a soluble problem though? I was thinking about adding a to_gpio function to struct irq_chip, as the inverse of struct gpio_chip's to_irq. Then, presumably any platform would be able to convert back from IRQ to GPIO, provided the platform called a new __irq_to_gpio from the platform-specific gpio_to_irq, just like most gpio_to_irq implementations defer to __gpio_to_irq. I ended up not doing that in this patchset, since Tegra's gpio_to_irq function already works for the IRQs/GPIOs that were relevant for my testing, and I wanted to post a simple patch first to driver discussion. > Here is the relevant section: > > Non-error values returned from irq_to_gpio() would most commonly be used > with gpio_get_value(), for example to initialize or update driver state > when the IRQ is edge-triggered. Note that some platforms don't support > this reverse mapping, so you should avoid using it. -- nvpublic

1 0

[alsa-devel] Has OMAP4 officially installed AIC31xx driver yet?
by hong zhang 05 Aug '11

05 Aug '11

Liam, I am wondering if OMAP4 has officially installed AIC31xx driver or not. If yes, where can I get it? See following , I have one aic31xx driver but it has problems to be second sound card if use sdp4430_snd_device = platform_device_alloc("soc-audio", -1); in sdp4430.c omap4_snd_device = platform_device_alloc("soc-audio", 1); in omap4_panda_aic31xx.c or sdp4430_snd_device = platform_device_alloc("soc-audio", 0); in sdp4430.c omap4_snd_device = platform_device_alloc("soc-audio", 1); in omap4_panda_aic31xx.c when mounting such as Freeing init memory: 200K init: ureadahead main process (796) terminated with status 5 Unable to handle kernel paging request at virtual address 72656767 pgd = dc3b4000 [72656767] *pgd=00000000 Internal error: Oops: 5 [#1] PREEMPT SMP last sysfs file: /sys/devices/virtual/block/ram11/removable Modules linked in: CPU: 0 Not tainted (2.6.35.7-ctbu-5-ga320686-dirty #19) PC is at snd_soc_info_enum_double+0x5c/0x6c LR is at snd_ctl_elem_info_user+0xc0/0x1d8 pc : [<c0324204>] lr : [<c0311bd0>] psr: 80000113 sp : dc3c1d18 ip : dc3c1d30 fp : dc3c1d24 r10: dc040dc0 r9 : dc9f297c r8 : 00000000 r7 : dcae1b80 r6 : c1105511 r5 : dc3c1d30 r4 : dc9f2800 r3 : 00000000 r2 : 72656767 r1 : 00000000 r0 : dc3c1d88 Flags: Nzcv IRQs on FIQs on Mode SVC_32 ISA ARM Segment user Control: 10c53c7d Table: 9c3b404a DAC: 00000015 Thanks. ---Henry

1 0

Re: [alsa-devel] Regression in 3.0-git after commit c82693db52beced0419cecf09a3c81adfe95a544 - sound does not work
by Takashi Iwai 04 Aug '11

04 Aug '11

At Thu, 04 Aug 2011 13:19:17 -0500, Larry Finger wrote: > > On 08/04/2011 12:55 PM, Takashi Iwai wrote: > > > > Thanks. Through a quick glance at these information, the problem > > looks like that both newly appearing "Headphone" and "Speaker" volumes > > are set to zero. Usually alsactl invoked by udev will set appropriate > > values for new elements, but in your case, it didn't work by some > > reason. > > > > Try to adjust them. (Note that you don't have to rebuild the kernel > > just for this test; pass model=auto option to snd-hda-intel module, > > then it'll switch to that mode.) > > By adding that option to /etc/modprobe.d/99-local.conf, and removing my patch, I > got new controls with the mixer. After fiddling with them a bit, I got working > sound. The issue is fixed. OK, thanks for a quick check! BTW, are you using systemd or sysvinit? The culprit might be the missing call of "alsactl init" at the startup, and it depends on the start-up system, of course. Takashi

1 0

Re: [alsa-devel] Regression in 3.0-git after commit c82693db52beced0419cecf09a3c81adfe95a544 - sound does not work
by Takashi Iwai 04 Aug '11

04 Aug '11

At Thu, 04 Aug 2011 11:07:13 -0500, Larry Finger wrote: > > On 08/04/2011 12:17 AM, Takashi Iwai wrote: > > > > And reverting the above fixes your problem, right? > > Then please give alsa-info.sh outputs on both working and non-working > > kernels (run with --no-upload option). > > Yes, reverting that one hunk fixes my problem. > > The alsa-info.sh outputs for the working and nonworking cases are attached. Thanks. Through a quick glance at these information, the problem looks like that both newly appearing "Headphone" and "Speaker" volumes are set to zero. Usually alsactl invoked by udev will set appropriate values for new elements, but in your case, it didn't work by some reason. Try to adjust them. (Note that you don't have to rebuild the kernel just for this test; pass model=auto option to snd-hda-intel module, then it'll switch to that mode.) thanks, Takashi

1 0

[alsa-devel] [PATCH] ALSA: hda: option to enable arbitrary buffer/period sizes
by Pierre-Louis Bossart 04 Aug '11

04 Aug '11

Add new parameter to disable rounding of buffer/period sizes to multiples of 128 bytes. This is more efficient in terms of memory access but isn't required by the HDA spec and prevents users from specifying exact period/buffer sizes. For example for 44.1kHz, a period size set to 20ms will be rounded to 19.59ms. Tested and enabled on Intel HDA controllers. Option is disabled by default for other controllers. Tested-by: Wu Fengguang <fengguang.wu(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- Documentation/sound/alsa/ALSA-Configuration.txt | 5 ++ sound/pci/hda/hda_intel.c | 68 ++++++++++++++++++----- 2 files changed, 58 insertions(+), 15 deletions(-) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 8975701..27126c4 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -886,6 +886,11 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. disable) power_save_controller - Reset HD-audio controller in power-saving mode (default = on) + align_buffer_size - Force rounding of buffer/period sizes to multiples + of 128 bytes. This is more efficient in terms of memory + access but isn't required by the HDA spec and prevents + users from specifying exact period/buffer sizes. + (default = on) This module supports multiple cards and autoprobe. diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index be69822..2a8bed9 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -116,6 +116,11 @@ module_param(power_save_controller, bool, 0644); MODULE_PARM_DESC(power_save_controller, "Reset controller in power save mode."); #endif +static int align_buffer_size = 1; +module_param(align_buffer_size, bool, 0644); +MODULE_PARM_DESC(align_buffer_size, + "Force buffer and period sizes to be multiple of 128 bytes."); + MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Intel, ICH6}," "{Intel, ICH6M}," @@ -481,6 +486,7 @@ enum { #define AZX_DCAPS_NO_64BIT (1 << 18) /* No 64bit address */ #define AZX_DCAPS_SYNC_WRITE (1 << 19) /* sync each cmd write */ #define AZX_DCAPS_OLD_SSYNC (1 << 20) /* Old SSYNC reg for ICH */ +#define AZX_DCAPS_BUFSIZE (1 << 21) /* no buffer size alignment */ /* quirks for ATI SB / AMD Hudson */ #define AZX_DCAPS_PRESET_ATI_SB \ @@ -1599,6 +1605,7 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; unsigned long flags; int err; + int buff_step; mutex_lock(&chip->open_mutex); azx_dev = azx_assign_device(chip, substream); @@ -1613,10 +1620,25 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) runtime->hw.rates = hinfo->rates; snd_pcm_limit_hw_rates(runtime); snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); + if (align_buffer_size) + /* constrain buffer sizes to be multiple of 128 + bytes. This is more efficient in terms of memory + access but isn't required by the HDA spec and + prevents users from specifying exact period/buffer + sizes. For example for 44.1kHz, a period size set + to 20ms will be rounded to 19.59ms. */ + buff_step = 128; + else + /* Don't enforce steps on buffer sizes, still need to + be multiple of 4 bytes (HDA spec). Tested on Intel + HDA controllers, may not work on all devices where + option needs to be disabled */ + buff_step = 4; + snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, - 128); + buff_step); snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, - 128); + buff_step); snd_hda_power_up(apcm->codec); err = hinfo->ops.open(hinfo, apcm->codec, substream); if (err < 0) { @@ -2616,6 +2638,10 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, gcap &= ~ICH6_GCAP_64OK; } + /* disable buffer size rounding to 128-byte multiples if supported */ + if (chip->driver_caps & AZX_DCAPS_BUFSIZE) + align_buffer_size = 0; + /* allow 64bit DMA address if supported by H/W */ if ((gcap & ICH6_GCAP_64OK) && !pci_set_dma_mask(pci, DMA_BIT_MASK(64))) pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(64)); @@ -2817,37 +2843,49 @@ static void __devexit azx_remove(struct pci_dev *pci) static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { /* CPT */ { PCI_DEVICE(0x8086, 0x1c20), - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP }, + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | + AZX_DCAPS_BUFSIZE }, /* PBG */ { PCI_DEVICE(0x8086, 0x1d20), - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP }, + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | + AZX_DCAPS_BUFSIZE}, /* Panther Point */ { PCI_DEVICE(0x8086, 0x1e20), - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP }, + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | + AZX_DCAPS_BUFSIZE}, /* SCH */ { PCI_DEVICE(0x8086, 0x811b), - .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP }, + .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | + AZX_DCAPS_BUFSIZE}, { PCI_DEVICE(0x8086, 0x2668), - .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ICH6 */ + .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | + AZX_DCAPS_BUFSIZE }, /* ICH6 */ { PCI_DEVICE(0x8086, 0x27d8), - .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ICH7 */ + .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | + AZX_DCAPS_BUFSIZE }, /* ICH7 */ { PCI_DEVICE(0x8086, 0x269a), - .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ESB2 */ + .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | + AZX_DCAPS_BUFSIZE }, /* ESB2 */ { PCI_DEVICE(0x8086, 0x284b), - .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ICH8 */ + .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | + AZX_DCAPS_BUFSIZE }, /* ICH8 */ { PCI_DEVICE(0x8086, 0x293e), - .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ICH9 */ + .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | + AZX_DCAPS_BUFSIZE }, /* ICH9 */ { PCI_DEVICE(0x8086, 0x293f), - .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ICH9 */ + .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | + AZX_DCAPS_BUFSIZE }, /* ICH9 */ { PCI_DEVICE(0x8086, 0x3a3e), - .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ICH10 */ + .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | + AZX_DCAPS_BUFSIZE }, /* ICH10 */ { PCI_DEVICE(0x8086, 0x3a6e), - .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ICH10 */ + .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | + AZX_DCAPS_BUFSIZE }, /* ICH10 */ /* Generic Intel */ { PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_ANY_ID), .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8, .class_mask = 0xffffff, - .driver_data = AZX_DRIVER_ICH }, + .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_BUFSIZE }, /* ATI SB 450/600/700/800/900 */ { PCI_DEVICE(0x1002, 0x437b), .driver_data = AZX_DRIVER_ATI | AZX_DCAPS_PRESET_ATI_SB }, -- 1.7.6

2 1

[alsa-devel] [PATCH] Fix Jack plugin
by Kevin Ross 04 Aug '11

04 Aug '11

I've uploaded a patch to ticket 2727, at https://bugtrack.alsa-project.org/alsa-bug/view.php?id=2727 My background: I'm a software developer, but I know nothing about ALSA development. But I use Jack, and needed to get the Jack plugin working. Since no one has touched the plugin in years, I figured I'd take a stab at fixing it myself. The symptom: Using the Jack plugin causes whatever program is outputting audio to hang after about 10-20 seconds. The cause: The plugin creates a pipe, so that the fd's could be polled to indicate completion of a transfer. However, the call to write() blocks when playing audio, as nothing actually reads from the pipe, and it fills up until it blocks. The solution: This is where my inexperience with ALSA shows up. I don't know the best way to fix it. Making the socket non-blocking fixes the problem, and testing with playback and capture both work properly for me now. However, comparing the code with that of pcm-pulse.c (for piping ALSA output to PulseAudio), it's obvious that the Jack plugin was a hack. It could use more work. However, my fix at least gets me running. The patch: --- alsa-plugins-1.0.23/jack/pcm_jack.c 2010-04-16 04:11:10.000000000 -0700 +++ alsa-plugins-1.0.23.new/jack/pcm_jack.c 2011-08-03 21:54:51.795378947 -0700 @@ -307,6 +308,18 @@ return 0; } +static int make_nonblock(int fd) { + int fl; + + if ((fl = fcntl(fd, F_GETFL)) < 0) + return fl; + + if (fl & O_NONBLOCK) + return 0; + + return fcntl(fd, F_SETFL, fl | O_NONBLOCK); +} + static int snd_pcm_jack_open(snd_pcm_t **pcmp, const char *name, snd_config_t *playback_conf, snd_config_t *capture_conf, @@ -363,6 +376,9 @@ socketpair(AF_LOCAL, SOCK_STREAM, 0, fd); + make_nonblock(fd[0]); + make_nonblock(fd[1]); + jack->fd = fd[0]; jack->io.version = SND_PCM_IOPLUG_VERSION;

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[alsa-devel] ASoC updates for 3.1
by Mark Brown 04 Aug '11

04 Aug '11

The following changes since commit 06132fdf63c7d9acc19f136623bf2201f49e73b5: ASoC: Fix txx9aclc.c build (2011-07-29 07:53:08 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-3.1 Mark Brown (1): ASoC: Disable wm_hubs periodic DC servo update Wolfram Sang (1): ASoC: sgtl5000: fix cache handling sound/soc/codecs/sgtl5000.c | 128 ++++++++++++------------------------------- sound/soc/codecs/wm_hubs.c | 3 +- 2 files changed, 36 insertions(+), 95 deletions(-)

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[alsa-devel] ALSA pcm1808-at91sam9g20 driver
by Sergio Arroutbi 04 Aug '11

04 Aug '11

Hello. I would like to develop a driver for and ADC PCM1808. This PCM is connected to the SSC of a CPU ATMEL SAM91G20 Which driver/drivers could I have as reference? Is pcm3008 the best codec to look at? Thanks a lot -- Sergio Arroutbi Braojos Ingeniero de Desarrollo Software Grupo GMIS SEPSA, SISTEMAS CONTROL E INFORMACIÓN sarroutbi(a)sepsa.es Tel: (+34) 91 495 70 00 - Ext. 1587 Fax: (+34) 91 692 63 37 C/ALBATROS, 7 - 9 Polígono Industrial "Pinto Estación" 28320 - Pinto (MADRID)

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[alsa-devel] EeePC 1005PE microphone is "stereo only", silent in mono
by Pavel Roskin 04 Aug '11

04 Aug '11

Hello! I have discovered that the microphone on my ASUS EeePC 1005PE only works in stereo. If recording in mono, the result is inaudible. When I open the stereo recording in audacity, I see that the two channels are in counterphase. If I convert stereo to mono in audacity, I get a straight line, and it's inaudible when played. The driver itself requires stereo recording: $ arecord --rate 44100 -D hw:0 -f S16_LE --channels 1 >test.wav Recording WAVE 'stdin' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono arecord: set_params:1065: Channels count non available However, pulse audio doesn't enforce it. So it's possible to record a mono file through pulse audio, and it's inaudible. Some programs simply assume a mono microphone and thus get no sound. I can work it around by setting the level on one of the channels to 100% and to 0% on the other in alsamixer. However, the GNOME volume manager removes that imbalance when adjusting the input level. I would prefer to have a robust solution. Perhaps one of the channels should be inverted. Or maybe the difference between the channels should be reported as mono input. I don't really care about stereo input. I'm using Linux 3.0 on Fedora 15. I checked out and tested the master branch of sound-2.6.git, and it has the same problem. It looks like the problem is known. I found this comment in alc882_quirks.c: /* DMIC fix * This laptop has a stereo digital microphone. The mics are only 1cm apart * which makes the stereo useless. However, either the mic or the ALC889 * makes the signal become a difference/sum signal instead of standard * stereo, which is annoying. So instead we flip this bit which makes the * codec replicate the sum signal to both channels, turning it into a * normal mono mic. */ My laptop has ALC269, bit it also used the DMIC quirk: SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005HA", ALC269_DMIC), That's the detailed lspci output: $ lspci -vvnn -s 00:1b.0 00:1b.0 Audio device [0403]: Intel Corporation N10/ICH 7 Family High Definition Audio Controller [8086:27d8] (rev 02) Subsystem: ASUSTeK Computer Inc. Device [1043:83ce] Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 32 bytes Interrupt: pin A routed to IRQ 45 Region 0: Memory at f7cf8000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: HDA Intel I'm not using any module parameters. No external devices are connected to the laptop. -- Regards, Pavel Roskin

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Re: [alsa-devel] [PATCH v3] pass ELD to HDMI/DP audio driver
by Wu Fengguang 04 Aug '11

04 Aug '11

> On Mon, 1 Aug 2011 21:51:40 +0800, Wu Fengguang <fengguang.wu(a)intel.com> wrote: > > > 1) intel_write_eld() is not called at all > > It seems we need to call intel_write_eld() in other places besides > > inside ->mode_set(). Is ->detect() the right place to do so? In > > other words, are there established connector<=>encoder mapping > > that can be queried inside intel_hdmi_detect()/intel_dp_detect()? > > The connector has a link to the encoder (struct intel_connector contains a > pointer to a struct intel_encoder). Right. I actually have this chunk. dmesg shows that in intel_hdmi_detect(), the drm_encoder object is there, however encoder->crtc is NULL at the time. @@ -269,6 +271,12 @@ intel_hdmi_detect(struct drm_connector * status = connector_status_connected; intel_hdmi->has_hdmi_sink = drm_detect_hdmi_monitor(edid); intel_hdmi->has_audio = drm_detect_monitor_audio(edid); + drm_edid_to_eld(connector, edid); + if (intel_hdmi->base.base.crtc) + intel_write_eld(&intel_hdmi->base.base, + &intel_hdmi->base.base.crtc->mode); + else + printk("intel_hdmi_detect: cannot write eld: NULL crtc\n"); } connector->display_info.raw_edid = NULL; kfree(edid); You may wonder why the mode parameter is needed in intel_write_eld(). This is because the ELD field aud_synch_delay (ie. A/V sync delay) may have different values in progressive/interleaved display modes. > > 2) intel_dp_detect() is called even though it's an HDMI monitor > > connected to an HDMI jack.. It may be a bug specific to the > > hardware I'm testing (attached its full dmesg). > > We run all of the hotplug functions when any connector change is > detected; easier than trying to track what happened from the bits > visible from the interrupt. Ah yes. I figured it out later. > I note that this patch does not include the necessary hooks for > Ivybridge; we'll need that included (and tested) for this to be > considered for kernel 3.1. I'd be surprised if it couldn't use > ironlake_write_eld just fine, so it should just be a matter of setting > the write_eld field and testing it on actual hardware. I tested Ivybridge yesterday and ironlake_write_eld seems to not work properly for Ivybridge. Need to double check its spec. Thanks, Fengguang

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[alsa-devel] [PATCH] ASoC: Allow userspace control of Speyside headphone output
by Mark Brown 04 Aug '11

04 Aug '11

In order to facilitate the widest range of use cases (especially things like speakerphone) allow the headphone output to be enabled and disabled by the application layer. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/speyside.c | 5 +---- 1 files changed, 1 insertions(+), 4 deletions(-) diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index 590e927..bfed1ff 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -125,10 +125,6 @@ static struct snd_soc_jack_pin speyside_headset_pins[] = { .pin = "Headset Mic", .mask = SND_JACK_MICROPHONE, }, - { - .pin = "Headphone", - .mask = SND_JACK_HEADPHONE, - }, }; /* Default the headphone selection to active high */ @@ -252,6 +248,7 @@ static const struct snd_kcontrol_new controls[] = { SOC_DAPM_PIN_SWITCH("Main AMIC"), SOC_DAPM_PIN_SWITCH("WM1250 Input"), SOC_DAPM_PIN_SWITCH("WM1250 Output"), + SOC_DAPM_PIN_SWITCH("Headphone"), }; static struct snd_soc_dapm_widget widgets[] = { -- 1.7.5.4

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[alsa-devel] memory leak in mmap ops
by Scott Jiang 04 Aug '11

04 Aug '11

Hi all, I found there is a memory leak about 128K every time in linux-2.6.34, but it doesn't exist in linux-3.0. If I use mmap ops in driver the bug will come out, while use copy ops no memory leak. This driver is based on asoc. I tested with aplay to play a wav file. Because there is a lot of changes in ALSA between 2.6.34 and 3.0, so can anyone who familiar with the change history tell me which patch has solved this bug? Thanks in advance. Scott

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Re: [alsa-devel] Regression in 3.0-git after commit c82693db52beced0419cecf09a3c81adfe95a544 - sound does not work
by Takashi Iwai 04 Aug '11

04 Aug '11

At Wed, 03 Aug 2011 16:41:08 -0500, Larry Finger wrote: > > In the latest git pull of 3.0-git from Linus's tree, the sound on my HP laptop > does not work. This problem was bisected to the commit in the subject. > > The lspci output for my sound device is: > > 00:07.0 Audio device [0403]: nVidia Corporation MCP67 High Definition Audio > [10de:055c] (rev a1) > Subsystem: Hewlett-Packard Company Device [103c:30d6] > Kernel driver in use: snd_hda_intel > > I further isolated the problem to the following hunk: > > diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c > index 6e90b6b..4ca880b 100644 > --- a/sound/pci/hda/patch_conexant.c > +++ b/sound/pci/hda/patch_conexant.c > @@ -1994,10 +1990,8 @@ static int patch_cxt5051(struct hda_codec *codec) > board_config = snd_hda_check_board_config(codec, CXT5051_MODELS, > cxt5051_models, > cxt5051_cfg_tbl); > -#if 0 /* use the old method just for safety */ > if (board_config < 0) > - board_config = CXT5051_AUTO; > -#endif > + board_config = CXT5051_AUTO; /* model=auto as default */ > if (board_config == CXT5051_AUTO) > return patch_conexant_auto(codec); > > In my case, snd_hda_check_board_config() returns -1. Thus, the new code returns > the value of patch_conexant_auto(), whereas the original version fell through to > the setup code below. Clearly, this is needed for my device. And reverting the above fixes your problem, right? Then please give alsa-info.sh outputs on both working and non-working kernels (run with --no-upload option). thanks, Takashi

1 0

[alsa-devel] [PATCH RESEND] ASoC: sgtl5000: fix cache handling
by Wolfram Sang 03 Aug '11

