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

[alsa-devel] RFC: Jack-detection control elements on HD-audio
by Takashi Iwai 03 Jul '09

03 Jul '09

Hi, yet another thing I've worked recently on is a bit more generalized framework for jack-sense reporting for HD-audio. This supersedes the existing jack-sense reporting via the input layer. In addition, it gives the corresponding control elements. For example, you'll have "Jack HP-Out" (or a bit more verbose like "Jack HP-Out at Ext Rear") control via control API as well as other mixer controls. This can notify when a jack is plugged/unplugged. But, another big win is that you can read the current pin sense just by reading this element. I think this is a sort of feature Lennart wanted for PA. Lennart, does it make sense? The test codes are already in topic/hda-jack branch on sound-unstable GIT tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable-2.6.git topic/hda-jack Any comments appreciated. This is also intended for 2.6.32, not for 2.6.31 merge, of course :) thanks, Takashi

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[alsa-devel] Problems with snd-hda-intel on hp 6930p (2.6.30)
by Rafael J. Wysocki 02 Jul '09

02 Jul '09

Hi Takashi, I have a problem with snd-hda-intel from 2.6.30 on an hp EliteBook 6930p, which is that the sound is not audible, although everything appears to work just fine. Except maybe that when the box is rebooted and KDE (3.5) is started, the master control is "muted" and the led indicator next to the keyboard shows "no sound". Then, changing the "muted" status of the master control doesn't change the state of the led indicator and vice versa (the two things appear to be totally independent of each other). Can you please tell me what I can do to debug this issue? Best, Rafael

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[alsa-devel] [PATCH] Line In for Acer Inspire 6530G model
by Tony Vroon 01 Jul '09

01 Jul '09

The Line In connector is set up as PIN_IN by default, using VREF_HIZ. It is connected to both ADCs, so add it to both input selectors. Also add the ability to use the input mix (on a SoundBlaster one would call this "What You Hear"). Signed-off-by: Tony Vroon <tony(a)linx.net> diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3345331..67dd883 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1560,18 +1560,22 @@ static struct hda_input_mux alc888_2_capture_sources[2] = { static struct hda_input_mux alc888_acer_aspire_6530_sources[2] = { /* Interal mic only available on one ADC */ { - .num_items = 3, + .num_items = 5, .items = { { "Ext Mic", 0x0 }, + { "Line In", 0x2 }, { "CD", 0x4 }, + { "Input Mix", 0xa }, { "Int Mic", 0xb }, }, }, { - .num_items = 2, + .num_items = 4, .items = { { "Ext Mic", 0x0 }, + { "Line In", 0x2 }, { "CD", 0x4 }, + { "Input Mix", 0xa }, }, } }; @@ -8189,6 +8193,8 @@ static struct snd_kcontrol_new alc888_acer_aspire_6530_mixer[] = { HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("LFE Playback Volume", 0x0f, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("LFE Playback Switch", 0x0f, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),

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[alsa-devel] [PATCH] alsa/soc: add locking to mpc5200-psc-ac97 driver
by Grant Likely 01 Jul '09

