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

[alsa-devel] RFC: ASoC driver for S3C24XX Simtec boards
by Ben Dooks 26 Jun '09

26 Jun '09

I'd like some comments on the following driver which is currently being developed to get ASoC working on the TLV320AIC23 based Simtec boards. My question is: 1) I'm not seeing a control to enable/disable the 'Speaker Amp', do I need to change the controls or how this is exported? 2) There are two boards with an GPIO AMP which has two gain controls. This is a TPA2001D1 which provides a gain of 6,12,18 or 23.5dB depending on the GPIO settings. [ http://focus.ti.com/docs/prod/folders/print/tpa2001d1.html ] This driver has yet to be tested on all our boards. /* sound/soc/s3c24xx/s3c24xx_simtec.c * * Copyright 2009 Simtec Electronics * * 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. */ #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/platform_device.h> #include <linux/gpio.h> #include <linux/clk.h> #include <linux/i2c.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> #include <sound/soc-dapm.h> #include <plat/audio-simtec.h> #include "s3c24xx-pcm.h" #include "s3c24xx-i2s.h" #include "../codecs/tlv320aic23.h" /* supported machines: * * Machine Connections AMP * ------- ----------- --- * BAST MIC, HPOUT, LOUT, LIN TPA2001D1 (HPOUTL,R) (gain hardwired) * VR1000 HPOUT, LIN None * VR2000 LIN, LOUT, MIC, HP LM4871 (HPOUTL,R) * DePicture LIN, LOUT, MIC, HP LM4871 (HPOUTL,R) * Anubis LIN, LOUT, MIC, HP TPA2001D1 (HPOUTL,R) */ static struct s3c24xx_audio_simtec_pdata *pdata; static struct clk *xtal_clk; /** * simtec_spk_event - handle power management event for speaker * @w: The widget for this event * @k: The sound control the event is for. * @event; The event being processed. * * Handle power management events to deal with the state of the * GPIO AMP attached to some boards. */ static int simtec_spk_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { int value = SND_SOC_DAPM_EVENT_ON(event) ? 0 : 1; gpio_set_value(pdata->amp_gpio, value); return 0; } static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_LINE("Line In", NULL), SND_SOC_DAPM_LINE("Line Out", NULL), SND_SOC_DAPM_MIC("Mic Jack", NULL), SND_SOC_DAPM_PGA_E("Speaker Amp", SND_SOC_NOPM, 0, 0, NULL, 0, simtec_spk_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), }; static const struct snd_soc_dapm_route audio_map[] = { { "Headphone Jack", NULL, "LHPOUT"}, { "Headphone Jack", NULL, "RHPOUT"}, { "Line Out", NULL, "LLOUT" }, { "Line Out", NULL, "RLOUT" }, { "LLINEIN", NULL, "Line In"}, { "RLINEIN", NULL, "Line In"}, { "MICIN", NULL, "Mic Jack"}, }; static const struct snd_soc_dapm_route amp_map[] = { { "Speaker Amp", NULL, "RHPOUT" }, { "Speaker Amp", NULL, "LHPOUT" }, }; static int spk_gain; /** * speaker_gain_get - read the speaker gain setting. * @kcontrol: The control for the speaker gain. * @ucontrol: The value that needs to be updated. * * Read the value for the AMP gain control. */ static int speaker_gain_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { ucontrol->value.integer.value[0] = spk_gain; return 0; } /** * speaker_gain_put - set the speaker gain setting. * @kcontrol: The control for the speaker gain. * @ucontrol: The value that needs to be set. * * Set the value of the speaker gain from the specified * @ucontrol setting. */ static int speaker_gain_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { int value = ucontrol->value.integer.value[0]; spk_gain = value; gpio_set_value_cansleep(pdata->amp_gain[0], value & 1); gpio_set_value_cansleep(pdata->amp_gain[1], value >> 1); return 0; } static const struct snd_kcontrol_new amp_controls[] = { SOC_SINGLE_EXT("Speaker Gain", 0, 0, 3, 0, speaker_gain_get, speaker_gain_put), }; /** * simtec_tlv320aic23_init - initialise and add controls * @codec; The codec instance to attach to. * * Attach our controls and configure the necessary codec * mappings for our sound card instance. */ static int simtec_tlv320aic23_init(struct snd_soc_codec *codec) { snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets, ARRAY_SIZE(tlv320aic23_dapm_widgets)); snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); if (pdata->amp_gpio > 0) { printk(KERN_DEBUG "%s: adding amp routes\n", __func__); snd_soc_dapm_add_routes(codec, amp_map, ARRAY_SIZE(amp_map)); snd_soc_dapm_enable_pin(codec, "Speaker Amp"); } if (pdata->amp_gain[0] > 0) { printk(KERN_DEBUG "%s: adding amp controls\n", __func__); snd_soc_add_controls(codec, amp_controls, ARRAY_SIZE(amp_controls)); } snd_soc_dapm_enable_pin(codec, "Headphone Jack"); snd_soc_dapm_enable_pin(codec, "Line In"); snd_soc_dapm_enable_pin(codec, "Line Out"); snd_soc_dapm_enable_pin(codec, "Mic Jack"); snd_soc_dapm_sync(codec); return 0; } #define CODEC_CLOCK 12000000 /** * simtec_hw_params - update hardware parameters * @substream: The audio substream instance. * @params: The parameters requested. * * Update the codec data routing and configuration settings * from the supplied data. */ static int simtec_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; int ret; /* Set the CODEC as the bus clock master, I2S */ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret) { printk(KERN_ERR "%s: failed set cpu dai format\n", __func__); return ret; } /* Set the CODEC as the bus clock master */ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret) { printk(KERN_ERR "%s: failed set codec dai format\n", __func__); return ret; } ret = snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN); if (ret) { printk(KERN_ERR "%s: failed setting codec sysclk\n", __func__); return ret; } if (pdata->use_mpllin) { ret = snd_soc_dai_set_sysclk(cpu_dai, S3C24XX_CLKSRC_MPLL, 0, SND_SOC_CLOCK_OUT); if (ret) { printk(KERN_ERR "%s: failed to set MPLLin as clksrc\n", __func__); return ret; } } if (pdata->output_cdclk) { int cdclk_scale; cdclk_scale = clk_get_rate(xtal_clk) / CODEC_CLOCK; cdclk_scale--; ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, cdclk_scale); } return 0; } static int simtec_hw_startup(struct snd_pcm_substream *substream) { /* call any board supplied startup code, this currently only * covers the bast/vr1000 which have a CPLD in the way of the * LRCLK */ if (pdata->startup) pdata->startup(); return 0; } static struct snd_soc_ops simtec_ops = { .startup = simtec_hw_startup, .hw_params = simtec_hw_params, }; static struct snd_soc_dai_link simtec_dai_aic23 = { .name = "tlv320aic23", .stream_name = "TLV320AIC23", .cpu_dai = &s3c24xx_i2s_dai, .codec_dai = &tlv320aic23_dai, .init = simtec_tlv320aic23_init, .ops = &simtec_ops, }; /* simtec audio machine driver */ static struct snd_soc_card snd_soc_machine_simtec_aic23 = { .name = "Simtec", .platform = &s3c24xx_soc_platform, .dai_link = &simtec_dai_aic23, .num_links = 1, }; /* simtec audio subsystem */ static struct snd_soc_device simtec_snd_devdata_aic23 = { .card = &snd_soc_machine_simtec_aic23, .codec_dev = &soc_codec_dev_tlv320aic23, }; /** * attach_gpio_amp - get and configure the necessary gpios * @dev: The device we're probing. * @pd: The platform data supplied by the board. * * If there is a GPIO based amplifier attached to the board, claim * the necessary GPIO lines for it, and set default values. */ static int attach_gpio_amp(struct device *dev, struct s3c24xx_audio_simtec_pdata *pd) { int ret; /* attach gpio amp gain (if any) */ if (pdata->amp_gain[0] > 0) { ret = gpio_request(pd->amp_gain[0], "gpio-amp-gain"); if (ret) { dev_err(dev, "cannot get amp gpio gain0\n"); return ret; } ret = gpio_request(pd->amp_gain[1], "gpio-amp-gain"); if (ret) { dev_err(dev, "cannot get amp gpio gain0\n"); gpio_free(pdata->amp_gain[0]); return ret; } gpio_direction_output(pd->amp_gain[0], 0); gpio_direction_output(pd->amp_gain[1], 0); } /* note, curently we assume GPA0 isn't valid amp */ if (pdata->amp_gpio > 0) { ret = gpio_request(pd->amp_gpio, "gpio-amp"); if (ret) { dev_err(dev, "cannot get amp gpio %d (%d)\n", pd->amp_gpio, ret); goto err_amp; } /* start with amp off */ gpio_direction_output(pd->amp_gpio, 1); } return 0; err_amp: if (pd->amp_gain[0] > 0) { gpio_free(pd->amp_gain[0]); gpio_free(pd->amp_gain[1]); } return ret; } static void detach_gpio_amp(struct s3c24xx_audio_simtec_pdata *pd) { if (pd->amp_gain[0] > 0) { gpio_free(pd->amp_gain[0]); gpio_free(pd->amp_gain[1]); } if (pd->amp_gpio > 0) gpio_free(pd->amp_gpio); } static int __devinit simtec_audio_probe(struct platform_device *pdev) { struct platform_device *snd_dev; int ret; dev_info(&pdev->dev, "probe called\n"); pdata = pdev->dev.platform_data; if (!pdata) { dev_err(&pdev->dev, "no platform data supplied\n"); return -EINVAL; } xtal_clk = clk_get(&pdev->dev, "xtal"); if (IS_ERR(xtal_clk)) { dev_err(&pdev->dev, "could not get clkout0\n"); return -EINVAL; } dev_info(&pdev->dev, "xtal rate is %ld\n", clk_get_rate(xtal_clk)); ret = attach_gpio_amp(&pdev->dev, pdata); if (ret) goto err_clk; snd_dev = platform_device_alloc("soc-audio", -1); if (!snd_dev) { dev_err(&pdev->dev, "failed to alloc soc-audio devicec\n"); ret = -ENOMEM; goto err_gpio; } platform_set_drvdata(snd_dev, &simtec_snd_devdata_aic23); simtec_snd_devdata_aic23.dev = &snd_dev->dev; ret = platform_device_add(snd_dev); if (ret) { dev_err(&pdev->dev, "failed to add soc-audio dev\n"); platform_device_put(snd_dev); goto err_clk; } platform_set_drvdata(pdev, snd_dev); return 0; err_gpio: detach_gpio_amp(pdata); err_clk: clk_put(xtal_clk); return ret; } static int __devexit simtec_audio_remove(struct platform_device *pdev) { struct platform_device *snd_dev = platform_get_drvdata(pdev); platform_device_unregister(snd_dev); detach_gpio_amp(pdata); clk_put(xtal_clk); return 0; } static struct platform_driver simtec_audio_platdrv = { .driver = { .owner = THIS_MODULE, .name = "s3c24xx-tlv320aic23", }, .probe = simtec_audio_probe, .remove = __devexit_p(simtec_audio_remove), }; MODULE_ALIAS("platform:s3c24xx-tlv320aic23"); static int __init simtec_init(void) { return platform_driver_register(&simtec_audio_platdrv); } static void __exit simtec_exit(void) { platform_driver_unregister(&simtec_audio_platdrv); } module_init(simtec_init); module_exit(simtec_exit); MODULE_AUTHOR("Ben Dooks <ben(a)simtec.co.uk>"); MODULE_DESCRIPTION("ALSA SoC Simtec Audio support"); MODULE_LICENSE("GPL");

3 3

Re: [alsa-devel] Regarding Sound on Acer Aspire 6935G
by Emilio López 26 Jun '09

26 Jun '09

Well, it seems my last email was too heavy with the 4 files attached. I have uploaded them to drop.io, here is the link: http://drop.io/alsaonaceraspire6930 Thanks! 2009/5/23 Emilio López <buhitoescolar(a)gmail.com> > Here you are, I ran alsa-info.sh in the old alsa (with model=acer) and > saved the output of the script when side was enabled and disabled, with > volumes 0 and 100. Hope you can work out the problem with this, do not > hesitate to ask for more debug info if you need. > > Thanks for all! > > Emilio > > PS: On my previous email (cited below) I wrote "Mic sensing" incorrectly, I > wanted to write "headphones sensing". > > 2009/5/23 Takashi Iwai <tiwai(a)suse.de> > > At Sat, 23 May 2009 12:19:12 -0300, >> Emilio López wrote: >> > >> > I reinstalled 1.0.20 today and almost everything started working when I >> set >> > model=acer-aspire-4930g. Mic sensing works, my mic now records too. But >> my >> > bass does not, and I cannot control it with "Side" anymore. >> > >> > Any idea on how to fix it? The audio setup for this laptop will be fully >> > working when I get the bass to play once again. >> >> Could you check with the older version and what "Side" control changes >> in hd-audio codec registers? You can just get alsa-info.sh output >> before and after changing the "Side" control value. >> >> >> thanks, >> >> Takashi >> > >

5 40

Re: [alsa-devel] Proposal for more reliable audio DMA.
by Mark Brown 26 Jun '09

26 Jun '09

On 24 Jun 2009, at 17:07, Jon Smirl <jonsmirl(a)gmail.com> wrote: > Trying to directly expose the ring buffer to the app seems like a good > > way to avoid a copy, but it isn't achieving that. pulse is not > decoding audio straight into the ring buffer, it decodes first and > then copies into the buffer. Pulse is using the timers to estimate the > destination for the copy. Move this copy down into the drivers, the > drivers know the correct destination for the copy. Pulse isn't just doing straight playback, a large part of what it's there for is to do software mixing. When you have multiple sources active pulse is going to be forced to do the copy as part of the mixing process so putting that bit of the buffer management in kernel won't help in the way you think it does. Part of what's going on here is that the kernel code is trying to give userspace access to the data for as long as possible.

4 7

[alsa-devel] [PATCH] ASoC: txx9aclc: dynamically allocate dmaengine devname
by Atsushi Nemoto 25 Jun '09

25 Jun '09

Use kasprintf to allocate temporary devname string instead of a fixed size string. This fixes "FIXME" introduced on removal of BUS_ID_SIZE. Signed-off-by: Atsushi Nemoto <anemo(a)mba.ocn.ne.jp> --- sound/soc/txx9/txx9aclc.c | 10 ++++++---- 1 files changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c index 938a58a..efed64b 100644 --- a/sound/soc/txx9/txx9aclc.c +++ b/sound/soc/txx9/txx9aclc.c @@ -297,15 +297,17 @@ static int txx9aclc_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, static bool filter(struct dma_chan *chan, void *param) { struct txx9aclc_dmadata *dmadata = param; - char devname[20 + 2]; /* FIXME: old BUS_ID_SIZE + 2 */ + char *devname; + bool found = false; - snprintf(devname, sizeof(devname), "%s.%d", dmadata->dma_res->name, + devname = kasprintf(GFP_KERNEL, "%s.%d", dmadata->dma_res->name, (int)dmadata->dma_res->start); if (strcmp(dev_name(chan->device->dev), devname) == 0) { chan->private = &dmadata->dma_slave; - return true; + found = true; } - return false; + kfree(devname); + return found; } static int txx9aclc_dma_init(struct txx9aclc_soc_device *dev, -- 1.5.6.5

2 1

[alsa-devel] Reenable support for Auzentech X-Fi Prelude in snd-ctxfi module
by Kamil Kaminski 25 Jun '09

25 Jun '09

Hi guys, I noticed that latest alsa-driver-unstable does not detect my card as if PCI IDs were discarded or purposely disabled to only work on Creative cards, the earlier snapshots used to work fine. This is what I get: ctatc: Card not recognised ctxfi: Something wrong!!! SB-XFi: probe of 0000:06:01.0 failed with error -2 This is the card: 06:01.0 Multimedia audio controller: Creative Labs SB X-Fi Subsystem: Auzentech, Inc. Device 0034 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64 (1000ns min, 1250ns max) Interrupt: pin A routed to IRQ 22 Region 0: I/O ports at 8f00 [size=32] Region 1: Memory at dfc00000 (64-bit, non-prefetchable) [size=2M] Region 3: Memory at d8000000 (64-bit, non-prefetchable) [size=64M] Capabilities: [40] Power Management version 2 Flags: PMEClk- DSI+ D1+ D2+ AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [50] MSI: Mask- 64bit+ Count=1/1 Enable- Address: 0000000000000000 Data: 0000 Kernel driver in use: CTALSA Kernel modules: ctxfi I'm currently using v1.0 ctxfi alsa driver from Creative, It would be nice to have Auzentech X-Fi Prelude working in 2.6.31 kernel ;) Thanks.

4 3

[alsa-devel] Proposal for more reliable audio DMA.
by Jon Smirl 25 Jun '09

25 Jun '09

This algorithm is fairly similar to what currently exists in ALSA with a few modifications. I'm trying to come up with guaranteed way to play glitch free audio. Does this algorithm work or could it be modified to work? The main change is to track a high resolution timer as the means of estimating where in the buffer to insert new, low latency data. This tracking is done in a timer tick interrupt. Jiffies are too coarse and not all hardware allows you to ask the current DMA position. The other change is to keep all of the buffers filled with silence if there isn't any pending data. (not doing this caused problem in my mpc5200 AC97 driver) Three buffers are used as a way to deterministically bound the DMA pointer without needing interrupts. --- Use three chained buffers. Buffer size is samples/tick (or maybe 1.5 samples/tick) Initialize buffers to silence. Set end of buffer three to automatically terminate DMA Set FIFO to minimum bus allows Fill buffer one with samples available, start playing. On timer tick call into driver.. If buffer one has finished playing move it to third position. New last buffer is set to automatically stop DMA. Return swap status to ALSA If swapped, fill buffer three with silence or pending data It is important to fill this buffer with silence if there is no pending data The size of the buffers needs to be large enough to cover worst case timer tick latency. Buffer two has to be large enough to ensure that the tick routine will notice buffer one has been played before buffer three starts. If two buffers have been completed when the tick runs, let the third buffer finish without swapping. Since you don't know where you are in the third buffer, it is unsafe to swap at this point. Playing the last buffer causes the end of buffer interrupt to happen. Callback into ALSA to alert it of the underrun and need to restart DMA. If this callback happens, make the buffers larger. Long term observation of buffer completion status in tick handler will allow accurate computation of samples/HPET unit. Record high accuracy timer (HPET) on tick. New low latency data that arrives can be inserted into the buffers dynamically. Use the high resolution timer source to estimate where to place it. Minimum FIFO ensures low latency play. Design a minimum power mode. Allows a huge FIFO to be loaded (like 128KB). Call into driver to reset to minimum latency, small FIFO mode. Advantages: 1) If ALSA goes away, hardware will stop on it's own with no noise. 2) No need to know the current position of DMA hardware. 3) Both low and high latency modes. 4) Audio interrupts only generated as error condition 5) Behavior is deterministic. Nothing is left to guess work. -- Jon Smirl jonsmirl(a)gmail.com

4 9

Re: [alsa-devel] ASoC updates for 2.6.31
by Mark Brown 25 Jun '09

25 Jun '09

The following changes since commit 28d0325ce6e0a52f53d8af687e6427fee59004d3: Linus Torvalds (1): Linux 2.6.31-rc1 are available in the git repository at: ssh://master.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31 Peter Ujfalusi (1): MAINTAINERS: Add entry for twl4030 series soc codec driver MAINTAINERS | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-)

2 1

[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 25 Jun '09

25 Jun '09

The following changes since commit 28d0325ce6e0a52f53d8af687e6427fee59004d3: Linus Torvalds (1): Linux 2.6.31-rc1 are available in the git repository at: ssh://master.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Barry Song (1): ASoC: Blackfin: convert internal names from bf52x to bf5xx Daniel Ribeiro (2): ASoC: Allow 32 bit registers for DAPM ASoC: remove duplicated code on pxa-ssp.c Lopez Cruz, Misael (4): ASoC: Zoom2: Add machine driver for Zoom2 board ASoC: TWL4030: Fix voice interface clock masters ASoC: TWL4030: Add AVADC Clock Priority ASoC: TWL4030: Correct bypass event for voice sidetone Mark Brown (21): ASoC: Sort DAPM power sequences while building lists ASoC: Factor out DAPM sequence execution ASoC: Coalesce register writes for DAPM sequences ASoC: Sort specialised mixers and muxes together ASoC: Coalesce power updates for DAPM widgets with events ASoC: Coalesce power updates for PGAs ASoC: Make DAPM power sequence lists local variables Merge branch 'dapm' into for-2.6.32 ASoC: Add WM8961 driver ASoC: Automatically control WM8903 sloping stopband filter ASoC: Automatically manage WM8900 sloping stopband filter ASoC: Automatically manage WM8350 sloping stopband filter ASoC: Add stub suspend and resume calls for ASoC subdevices ASoC: Add suspend and resume callbacks to Wolfson CODEC drivers ASoC: Regulator support for WM8580 ASoC: Fix shadowed variables in twl4030 ASoC: Staticise put_twl4030_opmode_enum_double() Merge branch 'wm8961' into for-2.6.32 ASoC: Add a shutdown callback ASoC: Add core suspend and resume callbacks to WM8961 Merge branch 'for-2.6.31' into for-2.6.32 Peter Ujfalusi (1): MAINTAINERS: Add entry for twl4030 series soc codec driver Philipp Zabel (1): ASoC: UDA1380: refactor device registration MAINTAINERS | 7 + include/sound/soc-dapm.h | 1 + include/sound/soc.h | 11 +- include/sound/uda1380.h | 22 + sound/soc/blackfin/bf5xx-ad73311.c | 16 +- sound/soc/blackfin/bf5xx-ssm2602.c | 16 +- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/twl4030.c | 38 +- sound/soc/codecs/uda1380.c | 313 +++++---- sound/soc/codecs/uda1380.h | 8 - sound/soc/codecs/wm8350.c | 40 +- sound/soc/codecs/wm8400.c | 17 + sound/soc/codecs/wm8580.c | 55 ++- sound/soc/codecs/wm8731.c | 39 ++ sound/soc/codecs/wm8753.c | 35 + sound/soc/codecs/wm8900.c | 30 +- sound/soc/codecs/wm8903.c | 27 +- sound/soc/codecs/wm8940.c | 17 + sound/soc/codecs/wm8960.c | 17 + sound/soc/codecs/wm8961.c | 1326 ++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8961.h | 866 +++++++++++++++++++++++ sound/soc/codecs/wm8988.c | 34 + sound/soc/codecs/wm9081.c | 17 + sound/soc/omap/Kconfig | 7 + sound/soc/omap/Makefile | 2 + sound/soc/omap/sdp3430.c | 2 +- sound/soc/omap/zoom2.c | 301 ++++++++ sound/soc/pxa/magician.c | 54 +- sound/soc/pxa/pxa-ssp.c | 50 +- sound/soc/soc-core.c | 83 ++- sound/soc/soc-dapm.c | 347 +++++++--- 32 files changed, 3452 insertions(+), 352 deletions(-) create mode 100644 include/sound/uda1380.h create mode 100644 sound/soc/codecs/wm8961.c create mode 100644 sound/soc/codecs/wm8961.h create mode 100644 sound/soc/omap/zoom2.c

