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[alsa-devel] [PATCH - hda 1/1] ALSA: hda - Add PCI quirk for 17" MacBook Pro 5, 2 model (July 2009) to use existing mb5 model
by murray.alex＠gmail.com 22 Jan '10

22 Jan '10

From: Alex Murray <murray.alex(a)gmail.com> Signed-off-by: Alex Murray <murray.alex(a)gmail.com> --- pci/hda/patch_realtek.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/pci/hda/patch_realtek.c b/pci/hda/patch_realtek.c index 7e0ea37..d044db9 100644 --- a/pci/hda/patch_realtek.c +++ b/pci/hda/patch_realtek.c @@ -9223,6 +9223,7 @@ static struct snd_pci_quirk alc882_ssid_cfg_tbl[] = { */ SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC885_MB5), SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC885_MB5), + SND_PCI_QUIRK(0x10ec, 0x0885, "MacbookPro 5,2", ALC885_MB5), {} /* terminator */ }; -- 1.6.3.3

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[alsa-devel] Update information: HD-Audio-models list
by Mark Brown 22 Jan '10

22 Jan '10

Sir/Madam, I have an update for the HD-Audio-Models.txt file. Under ALC262, the toshiba-s06 model setting is applicable to the Sony Vaio VGN-CS60B laptop, and may also be applicable to similar models. It also needs configuring in alsamixer to enable the internal microphone afterwards. Best regards, Mark Brown.

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Re: [alsa-devel] Reforming s3c2443 ac97 driver
by SeungHyun Choi 22 Jan '10

22 Jan '10

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[alsa-devel] [PATCH] ASoC: add helper macros to declare struct soc_enum instances
by Guennadi Liakhovetski 21 Jan '10

21 Jan '10

Several shortcuts for popular uses of some of SOC_ENUM_* and SOC_VALUE_ENUM_* macros. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> --- diff --git a/include/sound/soc.h b/include/sound/soc.h index 08909cc..9537144 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -169,6 +169,23 @@ .private_value = (unsigned long)&xenum } /* + * Simplified versions of above macros, declaring a struct and calculating + * ARRAY_SIZE internally + */ +#define SOC_ENUM_DOUBLE_DECL(name, xreg, xshift_l, xshift_r, xtexts) \ + struct soc_enum name = SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, \ + ARRAY_SIZE(xtexts), xtexts) +#define SOC_ENUM_SINGLE_DECL(name, xreg, xshift, xtexts) \ + SOC_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xtexts) +#define SOC_ENUM_SINGLE_EXT_DECL(name, xtexts) \ + struct soc_enum name = SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(xtexts), xtexts) +#define SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift_l, xshift_r, xmask, xtexts, xvalues) \ + struct soc_enum name = SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, \ + ARRAY_SIZE(xtexts), xtexts, xvalues) +#define SOC_VALUE_ENUM_SINGLE_DECL(name, xreg, xshift, xmask, xtexts, xvalues) \ + SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xmask, xtexts, xvalues) + +/* * Bias levels * * @ON: Bias is fully on for audio playback and capture operations.

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[alsa-devel] [PATCH 2/5] ASoC: DaVinci: Voice Codec Interface
by miguel.aguilar＠ridgerun.com 21 Jan '10

21 Jan '10

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> This patch adds the support for the interface needed by the DaVinci Voice Codec CQ93VC. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- sound/soc/davinci/Kconfig | 3 + sound/soc/davinci/Makefile | 2 + sound/soc/davinci/davinci-vcif.c | 272 ++++++++++++++++++++++++++++++++++++++ sound/soc/davinci/davinci-vcif.h | 28 ++++ 4 files changed, 305 insertions(+), 0 deletions(-) create mode 100644 sound/soc/davinci/davinci-vcif.c create mode 100644 sound/soc/davinci/davinci-vcif.h diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 047ee39..47e7cce 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -12,6 +12,9 @@ config SND_DAVINCI_SOC_I2S config SND_DAVINCI_SOC_MCASP tristate +config SND_DAVINCI_SOC_VCIF + tristate + config SND_DAVINCI_SOC_EVM tristate "SoC Audio support for DaVinci DM6446, DM355 or DM365 EVM" depends on SND_DAVINCI_SOC diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index a6939d7..a93679d 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -2,10 +2,12 @@ snd-soc-davinci-objs := davinci-pcm.o snd-soc-davinci-i2s-objs := davinci-i2s.o snd-soc-davinci-mcasp-objs:= davinci-mcasp.o +snd-soc-davinci-vcif-objs:= davinci-vcif.o obj-$(CONFIG_SND_DAVINCI_SOC) += snd-soc-davinci.o obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o obj-$(CONFIG_SND_DAVINCI_SOC_MCASP) += snd-soc-davinci-mcasp.o +obj-$(CONFIG_SND_DAVINCI_SOC_VCIF) += snd-soc-davinci-vcif.o # DAVINCI Machine Support snd-soc-evm-objs := davinci-evm.o diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c new file mode 100644 index 0000000..03f3feb --- /dev/null +++ b/sound/soc/davinci/davinci-vcif.c @@ -0,0 +1,272 @@ +/* + * ALSA SoC Voice Codec Interface for TI DAVINCI processor + * + * Copyright (C) 2010 Texas Instruments. + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/mfd/davinci_voicecodec.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include "davinci-pcm.h" +#include "davinci-i2s.h" +#include "davinci-vcif.h" + +#define MOD_REG_BIT(val, mask, set) do { \ + if (set) { \ + val |= mask; \ + } else { \ + val &= ~mask; \ + } \ +} while (0) + +struct davinci_vcif_dev { + struct davinci_vc *davinci_vc; + struct davinci_pcm_dma_params dma_params[2]; +}; + +static void davinci_vcif_start(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_vcif_dev *davinci_vcif_dev = + rtd->dai->cpu_dai->private_data; + struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc; + u32 w; + + /* Start the sample generator and enable transmitter/receiver */ + w = readl(davinci_vc->base + DAVINCI_VC_CTRL); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 1); + else + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 1); + + writel(w, davinci_vc->base + DAVINCI_VC_CTRL); +} + +static void davinci_vcif_stop(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_vcif_dev *davinci_vcif_dev = + rtd->dai->cpu_dai->private_data; + struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc; + u32 w; + + /* Reset transmitter/receiver and sample rate/frame sync generators */ + w = readl(davinci_vc->base + DAVINCI_VC_CTRL); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 0); + else + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 0); + + writel(w, davinci_vc->base + DAVINCI_VC_CTRL); +} + +static int davinci_vcif_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct davinci_vcif_dev *davinci_vcif_dev = dai->private_data; + struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc; + struct davinci_pcm_dma_params *dma_params = + &davinci_vcif_dev->dma_params[substream->stream]; + u32 w; + + /* Restart the codec before setup */ + davinci_vcif_stop(substream); + davinci_vcif_start(substream); + + /* General line settings */ + writel(DAVINCI_VC_CTRL_MASK, davinci_vc->base + DAVINCI_VC_CTRL); + + writel(DAVINCI_VC_INT_MASK, davinci_vc->base + DAVINCI_VC_INTCLR); + + writel(DAVINCI_VC_INT_MASK, davinci_vc->base + DAVINCI_VC_INTEN); + + w = readl(davinci_vc->base + DAVINCI_VC_CTRL); + + /* Determine xfer data type */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_U8: + dma_params->data_type = 0; + + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_BITS_8 | + DAVINCI_VC_CTRL_RD_UNSIGNED | + DAVINCI_VC_CTRL_WD_BITS_8 | + DAVINCI_VC_CTRL_WD_UNSIGNED, 1); + break; + case SNDRV_PCM_FORMAT_S8: + dma_params->data_type = 1; + + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_BITS_8 | + DAVINCI_VC_CTRL_WD_BITS_8, 1); + + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_UNSIGNED | + DAVINCI_VC_CTRL_WD_UNSIGNED, 0); + break; + case SNDRV_PCM_FORMAT_S16_LE: + dma_params->data_type = 2; + + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_BITS_8 | + DAVINCI_VC_CTRL_RD_UNSIGNED | + DAVINCI_VC_CTRL_WD_BITS_8 | + DAVINCI_VC_CTRL_WD_UNSIGNED, 0); + break; + default: + printk(KERN_WARNING "davinci-vcif: unsupported PCM format"); + return -EINVAL; + } + + dma_params->acnt = dma_params->data_type; + + writel(w, davinci_vc->base + DAVINCI_VC_CTRL); + + return 0; +} + +static int davinci_vcif_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + davinci_vcif_start(substream); + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + davinci_vcif_stop(substream); + break; + default: + ret = -EINVAL; + } + + return ret; +} + +#define DAVINCI_VCIF_RATES SNDRV_PCM_RATE_8000_48000 + +static struct snd_soc_dai_ops davinci_vcif_dai_ops = { + .trigger = davinci_vcif_trigger, + .hw_params = davinci_vcif_hw_params, +}; + +struct snd_soc_dai davinci_vcif_dai = { + .name = "davinci-vcif", + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = DAVINCI_VCIF_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = DAVINCI_VCIF_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = &davinci_vcif_dai_ops, + +}; +EXPORT_SYMBOL_GPL(davinci_vcif_dai); + +static int davinci_vcif_probe(struct platform_device *pdev) +{ + struct davinci_vc *davinci_vc = platform_get_drvdata(pdev); + struct davinci_vcif_dev *davinci_vcif_dev; + int ret; + + davinci_vcif_dev = kzalloc(sizeof(struct davinci_vcif_dev), GFP_KERNEL); + if (!davinci_vc) { + dev_dbg(&pdev->dev, + "could not allocate memory for private data\n"); + return -ENOMEM; + } + + /* DMA tx params */ + davinci_vcif_dev->davinci_vc = davinci_vc; + davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].channel = + davinci_vc->davinci_vcif.dma_tx_channel; + davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].dma_addr = + davinci_vc->davinci_vcif.dma_tx_addr; + + /* DMA rx params */ + davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = + davinci_vc->davinci_vcif.dma_rx_channel; + davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].dma_addr = + davinci_vc->davinci_vcif.dma_rx_addr; + + davinci_vcif_dai.dev = &pdev->dev; + davinci_vcif_dai.dma_data = davinci_vcif_dev->dma_params; + davinci_vcif_dai.private_data = davinci_vcif_dev; + + ret = snd_soc_register_dai(&davinci_vcif_dai); + if (ret != 0) { + dev_err(&pdev->dev, "could not register dai\n"); + goto fail; + } + + return 0; + +fail: + kfree(davinci_vcif_dev); + + return ret; +} + +static int davinci_vcif_remove(struct platform_device *pdev) +{ + snd_soc_unregister_dai(&davinci_vcif_dai); + + return 0; +} + +static struct platform_driver davinci_vcif_driver = { + .probe = davinci_vcif_probe, + .remove = davinci_vcif_remove, + .driver = { + .name = "davinci_vcif", + .owner = THIS_MODULE, + }, +}; + +static int __init davinci_vcif_init(void) +{ + return platform_driver_probe(&davinci_vcif_driver, davinci_vcif_probe); +} +module_init(davinci_vcif_init); + +static void __exit davinci_vcif_exit(void) +{ + platform_driver_unregister(&davinci_vcif_driver); +} +module_exit(davinci_vcif_exit); + +MODULE_AUTHOR("Miguel Aguilar"); +MODULE_DESCRIPTION("Texas Instruments DaVinci ASoC Voice Codec Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/davinci/davinci-vcif.h b/sound/soc/davinci/davinci-vcif.h new file mode 100644 index 0000000..571c994 --- /dev/null +++ b/sound/soc/davinci/davinci-vcif.h @@ -0,0 +1,28 @@ +/* + * ALSA SoC Voice Codec Interface for TI DAVINCI processor + * + * Copyright (C) 2010 Texas Instruments. + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef _DAVINCI_VCIF_H +#define _DAVINCI_VCIF_H + +extern struct snd_soc_dai davinci_vcif_dai; + +#endif -- 1.6.0.4

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[alsa-devel] Problem with PPH plugin
by Colin Guthrie 21 Jan '10

21 Jan '10

Hi, Just wanted to highlight a bug over on Mozilla relating to a crash in the PPH plugin. https://bugzilla.mozilla.org/show_bug.cgi?id=540460 It was initially though to be in PulseAudio, but on closer inspection this doesn't involve PA at all and in fact seems to relate to the PPH plugin from what I can gather. The backtrace is here: https://bug540460.bugzilla.mozilla.org/attachment.cgi?id=422264 The user did submit a bug report to the bugtracker (https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4878) but given previous experience there (see other mail!) I figured I'd post a message here to highlight it seeing as the reporter has already done a lot to try and work out what is going on and posting a message on his behalf seemed like a good idea :) Cheers All the best. -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mandriva Linux Contributor [http://www.mandriva.com/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/]

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[alsa-devel] ASoC updates for 2.6.34
by Mark Brown 21 Jan '10

21 Jan '10

The following changes since commit 9135f6db09aa9d186f05d41e9c516ec8e3550d78: Mark Brown (1): Merge branch 'mxc-audio' into for-2.6.34 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.34 Guennadi Liakhovetski (1): ASoC: fix compile breakage - add a missing header include Mark Brown (3): ASoC: Remove console DAPM debug code ASoC: Support turning off bias when the CODEC is idle ASoC: Use BIAS_OFF when idle for wm_hubs devices Peter Ujfalusi (3): ASoC: tlv320dac33: Correct the prefill number of samples ASoC: tlv320dac33: BCLK divider fix ASoC: tlv320dac33: Burst mode BCLK divider configuration include/sound/soc-dai.h | 2 + include/sound/soc.h | 2 + include/sound/tlv320dac33-plat.h | 1 + sound/soc/codecs/tlv320dac33.c | 24 +++++++-- sound/soc/codecs/wm_hubs.c | 6 ++ sound/soc/soc-dapm.c | 105 ++++++++++---------------------------- 6 files changed, 56 insertions(+), 84 deletions(-)

2 1

[alsa-devel] [PATCH] ASoC: Support turning off bias when the CODEC is idle
by Mark Brown 21 Jan '10

21 Jan '10

Currently ASoC always maintains the bias of the CODEC while the system is active. With older mobile CODECs this is required since the outputs are referenced to a non-zero voltage and enabling or disabling this voltage without audible pops or clicks in the output takes too long to do when starting or stopping audio. As a result of features such as ground referenced outputs and class D speaker drivers current generation devices are able to power on and off much more quickly without these system level issues so provide a new flag idle_bias_off in snd_soc_codec which will cause the core to turn off the CODEC bias. The distinction between STANDBY and OFF is still maintained. This is partly for consistency but also allows for potential future extensions such as per-machine overrides or deferring the bias removal. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg(a)slimlogic.co.uk> --- include/sound/soc.h | 2 ++ sound/soc/soc-dapm.c | 25 ++++++++++++++++++++++++- 2 files changed, 26 insertions(+), 1 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 08909cc..a8768ea 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -405,6 +405,8 @@ struct snd_soc_codec { short reg_cache_size; short reg_cache_step; + unsigned int idle_bias_off:1; /* Use BIAS_OFF instead of STANDBY */ + /* dapm */ u32 pop_time; struct list_head dapm_widgets; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d8e9374..6c33510 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1012,13 +1012,28 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) sys_power = 0; break; case SND_SOC_DAPM_STREAM_NOP: - sys_power = codec->bias_level != SND_SOC_BIAS_STANDBY; + switch (codec->bias_level) { + case SND_SOC_BIAS_STANDBY: + case SND_SOC_BIAS_OFF: + sys_power = 0; + break; + default: + sys_power = 1; + break; + } break; default: break; } } + if (sys_power && codec->bias_level == SND_SOC_BIAS_OFF) { + ret = snd_soc_dapm_set_bias_level(socdev, + SND_SOC_BIAS_STANDBY); + if (ret != 0) + pr_err("Failed to turn on bias: %d\n", ret); + } + /* If we're changing to all on or all off then prepare */ if ((sys_power && codec->bias_level == SND_SOC_BIAS_STANDBY) || (!sys_power && codec->bias_level == SND_SOC_BIAS_ON)) { @@ -1042,6 +1057,14 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) pr_err("Failed to apply standby bias: %d\n", ret); } + /* If we're in standby and can support bias off then do that */ + if (codec->bias_level == SND_SOC_BIAS_STANDBY && + codec->idle_bias_off) { + ret = snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_OFF); + if (ret != 0) + pr_err("Failed to turn off bias: %d\n", ret); + } + /* If we just powered up then move to active bias */ if (codec->bias_level == SND_SOC_BIAS_PREPARE && sys_power) { ret = snd_soc_dapm_set_bias_level(socdev, -- 1.6.6

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[alsa-devel] Reforming s3c2443 ac97 driver
by JASWINDERSINGH BRAR 21 Jan '10

21 Jan '10

Hi, Right now the S3C2443's AC97_ver2.0 controller driver isn't flexible enough to handle newer SoCs. In order to prepare ground for supporting 64xx and S5Pxxxx the extant driver needs to be reformed, and that is what i am planning to do next. I plan to do the following:- 1) User platform_driver structure to detect and manage AC97 controller platform devices. 2) Remove hardcoded parameters(DMA, IO address, IRQ etc) and pass via platform resources. 3) Initialize platform specific stuff by callback pointers to platform code(cfg_gpio) Objections, suggestions welcome. ------------------------------- Jaswinder Singh Brar Senior Engineer Linux OS Team Application Processor Development System LSI Division SAMSUNG Electronics Co., Ltd. --------------------------------

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[alsa-devel] snd_pcm_hw_params_free() immediately after snd_pcm_hw_params() ???
by Raymond Yau 21 Jan '10

21 Jan '10

Refer to http://www.alsa-project.org/~tiwai/writing-an-alsa-driver/ch05s06.html#pcm-… hw_free callback static int snd_xxx_hw_free(struct snd_pcm_substream *substream); This is called to release the resources allocated via *hw_params*. For example, releasing the buffer via snd_pcm_lib_malloc_pages() is done by calling the following: snd_pcm_lib_free_pages(substream); This function is always called before the close callback is called. Also, the callback may be called multiple times, too. Keep track whether the resource was already released. Nearly all the examples in those ALSA tutorials in the web sites have the code if ((err = snd_pcm_hw_params (playback_handle, hw_params)) < 0) { fprintf (stderr, "cannot set parameters (%s)\n", snd_strerror (err)); exit (1); } snd_pcm_hw_params_free (hw_params); This seem to violate "This function is always called before the close callback is called." and those hardware mixing sound cards (e.g cs46xx , au88x0, ... ) use snd_xxx_hw_free() to deallocate the hardware mixing path in the DSP. There are a number of audio package (e.g. gstreamer , wine , .... ) call snd_pcm_hw_params_free immediately after snd_pcm_hw_params() Is this a bug of snd_pcm_hw_params_free() not change the state of the PCM stream ? or snd_pcm_prepare() did not check the buffer is already released ?

2 2

[alsa-devel] [PATCH] ARM: DaVinci: Audio support for DA850/OMAP-L138 EVM
by Chaithrika U S 21 Jan '10

21 Jan '10

Define resources for McASP used on DA850/OMAP-L138 EVM, add platform device defintion and Pin Mux configurations. Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- Applies on DaVinci GIT tree. This patch is dependent on the following patch which I have submitted earlier: [PATCH v2] ARM: DaVinci: Audio support for DA830 EVM arch/arm/mach-davinci/board-da850-evm.c | 29 ++++++++++++++++++++++ arch/arm/mach-davinci/da850.c | 39 ++++++++++++++++++++++++++++++ arch/arm/mach-davinci/devices-da8xx.c | 34 +++++++++++++++++++++++++- arch/arm/mach-davinci/include/mach/asp.h | 7 +++++ arch/arm/mach-davinci/include/mach/mux.h | 25 +++++++++++++++++++ 5 files changed, 133 insertions(+), 1 deletions(-) diff --git a/arch/arm/mach-davinci/board-da850-evm.c b/arch/arm/mach-davinci/board-da850-evm.c index d989346..52bfe4c 100644 --- a/arch/arm/mach-davinci/board-da850-evm.c +++ b/arch/arm/mach-davinci/board-da850-evm.c @@ -38,6 +38,28 @@ static struct davinci_uart_config da850_evm_uart_config __initdata = { .enabled_uarts = 0x7, }; +/* davinci da850 evm audio machine driver */ +static u8 da850_iis_serializer_direction[] = { + INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, + INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, + INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, TX_MODE, + RX_MODE, INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, +}; + +static struct snd_platform_data da850_evm_snd_data = { + .tx_dma_offset = 0x2000, + .rx_dma_offset = 0x2000, + .op_mode = DAVINCI_MCASP_IIS_MODE, + .num_serializer = ARRAY_SIZE(da850_iis_serializer_direction), + .tdm_slots = 2, + .serial_dir = da850_iis_serializer_direction, + .eventq_no = EVENTQ_1, + .version = MCASP_VERSION_2, + .txnumevt = 1, + .rxnumevt = 1, +}; + + static __init void da850_evm_init(void) { struct davinci_soc_info *soc_info = &davinci_soc_info; @@ -86,6 +108,13 @@ static __init void da850_evm_init(void) */ __raw_writel(0, IO_ADDRESS(DA8XX_UART1_BASE) + 0x30); __raw_writel(0, IO_ADDRESS(DA8XX_UART0_BASE) + 0x30); + + ret = da8xx_pinmux_setup(da850_mcasp_pins); + if (ret) + pr_warning("da850_evm_init: mcasp mux setup failed: %d\n", + ret); + + da8xx_init_mcasp(0, &da850_evm_snd_data); } #ifdef CONFIG_SERIAL_8250_CONSOLE diff --git a/arch/arm/mach-davinci/da850.c b/arch/arm/mach-davinci/da850.c index 4a43ae2..e33b226 100644 --- a/arch/arm/mach-davinci/da850.c +++ b/arch/arm/mach-davinci/da850.c @@ -289,6 +289,13 @@ static struct clk emac_clk = { .lpsc = DA8XX_LPSC1_CPGMAC, }; +static struct clk mcasp_clk = { + .name = "mcasp", + .parent = &pll0_sysclk2, + .lpsc = DA8XX_LPSC1_McASP0, + .psc_ctlr = 1, +}; + static struct davinci_clk da850_clks[] = { CLK(NULL, "ref", &ref_clk), CLK(NULL, "pll0", &pll0_clk), @@ -326,6 +333,7 @@ static struct davinci_clk da850_clks[] = { CLK(NULL, "arm", &arm_clk), CLK(NULL, "rmii", &rmii_clk), CLK("davinci_emac.1", NULL, &emac_clk), + CLK("davinci-mcasp.0", NULL, &mcasp_clk), CLK(NULL, NULL, NULL), }; @@ -370,6 +378,30 @@ static const struct mux_config da850_pins[] = { MUX_CFG(DA850, MII_RXD_2, 3, 20, 15, 8, false) MUX_CFG(DA850, MII_RXD_1, 3, 24, 15, 8, false) MUX_CFG(DA850, MII_RXD_0, 3, 28, 15, 8, false) + /* McASP function */ + MUX_CFG(DA850, ACLKR, 0, 0, 15, 1, false) + MUX_CFG(DA850, ACLKX, 0, 4, 15, 1, false) + MUX_CFG(DA850, AFSR, 0, 8, 15, 1, false) + MUX_CFG(DA850, AFSX, 0, 12, 15, 1, false) + MUX_CFG(DA850, AHCLKR, 0, 16, 15, 1, false) + MUX_CFG(DA850, AHCLKX, 0, 20, 15, 1, false) + MUX_CFG(DA850, AMUTE, 0, 24, 15, 1, false) + MUX_CFG(DA850, AXR_15, 1, 0, 15, 1, false) + MUX_CFG(DA850, AXR_14, 1, 4, 15, 1, false) + MUX_CFG(DA850, AXR_13, 1, 8, 15, 1, false) + MUX_CFG(DA850, AXR_12, 1, 12, 15, 1, false) + MUX_CFG(DA850, AXR_11, 1, 16, 15, 1, false) + MUX_CFG(DA850, AXR_10, 1, 20, 15, 1, false) + MUX_CFG(DA850, AXR_9, 1, 24, 15, 1, false) + MUX_CFG(DA850, AXR_8, 1, 28, 15, 1, false) + MUX_CFG(DA850, AXR_7, 2, 0, 15, 1, false) + MUX_CFG(DA850, AXR_6, 2, 4, 15, 1, false) + MUX_CFG(DA850, AXR_5, 2, 8, 15, 1, false) + MUX_CFG(DA850, AXR_4, 2, 12, 15, 1, false) + MUX_CFG(DA850, AXR_3, 2, 16, 15, 1, false) + MUX_CFG(DA850, AXR_2, 2, 20, 15, 1, false) + MUX_CFG(DA850, AXR_1, 2, 24, 15, 1, false) + MUX_CFG(DA850, AXR_0, 2, 28, 15, 1, false) #endif }; @@ -406,6 +438,13 @@ const short da850_cpgmac_pins[] __initdata = { -1 }; +const short da850_mcasp_pins[] __initdata = { + DA850_AHCLKX, DA850_ACLKX, DA850_AFSX, + DA850_AHCLKR, DA850_ACLKR, DA850_AFSR, DA850_AMUTE, + DA850_AXR_11, DA850_AXR_12, + -1 +}; + /* FIQ are pri 0-1; otherwise 2-7, with 7 lowest priority */ static u8 da850_default_priorities[DA850_N_CP_INTC_IRQ] = { [IRQ_DA8XX_COMMTX] = 7, diff --git a/arch/arm/mach-davinci/devices-da8xx.c b/arch/arm/mach-davinci/devices-da8xx.c index fe0baaf..1882eef 100644 --- a/arch/arm/mach-davinci/devices-da8xx.c +++ b/arch/arm/mach-davinci/devices-da8xx.c @@ -309,6 +309,34 @@ static struct platform_device da830_mcasp1_device = { .resource = da830_mcasp1_resources, }; +static struct resource da850_mcasp_resources[] = { + { + .name = "mcasp", + .start = DAVINCI_DA8XX_MCASP0_REG_BASE, + .end = DAVINCI_DA8XX_MCASP0_REG_BASE + (SZ_1K * 12) - 1, + .flags = IORESOURCE_MEM, + }, + /* TX event */ + { + .start = DAVINCI_DA8XX_DMA_MCASP0_AXEVT, + .end = DAVINCI_DA8XX_DMA_MCASP0_AXEVT, + .flags = IORESOURCE_DMA, + }, + /* RX event */ + { + .start = DAVINCI_DA8XX_DMA_MCASP0_AREVT, + .end = DAVINCI_DA8XX_DMA_MCASP0_AREVT, + .flags = IORESOURCE_DMA, + }, +}; + +static struct platform_device da850_mcasp_device = { + .name = "davinci-mcasp", + .id = 0, + .num_resources = ARRAY_SIZE(da850_mcasp_resources), + .resource = da850_mcasp_resources, +}; + int __init da8xx_register_emac(void) { return platform_device_register(&da8xx_emac_device); @@ -316,8 +344,12 @@ int __init da8xx_register_emac(void) void __init da8xx_init_mcasp(int id, struct snd_platform_data *pdata) { - if (id == 1) { /* DA830/OMAP-L137 has 3 instances of McASP */ + /* DA830/OMAP-L137 has 3 instances of McASP */ + if (cpu_is_davinci_da830() && id == 1) { da830_mcasp1_device.dev.platform_data = pdata; platform_device_register(&da830_mcasp1_device); + } else if (cpu_is_davinci_da850()) { + da850_mcasp_device.dev.platform_data = pdata; + platform_device_register(&da850_mcasp_device); } } diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index f3c97ac..18e4ce3 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -15,6 +15,9 @@ #define DAVINCI_DM646X_MCASP0_REG_BASE 0x01D01000 #define DAVINCI_DM646X_MCASP1_REG_BASE 0x01D01800 +/* Bases of da850/da830 McASP0 register banks */ +#define DAVINCI_DA8XX_MCASP0_REG_BASE 0x01D00000 + /* Bases of da830 McASP1 register banks */ #define DAVINCI_DA830_MCASP1_REG_BASE 0x01D04000 @@ -29,6 +32,10 @@ #define DAVINCI_DM646X_DMA_MCASP0_AREVT0 9 #define DAVINCI_DM646X_DMA_MCASP1_AXEVT1 12 +/* EDMA channels of da850/da830 McASP0 */ +#define DAVINCI_DA8XX_DMA_MCASP0_AREVT 0 +#define DAVINCI_DA8XX_DMA_MCASP0_AXEVT 1 + /* EDMA channels of da830 McASP1 */ #define DAVINCI_DA830_DMA_MCASP1_AREVT 2 #define DAVINCI_DA830_DMA_MCASP1_AXEVT 3 diff --git a/arch/arm/mach-davinci/include/mach/mux.h b/arch/arm/mach-davinci/include/mach/mux.h index a676b2f..f50f5b2 100644 --- a/arch/arm/mach-davinci/include/mach/mux.h +++ b/arch/arm/mach-davinci/include/mach/mux.h @@ -748,6 +748,31 @@ enum davinci_da850_index { DA850_MII_RXD_2, DA850_MII_RXD_1, DA850_MII_RXD_0, + + /* McASP function */ + DA850_ACLKR, + DA850_ACLKX, + DA850_AFSR, + DA850_AFSX, + DA850_AHCLKR, + DA850_AHCLKX, + DA850_AMUTE, + DA850_AXR_15, + DA850_AXR_14, + DA850_AXR_13, + DA850_AXR_12, + DA850_AXR_11, + DA850_AXR_10, + DA850_AXR_9, + DA850_AXR_8, + DA850_AXR_7, + DA850_AXR_6, + DA850_AXR_5, + DA850_AXR_4, + DA850_AXR_3, + DA850_AXR_2, + DA850_AXR_1, + DA850_AXR_0, }; #ifdef CONFIG_DAVINCI_MUX -- 1.5.6

3 3

[alsa-devel] device naming
by Andrew Kohlsmith (Mailing List Account) 20 Jan '10

20 Jan '10

Is there a definitive guide to device naming with ALSA? I'm trying to use a single device for simultaneous capture and playback but opening different device names gives different results and I'm trying to determine which is the best to use. aplay -L output gives front, surround40, surround41, surround50, surround51, null and pulse. Trying to register a playback callback using "pulse" gives a "function not implemented" error, although I can open and set all the parameters I need (8kHz, 2 channels, 20ms buffers). I get the same result with "default". If I use "plughw:0,0" I can capture and play audio, but the driver (intel HDA) seems to be very prone to either not calling either the playback or capture callback, or throwing an xrun and not being able to recover. This is an otherwise idle core2 machine running Ubuntu 9.10. Playing with nice or ionice does not seem to affect this "brittleness." I do not have an ~/.asoundrc, nor do I have anything related to asound in /etc. Moving pulseaudio out of the way doesn't seem to do anything either. Is there a definitive list of devices and their device filenames so I can stop guessing? /proc/asound/cards just shows card 0 as an Intel HDA, and /proc/asound/devices shows the following: $ cat /proc/asound/devices 2: : timer 3: : sequencer 4: [ 0- 2]: digital audio capture 5: [ 0- 0]: digital audio playback 6: [ 0- 0]: digital audio capture 7: [ 0- 1]: hardware dependent 8: [ 0- 0]: hardware dependent 9: [ 0] : control (0-0 is the only device showing both playback and capture, which is why I'm using it, but how do I know that plughw is the correct device filename? I just guessed at it due to previous experience trying to get bt audio working with ALSA.) I'm sure I'm doing something incredibly stupid, but I can't seem to find out what it is. -A.

1 0

[alsa-devel] [PATCH 2/2] Davinci: DM365: Enable DaVinci Voice Codec support for DM365 EVM
by miguel.aguilar＠ridgerun.com 20 Jan '10

20 Jan '10

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> It sets up one single card for the DM365 EVM with two subdevices: the AIC3x and the Voice Codec. The general structures are defined at DM365 SoC file and the specific platform data structure for the EVM is defined at board file. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- arch/arm/mach-davinci/board-dm365-evm.c | 39 +++++++++++++++++++++++++++- arch/arm/mach-davinci/dm365.c | 37 ++++++++++++++++++++++++-- arch/arm/mach-davinci/include/mach/asp.h | 1 + arch/arm/mach-davinci/include/mach/dm365.h | 5 +++ arch/arm/mach-davinci/include/mach/mux.h | 2 + 5 files changed, 80 insertions(+), 4 deletions(-) diff --git a/arch/arm/mach-davinci/board-dm365-evm.c b/arch/arm/mach-davinci/board-dm365-evm.c index 8d23972..74f3519 100644 --- a/arch/arm/mach-davinci/board-dm365-evm.c +++ b/arch/arm/mach-davinci/board-dm365-evm.c @@ -175,7 +175,43 @@ static struct at24_platform_data eeprom_info = { .context = (void *)0x7f00, }; -static struct snd_platform_data dm365_evm_snd_data; +static int dm365_snd_dma_event_mux(int device, int enable, int playback) +{ + static int tx_dma_mux = 0; + static int rx_dma_mux = 0; + + if (!enable) { + if (playback) + tx_dma_mux = 0; + else + rx_dma_mux = 0; + return 0; + } + + if (playback) { + if (tx_dma_mux == 1) + return -EBUSY; + if (device == 0) + davinci_cfg_reg(DM365_EVT2_ASP_TX); + else + davinci_cfg_reg(DM365_EVT2_VC_TX); + tx_dma_mux = 1; + } else { + if (rx_dma_mux == 1) + return -EBUSY; + if (device == 0) + davinci_cfg_reg(DM365_EVT3_ASP_RX); + else + davinci_cfg_reg(DM365_EVT3_VC_RX); + rx_dma_mux = 1; + } + + return 0; +} + +static struct snd_platform_data dm365_evm_snd_data = { + .dma_event_mux = &dm365_snd_dma_event_mux, +}; static struct i2c_board_info i2c_info[] = { { @@ -513,6 +549,7 @@ static __init void dm365_evm_init(void) evm_init_cpld(); dm365_init_asp(&dm365_evm_snd_data); + dm365_init_vc(&dm365_evm_snd_data); dm365_init_rtc(); #ifdef CONFIG_KEYBOARD_DAVINCI diff --git a/arch/arm/mach-davinci/dm365.c b/arch/arm/mach-davinci/dm365.c index cc3bae4..024cbab 100644 --- a/arch/arm/mach-davinci/dm365.c +++ b/arch/arm/mach-davinci/dm365.c @@ -455,7 +455,7 @@ static struct davinci_clk dm365_clks[] = { CLK(NULL, "timer3", &timer3_clk), CLK(NULL, "usb", &usb_clk), CLK("davinci_emac.1", NULL, &emac_clk), - CLK("voice_codec", NULL, &voicecodec_clk), + CLK("davinci_voicecodec", NULL, &voicecodec_clk), CLK("davinci-asp.0", NULL, &asp0_clk), CLK(NULL, "rto", &rto_clk), CLK(NULL, "mjcp", &mjcp_clk), @@ -604,6 +604,8 @@ INT_CFG(DM365, INT_IMX1_DISABLE, 24, 1, 0, false) INT_CFG(DM365, INT_NSF_ENABLE, 25, 1, 1, false) INT_CFG(DM365, INT_NSF_DISABLE, 25, 1, 0, false) +EVT_CFG(DM365, EVT2_VC_TX, 0, 1, 1, false) +EVT_CFG(DM365, EVT3_VC_RX, 1, 1, 1, false) EVT_CFG(DM365, EVT2_ASP_TX, 0, 1, 0, false) EVT_CFG(DM365, EVT3_ASP_RX, 1, 1, 0, false) #endif @@ -835,6 +837,31 @@ static struct platform_device dm365_asp_device = { .resource = dm365_asp_resources, }; +static struct resource dm365_vc_resources[] = { + { + .start = DAVINCI_DM365_VC_BASE, + .end = DAVINCI_DM365_VC_BASE + SZ_1K - 1, + .flags = IORESOURCE_MEM, + }, + { + .start = DAVINCI_DMA_VC_TX, + .end = DAVINCI_DMA_VC_TX, + .flags = IORESOURCE_DMA, + }, + { + .start = DAVINCI_DMA_VC_RX, + .end = DAVINCI_DMA_VC_RX, + .flags = IORESOURCE_DMA, + }, +}; + +static struct platform_device dm365_vc_device = { + .name = "davinci_voicecodec", + .id = -1, + .num_resources = ARRAY_SIZE(dm365_vc_resources), + .resource = dm365_vc_resources, +}; + static struct resource dm365_rtc_resources[] = { { .start = DM365_RTC_BASE, @@ -985,12 +1012,16 @@ void __init dm365_init_asp(struct snd_platform_data *pdata) davinci_cfg_reg(DM365_MCBSP0_BDR); davinci_cfg_reg(DM365_MCBSP0_R); davinci_cfg_reg(DM365_MCBSP0_BFSR); - davinci_cfg_reg(DM365_EVT2_ASP_TX); - davinci_cfg_reg(DM365_EVT3_ASP_RX); dm365_asp_device.dev.platform_data = pdata; platform_device_register(&dm365_asp_device); } +void __init dm365_init_vc(struct snd_platform_data *pdata) +{ + dm365_vc_device.dev.platform_data = pdata; + platform_device_register(&dm365_vc_device); +} + void __init dm365_init_ks(struct davinci_ks_platform_data *pdata) { davinci_cfg_reg(DM365_KEYSCAN); diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index 834725f..6a175eb 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -50,6 +50,7 @@ #define DAVINCI_ASP1_TX_INT IRQ_MBXINT struct snd_platform_data { + int (*dma_event_mux)(int, int, int); u32 tx_dma_offset; u32 rx_dma_offset; enum dma_event_q eventq_no; /* event queue number */ diff --git a/arch/arm/mach-davinci/include/mach/dm365.h b/arch/arm/mach-davinci/include/mach/dm365.h index 3c07a88..50aeaec 100644 --- a/arch/arm/mach-davinci/include/mach/dm365.h +++ b/arch/arm/mach-davinci/include/mach/dm365.h @@ -31,8 +31,13 @@ #define DM365_RTC_BASE (0x01C69000) +#define DAVINCI_DM365_VC_BASE (0x01D0C000) +#define DAVINCI_DMA_VC_TX 2 +#define DAVINCI_DMA_VC_RX 3 + void __init dm365_init(void); void __init dm365_init_asp(struct snd_platform_data *pdata); +void __init dm365_init_vc(struct snd_platform_data *pdata); void __init dm365_init_ks(struct davinci_ks_platform_data *pdata); void __init dm365_init_rtc(void); diff --git a/arch/arm/mach-davinci/include/mach/mux.h b/arch/arm/mach-davinci/include/mach/mux.h index b60c693..2937c51 100644 --- a/arch/arm/mach-davinci/include/mach/mux.h +++ b/arch/arm/mach-davinci/include/mach/mux.h @@ -325,6 +325,8 @@ enum davinci_dm365_index { DM365_INT_NSF_DISABLE, /* EDMA event muxing */ + DM365_EVT2_VC_TX, + DM365_EVT3_VC_RX, DM365_EVT2_ASP_TX, DM365_EVT3_ASP_RX, DM365_EVT26_MMC0_RX, -- 1.6.0.4

4 13

[alsa-devel] [PATCH] Obsolete config in kernel source (USE_MIXARTLOADER)
by Christoph Egger 20 Jan '10

20 Jan '10

Hi all! As part of the VAMOS[0] research project at the University of Erlangen we're checking referential integrity between kernel KConfig options and in-code Conditional blocks. The config Variable USE_MIXARTLOADER has been non-existent for all git based linux kernels (2.6.12-rc) and seems to be no longer relevant. It is only referenced at one place in the Code (sound/pci/mixart/mixart_hwdep.c). The attached patch makes the behaviour more obvious as it is equivalent to the old one when assuming CONFIG_MIXARTLOADER as not defined. Please keep me informed of this patch getting confirmed / merged so we can keep track of it. Regards Christoph Egger

2 3

[alsa-devel] Conexant CX20561 on HP Pavilion dv2700
by Jacopo De Simoi 20 Jan '10

20 Jan '10

Dear alsa-devs I'm trying to have CX20561 in working order on my laptop; I'll be glad to send a patch once I figure out what is going on there, and for this I need your help. There are a number of issues, mostly regarding the internal mic, as I'm sure you know considering the bugs volume on the subject :) 1a) There are three capture sources listed in alsamixer (Digital excluded): Docking External Internal; the only one that has any influence in recording (both from internal and external mics) is "External" 1b) Why does Node 0x14 shows two volume controls? the first one doesn't seem to have any effect. 2) The input from the internal mic device has very low volume and is very noisy; I tried to optimize fiddling around with hda_verb and the values in alsa-info; the output of codec#0 below shows the optimal ones. (the only relevant ones were gains on node 0x17). I managed, however, to saturate it by blowing directly into it, which shows that it is not really an amplification issue. Rather the real issue here is noise; Audacity shows ~-24dB and in 16bits the noise background gets up to ~+/-500 versus +/-50 with an external mic. The internal mic is indeed a stereo mic; the left channel only produces noise, the right channel seems to work, with the caveats above. (this could be very well that my mic is broken, even if I remember that at some point during my fiddling with hda_verb it started to work properly :) in any case I noticed that the noise background is very much correlated between R and L; would it be possible to write an alsa-plugin to mix the two channels with an inversion to mix away the noise? It sounds fairly simple, but I couldn't find any documentation on how to make a plugin I'm willing to invest some time on fixing this, so maybe I can get some help here to start with. Judging from the patch_conexant code it should be at least possible to fix #1. I'd like to hear your thoughts about #2; especially about fiddling around with hda_verb or making the described alsa-plugin. Thanks a lot Jacopo De Simoi !!################################ !!ALSA Information Script v 0.4.58 !!################################ !!Script ran on: Wed Jan 20 08:49:43 UTC 2010 !!Linux Distribution !!------------------ Gentoo Base System release 2.0.0 !!DMI Information !!--------------- Manufacturer: Hewlett-Packard Product Name: HP Pavilion dv2700 Notebook PC !!Kernel Information !!------------------ Kernel release: 2.6.30-tuxonice-r9 Operating System: GNU/Linux Architecture: x86_64 Processor: AMD Turion(tm) 64 X2 TL-62 SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.22.1 Library version: 1.0.22 Utilities version: 1.0.22 !!Loaded ALSA modules !!------------------- snd_hda_intel !!Sound Servers on this system !!---------------------------- No sound servers found. !!Soundcards recognised by ALSA !!----------------------------- 0 [NVidia ]: HDA-Intel - HDA NVidia HDA NVidia at 0xfc480000 irq 20 !!PCI Soundcards installed in the system !!-------------------------------------- 00:07.0 Audio device: nVidia Corporation MCP67 High Definition Audio (rev a1) !!Advanced information - PCI Vendor/Device/Susbsystem ID's !!-------------------------------------------------------- 00:07.0 0403: 10de:055c (rev a1) Subsystem: 103c:30d6 !!Modprobe options (Sound related) !!-------------------------------- snd-hda-intel: model=laptop snd-hda-intel: model=auto !!Loaded sound module options !!-------------------------- !!Module: snd_hda_intel bdl_pos_adj : 32,-1,-1,-1,-1,-1,-1,-1 beep_mode : 1,1,1,1,1,1,1,1 enable : Y,Y,Y,Y,Y,Y,Y,Y enable_msi : -1 id : <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> index : -1,-1,-1,-1,-1,-1,-1,-1 model : laptop,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> patch : <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> position_fix : 0,0,0,0,0,0,0,0 power_save : 0 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1 probe_only : N,N,N,N,N,N,N,N single_cmd : N !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: Conexant CX20561 (Hermosa) Address: 0 Function Id: 0x2 Vendor Id: 0x14f15051 Subsystem Id: 0x103c30d6 Revision Id: 0x100000 Modem Function Group: 0x2 Default PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=4, o=0, i=0, unsolicited=1, wake=0 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 0x10 [Audio Output] wcaps 0xc1d: Stereo Amp-Out R/L Control: name="Master Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="CONEXANT Analog", type="Audio", device=0 Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=0 Amp-Out vals: [0x3b 0x3b] Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Node 0x11 [Audio Output] wcaps 0xc1d: Stereo Amp-Out R/L Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=0 Amp-Out vals: [0x4a 0x4a] Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Node 0x12 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Node 0x13 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x03, nsteps=0x03, stepsize=0x17, mute=0 Amp-Out vals: [0x00] Node 0x14 [Audio Input] wcaps 0x100d1b: Stereo Amp-In R/L Control: name="Internal Mic Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Internal Mic Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="External Mic Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="External Mic Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Device: name="CONEXANT Analog", type="Audio", device=0 Amp-In caps: ofs=0x4a, nsteps=0x50, stepsize=0x03, mute=0 Amp-In vals: [0x00 0x00] [0x50 0x50] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 2 0x1d 0x17* Node 0x15 [Audio Input] wcaps 0x100d1b: Stereo Amp-In R/L Control: name="Docking Mic Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Docking Mic Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x4a, nsteps=0x50, stepsize=0x03, mute=0 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 1 0x18 Node 0x16 [Pin Complex] wcaps 0x400581: Stereo Pincap 0x0000001c: OUT HP Detect Pin Default 0x01214030: [Jack] HP Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x3, Sequence = 0x0 Pin-ctls: 0xc0: OUT HP Unsolicited: tag=37, enabled=1 Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x17 [Pin Complex] wcaps 0x40048b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-In vals: [0x04 0x04] Pincap 0x00001224: IN Detect Vref caps: 50 80 Pin Default 0x01a19020: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=38, enabled=1 Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Node 0x18 [Pin Complex] wcaps 0x40048b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00001224: IN Detect Vref caps: 50 80 Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=39, enabled=1 Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Node 0x19 [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 2 0x10 0x11* Node 0x1a [Pin Complex] wcaps 0x400501: Stereo Pincap 0x00010010: OUT EAPD EAPD 0x0: Pin Default 0x92170110: [Fixed] Speaker at Int Front Conn = Analog, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x1b [Pin Complex] wcaps 0x400500: Mono Pincap 0x00010010: OUT EAPD EAPD 0x0: Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x1c [Pin Complex] wcaps 0x400701: Stereo Digital Control: name="IEC958 Playback Con Mask", index=0, device=0 Control: name="IEC958 Playback Pro Mask", index=0, device=0 Control: name="IEC958 Playback Default", index=0, device=0 Control: name="IEC958 Playback Switch", index=0, device=0 Control: name="IEC958 Default PCM Playback Switch", index=0, device=0 Device: name="Conexant Digital", type="SPDIF", device=1 Pincap 0x00000010: OUT Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 1 0x12 Node 0x1d [Pin Complex] wcaps 0x40040b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-In vals: [0x04 0x04] Pincap 0x00000020: IN Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Node 0x1e [Vendor Defined Widget] wcaps 0xf00000: Mono --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw---- 1 root audio 116, 0 Jan 19 20:26 /dev/snd/controlC0 crw-rw---- 1 root audio 116, 4 Jan 19 20:26 /dev/snd/hwC0D0 crw-rw---- 1 root audio 116, 24 Jan 20 09:31 /dev/snd/pcmC0D0c crw-rw---- 1 root audio 116, 16 Jan 20 09:34 /dev/snd/pcmC0D0p crw-rw---- 1 root audio 116, 17 Jan 19 20:26 /dev/snd/pcmC0D1p crw-rw---- 1 root audio 116, 1 Jan 19 20:26 /dev/snd/seq crw-rw---- 1 root audio 116, 33 Jan 19 20:26 /dev/snd/timer !!Aplay/Arecord output !!------------ APLAY **** List of PLAYBACK Hardware Devices **** card 0: NVidia [HDA NVidia], device 0: CONEXANT Analog [CONEXANT Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: NVidia [HDA NVidia], device 1: Conexant Digital [Conexant Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: NVidia [HDA NVidia], device 0: CONEXANT Analog [CONEXANT Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [NVidia] Card hw:0 'NVidia'/'HDA NVidia at 0xfc480000 irq 20' Mixer name : 'Conexant CX20561 (Hermosa)' Components : 'HDA:14f15051,103c30d6,00100000' Controls : 15 Simple ctrls : 8 Simple mixer control 'Master',0 Capabilities: pvolume pswitch pswitch-joined penum Playback channels: Front Left - Front Right Limits: Playback 0 - 74 Mono: Front Left: Playback 59 [80%] [-15.00dB] [off] Front Right: Playback 59 [80%] [-15.00dB] [off] Simple mixer control 'PCM',0 Capabilities: pvolume penum Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [0.00dB] Front Right: Playback 255 [100%] [0.00dB] Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [on] Simple mixer control 'IEC958 Default PCM',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [on] Simple mixer control 'Digital',0 Capabilities: cvolume penum Capture channels: Front Left - Front Right Limits: Capture 0 - 120 Front Left: Capture 0 [0%] [-30.00dB] Front Right: Capture 0 [0%] [-30.00dB] Simple mixer control 'Docking Mic',0 Capabilities: volume pswitch penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 80 Front Left: 0 [0%] [-74.00dB] Playback [on] Front Right: 0 [0%] [-74.00dB] Playback [on] Simple mixer control 'External Mic',0 Capabilities: volume pswitch penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 80 Front Left: 80 [100%] [6.00dB] Playback [on] Front Right: 80 [100%] [6.00dB] Playback [on] Simple mixer control 'Internal Mic',0 Capabilities: volume pswitch penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 80 Front Left: 0 [0%] [-74.00dB] Playback [on] Front Right: 0 [0%] [-74.00dB] Playback [on] !!Alsactl output !!------------- --startcollapse-- state.NVidia { control.1 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 80' comment.dbmin -7400 comment.dbmax 600 iface MIXER name 'Internal Mic Volume' value.0 0 value.1 0 } control.2 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Internal Mic Switch' value.0 true value.1 true } control.3 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 80' comment.dbmin -7400 comment.dbmax 600 iface MIXER name 'External Mic Volume' value.0 80 value.1 80 } control.4 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'External Mic Switch' value.0 true value.1 true } control.5 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 80' comment.dbmin -7400 comment.dbmax 600 iface MIXER name 'Docking Mic Volume' value.0 0 value.1 0 } control.6 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Docking Mic Switch' value.0 true value.1 true } control.7 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 74' comment.dbmin -7400 comment.dbmax 0 iface MIXER name 'Master Playback Volume' value.0 59 value.1 59 } control.8 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Master Playback Switch' value false } control.9 { comment.access read comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Con Mask' value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.10 { comment.access read comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Pro Mask' value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.11 { comment.access 'read write' comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Default' value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.12 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'IEC958 Playback Switch' value true } control.13 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'IEC958 Default PCM Playback Switch' value true } control.14 { comment.access 'read write user' comment.type INTEGER comment.count 2 comment.range '0 - 255' comment.tlv '0000000100000008ffffec1400000014' comment.dbmin -5100 comment.dbmax 0 iface MIXER name 'PCM Playback Volume' value.0 255 value.1 255 } control.15 { comment.access 'read write user' comment.type INTEGER comment.count 2 comment.range '0 - 120' comment.tlv '0000000100000008fffff44800000032' comment.dbmin -3000 comment.dbmax 3000 iface MIXER name 'Digital Capture Volume' value.0 0 value.1 0 } } --endcollapse-- !!All Loaded Modules !!------------------ Module ndiswrapper bnep sco rfcomm l2cap snd_seq snd_seq_device snd_hda_codec_conexant uvcvideo videodev snd_hda_intel snd_hda_codec snd_hwdep snd_pcm snd_timer sdhci_pci sdhci mmc_core v4l1_compat nvidia forcedeth snd btusb bluetooth v4l2_compat_ioctl32 led_class snd_page_alloc !!Sysfs Files !!----------- /sys/class/sound/hwC0D0/init_pin_configs: 0x16 0x01214030 0x17 0x01a19020 0x18 0x400001f0 0x19 0x400001f0 0x1a 0x92170110 0x1b 0x400001f0 0x1c 0x400001f0 0x1d 0x400001f0 /sys/class/sound/hwC0D0/driver_pin_configs: /sys/class/sound/hwC0D0/user_pin_configs: /sys/class/sound/hwC0D0/init_verbs: !!ALSA/HDA dmesg !!------------------ [ 8.457898] ACPI: PCI Interrupt Link [LAZA] enabled at IRQ 20 [ 8.457916] HDA Intel 0000:00:07.0: PCI INT A -> Link[LAZA] -> GSI 20 (level, low) -> IRQ 20 [ 8.457970] HDA Intel 0000:00:07.0: setting latency timer to 64 [ 8.499562] Linux video capture interface: v2.00 -- [ 8.548267] USB Video Class driver (v0.1.0) [ 9.660207] input: HDA NVidia Mic as /class/input/input10 [ 9.660575] input: HDA NVidia Mic as /class/input/input11 [ 9.660798] input: HDA NVidia Headphone as /class/input/input12 [ 10.842906] kjournald starting. Commit interval 5 seconds -- [18156.018398] forcedeth 0000:00:0a.0: PCI INT A disabled [18156.400042] HDA Intel 0000:00:07.0: PCI INT A disabled [18156.440030] ehci_hcd 0000:00:04.1: PCI INT B disabled -- [18157.038988] ehci_hcd 0000:00:04.1: restoring config space at offset 0x1 (was 0xb00006, writing 0xb00002) [18157.039014] HDA Intel 0000:00:07.0: restoring config space at offset 0xf (was 0x5020100, writing 0x502010a) [18157.039022] HDA Intel 0000:00:07.0: restoring config space at offset 0x4 (was 0x0, writing 0xfc480000) [18157.039027] HDA Intel 0000:00:07.0: restoring config space at offset 0x1 (was 0xb00000, writing 0xb00002) [18157.039057] ahci 0000:00:09.0: restoring config space at offset 0x1 (was 0xb00007, writing 0xb00407) -- [18157.150089] amd74xx 0000:00:06.0: BIOS didn't set cable bits correctly. Enabling workaround. [18157.150111] HDA Intel 0000:00:07.0: PCI INT A -> Link[LAZA] -> GSI 20 (level, low) -> IRQ 20 [18157.150115] HDA Intel 0000:00:07.0: setting latency timer to 64 [18157.850034] pci 0000:00:08.0: setting latency timer to 64