01 Jul '09

From: Grant Likely <grant.likely(a)secretlab.ca> AC97 bus register read/write hooks need to provide locking, but the mpc5200-psc-ac97 driver does not. This patch adds a mutex around the register access routines. Signed-off-by: Grant Likely <grant.likely(a)secretlab.ca> --- sound/soc/fsl/mpc5200_dma.c | 1 + sound/soc/fsl/mpc5200_dma.h | 1 + sound/soc/fsl/mpc5200_psc_ac97.c | 17 ++++++++++++++++- 3 files changed, 18 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index efec33a..f0a2d40 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -456,6 +456,7 @@ int mpc5200_audio_dma_create(struct of_device *op) return -ENODEV; spin_lock_init(&psc_dma->lock); + mutex_init(&psc_dma->mutex); psc_dma->id = be32_to_cpu(*prop); psc_dma->irq = irq; psc_dma->psc_regs = regs; diff --git a/sound/soc/fsl/mpc5200_dma.h b/sound/soc/fsl/mpc5200_dma.h index 2000803..8d396bb 100644 --- a/sound/soc/fsl/mpc5200_dma.h +++ b/sound/soc/fsl/mpc5200_dma.h @@ -55,6 +55,7 @@ struct psc_dma { unsigned int irq; struct device *dev; spinlock_t lock; + struct mutex mutex; u32 sicr; uint sysclk; int imr; diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c index 794a247..7eb5499 100644 --- a/sound/soc/fsl/mpc5200_psc_ac97.c +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -34,13 +34,20 @@ static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg) int status; unsigned int val; + mutex_lock(&psc_dma->mutex); + /* Wait for command send status zero = ready */ status = spin_event_timeout(!(in_be16(&psc_dma->psc_regs->sr_csr.status) & MPC52xx_PSC_SR_CMDSEND), 100, 0); if (status == 0) { pr_err("timeout on ac97 bus (rdy)\n"); + mutex_unlock(&psc_dma->mutex); return -ENODEV; } + + /* Force clear the data valid bit */ + in_be32(&psc_dma->psc_regs->ac97_data); + /* Send the read */ out_be32(&psc_dma->psc_regs->ac97_cmd, (1<<31) | ((reg & 0x7f) << 24)); @@ -50,16 +57,19 @@ static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg) if (status == 0) { pr_err("timeout on ac97 read (val) %x\n", in_be16(&psc_dma->psc_regs->sr_csr.status)); + mutex_unlock(&psc_dma->mutex); return -ENODEV; } /* Get the data */ val = in_be32(&psc_dma->psc_regs->ac97_data); if (((val >> 24) & 0x7f) != reg) { pr_err("reg echo error on ac97 read\n"); + mutex_unlock(&psc_dma->mutex); return -ENODEV; } val = (val >> 8) & 0xffff; + mutex_unlock(&psc_dma->mutex); return (unsigned short) val; } @@ -68,16 +78,21 @@ static void psc_ac97_write(struct snd_ac97 *ac97, { int status; + mutex_lock(&psc_dma->mutex); + /* Wait for command status zero = ready */ status = spin_event_timeout(!(in_be16(&psc_dma->psc_regs->sr_csr.status) & MPC52xx_PSC_SR_CMDSEND), 100, 0); if (status == 0) { pr_err("timeout on ac97 bus (write)\n"); - return; + goto out; } /* Write data */ out_be32(&psc_dma->psc_regs->ac97_cmd, ((reg & 0x7f) << 24) | (val << 8)); + + out: + mutex_unlock(&psc_dma->mutex); } static void psc_ac97_warm_reset(struct snd_ac97 *ac97)

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[alsa-devel] [PATCH 1/4] ASoC: pxa-ssp: enhance I2S and add Left_J support
by Eric Miao 01 Jul '09