1 0

[alsa-devel] [PATCH 0/1] Add maintainer for the twl4030 codec
by Peter Ujfalusi 25 Jun '09

25 Jun '09

Hello, Mark Brown kindly asked me to send the following patch to add myself to the MAINTAINERS file for the twl4030 series soc codec driver. I have thought about it and here is the patch. Thank you Mark for the suggestion, I would not have sent this patch otherwise. Thanks for Steve Sakoman to write the initial twl4030 codec driver. I'll try my best with the maintainer responsibility! --- Peter Ujfalusi (1): MAINTAINERS: Add entry for twl4030 series soc codec driver MAINTAINERS | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-)

3 3

[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 25 Jun '09

25 Jun '09

The following changes since commit 28d0325ce6e0a52f53d8af687e6427fee59004d3: Linus Torvalds (1): Linux 2.6.31-rc1 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Barry Song (1): ASoC: Blackfin: convert internal names from bf52x to bf5xx Daniel Ribeiro (2): ASoC: Allow 32 bit registers for DAPM ASoC: remove duplicated code on pxa-ssp.c Lopez Cruz, Misael (4): ASoC: Zoom2: Add machine driver for Zoom2 board ASoC: TWL4030: Fix voice interface clock masters ASoC: TWL4030: Add AVADC Clock Priority ASoC: TWL4030: Correct bypass event for voice sidetone Mark Brown (20): ASoC: Sort DAPM power sequences while building lists ASoC: Factor out DAPM sequence execution ASoC: Coalesce register writes for DAPM sequences ASoC: Sort specialised mixers and muxes together ASoC: Coalesce power updates for DAPM widgets with events ASoC: Coalesce power updates for PGAs ASoC: Make DAPM power sequence lists local variables Merge branch 'dapm' into for-2.6.32 ASoC: Add WM8961 driver ASoC: Automatically control WM8903 sloping stopband filter ASoC: Automatically manage WM8900 sloping stopband filter ASoC: Automatically manage WM8350 sloping stopband filter ASoC: Add stub suspend and resume calls for ASoC subdevices ASoC: Add suspend and resume callbacks to Wolfson CODEC drivers ASoC: Regulator support for WM8580 ASoC: Fix shadowed variables in twl4030 ASoC: Staticise put_twl4030_opmode_enum_double() Merge branch 'wm8961' into for-2.6.32 ASoC: Add a shutdown callback Merge branch 'topic/asoc' of git://git.kernel.org/.../tiwai/sound-2.6 into for-2.6.32 Philipp Zabel (1): ASoC: UDA1380: refactor device registration include/sound/soc-dapm.h | 1 + include/sound/soc.h | 11 +- include/sound/uda1380.h | 22 + sound/soc/blackfin/bf5xx-ad73311.c | 16 +- sound/soc/blackfin/bf5xx-ssm2602.c | 16 +- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/twl4030.c | 38 +- sound/soc/codecs/uda1380.c | 313 +++++---- sound/soc/codecs/uda1380.h | 8 - sound/soc/codecs/wm8350.c | 40 +- sound/soc/codecs/wm8400.c | 17 + sound/soc/codecs/wm8580.c | 55 ++- sound/soc/codecs/wm8731.c | 39 ++ sound/soc/codecs/wm8753.c | 35 + sound/soc/codecs/wm8900.c | 30 +- sound/soc/codecs/wm8903.c | 27 +- sound/soc/codecs/wm8940.c | 17 + sound/soc/codecs/wm8960.c | 17 + sound/soc/codecs/wm8961.c | 1309 ++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8961.h | 866 ++++++++++++++++++++++++ sound/soc/codecs/wm8988.c | 34 + sound/soc/codecs/wm9081.c | 17 + sound/soc/omap/Kconfig | 7 + sound/soc/omap/Makefile | 2 + sound/soc/omap/sdp3430.c | 2 +- sound/soc/omap/zoom2.c | 301 +++++++++ sound/soc/pxa/magician.c | 54 +- sound/soc/pxa/pxa-ssp.c | 50 +- sound/soc/soc-core.c | 83 ++- sound/soc/soc-dapm.c | 347 ++++++++--- 31 files changed, 3428 insertions(+), 352 deletions(-) create mode 100644 include/sound/uda1380.h create mode 100644 sound/soc/codecs/wm8961.c create mode 100644 sound/soc/codecs/wm8961.h create mode 100644 sound/soc/omap/zoom2.c

2 5

[alsa-devel] Support for MSM7 in ASoC
by Didier "Ptitjes" 25 Jun '09

25 Jun '09

Hello all, I'm working together with FreeSmartphone.org and SHR. We would like to support the MSM7-based phones. We already have the modem working, but audio is a show-stopper for us, as actual implementation of the audio subsystem in Android is in binary form, and does not us Alsa !!! Can someone look at the specs for the MSM7200A specs and tell whether it could be difficult to support its audio subsystem ? Thanks for your help, Didier.

3 3

Re: [alsa-devel] Reg: Implementation of GSM digital audio path
by Peter Ujfalusi 25 Jun '09

25 Jun '09

On Thursday 25 June 2009 14:11:04 ext shariff wrote: > Hi Peter, > > Thanks for quick response. I understood some what from your mail. > > I will explain, how my hardware is connected. > > > USB GSM chip -> McBSP3 > > McBSP2 -> TWL4030 > > Here, playback functionality will not be there in the GSM path. I > need to take audio data from GSM. and redirect this to TWL4030 codec. I see. Sorry I have confused things a bit, In this case you need to do what Mark has suggested: dummy dai for the codec, connect it to McBSP3 as it were a normal codec. Than you will have two PCM on the sound card: 0.0: TWL digital audio 0.1: Your codec arecord -Dplughw:0.1 | aplay -Dplughw:0.0 or arecord -Dhw:0.1 | aplay -Dhw:0.0 > > The thing is, How and where to implement this GSM path without > playback? as I am new to this ASoC framework. > > What I understood from ur mail is. Audio is being redirected from > the application space i.e arecord will record and store in buffer and this > buffer is > > being played using aplay to other hardware (twl4030). > > Can we handle the same thing in kernel space ( i.e in driver itself). You anyway have to copy from one buffer to another to be in the safe side (in theory you could use the same buffer for recording and playback, but it should be avoided). So you have to copy the samples from the capture buffer to the playback buffer. It really does not matter if it is done inside of the kernel or in user space. This kind of 'routing' clearly belongs to the user space. > > Regards, > Shariff -- Péter

2 1

[alsa-devel] Reg: Implementation of GSM digital audio path
by shariff 25 Jun '09

25 Jun '09

Hi all, Iam working on OMAP3 based custom board. On our board, GSM digital audio path is connected to McBSP3 and TWL4030 digital audio path is connected to McBSP2. I want to implement a module that transfers GSM audio data and store audio data into some buffer and redirect it to TWL4030 audio codec through McBSP2. I question is how can I implement this in ASoC? I studied ASoC framework, but I am unable to understand where to implement my stuff. Regards, Shariff

3 4

[alsa-devel] [PATCH 3/3] ASoC: TWL4030: Fix voice interface clock masters
by Lopez Cruz, Misael 25 Jun '09

25 Jun '09

Voice interface of twl4030 codec supports: CBM_CFM and CBS_CFS. It doesn't support CBS_CFM. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl4030.c | 2 +- sound/soc/omap/sdp3430.c | 2 +- sound/soc/omap/zoom2.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index fd5b30f..a6fb1d6 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1969,7 +1969,7 @@ static int twl4030_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFM: + case SND_SOC_DAIFMT_CBM_CFM: format &= ~(TWL4030_VIF_SLAVE_EN); break; case SND_SOC_DAIFMT_CBS_CFS: diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index b719e5d..c51594d 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -96,7 +96,7 @@ static int sdp3430_hw_voice_params(struct snd_pcm_substream *substream, ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF | - SND_SOC_DAIFMT_CBS_CFM); + SND_SOC_DAIFMT_CBM_CFM); if (ret) { printk(KERN_ERR "can't set codec DAI configuration\n"); return ret; diff --git a/sound/soc/omap/zoom2.c b/sound/soc/omap/zoom2.c index 17ae5c2..c84bc1c 100644 --- a/sound/soc/omap/zoom2.c +++ b/sound/soc/omap/zoom2.c @@ -89,7 +89,7 @@ static int zoom2_hw_voice_params(struct snd_pcm_substream *substream, ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF | - SND_SOC_DAIFMT_CBS_CFM); + SND_SOC_DAIFMT_CBM_CFM); if (ret) { printk(KERN_ERR "can't set codec DAI configuration\n"); return ret; -- 1.5.4.3

5 7

[alsa-devel] [PATCH] alsa: lx6464es - configure ethersound io channels
by Tim Blechmann 25 Jun '09

25 Jun '09

as long as the io channel number is not set by the driver, the card is not visible from the ethersound network Signed-off-by: Tim Blechmann <tim(a)klingt.org> --- sound/pci/lx6464es/lx6464es.c | 7 +++---- 1 files changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c index ccf1b38..c92fba5 100644 --- a/sound/pci/lx6464es/lx6464es.c +++ b/sound/pci/lx6464es/lx6464es.c @@ -654,13 +654,12 @@ static int __devinit lx_init_ethersound_config(struct lx6464es *chip) int i; u32 orig_conf_es = lx_dsp_reg_read(chip, eReg_CONFES); - u32 default_conf_es = (64 << IOCR_OUTPUTS_OFFSET) | + /* configure 64 io channels */ + u32 conf_es = (orig_conf_es & CONFES_READ_PART_MASK) | (64 << IOCR_INPUTS_OFFSET) | + (64 << IOCR_OUTPUTS_OFFSET) | (FREQ_RATIO_SINGLE_MODE << FREQ_RATIO_OFFSET); - u32 conf_es = (orig_conf_es & CONFES_READ_PART_MASK) - | (default_conf_es & CONFES_WRITE_PART_MASK); - snd_printdd("->lx_init_ethersound\n"); chip->freq_ratio = FREQ_RATIO_SINGLE_MODE; -- 1.6.2.1

2 1

[alsa-devel] Patch for suspend/resume in ICE1724 for Audiotrak Prodigy HD2
by Igor Chernyshev 25 Jun '09

25 Jun '09

Hi Alsa Developers! I've built a small HTPC and had to add suspend/resume support in ice1724 driver. There seem to be 3 existing bugs related to that: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4413 https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3748 https://bugtrack.alsa-project.org/alsa-bug/view.php?id=2314 Due to hardware (un)availability, I only enabled the fix for Audiotrak Prodigy HD2 card, which is installed in my HTPC. However, most of my code should be reusable in the future on other ice1724-based cards as well (as long as people add card-specific peices of code). The fix is currently based on ALSA 1.0.20 and works on my MythBuntu 9.04 HTPC (using 2.6.28-11 kernel). The following files have been modified: alsa-kernel/pci/ice1712/ice1712.h alsa-kernel/pci/ice1712/ice1724.c alsa-kernel/pci/ice1712/prodigy_hifi.c Now, I'm a complete newbie in Linux development and not sure I can safely rebuild latest Linux Kernel under MythBuntu, etc. Are there any resources for people like me to read? I want to provide the patch without destroying my computer a hunder times, but also make it more convenient and safer for ALSA maintainers. Thanks, Igor