1 1

[PATCH] config USE_MIXARTLOADER has never existed in KConfig for 2.6 linux kernels so this part of the check always evaluates to true thus removing the part of the check.
by Christoph Egger 20 Jan '10

20 Jan '10

Signed-off-by: Christoph Egger <siccegge(a)stud.informatik.uni-erlangen.de> --- sound/pci/mixart/mixart_hwdep.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/mixart/mixart_hwdep.c b/sound/pci/mixart/mixart_hwdep.c index 4cf4cd8..ef39a58 100644 --- a/sound/pci/mixart/mixart_hwdep.c +++ b/sound/pci/mixart/mixart_hwdep.c @@ -545,7 +545,7 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw #if defined(CONFIG_FW_LOADER) || defined(CONFIG_FW_LOADER_MODULE) -#if !defined(CONFIG_USE_MIXARTLOADER) && !defined(CONFIG_SND_MIXART) /* built-in kernel */ +#if !defined(CONFIG_SND_MIXART) /* built-in kernel */ #define SND_MIXART_FW_LOADER /* use the standard firmware loader */ #endif #endif -- 1.6.3.3 --+pHx0qQiF2pBVqBT--

1 0

[PATCH] config USE_MIXARTLOADER has never existed in KConfig for 2.6 linux kernels so this part of the check always evaluates to true thus removing the part of the check.
by Christoph Egger 20 Jan '10

20 Jan '10

The same is true for USE_PCXHRLOADER Signed-off-by: Christoph Egger <siccegge(a)stud.informatik.uni-erlangen.de> --- sound/pci/mixart/mixart_hwdep.c | 2 +- sound/pci/pcxhr/pcxhr_hwdep.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/mixart/mixart_hwdep.c b/sound/pci/mixart/mixart_hwdep.c index 4cf4cd8..ef39a58 100644 --- a/sound/pci/mixart/mixart_hwdep.c +++ b/sound/pci/mixart/mixart_hwdep.c @@ -545,7 +545,7 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw #if defined(CONFIG_FW_LOADER) || defined(CONFIG_FW_LOADER_MODULE) -#if !defined(CONFIG_USE_MIXARTLOADER) && !defined(CONFIG_SND_MIXART) /* built-in kernel */ +#if !defined(CONFIG_SND_MIXART) /* built-in kernel */ #define SND_MIXART_FW_LOADER /* use the standard firmware loader */ #endif #endif diff --git a/sound/pci/pcxhr/pcxhr_hwdep.c b/sound/pci/pcxhr/pcxhr_hwdep.c index 17cb123..16152ce 100644 --- a/sound/pci/pcxhr/pcxhr_hwdep.c +++ b/sound/pci/pcxhr/pcxhr_hwdep.c @@ -35,7 +35,7 @@ #if defined(CONFIG_FW_LOADER) || defined(CONFIG_FW_LOADER_MODULE) -#if !defined(CONFIG_USE_PCXHRLOADER) && !defined(CONFIG_SND_PCXHR) /* built-in kernel */ +#if !defined(CONFIG_SND_PCXHR) /* built-in kernel */ #define SND_PCXHR_FW_LOADER /* use the standard firmware loader */ #endif #endif -- 1.6.3.3 --liOOAslEiF7prFVr--

1 0

[PATCH] config USE_MIXARTLOADER has never existed in KConfig for 2.6 linux kernels so this part of the check always evaluates to true thus removing the part of the check.
by Christoph Egger 20 Jan '10

20 Jan '10

The same is true for USE_PCXHRLOADER and USE_VXLOADER Signed-off-by: Christoph Egger <siccegge(a)stud.informatik.uni-erlangen.de> --- include/sound/vx_core.h | 2 +- sound/pci/mixart/mixart_hwdep.c | 2 +- sound/pci/pcxhr/pcxhr_hwdep.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/include/sound/vx_core.h b/include/sound/vx_core.h index 5456343..c6835dd 100644 --- a/include/sound/vx_core.h +++ b/include/sound/vx_core.h @@ -28,7 +28,7 @@ #include <linux/interrupt.h> #if defined(CONFIG_FW_LOADER) || defined(CONFIG_FW_LOADER_MODULE) -#if !defined(CONFIG_USE_VXLOADER) && !defined(CONFIG_SND_VX_LIB) /* built-in kernel */ +#if !defined(CONFIG_SND_VX_LIB) /* built-in kernel */ #define SND_VX_FW_LOADER /* use the standard firmware loader */ #endif #endif diff --git a/sound/pci/mixart/mixart_hwdep.c b/sound/pci/mixart/mixart_hwdep.c index 4cf4cd8..ef39a58 100644 --- a/sound/pci/mixart/mixart_hwdep.c +++ b/sound/pci/mixart/mixart_hwdep.c @@ -545,7 +545,7 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw #if defined(CONFIG_FW_LOADER) || defined(CONFIG_FW_LOADER_MODULE) -#if !defined(CONFIG_USE_MIXARTLOADER) && !defined(CONFIG_SND_MIXART) /* built-in kernel */ +#if !defined(CONFIG_SND_MIXART) /* built-in kernel */ #define SND_MIXART_FW_LOADER /* use the standard firmware loader */ #endif #endif diff --git a/sound/pci/pcxhr/pcxhr_hwdep.c b/sound/pci/pcxhr/pcxhr_hwdep.c index 17cb123..16152ce 100644 --- a/sound/pci/pcxhr/pcxhr_hwdep.c +++ b/sound/pci/pcxhr/pcxhr_hwdep.c @@ -35,7 +35,7 @@ #if defined(CONFIG_FW_LOADER) || defined(CONFIG_FW_LOADER_MODULE) -#if !defined(CONFIG_USE_PCXHRLOADER) && !defined(CONFIG_SND_PCXHR) /* built-in kernel */ +#if !defined(CONFIG_SND_PCXHR) /* built-in kernel */ #define SND_PCXHR_FW_LOADER /* use the standard firmware loader */ #endif #endif -- 1.6.3.3 --bCsyhTFzCvuiizWE--

1 0

Re: [alsa-devel] [PATCH] ASoC: DaVinci: Fix stream restart error
by Chaithrika U S 20 Jan '10

20 Jan '10

All, Please ignore this patch, missed the 'To' address in the mail. I have re-submitted this patch. Regards, Chaithrika On Wed, Jan 20, 2010 at 17:04:45, alsa-devel(a)alsa-project.org wrote: > From: Chaithrika U S <chaithrika(a)ti.com> > > Sometimes after a suspend-resume cycle, the ALSA application > restarts the stream when resume fails and McASP fails to work > as the clock is not enabled. This patch corrects this bug. > > Testes on TI DA850/OMAP-L138 EVM. > > Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> > --- > Applies to ALSA GIT tree on branch topic/asoc at > http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=shortlog; > h=topic/asoc > > sound/soc/davinci/davinci-mcasp.c | 5 ++--- > 1 files changed, 2 insertions(+), 3 deletions(-) > > diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c > index a613bbb..ab6518d 100644 > --- a/sound/soc/davinci/davinci-mcasp.c > +++ b/sound/soc/davinci/davinci-mcasp.c > @@ -768,13 +768,12 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream, > > switch (cmd) { > case SNDRV_PCM_TRIGGER_RESUME: > + case SNDRV_PCM_TRIGGER_START: > + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: > if (!dev->clk_active) { > clk_enable(dev->clk); > dev->clk_active = 1; > } > - /* Fall through */ > - case SNDRV_PCM_TRIGGER_START: > - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: > davinci_mcasp_start(dev, substream->stream); > break; > > -- > 1.5.6 >

1 0

[alsa-devel] [PATCH 0/2] ASoC: tlv320dac33: Burst mode BCLK divider fixes
by Peter Ujfalusi 20 Jan '10

20 Jan '10

Hello, The following patches provides fixes for the BCLK clock divider configuration in tlv320dac33 codec. BCLK divider is used to specify the speed of the serial bus. The speed of the bus can be (must be) faster than the actual stream's sample rate in order to use the system in burst mode. With the introduction of mode7 FIFO mode, the BCLK handling got broken in this mode, the first patch fixes this issue. The second patch adds support for configuring the burst mode BCLK through platform data. --- Peter Ujfalusi (2): ASoC: tlv320dac33: BCLK divider fix ASoC: tlv320dac33: Burst mode BCLK divider configuration include/sound/tlv320dac33-plat.h | 1 + sound/soc/codecs/tlv320dac33.c | 22 +++++++++++++++++----- 2 files changed, 18 insertions(+), 5 deletions(-)

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[alsa-devel] Can we add optional parameters in snd_pcm_hw_params
by Raul Xiong 20 Jan '10

20 Jan '10

Hi all, In my current ALSA driver, I have defined a mode which will impact the configuration of the audio driver. The requirment is to ask application users to set this mode to driver. So I want to add a parameter in 'hwparams' of snd_pcm_hw_params(pcm_handle, hwparams). But it seems that ALSA have no such expansibility. Basically, my requirment is: There are two modes: ModeA and ModeB. I'd like application to set this mode before it begin playing. If you have other better methods to reach this requirment, it's also welcome. Thanks.

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[alsa-devel] State of the bugtracker
by Colin Guthrie 20 Jan '10

20 Jan '10

Hi, I remember a while back I asked about the bugtracker over at https://bugtrack.alsa-project.org/alsa-bug/my_view_page.php At the time it was generally regarded that the bugtracker was a bit of a black hole. I've generally avoided using it to report bugs, but I'd like to ask what it's current state is as I often have to direct users to *something* and generally go for the mailing list. So really this is just a "what is the state of the bug tracker" question. Do the main devs check it and triage it or is there no specific process? Cheers Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mandriva Linux Contributor [http://www.mandriva.com/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/]

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[alsa-devel] No alsa-driver snapshots since 2009-12-19?
by Aleksander Adamowski 20 Jan '10

20 Jan '10

Hi! Seems that alsa-driver snapshots have stopped being updated on http://ftp.kernel.org/pub/linux/kernel/people/tiwai/alsa/alsa-driver/ - the last one is from 2009-12-19. What happened? -- Best Regards, Aleksander Adamowski http://olo.org.pl

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[alsa-devel] [PATCH] Fix automute issue on HP T5735
by Lee Trager 20 Jan '10

20 Jan '10

The following patch fixes an issue where when you plug something into the headphone jack the internal speaker does not automute. This also will close alsa bug #0004766 Lee --- alsa-driver-orig/alsa-kernel/pci/hda/patch_realtek.c 2009-08-28 07:41:39.000000000 -0400 +++ alsa-driver/alsa-kernel/pci/hda/patch_realtek.c 2009-10-13 14:24:40.000000000 -0400 @@ -10028,7 +10028,9 @@ struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x15; - spec->autocfg.speaker_pins[0] = 0x0c; /* HACK: not actually a pin */ + spec->autocfg.speaker_pins[0] = 0x14; + + alc_automute_amp(codec); } static struct snd_kcontrol_new alc262_hp_t5735_mixer[] = {

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Re: [alsa-devel] [PATCH 1/2] ASoC: DaVinci: Voice Codec Support
by Miguel Aguilar 19 Jan '10

19 Jan '10

Hi Mark, I have been trying to test the AIC3x and the Voice Codec in the same kernel, however only the AIC3x is working, I'm having problem when I try to use the Voice Codec, I already have verified that the Voice Codec works since I tested alone. Please check the following output: root@dm365-evm:~# aplay -l **** List of PLAYBACK Hardware Devices **** card 0: EVM [DaVinci DM365 EVM], device 0: AIC3X tlv320aic3x-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: EVM [DaVinci DM365 EVM], device 1: CQ93 CQ93VC-1 [] Subdevices: 1/1 Subdevice #0: subdevice #0 root@dm365-evm:~# aplay -L default:CARD=EVM DaVinci DM365 EVM, Default Audio Device null Discard all samples (playback) or generate zero samples (capture) root@dm365-evm:~# cat /proc/asound/pcm 00-00: AIC3X tlv320aic3x-0 : : playback 1 : capture 1 00-01: CQ93 CQ93VC-1 : : playback 1 : capture 1 root@dm365-evm:~# cat /proc/asound/devices 0: [ 0] : control 16: [ 0- 0]: digital audio playback 17: [ 0- 1]: digital audio playback 24: [ 0- 0]: digital audio capture 25: [ 0- 1]: digital audio capture 33: : timer root@dm365-evm:~# cat /proc/asound/cards 0 [EVM ]: tlv320aic3x - DaVinci DM365 EVM DaVinci DM365 EVM (tlv320aic3x) root@dm365-evm:~# cat /proc/asound/timers G0: system timer : 10000.000us (10000000 ticks) P0-0-0: PCM playback 0-0-0 : SLAVE P0-0-1: PCM capture 0-0-1 : SLAVE P0-1-0: PCM playback 0-1-0 : SLAVE P0-1-1: PCM capture 0-1-1 : SLAVE root@dm365-evm:~# arecord -D plughw:0,0 test.wav Recording WAVE 'test.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono *** This is the error that I have. I not sure if it has to do with the root@dm365-evm:~# arecord -D plughw:0,1 test.wav asoc: can't open platform davinci-audio arecord: main:608: audio open error: No such device The following is the patch to the davinci-evm.c please check it if you see something wrong there: diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 7ccbe66..951d730 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -29,8 +29,10 @@ #include "../codecs/tlv320aic3x.h" #include "../codecs/spdif_transciever.h" +#include "../codecs/cq93vc.h" #include "davinci-pcm.h" #include "davinci-i2s.h" +#include "davinci-vcif.h" #include "davinci-mcasp.h" #define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \ @@ -151,6 +153,23 @@ static struct snd_soc_dai_link evm_dai = { .ops = &evm_ops, }; +static struct snd_soc_dai_link dm365_evm_dai[] = { + { + .name = "TLV320AIC3X", + .stream_name = "AIC3X", + .cpu_dai = &davinci_i2s_dai, + .codec_dai = &aic3x_dai, + .init = evm_aic3x_init, + .ops = &evm_ops, + }, + { + .name = "Voice Codec - CQ93VC", + .stream_name = "CQ93", + .cpu_dai = &davinci_vcif_dai, + .codec_dai = &cq93vc_dai, + }, +}; + static struct snd_soc_dai_link dm6467_evm_dai[] = { { .name = "TLV320AIC3X", @@ -177,7 +196,7 @@ static struct snd_soc_dai_link da8xx_evm_dai = { .ops = &evm_ops, }; -/* davinci dm6446, dm355 or dm365 evm audio machine driver */ +/* davinci dm6446, dm355 evm audio machine driver */ static struct snd_soc_card snd_soc_card_evm = { .name = "DaVinci EVM", .platform = &davinci_soc_platform, @@ -185,6 +204,14 @@ static struct snd_soc_card snd_soc_card_evm = { .num_links = 1, }; +/* davinci dm365 evm audio machine driver */ +static struct snd_soc_card dm365_snd_soc_card_evm = { + .name = "DaVinci DM365 EVM", + .platform = &davinci_soc_platform, + .dai_link = &dm365_evm_dai, + .num_links = ARRAY_SIZE(dm365_evm_dai), +}; + /* davinci dm6467 evm audio machine driver */ static struct snd_soc_card dm6467_snd_soc_card_evm = { .name = "DaVinci DM6467 EVM", @@ -217,6 +244,13 @@ static struct snd_soc_device evm_snd_devdata = { }; /* evm audio subsystem */ +static struct snd_soc_device dm365_evm_snd_devdata = { + .card = &dm365_snd_soc_card_evm, + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &aic3x_setup, +}; + +/* evm audio subsystem */ static struct snd_soc_device dm6467_evm_snd_devdata = { .card = &dm6467_snd_soc_card_evm, .codec_dev = &soc_codec_dev_aic3x, @@ -244,12 +278,15 @@ static int __init evm_init(void) int index; int ret; - if (machine_is_davinci_evm() || machine_is_davinci_dm365_evm()) { + if (machine_is_davinci_evm()) { evm_snd_dev_data = &evm_snd_devdata; index = 0; } else if (machine_is_davinci_dm355_evm()) { evm_snd_dev_data = &evm_snd_devdata; index = 1; + } else if (machine_is_davinci_dm365_evm()) { + evm_snd_dev_data = &dm365_evm_snd_devdata; + index = 0; } else if (machine_is_davinci_dm6467_evm()) { evm_snd_dev_data = &dm6467_evm_snd_devdata; index = 0; Thanks, Miguel Aguilar

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[alsa-devel] [PATCH] ASoC: fix compile breakage - add a missing header include
by Guennadi Liakhovetski 19 Jan '10

19 Jan '10

Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> --- It might well have been reported and / or fixed in some trees, sorry for the noise, if that's the case. include/sound/soc-dai.h | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index ca24e7f..061f16d 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -16,6 +16,8 @@ #include <linux/list.h> +#include <sound/soc.h> + struct snd_pcm_substream; /* -- 1.6.2.4

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[alsa-devel] [PATCH] ASoC: tlv320dac33: Correct the prefill number of samples
by Peter Ujfalusi 19 Jan '10

19 Jan '10

Set the prefill number of samples as the same as the lower threshold in mode7. In this way the codec will read the same amount of data on startup and during the running playback. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/tlv320dac33.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 2df9c20..65683aa 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -559,7 +559,7 @@ static inline void dac33_prefill_handler(struct tlv320dac33_priv *dac33) break; case DAC33_FIFO_MODE7: dac33_write16(codec, DAC33_PREFILL_MSB, - DAC33_THRREG(20)); + DAC33_THRREG(10)); break; default: dev_warn(codec->dev, "Unhandled FIFO mode: %d\n", -- 1.6.6

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[alsa-devel] About audio driver architecture
by Michael Qiu 19 Jan '10

19 Jan '10

Hi, I'm going to write an audio driver for a developping soc used for dvb set top box. As I've never done such things before, so I've some questions. 1. Seems the "ALSA Driver Documentation" is mainly for PCI(and for PC) based audio card. Where can I got a document for SOC which has no PCI bus. If there are some existing drivers it will be even better. 2. Is there any standard for such a audio driver which upper applications will depend on it? Or I can just follow ALSA? Best regards Michael Qiu

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Re: [alsa-devel] snd-hdsp driver error?
by Raymond Yau 19 Jan '10

19 Jan '10

2010/1/16 Lars Zobbe <fedesensfar(a)gmail.com> > Thansk a lot Raymond. That actually works! My HDSP now loads without mayor > problems. This is brilliant. I've been struggling for so long to make this > work and finally... > I recommend everyone with a RME HDSP, who wants to use pulseaudio, to check > out ticket 764. > Thanks again Raymond. > Lars. > > 2010/1/14 Raymond Yau <superquad.vortex2(a)gmail.com> > >> >> Refer to >> >> http://pulseaudio.org/ticket/764#comment:1 >> >> >> load-module module-alsa-sink device_id=0 tsched=0 >> > load-module module-alsa-source device_id=0 tsched=0 >> > if your hdsp works , this mean that this is a bug introduced at pulseaudio since 0.9.15 ( working profile ) I guess none of PA develpers had your card , if you want the pulseaudio bug fixed , you will need to post the output of full pulseaudio log of module-alsa-sink ( pulseaudio -k;pulseaudio -vvvvv ) and the outout of aplay -v -Dplughw:0,0 any.wav in the pulseaudio bug tracker so that PA developer can compare with the non-working pulseaudio log

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[alsa-devel] [PATCH 1/2] ASoC: DaVinci: Voice Codec Support
by miguel.aguilar＠ridgerun.com 18 Jan '10

18 Jan '10

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> There are two clients for the MFD driver: * Voice Codec Interface. * Voice Codec itself, CQ0093. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- drivers/mfd/Kconfig | 5 + drivers/mfd/Makefile | 1 + drivers/mfd/davinci_voicecodec.c | 216 ++++++++++++++++++++ include/linux/mfd/davinci_voicecodec.h | 113 +++++++++++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cq93vc.c | 342 ++++++++++++++++++++++++++++++++ sound/soc/codecs/cq93vc.h | 29 +++ sound/soc/davinci/Kconfig | 10 +- sound/soc/davinci/Makefile | 2 + sound/soc/davinci/davinci-vcif.h | 28 +++ 11 files changed, 750 insertions(+), 2 deletions(-) create mode 100644 drivers/mfd/davinci_voicecodec.c create mode 100644 include/linux/mfd/davinci_voicecodec.h create mode 100644 sound/soc/codecs/cq93vc.c create mode 100644 sound/soc/codecs/cq93vc.h create mode 100644 sound/soc/davinci/davinci-vcif.h diff --git a/drivers/mfd/Kconfig b/drivers/mfd/Kconfig index 8782978..efeba26 100644 --- a/drivers/mfd/Kconfig +++ b/drivers/mfd/Kconfig @@ -43,6 +43,11 @@ config MFD_SH_MOBILE_SDHI This driver supports the SDHI hardware block found in many SuperH Mobile SoCs. +config MFD_DAVINCI_VOICECODEC + bool "DaVinci Voice Codec" + help + This driver supports the DaVinci voice codec and its interface. + config MFD_DM355EVM_MSP bool "DaVinci DM355 EVM microcontroller" depends on I2C && MACH_DAVINCI_DM355_EVM diff --git a/drivers/mfd/Makefile b/drivers/mfd/Makefile index ca2f2c4..f5c617f 100644 --- a/drivers/mfd/Makefile +++ b/drivers/mfd/Makefile @@ -9,6 +9,7 @@ obj-$(CONFIG_MFD_SH_MOBILE_SDHI) += sh_mobile_sdhi.o obj-$(CONFIG_HTC_EGPIO) += htc-egpio.o obj-$(CONFIG_HTC_PASIC3) += htc-pasic3.o +obj-$(CONFIG_MFD_DAVINCI_VOICECODEC) += davinci_voicecodec.o obj-$(CONFIG_MFD_DM355EVM_MSP) += dm355evm_msp.o obj-$(CONFIG_MFD_T7L66XB) += t7l66xb.o diff --git a/drivers/mfd/davinci_voicecodec.c b/drivers/mfd/davinci_voicecodec.c new file mode 100644 index 0000000..94dbeaa --- /dev/null +++ b/drivers/mfd/davinci_voicecodec.c @@ -0,0 +1,216 @@ +/* + * DaVinci Voice Codec Core Interface for TI platforms + * + * Copyright (C) 2010 Texas Instruments, Inc + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/clk.h> + +#include <sound/pcm.h> + +#include <linux/mfd/core.h> +#include <linux/mfd/davinci_voicecodec.h> + +u32 davinci_vc_read(struct davinci_vc *davinci_vc, int reg) +{ + return __raw_readl(davinci_vc->base + reg); +} + +void davinci_vc_write(struct davinci_vc *davinci_vc, + int reg, u32 val) +{ + __raw_writel(val, davinci_vc->base + reg); +} + +static int davinci_vc_client_dev_register(struct davinci_vc *davinci_vc, + const char *name, + struct platform_device **pdev) +{ + int ret; + + *pdev = platform_device_alloc(name, -1); + if (pdev == NULL) { + dev_err(davinci_vc->dev, "failed to allocate %s\n", name); + return -ENODEV; + } + + (*pdev)->dev.parent = davinci_vc->dev; + platform_set_drvdata(*pdev, davinci_vc); + ret = platform_device_add(*pdev); + if (ret != 0) { + dev_err(davinci_vc->dev, "failed to register %s: %d\n", name, + ret); + platform_device_put(*pdev); + *pdev = NULL; + return ret; + } + + return 0; +} + +static int __init davinci_vc_probe(struct platform_device *pdev) +{ + struct davinci_vc *davinci_vc; + struct resource *res, *mem; + int ret; + + davinci_vc = kzalloc(sizeof(struct davinci_vc), GFP_KERNEL); + if (!davinci_vc) { + dev_dbg(&pdev->dev, + "could not allocate memory for private data\n"); + return -ENOMEM; + } + + davinci_vc->clk = clk_get(&pdev->dev, NULL); + if (IS_ERR(davinci_vc->clk)) { + dev_dbg(&pdev->dev, + "could not get the clock for voice codec\n"); + ret = -ENODEV; + goto fail1; + } + clk_enable(davinci_vc->clk); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(&pdev->dev, "no mem resource\n"); + ret = -ENODEV; + goto fail2; + } + + davinci_vc->pbase = res->start; + davinci_vc->base_size = resource_size(res); + + mem = request_mem_region(davinci_vc->pbase, davinci_vc->base_size, + pdev->name); + if (!mem) { + dev_err(&pdev->dev, "VCIF region already claimed\n"); + ret = -EBUSY; + goto fail2; + } + + davinci_vc->base = ioremap(davinci_vc->pbase, davinci_vc->base_size); + if (!davinci_vc->base) { + dev_err(&pdev->dev, "can't ioremap mem resource.\n"); + ret = -ENOMEM; + goto fail3; + } + + res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!res) { + dev_err(&pdev->dev, "no DMA resource\n"); + return -ENXIO; + } + + davinci_vc->davinci_vcif.dma_tx_channel = res->start; + davinci_vc->davinci_vcif.dma_tx_addr = + (dma_addr_t)(io_v2p(davinci_vc->base) + DAVINCI_VC_WFIFO); + + res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!res) { + dev_err(&pdev->dev, "no DMA resource\n"); + return -ENXIO; + } + + davinci_vc->davinci_vcif.dma_rx_channel = res->start; + davinci_vc->davinci_vcif.dma_rx_addr = + (dma_addr_t)(io_v2p(davinci_vc->base) + DAVINCI_VC_RFIFO); + + davinci_vc->dev = &pdev->dev; + + /* Register voice codec interface */ + ret = davinci_vc_client_dev_register(davinci_vc, "davinci_vcif", + &(davinci_vc->davinci_vcif.pdev)); + if (ret != 0) { + dev_err(&pdev->dev, "fail to register client device\n"); + goto fail4; + } + + /* Register voice codec */ + ret = davinci_vc_client_dev_register(davinci_vc, "cq93vc", + &(davinci_vc->cq93vc.pdev)); + if (ret != 0) { + dev_err(&pdev->dev, "fail to register client device\n"); + goto fail5; + } + + return 0; + +fail5: + platform_device_unregister(davinci_vc->davinci_vcif.pdev); +fail4: + iounmap(davinci_vc->base); +fail3: + release_mem_region(davinci_vc->pbase, davinci_vc->base_size); +fail2: + clk_disable(davinci_vc->clk); + clk_put(davinci_vc->clk); + davinci_vc->clk = NULL; +fail1: + kfree(davinci_vc); + + return ret; +} + +static int __devexit davinci_vc_remove(struct platform_device *pdev) +{ + struct davinci_vc *davinci_vc = platform_get_drvdata(pdev); + + platform_device_unregister(davinci_vc->cq93vc.pdev); + platform_device_unregister(davinci_vc->davinci_vcif.pdev); + + iounmap(davinci_vc->base); + release_mem_region(davinci_vc->pbase, davinci_vc->base_size); + + clk_disable(davinci_vc->clk); + clk_put(davinci_vc->clk); + davinci_vc->clk = NULL; + + kfree(davinci_vc); + + return 0; +} + +static struct platform_driver davinci_vc_driver = { + .driver = { + .name = "davinci_voicecodec", + .owner = THIS_MODULE, + }, + .remove = __devexit_p(davinci_vc_remove), +}; + +static int __init davinci_vc_init(void) +{ + return platform_driver_probe(&davinci_vc_driver, davinci_vc_probe); +} +module_init(davinci_vc_init); + +static void __exit davinci_vc_exit(void) +{ + platform_driver_unregister(&davinci_vc_driver); +} +module_exit(davinci_vc_exit); + +MODULE_AUTHOR("Miguel Aguilar"); +MODULE_DESCRIPTION("Texas Instruments DaVinci Voice Codec Core Interface"); +MODULE_LICENSE("GPL"); diff --git a/include/linux/mfd/davinci_voicecodec.h b/include/linux/mfd/davinci_voicecodec.h new file mode 100644 index 0000000..6a1757a --- /dev/null +++ b/include/linux/mfd/davinci_voicecodec.h @@ -0,0 +1,113 @@ +/* + * DaVinci Voice Codec Core Interface for TI platforms + * + * Copyright (C) 2010 Texas Instruments, Inc + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef __LINUX_MFD_DAVINCI_VOICECODEC_H_ +#define __LINUX_MFD_DAVINIC_VOICECODEC_H_ + +#include <linux/kernel.h> +#include <linux/platform_device.h> +#include <mach/edma.h> + +/* + * Register values. + */ +#define DAVINCI_VC_PID 0x00 +#define DAVINCI_VC_CTRL 0x04 +#define DAVINCI_VC_INTEN 0x08 +#define DAVINCI_VC_INTSTATUS 0x0c +#define DAVINCI_VC_INTCLR 0x10 +#define DAVINCI_VC_EMUL_CTRL 0x14 +#define DAVINCI_VC_RFIFO 0x20 +#define DAVINCI_VC_WFIFO 0x24 +#define DAVINCI_VC_FIFOSTAT 0x28 +#define DAVINCI_VC_TST_CTRL 0x2C +#define DAVINCI_VC_REG05 0x94 +#define DAVINCI_VC_REG09 0xA4 +#define DAVINCI_VC_REG12 0xB0 + +/* DAVINCI_VC_CTRL bit fields */ +#define DAVINCI_VC_CTRL_MASK 0x5500 +#define DAVINCI_VC_CTRL_RSTADC BIT(0) +#define DAVINCI_VC_CTRL_RSTDAC BIT(1) +#define DAVINCI_VC_CTRL_RD_BITS_8 BIT(4) +#define DAVINCI_VC_CTRL_RD_UNSIGNED BIT(5) +#define DAVINCI_VC_CTRL_WD_BITS_8 BIT(6) +#define DAVINCI_VC_CTRL_WD_UNSIGNED BIT(7) +#define DAVINCI_VC_CTRL_RFIFOEN BIT(8) +#define DAVINCI_VC_CTRL_RFIFOCL BIT(9) +#define DAVINCI_VC_CTRL_RFIFOMD_WORD_1 BIT(10) +#define DAVINCI_VC_CTRL_WFIFOEN BIT(12) +#define DAVINCI_VC_CTRL_WFIFOCL BIT(13) +#define DAVINCI_VC_CTRL_WFIFOMD_WORD_1 BIT(14) + +/* DAVINCI_VC_INT bit fields */ +#define DAVINCI_VC_INT_MASK 0x3F +#define DAVINCI_VC_INT_RDRDY_MASK BIT(0) +#define DAVINCI_VC_INT_RERROVF_MASK BIT(1) +#define DAVINCI_VC_INT_RERRUDR_MASK BIT(2) +#define DAVINCI_VC_INT_WDREQ_MASK BIT(3) +#define DAVINCI_VC_INT_WERROVF_MASKBIT BIT(4) +#define DAVINCI_VC_INT_WERRUDR_MASK BIT(5) + +/* DAVINCI_VC_REG05 bit fields */ +#define DAVINCI_VC_REG05_PGA_GAIN 0x07 + +/* DAVINCI_VC_REG09 bit fields */ +#define DAVINCI_VC_REG09_MUTE 0x40 +#define DAVINCI_VC_REG09_DIG_ATTEN 0x3F + +/* DAVINCI_VC_REG12 bit fields */ +#define DAVINCI_VC_REG12_POWER_ALL_ON 0xFD +#define DAVINCI_VC_REG12_POWER_ALL_OFF 0x00 + +struct davinci_vcif { + struct platform_device *pdev; + u32 dma_tx_channel; + u32 dma_rx_channel; + dma_addr_t dma_tx_addr; + dma_addr_t dma_rx_addr; +}; + +struct cq93vc { + struct platform_device *pdev; + struct snd_soc_codec *codec; + u32 sysclk; +}; + +struct davinci_vc; + +struct davinci_vc { + /* Device data */ + struct device *dev; + struct clk *clk; + + /* Memory resources */ + void __iomem *base; + resource_size_t pbase; + size_t base_size; + + /* Client devices */ + struct davinci_vcif davinci_vcif; + struct cq93vc cq93vc; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 52b005f..a759d06 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -21,6 +21,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4535 if I2C select SND_SOC_AK4642 if I2C select SND_SOC_AK4671 if I2C + select SND_SOC_CQ0093VC select SND_SOC_CS4270 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 @@ -108,6 +109,9 @@ config SND_SOC_AK4642 config SND_SOC_AK4671 tristate +config SND_SOC_CQ0093VC + tristate + # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index dbaecb1..674db53 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -8,6 +8,7 @@ snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o +snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs4270-objs := cs4270.o snd-soc-cx20442-objs := cx20442.o snd-soc-l3-objs := l3.o @@ -64,6 +65,7 @@ obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o +obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c new file mode 100644 index 0000000..f420e01 --- /dev/null +++ b/sound/soc/codecs/cq93vc.c @@ -0,0 +1,342 @@ +/* + * ALSA SoC CQ0093 Voice Codec Driver for DaVinci platforms + * + * Copyright (C) 2010 Texas Instruments, Inc + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/io.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/device.h> +#include <linux/clk.h> +#include <linux/mfd/davinci_voicecodec.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include <mach/dm365.h> + +#include "cq93vc.h" + +static inline unsigned int cq93vc_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + struct davinci_vc *davinci_vc = codec->control_data; + + return readl(davinci_vc->base + reg); +} + +static inline int cq93vc_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + struct davinci_vc *davinci_vc = codec->control_data; + + writel(value, davinci_vc->base + reg); + + return 0; +} + +static const struct snd_kcontrol_new cq93vc_snd_controls[] = { + SOC_SINGLE("PGA Capture Volume", DAVINCI_VC_REG05, 0, 0x03, 0), + SOC_SINGLE("Mono DAC Playback Volume", DAVINCI_VC_REG09, 0, 0x3f, 0), +}; + +static int cq93vc_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u8 reg = cq93vc_read(codec, DAVINCI_VC_REG09) & ~DAVINCI_VC_REG09_MUTE; + + if (mute) + cq93vc_write(codec, DAVINCI_VC_REG09, + reg | DAVINCI_VC_REG09_MUTE); + else + cq93vc_write(codec, DAVINCI_VC_REG09, reg); + + return 0; +} + +static int cq93vc_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(cq93vc_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&cq93vc_snd_controls[i], + codec, NULL)); + if (err < 0) + return err; + } + + return 0; +} + +static int cq93vc_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct davinci_vc *davinci_vc = codec->control_data; + + switch (freq) { + case 22579200: + case 27000000: + case 33868800: + davinci_vc->cq93vc.sysclk = freq; + return 0; + } + + return -EINVAL; +} + +static int cq93vc_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + cq93vc_write(codec, DAVINCI_VC_REG12, + DAVINCI_VC_REG12_POWER_ALL_ON); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + cq93vc_write(codec, DAVINCI_VC_REG12, + DAVINCI_VC_REG12_POWER_ALL_OFF); + break; + case SND_SOC_BIAS_OFF: + /* force all power off */ + cq93vc_write(codec, DAVINCI_VC_REG12, + DAVINCI_VC_REG12_POWER_ALL_OFF); + break; + } + codec->bias_level = level; + + return 0; +} + +#define CQ93VC_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000) +#define CQ93VC_FORMATS (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE) + +static struct snd_soc_dai_ops cq93vc_dai_ops = { + .digital_mute = cq93vc_mute, + .set_sysclk = cq93vc_set_dai_sysclk, +}; + +struct snd_soc_dai cq93vc_dai = { + .name = "CQ93VC", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CQ93VC_RATES, + .formats = CQ93VC_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CQ93VC_RATES, + .formats = CQ93VC_FORMATS,}, + .ops = &cq93vc_dai_ops, +}; +EXPORT_SYMBOL_GPL(cq93vc_dai); + +static int cq93vc_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + cq93vc_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int cq93vc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + cq93vc_set_bias_level(codec, codec->suspend_bias_level); + + return 0; +} + +static struct snd_soc_codec *cq93vc_codec; + +static int cq93vc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct device *dev = &pdev->dev; + struct snd_soc_codec *codec; + int ret; + + socdev->card->codec = cq93vc_codec; + codec = socdev->card->codec; + + /* Set the PGA Gain to 18 dB */ + cq93vc_write(codec, DAVINCI_VC_REG05, DAVINCI_VC_REG05_PGA_GAIN); + + /* Set the DAC digital attenuation to 0 dB */ + cq93vc_write(codec, DAVINCI_VC_REG09, DAVINCI_VC_REG09_DIG_ATTEN); + + /* Register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(dev, "%s: failed to create pcms\n", pdev->name); + return ret; + } + + /* Set controls */ + cq93vc_add_controls(codec); + + /* Off, with power on */ + cq93vc_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +static int cq93vc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + /* Power down chip */ + if (codec->control_data) + cq93vc_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_cq93vc = { + .probe = cq93vc_probe, + .remove = cq93vc_remove, + .suspend = cq93vc_suspend, + .resume = cq93vc_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_cq93vc); + +static __init int cq93vc_codec_probe(struct platform_device *pdev) +{ + struct davinci_vc *davinci_vc = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret; + + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) { + dev_dbg(davinci_vc->dev, + "could not allocate memory for codec data\n"); + return -ENOMEM; + } + + davinci_vc->cq93vc.codec = codec; + + cq93vc_dai.dev = &pdev->dev; + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + codec->dev = &pdev->dev; + codec->name = "CQ93VC"; + codec->owner = THIS_MODULE; + codec->read = cq93vc_read; + codec->write = cq93vc_write; + codec->set_bias_level = cq93vc_set_bias_level; + codec->dai = &cq93vc_dai; + codec->num_dai = 1; + codec->control_data = davinci_vc; + + cq93vc_codec = codec; + + ret = snd_soc_register_dai(&cq93vc_dai); + if (ret != 0) { + dev_err(davinci_vc->dev, "could register dai\n"); + goto fail1; + } + return 0; + +fail1: + kfree(codec); + cq93vc_codec = NULL; + + return ret; +} + +static int __devexit cq93vc_codec_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + snd_soc_unregister_dai(&cq93vc_dai); + + kfree(codec); + cq93vc_codec = NULL; + + return 0; +} + +#ifdef CONFIG_PM +static int cq93vc_codec_suspend(struct platform_device *pdev, pm_message_t m) +{ + return snd_soc_suspend_device(&pdev->dev); +} + +static int cq93vc_codec_resume(struct platform_device *pdev) +{ + return snd_soc_resume_device(&pdev->dev); +} +#else +#define cq93vc_codec_suspend NULL +#define cq93vc_codec_resume NULL +#endif + +static struct platform_driver cq93vc_codec_driver = { + .driver = { + .name = "cq93vc", + .owner = THIS_MODULE, + }, + .probe = cq93vc_codec_probe, + .remove = __devexit_p(cq93vc_codec_remove), + .suspend = cq93vc_codec_suspend, + .resume = cq93vc_codec_resume, +}; + +static __init int cq93vc_init(void) +{ + return platform_driver_probe(&cq93vc_codec_driver, cq93vc_codec_probe); +} +module_init(cq93vc_init); + +static __exit void cq93vc_exit(void) +{ + platform_driver_unregister(&cq93vc_codec_driver); +} +module_exit(cq93vc_exit); + +MODULE_DESCRIPTION("Texas Instruments DaVinci ASoC CQ0093 Voice Codec Driver"); +MODULE_AUTHOR("Miguel Aguilar"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cq93vc.h b/sound/soc/codecs/cq93vc.h new file mode 100644 index 0000000..845b196 --- /dev/null +++ b/sound/soc/codecs/cq93vc.h @@ -0,0 +1,29 @@ +/* + * ALSA SoC CQ0093 Voice Codec Driver for DaVinci platforms + * + * Copyright (C) 2010 Texas Instruments, Inc + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef _CQ93VC_H +#define _CQ93VC_H + +extern struct snd_soc_dai cq93vc_dai; +extern struct snd_soc_codec_device soc_codec_dev_cq93vc; + +#endif diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 047ee39..3aa04e6 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -12,15 +12,21 @@ config SND_DAVINCI_SOC_I2S config SND_DAVINCI_SOC_MCASP tristate +config SND_DAVINCI_SOC_VCIF + tristate + config SND_DAVINCI_SOC_EVM tristate "SoC Audio support for DaVinci DM6446, DM355 or DM365 EVM" depends on SND_DAVINCI_SOC - depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM || MACH_DAVINCI_DM365_EVM + depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM || MACH_DAVINCI_DM365_EVM select SND_DAVINCI_SOC_I2S select SND_SOC_TLV320AIC3X + select MFD_DAVINCI_VOICECODEC + select SND_DAVINCI_SOC_VCIF + select SND_SOC_CQ0093VC help Say Y if you want to add support for SoC audio on TI - DaVinci DM6446 or DM355 EVM platforms. + DaVinci DM6446, DM355 or DM365 EVM platforms. config SND_DM6467_SOC_EVM tristate "SoC Audio support for DaVinci DM6467 EVM" diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index a6939d7..a93679d 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -2,10 +2,12 @@ snd-soc-davinci-objs := davinci-pcm.o snd-soc-davinci-i2s-objs := davinci-i2s.o snd-soc-davinci-mcasp-objs:= davinci-mcasp.o +snd-soc-davinci-vcif-objs:= davinci-vcif.o obj-$(CONFIG_SND_DAVINCI_SOC) += snd-soc-davinci.o obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o obj-$(CONFIG_SND_DAVINCI_SOC_MCASP) += snd-soc-davinci-mcasp.o +obj-$(CONFIG_SND_DAVINCI_SOC_VCIF) += snd-soc-davinci-vcif.o # DAVINCI Machine Support snd-soc-evm-objs := davinci-evm.o diff --git a/sound/soc/davinci/davinci-vcif.h b/sound/soc/davinci/davinci-vcif.h new file mode 100644 index 0000000..571c994 --- /dev/null +++ b/sound/soc/davinci/davinci-vcif.h @@ -0,0 +1,28 @@ +/* + * ALSA SoC Voice Codec Interface for TI DAVINCI processor + * + * Copyright (C) 2010 Texas Instruments. + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef _DAVINCI_VCIF_H +#define _DAVINCI_VCIF_H + +extern struct snd_soc_dai davinci_vcif_dai; + +#endif -- 1.6.0.4

3 3

[alsa-devel] Echo Indigo card on Sabayon Gentoo
by S. Aguinaga 18 Jan '10

18 Jan '10

Hello Fellows, Just downloaded the latest (released) alsa-lib, alsa-driver, and alsa-utils, but my Indigo card does not work. It has been tested on the same system using the same drivers and on the same system using Fedora 10, that works fine. Do you know what I might have to do to get it working on Gentoo (sabayon)? When I type the following, this is what I get: ~/Downloads/alsa-utils-1.0.22/alsactl $ ./alsactl init 2 Unknown hardware: "Echo_Indigo IO" "" "" "0xecc0" "0x00a0" Hardware is initialized using a guess method Using aplay to list the device: card 2: IO [Indigo IO], device 0: PCM [Indigo IO] Subdevices: 8/8 .... etc. Using aplay to test the card here is the command: aplay -D plughw:2,0 <wav file> And this does not work, I have checked that things are ok with alsamixer, but still! Any help will be appreciated or let me know if this is best addressed by submitting a bug. // Salvador

2 2

[alsa-devel] WARNING: at sound/core/sound_oss.c:96 snd_oss_kernel_minor+0xdf/0x120 [snd]()
by John Kacur 18 Jan '10

18 Jan '10

The following was in my 2.6.33-rc3 dmesg (of course that doesn't mean that the problem didn't exist earlier too) WARNING: at /home/jkacur/rt.linux.git/sound/core/sound_oss.c:96 snd_oss_kernel_minor+0xdf/0x120 [snd]() Hardware name: 2241B48 BUG? (minor < 0 || minor >= 128) Modules linked in: snd_mixer_oss thinkpad_acpi(+) sdhci hwmon i2c_i801 firewire_ohci snd_pcm mmc_core snd_timer cfg80211 firewire_core btusb joydev iTCO_wdt bluetooth ata_generic e1000e i2c_core ppdev snd battery ac iTCO_vendor_support parport_pc ricoh_mmc intel_agp pata_acpi pcspkr parport tpm_tis video wmi output soundcore button tpm rfkill sr_mod tpm_bios snd_page_alloc crc_itu_t cdrom sg ahci libata sd_mod scsi_mod crc_t10dif xfs exportfs uhci_hcd ohci_hcd ehci_hcd [last unloaded: scsi_wait_scan] Pid: 1317, comm: modprobe Not tainted 2.6.33-rc3 #1 Call Trace: [<ffffffffa027bb25>] ? snd_oss_kernel_minor+0xdf/0x120 [snd] [<ffffffff8104338c>] warn_slowpath_common+0x7c/0xa9 [<ffffffff81043410>] warn_slowpath_fmt+0x41/0x43 [<ffffffffa027bb25>] snd_oss_kernel_minor+0xdf/0x120 [snd] [<ffffffffa027bc7e>] snd_register_oss_device+0x2c/0x21a [snd] [<ffffffffa0271e1e>] snd_mixer_oss_notify_handler+0xa5/0x2e5 [snd_mixer_oss] [<ffffffff81375e00>] ? __mutex_unlock_slowpath+0x5c/0x141 [<ffffffff810710d5>] ? trace_hardirqs_on_caller+0x11f/0x14a [<ffffffff8107110d>] ? trace_hardirqs_on+0xd/0xf sdhci-pci 0000:15:00.2: SDHCI controller found [1180:0822] (rev 21) sdhci-pci 0000:15:00.2: PCI INT C -> GSI 18 (level, low) -> IRQ 18 [<ffffffff81375ed3>] ? __mutex_unlock_slowpath+0x12f/0x141 sdhci-pci 0000:15:00.2: Will use DMA mode even though HW doesn't fully claim to support it. Registered led device: mmc0:: mmc0: SDHCI controller on PCI [0000:15:00.2] using DMA [<ffffffffa027628a>] snd_card_register+0x135/0x14b [snd] [<ffffffffa0370c88>] volume_init+0x335/0x3d2 [thinkpad_acpi] [<ffffffffa037289b>] thinkpad_acpi_module_init+0x61f/0xa4f [thinkpad_acpi] [<ffffffffa037227c>] ? thinkpad_acpi_module_init+0x0/0xa4f [thinkpad_acpi] [<ffffffff810001fa>] do_one_initcall+0x5f/0x154 [<ffffffff8107ff34>] sys_init_module+0xd7/0x234 [<ffffffff81002d5b>] system_call_fastpath+0x16/0x1b ---[ end trace a6d53dc318f09cb5 ]---

3 8

[alsa-devel] issue with sound stopping and not being able to restart
by Andrew Kohlsmith (Mailing List Account) 18 Jan '10

18 Jan '10

Good morning, I am new to ALSA development, but experienced in C. I wrote up a little test application which simply opens one playback and one capture device and spits noise into the playback and recording it on the capture device. I cross- correlate the data streams to determine the "depth" or latency of the pipe, end-to-end. I'm using playback and capture callbacks to handle the audio. The program seems to work reasonably well, but at seemingly random times either the playback or capture stream will "hiccup" (xrun) -- I am handling this case, but it's fairly often that the stream does NOT resume and I'm stuck with either no playback or no capture. I've looked at the example programs and the documentation and I do not see what it is I'm doing wrong. I'm posting my code here, hopefully someone can show me the error of my way. Capture/playback setup (error checking removed): snd_pcm_open(handle, fname, stream, SND_PCM_NONBLOCK); snd_pcm_hw_params_malloc(hwptr); snd_pcm_hw_params_any(h, hwp); snd_pcm_hw_params_set_access(h, hwp, SND_PCM_ACCESS_RW_INTERLEAVED); snd_pcm_hw_params_set_format(h, hwp, SND_PCM_FORMAT_S16_LE); snd_pcm_hw_params_set_rate(h, hwp, 8000, 0); snd_pcm_hw_params_set_channels(h, hwp, 2); snd_pcm_hw_params_set_buffer_size(h, hwp, BUF_SIZE); snd_pcm_hw_params(h, hwp); At this point I have the capture or playback device set up for 16 bit signed linear data in interleaved format and two channels. I've set up the buffers to be 160 frames, or 20ms. Each of these calls is checked for errors, and I do verify that the rate, buffer size and buffer time are what I expect. There is nothing unusual going on with these calls that I can determine. I set up the playback and capture callbacks thusly (again checking for errors) snd_async_add_pcm_handler(&ahp, hp, playback_callback, &pb_data); snd_async_add_pcm_handler(&ahc, hc, capture_callback, &cap_data); At this point I believe I'm ready to go, all I have to do is start playback and capture: if((fcount = snd_pcm_writei(h[PLAYBACK], pb_data.buf, BUF_SIZE)) != BUF_SIZE) { if(fcount < 0) { if((e = snd_pcm_recover(h[PLAYBACK], fcount, 0)) < 0) fprintf(stderr, "initial: can't writei: %s\n", snd_strerror(e)); } else { fprintf(stderr, "initial: short write: %d instead of %d\n", fcount, BUF_SIZE); } } if((e = snd_pcm_start(h[CAPTURE])) < 0) { fprintf(stderr, "capture prepare error: %s\n", snd_strerror(e)); return 1; } Now the playback and capture handlers are called by ALSA as required. The capture handler looks like this: /* capture callback. Called by ALSA whenever a period time is up */ void capture_callback(snd_async_handler_t *ah) { snd_pcm_t *h = snd_async_handler_get_pcm(ah); struct async_data *d = snd_async_handler_get_callback_private(ah); snd_pcm_sframes_t fa; short *p; int i, fc, e, faloops; struct timeval tv; static int c=0; /* read in whatever is available */ fa = snd_pcm_avail_update(h); faloops = 0; while(fa > 0) { if(fa > BUF_SIZE) fa = BUF_SIZE; if((fc = snd_pcm_readi(h, d->buf, fa)) != fa) { if(fc < 0) { if((e = snd_pcm_recover(h, e, 0)) < 0) { fprintf(stderr, "callback: can't readi: %s\n", snd_strerror(e)); } fc = 0; } else { fprintf(stderr, "callback: short read: %d instead of %ld\n", fc, fa); } } for(i=0,p=&d->buf[0]; i<fc; i++) { *d->ptr++ = *p++; p++; if(d->ptr > &in[MAX_BUFS * (BUF_SIZE - 1)]) { d->ptr = (short *)&in[0]; } } c += fc; if(c >= BUF_SIZE) { gettimeofday(&tv, NULL); c -= BUF_SIZE; in_step++; if(in_step >= MAX_BUFS) in_step = 0; } frames_cap += fc; fa = snd_pcm_avail_update(h); faloops++; }; cap_faloops=faloops; } The playback callback is very similar: /* playback callback. Called by ALSA whenever a period time is up */ void playback_callback(snd_async_handler_t *ah) { snd_pcm_t *h = snd_async_handler_get_pcm(ah); struct async_data *d = snd_async_handler_get_callback_private(ah); snd_pcm_sframes_t fa; int i, fc, e, faloops; static int c=0; struct timeval tv; fa = snd_pcm_avail_update(h); faloops = 0; while(fa > 0) { if(fa > BUF_SIZE) fa = BUF_SIZE; for(i=0; i<fa; i++) { d->buf[i*2] = awgn(&noise_source); d->buf[i*2+1] = awgn(&noise_source); } if((fc = snd_pcm_writei(h, d->buf, fa)) != fa) { if(fc < 0) { if((e = snd_pcm_recover(h, e, 0)) < 0) { fprintf(stderr, "callback: can't writei: error %x %s\n", e, snd_strerror(e)); } fc = 0; } else { fprintf(stderr, "callback: short write: %d instead of %ld\n", fc, fa); } } for(i=0; i<fc; i++) { *d->ptr++ = d->buf[i*2]; if(d->ptr > &out[MAX_BUFS * (BUF_SIZE - 1)]) { d->ptr = (short *)&out[0]; } } c += fc; if(c >= BUF_SIZE) { gettimeofday(&tv, NULL); c -= BUF_SIZE; out_step++; if(out_step >= MAX_BUFS) out_step = 0; } frames_pb += fc; fa = snd_pcm_avail_update(h); faloops++; }; pb_faloops = faloops; } That's really it; the main loop keeps track of the in_step and out_step variables and does the cross-correlation as the data comes in. Now it seems that it's the capture playback that is getting xruns, but I when capture xruns occur my playback callback seems to stop, and sometimes the entire program just hangs as if it is waiting for ALSA to call one of the callbacks. Can anyone shed any light on this? I'm certain it's something very simple and braindead, but I can't find it. Regards, Andrew

1 0

Re: [alsa-devel] ALSA DMIX support without MMAP ?
by ANISH KUMAR 18 Jan '10

18 Jan '10

> Actually i am working on soc platform(android-S3C6410) and wanted to > use DMIX and some good resampler for my software mixing instead of > current software mixing and resampling in upper layer(Above > alsa-lib).DMIX is working fine but this resampling is causing some > jitter.During rate conversion it looks for > libasound_module_rate_speexrate.so which i don't have so its using > default one causing the jitter. >>Have you considered using PulseAudio for this? Mark thanks for the suggestion and i checked out PulseAudio for my requirements.I found out that i don't need all the things in PulseAudio a. I just need good resampling code.Can i just use just the resampler code of pulseaudio(mine is mobile enviroment)? b. Just want a good mixer at ALSA-LIB level.As of now this is provided by android framework but i want this to be done by ALSA-LIB so that i can replace with some good mixer later on(if i find that some other mixer is good).DMIX is providing that but it's not that good. c.Jaroslav suggested to use sox resamplers but for that to work i have to get the sox resampler code(proper license) and need to build it for my system and link it with corresponding functions of ALSA-LIB so that they can call relevant functions for resampling.As i am working on mobile domain i think i need to do the integration of sox resampler code with ALSA-LIB for that to work?? or Can i just use the integrated sox library(some .so i guess) and change the asound.conf for using that library??Will this work?? d. I saw some check-ins related to Scenario Manager.As of now scenario manager functionality is also given in Android(we open a differnet device and the corresponding hooks for changing the routes dynamically) but i think if it given in alsa-lib switching will be much faster and this will increase the system performance.