01 Jul '09

Make the pxa I2S configuration generic, add support for Left_J, add support for variable frame width like 32fs, 48fs, 64fs and 96fs Signed-off-by: Paul Shen <bshen9(a)marvell.com> Signed-off-by: Eric Miao <eric.miao(a)marvell.com> Cc: Daniel Mack <daniel(a)caiaq.de> --- arch/arm/mach-pxa/include/mach/regs-ssp.h | 14 +++--- sound/soc/pxa/pxa-ssp.c | 62 ++++++++++++++-------------- sound/soc/pxa/pxa-ssp.h | 9 ++++ 3 files changed, 47 insertions(+), 38 deletions(-) diff --git a/arch/arm/mach-pxa/include/mach/regs-ssp.h b/arch/arm/mach-pxa/include/mach/regs-ssp.h index 6a2ed35..27f0cd4 100644 --- a/arch/arm/mach-pxa/include/mach/regs-ssp.h +++ b/arch/arm/mach-pxa/include/mach/regs-ssp.h @@ -108,21 +108,21 @@ #define SSSR_TINT (1 << 19) /* Receiver Time-out Interrupt */ #define SSSR_PINT (1 << 18) /* Peripheral Trailing Byte Interrupt */ -#if defined(CONFIG_PXA3xx) -#define SSPSP_EDMYSTOP(x) ((x) << 28) /* Extended Dummy Stop */ -#define SSPSP_EDMYSTRT(x) ((x) << 26) /* Extended Dummy Start */ -#endif - #define SSPSP_FSRT (1 << 25) /* Frame Sync Relative Timing */ -#define SSPSP_DMYSTOP(x) ((x) << 23) /* Dummy Stop */ #define SSPSP_SFRMWDTH(x) ((x) << 16) /* Serial Frame Width */ #define SSPSP_SFRMDLY(x) ((x) << 9) /* Serial Frame Delay */ -#define SSPSP_DMYSTRT(x) ((x) << 7) /* Dummy Start */ #define SSPSP_STRTDLY(x) ((x) << 4) /* Start Delay */ #define SSPSP_ETDS (1 << 3) /* End of Transfer data State */ #define SSPSP_SFRMP (1 << 2) /* Serial Frame Polarity */ #define SSPSP_SCMODE(x) ((x) << 0) /* Serial Bit Rate Clock Mode */ +/* NOTE: PXA3xx extends the bit number of dummy start and stop, the macros + * below are compatible with PXA25x/27x as long as the parameter is within + * the correct limits, driver code has to take care of this. + */ +#define SSPSP_DMYSTRT(x) ((((x) & 3) << 7) | ((((x) >> 2) & 3) << 26)) +#define SSPSP_DMYSTOP(x) ((((x) & 3) << 23) | ((((x) >> 2) & 7) << 28)) + #define SSACD_SCDB (1 << 3) /* SSPSYSCLK Divider Bypass */ #define SSACD_ACPS(x) ((x) << 4) /* Audio clock PLL select */ #define SSACD_ACDS(x) ((x) << 0) /* Audio clock divider select */ diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 6fc7876..2831c16 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -463,7 +463,8 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - sscr0 |= SSCR0_PSP; + case SND_SOC_DAIFMT_LEFT_J: + sscr0 |= SSCR0_PSP | SSCR0_MOD; sscr1 |= SSCR1_RWOT | SSCR1_TRAIL; /* See hw_params() */ @@ -541,6 +542,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, int chn = params_channels(params); u32 sscr0; u32 sspsp; + int frame_width; int width = snd_pcm_format_physical_width(params_format(params)); int ttsa = ssp_read_reg(ssp, SSTSA) & 0xf; @@ -585,40 +587,38 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - sspsp = ssp_read_reg(ssp, SSPSP); - - if ((ssp_get_scr(ssp) == 4) && (width == 16)) { - /* This is a special case where the bitclk is 64fs - * and we're not dealing with 2*32 bits of audio - * samples. - * - * The SSP values used for that are all found out by - * trying and failing a lot; some of the registers - * needed for that mode are only available on PXA3xx. - */ + sspsp = ssp_read_reg(ssp, SSPSP); + frame_width = PXA_SSP_FRM_WIDTH(priv->dai_fmt); -#ifdef CONFIG_PXA3xx - if (!cpu_is_pxa3xx()) - return -EINVAL; - - sspsp |= SSPSP_SFRMWDTH(width * 2); - sspsp |= SSPSP_SFRMDLY(width * 4); - sspsp |= SSPSP_EDMYSTOP(3); - sspsp |= SSPSP_DMYSTOP(3); - sspsp |= SSPSP_DMYSTRT(1); -#else + if (frame_width < width * 2) return -EINVAL; -#endif - } else { - /* The frame width is the width the LRCLK is - * asserted for; the delay is expressed in - * half cycle units. We need the extra cycle - * because the data starts clocking out one BCLK - * after LRCLK changes polarity. + + if (frame_width == width * 2) + /* frame width is exactly double of data sample width, + * use FSRT instead */ - sspsp |= SSPSP_SFRMWDTH(width + 1); - sspsp |= SSPSP_SFRMDLY((width + 1) * 2); + sspsp |= SSPSP_FSRT | SSPSP_SFRMWDTH(width); + else { sspsp |= SSPSP_DMYSTRT(1); + sspsp |= SSPSP_DMYSTOP((frame_width / 2 - width - 1)); + sspsp |= SSPSP_SFRMWDTH(frame_width / 2); + } + + ssp_write_reg(ssp, SSPSP, sspsp); + break; + + case SND_SOC_DAIFMT_LEFT_J: + sspsp = ssp_read_reg(ssp, SSPSP); + frame_width = PXA_SSP_FRM_WIDTH(priv->dai_fmt); + + if (frame_width < width * 2) + return -EINVAL; + + if (frame_width == width * 2) + sspsp |= SSPSP_SFRMWDTH(width); + else { + sspsp |= SSPSP_DMYSTOP((frame_width / 2 - width)); + sspsp |= SSPSP_SFRMWDTH(frame_width / 2); } ssp_write_reg(ssp, SSPSP, sspsp); diff --git a/sound/soc/pxa/pxa-ssp.h b/sound/soc/pxa/pxa-ssp.h index 91deadd..fda51d0 100644 --- a/sound/soc/pxa/pxa-ssp.h +++ b/sound/soc/pxa/pxa-ssp.h @@ -40,6 +40,15 @@ #define PXA_SSP_CLK_SCDB_1 1 #define PXA_SSP_CLK_SCDB_8 2 +/* frame size definitions for I2S and Left_J formats - default is + * 32fs, other possibilities are 48fs, 64fs and 96fs + */ +#define PXA_SSP_FRM_32FS (0 << 16) +#define PXA_SSP_FRM_48FS (1 << 16) +#define PXA_SSP_FRM_64FS (2 << 16) +#define PXA_SSP_FRM_96FS (3 << 16) +#define PXA_SSP_FRM_WIDTH(x) (((((x) >> 16) & 0x3) + 2) << 4) + #define PXA_SSP_PLL_OUT 0 extern struct snd_soc_dai pxa_ssp_dai[4]; -- 1.6.0.4

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Re: [alsa-devel] PC speaker beeps in 2.6.30?
by Michael Tokarev 30 Jun '09