2 9

[alsa-devel] [PATCH 19/19] sound: Use PCI_VDEVICE
by Joe Perches 25 Jun '09

25 Jun '09

Signed-off-by: Joe Perches <joe(a)perches.com> --- sound/oss/kahlua.c | 2 +- sound/pci/atiixp.c | 8 +++--- sound/pci/atiixp_modem.c | 4 +- sound/pci/au88x0/au8810.c | 3 +- sound/pci/au88x0/au8820.c | 3 +- sound/pci/au88x0/au8830.c | 3 +- sound/pci/cmipci.c | 10 ++++---- sound/pci/cs4281.c | 2 +- sound/pci/cs46xx/cs46xx.c | 6 ++-- sound/pci/ens1370.c | 6 ++-- sound/pci/es1938.c | 2 +- sound/pci/ice1712/ice1712.c | 2 +- sound/pci/ice1712/ice1724.c | 2 +- sound/pci/intel8x0.c | 46 +++++++++++++++++++++--------------------- sound/pci/intel8x0m.c | 34 +++++++++++++++--------------- sound/pci/mixart/mixart.c | 2 +- sound/pci/nm256/nm256.c | 6 ++-- sound/pci/rme32.c | 9 ++----- sound/pci/rme96.c | 12 +++------- sound/pci/sonicvibes.c | 2 +- sound/pci/via82xx.c | 4 +- sound/pci/via82xx_modem.c | 2 +- sound/pci/ymfpci/ymfpci.c | 12 +++++----- 23 files changed, 86 insertions(+), 96 deletions(-) diff --git a/sound/oss/kahlua.c b/sound/oss/kahlua.c index c180598..89466b0 100644 --- a/sound/oss/kahlua.c +++ b/sound/oss/kahlua.c @@ -199,7 +199,7 @@ MODULE_LICENSE("GPL"); */ static struct pci_device_id id_tbl[] = { - { PCI_VENDOR_ID_CYRIX, PCI_DEVICE_ID_CYRIX_5530_AUDIO, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, + { PCI_VDEVICE(CYRIX, PCI_DEVICE_ID_CYRIX_5530_AUDIO), 0 }, { } }; diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index 71515dd..d6752df 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -287,10 +287,10 @@ struct atiixp { /* */ static struct pci_device_id snd_atiixp_ids[] = { - { 0x1002, 0x4341, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* SB200 */ - { 0x1002, 0x4361, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* SB300 */ - { 0x1002, 0x4370, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* SB400 */ - { 0x1002, 0x4382, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* SB600 */ + { PCI_VDEVICE(ATI, 0x4341), 0 }, /* SB200 */ + { PCI_VDEVICE(ATI, 0x4361), 0 }, /* SB300 */ + { PCI_VDEVICE(ATI, 0x4370), 0 }, /* SB400 */ + { PCI_VDEVICE(ATI, 0x4382), 0 }, /* SB600 */ { 0, } }; diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index c3136cc..e7e147b 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -262,8 +262,8 @@ struct atiixp_modem { /* */ static struct pci_device_id snd_atiixp_ids[] = { - { 0x1002, 0x434d, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* SB200 */ - { 0x1002, 0x4378, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* SB400 */ + { PCI_VDEVICE(ATI, 0x434d), 0 }, /* SB200 */ + { PCI_VDEVICE(ATI, 0x4378), 0 }, /* SB400 */ { 0, } }; diff --git a/sound/pci/au88x0/au8810.c b/sound/pci/au88x0/au8810.c index fce22c7..c0e8c6b 100644 --- a/sound/pci/au88x0/au8810.c +++ b/sound/pci/au88x0/au8810.c @@ -1,8 +1,7 @@ #include "au8810.h" #include "au88x0.h" static struct pci_device_id snd_vortex_ids[] = { - {PCI_VENDOR_ID_AUREAL, PCI_DEVICE_ID_AUREAL_ADVANTAGE, - PCI_ANY_ID, PCI_ANY_ID, 0, 0, 1,}, + {PCI_VDEVICE(AUREAL, PCI_DEVICE_ID_AUREAL_ADVANTAGE), 1,}, {0,} }; diff --git a/sound/pci/au88x0/au8820.c b/sound/pci/au88x0/au8820.c index d1fbcce..a652733 100644 --- a/sound/pci/au88x0/au8820.c +++ b/sound/pci/au88x0/au8820.c @@ -1,8 +1,7 @@ #include "au8820.h" #include "au88x0.h" static struct pci_device_id snd_vortex_ids[] = { - {PCI_VENDOR_ID_AUREAL, PCI_DEVICE_ID_AUREAL_VORTEX_1, - PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0,}, + {PCI_VDEVICE(AUREAL, PCI_DEVICE_ID_AUREAL_VORTEX_1), 0,}, {0,} }; diff --git a/sound/pci/au88x0/au8830.c b/sound/pci/au88x0/au8830.c index d4f2717..6c702ad 100644 --- a/sound/pci/au88x0/au8830.c +++ b/sound/pci/au88x0/au8830.c @@ -1,8 +1,7 @@ #include "au8830.h" #include "au88x0.h" static struct pci_device_id snd_vortex_ids[] = { - {PCI_VENDOR_ID_AUREAL, PCI_DEVICE_ID_AUREAL_VORTEX_2, - PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0,}, + {PCI_VDEVICE(AUREAL, PCI_DEVICE_ID_AUREAL_VORTEX_2), 0,}, {0,} }; diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index 449fe02..ddcd4a9 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -2797,11 +2797,11 @@ static inline void snd_cmipci_proc_init(struct cmipci *cm) {} static struct pci_device_id snd_cmipci_ids[] = { - {PCI_VENDOR_ID_CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8338A, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0}, - {PCI_VENDOR_ID_CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8338B, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0}, - {PCI_VENDOR_ID_CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8738, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0}, - {PCI_VENDOR_ID_CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8738B, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0}, - {PCI_VENDOR_ID_AL, PCI_DEVICE_ID_CMEDIA_CM8738, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0}, + {PCI_VDEVICE(CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8338A), 0}, + {PCI_VDEVICE(CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8338B), 0}, + {PCI_VDEVICE(CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8738), 0}, + {PCI_VDEVICE(CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8738B), 0}, + {PCI_VDEVICE(AL, PCI_DEVICE_ID_CMEDIA_CM8738), 0}, {0,}, }; diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c index f6286f8..e2e0359 100644 --- a/sound/pci/cs4281.c +++ b/sound/pci/cs4281.c @@ -495,7 +495,7 @@ struct cs4281 { static irqreturn_t snd_cs4281_interrupt(int irq, void *dev_id); static struct pci_device_id snd_cs4281_ids[] = { - { 0x1013, 0x6005, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* CS4281 */ + { PCI_VDEVICE(CIRRUS, 0x6005), 0, }, /* CS4281 */ { 0, } }; diff --git a/sound/pci/cs46xx/cs46xx.c b/sound/pci/cs46xx/cs46xx.c index c9b3e3d..004a59a 100644 --- a/sound/pci/cs46xx/cs46xx.c +++ b/sound/pci/cs46xx/cs46xx.c @@ -65,9 +65,9 @@ module_param_array(mmap_valid, bool, NULL, 0444); MODULE_PARM_DESC(mmap_valid, "Support OSS mmap."); static struct pci_device_id snd_cs46xx_ids[] = { - { 0x1013, 0x6001, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* CS4280 */ - { 0x1013, 0x6003, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* CS4612 */ - { 0x1013, 0x6004, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* CS4615 */ + { PCI_VDEVICE(CIRRUS, 0x6001), 0, }, /* CS4280 */ + { PCI_VDEVICE(CIRRUS, 0x6003), 0, }, /* CS4612 */ + { PCI_VDEVICE(CIRRUS, 0x6004), 0, }, /* CS4615 */ { 0, } }; diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 18f4d1e..d589bbc 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -445,11 +445,11 @@ static irqreturn_t snd_audiopci_interrupt(int irq, void *dev_id); static struct pci_device_id snd_audiopci_ids[] = { #ifdef CHIP1370 - { 0x1274, 0x5000, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* ES1370 */ + { PCI_VDEVICE(ENSONIQ, 0x5000), 0, }, /* ES1370 */ #endif #ifdef CHIP1371 - { 0x1274, 0x1371, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* ES1371 */ - { 0x1274, 0x5880, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* ES1373 - CT5880 */ + { PCI_VDEVICE(ENSONIQ, 0x1371), 0, }, /* ES1371 */ + { PCI_VDEVICE(ENSONIQ, 0x5880), 0, }, /* ES1373 - CT5880 */ { 0x1102, 0x8938, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* Ectiva EV1938 */ #endif { 0, } diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c index fbd2ac0..b47232c 100644 --- a/sound/pci/es1938.c +++ b/sound/pci/es1938.c @@ -244,7 +244,7 @@ struct es1938 { static irqreturn_t snd_es1938_interrupt(int irq, void *dev_id); static struct pci_device_id snd_es1938_ids[] = { - { 0x125d, 0x1969, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* Solo-1 */ + { PCI_VDEVICE(ESS, 0x1969), 0, }, /* Solo-1 */ { 0, } }; diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 0d0cdbd..cecf1ff 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -107,7 +107,7 @@ MODULE_PARM_DESC(dxr_enable, "Enable DXR support for Terratec DMX6FIRE."); static const struct pci_device_id snd_ice1712_ids[] = { - { PCI_VENDOR_ID_ICE, PCI_DEVICE_ID_ICE_1712, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* ICE1712 */ + { PCI_VDEVICE(ICE, PCI_DEVICE_ID_ICE_1712), 0 }, /* ICE1712 */ { 0, } }; diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 36ade77..cc84a83 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -93,7 +93,7 @@ MODULE_PARM_DESC(model, "Use the given board model."); /* Both VT1720 and VT1724 have the same PCI IDs */ static const struct pci_device_id snd_vt1724_ids[] = { - { PCI_VENDOR_ID_ICE, PCI_DEVICE_ID_VT1724, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, + { PCI_VDEVICE(ICE, PCI_DEVICE_ID_VT1724), 0 }, { 0, } }; diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 8aa5687..171ada5 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -421,29 +421,29 @@ struct intel8x0 { }; static struct pci_device_id snd_intel8x0_ids[] = { - { 0x8086, 0x2415, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* 82801AA */ - { 0x8086, 0x2425, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* 82901AB */ - { 0x8086, 0x2445, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* 82801BA */ - { 0x8086, 0x2485, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* ICH3 */ - { 0x8086, 0x24c5, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL_ICH4 }, /* ICH4 */ - { 0x8086, 0x24d5, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL_ICH4 }, /* ICH5 */ - { 0x8086, 0x25a6, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL_ICH4 }, /* ESB */ - { 0x8086, 0x266e, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL_ICH4 }, /* ICH6 */ - { 0x8086, 0x27de, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL_ICH4 }, /* ICH7 */ - { 0x8086, 0x2698, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL_ICH4 }, /* ESB2 */ - { 0x8086, 0x7195, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* 440MX */ - { 0x1039, 0x7012, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_SIS }, /* SI7012 */ - { 0x10de, 0x01b1, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_NFORCE }, /* NFORCE */ - { 0x10de, 0x003a, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_NFORCE }, /* MCP04 */ - { 0x10de, 0x006a, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_NFORCE }, /* NFORCE2 */ - { 0x10de, 0x0059, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_NFORCE }, /* CK804 */ - { 0x10de, 0x008a, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_NFORCE }, /* CK8 */ - { 0x10de, 0x00da, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_NFORCE }, /* NFORCE3 */ - { 0x10de, 0x00ea, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_NFORCE }, /* CK8S */ - { 0x10de, 0x026b, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_NFORCE }, /* MCP51 */ - { 0x1022, 0x746d, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* AMD8111 */ - { 0x1022, 0x7445, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* AMD768 */ - { 0x10b9, 0x5455, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_ALI }, /* Ali5455 */ + { PCI_VDEVICE(INTEL, 0x2415), DEVICE_INTEL }, /* 82801AA */ + { PCI_VDEVICE(INTEL, 0x2425), DEVICE_INTEL }, /* 82901AB */ + { PCI_VDEVICE(INTEL, 0x2445), DEVICE_INTEL }, /* 82801BA */ + { PCI_VDEVICE(INTEL, 0x2485), DEVICE_INTEL }, /* ICH3 */ + { PCI_VDEVICE(INTEL, 0x24c5), DEVICE_INTEL_ICH4 }, /* ICH4 */ + { PCI_VDEVICE(INTEL, 0x24d5), DEVICE_INTEL_ICH4 }, /* ICH5 */ + { PCI_VDEVICE(INTEL, 0x25a6), DEVICE_INTEL_ICH4 }, /* ESB */ + { PCI_VDEVICE(INTEL, 0x266e), DEVICE_INTEL_ICH4 }, /* ICH6 */ + { PCI_VDEVICE(INTEL, 0x27de), DEVICE_INTEL_ICH4 }, /* ICH7 */ + { PCI_VDEVICE(INTEL, 0x2698), DEVICE_INTEL_ICH4 }, /* ESB2 */ + { PCI_VDEVICE(INTEL, 0x7195), DEVICE_INTEL }, /* 440MX */ + { PCI_VDEVICE(SI, 0x7012), DEVICE_SIS }, /* SI7012 */ + { PCI_VDEVICE(NVIDIA, 0x01b1), DEVICE_NFORCE }, /* NFORCE */ + { PCI_VDEVICE(NVIDIA, 0x003a), DEVICE_NFORCE }, /* MCP04 */ + { PCI_VDEVICE(NVIDIA, 0x006a), DEVICE_NFORCE }, /* NFORCE2 */ + { PCI_VDEVICE(NVIDIA, 0x0059), DEVICE_NFORCE }, /* CK804 */ + { PCI_VDEVICE(NVIDIA, 0x008a), DEVICE_NFORCE }, /* CK8 */ + { PCI_VDEVICE(NVIDIA, 0x00da), DEVICE_NFORCE }, /* NFORCE3 */ + { PCI_VDEVICE(NVIDIA, 0x00ea), DEVICE_NFORCE }, /* CK8S */ + { PCI_VDEVICE(NVIDIA, 0x026b), DEVICE_NFORCE }, /* MCP51 */ + { PCI_VDEVICE(AMD, 0x746d), DEVICE_INTEL }, /* AMD8111 */ + { PCI_VDEVICE(AMD, 0x7445), DEVICE_INTEL }, /* AMD768 */ + { PCI_VDEVICE(AL, 0x5455), DEVICE_ALI }, /* Ali5455 */ { 0, } }; diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index 6ec0fc5..9e7d12e 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -220,24 +220,24 @@ struct intel8x0m { }; static struct pci_device_id snd_intel8x0m_ids[] = { - { 0x8086, 0x2416, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* 82801AA */ - { 0x8086, 0x2426, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* 82901AB */ - { 0x8086, 0x2446, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* 82801BA */ - { 0x8086, 0x2486, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* ICH3 */ - { 0x8086, 0x24c6, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* ICH4 */ - { 0x8086, 0x24d6, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* ICH5 */ - { 0x8086, 0x266d, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* ICH6 */ - { 0x8086, 0x27dd, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* ICH7 */ - { 0x8086, 0x7196, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* 440MX */ - { 0x1022, 0x7446, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* AMD768 */ - { 0x1039, 0x7013, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_SIS }, /* SI7013 */ - { 0x10de, 0x01c1, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_NFORCE }, /* NFORCE */ - { 0x10de, 0x0069, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_NFORCE }, /* NFORCE2 */ - { 0x10de, 0x0089, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_NFORCE }, /* NFORCE2s */ - { 0x10de, 0x00d9, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_NFORCE }, /* NFORCE3 */ + { PCI_VDEVICE(INTEL, 0x2416), DEVICE_INTEL }, /* 82801AA */ + { PCI_VDEVICE(INTEL, 0x2426), DEVICE_INTEL }, /* 82901AB */ + { PCI_VDEVICE(INTEL, 0x2446), DEVICE_INTEL }, /* 82801BA */ + { PCI_VDEVICE(INTEL, 0x2486), DEVICE_INTEL }, /* ICH3 */ + { PCI_VDEVICE(INTEL, 0x24c6), DEVICE_INTEL }, /* ICH4 */ + { PCI_VDEVICE(INTEL, 0x24d6), DEVICE_INTEL }, /* ICH5 */ + { PCI_VDEVICE(INTEL, 0x266d), DEVICE_INTEL }, /* ICH6 */ + { PCI_VDEVICE(INTEL, 0x27dd), DEVICE_INTEL }, /* ICH7 */ + { PCI_VDEVICE(INTEL, 0x7196), DEVICE_INTEL }, /* 440MX */ + { PCI_VDEVICE(AMD, 0x7446), DEVICE_INTEL }, /* AMD768 */ + { PCI_VDEVICE(SI, 0x7013), DEVICE_SIS }, /* SI7013 */ + { PCI_VDEVICE(NVIDIA, 0x01c1), DEVICE_NFORCE }, /* NFORCE */ + { PCI_VDEVICE(NVIDIA, 0x0069), DEVICE_NFORCE }, /* NFORCE2 */ + { PCI_VDEVICE(NVIDIA, 0x0089), DEVICE_NFORCE }, /* NFORCE2s */ + { PCI_VDEVICE(NVIDIA, 0x00d9), DEVICE_NFORCE }, /* NFORCE3 */ #if 0 - { 0x1022, 0x746d, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* AMD8111 */ - { 0x10b9, 0x5455, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_ALI }, /* Ali5455 */ + { PCI_VDEVICE(AMD, 0x746d), DEVICE_INTEL }, /* AMD8111 */ + { PCI_VDEVICE(AL, 0x5455), DEVICE_ALI }, /* Ali5455 */ #endif { 0, } }; diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index 82bc5b9..a83d196 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -61,7 +61,7 @@ MODULE_PARM_DESC(enable, "Enable Digigram " CARD_NAME " soundcard."); */ static struct pci_device_id snd_mixart_ids[] = { - { 0x1057, 0x0003, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* MC8240 */ + { PCI_VDEVICE(MOTOROLA, 0x0003), 0, }, /* MC8240 */ { 0, } }; diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c index 522a040..97a0731 100644 --- a/sound/pci/nm256/nm256.c +++ b/sound/pci/nm256/nm256.c @@ -263,9 +263,9 @@ struct nm256 { * PCI ids */ static struct pci_device_id snd_nm256_ids[] = { - {PCI_VENDOR_ID_NEOMAGIC, PCI_DEVICE_ID_NEOMAGIC_NM256AV_AUDIO, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0}, - {PCI_VENDOR_ID_NEOMAGIC, PCI_DEVICE_ID_NEOMAGIC_NM256ZX_AUDIO, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0}, - {PCI_VENDOR_ID_NEOMAGIC, PCI_DEVICE_ID_NEOMAGIC_NM256XL_PLUS_AUDIO, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0}, + {PCI_VDEVICE(NEOMAGIC, PCI_DEVICE_ID_NEOMAGIC_NM256AV_AUDIO), 0}, + {PCI_VDEVICE(NEOMAGIC, PCI_DEVICE_ID_NEOMAGIC_NM256ZX_AUDIO), 0}, + {PCI_VDEVICE(NEOMAGIC, PCI_DEVICE_ID_NEOMAGIC_NM256XL_PLUS_AUDIO), 0}, {0,}, }; diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c index d7b966e..f977dba 100644 --- a/sound/pci/rme32.c +++ b/sound/pci/rme32.c @@ -227,12 +227,9 @@ struct rme32 { }; static struct pci_device_id snd_rme32_ids[] = { - {PCI_VENDOR_ID_XILINX_RME, PCI_DEVICE_ID_RME_DIGI32, - PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0,}, - {PCI_VENDOR_ID_XILINX_RME, PCI_DEVICE_ID_RME_DIGI32_8, - PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0,}, - {PCI_VENDOR_ID_XILINX_RME, PCI_DEVICE_ID_RME_DIGI32_PRO, - PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0,}, + {PCI_VDEVICE(XILINX_RME, PCI_DEVICE_ID_RME_DIGI32), 0,}, + {PCI_VDEVICE(XILINX_RME, PCI_DEVICE_ID_RME_DIGI32_8), 0,}, + {PCI_VDEVICE(XILINX_RME, PCI_DEVICE_ID_RME_DIGI32_PRO), 0,}, {0,} }; diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index 55fb1c1..c01296c 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -232,14 +232,10 @@ struct rme96 { }; static struct pci_device_id snd_rme96_ids[] = { - { PCI_VENDOR_ID_XILINX, PCI_DEVICE_ID_RME_DIGI96, - PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, - { PCI_VENDOR_ID_XILINX, PCI_DEVICE_ID_RME_DIGI96_8, - PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, - { PCI_VENDOR_ID_XILINX, PCI_DEVICE_ID_RME_DIGI96_8_PRO, - PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, - { PCI_VENDOR_ID_XILINX, PCI_DEVICE_ID_RME_DIGI96_8_PAD_OR_PST, - PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, + { PCI_VDEVICE(XILINX, PCI_DEVICE_ID_RME_DIGI96), 0, }, + { PCI_VDEVICE(XILINX, PCI_DEVICE_ID_RME_DIGI96_8), 0, }, + { PCI_VDEVICE(XILINX, PCI_DEVICE_ID_RME_DIGI96_8_PRO), 0, }, + { PCI_VDEVICE(XILINX, PCI_DEVICE_ID_RME_DIGI96_8_PAD_OR_PST), 0, }, { 0, } }; diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c index 7dc60ad..1f6406c 100644 --- a/sound/pci/sonicvibes.c +++ b/sound/pci/sonicvibes.c @@ -243,7 +243,7 @@ struct sonicvibes { }; static struct pci_device_id snd_sonic_ids[] = { - { 0x5333, 0xca00, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, + { PCI_VDEVICE(S3, 0xca00), 0, }, { 0, } }; diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 949fcaf..acfa476 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -402,9 +402,9 @@ struct via82xx { static struct pci_device_id snd_via82xx_ids[] = { /* 0x1106, 0x3058 */ - { PCI_VENDOR_ID_VIA, PCI_DEVICE_ID_VIA_82C686_5, PCI_ANY_ID, PCI_ANY_ID, 0, 0, TYPE_CARD_VIA686, }, /* 686A */ + { PCI_VDEVICE(VIA, PCI_DEVICE_ID_VIA_82C686_5), TYPE_CARD_VIA686, }, /* 686A */ /* 0x1106, 0x3059 */ - { PCI_VENDOR_ID_VIA, PCI_DEVICE_ID_VIA_8233_5, PCI_ANY_ID, PCI_ANY_ID, 0, 0, TYPE_CARD_VIA8233, }, /* VT8233 */ + { PCI_VDEVICE(VIA, PCI_DEVICE_ID_VIA_8233_5), TYPE_CARD_VIA8233, }, /* VT8233 */ { 0, } }; diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c index 0d54e35..47eb615 100644 --- a/sound/pci/via82xx_modem.c +++ b/sound/pci/via82xx_modem.c @@ -261,7 +261,7 @@ struct via82xx_modem { }; static struct pci_device_id snd_via82xx_modem_ids[] = { - { 0x1106, 0x3068, PCI_ANY_ID, PCI_ANY_ID, 0, 0, TYPE_CARD_VIA82XX_MODEM, }, + { PCI_VDEVICE(VIA, 0x3068), TYPE_CARD_VIA82XX_MODEM, }, { 0, } }; diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c index 4af6666..822b60c 100644 --- a/sound/pci/ymfpci/ymfpci.c +++ b/sound/pci/ymfpci/ymfpci.c @@ -67,12 +67,12 @@ module_param_array(rear_switch, bool, NULL, 0444); MODULE_PARM_DESC(rear_switch, "Enable shared rear/line-in switch"); static struct pci_device_id snd_ymfpci_ids[] = { - { 0x1073, 0x0004, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* YMF724 */ - { 0x1073, 0x000d, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* YMF724F */ - { 0x1073, 0x000a, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* YMF740 */ - { 0x1073, 0x000c, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* YMF740C */ - { 0x1073, 0x0010, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* YMF744 */ - { 0x1073, 0x0012, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* YMF754 */ + { PCI_VDEVICE(YAMAHA, 0x0004), 0, }, /* YMF724 */ + { PCI_VDEVICE(YAMAHA, 0x000d), 0, }, /* YMF724F */ + { PCI_VDEVICE(YAMAHA, 0x000a), 0, }, /* YMF740 */ + { PCI_VDEVICE(YAMAHA, 0x000c), 0, }, /* YMF740C */ + { PCI_VDEVICE(YAMAHA, 0x0010), 0, }, /* YMF744 */ + { PCI_VDEVICE(YAMAHA, 0x0012), 0, }, /* YMF754 */ { 0, } }; -- 1.6.3.1.10.g659a0.dirty

2 5

[alsa-devel] [PATCH] ASoC: Zoom2: Update twl4030_setup_data parameters
by Jorge Eduardo Candelaria 25 Jun '09

25 Jun '09

Add EXT_MUTE support in Zoom2 machine driver. Member ext_mute_gpio is passed to TWL4030 as an external mute control for headset power on/off sequence. This is necessary to further reduce pop noise problem. Gpio 153 is connected to signal EXT_MUTE in Zoom2 board. In addition, change ramp delay value to 3 (218/161/109 ms). With previous ramp delay value, pop noise was louder, and with any longer value beep tone can be observed. Signed-off-by: Jorge Eduardo Candelaria <x0107209(a)ti.com> --- sound/soc/omap/zoom2.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/omap/zoom2.c b/sound/soc/omap/zoom2.c index 3de6d2b..8941f45 100644 --- a/sound/soc/omap/zoom2.c +++ b/sound/soc/omap/zoom2.c @@ -236,8 +236,9 @@ static struct snd_soc_card snd_soc_zoom2 = { /* twl4030 setup */ static struct twl4030_setup_data twl4030_setup = { - .ramp_delay_value = 2, /* 81 ms */ + .ramp_delay_value = 3, /* 161 ms */ .sysclk = 26000, + .ext_mute_gpio = 153, }; /* Audio subsystem */ -- 1.6.0.4

1 0

[alsa-devel] Should my Creative X-fi card be loading the snd-hda-intel module
by Susan Cragin 24 Jun '09

24 Jun '09

Shouldn't it be loading snd-hda-codec-ca0110 instead? I am using the 6/24 stable snapshot. lspci -vv gives me this. 05:00.0 Audio device: Creative Labs [SB X-Fi Xtreme Audio] CA0110-IBG Subsystem: Creative Labs Device 0018 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64 (500ns min, 5000ns max) Interrupt: pin A routed to IRQ 18 Region 0: Memory at fdffc000 (32-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel

2 1

[alsa-devel] RFC: PCM extra attributes
by Takashi Iwai 24 Jun '09

24 Jun '09

Hi, this is yet another topic I'm (currently) working on -- the addition of PCM ioctls to get/set some extra attributes. Basically, it adds two simple ioctls for getting/setting extra attributes to the PCM substream. The attribute has a sort of TLV form, /* PCM extra attributes */ struct snd_pcm_attr { unsigned int type; /* SNDRC_PCM_TYPE_ATTR_XXX */ unsigned int len; /* GET R: the max elements in value array * W: the actually written # elements * SET R/W: # elements to store */ unsigned int value[0]; /* value(s) to read / write */ }; And corresponding two ioctls #define SNDRV_PCM_IOCTL_GET_ATTR _IOWR('A', 0x14, struct snd_pcm_attr) #define SNDRV_PCM_IOCTL_SET_ATTR _IOWR('A', 0x15, struct snd_pcm_attr) The caller sets type and len fields. For getting attributes, the len points the max buffer length, and in return, the actual size is stored there. For setting, it's the size to set. Now, for getting the control ids associated to a PCM substream, it'll be like unsigned int buf[32]; buf[0] = SNDRV_PCM_ATTR_ASSOC_CTLS; buf[1] = ARRAY_SIZE(buf) - 2; /* minus header size */ ioctl(fd, SNDRV_PCM_IOCTL_GET_ATTR, buf); The kernel side implementation (a core part) is attached below. In this prototype, each driver is supposed to allocate and set substream->assoc_ctls and substream->num_assoc_ctls appropriately. I chose it since the control ids are statically associated to the PCM substreams at the driver loading time. It could be, of course, a liked list with kmalloc, but I thought it's overkill. Well, that's a pretty subtle thing and can be changed at any time. But, we need to define ABI properly from the beginning. So, if you have any comments on ABI or on the whole design, please speak up. An important note is that I'm planning to use this framework for getting/setting the PCM (surround) channel mapping. It's straightforward for this ABI. The internal implementation may vary, but the interface is the question now. For example, one could imagine that the channel mapping belongs to hw_params. But, it looks difficult because of the nature of channel mapping; it doesn't match with a simple integer parameter as hw_parmas has. The PCM channel mapping feature will be certainly useful for PA (and even for normal apps with a new alsa-lib plugin to convert the correct the channel mapping properly on demand). So, this is rather more interesting stuff to me, too... thanks, Takashi --- diff --git a/include/sound/asound.h b/include/sound/asound.h index 82aed3f..159826b 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -445,6 +445,21 @@ struct snd_xfern { snd_pcm_uframes_t frames; }; +/* PCM extra attributes */ +struct snd_pcm_attr { + unsigned int type; /* SNDRC_PCM_TYPE_ATTR_XXX */ + unsigned int len; /* GET R: the max elements in value array + * W: the actually written # elements + * SET R/W: # elements to store + */ + unsigned int value[0]; /* value(s) to read / write */ +}; + +/* PCM attribute types */ +enum { + SNDRV_PCM_ATTR_ASSOC_CTLS, /* array of associated ctl ids */ +}; + enum { SNDRV_PCM_TSTAMP_TYPE_GETTIMEOFDAY = 0, /* gettimeofday equivalent */ SNDRV_PCM_TSTAMP_TYPE_MONOTONIC, /* posix_clock_monotonic equivalent */ @@ -459,6 +474,8 @@ enum { #define SNDRV_PCM_IOCTL_HW_PARAMS _IOWR('A', 0x11, struct snd_pcm_hw_params) #define SNDRV_PCM_IOCTL_HW_FREE _IO('A', 0x12) #define SNDRV_PCM_IOCTL_SW_PARAMS _IOWR('A', 0x13, struct snd_pcm_sw_params) +#define SNDRV_PCM_IOCTL_GET_ATTR _IOWR('A', 0x14, struct snd_pcm_attr) +#define SNDRV_PCM_IOCTL_SET_ATTR _IOWR('A', 0x15, struct snd_pcm_attr) #define SNDRV_PCM_IOCTL_STATUS _IOR('A', 0x20, struct snd_pcm_status) #define SNDRV_PCM_IOCTL_DELAY _IOR('A', 0x21, snd_pcm_sframes_t) #define SNDRV_PCM_IOCTL_HWSYNC _IO('A', 0x22) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 2389352..98416b0 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -392,6 +392,10 @@ struct snd_pcm_substream { struct snd_info_entry *proc_prealloc_entry; struct snd_info_entry *proc_prealloc_max_entry; #endif + /* associated controls */ + unsigned int num_assoc_ctls; + unsigned int *assoc_ctls; + /* misc flags */ unsigned int hw_opened: 1; }; diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 145931a..7a46674 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -748,6 +748,7 @@ static void snd_pcm_free_stream(struct snd_pcm_str * pstr) substream_next = substream->next; snd_pcm_timer_done(substream); snd_pcm_substream_proc_done(substream); + kfree(substream->assoc_ctls); kfree(substream); substream = substream_next; } diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c index 08bfed5..07512b3 100644 --- a/sound/core/pcm_compat.c +++ b/sound/core/pcm_compat.c @@ -485,6 +485,8 @@ static long snd_pcm_ioctl_compat(struct file *file, unsigned int cmd, unsigned l case SNDRV_PCM_IOCTL_TTSTAMP: case SNDRV_PCM_IOCTL_HWSYNC: case SNDRV_PCM_IOCTL_PREPARE: + case SNDRV_PCM_IOCTL_GET_ATTR: + case SNDRV_PCM_IOCTL_SET_ATTR: case SNDRV_PCM_IOCTL_RESET: case SNDRV_PCM_IOCTL_START: case SNDRV_PCM_IOCTL_DROP: diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 84da3ba..46129c7 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2492,6 +2492,43 @@ static int snd_pcm_tstamp(struct snd_pcm_substream *substream, int __user *_arg) return 0; } +/* store back the associated ctl ids to the given PCM substream */ +static int snd_pcm_get_attr_assoc_ctls(struct snd_pcm_substream *substream, + unsigned int __user *arg) +{ + unsigned int len, i; + + arg++; + if (get_user(len, arg)) + return -EFAULT; + for (i = 0; i < len && i < substream->num_assoc_ctls; i++) + put_user(substream->assoc_ctls[i], arg + i + 1); + put_user(i, arg); + return 0; +} + +/* GET_ATTR ioctl */ +static int snd_pcm_get_attr(struct snd_pcm_substream *substream, + unsigned int __user *arg) +{ + unsigned int type; + + if (get_user(type, arg)) + return -EFAULT; + switch (type) { + case SNDRV_PCM_ATTR_ASSOC_CTLS: + return snd_pcm_get_attr_assoc_ctls(substream, arg); + } + return -ENOENT; +} + +/* SET_ATTR ioctl */ +static int snd_pcm_set_attr(struct snd_pcm_substream *substream, + struct snd_pcm_attr __user *arg) +{ + return -ENOENT; +} + static int snd_pcm_common_ioctl1(struct file *file, struct snd_pcm_substream *substream, unsigned int cmd, void __user *arg) @@ -2519,6 +2556,10 @@ static int snd_pcm_common_ioctl1(struct file *file, return snd_pcm_channel_info_user(substream, arg); case SNDRV_PCM_IOCTL_PREPARE: return snd_pcm_prepare(substream, file); + case SNDRV_PCM_IOCTL_GET_ATTR: + return snd_pcm_get_attr(substream, arg); + case SNDRV_PCM_IOCTL_SET_ATTR: + return snd_pcm_set_attr(substream, arg); case SNDRV_PCM_IOCTL_RESET: return snd_pcm_reset(substream); case SNDRV_PCM_IOCTL_START:

3 17

Re: [alsa-devel] Should my Creative X-fi card be loading the snd-hda-intel module
by Susan Cragin 24 Jun '09

24 Jun '09

Bill, My creative x-fi card used to work great with the daily build, and the module I mentioned. I've been away for it for a couple of months because I need sound for wine programs and there were issues with wine sound, and I just got back. I was surprised to see such a big step backward. When I was last using a daily build, both unstable and stable worked for me. Susan Susan, I also had a CA110-IBG chip on my Creative X-Fi. Never could get it to work, not under Ubuntu or Fedora, not with the Creative-supplied driver and also not with ctxfi. So we dropped that card and went to using an Asus PCI-e sound card instead. As for what module should load, here is what Takashi said on June 11: Are you sure that your board has really emu20k1 or emu20k2 chip? X-Fi isn't always with these chips. Some are HD-audio only (thus snd-hda-intel is the right driver) and some are with CA0106 chip. -------- Original Message -------- Shouldn't it be loading snd-hda-codec-ca0110 instead? I am using the 6/24 stable snapshot. lspci -vv gives me this. 05:00.0 Audio device: Creative Labs [SB X-Fi Xtreme Audio] CA0110-IBG Subsystem: Creative Labs Device 0018 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64 (500ns min, 5000ns max) Interrupt: pin A routed to IRQ 18 Region 0: Memory at fdffc000 (32-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel

1 0

[alsa-devel] Acer Aspire 7730G support
by Denis Kup 24 Jun '09

24 Jun '09

Hey, when i first installed alsa to my laptop, there no sound from subwoofer, jacks were map very strange (2 surrounds for ex.). I try all models from the list, but all don't support my subwoofer. After a weak of hard work i made a patch that enables ACER_ASPIRE_7730G model for my laptop. It have ALC888S, 3 jacks, 6 channels, stereo internal frontal speakers and internal subwoofer.You can see acer.com for more info about hardware and jacks. It's the changes i made to /alsa-kernel/pci/hda/patch_realtek.c: > enum { > ................... > ALC888_ACER_ASPIRE_7730G, > ................... > }; > ////////////////////////////////////////////////////////////// > static struct hda_verb alc888_acer_aspire_7730G_verbs[] = { > {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /*CHANGE DIRECT FRONT TO BLACK JACK*/ > {0x17, AC_VERB_SET_CONNECT_SEL, 0x02}, /* CHANGE: ENABLES SUBWOOFER ON 7730*/ > {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, //CHANGE FOR AUTOMUTE > { } /* end */ > }; > > /*CHANGE ENABLE AUTOMUTING ON ACER 7730G > ////////////////////////////////////////////////////// > //////////////////////////////////////////////////////*/ > > static void alc888_acer_aspire_7730G_front_automute(struct hda_codec *codec) > { > unsigned int present; > > present = snd_hda_codec_read(codec, 0x15, 0, > AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; > snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, > HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); > snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 0, > HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); > } > > static void alc888_acer_aspire_7730G_unsol_event(struct hda_codec *codec, > unsigned int res) > { > if ((res >> 26) == ALC880_HP_EVENT) > alc888_acer_aspire_7730G_front_automute(codec); > } > > /*//////////////////////////////////////////////////// > ////////////////////////////////////////////////// > //////////////////////////////////////////////////////*/ > static struct alc_config_preset alc883_presets[] = { > > [ALC888_ACER_ASPIRE_7730G] = { > .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer }, > .init_verbs = { alc883_init_verbs, alc888_acer_aspire_7730G_verbs }, > .num_dacs = ARRAY_SIZE(alc883_dac_nids), > .dac_nids = alc883_dac_nids, > .dig_out_nid = ALC883_DIGOUT_NID, > .dig_in_nid = ALC883_DIGIN_NID, > .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes), > .channel_mode = alc883_3ST_6ch_modes, > .need_dac_fix = 1, > .input_mux = &alc883_capture_source, > .unsol_event = alc888_acer_aspire_7730G_unsol_event, /* CHANGE AUTOMUTE DECLARATION*/ > .init_hook = alc888_acer_aspire_7730G_front_automute, > }, > > /////////////////////////////////////////////////////////////// > > static const char *alc883_models[ALC883_MODEL_LAST] = { > ................................. > [ALC888_ACER_ASPIRE_7730G] = "acer-aspire-7730g", > ......................... > > //////////////////////////////// > static struct snd_pci_quirk alc883_cfg_tbl[] = { > ...................................... > SND_PCI_QUIRK(0x1025, 0x0142, "Acer Aspire 7730G", ALC888_ACER_ASPIRE_7730G), > .......................................... After changing file i install driver, and it automatically detect its model (maybe there are old settings from previous install, so i dont 100% sure that it was detected). With this settings subwoofer works perfect, internal stereo and subwoofer mute when headphone are plugged all works, except 1 thing: subwoofer uses center channel instead LFE. To fix it i use such asound.conf file: > pcm.upmix_20to51 { > type plug > slave.pcm lowpass_21to21 > slave.channels 3 > ttable { > 0.0 1 # left channel > 1.1 1 # right channel > 0.2 1 # mix left and right ... > 1.2 1 # ... channel for subwoofer > } > } > pcm.lowpass_21to21 { > type ladspa > slave.pcm upmix_21to51 > path "/usr/lib/ladspa" > channels 3 > plugins { > 0 { > id 1098 # Identity (Audio) (1098/identity_audio) > policy duplicate > input.bindings.0 "Input"; > output.bindings.0 "Output"; > } > 1 { > id 1671 # 4 Pole Low-Pass Filter with Resonance (FCRCIA) (1672/lp4pole_fcrcia_oa) > policy none > input.bindings.2 "Input"; > output.bindings.2 "Output"; > input { > controls [ 1000 2 ] > } > } > } > } > pcm.upmix_21to51 { > type plug > slave.pcm surround51 > slave.channels 6 > ttable { > 0.0 1 # front left > 1.1 1 # front right > 0.2 1 # rear left > 1.3 1 # rear right > 0.5 0.5 # swap channels > 1.5 0.5 > 2.4 1 > } > } > > pcm.swapCLFE_51 { > type plug > slave.pcm surround51 > slave.channels 6 > ttable { > 0.0 1 # front left > 1.1 1 # front right > 2.2 1 # rear left > 3.3 1 # rear right > 4.5 1 # swap channels > 5.4 1 > } > } I use upmix_21to51 and swapCLFE_51 devices in Amarok settings. Is there any chance to swap Center and LFE in patch_realtek.c? And can my 7730g config be used in future version of alsa? And how can I detect params for snd_pci_quirk alc883_cfg_tbl ? And some config settings for my laptop, if needed: > linux-wpg9:/home/dener # cat /proc/asound/card0/codec* > Codec: Realtek ALC888 > Address: 0 > Function Id: 0x1 > Vendor Id: 0x10ec0888 > Subsystem Id: 0x10250142 > Revision Id: 0x100202 > No Modem Function Group found > Default PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Default Amp-In caps: N/A > Default Amp-Out caps: N/A > GPIO: io=2, o=0, i=0, unsolicited=1, wake=1 > IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > Node 0x02 [Audio Output] wcaps 0x411: Stereo > Converter: stream=5, channel=0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x03 [Audio Output] wcaps 0x411: Stereo > Converter: stream=5, channel=0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x04 [Audio Output] wcaps 0x411: Stereo > Converter: stream=5, channel=0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x05 [Audio Output] wcaps 0x411: Stereo > Converter: stream=0, channel=0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x06 [Audio Output] wcaps 0x611: Stereo Digital > Converter: stream=0, channel=0 > Digital: Enabled GenLevel > Digital category: 0x2 > PCM: > rates [0x5e0]: 44100 48000 88200 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x08 [Audio Input] wcaps 0x10051b: Stereo Amp-In > Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 > Amp-In vals: [0x16 0x16] > Converter: stream=0, channel=0 > SDI-Select: 0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Connection: 1 > 0x23 > Node 0x09 [Audio Input] wcaps 0x10051b: Stereo Amp-In > Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 > Amp-In vals: [0x9f 0x9f] > Converter: stream=0, channel=0 > SDI-Select: 0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Connection: 1 > 0x22 > Node 0x0a [Audio Input] wcaps 0x100711: Stereo Digital > Converter: stream=0, channel=0 > SDI-Select: 0 > Digital: > Digital category: 0x0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Connection: 1 > 0x1f > Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 > Amp-In vals: [0x80 0x80] [0x1f 0x1f] [0x00 0x00] [0x80 0x80] [0x17 0x17] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] > Connection: 10 > 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 > Node 0x0c [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x00 0x00] [0x00 0x00] > Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 > Amp-Out vals: [0x12 0x12] > Connection: 2 > 0x02 0x0b > Node 0x0d [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x00 0x00] [0x00 0x00] > Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 > Amp-Out vals: [0x15 0x15] > Connection: 2 > 0x03 0x0b > Node 0x0e [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x00 0x00] [0x00 0x00] > Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 > Amp-Out vals: [0x15 0x15] > Connection: 2 > 0x04 0x0b > Node 0x0f [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x80 0x80] [0x80 0x80] > Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 > Amp-Out vals: [0x00 0x00] > Connection: 2 > 0x05 0x0b > Node 0x10 [Audio Output] wcaps 0x611: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > PCM: > rates [0x5e0]: 44100 48000 88200 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x11 [Pin Complex] wcaps 0x400780: Mono Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 1 > 0x10 > Node 0x12 [Pin Complex] wcaps 0x400401: Stereo > Pincap 0x00000020: IN > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Power: setting=D0, actual=D0 > Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x14 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x0001003e: IN OUT HP EAPD Detect Trigger > EAPD 0x0: > Pin Default 0x99130110: [Fixed] Speaker at Int ATAPI > Conn = ATAPI, Color = Unknown > DefAssociation = 0x1, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c* 0x0d 0x0e 0x0f 0x26 > Node 0x15 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x0001003e: IN OUT HP EAPD Detect Trigger > EAPD 0x0: > Pin Default 0x0221101f: [Jack] HP Out at Ext Front > Conn = 1/8, Color = Black > DefAssociation = 0x1, Sequence = 0xf > Pin-ctls: 0x40: OUT > Unsolicited: tag=04, enabled=1 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c* 0x0d 0x0e 0x0f 0x26 > Node 0x16 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x00000036: IN OUT Detect Trigger > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c 0x0d 0x0e* 0x0f 0x26 > Node 0x17 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x00000036: IN OUT Detect Trigger > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c 0x0d 0x0e* 0x0f 0x26 > Node 0x18 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x0000373e: IN OUT HP Detect Trigger > Vref caps: HIZ 50 GRD 80 100 > Pin Default 0x02a19c30: [Jack] Mic at Ext Front > Conn = 1/8, Color = Pink > DefAssociation = 0x3, Sequence = 0x0 > Pin-ctls: 0x40: OUT VREF_HIZ > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c 0x0d 0x0e* 0x0f 0x26 > Node 0x19 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x80 0x80] > Pincap 0x0000373e: IN OUT HP Detect Trigger > Vref caps: HIZ 50 GRD 80 100 > Pin Default 0x99a30931: [Fixed] Mic at Int ATAPI > Conn = ATAPI, Color = Unknown > DefAssociation = 0x3, Sequence = 0x1 > Misc = NO_PRESENCE > Pin-ctls: 0x24: IN VREF_80 > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c* 0x0d 0x0e 0x0f 0x26 > Node 0x1a [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x0000373e: IN OUT HP Detect Trigger > Vref caps: HIZ 50 GRD 80 100 > Pin Default 0x0281343f: [Jack] Line In at Ext Front > Conn = 1/8, Color = Blue > DefAssociation = 0x3, Sequence = 0xf > Pin-ctls: 0x40: OUT VREF_HIZ > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c 0x0d* 0x0e 0x0f 0x26 > Node 0x1b [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x0000373e: IN OUT HP Detect Trigger > Vref caps: HIZ 50 GRD 80 100 > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0xc0: OUT HP VREF_HIZ > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c* 0x0d 0x0e 0x0f 0x26 > Node 0x1c [Pin Complex] wcaps 0x400481: Stereo > Pincap 0x00000024: IN Detect > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x20: IN > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Node 0x1d [Pin Complex] wcaps 0x400400: Mono > Pincap 0x00000020: IN > Pin Default 0x4016852d: [N/A] Speaker at Ext N/A > Conn = Digital, Color = Purple > DefAssociation = 0x2, Sequence = 0xd > Misc = NO_PRESENCE > Pin-ctls: 0x20: IN > Power: setting=D0, actual=D0 > Node 0x1e [Pin Complex] wcaps 0x400780: Mono Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x02451120: [Jack] SPDIF Out at Ext Front > Conn = Optical, Color = Black > DefAssociation = 0x2, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 1 > 0x06 > Node 0x1f [Pin Complex] wcaps 0x400680: Mono Digital > Pincap 0x00000024: IN Detect > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x20: IN > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono > Processing caps: benign=0, ncoeff=25 > Processing Coefficient: 0x00 > Coefficient Index: 0x09 > Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x80 0x80] [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] > Connection: 12 > 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b 0x12 > Node 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] > Connection: 11 > 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b > Node 0x24 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x25 [Audio Output] wcaps 0x411: Stereo > Converter: stream=0, channel=0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x26 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x00 0x00] [0x80 0x80] > Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 > Amp-Out vals: [0x00 0x00] > Connection: 2 > 0x25 0x0b > Codec: LSI ID 1040 > Address: 1 > Function Id: 0x2 > Vendor Id: 0x11c11040 > Subsystem Id: 0x1025013d > Revision Id: 0x100200 > Modem Function Group: 0x1 > Codec: Nvidia MCP78 HDMI > Address: 3 > Function Id: 0x1 > Vendor Id: 0x10de0006 > Subsystem Id: 0x10de0101 > Revision Id: 0x100000 > No Modem Function Group found > Default PCM: > rates [0x0]: > bits [0x0]: > formats [0x0]: > Default Amp-In caps: N/A > Default Amp-Out caps: N/A > GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 > Node 0x04 [Audio Output] wcaps 0x211: Stereo Digital > Converter: stream=0, channel=0 > Digital: Enabled > Digital category: 0x0 > PCM: > rates [0xc0]: 48000 88200 > bits [0xf]: 8 16 20 24 > formats [0x1]: PCM > Node 0x05 [Pin Complex] wcaps 0x400381: Stereo Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x18560110: [Jack] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0x1, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Connection: 1 > 0x04 > Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > PCM: > rates [0xc0]: 48000 88200 > bits [0xf]: 8 16 20 24 > formats [0x1]: PCM > Node 0x07 [Pin Complex] wcaps 0x400381: Stereo Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x58560121: [N/A] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0x2, Sequence = 0x1 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Unsolicited: tag=00, enabled=0 > Connection: 1 > 0x06 > Node 0x08 [Audio Output] wcaps 0x211: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > PCM: > rates [0xc0]: 48000 88200 > bits [0xf]: 8 16 20 24 > formats [0x1]: PCM > Node 0x09 [Pin Complex] wcaps 0x400381: Stereo Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x58560122: [N/A] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0x2, Sequence = 0x2 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Unsolicited: tag=00, enabled=0 > Connection: 1 > 0x08 > Node 0x0a [Audio Output] wcaps 0x211: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > PCM: > rates [0xc0]: 48000 88200 > bits [0xf]: 8 16 20 24 > formats [0x1]: PCM > Node 0x0b [Pin Complex] wcaps 0x400381: Stereo Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x58560123: [N/A] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0x2, Sequence = 0x3 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Unsolicited: tag=00, enabled=0 > Connection: 1 > 0x0a > Node 0x0c [Audio Output] wcaps 0x211: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > PCM: > rates [0xc0]: 48000 88200 > bits [0xf]: 8 16 20 24 > formats [0x1]: PCM > Node 0x0d [Pin Complex] wcaps 0x400381: Stereo Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x58560124: [N/A] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0x2, Sequence = 0x4 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Unsolicited: tag=00, enabled=0 > Connection: 1 > 0x0c > > linux-wpg9:/home/dener # cat /proc/asound/version > Advanced Linux Sound Architecture Driver Version 1.0.20. > Compiled on Jun 14 2009 for kernel 2.6.27.23-0.1-pae (SMP). > > cat /proc/asound/devices > 0: [ 0] : control > 1: : sequencer > 4: [ 0- 0]: hardware dependent > 5: [ 0- 1]: hardware dependent > 7: [ 0- 3]: hardware dependent > 16: [ 0- 0]: digital audio playback > 17: [ 0- 1]: digital audio playback > 19: [ 0- 3]: digital audio playback > 24: [ 0- 0]: digital audio capture > 25: [ 0- 1]: digital audio capture > 26: [ 0- 2]: digital audio capture > 33: : timer > > cat /proc/asound/pcm > 00-00: ALC888 Analog : ALC888 Analog : playback 1 : capture 1 > 00-01: ALC888 Digital : ALC888 Digital : playback 1 : capture 1 > 00-02: ALC888 Analog : ALC888 Analog : capture 1 > 00-03: NVIDIA HDMI : NVIDIA HDMI : playback 1 > > lspci | grep -i audio > 00:1b.0 Audio device: Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 03) > I have 1.0.20 alsa version and OpenSUSE 11.1 distributive. I hope my materials are usefull PS I am from Russia, sorry for my bad English :)

2 1

[alsa-devel] Need help on mmap
by Christopher Tsui 24 Jun '09

24 Jun '09

Hi I am in the process of add mmap support in my ALSA driver. So this is what i have done so far In snd_pcm_hardware structure .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID In snd_pcm_ops structure .mmap = my_pcm_mmap So to test my code, I use aplay -M -Dhw:0 rock.wav and this is what I get ALSA lib pcm_mmap.c:378:(snd_pcm_mmap) mmap failed: Invalid argument aplay: set_params:1022: Unable to install hw params: ACCESS: MMAP_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 32 CHANNELS: 2 RATE: 44100 PERIOD_TIME: (27210 27211) PERIOD_SIZE: 1200 PERIOD_BYTES: 4800 PERIODS: 4 BUFFER_TIME: (108843 108844) BUFFER_SIZE: 4800 BUFFER_BYTES: 19200 TICK_TIME: 10000 What am I doing wrong?? Thanks in advance