03 Aug '11

Cache handling in this driver is broken. The chip has 16-bit registers, yet the register numbers also increase by 2 per register, i.e. there are only even-numbered registers. The cache in this driver, though, simply increments register numbers, so it does need some mapping as seen in sgtl5000_restore_regs(), note the '>> 1': snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL, cache[SGTL5000_CHIP_LINREG_CTRL >> 1]); That, of course, won't work with snd_soc_update_bits(). (Thus, we won't even notice the missing register 0x1c in the default regs which shifted all follwing registers to wrong values.) Noticed on the MX28EVK where enabling the regulators simply locked up the chip. Refactor the routines and use a properly sized default_regs array which matches the register layout of the underlying chip, i.e. create a truly flat cache. This also saves some code which should make up for the bigger array a little. When soc-core will somewhen have another cache type which handles a step size, this conversion will also ease the transition. Signed-off-by: Wolfram Sang <w.sang(a)pengutronix.de> Tested-by: Dong Aisheng <b29396(a)freescale.com> Tested-by: Shawn Guo <shawn.guo(a)linaro.org> Cc: Zeng Zhaoming <b32542(a)freescale.com> Cc: Liam Girdwood <lrg(a)ti.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- Resending as suggested in a previous thread. sound/soc/codecs/sgtl5000.c | 128 ++++++++++++------------------------------- 1 files changed, 35 insertions(+), 93 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 76258f2..7e4066e 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -33,73 +33,31 @@ #define SGTL5000_DAP_REG_OFFSET 0x0100 #define SGTL5000_MAX_REG_OFFSET 0x013A -/* default value of sgtl5000 registers except DAP */ -static const u16 sgtl5000_regs[SGTL5000_MAX_REG_OFFSET >> 1] = { - 0xa011, /* 0x0000, CHIP_ID. 11 stand for revison 17 */ - 0x0000, /* 0x0002, CHIP_DIG_POWER. */ - 0x0008, /* 0x0004, CHIP_CKL_CTRL */ - 0x0010, /* 0x0006, CHIP_I2S_CTRL */ - 0x0000, /* 0x0008, reserved */ - 0x0008, /* 0x000A, CHIP_SSS_CTRL */ - 0x0000, /* 0x000C, reserved */ - 0x020c, /* 0x000E, CHIP_ADCDAC_CTRL */ - 0x3c3c, /* 0x0010, CHIP_DAC_VOL */ - 0x0000, /* 0x0012, reserved */ - 0x015f, /* 0x0014, CHIP_PAD_STRENGTH */ - 0x0000, /* 0x0016, reserved */ - 0x0000, /* 0x0018, reserved */ - 0x0000, /* 0x001A, reserved */ - 0x0000, /* 0x001E, reserved */ - 0x0000, /* 0x0020, CHIP_ANA_ADC_CTRL */ - 0x1818, /* 0x0022, CHIP_ANA_HP_CTRL */ - 0x0111, /* 0x0024, CHIP_ANN_CTRL */ - 0x0000, /* 0x0026, CHIP_LINREG_CTRL */ - 0x0000, /* 0x0028, CHIP_REF_CTRL */ - 0x0000, /* 0x002A, CHIP_MIC_CTRL */ - 0x0000, /* 0x002C, CHIP_LINE_OUT_CTRL */ - 0x0404, /* 0x002E, CHIP_LINE_OUT_VOL */ - 0x7060, /* 0x0030, CHIP_ANA_POWER */ - 0x5000, /* 0x0032, CHIP_PLL_CTRL */ - 0x0000, /* 0x0034, CHIP_CLK_TOP_CTRL */ - 0x0000, /* 0x0036, CHIP_ANA_STATUS */ - 0x0000, /* 0x0038, reserved */ - 0x0000, /* 0x003A, CHIP_ANA_TEST2 */ - 0x0000, /* 0x003C, CHIP_SHORT_CTRL */ - 0x0000, /* reserved */ -}; - -/* default value of dap registers */ -static const u16 sgtl5000_dap_regs[] = { - 0x0000, /* 0x0100, DAP_CONTROL */ - 0x0000, /* 0x0102, DAP_PEQ */ - 0x0040, /* 0x0104, DAP_BASS_ENHANCE */ - 0x051f, /* 0x0106, DAP_BASS_ENHANCE_CTRL */ - 0x0000, /* 0x0108, DAP_AUDIO_EQ */ - 0x0040, /* 0x010A, DAP_SGTL_SURROUND */ - 0x0000, /* 0x010C, DAP_FILTER_COEF_ACCESS */ - 0x0000, /* 0x010E, DAP_COEF_WR_B0_MSB */ - 0x0000, /* 0x0110, DAP_COEF_WR_B0_LSB */ - 0x0000, /* 0x0112, reserved */ - 0x0000, /* 0x0114, reserved */ - 0x002f, /* 0x0116, DAP_AUDIO_EQ_BASS_BAND0 */ - 0x002f, /* 0x0118, DAP_AUDIO_EQ_BAND0 */ - 0x002f, /* 0x011A, DAP_AUDIO_EQ_BAND2 */ - 0x002f, /* 0x011C, DAP_AUDIO_EQ_BAND3 */ - 0x002f, /* 0x011E, DAP_AUDIO_EQ_TREBLE_BAND4 */ - 0x8000, /* 0x0120, DAP_MAIN_CHAN */ - 0x0000, /* 0x0122, DAP_MIX_CHAN */ - 0x0510, /* 0x0124, DAP_AVC_CTRL */ - 0x1473, /* 0x0126, DAP_AVC_THRESHOLD */ - 0x0028, /* 0x0128, DAP_AVC_ATTACK */ - 0x0050, /* 0x012A, DAP_AVC_DECAY */ - 0x0000, /* 0x012C, DAP_COEF_WR_B1_MSB */ - 0x0000, /* 0x012E, DAP_COEF_WR_B1_LSB */ - 0x0000, /* 0x0130, DAP_COEF_WR_B2_MSB */ - 0x0000, /* 0x0132, DAP_COEF_WR_B2_LSB */ - 0x0000, /* 0x0134, DAP_COEF_WR_A1_MSB */ - 0x0000, /* 0x0136, DAP_COEF_WR_A1_LSB */ - 0x0000, /* 0x0138, DAP_COEF_WR_A2_MSB */ - 0x0000, /* 0x013A, DAP_COEF_WR_A2_LSB */ +/* default value of sgtl5000 registers */ +static const u16 sgtl5000_regs[SGTL5000_MAX_REG_OFFSET] = { + [SGTL5000_CHIP_CLK_CTRL] = 0x0008, + [SGTL5000_CHIP_I2S_CTRL] = 0x0010, + [SGTL5000_CHIP_SSS_CTRL] = 0x0008, + [SGTL5000_CHIP_DAC_VOL] = 0x3c3c, + [SGTL5000_CHIP_PAD_STRENGTH] = 0x015f, + [SGTL5000_CHIP_ANA_HP_CTRL] = 0x1818, + [SGTL5000_CHIP_ANA_CTRL] = 0x0111, + [SGTL5000_CHIP_LINE_OUT_VOL] = 0x0404, + [SGTL5000_CHIP_ANA_POWER] = 0x7060, + [SGTL5000_CHIP_PLL_CTRL] = 0x5000, + [SGTL5000_DAP_BASS_ENHANCE] = 0x0040, + [SGTL5000_DAP_BASS_ENHANCE_CTRL] = 0x051f, + [SGTL5000_DAP_SURROUND] = 0x0040, + [SGTL5000_DAP_EQ_BASS_BAND0] = 0x002f, + [SGTL5000_DAP_EQ_BASS_BAND1] = 0x002f, + [SGTL5000_DAP_EQ_BASS_BAND2] = 0x002f, + [SGTL5000_DAP_EQ_BASS_BAND3] = 0x002f, + [SGTL5000_DAP_EQ_BASS_BAND4] = 0x002f, + [SGTL5000_DAP_MAIN_CHAN] = 0x8000, + [SGTL5000_DAP_AVC_CTRL] = 0x0510, + [SGTL5000_DAP_AVC_THRESHOLD] = 0x1473, + [SGTL5000_DAP_AVC_ATTACK] = 0x0028, + [SGTL5000_DAP_AVC_DECAY] = 0x0050, }; /* regulator supplies for sgtl5000, VDDD is an optional external supply */ @@ -1023,12 +981,10 @@ static int sgtl5000_suspend(struct snd_soc_codec *codec, pm_message_t state) static int sgtl5000_restore_regs(struct snd_soc_codec *codec) { u16 *cache = codec->reg_cache; - int i; - int regular_regs = SGTL5000_CHIP_SHORT_CTRL >> 1; + u16 reg; /* restore regular registers */ - for (i = 0; i < regular_regs; i++) { - int reg = i << 1; + for (reg = 0; reg <= SGTL5000_CHIP_SHORT_CTRL; reg += 2) { /* this regs depends on the others */ if (reg == SGTL5000_CHIP_ANA_POWER || @@ -1038,35 +994,31 @@ static int sgtl5000_restore_regs(struct snd_soc_codec *codec) reg == SGTL5000_CHIP_CLK_CTRL) continue; - snd_soc_write(codec, reg, cache[i]); + snd_soc_write(codec, reg, cache[reg]); } /* restore dap registers */ - for (i = SGTL5000_DAP_REG_OFFSET >> 1; - i < SGTL5000_MAX_REG_OFFSET >> 1; i++) { - int reg = i << 1; - - snd_soc_write(codec, reg, cache[i]); - } + for (reg = SGTL5000_DAP_REG_OFFSET; reg < SGTL5000_MAX_REG_OFFSET; reg += 2) + snd_soc_write(codec, reg, cache[reg]); /* * restore power and other regs according * to set_power() and set_clock() */ snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL, - cache[SGTL5000_CHIP_LINREG_CTRL >> 1]); + cache[SGTL5000_CHIP_LINREG_CTRL]); snd_soc_write(codec, SGTL5000_CHIP_ANA_POWER, - cache[SGTL5000_CHIP_ANA_POWER >> 1]); + cache[SGTL5000_CHIP_ANA_POWER]); snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, - cache[SGTL5000_CHIP_CLK_CTRL >> 1]); + cache[SGTL5000_CHIP_CLK_CTRL]); snd_soc_write(codec, SGTL5000_CHIP_REF_CTRL, - cache[SGTL5000_CHIP_REF_CTRL >> 1]); + cache[SGTL5000_CHIP_REF_CTRL]); snd_soc_write(codec, SGTL5000_CHIP_LINE_OUT_CTRL, - cache[SGTL5000_CHIP_LINE_OUT_CTRL >> 1]); + cache[SGTL5000_CHIP_LINE_OUT_CTRL]); return 0; } @@ -1454,16 +1406,6 @@ static __devinit int sgtl5000_i2c_probe(struct i2c_client *client, if (!sgtl5000) return -ENOMEM; - /* - * copy DAP default values to default value array. - * sgtl5000 register space has a big hole, merge it - * at init phase makes life easy. - * FIXME: should we drop 'const' of sgtl5000_regs? - */ - memcpy((void *)(&sgtl5000_regs[0] + (SGTL5000_DAP_REG_OFFSET >> 1)), - sgtl5000_dap_regs, - SGTL5000_MAX_REG_OFFSET - SGTL5000_DAP_REG_OFFSET); - i2c_set_clientdata(client, sgtl5000); ret = snd_soc_register_codec(&client->dev, -- 1.7.2.5

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[alsa-devel] I2C controller vs. WM8903 suspend ordering
by Stephen Warren 03 Aug '11

03 Aug '11

Mark, Liam, We're facing a suspend ordering issue on Tegra, when playing audio when suspend is initiated. Specifically, the I2C controller to which the WM8903 codec is attached is suspended before snd_soc_suspend is executed. This then prevents any I2C register accesses during snd_soc_suspend, so various analog paths are left active within the WM8903 during suspend, and this causes a pop noise during resume. (For reference, this is in the chromeos-2.6.38 kernel, with ASoC back- ported from a point prior to 2.6.39) We've found that switching the Tegra I2C controller from struct platform_driver.{suspend,resume} to .{suspend,resume}_noirq solves this. However, this seems like it's more of an accident than registering an explicit dependency, especially since most I2C controllers just use suspend rather than suspend_noirq. Is there a way to explicitly indicate that sound should be suspended before the I2C controller? I assume the issue is more that snd_soc_suspend shuts down various DAPM paths, which can't actually program the HW for this when this happens after I2C is shut down, rather than just the I2C controller being suspended before the WM8903 driver itself (which I assume doesn't happen, but didn't check). Thanks for any pointers! -- nvpublic

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[alsa-devel] Any patch for OMAP4 sound card on Linux 3.0.0-rc7?
by hong zhang 03 Aug '11

03 Aug '11

Team, Linux 3.0.0-rc7 complains following for OMAP4 on Linux 3.0.0-rc7 during boot SDP4430 SoC init Error: Driver 'hdmi-audio-dai' is already registered, aborting... Any patch available for the error or where I can look at? Thanks. ---henry

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Re: [alsa-devel] Copy ALSA buffer to the buffer allocated in my driver.
by Clemens Ladisch 02 Aug '11

02 Aug '11

`/ET!-i1S!-i wrote: > 1. I have no idea how to copy the alsa buffer to intermediate buffer (my > Buffer) for playback application . > 2. I have taken dummy driver code as example. > 3. Can you please help me. What code has to be included in the below copy > function to copy alsa buffer to my Buffer. Something like copy_from_user(). See sound/pci/rme9652/hdsp.c for an example. Regards, Clemens

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[PATCH] alsa-plugins: Pulse: only underrun if no more data has been written
by David Henningsson 02 Aug '11

02 Aug '11

If more data has already been written after the underrun, the underrun will automatically end and therefore we should not report it or restart the stream. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- pulse/pcm_pulse.c | 10 +++++++--- 1 files changed, 7 insertions(+), 3 deletions(-) diff --git a/pulse/pcm_pulse.c b/pulse/pcm_pulse.c index d6c6792..9c4a7e5 100644 --- a/pulse/pcm_pulse.c +++ b/pulse/pcm_pulse.c @@ -39,9 +39,10 @@ typedef struct snd_pcm_pulse { size_t last_size; size_t ptr; int underrun; - int handle_underrun; + int handle_underrun; /* can be 0=never, 1=always or 2=only if more data has not been written */ size_t offset; + int64_t written; pa_stream *stream; @@ -459,6 +460,7 @@ static snd_pcm_sframes_t pulse_write(snd_pcm_ioplug_t * io, } /* Make sure the buffer pointer is in sync */ + pcm->written += writebytes; pcm->last_size -= writebytes; ret = update_ptr(pcm); if (ret < 0) @@ -594,7 +596,8 @@ static void stream_underrun_cb(pa_stream * p, void *userdata) if (!pcm->p) return; - pcm->underrun = 1; + if (pcm->handle_underrun == 1 || pcm->written <= pa_stream_get_underflow_index(p)) + pcm->underrun = 1; } static void stream_latency_cb(pa_stream *p, void *userdata) { @@ -691,6 +694,7 @@ static int pulse_prepare(snd_pcm_ioplug_t * io) goto finish; } + pcm->written = 0; pa_stream_set_state_callback(pcm->stream, stream_state_cb, pcm); pa_stream_set_latency_update_callback(pcm->stream, stream_latency_cb, pcm); @@ -983,7 +987,7 @@ SND_PCM_PLUGIN_DEFINE_FUNC(pulse) const char *server = NULL; const char *device = NULL; const char *fallback_name = NULL; - int handle_underrun = 0; + int handle_underrun = 2; int err; snd_pcm_pulse_t *pcm; -- 1.7.4.1 --------------030800070508050501080505--

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Re: [alsa-devel] [PATCH 2/4] Fixes for "input: Add initial support for TWL6040 vibrator"
by Péter Ujfalusi 02 Aug '11

02 Aug '11

Hi Felipe, On Tuesday 02 August 2011 14:33:09 Balbi, Felipe wrote: > Hi, > > On Tue, Aug 02, 2011 at 02:28:42PM +0300, Peter Ujfalusi wrote: > > @@ -145,7 +143,7 @@ static int vibra_play(struct input_dev *input, void > > *data,> > > ret = queue_work(info->workqueue, &info->play_work); > > if (!ret) { > > > > - dev_err(&input->dev, "work is already on queue\n"); > > + dev_info(&input->dev, "work is already on queue\n"); > > why ??? Please do not waste your valuable time on this series. It was an error in my side to send it (just came back form vacation...). I have somehow ended up in a wrong directory when I typed git send-email, and sent the wrong set of patches. Not these, but the fixed/cleaned ones are already merged upstream. Sorry for the inconvenience I have caused. -- Péter

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Re: [alsa-devel] Copy ALSA buffer to the buffer allocated in my driver.
by Daniel Mack 02 Aug '11

02 Aug '11

Please don't drop any addresses from Cc:, and please quote original postings appropriately. On Tue, Aug 2, 2011 at 2:21 PM, `/ET!-i1S!-i <yethish7(a)gmail.com> wrote: > The driver i am writing is purely virtual . > > The role of my driver is to receive the audio data played in the application > program and stream it to remote system. Over the network? > For this purpose i need to copy the audio data from alsa buffer to my > Buffer. I have no idea how this can be done. So i am looking for sample code > to copy the audio data to my Buffer. I'd bet there is no need to write any ALSA driver for this. You can either use the ALSA plugin mechanism to achive this, or even some higher level. PulseAudio for example allows seamless streaming over the network out of the box. Why are you confident that you need to write a driver? Daniel

1 0

[alsa-devel] Copy ALSA buffer to the buffer allocated in my driver.
by `/ET!-i1S!-i 02 Aug '11

02 Aug '11

Hi all, I am writing my own alsa driver for playback application.1. I have no idea how to copy the alsa buffer to intermediate buffer (my Buffer) . 2. I have taken dummy driver code as example. 3. Can anyone help me. What code has to be included in the dummy driver to copy alsa buffer to intermediate buffer. Regards Cherry

2 1

[alsa-devel] [PATCH 1/3] mfd: twl6040: Add initial support
by Peter Ujfalusi 02 Aug '11