2 1

[alsa-devel] [PATCH] add atomic operations for Blackfin parts
by Mike Frysinger 18 Jan '10

18 Jan '10

Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- include/iatomic.h | 23 +++++++++++++++++++++++ 1 files changed, 23 insertions(+), 0 deletions(-) diff --git a/include/iatomic.h b/include/iatomic.h index 8f6ec22..e92dbfd 100644 --- a/include/iatomic.h +++ b/include/iatomic.h @@ -1079,6 +1079,29 @@ static __inline__ int atomic_sub_return(int i, volatile atomic_t *v) #endif /* __sh__ */ +#ifdef __bfin__ + +#include <bfin_fixed_code.h> + +typedef struct { volatile int counter; } atomic_t; + +#define ATOMIC_INIT(i) { (i) } + +#define atomic_read(v) ((v)->counter) +#define atomic_set(v,i) (((v)->counter) = (i)) +#define atomic_add(i,v) bfin_atomic_add32(&(v)->counter, i) +#define atomic_sub(i,v) bfin_atomic_sub32(&(v)->counter, i) +#define atomic_inc(v) bfin_atomic_inc32(&(v)->counter); +#define atomic_dec(v) bfin_atomic_dec32(&(v)->counter); + +#define mb() __asm__ __volatile__ ("" : : : "memory") +#define rmb() mb() +#define wmb() mb() + +#define IATOMIC_DEFINED 1 + +#endif /* __bfin__ */ + #ifndef IATOMIC_DEFINED /* * non supported architecture. -- 1.6.6

3 4

[alsa-devel] ASoC updates for 2.6.34
by Mark Brown 18 Jan '10

18 Jan '10

The following changes since commit 735fe4cfbc3cedea41bd0ed31955054dae6beb46: Mark Brown (1): ASoC: Add missing __devexit and __devinit annotations are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.34 Daniel Mack (3): ASoC: ak4104: allow more sample rates ASoC: cs4270: allow passing freq=0 in set_dai_sysclk() ASoC: support more sample rates on raumfeld devices Mark Brown (6): ASoC: Fix i.MX audio build for i.MX3x ASoC: Convert new i.MX SSI driver to use static DAI array ASoC: i.MX SSI driver does not yet support master mode ASoC: Remove old i.MX driver code ASoC: Mark new i.MX drivers as BROKEN until arch/arm merged Merge branch 'mxc-audio' into for-2.6.34 Peter Ujfalusi (1): ASoC: TWL4030: Replace comma with semicolon in probe function Sascha Hauer (1): ASoC: Add a new imx-ssi sound driver Thomas Weber (1): ASoC: TWL4030: Fix typo in comment in header file arch/arm/plat-mxc/Makefile | 4 + arch/arm/plat-mxc/ssi-fiq-ksym.c | 20 + arch/arm/plat-mxc/ssi-fiq.S | 134 ++++++ sound/soc/codecs/ak4104.c | 4 +- sound/soc/codecs/cs4270.c | 38 ++- sound/soc/codecs/twl4030.c | 2 +- sound/soc/codecs/twl4030.h | 2 +- sound/soc/imx/Kconfig | 20 +- sound/soc/imx/Makefile | 12 +- sound/soc/imx/imx-pcm-dma-mx2.c | 313 ++++++++++++++ sound/soc/imx/imx-pcm-fiq.c | 277 ++++++++++++ sound/soc/imx/imx-ssi.c | 758 +++++++++++++++++++++++++++++++++ sound/soc/imx/imx-ssi.h | 237 +++++++++++ sound/soc/imx/mx1_mx2-pcm.c | 488 --------------------- sound/soc/imx/mx1_mx2-pcm.h | 26 -- sound/soc/imx/mx27vis_wm8974.c | 318 -------------- sound/soc/imx/mxc-ssi.c | 860 -------------------------------------- sound/soc/imx/mxc-ssi.h | 238 ----------- sound/soc/pxa/raumfeld.c | 61 ++- 19 files changed, 1823 insertions(+), 1989 deletions(-) create mode 100644 arch/arm/plat-mxc/ssi-fiq-ksym.c create mode 100644 arch/arm/plat-mxc/ssi-fiq.S create mode 100644 sound/soc/imx/imx-pcm-dma-mx2.c create mode 100644 sound/soc/imx/imx-pcm-fiq.c create mode 100644 sound/soc/imx/imx-ssi.c create mode 100644 sound/soc/imx/imx-ssi.h delete mode 100644 sound/soc/imx/mx1_mx2-pcm.c delete mode 100644 sound/soc/imx/mx1_mx2-pcm.h delete mode 100644 sound/soc/imx/mx27vis_wm8974.c delete mode 100644 sound/soc/imx/mxc-ssi.c delete mode 100644 sound/soc/imx/mxc-ssi.h

2 1

[alsa-devel] MSM DSP
by GNUtoo 17 Jan '10

17 Jan '10

hi, I've bought an HTC dream,before buying I looked at the feasibility of porting GNU/Linux on it(so not Android). There is a patch(for maing the driver standard) or a standard(everything is non-standard in android) driver for most of the components. The biggest problem seems the sound. So knowing that codeaurora( https://www.codeaurora.org/gitweb/quic/le/?p=kernel/msm.git ) had alsa in its kenrel I first tried to compile that kernel...but due to numerous bugs(It didn't compile and needed too much fixing) in it android-msm-2.6.29 branch I decided to import alsa in the linuxtogo kernel,and I pushed the sources at a temporary htc-alsa branch at git://gitorious.org/replicant/gnulinuxkernel.git (it's called gnulinux in order to differentiate from an android kernel,I didn't find a good name for it) from the commit message available here : http://gitorious.org/replicant/gnulinuxkernel/commit/92945ccd0921e940fc2675… I had to change snd_rpc_ids.vers = 0x00020001 into snd_rpc_ids.vers = 0xaa2b1a44 ,and to make minor build system changes and I had aplay working rapidely... but only aplay... mplayer was waiting indefinitely for the following system call: ioctl(6, 0x400c4150 ls -l /proc/$(pido mplayer)/fd/6 gave /dev/snd/pcmC0D0p and 0x400c4150 gave that: SNDRV_PCM_IOCTL_WRITEI_FRAMES = _IOW('A', 0x50, struct snd_xferi) Then with printks(I had no serial cable(I've looked for a craddle/docking station with the special HTC connector for easy soldering...but I didn't found one) so I couldn't use kgdb) I went until that point: rc = wait_event_interruptible(the_locks.write_wait, (frame->used == 0)|| (prtd->stopped)); That's in the alsa_send_buffer function in /sound/soc/msm/msm-pcm.c Then I continued the tracing puting printks everywhere...and I found that: with aplay: msm_pcm_playback_copy in the /sound/soc/msm/msm7k-pcm.c was called rc = alsa_send_buffer(prtd, buf, fbytes, NULL); was called once inside alsa_send_buffer (in /sound/soc/msm/msm-pcm.c) the count loop was only executed once because count was exhausted on the first run then copy_count is incremented and the audio is configured then at second call of the alsa_send_buffer the audio is configured and works. here's an output of the aplay parameters: $ aplay -v 05\ -\ Jet\ Set.wav Playing WAVE '05 - Jet Set.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Plug PCM: Hardware PCM card 0 'msm-audio' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 4800 period_size : 1200 period_time : 27210 tick_time : 0 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 1200 xfer_align : 1200 start_threshold : 4800 stop_threshold : 4800 silence_threshold: 0 silence_size : 0 boundary : 1258291200 note that frame = prtd->out + prtd->out_tail; makes frame->used have a bad value the first time it is run but as the count is only run once it doesn't import. the next runs prtd->out + prtd->out_tail have a good value with mplayer the story was quite different,even if the beginning was the same : msm_pcm_playback_copy in the /sound/soc/msm/msm7k-pcm.c was also called,rc = alsa_send_buffer(prtd, buf, fbytes, NULL); was also called inside alsa_send_buffer (in /sound/soc/msm/msm-pcm.c) but once in alsa_send_buffer,more precisely under the count loop the story changes: count is not exhausted the first time,because in "xfer = count > frame->size ? frame->size : count;" the smallest between count and frame->size is assigned,and so after it does "count -= xfer" and so count is not 0,because the buffer size detected by mplayer is 19200 insterad of 4800 so here what it does in detail: it passes 19200 as count here: fbytes = frames_to_bytes(runtime, frames); rc = alsa_send_buffer(prtd, buf, fbytes, NULL); that goes into that: ssize_t alsa_send_buffer(struct msm_audio *prtd, const char __user *buf, size_t count, loff_t *pos) and then the count variable is 19200 in alsa_send_buffer then there is xfer = count > frame->size ? frame->size : count; so it assigns the minimum between count and frame->size to xfer count is still 19200 frame->size is 4800 so xfer becomes 4800 then count -= xfer decrement count but only of 4800 so count is not exhausted and runs another time then : frame = prtd->out + prtd->out_tail; makes frame->used have a bad value,and then it blocks here: rc = wait_event_interruptible(the_locks.write_wait,(frame->used == 0) || (prtd->stopped)); because frame->used is not 0. here an mplayer output: Forced audio codec: mad Opening audio decoder: [pcm] Uncompressed PCM audio decoder dec_audio: Allocating 2048 + 65536 = 67584 bytes for output buffer. AUDIO: 44100 Hz, 2 ch, s16le, 1411.2 kbit/100.00% (ratio: 176400->176400) Selected audio codec: [pcm] afm: pcm (Uncompressed PCM) ========================================================================== Building audio filter chain for 44100Hz/2ch/s16le -> 0Hz/0ch/??... [libaf] Adding filter dummy [dummy] Was reinitialized: 44100Hz/2ch/s16le [dummy] Was reinitialized: 44100Hz/2ch/s16le Trying preferred audio driver 'alsa', options '[none]' alsa-init: requested format: 44100 Hz, 2 channels, 9 alsa-init: using ALSA 1.0.15 alsa-init: setup for 1/2 channel(s) alsa-init: using device default alsa-init: pcm opened in blocking mode alsa-init: got buffersize=19200 alsa-init: got period size 300 alsa: 44100 Hz/2 channels/4 bpf/19200 bytes buffer/Signed 16 bit Little Endian AO: [alsa] 44100Hz 2ch s16le (2 bytes per sample) AO: Description: ALSA-0.9.x-1.x audio output AO: Author: Alex Beregszaszi, Zsolt Barat <joy(a)streamminister.de> AO: Comment: under developement Building audio filter chain for 44100Hz/2ch/s16le -> 44100Hz/2ch/s16le... [dummy] Was reinitialized: 44100Hz/2ch/s16le [dummy] Was reinitialized: 44100Hz/2ch/s16le Video: no video Freeing 0 unused video chunks. Starting playback... Increasing filtered audio buffer size from 0 to 21248 so the quick solution was to force mplayer to use 4800 as buffer size with something like that: diff --git a/sound/soc/msm/msm7k-pcm.c b/sound/soc/msm/msm7k-pcm.c index 38e8283..bfbc66d 100644 --- a/sound/soc/msm/msm7k-pcm.c +++ b/sound/soc/msm/msm7k-pcm.c @@ -115,7 +115,7 @@ static struct snd_pcm_hardware msm_pcm_playback_hardware = { .rate_max = USE_RATE_MAX, .channels_min = USE_CHANNELS_MIN, .channels_max = USE_CHANNELS_MAX, - .buffer_bytes_max = MAX_BUFFER_PLAYBACK_SIZE, + .buffer_bytes_max = 4800, .period_bytes_min = 64, .period_bytes_max = MAX_PERIOD_SIZE, .periods_min = USE_PERIODS_MIN, but then mplayer seem to loop on an ioctl: mplayer is frozen here and doesn't respond to other input than ctrl+c: 0.0 (00.0) of 206.0 (03:26.0) ??,?% strace shows a lot of : nanosleep followed by SNDRV_PCM_IOCTL_STATUS (ioctl(6, 0x806c4120, 0xbe8bc9c8) = 0) an analyse of the status shows a constant : status->delay is 1200,and status->avail is 0,they don't vary over time and are printed just after that code in snd_pcm_status in sound/core/pcm_native.c if (runtime->status->state == SNDRV_PCM_STATE_RUNNING || runtime->status->state == SNDRV_PCM_STATE_DRAINING) status->delay = runtime->buffer_size - status->avail; else status->delay = 0; I was told in #mplayerdev that it was because the DSP didn't play the frames...indeed I hear no sound.... So how should I fix that mess? *did the quick and dirty fix cause the second problem? *should I use the DSP directly and make a pulseaudio plugin that talks to the android driver, instead(the driver seems very buggy) The good would be support for mp3 and similar things within the DSP,and that it should be more stable. The bad is that I already spent *a lot* of time with the alsa kernel driver...and that writing that will also require some userspaces changes such as: *modification of the phone routing in the "FSO" ( http://www.freesmartphone.org/index.php/Main_Page ) dbus framework to use pulseaudio *would require changes in the build system for using pulseaudio(openembedded(http://wiki.openembedded.net/index.php/Main_Page )) *may be slow...openmoko stopped to use pulseaudio because of speed issues. Denis

7 18

[alsa-devel] Audio buffer mmap-ing on sh arch vs. cache aliasing problem
by Pawel MOLL 16 Jan '10

16 Jan '10

Hi Guys, Recently I was investigating an issue with capturing audio using USB Audio Class device on a sh4-based board. "Bad voice quality" was reported... Finally I have traced the problem to something which is (unfortunately) well known to sh developers as a D-cache aliasing (or synonym) problem. Briefly speaking: due to some MMU design decisions, one can have two different virtual address pointing to the same physical location, which is fine, but going via different cache slots! So if there was a value of "0" in the memory and user "A" will write "1" there, user "B" will still read "0"... The solution is to ensure all TLB entries (so virtual memory areas) are beginning from a 16kB-aligned virtual address. Otherwise it is necessary to flush the cache between accesses from "A" and "B" sides. And now. The USB Audio Class driver (sound/usb/usbaudio.c) is allocating the sound buffer like this... static int snd_pcm_alloc_vmalloc_buffer(struct snd_pcm_substream *subs, size_t size) { [...] runtime->dma_area = vmalloc(size); [...] } ... and vmalloc will return a page(4k)-aligned pointer, possibly not 16k-aligned one. This is the source of all evil ;-) When using RW transfers everything is fine, as data is memcopied between two independent buffers. In a case of MMAP mode we will end up with such a situation: 1. The driver will memcpy data from an URB to the buffer. It will populate several cache lines. Let's call it a "kernel cache". 2. As the library has mapped the non-16k-aligned address via different cache line ("user cache"), it will read some rubbish from the physical memory, populating the "user cache" by the way. 3. Some time later the "kernel cache" is flushed, writing the data to the memory. 4. Some new data from URB is entering "kernel cache". 5. The library will access mmap-ed area again, going via the "user cache", which may or may not reflect the correct data (depending on the fact was the "user cache" flushed or not) etc. Of course this cycle is completely not deterministic, so some of the "kernel cache" lines will be flushed before being accessed from user space, other not... The final effect is... hmm... bizarre :-) First, of course, you will get a (probably loud) glitch (rubbish from buffer's underlying memory, before the first valid data is written back), and then something that could be described as an "echo" ;-) I mean - you are capturing "1 2 3 4 5 6 7..." and the result is (G stands for Glitch ;-) "G 1 23 3 4 4 6 67..." As a quick-and-dirty work-around I have modified snd_pcm_alloc_vmalloc_buffer() to allocate always 12kB more and then use 16k-aligned pointer: static int snd_pcm_alloc_vmalloc_buffer(struct snd_pcm_substream *subs, size_t size) { struct snd_pcm_runtime *runtime = subs->runtime; if (runtime->dma_addr) { if (runtime->dma_bytes >= size) return 0; /* already large enough */ vfree((void *)runtime->dma_addr); } runtime->dma_addr = (unsigned long)vmalloc(size + 12 * 1024); if (!runtime->dma_addr) return -ENOMEM; runtime->dma_area = (void *)ALIGN(runtime->dma_addr, 16 * 1024); runtime->dma_bytes = size; return 0; } Of course it cannot be regarded as anything more as a hack. And definitely not as a solution... So my question is: Any idea how the "proper solution" should look like? I see no obvious method to get an arch-independent "mmap-compliant" buffer. This problem was found on a sh arch, using an USB Audio Class device, however out there some other architectures are suffering from this MMU "feature" as well (ie. see http://docs.hp.com/en/B3906-90006/ch07s09.html) and possibly other drivers could behave "wrongly" (quoted as the driver is actually innocent...) To be honest I just have absolutely no idea what to do with this all! :-O I hope I was clear enough in the description... Any feedback, advise, idea etc. will be more than appreciated. Cheers Paweł

3 11

[alsa-devel] [PATCH] - fix speaker output for toshiba nb205 netbooks
by Nickolas 16 Jan '10

16 Jan '10

this is a quick fix i wrote up a couple days ago to fix speaker output on the Realtek ALC272 codec present in toshiba nb205 netbooks. presently, speaker output doesn't work at all on this model, regardless of settings in the mixer. this is because the current driver thinks the speaker is at nid 0x0c instead of the correct 0x17. i'm not too familiar with this hardware, so i'm not sure that this patch won't have any negative effects for other hardware. it also seems like a messy way of fixing the problem. any comments or suggestions are welcome, as is testing for those who have this board. diffed against the latest git. Thanks, Nick -- Signed-off-by: Nickolas Lloyd <ultrageek.lloyd(a)gmail.com> --- alsa-driver/sound/pci/hda/patch_realtek.c 2010-01-12 08:03:28.000000000 -0500 +++ alsa-driver-b/sound/pci/hda/patch_realtek.c 2010-01-12 16:27:48.000000000 -0500 @@ -192,6 +192,7 @@ enum { ALC663_ASUS_MODE8, ALC272_DELL, ALC272_DELL_ZM1, + ALC272_TOSHIBA_N200, ALC272_SAMSUNG_NC10, ALC662_AUTO, ALC662_MODEL_LAST, @@ -16725,6 +16726,11 @@ static struct hda_verb alc272_dell_init_ {} }; +static struct hda_verb alc272_toshiba_init_verbs[] = { + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {} +}; + static struct hda_verb alc663_mode7_init_verbs[] = { {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, @@ -17255,6 +17261,36 @@ static void alc663_g50v_inithook(struct alc_mic_automute(codec); } +static void alc272_toshiba_speaker_automute(struct hda_codec *codec) +{ + unsigned int present; + unsigned char bits; + + present = snd_hda_jack_detect(codec, 0x21); + bits = present ? HDA_AMP_MUTE : 0; + snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); +} + +static void alc272_toshiba_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case ALC880_HP_EVENT: + alc272_toshiba_speaker_automute(codec); + break; + case ALC880_MIC_EVENT: + alc_mic_automute(codec); + break; + } +} + +static void alc272_toshiba_init_hook(struct hda_codec *codec) +{ + alc272_toshiba_speaker_automute(codec); + alc_mic_automute(codec); +} + static struct snd_kcontrol_new alc662_ecs_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x02, 0x0, HDA_OUTPUT), ALC262_HIPPO_MASTER_SWITCH, @@ -17391,7 +17427,7 @@ static struct snd_pci_quirk alc662_cfg_t SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS), SND_PCI_QUIRK(0x105b, 0x0d47, "Foxconn 45CMX/45GMX/45CMX-K", ALC662_3ST_6ch_DIG), - SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB200", ALC663_ASUS_MODE4), + SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB200", ALC272_TOSHIBA_N200), SND_PCI_QUIRK(0x144d, 0xca00, "Samsung NC10", ALC272_SAMSUNG_NC10), SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L", ALC662_3ST_6ch_DIG), @@ -17696,6 +17732,22 @@ static struct alc_config_preset alc662_p .setup = alc663_m51va_setup, .init_hook = alc663_m51va_inithook, }, + [ALC272_TOSHIBA_N200] = { + .mixers = { alc663_asus_21jd_clfe_mixer }, + .cap_mixer = alc662_auto_capture_mixer, + .init_verbs = { alc662_init_verbs, + alc663_21jd_amic_init_verbs, + alc272_toshiba_init_verbs }, + .num_dacs = ARRAY_SIZE(alc662_dac_nids), + .hp_nid = 0x03, + .dac_nids = alc662_dac_nids, + .dig_out_nid = ALC662_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), + .channel_mode = alc662_3ST_2ch_modes, + .unsol_event = alc272_toshiba_unsol_event, + .setup = alc663_mode4_setup, + .init_hook = alc272_toshiba_init_hook, + }, [ALC272_SAMSUNG_NC10] = { .mixers = { alc272_nc10_mixer }, .init_verbs = { alc662_init_verbs,

2 5

[alsa-devel] Audio suspend/resume status on OMAP3 based platforms
by Aggarwal, Anuj 15 Jan '10

15 Jan '10

Hi, I want to check the status of suspend/resume functionality when audio subsystem is being used. Specifically, has some one tried the following tests on their omap3 based platforms: a) echo mem > /sys/power/state, when audio playback is running in background b) same command, when audio capture is running in background b) same command, when audio loopback is running in background I am facing some issues with (b) & (c) so wanted to confirm whether someone else has faced similar problem or not. Regards, Anuj Aggarwal

5 6

[alsa-devel] [PATCH] ALSA: ASoC: support more sample rates on raumfeld devices
by Daniel Mack 15 Jan '10

15 Jan '10

Add support for sample rates other than 44100Khz on raumfeld audio devices. Force the codec's supported sample rates as the table of supported rates which is calculated at run-time does not suffice: if we call snd_soc_dai_set_sysclk() at stream startup, half of the possible rates are missing. Use the external audio clock generator to provide double data rate clocks as the PXA's internal baud generator does anything but what's described in the datasheets. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Timur Tabi <timur(a)freescale.com> Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- sound/soc/pxa/raumfeld.c | 61 ++++++++++++++++++++++++++++++---------------- 1 files changed, 40 insertions(+), 21 deletions(-) diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c index e713181..8d65a49 100644 --- a/sound/soc/pxa/raumfeld.c +++ b/sound/soc/pxa/raumfeld.c @@ -41,7 +41,9 @@ static struct i2c_board_info max9486_hwmon_info = { }; #define MAX9485_MCLK_FREQ_112896 0x22 -#define MAX9485_MCLK_FREQ_122880 0x23 +#define MAX9485_MCLK_FREQ_122880 0x23 +#define MAX9485_MCLK_FREQ_225792 0x32 +#define MAX9485_MCLK_FREQ_245760 0x33 static void set_max9485_clk(char clk) { @@ -71,9 +73,18 @@ static int raumfeld_cs4270_startup(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - set_max9485_clk(MAX9485_MCLK_FREQ_112896); + codec_dai->playback.rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000; + codec_dai->playback.rate_min = 44100; + codec_dai->playback.rate_max = 48000; - return snd_soc_dai_set_sysclk(codec_dai, 0, 11289600, 0); + codec_dai->capture.rates = codec_dai->playback.rates; + codec_dai->capture.rate_min = codec_dai->playback.rate_min; + codec_dai->capture.rate_max = codec_dai->playback.rate_max; + + return 0; } static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream, @@ -86,20 +97,24 @@ static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream, int ret = 0; switch (params_rate(params)) { - case 8000: - case 16000: + case 44100: + set_max9485_clk(MAX9485_MCLK_FREQ_112896); + clk = 11289600; + break; case 48000: - case 96000: set_max9485_clk(MAX9485_MCLK_FREQ_122880); clk = 12288000; break; - case 11025: - case 22050: - case 44100: case 88200: - set_max9485_clk(MAX9485_MCLK_FREQ_112896); - clk = 11289600; + set_max9485_clk(MAX9485_MCLK_FREQ_225792); + clk = 22579200; + break; + case 96000: + set_max9485_clk(MAX9485_MCLK_FREQ_245760); + clk = 24576000; break; + default: + return -EINVAL; } fmt = SND_SOC_DAIFMT_I2S | @@ -128,7 +143,7 @@ static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, 0, 1); + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, clk, 1); if (ret < 0) return ret; @@ -181,20 +196,24 @@ static int raumfeld_ak4104_hw_params(struct snd_pcm_substream *substream, int fmt, ret = 0, clk = 0; switch (params_rate(params)) { - case 8000: - case 16000: + case 44100: + set_max9485_clk(MAX9485_MCLK_FREQ_112896); + clk = 11289600; + break; case 48000: - case 96000: set_max9485_clk(MAX9485_MCLK_FREQ_122880); clk = 12288000; break; - case 11025: - case 22050: - case 44100: case 88200: - set_max9485_clk(MAX9485_MCLK_FREQ_112896); - clk = 11289600; + set_max9485_clk(MAX9485_MCLK_FREQ_225792); + clk = 22579200; + break; + case 96000: + set_max9485_clk(MAX9485_MCLK_FREQ_245760); + clk = 24576000; break; + default: + return -EINVAL; } fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF; @@ -217,7 +236,7 @@ static int raumfeld_ak4104_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, 0, 1); + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, clk, 1); if (ret < 0) return ret; -- 1.6.5.2

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[alsa-devel] [PATCH] ALSA: Fix mute led GPIO on HP dv-series notebooks.
by Kunal Gangakhedkar 15 Jan '10

15 Jan '10

On my laptop (HP dv6-1110ax), there are no OEM strings in SMBIOS of type "HP_Mute_LED*". Hence, the GPIO for the mute button LED doesn't get set properly. I didn't find the strings in my cousin's laptop (HP dv9500t CTO) either. As per the documentation of find_mute_led_gpio(), these strings occur in HP B-series systems - so, before scanning the SMBIOS strings, we need to check if we're dealing with a B-series system. Need to get confirmation from HP if this logic takes care of all the systems. I'm trying to poke a friend there. Please let me know if this looks OK or needs changes. Signed-off-by: Kunal Gangakhedkar <kunal.gangakhedkar(a)gmail.com> --- sound/pci/hda/patch_sigmatel.c | 82 ++++++++++++++++++++++++++++++---------- 1 files changed, 62 insertions(+), 20 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 117919a..d421b3b 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4737,6 +4737,8 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res) } } +static int hp_blike_system(u32 subsystem_id); + /* * This method searches for the mute LED GPIO configuration * provided as OEM string in SMBIOS. The format of that string @@ -4748,6 +4750,19 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res) * * So, HP B-series like systems may have HP_Mute_LED_0 (current models) * or HP_Mute_LED_0_3 (future models) OEM SMBIOS strings + * + * + * The dv-series laptops don't seem to have the HP_Mute_LED* strings in + * SMBIOS - at least the ones I have seen do not have them - which include + * my own system (HP Pavilion dv6-1110ax) and my cousin's + * HP Pavilion dv9500t CTO. + * Need more information on whether it is true across the entire series. + * + * The "else" could be converted into a conditional check for dv-series - + * say, hp_dv_series_system() - just like hp_blike_system(); however, + * I don't have all the PCI ids to identify the dv-series machines - maybe, + * we can add that later. + * -- kunal */ static int find_mute_led_gpio(struct hda_codec *codec) { @@ -4755,29 +4770,54 @@ static int find_mute_led_gpio(struct hda_codec *codec) const struct dmi_device *dev = NULL; if ((codec->subsystem_id >> 16) == PCI_VENDOR_ID_HP) { - while ((dev = dmi_find_device(DMI_DEV_TYPE_OEM_STRING, + if (hp_blike_system(codec->subsystem_id)) { + while ((dev = dmi_find_device(DMI_DEV_TYPE_OEM_STRING, NULL, dev))) { - if (sscanf(dev->name, "HP_Mute_LED_%d_%d", - &spec->gpio_led_polarity, - &spec->gpio_led) == 2) { - spec->gpio_led = 1 << spec->gpio_led; - return 1; - } - if (sscanf(dev->name, "HP_Mute_LED_%d", - &spec->gpio_led_polarity) == 1) { - switch (codec->vendor_id) { - case 0x111d7608: - /* GPIO 0 */ - spec->gpio_led = 0x01; - return 1; - case 0x111d7600: - case 0x111d7601: - case 0x111d7602: - case 0x111d7603: - /* GPIO 3 */ - spec->gpio_led = 0x08; + if (sscanf(dev->name, "HP_Mute_LED_%d_%d", + &spec->gpio_led_polarity, + &spec->gpio_led) == 2) { + spec->gpio_led = 1 << spec->gpio_led; return 1; } + if (sscanf(dev->name, "HP_Mute_LED_%d", + &spec->gpio_led_polarity) == 1) { + switch (codec->vendor_id) { + case 0x111d7608: + /* GPIO 0 */ + spec->gpio_led = 0x01; + return 1; + case 0x111d7600: + case 0x111d7601: + case 0x111d7602: + case 0x111d7603: + /* GPIO 3 */ + spec->gpio_led = 0x08; + return 1; + } + } + } + } else { + /* + * Pavilion dv series notebooks do not seem to have + * HP_Mute_LED* in SMBIOS. + * If HP decides to add it in dv series, + * we'll need to fix this part + * - or merge it with the B-like check above. + */ + switch (codec->vendor_id) { + case 0x111d7608: + /* GPIO 0 */ + spec->gpio_led = 0x01; + spec->gpio_led_polarity |= spec->gpio_led; + return 1; + case 0x111d7600: + case 0x111d7601: + case 0x111d7602: + case 0x111d7603: + /* GPIO 3 */ + spec->gpio_led = 0x08; + spec->gpio_led_polarity |= spec->gpio_led; + return 1; } } } @@ -5603,6 +5643,8 @@ again: spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids); spec->num_smuxes = stac92hd71bxx_connected_smuxes(codec, 0x1e); + snd_printd("Found board config: %d\n", spec->board_config); + switch (spec->board_config) { case STAC_HP_M4: /* enable internal microphone */ -- 1.6.6.60.gc2ff1

2 8

[alsa-devel] [PATCH] alsa: use subsys_initcall for sound core instead of module_init
by Thadeu Lima de Souza Cascardo 15 Jan '10

15 Jan '10

This is needed for built-in drivers which are built before the sound directory, like thinkpad_acpi. Otherwise, registering a card fails. Signed-off-by: Thadeu Lima de Souza Cascardo <cascardo(a)holoscopio.com> --- sound/core/sound.c | 4 ++-- sound/sound_core.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/core/sound.c b/sound/core/sound.c index 7872a02..563d196 100644 --- a/sound/core/sound.c +++ b/sound/core/sound.c @@ -468,5 +468,5 @@ static void __exit alsa_sound_exit(void) unregister_chrdev(major, "alsa"); } -module_init(alsa_sound_init) -module_exit(alsa_sound_exit) +subsys_initcall(alsa_sound_init); +module_exit(alsa_sound_exit); diff --git a/sound/sound_core.c b/sound/sound_core.c index dbca7c9..7c2d677 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -61,7 +61,7 @@ static void __exit cleanup_soundcore(void) class_destroy(sound_class); } -module_init(init_soundcore); +subsys_initcall(init_soundcore); module_exit(cleanup_soundcore); -- 1.6.6

2 6

Re: [alsa-devel] ALSA DMIX support without MMAP ?
by ANISH KUMAR 15 Jan '10

15 Jan '10

>For dmix, the stop_threshold should be equal to boundary value, thus XRUN >check should never be true. If you got xrun with dmix, something is wrong. DMIX is working now. > but when i am > playing with Plug PCM there is some noise in the song why????Is rate > conversion algoritm not able to resample it properly? >The default linear resampler is not very good quality. You may use some >sox resamplers from the alsa-plugins package, but they're more CPU >intensive. Check google. Actually i am working on soc platform(android-S3C6410) and wanted to use DMIX and some good resampler for my software mixing instead of current software mixing and resampling in upper layer(Above alsa-lib).DMIX is working fine but this resampling is causing some jitter.During rate conversion it looks for libasound_module_rate_speexrate.so which i don't have so its using default one causing the jitter. Can i build this .so file (as current way of alsa-plugin downloading will not work for me)?For sox resampler i guess i need to use differnet .so and corresponding changes in asound.conf file?? Can i get speex or sox resampler code so that i can build it for my system????? This code(speex or sox) i guess would have been integrated with alsa-lib so that alsa-lib can call corresponding functions of resampler source code. OR simply point me to url as googling is not helping.Everywhere i am getting this "you need to download plugins".This will not work in my case.

2 1

[alsa-devel] soc-cache problem
by Cliff Cai 15 Jan '10

15 Jan '10

Hi all, Nowadays, soc tries to set up standard I/O functions with same addr_bits and data_bits for a codec. My problem is some codec might have registers with different data length.i.e.ADAU1761, Most of its registers are 8 bits long except PLL register,which is a 8 bytes long one.In order to set/read PLL , a block of data(8 bytes) needs to be transfer to/from the PLL register at a time. So, if soc-cache needs to consider this kind of case, or just keep it in local driver? Thanks Cliff

2 1

Re: [alsa-devel] VGN-P11Z_G (hda-intel ALC262) internal microphone not working
by Daniel Chen 15 Jan '10

15 Jan '10

Hi, Takashi, On Wed, Jan 13, 2010 at 12:28 PM, Christian Stöveken <christian.stoeveken(a)gmail.com> wrote: > sorry to get back to you so late - I tried again 5 minutes ago and it is now > working with the false parameter removed (/etc/modules:9: snd-hda-intel > model=vaio) > > working "model=auto": > http://www.alsa-project.org/db/?f=67f6a692f0ccd70d6893665d36c8b90a51f4a577 > > also with "power_save=10 power_save_controller=N model=auto": > http://www.alsa-project.org/db/?f=618f4d4ae530957b4b6d2b66751871d4073df685 > > I hope this is how you expected it to look like. > Also the headphones now work correctly. Attached is a patch against current sound-2.6 master HEAD that: * adds Christian's SSID to the relevant quirk table, * prefers BIOS autoprobing for existing Sony VAIO models matching 0x104d90xx. The second change was made to ease backing out the change in case of regressions. Best, -Dan

3 6

Re: [alsa-devel] [PATCH 0/5] PCM mmap (temporary) fixes for non-coherent architectures
by Takashi Iwai 15 Jan '10

15 Jan '10

At Fri, 1 Jan 2010 20:31:30 +0100, Andreas Mohr wrote: > > Hi, > > I've tried this patch set (with the typo-corrected part 4) on my ASUS > WL-500gP v2 MIPSEL via a backport to 2.6.31.9, but all I get is a > small blip of the sound I wanted to play, and then the system is fubar > (I believe just the same thing as what happened without having this patch > applied). As I mentioned in the previous followup, if your device is a USB-audio, the patch doesn't help because it's for devices with buffers using dma_alloc_coherent(). For USB-audio, it uses vmalloc for an intermediate buffer. Maybe this should be changed to dma_*() stuff for such architectures. Nevertheless, I don't know whether the crash is related with the audio part... Takashi

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Re: [alsa-devel] hda_intel.c patch requested for Teradici
by Maher Fahmi 15 Jan '10

15 Jan '10

ALSA Team, Sorry to bother you with this again. I haven't yet received confirmation (or rejection) from the moderator, and just wondering if there's anything else I can/should do. Best regards & Happy New Year, Maher. ________________________________ From: Maher Fahmi Sent: Friday, December 04, 2009 1:41 PM To: 'alsa-devel(a)alsa-project.org' Subject: hda_intel.c patch requested for Teradici Alsa Team, Enclosed is a patch to the hda_intel.c file from the alsa-driver-1.0.21 package. I tried to submit this using git, but ran into some problems that exceed my limited understanding of Linux. I hope this will suffice for you. Summary: Fix to a HW issue in which the 64OK register bit of the GCAP register is incorrectly asserted. Signed-off-by: Maher Fahmi mfahmi(a)teradici.com Thanks very much, Maher.

2 1

[alsa-devel] snd_pcm_avail_update returns 0
by Singaravelan Nallasellan 14 Jan '10

14 Jan '10

Hi, Our audio driver uses the indirect PCM data transfer mechanism to transfer the buffer pointers and the size of the buffer to the Audio engine which DMAs the PCM data to the sound card. The driver uses the .ack callback to receive the buffer pointer and the size. It also sends it to the audio engine. We are facing a problem when we play long wav file using our driver. The application has written the PCM data and the ring buffer is full. snd_pcm_avail_update returns 0 sporadically. But the .ack callback was not called. The playback is stuck. It is not progressing. I have tried with pulseaudio (0.9.15) and with my own player. Both gives the same result. I would appreciate if somebody could provide some pointer to identify the cause of this issue. Let me know if anybody needs more info. Thanks Sing

2 5

Re: [alsa-devel] snd-hdsp driver error?
by Raymond Yau 14 Jan '10

14 Jan '10

AFAIK , alsa-lib use the driver name to find the name of corresponding conf file from /usr/share/alsa/cards/aliases.conf In your case , H-DSP.conf does not exist and there is no aliases in /usr/share/alsa/cards/aliases.conf You have to ask Takashi or Jaroslav to fix this issue for your card open("/usr/share/alsa/cards/aliases.conf", O_RDONLY) = 3 access("/usr/share/alsa/cards/H-DSP.conf", R_OK) = -1 ENOENT (No such file or directory) open("/dev/snd/controlC1", O_RDONLY) = 3 close(3) = 0 write(2, "ALSA lib confmisc.c:1286:(snd_fu"..., 42ALSA lib confmisc.c:1286:(snd_func_refer) ) = 42 write(2, "Unable to find definition 'cards"..., 58Unable to find definition 'cards.H-DSP.pcm.front.0:CARD=0') = 58 write(2, "\n", 1 2009/11/7 Lars Zobbe <fedesensfar(a)gmail.com> > It is a clean install. > lars@lars-desktop:~$ lsb_release -a && uname -a > No LSB modules are available. > Distributor ID: Ubuntu > Description: Ubuntu 9.10 > Release: 9.10 > Codename: karmic > Linux lars-desktop 2.6.31-14-generic #48-Ubuntu SMP Fri Oct 16 14:05:01 UTC > 2009 x86_64 GNU/Linux > > It didn't work with pulseaudio in any previos version (of ubuntu or any > other distro I've tried) either. > And the standard answer from people "out there" is "forget about pulse it > doesn't work with the HDSP" f.x. this guy > http://ubuntuforums.org/showpost.php?p=7578331&postcount=10 > And that's why I contacted pulseaudio developers first when I found out the > it was because snd_hdsp didn't load with the preferred udev but only with > module-detect (se my first post). They said that I should inform Alsa > developers as there seems to be a problem with the alsa driver. > > I've attached the strace output. > > Thanks. > Lars. > > > > > On Fri, Nov 6, 2009 at 2:53 AM, Raymond Yau <superquad.vortex2(a)gmail.com>wrote: > >> Is it a clean install of Ubuntu 9.10 or upgrade from previous Ubuntu ? >> >> you may need to find out why aplay -v -Dfront 32bit.wav fail ? >> >> strace aplay -v -Dfront 32bit.wav >> >> >> >> >> >> !!Linux Distribution >> >> !!------------------ >> >> Ubuntu 9.10 \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu 9.10" >> >> !!Sound Servers on this system >> !!---------------------------- >> >> Pulseaudio: >> Installed - Yes (/usr/bin/pulseaudio) >> >> >> Running - Yes >> >> ESound Daemon: >> Installed - Yes (/usr/bin/esd) >> Running - No >> >> Jack: >> Installed - Yes (/usr/bin/jackd) >> Running - No >> >> >> !!ALSA configuration files >> >> >> !!------------------------ >> >> !!User specific config file (~/.asoundrc) >> >> pcm.!default { >> type plug >> slave { pcm "jack" } >> } >> >> pcm.jack { >> type jack >> playback_ports { >> >> 0 alsa_pcm:playback_1 >> >> 1 alsa_pcm:playback_2 >> } >> capture_ports { >> 0 alsa_pcm:capture_1 >> 1 alsa_pcm:capture_2 >> } >> } >> >> ctl.mixer0 { >> type hw >> card 0 >> } >> >> >> lars@lars-desktop:~$ aplay -v -Dfront:0 32bit.wav >> ALSA lib confmisc.c:1286:(snd_func_refer) Unable to find definition >> 'cards.H-DSP.pcm.front.0:CARD=0' >> ALSA lib conf.c:3513:(_snd_config_evaluate) function snd_func_refer >> returned error: No such file or directory >> ALSA lib conf.c:3985:(snd_config_expand) Evaluate error: No such file or >> directory >> ALSA lib pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM front:0 >> aplay: main:608: audio open error: No such file or directory >> >> >> Did PA developer tell you not to load oss module ? >> >> >> >> D: alsa-util.c: Checking for subset analog-mono (hw:%f) >> D: alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ... >> D: alsa-util.c: Managed to open hw:0 >> D: alsa-util.c: snd_pcm_hw_params_set_access() failed: Invalid argument >> D: alsa-util.c: Trying hw:0 without SND_PCM_NO_AUTO_FORMAT ... >> D: alsa-util.c: Managed to open hw:0 >> D: alsa-util.c: snd_pcm_hw_params_set_access() failed: Invalid argument >> D: alsa-util.c: Trying plug:hw:0 with SND_PCM_NO_AUTO_FORMAT ... >> D: alsa-util.c: Managed to open plug:hw:0 >> D: alsa-util.c: snd_pcm_hw_params_set_format(Signed 16 bit Little Endian) >> failed: Invalid argument >> D: alsa-util.c: snd_pcm_hw_params_set_format(Signed 16 bit Big Endian) >> failed: Invalid argument >> D: alsa-util.c: snd_pcm_hw_params_set_format(Float 32 bit Little Endian) >> failed: Invalid argument >> D: alsa-util.c: snd_pcm_hw_params_set_format(Float 32 bit Big Endian) >> failed: Invalid argument >> D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument >> D: alsa-util.c: Trying plug:hw:0 without SND_PCM_NO_AUTO_FORMAT ... >> D: alsa-util.c: Managed to open plug:hw:0 >> D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument >> I: alsa-util.c: Failed to set hardware parameters on plug:hw:0: Invalid >> argument >> D: alsa-util.c: Trying hw:0 as last resort... >> D: alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ... >> D: alsa-util.c: Managed to open hw:0 >> D: alsa-util.c: snd_pcm_hw_params_set_access() failed: Invalid argument >> D: alsa-util.c: Trying hw:0 without SND_PCM_NO_AUTO_FORMAT ... >> D: alsa-util.c: Managed to open hw:0 >> D: alsa-util.c: snd_pcm_hw_params_set_access() failed: Invalid argument >> D: alsa-util.c: Trying plug:hw:0 with SND_PCM_NO_AUTO_FORMAT ... >> D: alsa-util.c: Managed to open plug:hw:0 >> D: alsa-util.c: snd_pcm_hw_params_set_format(Signed 16 bit Little Endian) >> failed: Invalid argument >> D: alsa-util.c: snd_pcm_hw_params_set_format(Signed 16 bit Big Endian) >> failed: Invalid argument >> D: alsa-util.c: snd_pcm_hw_params_set_format(Float 32 bit Little Endian) >> failed: Invalid argument >> D: alsa-util.c: snd_pcm_hw_params_set_format(Float 32 bit Big Endian) >> failed: Invalid argument >> D: alsa-util.c: Maximum hw buffer size is 371 ms >> I: (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed >> I: (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed >> I: (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed >> I: (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed >> D: alsa-util.c: Set neither period nor buffer size. >> I: (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed >> I: alsa-util.c: snd_pcm_hw_params failed: Device or resource busy >> D: alsa-util.c: Trying plug:hw:0 without SND_PCM_NO_AUTO_FORMAT ... >> D: alsa-util.c: Managed to open plug:hw:0 >> D: alsa-util.c: Maximum hw buffer size is 371 ms >> I: (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed >> I: (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed >> I: (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed >> I: (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed >> D: alsa-util.c: Set neither period nor buffer size. >> I: (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed >> I: alsa-util.c: snd_pcm_hw_params failed: Device or resource busy >> I: alsa-util.c: Failed to set hardware parameters on plug:hw:0: Device or >> resource busy >> E: module.c: Failed to load module "module-alsa-source" (argument: >> "device_id=0"): initialization failed. >> W: module.c: module-oss is deprecated: Please use module-alsa-card instead >> of module-oss! >> D: oss-util.c: capabilities: MMAP REALTIME TRIGGER >> I: module-oss.c: Device opened in O_RDONLY mode. >> D: oss-util.c: Asking for 4 fragments of size 512 (requested 880) >> E: oss-util.c: SNDCTL_DSP_SETFMT: Device or resource busy >> E: module.c: Failed to load module "module-oss" (argument: >> "device=/dev/dsp"): initialization failed. >> W: module-detect.c: failed to detect any sound hardware. >> >> >> 2009/11/6 Lars Zobbe <fedesensfar(a)gmail.com> >> >> Sorry. I didn't realise that it was a bash script. Here is the link to my >>> alsa-info >>> >>> http://www.alsa-project.org/db/?f=d3728557be71067e2ed0920747118ae7efff2829 >>> >>> Thanks again. >>> Lars. >>> >>> >>> On Thu, Nov 5, 2009 at 5:29 AM, Raymond Yau <superquad.vortex2(a)gmail.com >>> > wrote: >>> >>>> Please provide the full pulseaudio log and output of alsa-info.sh >>>> >>>> http://www.pulseaudio.org/wiki/Troubleshooting >>>> >>>> pulseaudio -k; pulseaudio -vvv >>>> >>>> and >>>> >>>> aplay -v -Dfront:0 32bit.wav >>>> >>>> aplay -v -M -Dfront:0 32bit.wav >>>> >>>> aplay -v -Dplughw:0 any.wav >>>> >>>> aplay -v -M -Dplughw:0 any.wav >>>> >>>> >>>> 2009/11/4 Lars Zobbe <fedesensfar(a)gmail.com> >>>> >>>> > Hi Raymond. >>>> > >>>> > I so sorry Raymond. But I'm not sure what to do with this info. Is >>>> there >>>> > something more I have to tell you? Is there anything I can do to make >>>> > pulseaudio able to detect via udev? Can I copy-paste that bit of code >>>> you >>>> > sent me somewhere? >>>> > Please tell me if there's more you need to know. >>>> > Greetings. >>>> > Lars. >>>> > >>>> > >>>> > >>>> > >>>> > On Tue, Nov 3, 2009 at 7:48 AM, Raymond Yau < >>>> superquad.vortex2(a)gmail.com>wrote: >>>> > >>>> >> I guess route plugin used by RME96*.conf does not support mmap when >>>> there >>>> >> is >>>> >> channel conversion since minimum channels supported by hdsp is 14 >>>> >> >>>> >> >>>> >> static struct snd_pcm_hardware snd_hdsp_playback_subinfo = >>>> >> { >>>> >> .info = (SNDRV_PCM_INFO_MMAP | >>>> >> SNDRV_PCM_INFO_MMAP_VALID | >>>> >> SNDRV_PCM_INFO_NONINTERLEAVED | >>>> >> SNDRV_PCM_INFO_SYNC_START | >>>> >> SNDRV_PCM_INFO_DOUBLE), >>>> >> #ifdef SNDRV_BIG_ENDIAN >>>> >> .formats = SNDRV_PCM_FMTBIT_S32_BE, >>>> >> #else >>>> >> .formats = SNDRV_PCM_FMTBIT_S32_LE, >>>> >> #endif >>>> >> .rates = (SNDRV_PCM_RATE_32000 | >>>> >> SNDRV_PCM_RATE_44100 | >>>> >> SNDRV_PCM_RATE_48000 | >>>> >> SNDRV_PCM_RATE_64000 | >>>> >> SNDRV_PCM_RATE_88200 | >>>> >> SNDRV_PCM_RATE_96000), >>>> >> .rate_min = 32000, >>>> >> .rate_max = 96000, >>>> >> .channels_min = 14, >>>> >> .channels_max = HDSP_MAX_CHANNELS, >>>> >> .buffer_bytes_max = HDSP_CHANNEL_BUFFER_BYTES * HDSP_MAX_CHANNELS, >>>> >> .period_bytes_min = (64 * 4) * 10, >>>> >> .period_bytes_max = (8192 * 4) * HDSP_MAX_CHANNELS, >>>> >> .periods_min = 2, >>>> >> .periods_max = 2, >>>> >> .fifo_size = 0 >>>> >> }; >>>> >> >>>> >> 2009/11/2 Lars Zobbe <fedesensfar(a)gmail.com> >>>> >> >>>> >> > Hi. >>>> >> > >>>> >> > Talking to one of the pulseaudio developers (*Colin Guthrie*) about >>>> a >>>> >> > problem with pulseaudio loading modules and my sound card (RME HDSP >>>> >> 9632), >>>> >> > I >>>> >> > was adviced to inform you about the problem. >>>> >> > >>>> >> > I'm on ubuntu 9.10, 64bit, kernel 2.6.31 with an amd64 x2 3800+ >>>> >> processor. >>>> >> > >>>> >> > >>>> >> > Symptoms are: >>>> >> > Can only make pulseaudio load modules with the deprecated >>>> load-module >>>> >> > module-detect command, module-udev-detect doesn't detect anything >>>> (no >>>> >> sound >>>> >> > - only null sink output in padevchooser). >>>> >> > I get terrible stuttering. >>>> >> > I can remove pulseaudio and make things work, more or less. >>>> However, >>>> >> that >>>> >> > leaves me without software mixing and apps fighting for the rights >>>> to >>>> >> make >>>> >> > sound. And more inconveniences (fx. no volume-control). >>>> >> > I can play sound from audacity without stuttering by choosing the >>>> >> cardname >>>> >> > directly in output HDSP, not ALSA, not pulseaudio. >>>> >> > >>>> >> > >>>> >> > When trying to run alsa-info.sh I get a lot of "not found" errors. >>>> So >>>> >> > you'll >>>> >> > get some manual info (they might have moved som files around in >>>> ubuntu >>>> >> > 9.10). >>>> >> > >>>> >> > This is part of my syslog during a boot. (I did the formatting to >>>> bold). >>>> >> > >>>> >> > Nov 2 10:57:32 lars-desktop pulseaudio[1524]: module.c: >>>> module-detect >>>> >> is >>>> >> > deprecated: Please use module-udev-detect instead of module-detect! >>>> >> > Nov 2 10:57:37 lars-desktop pulseaudio[1524]: ratelimit.c: 27 >>>> events >>>> >> > suppressed >>>> >> > Nov 2 10:57:39 lars-desktop pulseaudio[1612]: module.c: >>>> module-detect >>>> >> is >>>> >> > deprecated: Please use module-udev-detect instead of module-detect! >>>> >> > Nov 2 10:57:44 lars-desktop pulseaudio[1612]: ratelimit.c: 23 >>>> events >>>> >> > suppressed >>>> >> > Nov 2 10:57:44 lars-desktop pulseaudio[1612]: alsa-sink.c: ALSA >>>> woke us >>>> >> up >>>> >> > to write new data to the device, but there was actually nothing to >>>> >> write! >>>> >> > Nov 2 10:57:44 lars-desktop pulseaudio[1612]: alsa-sink.c: *Most >>>> likely >>>> >> > this is a bug in the ALSA driver 'snd_hdsp'. Please report this >>>> issue to >>>> >> > the >>>> >> > ALSA developers.* >>>> >> > Nov 2 10:57:44 lars-desktop pulseaudio[1612]: alsa-sink.c: We were >>>> >> woken >>>> >> > up >>>> >> > with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 >>>> or >>>> >> > another value < min_avail. >>>> >> > Nov 2 10:57:49 lars-desktop pulseaudio[1612]: ratelimit.c: 99 >>>> events >>>> >> > suppressed >>>> >> > >>>> >> > Later if I try to play audio I get terrible skipping or stuttering >>>> and >>>> >> my >>>> >> > syslog (in less than a minute) writes: >>>> >> > Nov 2 11:55:16 lars-desktop pulseaudio[1612]: ratelimit.c: 16 >>>> events >>>> >> > suppressed >>>> >> > Nov 2 11:55:21 lars-desktop pulseaudio[1612]: ratelimit.c: 34 >>>> events >>>> >> > suppressed >>>> >> > Nov 2 11:55:26 lars-desktop pulseaudio[1612]: ratelimit.c: 70 >>>> events >>>> >> > suppressed >>>> >> > Nov 2 11:55:31 lars-desktop pulseaudio[1612]: ratelimit.c: 29 >>>> events >>>> >> > suppressed >>>> >> > Nov 2 11:55:36 lars-desktop pulseaudio[1612]: ratelimit.c: 43 >>>> events >>>> >> > suppressed >>>> >> > Nov 2 11:55:41 lars-desktop pulseaudio[1612]: ratelimit.c: 161 >>>> events >>>> >> > suppressed >>>> >> > Nov 2 11:55:46 lars-desktop pulseaudio[1612]: ratelimit.c: 28 >>>> events >>>> >> > suppressed >>>> >> > Nov 2 11:55:51 lars-desktop pulseaudio[1612]: ratelimit.c: 18 >>>> events >>>> >> > suppressed >>>> >> > Nov 2 11:55:56 lars-desktop pulseaudio[1612]: ratelimit.c: 16 >>>> events >>>> >> > suppressed >>>> >> > >>>> >> > Here's the relevant part (concerning my sound card) of lspci -vvv >>>> >> > >>>> >> > 02:09.0 Multimedia audio controller: Xilinx Corporation RME >>>> Hammerfall >>>> >> DSP >>>> >> > (rev 9a) >>>> >> > 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: 255 >>>> >> > Interrupt: pin A routed to IRQ 19 >>>> >> > Region 0: Memory at fdff0000 (32-bit, non-prefetchable) >>>> [size=64K] >>>> >> > Kernel driver in use: RME Hammerfall DSP >>>> >> > Kernel modules: snd-hdsp >>>> >> > >>>> >> > And: >>>> >> > lars@lars-desktop:~$ aplay -l >>>> >> > **** List of PLAYBACK Hardware Devices **** >>>> >> > card 0: DSP [Hammerfall DSP], device 0: RME Hammerfall HDSP 9632 >>>> [RME >>>> >> > Hammerfall HDSP 9632] >>>> >> > Subdevices: 1/1 >>>> >> > Subdevice #0: subdevice #0 >>>> >> > >>>> >> > >>>> >> > lars@lars-desktop:~$ cat /proc/asound/version >>>> >> > Advanced Linux Sound Architecture Driver Version 1.0.21. >>>> >> > Compiled on Oct 15 2009 for kernel 2.6.31-14-generic (SMP). >>>> >> > >>>> >> > lars@lars-desktop:~$ cat /proc/asound/cards >>>> >> > 0 [DSP ]: H-DSP - Hammerfall DSP >>>> >> > RME Hammerfall HDSP 9632 at 0xfdff0000, irq 19 >>>> >> > >>>> >> > >>>> >> > lars@lars-desktop:~$ cat /proc/asound/modules >>>> >> > 0 snd_hdsp >>>> >> > >>>> >> > I don't have .asound.conf files. >>>> >> > >>>> >> > lars@lars-desktop:~$ hdsploader >>>> >> > hdsploader - firmware loader for RME Hammerfall DSP cards >>>> >> > Looking for HDSP + Multiface or Digiface cards : >>>> >> > Card 0 : RME Hammerfall HDSP 9632 at 0xfdff0000, irq 19 >>>> >> > >>>> >> > And of course. My computer makes no sound if I don't load the >>>> hdspmixer >>>> >> at >>>> >> > startup. I guess that's just normal for this card though. >>>> >> > >>>> >> > Please let me know, if you need more info. I really hope you can >>>> help >>>> >> out >>>> >> > here. And that this might help you to. >>>> >> > >>>> >> > Thanks for the great work you do. Greetings. >>>> >> > Lars Zobbe. >>>> >> > _______________________________________________ >>>> >> > Alsa-devel mailing list >>>> >> > Alsa-devel(a)alsa-project.org >>>> >> > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >>>> >> > >>>> >> _______________________________________________ >>>> >> Alsa-devel mailing list >>>> >> Alsa-devel(a)alsa-project.org >>>> >> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >>>> >> >>>> > >>>> > >>>> _______________________________________________ >>>> Alsa-devel mailing list >>>> Alsa-devel(a)alsa-project.org >>>> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >>>> >>> >>> >> >

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[alsa-devel] [PATCH] ASoC: TWL4030: Fix typo in comment in header file
by Thomas Weber 14 Jan '10

14 Jan '10

Signed-off-by: Thomas Weber <weber(a)corscience.de> --- sound/soc/codecs/twl4030.h | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index dd6396e..f206d24 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -25,7 +25,7 @@ /* Register descriptions are here */ #include <linux/mfd/twl4030-codec.h> -/* Sgadow register used by the audio driver */ +/* Shadow register used by the audio driver */ #define TWL4030_REG_SW_SHADOW 0x4A #define TWL4030_CACHEREGNUM (TWL4030_REG_SW_SHADOW + 1) -- 1.6.4.4

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Re: [alsa-devel] ALSA DMIX support without MMAP ?
by ANISH KUMAR 14 Jan '10

14 Jan '10

>mmap is supported by the Samsung drivers. Thanks for this information which i overlooked. >This looks like the dmix data rates are getting out of sync with what >the hardware is doing. Enabling CONFIG_SND_PCM_XRUN_DEBUG will produce >more diagnostic information from the driver side I got the problem and after doing the changes DMIX plugin is working very fine. As you know XRUN was happening and this happening in snd_pcm_update_hw_ptr_post function. This happened because stop_threshold is increased suddenly(Is it done by dmix? If yes then why ?) ?????? In normal condtions this would have been fine but in my code i was multiplying it by some amount causing it to wrap around(stop_threshold=0 );Causing XRUN.So i just removed this multiplying factor. One more Rate conversion query: ./alsa_aplay -v -D"plug:'dmix:RATE=44100'" bond.wav Playing WAVE 'bond.wav' : Unsigned 8 bit, Rate 11025 Hz, Mono Plug PCM: Rate conversion PCM (44100, sformat=U8) As you can see bond.wav is sampled at 11025k but when i am playing with Plug PCM there is some noise in the song why????Is rate conversion algoritm not able to resample it properly? If i play with ./alsa_aplay -v -D"plug:'dmix:RATE=11025'" bond.wav then no noise.Probably because no resampling happening? Mark Brown Heartfelt Thanks for your time.