2 1

[alsa-devel] Getting board to work on 2.6.30
by Tony Hoyle 24 Jun '09

24 Jun '09

Hi, I'm trying to get sound working on a board I'm working with. The sound works correctly on 2.6.24.7 but not 2.6.30. I has an intel HDA (Poulson) chipset and a sigmatel 9202 codec. Physically there's only one output - the speakers, and no input, so from a testing point of view it either works or it doesn't. Kernel output from both looks similar: 2.6.24.7: [ 19.942029] ALSA sound/pci/hda/hda_intel.c:916: HDA snoop disabled, try to enable ... OK [ 20.157458] ALSA sound/pci/hda/hda_codec.c:2760: autoconfig: line_outs=1 (0xd/0x0/0x0/0x0/0x0) [ 20.157467] ALSA sound/pci/hda/hda_codec.c:2764: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 20.157474] ALSA sound/pci/hda/hda_codec.c:2768: hp_outs=1 (0xa/0x0/0x0/0x0/0x0) [ 20.157481] ALSA sound/pci/hda/hda_codec.c:2776: inputs: mic=0x0, fmic=0xb, line=0xc, fline=0x0, cd=0x11, aux=0x0 2.6.30 (running alsa snapshot from last night): 19.379811] ALSA sound/pci/hda/hda_codec.c:3795: autoconfig: line_outs=1 (0x10/0x0/0x0/0x0/0x0) [ 19.379824] ALSA sound/pci/hda/hda_codec.c:3799: speaker_outs=2 (0xa/0x7/0x0/0x0/0x0) [ 19.379833] ALSA sound/pci/hda/hda_codec.c:3803: hp_outs=1 (0xd/0x0/0x0/0x0/0x0) [ 19.379842] ALSA sound/pci/hda/hda_codec.c:3804: mono: mono_out=0x10 [ 19.379849] ALSA sound/pci/hda/hda_codec.c:3807: dig-out=0x8/0x0 [ 19.379856] ALSA sound/pci/hda/hda_codec.c:3815: inputs: mic=0xb, fmic=0x0, line=0xc, fline=0x0, cd=0x11, aux=0x0 [ 19.390010] input: HDA Intel MID CD at Ext Rear Jack as /class/input/input3 [ 19.401109] input: HDA Intel MID Line In at Ext Rear Jack as /class/input/input4 [ 19.414726] input: HDA Intel MID Mic at Ext Rear Jack as /class/input/input5 [ 19.430427] input: HDA Intel MID Line Out at Ext Rear Jack as /class/input/input6 [ 19.447561] input: HDA Intel MID HP Out at Ext Front Jack as /class/input/input7 It seems to be creating the outputs as inputs? This seems wrong. I've tried every model setting I could find and reckon I need to create a new one.. but don't actually know what to create to get things working - there are lots of differences in the alsa between the two kernels and it isn't something that looks easy to pin down short of going nuts with git bisect. Can someone give me a pointer where to start looking at this? Tony Codec: SigmaTel STAC9202 Address: 0 Function Id: 0x1 Vendor Id: 0x83847632 Subsystem Id: 0x00000100 Revision Id: 0x100101 No Modem Function Group found Default PCM: rates [0x7e0]: 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 GPIO: io=4, o=0, i=0, unsolicited=1, wake=1 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0xd0401: Stereo Converter: stream=0, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x03 [Audio Input] wcaps 0x1d0541: Stereo Converter: stream=0, channel=0 SDI-Select: 0 Power: setting=D0, actual=D0 Delay: 13 samples Connection: 1 0x14 Processing caps: benign=0, ncoeff=0 Node 0x04 [Audio Input] wcaps 0x140311: Stereo Digital Converter: stream=0, channel=0 SDI-Select: 0 Digital: Digital category: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Delay: 4 samples Connection: 1 0x07 Node 0x05 [Audio Output] wcaps 0x40211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x1e0]: 44100 48000 88200 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Delay: 4 samples Node 0x06 [Audio Selector] wcaps 0x300901: Stereo R/L Connection: 3 0x02* 0x07 0x14 Node 0x07 [Pin Complex] wcaps 0x430681: Stereo Digital Pincap 0x00010024: IN EAPD Detect EAPD 0x0: Pin Default 0x01110014: [Jack] Speaker at Ext Rear Conn = 1/8, Color = Unknown DefAssociation = 0x1, Sequence = 0x4 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Delay: 3 samples Node 0x08 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x01410013: [Jack] SPDIF Out at Ext Rear Conn = 1/8, Color = Unknown DefAssociation = 0x1, Sequence = 0x3 Pin-ctls: 0x40: OUT Connection: 2 0x05* 0x14 Node 0x09 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=0 Amp-Out vals: [0x07 0x07] Connection: 1 0x0f Node 0x0a [Pin Complex] wcaps 0x400181: Stereo Pincap 0x0000003f: IN OUT HP Detect Trigger ImpSense Pin Default 0x02111011: [Jack] Speaker at Ext Front Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0x1 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x0e Node 0x0b [Pin Complex] wcaps 0x400181: Stereo Pincap 0x00001737: IN OUT Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x01a10021: [Jack] Mic at Ext Rear Conn = 1/8, Color = Unknown DefAssociation = 0x2, Sequence = 0x1 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=02, enabled=1 Connection: 1 0x0e Node 0x0c [Pin Complex] wcaps 0x400181: Stereo Pincap 0x00001737: IN OUT Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x01810022: [Jack] Line In at Ext Rear Conn = 1/8, Color = Unknown DefAssociation = 0x2, Sequence = 0x2 Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=03, enabled=1 Connection: 1 0x0e Node 0x0d [Pin Complex] wcaps 0x400181: Stereo Pincap 0x0000003f: IN OUT HP Detect Trigger ImpSense Pin Default 0x02214012: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x2 Pin-ctls: 0x00: Unsolicited: tag=01, enabled=1 Connection: 1 0x0e Node 0x0e [Audio Selector] wcaps 0x300105: Stereo Amp-Out Amp-Out caps: N/A Amp-Out vals: [0x17 0x17] Connection: 1 0x06 Node 0x0f [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-Out vals: [0x00 0x00] Connection: 5 0x0b* 0x0c 0x0d 0x0a 0x11 Node 0x10 [Pin Complex] wcaps 0x400104: Mono Amp-Out Amp-Out caps: N/A Amp-Out vals: [0x00] Pincap 0x00000010: OUT Pin Default 0x01010031: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Unknown DefAssociation = 0x3, Sequence = 0x1 Pin-ctls: 0x40: OUT Connection: 1 0x12 Node 0x11 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x01310023: [Jack] CD at Ext Rear Conn = 1/8, Color = Unknown DefAssociation = 0x2, Sequence = 0x3 Pin-ctls: 0x20: IN Node 0x12 [Audio Mixer] wcaps 0x200100: Mono Connection: 1 0x06 Node 0x13 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x03, nsteps=0x03, stepsize=0x17, mute=1 Amp-Out vals: [0x00] Node 0x14 [Audio Selector] wcaps 0x30090d: Stereo Amp-Out R/L Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Connection: 2 0x09* 0x15 Node 0x15 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x01d10024: [Jack] Digitial In at Ext Rear Conn = 1/8, Color = Unknown DefAssociation = 0x2, Sequence = 0x4 Pin-ctls: 0x20: IN

2 2

[alsa-devel] [PATCH 1/3] ASoC: TWL4030: Enable voice filters for voice sidetone
by Lopez Cruz, Misael 24 Jun '09

24 Jun '09

Digital voice loopback (sidetone) requires voice filters to be enabled: VTXL, VTXR, VRX. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl4030.c | 67 +++++++++++++++++++++++++------------------- 1 files changed, 38 insertions(+), 29 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 4dbb853..780217a 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -332,6 +332,28 @@ static void twl4030_power_down(struct snd_soc_codec *codec) twl4030_codec_enable(codec, 0); } +/* In case of voice mode, the RX1 L(VRX) for downlink and the TX2 L/R + * (VTXL, VTXR) for uplink has to be enabled/disabled. */ +static void twl4030_voice_enable(struct snd_soc_codec *codec, int direction, + int enable) +{ + u8 reg, mask; + + reg = twl4030_read_reg_cache(codec, TWL4030_REG_OPTION); + + if (direction == SNDRV_PCM_STREAM_PLAYBACK) + mask = TWL4030_ARXL1_VRX_EN; + else + mask = TWL4030_ATXL2_VTXL_EN | TWL4030_ATXR2_VTXR_EN; + + if (enable) + reg |= mask; + else + reg &= ~mask; + + twl4030_write(codec, TWL4030_REG_OPTION, reg); +} + /* Earpiece */ static const struct snd_kcontrol_new twl4030_dapm_earpiece_controls[] = { SOC_DAPM_SINGLE("Voice", TWL4030_REG_EAR_CTL, 0, 1, 0), @@ -712,7 +734,22 @@ static int bypass_event(struct snd_soc_dapm_widget *w, reg = twl4030_read_reg_cache(w->codec, m->reg); - if (m->reg <= TWL4030_REG_ARXR2_APGA_CTL) { + if (m->reg == TWL4030_REG_VSTPGA) { + /* Voice digital bypass */ + if (reg) { + twl4030->bypass_state |= (1 << 5); + twl4030_voice_enable(w->codec, + SNDRV_PCM_STREAM_PLAYBACK, 1); + twl4030_voice_enable(w->codec, + SNDRV_PCM_STREAM_CAPTURE, 1); + } else { + twl4030->bypass_state &= ~(1 << 5); + twl4030_voice_enable(w->codec, + SNDRV_PCM_STREAM_PLAYBACK, 0); + twl4030_voice_enable(w->codec, + SNDRV_PCM_STREAM_CAPTURE, 0); + } + } else if (m->reg <= TWL4030_REG_ARXR2_APGA_CTL) { /* Analog bypass */ if (reg & (1 << m->shift)) twl4030->bypass_state |= @@ -726,12 +763,6 @@ static int bypass_event(struct snd_soc_dapm_widget *w, twl4030->bypass_state |= (1 << 4); else twl4030->bypass_state &= ~(1 << 4); - } else if (m->reg == TWL4030_REG_VSTPGA) { - /* Voice digital bypass */ - if (reg) - twl4030->bypass_state |= (1 << 5); - else - twl4030->bypass_state &= ~(1 << 5); } else { /* Digital bypass */ if (reg & (0x7 << m->shift)) @@ -1806,28 +1837,6 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } -/* In case of voice mode, the RX1 L(VRX) for downlink and the TX2 L/R - * (VTXL, VTXR) for uplink has to be enabled/disabled. */ -static void twl4030_voice_enable(struct snd_soc_codec *codec, int direction, - int enable) -{ - u8 reg, mask; - - reg = twl4030_read_reg_cache(codec, TWL4030_REG_OPTION); - - if (direction == SNDRV_PCM_STREAM_PLAYBACK) - mask = TWL4030_ARXL1_VRX_EN; - else - mask = TWL4030_ATXL2_VTXL_EN | TWL4030_ATXR2_VTXR_EN; - - if (enable) - reg |= mask; - else - reg &= ~mask; - - twl4030_write(codec, TWL4030_REG_OPTION, reg); -} - static int twl4030_voice_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { -- 1.5.4.3

3 10

[alsa-devel] USB microphone not working. No sound, although device visible.
by Shapeshifter 23 Jun '09

23 Jun '09

Hi there. Following problem, details below: PROBLEM: I can't record anything using the microphone which is integrated into my USB webcam. The cam shows up as a device, and I can adjust the only available mixer which is for <Mic> in [Capture]. It's not muted and it's maxed out. DETAILS and PASTES: Alsa-info output: http://www.alsa-project.org/db/?f=f12fd357bc269d27ebdb4b5d562227bbe22a1086 You might notice the device comes up as [default]. If I run... $ arecord -Dplughw:1,0 -f cd -vv /dev/null or $ arecord -Ddefault -f cd -vv /dev/null then it out puts a lot of stuff (both the same) which I pasted in the below pastebin, but at the bottom it doesn't show any #'s and %'s to indicate the volume. The blue light on the webcam also turns on. If I save to a file instead of /dev/null and aplay it, it's just silence. Here's the output from the above command: http://pastie.org/521697 Although if I run $ arecord -Dhw:1,0 -f cd -vv /dev/null then it fails with: -> Recording WAVE '/dev/null' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo -> arecord: set_params:984: Channels count non available" I made a screenshot illustrating that everything is turned up and unmuted: http://omploader.org/vMXZldg ALREADY TRIED: - Switching the two devices so that 0 becomes the webcam and 1 the intel-hda through modprobe.conf but that had no other effect other than the switching. - Playing a lot with the device, muting, triplechecking volumes etc. etc. - Trying recording something in audacity. First time, the light goes on but it doesn't record, on subsequent tries, audacity spits an error stating "error opening sound device". - Trying it in linphone, light turns on, but other party can't hear anything. Any ideas how to fix hits? Btw, it used to work in ubuntu 9.04 for some reason. Tried it on two Archlinux boxes, it's exactly the same on both. But everything works really, everything is very up to date (I'm sure ubuntu has older versions of stuff). Well, all the info is in the alsa-info link above. Hope I can work this out! Thanks for any input! Regards.

1 0

[alsa-devel] ASoC CODEC power management on reboot
by Ben Dooks 23 Jun '09

23 Jun '09

I've run into a problem where when the system reboots the audio starts playing random data. I suspect that this is due to the CODEC being left powered and when the during/after the reset process the GPIO lines which form the I2S bus are left floating. Should the ASoC (possibly DAPM) layer sort out powering down the CODEC before reboot, or should it be down to the codec or the mapping driver to do this? -- Ben Q: What's a light-year? A: One-third less calories than a regular year.

2 3

[alsa-devel] Please help in adding ams-delta support to ASoC
by Janusz Krzysztofik 23 Jun '09

23 Jun '09

Hi, I am trying to add sound support for ams-delta omap machine, yet without much success. Three years ago, Mark Underwood created an omap-alsa compatible driver that basically worked[1]. It was derieved from similiar driver for aic23 codec found on omap osk machine. It looks like Mark has never managed to finish his work. It's not clear for me if he has found that getting a working fullduplex sound is technically impossible on a voice modem codec, shared among modem and cpu, controlable only from modem side afaik, but I have decided to give it a try. Since Mark's initial work, omap-alsa framework has been depreciated in favour of soc-omap. API changes are so significant that Mark's code is rather not useable directly any more. However, I am trying to use it as a starting point, by comparing it against it's prototype osk/aic23 code. Following Mark, I am trying to derieve the new ams-delta sound driver from current asoc driver for omap osk9512. For codec part, I decided to base my work on much more simple ad73311 rather that tlv320aic23. Comparing Mark's code agaist it's osk/aic23 prototype, I can see the folowing significant changes: 1. rate tables/bitmaps found in hw_constraint_rates, snd_omap_alsa_playback and snd_omap_alsa_capture structures limited to 8kHz, 2. hardware related code found in codec_configure_dev(), codec_clock_on() and codec_clock_off() callback functions replaced with ams-delta hardware specific code that switches the codec DAI pins from modem chip to mcbsp cpu interface and back, 3. codec_set_samplerate() and all mixer related functions replaced with stubs. 4. the following McBSP register settings changes: - .xcr2 = XPHASE | XFRLEN2(OMAP_MCBSP_WORD_8) | - XWDLEN2(OMAP_MCBSP_WORD_16) | XDATDLY(0) | XFIG, + .xcr2 = XPHASE | XWDLEN2(OMAP_MCBSP_WORD_16) | XFRLEN2(0), - .srgr1 = FWID(DEFAULT_BITPERSAMPLE - 1), + .srgr1 = CLKGDV(0), - .srgr2 = GSYNC | CLKSP | FSGM | FPER(DEFAULT_BITPERSAMPLE * 2 - 1), + .srgr2 = GSYNC, - .pcr0 = CLKXP | CLKRP, /* mcbsp: slave */ I have found points 1. to 3. rather trivial to implement in new framework. Regarding point 4., it looks like detailed register settings are now done inside omap-mcbsp.c, based on machine specified format, so I have to find out what format should be specified for ams-delta. Please correct me if I am missing something. There was one more modification in Mark's code, addressing dma chaining problem on omap15xx hardware, but as far as I could see, the problem was already solved in the asoc omap framework. Initially, I based my work on ompenembedded provided linux-omap.git revision 90e758af52ba803cba233fabee81176d99589f09. The results were rather poor - total system hangup after first device access, with no single message. So I have switched to linux-2.6.30-rc5 and now I can safely access the device, however it does not work as expected. aplay and arecord wait forever, cat to/from /dev/dsp breaks with hardware error messgae. DMA interrput counters stay at 0. However, codec switching that I do from machine->ops->startup/shutdown seems working, as modem stops producing any sounds while the alsa device is in use and gets back thereafter. First of all, I'd like to make sure if my problem is related to my code only. As I am new in these areas, I would like to ask you if the omap asoc framework is stable enough to relay on. If yes, could you please look at my dirty code (attached) an give me some hints? I can provide you with more information if necessary. Regards, Janusz [1] http://www.earth.li/pipermail/e3-hacking/2006-April/000481.html

6 52

[alsa-devel] Indirect Callback with ack is called with no data
by Singaravelan Nallasellan 23 Jun '09

23 Jun '09

I am using indirect callback mechanism inside ack. I see that ack does not call my indirect callback pointer. Going through the code, I suspect that the appl_ptr is not being incremented and there is no data for the indirect callback to be called. After few acks, I get the indirect callback called. With aplay, I am able to play the wav file. But with flash, there is not data being sent to the driver. Please help me with any clues. Thanks in advance. Sing _________________________________________________________________ Insert movie times and more without leaving Hotmail®. http://windowslive.com/Tutorial/Hotmail/QuickAdd?ocid=TXT_TAGLM_WL_HM_Tutor…

3 10

Re: [alsa-devel] alc889
by Takashi Iwai 23 Jun '09

23 Jun '09

At Tue, 23 Jun 2009 00:05:42 +0200, Stefan Krastanov wrote: > > Hello, > > Thank you for your fast response, I appreciate it, knowing you don't have all > that free time. > > As I have said my hardware configuration is: laptop with alc889 soundcard > - two front > speakers > - one > sub-woofer > - one > internal microphone > - jacks > with jacksensing (tested on windows) - mic, line-in, headphones > - digital > IO > - > potentiometer for volume control > > I installed the latest snapshot using ubuntu's automated script (I manually > verified that it's the latest) and used model=auto. The results were the same > when I tried it few months ago: > - "Front" controls the front speakers OK > - "Front" also controls headphones problem > - "Headphones" slider does not control anything problem > - No control slider for the build-in sub-woofer problem > - microphones working (external and internal) OK > - potentiometer for volume control not working problem > - Line-in not tested, digital IO not tested not tested > - jack sensing on the headphones is not working problem > - jack sensing on microphone and line in not tested not tested > > Attached to the message you can find the output from alsa-info. Hm, how about model=fujitsu-pi2515 or model=fujitsu-xa3530 option? Just a wild guess from the PCI SSID, though. Takashi

2 2

[alsa-devel] Noisy USB audio stream using MPlayer
by Eran Ben-Avi 23 Jun '09

23 Jun '09

Hi, I encountered some issue with my USB audio device and couldn't find any similar issue in previous mailing lists, any help will be appreciated. When playing an audio stream using MPlayer Ver1.0rc2-4.1.2 (tried different sample rates 44K, 48K and different files WAV , MP3) using my Creative SB0271 USB device on ARM based system(Marvell Kirkwood SoC) and kernel 2.6.22.18, I hear an interfering constant noise additionally to the original audio stream. At the beginning I heard this noise also when I used aplay(for WAV) but after I reinstalled the ALSA-LIB and ALSA-UTILS(version 1.0.14rc4), the aplay suddenly started to work smooth. I tested different USB audio device(AKM) but got same noisy results. Another piece of information I can provide is that increasing the CPU utilization eliminates the noise completely - maybe a timing issue ??? Any ideas ? Thanks, Eran.

2 3

[alsa-devel] [PATCH] ASoC: TWL4030: Correct bypass event for voice sidetone
by Lopez Cruz, Misael 23 Jun '09

23 Jun '09

Event for voice sidetone was being interpreted as an analog HiFi bypass event because VSTPGA register offset is less than ARXR2_APGA_CTL offset. Reordering the register checks allows to handle voice digital bypass event properly. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl4030.c | 20 +++++++++++++------- 1 files changed, 13 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index f916a9a..df42fa2 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -712,7 +712,19 @@ static int bypass_event(struct snd_soc_dapm_widget *w, reg = twl4030_read_reg_cache(w->codec, m->reg); - if (m->reg <= TWL4030_REG_ARXR2_APGA_CTL) { + /* + * bypass_state[0:3] - analog HiFi bypass + * bypass_state[4] - analog voice bypass + * bypass_state[5] - digital voice bypass + * bypass_state[6:7] - digital HiFi bypass + */ + if (m->reg == TWL4030_REG_VSTPGA) { + /* Voice digital bypass */ + if (reg) + twl4030->bypass_state |= (1 << 5); + else + twl4030->bypass_state &= ~(1 << 5); + } else if (m->reg <= TWL4030_REG_ARXR2_APGA_CTL) { /* Analog bypass */ if (reg & (1 << m->shift)) twl4030->bypass_state |= @@ -726,12 +738,6 @@ static int bypass_event(struct snd_soc_dapm_widget *w, twl4030->bypass_state |= (1 << 4); else twl4030->bypass_state &= ~(1 << 4); - } else if (m->reg == TWL4030_REG_VSTPGA) { - /* Voice digital bypass */ - if (reg) - twl4030->bypass_state |= (1 << 5); - else - twl4030->bypass_state &= ~(1 << 5); } else { /* Digital bypass */ if (reg & (0x7 << m->shift)) -- 1.5.4.3

3 2

[alsa-devel] [PATCH] ASoC: TWL4030: Add AVADC Clock Priority
by Lopez Cruz, Misael 23 Jun '09

23 Jun '09

AVDAC clk priority allows to determine the path ADC must be connected when the codec is in option2 and both HiFi and Voice paths are enabled. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl4030.c | 12 ++++++++++++ 1 files changed, 12 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 780217a..42cdcce 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1036,6 +1036,16 @@ static DECLARE_TLV_DB_SCALE(digital_capture_tlv, 0, 100, 0); */ static DECLARE_TLV_DB_SCALE(input_gain_tlv, 0, 600, 0); +/* AVADC clock priority */ +static const char *twl4030_avadc_clk_priority_texts[] = { + "Voice high priority", "HiFi high priority" +}; + +static const struct soc_enum twl4030_avadc_clk_priority_enum = + SOC_ENUM_SINGLE(TWL4030_REG_AVADC_CTL, 2, + ARRAY_SIZE(twl4030_avadc_clk_priority_texts), + twl4030_avadc_clk_priority_texts); + static const char *twl4030_rampdelay_texts[] = { "27/20/14 ms", "55/40/27 ms", "109/81/55 ms", "218/161/109 ms", "437/323/218 ms", "874/645/437 ms", "1748/1291/874 ms", @@ -1137,6 +1147,8 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = { SOC_DOUBLE_TLV("Analog Capture Volume", TWL4030_REG_ANAMIC_GAIN, 0, 3, 5, 0, input_gain_tlv), + SOC_ENUM("AVADC Clock Priority", twl4030_avadc_clk_priority_enum), + SOC_ENUM("HS ramp delay", twl4030_rampdelay_enum), SOC_ENUM("Vibra H-bridge mode", twl4030_vibradirmode_enum), -- 1.5.4.3