02 Aug '11

From: Misael Lopez Cruz <misael.lopez(a)ti.com> TWL6040 IC provides analog high-end audio codec functions for handset applications. It contains several audio analog inputs and outputs as well as vibrator support. It's connected to the host processor via PDM interface for audio data communication. The audio modules are controlled by internal registers that can be accessed by I2C and PDM interface. TWL6040 MFD will be registered as a child of TWL-CORE, and will have two children of its own: twl6040-codec and twl6040-vibra. This driver is based on TWL4030 and WM8350 MFD drivers. Signed-off-by: Misael Lopez Cruz <misael.lopez(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> --- arch/arm/plat-omap/include/plat/irqs.h | 12 +- drivers/mfd/Kconfig | 6 + drivers/mfd/Makefile | 1 + drivers/mfd/twl-core.c | 4 +- drivers/mfd/twl6040-codec.c | 587 ++++++++++++++++++++++++++++++++ drivers/mfd/twl6040-irq.c | 205 +++++++++++ include/linux/i2c/twl.h | 1 + include/linux/mfd/twl6040-codec.h | 260 ++++++++++++++ 8 files changed, 1072 insertions(+), 4 deletions(-) create mode 100644 drivers/mfd/twl6040-codec.c create mode 100644 drivers/mfd/twl6040-irq.c create mode 100644 include/linux/mfd/twl6040-codec.h diff --git a/arch/arm/plat-omap/include/plat/irqs.h b/arch/arm/plat-omap/include/plat/irqs.h index 5a25098..2cfba51 100644 --- a/arch/arm/plat-omap/include/plat/irqs.h +++ b/arch/arm/plat-omap/include/plat/irqs.h @@ -407,11 +407,19 @@ #endif #define TWL6030_IRQ_END (TWL6030_IRQ_BASE + TWL6030_BASE_NR_IRQS) +#define TWL6040_CODEC_IRQ_BASE TWL6030_IRQ_END +#ifdef CONFIG_TWL6040_CODEC +#define TWL6040_CODEC_NR_IRQS 6 +#else +#define TWL6040_CODEC_NR_IRQS 0 +#endif +#define TWL6040_CODEC_IRQ_END (TWL6040_CODEC_IRQ_BASE + TWL6040_CODEC_NR_IRQS) + /* Total number of interrupts depends on the enabled blocks above */ -#if (TWL4030_GPIO_IRQ_END > TWL6030_IRQ_END) +#if (TWL4030_GPIO_IRQ_END > TWL6040_CODEC_IRQ_END) #define TWL_IRQ_END TWL4030_GPIO_IRQ_END #else -#define TWL_IRQ_END TWL6030_IRQ_END +#define TWL_IRQ_END TWL6040_CODEC_IRQ_END #endif /* GPMC related */ diff --git a/drivers/mfd/Kconfig b/drivers/mfd/Kconfig index 2b4cae0..c05ec17 100644 --- a/drivers/mfd/Kconfig +++ b/drivers/mfd/Kconfig @@ -233,6 +233,12 @@ config TWL6030_PWM Say yes here if you want support for TWL6030 PWM. This is used to control charging LED brightness. +config TWL6040_CODEC + bool + depends on TWL4030_CORE && GENERIC_HARDIRQS + select MFD_CORE + default n + config MFD_STMPE bool "Support STMicroelectronics STMPE" depends on I2C=y && GENERIC_HARDIRQS diff --git a/drivers/mfd/Makefile b/drivers/mfd/Makefile index f4afcc8..1a04029 100644 --- a/drivers/mfd/Makefile +++ b/drivers/mfd/Makefile @@ -42,6 +42,7 @@ obj-$(CONFIG_TWL4030_MADC) += twl4030-madc.o obj-$(CONFIG_TWL4030_POWER) += twl4030-power.o obj-$(CONFIG_TWL4030_CODEC) += twl4030-codec.o obj-$(CONFIG_TWL6030_PWM) += twl6030-pwm.o +obj-$(CONFIG_TWL6040_CODEC) += twl6040-codec.o twl6040-irq.o obj-$(CONFIG_MFD_MC13XXX) += mc13xxx-core.o diff --git a/drivers/mfd/twl-core.c b/drivers/mfd/twl-core.c index 2bd9e06..cb37c08 100644 --- a/drivers/mfd/twl-core.c +++ b/drivers/mfd/twl-core.c @@ -110,7 +110,7 @@ #endif #if defined(CONFIG_TWL4030_CODEC) || defined(CONFIG_TWL4030_CODEC_MODULE) ||\ - defined(CONFIG_SND_SOC_TWL6040) || defined(CONFIG_SND_SOC_TWL6040_MODULE) + defined(CONFIG_TWL6040_CODEC) || defined(CONFIG_TWL6040_CODEC_MODULE) #define twl_has_codec() true #else #define twl_has_codec() false @@ -801,7 +801,7 @@ add_children(struct twl4030_platform_data *pdata, unsigned long features) /* Phoenix codec driver is probed directly atm */ if (twl_has_codec() && pdata->codec && twl_class_is_6030()) { sub_chip_id = twl_map[TWL_MODULE_AUDIO_VOICE].sid; - child = add_child(sub_chip_id, "twl6040-codec", + child = add_child(sub_chip_id, "twl6040-audio", pdata->codec, sizeof(*pdata->codec), false, 0, 0); if (IS_ERR(child)) diff --git a/drivers/mfd/twl6040-codec.c b/drivers/mfd/twl6040-codec.c new file mode 100644 index 0000000..a40cd07 --- /dev/null +++ b/drivers/mfd/twl6040-codec.c @@ -0,0 +1,587 @@ +/* + * MFD driver for TWL6040 codec submodule + * + * Authors: Misael Lopez Cruz <misael.lopez(a)ti.com> + * Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> + * + * Copyright: (C) 2011 Texas Instruments, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/module.h> +#include <linux/types.h> +#include <linux/slab.h> +#include <linux/kernel.h> +#include <linux/platform_device.h> +#include <linux/gpio.h> +#include <linux/delay.h> +#include <linux/i2c/twl.h> +#include <linux/mfd/core.h> +#include <linux/mfd/twl6040-codec.h> + +static struct platform_device *twl6040_dev; + +int twl6040_reg_read(struct twl6040 *twl6040, unsigned int reg) +{ + int ret; + u8 val = 0; + + mutex_lock(&twl6040->io_mutex); + ret = twl_i2c_read_u8(TWL_MODULE_AUDIO_VOICE, &val, reg); + if (ret < 0) { + mutex_unlock(&twl6040->io_mutex); + return ret; + } + mutex_unlock(&twl6040->io_mutex); + + return val; +} +EXPORT_SYMBOL(twl6040_reg_read); + +int twl6040_reg_write(struct twl6040 *twl6040, unsigned int reg, u8 val) +{ + int ret; + + mutex_lock(&twl6040->io_mutex); + ret = twl_i2c_write_u8(TWL_MODULE_AUDIO_VOICE, val, reg); + mutex_unlock(&twl6040->io_mutex); + + return ret; +} +EXPORT_SYMBOL(twl6040_reg_write); + +int twl6040_set_bits(struct twl6040 *twl6040, unsigned int reg, u8 mask) +{ + int ret; + u8 val; + + mutex_lock(&twl6040->io_mutex); + ret = twl_i2c_read_u8(TWL_MODULE_AUDIO_VOICE, &val, reg); + if (ret) + goto out; + + val |= mask; + ret = twl_i2c_write_u8(TWL_MODULE_AUDIO_VOICE, val, reg); +out: + mutex_unlock(&twl6040->io_mutex); + return ret; +} +EXPORT_SYMBOL(twl6040_set_bits); + +int twl6040_clear_bits(struct twl6040 *twl6040, unsigned int reg, u8 mask) +{ + int ret; + u8 val; + + mutex_lock(&twl6040->io_mutex); + ret = twl_i2c_read_u8(TWL_MODULE_AUDIO_VOICE, &val, reg); + if (ret) + goto out; + + val &= ~mask; + ret = twl_i2c_write_u8(TWL_MODULE_AUDIO_VOICE, val, reg); +out: + mutex_unlock(&twl6040->io_mutex); + return ret; +} +EXPORT_SYMBOL(twl6040_clear_bits); + +/* twl6040 codec manual power-up sequence */ +static int twl6040_power_up(struct twl6040 *twl6040) +{ + u8 ldoctl, ncpctl, lppllctl; + int ret; + + /* enable high-side LDO, reference system and internal oscillator */ + ldoctl = TWL6040_HSLDOENA | TWL6040_REFENA | TWL6040_OSCENA; + ret = twl6040_reg_write(twl6040, TWL6040_REG_LDOCTL, ldoctl); + if (ret) + return ret; + usleep_range(10000, 10500); + + /* enable negative charge pump */ + ncpctl = TWL6040_NCPENA; + ret = twl6040_reg_write(twl6040, TWL6040_REG_NCPCTL, ncpctl); + if (ret) + goto ncp_err; + usleep_range(1000, 1500); + + /* enable low-side LDO */ + ldoctl |= TWL6040_LSLDOENA; + ret = twl6040_reg_write(twl6040, TWL6040_REG_LDOCTL, ldoctl); + if (ret) + goto lsldo_err; + usleep_range(1000, 1500); + + /* enable low-power PLL */ + lppllctl = TWL6040_LPLLENA; + ret = twl6040_reg_write(twl6040, TWL6040_REG_LPPLLCTL, lppllctl); + if (ret) + goto lppll_err; + usleep_range(5000, 5500); + + /* disable internal oscillator */ + ldoctl &= ~TWL6040_OSCENA; + ret = twl6040_reg_write(twl6040, TWL6040_REG_LDOCTL, ldoctl); + if (ret) + goto osc_err; + + return 0; + +osc_err: + lppllctl &= ~TWL6040_LPLLENA; + twl6040_reg_write(twl6040, TWL6040_REG_LPPLLCTL, lppllctl); +lppll_err: + ldoctl &= ~TWL6040_LSLDOENA; + twl6040_reg_write(twl6040, TWL6040_REG_LDOCTL, ldoctl); +lsldo_err: + ncpctl &= ~TWL6040_NCPENA; + twl6040_reg_write(twl6040, TWL6040_REG_NCPCTL, ncpctl); +ncp_err: + ldoctl &= ~(TWL6040_HSLDOENA | TWL6040_REFENA | TWL6040_OSCENA); + twl6040_reg_write(twl6040, TWL6040_REG_LDOCTL, ldoctl); + + return ret; +} + +/* twl6040 codec manual power-down sequence */ +static void twl6040_power_down(struct twl6040 *twl6040) +{ + u8 ncpctl, ldoctl, lppllctl; + + ncpctl = twl6040_reg_read(twl6040, TWL6040_REG_NCPCTL); + ldoctl = twl6040_reg_read(twl6040, TWL6040_REG_LDOCTL); + lppllctl = twl6040_reg_read(twl6040, TWL6040_REG_LPPLLCTL); + + /* enable internal oscillator */ + ldoctl |= TWL6040_OSCENA; + twl6040_reg_write(twl6040, TWL6040_REG_LDOCTL, ldoctl); + usleep_range(1000, 1500); + + /* disable low-power PLL */ + lppllctl &= ~TWL6040_LPLLENA; + twl6040_reg_write(twl6040, TWL6040_REG_LPPLLCTL, lppllctl); + + /* disable low-side LDO */ + ldoctl &= ~TWL6040_LSLDOENA; + twl6040_reg_write(twl6040, TWL6040_REG_LDOCTL, ldoctl); + + /* disable negative charge pump */ + ncpctl &= ~TWL6040_NCPENA; + twl6040_reg_write(twl6040, TWL6040_REG_NCPCTL, ncpctl); + + /* disable high-side LDO, reference system and internal oscillator */ + ldoctl &= ~(TWL6040_HSLDOENA | TWL6040_REFENA | TWL6040_OSCENA); + twl6040_reg_write(twl6040, TWL6040_REG_LDOCTL, ldoctl); +} + +static irqreturn_t twl6040_naudint_handler(int irq, void *data) +{ + struct twl6040 *twl6040 = data; + u8 intid; + + intid = twl6040_reg_read(twl6040, TWL6040_REG_INTID); + + if (intid & TWL6040_READYINT) + complete(&twl6040->ready); + + return IRQ_HANDLED; +} + +static int twl6040_power_up_completion(struct twl6040 *twl6040, + int naudint) +{ + int time_left; + u8 intid; + + time_left = wait_for_completion_timeout(&twl6040->ready, + msecs_to_jiffies(144)); + if (!time_left) { + intid = twl6040_reg_read(twl6040, TWL6040_REG_INTID); + if (!(intid & TWL6040_READYINT)) { + dev_err(&twl6040_dev->dev, + "timeout waiting for READYINT\n"); + return -ETIMEDOUT; + } + } + + return 0; +} + +int twl6040_power(struct twl6040 *twl6040, int on) +{ + int audpwron = twl6040->audpwron; + int naudint = twl6040->irq; + int ret = 0; + + mutex_lock(&twl6040->mutex); + + if (on) { + /* already powered-up */ + if (twl6040->power_count++) + goto out; + + if (gpio_is_valid(audpwron)) { + /* use AUDPWRON line */ + gpio_set_value(audpwron, 1); + /* wait for power-up completion */ + ret = twl6040_power_up_completion(twl6040, naudint); + if (ret) { + dev_err(&twl6040_dev->dev, + "automatic power-down failed\n"); + twl6040->power_count = 0; + goto out; + } + } else { + /* use manual power-up sequence */ + ret = twl6040_power_up(twl6040); + if (ret) { + dev_err(&twl6040_dev->dev, + "manual power-up failed\n"); + twl6040->power_count = 0; + goto out; + } + } + twl6040->pll = TWL6040_LPPLL_ID; + twl6040->sysclk = 19200000; + } else { + /* already powered-down */ + if (!twl6040->power_count) { + dev_err(&twl6040_dev->dev, + "device is already powered-off\n"); + ret = -EPERM; + goto out; + } + + if (--twl6040->power_count) + goto out; + + if (gpio_is_valid(audpwron)) { + /* use AUDPWRON line */ + gpio_set_value(audpwron, 0); + + /* power-down sequence latency */ + udelay(500); + } else { + /* use manual power-down sequence */ + twl6040_power_down(twl6040); + } + twl6040->pll = TWL6040_NOPLL_ID; + twl6040->sysclk = 0; + } + +out: + mutex_unlock(&twl6040->mutex); + return ret; +} +EXPORT_SYMBOL(twl6040_power); + +int twl6040_is_powered(struct twl6040 *twl6040) +{ + return twl6040->power_count; +} +EXPORT_SYMBOL(twl6040_is_powered); + +int twl6040_set_pll(struct twl6040 *twl6040, enum twl6040_pll_id id, + unsigned int freq_in, unsigned int freq_out) +{ + u8 hppllctl, lppllctl; + int ret = 0; + + mutex_lock(&twl6040->mutex); + + hppllctl = twl6040_reg_read(twl6040, TWL6040_REG_HPPLLCTL); + lppllctl = twl6040_reg_read(twl6040, TWL6040_REG_LPPLLCTL); + + switch (id) { + case TWL6040_LPPLL_ID: + /* low-power PLL divider */ + switch (freq_out) { + case 17640000: + lppllctl |= TWL6040_LPLLFIN; + break; + case 19200000: + lppllctl &= ~TWL6040_LPLLFIN; + break; + default: + dev_err(&twl6040_dev->dev, + "freq_out %d not supported\n", freq_out); + ret = -EINVAL; + goto pll_out; + } + twl6040_reg_write(twl6040, TWL6040_REG_LPPLLCTL, lppllctl); + + switch (freq_in) { + case 32768: + lppllctl |= TWL6040_LPLLENA; + twl6040_reg_write(twl6040, TWL6040_REG_LPPLLCTL, + lppllctl); + mdelay(5); + lppllctl &= ~TWL6040_HPLLSEL; + twl6040_reg_write(twl6040, TWL6040_REG_LPPLLCTL, + lppllctl); + hppllctl &= ~TWL6040_HPLLENA; + twl6040_reg_write(twl6040, TWL6040_REG_HPPLLCTL, + hppllctl); + break; + default: + dev_err(&twl6040_dev->dev, + "freq_in %d not supported\n", freq_in); + ret = -EINVAL; + goto pll_out; + } + + twl6040->pll = TWL6040_LPPLL_ID; + break; + case TWL6040_HPPLL_ID: + /* high-performance PLL can provide only 19.2 MHz */ + if (freq_out != 19200000) { + dev_err(&twl6040_dev->dev, + "freq_out %d not supported\n", freq_out); + ret = -EINVAL; + goto pll_out; + } + + hppllctl &= ~TWL6040_MCLK_MSK; + + switch (freq_in) { + case 12000000: + /* PLL enabled, active mode */ + hppllctl |= TWL6040_MCLK_12000KHZ | + TWL6040_HPLLENA; + break; + case 19200000: + /* + * PLL disabled + * (enable PLL if MCLK jitter quality + * doesn't meet specification) + */ + hppllctl |= TWL6040_MCLK_19200KHZ; + break; + case 26000000: + /* PLL enabled, active mode */ + hppllctl |= TWL6040_MCLK_26000KHZ | + TWL6040_HPLLENA; + break; + case 38400000: + /* PLL enabled, active mode */ + hppllctl |= TWL6040_MCLK_38400KHZ | + TWL6040_HPLLENA; + break; + default: + dev_err(&twl6040_dev->dev, + "freq_in %d not supported\n", freq_in); + ret = -EINVAL; + goto pll_out; + } + + /* enable clock slicer to ensure input waveform is square */ + hppllctl |= TWL6040_HPLLSQRENA; + + twl6040_reg_write(twl6040, TWL6040_REG_HPPLLCTL, hppllctl); + udelay(500); + lppllctl |= TWL6040_HPLLSEL; + twl6040_reg_write(twl6040, TWL6040_REG_LPPLLCTL, lppllctl); + lppllctl &= ~TWL6040_LPLLENA; + twl6040_reg_write(twl6040, TWL6040_REG_LPPLLCTL, lppllctl); + + twl6040->pll = TWL6040_HPPLL_ID; + break; + default: + dev_err(&twl6040_dev->dev, "unknown pll id %d\n", id); + ret = -EINVAL; + goto pll_out; + } + + twl6040->sysclk = freq_out; + +pll_out: + mutex_unlock(&twl6040->mutex); + return ret; +} +EXPORT_SYMBOL(twl6040_set_pll); + +enum twl6040_pll_id twl6040_get_pll(struct twl6040 *twl6040) +{ + return twl6040->pll; +} +EXPORT_SYMBOL(twl6040_get_pll); + +unsigned int twl6040_get_sysclk(struct twl6040 *twl6040) +{ + return twl6040->sysclk; +} +EXPORT_SYMBOL(twl6040_get_sysclk); + +static int __devinit twl6040_probe(struct platform_device *pdev) +{ + struct twl4030_codec_data *pdata = pdev->dev.platform_data; + struct twl6040 *twl6040; + struct mfd_cell *cell = NULL; + int ret, children = 0; + + if (!pdata) { + dev_err(&pdev->dev, "Platform data is missing\n"); + return -EINVAL; + } + + twl6040 = kzalloc(sizeof(struct twl6040), GFP_KERNEL); + if (!twl6040) + return -ENOMEM; + + platform_set_drvdata(pdev, twl6040); + + twl6040_dev = pdev; + twl6040->dev = &pdev->dev; + twl6040->audpwron = pdata->audpwron_gpio; + twl6040->irq = pdata->naudint_irq; + twl6040->irq_base = pdata->irq_base; + + mutex_init(&twl6040->mutex); + mutex_init(&twl6040->io_mutex); + init_completion(&twl6040->ready); + + twl6040->rev = twl6040_reg_read(twl6040, TWL6040_REG_ASICREV); + + if (gpio_is_valid(twl6040->audpwron)) { + ret = gpio_request(twl6040->audpwron, "audpwron"); + if (ret) + goto gpio1_err; + + ret = gpio_direction_output(twl6040->audpwron, 0); + if (ret) + goto gpio2_err; + } + + /* ERRATA: Automatic power-up is not possible in ES1.0 */ + if (twl6040_get_rev(twl6040) == TWL6040_REV_ES1_0) + twl6040->audpwron = -EINVAL; + + if (twl6040->irq) { + /* codec interrupt */ + ret = twl6040_irq_init(twl6040); + if (ret) + goto gpio2_err; + + ret = twl6040_request_irq(twl6040, TWL6040_IRQ_READY, + twl6040_naudint_handler, 0, + "twl6040_irq_ready", twl6040); + if (ret) { + dev_err(twl6040->dev, "READY IRQ request failed: %d\n", + ret); + goto irq_err; + } + } + + /* dual-access registers controlled by I2C only */ + twl6040_set_bits(twl6040, TWL6040_REG_ACCCTL, TWL6040_I2CSEL); + + if (pdata->audio) { + cell = &twl6040->cells[children]; + cell->name = "twl6040-codec"; + cell->platform_data = pdata->audio; + cell->pdata_size = sizeof(*pdata->audio); + children++; + } + + if (pdata->vibra) { + cell = &twl6040->cells[children]; + cell->name = "twl6040-vibra"; + cell->platform_data = pdata->vibra; + cell->pdata_size = sizeof(*pdata->vibra); + children++; + } + + if (children) { + ret = mfd_add_devices(&pdev->dev, pdev->id, twl6040->cells, + children, NULL, 0); + if (ret) + goto mfd_err; + } else { + dev_err(&pdev->dev, "No platform data found for children\n"); + ret = -ENODEV; + goto mfd_err; + } + + return 0; + +mfd_err: + if (twl6040->irq) + twl6040_free_irq(twl6040, TWL6040_IRQ_READY, twl6040); +irq_err: + if (twl6040->irq) + twl6040_irq_exit(twl6040); +gpio2_err: + if (gpio_is_valid(twl6040->audpwron)) + gpio_free(twl6040->audpwron); +gpio1_err: + platform_set_drvdata(pdev, NULL); + kfree(twl6040); + twl6040_dev = NULL; + return ret; +} + +static int __devexit twl6040_remove(struct platform_device *pdev) +{ + struct twl6040 *twl6040 = platform_get_drvdata(pdev); + + if (twl6040_is_powered(twl6040)) + twl6040_power(twl6040, 0); + + if (gpio_is_valid(twl6040->audpwron)) + gpio_free(twl6040->audpwron); + + twl6040_free_irq(twl6040, TWL6040_IRQ_READY, twl6040); + + if (twl6040->irq) + twl6040_irq_exit(twl6040); + + mfd_remove_devices(&pdev->dev); + platform_set_drvdata(pdev, NULL); + kfree(twl6040); + twl6040_dev = NULL; + + return 0; +} + +static struct platform_driver twl6040_driver = { + .probe = twl6040_probe, + .remove = __devexit_p(twl6040_remove), + .driver = { + .owner = THIS_MODULE, + .name = "twl6040-audio", + }, +}; + +static int __devinit twl6040_init(void) +{ + return platform_driver_register(&twl6040_driver); +} +module_init(twl6040_init); + +static void __devexit twl6040_exit(void) +{ + platform_driver_unregister(&twl6040_driver); +} + +module_exit(twl6040_exit); + +MODULE_DESCRIPTION("TWL6040 MFD"); +MODULE_AUTHOR("Misael Lopez Cruz <misael.lopez(a)ti.com>"); +MODULE_AUTHOR("Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:twl6040-audio"); diff --git a/drivers/mfd/twl6040-irq.c b/drivers/mfd/twl6040-irq.c new file mode 100644 index 0000000..ac776be --- /dev/null +++ b/drivers/mfd/twl6040-irq.c @@ -0,0 +1,205 @@ +/* + * Interrupt controller support for TWL6040 + * + * Author: Misael Lopez Cruz <misael.lopez(a)ti.com> + * + * Copyright: (C) 2011 Texas Instruments, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/irq.h> +#include <linux/interrupt.h> +#include <linux/mfd/core.h> +#include <linux/mfd/twl6040-codec.h> + +struct twl6040_irq_data { + int mask; + int status; +}; + +static struct twl6040_irq_data twl6040_irqs[] = { + { + .mask = TWL6040_THMSK, + .status = TWL6040_THINT, + }, + { + .mask = TWL6040_PLUGMSK, + .status = TWL6040_PLUGINT | TWL6040_UNPLUGINT, + }, + { + .mask = TWL6040_HOOKMSK, + .status = TWL6040_HOOKINT, + }, + { + .mask = TWL6040_HFMSK, + .status = TWL6040_HFINT, + }, + { + .mask = TWL6040_VIBMSK, + .status = TWL6040_VIBINT, + }, + { + .mask = TWL6040_READYMSK, + .status = TWL6040_READYINT, + }, +}; + +static inline +struct twl6040_irq_data *irq_to_twl6040_irq(struct twl6040 *twl6040, + int irq) +{ + return &twl6040_irqs[irq - twl6040->irq_base]; +} + +static void twl6040_irq_lock(struct irq_data *data) +{ + struct twl6040 *twl6040 = irq_data_get_irq_chip_data(data); + + mutex_lock(&twl6040->irq_mutex); +} + +static void twl6040_irq_sync_unlock(struct irq_data *data) +{ + struct twl6040 *twl6040 = irq_data_get_irq_chip_data(data); + + /* write back to hardware any change in irq mask */ + if (twl6040->irq_masks_cur != twl6040->irq_masks_cache) { + twl6040->irq_masks_cache = twl6040->irq_masks_cur; + twl6040_reg_write(twl6040, TWL6040_REG_INTMR, + twl6040->irq_masks_cur); + } + + mutex_unlock(&twl6040->irq_mutex); +} + +static void twl6040_irq_enable(struct irq_data *data) +{ + struct twl6040 *twl6040 = irq_data_get_irq_chip_data(data); + struct twl6040_irq_data *irq_data = irq_to_twl6040_irq(twl6040, + data->irq); + + twl6040->irq_masks_cur &= ~irq_data->mask; +} + +static void twl6040_irq_disable(struct irq_data *data) +{ + struct twl6040 *twl6040 = irq_data_get_irq_chip_data(data); + struct twl6040_irq_data *irq_data = irq_to_twl6040_irq(twl6040, + data->irq); + + twl6040->irq_masks_cur |= irq_data->mask; +} + +static struct irq_chip twl6040_irq_chip = { + .name = "twl6040", + .irq_bus_lock = twl6040_irq_lock, + .irq_bus_sync_unlock = twl6040_irq_sync_unlock, + .irq_enable = twl6040_irq_enable, + .irq_disable = twl6040_irq_disable, +}; + +static irqreturn_t twl6040_irq_thread(int irq, void *data) +{ + struct twl6040 *twl6040 = data; + u8 intid; + int i; + + intid = twl6040_reg_read(twl6040, TWL6040_REG_INTID); + + /* apply masking and report (backwards to handle READYINT first) */ + for (i = ARRAY_SIZE(twl6040_irqs) - 1; i >= 0; i--) { + if (twl6040->irq_masks_cur & twl6040_irqs[i].mask) + intid &= ~twl6040_irqs[i].status; + if (intid & twl6040_irqs[i].status) + handle_nested_irq(twl6040->irq_base + i); + } + + /* ack unmasked irqs */ + twl6040_reg_write(twl6040, TWL6040_REG_INTID, intid); + + return IRQ_HANDLED; +} + +int twl6040_irq_init(struct twl6040 *twl6040) +{ + int cur_irq, ret; + u8 val; + + mutex_init(&twl6040->irq_mutex); + + /* mask the individual interrupt sources */ + twl6040->irq_masks_cur = TWL6040_ALLINT_MSK; + twl6040->irq_masks_cache = TWL6040_ALLINT_MSK; + twl6040_reg_write(twl6040, TWL6040_REG_INTMR, TWL6040_ALLINT_MSK); + + if (!twl6040->irq) { + dev_warn(twl6040->dev, + "no interrupt specified, no interrupts\n"); + twl6040->irq_base = 0; + return 0; + } + + if (!twl6040->irq_base) { + dev_err(twl6040->dev, + "no interrupt base specified, no interrupts\n"); + return 0; + } + + /* Register them with genirq */ + for (cur_irq = twl6040->irq_base; + cur_irq < twl6040->irq_base + ARRAY_SIZE(twl6040_irqs); + cur_irq++) { + irq_set_chip_data(cur_irq, twl6040); + irq_set_chip_and_handler(cur_irq, &twl6040_irq_chip, + handle_level_irq); + irq_set_nested_thread(cur_irq, 1); + + /* ARM needs us to explicitly flag the IRQ as valid + * and will set them noprobe when we do so. */ +#ifdef CONFIG_ARM + set_irq_flags(cur_irq, IRQF_VALID); +#else + irq_set_noprobe(cur_irq); +#endif + } + + ret = request_threaded_irq(twl6040->irq, NULL, twl6040_irq_thread, + IRQF_ONESHOT, "twl6040", twl6040); + if (ret) { + dev_err(twl6040->dev, "failed to request IRQ %d: %d\n", + twl6040->irq, ret); + return ret; + } + + /* reset interrupts */ + val = twl6040_reg_read(twl6040, TWL6040_REG_INTID); + + /* interrupts cleared on write */ + twl6040_clear_bits(twl6040, TWL6040_REG_ACCCTL, TWL6040_INTCLRMODE); + + return 0; +} +EXPORT_SYMBOL(twl6040_irq_init); + +void twl6040_irq_exit(struct twl6040 *twl6040) +{ + if (twl6040->irq) + free_irq(twl6040->irq, twl6040); +} +EXPORT_SYMBOL(twl6040_irq_exit); diff --git a/include/linux/i2c/twl.h b/include/linux/i2c/twl.h index 314218e..9b1ae23 100644 --- a/include/linux/i2c/twl.h +++ b/include/linux/i2c/twl.h @@ -675,6 +675,7 @@ struct twl4030_codec_data { /* twl6040 */ int audpwron_gpio; /* audio power-on gpio */ int naudint_irq; /* audio interrupt */ + unsigned int irq_base; }; struct twl4030_platform_data { diff --git a/include/linux/mfd/twl6040-codec.h b/include/linux/mfd/twl6040-codec.h new file mode 100644 index 0000000..2a41cd4 --- /dev/null +++ b/include/linux/mfd/twl6040-codec.h @@ -0,0 +1,260 @@ +/* + * MFD driver for twl6040 codec submodule + * + * Authors: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> + * Misael Lopez Cruz <misael.lopez(a)ti.com> + * + * Copyright: (C) 2011 Texas Instruments, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __TWL6040_CODEC_H__ +#define __TWL6040_CODEC_H__ + +#include <linux/interrupt.h> +#include <linux/mfd/core.h> + +#define TWL6040_REG_ASICID 0x01 +#define TWL6040_REG_ASICREV 0x02 +#define TWL6040_REG_INTID 0x03 +#define TWL6040_REG_INTMR 0x04 +#define TWL6040_REG_NCPCTL 0x05 +#define TWL6040_REG_LDOCTL 0x06 +#define TWL6040_REG_HPPLLCTL 0x07 +#define TWL6040_REG_LPPLLCTL 0x08 +#define TWL6040_REG_LPPLLDIV 0x09 +#define TWL6040_REG_AMICBCTL 0x0A +#define TWL6040_REG_DMICBCTL 0x0B +#define TWL6040_REG_MICLCTL 0x0C +#define TWL6040_REG_MICRCTL 0x0D +#define TWL6040_REG_MICGAIN 0x0E +#define TWL6040_REG_LINEGAIN 0x0F +#define TWL6040_REG_HSLCTL 0x10 +#define TWL6040_REG_HSRCTL 0x11 +#define TWL6040_REG_HSGAIN 0x12 +#define TWL6040_REG_EARCTL 0x13 +#define TWL6040_REG_HFLCTL 0x14 +#define TWL6040_REG_HFLGAIN 0x15 +#define TWL6040_REG_HFRCTL 0x16 +#define TWL6040_REG_HFRGAIN 0x17 +#define TWL6040_REG_VIBCTLL 0x18 +#define TWL6040_REG_VIBDATL 0x19 +#define TWL6040_REG_VIBCTLR 0x1A +#define TWL6040_REG_VIBDATR 0x1B +#define TWL6040_REG_HKCTL1 0x1C +#define TWL6040_REG_HKCTL2 0x1D +#define TWL6040_REG_GPOCTL 0x1E +#define TWL6040_REG_ALB 0x1F +#define TWL6040_REG_DLB 0x20 +#define TWL6040_REG_TRIM1 0x28 +#define TWL6040_REG_TRIM2 0x29 +#define TWL6040_REG_TRIM3 0x2A +#define TWL6040_REG_HSOTRIM 0x2B +#define TWL6040_REG_HFOTRIM 0x2C +#define TWL6040_REG_ACCCTL 0x2D +#define TWL6040_REG_STATUS 0x2E + +#define TWL6040_CACHEREGNUM (TWL6040_REG_STATUS + 1) + +#define TWL6040_VIOREGNUM 18 +#define TWL6040_VDDREGNUM 21 + +/* INTID (0x03) fields */ + +#define TWL6040_THINT 0x01 +#define TWL6040_PLUGINT 0x02 +#define TWL6040_UNPLUGINT 0x04 +#define TWL6040_HOOKINT 0x08 +#define TWL6040_HFINT 0x10 +#define TWL6040_VIBINT 0x20 +#define TWL6040_READYINT 0x40 + +/* INTMR (0x04) fields */ + +#define TWL6040_THMSK 0x01 +#define TWL6040_PLUGMSK 0x02 +#define TWL6040_HOOKMSK 0x08 +#define TWL6040_HFMSK 0x10 +#define TWL6040_VIBMSK 0x20 +#define TWL6040_READYMSK 0x40 +#define TWL6040_ALLINT_MSK 0x7B + +/* NCPCTL (0x05) fields */ + +#define TWL6040_NCPENA 0x01 +#define TWL6040_NCPOPEN 0x40 + +/* LDOCTL (0x06) fields */ + +#define TWL6040_LSLDOENA 0x01 +#define TWL6040_HSLDOENA 0x04 +#define TWL6040_REFENA 0x40 +#define TWL6040_OSCENA 0x80 + +/* HPPLLCTL (0x07) fields */ + +#define TWL6040_HPLLENA 0x01 +#define TWL6040_HPLLRST 0x02 +#define TWL6040_HPLLBP 0x04 +#define TWL6040_HPLLSQRENA 0x08 +#define TWL6040_MCLK_12000KHZ (0 << 5) +#define TWL6040_MCLK_19200KHZ (1 << 5) +#define TWL6040_MCLK_26000KHZ (2 << 5) +#define TWL6040_MCLK_38400KHZ (3 << 5) +#define TWL6040_MCLK_MSK 0x60 + +/* LPPLLCTL (0x08) fields */ + +#define TWL6040_LPLLENA 0x01 +#define TWL6040_LPLLRST 0x02 +#define TWL6040_LPLLSEL 0x04 +#define TWL6040_LPLLFIN 0x08 +#define TWL6040_HPLLSEL 0x10 + +/* HSLCTL (0x10) fields */ + +#define TWL6040_HSDACMODEL 0x02 +#define TWL6040_HSDRVMODEL 0x08 + +/* HSRCTL (0x11) fields */ + +#define TWL6040_HSDACMODER 0x02 +#define TWL6040_HSDRVMODER 0x08 + +/* VIBCTLL (0x18) fields */ + +#define TWL6040_VIBENAL 0x01 +#define TWL6040_VIBCTRLL 0x04 +#define TWL6040_VIBCTRLLP 0x08 +#define TWL6040_VIBCTRLLN 0x10 + +/* VIBDATL (0x19) fields */ + +#define TWL6040_VIBDAT_MAX 0x64 + +/* VIBCTLR (0x1A) fields */ + +#define TWL6040_VIBENAR 0x01 +#define TWL6040_VIBCTRLR 0x04 +#define TWL6040_VIBCTRLRP 0x08 +#define TWL6040_VIBCTRLRN 0x10 + +/* GPOCTL (0x1E) fields */ + +#define TWL6040_GPO1 0x01 +#define TWL6040_GPO2 0x02 +#define TWL6040_GPO3 0x03 + +/* ACCCTL (0x2D) fields */ + +#define TWL6040_I2CSEL 0x01 +#define TWL6040_RESETSPLIT 0x04 +#define TWL6040_INTCLRMODE 0x08 + +#define TWL6040_SYSCLK_SEL_LPPLL 1 +#define TWL6040_SYSCLK_SEL_HPPLL 2 + +/* STATUS (0x2E) fields */ + +#define TWL6040_PLUGCOMP 0x02 +#define TWL6040_VIBLOCDET 0x10 +#define TWL6040_VIBROCDET 0x20 + +#define TWL6040_CELLS 2 + +#define TWL6040_REV_ES1_0 0x00 +#define TWL6040_REV_ES1_1 0x01 +#define TWL6040_REV_ES1_2 0x02 + +#define TWL6040_IRQ_TH 0 +#define TWL6040_IRQ_PLUG 1 +#define TWL6040_IRQ_HOOK 2 +#define TWL6040_IRQ_HF 3 +#define TWL6040_IRQ_VIB 4 +#define TWL6040_IRQ_READY 5 + +enum twl6040_pll_id { + TWL6040_NOPLL_ID, + TWL6040_LPPLL_ID, + TWL6040_HPPLL_ID, +}; + +struct twl6040 { + struct device *dev; + struct mutex mutex; + struct mutex io_mutex; + struct mutex irq_mutex; + struct mfd_cell cells[TWL6040_CELLS]; + struct completion ready; + + int audpwron; + int power_count; + int rev; + + enum twl6040_pll_id pll; + unsigned int sysclk; + + unsigned int irq; + unsigned int irq_base; + u8 irq_masks_cur; + u8 irq_masks_cache; +}; + +static inline int twl6040_get_rev(struct twl6040 *twl6040) +{ + return twl6040->rev; +} + +static inline int twl6040_request_irq(struct twl6040 *twl6040, int irq, + irq_handler_t handler, + unsigned long irqflags, + const char *name, + void *data) +{ + if (!twl6040->irq_base) + return -EINVAL; + + return request_threaded_irq(twl6040->irq_base + irq, NULL, handler, + irqflags, name, data); +} + +static inline void twl6040_free_irq(struct twl6040 *twl6040, int irq, + void *data) +{ + if (!twl6040->irq_base) + return; + + free_irq(twl6040->irq_base + irq, data); +} + +int twl6040_reg_read(struct twl6040 *twl6040, unsigned int reg); +int twl6040_reg_write(struct twl6040 *twl6040, unsigned int reg, + u8 val); +int twl6040_set_bits(struct twl6040 *twl6040, unsigned int reg, + u8 mask); +int twl6040_clear_bits(struct twl6040 *twl6040, unsigned int reg, + u8 mask); +int twl6040_power(struct twl6040 *twl6040, int on); +int twl6040_is_powered(struct twl6040 *twl6040); +int twl6040_set_pll(struct twl6040 *twl6040, enum twl6040_pll_id id, + unsigned int freq_in, unsigned int freq_out); +enum twl6040_pll_id twl6040_get_pll(struct twl6040 *twl6040); +unsigned int twl6040_get_sysclk(struct twl6040 *twl6040); +int twl6040_irq_init(struct twl6040 *twl6040); +void twl6040_irq_exit(struct twl6040 *twl6040); + +#endif /* End of __TWL6040_CODEC_H__ */ -- 1.7.5.rc3