2 1

Re: [alsa-devel] ALSA DMIX support without MMAP ?
by ANISH KUMAR 14 Jan '10

14 Jan '10

2 1

[alsa-devel] Linux kernel 2.6.32.3 bug report
by d binderman 14 Jan '10

14 Jan '10

Hello there, I just ran the sourceforge tool cppcheck over the source code of the Linux kernel 2.6.32.3 It said [./sound/soc/s3c24xx/s3c24xx-pcm.c:128]: (error) Possible null pointer dereference: substream - otherwise it is redundant to check if substream is null at line 130 I checked the source code and I agree with the compiler. Suggest code rework. Regards David Binderman _________________________________________________________________ Do you have a story that started on Hotmail? Tell us now http://clk.atdmt.com/UKM/go/195013117/direct/01/

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[alsa-devel] capturing data from the microphone
by Riccardo Magliocchetti 14 Jan '10

14 Jan '10

[the same message is waiting in moderator queue, sorry if you receive it two times] Hello, i'm trying to capture audio data from the microphone, but i'm not able to read any frames. The frames are always 0 and delay is fixed to 2730. I've tried calling snd_pcm_avail_delay, snd_pcm_avail_update but does not make a difference. The soundcard is driven by snd_hda_intel. The code is more or less the same as Paul Davis' tutorial on using the alsa api. This is the code for init and the loop that tries to read the data: #define MIC_BUFSIZE 4096 static gpointer snd_pcm_read(gpointer data) { int error; snd_pcm_sframes_t frames = MIC_BUFSIZE; while (TRUE) { if ((error = snd_pcm_wait (pcm_handle, 1000)) < 0) { g_printerr("Failed to poll: %s\n", snd_strerror(error)); continue; } if ((error = snd_pcm_avail(pcm_handle)) < 0) { if (error == -EPIPE) { g_printerr("xrun! %s\n", snd_strerror(error)); continue; //return GINT_TO_POINTER(FALSE); } else { g_printerr("alsa_pcm_avail_update error %s\n", snd_strerror(error)); continue; //return GINT_TO_POINTER(FALSE); } } frames = error; if (frames == 0) continue; frames = frames > MIC_BUFSIZE ? MIC_BUFSIZE : frames; g_printerr ("frames: %ld\n", frames); g_static_mutex_lock(&mutex); error = snd_pcm_readi(pcm_handle, Mic_Buffer[Mic_WriteBuf], frames); g_static_mutex_unlock(&mutex); if (error < 0) error = snd_pcm_recover(pcm_handle, error, 0); if (error < 0) { LOG("snd_pcm_readi FAIL!: %s\n", snd_strerror(error)); } } return GINT_TO_POINTER(TRUE); } BOOL Mic_Init() { snd_pcm_hw_params_t *hwparams; snd_pcm_sw_params_t *swparams; int err; if (Mic_Inited) return TRUE; // Open the default sound card in capture if ((err = snd_pcm_open(&pcm_handle, "default", SND_PCM_STREAM_CAPTURE, /*SND_PCM_NONBLOCK*/ 0)) < 0) { g_printerr("Failed to open device: %s\n", snd_strerror(err)); return FALSE; } // Allocate the snd_pcm_hw_params_t structure and fill it. snd_pcm_hw_params_alloca(&hwparams); if ((err = snd_pcm_hw_params_any(pcm_handle, hwparams)) < 0) { g_printerr("Failed to setup hw parameters: %s\n", snd_strerror(err)); return FALSE; } //Set the access if ((err = snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) { g_printerr("Failed to set access: %s\n", snd_strerror(err)); return FALSE; } //dir 0 == exacte (Rate = 16K exacte) if ((err = snd_pcm_hw_params_set_rate(pcm_handle, hwparams, 16000, 0)) < 0) { g_printerr("Failed to set rate: %s\n", snd_strerror(err)); return FALSE; } /* Set sample format */ if ((err = snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S8)) < 0) { g_printerr("Failed to set format: %s\n", snd_strerror(err)); return FALSE; } // Set one channel (mono) if ((err = snd_pcm_hw_params_set_channels(pcm_handle, hwparams, 1)) < 0) { g_printerr("Failed to set channels: %s\n", snd_strerror(err)); return FALSE; } //Set the params if ((err = snd_pcm_hw_params(pcm_handle, hwparams)) < 0) { g_printerr("Failed to set hw parameters: %s\n", snd_strerror(err)); return FALSE; } snd_pcm_sw_params_alloca(&swparams); if ((err = snd_pcm_sw_params_current (pcm_handle, swparams)) < 0) { g_printerr("Failed to set current sw parameters: %s\n", snd_strerror(err)); return FALSE; } if ((err = snd_pcm_sw_params_set_avail_min (pcm_handle, swparams, MIC_BUFSIZE)) < 0) { g_printerr("Failed to set minimum available count: %s\n", snd_strerror(err)); return FALSE; } if ((err = snd_pcm_sw_params_set_start_threshold (pcm_handle, swparams, 0U)) < 0) { g_printerr("Failed to set start mode: %s\n", snd_strerror(err)); return FALSE; } if ((err = snd_pcm_sw_params (pcm_handle, swparams)) < 0) { g_printerr("Failed to set sw parameters: %s\n", snd_strerror(err)); return FALSE; } if ((err = snd_pcm_prepare (pcm_handle)) < 0) { g_printerr("Failed to prepare audio interface to use: %s\n", snd_strerror(err)); return FALSE; } Mic_Inited = TRUE; Mic_Reset(); mic_reader = g_thread_create(snd_pcm_read, NULL, TRUE, NULL); return TRUE; } thanks, Riccardo

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[alsa-devel] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write!
by Konrad Schlichtherle 14 Jan '10

14 Jan '10

Hello, sometimes it occure that my speakers do not play sound although the speakers should play music with VLC. Then i took a look at /var/log/messages and found this: Jan 13 17:01:39 riker pulseaudio[3335]: alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write! Jan 13 17:01:39 riker pulseaudio[3335]: alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. Jan 13 17:01:39 riker pulseaudio[3335]: alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. The output of utils_alsa-info.sh is there: http://www.alsa-project.org/db/?f=1767e9bc745724a67c908549d20c137375d0605e I didn't know wether this is a bug or a misconfiguration of my openSUSE 11.2 Konrad

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Re: [alsa-devel] [PATCH] OMAP: Exporting functions doing common register access
by Aggarwal, Anuj 14 Jan '10

14 Jan '10

> -----Original Message----- > From: Paul Walmsley [mailto:paul@pwsan.com] > Sent: Tuesday, November 17, 2009 2:46 AM > To: Aggarwal, Anuj > Cc: linux-omap(a)vger.kernel.org > Subject: Re: [PATCH] OMAP: Exporting functions doing common register > access > > Hello Anuj, > > On Mon, 16 Nov 2009, Anuj Aggarwal wrote: > > > These functions need to be exported so that drivers (e.g. McBSP) can > > be configured as modules. > > > > McBSP driver gets built as a module when ASoC driver for OMAP3 EVM > > is configured as module. McBSP driver uses functions like > > omap_ctrl_readl/omap_ctrl_writel, which are defined in control.c > > file but not exported. Without that, McBSP driver fails to build as > > a module. > > > > Signed-off-by: Anuj Aggarwal <anuj.aggarwal(a)ti.com> > > This will prevent the McBSP driver from being usable across chips. For > example, if a future OMAP comes out with McBSP clock source switching > implemented in a different way, or if a DaVinci chip comes out with a > McBSP block but no SCM block, this either won't build or won't run. > > A better approach would be to move the OMAP2/3-specific functions from the > McBSP driver into the OMAP McBSP integration code, > arch/arm/*omap*/mcbsp.c, and to define function pointers in > struct platform_data for these functions, and to pass them from the > integration code into the device driver code. That will make these > drivers independent of chip integration details. Sorry to confuse you, I was talking about the platform specific ASoC-McBSP code for OMAP (file - sound/soc/omap/omap-mcbsp.c) platform. The McBSP driver - both platform and machine specific - doesn't use the functions exported by the patch. Including alsa-devel list too... > > > - Paul

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[alsa-devel] Newbie question: Sending untimed MIDI data to ALSA seq
by Josh Lehan 13 Jan '10

13 Jan '10

Hello. Newbie question: I'm wondering about sending untimed MIDI data to the ALSA sequencer subsystem. The application would be for a "beep"-like program, from the command line, that instead of making beeps, would play MIDI instruments to make better-sounding beeps. I have TiMidity hooked up into ALSA, as a softsynth, and it works just fine for playing MIDI data sent to it by ALSA. The "aplaymidi" command works for .MID files, and the "vkeybd" program works for simulating a keyboard and sending in untimed MIDI data. The "aseqview" program also works great for watching the notes being played, as they go by. So, my MIDI software stack appears to be in order. I don't have a hardware synthesizer hooked up at the moment (my Kawai K1 is showing its age, and a few keys are broken). I also looked at "amidi", which features a -S option to send in untimed MIDI bytes, but unfortunately, that addresses an entirely different namespace. Evidently, it only speaks to "RawMIDI" devices, of which TiMidity is not. So, I can't generate untimed MIDI data on my own. I was thinking of writing a small "midibeep" program that would do this, similar to what vkeybd does, but without requiring any user interaction. Does such a program already exist? Thanks! Josh

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[alsa-devel] Mic Boost control problem with snd-hda-intel codec AL662 rev 1
by Anisse Astier 13 Jan '10

13 Jan '10

Hi, The Mic Boost control on a laptop seems to behave incorrectly. The control has the flag "volume" instead of "cvolume", causing it to appear in both the "Playback" and "Capture" tabs of alsamixer. This wouldn't be a problem if it would just work in the Capture tab: it doesn't. When I modify it in Capture, it doesn't retain its value: it's gone after i close alsamixer. But if i modify it in "Playback" it will retain its value and the value will be mirrored by the Mic Boost in Capture (both Mic Boost seem to reflect the same control, except one is RO, the other is RW). It's a minor a annoyance, but I'd like to fix it and have "Mic Boost" appear only as a Capture control and retaining its value. Maybe I should create a new mixer configuration for this hardware in patch_realtek.c, but I don't know where to start. Bios autoprobing works for everything but just what I mentioned. My alsa-info.sh : http://www.alsa-project.org/db/?f=fc17b509e532fc366eb2d5fe5ec9e6730c96c837 Any ideas? Where should I start? Regards, Anisse

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[alsa-devel] [PATCH] ASoC: TWL4030: Replace comma with semicolon in probe function
by Peter Ujfalusi 13 Jan '10

13 Jan '10

The codec structure initialization statements should be separated by semicolons. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/twl4030.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 2a27f7b..74f0d65 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -2192,7 +2192,7 @@ static int __devinit twl4030_codec_probe(struct platform_device *pdev) codec->write = twl4030_write; codec->set_bias_level = twl4030_set_bias_level; codec->dai = twl4030_dai; - codec->num_dai = ARRAY_SIZE(twl4030_dai), + codec->num_dai = ARRAY_SIZE(twl4030_dai); codec->reg_cache_size = sizeof(twl4030_reg); codec->reg_cache = kmemdup(twl4030_reg, sizeof(twl4030_reg), GFP_KERNEL); -- 1.6.6

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[alsa-devel] [PATCH] ALSA: ASoC: ak4104: allow more sample rates
by Daniel Mack 13 Jan '10

13 Jan '10

The transmitter supports all sample rates up to 192KHz, so the driver should not give a limit. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/ak4104.c | 4 +--- 1 files changed, 1 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index 3a14c6f..b9ef7e4 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -185,9 +185,7 @@ struct snd_soc_dai ak4104_dai = { .stream_name = "Playback", .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_32000, + .rates = SNDRV_PCM_RATE_8000_192000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE -- 1.6.5.2

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[alsa-devel] Personal invitation from Kunal Narkhede
by Kunal Narkhede 13 Jan '10

13 Jan '10

Hi, On 12/24/2009 9:02:23 AM, you were invited to join Kunal Narkhede's UNYK address book so he/she would always have access to your contact info and you to his/hers. To accept his/her request, Click here. http://www.unyk.com/ml/65/5/?i=C7B72DD0179E4FD7A6DED23A2976A060&s=17267503 UNYK is a smart and simple way to manage your contacts so you never lose anyones contact information again. No more worrying about your contacts' info. From now on, let them do it for you. By changing their information in UNYK.com, you address book will be automatically updated. By changing your information in UNYK.com, their address book will be automatically updated. Its so simple and its free... Already 25 million users. Please do not reply to this email. Questions ? Contact us - http://unyk.com/en/Contact UNYK Inc., 1010 de Serigny, suite 820, Longueuil, QC, J4K 5G7, Canada. I no longer wish to receive invitations to join UNYK from Kunal Narkhede: Unsubscribe http://www.unyk.com/ml/245/74/?mid=9E98A4AA807983A0&email=alsa-devel@alsa-p… I never again wish to receive invitation to join UNYK: Unsubscribe http://www.unyk.com/ml/65/6/?i=C7B72DD0179E4FD7A6DED23A2976A060&s=17267503 UNYK, the first smart address book that updates itself!

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[alsa-devel] [PATCH - hda 1/1] ALSA: hda - Improved MacBook (Pro) 5, 1 / 5, 2 support
by murray.alex＠gmail.com 13 Jan '10

13 Jan '10

From: Alex Murray <murray.alex(a)gmail.com> This patch adds support for automatically muting the speakers when headphones are inserted, as well as relabelling the headphone widgets from the non-standard "HP" to the standard "Headphone" for the mb5 model. Signed-off-by: Alex Murray <murray.alex(a)gmail.com> --- pci/hda/patch_realtek.c | 28 ++++++++++++++++++++++++++-- 1 files changed, 26 insertions(+), 2 deletions(-) diff --git a/pci/hda/patch_realtek.c b/pci/hda/patch_realtek.c index f858dcd..bd65e47 100644 --- a/pci/hda/patch_realtek.c +++ b/pci/hda/patch_realtek.c @@ -7201,8 +7201,8 @@ static struct snd_kcontrol_new alc885_mb5_mixer[] = { HDA_BIND_MUTE ("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("LFE Playback Volume", 0x0e, 0x00, HDA_OUTPUT), HDA_BIND_MUTE ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("HP Playback Volume", 0x0f, 0x00, HDA_OUTPUT), - HDA_BIND_MUTE ("HP Playback Switch", 0x0f, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0f, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("Headphone Playback Switch", 0x0f, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), @@ -7603,6 +7603,7 @@ static struct hda_verb alc885_mb5_init_verbs[] = { {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x14, AC_VERB_SET_CONNECT_SEL, 0x03}, + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, /* Front Mic pin: input vref at 80% */ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, @@ -7787,6 +7788,27 @@ static void alc885_mbp3_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; } +static void alc885_mb5_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x14, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_amp_stereo(codec, 0x18, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + snd_hda_codec_amp_stereo(codec, 0x1a, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + +} + +static void alc885_mb5_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + /* Headphone insertion or removal. */ + if ((res >> 26) == ALC880_HP_EVENT) + alc885_mb5_automute(codec); +} + static void alc885_imac91_automute(struct hda_codec *codec) { unsigned int present; @@ -9233,6 +9255,8 @@ static struct alc_config_preset alc882_presets[] = { .input_mux = &mb5_capture_source, .dig_out_nid = ALC882_DIGOUT_NID, .dig_in_nid = ALC882_DIGIN_NID, + .unsol_event = alc885_mb5_unsol_event, + .init_hook = alc885_mb5_automute, }, [ALC885_MACPRO] = { .mixers = { alc882_macpro_mixer }, -- 1.6.3.3

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[alsa-devel] ALSA DMIX support without MMAP ?
by ANISH KUMAR 13 Jan '10

13 Jan '10

Hi, I am working on android and and in our device we use ALSA. I have enabled dmix support for andoid as by default android doesn't have this plugin enabled and soon will post it in android forum however i have run into some problems.Hoping takashi can help me out on this. Problem is when i am trying to use dmix plugin i am not getting any sound (logs shown below).Is it because i don't have mmap mode support on my device??? If yes then can't dmix work without mmap mode??? Logs shown below: Entered s3c24xx_pcm_hw_params, Entered s3c24xx_pcm_hw_params, line 330 Entered s3c24xx_pcm_mmap Entered s3c24xx_pcm_prepare Entered s3c24xx_pcm_enqueue Entered s3c24xx_pcm_trigger Entered s3c24xx_pcm_pointer Entered s3c24xx_pcm_pointer res_value[0] snd_pcm_update_hw_ptr_pos xun: [0]occured XRUN! Normal playback and recording is fine if i dont use dmix plugin.During my normal playback and recording i don't use mmap as i preallocate DMA memory and use them. Regards, anish

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[alsa-devel] [2.6.33-rc3] HDA works in polling mode
by Tvrtko Ursulin 13 Jan '10

13 Jan '10

Hi, I have a AMD 785G motherboard where there is a worrying situation logged by ALSA drivers. Sound nevertheless works. lspci: 00:14.2 Audio device: ATI Technologies Inc SBx00 Azalia (Intel HDA) dmesg: HDA Intel 0000:01:05.1: PCI INT B disabled HDA Intel 0000:00:14.2: PCI INT A disabled HDA Intel 0000:00:14.2: PCI INT A -> GSI 16 (level, low) -> IRQ 16 HDA Intel 0000:01:05.1: PCI INT B -> GSI 19 (level, low) -> IRQ 19 HDA Intel 0000:01:05.1: irq 28 for MSI/MSI-X HDA Intel 0000:01:05.1: setting latency timer to 64 hda-intel: azx_get_response timeout, switching to polling mode: last cmd=0x000f0000 I tried position_fix=0, 1 and 2 with no difference. I also tried enable_msi=0 to make it use IO-APIC interrupt. Then this error is not displayed, sound works, but again no interrupts are generated at least count is not growing in /proc/interrupts. Please tell me what else I can try and what debugging info you need. Thanks, Tvrtko

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[alsa-devel] No CD audio with kernel 2.6.32.3
by Chris Clayton 12 Jan '10

12 Jan '10

Hi, Please cc me on any reply - I'm not subscribed. Except for CD audio, sound is working fine on my new HP G61 laptop with kernel 2.6.32.3. I can play CD audio fine in Windows. IIRC, on my (very) old Dell laptop, alsamixer showed a CD audio device, but that's not the case on the G61. I have asla built as modules. Diagnostics: [chris:~]$ lsmod | grep snd snd_hda_codec_intelhdmi 10250 1 snd_hda_codec_idt 40146 1 snd_hda_intel 17040 1 snd_hda_codec 46119 3 snd_hda_codec_intelhdmi,snd_hda_codec_idt,snd_hda_intel snd_pcm 51424 2 snd_hda_intel,snd_hda_codec snd_timer 15133 1 snd_pcm snd 37459 8 snd_hda_codec_intelhdmi,snd_hda_codec_idt,snd_hda_intel,snd_hda_codec,snd_pcm,snd_timer snd_page_alloc 5657 2 snd_hda_intel,snd_pcm [chris:~]$ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 3: INTEL HDMI [INTEL HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 [chris:~]$ amixer controls numid=10,iface=MIXER,name='Master Playback Switch' numid=9,iface=MIXER,name='Master Playback Volume' numid=5,iface=MIXER,name='Headphone Playback Switch' numid=4,iface=MIXER,name='Headphone Playback Volume' numid=15,iface=MIXER,name='PCM Playback Volume' numid=3,iface=MIXER,name='Mic Jack Mode' numid=7,iface=MIXER,name='Capture Switch' numid=6,iface=MIXER,name='Capture Volume' numid=11,iface=MIXER,name='IEC958 Playback Con Mask' numid=12,iface=MIXER,name='IEC958 Playback Pro Mask' numid=13,iface=MIXER,name='IEC958 Playback Default' numid=14,iface=MIXER,name='IEC958 Playback Switch' numid=8,iface=MIXER,name='Mux Capture Volume' numid=2,iface=MIXER,name='Speaker Playback Switch' numid=1,iface=MIXER,name='Speaker Playback Volume' [chris:~]$ amixer scontrols Simple mixer control 'Master',0 Simple mixer control 'Headphone',0 Simple mixer control 'Speaker',0 Simple mixer control 'PCM',0 Simple mixer control 'Mic Jack Mode',0 Simple mixer control 'IEC958',0 Simple mixer control 'Capture',0 Simple mixer control 'Mux',0 Extract from dmesg: HDA Intel 0000:00:1b.0: PCI INT A -> GSI 22 (level, low) -> IRQ 22 HDA Intel 0000:00:1b.0: setting latency timer to 64 Let me know if there is any other information I can provide. Thanks Chris -- No, Sir; there is nothing which has yet been contrived by man, by which so much happiness is produced as by a good tavern or inn - Doctor Samuel Johnson

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[alsa-devel] X-Fi notebook support
by The Source 12 Jan '10

12 Jan '10

Hello. I have some info on that card, but not much. I have discovered that this card is detected connected via USB rather than PCI-E (the slot is ExpressCard). snd_ctxfi uses PCI interfaces right? Can X-Fi support be also added to snd_usb_audio? Or can snd_ctxfi support USB? lsusb -v output on that card is attached (sorry for duplicate mail, forgot to attach it first time). I now have this card and willing to help to test (or perhaps create) proper driver.

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[alsa-devel] ASoC updates for 2.6.33
by Mark Brown 12 Jan '10

12 Jan '10

The following changes since commit 7f50548abb5454bd82c25aae15f0a3bf6a530f46: Mark Brown (1): Merge commit 'v2.6.33-rc2' into for-2.6.33 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.33 Manuel Lauss (1): ASoC: fixup oops in generic AC97 codec glue Mark Brown (1): ASoC: Fix WM8350 DSP mode B configuration sound/soc/codecs/ac97.c | 6 ++++++ sound/soc/codecs/wm8350.c | 2 +- 2 files changed, 7 insertions(+), 1 deletions(-)

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[alsa-devel] hda-intel: no codecs found
by Ozan Çağlayan 12 Jan '10

12 Jan '10

Hi, Does that warning can be interpreted as a hardware failure? I'm reverting to older alsa-driver snapshots that I'm sure was working OK but still having this warning. I'm suspecting the power outage that the system suffered on Friday. Is there a way to understand whether it's driver related or HW related? Thanks, Ozan

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[alsa-devel] [PATCH 1/4] ASoC: Add a new imx-ssi sound driver
by Mark Brown 11 Jan '10