3 2

[alsa-devel] Nmode issues
by Dmitri Seletski 23 Jun '09

23 Jun '09

Hello Guys. I got upgraded my kernel to uname -a Linux dimko 2.6.30 #3 SMP PREEMPT Mon Jun 22 23:58:16 IST 2009 x86_64 AMD Athlon(tm) 64 X2 Dual Core Processor 6000+ AuthenticAMD GNU/Linux i was able to make files now!(previously i couldnt) make make install modprobe ath9k [ 941.365887] f71882fg: Not a Fintek device [ 941.365932] f71882fg: Found f71882fg chip at 0xa00, revision 32 [ 941.366116] f71882fg f71882fg.2560: Fan: 1 is in duty-cycle mode [ 941.366118] f71882fg f71882fg.2560: Fan: 2 is in duty-cycle mode [ 941.366119] f71882fg f71882fg.2560: Fan: 3 is in duty-cycle mode [ 941.366121] f71882fg f71882fg.2560: Fan: 4 is in duty-cycle mode [ 941.377478] k8temp 0000:00:18.3: Temperature readouts might be wrong - check erratum #141 [ 956.642074] IRQ 19/nvidia: IRQF_DISABLED is not guaranteed on shared IRQs [ 957.433234] IRQ 19/nvidia: IRQF_DISABLED is not guaranteed on shared IRQs [ 1415.180230] AV200 0000:09:04.0: PCI INT A -> Link[LN4A] -> GSI 16 (level, low) -> IRQ 16 [ 1509.674361] cfg80211: Calling CRDA to update world regulatory domain [ 1509.674399] general protection fault: 0000 [#1] PREEMPT SMP [ 1509.674411] last sysfs file: /sys/devices/pci0000:00/0000:00:04.1/usb2/2-1/2-1:1.0/host8/target8:0:0/8:0:0:0/block/sdd/sdd1/stat [ 1509.674418] CPU 1 [ 1509.674422] Modules linked in: cfg80211 rfkill_backport snd_seq snd_virtuoso snd_oxygen_lib snd_pcm snd_timer snd_page_alloc snd_mpu401_uart snd_rawmidi snd_seq_device snd soundcore k8temp f71882fg hwmon i2c_dev nvidia(P) i2c_core [ 1509.674464] Pid: 29571, comm: modprobe Tainted: P 2.6.30 #3 MS-7375 [ 1509.674470] RIP: 0010:[<ffffffff810710f1>] [<ffffffff810710f1>] module_put+0x31/0x80 [ 1509.674488] RSP: 0018:ffff880119cebf38 EFLAGS: 00010287 [ 1509.674493] RAX: 0000000000000001 RBX: ffffffffa0886d00 RCX: 0000000000000000 [ 1509.674498] RDX: ffff10013a92b6c0 RSI: 0000000000000000 RDI: 0000000000000001 [ 1509.674503] RBP: ffff880119cebf48 R08: 0000000000000000 R09: 0000000000000000 [ 1509.674508] R10: 0000000000000000 R11: 0000000000000000 R12: 0000000000000000 [ 1509.674513] R13: 00007f4dae114010 R14: 00000000006da7f0 R15: 00000000006da808 [ 1509.674519] FS: 00007f4dae1766f0(0000) GS:ffff88002803b000(0000) knlGS:0000000000000000 [ 1509.674524] CS: 0010 DS: 0000 ES: 0000 CR0: 000000008005003b [ 1509.674529] CR2: 00007fb3a126d098 CR3: 000000010f527000 CR4: 00000000000006e0 [ 1509.674534] DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 [ 1509.674539] DR3: 0000000000000000 DR6: 00000000ffff0ff0 DR7: 0000000000000400 [ 1509.674545] Process modprobe (pid: 29571, threadinfo ffff880119cea000, task ffff880119d0c380) [ 1509.674549] Stack: [ 1509.674552] ffffffffa0886d00 ffffffffa0886d00 ffff880119cebf78 ffffffff810730ce [ 1509.674560] 0000000000000000 00000000006da680 0000000000000000 00000000006da6e0 [ 1509.674569] 00000000006da680 ffffffff8100beeb 0000000000000246 00000000006da6c0 [ 1509.674578] Call Trace: [ 1509.674583] [<ffffffff810730ce>] sys_init_module+0x10e/0x200 [ 1509.674592] [<ffffffff8100beeb>] system_call_fastpath+0x16/0x1b [ 1509.674604] Code: 89 fb 48 83 ec 08 48 85 ff 74 44 bf 01 00 00 00 e8 25 d5 fc ff e8 d0 48 15 00 48 98 48 8b 93 f0 01 00 00 48 03 14 c5 20 70 67 81 <48> ff 0a 83 3b 02 74 23 bf 01 00 00 00 e8 6d d4 fc ff 65 48 8b [ 1509.674680] RIP [<ffffffff810710f1>] module_put+0x31/0x80 [ 1509.674689] RSP <ffff880119cebf38> [ 1509.674695] ---[ end trace 6aaa73f49bc73faf ]--- [ 1509.674701] note: modprobe[29571] exited with preempt_count 1 after i tried to unload drivers(dont remember nvidia ones or ath9k ones) machine became unresponsive, i had to raise elephants. after which new kernel did not reboot properly. recompiled, it from working kernel, relaoded machine, everything seesm to be working now however, i still get same issue. pwd /home/dimko/compat-wireless-2009-06-20 tcp dump -i wlan0 [start] ip link set wlan0 start iwconfig wlan0 essid dlink dmesg 761.433090] wlan0: authenticated [ 761.433094] wlan0: associate with AP 00:1e:58:b4:f6:83 [ 761.633264] wlan0: associate with AP 00:1e:58:b4:f6:83 [ 761.640078] wlan0: deauthenticated (Reason: 9) [ 762.640264] wlan0: direct probe to AP 00:1e:58:b4:f6:83 try 1 [ 762.643782] wlan0 direct probe responded [ 762.643787] wlan0: authenticate with AP 00:1e:58:b4:f6:83 [ 762.648350] wlan0: authenticated [ 762.648354] wlan0: associate with AP 00:1e:58:b4:f6:83 [ 762.848263] wlan0: associate with AP 00:1e:58:b4:f6:83 [ 762.855024] wlan0: deauthenticated (Reason: 9) i tcpdum [stop] tcpdump -i wlan0 tcpdump: WARNING: wlan0: no IPv4 address assigned tcpdump: verbose output suppressed, use -v or -vv for full protocol decode listening on wlan0, link-type EN10MB (Ethernet), capture size 96 bytes ^C 0 packets captured 0 packets received by filter 0 packets dropped by kernel Same problem persist... pstree -p init(1)-+-acpid(4077) |-agetty(4579) |-agetty(4580) |-agetty(4581) |-agetty(4582) |-bonobo-activati(5073)---{bonobo-activati}(5074) |-console-kit-dae(4607)-+-{console-kit-dae}(4611) | |-{console-kit-dae}(4615) | |-{console-kit-dae}(4616) | |-{console-kit-dae}(4617) | |-{console-kit-dae}(4618) | |-{console-kit-dae}(4620) | |-{console-kit-dae}(4621) | |-{console-kit-dae}(4622) | |-{console-kit-dae}(4623) | |-{console-kit-dae}(4624) | |-{console-kit-dae}(4625) | |-{console-kit-dae}(4626) | |-{console-kit-dae}(4627) | |-{console-kit-dae}(4628) | |-{console-kit-dae}(4629) | |-{console-kit-dae}(4630) | |-{console-kit-dae}(4631) | |-{console-kit-dae}(4632) | |-{console-kit-dae}(4633) | |-{console-kit-dae}(4634) | |-{console-kit-dae}(4635) | |-{console-kit-dae}(4636) | |-{console-kit-dae}(4637) | |-{console-kit-dae}(4638) | |-{console-kit-dae}(4639) | |-{console-kit-dae}(4640) | |-{console-kit-dae}(4641) | |-{console-kit-dae}(4642) | |-{console-kit-dae}(4643) | |-{console-kit-dae}(4644) | |-{console-kit-dae}(4645) | |-{console-kit-dae}(4646) | |-{console-kit-dae}(4647) | |-{console-kit-dae}(4648) | |-{console-kit-dae}(4649) | |-{console-kit-dae}(4650) | |-{console-kit-dae}(4651) | |-{console-kit-dae}(4652) | |-{console-kit-dae}(4653) | |-{console-kit-dae}(4654) | |-{console-kit-dae}(4655) | |-{console-kit-dae}(4656) | |-{console-kit-dae}(4657) | |-{console-kit-dae}(4658) | |-{console-kit-dae}(4659) | |-{console-kit-dae}(4660) | |-{console-kit-dae}(4661) | |-{console-kit-dae}(4662) | |-{console-kit-dae}(4663) | |-{console-kit-dae}(4664) | |-{console-kit-dae}(4665) | |-{console-kit-dae}(4666) | |-{console-kit-dae}(4667) | |-{console-kit-dae}(4668) | |-{console-kit-dae}(4669) | |-{console-kit-dae}(4670) | |-{console-kit-dae}(4671) | |-{console-kit-dae}(4672) | |-{console-kit-dae}(4673) | |-{console-kit-dae}(4877) | |-{console-kit-dae}(4926) | `-{console-kit-dae}(4983) |-dbus-daemon(4233) |-dbus-daemon(5021) |-dbus-launch(5020) |-dhcpcd(3192) |-exaile(5149) |-gconfd-2(5027) |-gdm(4979)---gdm(4980)-+-X(4982) | `-gnome-session(5005)-+-gnome-panel(5070) | |-metacity(5066) | |-nautilus(5077) | |-seahorse-agent(5035) | |-ssh-agent(5025) | |-vino-server(5084) | `-{gnome-session}(5047) |-gnome-dictionar(5098) |-gnome-keyring-d(5044) |-gnome-power-man(5103) |-gnome-screensav(5065) |-gnome-settings-(5048)---{gnome-settings-}(5052) |-gnome-terminal(5163)-+-bash(5166)---su(5172)---bash(5175) | |-bash(5218)---su(5223)---bash(5226)---pstree(52+ | |-gnome-pty-helpe(5165) | `-{gnome-terminal}(5168) |-gnome-volume-ma(5104) |-gvfs-hal-volume(5102) |-gvfsd(5037) |-gvfsd-burn(5119) |-gvfsd-trash(5107) |-hald(4292)---hald-runner(4293)-+-hald-addon-acpi(4307) | |-hald-addon-cpuf(4306) | |-hald-addon-inpu(4298) | `-hald-addon-stor(4317) |-login(4577)---bash(4878) |-login(4578)---bash(4921) |-mdadm(2738) |-mixer_applet2(5090)---{mixer_applet2}(5117) |-mozilla-launche(5121)---thunderbird-bin(5131)-+-{thunderbird-bin}(513+ | |-{thunderbird-bin}(513+ | |-{thunderbird-bin}(513+ | |-{thunderbird-bin}(513+ | |-{thunderbird-bin}(513+ | |-{thunderbird-bin}(514+ | |-{thunderbird-bin}(514+ | `-{thunderbird-bin}(514+ |-multiload-apple(5080) |-smartd(4507) |-syslog-ng(4018) `-udevd(932) lsmod Module Size Used by ath9k 245620 0 mac80211 188280 1 ath9k ath 9512 1 ath9k cfg80211 98080 3 ath9k,mac80211,ath rfkill_backport 21520 1 cfg80211 nvidia 8121160 36 f71882fg 28632 0 snd_seq 62016 0 output 3432 0 ext4 241640 1 jbd2 63000 1 ext4 crc16 1960 1 ext4 snd_virtuoso 14564 1 snd_oxygen_lib 34344 1 snd_virtuoso snd_pcm 88816 1 snd_oxygen_lib snd_timer 24536 2 snd_seq,snd_pcm snd_page_alloc 9976 1 snd_pcm snd_mpu401_uart 8424 1 snd_oxygen_lib snd_rawmidi 23336 1 snd_mpu401_uart snd_seq_device 7772 2 snd_seq,snd_rawmidi snd 60968 10 snd_seq,snd_virtuoso,snd_oxygen_lib,snd_pcm,snd_timer,snd_mpu401_uart,snd_rawmidi,snd_seq_device r8169 37100 0 k8temp 5608 0 hwmon 2920 2 f71882fg,k8temp soundcore 1648 1 snd i2c_nforce2 8336 0 i2c_core 27328 2 nvidia,i2c_nforce2

2 1

[alsa-devel] alc889
by Stefan Krastanov 22 Jun '09

22 Jun '09

Hi I would like to help (or at least help myself) with the creation of a model config for alc889. I'm not a sound developer. I'm using a laptop with alc889 based sound card. The reported pins are really messed up (it's hard not to use more profane expression). My laptop is Fujitsu-Siemens Amilo xi but the problems are the same as here: https://answers.launchpad.net/ubuntu/+source/alsa-driver/+question/71205 or here: http://mailman.alsa-project.org/pipermail/alsa-devel/2008-October/011295.ht… namely: 1.- no jack sensing (extremely annoying - for two years allready) 2.- misnamed mixers in volume control for gnome or alsamixer (example: front instead of headphone) 3.- the volume-control potentiometer on the side of the laptop is not working 4.- I need to mute two different channel at the same time (center and LFE) to mute one physical output (build-in sub-woofer) 5.- A great deal of other small annoyances destroying the user experience I'm using model 3stack-dig and the optical output is working (at least is lighted up - I do not have the hardware to check it) My questions are: - What I need to do, to create new model option based on 3stack-dig with different pin mapping? - Is it possible that my present configuration (3stack-dig) is using pins that should not be used? I'm asking in search of an explanation of problem 4. - What I need to do to rename the controls in alsamixer? - What to do to repair jack sensing? - that's the most important of all - What to do to repair the potentiometer for volume control? I have seen that hda-analyser can create diff files, but where should they be used. (And why is hda-analyser not working - I was forced to use http://ubuntuforums.org/showpost.php?p=7227626&postcount=33. Is it that much to expect working tools on the official website. I understand that small bug will always exist, but that one is there for at least 10 months) I won't be surprised if many of my questions are answered in the documentation, but I was unable to find anything useful. It would be enough to point me a place to start reading. I know you guys are working hard and you have done great things, but it's really hard to find something in the documentation(which is occasionally quite old) Best regards and thank you for the great work Stefan Krastanov

2 1

[alsa-devel] [PATCH] sound: remove redundant tests on unsigned
by Roel Kluin 22 Jun '09

22 Jun '09

reg and channel_id are unsigned. When negative they are wrapped and caught by the other test. Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- sound/pci/ca0106/ca0106_proc.c | 2 +- sound/pci/emu10k1/emu10k1x.c | 3 +-- sound/pci/emu10k1/emuproc.c | 2 +- sound/soc/codecs/tlv320aic23.c | 2 +- 4 files changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/pci/ca0106/ca0106_proc.c b/sound/pci/ca0106/ca0106_proc.c index c62b7d1..2deac54 100644 --- a/sound/pci/ca0106/ca0106_proc.c +++ b/sound/pci/ca0106/ca0106_proc.c @@ -304,7 +304,7 @@ static void snd_ca0106_proc_reg_write32(struct snd_info_entry *entry, while (!snd_info_get_line(buffer, line, sizeof(line))) { if (sscanf(line, "%x %x", &reg, &val) != 2) continue; - if ((reg < 0x40) && (reg >=0) && (val <= 0xffffffff) ) { + if ((reg < 0x40) && (val <= 0xffffffff)) { spin_lock_irqsave(&emu->emu_lock, flags); outl(val, emu->port + (reg & 0xfffffffc)); spin_unlock_irqrestore(&emu->emu_lock, flags); diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index 4d3ad79..957b130 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -1040,8 +1040,7 @@ static void snd_emu10k1x_proc_reg_write(struct snd_info_entry *entry, if (sscanf(line, "%x %x %x", &reg, &channel_id, &val) != 3) continue; - if ((reg < 0x49) && (reg >= 0) && (val <= 0xffffffff) - && (channel_id >= 0) && (channel_id <= 2) ) + if ((reg < 0x49) && (val <= 0xffffffff) && (channel_id <= 2)) snd_emu10k1x_ptr_write(emu, reg, channel_id, val); } } diff --git a/sound/pci/emu10k1/emuproc.c b/sound/pci/emu10k1/emuproc.c index 216f974..6ad3825 100644 --- a/sound/pci/emu10k1/emuproc.c +++ b/sound/pci/emu10k1/emuproc.c @@ -527,7 +527,7 @@ static void snd_emu_proc_ptr_reg_write(struct snd_info_entry *entry, while (!snd_info_get_line(buffer, line, sizeof(line))) { if (sscanf(line, "%x %x %x", &reg, &channel_id, &val) != 3) continue; - if ((reg < 0xa0) && (reg >= 0) && (val <= 0xffffffff) && (channel_id >= 0) && (channel_id <= 3) ) + if ((reg < 0xa0) && (val <= 0xffffffff) && (channel_id <= 3)) snd_ptr_write(emu, iobase, reg, channel_id, val); } } diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 0b8dcb5..6f8cd8d 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -85,7 +85,7 @@ static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg, * of data into val */ - if ((reg < 0 || reg > 9) && (reg != 15)) { + if ((reg > 9) && (reg != 15)) { printk(KERN_WARNING "%s Invalid register R%u\n", __func__, reg); return -1; }

2 1

Re: [alsa-devel] [PATCH] New ASoC Drivers for ADI AD1938 codec
by Mark Brown 22 Jun '09

22 Jun '09

On Fri, Jun 19, 2009 at 05:28:15PM +0800, Barry Song wrote: > 1. add AD1938 codec driver (codec) > 2. add blackfin SPORT-TDM DAI and PCM driver (platform) > 3. add bf5xx board with AD1938 driver (machine) As Liam said you really need to submit this as a patch series rather than as a single big patch - as your commit log here indicates you've got several different things going on here. > +++ b/include/sound/soc-dai.h > @@ -30,6 +30,7 @@ struct snd_pcm_substream; > #define SND_SOC_DAIFMT_DSP_A 3 /* L data msb after FRM LRC */ > #define SND_SOC_DAIFMT_DSP_B 4 /* L data msb during FRM LRC */ > #define SND_SOC_DAIFMT_AC97 5 /* AC97 */ > +#define SND_SOC_DAIFMT_SPORT_TDM 6 /* SPORT TDM for ADI parts */ If you're going to add a new DAI format that really needs more explanation than this explaining what the DAI format is. It'd be very surprising to see hardware needing a new format. Looking at the datasheet for the ad1938 it appears that the actual format here is just normal I2S with TDM. This does not need a new DAI format or new CPU DAI, you just need to add suport for TDM to the Blackfin I2S driver. The format is fairly standard and implemented by a number of other devices. See set_tdm_slot() for setting up the higher channel counts - there's some ongoing revisions to that API so you'll want to also ensure that the code is set up so that it can cope with specification of the sample width for each slot in set_tdm_slot(). Given this I've only looked at the CODEC driver below. > diff --git a/sound/soc/codecs/ad1938.c b/sound/soc/codecs/ad1938.c > new file mode 100644 > index 0000000..9aa78e1 > --- /dev/null > +++ b/sound/soc/codecs/ad1938.c > + * > + * Revision history > + * 4 June 2009 Initial version. Don't include this, git provides code history for us. > +struct snd_soc_device *ad1938_socdev; > + > +/* dac de-emphasis enum control */ > +static const char *ad1938_deemp[] = {"flat", "48kHz", "44.1kHz", "32kHz"}; For consistency with other drivers "flat" should be "None". > +/* AD1938 volume/mute/de-emphasis etc. controls */ > +static const struct snd_kcontrol_new ad1938_snd_controls[] = { > + /* DAC volume control */ > + SOC_SINGLE("DAC L1 Volume", AD1938_DAC_L1_VOL, 0, 0xFF, 1), > + SOC_SINGLE("DAC R1 Volume", AD1938_DAC_R1_VOL, 0, 0xFF, 1), These (and the other stereo pairs below) should be SOC_DOUBLE_R(). This allows ALSA to represent them as stereo controls to applications rather than as two separate controls. You should also provide TLV information so actually SOC_DOUBLE_R_TLV() if possible. > + /* DAC mute control */ > + SOC_SINGLE("DAC L1 Switch", AD1938_DAC_CHNL_MUTE, 0, 1, 1), > + SOC_SINGLE("DAC R1 Switch", AD1938_DAC_CHNL_MUTE, 1, 1, 1), These should be stereo controls too - SOC_DOUBLE() since they're in the same register. > + /* ADC mute control */ > + SOC_SINGLE("ADC L1 Switch", AD1938_ADC_CTRL0, ADC0_MUTE, 1, 1), > + SOC_SINGLE("ADC R1 Switch", AD1938_ADC_CTRL0, ADC1_MUTE, 1, 1), These too. > + /* DAC de-emphasis */ > + SOC_ENUM("Playback Deemphasis", ad1938_enum[0]), Don't put your enums in an array, use named variables for them. This makes drivers easier to maintian when you get a lot of enums. > +static int ad1938_add_controls(struct snd_soc_codec *codec) > +{ > + int err, i; > + > + for (i = 0; i < ARRAY_SIZE(ad1938_snd_controls); i++) { > + err = snd_ctl_add(codec->card, > + snd_soc_cnew(&ad1938_snd_controls[i], codec, NULL)); Use snd_soc_add_controls() here - you can replace the entire function with a call to that. > +/* dac/adc/pll poweron/off functions */ > +static int ad1938_dac_powerctrl(struct snd_soc_codec *codec, int cmd) > +{ > + int reg; > + > + reg = codec->read(codec, AD1938_DAC_CTRL0); > + if (cmd) > + reg &= ~DAC_POWERDOWN; > + else > + reg |= DAC_POWERDOWN; > + codec->write(codec, AD1938_DAC_CTRL0, reg); This should be handled by DAPM - either have a single DAC widget representing all the channels (since you don't appear to have independant control anyway) or have a bunch of dummy DAC widgets and a supply widget representing the actual power control. The same thing applies to the ADCs. > +static int ad1938_set_pll(struct snd_soc_dai *codec_dai, > + int pll_id, unsigned int freq_in, unsigned int freq_out) > +{ > + struct snd_soc_codec *codec = codec_dai->codec; > + > + if (freq_out) > + ad1938_pll_powerctrl(codec, 1); > + else { > + /* playing while recording, framework will poweroff-poweron pll redundantly */ > + if ((!codec_dai->capture.active) && (!codec_dai->playback.active)) > + ad1938_pll_powerctrl(codec, 0); > + } Hrm. This appears to completely ignore the frequencies supplied for the PLL and just provide power control. I suspect that you can just handle the PLL as a SND_SOC_DAPM_SUPPLY(), there seems to be no need to expose the set_pll() operation and make machine drivers call it given that there isn't any frequency configuration going on. > +static int ad1938_mute(struct snd_soc_dai *dai, int mute) > +{ > + struct snd_soc_codec *codec = dai->codec; > + > + if (!mute) > + codec->write(codec, AD1938_DAC_CHNL_MUTE, 0); > + else > + codec->write(codec, AD1938_DAC_CHNL_MUTE, 0xff); > + > + return 0; > +} This isn't going to play well with the explicit mute controls you've got above - it's writing to the same register bits without any coordination. One or the other set of controls ought to be removed. > +static int ad1938_tdm_set(struct snd_soc_codec *codec) > +{ > + codec->write(codec, AD1938_DAC_CTRL0, (codec->read(codec, AD1938_DAC_CTRL0) & > + (~DAC_SERFMT_MASK)) | DAC_SERFMT_TDM); > + codec->write(codec, AD1938_DAC_CTRL1, 0x84); /* invert bclk, 256bclk/frame, latch in mid */ > + codec->write(codec, AD1938_ADC_CTRL1, 0x43); /* sata delay=1, adc aux mode */ > + codec->write(codec, AD1938_ADC_CTRL2, 0x6F); /* left high, driver on rising edge */ > + > + return 0; > +} If you use set_tdm_slot() then the BCLK/frame ratio will be set by that. Inversion of BCLK (and any other clocks) should be handled by the set_dai_fmt() operation based on the machine driver request rather than done unconditionally. > + /* bit size */ > + switch (params_format(params)) { > + case SNDRV_PCM_FORMAT_S16_LE: > + word_len = 3; > + break; Once you implement set_tdm_slot() you should allow the word length to be configured there if it's called or otherwise keep this code here - see Daniel Ribeiro's patche "change set_tdm_slot api to allow slot_width override" posted to the ALSA list this week. > +static int __devinit ad1938_spi_probe(struct spi_device *spi) > +{ > + spi->dev.power.power_state = PMSG_ON; > + ad1938_socdev->card->codec->control_data = spi; > + > + return 0; > +} > + > +static int __devexit ad1938_spi_remove(struct spi_device *spi) > +{ > + return 0; > +} Your device probing should all be restructured so that the SPI device for the CODEC is registered as any other SPI device rather than being set up as part of probing the ASoC device. See the wm8731 driver for an example of doing this for a SPI device. This will require that the arch code for any systems with the ad1938 do the setup of the device. > + .name = "AD1938", > + .playback = { > + .stream_name = "Playback", > + .channels_min = 2, > + .channels_max = 8, > + .rates = SNDRV_PCM_RATE_48000, > + .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, }, Please keep your lines to under 80 columns. > +#define AD1938_PLL_CLK_CTRL0 0 > +#define PLL_POWERDOWN 0x01 > +#define AD1938_PLL_CLK_CTRL1 1 > +#define AD1938_DAC_CTRL0 2 > +#define DAC_POWERDOWN 0x01 > +#define DAC_SERFMT_MASK 0xC0 > +#define DAC_SERFMT_STEREO (0 << 6) > +#define DAC_SERFMT_TDM (1 << 6) These defines need namespacing if they're going to appear in the headers - everything should have the AD1938_ prefix.