30 Jun '09

[Adding some more Cc's...] Am I the only concerned about this? My 2.6.30 is still silent w.r.t. old good PC speaker beeps, and I wasn't able to make it to produce any sound. Yes, as pointed out by others there is a control now, in alsa, and I can hear ugly and scary beeps from my stereo speakers (when they're turned on and when the control is un-muted). But that's.. not a solution/answer to the original question... ;) Thanks! Michael Tokarev wrote at Mon, 15 Jun 2009 18:41:42 +0400: > Anyone know what happened with PC speaker in 2.6.30? > Basically, it does not work anymore here, with the > config which is very similar to the one used for > previous 2.6.29 kernel (where PC speaker worked). > At least I don't see any relevant differences and > the (seemengly) relevant symbols -- which are > INPUT_PCSPKR and PCSPKR_PLATFORM -- are both set > to 'y' for both 2.6.29 and 2.6.30. > > Help? :) > > Thanks. > > /mjt -- 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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Re: [alsa-devel] Laptop speakers nonfunctional, jacks functional
by Takashi Iwai 30 Jun '09

30 Jun '09

At Mon, 29 Jun 2009 20:06:47 +0200, Ludwig Maes wrote: > > Tried the snapshot, > still the same, alsa-info.txt attached Hm, just through a quick look at the output, there is no obvious problem. It's only the problem from speaker, not about the headphone, right? If so, the problem is likely the missing amplifier setup, which is usually powered up via COEF verbs or GPIO verbs (ALC888/1200 have no EAPD verbs). Try to install hda-verb program and check the command below. # hda-verb /dev/snd/hwC0D0 0x20 SET_COEF_INDEX 7 # hda-verb /dev/snd/hwC0D0 0x20 GET_PROC_COEF 0 Which value do you get? What happens if you set # hda-verb /dev/snd/hwC0D0 0x20 SET_COEF_INDEX 7 # hda-verb /dev/snd/hwC0D0 0x20 SET_PROC_COEF 0x3060 or # hda-verb /dev/snd/hwC0D0 0x20 SET_COEF_INDEX 7 # hda-verb /dev/snd/hwC0D0 0x20 SET_PROC_COEF 0x3070 or # hda-verb /dev/snd/hwC0D0 0x20 SET_COEF_INDEX 7 # hda-verb /dev/snd/hwC0D0 0x20 SET_PROC_COEF 0x0830 or so? Takashi

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[alsa-devel] Understanding _snd_pcm_channel_area Struct (PCM Interface)!
by Guilherme 30 Jun '09

30 Jun '09

Hi all. I am a bit confused on what I get with this function. I finished my first project but I did not understand what exactly the _snd_pcm_channel_area at http://www.alsa-project.org/alsa-doc/alsa-lib/struct__snd__pcm__channel__ar… does. There are 3 fields in the documentation. addr, first and step. It seems that addr is the memory address of the channel samples... so far so good. The "step" is the distance between 2 sample (but I am not sure) and the least, "first", I could not figure out what does it means. Could I get some help from here? Thanks in advanced! Tks! ------------------- Guilherme Longo Dept. Eng. da Computação Unaerp Linux User - #484927 *Before Asking http://www.istf.com.br/?page=perguntas !- I'd rather die on my feet than live on my knees -!

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[alsa-devel] Merged patch_alc882() and patch_alc883()
by Takashi Iwai 30 Jun '09

30 Jun '09

Hi, I hacked around patch_realtek.c to clean it up a bit. As a result, now patch_alc883() gets merged to patch_alc882(). This cuts off lots of lines. I think this action is mostly OK as long as I tested with hda-emu. But, if you find any problems with this, please let me know. At best, get alsa-info.sh output before updating. It's always recommended to keep alsa-info.sh output once on a working system. This will make easier to track down the problem. thanks, Takashi

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[alsa-devel] External PCM hook SO is dlclosed after calling install, making it hard to have hooks in it
by Paul Fertser 30 Jun '09

30 Jun '09

Hi, I've tried to implement an external shared object to be able to execute arbitrary functions on PCM device opening and closing. After quite some source code reading i had an almost working but segfaulting example. This segfault is because snd_dlclose is called right after calling a hook install function and therefore i can't snd_pcm_hook_add functions from the same shared object. To me it looks like a bug but probably i just don't see how this functionality is intended to be used. Takashi, git-blame showed that it's you who stuffed this dlclose call that bothers me ;) TIA Background information: We're working on audio subsystem for OpenMoko Freerunner. It's an idea of Joerg Reisenweber to have virtual alsa devices for every reasonable use-case (like dedicated device for stereout, dedicated device for recording from gsm etc), for that we need a hook on every device opening and closing mostly to set up routing inside WM8753 codec. ctl_elems is not enough because we need a more complex locking scheme (with priorities, some kind of callbacks to the apps using particular devices etc) and other flexibility. -- Be free, use free (http://www.gnu.org/philosophy/free-sw.html) software! mailto:fercerpav@gmail.com

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