11 Jan '10

From: Sascha Hauer <s.hauer(a)pengutronix.de> The old driver has the number of SSI units in the system hardcoded, does not make use of the device model and works only on i.MX21/27. This driver replaces it. It works in DMA mode on i.MX21/27 and using an FIQ handler on other systems. It also supports AC97 mode of the SSI units. Signed-off-by: Sascha Hauer <s.hauer(a)pengutronix.de> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- arch/arm/plat-mxc/Makefile | 6 + arch/arm/plat-mxc/ssi-fiq-ksym.c | 20 + arch/arm/plat-mxc/ssi-fiq.S | 134 +++++++ sound/soc/imx/Kconfig | 20 +- sound/soc/imx/Makefile | 12 +- sound/soc/imx/imx-pcm-dma-mx2.c | 313 ++++++++++++++++ sound/soc/imx/imx-pcm-fiq.c | 277 ++++++++++++++ sound/soc/imx/imx-ssi.c | 762 ++++++++++++++++++++++++++++++++++++++ sound/soc/imx/imx-ssi.h | 238 ++++++++++++ 9 files changed, 1762 insertions(+), 20 deletions(-) create mode 100644 arch/arm/plat-mxc/ssi-fiq-ksym.c create mode 100644 arch/arm/plat-mxc/ssi-fiq.S create mode 100644 sound/soc/imx/imx-pcm-dma-mx2.c create mode 100644 sound/soc/imx/imx-pcm-fiq.c create mode 100644 sound/soc/imx/imx-ssi.c create mode 100644 sound/soc/imx/imx-ssi.h diff --git a/arch/arm/plat-mxc/Makefile b/arch/arm/plat-mxc/Makefile index e3212c8..b0b9fc3 100644 --- a/arch/arm/plat-mxc/Makefile +++ b/arch/arm/plat-mxc/Makefile @@ -9,3 +9,9 @@ obj-$(CONFIG_ARCH_MX1) += iomux-mx1-mx2.o dma-mx1-mx2.o obj-$(CONFIG_ARCH_MX2) += iomux-mx1-mx2.o dma-mx1-mx2.o obj-$(CONFIG_ARCH_MXC_IOMUX_V3) += iomux-v3.o obj-$(CONFIG_MXC_PWM) += pwm.o +obj-$(CONFIG_ARCH_MXC_AUDMUX_V1) += audmux-v1.o +obj-$(CONFIG_ARCH_MXC_AUDMUX_V2) += audmux-v2.o +ifdef CONFIG_SND_IMX_SOC +obj-y += ssi-fiq.o +obj-y += ssi-fiq-ksym.o +endif diff --git a/arch/arm/plat-mxc/ssi-fiq-ksym.c b/arch/arm/plat-mxc/ssi-fiq-ksym.c new file mode 100644 index 0000000..b5fad45 --- /dev/null +++ b/arch/arm/plat-mxc/ssi-fiq-ksym.c @@ -0,0 +1,20 @@ +/* + * Exported ksyms for the SSI FIQ handler + * + * Copyright (C) 2009, Sascha Hauer <s.hauer(a)pengutronix.de> + * + * 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 <mach/ssi.h> + +EXPORT_SYMBOL(imx_ssi_fiq_tx_buffer); +EXPORT_SYMBOL(imx_ssi_fiq_rx_buffer); +EXPORT_SYMBOL(imx_ssi_fiq_start); +EXPORT_SYMBOL(imx_ssi_fiq_end); +EXPORT_SYMBOL(imx_ssi_fiq_base); + diff --git a/arch/arm/plat-mxc/ssi-fiq.S b/arch/arm/plat-mxc/ssi-fiq.S new file mode 100644 index 0000000..4ddce56 --- /dev/null +++ b/arch/arm/plat-mxc/ssi-fiq.S @@ -0,0 +1,134 @@ +/* + * Copyright (C) 2009 Sascha Hauer <s.hauer(a)pengutronix.de> + * + * 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/linkage.h> +#include <asm/assembler.h> + +/* + * r8 = bit 0-15: tx offset, bit 16-31: tx buffer size + * r9 = bit 0-15: rx offset, bit 16-31: rx buffer size + */ + +#define SSI_STX0 0x00 +#define SSI_SRX0 0x08 +#define SSI_SISR 0x14 +#define SSI_SIER 0x18 +#define SSI_SACNT 0x38 + +#define SSI_SACNT_AC97EN (1 << 0) + +#define SSI_SIER_TFE0_EN (1 << 0) +#define SSI_SISR_TFE0 (1 << 0) +#define SSI_SISR_RFF0 (1 << 2) +#define SSI_SIER_RFF0_EN (1 << 2) + + .text + .global imx_ssi_fiq_start + .global imx_ssi_fiq_end + .global imx_ssi_fiq_base + .global imx_ssi_fiq_rx_buffer + .global imx_ssi_fiq_tx_buffer + +imx_ssi_fiq_start: + ldr r12, imx_ssi_fiq_base + + /* TX */ + ldr r11, imx_ssi_fiq_tx_buffer + + /* shall we send? */ + ldr r13, [r12, #SSI_SIER] + tst r13, #SSI_SIER_TFE0_EN + beq 1f + + /* TX FIFO empty? */ + ldr r13, [r12, #SSI_SISR] + tst r13, #SSI_SISR_TFE0 + beq 1f + + mov r10, #0x10000 + sub r10, #1 + and r10, r10, r8 /* r10: current buffer offset */ + + add r11, r11, r10 + + ldrh r13, [r11] + strh r13, [r12, #SSI_STX0] + + ldrh r13, [r11, #2] + strh r13, [r12, #SSI_STX0] + + ldrh r13, [r11, #4] + strh r13, [r12, #SSI_STX0] + + ldrh r13, [r11, #6] + strh r13, [r12, #SSI_STX0] + + add r10, #8 + lsr r13, r8, #16 /* r13: buffer size */ + cmp r10, r13 + lslgt r8, r13, #16 + addle r8, #8 +1: + /* RX */ + + /* shall we receive? */ + ldr r13, [r12, #SSI_SIER] + tst r13, #SSI_SIER_RFF0_EN + beq 1f + + /* RX FIFO full? */ + ldr r13, [r12, #SSI_SISR] + tst r13, #SSI_SISR_RFF0 + beq 1f + + ldr r11, imx_ssi_fiq_rx_buffer + + mov r10, #0x10000 + sub r10, #1 + and r10, r10, r9 /* r10: current buffer offset */ + + add r11, r11, r10 + + ldr r13, [r12, #SSI_SACNT] + tst r13, #SSI_SACNT_AC97EN + + ldr r13, [r12, #SSI_SRX0] + strh r13, [r11] + + ldr r13, [r12, #SSI_SRX0] + strh r13, [r11, #2] + + /* dummy read to skip slot 12 */ + ldrne r13, [r12, #SSI_SRX0] + + ldr r13, [r12, #SSI_SRX0] + strh r13, [r11, #4] + + ldr r13, [r12, #SSI_SRX0] + strh r13, [r11, #6] + + /* dummy read to skip slot 12 */ + ldrne r13, [r12, #SSI_SRX0] + + add r10, #8 + lsr r13, r9, #16 /* r13: buffer size */ + cmp r10, r13 + lslgt r9, r13, #16 + addle r9, #8 + +1: + @ return from FIQ + subs pc, lr, #4 +imx_ssi_fiq_base: + .word 0x0 +imx_ssi_fiq_rx_buffer: + .word 0x0 +imx_ssi_fiq_tx_buffer: + .word 0x0 +imx_ssi_fiq_end: + diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index a700562..84a25e6 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -1,21 +1,13 @@ -config SND_MX1_MX2_SOC - tristate "SoC Audio for Freecale i.MX1x i.MX2x CPUs" - depends on ARCH_MX2 || ARCH_MX1 +config SND_IMX_SOC + tristate "SoC Audio for Freecale i.MX CPUs" + depends on ARCH_MXC select SND_PCM + select FIQ + select SND_SOC_AC97_BUS help Say Y or M if you want to add support for codecs attached to - the MX1 or MX2 SSI interface. + the i.MX SSI interface. config SND_MXC_SOC_SSI tristate -config SND_SOC_MX27VIS_WM8974 - tristate "SoC Audio support for MX27 - WM8974 Visstrim_sm10 board" - depends on SND_MX1_MX2_SOC && MACH_MX27 && MACH_IMX27_VISSTRIM_M10 - select SND_MXC_SOC_SSI - select SND_SOC_WM8974 - help - Say Y if you want to add support for SoC audio on Visstrim SM10 - board with WM8974. - - diff --git a/sound/soc/imx/Makefile b/sound/soc/imx/Makefile index c2ffd2c..4bde34a 100644 --- a/sound/soc/imx/Makefile +++ b/sound/soc/imx/Makefile @@ -1,10 +1,10 @@ # i.MX Platform Support -snd-soc-mx1_mx2-objs := mx1_mx2-pcm.o -snd-soc-mxc-ssi-objs := mxc-ssi.o +snd-soc-imx-objs := imx-ssi.o imx-pcm-fiq.o imx-pcm-dma-mx2.o -obj-$(CONFIG_SND_MX1_MX2_SOC) += snd-soc-mx1_mx2.o -obj-$(CONFIG_SND_MXC_SOC_SSI) += snd-soc-mxc-ssi.o +ifdef CONFIG_MACH_MX27 +snd-soc-imx-objs += imx-pcm-dma-mx2.o +endif + +obj-$(CONFIG_SND_IMX_SOC) += snd-soc-imx.o # i.MX Machine Support -snd-soc-mx27vis-wm8974-objs := mx27vis_wm8974.o -obj-$(CONFIG_SND_SOC_MX27VIS_WM8974) += snd-soc-mx27vis-wm8974.o diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c new file mode 100644 index 0000000..19452e4 --- /dev/null +++ b/sound/soc/imx/imx-pcm-dma-mx2.c @@ -0,0 +1,313 @@ +/* + * imx-pcm-dma-mx2.c -- ALSA Soc Audio Layer + * + * Copyright 2009 Sascha Hauer <s.hauer(a)pengutronix.de> + * + * This code is based on code copyrighted by Freescale, + * Liam Girdwood, Javier Martin and probably others. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/device.h> +#include <linux/dma-mapping.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/platform_device.h> + +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <mach/dma-mx1-mx2.h> + +#include "imx-ssi.h" + +struct imx_pcm_runtime_data { + int sg_count; + struct scatterlist *sg_list; + int period; + int periods; + unsigned long dma_addr; + int dma; + struct snd_pcm_substream *substream; + unsigned long offset; + unsigned long size; + unsigned long period_cnt; + void *buf; + int period_time; +}; + +/* Called by the DMA framework when a period has elapsed */ +static void imx_ssi_dma_progression(int channel, void *data, + struct scatterlist *sg) +{ + struct snd_pcm_substream *substream = data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + + if (!sg) + return; + + runtime = iprtd->substream->runtime; + + iprtd->offset = sg->dma_address - runtime->dma_addr; + + snd_pcm_period_elapsed(iprtd->substream); +} + +static void imx_ssi_dma_callback(int channel, void *data) +{ + pr_err("%s shouldn't be called\n", __func__); +} + +static void snd_imx_dma_err_callback(int channel, void *data, int err) +{ + pr_err("DMA error callback called\n"); + + pr_err("DMA timeout on channel %d -%s%s%s%s\n", + channel, + err & IMX_DMA_ERR_BURST ? " burst" : "", + err & IMX_DMA_ERR_REQUEST ? " request" : "", + err & IMX_DMA_ERR_TRANSFER ? " transfer" : "", + err & IMX_DMA_ERR_BUFFER ? " buffer" : ""); +} + +static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct imx_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + int ret; + + iprtd->dma = imx_dma_request_by_prio(DRV_NAME, DMA_PRIO_HIGH); + if (iprtd->dma < 0) { + pr_err("Failed to claim the audio DMA\n"); + return -ENODEV; + } + + ret = imx_dma_setup_handlers(iprtd->dma, + imx_ssi_dma_callback, + snd_imx_dma_err_callback, substream); + if (ret) + goto out; + + ret = imx_dma_setup_progression_handler(iprtd->dma, + imx_ssi_dma_progression); + if (ret) { + pr_err("Failed to setup the DMA handler\n"); + goto out; + } + + ret = imx_dma_config_channel(iprtd->dma, + IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO, + IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR, + dma_params->dma, 1); + if (ret < 0) { + pr_err("Cannot configure DMA channel: %d\n", ret); + goto out; + } + + imx_dma_config_burstlen(iprtd->dma, dma_params->burstsize * 2); + + return 0; +out: + imx_dma_free(iprtd->dma); + return ret; +} + +static int snd_imx_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + int i; + unsigned long dma_addr; + + imx_ssi_dma_alloc(substream); + + iprtd->size = params_buffer_bytes(params); + iprtd->periods = params_periods(params); + iprtd->period = params_period_bytes(params); + iprtd->offset = 0; + iprtd->period_time = HZ / (params_rate(params) / + params_period_size(params)); + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + + if (iprtd->sg_count != iprtd->periods) { + kfree(iprtd->sg_list); + + iprtd->sg_list = kcalloc(iprtd->periods + 1, + sizeof(struct scatterlist), GFP_KERNEL); + if (!iprtd->sg_list) + return -ENOMEM; + iprtd->sg_count = iprtd->periods + 1; + } + + sg_init_table(iprtd->sg_list, iprtd->sg_count); + dma_addr = runtime->dma_addr; + + for (i = 0; i < iprtd->periods; i++) { + iprtd->sg_list[i].page_link = 0; + iprtd->sg_list[i].offset = 0; + iprtd->sg_list[i].dma_address = dma_addr; + iprtd->sg_list[i].length = iprtd->period; + dma_addr += iprtd->period; + } + + /* close the loop */ + iprtd->sg_list[iprtd->sg_count - 1].offset = 0; + iprtd->sg_list[iprtd->sg_count - 1].length = 0; + iprtd->sg_list[iprtd->sg_count - 1].page_link = + ((unsigned long) iprtd->sg_list | 0x01) & ~0x02; + return 0; +} + +static int snd_imx_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + + if (iprtd->dma >= 0) { + imx_dma_free(iprtd->dma); + iprtd->dma = -EINVAL; + } + + kfree(iprtd->sg_list); + iprtd->sg_list = NULL; + + return 0; +} + +static int snd_imx_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct imx_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + int err; + + iprtd->substream = substream; + iprtd->buf = (unsigned int *)substream->dma_buffer.area; + iprtd->period_cnt = 0; + + pr_debug("%s: buf: %p period: %d periods: %d\n", + __func__, iprtd->buf, iprtd->period, iprtd->periods); + + err = imx_dma_setup_sg(iprtd->dma, iprtd->sg_list, iprtd->sg_count, + IMX_DMA_LENGTH_LOOP, dma_params->dma_addr, + substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + DMA_MODE_WRITE : DMA_MODE_READ); + if (err) + return err; + + return 0; +} + +static int snd_imx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + imx_dma_enable(iprtd->dma); + + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + imx_dma_disable(iprtd->dma); + + break; + default: + return -EINVAL; + } + + return 0; +} + +static snd_pcm_uframes_t snd_imx_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + + return bytes_to_frames(substream->runtime, iprtd->offset); +} + +static struct snd_pcm_hardware snd_imx_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rate_min = 8000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = IMX_SSI_DMABUF_SIZE, + .period_bytes_min = 128, + .period_bytes_max = 16 * 1024, + .periods_min = 2, + .periods_max = 255, + .fifo_size = 0, +}; + +static int snd_imx_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd; + int ret; + + iprtd = kzalloc(sizeof(*iprtd), GFP_KERNEL); + runtime->private_data = iprtd; + + ret = snd_pcm_hw_constraint_integer(substream->runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + return ret; + + snd_soc_set_runtime_hwparams(substream, &snd_imx_hardware); + return 0; +} + +static struct snd_pcm_ops imx_pcm_ops = { + .open = snd_imx_open, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_imx_pcm_hw_params, + .hw_free = snd_imx_pcm_hw_free, + .prepare = snd_imx_pcm_prepare, + .trigger = snd_imx_pcm_trigger, + .pointer = snd_imx_pcm_pointer, + .mmap = snd_imx_pcm_mmap, +}; + +static struct snd_soc_platform imx_soc_platform_dma = { + .name = "imx-audio", + .pcm_ops = &imx_pcm_ops, + .pcm_new = imx_pcm_new, + .pcm_free = imx_pcm_free, +}; + +struct snd_soc_platform *imx_ssi_dma_mx2_init(struct platform_device *pdev, + struct imx_ssi *ssi) +{ + ssi->dma_params_tx.burstsize = DMA_TXFIFO_BURST; + ssi->dma_params_rx.burstsize = DMA_RXFIFO_BURST; + + return &imx_soc_platform_dma; +} + diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c new file mode 100644 index 0000000..5532579 --- /dev/null +++ b/sound/soc/imx/imx-pcm-fiq.c @@ -0,0 +1,277 @@ +/* + * imx-pcm-fiq.c -- ALSA Soc Audio Layer + * + * Copyright 2009 Sascha Hauer <s.hauer(a)pengutronix.de> + * + * This code is based on code copyrighted by Freescale, + * Liam Girdwood, Javier Martin and probably others. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/device.h> +#include <linux/dma-mapping.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/platform_device.h> + +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <asm/fiq.h> + +#include <mach/ssi.h> + +#include "imx-ssi.h" + +struct imx_pcm_runtime_data { + int period; + int periods; + unsigned long dma_addr; + int dma; + unsigned long offset; + unsigned long size; + unsigned long period_cnt; + void *buf; + struct timer_list timer; + int period_time; +}; + +static void imx_ssi_timer_callback(unsigned long data) +{ + struct snd_pcm_substream *substream = (void *)data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + struct pt_regs regs; + + get_fiq_regs(&regs); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + iprtd->offset = regs.ARM_r8 & 0xffff; + else + iprtd->offset = regs.ARM_r9 & 0xffff; + + iprtd->timer.expires = jiffies + iprtd->period_time; + add_timer(&iprtd->timer); + snd_pcm_period_elapsed(substream); +} + +static struct fiq_handler fh = { + .name = DRV_NAME, +}; + +static int snd_imx_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + + iprtd->size = params_buffer_bytes(params); + iprtd->periods = params_periods(params); + iprtd->period = params_period_bytes(params); + iprtd->offset = 0; + iprtd->period_time = HZ / (params_rate(params) / params_period_size(params)); + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + + return 0; +} + +static int snd_imx_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + struct pt_regs regs; + + get_fiq_regs(&regs); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + regs.ARM_r8 = (iprtd->period * iprtd->periods - 1) << 16; + else + regs.ARM_r9 = (iprtd->period * iprtd->periods - 1) << 16; + + set_fiq_regs(&regs); + + return 0; +} + +static int fiq_enable; +static int imx_pcm_fiq; + +static int snd_imx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + iprtd->timer.expires = jiffies + iprtd->period_time; + add_timer(&iprtd->timer); + if (++fiq_enable == 1) + enable_fiq(imx_pcm_fiq); + + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + del_timer(&iprtd->timer); + if (--fiq_enable == 0) + disable_fiq(imx_pcm_fiq); + + + break; + default: + return -EINVAL; + } + + return 0; +} + +static snd_pcm_uframes_t snd_imx_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + + return bytes_to_frames(substream->runtime, iprtd->offset); +} + +static struct snd_pcm_hardware snd_imx_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rate_min = 8000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = IMX_SSI_DMABUF_SIZE, + .period_bytes_min = 128, + .period_bytes_max = 16 * 1024, + .periods_min = 2, + .periods_max = 255, + .fifo_size = 0, +}; + +static int snd_imx_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd; + int ret; + + iprtd = kzalloc(sizeof(*iprtd), GFP_KERNEL); + runtime->private_data = iprtd; + + init_timer(&iprtd->timer); + iprtd->timer.data = (unsigned long)substream; + iprtd->timer.function = imx_ssi_timer_callback; + + ret = snd_pcm_hw_constraint_integer(substream->runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + return ret; + + snd_soc_set_runtime_hwparams(substream, &snd_imx_hardware); + return 0; +} + +static int snd_imx_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + + del_timer_sync(&iprtd->timer); + kfree(iprtd); + + return 0; +} + +static struct snd_pcm_ops imx_pcm_ops = { + .open = snd_imx_open, + .close = snd_imx_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_imx_pcm_hw_params, + .prepare = snd_imx_pcm_prepare, + .trigger = snd_imx_pcm_trigger, + .pointer = snd_imx_pcm_pointer, + .mmap = snd_imx_pcm_mmap, +}; + +static int imx_pcm_fiq_new(struct snd_card *card, struct snd_soc_dai *dai, + struct snd_pcm *pcm) +{ + int ret; + + ret = imx_pcm_new(card, dai, pcm); + if (ret) + return ret; + + if (dai->playback.channels_min) { + struct snd_pcm_substream *substream = + pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + + imx_ssi_fiq_tx_buffer = (unsigned long)buf->area; + } + + if (dai->capture.channels_min) { + struct snd_pcm_substream *substream = + pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + + imx_ssi_fiq_rx_buffer = (unsigned long)buf->area; + } + + set_fiq_handler(&imx_ssi_fiq_start, + &imx_ssi_fiq_end - &imx_ssi_fiq_start); + + return 0; +} + +static struct snd_soc_platform imx_soc_platform_fiq = { + .pcm_ops = &imx_pcm_ops, + .pcm_new = imx_pcm_fiq_new, + .pcm_free = imx_pcm_free, +}; + +struct snd_soc_platform *imx_ssi_fiq_init(struct platform_device *pdev, + struct imx_ssi *ssi) +{ + int ret = 0; + + ret = claim_fiq(&fh); + if (ret) { + dev_err(&pdev->dev, "failed to claim fiq: %d", ret); + return ERR_PTR(ret); + } + + mxc_set_irq_fiq(ssi->irq, 1); + + imx_pcm_fiq = ssi->irq; + + imx_ssi_fiq_base = (unsigned long)ssi->base; + + ssi->dma_params_tx.burstsize = 4; + ssi->dma_params_rx.burstsize = 6; + + return &imx_soc_platform_fiq; +} + +void imx_ssi_fiq_exit(struct platform_device *pdev, + struct imx_ssi *ssi) +{ + mxc_set_irq_fiq(ssi->irq, 0); + release_fiq(&fh); +} + diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c new file mode 100644 index 0000000..c57a11f --- /dev/null +++ b/sound/soc/imx/imx-ssi.c @@ -0,0 +1,762 @@ +/* + * imx-ssi.c -- ALSA Soc Audio Layer + * + * Copyright 2009 Sascha Hauer <s.hauer(a)pengutronix.de> + * + * This code is based on code copyrighted by Freescale, + * Liam Girdwood, Javier Martin and probably others. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + * + * The i.MX SSI core has some nasty limitations in AC97 mode. While most + * sane processor vendors have a FIFO per AC97 slot, the i.MX has only + * one FIFO which combines all valid receive slots. We cannot even select + * which slots we want to receive. The WM9712 with which this driver + * was developped with always sends GPIO status data in slot 12 which + * we receive in our (PCM-) data stream. The only chance we have is to + * manually skip this data in the FIQ handler. With sampling rates different + * from 48000Hz not every frame has valid receive data, so the ratio + * between pcm data and GPIO status data changes. Our FIQ handler is not + * able to handle this, hence this driver only works with 48000Hz sampling + * rate. + * Reading and writing AC97 registers is another challange. The core + * provides us status bits when the read register is updated with *another* + * value. When we read the same register two times (and the register still + * contains the same value) these status bits are not set. We work + * around this by not polling these bits but only wait a fixed delay. + * + */ + +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/device.h> +#include <linux/dma-mapping.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/platform_device.h> + +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <mach/ssi.h> +#include <mach/hardware.h> + +#include "imx-ssi.h" + +#define SSI_SACNT_DEFAULT (SSI_SACNT_AC97EN | SSI_SACNT_FV) + +/* + * SSI Network Mode or TDM slots configuration. + * Should only be called when port is inactive (i.e. SSIEN = 0). + */ +static int imx_ssi_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) +{ + struct imx_ssi *ssi = container_of(cpu_dai, struct imx_ssi, dai); + u32 sccr; + + sccr = readl(ssi->base + SSI_STCCR); + sccr &= ~SSI_STCCR_DC_MASK; + sccr |= SSI_STCCR_DC(slots - 1); + writel(sccr, ssi->base + SSI_STCCR); + + sccr = readl(ssi->base + SSI_SRCCR); + sccr &= ~SSI_STCCR_DC_MASK; + sccr |= SSI_STCCR_DC(slots - 1); + writel(sccr, ssi->base + SSI_SRCCR); + + writel(tx_mask, ssi->base + SSI_STMSK); + writel(rx_mask, ssi->base + SSI_SRMSK); + + return 0; +} + +/* + * SSI DAI format configuration. + * Should only be called when port is inactive (i.e. SSIEN = 0). + * Note: We don't use the I2S modes but instead manually configure the + * SSI for I2S because the I2S mode is only a register preset. + */ +static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) +{ + struct imx_ssi *ssi = container_of(cpu_dai, struct imx_ssi, dai); + u32 strcr = 0, scr; + + scr = readl(ssi->base + SSI_SCR) & ~(SSI_SCR_SYN | SSI_SCR_NET); + + /* DAI mode */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + /* data on rising edge of bclk, frame low 1clk before data */ + strcr |= SSI_STCR_TFSI | SSI_STCR_TEFS | SSI_STCR_TXBIT0; + scr |= SSI_SCR_NET; + break; + case SND_SOC_DAIFMT_LEFT_J: + /* data on rising edge of bclk, frame high with data */ + strcr |= SSI_STCR_TXBIT0; + break; + case SND_SOC_DAIFMT_DSP_B: + /* data on rising edge of bclk, frame high with data */ + strcr |= SSI_STCR_TFSL; + break; + case SND_SOC_DAIFMT_DSP_A: + /* data on rising edge of bclk, frame high 1clk before data */ + strcr |= SSI_STCR_TFSL | SSI_STCR_TEFS; + break; + } + + /* DAI clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_IF: + strcr |= SSI_STCR_TFSI; + strcr &= ~SSI_STCR_TSCKP; + break; + case SND_SOC_DAIFMT_IB_NF: + strcr &= ~(SSI_STCR_TSCKP | SSI_STCR_TFSI); + break; + case SND_SOC_DAIFMT_NB_IF: + strcr |= SSI_STCR_TFSI | SSI_STCR_TSCKP; + break; + case SND_SOC_DAIFMT_NB_NF: + strcr &= ~SSI_STCR_TFSI; + strcr |= SSI_STCR_TSCKP; + break; + } + + /* DAI clock master masks */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + strcr |= SSI_STCR_TFDIR | SSI_STCR_TXDIR; + break; + case SND_SOC_DAIFMT_CBM_CFS: + strcr |= SSI_STCR_TFDIR; + break; + case SND_SOC_DAIFMT_CBS_CFM: + strcr |= SSI_STCR_TXDIR; + break; + } + + strcr |= SSI_STCR_TFEN0; + + writel(strcr, ssi->base + SSI_STCR); + writel(strcr, ssi->base + SSI_SRCR); + writel(scr, ssi->base + SSI_SCR); + + return 0; +} + +/* + * SSI system clock configuration. + * Should only be called when port is inactive (i.e. SSIEN = 0). + */ +static int imx_ssi_set_dai_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + struct imx_ssi *ssi = container_of(cpu_dai, struct imx_ssi, dai); + u32 scr; + + scr = readl(ssi->base + SSI_SCR); + + switch (clk_id) { + case IMX_SSP_SYS_CLK: + if (dir == SND_SOC_CLOCK_OUT) + scr |= SSI_SCR_SYS_CLK_EN; + else + scr &= ~SSI_SCR_SYS_CLK_EN; + break; + default: + return -EINVAL; + } + + writel(scr, ssi->base + SSI_SCR); + + return 0; +} + +/* + * SSI Clock dividers + * Should only be called when port is inactive (i.e. SSIEN = 0). + */ +static int imx_ssi_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct imx_ssi *ssi = container_of(cpu_dai, struct imx_ssi, dai); + u32 stccr, srccr; + + stccr = readl(ssi->base + SSI_STCCR); + srccr = readl(ssi->base + SSI_SRCCR); + + switch (div_id) { + case IMX_SSI_TX_DIV_2: + stccr &= ~SSI_STCCR_DIV2; + stccr |= div; + break; + case IMX_SSI_TX_DIV_PSR: + stccr &= ~SSI_STCCR_PSR; + stccr |= div; + break; + case IMX_SSI_TX_DIV_PM: + stccr &= ~0xff; + stccr |= SSI_STCCR_PM(div); + break; + case IMX_SSI_RX_DIV_2: + stccr &= ~SSI_STCCR_DIV2; + stccr |= div; + break; + case IMX_SSI_RX_DIV_PSR: + stccr &= ~SSI_STCCR_PSR; + stccr |= div; + break; + case IMX_SSI_RX_DIV_PM: + stccr &= ~0xff; + stccr |= SSI_STCCR_PM(div); + break; + default: + return -EINVAL; + } + + writel(stccr, ssi->base + SSI_STCCR); + writel(srccr, ssi->base + SSI_SRCCR); + + return 0; +} + +/* + * Should only be called when port is inactive (i.e. SSIEN = 0), + * although can be called multiple times by upper layers. + */ +static int imx_ssi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + struct imx_ssi *ssi = container_of(cpu_dai, struct imx_ssi, dai); + u32 reg, sccr; + + /* Tx/Rx config */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + reg = SSI_STCCR; + cpu_dai->dma_data = &ssi->dma_params_tx; + } else { + reg = SSI_SRCCR; + cpu_dai->dma_data = &ssi->dma_params_rx; + } + + sccr = readl(ssi->base + reg) & ~SSI_STCCR_WL_MASK; + + /* DAI data (word) size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + sccr |= SSI_SRCCR_WL(16); + break; + case SNDRV_PCM_FORMAT_S20_3LE: + sccr |= SSI_SRCCR_WL(20); + break; + case SNDRV_PCM_FORMAT_S24_LE: + sccr |= SSI_SRCCR_WL(24); + break; + } + + writel(sccr, ssi->base + reg); + + return 0; +} + +static int imx_ssi_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct imx_ssi *ssi = container_of(cpu_dai, struct imx_ssi, dai); + unsigned int sier_bits, sier; + unsigned int scr; + + scr = readl(ssi->base + SSI_SCR); + sier = readl(ssi->base + SSI_SIER); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (ssi->flags & IMX_SSI_DMA) + sier_bits = SSI_SIER_TDMAE; + else + sier_bits = SSI_SIER_TIE | SSI_SIER_TFE0_EN; + } else { + if (ssi->flags & IMX_SSI_DMA) + sier_bits = SSI_SIER_RDMAE; + else + sier_bits = SSI_SIER_RIE | SSI_SIER_RFF0_EN; + } + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + scr |= SSI_SCR_TE; + else + scr |= SSI_SCR_RE; + sier |= sier_bits; + + if (++ssi->enabled == 1) + scr |= SSI_SCR_SSIEN; + + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + scr &= ~SSI_SCR_TE; + else + scr &= ~SSI_SCR_RE; + sier &= ~sier_bits; + + if (--ssi->enabled == 0) + scr &= ~SSI_SCR_SSIEN; + + break; + default: + return -EINVAL; + } + + if (!(ssi->flags & IMX_SSI_USE_AC97)) + /* rx/tx are always enabled to access ac97 registers */ + writel(scr, ssi->base + SSI_SCR); + + writel(sier, ssi->base + SSI_SIER); + + return 0; +} + +static struct snd_soc_dai_ops imx_ssi_pcm_dai_ops = { + .hw_params = imx_ssi_hw_params, + .set_fmt = imx_ssi_set_dai_fmt, + .set_clkdiv = imx_ssi_set_dai_clkdiv, + .set_sysclk = imx_ssi_set_dai_sysclk, + .set_tdm_slot = imx_ssi_set_dai_tdm_slot, + .trigger = imx_ssi_trigger, +}; + +static struct snd_soc_dai imx_ssi_dai = { + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &imx_ssi_pcm_dai_ops, +}; + +int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + int ret; + + ret = dma_mmap_coherent(NULL, vma, runtime->dma_area, + runtime->dma_addr, runtime->dma_bytes); + + pr_debug("%s: ret: %d %p 0x%08x 0x%08x\n", __func__, ret, + runtime->dma_area, + runtime->dma_addr, + runtime->dma_bytes); + return ret; +} + +static int imx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = IMX_SSI_DMABUF_SIZE; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->area = dma_alloc_writecombine(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + if (!buf->area) + return -ENOMEM; + buf->bytes = size; + + return 0; +} + +static u64 imx_pcm_dmamask = DMA_BIT_MASK(32); + +int imx_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, + struct snd_pcm *pcm) +{ + + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &imx_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); + if (dai->playback.channels_min) { + ret = imx_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + } + + if (dai->capture.channels_min) { + ret = imx_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + } + +out: + return ret; +} + +void imx_pcm_free(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + + dma_free_writecombine(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + +struct snd_soc_platform imx_soc_platform = { + .name = "imx-audio", +}; +EXPORT_SYMBOL_GPL(imx_soc_platform); + +static struct snd_soc_dai imx_ac97_dai = { + .name = "AC97", + .ac97_control = 1, + .playback = { + .stream_name = "AC97 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "AC97 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &imx_ssi_pcm_dai_ops, +}; + +static void setup_channel_to_ac97(struct imx_ssi *imx_ssi) +{ + void __iomem *base = imx_ssi->base; + + writel(0x0, base + SSI_SCR); + writel(0x0, base + SSI_STCR); + writel(0x0, base + SSI_SRCR); + + writel(SSI_SCR_SYN | SSI_SCR_NET, base + SSI_SCR); + + writel(SSI_SFCSR_RFWM0(8) | + SSI_SFCSR_TFWM0(8) | + SSI_SFCSR_RFWM1(8) | + SSI_SFCSR_TFWM1(8), base + SSI_SFCSR); + + writel(SSI_STCCR_WL(16) | SSI_STCCR_DC(12), base + SSI_STCCR); + writel(SSI_STCCR_WL(16) | SSI_STCCR_DC(12), base + SSI_SRCCR); + + writel(SSI_SCR_SYN | SSI_SCR_NET | SSI_SCR_SSIEN, base + SSI_SCR); + writel(SSI_SOR_WAIT(3), base + SSI_SOR); + + writel(SSI_SCR_SYN | SSI_SCR_NET | SSI_SCR_SSIEN | + SSI_SCR_TE | SSI_SCR_RE, + base + SSI_SCR); + + writel(SSI_SACNT_DEFAULT, base + SSI_SACNT); + writel(0xff, base + SSI_SACCDIS); + writel(0x300, base + SSI_SACCEN); +} + +static struct imx_ssi *ac97_ssi; + +static void imx_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, + unsigned short val) +{ + struct imx_ssi *imx_ssi = ac97_ssi; + void __iomem *base = imx_ssi->base; + unsigned int lreg; + unsigned int lval; + + if (reg > 0x7f) + return; + + pr_debug("%s: 0x%02x 0x%04x\n", __func__, reg, val); + + lreg = reg << 12; + writel(lreg, base + SSI_SACADD); + + lval = val << 4; + writel(lval , base + SSI_SACDAT); + + writel(SSI_SACNT_DEFAULT | SSI_SACNT_WR, base + SSI_SACNT); + udelay(100); +} + +static unsigned short imx_ssi_ac97_read(struct snd_ac97 *ac97, + unsigned short reg) +{ + struct imx_ssi *imx_ssi = ac97_ssi; + void __iomem *base = imx_ssi->base; + + unsigned short val = -1; + unsigned int lreg; + + lreg = (reg & 0x7f) << 12 ; + writel(lreg, base + SSI_SACADD); + writel(SSI_SACNT_DEFAULT | SSI_SACNT_RD, base + SSI_SACNT); + + udelay(100); + + val = (readl(base + SSI_SACDAT) >> 4) & 0xffff; + + pr_debug("%s: 0x%02x 0x%04x\n", __func__, reg, val); + + return val; +} + +static void imx_ssi_ac97_reset(struct snd_ac97 *ac97) +{ + struct imx_ssi *imx_ssi = ac97_ssi; + + if (imx_ssi->ac97_reset) + imx_ssi->ac97_reset(ac97); +} + +static void imx_ssi_ac97_warm_reset(struct snd_ac97 *ac97) +{ + struct imx_ssi *imx_ssi = ac97_ssi; + + if (imx_ssi->ac97_warm_reset) + imx_ssi->ac97_warm_reset(ac97); +} + +struct snd_ac97_bus_ops soc_ac97_ops = { + .read = imx_ssi_ac97_read, + .write = imx_ssi_ac97_write, + .reset = imx_ssi_ac97_reset, + .warm_reset = imx_ssi_ac97_warm_reset +}; +EXPORT_SYMBOL_GPL(soc_ac97_ops); + +struct snd_soc_dai *imx_ssi_pcm_dai[2]; +EXPORT_SYMBOL_GPL(imx_ssi_pcm_dai); + +static int imx_ssi_probe(struct platform_device *pdev) +{ + struct resource *res; + struct imx_ssi *ssi; + struct imx_ssi_platform_data *pdata = pdev->dev.platform_data; + struct snd_soc_platform *platform; + int ret = 0; + unsigned int val; + + ssi = kzalloc(sizeof(*ssi), GFP_KERNEL); + if (!ssi) + return -ENOMEM; + + if (pdata) { + ssi->ac97_reset = pdata->ac97_reset; + ssi->ac97_warm_reset = pdata->ac97_warm_reset; + ssi->flags = pdata->flags; + } + + imx_ssi_pcm_dai[pdev->id] = &ssi->dai; + + ssi->irq = platform_get_irq(pdev, 0); + + ssi->clk = clk_get(&pdev->dev, NULL); + if (IS_ERR(ssi->clk)) { + ret = PTR_ERR(ssi->clk); + dev_err(&pdev->dev, "Cannot get the clock: %d\n", + ret); + goto failed_clk; + } + clk_enable(ssi->clk); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + ret = -ENODEV; + goto failed_get_resource; + } + + if (!request_mem_region(res->start, resource_size(res), DRV_NAME)) { + dev_err(&pdev->dev, "request_mem_region failed\n"); + ret = -EBUSY; + goto failed_get_resource; + } + + ssi->base = ioremap(res->start, resource_size(res)); + if (!ssi->base) { + dev_err(&pdev->dev, "ioremap failed\n"); + ret = -ENODEV; + goto failed_ioremap; + } + + if (ssi->flags & IMX_SSI_USE_AC97) { + if (ac97_ssi) { + ret = -EBUSY; + goto failed_ac97; + } + ac97_ssi = ssi; + setup_channel_to_ac97(ssi); + memcpy(&ssi->dai, &imx_ac97_dai, sizeof(imx_ac97_dai)); + } else + memcpy(&ssi->dai, &imx_ssi_dai, sizeof(imx_ssi_dai)); + + ssi->dai.id = pdev->id; + ssi->dai.dev = &pdev->dev; + ssi->dai.name = kasprintf(GFP_KERNEL, "imx-ssi.%d", pdev->id); + + writel(0x0, ssi->base + SSI_SIER); + + ssi->dma_params_rx.dma_addr = res->start + SSI_SRX0; + ssi->dma_params_tx.dma_addr = res->start + SSI_STX0; + + res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "tx0"); + if (res) + ssi->dma_params_tx.dma = res->start; + + res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "rx0"); + if (res) + ssi->dma_params_rx.dma = res->start; + + ssi->dai.id = pdev->id; + ssi->dai.dev = &pdev->dev; + ssi->dai.name = kasprintf(GFP_KERNEL, "imx-ssi.%d", pdev->id); + + if ((cpu_is_mx27() || cpu_is_mx21()) && + !(ssi->flags & IMX_SSI_USE_AC97)) { + ssi->flags |= IMX_SSI_DMA; + platform = imx_ssi_dma_mx2_init(pdev, ssi); + } else + platform = imx_ssi_fiq_init(pdev, ssi); + + imx_soc_platform.pcm_ops = platform->pcm_ops; + imx_soc_platform.pcm_new = platform->pcm_new; + imx_soc_platform.pcm_free = platform->pcm_free; + + val = SSI_SFCSR_TFWM0(ssi->dma_params_tx.burstsize) | + SSI_SFCSR_RFWM0(ssi->dma_params_rx.burstsize); + writel(val, ssi->base + SSI_SFCSR); + + ret = snd_soc_register_dai(&ssi->dai); + if (ret) { + dev_err(&pdev->dev, "register DAI failed\n"); + goto failed_register; + } + + platform_set_drvdata(pdev, ssi); + + return 0; + +failed_register: +failed_ac97: + iounmap(ssi->base); +failed_ioremap: + release_mem_region(res->start, resource_size(res)); +failed_get_resource: + clk_disable(ssi->clk); + clk_put(ssi->clk); +failed_clk: + kfree(ssi); + + return ret; +} + +static int __devexit imx_ssi_remove(struct platform_device *pdev) +{ + struct resource *res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + struct imx_ssi *ssi = platform_get_drvdata(pdev); + + snd_soc_unregister_dai(&ssi->dai); + + if (ssi->flags & IMX_SSI_USE_AC97) + ac97_ssi = NULL; + + if (!(ssi->flags & IMX_SSI_DMA)) + imx_ssi_fiq_exit(pdev, ssi); + + iounmap(ssi->base); + release_mem_region(res->start, resource_size(res)); + clk_disable(ssi->clk); + clk_put(ssi->clk); + kfree(ssi); + + return 0; +} + +static struct platform_driver imx_ssi_driver = { + .probe = imx_ssi_probe, + .remove = __devexit_p(imx_ssi_remove), + + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + }, +}; + +static int __init imx_ssi_init(void) +{ + int ret; + + ret = snd_soc_register_platform(&imx_soc_platform); + if (ret) { + pr_err("failed to register soc platform: %d\n", ret); + return ret; + } + + ret = platform_driver_register(&imx_ssi_driver); + if (ret) { + snd_soc_unregister_platform(&imx_soc_platform); + return ret; + } + + return 0; +} + +static void __exit imx_ssi_exit(void) +{ + platform_driver_unregister(&imx_ssi_driver); + snd_soc_unregister_platform(&imx_soc_platform); +} + +module_init(imx_ssi_init); +module_exit(imx_ssi_exit); + +/* Module information */ +MODULE_AUTHOR("Sascha Hauer, <s.hauer(a)pengutronix.de>"); +MODULE_DESCRIPTION("i.MX I2S/ac97 SoC Interface"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/imx/imx-ssi.h b/sound/soc/imx/imx-ssi.h new file mode 100644 index 0000000..cb2c81f --- /dev/null +++ b/sound/soc/imx/imx-ssi.h @@ -0,0 +1,238 @@ +/* + * 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. + */ + +#ifndef _IMX_SSI_H +#define _IMX_SSI_H + +#define SSI_STX0 0x00 +#define SSI_STX1 0x04 +#define SSI_SRX0 0x08 +#define SSI_SRX1 0x0c + +#define SSI_SCR 0x10 +#define SSI_SCR_CLK_IST (1 << 9) +#define SSI_SCR_CLK_IST_SHIFT 9 +#define SSI_SCR_TCH_EN (1 << 8) +#define SSI_SCR_SYS_CLK_EN (1 << 7) +#define SSI_SCR_I2S_MODE_NORM (0 << 5) +#define SSI_SCR_I2S_MODE_MSTR (1 << 5) +#define SSI_SCR_I2S_MODE_SLAVE (2 << 5) +#define SSI_I2S_MODE_MASK (3 << 5) +#define SSI_SCR_SYN (1 << 4) +#define SSI_SCR_NET (1 << 3) +#define SSI_SCR_RE (1 << 2) +#define SSI_SCR_TE (1 << 1) +#define SSI_SCR_SSIEN (1 << 0) + +#define SSI_SISR 0x14 +#define SSI_SISR_MASK ((1 << 19) - 1) +#define SSI_SISR_CMDAU (1 << 18) +#define SSI_SISR_CMDDU (1 << 17) +#define SSI_SISR_RXT (1 << 16) +#define SSI_SISR_RDR1 (1 << 15) +#define SSI_SISR_RDR0 (1 << 14) +#define SSI_SISR_TDE1 (1 << 13) +#define SSI_SISR_TDE0 (1 << 12) +#define SSI_SISR_ROE1 (1 << 11) +#define SSI_SISR_ROE0 (1 << 10) +#define SSI_SISR_TUE1 (1 << 9) +#define SSI_SISR_TUE0 (1 << 8) +#define SSI_SISR_TFS (1 << 7) +#define SSI_SISR_RFS (1 << 6) +#define SSI_SISR_TLS (1 << 5) +#define SSI_SISR_RLS (1 << 4) +#define SSI_SISR_RFF1 (1 << 3) +#define SSI_SISR_RFF0 (1 << 2) +#define SSI_SISR_TFE1 (1 << 1) +#define SSI_SISR_TFE0 (1 << 0) + +#define SSI_SIER 0x18 +#define SSI_SIER_RDMAE (1 << 22) +#define SSI_SIER_RIE (1 << 21) +#define SSI_SIER_TDMAE (1 << 20) +#define SSI_SIER_TIE (1 << 19) +#define SSI_SIER_CMDAU_EN (1 << 18) +#define SSI_SIER_CMDDU_EN (1 << 17) +#define SSI_SIER_RXT_EN (1 << 16) +#define SSI_SIER_RDR1_EN (1 << 15) +#define SSI_SIER_RDR0_EN (1 << 14) +#define SSI_SIER_TDE1_EN (1 << 13) +#define SSI_SIER_TDE0_EN (1 << 12) +#define SSI_SIER_ROE1_EN (1 << 11) +#define SSI_SIER_ROE0_EN (1 << 10) +#define SSI_SIER_TUE1_EN (1 << 9) +#define SSI_SIER_TUE0_EN (1 << 8) +#define SSI_SIER_TFS_EN (1 << 7) +#define SSI_SIER_RFS_EN (1 << 6) +#define SSI_SIER_TLS_EN (1 << 5) +#define SSI_SIER_RLS_EN (1 << 4) +#define SSI_SIER_RFF1_EN (1 << 3) +#define SSI_SIER_RFF0_EN (1 << 2) +#define SSI_SIER_TFE1_EN (1 << 1) +#define SSI_SIER_TFE0_EN (1 << 0) + +#define SSI_STCR 0x1c +#define SSI_STCR_TXBIT0 (1 << 9) +#define SSI_STCR_TFEN1 (1 << 8) +#define SSI_STCR_TFEN0 (1 << 7) +#define SSI_FIFO_ENABLE_0_SHIFT 7 +#define SSI_STCR_TFDIR (1 << 6) +#define SSI_STCR_TXDIR (1 << 5) +#define SSI_STCR_TSHFD (1 << 4) +#define SSI_STCR_TSCKP (1 << 3) +#define SSI_STCR_TFSI (1 << 2) +#define SSI_STCR_TFSL (1 << 1) +#define SSI_STCR_TEFS (1 << 0) + +#define SSI_SRCR 0x20 +#define SSI_SRCR_RXBIT0 (1 << 9) +#define SSI_SRCR_RFEN1 (1 << 8) +#define SSI_SRCR_RFEN0 (1 << 7) +#define SSI_FIFO_ENABLE_0_SHIFT 7 +#define SSI_SRCR_RFDIR (1 << 6) +#define SSI_SRCR_RXDIR (1 << 5) +#define SSI_SRCR_RSHFD (1 << 4) +#define SSI_SRCR_RSCKP (1 << 3) +#define SSI_SRCR_RFSI (1 << 2) +#define SSI_SRCR_RFSL (1 << 1) +#define SSI_SRCR_REFS (1 << 0) + +#define SSI_SRCCR 0x28 +#define SSI_SRCCR_DIV2 (1 << 18) +#define SSI_SRCCR_PSR (1 << 17) +#define SSI_SRCCR_WL(x) ((((x) - 2) >> 1) << 13) +#define SSI_SRCCR_DC(x) (((x) & 0x1f) << 8) +#define SSI_SRCCR_PM(x) (((x) & 0xff) << 0) +#define SSI_SRCCR_WL_MASK (0xf << 13) +#define SSI_SRCCR_DC_MASK (0x1f << 8) +#define SSI_SRCCR_PM_MASK (0xff << 0) + +#define SSI_STCCR 0x24 +#define SSI_STCCR_DIV2 (1 << 18) +#define SSI_STCCR_PSR (1 << 17) +#define SSI_STCCR_WL(x) ((((x) - 2) >> 1) << 13) +#define SSI_STCCR_DC(x) (((x) & 0x1f) << 8) +#define SSI_STCCR_PM(x) (((x) & 0xff) << 0) +#define SSI_STCCR_WL_MASK (0xf << 13) +#define SSI_STCCR_DC_MASK (0x1f << 8) +#define SSI_STCCR_PM_MASK (0xff << 0) + +#define SSI_SFCSR 0x2c +#define SSI_SFCSR_RFCNT1(x) (((x) & 0xf) << 28) +#define SSI_RX_FIFO_1_COUNT_SHIFT 28 +#define SSI_SFCSR_TFCNT1(x) (((x) & 0xf) << 24) +#define SSI_TX_FIFO_1_COUNT_SHIFT 24 +#define SSI_SFCSR_RFWM1(x) (((x) & 0xf) << 20) +#define SSI_SFCSR_TFWM1(x) (((x) & 0xf) << 16) +#define SSI_SFCSR_RFCNT0(x) (((x) & 0xf) << 12) +#define SSI_RX_FIFO_0_COUNT_SHIFT 12 +#define SSI_SFCSR_TFCNT0(x) (((x) & 0xf) << 8) +#define SSI_TX_FIFO_0_COUNT_SHIFT 8 +#define SSI_SFCSR_RFWM0(x) (((x) & 0xf) << 4) +#define SSI_SFCSR_TFWM0(x) (((x) & 0xf) << 0) +#define SSI_SFCSR_RFWM0_MASK (0xf << 4) +#define SSI_SFCSR_TFWM0_MASK (0xf << 0) + +#define SSI_STR 0x30 +#define SSI_STR_TEST (1 << 15) +#define SSI_STR_RCK2TCK (1 << 14) +#define SSI_STR_RFS2TFS (1 << 13) +#define SSI_STR_RXSTATE(x) (((x) & 0xf) << 8) +#define SSI_STR_TXD2RXD (1 << 7) +#define SSI_STR_TCK2RCK (1 << 6) +#define SSI_STR_TFS2RFS (1 << 5) +#define SSI_STR_TXSTATE(x) (((x) & 0xf) << 0) + +#define SSI_SOR 0x34 +#define SSI_SOR_CLKOFF (1 << 6) +#define SSI_SOR_RX_CLR (1 << 5) +#define SSI_SOR_TX_CLR (1 << 4) +#define SSI_SOR_INIT (1 << 3) +#define SSI_SOR_WAIT(x) (((x) & 0x3) << 1) +#define SSI_SOR_WAIT_MASK (0x3 << 1) +#define SSI_SOR_SYNRST (1 << 0) + +#define SSI_SACNT 0x38 +#define SSI_SACNT_FRDIV(x) (((x) & 0x3f) << 5) +#define SSI_SACNT_WR (1 << 4) +#define SSI_SACNT_RD (1 << 3) +#define SSI_SACNT_TIF (1 << 2) +#define SSI_SACNT_FV (1 << 1) +#define SSI_SACNT_AC97EN (1 << 0) + +#define SSI_SACADD 0x3c +#define SSI_SACDAT 0x40 +#define SSI_SATAG 0x44 +#define SSI_STMSK 0x48 +#define SSI_SRMSK 0x4c +#define SSI_SACCST 0x50 +#define SSI_SACCEN 0x54 +#define SSI_SACCDIS 0x58 + +/* SSI clock sources */ +#define IMX_SSP_SYS_CLK 0 + +/* SSI audio dividers */ +#define IMX_SSI_TX_DIV_2 0 +#define IMX_SSI_TX_DIV_PSR 1 +#define IMX_SSI_TX_DIV_PM 2 +#define IMX_SSI_RX_DIV_2 3 +#define IMX_SSI_RX_DIV_PSR 4 +#define IMX_SSI_RX_DIV_PM 5 + +extern struct snd_soc_dai *imx_ssi_pcm_dai[2]; +extern struct snd_soc_platform imx_soc_platform; + +#define DRV_NAME "imx-ssi" + +struct imx_pcm_dma_params { + int dma; + unsigned long dma_addr; + int burstsize; +}; + +struct imx_ssi { + struct snd_soc_dai dai; + struct platform_device *ac97_dev; + + struct snd_soc_device imx_ac97; + struct clk *clk; + void __iomem *base; + int irq; + int fiq_enable; + unsigned int offset; + + unsigned int flags; + + void (*ac97_reset) (struct snd_ac97 *ac97); + void (*ac97_warm_reset)(struct snd_ac97 *ac97); + + struct imx_pcm_dma_params dma_params_rx; + struct imx_pcm_dma_params dma_params_tx; + + int enabled; +}; + +struct snd_soc_platform *imx_ssi_fiq_init(struct platform_device *pdev, + struct imx_ssi *ssi); +void imx_ssi_fiq_exit(struct platform_device *pdev, struct imx_ssi *ssi); +struct snd_soc_platform *imx_ssi_dma_mx2_init(struct platform_device *pdev, + struct imx_ssi *ssi); + +int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma); +int imx_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, + struct snd_pcm *pcm); +void imx_pcm_free(struct snd_pcm *pcm); + +/* + * Do not change this as the FIQ handler depends on this size + */ +#define IMX_SSI_DMABUF_SIZE (64 * 1024) + +#define DMA_RXFIFO_BURST 0x4 +#define DMA_TXFIFO_BURST 0x6 + +#endif /* _IMX_SSI_H */ -- 1.6.6

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[alsa-devel] [PATCH] ASoC: tpa6130a2: Support for tpa6140's regulators
by Ilkka Koskinen 11 Jan '10

11 Jan '10

tpa6140a2 uses different names for the regulators. Signed-off-by: Ilkka Koskinen <ilkka.koskinen(a)nokia.com> --- include/sound/tpa6130a2-plat.h | 6 ++++++ sound/soc/codecs/tpa6130a2.c | 22 ++++++++++++++++++++-- 2 files changed, 26 insertions(+), 2 deletions(-) diff --git a/include/sound/tpa6130a2-plat.h b/include/sound/tpa6130a2-plat.h index e8c901e..e29fde6 100644 --- a/include/sound/tpa6130a2-plat.h +++ b/include/sound/tpa6130a2-plat.h @@ -23,7 +23,13 @@ #ifndef TPA6130A2_PLAT_H #define TPA6130A2_PLAT_H +enum tpa_model { + TPA6130A2, + TPA6140A2, +}; + struct tpa6130a2_platform_data { + enum tpa_model id; int power_gpio; }; diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 8e98ccf..8b27281 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -41,6 +41,11 @@ static const char *tpa6130a2_supply_names[TPA6130A2_NUM_SUPPLIES] = { "Vdd", }; +static const char *tpa6140a2_supply_names[TPA6130A2_NUM_SUPPLIES] = { + "HPVdd", + "AVdd", +}; + /* This struct is used to save the context */ struct tpa6130a2_data { struct mutex mutex; @@ -420,8 +425,21 @@ static int tpa6130a2_probe(struct i2c_client *client, gpio_direction_output(data->power_gpio, 0); } - for (i = 0; i < ARRAY_SIZE(data->supplies); i++) - data->supplies[i].supply = tpa6130a2_supply_names[i]; + switch (pdata->id) { + case TPA6130A2: + for (i = 0; i < ARRAY_SIZE(data->supplies); i++) + data->supplies[i].supply = tpa6130a2_supply_names[i]; + break; + case TPA6140A2: + for (i = 0; i < ARRAY_SIZE(data->supplies); i++) + data->supplies[i].supply = tpa6140a2_supply_names[i];; + break; + default: + dev_warn(dev, "Unknown TPA model (%d). Assuming 6130A2\n", + pdata->id); + for (i = 0; i < ARRAY_SIZE(data->supplies); i++) + data->supplies[i].supply = tpa6130a2_supply_names[i]; + } ret = regulator_bulk_get(dev, ARRAY_SIZE(data->supplies), data->supplies); -- 1.6.0.4

3 2

[alsa-devel] Problems with snd_pcm_wait(..)
by Markus Luttenberger 11 Jan '10

11 Jan '10

Hey, I tried to implement a callback functionality with ALSA based on the example given at http://www.cs.utk.edu/~vose/linux/equalarea.com/alsa-audio.html#interruptex. Especially the while-loop is basically the same in my program. However, it doesn't work. I can trace the program to snd_pcm_wait(..) which just waits endlessly. The whole thing is called within a thread. The thread is still accessible from elsewhere in the pogram. It's just the snd_pcm_wait(..)-function which does not return although a time-out of 1000 ms is given. If I don't wait for the device I get the following assertion: pcm.c:1079: snd_pcm_writei: Assertion `size == 0 || buffer' failed. Any ideas what could be wrong? greetings.. ..Markus

2 2

[alsa-devel] alsa-util.c: snd_pcm_avail() returned a value that is exceptionally large: 349696 bytes (1821 ms).
by Chris 11 Jan '10

11 Jan '10

Mandriva 2010.0, Gnome 2.28, pulseaudio 0.9.19-7mdv2010.0. With pulseaudio enabled in MCC periodically the hd light will constantly be on for a period of 4 or 5 minutes and system will slow down to a crawl. Also these comments will be in syslog: pulseaudio[4331]: alsa-util.c: snd_pcm_avail() returned a value that is exceptionally large: 349696 bytes (1821 ms). Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: Most likely this is a bug in the ALSA driver 'snd_via82xx'. Please report this issue to the ALSA developers. Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: snd_pcm_dump(): Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: Hardware PCM card 1 'VIA 8237' device 0 subdevice 0 Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: Its setup is: Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: stream : PLAYBACK Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: access : MMAP_INTERLEAVED Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: format : S16_LE Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: subformat : STD Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: channels : 2 Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: rate : 48000 Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: exact rate : 48000 (48000/1) Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: msbits : 16 Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: buffer_size : 16384 Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: period_size : 8192 Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: period_time : 170666 Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: tstamp_mode : ENABLE Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: period_step : 1 Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: avail_min : 9665 Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: period_event : 0 Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: start_threshold : -1 Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: stop_threshold : 1073741824 Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: silence_threshold: 0 Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: silence_size : 0 Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: boundary : 1073741824 Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: appl_ptr : 767919576 Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: hw_ptr : 767990616 Dec 14 20:26:29 localhost pulseaudio[4331]: ratelimit.c: 7 events suppressed Dec 14 20:26:31 localhost pulseaudio[4331]: asyncq.c: q overrun, queuing locally I also see this quite frequently: Dec 14 20:15:29 localhost pulseaudio[4331]: asyncq.c: q overrun, queuing locally Dec 14 20:15:34 localhost pulseaudio[4331]: ratelimit.c: 657 events suppressed Dec 14 20:15:34 localhost pulseaudio[4331]: asyncq.c: q overrun, queuing locally Dec 14 20:15:39 localhost pulseaudio[4331]: ratelimit.c: 2508 events suppressed Dec 14 20:15:39 localhost pulseaudio[4331]: asyncq.c: q overrun, queuing locally Dec 14 20:15:44 localhost pulseaudio[4331]: ratelimit.c: 3530 events suppressed Dec 14 20:15:44 localhost pulseaudio[4331]: asyncq.c: q overrun, queuing locally Dec 14 20:15:44 localhost pulseaudio[4331]: asyncq.c: q overrun, queuing locally Dec 14 20:15:49 localhost pulseaudio[4331]: ratelimit.c: 2672 events suppressed Dec 14 20:15:49 localhost pulseaudio[4331]: asyncq.c: q overrun, queuing locally Dec 14 20:16:05 localhost pulseaudio[4331]: ratelimit.c: 2244 events suppressed Dec 14 20:16:18 localhost pulseaudio[4331]: ratelimit.c: 1 events suppressed Dec 14 20:16:23 localhost pulseaudio[4331]: ratelimit.c: 1 events suppressed Dec 14 20:16:34 localhost pulseaudio[4331]: ratelimit.c: 5 events suppressed Dec 14 20:16:39 localhost pulseaudio[4331]: ratelimit.c: 9 events suppressed Dec 14 20:16:45 localhost pulseaudio[4331]: ratelimit.c: 11 events suppressed Dec 14 20:16:50 localhost pulseaudio[4331]: ratelimit.c: 13 events suppressed Dec 14 20:16:55 localhost pulseaudio[4331]: ratelimit.c: 16 events suppressed Dec 14 20:17:01 localhost pulseaudio[4331]: ratelimit.c: 17 events suppressed Dec 14 20:17:06 localhost pulseaudio[4331]: ratelimit.c: 23 events suppressed Dec 14 20:17:11 localhost pulseaudio[4331]: ratelimit.c: 19 events suppressed Dec 14 20:17:31 localhost pulseaudio[4331]: ratelimit.c: 23 events suppressed Dec 14 20:17:36 localhost pulseaudio[4331]: ratelimit.c: 19 events suppressed Dec 14 20:17:42 localhost pulseaudio[4331]: ratelimit.c: 21 events suppressed Dec 14 20:17:47 localhost pulseaudio[4331]: ratelimit.c: 19 events suppressed I have pulseaudio enabled but have glitch-free mode disabled. Any assistance in troubleshooting this would be appreciated. Chris -- KeyID 0xE372A7DA98E6705C

3 22

[alsa-devel] [PATCH] ac97: add AC97 STMicroelectronics' codecs
by Krzysztof Helt 10 Jan '10

10 Jan '10

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Add the STMicroelectronics ST7597 codec and an unknown codec from the same manufacturer found on the Creative SB 128 card (CT4810). Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- Datasheet of the ST7597 can be found on the web (various datasheet sites have copies). sound/pci/ac97/ac97_codec.c | 10 ++++++++++ sound/pci/ac97/ac97_id.h | 2 ++ 2 files changed, 12 insertions(+), 0 deletions(-) diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index c119206..a7630e9 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -83,6 +83,7 @@ static const struct ac97_codec_id snd_ac97_codec_id_vendors[] = { { 0x4e534300, 0xffffff00, "National Semiconductor", NULL, NULL }, { 0x50534300, 0xffffff00, "Philips", NULL, NULL }, { 0x53494c00, 0xffffff00, "Silicon Laboratory", NULL, NULL }, +{ 0x53544d00, 0xffffff00, "STMicroelectronics", NULL, NULL }, { 0x54524100, 0xffffff00, "TriTech", NULL, NULL }, { 0x54584e00, 0xffffff00, "Texas Instruments", NULL, NULL }, { 0x56494100, 0xffffff00, "VIA Technologies", NULL, NULL }, @@ -161,6 +162,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = { { 0x4e534350, 0xffffffff, "LM4550", patch_lm4550, NULL }, // volume wrap fix { 0x50534304, 0xffffffff, "UCB1400", patch_ucb1400, NULL }, { 0x53494c20, 0xffffffe0, "Si3036,8", mpatch_si3036, mpatch_si3036, AC97_MODEM_PATCH }, +{ 0x53544d02, 0xffffffff, "ST7597", NULL, NULL }, { 0x54524102, 0xffffffff, "TR28022", NULL, NULL }, { 0x54524103, 0xffffffff, "TR28023", NULL, NULL }, { 0x54524106, 0xffffffff, "TR28026", NULL, NULL }, @@ -213,6 +215,14 @@ static int snd_ac97_valid_reg(struct snd_ac97 *ac97, unsigned short reg) { /* filter some registers for buggy codecs */ switch (ac97->id) { + case AC97_ID_ST_AC97_ID4: + if (reg == 0x08) + return 0; + /* fall through */ + case AC97_ID_ST7597: + if (reg == 0x22 || reg == 0x7a) + return 1; + /* fall through */ case AC97_ID_AK4540: case AC97_ID_AK4542: if (reg <= 0x1c || reg == 0x20 || reg == 0x26 || reg >= 0x7c) diff --git a/sound/pci/ac97/ac97_id.h b/sound/pci/ac97/ac97_id.h index c129492..d603147 100644 --- a/sound/pci/ac97/ac97_id.h +++ b/sound/pci/ac97/ac97_id.h @@ -62,3 +62,5 @@ #define AC97_ID_CM9761_78 0x434d4978 #define AC97_ID_CM9761_82 0x434d4982 #define AC97_ID_CM9761_83 0x434d4983 +#define AC97_ID_ST7597 0x53544d02 +#define AC97_ID_ST_AC97_ID4 0x53544d04 -- 1.6.4

2 1

[alsa-devel] [PATCH] ALSA: ac97: Add Dell Dimension 2400 to Headphone/Line Jack Sense blacklist
by Daniel Chen 10 Jan '10

10 Jan '10

Hi, Attached is a patch against current sound-2.6 master HEAD fixing inaudible playback by default. Thanks, -Dan

2 1

[alsa-devel] aplay - Audio Surveilance
by John Sauter 10 Jan '10

10 Jan '10

I have added some features to aplay to make audio surveillance easier: --process-id-file writes the PID to a file so another program can signal aplay to close its capture file and open another one without missing any sound. --max-file-time causes the close and re-open to happen automatically. --use-strftime allows capture files to be named for their start times. Details are in the man file. For ease of review, this patch is divided into three parts. Part 1 is just --process-id-file, which is intrusive because it must intercept the exit points to delete the file. Part 2 is the remainder of the new features in aplay.c. Part 3 updates the man file. John Sauter (John_Sauter(a)systemeyescomputerstore.com)

2 1

[alsa-devel] VGN-P11Z_G (hda-intel ALC262) internal microphone not working
by Christian Stöveken 09 Jan '10

09 Jan '10

I'm running Ubuntu Karmic on a Vaio P11z and wanted to ask if there's anything I can do to help get the not working microphone going for me and all P11z users ;-) Manufacturer: Sony Corporation Product Name: VGN-P11Z_G Soundcards recognised by ALSA: HDA-Intel - HDA Intel MID Codec: Realtek ALC262 I already upgraded to alsa 1.0.21 using that http://www.webupd8.org/2009/11/ubuntu-karmic-upgrade-alsa-to-1021-from.html… and am atm on 2.6.33-020633rc3-generic. alsa-info: -> http://www.alsa-project.org/db/?f=35205f955e08e7e16b5274862f6b376b890731e3 -> toshiba-s06 (somewhat working) http://www.alsa-project.org/db/?f=8ae0ab7b6672e41550651dbda63587438c93756f the issues: - Microphone isn't working with toshiba-s06 it is sometimes working right after powering up, sometimes I have to force-reload alsa. - Sound also comes from the speakers with plugged-in headphones sony-assamd / toshiba-s06 (and multiple others fix that issue) Regards, Chris <http://mailman.alsa-project.org/mailman/listinfo/alsa-devel>

2 3

[PATCH] pulse: Fix invalid buffer pointer return value
by David Henningsson 09 Jan '10

09 Jan '10

This patch improves recovering from underruns, and prevents hangs inside snd_pcm_write* and snd_pcm_read* due to snd_pcm_avail* returning too low values. It especially helps low latency situations. Signed-off-by: David Henningsson <diwic(a)ubuntu.com> --- pulse/pcm_pulse.c | 15 ++++++++++++++- 1 files changed, 14 insertions(+), 1 deletions(-) Index: alsa-plugins-1.0.22/pulse/pcm_pulse.c =================================================================== --- alsa-plugins-1.0.22.orig/pulse/pcm_pulse.c 2010-02-28 17:12:02.000000000 -0500 +++ alsa-plugins-1.0.22/pulse/pcm_pulse.c 2010-02-28 17:12:25.000000000 -0500 @@ -90,6 +90,10 @@ if (pcm->io.stream == SND_PCM_STREAM_CAPTURE) size -= pcm->offset; + /* Prevent accidental overrun of the fake ringbuffer */ + if (size >= pcm->buffer_attr.tlength) + size = pcm->buffer_attr.tlength-1; + if (size > pcm->last_size) { pcm->ptr += size - pcm->last_size; pcm->ptr %= pcm->buffer_attr.tlength; @@ -424,6 +428,7 @@ snd_pcm_pulse_t *pcm = io->private_data; const char *buf; snd_pcm_sframes_t ret = 0; + size_t writebytes; assert(pcm); @@ -445,13 +450,15 @@ (char *) areas->addr + (areas->first + areas->step * offset) / 8; - ret = pa_stream_write(pcm->stream, buf, size * pcm->frame_size, NULL, 0, 0); + writebytes = size * pcm->frame_size; + ret = pa_stream_write(pcm->stream, buf, writebytes, NULL, 0, 0); if (ret < 0) { ret = -EIO; goto finish; } /* Make sure the buffer pointer is in sync */ + pcm->last_size -= writebytes; ret = update_ptr(pcm); if (ret < 0) goto finish; @@ -528,6 +535,7 @@ dst_buf = (char *) dst_buf + frag_length; remain_size -= frag_length; + pcm->last_size -= frag_length; } /* Make sure the buffer pointer is in sync */ @@ -730,6 +738,11 @@ pcm->offset = 0; /*pcm->underrun = 0;*/ + /* Reset fake ringbuffer */ + pcm->last_size = 0; + pcm->ptr = 0; + update_ptr(pcm); + finish: pa_threaded_mainloop_unlock(pcm->p->mainloop); --------------040006090607070407080605--

1 0

[alsa-devel] [RFC] audio surveilance
by John Sauter 08 Jan '10

08 Jan '10

When doing audio surveillance, you want to record all the sound, but occasionally close the old output file and open a new one, so you can listen to a captured event without disturbing the recording of future sounds. I have some ideas for additions to aplay to meet the needs of audio surveillance, and would appreciate feedback from the alsa developers. My proposed patches are attached. They have been tested only on GNU/Linux 64-bit. John Sauter (John_Sauter(a)systemeyescomputerstore.com)

2 3

Re: [alsa-devel] alsa support for HDA ALC889A nVidia Corporation MCP55 High Definition Audio EVGA 122-CK-NF68
by Oliver Stieber 08 Jan '10

08 Jan '10

just quit x and tried mb3, no sound. module snd-hda-intel did unload ok though (often the module is still showen as used when it's not) i wrote a little bash script to try every model listed in the source code, and mb5 was the only one that workedish and i had lots of kernel mess along the way. On Jan 8, 2010 3:46 PM, "Justin P. Mattock" <justinmattock(a)gmail.com> wrote: On 01/07/10 16:39, Oliver Stieber wrote: > > Hi, > > I'm having some issues, (and think I've discove... using model=mb3 gives no effect? Justin P. Mattock

2 1

[alsa-devel] [PATCH V2] Add i.MX sound support
by Sascha Hauer 08 Jan '10

08 Jan '10

Hi All, Here is the second version of the patches to add i.MX sound support. This time I haven't included ARM related patches, so this series should be safe to apply. I skipped the mc13783 driver since I'm not ready integrating the comments from Mark. The main change in the driver is that I splitted out the transport from the driver, so it should be easy to add another transport layer to the driver when SDMA support is merged. I hopefully also addressed the other comments from Mark. Sascha

3 20

[alsa-devel] Speakers won't turn off, when headphones are plugged in
by Joern Bernhardt 08 Jan '10

08 Jan '10

Hi there, my speakers won't turn off, when I plug in headphones. I've tried all of these models: ref Reference board ref-no-jd Reference board without HP/Mic jack detection 3stack D965 3stack 5stack D965 5stack SPDIF 5stack-no-fp D965 5stack without front panel dell-3stack Dell Dimension E520 dell-bios Fixes with Dell BIOS setup auto BIOS setup (default) Some of these only had sound when headphones were plugged in (speakers didn't work at all) or when they were plugged in the microphone jack. My laptop is a HP HDX 9000 and my alsa-info script output is here: http://www.alsa-project.org/db/?f=b5a4386553266384382868ddef56553d643e7b4a Thanks for helping, Joern

2 1

[alsa-devel] alsa support for HDA ALC889A nVidia Corporation MCP55 High Definition Audio EVGA 122-CK-NF68
by Oliver Stieber 08 Jan '10