4 10

[PATCH] ASoC: Add a shutdown callback
by Mark Brown 22 Jun '09

22 Jun '09

Ensure that the audio subsystem is powered down cleanly when the system shuts down by providing a shutdown operation. This ensures that all the components have been returned to an off state cleanly which should avoid audio issues from partially charged capacitors or noise on digital inputs if the system is restarted quickly. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 1 + sound/soc/soc-core.c | 16 ++++++++++++++++ sound/soc/soc-dapm.c | 29 +++++++++++++++++++++++++++++ 3 files changed, 46 insertions(+), 0 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index ec8a45f..35814ce 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -279,6 +279,7 @@ int snd_soc_dapm_add_routes(struct snd_soc_codec *codec, /* dapm events */ int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, char *stream, int event); +void snd_soc_dapm_shutdown(struct snd_soc_device *socdev); /* dapm sys fs - used by the core */ int snd_soc_dapm_sys_add(struct device *dev); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4414117..55d45c4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1020,6 +1020,21 @@ static int soc_remove(struct platform_device *pdev) return 0; } +static void soc_shutdown(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_card *card = socdev->card; + + if (!card->instantiated) + return; + + /* Flush out pmdown_time work - we actually do want to run it + * now, we're shutting down so no imminent restart. */ + run_delayed_work(&card->delayed_work); + + snd_soc_dapm_shutdown(socdev); +} + /* ASoC platform driver */ static struct platform_driver soc_driver = { .driver = { @@ -1030,6 +1045,7 @@ static struct platform_driver soc_driver = { .remove = soc_remove, .suspend = soc_suspend, .resume = soc_resume, + .shutdown = soc_shutdown, }; /* create a new pcm */ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 6534359..b9129ef 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2032,6 +2032,35 @@ void snd_soc_dapm_free(struct snd_soc_device *socdev) } EXPORT_SYMBOL_GPL(snd_soc_dapm_free); +/* + * snd_soc_dapm_shutdown - callback for system shutdown + */ +void snd_soc_dapm_shutdown(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->card->codec; + struct snd_soc_dapm_widget *w; + LIST_HEAD(down_list); + int powerdown = 0; + + list_for_each_entry(w, &codec->dapm_widgets, list) { + if (w->power) { + dapm_seq_insert(w, &down_list, dapm_down_seq); + powerdown = 1; + } + } + + /* If there were no widgets to power down we're already in + * standby. + */ + if (powerdown) { + snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_PREPARE); + dapm_seq_run(codec, &down_list, 0, dapm_down_seq); + snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_STANDBY); + } + + snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_OFF); +} + /* Module information */ MODULE_AUTHOR("Liam Girdwood, lrg(a)slimlogic.co.uk"); MODULE_DESCRIPTION("Dynamic Audio Power Management core for ALSA SoC"); -- 1.6.3.1

1 0

[alsa-devel] [PATCH 2/3] ASoC: TWL4030: Add AVADC Clock Priority
by Lopez Cruz, Misael 22 Jun '09

22 Jun '09

AVDAC clk priority allows to determine the path ADC must be connected when the codec is in option2 and both audio and voice paths are enabled. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl4030.c | 12 ++++++++++++ 1 files changed, 12 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 780217a..fd5b30f 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1036,6 +1036,16 @@ static DECLARE_TLV_DB_SCALE(digital_capture_tlv, 0, 100, 0); */ static DECLARE_TLV_DB_SCALE(input_gain_tlv, 0, 600, 0); +/* AVADC clock priority */ +static const char *twl4030_avadc_clk_priority_texts[] = { + "Voice high priority", "Audio high priority" +}; + +static const struct soc_enum twl4030_avadc_clk_priority_enum = + SOC_ENUM_SINGLE(TWL4030_REG_AVADC_CTL, 2, + ARRAY_SIZE(twl4030_avadc_clk_priority_texts), + twl4030_avadc_clk_priority_texts); + static const char *twl4030_rampdelay_texts[] = { "27/20/14 ms", "55/40/27 ms", "109/81/55 ms", "218/161/109 ms", "437/323/218 ms", "874/645/437 ms", "1748/1291/874 ms", @@ -1137,6 +1147,8 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = { SOC_DOUBLE_TLV("Analog Capture Volume", TWL4030_REG_ANAMIC_GAIN, 0, 3, 5, 0, input_gain_tlv), + SOC_ENUM("AVADC Clock Priority", twl4030_avadc_clk_priority_enum), + SOC_ENUM("HS ramp delay", twl4030_rampdelay_enum), SOC_ENUM("Vibra H-bridge mode", twl4030_vibradirmode_enum), -- 1.5.4.3

3 3

[alsa-devel] ASoC updates for 2.6.31
by Mark Brown 22 Jun '09

22 Jun '09

The following changes since commit ef39412622b6e8f09c383de9565b07e93553fc27: Takashi Iwai (1): ASoC: Kill BUS_ID_SIZE are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31 Barry Song (1): ASoC: Blackfin: update the bf5xx_i2s_resume parameters Cliff Cai (1): ASoC: Blackfin: keep better track of SPORT configuration state sound/soc/blackfin/bf5xx-i2s.c | 10 +++++++--- 1 files changed, 7 insertions(+), 3 deletions(-)

2 1

[alsa-devel] [PATCH] via82xx: add option to disable 500ms delay in snd_via82xx_codec_wait
by Simon Arlott 22 Jun '09

22 Jun '09

There's a large 500ms delay in snd_via82xx_codec_wait() that, at least on my hardware, appears to be unnecessary. The rest of the init of the card works without logging any warnings or errors and both audio and mixer settings work. This adds an "nodelay" parameter to disable this (undocumented in the code) large delay improving bootup time by 489-500ms. [ 1.034217] initcall alsa_card_via82xx_init+0x0/0x16 returned 0 after 505757 usecs vs. [ 0.533136] initcall alsa_card_via82xx_init+0x0/0x16 returned 0 after 15915 usecs Signed-off-by: Simon Arlott <simon(a)fire.lp0.eu> --- sound/pci/via82xx.c | 6 +++++- 1 files changed, 5 insertions(+), 1 deletions(-) diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 1ef58c5..949fcaf 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -85,6 +85,7 @@ static int joystick; static int ac97_clock = 48000; static char *ac97_quirk; static int dxs_support; +static int nodelay; module_param(index, int, 0444); MODULE_PARM_DESC(index, "Index value for VIA 82xx bridge."); @@ -102,6 +103,8 @@ module_param(ac97_quirk, charp, 0444); MODULE_PARM_DESC(ac97_quirk, "AC'97 workaround for strange hardware."); module_param(dxs_support, int, 0444); MODULE_PARM_DESC(dxs_support, "Support for DXS channels (0 = auto, 1 = enable, 2 = disable, 3 = 48k only, 4 = no VRA, 5 = enable any sample rate)"); +module_param(nodelay, int, 0444); +MODULE_PARM_DESC(nodelay, "Disable 500ms init delay"); /* just for backward compatibility */ static int enable; @@ -549,7 +552,8 @@ static void snd_via82xx_codec_wait(struct snd_ac97 *ac97) int err; err = snd_via82xx_codec_ready(chip, ac97->num); /* here we need to wait fairly for long time.. */ - msleep(500); + if (!nodelay) + msleep(500); } static void snd_via82xx_codec_write(struct snd_ac97 *ac97, -- 1.6.3.1 -- Simon Arlott

2 1

[alsa-devel] Need help on MMAP
by Christopher Tsui 22 Jun '09

22 Jun '09

Hi I am in the process of add mmap support in my ALSA driver. So this is what i have done so far In snd_pcm_hardware structure .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID In snd_pcm_ops structure .mmap = my_pcm_mmap So to test my code, I use aplay -M -Dhw:0 rock.wav and this is what I get ALSA lib pcm_mmap.c:378:(snd_pcm_mmap) mmap failed: Invalid argument aplay: set_params:1022: Unable to install hw params: ACCESS: MMAP_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 32 CHANNELS: 2 RATE: 44100 PERIOD_TIME: (27210 27211) PERIOD_SIZE: 1200 PERIOD_BYTES: 4800 PERIODS: 4 BUFFER_TIME: (108843 108844) BUFFER_SIZE: 4800 BUFFER_BYTES: 19200 TICK_TIME: 10000 What am I doing wrong?? Thanks in advance

1 0

[alsa-devel] Acer 7730G support for ALSA.
by Denis Kup 21 Jun '09

21 Jun '09

Hey, when i first installed alsa to my laptop, there no sound from subwoofer, jacks were map very strange (2 surrounds for ex.). I try all models from the list, but all don't support my subwoofer. After a weak of hard work i made a patch that enables ACER_ASPIRE_7730G model for my laptop. It have ALC888S, 3 jacks, 6 channels, stereo internal frontal speakers and internal subwoofer.You can see acer.com for more info about hardware and jacks. It's the changes i made to /alsa-kernel/pci/hda/patch_realtek.c: > enum { > ................... > ALC888_ACER_ASPIRE_7730G, > ................... > }; > ////////////////////////////////////////////////////////////// > static struct hda_verb alc888_acer_aspire_7730G_verbs[] = { > {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /*CHANGE DIRECT FRONT TO BLACK JACK*/ > {0x17, AC_VERB_SET_CONNECT_SEL, 0x02}, /* CHANGE: ENABLES SUBWOOFER ON 7730*/ > {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, //CHANGE FOR AUTOMUTE > { } /* end */ > }; > > /*CHANGE ENABLE AUTOMUTING ON ACER 7730G > ////////////////////////////////////////////////////// > //////////////////////////////////////////////////////*/ > > static void alc888_acer_aspire_7730G_front_automute(struct hda_codec *codec) > { > unsigned int present; > > present = snd_hda_codec_read(codec, 0x15, 0, > AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; > snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, > HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); > snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 0, > HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); > } > > static void alc888_acer_aspire_7730G_unsol_event(struct hda_codec *codec, > unsigned int res) > { > if ((res >> 26) == ALC880_HP_EVENT) > alc888_acer_aspire_7730G_front_automute(codec); > } > > /*//////////////////////////////////////////////////// > ////////////////////////////////////////////////// > //////////////////////////////////////////////////////*/ > static struct alc_config_preset alc883_presets[] = { > > [ALC888_ACER_ASPIRE_7730G] = { > .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer }, > .init_verbs = { alc883_init_verbs, alc888_acer_aspire_7730G_verbs }, > .num_dacs = ARRAY_SIZE(alc883_dac_nids), > .dac_nids = alc883_dac_nids, > .dig_out_nid = ALC883_DIGOUT_NID, > .dig_in_nid = ALC883_DIGIN_NID, > .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes), > .channel_mode = alc883_3ST_6ch_modes, > .need_dac_fix = 1, > .input_mux = &alc883_capture_source, > .unsol_event = alc888_acer_aspire_7730G_unsol_event, /* CHANGE AUTOMUTE DECLARATION*/ > .init_hook = alc888_acer_aspire_7730G_front_automute, > }, > > /////////////////////////////////////////////////////////////// > > static const char *alc883_models[ALC883_MODEL_LAST] = { > ................................. > [ALC888_ACER_ASPIRE_7730G] = "acer-aspire-7730g", > ......................... > > //////////////////////////////// > static struct snd_pci_quirk alc883_cfg_tbl[] = { > ...................................... > SND_PCI_QUIRK(0x1025, 0x0142, "Acer Aspire 7730G", ALC888_ACER_ASPIRE_7730G), > .......................................... After changing file i install driver, and it automatically detect its model (maybe there are old settings from previous install, so i dont 100% sure that it was detected). With this settings subwoofer works perfect, internal stereo and subwoofer mute when headphone are plugged all works, except 1 thing: subwoofer uses center channel instead LFE. To fix it i use such asound.conf file: > pcm.upmix_20to51 { > type plug > slave.pcm lowpass_21to21 > slave.channels 3 > ttable { > 0.0 1 # left channel > 1.1 1 # right channel > 0.2 1 # mix left and right ... > 1.2 1 # ... channel for subwoofer > } > } > pcm.lowpass_21to21 { > type ladspa > slave.pcm upmix_21to51 > path "/usr/lib/ladspa" > channels 3 > plugins { > 0 { > id 1098 # Identity (Audio) (1098/identity_audio) > policy duplicate > input.bindings.0 "Input"; > output.bindings.0 "Output"; > } > 1 { > id 1671 # 4 Pole Low-Pass Filter with Resonance (FCRCIA) (1672/lp4pole_fcrcia_oa) > policy none > input.bindings.2 "Input"; > output.bindings.2 "Output"; > input { > controls [ 1000 2 ] > } > } > } > } > pcm.upmix_21to51 { > type plug > slave.pcm surround51 > slave.channels 6 > ttable { > 0.0 1 # front left > 1.1 1 # front right > 0.2 1 # rear left > 1.3 1 # rear right > 0.5 0.5 # swap channels > 1.5 0.5 > 2.4 1 > } > } > > pcm.swapCLFE_51 { > type plug > slave.pcm surround51 > slave.channels 6 > ttable { > 0.0 1 # front left > 1.1 1 # front right > 2.2 1 # rear left > 3.3 1 # rear right > 4.5 1 # swap channels > 5.4 1 > } > } I use upmix_21to51 and swapCLFE_51 devices in Amarok settings. Is there any chance to swap Center and LFE in patch_realtek.c? And can my 7730g config be used in future version of alsa? And how can I detect params for snd_pci_quirk alc883_cfg_tbl ? And some config settings for my laptop, if needed: > linux-wpg9:/home/dener # cat /proc/asound/card0/codec* > Codec: Realtek ALC888 > Address: 0 > Function Id: 0x1 > Vendor Id: 0x10ec0888 > Subsystem Id: 0x10250142 > Revision Id: 0x100202 > No Modem Function Group found > Default PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Default Amp-In caps: N/A > Default Amp-Out caps: N/A > GPIO: io=2, o=0, i=0, unsolicited=1, wake=1 > IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > Node 0x02 [Audio Output] wcaps 0x411: Stereo > Converter: stream=5, channel=0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x03 [Audio Output] wcaps 0x411: Stereo > Converter: stream=5, channel=0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x04 [Audio Output] wcaps 0x411: Stereo > Converter: stream=5, channel=0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x05 [Audio Output] wcaps 0x411: Stereo > Converter: stream=0, channel=0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x06 [Audio Output] wcaps 0x611: Stereo Digital > Converter: stream=0, channel=0 > Digital: Enabled GenLevel > Digital category: 0x2 > PCM: > rates [0x5e0]: 44100 48000 88200 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x08 [Audio Input] wcaps 0x10051b: Stereo Amp-In > Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 > Amp-In vals: [0x16 0x16] > Converter: stream=0, channel=0 > SDI-Select: 0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Connection: 1 > 0x23 > Node 0x09 [Audio Input] wcaps 0x10051b: Stereo Amp-In > Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 > Amp-In vals: [0x9f 0x9f] > Converter: stream=0, channel=0 > SDI-Select: 0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Connection: 1 > 0x22 > Node 0x0a [Audio Input] wcaps 0x100711: Stereo Digital > Converter: stream=0, channel=0 > SDI-Select: 0 > Digital: > Digital category: 0x0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Connection: 1 > 0x1f > Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 > Amp-In vals: [0x80 0x80] [0x1f 0x1f] [0x00 0x00] [0x80 0x80] [0x17 0x17] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] > Connection: 10 > 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 > Node 0x0c [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x00 0x00] [0x00 0x00] > Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 > Amp-Out vals: [0x12 0x12] > Connection: 2 > 0x02 0x0b > Node 0x0d [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x00 0x00] [0x00 0x00] > Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 > Amp-Out vals: [0x15 0x15] > Connection: 2 > 0x03 0x0b > Node 0x0e [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x00 0x00] [0x00 0x00] > Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 > Amp-Out vals: [0x15 0x15] > Connection: 2 > 0x04 0x0b > Node 0x0f [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x80 0x80] [0x80 0x80] > Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 > Amp-Out vals: [0x00 0x00] > Connection: 2 > 0x05 0x0b > Node 0x10 [Audio Output] wcaps 0x611: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > PCM: > rates [0x5e0]: 44100 48000 88200 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x11 [Pin Complex] wcaps 0x400780: Mono Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 1 > 0x10 > Node 0x12 [Pin Complex] wcaps 0x400401: Stereo > Pincap 0x00000020: IN > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Power: setting=D0, actual=D0 > Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x14 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x0001003e: IN OUT HP EAPD Detect Trigger > EAPD 0x0: > Pin Default 0x99130110: [Fixed] Speaker at Int ATAPI > Conn = ATAPI, Color = Unknown > DefAssociation = 0x1, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c* 0x0d 0x0e 0x0f 0x26 > Node 0x15 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x0001003e: IN OUT HP EAPD Detect Trigger > EAPD 0x0: > Pin Default 0x0221101f: [Jack] HP Out at Ext Front > Conn = 1/8, Color = Black > DefAssociation = 0x1, Sequence = 0xf > Pin-ctls: 0x40: OUT > Unsolicited: tag=04, enabled=1 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c* 0x0d 0x0e 0x0f 0x26 > Node 0x16 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x00000036: IN OUT Detect Trigger > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c 0x0d 0x0e* 0x0f 0x26 > Node 0x17 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x00000036: IN OUT Detect Trigger > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c 0x0d 0x0e* 0x0f 0x26 > Node 0x18 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x0000373e: IN OUT HP Detect Trigger > Vref caps: HIZ 50 GRD 80 100 > Pin Default 0x02a19c30: [Jack] Mic at Ext Front > Conn = 1/8, Color = Pink > DefAssociation = 0x3, Sequence = 0x0 > Pin-ctls: 0x40: OUT VREF_HIZ > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c 0x0d 0x0e* 0x0f 0x26 > Node 0x19 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x80 0x80] > Pincap 0x0000373e: IN OUT HP Detect Trigger > Vref caps: HIZ 50 GRD 80 100 > Pin Default 0x99a30931: [Fixed] Mic at Int ATAPI > Conn = ATAPI, Color = Unknown > DefAssociation = 0x3, Sequence = 0x1 > Misc = NO_PRESENCE > Pin-ctls: 0x24: IN VREF_80 > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c* 0x0d 0x0e 0x0f 0x26 > Node 0x1a [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x0000373e: IN OUT HP Detect Trigger > Vref caps: HIZ 50 GRD 80 100 > Pin Default 0x0281343f: [Jack] Line In at Ext Front > Conn = 1/8, Color = Blue > DefAssociation = 0x3, Sequence = 0xf > Pin-ctls: 0x40: OUT VREF_HIZ > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c 0x0d* 0x0e 0x0f 0x26 > Node 0x1b [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x0000373e: IN OUT HP Detect Trigger > Vref caps: HIZ 50 GRD 80 100 > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0xc0: OUT HP VREF_HIZ > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c* 0x0d 0x0e 0x0f 0x26 > Node 0x1c [Pin Complex] wcaps 0x400481: Stereo > Pincap 0x00000024: IN Detect > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x20: IN > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Node 0x1d [Pin Complex] wcaps 0x400400: Mono > Pincap 0x00000020: IN > Pin Default 0x4016852d: [N/A] Speaker at Ext N/A > Conn = Digital, Color = Purple > DefAssociation = 0x2, Sequence = 0xd > Misc = NO_PRESENCE > Pin-ctls: 0x20: IN > Power: setting=D0, actual=D0 > Node 0x1e [Pin Complex] wcaps 0x400780: Mono Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x02451120: [Jack] SPDIF Out at Ext Front > Conn = Optical, Color = Black > DefAssociation = 0x2, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 1 > 0x06 > Node 0x1f [Pin Complex] wcaps 0x400680: Mono Digital > Pincap 0x00000024: IN Detect > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x20: IN > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono > Processing caps: benign=0, ncoeff=25 > Processing Coefficient: 0x00 > Coefficient Index: 0x09 > Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x80 0x80] [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] > Connection: 12 > 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b 0x12 > Node 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] > Connection: 11 > 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b > Node 0x24 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x25 [Audio Output] wcaps 0x411: Stereo > Converter: stream=0, channel=0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x26 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x00 0x00] [0x80 0x80] > Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 > Amp-Out vals: [0x00 0x00] > Connection: 2 > 0x25 0x0b > Codec: LSI ID 1040 > Address: 1 > Function Id: 0x2 > Vendor Id: 0x11c11040 > Subsystem Id: 0x1025013d > Revision Id: 0x100200 > Modem Function Group: 0x1 > Codec: Nvidia MCP78 HDMI > Address: 3 > Function Id: 0x1 > Vendor Id: 0x10de0006 > Subsystem Id: 0x10de0101 > Revision Id: 0x100000 > No Modem Function Group found > Default PCM: > rates [0x0]: > bits [0x0]: > formats [0x0]: > Default Amp-In caps: N/A > Default Amp-Out caps: N/A > GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 > Node 0x04 [Audio Output] wcaps 0x211: Stereo Digital > Converter: stream=0, channel=0 > Digital: Enabled > Digital category: 0x0 > PCM: > rates [0xc0]: 48000 88200 > bits [0xf]: 8 16 20 24 > formats [0x1]: PCM > Node 0x05 [Pin Complex] wcaps 0x400381: Stereo Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x18560110: [Jack] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0x1, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Connection: 1 > 0x04 > Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > PCM: > rates [0xc0]: 48000 88200 > bits [0xf]: 8 16 20 24 > formats [0x1]: PCM > Node 0x07 [Pin Complex] wcaps 0x400381: Stereo Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x58560121: [N/A] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0x2, Sequence = 0x1 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Unsolicited: tag=00, enabled=0 > Connection: 1 > 0x06 > Node 0x08 [Audio Output] wcaps 0x211: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > PCM: > rates [0xc0]: 48000 88200 > bits [0xf]: 8 16 20 24 > formats [0x1]: PCM > Node 0x09 [Pin Complex] wcaps 0x400381: Stereo Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x58560122: [N/A] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0x2, Sequence = 0x2 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Unsolicited: tag=00, enabled=0 > Connection: 1 > 0x08 > Node 0x0a [Audio Output] wcaps 0x211: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > PCM: > rates [0xc0]: 48000 88200 > bits [0xf]: 8 16 20 24 > formats [0x1]: PCM > Node 0x0b [Pin Complex] wcaps 0x400381: Stereo Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x58560123: [N/A] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0x2, Sequence = 0x3 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Unsolicited: tag=00, enabled=0 > Connection: 1 > 0x0a > Node 0x0c [Audio Output] wcaps 0x211: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > PCM: > rates [0xc0]: 48000 88200 > bits [0xf]: 8 16 20 24 > formats [0x1]: PCM > Node 0x0d [Pin Complex] wcaps 0x400381: Stereo Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x58560124: [N/A] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0x2, Sequence = 0x4 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Unsolicited: tag=00, enabled=0 > Connection: 1 > 0x0c > > linux-wpg9:/home/dener # cat /proc/asound/version > Advanced Linux Sound Architecture Driver Version 1.0.20. > Compiled on Jun 14 2009 for kernel 2.6.27.23-0.1-pae (SMP). > > cat /proc/asound/devices > 0: [ 0] : control > 1: : sequencer > 4: [ 0- 0]: hardware dependent > 5: [ 0- 1]: hardware dependent > 7: [ 0- 3]: hardware dependent > 16: [ 0- 0]: digital audio playback > 17: [ 0- 1]: digital audio playback > 19: [ 0- 3]: digital audio playback > 24: [ 0- 0]: digital audio capture > 25: [ 0- 1]: digital audio capture > 26: [ 0- 2]: digital audio capture > 33: : timer > > cat /proc/asound/pcm > 00-00: ALC888 Analog : ALC888 Analog : playback 1 : capture 1 > 00-01: ALC888 Digital : ALC888 Digital : playback 1 : capture 1 > 00-02: ALC888 Analog : ALC888 Analog : capture 1 > 00-03: NVIDIA HDMI : NVIDIA HDMI : playback 1 > > lspci | grep -i audio > 00:1b.0 Audio device: Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 03) > I have 1.0.20 alsa version and OpenSUSE 11.1 distributive. I hope my materials are usefull PS I am from Russia, sorry for my bad English :)