1 6

[alsa-devel] Timer interface
by Nenad Sljivic 02 Aug '11

02 Aug '11

Hi, I am extending my ALSA driver in order to support software mixing. Driver currently supports both MMAP and RW access modes. However, I am still not able to use dmix since it requires timer interface to be implemented. My hardware does not generate interrupts at the period boundary but issues timer interrupts at a fixed timer rate. Could you please point me to the document or sample driver that can help me to fully understand timer interface concept (especially for High frequency timer interrupts)? Thanks, Nenad

2 4

[alsa-devel] [PATCH 1/3] ASoC: Add device tree binding for WM8510
by Mark Brown 02 Aug '11

02 Aug '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- Documentation/devicetree/bindings/sound/wm8510.txt | 18 ++++++++++++++++++ sound/soc/codecs/wm8510.c | 8 ++++++++ 2 files changed, 26 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/wm8510.txt diff --git a/Documentation/devicetree/bindings/sound/wm8510.txt b/Documentation/devicetree/bindings/sound/wm8510.txt new file mode 100644 index 0000000..fa1a32b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8510.txt @@ -0,0 +1,18 @@ +WM8510 audio CODEC + +This device supports both I2C and SPI (configured with pin strapping +on the board). + +Required properties: + + - compatible : "wlf,wm8510" + + - reg : the I2C address of the device for I2C, the chip select + number for SPI. + +Example: + +codec: wm8510@1a { + compatible = "wlf,wm8510"; + reg = <0x1a>; +}; diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index db0dced..55a4c83 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -20,6 +20,7 @@ #include <linux/platform_device.h> #include <linux/spi/spi.h> #include <linux/slab.h> +#include <linux/of_device.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -598,6 +599,11 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8510 = { .reg_cache_default =wm8510_reg, }; +static const struct of_device_id wm8510_of_match[] = { + { .compatible = "wlf,wm8510" }, + { }, +}; + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8510_spi_probe(struct spi_device *spi) { @@ -628,6 +634,7 @@ static struct spi_driver wm8510_spi_driver = { .driver = { .name = "wm8510", .owner = THIS_MODULE, + .of_match_table = wm8510_of_match, }, .probe = wm8510_spi_probe, .remove = __devexit_p(wm8510_spi_remove), @@ -671,6 +678,7 @@ static struct i2c_driver wm8510_i2c_driver = { .driver = { .name = "wm8510-codec", .owner = THIS_MODULE, + .of_match_table = wm8510_of_match, }, .probe = wm8510_i2c_probe, .remove = __devexit_p(wm8510_i2c_remove), -- 1.7.5.4

2 3

[alsa-devel] [PATCH 1/2] ASoC: Remove -codec from WM8523 driver name
by Mark Brown 02 Aug '11

02 Aug '11

It's redundant to specify it. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8523.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 4fd4d8d..1312009 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -551,7 +551,7 @@ MODULE_DEVICE_TABLE(i2c, wm8523_i2c_id); static struct i2c_driver wm8523_i2c_driver = { .driver = { - .name = "wm8523-codec", + .name = "wm8523", .owner = THIS_MODULE, }, .probe = wm8523_i2c_probe, -- 1.7.5.4

2 3

[alsa-devel] [PATCH] ALSA: asihpi - Clarify adapter index validity check.
by linux＠audioscience.com 02 Aug '11

02 Aug '11

From: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> Avoids assigning possibly invalid address to pa, even if it is never dereferenced. Signed-off-by: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> --- sound/pci/asihpi/hpioctl.c | 19 ++++++++++++------- 1 files changed, 12 insertions(+), 7 deletions(-) diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c index 9683f84..a32502e 100644 --- a/sound/pci/asihpi/hpioctl.c +++ b/sound/pci/asihpi/hpioctl.c @@ -177,16 +177,21 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg) } else { u16 __user *ptr = NULL; u32 size = 0; - + u32 adapter_present; /* -1=no data 0=read from user mem, 1=write to user mem */ int wrflag = -1; - u32 adapter = hm->h.adapter_index; - struct hpi_adapter *pa = &adapters[adapter]; + struct hpi_adapter *pa; + + if (hm->h.adapter_index < HPI_MAX_ADAPTERS) { + pa = &adapters[hm->h.adapter_index]; + adapter_present = pa->type; + } else { + adapter_present = 0; + } - if ((adapter >= HPI_MAX_ADAPTERS) || (!pa->type)) { - hpi_init_response(&hr->r0, HPI_OBJ_ADAPTER, - HPI_ADAPTER_OPEN, - HPI_ERROR_BAD_ADAPTER_NUMBER); + if (!adapter_present) { + hpi_init_response(&hr->r0, hm->h.object, + hm->h.function, HPI_ERROR_BAD_ADAPTER_NUMBER); uncopied_bytes = copy_to_user(puhr, hr, sizeof(hr->h)); -- 1.7.0.4

3 4

[alsa-devel] [PATCH 1/1] ASoC: sgtl5000: add one missed cache reg
by Dong Aisheng 01 Aug '11

01 Aug '11

Signed-off-by: Dong Aisheng <b29396(a)freescale.com> Signed-off-by: Zeng Zhaoming <zhaoming.zeng(a)freescale.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lrg(a)ti.com> Cc: Sascha Hauer <s.hauer(a)pengutronix.de> --- sound/soc/codecs/sgtl5000.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 76258f2..5a0d8e4 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -49,6 +49,7 @@ static const u16 sgtl5000_regs[SGTL5000_MAX_REG_OFFSET >> 1] = { 0x0000, /* 0x0016, reserved */ 0x0000, /* 0x0018, reserved */ 0x0000, /* 0x001A, reserved */ + 0x0000, /* 0x001C, reserved */ 0x0000, /* 0x001E, reserved */ 0x0000, /* 0x0020, CHIP_ANA_ADC_CTRL */ 0x1818, /* 0x0022, CHIP_ANA_HP_CTRL */ -- 1.7.0.4

4 3

[alsa-devel] [PATCH 1/2] ASoC: Implement new DC servo readback mode for late WM8994 revisions
by Mark Brown 01 Aug '11

01 Aug '11

Later WM8994 devices implement a new DC servo readback mode with the register used to access the offset moved to register 0x59. Implement support for this and enable it on the appropriate devices. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/linux/mfd/wm8994/registers.h | 1 + sound/soc/codecs/wm8994.c | 3 ++- sound/soc/codecs/wm_hubs.c | 19 +++++++++++++++---- 3 files changed, 18 insertions(+), 5 deletions(-) diff --git a/include/linux/mfd/wm8994/registers.h b/include/linux/mfd/wm8994/registers.h index f3ee842..6152914 100644 --- a/include/linux/mfd/wm8994/registers.h +++ b/include/linux/mfd/wm8994/registers.h @@ -72,6 +72,7 @@ #define WM8994_DC_SERVO_2 0x55 #define WM8994_DC_SERVO_4 0x57 #define WM8994_DC_SERVO_READBACK 0x58 +#define WM8994_DC_SERVO_4E 0x59 #define WM8994_ANALOGUE_HP_1 0x60 #define WM8958_MIC_DETECT_1 0xD0 #define WM8958_MIC_DETECT_2 0xD1 diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index cc8222c..e1e9aca 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -107,6 +107,7 @@ static int wm8994_volatile(struct snd_soc_codec *codec, unsigned int reg) case WM8994_LDO_2: case WM8958_DSP2_EXECCONTROL: case WM8958_MIC_DETECT_3: + case WM8994_DC_SERVO_4E: return 1; default: return 0; @@ -2965,7 +2966,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994->hubs.series_startup = 1; break; default: - wm8994->hubs.dcs_readback_mode = 1; + wm8994->hubs.dcs_readback_mode = 2; break; } diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 4416a76..524bdc6 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -18,6 +18,7 @@ #include <linux/pm.h> #include <linux/i2c.h> #include <linux/platform_device.h> +#include <linux/mfd/wm8994/registers.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -116,14 +117,23 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) { struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); s8 offset; - u16 reg, reg_l, reg_r, dcs_cfg; + u16 reg, reg_l, reg_r, dcs_cfg, dcs_reg; + + switch (hubs->dcs_readback_mode) { + case 2: + dcs_reg = WM8994_DC_SERVO_4E; + break; + default: + dcs_reg = WM8993_DC_SERVO_3; + break; + } /* If we're using a digital only path and have a previously * callibrated DC servo offset stored then use that. */ if (hubs->class_w && hubs->class_w_dcs) { dev_dbg(codec->dev, "Using cached DC servo offset %x\n", hubs->class_w_dcs); - snd_soc_write(codec, WM8993_DC_SERVO_3, hubs->class_w_dcs); + snd_soc_write(codec, dcs_reg, hubs->class_w_dcs); wait_for_dc_servo(codec, WM8993_DCS_TRIG_DAC_WR_0 | WM8993_DCS_TRIG_DAC_WR_1); @@ -154,8 +164,9 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) reg_r = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_2) & WM8993_DCS_INTEG_CHAN_1_MASK; break; + case 2: case 1: - reg = snd_soc_read(codec, WM8993_DC_SERVO_3); + reg = snd_soc_read(codec, dcs_reg); reg_r = (reg & WM8993_DCS_DAC_WR_VAL_1_MASK) >> WM8993_DCS_DAC_WR_VAL_1_SHIFT; reg_l = reg & WM8993_DCS_DAC_WR_VAL_0_MASK; @@ -185,7 +196,7 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) dev_dbg(codec->dev, "DCS result: %x\n", dcs_cfg); /* Do it */ - snd_soc_write(codec, WM8993_DC_SERVO_3, dcs_cfg); + snd_soc_write(codec, dcs_reg, dcs_cfg); wait_for_dc_servo(codec, WM8993_DCS_TRIG_DAC_WR_0 | WM8993_DCS_TRIG_DAC_WR_1); -- 1.7.5.4

2 2

[alsa-devel] [PATCH] ASoC: Trivial formatting fix in soc-core.c
by Mark Brown 01 Aug '11

01 Aug '11

Utterly trivial but it annoys me. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 83ad8ca..e2ce248 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -105,7 +105,7 @@ static int format_register_str(struct snd_soc_codec *codec, if (wordsize + regsize + 2 + 1 != len) return -EINVAL; - ret = snd_soc_read(codec , reg); + ret = snd_soc_read(codec, reg); if (ret < 0) { memset(regbuf, 'X', regsize); regbuf[regsize] = '\0'; -- 1.7.5.4

2 1

[alsa-devel] Some more on MSI GX740 and HDA Intel (bug 5321)
by K900 01 Aug '11

01 Aug '11

Sorry for disappearing last time, I've been quite busy at the time. But I managed to find a bit more on my laptop's sound card and how to make it work. I've found two nodes that actually produce (some) sound when connected to a mixer. 0x1b pin complex is connected to front speakers and 0x16 to the center/LFE thingy. However, I can only get sound from them when I connect them to 0x0c and 0x0e mixers, that both (why?) output the sound from front channels. So, there's no mixer that outputs rear or LFE channels. To be honest, I don't understand anything further than this, so can someone please help me :) I'll try to be around on IRC most of the time. Also, my JID is k900(a)neko.im. // Note: it seems my last mail was lost somewhere. Correct me if I'm wrong.

1 0

Re: [alsa-devel] Copy ALSA buffer to the buffer allocated in my driver.
by Clemens Ladisch 01 Aug '11

01 Aug '11

`/ET!-i1S!-i wrote: > 1. I guess pulse audio inturn calls Alsa and i want to write a Alsa driver. But the sound cards on the networked PCs already have drivers. > The problem i am facing is receiving Alsa buffer to my buffer Your driver's buffer _is_ the ALSA buffer. > and updating the buffer position to middle layer. The hardware position is returned by the pointer callback. The software position should be of no concern to the driver. > 2. Is it substream->runtime->dma_area or substream->dma_buffer.area holds > the audio data? What is the difference between them? These fields are controlled by your driver, although typically they are set by ALSA helper functions. substream->dma_buffer is the preallocated buffer; if your driver doesn't do preallocation, it is not set. The actual buffer, if it exists and is accessible by the CPU from kernel space, is in substream->runtime->dma_area. Regards, Clemens

1 0

Re: [alsa-devel] [PATCH] Sound: AudioScience HPI: Don't leak firmware if mem alloc fails
by Takashi Iwai 01 Aug '11

01 Aug '11

At Sun, 31 Jul 2011 23:16:43 +0200 (CEST), Jesper Juhl wrote: > > On Fri, 29 Jul 2011, Jesper Juhl wrote: > > > On Fri, 29 Jul 2011, Takashi Iwai wrote: > > > > > At Thu, 28 Jul 2011 21:48:27 +0200 (CEST), > > > Jesper Juhl wrote: > > > > > > > > We leak the memory allocated to 'firmware' when we fail to > > > > release_firmware() after a kmalloc() failure in hpi_dsp_code_open(). > > > > This patch should take care of the leak. > > > > > > > > Signed-off-by: Jesper Juhl <jj(a)chaosbits.net> > > > > --- > > > > sound/pci/asihpi/hpidspcd.c | 4 +++- > > > > 1 files changed, 3 insertions(+), 1 deletions(-) > > > > > > > > Build tested only. > > > > > > > > diff --git a/sound/pci/asihpi/hpidspcd.c b/sound/pci/asihpi/hpidspcd.c > > > > index 3a7afa3..8d261ef 100644 > > > > --- a/sound/pci/asihpi/hpidspcd.c > > > > +++ b/sound/pci/asihpi/hpidspcd.c > > > > @@ -85,8 +85,10 @@ short hpi_dsp_code_open(u32 adapter, void *os_data, struct dsp_code *dsp_code, > > > > > > > > HPI_DEBUG_LOG(DEBUG, "dsp code %s opened\n", fw_name); > > > > dsp_code->pvt = kmalloc(sizeof(*dsp_code->pvt), GFP_KERNEL); > > > > - if (!dsp_code->pvt) > > > > + if (!dsp_code->pvt) { > > > > + release_firmware(firmware); > > > > return HPI_ERROR_MEMORY_ALLOC; > > > > + } > > > > > > IMO, it's better to use the same error handling like others, i.e. > > > "goto error2". But you want to pass a different error code, so we'd > > > need to introduce an error code variable there... > > > > > Thank you for the feedback. I'll cook up a better patch tonight. > > > > > > Sorry about the delay - had non-computer things to attend to. > > I hope this is more to your liking :-) > > > From: Jesper Juhl <jj(a)chaosbits.net> > > We leak the memory allocated to 'firmware' when we fail to > release_firmware() after a kmalloc() failure in hpi_dsp_code_open(). > This patch should take care of the leak. > > Signed-off-by: Jesper Juhl <jj(a)chaosbits.net> Applied now. Thanks. Takashi > --- > sound/pci/asihpi/hpidspcd.c | 9 ++++++--- > 1 files changed, 6 insertions(+), 3 deletions(-) > > diff --git a/sound/pci/asihpi/hpidspcd.c b/sound/pci/asihpi/hpidspcd.c > index 3a7afa3..71d32c8 100644 > --- a/sound/pci/asihpi/hpidspcd.c > +++ b/sound/pci/asihpi/hpidspcd.c > @@ -43,6 +43,7 @@ short hpi_dsp_code_open(u32 adapter, void *os_data, struct dsp_code *dsp_code, > struct pci_dev *dev = os_data; > struct code_header header; > char fw_name[20]; > + short err_ret = HPI_ERROR_DSP_FILE_NOT_FOUND; > int err; > > sprintf(fw_name, "asihpi/dsp%04x.bin", adapter); > @@ -85,8 +86,10 @@ short hpi_dsp_code_open(u32 adapter, void *os_data, struct dsp_code *dsp_code, > > HPI_DEBUG_LOG(DEBUG, "dsp code %s opened\n", fw_name); > dsp_code->pvt = kmalloc(sizeof(*dsp_code->pvt), GFP_KERNEL); > - if (!dsp_code->pvt) > - return HPI_ERROR_MEMORY_ALLOC; > + if (!dsp_code->pvt) { > + err_ret = HPI_ERROR_MEMORY_ALLOC; > + goto error2; > + } > > dsp_code->pvt->dev = dev; > dsp_code->pvt->firmware = firmware; > @@ -99,7 +102,7 @@ error2: > release_firmware(firmware); > error1: > dsp_code->block_length = 0; > - return HPI_ERROR_DSP_FILE_NOT_FOUND; > + return err_ret; > } > > /*-------------------------------------------------------------------*/ > -- > 1.7.6 > > > -- > Jesper Juhl <jj(a)chaosbits.net> http://www.chaosbits.net/ > Don't top-post http://www.catb.org/jargon/html/T/top-post.html > Plain text mails only, please. >

1 0

[alsa-devel] [PATCH -next] sound: rtctimer.c needs module.h
by Randy Dunlap 30 Jul '11

30 Jul '11

From: Randy Dunlap <rdunlap(a)xenotime.net> rtctimer.c uses interfaces from linux/module.h, so it should include that file. This fixes build errors. Signed-off-by: Randy Dunlap <rdunlap(a)xenotime.net> --- sound/core/rtctimer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- linux-next-20110729.orig/sound/core/rtctimer.c +++ linux-next-20110729/sound/core/rtctimer.c @@ -22,7 +22,7 @@ #include <linux/init.h> #include <linux/interrupt.h> -#include <linux/moduleparam.h> +#include <linux/module.h> #include <linux/log2.h> #include <sound/core.h> #include <sound/timer.h>

2 1

Re: [alsa-devel] [RFC PATCH] ARM ASoC: add sound driver for imx27pdk using mc13783 codec
by Jürgen Lambrecht 29 Jul '11

29 Jul '11

2011/7/27 Philippe Rétornaz <philippe.retornaz(a)epfl.ch>: > Hi > > Le vendredi 8 juillet 2011 16:19:22, Lambrecht Jürgen a écrit : >> indeed, doesn't work at all. I know it used to work on 2.6.34. >> We also have a fsl linux 2.6.22 were it works. >> I must say: when i use aplay to play a mono file, then there is no >> error, it just never ends, and no sound also. >> For a stereo file, it gives the error. >> Mark also that when I use madplay (v0.15.2) (or mp3play) to play a mono >> or stereo mp3, it allocates 3 DMA channels, but only frees one after the >> error. >> > > Did you made any progress on this ? Yes. Sascha fixed it, and will commit it. Regards, Jürgen > > Regards, > > Philippe > > >

1 0

[alsa-devel] [PATCH 0/2] hdsploader/hdspmixer: Fix RPM constant
by Adrian Knoth 29 Jul '11