08 Jan '10

Hi, I'm having some issues, (and think I've discovered some bugs/ performance features) in alsa with my card. Certainly sound used to work on front stereo speakers with mode=auto in ubuntu 9, but had to change to model=mb5 and have severe stability issues (alsa no longer works, kernel gets trashed) when I do things like try to change volume, and kde volume changer doesn't work unless I use the mixer etc... Also, this is the only model where I get sound, other models trash kernel space etc... With this model I can't unload the module as it says it's still in use even with the sound app has terminated (didn't have this problem under ubuntu 9) Have some dmesg dumps in a bug I filed (will dig it out). Anyhow, I'm more than happy to have a crack at getting my sound card working perfectly. Oliver. For starters.... alsa-info.sh is here. http://www.alsa-project.org/db/?f=a65b159863be66dc66863ec82fb65d616bd2fd52 !!################################ !!ALSA Information Script v 0.4.58 !!################################ !!Script ran on: Fri Jan 8 00:30:41 UTC 2010 !!Linux Distribution !!------------------ Ubuntu 9.10 \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu 9.10" !!DMI Information !!--------------- Manufacturer: EVGA Product Name: 122-CK-NF68 !!Kernel Information !!------------------ Kernel release: 2.6.32-0.dmz.2-liquorix-amd64 Operating System: GNU/Linux Architecture: x86_64 Processor: unknown SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.22 Library version: 1.0.20 Utilities version: 1.0.22 !!Loaded ALSA modules !!------------------- snd_hda_intel !!Sound Servers on this system !!---------------------------- aRts: Installed - Yes (/usr/bin/artsd) Running - No Jack: Installed - Yes (/usr/bin/jackd) Running - No !!Soundcards recognised by ALSA !!----------------------------- 0 [NVidia ]: HDA-Intel - HDA NVidia HDA NVidia at 0xcfff0000 irq 25 !!PCI Soundcards installed in the system !!-------------------------------------- 00:0f.1 Audio device: nVidia Corporation MCP55 High Definition Audio (rev a2) !!Advanced information - PCI Vendor/Device/Susbsystem ID's !!-------------------------------------------------------- 00:0f.1 0403: 10de:0371 (rev a2) Subsystem: 10de:c55e -- Subsystem: 10b0:0401 Control: I/O Mem BusMaster SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- !!Modprobe options (Sound related) !!-------------------------------- snd-atiixp-modem: index=-2 snd-intel8x0m: index=-2 snd-via82xx-modem: index=-2 snd-usb-audio: index=-2 snd-usb-us122l: index=-2 snd-usb-usx2y: index=-2 snd-usb-caiaq: index=-2 snd-cmipci: mpu_port=0x330 fm_port=0x388 snd-pcsp: index=-2 snd-hda-intel: power_save=0 model=mb5 !!Loaded sound module options !!-------------------------- !!Module: snd_hda_intel bdl_pos_adj : 32,-1,-1,-1,-1,-1,-1,-1 beep_mode : 1,1,1,1,1,1,1,1 enable : Y,Y,Y,Y,Y,Y,Y,Y enable_msi : -1 id : <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> index : -1,-1,-1,-1,-1,-1,-1,-1 model : mb5,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> patch : <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> position_fix : 0,0,0,0,0,0,0,0 power_save : 0 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1 probe_only : N,N,N,N,N,N,N,N single_cmd : N !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: Realtek ALC889A Address: 0 Function Id: 0x1 Vendor Id: 0x10ec0885 Subsystem Id: 0x10ece601 Revision Id: 0x100101 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=0 IO[0]: enable=1, dir=1, wake=0, sticky=0, data=1, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0x11: Stereo Device: name="ALC889A Analog", type="Audio", device=0 Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x05 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital Control: name="IEC958 Playback Con Mask", index=0, device=0 Control: name="IEC958 Playback Pro Mask", index=0, device=0 Control: name="IEC958 Playback Default", index=0, device=0 Control: name="IEC958 Playback Switch", index=0, device=0 Control: name="IEC958 Default PCM Playback Switch", index=0, device=0 Device: name="ALC889A Digital", type="SPDIF", device=1 Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x07 [Audio Input] wcaps 0x10011b: Stereo Amp-In Control: name="Capture Switch", index=0, device=0 Control: name="Capture Volume", index=0, device=0 Device: name="ALC889A Analog", type="Audio", device=0 Amp-In caps: ofs=0x10, nsteps=0x2e, stepsize=0x03, mute=1 Amp-In vals: [0x1c 0x1c] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Connection: 1 0x24 Node 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In Control: name="Capture Switch", index=1, device=0 Control: name="Capture Volume", index=1, device=0 Device: name="ALC889A Analog", type="Audio", device=2 Amp-In caps: ofs=0x10, nsteps=0x2e, stepsize=0x03, mute=1 Amp-In vals: [0x80 0x80] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In Control: name="Capture Switch", index=2, device=0 Control: name="Capture Volume", index=2, device=0 Amp-In caps: ofs=0x10, nsteps=0x2e, stepsize=0x03, mute=1 Amp-In vals: [0x80 0x80] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Connection: 1 0x22 Node 0x0a [Audio Input] wcaps 0x100391: Stereo Digital Control: name="IEC958 Capture Switch", index=0, device=0 Control: name="IEC958 Capture Default", index=0, device=0 Device: name="ALC889A Digital", type="SPDIF", device=1 Converter: stream=0, channel=0 SDI-Select: 0 Digital: Digital category: 0x0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Unsolicited: tag=00, enabled=0 Connection: 1 0x1f Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="Line Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Control: name="Line Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Control: name="Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Beep Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=5, ofs=0 Control: name="Beep Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=5, ofs=0 Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x97 0x97] [0x80 0x80] [0x80 0x80] [0x97 0x97] [0x97 0x97] [0x00 0x00] [0x97 0x97] [0x97 0x97] [0x97 0x97] [0x97 0x97] Connection: 10 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 Node 0x0c [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="Front Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x2c 0x2c] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="Surround Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x1d 0x1d] Connection: 2 0x03 0x0b Node 0x0e [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="LFE Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x19 0x00] Connection: 2 0x04 0x0b Node 0x0f [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="HP Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x00 0x00] Connection: 2 0x05 0x0b Node 0x10 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x12 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x14 [Pin Complex] wcaps 0x40018f: 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 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x01014010: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d 0x0e 0x0f* 0x26 Node 0x15 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Line Boost", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 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 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x01011012: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0x2 Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d* 0x0e 0x0f 0x26 Node 0x16 [Pin Complex] wcaps 0x40018f: 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 0x0000003c: IN OUT HP Detect Pin Default 0x01016011: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Orange DefAssociation = 0x1, Sequence = 0x1 Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d 0x0e* 0x0f 0x26 Node 0x17 [Pin Complex] wcaps 0x40018f: 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 0x0000003c: IN OUT HP Detect Pin Default 0x01012014: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Grey DefAssociation = 0x1, Sequence = 0x4 Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d 0x0e 0x0f* 0x26 Node 0x18 [Pin Complex] wcaps 0x40018f: 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 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x01a19830: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x3, Sequence = 0x0 Pin-ctls: 0x41: OUT VREF_50 Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x19 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Mic Boost", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 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 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x02a1983f: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0x3, Sequence = 0xf Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1a [Pin Complex] wcaps 0x40018f: 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 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x01813031: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x3, Sequence = 0x1 Pin-ctls: 0x41: OUT VREF_50 Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d 0x0e* 0x0f 0x26 Node 0x1b [Pin Complex] wcaps 0x40018f: 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 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x02214c1f: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1c [Pin Complex] wcaps 0x400001: 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: 0x20: IN Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0x99830132: [Fixed] Line In at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0x3, Sequence = 0x2 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital Pincap 0x00000010: OUT Pin Default 0x01441120: [Jack] SPDIF Out at Ext Rear Conn = RCA, Color = Black DefAssociation = 0x2, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x06 Node 0x1f [Pin Complex] wcaps 0x400200: Mono Digital Pincap 0x00000020: IN Pin Default 0x01c46140: [Jack] SPDIF In at Ext Rear Conn = RCA, Color = Orange DefAssociation = 0x4, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=17 Processing Coefficient: 0xc128 Coefficient Index: 0x02 Node 0x21 [Volume Knob Widget] wcaps 0x600080: Mono Volume-Knob: delta=0, steps=32, direct=0, val=64 Unsolicited: tag=00, enabled=0 Connection: 0 Node 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="Input Source", index=0, device=0 Control: name="Input Source", index=1, device=0 Control: name="Input Source", index=2, device=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [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 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="Input Source", index=0, device=0 Control: name="Input Source", index=1, device=0 Control: name="Input Source", index=2, device=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [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 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="Input Source", index=0, device=0 Control: name="Input Source", index=1, device=0 Control: name="Input Source", index=2, device=0 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 0x25 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM 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=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] Connection: 2 0x25 0x0b --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw---- 1 root audio 116, 0 Jan 7 22:48 /dev/snd/controlC0 crw-rw---- 1 root audio 116, 4 Jan 7 22:48 /dev/snd/hwC0D0 crw-rw---- 1 root audio 116, 24 Jan 7 22:48 /dev/snd/pcmC0D0c crw-rw---- 1 root audio 116, 16 Jan 7 23:01 /dev/snd/pcmC0D0p crw-rw---- 1 root audio 116, 25 Jan 7 22:48 /dev/snd/pcmC0D1c crw-rw---- 1 root audio 116, 17 Jan 7 22:48 /dev/snd/pcmC0D1p crw-rw---- 1 root audio 116, 26 Jan 7 22:48 /dev/snd/pcmC0D2c crw-rw---- 1 root audio 116, 1 Jan 7 22:47 /dev/snd/seq crw-rw---- 1 root audio 116, 33 Jan 7 22:47 /dev/snd/timer /dev/snd/by-path: total 0 drwxr-xr-x 2 root root 60 Jan 7 22:48 . drwxr-xr-x 3 root root 240 Jan 7 22:48 .. lrwxrwxrwx 1 root root 12 Jan 7 22:48 pci-0000:00:0f.1 -> ../controlC0 !!Aplay/Arecord output !!------------ APLAY **** List of PLAYBACK Hardware Devices **** card 0: NVidia [HDA NVidia], device 0: ALC889A Analog [ALC889A Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: NVidia [HDA NVidia], device 1: ALC889A Digital [ALC889A Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: NVidia [HDA NVidia], device 0: ALC889A Analog [ALC889A Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: NVidia [HDA NVidia], device 1: ALC889A Digital [ALC889A Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: NVidia [HDA NVidia], device 2: ALC889A Analog [ALC889A Analog] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 !!Amixer output !!------------- !!-------Mixer controls for card 0 [NVidia] Card hw:0 'NVidia'/'HDA NVidia at 0xcfff0000 irq 25' Mixer name : 'Realtek ALC889A' Components : 'HDA:10ec0885,10ece601,00100101' Controls : 36 Simple ctrls : 20 Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 64 Mono: Playback 44 [69%] [-20.00dB] [on] Simple mixer control 'PCM',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [0.00dB] Front Right: Playback 255 [100%] [0.00dB] Simple mixer control 'Front',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 64 [100%] [0.00dB] [on] Front Right: Playback 64 [100%] [0.00dB] [on] Simple mixer control 'Surround',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 49 [77%] [-15.00dB] [on] Front Right: Playback 49 [77%] [-15.00dB] [on] Simple mixer control 'LFE',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 45 [70%] [-19.00dB] [on] Front Right: Playback 0 [0%] [-64.00dB] [on] Simple mixer control 'Line',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Line Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] Front Right: 0 [0%] Simple mixer control 'Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] Front Right: 0 [0%] Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined cswitch cswitch-joined Playback channels: Mono Capture channels: Mono Mono: Playback [off] Capture [off] Simple mixer control 'IEC958 Default PCM',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 46 Front Left: Capture 28 [61%] [12.00dB] [on] Front Right: Capture 28 [61%] [12.00dB] [on] Simple mixer control 'Capture',1 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 46 Front Left: Capture 0 [0%] [-16.00dB] [off] Front Right: Capture 0 [0%] [-16.00dB] [off] Simple mixer control 'Capture',2 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 46 Front Left: Capture 0 [0%] [-16.00dB] [off] Front Right: Capture 0 [0%] [-16.00dB] [off] Simple mixer control 'Beep',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [on] Front Right: Playback 0 [0%] [-34.50dB] [on] Simple mixer control 'Channel Mode',0 Capabilities: enum Items: '2ch' '6ch' Item0: '2ch' Simple mixer control 'HP',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 0 [0%] [-64.00dB] [on] Front Right: Playback 0 [0%] [-64.00dB] [on] Simple mixer control 'Input Source',0 Capabilities: cenum Items: 'Mic' 'Line' 'CD' Item0: 'Mic' Simple mixer control 'Input Source',1 Capabilities: cenum Items: 'Mic' 'Line' 'CD' Item0: 'Mic' Simple mixer control 'Input Source',2 Capabilities: cenum Items: 'Mic' 'Line' 'CD' Item0: 'Mic' !!Alsactl output !!------------- --startcollapse-- state.NVidia { control.1 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 64' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'Front Playback Volume' value.0 64 value.1 64 } control.2 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Front Playback Switch' value.0 true value.1 true } control.3 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 64' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'Surround Playback Volume' value.0 49 value.1 49 } control.4 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Surround Playback Switch' value.0 true value.1 true } control.5 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 64' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'LFE Playback Volume' value.0 45 value.1 0 } control.6 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'LFE Playback Switch' value.0 true value.1 true } control.7 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 64' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'HP Playback Volume' value.0 0 value.1 0 } control.8 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'HP Playback Switch' value.0 true value.1 true } control.9 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'Line Playback Volume' value.0 0 value.1 0 } control.10 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Line Playback Switch' value.0 false value.1 false } control.11 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'Mic Playback Volume' value.0 0 value.1 0 } control.12 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Mic Playback Switch' value.0 false value.1 false } control.13 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 3' comment.dbmin 0 comment.dbmax 3000 iface MIXER name 'Line Boost' value.0 0 value.1 0 } control.14 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 3' comment.dbmin 0 comment.dbmax 3000 iface MIXER name 'Mic Boost' value.0 0 value.1 0 } control.15 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 '2ch' comment.item.1 '6ch' iface MIXER name 'Channel Mode' value '2ch' } control.16 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Capture Switch' value.0 true value.1 true } control.17 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Capture Switch' index 1 value.0 false value.1 false } control.18 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Capture Switch' index 2 value.0 false value.1 false } control.19 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 46' comment.dbmin -1600 comment.dbmax 3000 iface MIXER name 'Capture Volume' value.0 28 value.1 28 } control.20 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 46' comment.dbmin -1600 comment.dbmax 3000 iface MIXER name 'Capture Volume' index 1 value.0 0 value.1 0 } control.21 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 46' comment.dbmin -1600 comment.dbmax 3000 iface MIXER name 'Capture Volume' index 2 value.0 0 value.1 0 } control.22 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 Mic comment.item.1 Line comment.item.2 CD iface MIXER name 'Input Source' value Mic } control.23 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 Mic comment.item.1 Line comment.item.2 CD iface MIXER name 'Input Source' index 1 value Mic } control.24 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 Mic comment.item.1 Line comment.item.2 CD iface MIXER name 'Input Source' index 2 value Mic } control.25 { comment.access read comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Con Mask' value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.26 { comment.access read comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Pro Mask' value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.27 { comment.access 'read write' comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Default' value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.28 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'IEC958 Playback Switch' value false } control.29 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'IEC958 Default PCM Playback Switch' value true } control.30 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'IEC958 Capture Switch' value false } control.31 { comment.access read comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Capture Default' value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.32 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'Beep Playback Volume' value.0 0 value.1 0 } control.33 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Beep Playback Switch' value.0 true value.1 true } control.34 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 64' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'Master Playback Volume' value 44 } control.35 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Master Playback Switch' value true } control.36 { comment.access 'read write user' comment.type INTEGER comment.count 2 comment.range '0 - 255' comment.tlv '0000000100000008ffffec1400000014' comment.dbmin -5100 comment.dbmax 0 iface MIXER name 'PCM Playback Volume' value.0 255 value.1 255 } } --endcollapse-- !!All Loaded Modules !!------------------ Module binfmt_misc ppdev rfcomm sco bridge stp bnep l2cap crc16 ipt_ULOG battery cpufreq_powersave cpufreq_performance af_packet usblp ipv6 snd_hda_codec_realtek snd_hda_intel arc4 snd_hda_codec ecb snd_hwdep snd_pcm_oss snd_mixer_oss iptable_filter ip_tables x_tables b43 snd_pcm rng_core snd_seq_oss snd_seq_midi_event snd_seq mac80211 snd_timer snd_seq_device btusb snd cfg80211 soundcore i2c_nforce2 snd_page_alloc led_class bluetooth rfkill lp evdev rtc_cmos processor parport ext3 jbd mbcache sr_mod usbhid hid sg cdrom sd_mod crc_t10dif ata_generic usb_storage fan ohci_hcd pata_amd fuse ssb pata_acpi ehci_hcd sata_nv floppy nvidia mmc_core pcmcia forcedeth pcmcia_core usbcore nls_base i2c_core libata thermal scsi_mod button !!Sysfs Files !!----------- /sys/class/sound/hwC0D0/init_pin_configs: 0x14 0x01014010 0x15 0x01011012 0x16 0x01016011 0x17 0x01012014 0x18 0x01a19830 0x19 0x02a1983f 0x1a 0x01813031 0x1b 0x02214c1f 0x1c 0x411111f0 0x1d 0x99830132 0x1e 0x01441120 0x1f 0x01c46140 /sys/class/sound/hwC0D0/driver_pin_configs: /sys/class/sound/hwC0D0/user_pin_configs: /sys/class/sound/hwC0D0/init_verbs: !!ALSA/HDA dmesg !!------------------ ACPI: PCI Interrupt Link [AAZA] enabled at IRQ 20 HDA Intel 0000:00:0f.1: PCI INT B -> Link[AAZA] -> GSI 20 (level, low) -> IRQ 20 alloc irq_desc for 25 on node -1 alloc kstat_irqs on node -1 HDA Intel 0000:00:0f.1: irq 25 for MSI/MSI-X HDA Intel 0000:00:0f.1: setting latency timer to 64 input: HDA Digital PCBeep as /devices/pci0000:00/0000:00:0f.1/input/input5 EXT3 FS on sda3, internal journal

2 1

[alsa-devel] [PATCH] sbawe: fix memory detection part 2
by Krzysztof Helt 08 Jan '10

08 Jan '10

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> The patch "sbawe: fix memory detection" fixed detection for memoryless SB32 cards but broke detection of memory above 512KB. This patch fixes the regression. The patch has been tested on the SB32 card (CT3670) with 0MB, 2MB and 8MB memory installed. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- sound/isa/sb/emu8000.c | 11 +++++++---- 1 files changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/isa/sb/emu8000.c b/sound/isa/sb/emu8000.c index 751762f..0c40951 100644 --- a/sound/isa/sb/emu8000.c +++ b/sound/isa/sb/emu8000.c @@ -377,12 +377,13 @@ init_arrays(struct snd_emu8000 *emu) static void __devinit size_dram(struct snd_emu8000 *emu) { - int i, size; + int i, size, detected_size; if (emu->dram_checked) return; size = 0; + detected_size = 0; /* write out a magic number */ snd_emu8000_dma_chan(emu, 0, EMU8000_RAM_WRITE); @@ -393,6 +394,8 @@ size_dram(struct snd_emu8000 *emu) while (size < EMU8000_MAX_DRAM) { + size += 512 * 1024; /* increment 512kbytes */ + /* Write a unique data on the test address. * if the address is out of range, the data is written on * 0x200000(=EMU8000_DRAM_OFFSET). Then the id word is @@ -414,7 +417,7 @@ size_dram(struct snd_emu8000 *emu) if (EMU8000_SMLD_READ(emu) != UNIQUE_ID2) break; /* no memory at this address */ - size += 512 * 1024; /* increment 512kbytes */ + detected_size = size; snd_emu8000_read_wait(emu); @@ -442,9 +445,9 @@ size_dram(struct snd_emu8000 *emu) snd_emu8000_dma_chan(emu, 1, EMU8000_RAM_CLOSE); snd_printdd("EMU8000 [0x%lx]: %d Kb on-board memory detected\n", - emu->port1, size/1024); + emu->port1, detected_size/1024); - emu->mem_size = size; + emu->mem_size = detected_size; emu->dram_checked = 1; } -- 1.6.4 ---------------------------------------------------------------------- Sypie, sypie, ale czy sypac bedzie nadal? Sprawdz pogode >> http://link.interia.pl/f2530

2 1

[alsa-devel] [PATCH 2/2] ALSA: hda - support OLPC XO-1.5 DC input
by Daniel Drake 08 Jan '10

08 Jan '10

The XO's audio hardware is wired up to allow DC sensors (e.g. light sensors, thermistors, etc) to be plugged in through the microphone jack. Add sound mixer controls to allow this mode to be enabled and tweaked. Signed-off-by: Daniel Drake <dsd(a)laptop.org> --- sound/pci/hda/patch_conexant.c | 213 +++++++++++++++++++++++++++++++++++----- 1 files changed, 190 insertions(+), 23 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 9ad1219..ffc609e 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -117,6 +117,16 @@ struct conexant_spec { unsigned int recording; void (*capture_prepare)(struct hda_codec *codec); void (*capture_cleanup)(struct hda_codec *codec); + + /* OLPC XO-1.5 supports DC input mode (e.g. for use with analog sensors) + * through the microphone jack. + * When the user enables this through a mixer switch, both internal and + * external microphones are disabled. Gain is fixed at 0dB. In this mode, + * we also allow the bias to be configured through a separate mixer + * control. */ + unsigned int dc_enable; + unsigned int dc_input_bias; /* offset into cxt5066_olpc_dc_bias */ + unsigned int mic_boost; /* offset into cxt5066_analog_mic_boost */ }; static int conexant_playback_pcm_open(struct hda_pcm_stream *hinfo, @@ -1974,6 +1984,26 @@ static int cxt5066_hp_master_sw_put(struct snd_kcontrol *kcontrol, return 1; } +static const struct hda_input_mux cxt5066_olpc_dc_bias = { + .num_items = 3, + .items = { + { "Off", PIN_IN }, + { "50%", PIN_VREF50 }, + { "80%", PIN_VREF80 }, + }, +}; + +static int cxt5066_set_olpc_dc_bias(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + /* Even though port F is the DC input, the bias is controlled on port B. + * we also leave that port as an active input (but unselected) in DC mode + * just in case that is necessary to make the bias setting take effect. */ + return snd_hda_codec_write_cache(codec, 0x1a, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + cxt5066_olpc_dc_bias.items[spec->dc_input_bias].index); +} + /* OLPC defers mic widget control until when capture is started because the * microphone LED comes on as soon as these settings are put in place. if we * did this before recording, it would give the false indication that recording @@ -1984,6 +2014,27 @@ static void cxt5066_olpc_select_mic(struct hda_codec *codec) if (!spec->recording) return; + if (spec->dc_enable) { + /* in DC mode we ignore presence detection and just use the jack + * through our special DC port */ + const struct hda_verb enable_dc_mode[] = { + /* disble internal mic, port C */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* enable DC capture, port F */ + {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {}, + }; + + snd_hda_sequence_write(codec, enable_dc_mode); + /* port B input disabled (and bias set) through the following call */ + cxt5066_set_olpc_dc_bias(codec); + return; + } + + /* disable DC (port F) */ + snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0); + /* external mic, port B */ snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, spec->ext_mic_present ? CXT5066_OLPC_EXT_MIC_BIAS : 0); @@ -1999,6 +2050,9 @@ static void cxt5066_olpc_automic(struct hda_codec *codec) struct conexant_spec *spec = codec->spec; unsigned int present; + if (spec->dc_enable) /* don't do presence detection in DC mode */ + return; + present = snd_hda_codec_read(codec, 0x1a, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; if (present) @@ -2073,13 +2127,16 @@ static void cxt5066_hp_automute(struct hda_codec *codec) /* unsolicited event for jack sensing */ static void cxt5066_olpc_unsol_event(struct hda_codec *codec, unsigned int res) { + struct conexant_spec *spec = codec->spec; snd_printdd("CXT5066: unsol event %x (%x)\n", res, res >> 26); switch (res >> 26) { case CONEXANT_HP_EVENT: cxt5066_hp_automute(codec); break; case CONEXANT_MIC_EVENT: - cxt5066_olpc_automic(codec); + /* ignore mic events in DC mode; we're always using the jack */ + if (!spec->dc_enable) + cxt5066_olpc_automic(codec); break; } } @@ -2109,6 +2166,15 @@ static const struct hda_input_mux cxt5066_analog_mic_boost = { }, }; +static int cxt5066_set_mic_boost(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + return snd_hda_codec_write_cache(codec, 0x17, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | AC_AMP_SET_OUTPUT | + cxt5066_analog_mic_boost.items[spec->mic_boost].index); +} + static int cxt5066_mic_boost_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -2119,15 +2185,8 @@ static int cxt5066_mic_boost_mux_enum_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - int val; - hda_nid_t nid = kcontrol->private_value & 0xff; - int inout = (kcontrol->private_value & 0x100) ? - AC_AMP_GET_INPUT : AC_AMP_GET_OUTPUT; - - val = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_AMP_GAIN_MUTE, inout); - - ucontrol->value.enumerated.item[0] = val & AC_AMP_GAIN; + struct conexant_spec *spec = codec->spec; + ucontrol->value.enumerated.item[0] = spec->mic_boost; return 0; } @@ -2135,23 +2194,101 @@ static int cxt5066_mic_boost_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct conexant_spec *spec = codec->spec; const struct hda_input_mux *imux = &cxt5066_analog_mic_boost; unsigned int idx; - hda_nid_t nid = kcontrol->private_value & 0xff; - int inout = (kcontrol->private_value & 0x100) ? - AC_AMP_SET_INPUT : AC_AMP_SET_OUTPUT; + idx = ucontrol->value.enumerated.item[0]; + if (idx >= imux->num_items) + idx = imux->num_items - 1; + + spec->mic_boost = idx; + if (!spec->dc_enable) + cxt5066_set_mic_boost(codec); + return 1; +} + +static void cxt5066_enable_dc(struct hda_codec *codec) +{ + const struct hda_verb enable_dc_mode[] = { + /* disable gain */ + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + + /* switch to DC input */ + {0x17, AC_VERB_SET_CONNECT_SEL, 3}, + {} + }; + + /* configure as input source */ + snd_hda_sequence_write(codec, enable_dc_mode); + cxt5066_olpc_select_mic(codec); /* also sets configured bias */ +} + +static void cxt5066_disable_dc(struct hda_codec *codec) +{ + /* reconfigure input source */ + cxt5066_set_mic_boost(codec); + /* automic also selects the right mic if we're recording */ + cxt5066_olpc_automic(codec); +} + +static int cxt5066_olpc_dc_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct conexant_spec *spec = codec->spec; + ucontrol->value.integer.value[0] = spec->dc_enable; + return 0; +} - if (!imux->num_items) +static int cxt5066_olpc_dc_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct conexant_spec *spec = codec->spec; + int dc_enable = !!ucontrol->value.integer.value[0]; + + if (dc_enable == spec->dc_enable) return 0; + + spec->dc_enable = dc_enable; + if (dc_enable) + cxt5066_enable_dc(codec); + else + cxt5066_disable_dc(codec); + + return 1; +} + +static int cxt5066_olpc_dc_bias_enum_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + return snd_hda_input_mux_info(&cxt5066_olpc_dc_bias, uinfo); +} + +static int cxt5066_olpc_dc_bias_enum_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct conexant_spec *spec = codec->spec; + ucontrol->value.enumerated.item[0] = spec->dc_input_bias; + return 0; +} + +static int cxt5066_olpc_dc_bias_enum_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct conexant_spec *spec = codec->spec; + const struct hda_input_mux *imux = &cxt5066_analog_mic_boost; + unsigned int idx; + idx = ucontrol->value.enumerated.item[0]; if (idx >= imux->num_items) idx = imux->num_items - 1; - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_AMP_GAIN_MUTE, - AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | inout | - imux->items[idx].index); - + spec->dc_input_bias = idx; + if (spec->dc_enable) + cxt5066_set_olpc_dc_bias(codec); return 1; } @@ -2173,6 +2310,9 @@ static void cxt5066_olpc_capture_cleanup(struct hda_codec *codec) /* disble internal mic, port C */ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* disable DC capture, port F */ + {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, {}, }; @@ -2231,6 +2371,24 @@ static struct snd_kcontrol_new cxt5066_mixer_master_olpc[] = { {} }; +static struct snd_kcontrol_new cxt5066_mixer_olpc_dc[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "DC Mode Enable Switch", + .info = snd_ctl_boolean_mono_info, + .get = cxt5066_olpc_dc_get, + .put = cxt5066_olpc_dc_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "DC Input Bias Enum", + .info = cxt5066_olpc_dc_bias_enum_info, + .get = cxt5066_olpc_dc_bias_enum_get, + .put = cxt5066_olpc_dc_bias_enum_put, + }, + {} +}; + static struct snd_kcontrol_new cxt5066_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -2243,11 +2401,10 @@ static struct snd_kcontrol_new cxt5066_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Ext Mic Boost Capture Enum", + .name = "Analog Mic Boost Capture Enum", .info = cxt5066_mic_boost_mux_enum_info, .get = cxt5066_mic_boost_mux_enum_get, .put = cxt5066_mic_boost_mux_enum_put, - .private_value = 0x17, }, HDA_BIND_VOL("Capture Volume", &cxt5066_bind_capture_vol_others), @@ -2341,7 +2498,7 @@ static struct hda_verb cxt5066_init_verbs_olpc[] = { {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, {0x1d, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ - /* Port F: unused */ + /* Port F: external DC input through microphone port */ {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, /* Port G: internal speakers */ @@ -2462,15 +2619,22 @@ static int cxt5066_init(struct hda_codec *codec) if (spec->dell_vostro) cxt5066_vostro_automic(codec); } + cxt5066_set_mic_boost(codec); return 0; } static int cxt5066_olpc_init(struct hda_codec *codec) { + struct conexant_spec *spec = codec->spec; snd_printdd("CXT5066: init\n"); conexant_init(codec); cxt5066_hp_automute(codec); - cxt5066_olpc_automic(codec); + if (!spec->dc_enable) { + cxt5066_set_mic_boost(codec); + cxt5066_olpc_automic(codec); + } else { + cxt5066_enable_dc(codec); + } return 0; } @@ -2553,8 +2717,10 @@ static int patch_cxt5066(struct hda_codec *codec) codec->patch_ops.unsol_event = cxt5066_olpc_unsol_event; spec->init_verbs[0] = cxt5066_init_verbs_olpc; spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc; + spec->mixers[spec->num_mixers++] = cxt5066_mixer_olpc_dc; spec->mixers[spec->num_mixers++] = cxt5066_mixers; spec->port_d_mode = 0; + spec->mic_boost = 3; /* default 30dB gain */ /* no S/PDIF out */ spec->multiout.dig_out_nid = 0; @@ -2576,6 +2742,7 @@ static int patch_cxt5066(struct hda_codec *codec) spec->mixers[spec->num_mixers++] = cxt5066_vostro_mixers; spec->port_d_mode = 0; spec->dell_vostro = 1; + spec->mic_boost = 3; /* default 30dB gain */ snd_hda_attach_beep_device(codec, 0x13); /* no S/PDIF out */ -- 1.6.2.5

2 1

[alsa-devel] [PATCH 1/2] ALSA: hda - Configure XO-1.5 microphones at capture time
by Daniel Drake 08 Jan '10

08 Jan '10

The XO-1.5 has a microphone LED designed to indicate to the user when something is being recorded. This light is controlled by the microphone bias voltage and it is currently coming on all the time. This patch defers the microphone port configuration until when recording is actually taking place, fixing the behaviour of the LED. Signed-off-by: Daniel Drake <dsd(a)laptop.org> --- sound/pci/hda/patch_conexant.c | 125 +++++++++++++++++++++++++++------------- 1 files changed, 85 insertions(+), 40 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index c578c28..9ad1219 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -111,8 +111,12 @@ struct conexant_spec { unsigned int dell_automute; unsigned int port_d_mode; - unsigned char ext_mic_bias; unsigned int dell_vostro; + + unsigned int ext_mic_present; + unsigned int recording; + void (*capture_prepare)(struct hda_codec *codec); + void (*capture_cleanup)(struct hda_codec *codec); }; static int conexant_playback_pcm_open(struct hda_pcm_stream *hinfo, @@ -185,6 +189,8 @@ static int conexant_capture_pcm_prepare(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct conexant_spec *spec = codec->spec; + if (spec->capture_prepare) + spec->capture_prepare(codec); snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number], stream_tag, 0, format); return 0; @@ -196,6 +202,8 @@ static int conexant_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, { struct conexant_spec *spec = codec->spec; snd_hda_codec_cleanup_stream(codec, spec->adc_nids[substream->number]); + if (spec->capture_cleanup) + spec->capture_cleanup(codec); return 0; } @@ -1966,53 +1974,53 @@ static int cxt5066_hp_master_sw_put(struct snd_kcontrol *kcontrol, return 1; } -/* toggle input of built-in and mic jack appropriately */ -static void cxt5066_automic(struct hda_codec *codec) +/* OLPC defers mic widget control until when capture is started because the + * microphone LED comes on as soon as these settings are put in place. if we + * did this before recording, it would give the false indication that recording + * is happening when it is not. */ +static void cxt5066_olpc_select_mic(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; - struct hda_verb ext_mic_present[] = { - /* enable external mic, port B */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, spec->ext_mic_bias}, - - /* switch to external mic input */ - {0x17, AC_VERB_SET_CONNECT_SEL, 0}, + if (!spec->recording) + return; - /* disable internal mic, port C */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {} - }; - static struct hda_verb ext_mic_absent[] = { - /* enable internal mic, port C */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + /* external mic, port B */ + snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + spec->ext_mic_present ? CXT5066_OLPC_EXT_MIC_BIAS : 0); - /* switch to internal mic input */ - {0x17, AC_VERB_SET_CONNECT_SEL, 1}, + /* internal mic, port C */ + snd_hda_codec_write(codec, 0x1b, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + spec->ext_mic_present ? 0 : PIN_VREF80); +} - /* disable external mic, port B */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {} - }; +/* toggle input of built-in and mic jack appropriately */ +static void cxt5066_olpc_automic(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; unsigned int present; - present = snd_hda_jack_detect(codec, 0x1a); - if (present) { + present = snd_hda_codec_read(codec, 0x1a, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + if (present) snd_printdd("CXT5066: external microphone detected\n"); - snd_hda_sequence_write(codec, ext_mic_present); - } else { + else snd_printdd("CXT5066: external microphone absent\n"); - snd_hda_sequence_write(codec, ext_mic_absent); - } + + snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_CONNECT_SEL, + present ? 0 : 1); + spec->ext_mic_present = !!present; + + cxt5066_olpc_select_mic(codec); } /* toggle input of built-in digital mic and mic jack appropriately */ static void cxt5066_vostro_automic(struct hda_codec *codec) { - struct conexant_spec *spec = codec->spec; unsigned int present; struct hda_verb ext_mic_present[] = { /* enable external mic, port B */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, spec->ext_mic_bias}, + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* switch to external mic input */ {0x17, AC_VERB_SET_CONNECT_SEL, 0}, @@ -2063,7 +2071,7 @@ static void cxt5066_hp_automute(struct hda_codec *codec) } /* unsolicited event for jack sensing */ -static void cxt5066_unsol_event(struct hda_codec *codec, unsigned int res) +static void cxt5066_olpc_unsol_event(struct hda_codec *codec, unsigned int res) { snd_printdd("CXT5066: unsol event %x (%x)\n", res, res >> 26); switch (res >> 26) { @@ -2071,7 +2079,7 @@ static void cxt5066_unsol_event(struct hda_codec *codec, unsigned int res) cxt5066_hp_automute(codec); break; case CONEXANT_MIC_EVENT: - cxt5066_automic(codec); + cxt5066_olpc_automic(codec); break; } } @@ -2147,6 +2155,31 @@ static int cxt5066_mic_boost_mux_enum_put(struct snd_kcontrol *kcontrol, return 1; } +static void cxt5066_olpc_capture_prepare(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + /* mark as recording and configure the microphone widget so that the + * recording LED comes on. */ + spec->recording = 1; + cxt5066_olpc_select_mic(codec); +} + +static void cxt5066_olpc_capture_cleanup(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + const struct hda_verb disable_mics[] = { + /* disable external mic, port B */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* disble internal mic, port C */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {}, + }; + + snd_hda_sequence_write(codec, disable_mics); + spec->recording = 0; +} + static struct hda_input_mux cxt5066_capture_source = { .num_items = 4, .items = { @@ -2296,10 +2329,10 @@ static struct hda_verb cxt5066_init_verbs_olpc[] = { {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ /* Port B: external microphone */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, CXT5066_OLPC_EXT_MIC_BIAS}, + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, /* Port C: internal microphone */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, /* Port D: unused */ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, @@ -2428,12 +2461,19 @@ static int cxt5066_init(struct hda_codec *codec) cxt5066_hp_automute(codec); if (spec->dell_vostro) cxt5066_vostro_automic(codec); - else - cxt5066_automic(codec); } return 0; } +static int cxt5066_olpc_init(struct hda_codec *codec) +{ + snd_printdd("CXT5066: init\n"); + conexant_init(codec); + cxt5066_hp_automute(codec); + cxt5066_olpc_automic(codec); + return 0; +} + enum { CXT5066_LAPTOP, /* Laptops w/ EAPD support */ CXT5066_DELL_LAPTOP, /* Dell Laptop */ @@ -2470,7 +2510,7 @@ static int patch_cxt5066(struct hda_codec *codec) codec->spec = spec; codec->patch_ops = conexant_patch_ops; - codec->patch_ops.init = cxt5066_init; + codec->patch_ops.init = conexant_init; spec->dell_automute = 0; spec->multiout.max_channels = 2; @@ -2483,7 +2523,6 @@ static int patch_cxt5066(struct hda_codec *codec) spec->input_mux = &cxt5066_capture_source; spec->port_d_mode = PIN_HP; - spec->ext_mic_bias = PIN_VREF80; spec->num_init_verbs = 1; spec->init_verbs[0] = cxt5066_init_verbs; @@ -2510,20 +2549,26 @@ static int patch_cxt5066(struct hda_codec *codec) spec->dell_automute = 1; break; case CXT5066_OLPC_XO_1_5: - codec->patch_ops.unsol_event = cxt5066_unsol_event; + codec->patch_ops.init = cxt5066_olpc_init; + codec->patch_ops.unsol_event = cxt5066_olpc_unsol_event; spec->init_verbs[0] = cxt5066_init_verbs_olpc; spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc; spec->mixers[spec->num_mixers++] = cxt5066_mixers; spec->port_d_mode = 0; - spec->ext_mic_bias = CXT5066_OLPC_EXT_MIC_BIAS; /* no S/PDIF out */ spec->multiout.dig_out_nid = 0; /* input source automatically selected */ spec->input_mux = NULL; + + /* our capture hooks which allow us to turn on the microphone LED + * at the right time */ + spec->capture_prepare = cxt5066_olpc_capture_prepare; + spec->capture_cleanup = cxt5066_olpc_capture_cleanup; break; case CXT5066_DELL_VOSTO: + codec->patch_ops.init = cxt5066_init; codec->patch_ops.unsol_event = cxt5066_vostro_event; spec->init_verbs[0] = cxt5066_init_verbs_vostro; spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc; -- 1.6.2.5

2 1

[alsa-devel] [PATCH] ac97: merge WM9703 and WM9705 ops
by Krzysztof Helt 08 Jan '10

08 Jan '10

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> The WM9705 and WM9703 ops are the same actually so use the same code for both. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- I have another problem with the AC97. I have a Creative SB 128 card (model CT4810). It has the ST Micro AC97 codec. There is no ST Micro AC97 codec defined in the ALSA code. The sound plays ok, but reading of certain registers (e.g. whole range from 0x2a to 0x7a) ends with timeout. The codec ID (registers 0x7c and 0x7e) is "STM\04" (0x53544d04). The codec info is 20-bit DAC, 18-bit ADC, no 3D and reading of the tone control register (0x8) end with the timeout. I put photos on the card here: http://picasaweb.google.com/krzysztof.h1/PCISoundCards# Any help with identification is appreciated. Is there any list of codecs used by Creative in the SB 128/64/PCI cards? I suppose there were only few used. Regards, Krzysztof sound/pci/ac97/ac97_patch.c | 17 +---------------- 1 files changed, 1 insertions(+), 16 deletions(-) diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 139cf3b..e288a55 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -544,25 +544,10 @@ static int patch_wolfson04(struct snd_ac97 * ac97) return 0; } -static int patch_wolfson_wm9705_specific(struct snd_ac97 * ac97) -{ - int err, i; - for (i = 0; i < ARRAY_SIZE(wm97xx_snd_ac97_controls); i++) { - if ((err = snd_ctl_add(ac97->bus->card, snd_ac97_cnew(&wm97xx_snd_ac97_controls[i], ac97))) < 0) - return err; - } - snd_ac97_write_cache(ac97, 0x72, 0x0808); - return 0; -} - -static struct snd_ac97_build_ops patch_wolfson_wm9705_ops = { - .build_specific = patch_wolfson_wm9705_specific, -}; - static int patch_wolfson05(struct snd_ac97 * ac97) { /* WM9705, WM9710 */ - ac97->build_ops = &patch_wolfson_wm9705_ops; + ac97->build_ops = &patch_wolfson_wm9703_ops; #ifdef CONFIG_TOUCHSCREEN_WM9705 /* WM9705 touchscreen uses AUX and VIDEO for touch */ ac97->flags |= AC97_HAS_NO_VIDEO | AC97_HAS_NO_AUX; -- 1.6.4 ---------------------------------------------------------------------- Co Ci sie dzisiaj snilo? Czy wiesz, co oznacza Twoj sen? Sprawdz w senniku >> http://link.interia.pl/f2532

2 1

[alsa-devel] ALSA Multichannel through HDMI
by John Ettedgui 08 Jan '10

08 Jan '10

Hello, (since I did not get any feedback on the user ml, I'm trying it here, sorry...) I am using the HDMI output of my radeonhd 4670 to get sound, but I cannot get ALSA to work with (5.1) multichannel, it only outputs to the front left and right speakers (checked by speaker-test). I tried following this howto: http://www.cse.ohio-state.edu/~bondhugu/alsamch.shtml but it did not do much for me... and whenever I tried to add slave.channels 6, I just could not use alsa anymore, it always complained about this line... I tried channels 6 as part of the slave.pcm{} too and got the same result... What should I look into? I tried with 2.6.32 and alsa-lib 1.0.21.a, and then 2.6.33-rc3 (stil using alsa-driver 1.0.21 it seems) and alsa-lib 1.0.22 and no success. I am using the radeonhd xorg driver to get HDMI audio. I compiled alsa-driver 1.0.22.1 on top of a 2.6.32.2 kernel and the result is still the same although this time I did not play with the asound.conf file. Pulse audio on top of alsa does give me a 5.0 setup or at least I can hear that from the speaker-test. Thank you, John

1 0

[alsa-devel] [PATCH, beta] Echoaudio, add suspend/resume support
by Giuliano Pochini 08 Jan '10

08 Jan '10

The following patched add suspend/resume support to all echoaudio drivers. They seem to work fine, but more testing is needed. Some testing have been performed on an IndigoIOx and a Layla3G (thanks to Mark Hills). The most important part which has not been tested is rawmidi. Is someone willing to try ? The following patches are against alsa-driver-1.0.22.1, but should be possible to apply them cleanly on other versions as well because no changes have been done recently. 1/4 Add firmware cache #1: When CONFIG_PM is enabled the firmware cannot be released because the driver may need it again to resume the card. With this patch the firmware is passed as an index of the struct firmware card_fw[] in place of a pointer. That same index is then uses to locate the firmware in the firmware cache. 2/4 Add firmware cache #2: This patch changes get_firmware(), free_firmware() and adds free_firmware_cache(). The first two functions implement a very simple cache and the latter is used to actually release all the stored firmwares when the module is unloaded. When CONFIG_PM is not enabled those functions act as before and free_firmware_cache() is a nop. 3/4 Add suspend support: This patch adds snd_echo_suspend() and snd_echo_resume() callbacks and suspend/resume trigger event handlers. It also makes hardware initialization and mixer setup two clearly distinct stages, so they can be used also in the resume procedure without messing the settings up. As a bonus all init_hw() functions (one for each card) are a little shorter and cleaner. 4/4 patch the patch: Updates pci/echoaudio/echoaudio.patch The patches are here, in the same order as above: http://www.webalice.it/g_pochini/ead/fw_cache_1.patch http://www.webalice.it/g_pochini/ead/fw_cache_2.patch http://www.webalice.it/g_pochini/ead/pm_suspend_10.patch http://www.webalice.it/g_pochini/ead/echoaudio.patch.patch Or if you prefer a single patch (54KB): http://www.webalice.it/g_pochini/ead/all_echoaudio_suspend.patch Please test and report, especially the rawmidi part. TIA. -- Giuliano.

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[alsa-devel] [PATCH 0/2] Add Voice Codec support for DaVinci platforms
by Miguel Aguilar 08 Jan '10

08 Jan '10

Hi, The following patch series adds support for the Voice Codec in the Davinci platform and in this particular case for the DM365 EVM. The second patch of the series configures the DM365 EVM sound card with two audi subdevices the AIC3x codec and the voice codec. The voice codec driver was tested on the DM365 EVM, however when the AIC and Voice Codec are defined as subdevices of the sound card doesn't work properly, so please check the file sound/soc/davinci/davinci-evm.c. Thanks, -- Miguel Angel Aguilar Ulloa Embedded Software Engineer RidgeRun Embedded Solutions miguel.aguilar(a)ridgerun.com

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[alsa-devel] issue with sound stopping and not being able to restart
by Andrew Kohlsmith (Mailing List Account) 07 Jan '10

07 Jan '10

Good afternoon, all, I am new to ALSA development, but experienced in C. I wrote up a little test application which simply opens one playback and one capture device and spits noise into the playback and recording it on the capture device. I cross- correlate the data streams to determine the "depth" or latency of the pipe, end-to-end. I'm using playback and capture callbacks to handle the audio. The program seems to work reasonably well, but at seemingly random times either the playback or capture stream will "hiccup" (xrun) -- I am handling this case, but it's fairly often that the stream does NOT resume and I'm stuck with either no playback or no capture. I've looked at the example programs and the documentation and I do not see what it is I'm doing wrong. I'm posting my code here, hopefully someone can show me the error of my way. Capture/playback setup (error checking removed): snd_pcm_open(handle, fname, stream, SND_PCM_NONBLOCK); snd_pcm_hw_params_malloc(hwptr); snd_pcm_hw_params_any(h, hwp); snd_pcm_hw_params_set_access(h, hwp, SND_PCM_ACCESS_RW_INTERLEAVED); snd_pcm_hw_params_set_format(h, hwp, SND_PCM_FORMAT_S16_LE); snd_pcm_hw_params_set_rate(h, hwp, 8000, 0); snd_pcm_hw_params_set_channels(h, hwp, 2); snd_pcm_hw_params_set_buffer_size(h, hwp, BUF_SIZE); snd_pcm_hw_params(h, hwp); At this point I have the capture or playback device set up for 16 bit signed linear data in interleaved format and two channels. I've set up the buffers to be 160 frames, or 20ms. Each of these calls is checked for errors, and I do verify that the rate, buffer size and buffer time are what I expect. There is nothing unusual going on with these calls that I can determine. I set up the playback and capture callbacks thusly (again checking for errors) snd_async_add_pcm_handler(&ahp, hp, playback_callback, &pb_data); snd_async_add_pcm_handler(&ahc, hc, capture_callback, &cap_data); At this point I believe I'm ready to go, all I have to do is start playback and capture: if((fcount = snd_pcm_writei(h[PLAYBACK], pb_data.buf, BUF_SIZE)) != BUF_SIZE) { if(fcount < 0) { if((e = snd_pcm_recover(h[PLAYBACK], fcount, 0)) < 0) fprintf(stderr, "initial: can't writei: %s\n", snd_strerror(e)); } else { fprintf(stderr, "initial: short write: %d instead of %d\n", fcount, BUF_SIZE); } } if((e = snd_pcm_start(h[CAPTURE])) < 0) { fprintf(stderr, "capture prepare error: %s\n", snd_strerror(e)); return 1; } Now the playback and capture handlers are called by ALSA as required. The capture handler looks like this: /* capture callback. Called by ALSA whenever a period time is up */ void capture_callback(snd_async_handler_t *ah) { snd_pcm_t *h = snd_async_handler_get_pcm(ah); struct async_data *d = snd_async_handler_get_callback_private(ah); snd_pcm_sframes_t fa; short *p; int i, fc, e, faloops; struct timeval tv; static int c=0; /* read in whatever is available */ fa = snd_pcm_avail_update(h); faloops = 0; while(fa > 0) { if(fa > BUF_SIZE) fa = BUF_SIZE; if((fc = snd_pcm_readi(h, d->buf, fa)) != fa) { if(fc < 0) { if((e = snd_pcm_recover(h, e, 0)) < 0) { fprintf(stderr, "callback: can't readi: %s\n", snd_strerror(e)); } fc = 0; } else { fprintf(stderr, "callback: short read: %d instead of %ld\n", fc, fa); } } for(i=0,p=&d->buf[0]; i<fc; i++) { *d->ptr++ = *p++; p++; if(d->ptr > &in[MAX_BUFS * (BUF_SIZE - 1)]) { d->ptr = (short *)&in[0]; } } c += fc; if(c >= BUF_SIZE) { gettimeofday(&tv, NULL); c -= BUF_SIZE; in_step++; if(in_step >= MAX_BUFS) in_step = 0; } frames_cap += fc; fa = snd_pcm_avail_update(h); faloops++; }; cap_faloops=faloops; } The playback callback is very similar: /* playback callback. Called by ALSA whenever a period time is up */ void playback_callback(snd_async_handler_t *ah) { snd_pcm_t *h = snd_async_handler_get_pcm(ah); struct async_data *d = snd_async_handler_get_callback_private(ah); snd_pcm_sframes_t fa; int i, fc, e, faloops; static int c=0; struct timeval tv; fa = snd_pcm_avail_update(h); faloops = 0; while(fa > 0) { if(fa > BUF_SIZE) fa = BUF_SIZE; for(i=0; i<fa; i++) { d->buf[i*2] = awgn(&noise_source); d->buf[i*2+1] = awgn(&noise_source); } if((fc = snd_pcm_writei(h, d->buf, fa)) != fa) { if(fc < 0) { if((e = snd_pcm_recover(h, e, 0)) < 0) { fprintf(stderr, "callback: can't writei: error %x %s\n", e, snd_strerror(e)); } fc = 0; } else { fprintf(stderr, "callback: short write: %d instead of %ld\n", fc, fa); } } for(i=0; i<fc; i++) { *d->ptr++ = d->buf[i*2]; if(d->ptr > &out[MAX_BUFS * (BUF_SIZE - 1)]) { d->ptr = (short *)&out[0]; } } c += fc; if(c >= BUF_SIZE) { gettimeofday(&tv, NULL); c -= BUF_SIZE; out_step++; if(out_step >= MAX_BUFS) out_step = 0; } frames_pb += fc; fa = snd_pcm_avail_update(h); faloops++; }; pb_faloops = faloops; } That's really it; the main loop keeps track of the in_step and out_step variables and does the cross-correlation as the data comes in. Now it seems that it's the capture playback that is getting xruns, but I when capture xruns occur my playback callback seems to stop, and sometimes the entire program just hangs as if it is waiting for ALSA to call one of the callbacks. Can anyone shed any light on this? I'm certain it's something very simple and braindead, but I can't find it. Regards, Andrew

1 0

Re: [alsa-devel] playback delay and problems with simultaneous playback/capture
by Max Klein 07 Jan '10

07 Jan '10

concerning the first point, I did some measurements... the file I used has a length of 1488 ms trigger start jiffies=7466937 trigger stop jiffies=7468172 difference = 1235 (1235 ms) pcm prepare jiffies 2=6761125 trigger start jiffies=6761126 trigger stop jiffies=6762361 difference = 1235 (1235 ms) trigger stopped, buffer empty, stop decoder jiffies=6762684 difference = 323 (323 ms) so I was wondering, why the difference between trigger start - stop is not 1488ms but 1235ms. I still guess that it may is a wrongly updated pcm pointer? Thanks for your help, Max -- Jetzt kostenlos herunterladen: Internet Explorer 8 und Mozilla Firefox 3.5 - sicherer, schneller und einfacher! http://portal.gmx.net/de/go/atbrowser

1 0

[alsa-devel] question on dapm_reg_event()
by Haojian Zhuang 06 Jan '10

06 Jan '10

Hi all, I just have a question on dapm code. In dapm_reg_event(), -(w->reg + 1) is input parameter of snd_soc_update_bits(). Why we need to use negative value as input parameter at here? Why do we not use w->reg at here? Thanks Haojian

3 3

[alsa-devel] playback delay and problems with simultaneous playback/capture
by mein_name＠ist-einmalig.de 06 Jan '10

06 Jan '10

I am currently testing my driver but I've got some problems with the "pcm machine"...as it is my first ALSA driver I am open for every hint. The code currently is not conform to linux kernel coding style, I will do all such cosmetically stuff after functionality is given. 1. Playing a file works fine, but when the ALSA middle layer triggers stop command, the DSP still has data to play, so sending a stop signal to the DSP at this moment would result in breaking the playback. The strange thing here is, that it seems there are 6 frames missing, but the buffer of the DSP is only about 3 frames long. I am not sure if the problem here is, that I am not correctly "calculating" the pcm_pointer: static snd_pcm_uframes_t snd_card_mychip_pcm_pointer(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_mychip_pcm *dpcm = runtime->private_data; /* get the current hardware pointer */ // TODO: calc correct return value print_log(DEBUGC "pcm pointer: ret b=%lu\n", bytes_to_frames(runtime, dpcm->pcm_buf_pos)); return bytes_to_frames(runtime, dpcm->pcm_buf_pos); } 2. when playing and capturing data simultaneously, I get troubles in both. Playback or Capture alone is working, but if both come togheter, troubles occur. The data-transfer from ALSA to DSP and DSP to ALSA I am doing in the timer function, I am not sure if this could be the problem...but to send data to the DSP I have to check a flag if the DSP is able to receive data, if so, I can send data, otherwise I have to wait until DSP is ready to receive data. That is done with dsp_sendstream. When capturing, the DSP is sending an interrupt every captured period that is available, so I store that in a buffer from where it got read frequently via the timer function (done by readbuf)...everytime when going through this function I think that this way may work, as it does normally, but I am absolutely not sure if this is a "right" way. I know that there are so called "copy" and "silence" callbacks, but using the copy callback for playback I lost frames, so I did it like it is now... static void snd_card_mychip_pcm_timer_function(unsigned long data) { struct snd_mychip_pcm *dpcm = (struct snd_mychip_pcm *)data; unsigned long flags; int count, pos, status; // disable interrupts spin_lock_irqsave(&dpcm->lock, flags); // restart the timer to be called again after one jiffie = 1 ms dpcm->timer.expires = 1 + jiffies; add_timer(&dpcm->timer); if (dpcm->substream->stream == 0) { // playback stream // increase counters by bytes per jiffie if (dpcm->sendingnotpossible == 0) { dpcm->pcm_irq_pos += dpcm->pcm_jiffie; //0..MAX_PERIOD_SIZE step=jiffie dpcm->pcm_buf_pos += dpcm->pcm_jiffie; //0..MAX_BUFFER_SIZE step=jiffie } // if pcm_count has been reached, call snd_pcm_period_elapsed // to signal the end of one period (e.g. 288 for mpg or 768 for wav) if (dpcm->pcm_irq_pos >= dpcm->pcm_count) { // >= MAX_PERIOD_SIZE? count=dpcm->pcm_count; // number of bytes to be sent pos=dpcm->pcm_buf_pos-count; // start offset in DMA buffer // try to send data to dsp status = dsp_sendstream(dpcm->substream->pcm->device, dpcm->substream->runtime->dma_area+pos, count); if (status != 0) { // sending not possible dpcm->sendingnotpossible=1; } else { // sending successful dpcm->sendingnotpossible=0; dpcm->sentframes++; if (dpcm->sentframes == 3) // 3 frames in buffer so we can start playback { // start decoder playback status=dsp_decod_action(dpcm->substream->pcm->device, DECOD_START); print_log(DEBUGA "reply of dsp_decod_action = %i\n",status); } // wrap counter if end of buffer is reached dpcm->pcm_buf_pos %= dpcm->pcm_size; // wrap this counter too dpcm->pcm_irq_pos %= dpcm->pcm_count; // tell the PCM middle layer when the buffer position goes across the prescribed period size snd_pcm_period_elapsed(dpcm->substream); print_log(DEBUGC "timer_function: Period elapsed (p) - pcm_count=%i - pcm_size=%i - src=%p - jiffie=%i\n",dpcm->pcm_count,dpcm->pcm_size,dpcm->substream->runtime->dma_area,dpcm->pcm_jiffie); } // enable interrupts spin_unlock_irqrestore(&dpcm->lock, flags); } else { // wrap counter if end of buffer is reached dpcm->pcm_buf_pos %= dpcm->pcm_size; // enable interrupts spin_unlock_irqrestore(&dpcm->lock, flags); } } else if (dpcm->substream->stream == 1) { // capture stream count = dpcm->pcm_count; // number of bytes to be captured pos = dpcm->pcm_buf_pos; // start offset in DMA buffer // read from HIF buffer status = readbuf(dpcm->substream->pcm->device, dpcm->substream->runtime->dma_area+pos, count); if (status == 0) { // we got new data from HIF buffer => increase pointers.. dpcm->pcm_buf_pos += count; // wrap counter if end of buffer is reached dpcm->pcm_buf_pos %= dpcm->pcm_size; // wrap this counter too //dpcm->pcm_irq_pos %= dpcm->pcm_count; // tell the PCM middle layer when the buffer position goes across the prescribed period size snd_pcm_period_elapsed(dpcm->substream); print_log(DEBUG "timer_function: Period elapsed (c) - pcm_count=%i - pcm_size=%i - src=%p - jiffie=%i\n",dpcm->pcm_count,dpcm->pcm_size,dpcm->substream->runtime->dma_area,dpcm->pcm_jiffie); } // enable interrupts spin_unlock_irqrestore(&dpcm->lock, flags); } } 3. Sometimes I am getting underruns with aplay or xruns with arecord. Currently I am not quite sure why they occur, but I guess this has to do with some of above problems? -bash-3.2# aplay 20575_24k_100.wav Playing WAVE '20575_24k_100.wav ' : Signed 16 bit Little Endian, Rate 24000 Hz, Mono underrun!!! (at least 8.019 ms long) As a reference I used the dummy driver and the tutorial Writing an ALSA Driver. If there are some other drivers that may be similar and I can look at, I am thankful for every hint! So far thanks for helping me! Greetings, Max -- Jetzt kostenlos herunterladen: Internet Explorer 8 und Mozilla Firefox 3.5 - sicherer, schneller und einfacher! http://portal.gmx.net/de/go/chbrowser

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[alsa-devel] Sound still not working after 1.0.22.1 update
by Daniel Grandja 06 Jan '10

06 Jan '10

I have managed to Download / Compile / Install the upgrade, but still no sound, and now i can't log in to Gnome (Xfce still works) http://www.alsa-project.org/db/?f=f31c932fe24d1ad6a8d67189eb2e984f03545be8 i have a VIA VT82xx based onboard card any ideas as to what has gone wrong ? from the log i noticed that the driver version was still 1.0.20 is this correct ? i wonder why that didn't update ? can i force it somehow ? cheers Daniel

2 1

Re: [alsa-devel] capturing data from the microphone
by Mark Brown 06 Jan '10

06 Jan '10

On Wed, Jan 06, 2010 at 11:44:04AM +0100, Marc Garnier wrote: > OK, well, I work on a custom device platform based on an Atmel > at91sam9261. I wrote an alsa driver composed of 2 files CCing in the list and Sedji, who wrote the driver. Please also try avoid top posting - Linux lists generally intersperse replies with the relevant context from the original message since it makes mails much more legible. > But when I want to record a pcm stream I have this error: > # arecord -v -c 1 -t wav -f S16_LE -r 8000 -d 10 input.wav > arecord: pcm_read:1629: read error: Input/output error [With an AT91 in slave mode...] > and actually no interrupt occur and I > wonder why... Can you see clocks on the audio interface? Have you checked that the pins on the processor are configured to route the signals to the correct IP block?