1 0

[alsa-devel] [PATCH] Acer Inspire 6530G model for Realtek ALC888
by Tony Vroon 21 Jun '09

21 Jun '09

The selected 4930G model seemed to keep the subwoofer 'tuba' function from operating correctly. Removing the existing PCI ID match made this work again, but it was mapped to 'Side' instead of to LFE as one would expect. This attempts to enable all functionality and keep the amount of available mixer sliders low. Any slider that had no audible effect on the output audio has been removed, and as such EAPD is not currently enabled. Signed-off-by: Tony Vroon <tony(a)linx.net> --- linux-2.6.30-git14/sound/pci/hda/patch_realtek.c.orig 2009-06-20 21:55:53.000000000 +0100 +++ linux-2.6.30-git14/sound/pci/hda/patch_realtek.c 2009-06-21 00:19:53.000000000 +0100 @@ -224,6 +224,7 @@ ALC883_ACER, ALC883_ACER_ASPIRE, ALC888_ACER_ASPIRE_4930G, + ALC888_ACER_ASPIRE_6530G, ALC888_ACER_ASPIRE_8930G, ALC883_MEDION, ALC883_MEDION_MD2, @@ -1471,6 +1472,25 @@ }; /* + * ALC888 Acer Aspire 6530G model + */ + +static struct hda_verb alc888_acer_aspire_6530g_verbs[] = { +/* Bias voltage on for external mic port */ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN | PIN_VREF80}, +/* Enable unsolicited event for HP jack */ + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, +/* Enable speaker output */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, +/* Enable headphone output */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | PIN_HP}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + { } +}; + +/* * ALC889 Acer Aspire 8930G model */ @@ -1544,6 +1564,25 @@ } }; +static struct hda_input_mux alc888_acer_aspire_6530_sources[2] = { + /* Interal mic only available on one ADC */ + { + .num_items = 3, + .items = { + { "Ext Mic", 0x0 }, + { "CD", 0x4 }, + { "Int Mic", 0xb }, + }, + }, + { + .num_items = 2, + .items = { + { "Ext Mic", 0x0 }, + { "CD", 0x4 }, + }, + } +}; + static struct hda_input_mux alc889_capture_sources[3] = { /* Digital mic only available on first "ADC" */ { @@ -8153,6 +8192,19 @@ { } /* end */ }; +static struct snd_kcontrol_new alc888_acer_aspire_6530_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + 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("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), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + { } /* end */ +}; + static struct snd_kcontrol_new alc888_lenovo_sky_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -9021,7 +9073,7 @@ SND_PCI_QUIRK(0x1025, 0x015e, "Acer Aspire 6930G", ALC888_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0x0166, "Acer Aspire 6530G", - ALC888_ACER_ASPIRE_4930G), + ALC888_ACER_ASPIRE_6530G), /* default Acer -- disabled as it causes more problems. * model=auto should work fine now */ @@ -9255,6 +9307,24 @@ .unsol_event = alc_automute_amp_unsol_event, .init_hook = alc888_acer_aspire_4930g_init_hook, }, + [ALC888_ACER_ASPIRE_6530G] = { + .mixers = { alc888_acer_aspire_6530_mixer }, + .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs, + alc888_acer_aspire_6530g_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev), + .adc_nids = alc883_adc_nids_rev, + .capsrc_nids = alc883_capsrc_nids_rev, + .dig_out_nid = ALC883_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), + .channel_mode = alc883_3ST_2ch_modes, + .num_mux_defs = + ARRAY_SIZE(alc888_2_capture_sources), + .input_mux = alc888_acer_aspire_6530_sources, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc888_acer_aspire_4930g_init_hook, + }, [ALC888_ACER_ASPIRE_8930G] = { .mixers = { alc888_base_mixer, alc883_chmode_mixer },

2 1

[alsa-devel] [PATCH] ASoC: Blackfin: convert internal names from bf52x to bf5xx
by Mike Frysinger 20 Jun '09

20 Jun '09

From: Barry Song <barry.song(a)analog.com> These drivers aren't BF52x specific, so don't use bf52x in the names. Signed-off-by: Barry Song <barry.song(a)analog.com> Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- sound/soc/blackfin/bf5xx-ad73311.c | 16 ++++++++-------- sound/soc/blackfin/bf5xx-ssm2602.c | 16 ++++++++-------- 2 files changed, 16 insertions(+), 16 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c index edfbdc0..9825b71 100644 --- a/sound/soc/blackfin/bf5xx-ad73311.c +++ b/sound/soc/blackfin/bf5xx-ad73311.c @@ -203,23 +203,23 @@ static struct snd_soc_device bf5xx_ad73311_snd_devdata = { .codec_dev = &soc_codec_dev_ad73311, }; -static struct platform_device *bf52x_ad73311_snd_device; +static struct platform_device *bf5xx_ad73311_snd_device; static int __init bf5xx_ad73311_init(void) { int ret; pr_debug("%s enter\n", __func__); - bf52x_ad73311_snd_device = platform_device_alloc("soc-audio", -1); - if (!bf52x_ad73311_snd_device) + bf5xx_ad73311_snd_device = platform_device_alloc("soc-audio", -1); + if (!bf5xx_ad73311_snd_device) return -ENOMEM; - platform_set_drvdata(bf52x_ad73311_snd_device, &bf5xx_ad73311_snd_devdata); - bf5xx_ad73311_snd_devdata.dev = &bf52x_ad73311_snd_device->dev; - ret = platform_device_add(bf52x_ad73311_snd_device); + platform_set_drvdata(bf5xx_ad73311_snd_device, &bf5xx_ad73311_snd_devdata); + bf5xx_ad73311_snd_devdata.dev = &bf5xx_ad73311_snd_device->dev; + ret = platform_device_add(bf5xx_ad73311_snd_device); if (ret) - platform_device_put(bf52x_ad73311_snd_device); + platform_device_put(bf5xx_ad73311_snd_device); return ret; } @@ -227,7 +227,7 @@ static int __init bf5xx_ad73311_init(void) static void __exit bf5xx_ad73311_exit(void) { pr_debug("%s enter\n", __func__); - platform_device_unregister(bf52x_ad73311_snd_device); + platform_device_unregister(bf5xx_ad73311_snd_device); } module_init(bf5xx_ad73311_init); diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c index bc0cdde..3a00fa4 100644 --- a/sound/soc/blackfin/bf5xx-ssm2602.c +++ b/sound/soc/blackfin/bf5xx-ssm2602.c @@ -148,24 +148,24 @@ static struct snd_soc_device bf5xx_ssm2602_snd_devdata = { .codec_data = &bf5xx_ssm2602_setup, }; -static struct platform_device *bf52x_ssm2602_snd_device; +static struct platform_device *bf5xx_ssm2602_snd_device; static int __init bf5xx_ssm2602_init(void) { int ret; pr_debug("%s enter\n", __func__); - bf52x_ssm2602_snd_device = platform_device_alloc("soc-audio", -1); - if (!bf52x_ssm2602_snd_device) + bf5xx_ssm2602_snd_device = platform_device_alloc("soc-audio", -1); + if (!bf5xx_ssm2602_snd_device) return -ENOMEM; - platform_set_drvdata(bf52x_ssm2602_snd_device, + platform_set_drvdata(bf5xx_ssm2602_snd_device, &bf5xx_ssm2602_snd_devdata); - bf5xx_ssm2602_snd_devdata.dev = &bf52x_ssm2602_snd_device->dev; - ret = platform_device_add(bf52x_ssm2602_snd_device); + bf5xx_ssm2602_snd_devdata.dev = &bf5xx_ssm2602_snd_device->dev; + ret = platform_device_add(bf5xx_ssm2602_snd_device); if (ret) - platform_device_put(bf52x_ssm2602_snd_device); + platform_device_put(bf5xx_ssm2602_snd_device); return ret; } @@ -173,7 +173,7 @@ static int __init bf5xx_ssm2602_init(void) static void __exit bf5xx_ssm2602_exit(void) { pr_debug("%s enter\n", __func__); - platform_device_unregister(bf52x_ssm2602_snd_device); + platform_device_unregister(bf5xx_ssm2602_snd_device); } module_init(bf5xx_ssm2602_init); -- 1.6.3.1

3 4

[alsa-devel] [PATCH 1/5] ASoC codec: SSM2602: remove unsupported sample rates
by Mike Frysinger 20 Jun '09

20 Jun '09

From: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- sound/soc/codecs/ssm2602.c | 8 +++----- 1 files changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 87f606c..d6af069 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -497,11 +497,9 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec, return 0; } -#define SSM2602_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ - SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ - SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\ - SNDRV_PCM_RATE_96000) +#define SSM2602_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_32000 |\ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) #define SSM2602_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -- 1.6.3.1

7 40

[alsa-devel] The new mixer logic in PA
by Lennart Poettering 20 Jun '09

20 Jun '09

Heya! During the last weeks I have been reworking the ALSA mixer and device enumeration handling in PA to be more flexible and support more features of usual mixers/cards then it previously did. (Previously we just picked one simple mixer element and used it for volume and muting; 'Master' for output; 'Capture' for input; and the ALSA device name list we used was hardcoded). I just landed all of this in PA, and would like to pass this at least roughly by you folks, hoping for comments. So here's a rough overview of what's going on: The current ALSA mixer APIs have been criticized in the past (including by me) for their complexity and stacking of abstraction layers. Some folks (including me) suggested reworking the ALSA mixer APIs to simplify the APIs. After doing my research I am not so sure anymore that this really would be of much benefit -- at least for software like PA. The reason for that is simply that PA has very specific needs on the mixer interfacing: we are only interested in a subset of the potential functionality: a) we want to do hw volume control when supported for a specific snd_pcm_t with good granularity and range, b) we want mute control, and c) we want to select the output/input port for a snd_pcm_t (i.e. line-in vs. mic/speaker vs. headset). And that's about it. We generally don't want to support any fancier setups even if the hardware would provide it: we don't want simulatenous output to multiple distinct outports, we want no hw mixing, we don't want to support legacy input sources such as CD/Phone/PCSpk..., no "3D" features, not even input feedback. OTOH PA can do stuff other clients cannot do, e.g. extending in softwre the capabilities of the hw if possible and sensible. e.g. we can extend the volume control range/granularity in software beyond the discrete steps the hw provides. It has been suggested to make queryable how mixer elements relate to each other via some mixer API, possibly using the HDA codec graph data. While that would be nice to have I believe that this information would be much more than PA could ever need, and also certainly not help simplicity of an API. I am hence not asking for that, what I do ask for however is that a couple of assumptions I make stay valid. (see below) Given that what PA needs and doesn't need is very different from the needs of almost all other applications needed further abstraction can happen in PA, and doesn't need to be done in ALSA. Which is basically the reason why I decided to do all this mixer abstraction work inside of PA, instead of trying to get that into ALSA. To be able to control more than just one element, PA needs to have some minimal clue how elements are related to each other. ALSA (as mentioned) currently doesn't provide that info, so PA must guess. Hardcoding assumptions about how mixer elements relate to each other of course sucks, which is why I chose to define the list of elements PA should control plus minimal information about how they work together in config files that can be updated more easily and that during runtime. I tried to encode only minimal and the most obvious assumptions. These config files simply list a series of (simple) mixer controls and the way how to initialize them/expose them in PA. Switches can either be bound to always off or on, or be used to follow PA's mute status, or be exposed as distinct options in the UI (which however almost no switch should do). Volumes can either be bound to the minimal values, or to 0dB or be integrated into the volume slider PA exposes. Enumeration values can be exposed in the UI, too, but only if that makes sense. The most interesting part of that is probably the volume handling: for that we go from top to botton in the list of elements in those config files and try to set the volume on the first element marked for it, then divide the actual volume set from our target volume and find the next volume element and apply the remaining volume there, and so on. What's left we then do in attenuate software. All this mixer handling assumes that PA is the only one in charge of volume control. Also, I assume that while I now control a substantial part of the available mixer controls that those I don't control will be initialized properly by "alsactl init". Here's an example how that looks like, for implementing a Mic path (which is the most complex path, due to the input selection stuff) http://git.0pointer.de/?p=pulseaudio.git;a=blob;f=src/modules/alsa/mixer/pa… http://git.0pointer.de/?p=pulseaudio.git;a=blob;f=src/modules/alsa/mixer/pa… http://git.0pointer.de/?p=pulseaudio.git;a=blob;f=src/modules/alsa/mixer/pa… To simplifiy these rules a bit I'd like to see some cleanups in the names used for controls: - Some drivers have a "Input Source Select" enumeration, others a "Input Source" and even others "Capture Source". These could be named the same, can't they? Or is there any difference in their behaviour from an apps PoV? (There's also a "Mic Select" enum, which however probably makes sense to keep seperate given that it the name suggests it is only useful if "Mic" is activated in a capture group) - The options of said enums are really chaotically named. That could really be cleaned up, and would probably be a trivial fix. - Since the various "boost" switches effectively control the volume I'd prefer if they'd be exposed as volume controls with dB info attached. Some Boosts encode the dB info in the name, which sucks. - Some cards have an element "TV Tuner", others "Video". I think that could be merged. I'll prepare a patch for at least some of these suggestions. Of course renaming controls on a big scale will break alsactl restore. Is that a problem? There's one assumption I am currently encoding in the files I am not really sure about. Which is that I assume that the volume level of the surround outputs is influenced by the following volume elements if that they exist: PCM -> Master -> Front, Rear, Surround, Side, Center, LFE while if switched to headphones I assume: PCM -> Master -> Headphone May I assume that? Or is sometimes the routing PCM -> Headphones, and 'Master' does not have influence? There's something else I was wondering, in contrast to the usual code that makes use of the ALSA mixer's dB data I use them to do actual calculations while previously they were just used for "enriching" the information shown on the screen. I wonder if this is going to backfire? How reliable is the dB data actually? [1] For more information on how all this really works you might find this mail I posted to the PA ML interesting: https://tango.0pointer.de/pipermail/pulseaudio-discuss/2009-June/004229.html It goes more into details in things which are relevant for PA folks, probably not so much for the ALSA folks. I'd be very interested in opinions on all of this! Thanks, Lennart [1] To verify and measure the dB data I actually wrote a little tool that can measure the data, which seemed to yield quite correct information on my HDA chip: http://git.0pointer.de/?p=dbmeasure.git;a=tree http://git.0pointer.de/?p=dbmeasure.git;a=blob;f=README -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net http://0pointer.net/lennart/ GnuPG 0x1A015CC4

2 1

[alsa-devel] Why analog loopback of sigmatel devices got removed?
by Maxim Levitsky 20 Jun '09

20 Jun '09

Hi, Due to lack of time, I didn't update kernel on my desktop system. Now I notice that analog loopback go removed. Why? Searching around I found this: http://article.gmane.org/gmane.linux.alsa.devel/59878 * Removed analog loopback mixer items, not mentioned in sigmatel datasheets and reported to break sound support. First they are mention. It took me lot of time to find them. I only have added the loopback items for devices that have this feature in datasheet. Then how it can break sound support? could you explain? I need this feature since I use it with my TV card, and I know more users that use that feature. I don't know about other devices, but I attach the my patch that was included in kernel for this feature. Best regards, Maxim Levitsky

3 13

[alsa-devel] [Bug] ICE1712 analog outputs are unusable
by Peter Lawler 20 Jun '09

20 Jun '09

http://mailman.alsa-project.org/pipermail/alsa-devel/2009-January/014108.ht… https://bugzilla.redhat.com/show_bug.cgi?id=499435 http://mailman.alsa-project.org/pipermail/alsa-devel/2009-June/018616.html The original patch to fix this was declined and thus the issue has been left to fester. I'm available to help test out any new patch proposals etc. Please don't hesitate to contact me etc. Pete.

1 0

[alsa-devel] IDT quirks...where?
by Adam Williamson 20 Jun '09

20 Jun '09

Hi, guys. Following up a Fedora forum quirk report: http://forums.fedoraforum.org/showpost.php?p=1229209&postcount=16 elsewhere in the thread, the reporter is instructed to try the hp-dv5 model parameter, and reports that this works. The alsa-info output shows the subsystem ID as 103c:3621, and the codec as IDT 92HD71B7X. This is one I haven't seen before. I wanted to write up a bug report with a patch to add a quirk for this chipset, but I can't see which of the quirk files it should go into - none of the patch_*.c files seems to cover IDT codecs. What's the story with these? Where do their quirks go? Does current ALSA have one already? (I did try searching through each of the patch_*.c files for 3621 and found nothing, but I may have missed something). Thanks! -- Adam Williamson Fedora QA Community Monkey IRC: adamw | Fedora Talk: adamwill AT fedoraproject DOT org http://www.happyassassin.net

2 2