29 Jul '11

As noticed by Takashi, the RPM constant needs to be 4, not 5. For hdspmixer, this doesn't matter, for hdsploader, it's essential. Adrian Knoth (2): hdsploader: Fix temporary RPM constant hdspmixer: Fix temporary RPM constant hdsploader/hdsploader.c | 2 +- hdspmixer/src/HDSPMixerCard.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) -- 1.7.5.4

2 3

[alsa-devel] [PATCH] ASoC: sgtl5000: add device tree probe support
by Shawn Guo 29 Jul '11

29 Jul '11

It adds device tree probe support for sgtl5000 driver. Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> Cc: Grant Likely <grant.likely(a)secretlab.ca> Cc: Zeng Zhaoming <zhaoming.zeng(a)freescale.com> Cc: Liam Girdwood <lrg(a)ti.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- .../bindings/sound/soc/codecs/fsl-sgtl5000.txt | 11 +++++++++++ sound/soc/codecs/sgtl5000.c | 8 ++++++++ 2 files changed, 19 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/soc/codecs/fsl-sgtl5000.txt diff --git a/Documentation/devicetree/bindings/sound/soc/codecs/fsl-sgtl5000.txt b/Documentation/devicetree/bindings/sound/soc/codecs/fsl-sgtl5000.txt new file mode 100644 index 0000000..2c3cd41 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/soc/codecs/fsl-sgtl5000.txt @@ -0,0 +1,11 @@ +* Freescale SGTL5000 Stereo Codec + +Required properties: +- compatible : "fsl,sgtl5000". + +Example: + +codec: sgtl5000@0a { + compatible = "fsl,sgtl5000"; + reg = <0x0a>; +}; diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 76258f2..cf6eea8 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -20,6 +20,7 @@ #include <linux/regulator/driver.h> #include <linux/regulator/machine.h> #include <linux/regulator/consumer.h> +#include <linux/of_device.h> #include <sound/core.h> #include <sound/tlv.h> #include <sound/pcm.h> @@ -1494,10 +1495,17 @@ static const struct i2c_device_id sgtl5000_id[] = { MODULE_DEVICE_TABLE(i2c, sgtl5000_id); +static const struct of_device_id sgtl5000_dt_ids[] = { + { .compatible = "fsl,sgtl5000", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(i2c, sgtl5000_dt_ids); + static struct i2c_driver sgtl5000_i2c_driver = { .driver = { .name = "sgtl5000", .owner = THIS_MODULE, + .of_match_table = sgtl5000_dt_ids, }, .probe = sgtl5000_i2c_probe, .remove = __devexit_p(sgtl5000_i2c_remove), -- 1.7.4.1

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Re: [alsa-devel] Copy ALSA buffer to the buffer allocated in my driver.
by Clemens Ladisch 29 Jul '11

29 Jul '11

`/ET!-i1S!-i wrote: > Actually i need to stream the audio data to other system through kernel > socket. Why aren't you using PulseAudio? Regards, Clemens

1 0

[alsa-devel] [PATCH 0/3] [RME - HDSP] Userlevel support for RPM
by Adrian Knoth 29 Jul '11

29 Jul '11

Hi! Here's a series of recent work on RME RPM support. Confirmed to be working by our one and only user so far. ;) As soon as alsa-lib supports the RPM (it's a single define), I'll drop the temporary define in both, hdsploader and hdspmixer. RPM users: note that the current kernel code seems to be broken and does not correctly detect your RPM box. More magic could be added to hdsploader to provide a temporary workaround. Cheers Adrian Knoth (3): hdspmixer: Add support for RME RPM hdsploader: Add support for RME RPM hdsploader: Temporary define for RPM hdsploader/hdsploader.c | 7 +++++++ hdspmixer/src/HDSPMixerCard.cxx | 11 +++++++++++ hdspmixer/src/HDSPMixerCard.h | 5 +++++ hdspmixer/src/HDSPMixerOutput.cxx | 15 +++++++++++++++ hdspmixer/src/HDSPMixerSelector.cxx | 7 +++++++ hdspmixer/src/HDSPMixerWindow.cxx | 4 ++++ hdspmixer/src/channelmap.cxx | 12 ++++++++++++ hdspmixer/src/channelmap.h | 5 +++++ hdspmixer/src/hdspmixer.cxx | 4 ++++ 9 files changed, 70 insertions(+), 0 deletions(-) -- 1.7.5.4

2 9

[alsa-devel] [PATCH] ASoC: Parse board ID/revision information from WM1250-EV1 board
by Mark Brown 29 Jul '11

29 Jul '11

The WM1250-EV1 board has an ID chip on it, check the board ID and display the board revision during startup. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm1250-ev1.c | 20 +++++++++++++++++++- 1 files changed, 19 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c index bcc2089..bbcf9ec 100644 --- a/sound/soc/codecs/wm1250-ev1.c +++ b/sound/soc/codecs/wm1250-ev1.c @@ -56,8 +56,26 @@ static struct snd_soc_codec_driver soc_codec_dev_wm1250_ev1 = { }; static int __devinit wm1250_ev1_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) + const struct i2c_device_id *i2c_id) { + int ret, id, board, rev; + + board = i2c_smbus_read_byte_data(i2c, 0); + if (board < 0) { + dev_err(&i2c->dev, "Failed to read ID: %d\n", ret); + return ret; + } + + id = (board & 0xfe) >> 2; + rev = board & 0x3; + + if (id != 1) { + dev_err(&i2c->dev, "Unknown board ID %d\n", id); + return -ENODEV; + } + + dev_info(&i2c->dev, "revision %d\n", rev); + return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm1250_ev1, &wm1250_ev1_dai, 1); } -- 1.7.5.4

1 0

[alsa-devel] Copy ALSA buffer to the buffer allocated in my driver.
by `/ET!-i1S!-i 29 Jul '11

29 Jul '11

Hi everyone. I'm writing a virtual ALSA driver for a playback device, with the help of dummy driver available at the linux kernel source code. I need to manually copy data from the ALSA buffer to my buffer. I have no idea how to do this, since dummy driver has completely discarded handling of buffers. So can anyone please provide a sample code to copy the audio data to the my buffer allocated in the driver. Is it ok to use the hrtimer interrupts (which is used in dummy driver) or mutex / spinlocks for my driver. Best regards, cherry

2 1

[alsa-devel] mmap ops in snd_pcm_ops
by Scott Jiang 29 Jul '11

29 Jul '11

Hi all, I found there is a memory leak about 128K every time in linux-2.6.34, but it doesn't exist in linux-3.0. If I use mmap ops in driver the bug will come out, while use copy ops no memory leak. I tested with aplay to play a wav file. Because there is a lot of changes in ALSA between 2.6.34 and 3.0, so can anyone familiar with the change history tell me which patch has solved this bug? Thanks in advance. Scott

1 0

[alsa-devel] [PATCH] ASoC: Disable wm_hubs periodic DC servo update
by Mark Brown 29 Jul '11

29 Jul '11

This does not function correctly in all circumstances so disable the periodic updates unconditionally for stable; a future patch will reenable where appropriate. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_hubs.c | 3 +-- 1 files changed, 1 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 4de6794..4416a76 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -440,9 +440,8 @@ static int hp_event(struct snd_soc_dapm_widget *w, reg |= WM8993_HPOUT1L_DLY | WM8993_HPOUT1R_DLY; snd_soc_write(codec, WM8993_ANALOGUE_HP_0, reg); - /* Smallest supported update interval */ snd_soc_update_bits(codec, WM8993_DC_SERVO_1, - WM8993_DCS_TIMER_PERIOD_01_MASK, 1); + WM8993_DCS_TIMER_PERIOD_01_MASK, 0); calibrate_dc_servo(codec); -- 1.7.5.4

2 1

[alsa-devel] [PATCH] SOUND: Fix txx9aclc.c build
by Ralf Baechle 29 Jul '11

29 Jul '11

552d1ef6b5a98d7b95959d5b139071e3c90cebf1 [ASoC: core - Optimise and refactor pcm_new() to pass only rtd] breaks compilation of txx9aclc.c: CC [M] sound/soc/txx9/txx9aclc.o /home/ralf/src/linux/linux-mips/sound/soc/txx9/txx9aclc.c: In function 'txx9aclc_pcm_new': /home/ralf/src/linux/linux-mips/sound/soc/txx9/txx9aclc.c:318:3: error: 'card' undeclared (first use in this function) /home/ralf/src/linux/linux-mips/sound/soc/txx9/txx9aclc.c:318:3: note: each undeclared identifier is reported only once for each function it appears in make[5]: *** [sound/soc/txx9/txx9aclc.o] Error 1 Fixed by providing a definition for card. Signed-off-by: Ralf Baechle <ralf(a)linux-mips.org> sound/soc/txx9/txx9aclc.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c index 34aa972..3de99af 100644 --- a/sound/soc/txx9/txx9aclc.c +++ b/sound/soc/txx9/txx9aclc.c @@ -290,6 +290,7 @@ static void txx9aclc_pcm_free_dma_buffers(struct snd_pcm *pcm) static int txx9aclc_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; struct snd_soc_dai *dai = rtd->cpu_dai; struct snd_pcm *pcm = rtd->pcm; struct platform_device *pdev = to_platform_device(dai->platform->dev);

2 1

[alsa-devel] [PATCH 0/3] [RME - HDSPM/MADI] Speed mode selector
by Adrian Knoth 29 Jul '11

29 Jul '11

Hi! Here's another small series of patches for the RME MADI cards. Most prominently, we now provide a control element to the user to explicitly select a frequency multiplier for the MADI link. This way, one can run applications at 64kHz and above (up to 192kHz) even when the card is not the clock master (read: is in slave mode). Cheers Adrian Knoth (3): ALSA: hdspm - Provide MADI speed mode selector on RME MADI and MADIface ALSA: hdspm - Fix reported external sample rate on RME MADI and MADIface ALSA: hdspm - Add firmware revision 0xcc for RME MADI sound/pci/rme9652/hdspm.c | 109 ++++++++++++++++++++++++++++++++++++++++++++- 1 files changed, 107 insertions(+), 2 deletions(-) -- 1.7.5.4

2 4

[alsa-devel] [PATCH alsa-lib 1/1] Add RME RPM to HDSP_IO_Type.
by Adrian Knoth 29 Jul '11

29 Jul '11

The hdsp.c driver supports the RME RPM devices, but we also need this information in alsa-lib for tools like hdsploader and hdspmixer. Signed-off-by: Adrian Knoth <adi(a)drcomp.erfurt.thur.de> diff --git a/include/sound/hdsp.h b/include/sound/hdsp.h index 0fc9e98..5adaf7b 100644 --- a/include/sound/hdsp.h +++ b/include/sound/hdsp.h @@ -28,6 +28,7 @@ typedef enum { Multiface, H9652, H9632, + RPM, Undefined, } HDSP_IO_Type; -- 1.7.5.4

2 1

Re: [alsa-devel] [PATCH] Sound: AudioScience HPI: Don't leak firmware if mem alloc fails
by Takashi Iwai 29 Jul '11

29 Jul '11

At Thu, 28 Jul 2011 21:48:27 +0200 (CEST), Jesper Juhl wrote: > > We leak the memory allocated to 'firmware' when we fail to > release_firmware() after a kmalloc() failure in hpi_dsp_code_open(). > This patch should take care of the leak. > > Signed-off-by: Jesper Juhl <jj(a)chaosbits.net> > --- > sound/pci/asihpi/hpidspcd.c | 4 +++- > 1 files changed, 3 insertions(+), 1 deletions(-) > > Build tested only. > > diff --git a/sound/pci/asihpi/hpidspcd.c b/sound/pci/asihpi/hpidspcd.c > index 3a7afa3..8d261ef 100644 > --- a/sound/pci/asihpi/hpidspcd.c > +++ b/sound/pci/asihpi/hpidspcd.c > @@ -85,8 +85,10 @@ short hpi_dsp_code_open(u32 adapter, void *os_data, struct dsp_code *dsp_code, > > HPI_DEBUG_LOG(DEBUG, "dsp code %s opened\n", fw_name); > dsp_code->pvt = kmalloc(sizeof(*dsp_code->pvt), GFP_KERNEL); > - if (!dsp_code->pvt) > + if (!dsp_code->pvt) { > + release_firmware(firmware); > return HPI_ERROR_MEMORY_ALLOC; > + } IMO, it's better to use the same error handling like others, i.e. "goto error2". But you want to pass a different error code, so we'd need to introduce an error code variable there... Takashi

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Re: [alsa-devel] [PATCH] ASoC: Add device tree binding for WM8731
by Grant Likely 29 Jul '11

29 Jul '11

On Tue, Jul 26, 2011 at 09:02:11PM +0100, Mark Brown wrote: > Tested with the famous "hey, look! this compiles" test plan. > > Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> > --- > Documentation/devicetree/bindings/sound/wm8731.txt | 18 ++++++++++++++++++ > sound/soc/codecs/wm8731.c | 12 ++++++++++++ > 2 files changed, 30 insertions(+), 0 deletions(-) > create mode 100644 Documentation/devicetree/bindings/sound/wm8731.txt > > diff --git a/Documentation/devicetree/bindings/sound/wm8731.txt b/Documentation/devicetree/bindings/sound/wm8731.txt > new file mode 100644 > index 0000000..15f7004 > --- /dev/null > +++ b/Documentation/devicetree/bindings/sound/wm8731.txt > @@ -0,0 +1,18 @@ > +WM8731 audio CODEC > + > +This device supports both I2C and SPI (configured with pin strapping > +on the board). > + > +Required properties: > + > + - compatible : "wlf,wm8731" > + > + - reg : the I2C address of the device for I2C, the chip select > + number for SPI. > + > +Example: > + > +codec: wm8731@1a { > + compatible = "wlf,wm8731"; > + reg = <0x1a>; > +}; Acked by: Grant Likely <grant.likely(a)secretlab.ca> Although I'm sure some actual testing would be valuable before it is actually merged. :-) And there is one comment below. > diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c > index 76b4361..5ecc05a 100644 > --- a/sound/soc/codecs/wm8731.c > +++ b/sound/soc/codecs/wm8731.c > @@ -22,6 +22,7 @@ > #include <linux/platform_device.h> > #include <linux/regulator/consumer.h> > #include <linux/spi/spi.h> > +#include <linux/of_device.h> > #include <sound/core.h> > #include <sound/pcm.h> > #include <sound/pcm_params.h> > @@ -607,6 +608,11 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8731 = { > .num_dapm_routes = ARRAY_SIZE(wm8731_intercon), > }; > > +static const struct of_device_id wm8731_of_match[] = { > + { .compatible = "wlf,wm8731", }, > + { } > +}; > + > #if defined(CONFIG_SPI_MASTER) > static int __devinit wm8731_spi_probe(struct spi_device *spi) > { > @@ -634,10 +640,13 @@ static int __devexit wm8731_spi_remove(struct spi_device *spi) > return 0; > } > > +MODULE_DEVICE_TABLE(spi, wm8731_of_match); MODULE_DEVICE_TABLE(of, wm8731_of_match); > + > static struct spi_driver wm8731_spi_driver = { > .driver = { > .name = "wm8731", > .owner = THIS_MODULE, > + .of_match_table = wm8731_of_match, > }, > .probe = wm8731_spi_probe, > .remove = __devexit_p(wm8731_spi_remove), > @@ -678,10 +687,13 @@ static const struct i2c_device_id wm8731_i2c_id[] = { > }; > MODULE_DEVICE_TABLE(i2c, wm8731_i2c_id); > > +MODULE_DEVICE_TABLE(i2c, wm8731_of_match); > + And this line can be dropped. i2c module info is separate from i2c and spi tables (although there may be some breakage here that need to be looked at. Right now I don't think the OF device ids get exported to udev for non-platform_devices. It needs some investigation.) > static struct i2c_driver wm8731_i2c_driver = { > .driver = { > .name = "wm8731", > .owner = THIS_MODULE, > + .of_match_table = wm8731_of_match, > }, > .probe = wm8731_i2c_probe, > .remove = __devexit_p(wm8731_i2c_remove), > -- > 1.7.5.4 > > _______________________________________________ > devicetree-discuss mailing list > devicetree-discuss(a)lists.ozlabs.org > https://lists.ozlabs.org/listinfo/devicetree-discuss

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[alsa-devel] [RME - hdsp] Multiface/Digiface/RPM detection
by Adrian Knoth 28 Jul '11

28 Jul '11

Hi! I've recently added support for the RME RPM to hdspmixer. Unfortunately, I don't have one, it's been done blindly with user feedback. This very user now reports that he needs to upload the device firmware from windows. I've checked hdsploader, and of course, it needs patching. I'll take care in a second. More surprisingly, though, the kernel wasn't able to upload the firmware itself, because it fails to detect the RPM and hence tries to upload a multiface firmware. After reading the kernel source, I think the code in hdsp.c is wrong: if (hdsp_fifo_wait(hdsp, 0, HDSP_SHORT_WAIT)) { hdsp_write(hdsp, HDSP_control2Reg, HDSP_VERSION_BIT); hdsp_write(hdsp, HDSP_control2Reg, HDSP_S_LOAD); if (hdsp_fifo_wait(hdsp, 0, HDSP_SHORT_WAIT)) hdsp->io_type = RPM; else hdsp->io_type = Multiface; } else { hdsp->io_type = Digiface; } Who here owns a Digiface and can confirm or deny that the kernel correctly detects it as Digiface? Same for Multiface, though I guess since it's more or less the default, users wouldn't notice it. What's wrong with the code above? I think all occurrences of HDSP_control2Reg in hdsp_check_for_iobox need to be changed to HDSP_controlRegister and the second hdsp_fifo_wait needs to be inverted. But this is pure guesswork. If I come up with a patch, who here has a RPM, Digiface or Multiface to test it? TIA

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Re: [alsa-devel] Build regressions/improvements in v3.0-rc7 (sound/isa/)
by Randy Dunlap 28 Jul '11

28 Jul '11

On Thu, 28 Jul 2011 14:24:49 -0400 Arnaud Lacombe wrote: > Hi, > > On Thu, Jul 28, 2011 at 2:11 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Thu, 28 Jul 2011 13:06:19 -0400, > > Arnaud Lacombe wrote: > >> > >> Hi, > >> > >> On Thu, Jul 28, 2011 at 12:44 PM, Randy Dunlap <rdunlap(a)xenotime.net> wrote: > >> > On Thu, 28 Jul 2011 12:36:27 -0400 Arnaud Lacombe wrote: > >> > > >> >> Hi, > >> >> > >> >> On Thu, Jul 28, 2011 at 12:21 PM, Randy Dunlap <rdunlap(a)xenotime.net> wrote: > >> >> > On Thu, 28 Jul 2011 11:17:09 +1000 Stephen Rothwell wrote: > >> >> > > >> >> >> Hi Randy, > >> >> >> > >> >> >> On Wed, 27 Jul 2011 12:13:20 -0700 Randy Dunlap <rdunlap(a)xenotime.net> wrote: > >> >> >> > > >> >> >> > On Mon, 25 Jul 2011 09:10:03 -0700 Randy Dunlap wrote: > >> >> >> > > >> >> >> > > On Mon, 25 Jul 2011 09:37:53 +0200 Takashi Iwai wrote: > >> >> >> > > > >> >> >> > > > At Sun, 24 Jul 2011 12:26:24 -0700, > >> >> >> > > > Randy Dunlap wrote: > >> >> >> > > > > > >> >> >> > > > > On Wed, 13 Jul 2011 21:33:30 +0200 Geert Uytterhoeven wrote: > >> >> >> > > > > > >> >> >> > > > > > + src/sound/isa/ad1816a/ad1816a_lib.c: error: implicit declaration of function 'snd_dma_disable': => 544 > >> >> >> > > > > > + src/sound/isa/ad1816a/ad1816a_lib.c: error: implicit declaration of function 'snd_dma_pointer': => 302 > >> >> >> > > > > > + src/sound/isa/ad1816a/ad1816a_lib.c: error: implicit declaration of function 'snd_dma_program': => 244 > >> >> >> > > > > > + src/sound/isa/es1688/es1688_lib.c: error: implicit declaration of function 'snd_dma_pointer': => 509 > >> >> >> > > > > > + src/sound/isa/es1688/es1688_lib.c: error: implicit declaration of function 'snd_dma_program': => 417 > >> >> >> > > > > > + src/sound/isa/gus/gus_dma.c: error: implicit declaration of function 'snd_dma_disable': => 177 > >> >> >> > > > > > + src/sound/isa/gus/gus_dma.c: error: implicit declaration of function 'snd_dma_program': => 79 > >> >> >> > > > > > + src/sound/isa/gus/gus_pcm.c: error: implicit declaration of function 'snd_dma_pointer': => 619 > >> >> >> > > > > > + src/sound/isa/gus/gus_pcm.c: error: implicit declaration of function 'snd_dma_program': => 591 > >> >> >> > > > > > + src/sound/isa/sb/sb16_main.c: error: implicit declaration of function 'snd_dma_pointer': => 456 > >> >> >> > > > > > + src/sound/isa/sb/sb16_main.c: error: implicit declaration of function 'snd_dma_program': => 276 > >> >> >> > > > > > + src/sound/isa/sb/sb8_main.c: error: implicit declaration of function 'snd_dma_pointer': => 425 > >> >> >> > > > > > + src/sound/isa/sb/sb8_main.c: error: implicit declaration of function 'snd_dma_program': => 172 > >> >> >> > > > > > + src/sound/isa/sscape.c: error: implicit declaration of function 'snd_dma_program': => 481 > >> >> >> > > > > > + src/sound/isa/wss/wss_lib.c: error: implicit declaration of function 'snd_dma_disable': => 1695 > >> >> >> > > > > > + src/sound/isa/wss/wss_lib.c: error: implicit declaration of function 'snd_dma_pointer': => 1160 > >> >> >> > > > > > + src/sound/isa/wss/wss_lib.c: error: implicit declaration of function 'snd_dma_program': => 1025 > >> >> >> > > > > > >> >> >> > > > > All of these source files do #include <sound/core.h>, > >> >> >> > > > > but these missing/implicit functions are conditional: > >> >> >> > > > > > >> >> >> > > > > #ifdef CONFIG_ISA_DMA_API > >> >> >> > > > > #define DMA_MODE_NO_ENABLE 0x0100 > >> >> >> > > > > > >> >> >> > > > > void snd_dma_program(unsigned long dma, unsigned long addr, unsigned int size, unsigned short mode); > >> >> >> > > > > void snd_dma_disable(unsigned long dma); > >> >> >> > > > > unsigned int snd_dma_pointer(unsigned long dma, unsigned int size); > >> >> >> > > > > #endif > >> >> >> > > > > > >> >> >> > > > > so should these drivers all depend on ISA_DMA_API ? > >> >> >> > > > > >> >> >> > > > Yes, these are ISA sound cards, so they must have CONFIG_ISA and > >> >> >> > > > CONFIG_ISA_DMA. Although CONFIG_SND_SB8_DSP, CONFIG_SND_SB16_DSP > >> >> >> > > > and CONFIG_SND_WSS_LIB have no direct dependency, but the drivers > >> >> >> > > > selecting these are all in "if SND_ISA" in sound/isa/Kconfig. > >> >> >> > > > >> >> >> > > Hm, yes, they are. > >> >> >> > > > >> >> >> > > > > or should only pieces of each driver depend on ISA_DMA_API ? > >> >> >> > > > > >> >> >> > > > I wonder which configuration makes it possible. > >> >> >> > > > Did you see a similar problem with randconfig? > >> >> >> > > > >> >> >> > > No. This parisc allmodconfig file from linux-next builds has > >> >> >> > > CONFIG_SND_ISA=y but it does not have CONFIG_ISA_DMA_API enabled. Weird. > >> >> >> > > > >> >> >> > > Geert, do you know how this happens? > >> >> >> > > > >> >> >> > > parisc config: http://kisskb.ellerman.id.au/kisskb/buildresult/4323458/config/ > >> >> >> > > from this failed build: http://kisskb.ellerman.id.au/kisskb/buildresult/4323458/ > >> >> >> > > >> >> >> > How is CONFIG_ISA_DMA_API enabled in parisc allmodconfig? > >> >> >> > I don't see how that happens, and it's causing build errors... > >> >> >> > >> >> >> Sorry, but I don't understand the question. The problam is that > >> >> >> CONFIG_ISA_DMA_API is not enabled, right? On parisc, it is just (from > >> >> >> arch/parisc/Kconfig) > >> >> >> > >> >> >> config ISA_DMA_API > >> >> >> bool > >> >> >> > >> >> >> and nothing selects it. However, I would have expected an allmodconfig > >> >> >> to set it to y. Clearly I don't understan how allmodconfig works any > >> >> >> more :-( > >> >> >> > >> >> >> Michal, is my understanding correct? Is this a bug in allmodconfig? > >> >> > > >> >> > When I do: > >> >> > $ make ARCH=parisc O=PAR allmodconfig > >> >> > > >> >> > I don't get CONFIG_ISA_DMA_API enabled, so the question is simply > >> >> > how is the linux-next build system ending up with CONFIG_ISA_DMA_API enabled? > >> >> > > >> >> Where do you see that "linux-next build system ending up with > >> >> CONFIG_ISA_DMA_API enabled" ? > >> >> > >> >> From what I understood of the problem, the issue is that > >> >> CONFIG_ISA_DMA_API does _not_ ends up enabled. > >> > > >> > Thanks for catching me. > >> > > >> > The problem (second try) is that ISA SND drivers are being built (on parisc) > >> > even though ISA_DMA_API is not enabled -- and should not be enabled, > >> > so the ISA SND drivers should not be built. I think I have it right this time. > >> > > >> > From the parisc .config file listed above: > >> > > >> > CONFIG_SND_ISA=y > >> > CONFIG_SND_ADLIB=m > >> > CONFIG_SND_AD1816A=m > >> > CONFIG_SND_AD1848=m > >> > CONFIG_SND_ALS100=m > >> > CONFIG_SND_AZT1605=m > >> > CONFIG_SND_AZT2316=m > >> > CONFIG_SND_AZT2320=m > >> > CONFIG_SND_CMI8330=m > >> > > >> > > >> > etc. But none of these should be enabled. AFAIK. > >> > So how does this happen? > >> > > >> warning: (RADIO_MIROPCM20) selects SND_ISA which has unmet direct > >> dependencies (SOUND && !M68K && SND && ISA && ISA_DMA_API) > > > > Hm, I thought this was fixed by > > commit 4756fc64ecd628efa6f8296711d7dbcefabf25dd > > Author: Ralf Baechle <ralf(a)linux-mips.org> > > Date: Fri Jun 24 10:30:09 2011 -0300 > > > > [media] MEDIA: Fix non-ISA_DMA_API link failure of sound code > > > > OK, it landed after 3.0-rc7, so this problem must be fixed now? > % git checkout v3.0-rc7 > % make ARCH=parisc allmodconfig > [...] > warning: (RADIO_WL128X) selects TI_ST which has unmet direct > dependencies (MISC_DEVICES && NET && GPIOLIB) > warning: (RADIO_MIROPCM20) selects SND_ISA which has unmet direct > dependencies (SOUND && !M68K && SND && ISA && ISA_DMA_API) > warning: (LOCKDEP && FAULT_INJECTION_STACKTRACE_FILTER && LATENCYTOP > && FUNCTION_TRACER && KMEMCHECK) selects FRAME_POINTER which has unmet > direct dependencies (DEBUG_KERNEL && (CRIS || M68K || FRV || UML || > AVR32 || SUPERH || BLACKFIN || MN10300) || ARCH_WANT_FRAME_POINTERS) > # > # configuration written to .config > # > % grep -c SND_ISA .config > 1 > > % git checkout v3.0 > Previous HEAD position was 620917d... Linux 3.0-rc7 > HEAD is now at 02f8c6a... Linux 3.0 > % make ARCH=parisc allmodconfig > [...] > warning: (RADIO_WL128X) selects TI_ST which has unmet direct > dependencies (MISC_DEVICES && NET && GPIOLIB) > warning: (LOCKDEP && FAULT_INJECTION_STACKTRACE_FILTER && LATENCYTOP > && FUNCTION_TRACER && KMEMCHECK) selects FRAME_POINTER which has unmet > direct dependencies (DEBUG_KERNEL && (CRIS || M68K || FRV || UML || > AVR32 || SUPERH || BLACKFIN || MN10300) || ARCH_WANT_FRAME_POINTERS) > # > # configuration written to .config > # > % grep -c SND_ISA .config > 0 > > Look good to me. Great, thanks everyone. Yes, I was working on old data. Sorry about that. Anyway, lots of build errors now fixed by Ralf. --- ~Randy *** Remember to use Documentation/SubmitChecklist when testing your code ***