2 1

[alsa-devel] ASoC updates for 2.6.33
by Mark Brown 05 Jan '10

05 Jan '10

The following changes since commit c5974b835a909ff15c3b7e6cf6789b5eb919f419: Linus Torvalds (1): Merge branch 'limits_cleanup' of git://decibel.fi.muni.cz/~xslaby/linux are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.33 Takashi is on holiday for the first few weeks of the year so he said to send direct pull requests in the interim. The patch from Morimoto-san is finishing off a cross-tree merge while the others are driver-specific fixes. Guennadi Liakhovetski (1): ASoC: fix params_rate() macro use in several codecs Kuninori Morimoto (1): ASoC: fsi-ak4642: Remove ak4642_add_i2c_device Manuel Lauss (1): ASoC: fixup oops in generic AC97 codec glue Mark Brown (1): Merge commit 'v2.6.33-rc2' into for-2.6.33 sound/soc/codecs/ac97.c | 6 ++++++ sound/soc/codecs/wm8510.c | 14 +++++++------- sound/soc/codecs/wm8940.c | 14 +++++++------- sound/soc/codecs/wm8974.c | 14 +++++++------- sound/soc/sh/fsi-ak4642.c | 30 ------------------------------ 5 files changed, 27 insertions(+), 51 deletions(-)

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[alsa-devel] [PATCH] ASoc: tpa6130a2: Remove unnecessary variable
by Ilkka Koskinen 04 Jan '10

04 Jan '10

Signed-off-by: Ilkka Koskinen <ilkka.koskinen(a)nokia.com> --- sound/soc/codecs/tpa6130a2.c | 4 ---- 1 files changed, 0 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 0eb33d4..8e98ccf 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -267,12 +267,8 @@ static const struct snd_kcontrol_new tpa6130a2_controls[] = { */ static void tpa6130a2_channel_enable(u8 channel, int enable) { - struct tpa6130a2_data *data; u8 val; - BUG_ON(tpa6130a2_client == NULL); - data = i2c_get_clientdata(tpa6130a2_client); - if (enable) { /* Enable channel */ /* Enable amplifier */ -- 1.6.0.4

3 2

[alsa-devel] [PATCH] ASoC: fixup oops in generic AC97 codec glue
by Manuel Lauss 04 Jan '10

04 Jan '10

Initialize the glue by calling snd_soc_new_ac97_codec() as is done in other ASoC AC97 codecs. Fixes an oops caused by dereferencing uninitialized members in snd_soc_new_pcms(). Run-tested on Au1250. Signed-off-by: Manuel Lauss <manuel.lauss(a)gmail.com> --- Fixes the oops I reported here: http://mailman.alsa-project.org/pipermail/alsa-devel/2009-December/023995.h… sound/soc/codecs/ac97.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 69bd0ac..a1bbe16 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -102,6 +102,12 @@ static int ac97_soc_probe(struct platform_device *pdev) INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); + ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); + if (ret < 0) { + printk(KERN_ERR "ASoC: failed to init gen ac97 glue\n"); + goto err; + } + /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) -- 1.6.6.rc3

2 1

[alsa-devel] Underrun problem and some basic queries
by ANISH KUMAR 04 Jan '10

04 Jan '10

1 0

[alsa-devel] [PATCH 0/5] ASoC: tlv320dac33: Adding new FIFO mode
by Peter Ujfalusi 04 Jan '10

04 Jan '10

Hello, the following series adds new FIFO operating mode to the tlv320dac33 codec driver. To be able to add new FIFO mode the code has to be restructured in order to keep it readable after the new mode is added. Since both of the support FIFO mode in the driver needs the codec to be the master, an additional check is added to avoid misconfiguration. --- Peter Ujfalusi (5): ASoC: tlv320dac33: Change nsample switch to FIFO mode enum ASoC: tlv320dac33: Introduce prefill and playback state handlers ASoC: tlv320dac33: Clean up the hardware configuration code ASoC: tlv320dac33: Add new FIFO mode: mode 7 ASoC: tlv320dac33: Safety check for codec slave mode sound/soc/codecs/tlv320dac33.c | 190 ++++++++++++++++++++++++++++++++-------- 1 files changed, 153 insertions(+), 37 deletions(-)

2 8

[alsa-devel] Underrun problem and some basic queries
by ANISH KUMAR 04 Jan '10

04 Jan '10

1 0

[alsa-devel] Echoaudio, suspend/resume question on rawmidi device
by Giuliano Pochini 03 Jan '10

03 Jan '10

I have some patches that add PM support to all echoaudio drivers. It seems to work fine, but it has been tested on two cards only and the rawmidi device hasn't been tested at all. I can't test it because suspend-to-anything does not work on my pmac. I want to be sure at least it cannot deadlock. * On suspend: Rawmidi input is interrupt-driven. The driver tells the card to stop accepting data. Rawmidi output is timer-driven. The driver uses a timer to check if the card's output buffer has room for more data to send. Do del_timer_sync() and wait until all buffered data is sent, which may take 20ms (64 bytes at 31250bps). * On resume: Input: Re-enable midi in. Output: Restart the timer. Do you see any race condition or weak points in this scheme, especially in the midi-out part ? Is del_timer_sync() safe ? -- Giuliano.

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[alsa-devel] Cross-compile problem with alsa-driver version 1.0.22
by Jie Jiang 03 Jan '10

03 Jan '10

Hi all, Recently I downloaded alsa-driver 1.0.22 and tried to build for i.MX27 platform. The "INSTALL" file enclosed in this release requires "--with-cross=arm-linux-" configuration option to cross-compile alsa driver for this ARM platform. Here is my build process #./configure --with-cross=arm-linux- --with-kernel=/home/working/bsp/Kernel/linux-2.6.19.2 --with-build=/home/working/bsp/ Kernel/linux-2.6.19.2 --with-moddir=/home/working/bsp/tmp --with-debug=full --enable-dynamic-minors --with-cards=all checking for gcc... arm-linux-gcc checking for C compiler default output file name... a.out checking whether the C compiler works... configure: error: cannot run C compiled programs. If you meant to cross compile, use `--host'. See `config.log' for more details. It seems that "--host=arm-linux" should be provided to configure script. However, when given the --host option, in the generated "Makefile.conf", I saw the following outputs, which are very confusing: NEW_KBUILD = mainsrcdir = . MAINSRCDIR = /work/63/alsa/alsa-driver-1.0.22 CROSS_COMPILE ?= arm-linux- ARCH ?= ifdef NEW_KBUILD CC = $(CROSS_COMPILE)arm-linux-gcc CPP = $(CROSS_COMPILE)arm-linux-gcc -E export CROSS_COMPILE ARCH else AS = $(CROSS_COMPILE)as LD = $(CROSS_COMPILE)ld CC = $(CROSS_COMPILE)arm-linux-gcc CPP = $(CROSS_COMPILE)arm-linux-gcc -E AR = $(CROSS_COMPILE)ar MODFLAGS = -DMODULE=1 CFLAGS_KERNEL = GENKSYMS = /sbin/genksyms -k 0.0.0 endif Obviously, the CC and CPP variables are not expected as "arm-linux-gcc". (There is one more $(CROSS_COMPILE)!) I overcome this problem by providing CC and CROSS_COMPILE variables with configure script: CC=gcc CROSS_COMPILE=arm-linux-gcc ./configure --with-cross=arm-linux- --with-kernel=/home/working/bsp/Kernel/linux-2.6.19.2 --with-build=/home/working/bsp/Kernel/linux-2.6.19.2 --with-moddir=/home/working/bsp/tmp --with-debug=full --enable-dynamic-minors --with-cards=all Then things become OK. What's wrong with the "INSTALL" file? Another problem. Even if "--with-cards=all" is given, codes in soc/codecs, soc/imx, and many other subdirs are not built. It seems that the configure script sets some variables according to the kernel version and target platform and outputs into "toplevel.config". Is it possible to manually edit "toplevel.config" file (after configure process) to get soc/codecs, soc/imx codes compiled? Regards, Jie

2 1

[alsa-devel] Trouble with sound over HDMI with Nvidia MCP7A HDMI
by Austin Robertson 03 Jan '10

03 Jan '10

I am having trouble getting sound over hdmi with my Nvidia MCP7A HDMI. The strange thing is that everything works fine in a Live disc, but once I do a fresh install of that Live disc I get no sound, even though my configuration is exactly the same (as far as I can tell). I've checked the little things like alsamixer volumes, aplay -l, etc etc. Everything looks good to me. I can reproduce on multiple TVs so its not the receiver. Video drivers are the same in both live version and installed. aplay -l detects my device. sound/pci/hda/patch_nvhdmi.c already has my correct vendor id in it. Perhaps the following info can help figure out what my installed environment is lacking. I've been at it for days now and I'm stumped! Any ideas? *Live Disc Info* XMBC Live 9.11 (basically a optimized version of Karmic configured to start XBMC, or Xbox Media Center, see http://www.xbmc.org for more info) NOTE: My problem is not specific to the XBMC program. Things like aplay and speaker-test don't work either. *Hardware* Acer Aspire Revo R1600 http://www.newegg.com/Product/Product.aspx?Item=N82E16883103228&Tpk=AR1600 *XBMC Live installed to hard disk via Ubuntu installer* (This setup gives me no sound over HDMI) (This is a fresh install. No additional changes were made) alsa-info.sh http://www.alsa-project.org/db/?f=ab7cef7c7233f8c42756d36e362df684b9147d38 xbmc@XBMCLive:~$ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: NVidia [HDA NVidia], device 0: ALC662 rev1 Analog [ALC662 rev1 Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 card 0: NVidia [HDA NVidia], device 3: NVIDIA HDMI [NVIDIA HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 xbmc@XBMCLive:~$ aplay -L default:CARD=NVidia HDA NVidia, ALC662 rev1 Analog Default Audio Device front:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog Front speakers surround40:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers hdmi:CARD=NVidia,DEV=0 HDA NVidia, NVIDIA HDMI HDMI Audio Output null Discard all samples (playback) or generate zero samples (capture) * ### speaker-test executes but NO SOUND over hdmi*. *I've also tried other devices like -D hdmi:0 -D hw:0,3 -D hdmi -D plug:hdmi, etc, etc* xbmc@XBMCLive:~$ speaker-test -D plughw:0,3 -c2 speaker-test 1.0.20 Playback device is plughw:0,3 Stream parameters are 48000Hz, S16_LE, 2 channels Using 16 octaves of pink noise Rate set to 48000Hz (requested 48000Hz) Buffer size range from 64 to 16384 Period size range from 32 to 8192 Using max buffer size 16384 Periods = 4 was set period_size = 4096 was set buffer_size = 16384 0 - Front Left 1 - Front Right Time per period = 5.637709 0 - Front Left xbmc@XBMCLive:~$ cat /proc/driver/nvidia/version NVRM version: NVIDIA UNIX x86 Kernel Module 190.53 Tue Dec 8 18:51:41 PST 2009 GCC version: gcc version 4.4.1 (Ubuntu 4.4.1-4ubuntu8) * XMBC Live Disc * ** (This setup gives me audio over hdmi perfectly) (This is running as a Live disc, ie not installed) ** alsa-info.sh http://www.alsa-project.org/db/?f=c95bebdc433f15147fb0be415be4e61e4d0b72ad [ ^<http://www.alsa-project.org/db/?f=c95bebdc433f15147fb0be415be4e61e4d0b72ad> ] xbmc@XBMCLive:~$ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: NVidia [HDA NVidia], device 0: ALC662 rev1 Analog [ALC662 rev1 Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 card 0: NVidia [HDA NVidia], device 3: NVIDIA HDMI [NVIDIA HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 xbmc@XBMCLive:~$ aplay -L default:CARD=NVidia HDA NVidia, ALC662 rev1 Analog Default Audio Device front:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog Front speakers surround40:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers hdmi:CARD=NVidia,DEV=0 HDA NVidia, NVIDIA HDMI HDMI Audio Output null Discard all samples (playback) or generate zero samples (capture) ###* speaker-test plays audio over hdmi perfectly. other programs play audio over hdmi.* xbmc@XBMCLive:~$ speaker-test -D plughw:0,3 -c2 speaker-test 1.0.20 Playback device is plughw:0,3 Stream parameters are 48000Hz, S16_LE, 2 channels Using 16 octaves of pink noise Rate set to 48000Hz (requested 48000Hz) Buffer size range from 64 to 16384 Period size range from 32 to 8192 Using max buffer size 16384 Periods = 4 was set period_size = 4096 was set buffer_size = 16384 0 - Front Left 1 - Front Right Time per period = 5.638693 xbmc@XBMCLive:~$ cat /proc/driver/nvidia/version NVRM version: NVIDIA UNIX x86 Kernel Module 190.53 Tue Dec 8 18:51:41 PST 2009 GCC version: gcc version 4.4.1 (Ubuntu 4.4.1-4ubuntu8) *Additional Information * I had an idea to run a diff on the two alsa-info.sh outputs of my installed and live version. (non-working and working, see above) I figured that might give me a clue on what I was missing. Most things should be identical with a few exceptions like timestamps, etc. MacBook-Austin:Desktop Austin$ diff -crB alsa-info_installed.txt alsa-info_live.txt > alsa-infodiff.diff alsa-info_installed.txt: http://pastebin.com/f63afa045 (non-working, same as above) alsa-info_live.txt: http://pastebin.com/f7b47d46d (working, same as above) alsa-infodiff.diff: http://pastebin.com/f79cf9c15 Only a few things caught my attention. At first I thought I had my playback volumes muted (line 525,532) according to the diff. But then I realized that was just for the mic, so that shouldn't matter. The only other thing that I saw was that my Amp-Out vals differed (line 123, 129). Perhaps, a more trained eye will have better luck. Bug has also been reported here: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4865

1 0

[alsa-devel] Cross-compile problem with alsa-driver version 1.0.22
by Jie Jiang 03 Jan '10

03 Jan '10

Hi all, Recently I downloaded alsa-driver 1.0.22 and tried to build for i.MX27 platform. The "INSTALL" file enclosed in this release requires "--with-cross=arm-linux-" configuration option to cross-compile alsa driver for this ARM platform. Here is my build process #./configure --with-cross=arm-linux- --with-kernel=/home/working/bsp/Kernel/linux-2.6.19.2 --with-build=/home/working/bsp/ Kernel/linux-2.6.19.2 --with-moddir=/home/working/bsp/tmp --with-debug=full --enable-dynamic-minors --with-cards=all checking for gcc... arm-linux-gcc checking for C compiler default output file name... a.out checking whether the C compiler works... configure: error: cannot run C compiled programs. If you meant to cross compile, use `--host'. See `config.log' for more details. It seems that "--host=arm-linux" should be provided to configure script. However, when given the --host option, in the generated "Makefile.conf", I saw the following outputs, which are very confusing: NEW_KBUILD = mainsrcdir = . MAINSRCDIR = /work/63/alsa/alsa-driver-1.0.22 CROSS_COMPILE ?= arm-linux- ARCH ?= ifdef NEW_KBUILD CC = $(CROSS_COMPILE)arm-linux-gcc CPP = $(CROSS_COMPILE)arm-linux-gcc -E export CROSS_COMPILE ARCH else AS = $(CROSS_COMPILE)as LD = $(CROSS_COMPILE)ld CC = $(CROSS_COMPILE)arm-linux-gcc CPP = $(CROSS_COMPILE)arm-linux-gcc -E AR = $(CROSS_COMPILE)ar MODFLAGS = -DMODULE=1 CFLAGS_KERNEL = GENKSYMS = /sbin/genksyms -k 0.0.0 endif Obviously, the CC and CPP variables are not expected as "arm-linux-gcc". (There is one more $(CROSS_COMPILE)!) I overcome this problem by providing CC and CROSS_COMPILE variables with configure script: CC=gcc CROSS_COMPILE=arm-linux-gcc ./configure --with-cross=arm-linux- --with-kernel=/home/working/bsp/Kernel/linux-2.6.19.2 --with-build=/home/working/bsp/Kernel/linux-2.6.19.2 --with-moddir=/home/working/bsp/tmp --with-debug=full --enable-dynamic-minors --with-cards=all Then things become OK. What's wrong with the "INSTALL" file? Another problem. Even if "--with-cards=all" is given, codes in soc/codecs, soc/imx, and many other subdirs are not built. Is this because my target kernel version is too low (2.6.19.2)? It seems that the configure script sets some variables according to the kernel version and target platform and outputs into "toplevel.config". Is it possible to manually edit "toplevel.config" file (after configure process) to get soc/codecs, soc/imx codes compiled? Regards, Jie

1 0

[alsa-devel] Git repository from which Takashi's snapshot is taken.
by Wael Nasreddine (aka eMxyzptlk) 02 Jan '10

02 Jan '10

Hello, I would like to patch the alsa with the patch given in this message[0], which at the time, applies to Takashi's snapshot[1], but it doesn't apply at the moment, in an effort to help me maintain it I would like to apply it to the git repository from which the snapshot has been made. Can someone give me the url to the repository from which the snapshot is taken? Thanks P.S: Happy New Year!! Footnotes: ---------- [0]: http://mailman.alsa-project.org/pipermail/alsa-devel/2009-November/023203.h… [1]: ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver-sna… -- Wael Nasreddine Weem Chief-Development Officer - http://www.weem.com Blog : http://wael.nasreddine.com E-mail : wael.nasreddine(a)weem.com gTalk : wael.nasreddine(a)gmail.com Tel : +33.6.32.94.70.13 Skype : eMxyzptlk Twitter : @eMxyzptlk PGP: 1024D/C8DD18A2 06F6 1622 4BC8 4CEB D724 DE12 5565 3945 C8DD 18A2 .: An infinite number of monkeys typing into GNU emacs, would never make a good program. (L. Torvalds 1995) :. /<o>\ If entropy is increasing, where is it coming from?

2 1

[alsa-devel] How to compile 1.0.22 drivers with 2.6.33-rc?
by Elimar Riesebieter 31 Dec '09

31 Dec '09

Hi all, to built alsa-driver 1.0.22.1 against 2.6.33-rc2 I had to copy files of kernels's include/generated to include/linux. Is there a more elegant way to improve that? I want to prepare a Debian upload. So users which are running 2.6.33 can't compile the 1.0.22 drivers without manipulating the kernel built dir. Any hints? Thanks Elimar -- On the keyboard of life you have always to keep a finger at the escape key;-)

1 0

[alsa-devel] [PATCH v2] soc-cache: cleanup training whitespace and coding style
by Barry Song 31 Dec '09

31 Dec '09

Signed-off-by: Barry Song <21cnbao(a)gmail.com> --- sound/soc/soc-cache.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index d2505e8..02c2357 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -182,7 +182,7 @@ static struct { { .addr_bits = 7, .data_bits = 9, .write = snd_soc_7_9_write, .read = snd_soc_7_9_read, - .spi_write = snd_soc_7_9_spi_write + .spi_write = snd_soc_7_9_spi_write, }, { .addr_bits = 8, .data_bits = 8, -- 1.5.6.3

2 1

Re: [alsa-devel] hw_ptr accuracy with intel8x0 -> PulseAudio issue
by Raymond Yau 31 Dec '09

31 Dec '09

2009/12/4 pl bossart <bossart.nospam(a)gmail.com> > Hi Raymond, > comments below. > - PL > > On Wed, Dec 2, 2009 at 8:26 PM, Raymond Yau <superquad.vortex2(a)gmail.com> > wrote: > > 1. D: alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ... > > 2. D: alsa-util.c: Managed to open front:0 > > 3. D: alsa-util.c: Maximum hw buffer size is 371 ms > > 4. D: alsa-util.c: Set buffer size first (to 4408 samples), period size > > second (to 1102 samples) > > > > What is the purpose of these buffer-size and period size during the > probing > > phase ? > > The PA server seem use different values > > This is a red herring, PulseAudio tries to open multiple profiles and > finds out which ones make sense. You want to look at the result below > which is the final configuration. > > > 1. I: sink.c: Created sink 0 > "alsa_output.pci-0000_00_1e.2.analog-stereo" > > with sample spec s16le 2ch 44100Hz and channel map > front-left,front-right > > 2. I: sink.c: alsa.resolution_bits = "16" > > 3. I: sink.c: device.api = "alsa" > > 4. I: sink.c: device.class = "sound" > > 5. I: sink.c: alsa.class = "generic" > > 6. I: sink.c: alsa.subclass = "generic-mix" > > 7. I: sink.c: alsa.name = "Intel ICH6" > > 8. I: sink.c: alsa.id = "Intel ICH" > > 9. I: sink.c: alsa.subdevice = "0" > > 10. I: sink.c: alsa.subdevice_name = "subdevice #0" > > 11. I: sink.c: alsa.device = "0" > > 12. I: sink.c: alsa.card = "0" > > 13. I: sink.c: alsa.card_name = "Intel ICH6" > > 14. I: sink.c: alsa.long_card_name = "Intel ICH6 with AD1981B at > irq > > 22" > > 15. I: sink.c: alsa.driver_name = "snd_intel8x0" > > 16. I: sink.c: device.bus_path = "pci-0000:00:1e.2" > > 17. I: sink.c: sysfs.path = > > "/devices/pci0000:00/0000:00:1e.2/sound/card0" > > 18. I: sink.c: device.bus = "pci" > > 19. I: sink.c: device.vendor.id = "8086" > > 20. I: sink.c: device.vendor.name = "Intel Corporation" > > 21. I: sink.c: device.product.id = "266e" > > 22. I: sink.c: device.product.name = "82801FB/FBM/FR/FW/FRW (ICH6 > > Family) AC'97 Audio Controller" > > 23. I: sink.c: device.form_factor = "internal" > > 24. I: sink.c: hal.udi = > > "/org/freedesktop/Hal/devices/pci_8086_266e_sound_card_0" > > 25. I: sink.c: hal.product = "Intel ICH6 with AD1981B Sound Card" > > 26. I: sink.c: hal.card_id = "Intel ICH6 with AD1981B" > > 27. I: sink.c: device.string = "front:0" > > 28. I: sink.c: device.buffering.buffer_size = "65536" > > 29. I: sink.c: device.buffering.fragment_size = "65536" > > 30. I: sink.c: device.access_mode = "mmap+timer" > > 31. I: sink.c: device.profile.name = "analog-stereo" > > 32. I: sink.c: device.profile.description = "Analog Stereo" > > 33. I: sink.c: device.description = "Internal Audio Analog Stereo" > > 34. I: sink.c: alsa.mixer_name = "Analog Devices AD1981B" > > 35. I: sink.c: alsa.components = "AC97a:41445374" > > 36. I: sink.c: module-udev-detect.discovered = "1" > > 37. I: sink.c: device.icon_name = "audio-card-pci" > > > intel8x0 does not has hdmi , it seem PA hardcode hdmi for all sound cards > > instead of using namehint for playback device > > Yes, PA will try all possible profiles. Not that smart but unrelated > to my problem. > > > > Is it possible to add timestamp in alsa-sink.c: Wrote 31792 bytes (of > > possible 31792 bytes) statement for debugging > > Not sure why this is needed. In that case the driver reports 7948 > frames are available, which translates to 31792 bytes. > The problem shows in the next call to snd_pcm_avail down to the kernel > level, PA only uses the misleading information reported by the driver. > > 17. hwptr 24351 buffer 16384 appl 24332, avail=16403 appl pointer is behind hwprtr ( underrun ) , need timestamp to find out whether PA is unable to write audio data in time since > 13. D: alsa-sink.c: Not filling up, because too early. > 14. D: alsa-sink.c: Wakeup from ALSA! > 15. D: alsa-sink.c: Loop > 16. D: alsa-sink.c: Buffer time: 371 ms; Sleep time: 351 ms; Process > time: 20 ms did it adjust the sleeping time when it think too early ? > > > 1. D: alsa-sink.c: Loop > > 2. D: alsa-sink.c: Buffer time: 371 ms; Sleep time: 351 ms; Process > time: > > 20 ms > > 3. hwptr 7948 buffer 16384 appl 16384, avail=7948 > > 4. D: alsa-sink.c: avail: 31792 (samples 7948) > > 5. D: alsa-sink.c: 191.29 ms left to play; inc threshold = 0.00 ms; dec > > threshold = 100.00 ms > > 6. D: alsa-sink.c: Filling up > > 7. hwptr 7948 buffer 16384 appl 16384, avail=7948 > > 8. hwptr 7948 buffer 16384 appl 16384, avail=7948 > > 9. D: alsa-sink.c: Wrote 31792 bytes (of possible 31792 bytes) > > 10. hwptr 7960 buffer 16384 appl 24332, avail=12 > > 11. D: alsa-sink.c: avail: 48 (samples 12) > > 12. D: alsa-sink.c: 371.25 ms left to play; inc threshold = 0.00 ms; > dec > > threshold = 100.00 ms > > 13. D: alsa-sink.c: Not filling up, because too early. > > 14. D: alsa-sink.c: Wakeup from ALSA! > > 15. D: alsa-sink.c: Loop > > 16. D: alsa-sink.c: Buffer time: 371 ms; Sleep time: 351 ms; Process > > time: 20 ms > > 17. hwptr 24351 buffer 16384 appl 24332, avail=16403 > > 18. E: alsa-util.c: avail 16403 delay -19 > > 19. D: alsa-sink.c: avail: 65536 (samples 16384) > > 20. D: alsa-sink.c: 0.00 ms left to play; inc threshold = 0.00 ms; dec > > threshold = 100.00 ms > > 21. D: alsa-sink.c: Filling up >

4 16

[alsa-devel] [PATCH 1/3] ASoC: fsi: Add over_period flag to prevent the misunderstanding
by Kuninori Morimoto 30 Dec '09

30 Dec '09

Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- sound/soc/sh/fsi.c | 20 ++++++++++++-------- 1 files changed, 12 insertions(+), 8 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 497c5e0..4337afc 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -373,14 +373,16 @@ static int fsi_data_push(struct fsi_priv *fsi) int fifo_free; int width; u8 *start; - int i; + int i, over_period; if (!fsi || !fsi->substream || !fsi->substream->runtime) return -EINVAL; - runtime = fsi->substream->runtime; + over_period = 0; + substream = fsi->substream; + runtime = substream->runtime; /* FSI FIFO has limit. * So, this driver can not send periods data at a time @@ -388,7 +390,7 @@ static int fsi_data_push(struct fsi_priv *fsi) if (fsi->byte_offset >= fsi->period_len * (fsi->periods + 1)) { - substream = fsi->substream; + over_period = 1; fsi->periods = (fsi->periods + 1) % runtime->periods; if (0 == fsi->periods) @@ -429,7 +431,7 @@ static int fsi_data_push(struct fsi_priv *fsi) fsi_irq_enable(fsi, 1); - if (substream) + if (over_period) snd_pcm_period_elapsed(substream); return 0; @@ -443,14 +445,16 @@ static int fsi_data_pop(struct fsi_priv *fsi) int fifo_fill; int width; u8 *start; - int i; + int i, over_period; if (!fsi || !fsi->substream || !fsi->substream->runtime) return -EINVAL; - runtime = fsi->substream->runtime; + over_period = 0; + substream = fsi->substream; + runtime = substream->runtime; /* FSI FIFO has limit. * So, this driver can not send periods data at a time @@ -458,7 +462,7 @@ static int fsi_data_pop(struct fsi_priv *fsi) if (fsi->byte_offset >= fsi->period_len * (fsi->periods + 1)) { - substream = fsi->substream; + over_period = 1; fsi->periods = (fsi->periods + 1) % runtime->periods; if (0 == fsi->periods) @@ -498,7 +502,7 @@ static int fsi_data_pop(struct fsi_priv *fsi) fsi_irq_enable(fsi, 0); - if (substream) + if (over_period) snd_pcm_period_elapsed(substream); return 0; -- 1.6.3.3

2 1

[alsa-devel] [PATCH 0/3] ad1836/ad1938: PM and typo fix/cleanup
by Barry Song 30 Dec '09

30 Dec '09

Hi Mark, The following 3 patches fix/cleanup PM and typo of ad1836/ad1938 codec driver: 1. ad1836: reset and restore clock control mode in suspend/resume entry 2. ad1938: fix typo, rename mask to rx_mask for ad1938_set_tdm_slot 3. ad1938:let soc-core dapm handle PLL power As I will send patches special for moving to soc-cache, in patch 1, I still use the old way to read/write codec registers. Thanks Barry

2 4

[alsa-devel] ALSA: HDA VIA VT2020 support
by draco 30 Dec '09

30 Dec '09

Hello, I saw in the devel archive that the support for VIA VT2020 chipset have been added and merged to 1.0.22 released. (cf. http://mailman.alsa-project.org/pipermail/alsa-devel/2009-October/022183.ht… and http://www.alsa-project.org/main/index.php/Changes_v1.0.21_v1.0.22 ) Unfortunately, despite all was OK from the 1.0.21+git devel snapshot few month ago, the alsa-driver v1.0.22 doesn't work. I tried different snapshot from 1.0.22, and sent a bug report here : https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4857 I don't know what I can do more to resolve the bug, but I can make some other tests and give you more log if needed. NB : I don't subscribe the devel mailing list, but I am "monitoring" the above bug report. Thank's in advance. Best Regards, draco

2 2

[alsa-devel] [PATCH] soc-cache: fix training whitespace checkpatch issue
by Barry Song 30 Dec '09

30 Dec '09

Signed-off-by: Barry Song <21cnbao(a)gmail.com> --- sound/soc/soc-cache.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index d2505e8..906878e 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -182,7 +182,7 @@ static struct { { .addr_bits = 7, .data_bits = 9, .write = snd_soc_7_9_write, .read = snd_soc_7_9_read, - .spi_write = snd_soc_7_9_spi_write + .spi_write = snd_soc_7_9_spi_write }, { .addr_bits = 8, .data_bits = 8, -- 1.5.6.3

2 2

[alsa-devel] [PATCH] ASoC: fsi-ak4642: Remove ak4642_add_i2c_device
by Kuninori Morimoto 30 Dec '09

30 Dec '09

I2C devices should be registered when platform board setting in latest ASoC. Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- sound/soc/sh/fsi-ak4642.c | 30 ------------------------------ 1 files changed, 0 insertions(+), 30 deletions(-) diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c index c7af097..5263ab1 100644 --- a/sound/soc/sh/fsi-ak4642.c +++ b/sound/soc/sh/fsi-ak4642.c @@ -42,42 +42,12 @@ static struct snd_soc_device fsi_snd_devdata = { .codec_dev = &soc_codec_dev_ak4642, }; -#define AK4642_BUS 0 -#define AK4642_ADR 0x12 -static int ak4642_add_i2c_device(void) -{ - struct i2c_board_info info; - struct i2c_adapter *adapter; - struct i2c_client *client; - - memset(&info, 0, sizeof(struct i2c_board_info)); - info.addr = AK4642_ADR; - strlcpy(info.type, "ak4642", I2C_NAME_SIZE); - - adapter = i2c_get_adapter(AK4642_BUS); - if (!adapter) { - printk(KERN_DEBUG "can't get i2c adapter\n"); - return -ENODEV; - } - - client = i2c_new_device(adapter, &info); - i2c_put_adapter(adapter); - if (!client) { - printk(KERN_DEBUG "can't add i2c device\n"); - return -ENODEV; - } - - return 0; -} - static struct platform_device *fsi_snd_device; static int __init fsi_ak4642_init(void) { int ret = -ENOMEM; - ak4642_add_i2c_device(); - fsi_snd_device = platform_device_alloc("soc-audio", -1); if (!fsi_snd_device) goto out; -- 1.6.3.3

3 7

[alsa-devel] X-Fi notebook support
by The Source 30 Dec '09

30 Dec '09

Hello. I have some info on that card, but not much. I have discovered that this card is detected connected via USB rather than PCI-E (the slot is ExpressCard). snd_ctxfi uses PCI interfaces right? Can X-Fi support be also added to snd_usb_audio? Or can snd_ctxfi support USB?

1 0

[alsa-devel] Front jack not working with hda-intel driver SigmaTel STAC9271D codec
by David Mohr 30 Dec '09

30 Dec '09

Hi, I'm running ubuntu 9.10. I had to force model=5stack to get any audio output at all. While the jack in the back now works, I still don't get any sound on the front jack. I haven't found any options in alsamixer, and switching the output mode in pavucontrol to the headphones just makes the sound disappear from the back jack, but I still get nothing in the front. I uploaded my alsa-info.sh output here: http://www.alsa-project.org/db/?f=3ed2ab9e7732bf08d2bbfbe8adb74bf037d9e618 Can I fix this somehow, or is it a bug? And of course thanks everyone for the usually great working alsa system. ~David

2 3

[alsa-devel] ASoC updates for 2.6.33
by Mark Brown 29 Dec '09

29 Dec '09

The following changes since commit 6b7b284958d47b77d06745b36bc7f36dab769d9b: Linus Torvalds (1): Linux 2.6.33-rc2 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.33 The change in the branch is much bigger than the diffstat shows due to the merge up of Linus' tree (which now builds for ARM). Guennadi Liakhovetski (1): ASoC: fix params_rate() macro use in several codecs Kuninori Morimoto (1): ASoC: fsi-ak4642: Remove ak4642_add_i2c_device Mark Brown (1): Merge commit 'v2.6.33-rc2' into for-2.6.33 sound/soc/codecs/wm8510.c | 14 +++++++------- sound/soc/codecs/wm8940.c | 14 +++++++------- sound/soc/codecs/wm8974.c | 14 +++++++------- sound/soc/sh/fsi-ak4642.c | 30 ------------------------------ 4 files changed, 21 insertions(+), 51 deletions(-)

2 1

[alsa-devel] DMA and delay feedback
by James Courtier-Dutton 29 Dec '09

29 Dec '09

Hi, Would there be benefit in separating the DMA and the transfer of audio samples to the sound card hardware from the delay and hw position feedback. For example, If the playback position is 0, the hardware has probably already transferred one period of the sound card. So, one cannot write to playback position 2 and expect it to be output to the sound card if the period size is 1024. For transfer purposes, one only needs to know that DMA transfer is complete on period X so that period X can now be over-written. I believe that this is what the DMA interrupt indicates. One needs to reliably detect a missed interrupt. It one took this approach, one might be able to hide the "missed interrupt" problem. When the interrupt finally did arrive, we would know where the next free period X is, and continue writing samples to that, irrespective of what went wrong before. This would allow for auto-recovery of the audio stream without having to stop and start the ring buffers again. One then needs to determine how one can get accurate "delay" values back from all sound card types. The "delay" value is used for applications like ensuring that audio and video play in sync. The problem ones at the moment appear to be the intel8x0 and hd-audio ones. There may be benefits from making the "delay" value return in nanoseconds instead of samples, particularly if gettimeofdate() is being used to try to produce accurate "delay" values. Kind Regards James

4 6

[alsa-devel] frequency analysis with alsa and fftw
by Markus Vogl 28 Dec '09

28 Dec '09

hello, i´m absolute newbie in programming alsa, but for my school project i have to do so. my project is to analyse the mic input frequency from my asus eepc 1008ha with ubuntu 9.10 and show it in a terminal window. i would use my acoustic 6 string guitar to play in front of my netbook. i´ve testet the howto from linuxjornal.com .. http://www.linuxjournal.com/article/6735 ...but i´ve no plan to use the mic input and then analyse it with fftw. regards markus -- Jetzt kostenlos herunterladen: Internet Explorer 8 und Mozilla Firefox 3.5 - sicherer, schneller und einfacher! http://portal.gmx.net/de/go/atbrowser

2 4

[alsa-devel] [PATCH - usb/usbaudio.c 1/1] sound/usb: Relax urb data align. restriction for HVR-950Q and HVR-850 only
by John S Gruber 28 Dec '09

28 Dec '09

Addressing audio quality problem. In sound/usb/usbaudio.c, for the Hauppage HVR-950Q only, change retire_capture_urb to copy the entire byte stream while still counting entire audio frames. urbs unaligned on channel sample boundaries are still truncated to the next lowest stride (audio slot) size to try to retain channel alignment in cases of data loss over usb. With the HVR950Q the left and right channel samples can be split between two different urbs. Throwing away extra channel samples causes a sound quality problem for stereo streams as the left and right channels are swapped repeatedly. BugLink: https://bugs.launchpad.net/ubuntu/+bug/495745 Signed-off-by: John S. Gruber <JohnSGruber(a)gmail.com> --- usb/usbaudio.c | 53 ++++++++++++++++++++++++++++++++++++++++------------- 1 files changed, 40 insertions(+), 13 deletions(-) diff --git a/usb/usbaudio.c b/usb/usbaudio.c index b074a59..d450b23 100644 --- a/usb/usbaudio.c +++ b/usb/usbaudio.c @@ -108,6 +108,7 @@ MODULE_PARM_DESC(ignore_ctl_error, #define MAX_URBS 8 #define SYNC_URBS 4 /* always four urbs for sync */ #define MAX_QUEUE 24 /* try not to exceed this queue length, in ms */ +#define ALLOW_SUBSLOT_BOUNDARIES 0x01 /* quirk */ struct audioformat { struct list_head list; @@ -127,6 +128,7 @@ struct audioformat { unsigned int rate_min, rate_max; /* min/max rates */ unsigned int nr_rates; /* number of rate table entries */ unsigned int *rate_table; /* rate table */ + unsigned int txfr_quirks; /* transfer quirks */ }; struct snd_usb_substream; @@ -175,6 +177,8 @@ struct snd_usb_substream { unsigned int running: 1; /* running status */ unsigned int hwptr_done; /* processed frame position in the buffer */ + unsigned int byteptr; /* position, in bytes, of next move */ + unsigned int txfr_quirks; /* substream transfer quirks */ unsigned int transfer_done; /* processed frames since last period update */ unsigned long active_mask; /* bitmask of active urbs */ unsigned long unlink_mask; /* bitmask of unlinked urbs */ @@ -343,7 +347,7 @@ static int retire_capture_urb(struct snd_usb_substream *subs, unsigned long flags; unsigned char *cp; int i; - unsigned int stride, len, oldptr; + unsigned int stride, len, bytelen, oldbyteptr; int period_elapsed = 0; stride = runtime->frame_bits >> 3; @@ -354,29 +358,39 @@ static int retire_capture_urb(struct snd_usb_substream *subs, snd_printd(KERN_ERR "frame %d active: %d\n", i, urb->iso_frame_desc[i].status); // continue; } - len = urb->iso_frame_desc[i].actual_length / stride; - if (! len) + bytelen = (urb->iso_frame_desc[i].actual_length); + if (!bytelen) continue; + if (bytelen%(runtime->sample_bits>>3) != 0) { + int oldbytelen = bytelen; + bytelen = ((bytelen/stride)*stride); + snd_printd(KERN_ERR "Corrected urb data len. %d->%d\n", + oldbytelen, bytelen); + } + if (!(subs->txfr_quirks & ALLOW_SUBSLOT_BOUNDARIES)) + bytelen = (bytelen/stride)*stride; /* update the current pointer */ spin_lock_irqsave(&subs->lock, flags); - oldptr = subs->hwptr_done; - subs->hwptr_done += len; - if (subs->hwptr_done >= runtime->buffer_size) - subs->hwptr_done -= runtime->buffer_size; + len = (bytelen + (subs->byteptr % stride)) / stride; subs->transfer_done += len; if (subs->transfer_done >= runtime->period_size) { subs->transfer_done -= runtime->period_size; period_elapsed = 1; } + oldbyteptr = subs->byteptr; + subs->byteptr += bytelen; + if (subs->byteptr >= runtime->buffer_size*stride) + subs->byteptr -= runtime->buffer_size*stride; + subs->hwptr_done = subs->byteptr/stride; spin_unlock_irqrestore(&subs->lock, flags); /* copy a data chunk */ - if (oldptr + len > runtime->buffer_size) { - unsigned int cnt = runtime->buffer_size - oldptr; - unsigned int blen = cnt * stride; - memcpy(runtime->dma_area + oldptr * stride, cp, blen); - memcpy(runtime->dma_area, cp + blen, len * stride - blen); + if ((oldbyteptr + bytelen) > (runtime->buffer_size*stride)) { + unsigned int blen; + blen = (runtime->buffer_size*stride) - oldbyteptr; + memcpy(runtime->dma_area + oldbyteptr, cp, blen); + memcpy(runtime->dma_area, cp + blen, bytelen - blen); } else { - memcpy(runtime->dma_area + oldptr * stride, cp, len * stride); + memcpy(runtime->dma_area + oldbyteptr, cp, bytelen); } } if (period_elapsed) @@ -1363,6 +1377,7 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) subs->syncpipe = subs->syncinterval = 0; subs->maxpacksize = fmt->maxpacksize; subs->fill_max = 0; + subs->txfr_quirks = fmt->txfr_quirks; /* we need a sync pipe in async OUT or adaptive IN mode */ /* check the number of EP, since some devices have broken @@ -1532,6 +1547,7 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) /* reset the pointer */ subs->hwptr_done = 0; subs->transfer_done = 0; + subs->byteptr = 0; subs->phase = 0; runtime->delay = 0; @@ -2776,6 +2792,7 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) fp->maxpacksize = (((fp->maxpacksize >> 11) & 3) + 1) * (fp->maxpacksize & 0x7ff); fp->attributes = csep ? csep[3] : 0; + fp->txfr_quirks = 0; /* some quirks for attributes here */ @@ -2804,6 +2821,16 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) else fp->ep_attr |= EP_ATTR_SYNC; break; + case USB_ID(0x2040, 0x7200): /* Hauppage hvr950Q */ + case USB_ID(0x2040, 0x7221): /* Hauppage hvr950Q */ + case USB_ID(0x2040, 0x7222): /* Hauppage hvr950Q */ + case USB_ID(0x2040, 0x7223): /* Hauppage hvr950Q */ + case USB_ID(0x2040, 0x7224): /* Hauppage hvr950Q */ + case USB_ID(0x2040, 0x7225): /* Hauppage hvr950Q */ + case USB_ID(0x2040, 0x7230): /* Hauppage hvr850 */ + case USB_ID(0x2040, 0x7250): /* Hauppage hvr950Q */ + fp->txfr_quirks |= ALLOW_SUBSLOT_BOUNDARIES; + break; } /* ok, let's parse further... */ -- 1.6.3.3 I'm new to changing the kernel. Normally the driver should transfer only whole audio frames, disregarding fragments, in order to keep channel and sample synchronization in the face of data loss on the bus. That's how the current driver works. The patch relaxes this (only) for devices with the quirk. An earlier version of this patch was posted recently to the linux-media list. No one there suggested a way to make the au0828 chip in this equipment behave better. They kindly forwarded me to this mailing list for review and consideration of the patch. To test the patch with a non-quirk device run it on a busy or problematic bus and listen for audio switching channels. To test with a HVR-950Q use composite input and disconnect either the left or right channel. Without the patch the audio will alternate between channels quickly, most easily heard with headphones or earphones, producing a flutter sound, With the patch the remaining test audio channel appears only on one channel and is steady.

5 8

[alsa-devel] [PATCH] ALSA: test off by one in setsamplerate()
by Roel Kluin 28 Dec '09

28 Dec '09

With `while (i++ < MAX_WRITE_RETRY)' i reaches MAX_WRITE_RETRY + 1 after the loop Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- sound/pci/riptide/riptide.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index b5ca02e..e66ef2b 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -1058,7 +1058,7 @@ setsamplerate(struct cmdif *cif, unsigned char *intdec, unsigned int rate) rptr.retwords[2] != M && rptr.retwords[3] != N && i++ < MAX_WRITE_RETRY); - if (i == MAX_WRITE_RETRY) { + if (i > MAX_WRITE_RETRY) { snd_printdd("sent samplerate %d: %d failed\n", *intdec, rate); return -EIO;

2 1

[alsa-devel] [PATCH] ALSA: atiixp: Specify codec for Foxconn RC4107MA-RS2
by Daniel Chen 28 Dec '09

28 Dec '09

Hi, Attached is a patch against sound-2.6.git and also targeted for stable to make sound audible by default for this mainboard. Thanks, Dan

2 1

[alsa-devel] [PATCH 1/1] Added functionality for E-mu 0404USB/0202USB/TrackerPre
by Zerg Cannibal 28 Dec '09

28 Dec '09

Added functionality: 1) Extension Units support (all XU settings now available at alsamixer, kmix, etc): - "AnalogueIn soft limiter" switch; - "Sample rate" selector (values 0,1,2,3,4,5 corresponds to 44.1 48 ... 192 kHz); - "DigitalIn CLK source" selector (internal/external) (**); - "DigitalOut format SPDIF/AC3" switch (**); (**)E-mu-0404usb only. 2) Automatic device sample rate adjustment depending on substream samplerate for both capture and playback substream. Patch is based on Mac OS-X driver (http://sourceforge.net/projects/zaudiodrivermac/) and successfully tested by Ubuntu community users for 0404usb and 0202usb. Signed-off-by: CannibalZerg <cnb_zerg(a)yahoo.com> --- sound/usb/usbaudio.c | 44 ++++++++++++++++++++++++++++++ sound/usb/usbaudio.h | 2 + sound/usb/usbmixer.c | 73 +++++++++++++++++++++++++++++++++++++++++++++++--- 3 files changed, 115 insertions(+), 4 deletions(-) diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 31b63ea..d8eb029 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -1271,6 +1271,42 @@ static int init_usb_sample_rate(struct usb_device *dev, int iface, } /* + * For E-Mu 0404USB/0202USB/TrackerPre sample rate should be set for device, + * not for interface. +*/ +static void set_format_emu_quirk(struct snd_usb_substream *subs, struct audioformat *fmt) +{ + unsigned char emu_samplerate_id = 0; + + /* When capture is active sample rate shouldn't be changed by playback substream*/ + if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) { + if (subs->stream->substream[SNDRV_PCM_STREAM_CAPTURE].interface != -1) + return; + } + + switch (fmt->rate_min) { + case 48000: + emu_samplerate_id = 1; + break; + case 88200: + emu_samplerate_id = 2; + break; + case 96000: + emu_samplerate_id = 3; + break; + case 176400: + emu_samplerate_id = 4; + break; + case 192000: + emu_samplerate_id = 5; + break; + default: + emu_samplerate_id = 0; + } + snd_emuusb_set_samplerate(subs->stream->chip, emu_samplerate_id); +} + +/* * find a matching format and set up the interface */ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) @@ -1383,6 +1419,14 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) subs->cur_audiofmt = fmt; + switch (subs->stream->chip->usb_id) { + case USB_ID(0x041e, 0x3f02): /* E-Mu 0202 USB */ + case USB_ID(0x041e, 0x3f04): /* E-Mu 0404 USB */ + case USB_ID(0x041e, 0x3f0a): /* E-Mu Tracker Pre */ + set_format_emu_quirk(subs, fmt); + break; + } + #if 0 printk(KERN_DEBUG "setting done: format = %d, rate = %d..%d, channels = %d\n", diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 9826337..8558773 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -233,6 +233,8 @@ void snd_usbmidi_input_stop(struct list_head* p); void snd_usbmidi_input_start(struct list_head* p); void snd_usbmidi_disconnect(struct list_head *p); +void snd_emuusb_set_samplerate(struct snd_usb_audio *chip, unsigned char samplerate_id); + /* * retrieve usb_interface descriptor from the host interface * (conditional for compatibility with the older API) diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index c998220..400208c 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -186,6 +186,21 @@ enum { USB_PROC_DCR_RELEASE = 6, }; +/*E-mu 0202(0404) eXtension Unit(XU) control*/ +enum { + USB_XU_CLOCK_RATE = 0xe301, + USB_XU_CLOCK_SOURCE = 0xe302, + USB_XU_DIGITAL_IO_STATUS = 0xe303, + USB_XU_DEVICE_OPTIONS = 0xe304, + USB_XU_DIRECT_MONITORING = 0xe305, + USB_XU_METERING = 0xe306 +}; +enum { + USB_XU_CLOCK_SOURCE_SELECTOR = 0x02, /* clock source*/ + USB_XU_CLOCK_RATE_SELECTOR = 0x03, /* clock rate */ + USB_XU_DIGITAL_FORMAT_SELECTOR = 0x01, /* the spdif format */ + USB_XU_SOFT_LIMIT_SELECTOR = 0x03 /* soft limiter */ +}; /* * manual mapping of mixer names @@ -1330,7 +1345,32 @@ static struct procunit_info procunits[] = { { USB_PROC_DCR, "DCR", dcr_proc_info }, { 0 }, }; - +/* + * predefined data for extension units + */ +static struct procunit_value_info clock_rate_xu_info[] = { + { USB_XU_CLOCK_RATE_SELECTOR, "Selector", USB_MIXER_U8, 0}, + { 0 } +}; +static struct procunit_value_info clock_source_xu_info[] = { + { USB_XU_CLOCK_SOURCE_SELECTOR, "External", USB_MIXER_BOOLEAN}, + { 0 } +}; +static struct procunit_value_info spdif_format_xu_info[] = { + { USB_XU_DIGITAL_FORMAT_SELECTOR, "SPDIF/AC3", USB_MIXER_BOOLEAN}, + { 0 } +}; +static struct procunit_value_info soft_limit_xu_info[] = { + { USB_XU_SOFT_LIMIT_SELECTOR, " ", USB_MIXER_BOOLEAN}, + { 0 } +}; +static struct procunit_info extunits[] = { + { USB_XU_CLOCK_RATE, "Clock rate", clock_rate_xu_info }, + { USB_XU_CLOCK_SOURCE, "DigitalIn CLK source", clock_source_xu_info }, + { USB_XU_DIGITAL_IO_STATUS, "DigitalOut format:", spdif_format_xu_info }, + { USB_XU_DEVICE_OPTIONS, "AnalogueIn Soft Limit", soft_limit_xu_info }, + { 0 } +}; /* * build a processing/extension unit */ @@ -1391,8 +1431,16 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, unsigned cval->max = dsc[15]; cval->res = 1; cval->initialized = 1; - } else - get_min_max(cval, valinfo->min_value); + } else { + if (type == USB_XU_CLOCK_RATE) { + /*E-Mu USB 0404/0202/TrackerPre samplerate control quirk*/ + cval->min = 0; + cval->max = 5; + cval->res = 1; + cval->initialized = 1; + } else + get_min_max(cval, valinfo->min_value); + } kctl = snd_ctl_new1(&mixer_procunit_ctl, cval); if (! kctl) { @@ -1433,7 +1481,7 @@ static int parse_audio_processing_unit(struct mixer_build *state, int unitid, un static int parse_audio_extension_unit(struct mixer_build *state, int unitid, unsigned char *desc) { - return build_audio_procunit(state, unitid, desc, NULL, "Extension Unit"); + return build_audio_procunit(state, unitid, desc, extunits, "Extension Unit"); } @@ -2109,6 +2157,23 @@ static int snd_xonar_u1_controls_create(struct usb_mixer_interface *mixer) return 0; } +void snd_emuusb_set_samplerate(struct snd_usb_audio *chip, unsigned char samplerate_id) +{ + struct usb_mixer_interface *mixer; + struct list_head *p; + struct usb_mixer_elem_info *cval; + int unitid = 12; /*SampleRate ExtensionUnit ID*/ + list_for_each(p, &chip->mixer_list) { + mixer = list_entry(p, struct usb_mixer_interface, list); + cval = mixer->id_elems[unitid]; + if (cval) { + set_cur_ctl_value(cval, cval->control << 8, samplerate_id); + snd_usb_mixer_notify_id(mixer, unitid); + } + break; + } +} + int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, int ignore_error) { -- 1.6.5.6

4 7

[alsa-devel] Use of default device with snd_pcm_link()
by Devin Heitmueller 28 Dec '09

28 Dec '09

I am working on some code which captures from one PCM device and plays to another, based off of the latency.c found in alsa-lib. I noticed something strange: The code works if I set the capture device to hw:1,0 and playback device to hw:0,0. However, if I change the playback device to "default:0", the snd_pcm_link() call fails with -ENOSYS (error 38). If I dump out the properties, I can see that the the default device *is* mapped to device 0,0, so it's not clear why the behaviour would differ. My goal was just to playback to whatever the default device is, and I was under the impression that default:0 should be equivalent to "hw:0,0". Can anyone offer any brief explanation as to why the snd_pcm_link() call would fail? Thanks in advance, Devin -- Devin J. Heitmueller - Kernel Labs http://www.kernellabs.com

2 1

[alsa-devel] [PATCH 1/2] OMAP3: MCBSP: Suspend/resume failure fix
by Jane Wang 28 Dec '09

28 Dec '09

McBSP fucntional clock is not disabled when suspend is triggerd which prevents PER domain entering target low-power state. To fix the problem: 1. Disable/enable McBSP functional clock for suspend/resume. 2. In addition, reconfigure McBSP, this is needed when systerm comes out of OFF state. Signed-off-by: Jane Wang <jwang(a)ti.com> --- arch/arm/plat-omap/include/mach/mcbsp.h | 2 ++ arch/arm/plat-omap/mcbsp.c | 27 +++++++++++++++++++++++++++ sound/soc/omap/omap-mcbsp.c | 14 ++++++++++++-- 3 files changed, 41 insertions(+), 2 deletions(-) diff --git a/arch/arm/plat-omap/include/mach/mcbsp.h b/arch/arm/plat-omap/include/mach/mcbsp.h index e0d6eca..94e1c66 100644 --- a/arch/arm/plat-omap/include/mach/mcbsp.h +++ b/arch/arm/plat-omap/include/mach/mcbsp.h @@ -440,6 +440,8 @@ static inline int omap_mcbsp_get_dma_op_mode(unsigned int id) { return 0; } #endif int omap_mcbsp_request(unsigned int id); void omap_mcbsp_free(unsigned int id); +void omap_mcbsp_disable_fclk(unsigned int id); +void omap_mcbsp_enable_fclk(unsigned int id); void omap_mcbsp_start(unsigned int id, int tx, int rx); void omap_mcbsp_stop(unsigned int id, int tx, int rx); void omap_mcbsp_xmit_word(unsigned int id, u32 word); diff --git a/arch/arm/plat-omap/mcbsp.c b/arch/arm/plat-omap/mcbsp.c index e664b91..bfe3c61 100644 --- a/arch/arm/plat-omap/mcbsp.c +++ b/arch/arm/plat-omap/mcbsp.c @@ -459,6 +459,33 @@ int omap_mcbsp_request(unsigned int id) } EXPORT_SYMBOL(omap_mcbsp_request); +void omap_mcbsp_disable_fclk(unsigned int id) +{ + struct omap_mcbsp *mcbsp; + + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return; + } + mcbsp = id_to_mcbsp_ptr(id); + clk_disable(mcbsp->fclk); +} +EXPORT_SYMBOL(omap_mcbsp_disable_fclk); + + +void omap_mcbsp_enable_fclk(unsigned int id) +{ + struct omap_mcbsp *mcbsp; + + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return; + } + mcbsp = id_to_mcbsp_ptr(id); + clk_enable(mcbsp->fclk); +} +EXPORT_SYMBOL(omap_mcbsp_enable_fclk); + void omap_mcbsp_free(unsigned int id) { struct omap_mcbsp *mcbsp; diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 45be942..e56b5ff 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -229,18 +229,28 @@ static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd, switch (cmd) { case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: mcbsp_data->active++; omap_mcbsp_start(mcbsp_data->bus_id, play, !play); break; + case SNDRV_PCM_TRIGGER_RESUME: + mcbsp_data->active++; + omap_mcbsp_enable_fclk(mcbsp_data->bus_id); + omap_mcbsp_config(mcbsp_data->bus_id, + &mcbsp_data->regs); + omap_mcbsp_start(mcbsp_data->bus_id, play, !play); + break; case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: omap_mcbsp_stop(mcbsp_data->bus_id, play, !play); mcbsp_data->active--; break; + case SNDRV_PCM_TRIGGER_SUSPEND: + omap_mcbsp_stop(mcbsp_data->bus_id, play, !play); + omap_mcbsp_disable_fclk(mcbsp_data->bus_id); + mcbsp_data->active--; + break; default: err = -EINVAL; } -- 1.5.4.3