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Re: [alsa-devel] Build regressions/improvements in v3.0-rc7 (sound/isa/)
by Takashi Iwai 28 Jul '11

28 Jul '11

At Thu, 28 Jul 2011 13:06:19 -0400, Arnaud Lacombe wrote: > > Hi, > > On Thu, Jul 28, 2011 at 12:44 PM, Randy Dunlap <rdunlap(a)xenotime.net> wrote: > > On Thu, 28 Jul 2011 12:36:27 -0400 Arnaud Lacombe wrote: > > > >> Hi, > >> > >> On Thu, Jul 28, 2011 at 12:21 PM, Randy Dunlap <rdunlap(a)xenotime.net> wrote: > >> > On Thu, 28 Jul 2011 11:17:09 +1000 Stephen Rothwell wrote: > >> > > >> >> Hi Randy, > >> >> > >> >> On Wed, 27 Jul 2011 12:13:20 -0700 Randy Dunlap <rdunlap(a)xenotime.net> wrote: > >> >> > > >> >> > On Mon, 25 Jul 2011 09:10:03 -0700 Randy Dunlap wrote: > >> >> > > >> >> > > On Mon, 25 Jul 2011 09:37:53 +0200 Takashi Iwai wrote: > >> >> > > > >> >> > > > At Sun, 24 Jul 2011 12:26:24 -0700, > >> >> > > > Randy Dunlap wrote: > >> >> > > > > > >> >> > > > > On Wed, 13 Jul 2011 21:33:30 +0200 Geert Uytterhoeven wrote: > >> >> > > > > > >> >> > > > > > + src/sound/isa/ad1816a/ad1816a_lib.c: error: implicit declaration of function 'snd_dma_disable': => 544 > >> >> > > > > > + src/sound/isa/ad1816a/ad1816a_lib.c: error: implicit declaration of function 'snd_dma_pointer': => 302 > >> >> > > > > > + src/sound/isa/ad1816a/ad1816a_lib.c: error: implicit declaration of function 'snd_dma_program': => 244 > >> >> > > > > > + src/sound/isa/es1688/es1688_lib.c: error: implicit declaration of function 'snd_dma_pointer': => 509 > >> >> > > > > > + src/sound/isa/es1688/es1688_lib.c: error: implicit declaration of function 'snd_dma_program': => 417 > >> >> > > > > > + src/sound/isa/gus/gus_dma.c: error: implicit declaration of function 'snd_dma_disable': => 177 > >> >> > > > > > + src/sound/isa/gus/gus_dma.c: error: implicit declaration of function 'snd_dma_program': => 79 > >> >> > > > > > + src/sound/isa/gus/gus_pcm.c: error: implicit declaration of function 'snd_dma_pointer': => 619 > >> >> > > > > > + src/sound/isa/gus/gus_pcm.c: error: implicit declaration of function 'snd_dma_program': => 591 > >> >> > > > > > + src/sound/isa/sb/sb16_main.c: error: implicit declaration of function 'snd_dma_pointer': => 456 > >> >> > > > > > + src/sound/isa/sb/sb16_main.c: error: implicit declaration of function 'snd_dma_program': => 276 > >> >> > > > > > + src/sound/isa/sb/sb8_main.c: error: implicit declaration of function 'snd_dma_pointer': => 425 > >> >> > > > > > + src/sound/isa/sb/sb8_main.c: error: implicit declaration of function 'snd_dma_program': => 172 > >> >> > > > > > + src/sound/isa/sscape.c: error: implicit declaration of function 'snd_dma_program': => 481 > >> >> > > > > > + src/sound/isa/wss/wss_lib.c: error: implicit declaration of function 'snd_dma_disable': => 1695 > >> >> > > > > > + src/sound/isa/wss/wss_lib.c: error: implicit declaration of function 'snd_dma_pointer': => 1160 > >> >> > > > > > + src/sound/isa/wss/wss_lib.c: error: implicit declaration of function 'snd_dma_program': => 1025 > >> >> > > > > > >> >> > > > > All of these source files do #include <sound/core.h>, > >> >> > > > > but these missing/implicit functions are conditional: > >> >> > > > > > >> >> > > > > #ifdef CONFIG_ISA_DMA_API > >> >> > > > > #define DMA_MODE_NO_ENABLE 0x0100 > >> >> > > > > > >> >> > > > > void snd_dma_program(unsigned long dma, unsigned long addr, unsigned int size, unsigned short mode); > >> >> > > > > void snd_dma_disable(unsigned long dma); > >> >> > > > > unsigned int snd_dma_pointer(unsigned long dma, unsigned int size); > >> >> > > > > #endif > >> >> > > > > > >> >> > > > > so should these drivers all depend on ISA_DMA_API ? > >> >> > > > > >> >> > > > Yes, these are ISA sound cards, so they must have CONFIG_ISA and > >> >> > > > CONFIG_ISA_DMA. Although CONFIG_SND_SB8_DSP, CONFIG_SND_SB16_DSP > >> >> > > > and CONFIG_SND_WSS_LIB have no direct dependency, but the drivers > >> >> > > > selecting these are all in "if SND_ISA" in sound/isa/Kconfig. > >> >> > > > >> >> > > Hm, yes, they are. > >> >> > > > >> >> > > > > or should only pieces of each driver depend on ISA_DMA_API ? > >> >> > > > > >> >> > > > I wonder which configuration makes it possible. > >> >> > > > Did you see a similar problem with randconfig? > >> >> > > > >> >> > > No. This parisc allmodconfig file from linux-next builds has > >> >> > > CONFIG_SND_ISA=y but it does not have CONFIG_ISA_DMA_API enabled. Weird. > >> >> > > > >> >> > > Geert, do you know how this happens? > >> >> > > > >> >> > > parisc config: http://kisskb.ellerman.id.au/kisskb/buildresult/4323458/config/ > >> >> > > from this failed build: http://kisskb.ellerman.id.au/kisskb/buildresult/4323458/ > >> >> > > >> >> > How is CONFIG_ISA_DMA_API enabled in parisc allmodconfig? > >> >> > I don't see how that happens, and it's causing build errors... > >> >> > >> >> Sorry, but I don't understand the question. The problam is that > >> >> CONFIG_ISA_DMA_API is not enabled, right? On parisc, it is just (from > >> >> arch/parisc/Kconfig) > >> >> > >> >> config ISA_DMA_API > >> >> bool > >> >> > >> >> and nothing selects it. However, I would have expected an allmodconfig > >> >> to set it to y. Clearly I don't understan how allmodconfig works any > >> >> more :-( > >> >> > >> >> Michal, is my understanding correct? Is this a bug in allmodconfig? > >> > > >> > When I do: > >> > $ make ARCH=parisc O=PAR allmodconfig > >> > > >> > I don't get CONFIG_ISA_DMA_API enabled, so the question is simply > >> > how is the linux-next build system ending up with CONFIG_ISA_DMA_API enabled? > >> > > >> Where do you see that "linux-next build system ending up with > >> CONFIG_ISA_DMA_API enabled" ? > >> > >> From what I understood of the problem, the issue is that > >> CONFIG_ISA_DMA_API does _not_ ends up enabled. > > > > Thanks for catching me. > > > > The problem (second try) is that ISA SND drivers are being built (on parisc) > > even though ISA_DMA_API is not enabled -- and should not be enabled, > > so the ISA SND drivers should not be built. I think I have it right this time. > > > > From the parisc .config file listed above: > > > > CONFIG_SND_ISA=y > > CONFIG_SND_ADLIB=m > > CONFIG_SND_AD1816A=m > > CONFIG_SND_AD1848=m > > CONFIG_SND_ALS100=m > > CONFIG_SND_AZT1605=m > > CONFIG_SND_AZT2316=m > > CONFIG_SND_AZT2320=m > > CONFIG_SND_CMI8330=m > > > > > > etc. But none of these should be enabled. AFAIK. > > So how does this happen? > > > warning: (RADIO_MIROPCM20) selects SND_ISA which has unmet direct > dependencies (SOUND && !M68K && SND && ISA && ISA_DMA_API) Hm, I thought this was fixed by commit 4756fc64ecd628efa6f8296711d7dbcefabf25dd Author: Ralf Baechle <ralf(a)linux-mips.org> Date: Fri Jun 24 10:30:09 2011 -0300 [media] MEDIA: Fix non-ISA_DMA_API link failure of sound code OK, it landed after 3.0-rc7, so this problem must be fixed now? I'm not sure at which point this thread is focused on. Takashi

1 0

Re: [alsa-devel] Build regressions/improvements in v3.0-rc7 (sound/isa/)
by Randy Dunlap 28 Jul '11

28 Jul '11

On Thu, 28 Jul 2011 12:36:27 -0400 Arnaud Lacombe wrote: > Hi, >=20 > On Thu, Jul 28, 2011 at 12:21 PM, Randy Dunlap <rdunlap(a)xenotime.net> wro= te: > > On Thu, 28 Jul 2011 11:17:09 +1000 Stephen Rothwell wrote: > > > >> Hi Randy, > >> > >> On Wed, 27 Jul 2011 12:13:20 -0700 Randy Dunlap <rdunlap(a)xenotime.net>= wrote: > >> > > >> > On Mon, 25 Jul 2011 09:10:03 -0700 Randy Dunlap wrote: > >> > > >> > > On Mon, 25 Jul 2011 09:37:53 +0200 Takashi Iwai wrote: > >> > > > >> > > > At Sun, 24 Jul 2011 12:26:24 -0700, > >> > > > Randy Dunlap wrote: > >> > > > > > >> > > > > On Wed, 13 Jul 2011 21:33:30 +0200 Geert Uytterhoeven wrote: > >> > > > > > >> > > > > > =A0 + src/sound/isa/ad1816a/ad1816a_lib.c: error: implicit d= eclaration of function 'snd_dma_disable': =A0=3D> 544 > >> > > > > > =A0 + src/sound/isa/ad1816a/ad1816a_lib.c: error: implicit d= eclaration of function 'snd_dma_pointer': =A0=3D> 302 > >> > > > > > =A0 + src/sound/isa/ad1816a/ad1816a_lib.c: error: implicit d= eclaration of function 'snd_dma_program': =A0=3D> 244 > >> > > > > > =A0 + src/sound/isa/es1688/es1688_lib.c: error: implicit dec= laration of function 'snd_dma_pointer': =A0=3D> 509 > >> > > > > > =A0 + src/sound/isa/es1688/es1688_lib.c: error: implicit dec= laration of function 'snd_dma_program': =A0=3D> 417 > >> > > > > > =A0 + src/sound/isa/gus/gus_dma.c: error: implicit declarati= on of function 'snd_dma_disable': =A0=3D> 177 > >> > > > > > =A0 + src/sound/isa/gus/gus_dma.c: error: implicit declarati= on of function 'snd_dma_program': =A0=3D> 79 > >> > > > > > =A0 + src/sound/isa/gus/gus_pcm.c: error: implicit declarati= on of function 'snd_dma_pointer': =A0=3D> 619 > >> > > > > > =A0 + src/sound/isa/gus/gus_pcm.c: error: implicit declarati= on of function 'snd_dma_program': =A0=3D> 591 > >> > > > > > =A0 + src/sound/isa/sb/sb16_main.c: error: implicit declarat= ion of function 'snd_dma_pointer': =A0=3D> 456 > >> > > > > > =A0 + src/sound/isa/sb/sb16_main.c: error: implicit declarat= ion of function 'snd_dma_program': =A0=3D> 276 > >> > > > > > =A0 + src/sound/isa/sb/sb8_main.c: error: implicit declarati= on of function 'snd_dma_pointer': =A0=3D> 425 > >> > > > > > =A0 + src/sound/isa/sb/sb8_main.c: error: implicit declarati= on of function 'snd_dma_program': =A0=3D> 172 > >> > > > > > =A0 + src/sound/isa/sscape.c: error: implicit declaration of= function 'snd_dma_program': =A0=3D> 481 > >> > > > > > =A0 + src/sound/isa/wss/wss_lib.c: error: implicit declarati= on of function 'snd_dma_disable': =A0=3D> 1695 > >> > > > > > =A0 + src/sound/isa/wss/wss_lib.c: error: implicit declarati= on of function 'snd_dma_pointer': =A0=3D> 1160 > >> > > > > > =A0 + src/sound/isa/wss/wss_lib.c: error: implicit declarati= on of function 'snd_dma_program': =A0=3D> 1025 > >> > > > > > >> > > > > All of these source files do #include <sound/core.h>, > >> > > > > but these missing/implicit functions are conditional: > >> > > > > > >> > > > > #ifdef CONFIG_ISA_DMA_API > >> > > > > #define DMA_MODE_NO_ENABLE =A0 =A00x0100 > >> > > > > > >> > > > > void snd_dma_program(unsigned long dma, unsigned long addr, un= signed int size, unsigned short mode); > >> > > > > void snd_dma_disable(unsigned long dma); > >> > > > > unsigned int snd_dma_pointer(unsigned long dma, unsigned int s= ize); > >> > > > > #endif > >> > > > > > >> > > > > so should these drivers all depend on ISA_DMA_API ? > >> > > > > >> > > > Yes, these are ISA sound cards, so they must have CONFIG_ISA and > >> > > > CONFIG_ISA_DMA. =A0Although CONFIG_SND_SB8_DSP, CONFIG_SND_SB16_= DSP > >> > > > and CONFIG_SND_WSS_LIB have no direct dependency, but the drivers > >> > > > selecting these are all in "if SND_ISA" in sound/isa/Kconfig. > >> > > > >> > > Hm, yes, they are. > >> > > > >> > > > > or should only pieces of each driver depend on ISA_DMA_API ? > >> > > > > >> > > > I wonder which configuration makes it possible. > >> > > > Did you see a similar problem with randconfig? > >> > > > >> > > No. =A0This parisc allmodconfig file from linux-next builds has > >> > > CONFIG_SND_ISA=3Dy but it does not have CONFIG_ISA_DMA_API enabled= . =A0Weird. > >> > > > >> > > Geert, do you know how this happens? > >> > > > >> > > parisc config: =A0http://kisskb.ellerman.id.au/kisskb/buildresult/= 4323458/config/ > >> > > from this failed build: =A0http://kisskb.ellerman.id.au/kisskb/bui= ldresult/4323458/ > >> > > >> > How is CONFIG_ISA_DMA_API enabled in parisc allmodconfig? > >> > I don't see how that happens, and it's causing build errors... > >> > >> Sorry, but I don't understand the question. =A0The problam is that > >> CONFIG_ISA_DMA_API is not enabled, right? =A0On parisc, it is just (fr= om > >> arch/parisc/Kconfig) > >> > >> config ISA_DMA_API > >> =A0 =A0 =A0 bool > >> > >> and nothing selects it. =A0However, I would have expected an allmodcon= fig > >> to set it to y. =A0Clearly I don't understan how allmodconfig works any > >> more :-( > >> > >> Michal, is my understanding correct? =A0Is this a bug in allmodconfig? > > > > When I do: > > $ make ARCH=3Dparisc O=3DPAR allmodconfig > > > > I don't get CONFIG_ISA_DMA_API enabled, so the question is simply > > how is the linux-next build system ending up with CONFIG_ISA_DMA_API en= abled? > > > Where do you see that "linux-next build system ending up with > CONFIG_ISA_DMA_API enabled" ? >=20 > From what I understood of the problem, the issue is that > CONFIG_ISA_DMA_API does _not_ ends up enabled. Thanks for catching me. The problem (second try) is that ISA SND drivers are being built (on parisc) even though ISA_DMA_API is not enabled -- and should not be enabled, so the ISA SND drivers should not be built. I think I have it right this t= ime. _From the parisc .config file listed above: CONFIG_SND_ISA=3Dy CONFIG_SND_ADLIB=3Dm CONFIG_SND_AD1816A=3Dm CONFIG_SND_AD1848=3Dm CONFIG_SND_ALS100=3Dm CONFIG_SND_AZT1605=3Dm CONFIG_SND_AZT2316=3Dm CONFIG_SND_AZT2320=3Dm CONFIG_SND_CMI8330=3Dm etc. But none of these should be enabled. AFAIK. So how does this happen? > - Arnaud >=20 > > What commands are used to cause this? --- ~Randy *** Remember to use Documentation/SubmitChecklist when testing your code ***

1 0

Re: [alsa-devel] Build regressions/improvements in v3.0-rc7 (sound/isa/)
by Randy Dunlap 28 Jul '11

28 Jul '11

On Wed, 13 Jul 2011 21:33:30 +0200 Geert Uytterhoeven wrote: > + src/sound/isa/ad1816a/ad1816a_lib.c: error: implicit declaration of function 'snd_dma_disable': => 544 > + src/sound/isa/ad1816a/ad1816a_lib.c: error: implicit declaration of function 'snd_dma_pointer': => 302 > + src/sound/isa/ad1816a/ad1816a_lib.c: error: implicit declaration of function 'snd_dma_program': => 244 > + src/sound/isa/es1688/es1688_lib.c: error: implicit declaration of function 'snd_dma_pointer': => 509 > + src/sound/isa/es1688/es1688_lib.c: error: implicit declaration of function 'snd_dma_program': => 417 > + src/sound/isa/gus/gus_dma.c: error: implicit declaration of function 'snd_dma_disable': => 177 > + src/sound/isa/gus/gus_dma.c: error: implicit declaration of function 'snd_dma_program': => 79 > + src/sound/isa/gus/gus_pcm.c: error: implicit declaration of function 'snd_dma_pointer': => 619 > + src/sound/isa/gus/gus_pcm.c: error: implicit declaration of function 'snd_dma_program': => 591 > + src/sound/isa/sb/sb16_main.c: error: implicit declaration of function 'snd_dma_pointer': => 456 > + src/sound/isa/sb/sb16_main.c: error: implicit declaration of function 'snd_dma_program': => 276 > + src/sound/isa/sb/sb8_main.c: error: implicit declaration of function 'snd_dma_pointer': => 425 > + src/sound/isa/sb/sb8_main.c: error: implicit declaration of function 'snd_dma_program': => 172 > + src/sound/isa/sscape.c: error: implicit declaration of function 'snd_dma_program': => 481 > + src/sound/isa/wss/wss_lib.c: error: implicit declaration of function 'snd_dma_disable': => 1695 > + src/sound/isa/wss/wss_lib.c: error: implicit declaration of function 'snd_dma_pointer': => 1160 > + src/sound/isa/wss/wss_lib.c: error: implicit declaration of function 'snd_dma_program': => 1025 All of these source files do #include <sound/core.h>, but these missing/implicit functions are conditional: #ifdef CONFIG_ISA_DMA_API #define DMA_MODE_NO_ENABLE 0x0100 void snd_dma_program(unsigned long dma, unsigned long addr, unsigned int size, unsigned short mode); void snd_dma_disable(unsigned long dma); unsigned int snd_dma_pointer(unsigned long dma, unsigned int size); #endif so should these drivers all depend on ISA_DMA_API ? or should only pieces of each driver depend on ISA_DMA_API ? --- ~Randy *** Remember to use Documentation/SubmitChecklist when testing your code ***

3 5

[alsa-devel] Hardware audio decoder driver design
by Nikita V. Youshchenko 28 Jul '11

28 Jul '11

Hello. Soon I will work on a linux kernel driver for a custom audio decoder device that is being developed by a company I work for. If not going into details, that devices reads A52-encoded stream from system memory, and writes raw pcm stream to system memory. Simplest thing to do is - implement a character device, where user-space will write encoded stream, and from where user-space will read decoded stream. (driver for similar hardware, located at http://sourceforge.net/projects/vs10xx/, does that). However, perhaps a better architecture (e.g. in-kernel intergation with an audio sink) is possible? I'm looking for any related information - e.g. ideas on what interface to implement, examples of drivers for similar devices, etc. Thanks on any hints. Nikita Youshchenko, embedded linux developer.

3 2

[alsa-devel] hda-intel and runtime power-management
by Tatulea, Dragos 28 Jul '11

28 Jul '11

Hi, I'm in the process of enabling pm runtime pm for the hda_intel driver, not being sure if this is a sane task :). Noticed that CONFIG_SND_HDA_POWER_SAVE disables codecs and chip when they are not in use, but the power gain is not noticeable on my hw. On the other hand, a full suspend (azx_suspend) does save a few hundred mW's. I am aware that this is hw specific, but I would like to take advantage of the full suspend. Suspend/resume latency is below jiffy resolution. How about a more aggressive scheme?: allow runtime power management (with azx_suspend/resume as runtime pm ops) to kick in when all streams are stopped/suspended/paused. And resume upon stream is start/resume. I don't have enough context into ALSA & sound drivers so I need a reality check: is this crazy/stupid? Thanks, Dragos

2 1

[alsa-devel] [PATCH 5/6] sound/core/pcm_compat.c: adjust array index
by Julia Lawall 28 Jul '11

28 Jul '11

From: Julia Lawall <julia(a)diku.dk> Convert array index from the loop bound to the loop index. A simplified version of the semantic patch that fixes this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> @@ expression e1,e2,ar; @@ for(e1 = 0; e1 < e2; e1++) { <... ar[ - e2 + e1 ] ...> } // </smpl> Signed-off-by: Julia Lawall <julia(a)diku.dk> --- Not tested. sound/core/pcm_compat.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff -u -p a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c --- a/sound/core/pcm_compat.c +++ b/sound/core/pcm_compat.c @@ -342,7 +342,7 @@ static int snd_pcm_ioctl_xfern_compat(st kfree(bufs); return -EFAULT; } - bufs[ch] = compat_ptr(ptr); + bufs[i] = compat_ptr(ptr); bufptr++; } if (dir == SNDRV_PCM_STREAM_PLAYBACK)

2 1

Re: [alsa-devel] Realtek ALC889: HDA Intel and kernel 3.0 gives choppy sound
by Takashi Iwai 28 Jul '11

28 Jul '11

At Tue, 26 Jul 2011 23:14:47 +0200, Colomban Wendling wrote: > > Hi, > > Following a bug report started in Debian BTS, in bug 635362 [1], I was > asked to post here after testing upstream kernels and reproducing the bug. > > So, here's the problem: my HDA audio chip [2] started to output very > choppy sound with Linux kernel 3.0, while it worked perfectly well in > previous versions -- latest two I used being 2.6.38 and 2.6.39 from > Debian. Please note that simply booting on 2.6.39 works as usual, so > it's IMHO definitely a driver regression. Do you have 2.6.39 kernel messages? I see the warnings in your dmesg below in 3.0* kernels. Did you get it with the working kernel as well? > [ 0.000000] ------------[ cut here ]------------ > [ 0.000000] WARNING: at drivers/pci/dmar.c:634 warn_invalid_dmar+0x77/0x85() > [ 0.000000] Hardware name: MS-7636 > [ 0.000000] Your BIOS is broken; DMAR reported at address fed93000 returns all ones! > [ 0.000000] BIOS vendor: American Megatrends Inc.; Ver: V1.1; Product Version: 1.0 > [ 0.000000] Modules linked in: > [ 0.000000] Pid: 0, comm: swapper Not tainted 3.0.0 #4 > [ 0.000000] Call Trace: > [ 0.000000] [<ffffffff81043359>] ? warn_slowpath_common+0x78/0x8c > [ 0.000000] [<ffffffff810433bb>] ? warn_slowpath_fmt_taint+0x3d/0x42 > [ 0.000000] [<ffffffff81318a7e>] ? __pte+0x7/0x8 > [ 0.000000] [<ffffffff816b2a2d>] ? __early_set_fixmap+0x89/0x8d > [ 0.000000] [<ffffffff81318a7e>] ? __pte+0x7/0x8 > [ 0.000000] [<ffffffff816b2a2d>] ? __early_set_fixmap+0x89/0x8d > [ 0.000000] [<ffffffff81318a7e>] ? __pte+0x7/0x8 > [ 0.000000] [<ffffffff811b72c9>] ? warn_invalid_dmar+0x77/0x85 > [ 0.000000] [<ffffffff816b2e3a>] ? early_iounmap+0xc5/0x114 > [ 0.000000] [<ffffffff816c6f13>] ? check_zero_address+0xb0/0xeb > [ 0.000000] [<ffffffff81327736>] ? bad_to_user+0x620/0x620 > [ 0.000000] [<ffffffff816c6f5f>] ? detect_intel_iommu+0x11/0xa6 > [ 0.000000] [<ffffffff816a5b29>] ? pci_iommu_alloc+0x3f/0x67 > [ 0.000000] [<ffffffff816b280c>] ? mem_init+0x14/0xe5 > [ 0.000000] [<ffffffff8169f94e>] ? start_kernel+0x1d0/0x3c8 > [ 0.000000] [<ffffffff8169f140>] ? early_idt_handlers+0x140/0x140 > [ 0.000000] [<ffffffff8169f3c4>] ? x86_64_start_kernel+0x104/0x111 > [ 0.000000] ---[ end trace a7919e7f17c0a725 ]--- Also, you can replace the files in sound/pci/hda/* from 2.6.39 kernel tree. If this fixes the problem, it's a sound driver bug. If not, it's anything else, e.g. the problem in more core side, e.g. IOMMU like above or the timer change, etc. Takashi

2 8

[alsa-devel] [PATCH v5 0/4] ASoC: mxs: add audio support
by Dong Aisheng 28 Jul '11

28 Jul '11

According to Liam's comment, i redo the patch set based on the for-next branch on Mark's sound-2.6 tree. The based one is [PATCH v4 00/10] ARM: mxs: add audio support. For asoc part (1-4), only one change is updating mxs_pcm_new since this API has been updated to pass in the runtime data only For mxs part, no changes made. This patch series is to add audio support based on saif for mxs platform. The driver only supports playback currently. For capturing, due to hardware limitation that we have to use two saif instances to implement full duplex (playback & recording), we need to figure out a good design to fit in ASoC as the following work. In addition, we simply just support master mode since saif tx can only work as a master due to hw limitation(rx can work in both master & slave mode). Tested on MX28EVK. For the ASoC-patches (1-4): Tested-by: Wolfram Sang <w.sang(a)pengutronix.de> For the mach-mxs-Patches (5-10): Reviewed-by: Wolfram Sang <w.sang(a)pengutronix.de> Changes since v4: * ASoC: mxs: add mxs-pcm driver update mxs_pcm_new due to this API's parameter number is changed. Changes since v3: This version only has one patch changed with another two for rebase the changed one. * [PATCH v4 06/10] ARM: mxs: add saif device Drop DIGCTRL-handling based on Wolfram's comment * [PATCH v4 07/10] ARM: mxs: add sgtl5000 i2c device Resend for rebase [PATCH v4 06/10] ARM: mxs: add saif device * [PATCH v4 08/10] ARM: mxs: add mxs-sgtl5000 device Resend for rebase [PATCH v4 06/10] ARM: mxs: add saif device changes since v2: * mxs-pcm - fix wrong periods_max. - For robustness replace the channels_min check with a check for the relevant substream being present * mxs-saif - Add IRQ_NONE handle - improve mxs_saif_set_clk to make it more easy to read - correct return value - request_irq sequence changed to after ioremap * mxs-sgtl5000 - clean up a few unused code * Kconfig&Makefile - SND_SOC_MXS_SGTL5000 should depend on I2C in Kconfig * mxs - Select MXS_I2C in kconfig for mxs-sgtl5000 changes since v1: * export saif mclk control interface to machine driver. Machine driver can decide whether it needs to use mclk supplied by saif according to its requirement. * error checking improvement * remove a few unneccesary code Dong Aisheng (4): ASoC: mxs: add mxs-pcm driver ASoC: mxs: add mxs-saif driver ASoC: mxs: add mxs-sgtl5000 machine driver ASoC: mxs: add asoc configuration files sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/mxs/Kconfig | 20 ++ sound/soc/mxs/Makefile | 10 + sound/soc/mxs/mxs-pcm.c | 359 ++++++++++++++++++++++ sound/soc/mxs/mxs-pcm.h | 43 +++ sound/soc/mxs/mxs-saif.c | 677 ++++++++++++++++++++++++++++++++++++++++++ sound/soc/mxs/mxs-saif.h | 130 ++++++++ sound/soc/mxs/mxs-sgtl5000.c | 165 ++++++++++ 9 files changed, 1406 insertions(+), 0 deletions(-) create mode 100644 sound/soc/mxs/Kconfig create mode 100644 sound/soc/mxs/Makefile create mode 100644 sound/soc/mxs/mxs-pcm.c create mode 100644 sound/soc/mxs/mxs-pcm.h create mode 100644 sound/soc/mxs/mxs-saif.c create mode 100644 sound/soc/mxs/mxs-saif.h create mode 100644 sound/soc/mxs/mxs-sgtl5000.c

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[alsa-devel] [RME - hdsp] alsa-lib's hdsp.h out of sync
by Adrian Knoth 28 Jul '11

28 Jul '11

Hi! While hacking on RPM support for hdspmixer, I've noticed that alsa-lib's hdsp.h doesn't reflect the kernel's hdsp.h: RPM is missing in HDSP_IO_Type. AFAIK, somebody has to regenerate alsa-lib's hdsp.h, right? Please do so. ;) Cheers

2 3

Re: [alsa-devel] [patch] ALSA: asihpi - off by one in asihpi_hpi_ioctl()
by Eliot Blennerhassett 28 Jul '11

28 Jul '11

On 28/07/11 00:55, Dan Rosenberg wrote: > On Wed, 2011-07-27 at 15:02 +0300, Dan Carpenter wrote: >> "adapter" is used as an array index in the adapters[] array so >> the off by one would make us read past the end. >> > > Agreed. I also don't like the fact that the "pa" pointer can be set to > an arbitrary address because the index isn't checked until after its > assignment. Even though the fix to the check prevents this pointer from > being dereferenced if it's out-of-bounds, it's still wrong. :( I have to agree. I'll submit another patch that avoids doing this. > I agree. This code seems to make assumptions in more than one place > that the adapters array is fully populated with non-NULL elements. At a > glance, I can't see where such initialization occurs though. It does, in void __init asihpi_init(void) ... memset(adapters, 0, sizeof(adapters)); -- Eliot Blennerhassett AudioScience Inc.

1 0

[alsa-devel] [PATCH V2 0/3] ASoC: au1x: update PSC AC97/I2S code
by Manuel Lauss 27 Jul '11

27 Jul '11

The following two patches depend on patchset "ASoC for Alchemy Au1000/1500/1100" to apply and work. Patch #1 moves the DBDMA (PCM) device registration from the AC97/I2S drivers into the board code. Patch #2 changes use of "soc-audio" platform device int db1200 machine code to the new way. Patch #3 removes the use of custom PCM_TX/RX constants in favour of the established SNDRV_PCM_SUBTREAM_PLAYBACK/CAPTURE. Changes since V1: - added patch #3 - tidied patch #1 a bit more. Run-tested on DB1200, DB1300 and DB1550 boards. Manuel Lauss (3): ASoC: au1x: remove automatic DMA device registration from PSC drivers ASoC: au1x: update db1200 machine to the new way of things ASoC: au1x: use substream stream info directly arch/mips/alchemy/devboards/db1200/platform.c | 16 +++++ sound/soc/au1x/db1200.c | 64 ++++++++++++------ sound/soc/au1x/dbdma2.c | 91 +++++-------------------- sound/soc/au1x/psc-ac97.c | 48 +++++++++----- sound/soc/au1x/psc-i2s.c | 42 +++++++---- sound/soc/au1x/psc.h | 11 --- 6 files changed, 133 insertions(+), 139 deletions(-) -- 1.7.6 -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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[alsa-devel] [PATCH V5 0/3] ASoC for Alchemy Au1000/1500/1100
by Manuel Lauss 27 Jul '11

27 Jul '11

Hello, Here's V5 of the AC97/I2S ASoC patchset for early Alchemy chips and their respective evaluation boards. The patches are largely based on the old mips/au1x00.c driver which they replace. AC97 Tested on a Db1500 development board; I2S untested since none of the testboards I have actually have an I2S codec (just testpoints). Changes since V4: - incorporate feedback from Liam Girdwood. - rediff against latest ASoC changes Changes since V3: - dropped the hunk which removed the I2S constants from the au1000.h header to avoid merge conflicts with other patches. - use the context structure declared in psc.h. Follow-up patches for psc* code depend on this. Changes since V2: - implemented changes after feedback from Lars-Peter Clausen: * split patch 1 in two, one for the ASoC drivers, and a separate for DB1000 machine code. * get rid of automatic dma device registration * tidied the I2S/AC97 sources - mark sound/mips/au1x00.c as DEPRECATED instead of removing it outright. Changes since V1: - added untested I2S controller driver for completeness, removed the audio defines from the au1000 header. Manuel Lauss (3): ASoC: Alchemy AC97C/I2SC audio support ASoC: Add a DB1x00 AC97 machine driver ALSA: deprecate MIPS AU1X00 AC97 driver arch/mips/alchemy/devboards/db1x00/platform.c | 48 ++++ sound/mips/Kconfig | 5 +- sound/soc/au1x/Kconfig | 28 ++ sound/soc/au1x/Makefile | 10 + sound/soc/au1x/ac97c.c | 365 ++++++++++++++++++++++++ sound/soc/au1x/db1000.c | 75 +++++ sound/soc/au1x/dma.c | 377 +++++++++++++++++++++++++ sound/soc/au1x/i2sc.c | 347 +++++++++++++++++++++++ sound/soc/au1x/psc.h | 19 +- 9 files changed, 1264 insertions(+), 10 deletions(-) create mode 100644 sound/soc/au1x/ac97c.c create mode 100644 sound/soc/au1x/db1000.c create mode 100644 sound/soc/au1x/dma.c create mode 100644 sound/soc/au1x/i2sc.c -- 1.7.6 -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

4 11

Re: [alsa-devel] [PATCH 1/1] oss: rename local change_bits to avoid powerpc bitsops.h definition
by Takashi Iwai 27 Jul '11

27 Jul '11

At Wed, 27 Jul 2011 17:48:41 +0100, Andy Whitcroft wrote: > > This collides with powerpc exported functions from bitops.h. Rename the > local copy in the oss soundblaster mixer and ad1848 driver. > > Signed-off-by: Andy Whitcroft <apw(a)canonical.com> Applied now. Thanks. Takashi > --- > sound/oss/ad1848.c | 6 +++--- > sound/oss/sb_mixer.c | 6 +++--- > 2 files changed, 6 insertions(+), 6 deletions(-) > > diff --git a/sound/oss/ad1848.c b/sound/oss/ad1848.c > index 4d2a6ae..8415158 100644 > --- a/sound/oss/ad1848.c > +++ b/sound/oss/ad1848.c > @@ -458,7 +458,7 @@ static int ad1848_set_recmask(ad1848_info * devc, int mask) > return mask; > } > > -static void change_bits(ad1848_info * devc, unsigned char *regval, > +static void oss_change_bits(ad1848_info * devc, unsigned char *regval, > unsigned char *muteval, int dev, int chn, int newval) > { > unsigned char mask; > @@ -516,10 +516,10 @@ static void ad1848_mixer_set_channel(ad1848_info *devc, int dev, int value, int > > if (muteregoffs != regoffs) { > muteval = ad_read(devc, muteregoffs); > - change_bits(devc, &val, &muteval, dev, channel, value); > + oss_change_bits(devc, &val, &muteval, dev, channel, value); > } > else > - change_bits(devc, &val, &val, dev, channel, value); > + oss_change_bits(devc, &val, &val, dev, channel, value); > > spin_lock_irqsave(&devc->lock,flags); > ad_write(devc, regoffs, val); > diff --git a/sound/oss/sb_mixer.c b/sound/oss/sb_mixer.c > index 2039d31..0266088 100644 > --- a/sound/oss/sb_mixer.c > +++ b/sound/oss/sb_mixer.c > @@ -232,7 +232,7 @@ static int detect_mixer(sb_devc * devc) > return 1; > } > > -static void change_bits(sb_devc * devc, unsigned char *regval, int dev, int chn, int newval) > +static void oss_change_bits(sb_devc * devc, unsigned char *regval, int dev, int chn, int newval) > { > unsigned char mask; > int shift; > @@ -284,7 +284,7 @@ int sb_common_mixer_set(sb_devc * devc, int dev, int left, int right) > return -EINVAL; > > val = sb_getmixer(devc, regoffs); > - change_bits(devc, &val, dev, LEFT_CHN, left); > + oss_change_bits(devc, &val, dev, LEFT_CHN, left); > > if ((*devc->iomap)[dev][RIGHT_CHN].regno != regoffs) /* > * Change register > @@ -304,7 +304,7 @@ int sb_common_mixer_set(sb_devc * devc, int dev, int left, int right) > * Read the new one > */ > } > - change_bits(devc, &val, dev, RIGHT_CHN, right); > + oss_change_bits(devc, &val, dev, RIGHT_CHN, right); > > sb_setmixer(devc, regoffs, val); > > -- > 1.7.4.1 >

1 0

[alsa-devel] Searching an argument in mailing post
by Ru Vuott 27 Jul '11

27 Jul '11

Hello, how can I do searching an particular argument in Alsa-mailing list post ?

1 0

[alsa-devel] Problem trying to add ASoC ADAV801 support to my kernel
by Wayne Tams 27 Jul '11

27 Jul '11

Hi, I am trying to add support for the ADAV801 codec to my 2.6.35.3 kernel. I came across this patch http://article.gmane.org/gmane.linux.alsa.devel/86474 obviously I had to make some modifications so it would correctly patch my source. However when I come to build the kernel I'm getting a whole host of errors and I am little unsure on how to proceed, could anyone maybe provide some guidance on the possible causes of my errors or maybe there is a more update driver available. Many thanks W wa CC sound/soc/codecs/adav80x.o ^[sound/soc/codecs/adav80x.c: In function 'adav80x_set_sysclk': sound/soc/codecs/adav80x.c:556:27: error: 'struct snd_soc_codec' has no member named 'dapm' sound/soc/codecs/adav80x.c: In function 'adav80x_set_pll': sound/soc/codecs/adav80x.c:630:27: error: 'struct snd_soc_codec' has no member named 'dapm' sound/soc/codecs/adav80x.c: In function 'adav80x_set_bias_level': sound/soc/codecs/adav80x.c:654:7: error: 'struct snd_soc_codec' has no member named 'dapm' sound/soc/codecs/adav80x.c: At top level: sound/soc/codecs/adav80x.c:698:34: error: array type has incomplete element type sound/soc/codecs/adav80x.c:700:3: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:700:3: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:701:3: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:701:3: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:702:3: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:702:3: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:703:4: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:703:4: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:704:4: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:704:4: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:705:4: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:705:4: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:706:4: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:706:4: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:707:4: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:707:4: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:709:3: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:709:3: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:710:4: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:710:4: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:711:4: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:711:4: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:712:4: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:712:4: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:713:4: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:713:4: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:714:4: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:714:4: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:716:3: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:716:3: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:719:3: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:719:3: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:720:3: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:720:3: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:721:3: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:721:3: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:722:4: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:722:4: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:723:4: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:723:4: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:724:4: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:724:4: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:725:4: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:725:4: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:726:4: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:726:4: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:728:3: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:728:3: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:729:4: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:729:4: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:730:4: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:730:4: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:731:4: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:731:4: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:732:4: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:732:4: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:733:4: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:733:4: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:735:3: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:735:3: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c: In function 'adav80x_resume': sound/soc/codecs/adav80x.c:767:2: error: implicit declaration of function 'snd_soc_cache_sync' sound/soc/codecs/adav80x.c: At top level: sound/soc/codecs/adav80x.c:777:15: error: variable 'adav80x_codec_driver' has initializer but incomplete type sound/soc/codecs/adav80x.c:778:2: error: unknown field 'probe' specified in initializer sound/soc/codecs/adav80x.c:778:2: warning: excess elements in struct initializer sound/soc/codecs/adav80x.c:778:2: warning: (near initialization for 'adav80x_codec_driver') sound/soc/codecs/adav80x.c:779:2: error: unknown field 'remove' specified in initializer sound/soc/codecs/adav80x.c:779:2: warning: excess elements in struct initializer sound/soc/codecs/adav80x.c:779:2: warning: (near initialization for 'adav80x_codec_driver') sound/soc/codecs/adav80x.c:780:2: error: unknown field 'suspend' specified in initializer sound/soc/codecs/adav80x.c:780:2: warning: excess elements in struct initializer sound/soc/codecs/adav80x.c:780:2: warning: (near initialization for 'adav80x_codec_driver') sound/soc/codecs/adav80x.c:781:2: error: unknown field 'resume' specified in initializer sound/soc/codecs/adav80x.c:781:2: warning: excess elements in struct initializer sound/soc/codecs/adav80x.c:781:2: warning: (near initialization for 'adav80x_codec_driver') sound/soc/codecs/adav80x.c:782:2: error: unknown field 'set_bias_level' specified in initializer sound/soc/codecs/adav80x.c:782:2: warning: excess elements in struct initializer sound/soc/codecs/adav80x.c:782:2: warning: (near initialization for 'adav80x_codec_driver') sound/soc/codecs/adav80x.c:784:2: error: unknown field 'set_pll' specified in initializer sound/soc/codecs/adav80x.c:784:2: warning: excess elements in struct initializer sound/soc/codecs/adav80x.c:784:2: warning: (near initialization for 'adav80x_codec_driver') sound/soc/codecs/adav80x.c:785:2: error: unknown field 'set_sysclk' specified in initializer sound/soc/codecs/adav80x.c:785:2: warning: excess elements in struct initializer sound/soc/codecs/adav80x.c:785:2: warning: (near initialization for 'adav80x_codec_driver') sound/soc/codecs/adav80x.c:787:2: error: unknown field 'reg_word_size' specified in initializer sound/soc/codecs/adav80x.c:787:2: warning: excess elements in struct initializer sound/soc/codecs/adav80x.c:787:2: warning: (near initialization for 'adav80x_codec_driver') sound/soc/codecs/adav80x.c:788:2: error: unknown field 'reg_cache_size' specified in initializer sound/soc/codecs/adav80x.c:788:103: warning: excess elements in struct initializer sound/soc/codecs/adav80x.c:788:103: warning: (near initialization for 'adav80x_codec_driver') sound/soc/codecs/adav80x.c:789:2: error: unknown field 'reg_cache_default' specified in initializer sound/soc/codecs/adav80x.c:789:2: warning: excess elements in struct initializer sound/soc/codecs/adav80x.c:789:2: warning: (near initialization for 'adav80x_codec_driver') sound/soc/codecs/adav80x.c:791:2: error: unknown field 'controls' specified in initializer sound/soc/codecs/adav80x.c:791:2: warning: excess elements in struct initializer sound/soc/codecs/adav80x.c:791:2: warning: (near initialization for 'adav80x_codec_driver') sound/soc/codecs/adav80x.c:792:2: error: unknown field 'num_controls' specified in initializer sound/soc/codecs/adav80x.c:792:93: warning: excess elements in struct initializer sound/soc/codecs/adav80x.c:792:93: warning: (near initialization for 'adav80x_codec_driver') sound/soc/codecs/adav80x.c:793:2: error: unknown field 'dapm_widgets' specified in initializer sound/soc/codecs/adav80x.c:793:2: warning: excess elements in struct initializer sound/soc/codecs/adav80x.c:793:2: warning: (near initialization for 'adav80x_codec_driver') sound/soc/codecs/adav80x.c:794:2: error: unknown field 'num_dapm_widgets' specified in initializer sound/soc/codecs/adav80x.c:794:105: warning: excess elements in struct initializer sound/soc/codecs/adav80x.c:794:105: warning: (near initialization for 'adav80x_codec_driver') sound/soc/codecs/adav80x.c:795:2: error: unknown field 'dapm_routes' specified in initializer sound/soc/codecs/adav80x.c:795:2: warning: excess elements in struct initializer sound/soc/codecs/adav80x.c:795:2: warning: (near initialization for 'adav80x_codec_driver') sound/soc/codecs/adav80x.c:796:2: error: unknown field 'num_dapm_routes' specified in initializer sound/soc/codecs/adav80x.c:796:102: warning: excess elements in struct initializer sound/soc/codecs/adav80x.c:796:102: warning: (near initialization for 'adav80x_codec_driver') sound/soc/codecs/adav80x.c: In function 'adav80x_bus_probe': sound/soc/codecs/adav80x.c:813:84: warning: type defaults to 'int' in type name sound/soc/codecs/adav80x.c:813:84: warning: type defaults to 'int' in type name sound/soc/codecs/adav80x.c:813:84: error: negative width in bit-field '<anonymous>' sound/soc/codecs/adav80x.c:813:84: warning: passing argument 1 of 'snd_soc_register_codec' from incompatible pointer type include/sound/soc.h:239:5: note: expected 'struct snd_soc_codec *' but argument is of type 'struct device *' sound/soc/codecs/adav80x.c:813:84: error: too many arguments to function 'snd_soc_register_codec' include/sound/soc.h:239:5: note: declared here sound/soc/codecs/adav80x.c: In function 'adav80x_bus_remove': sound/soc/codecs/adav80x.c:822:2: warning: passing argument 1 of 'snd_soc_unregister_codec' from incompatible pointer type include/sound/soc.h:240:6: note: expected 'struct snd_soc_codec *' but argument is of type 'struct device *' sound/soc/codecs/adav80x.c: At top level: sound/soc/codecs/adav80x.c:698:34: warning: 'adav80x_dais' defined but not used make[4]: *** [sound/soc/codecs/adav80x.o] Error 1 make[3]: *** [sound/soc/codecs] Error 2 make[2]: *** [sound/soc] Error 2 make[1]: *** [sound] Error 2 make[1]: Leaving directory `/home/wayne/sources/k/kernel/linux-2.6.35.3-imx' make: *** [/home/wayne/sources/k/kernel/linux-2.6.35.3-imx/.stamp_image_built] Error 2

2 2

[alsa-devel] [patch] ALSA: asihpi - off by one in asihpi_hpi_ioctl()
by Dan Carpenter 27 Jul '11

27 Jul '11

"adapter" is used as an array index in the adapters[] array so the off by one would make us read past the end. Signed-off-by: Dan Carpenter <error27(a)gmail.com> --- 1c073b67979 "ALSA: asihpi - Remove spurious adapter index check" reverted Dan Rosenburg's check that would have prevented the overflow here. Also it moved the initialization of "pa" down a couple lines so I'm concerned there may be a bogus derereference here when we check pa->type. I don't have the hardware, so I can't test this. diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c index 65fcf47..7ba7073 100644 --- a/sound/pci/asihpi/hpioctl.c +++ b/sound/pci/asihpi/hpioctl.c @@ -183,7 +183,7 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg) int wrflag = -1; u32 adapter = hm->h.adapter_index; - if ((adapter > HPI_MAX_ADAPTERS) || (!pa->type)) { + if ((adapter >= HPI_MAX_ADAPTERS) || (!pa->type)) { hpi_init_response(&hr->r0, HPI_OBJ_ADAPTER, HPI_ADAPTER_OPEN, HPI_ERROR_BAD_ADAPTER_NUMBER);

2 1

Re: [alsa-devel] [patch] ALSA: asihpi - off by one in asihpi_hpi_ioctl()
by Takashi Iwai 27 Jul '11

27 Jul '11

At Wed, 27 Jul 2011 09:02:13 -0400, Dan Rosenberg wrote: > > > > > Also it moved the initialization of "pa" down a couple lines so I'm > > > concerned there may be a bogus derereference here when we check > > > pa->type. I don't have the hardware, so I can't test this. > > > > > > > I agree. This code seems to make assumptions in more than one place > > that the adapters array is fully populated with non-NULL elements. At a > > glance, I can't see where such initialization occurs though. > > > > I hadn't read the updated code fully, so I missed what you meant. Yes, > this is definitely a NULL dereference, since "pa" is initialized to NULL > and not changed until after this dereference. NULL dereference was already fixed today in sound git tree by commit 767cd365b22820df07b962b49ce04b220b98e537. It'll be included in the pull request in the next days, maybe tomorrow. thanks, Takashi

1 0

Re: [alsa-devel] [PATCH 7/9] ASoC: Silence DEBUG_STRICT_USER_COPY_CHECKS=y warning
by Mark Brown 27 Jul '11

27 Jul '11

On Thu, May 12, 2011 at 04:50:10PM -0700, Stephen Boyd wrote: > Enabling DEBUG_STRICT_USER_COPY_CHECKS causes the following > warning: Applied, thanks.

1 0

[alsa-devel] [PATCH] ASoC: Ensure we delay long enough for WM8994 FLL to lock when starting
by Mark Brown 27 Jul '11

27 Jul '11

This delay is very conservative. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 38abdff..2eb4594 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1725,6 +1725,8 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_1 + reg_offset, WM8994_FLL1_ENA | WM8994_FLL1_FRAC, reg); + + msleep(5); } wm8994->fll[id].in = freq_in; -- 1.7.5.4

3 5

[alsa-devel] [PATCH] ASoC: dapm - Refactor widget IO functions in preparation for platform widgets.
by Liam Girdwood 27 Jul '11

27 Jul '11

Currently widget IO is tightly coupled to the CODEC drivers. Future platform DSP devices have mixer components that can alter power usage and hence require full DAPM support. This provides a generic widget IO operation wrapper in preparation for future patches that implement platform driver DAPM. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/soc-dapm.c | 55 ++++++++++++++++++++++++++++++++++++++++++++----- 1 files changed, 49 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 34106bc..7b7af9f 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -124,6 +124,49 @@ static inline struct snd_soc_dapm_widget *dapm_cnew_widget( return kmemdup(_widget, sizeof(*_widget), GFP_KERNEL); } +static int soc_widget_read(struct snd_soc_dapm_widget *w, int reg) +{ + if (w->codec) + return snd_soc_read(w->codec, reg); + return 0; +} + +static int soc_widget_write(struct snd_soc_dapm_widget *w,int reg, int val) +{ + if (w->codec) + return snd_soc_write(w->codec, reg, val); + return 0; +} + +int soc_widget_update_bits(struct snd_soc_dapm_widget *w, unsigned short reg, + unsigned int mask, unsigned int value) +{ + int change; + unsigned int old, new; + + old = soc_widget_read(w, reg); + new = (old & ~mask) | value; + change = old != new; + if (change) + soc_widget_write(w, reg, new); + + return change; +} + +int soc_widget_test_bits(struct snd_soc_dapm_widget *w, unsigned short reg, + unsigned int mask, unsigned int value) +{ + int change; + unsigned int old, new; + + old = soc_widget_read(w, reg); + new = (old & ~mask) | value; + change = old != new; + + return change; +} + + /** * snd_soc_dapm_set_bias_level - set the bias level for the system * @dapm: DAPM context @@ -181,7 +224,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; - val = snd_soc_read(w->codec, reg); + val = soc_widget_read(w, reg); val = (val >> shift) & mask; if ((invert && !val) || (!invert && val)) @@ -197,7 +240,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, for (bitmask = 1; bitmask < e->max; bitmask <<= 1) ; - val = snd_soc_read(w->codec, e->reg); + val = soc_widget_read(w, e->reg); item = (val >> e->shift_l) & (bitmask - 1); p->connect = 0; @@ -227,7 +270,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, w->kcontrol_news[i].private_value; int val, item; - val = snd_soc_read(w->codec, e->reg); + val = soc_widget_read(w, e->reg); val = (val >> e->shift_l) & e->mask; for (item = 0; item < e->max; item++) { if (val == e->values[item]) @@ -669,7 +712,7 @@ int dapm_reg_event(struct snd_soc_dapm_widget *w, else val = w->off_val; - snd_soc_update_bits(w->codec, -(w->reg + 1), + soc_widget_update_bits(w, -(w->reg + 1), w->mask << w->shift, val << w->shift); return 0; @@ -877,7 +920,7 @@ static void dapm_seq_run_coalesced(struct snd_soc_dapm_context *dapm, "pop test : Applying 0x%x/0x%x to %x in %dms\n", value, mask, reg, card->pop_time); pop_wait(card->pop_time); - snd_soc_update_bits(dapm->codec, reg, mask, value); + soc_widget_update_bits(w, reg, mask, value); } list_for_each_entry(w, pending, power_list) { @@ -1878,7 +1921,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) /* Read the initial power state from the device */ if (w->reg >= 0) { - val = snd_soc_read(w->codec, w->reg); + val = soc_widget_read(w, w->reg); val &= 1 << w->shift; if (w->invert) val = !val; -- 1.7.4.1

3 4

[alsa-devel] Event or signal for microphone connect/disconnect
by Putinei Ionut 27 Jul '11

27 Jul '11

Hi, I have 2 microphones on my laptop: 1 pluggable on the mic port and 1 on the surface . Is there any way to receive events/signal that the microphone on the mic port is connected/disconnected. I want to implement this feature in a recording program. Thank you Ionut Putinei

2 1

[alsa-devel] Rename Tegra support to Tegra20?
by Stephen Warren 27 Jul '11

27 Jul '11

Mark, Liam, The existing ASoC Tegra support is specifically for Tegra20. It turns out that Tegra30 has pretty different hardware and requires different drivers. A couple questions: * The new drivers I've written use "tegra30" in Kconfig variable names, filenames, symbol names, etc. Do you think it's a good idea to rename the existing Tegra20 driver to basically s/tegra/tegra20/ everywhere so everything is versioned, or do you think that's just unnecessary churn? (My motivation is kinda like the device tree compatible properties always being versioned) * I assume there's little point my posting Tegra30 ASoC drivers for inclusion in mainline, since there's no Tegra30 CPU support at all there yet? Thanks. -- nvpublic

2 1

[alsa-devel] [PATCH] ASoC: Add device tree binding for WM8731
by Mark Brown 27 Jul '11

27 Jul '11

Tested with the famous "hey, look! this compiles" test plan. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- Documentation/devicetree/bindings/sound/wm8731.txt | 18 ++++++++++++++++++ sound/soc/codecs/wm8731.c | 12 ++++++++++++ 2 files changed, 30 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/wm8731.txt diff --git a/Documentation/devicetree/bindings/sound/wm8731.txt b/Documentation/devicetree/bindings/sound/wm8731.txt new file mode 100644 index 0000000..15f7004 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8731.txt @@ -0,0 +1,18 @@ +WM8731 audio CODEC + +This device supports both I2C and SPI (configured with pin strapping +on the board). + +Required properties: + + - compatible : "wlf,wm8731" + + - reg : the I2C address of the device for I2C, the chip select + number for SPI. + +Example: + +codec: wm8731@1a { + compatible = "wlf,wm8731"; + reg = <0x1a>; +}; diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 76b4361..5ecc05a 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -22,6 +22,7 @@ #include <linux/platform_device.h> #include <linux/regulator/consumer.h> #include <linux/spi/spi.h> +#include <linux/of_device.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -607,6 +608,11 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8731 = { .num_dapm_routes = ARRAY_SIZE(wm8731_intercon), }; +static const struct of_device_id wm8731_of_match[] = { + { .compatible = "wlf,wm8731", }, + { } +}; + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8731_spi_probe(struct spi_device *spi) { @@ -634,10 +640,13 @@ static int __devexit wm8731_spi_remove(struct spi_device *spi) return 0; } +MODULE_DEVICE_TABLE(spi, wm8731_of_match); + static struct spi_driver wm8731_spi_driver = { .driver = { .name = "wm8731", .owner = THIS_MODULE, + .of_match_table = wm8731_of_match, }, .probe = wm8731_spi_probe, .remove = __devexit_p(wm8731_spi_remove), @@ -678,10 +687,13 @@ static const struct i2c_device_id wm8731_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, wm8731_i2c_id); +MODULE_DEVICE_TABLE(i2c, wm8731_of_match); + static struct i2c_driver wm8731_i2c_driver = { .driver = { .name = "wm8731", .owner = THIS_MODULE, + .of_match_table = wm8731_of_match, }, .probe = wm8731_i2c_probe, .remove = __devexit_p(wm8731_i2c_remove), -- 1.7.5.4

2 1

[alsa-devel] Supporting an unusual device
by Andrew Murray 27 Jul '11

27 Jul '11

Hello, I'm writing support for a slightly unusual device (which I intend to share later) and would like feedback on the best approach to take. The device is for a SOC which has two identical I2C connected codecs. Most codec drivers contain a snd_soc_dai structure - but as this contains instance specific data it would be necessary instead to produce (kmalloc) such structures for each instance (to ensure that both codecs can be treated and addressed independently). I.e. it seems that some codec drivers have been written with the assumption that there will only be one present on a system at any time. My view would be to implement such functionality (kmalloc'ing snd_soc_dai's) in one of the various platform specific directories as appropriate - e.g. atmel, blackfin, davinci, etc. Is this the best way forward? Are there any examples I've missed which do support multiple instances of the same codec? Also - is it recommended to expose these two codecs to the outside world as two snd_soc_card devices - or a single card with multiple dai_links? Thanks, Andrew Murray

2 2

[alsa-devel] [PATCH] ALSA: asihpi - bug fix pa use before init.
by linux＠audioscience.com 27 Jul '11

27 Jul '11

From: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> Fixes bug introduced by 1c073b67. Also declare pa local to block in which it is used. Signed-off-by: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> --- sound/pci/asihpi/hpioctl.c | 11 ++++------- 1 files changed, 4 insertions(+), 7 deletions(-) diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c index 65fcf47..e0cff0c 100644 --- a/sound/pci/asihpi/hpioctl.c +++ b/sound/pci/asihpi/hpioctl.c @@ -107,7 +107,6 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg) union hpi_response_buffer_v1 *hr; u16 res_max_size; u32 uncopied_bytes; - struct hpi_adapter *pa = NULL; int err = 0; if (cmd != HPI_IOCTL_LINUX) @@ -182,6 +181,7 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg) /* -1=no data 0=read from user mem, 1=write to user mem */ int wrflag = -1; u32 adapter = hm->h.adapter_index; + struct hpi_adapter *pa = &adapters[adapter]; if ((adapter > HPI_MAX_ADAPTERS) || (!pa->type)) { hpi_init_response(&hr->r0, HPI_OBJ_ADAPTER, @@ -197,9 +197,7 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg) goto out; } - pa = &adapters[adapter]; - - if (mutex_lock_interruptible(&adapters[adapter].mutex)) { + if (mutex_lock_interruptible(&pa->mutex)) { err = -EINTR; goto out; } @@ -235,8 +233,7 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg) "stream buffer size %d\n", size); - mutex_unlock(&adapters - [adapter].mutex); + mutex_unlock(&pa->mutex); err = -EINVAL; goto out; } @@ -277,7 +274,7 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg) uncopied_bytes, size); } - mutex_unlock(&adapters[adapter].mutex); + mutex_unlock(&pa->mutex); } /* on return response size must be set */ -- 1.7.0.4

2 1

Re: [alsa-devel] Bug#635362: Realtek ALC889: HDA Intel and kernel 3.0 gives choppy sound
by Julian Sikorski 27 Jul '11

27 Jul '11

W dniu 26.07.2011 23:14, Colomban Wendling pisze: > Hi, > > Following a bug report started in Debian BTS, in bug 635362 [1], I was > asked to post here after testing upstream kernels and reproducing the bug. > > So, here's the problem: my HDA audio chip [2] started to output very > choppy sound with Linux kernel 3.0, while it worked perfectly well in > previous versions -- latest two I used being 2.6.38 and 2.6.39 from > Debian. Please note that simply booting on 2.6.39 works as usual, so > it's IMHO definitely a driver regression. > > I tried Debian's 3.0 as well as mainline 3.0 and git5 snapshots, and all > 3 were affected by the problem -- though the kernel log isn't exactly > the same. > > I attach three dmesg outputs, always took after trying to play a sound > (and hearing it choppy, it happens every time): > > dmesg-3.0.0-mainline: dmesg when running 3.0 mainline > dmesg-3.0.0-git5: when running 3.0-git5 snapshot > dmesg-3.0.0-git5-2: same as dmesg-3.0.0-git5 but configured without > CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS (doesn't change anything) > > If you want to see dmesg from the 2.6.39 kernel (where sound works well) > or Debian's 3.0, see respectively [3] and [4]. > > > Thanks by advance, and if you need any further details or testing, > please let me know. > > Regards, > Colomban > > > [1] http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=635362 > [2] Realtek ALC889 on an MSI H55M-E33, see lspci in Debian bug [1] for > details > [3] > http://bugs.debian.org/cgi-bin/bugreport.cgi?msg=15;filename=dmesg-2.6.39-2… > [4] > http://bugs.debian.org/cgi-bin/bugreport.cgi?msg=15;filename=dmesg-3.0.0-1-… I am probably wrong, but did you try setting CONFIG_SND_HDA_BUFFER_SIZE to a larger value? Julian

2 2

[alsa-devel] [PATCH v3] ASoC: dapm - Add API call to query the widgets of valid DAPM paths.
by Liam Girdwood 26 Jul '11

26 Jul '11

In preparation for Dynamic PCM (AKA DSP) support. Add a DAPM API call to determine whether a DAPM audio path is valid between source and sink widgets and return the path widgets. This also takes into account all kcontrol mux and mixer settings in between the source and sink widgets to validate the audio path. This will be used by the Dynamic PCM core to determine the runtime DAI mappings between FE and BE DAIs in order to run PCM operations. This modifies the existing is_connected_input_ep() and is_connected_output_ep() by adding a new parameter to store any valid widgets in a list. We also add some extra debug to provide further insight into the DAPM path and widget discovery. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc-dapm.h | 6 ++ sound/soc/soc-dapm.c | 184 ++++++++++++++++++++++++++++++++++++++++++---- 2 files changed, 176 insertions(+), 14 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 5833f01..a56ead2 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -310,6 +310,7 @@ struct snd_soc_dapm_path; struct snd_soc_dapm_pin; struct snd_soc_dapm_route; struct snd_soc_dapm_context; +struct snd_soc_dapm_widget_list; int dapm_reg_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); @@ -375,6 +376,11 @@ int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm, int snd_soc_dapm_ignore_suspend(struct snd_soc_dapm_context *dapm, const char *pin); +/* dapm path query */ +int snd_soc_dapm_get_connected_widgets(struct snd_soc_dapm_context *dapm, + const char *stream_name, struct snd_soc_dapm_widget_list **list, + int stream); + /* dapm widget types */ enum snd_soc_dapm_type { snd_soc_dapm_input = 0, /* input pin */ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 80f97b2..120df5b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -605,11 +605,51 @@ static int snd_soc_dapm_suspend_check(struct snd_soc_dapm_widget *widget) } } +/* add widget to list if it's not already in the list */ +static int dapm_list_add_widget(struct snd_soc_dapm_widget_list **list, + struct snd_soc_dapm_widget *w) +{ + struct snd_soc_dapm_widget_list *wlist; + int wlistsize, wlistentries, i; + + if (*list == NULL) + return -EINVAL; + + wlist = *list; + + /* is this widget already in the list */ + for (i = 0; i < wlist->num_widgets; i++) { + if (wlist->widgets[i] == w) + return 0; + } + + /* allocate some new space */ + wlistentries = wlist->num_widgets + 1; + wlistsize = sizeof(struct snd_soc_dapm_widget_list) + + wlistentries * sizeof(struct snd_soc_dapm_widget *); + *list = krealloc(wlist, wlistsize, GFP_KERNEL); + if (*list == NULL) { + dev_err(w->dapm->dev, "can't allocate widget list for %s\n", + w->name); + return -ENOMEM; + } + wlist = *list; + + /* insert the widget */ + dev_dbg(w->dapm->dev, "added %s in widget list pos %d\n", + w->name, wlist->num_widgets); + + wlist->widgets[wlist->num_widgets] = w; + wlist->num_widgets++; + return 1; +} + /* * Recursively check for a completed path to an active or physically connected * output widget. Returns number of complete paths. */ -static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) +static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, + struct snd_soc_dapm_widget_list **list) { struct snd_soc_dapm_path *path; int con = 0; @@ -638,15 +678,35 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) } list_for_each_entry(path, &widget->sinks, list_source) { - if (path->weak) - continue; if (path->walked) continue; + dev_dbg(widget->dapm->dev," %c : %s -> %s -> %s : %c\n", + path->sink && path->connect ? '*' : ' ', + widget->name, path->name, path->sink->name, + path->weak ? 'w': ' '); + + if (path->weak) + continue; + if (path->sink && path->connect) { + path->walked = 1; - con += is_connected_output_ep(path->sink); + + /* do we need to add this widget to the list ? */ + if (list) { + int err; + + err = dapm_list_add_widget(list, path->sink); + if (err < 0) { + dev_err(widget->dapm->dev, "could not add widget %s\n", + widget->name); + return con; + } + } + + con += is_connected_output_ep(path->sink, list); } } @@ -657,7 +717,8 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) * Recursively check for a completed path to an active or physically connected * input widget. Returns number of complete paths. */ -static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) +static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, + struct snd_soc_dapm_widget_list **list) { struct snd_soc_dapm_path *path; int con = 0; @@ -691,21 +752,116 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) } list_for_each_entry(path, &widget->sources, list_sink) { - if (path->weak) - continue; if (path->walked) continue; + if (path->source) + dev_vdbg(widget->dapm->dev," %c : %s <- %s <- %s : %c\n", + path->source && path->connect ? '*' : ' ', + widget->name, path->name, path->source->name, + path->weak ? 'w': ' '); + + if (path->weak) + continue; + if (path->source && path->connect) { + path->walked = 1; - con += is_connected_input_ep(path->source); + + /* do we need to add this widget to the list ? */ + if (list) { + int err; + + err = dapm_list_add_widget(list, path->source); + if (err < 0) { + dev_err(widget->dapm->dev, "could not add widget %s\n", + widget->name); + return con; + } + } + + con += is_connected_input_ep(path->source, list); } } return con; } +static int dapm_get_playback_paths(struct snd_soc_dapm_context *dapm, + struct snd_soc_dapm_widget *root, + struct snd_soc_dapm_widget_list **list) +{ + int paths; + + dev_dbg(dapm->dev, "Playback: checking paths from %s\n",root->name); + paths = is_connected_output_ep(root, list); + dev_dbg(dapm->dev, "Playback: found %d paths from %s\n", paths, root->name); + + return paths; +} + +static int dapm_get_capture_paths(struct snd_soc_dapm_context *dapm, + struct snd_soc_dapm_widget *root, + struct snd_soc_dapm_widget_list **list) +{ + int paths; + + dev_dbg(dapm->dev, "Capture: checking paths from %s\n",root->name); + paths = is_connected_input_ep(root, list); + dev_dbg(dapm->dev, "Capture: found %d paths from %s\n", paths, root->name); + + return paths; +} + +/** + * snd_soc_dapm_get_connected_widgets - query audio path and it's widgets. + * @dapm: the dapm context. + * @stream_name: stream name. + * @list: list of active widgets for this stream. + * @stream: stream direction. + * + * Queries DAPM graph as to whether an valid audio stream path exists for + * the initial stream specified by name. This takes into account + * current mixer and mux kcontrol settings. Creates list of valid widgets. + * + * Returns the number of valid paths or negative error. + */ +int snd_soc_dapm_get_connected_widgets(struct snd_soc_dapm_context *dapm, + const char *stream_name, struct snd_soc_dapm_widget_list **list, + int stream) +{ + struct snd_soc_dapm_widget *w; + enum snd_soc_dapm_type type; + int paths; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + type = snd_soc_dapm_aif_in; + else + type = snd_soc_dapm_aif_out; + + /* get stream root widget AIF from stream string and direction */ + list_for_each_entry(w, &dapm->card->widgets, list) { + + if (w->id != type) + continue; + + if (!strcmp(w->sname, stream_name)) + goto found; + } + dev_err(dapm->dev, "root widget for %s not found\n", stream_name); + return 0; + +found: + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + paths = dapm_get_playback_paths(dapm, w, list); + else + paths = dapm_get_capture_paths(dapm, w, list); + + dapm_clear_walk(dapm); + return paths; +} + /* * Handler for generic register modifier widget. */ @@ -732,9 +888,9 @@ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w) { int in, out; - in = is_connected_input_ep(w); + in = is_connected_input_ep(w, NULL); dapm_clear_walk(w->dapm); - out = is_connected_output_ep(w); + out = is_connected_output_ep(w, NULL); dapm_clear_walk(w->dapm); return out != 0 && in != 0; } @@ -745,7 +901,7 @@ static int dapm_adc_check_power(struct snd_soc_dapm_widget *w) int in; if (w->active) { - in = is_connected_input_ep(w); + in = is_connected_input_ep(w, NULL); dapm_clear_walk(w->dapm); return in != 0; } else { @@ -759,7 +915,7 @@ static int dapm_dac_check_power(struct snd_soc_dapm_widget *w) int out; if (w->active) { - out = is_connected_output_ep(w); + out = is_connected_output_ep(w, NULL); dapm_clear_walk(w->dapm); return out != 0; } else { @@ -1315,9 +1471,9 @@ static ssize_t dapm_widget_power_read_file(struct file *file, if (!buf) return -ENOMEM; - in = is_connected_input_ep(w); + in = is_connected_input_ep(w, NULL); dapm_clear_walk(w->dapm); - out = is_connected_output_ep(w); + out = is_connected_output_ep(w, NULL); dapm_clear_walk(w->dapm); ret = snprintf(buf, PAGE_SIZE, "%s: %s in %d out %d", -- 1.7.4.1

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[alsa-devel] CX20590 - problems
by wouerner 26 Jul '11

26 Jul '11

hi, i'm from brazil, sorry bad english I have sony vaio vpceg15fb, internal mic - not work headphone jack - not work front speaker - work fine mic jack - work fine shell _> head -n 1 /proc/asound/card0/codec* ==> /proc/asound/card0/codec#0 <== Codec: Conexant CX20590 ==> /proc/asound/card0/codec#3 <== Codec: Intel CougarPoint HDMI not supported yet? i tried all tutorials in internet. thanks.

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