8 15

[alsa-devel] [PATCH 3/3] ASoC: fsi: Add over/under run error settlement
by Kuninori Morimoto 28 Dec '09

28 Dec '09

Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- sound/soc/sh/fsi.c | 29 +++++++++++++++++++++++++---- 1 files changed, 25 insertions(+), 4 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 099fc8c..5f9f269 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -67,6 +67,7 @@ /* DOFF_ST */ #define ERR_OVER 0x00000010 #define ERR_UNDER 0x00000001 +#define ST_ERR (ERR_OVER | ERR_UNDER) /* CLK_RST */ #define B_CLK 0x00000010 @@ -375,11 +376,12 @@ static int fsi_data_push(struct fsi_priv *fsi) { struct snd_pcm_runtime *runtime; struct snd_pcm_substream *substream = NULL; + u32 status; int send; int fifo_free; int width; u8 *start; - int i, over_period; + int i, ret, over_period; if (!fsi || !fsi->substream || @@ -435,23 +437,33 @@ static int fsi_data_push(struct fsi_priv *fsi) fsi->byte_offset += send * width; + ret = 0; + status = fsi_reg_read(fsi, DOFF_ST); + if (status & ERR_OVER) { + struct snd_soc_dai *dai = fsi_get_dai(substream); + dev_err(dai->dev, "over run error\n"); + fsi_reg_write(fsi, DOFF_ST, status & ~ST_ERR); + ret = -EIO; + } + fsi_irq_enable(fsi, 1); if (over_period) snd_pcm_period_elapsed(substream); - return 0; + return ret; } static int fsi_data_pop(struct fsi_priv *fsi) { struct snd_pcm_runtime *runtime; struct snd_pcm_substream *substream = NULL; + u32 status; int free; int fifo_fill; int width; u8 *start; - int i, over_period; + int i, ret, over_period; if (!fsi || !fsi->substream || @@ -506,12 +518,21 @@ static int fsi_data_pop(struct fsi_priv *fsi) fsi->byte_offset += fifo_fill * width; + ret = 0; + status = fsi_reg_read(fsi, DIFF_ST); + if (status & ERR_UNDER) { + struct snd_soc_dai *dai = fsi_get_dai(substream); + dev_err(dai->dev, "under run error\n"); + fsi_reg_write(fsi, DIFF_ST, status & ~ST_ERR); + ret = -EIO; + } + fsi_irq_enable(fsi, 0); if (over_period) snd_pcm_period_elapsed(substream); - return 0; + return ret; } static irqreturn_t fsi_interrupt(int irq, void *data) -- 1.6.3.3

1 0

[alsa-devel] [PATCH 2/3] ASoC: fsi: Add fsi_get_dai to get snd_soc_dai
by Kuninori Morimoto 28 Dec '09

28 Dec '09

Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- sound/soc/sh/fsi.c | 10 ++++++++-- 1 files changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 4337afc..099fc8c 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -210,11 +210,17 @@ static int fsi_is_port_a(struct fsi_priv *fsi) return fsi->master->base == fsi->base; } -static struct fsi_priv *fsi_get_priv(struct snd_pcm_substream *substream) +static struct snd_soc_dai *fsi_get_dai(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai_link *machine = rtd->dai; - struct snd_soc_dai *dai = machine->cpu_dai; + + return machine->cpu_dai; +} + +static struct fsi_priv *fsi_get_priv(struct snd_pcm_substream *substream) +{ + struct snd_soc_dai *dai = fsi_get_dai(substream); return dai->private_data; } -- 1.6.3.3

1 0

[alsa-devel] [PATCH 0/3] ASoC: fsi: Add over/under run error operatoin
by Kuninori Morimoto 28 Dec '09

28 Dec '09

Dear Mark These patches add over/under run error settlement support Kuninori Morimoto (3): ASoC: fsi: Add over_period flag to prevent the misunderstanding ASoC: fsi: Add fsi_get_dai to get snd_soc_dai ASoC: fsi: Add over/under run error settlement 1st patch modify un-understandable operation. 2nd patch add new function for 3rd patch Best regards -- Kuninori Morimoto

1 0

[alsa-devel] [PATCH 3/3] sound: usb-audio: relax urb data align. restriction HVR-950Q and HVR-850 only
by johnsgruber＠gmail.com 27 Dec '09

27 Dec '09

From: John S. Gruber <JohnSGruber(a)gmail.com> Detect the HVR-950Q HVR-850 urb data alignment quirk using usbquirk.h rather than using a case statement in snd_usb_audio_probe. Signed-off-by: John S. Gruber <JohnSGruber(a)gmail.com> --- sound/usb/usbaudio.c | 30 ++++++------ sound/usb/usbaudio.h | 1 + sound/usb/usbquirks.h | 114 +++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 130 insertions(+), 15 deletions(-) diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index be6f017..0cfff95 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -3152,6 +3152,18 @@ static int ignore_interface_quirk(struct snd_usb_audio *chip, return 0; } +/* + * Allow alignment on audio sub-slot (channel samples) rather than + * on audio slots (audio frames) + */ +static int create_align_transfer_quirk(struct snd_usb_audio *chip, + struct usb_interface *iface, + const struct snd_usb_audio_quirk *quirk) +{ + chip->txfr_quirk = 1; + return 1; /* Continue with creating streams and mixer */ +} + /* * boot quirks @@ -3326,7 +3338,8 @@ static int snd_usb_create_quirk(struct snd_usb_audio *chip, [QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_audio_quirk, [QUIRK_AUDIO_FIXED_ENDPOINT] = create_fixed_stream_quirk, [QUIRK_AUDIO_EDIROL_UA1000] = create_ua1000_quirk, - [QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk + [QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk, + [QUIRK_AUDIO_ALIGN_TRANSFER] = create_align_transfer_quirk }; if (quirk->type < QUIRK_TYPE_COUNT) { @@ -3580,20 +3593,7 @@ static void *snd_usb_audio_probe(struct usb_device *dev, } } - switch (chip->usb_id) { - case USB_ID(0x2040, 0x7200): /* Hauppage hvr950Q */ - case USB_ID(0x2040, 0x7221): /* Hauppage hvr950Q */ - case USB_ID(0x2040, 0x7222): /* Hauppage hvr950Q */ - case USB_ID(0x2040, 0x7223): /* Hauppage hvr950Q */ - case USB_ID(0x2040, 0x7224): /* Hauppage hvr950Q */ - case USB_ID(0x2040, 0x7225): /* Hauppage hvr950Q */ - case USB_ID(0x2040, 0x7230): /* Hauppage hvr850 */ - case USB_ID(0x2040, 0x7250): /* Hauppage hvr950Q */ - chip->txfr_quirk = 1; - break; - default: - chip->txfr_quirk = 0; - } + chip->txfr_quirk = 0; err = 1; /* continue */ if (quirk && quirk->ifnum != QUIRK_NO_INTERFACE) { /* need some special handlings */ diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 228da06..9c88f60 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -161,6 +161,7 @@ enum quirk_type { QUIRK_AUDIO_FIXED_ENDPOINT, QUIRK_AUDIO_EDIROL_UA1000, QUIRK_AUDIO_EDIROL_UAXX, + QUIRK_AUDIO_ALIGN_TRANSFER, QUIRK_TYPE_COUNT }; diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index bd6706c..65bbd22 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -2074,6 +2074,120 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +/* Hauppauge HVR-950Q and HVR-850 */ +{ + USB_DEVICE_VENDOR_SPEC(0x2040, 0x7200), + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Hauppauge", + .product_name = "HVR-950Q", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_AUDIO_ALIGN_TRANSFER, + } +}, +{ + USB_DEVICE_VENDOR_SPEC(0x2040, 0x7201), + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Hauppauge", + .product_name = "HVR-950Q", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_AUDIO_ALIGN_TRANSFER, + } +}, +{ + USB_DEVICE_VENDOR_SPEC(0x2040, 0x7202), + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Hauppauge", + .product_name = "HVR-950Q", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_AUDIO_ALIGN_TRANSFER, + } +}, +{ + USB_DEVICE_VENDOR_SPEC(0x2040, 0x7203), + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Hauppauge", + .product_name = "HVR-950Q", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_AUDIO_ALIGN_TRANSFER, + } +}, +{ + USB_DEVICE_VENDOR_SPEC(0x2040, 0x7204), + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Hauppauge", + .product_name = "HVR-950Q", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_AUDIO_ALIGN_TRANSFER, + } +}, +{ + USB_DEVICE_VENDOR_SPEC(0x2040, 0x7205), + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Hauppauge", + .product_name = "HVR-950Q", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_AUDIO_ALIGN_TRANSFER, + } +}, +{ + USB_DEVICE_VENDOR_SPEC(0x2040, 0x7250), + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Hauppauge", + .product_name = "HVR-950Q", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_AUDIO_ALIGN_TRANSFER, + } +}, +{ + USB_DEVICE_VENDOR_SPEC(0x2040, 0x7230), + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Hauppauge", + .product_name = "HVR-850", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_AUDIO_ALIGN_TRANSFER, + } +}, + { /* * Some USB MIDI devices don't have an audio control interface, -- 1.6.3.3

1 0

[alsa-devel] [PATCH 2/3] sound: usb-audio: relax urb data align. restriction HVR-950Q and HVR-850 only
by johnsgruber＠gmail.com 27 Dec '09

27 Dec '09

From: John S. Gruber <JohnSGruber(a)gmail.com> Addressing audio quality problem. In sound/usb/usbaudio.c, for the Hauppage HVR-950Q and HVR-850 only, change retire_capture_urb to allow transfers on audio sub-slot boundaries rather than audio slots boundaries. With these devices the left and right channel samples can be split between two different urbs. Throwing away extra channel samples causes a sound quality problem for stereo streams as the left and right channels are swapped repeatedly, perhaps many times per second. Urbs unaligned on sub-slot boundaries are still truncated to the next lowest stride (audio slot) to retain synchronization on samples even though left/right channel synchronization may be lost in this case. Detect the quirk using a case statement in snd_usb_audio_probe. BugLink: https://bugs.launchpad.net/ubuntu/+bug/495745 Signed-off-by: John S. Gruber <JohnSGruber(a)gmail.com> --- sound/usb/usbaudio.c | 29 +++++++++++++++++++++++++++-- sound/usb/usbaudio.h | 1 + 2 files changed, 28 insertions(+), 2 deletions(-) diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 1ba8f41..be6f017 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -169,6 +169,7 @@ struct snd_usb_substream { unsigned int curpacksize; /* current packet size in bytes (for capture) */ unsigned int curframesize; /* current packet size in frames (for capture) */ unsigned int fill_max: 1; /* fill max packet size always */ + unsigned int txfr_quirk:1; /* allow sub-frame alignment */ unsigned int fmt_type; /* USB audio format type (1-3) */ unsigned int running: 1; /* running status */ @@ -353,14 +354,23 @@ static int retire_capture_urb(struct snd_usb_substream *subs, snd_printd(KERN_ERR "frame %d active: %d\n", i, urb->iso_frame_desc[i].status); // continue; } - frames = urb->iso_frame_desc[i].actual_length / stride; - bytes = frames * stride; + bytes = urb->iso_frame_desc[i].actual_length; + frames = bytes / stride; + if (!subs->txfr_quirk) + bytes = frames * stride; + if (bytes % (runtime->sample_bits >> 3) != 0) { + int oldbytes = bytes; + bytes = frames * stride; + snd_printdd(KERN_ERR "Corrected urb data len. %d->%d\n", + oldbytes, bytes); + } /* update the current pointer */ spin_lock_irqsave(&subs->lock, flags); oldptr = subs->hwptr_done; subs->hwptr_done += bytes; if (subs->hwptr_done >= runtime->buffer_size * stride) subs->hwptr_done -= runtime->buffer_size * stride; + frames = (bytes + (oldptr % stride)) / stride; subs->transfer_done += frames; if (subs->transfer_done >= runtime->period_size) { subs->transfer_done -= runtime->period_size; @@ -2189,6 +2199,7 @@ static void init_substream(struct snd_usb_stream *as, int stream, struct audiofo subs->stream = as; subs->direction = stream; subs->dev = as->chip->dev; + subs->txfr_quirk = as->chip->txfr_quirk; if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL) { subs->ops = audio_urb_ops[stream]; } else { @@ -3569,6 +3580,20 @@ static void *snd_usb_audio_probe(struct usb_device *dev, } } + switch (chip->usb_id) { + case USB_ID(0x2040, 0x7200): /* Hauppage hvr950Q */ + case USB_ID(0x2040, 0x7221): /* Hauppage hvr950Q */ + case USB_ID(0x2040, 0x7222): /* Hauppage hvr950Q */ + case USB_ID(0x2040, 0x7223): /* Hauppage hvr950Q */ + case USB_ID(0x2040, 0x7224): /* Hauppage hvr950Q */ + case USB_ID(0x2040, 0x7225): /* Hauppage hvr950Q */ + case USB_ID(0x2040, 0x7230): /* Hauppage hvr850 */ + case USB_ID(0x2040, 0x7250): /* Hauppage hvr950Q */ + chip->txfr_quirk = 1; + break; + default: + chip->txfr_quirk = 0; + } err = 1; /* continue */ if (quirk && quirk->ifnum != QUIRK_NO_INTERFACE) { /* need some special handlings */ diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 9826337..228da06 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -125,6 +125,7 @@ struct snd_usb_audio { struct snd_card *card; u32 usb_id; int shutdown; + unsigned int txfr_quirk:1; /* Subframe boundaries on transfers */ int num_interfaces; int num_suspended_intf; -- 1.6.3.3

1 0

[alsa-devel] [PATCH 1/3] sound: usb-audio: make buffer pointer based on bytes instead on frames
by johnsgruber＠gmail.com 27 Dec '09

27 Dec '09

From: Clemens Ladisch <clemens(a)ladisch.de> Since there are devices that do not align the size of their data packets to frame boundaries, the driver needs to be able to keep track of partial frames. This patch prepares for support for such devices by changing the hwptr_done variable from a frame counter to a byte counter. Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> --- sound/usb/usbaudio.c | 76 ++++++++++++++++++++++++------------------------- 1 files changed, 37 insertions(+), 39 deletions(-) diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 31b63ea..1ba8f41 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -173,7 +173,7 @@ struct snd_usb_substream { unsigned int running: 1; /* running status */ - unsigned int hwptr_done; /* processed frame position in the buffer */ + unsigned int hwptr_done; /* processed byte position in the buffer */ unsigned int transfer_done; /* processed frames since last period update */ unsigned long active_mask; /* bitmask of active urbs */ unsigned long unlink_mask; /* bitmask of unlinked urbs */ @@ -342,7 +342,7 @@ static int retire_capture_urb(struct snd_usb_substream *subs, unsigned long flags; unsigned char *cp; int i; - unsigned int stride, len, oldptr; + unsigned int stride, frames, bytes, oldptr; int period_elapsed = 0; stride = runtime->frame_bits >> 3; @@ -353,29 +353,28 @@ static int retire_capture_urb(struct snd_usb_substream *subs, snd_printd(KERN_ERR "frame %d active: %d\n", i, urb->iso_frame_desc[i].status); // continue; } - len = urb->iso_frame_desc[i].actual_length / stride; - if (! len) - continue; + frames = urb->iso_frame_desc[i].actual_length / stride; + bytes = frames * stride; /* update the current pointer */ spin_lock_irqsave(&subs->lock, flags); oldptr = subs->hwptr_done; - subs->hwptr_done += len; - if (subs->hwptr_done >= runtime->buffer_size) - subs->hwptr_done -= runtime->buffer_size; - subs->transfer_done += len; + subs->hwptr_done += bytes; + if (subs->hwptr_done >= runtime->buffer_size * stride) + subs->hwptr_done -= runtime->buffer_size * stride; + subs->transfer_done += frames; if (subs->transfer_done >= runtime->period_size) { subs->transfer_done -= runtime->period_size; period_elapsed = 1; } spin_unlock_irqrestore(&subs->lock, flags); /* copy a data chunk */ - if (oldptr + len > runtime->buffer_size) { - unsigned int cnt = runtime->buffer_size - oldptr; - unsigned int blen = cnt * stride; - memcpy(runtime->dma_area + oldptr * stride, cp, blen); - memcpy(runtime->dma_area, cp + blen, len * stride - blen); + if (oldptr + bytes > runtime->buffer_size * stride) { + unsigned int bytes1 = + runtime->buffer_size * stride - oldptr; + memcpy(runtime->dma_area + oldptr, cp, bytes1); + memcpy(runtime->dma_area, cp + bytes1, bytes - bytes1); } else { - memcpy(runtime->dma_area + oldptr * stride, cp, len * stride); + memcpy(runtime->dma_area + oldptr, cp, bytes); } } if (period_elapsed) @@ -562,24 +561,24 @@ static int prepare_playback_urb(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime, struct urb *urb) { - int i, stride, offs; - unsigned int counts; + int i, stride; + unsigned int counts, frames, bytes; unsigned long flags; int period_elapsed = 0; struct snd_urb_ctx *ctx = urb->context; stride = runtime->frame_bits >> 3; - offs = 0; + frames = 0; urb->dev = ctx->subs->dev; /* we need to set this at each time */ urb->number_of_packets = 0; spin_lock_irqsave(&subs->lock, flags); for (i = 0; i < ctx->packets; i++) { counts = snd_usb_audio_next_packet_size(subs); /* set up descriptor */ - urb->iso_frame_desc[i].offset = offs * stride; + urb->iso_frame_desc[i].offset = frames * stride; urb->iso_frame_desc[i].length = counts * stride; - offs += counts; + frames += counts; urb->number_of_packets++; subs->transfer_done += counts; if (subs->transfer_done >= runtime->period_size) { @@ -589,7 +588,7 @@ static int prepare_playback_urb(struct snd_usb_substream *subs, if (subs->transfer_done > 0) { /* FIXME: fill-max mode is not * supported yet */ - offs -= subs->transfer_done; + frames -= subs->transfer_done; counts -= subs->transfer_done; urb->iso_frame_desc[i].length = counts * stride; @@ -599,7 +598,7 @@ static int prepare_playback_urb(struct snd_usb_substream *subs, if (i < ctx->packets) { /* add a transfer delimiter */ urb->iso_frame_desc[i].offset = - offs * stride; + frames * stride; urb->iso_frame_desc[i].length = 0; urb->number_of_packets++; } @@ -609,26 +608,25 @@ static int prepare_playback_urb(struct snd_usb_substream *subs, if (period_elapsed) /* finish at the period boundary */ break; } - if (subs->hwptr_done + offs > runtime->buffer_size) { + bytes = frames * stride; + if (subs->hwptr_done + bytes > runtime->buffer_size * stride) { /* err, the transferred area goes over buffer boundary. */ - unsigned int len = runtime->buffer_size - subs->hwptr_done; + unsigned int bytes1 = + runtime->buffer_size * stride - subs->hwptr_done; memcpy(urb->transfer_buffer, - runtime->dma_area + subs->hwptr_done * stride, - len * stride); - memcpy(urb->transfer_buffer + len * stride, - runtime->dma_area, - (offs - len) * stride); + runtime->dma_area + subs->hwptr_done, bytes1); + memcpy(urb->transfer_buffer + bytes1, + runtime->dma_area, bytes - bytes1); } else { memcpy(urb->transfer_buffer, - runtime->dma_area + subs->hwptr_done * stride, - offs * stride); + runtime->dma_area + subs->hwptr_done, bytes); } - subs->hwptr_done += offs; - if (subs->hwptr_done >= runtime->buffer_size) - subs->hwptr_done -= runtime->buffer_size; - runtime->delay += offs; + subs->hwptr_done += bytes; + if (subs->hwptr_done >= runtime->buffer_size * stride) + subs->hwptr_done -= runtime->buffer_size * stride; + runtime->delay += frames; spin_unlock_irqrestore(&subs->lock, flags); - urb->transfer_buffer_length = offs * stride; + urb->transfer_buffer_length = bytes; if (period_elapsed) snd_pcm_period_elapsed(subs->pcm_substream); return 0; @@ -901,18 +899,18 @@ static int wait_clear_urbs(struct snd_usb_substream *subs) /* - * return the current pcm pointer. just return the hwptr_done value. + * return the current pcm pointer. just based on the hwptr_done value. */ static snd_pcm_uframes_t snd_usb_pcm_pointer(struct snd_pcm_substream *substream) { struct snd_usb_substream *subs; - snd_pcm_uframes_t hwptr_done; + unsigned int hwptr_done; subs = (struct snd_usb_substream *)substream->runtime->private_data; spin_lock(&subs->lock); hwptr_done = subs->hwptr_done; spin_unlock(&subs->lock); - return hwptr_done; + return hwptr_done / (substream->runtime->frame_bits >> 3); } -- 1.6.3.3

1 0

[alsa-devel] CONFIG_GCD is never defined in the latest alsa-driver
by Ozan Çağlayan 26 Dec '09

26 Dec '09

Hi, Compiling the latest alsa-driver with kernel > 2.6.31 results in gcd() symbol duplication. Although there's a check in configure for linux/gcd.h, no boolean is adjusted according to that. In acore/wrappers.c, gcd() is exported if CONFIG_GCD is not defined which seems always the case for now. Something like the following fixes the problem for me: Index: alsa-driver/acore/wrappers.c =================================================================== --- alsa-driver.orig/acore/wrappers.c +++ alsa-driver/acore/wrappers.c @@ -339,7 +339,7 @@ char *compat_skip_spaces(const char *str EXPORT_SYMBOL(compat_skip_spaces); #endif /* < 2.6.33 */ -#ifndef CONFIG_GCD +#if LINUX_VERSION_CODE < KERNEL_VERSION(2, 6, 31) /* Greatest common divisor */ unsigned long gcd(unsigned long a, unsigned long b) { @@ -356,4 +356,4 @@ unsigned long gcd(unsigned long a, unsig return b; } EXPORT_SYMBOL(gcd); -#endif /* !CONFIG_GCD */ +#endif Thanks!

2 3

[PATCH] Don't define gcd() when already exists
by Takashi Iwai 26 Dec '09

26 Dec '09

Define compatible gcd() only when linux/gcd. doesn't exist. CONFIG_GCD isn't defined for 2.6.31/32, so it can'be used reliablty as the compile condition. Reported-by: Ozan Çağlayan <ozan(a)pardus.org.tr> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- acore/wrappers.c | 3 +++ include/gcd_compat.h | 1 + 2 files changed, 4 insertions(+), 0 deletions(-) diff --git a/acore/wrappers.c b/acore/wrappers.c index e9478bd..68750a7 100644 --- a/acore/wrappers.c +++ b/acore/wrappers.c @@ -340,6 +340,8 @@ EXPORT_SYMBOL(compat_skip_spaces); #endif /* < 2.6.33 */ #ifndef CONFIG_GCD +#include <linux/gcd.h> +#ifdef CONFIG_SND_COMPAT_GCD /* Greatest common divisor */ unsigned long gcd(unsigned long a, unsigned long b) { @@ -356,4 +358,5 @@ unsigned long gcd(unsigned long a, unsigned long b) return b; } EXPORT_SYMBOL(gcd); +#endif /* CONFIG_SND_COMPAT_GCD */ #endif /* !CONFIG_GCD */ diff --git a/include/gcd_compat.h b/include/gcd_compat.h index beeda05..daedab1 100644 --- a/include/gcd_compat.h +++ b/include/gcd_compat.h @@ -1,6 +1,7 @@ #ifndef _LINUX_GCD_H #define _LINUX_GCD_H +#define CONFIG_SND_COMPAT_GCD unsigned long gcd(unsigned long a, unsigned long b); #endif /* _LINUX_GCD_H */ -- 1.6.5.7

1 0

[alsa-devel] [PATCH] sound/arm: Fix build failure caused by missing struct aaci definition
by Peter Huewe 26 Dec '09

26 Dec '09

From: Peter Huewe <peterhuewe(a)gmx.de> This patch fixes a build failure introduced by the patch ALSA: AACI: factor common hw_params logic into aaci_pcm_hw_params [1] by adding/moving the aaci struct to the right position. The patch mentioned above merged common source parts into one function, but unfortunately left out the aaci struct and consequently caused a build failure e.g. for arm versatile_config [2] References: [1] http://git.kernel.org/?p=linux/kernel/git/torvalds/linux-2.6.git;a=commitdi… [2] http://kisskb.ellerman.id.au/kisskb/buildresult/1893605/ Patch against Linus' tree. Signed-off-by: Peter Huewe <peterhuewe(a)gmx.de> --- sound/arm/aaci.c | 3 +-- 1 files changed, 1 insertions(+), 2 deletions(-) diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index c569986..656e474 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -441,6 +441,7 @@ static int aaci_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { int err; + struct aaci *aaci = substream->private_data; aaci_pcm_hw_free(substream); if (aacirun->pcm_open) { @@ -560,7 +561,6 @@ static int aaci_pcm_open(struct snd_pcm_substream *substream) static int aaci_pcm_playback_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct aaci *aaci = substream->private_data; struct aaci_runtime *aacirun = substream->runtime->private_data; unsigned int channels = params_channels(params); int ret; @@ -659,7 +659,6 @@ static struct snd_pcm_ops aaci_playback_ops = { static int aaci_pcm_capture_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct aaci *aaci = substream->private_data; struct aaci_runtime *aacirun = substream->runtime->private_data; int ret; -- 1.6.4.4

2 1

Re: [alsa-devel] [PATCH] Recognise and use subdevice 0x3061 found in some HP dv6 notebooks.
by Kunal Gangakhedkar 26 Dec '09

26 Dec '09

On Wednesday 23 Dec 2009 12:02:47 am Kunal Gangakhedkar wrote: > On Monday 14 Dec 2009 4:02:58 pm Takashi Iwai wrote: > > At Mon, 14 Dec 2009 15:53:54 +0530, > > Kunal Gangakhedkar wrote: > > > > > > On Monday 14 Dec 2009 3:37:14 pm Takashi Iwai wrote: > > > > At Mon, 14 Dec 2009 00:29:42 +0530, > > > > Kunal Gangakhedkar wrote: > > > > > > > > > > Obvious patch. > > > > > Recently bought an HP Pavilion dv6-1110ax notebook. The modparm 'model=hp-dv5' > > > > > works, but I thought it was better if it were natively supported. > > > > > > > > > > Also, the 'hp_detect=1' hint > > > > > (echo "hp_detect = 1" > /sys/class/sound/hwC0D0/hints) works nicely to mute the > > > > > speakers when using the headphone jack(s). Is there any way to automatically > > > > > enable this? > > > > > > > > This is weird. hp_detect is basically always set unless you clear > > > > explicitly. I guess it's simply a known IRQ problem on HP dv > > > > laptops. It requires MSI. > > > > > > > > Anyway, could you test the very latest kernel without your patch and > > > > confirm that the change is still needed? > > > > > > > > > > The patch wasn't about hp_detect. It was to get the subdev recognised natively so that > > > you don't need the 'model=hp-dv5' modparm. > > > > Yeah, I know. But the speaker-muting should work even without > > hp_detect=1 hints. The IRQ thingy was only about the auto-muting. > > > > Also, with the latest version, the device might work even without the > > extra model option (as the auto-parser got improved). I'd like to > > avoid unneeded quirk lines as much as possible, so that's why I asked > > to check the latest one. > > > > > I'll try with enable_msi modparm as well without trying the hp_detect hint and let you know > > > about it. Will also try the latest kernel. > > > > Thanks! > > > > > > Takashi > > > > Sorry for the late reply - was busy with office work. > Just built the kernel from your git (sound-2.6.git) master > (as of commit 07e23a669544942a9154b64c583bdef014f66a15). > > This new driver didn't require 'model=hp-dv5' modparm - it did detect and > use my card properly; so, my patch is not needed anymore, I guess :) > > However, the hp_detect=1 problem still persists :(. > I had to use the "echo 'hp_detect=1' > /sys/class/sound/hwC0D0/hints" > method to get the auto-muting to work. > It's also true with enable_msi=1 modparm. > > Anything more you want me to try? > > Thanks, > Kunal > Also, "spec->gpio_led = 0x08" in patch_sigmatel.c seems to enable the colour changing (orange/white) of the mute/unmute button. I got this info from ubuntu's lucid kernel which somehow enabled the behaviour. But, it was missing from the sound-2.6.git master kernel that I built. I'll do more digging to see why it isn't working in the git kernel. This is the code that I copied from lucid kernel into linux-backports-modules that enabled the led behaviour: if ((codec->subsystem_id >> 16) == PCI_VENDOR_ID_HP) { const struct dmi_device *dev = NULL; while ((dev = dmi_find_device(DMI_DEV_TYPE_OEM_STRING, NULL, dev))) { if (strcmp(dev->name, "HP_Mute_LED_1")) { switch (codec->vendor_id) { case 0x111d7608: spec->gpio_led = 0x01; break; case 0x111d7600: case 0x111d7601: case 0x111d7602: case 0x111d7603: /* <-- My codec STAC 92HD75B3X5 */ spec->gpio_led = 0x08; break; } break; } } } Thanks, Kunal

2 4

[alsa-devel] [PATCH] jazz16: refine dma and irq selection
by Krzysztof Helt 25 Dec '09

25 Dec '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Narrow the dma and irq selection after the DOS driver. Add ALSA configuration description as well. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- The dma8 must be 8-bit otherwise the 8-bit sound played through 16-bit dma has metalic noise. The 8-bit DMA cannot play 16-bit sound. Documentation/sound/alsa/ALSA-Configuration.txt | 15 +++++++++++++++ sound/isa/sb/jazz16.c | 21 ++++++++++++++++++++- sound/isa/sb/sb8_main.c | 3 ++- 3 files changed, 37 insertions(+), 2 deletions(-) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 8923597..92b0975 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -1123,6 +1123,21 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. This module supports multiple cards, autoprobe and ISA PnP. + Module snd-jazz16 + ------------------- + + Module for Media Vision Jazz16 chipset. The chipset consists of 3 chips: + MVD1216 + MVA416 + MVA514. + + port - port # for SB DSP chip (0x210,0x220,0x230,0x240,0x250,0x260) + irq - IRQ # for SB DSP chip (3,5,7,9,10,15) + dma8 - DMA # for SB DSP chip (1,3) + dma16 - DMA # for SB DSP chip (5,7) + mpu_port - MPU-401 port # (0x300,0x310,0x320,0x330) + mpu_irq - MPU-401 irq # (2,3,5,7) + + This module supports multiple cards. + Module snd-korg1212 ------------------- diff --git a/sound/isa/sb/jazz16.c b/sound/isa/sb/jazz16.c index d52966b..8d21a3f 100644 --- a/sound/isa/sb/jazz16.c +++ b/sound/isa/sb/jazz16.c @@ -189,10 +189,29 @@ static int __devinit snd_jazz16_match(struct device *devptr, unsigned int dev) if (port[dev] == SNDRV_AUTO_PORT) { snd_printk(KERN_ERR "please specify port\n"); return 0; + } else if (port[dev] == 0x200 || (port[dev] & ~0x270)) { + snd_printk(KERN_ERR "incorrect port specified\n"); + return 0; + } + if (dma8[dev] != SNDRV_AUTO_DMA && + dma8[dev] != 1 && dma8[dev] != 3) { + snd_printk(KERN_ERR "dma8 must be 1 or 3\n"); + return 0; } if (dma16[dev] != SNDRV_AUTO_DMA && dma16[dev] != 5 && dma16[dev] != 7) { - snd_printk(KERN_ERR "dma16 must be 5 or 7"); + snd_printk(KERN_ERR "dma16 must be 5 or 7\n"); + return 0; + } + if (mpu_port[dev] != SNDRV_AUTO_PORT && + (mpu_port[dev] & ~0x030) != 0x300) { + snd_printk(KERN_ERR "incorrect mpu_port specified\n"); + return 0; + } + if (mpu_irq[dev] != SNDRV_AUTO_DMA && + mpu_irq[dev] != 2 && mpu_irq[dev] != 3 && + mpu_irq[dev] != 5 && mpu_irq[dev] != 7) { + snd_printk(KERN_ERR "mpu_irq must be 2, 3, 5 or 7\n"); return 0; } return 1; diff --git a/sound/isa/sb/sb8_main.c b/sound/isa/sb/sb8_main.c index 3222aed..7d84c9f 100644 --- a/sound/isa/sb/sb8_main.c +++ b/sound/isa/sb/sb8_main.c @@ -510,7 +510,8 @@ static int snd_sb8_open(struct snd_pcm_substream *substream) } switch (chip->hardware) { case SB_HW_JAZZ16: - runtime->hw.formats |= SNDRV_PCM_FMTBIT_S16_LE; + if (chip->dma16 == 5 || chip->dma16 == 7) + runtime->hw.formats |= SNDRV_PCM_FMTBIT_S16_LE; runtime->hw.rates |= SNDRV_PCM_RATE_8000_48000; runtime->hw.rate_min = 4000; runtime->hw.rate_max = 50000; -- 1.6.4 ---------------------------------------------------------------------- Sprawdz sekrety swojego biorytmu. Kliknij >> http://link.interia.pl/f2534

2 1

[alsa-devel] [PATCH] hda - HDMI sticky stream tag support
by Wu Fengguang 25 Dec '09

25 Dec '09

When we run the following commands in turn (with CONFIG_SND_HDA_POWER_SAVE_DEFAULT=0), speaker-test -Dhw:0,3 -c2 -twav # HDMI speaker-test -Dhw:0,0 -c2 -twav # Analog The second command will produce sound in the analog lineout _as well as_ HDMI sink. The root cause is, device 0 "reuses" the same stream tag that was used by device 3, and the "intelhdmi - sticky stream id" patch leaves the HDMI codec in a functional state. So the HDMI codec happily accepts the audio samples which reuse its stream tag. The proposed solution is to remember the last device each azx_dev was assigned to, and prefer to 1) reuse the azx_dev (and hence the stream tag) the HDMI codec last used 2) or assign a never-used azx_dev for HDMI With this patch and the above two speaker-test commands, HDMI codec will use stream tag 8 and Analog codec will use 5. The stream tag used by HDMI codec won't be reused by others, as long as we don't run out of the 4 playback azx_dev's. The legacy Analog codec will continue to use stream tag 5 because its device id is 0 (this is a bit tricky). Signed-off-by: Wu Fengguang <fengguang.wu(a)intel.com> --- sound/pci/hda/hda_intel.c | 21 +++++++++++++++------ 1 file changed, 15 insertions(+), 6 deletions(-) --- sound-2.6.orig/sound/pci/hda/hda_intel.c 2009-12-25 12:17:28.000000000 +0800 +++ sound-2.6/sound/pci/hda/hda_intel.c 2009-12-25 12:41:02.000000000 +0800 @@ -356,6 +356,7 @@ struct azx_dev { */ unsigned char stream_tag; /* assigned stream */ unsigned char index; /* stream index */ + int device; /* last device number assigned to */ unsigned int opened :1; unsigned int running :1; @@ -1441,10 +1442,13 @@ static int __devinit azx_codec_configure */ /* assign a stream for the PCM */ -static inline struct azx_dev *azx_assign_device(struct azx *chip, int stream) +static inline struct azx_dev * +azx_assign_device(struct azx *chip, struct snd_pcm_substream *substream) { int dev, i, nums; - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + struct azx_dev *res = NULL; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { dev = chip->playback_index_offset; nums = chip->playback_streams; } else { @@ -1453,10 +1457,15 @@ static inline struct azx_dev *azx_assign } for (i = 0; i < nums; i++, dev++) if (!chip->azx_dev[dev].opened) { - chip->azx_dev[dev].opened = 1; - return &chip->azx_dev[dev]; + res = &chip->azx_dev[dev]; + if (res->device == substream->pcm->device) + break; } - return NULL; + if (res) { + res->opened = 1; + res->device = substream->pcm->device; + } + return res; } /* release the assigned stream */ @@ -1505,7 +1514,7 @@ static int azx_pcm_open(struct snd_pcm_s int err; mutex_lock(&chip->open_mutex); - azx_dev = azx_assign_device(chip, substream->stream); + azx_dev = azx_assign_device(chip, substream); if (azx_dev == NULL) { mutex_unlock(&chip->open_mutex); return -EBUSY;

2 1

Re: [alsa-devel] [PATCH] Fix indentation in pcm_native.c
by Takashi Iwai 25 Dec '09

25 Dec '09

At Thu, 24 Dec 2009 01:17:46 +0100 (CET), Guennadi Liakhovetski wrote: > > Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> Applied now. Thanks. Takashi > --- > diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c > index ab73edf..cf57964 100644 > --- a/sound/core/pcm_native.c > +++ b/sound/core/pcm_native.c > @@ -1917,13 +1917,13 @@ int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream) > > err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_RATE, > hw->rate_min, hw->rate_max); > - if (err < 0) > - return err; > + if (err < 0) > + return err; > > err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, > hw->period_bytes_min, hw->period_bytes_max); > - if (err < 0) > - return err; > + if (err < 0) > + return err; > > err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIODS, > hw->periods_min, hw->periods_max); >

1 0

[alsa-devel] Personal invitation from Kunal Narkhede
by Kunal Narkhede 24 Dec '09

24 Dec '09

If you are not able to click these links, click the following: « Show content » and try again. E-mail sent 12/24/2009 9:01:58 AM, by Kunal Narkhede: Hello, Finally, a smart and simple way to manage your contacts! With UNYK, I put all my contacts together in <b>one address book</b> that is <b>automatically updated</b>. One of my contacts changes his or her information at UNYK.com: My address book is updated. I change information at UNYK.com: My contacts address books are updated. Simple, but life-changing! Can I add you as one of my contacts? To accept, click here! http://www.unyk.com/ml/245/1/?i=C7B72DD0179E4FD7A6DED23A2976A060&s=17267503 You too can create your own smart address book. Kunal Narkhede UNYK : ESJ 288 Please do not reply to this email directly. Questions? Contact us - http://unyk.com/en/Contact UNYK Inc., 1010 de Serigny, suite 820, Longueuil, QC, J4K 5G7, Canada. I no longer wish to receive invitations to join UNYK from Kunal Narkhede: Unsubscribe http://www.unyk.com/ml/245/74/?mid=9E98A4AA807983A0&email=alsa-devel@alsa-p… I never again wish to receive invitation to join UNYK: Unsubscribe http://www.unyk.com/ml/245/2/?i=C7B72DD0179E4FD7A6DED23A2976A060&s=17267503 UNYK, the first smart address book that updates itself!

1 0

Re: [alsa-devel] [PATCH] ASoC: fix params_rate() macro use in several codecs
by Mark Brown 24 Dec '09

24 Dec '09

On Thu, Dec 24, 2009 at 01:13:51AM +0100, Guennadi Liakhovetski wrote: > Sevelar ASoC codec drivers wrongly assume, that the params_rate() macro > returns one of SNDRV_PCM_RATE_* defines instead of the actual numerical > sampling rate. Fix them. > > Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> Applied, thanks.

1 0

Re: [alsa-devel] [PATCH] ASoC: fix params_rate() macro use in several codecs
by Liam Girdwood 24 Dec '09

24 Dec '09

On Thu, 2009-12-24 at 01:13 +0100, Guennadi Liakhovetski wrote: > Sevelar ASoC codec drivers wrongly assume, that the params_rate() macro > returns one of SNDRV_PCM_RATE_* defines instead of the actual numerical > sampling rate. Fix them. > > Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> > --- Acked-by: Liam Girdwood <lrg(a)slimlogic.co.uk>

1 0

[alsa-devel] need help with io plugin programming: how to add delay ?
by Stefan Schoenleitner 23 Dec '09

23 Dec '09

Hi, I would like to write a simple alsa io plugin that takes some amount of PCM samples and sends them to a socket in a timely manner (this would be playback). Later on it should work the other way round as well (recording). (Well, actually I have been working on this for a month already but so far I still did not manage to get a working solution.) However, if the PCM stream comes from something prerecorded (i.e. a wav file) there needs to be some kind of delay. If for example the wav file would have a sampling rate of 8 kHz, 16 bit resolution and one period should consist of 160 samples, it would be necessary to write on period exactly each 20ms. For this reason delay needs to be added somewhere so that each 20ms exactly one period is begin sent to the socket. Otherwise all the PCM samples would be just immediately written to the socket without any delay. I read a lot of other alsa plugin code but still I have found no way to properly solve the delay problem. * Where and how do I need to add delay ?? My first approach was to add delay by sleeping in the transfer callback function. After I had implemented that solution (and it did not work that well) I was told that this is not the proper way to do it. * What is the proper way then ? I was told that the alsa plugin framework is poll based and I should come up with a poll based solution. However, I can not just let the alsa io plugin poll for POLLOUT on the socket as this would not add any delay at all. (It would just wait until new data can be written to the socket.) After looking at the bluetooth alsa plugin I came up with another idea (and my second approach): Basically it should be possible to create an independent timing thread in the alsa plugin that writes dummy data (e.g. a single byte) to a pipe each period (thus in my case each 20ms). The plugin would then poll for POLLIN on the above mentioned pipe and poll for POLLOUT on the socket at the same time. This way the transfer callback function would be called if and only if the following two conditions are met: * 20 ms period time has passed * the socket can accept new data * Would this solution be the proper way ? Another solution (my third one already) would be to introduce *no* delay in the alsa plugin itself, but do everything in the application that is listening on the socket. For this reason the application would read one period, process the data and then sleep for the rest of the time until the time for one period (i.e. 20ms) is over. After that the next period is read from the socket and so forth. The alsa plugin would then "automatically" write new data to the socket each time the application listening to that socket reads data and there is free space in the socket send buffer. * Once again, would this be a proper solution ? * If so, how do I know when to increment the hardware pointer ? (- After the send() call in the transfer callback function ? - Using an independent thread in the plugin that just increments the hardware pointer each 20ms ?) - something even more complex ?) My guess is that the third solution (delay is added in the application that listens to the socket) might work well as the delay is added at the end of the "audio processing chain" which makes it less sensitive to delays that are introduced in between. * Is this correct ? I would appreciate any help as I have invested way too much time in this already. cheers, stefan

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[alsa-devel] ASoC update for 2.6.33
by Mark Brown 23 Dec '09

23 Dec '09

The following changes since commit 1628af5adf64cc2960bce81009f119de822f876e: Guennadi Liakhovetski (1): ASoC: add missing parameter to mx27vis_hifi_hw_free() are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.33 Eric Millbrandt (1): ASoC: Do not write to invalid registers on the wm9712. sound/soc/codecs/wm9712.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-)

2 1

Re: [alsa-devel] Mic not working on Dell Vostro 1700 by default
by Takashi Iwai 23 Dec '09

23 Dec '09

At Wed, 23 Dec 2009 17:47:21 +0100, Anisse Astier wrote: > > On Wed, 23 Dec 2009 17:28:45 +0100, Anisse Astier <anisse(a)astier.eu> wrote : > > > > > I was confused previously, and i didn't know a pci device could have a > > subsystem vendor/device with id's. > > > > I cleared my head, and did this patch. It's on top of alsa-kernel tree. > > > > Compiling now, I don't see why it wouldn't work, I'll give you an update > > ASAP. > > I works as expected. OK, applied now. Thanks. Takashi

1 0

[alsa-devel] [PATCH 1/1] Do not write to invalid registers on the wm9712.
by Eric Millbrandt 23 Dec '09

23 Dec '09

This patch fixes a bug where "virtual" registers were being written to the ac97 bus. This was causing unrelated registers to become corrupted (headphone 0x04, touchscreen 0x78, etc). This patch duplicates protection that was included in the wm9713 driver. Signed-off-by: Eric Millbrandt <emillbrandt(a)dekaresearch.com> --- sound/soc/codecs/wm9712.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 0ac1215..e237bf6 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -463,7 +463,8 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, { u16 *cache = codec->reg_cache; - soc_ac97_ops.write(codec->ac97, reg, val); + if (reg < 0x7c) + soc_ac97_ops.write(codec->ac97, reg, val); reg = reg >> 1; if (reg < (ARRAY_SIZE(wm9712_reg))) cache[reg] = val; -- 1.6.3.1 This e-mail and the information, including any attachments, it contains are intended to be a confidential communication only to the person or entity to whom it is addressed and may contain information that is privileged. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please immediately notify the sender and destroy the original message. Thank you. Please consider the environment before printing this email.

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Re: [alsa-devel] Mic not working on Dell Vostro 1700 by default
by Takashi Iwai 23 Dec '09

23 Dec '09

At Wed, 23 Dec 2009 14:51:25 +0100, Anisse Astier wrote: > > On Wed, 23 Dec 2009 08:31:40 +0100, Takashi Iwai <tiwai(a)suse.de> wrote : > > > At Tue, 22 Dec 2009 18:50:28 +0100, > > Anisse Astier wrote: > > > > > > On Tue, 22 Dec 2009 18:44:04 +0100, Anisse Astier <anisse(a)astier.eu> wrote : > > > > > > > Hi, > > > > > > > > (i'm not subscribed to the list, please cc me in replies) > > > > > > > > So I'm on a Dell Vostro 1700 with latest BIOS (A07). In the default > > > > configuration, the mic is not working. > > > > > > > > But if I append model=dell-m42 when loading snd-hda-intel, it works. > > > > > > > > My alsa-info: > > > > http://www.alsa-project.org/db/?f=8d6f7d613b909c0eef67f93e0e86bf13476e9e54 > > > > > > > > alsa-info with option model=dell-m42 > > > > http://www.alsa-project.org/db/?f=233cee7ffea01b9f46451e64c95c4fc482ab4ee4 > > > > > > > > I don't know if dell-m42 is enough, I chose it because it was used for the > > > > Dell Vostro 1500, for which there was a pci quirk in patch_sigmatel.c, > > > > and that seemed close enough. Everything seems to work properly though. > > > > > > > > I don't know if it's possible to add the same pci-quirk (or a similar one) > > > > in patch_sigmatel.c , or any dmi-based quirk, but I'd like to help in > > > > doing so. > > > > > > > > I compiled a working hda-verb should some remote debugging be needed. > > > > > > > > > > I forgot to add that this is also working with dell-m44. A pointer: > > > https://wiki.ubuntu.com/InstallingUbuntuOnADellVostro1700#AUDIO : > > > suggestion to add this option by default on ubuntu. > > > > So, which is better, m42 or m44? > > This can be checked only with a real hardware... > > > > I did some tests today, and I'm unable to tell. Both behave the same with > external headphones, internal speakers, external mic, internal mic. Jack > detection works flawlessly with both. From time to time the codec will > timeout and driver switch to polling mode, but it's independent from the > option model. > I really don't know what else to test. Maybe the different is either the digital I/O or the docking station. > But IMO, we should go with m42. According to patch_sigmatel.c dell-m42 is > made for Vostro 1500, and this should be tho closest to the 1700. It > seems they have the same motherboard. I attached a diff between the two > laptops' lspci. Only the graphics card changes. OK, then let's take dell-m42. thanks, Takashi

1 0

Re: [alsa-devel] [2.6.33-rc1] [SOUND] HDA: high cpu usage, noise
by Takashi Iwai 23 Dec '09

23 Dec '09

At Wed, 23 Dec 2009 15:27:08 +0100, Éric Piel wrote: > > Op 22-12-09 10:17, Takashi Iwai schreef: > : > >>> Another thing to try is to replace the whole HD-audio stack with the > >>> last working one (was it 2.6.32?), and check whether it works with > >>> 2.6.32 core. > >> Could you give more details on how to do this? > >> Does it consist in taking the sound/ directory of 2.6.32 and copying it > >> over 2.6.33-rc1 ? Or less files? > > > > I'd try first only sound/pci/hda/* files. > > > Brute-force reverting these files to the version of 2.6.32.2 does fix > the problem. OK, that's good to know. > When I find time, I'll try to git bisect on this particular directory... > but anyone willing to try to beat me at this is welcome, as Christmas is > going to keep me rather busy ;-) bisecting would be indeed helpful. It needs to be done only in the above directory, so there shouldn't be much. thanks, Takashi

1 0

[alsa-devel] HDA model specification and a topic/hda bug?
by Paul "TBBle" Hampson 23 Dec '09

23 Dec '09

Hi, I've a couple of questions. Firstly, I've been through "MORE NOTES ON HD-AUDIO DRIVER" and was able to use codecgraph and hda-analyzer to identify non-disabled speakers on my laptop (the laptop claims four speakers plus a subwoofer, Alsa's autodetect only enables the subwoofer for output) but I'm having trouble working out how to convert this knowledge into either changes to patch_realtek.c or even a patch file to feed the driver, so that it comes up correctly by default. I do have a printout of the Intel HDA spec's codec layout chapter, but I haven't read it yet. I'm hoping not to have to, but I suspect I will. Anyway, I've been tracking topic/hda from the Alsa git repository, hoping it'll magically fix my sound-card detection, but I'm having a problem with 5b0cb1d850c26893b1468b3a519433a1b7a176be (ALSA: hda - add more NID->Control mapping) which I believe is part of the 1.0.22 release. For reference, it's an MSI GT725 laptop... Basically, snd_hda_add_nids is being called with an array of three nids, the third of which is 0. I've been throwing snd_kprintd statements at it, and it seems that I see input_mux->num_items as 3 (which I _think_ is correct, I have a buit-in microphone, microphone jack and line-in jack) but I only have two analog aud_in nodes (0x8 and 0x9, 0x7 is a vendor node). _From the look of it, this call to snd_hda_add_nids is wrong: err = snd_hda_add_nids(codec, kctl, i, spec->capsrc_nids, spec->input_mux->num_items); it maybe should be err = snd_hda_add_nids(codec, kctl, i, spec->capsrc_nids, spec->num_adc_nids); if I'm understanding patch_alc882 correctly, although that looks wrong in pretty much every other use I can see. It's prolly me not understanding how the HDA stuff ties together very well, but right now the inbuilt sound fails to load with a -22 (-EINVAL) error, which is snd_hda_add_nid rejecting a nid <= 0. I suspect http://bugs.gentoo.org/show_bug.cgi?id=297454 is the same problem, and https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4846 looks like it too. On searching, https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4841 is exactly the same problem, and the same model laptop as mine. I'm happy to do debugging and whatnot, since I hope to actually work out the correct patch overrides to get my laptop speakers properly working so any debugging I do for this problem will hopefully improve my understanding of the HDA driver.

2 1

Re: [alsa-devel] Mic not working on Dell Vostro 1700 by default
by Takashi Iwai 23 Dec '09

23 Dec '09

At Tue, 22 Dec 2009 18:50:28 +0100, Anisse Astier wrote: > > On Tue, 22 Dec 2009 18:44:04 +0100, Anisse Astier <anisse(a)astier.eu> wrote : > > > Hi, > > > > (i'm not subscribed to the list, please cc me in replies) > > > > So I'm on a Dell Vostro 1700 with latest BIOS (A07). In the default > > configuration, the mic is not working. > > > > But if I append model=dell-m42 when loading snd-hda-intel, it works. > > > > My alsa-info: > > http://www.alsa-project.org/db/?f=8d6f7d613b909c0eef67f93e0e86bf13476e9e54 > > > > alsa-info with option model=dell-m42 > > http://www.alsa-project.org/db/?f=233cee7ffea01b9f46451e64c95c4fc482ab4ee4 > > > > I don't know if dell-m42 is enough, I chose it because it was used for the > > Dell Vostro 1500, for which there was a pci quirk in patch_sigmatel.c, > > and that seemed close enough. Everything seems to work properly though. > > > > I don't know if it's possible to add the same pci-quirk (or a similar one) > > in patch_sigmatel.c , or any dmi-based quirk, but I'd like to help in > > doing so. > > > > I compiled a working hda-verb should some remote debugging be needed. > > > > I forgot to add that this is also working with dell-m44. A pointer: > https://wiki.ubuntu.com/InstallingUbuntuOnADellVostro1700#AUDIO : > suggestion to add this option by default on ubuntu. So, which is better, m42 or m44? This can be checked only with a real hardware... thanks, Takashi

1 0

Re: [alsa-devel] Installing a "different" version of Alsa
by Brian Welch 23 Dec '09

23 Dec '09

Message: 1 Date: Sun, 20 Dec 2009 12:47:28 +0000 From: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Subject: Re: [alsa-devel] Installing a "different" version of Alsa To: Brian Welch <Brian-Welch(a)cox.net> Cc: alsa-devel(a)alsa-project.org Message-ID: <20091220124727.GA2881(a)sirena.org.uk> Content-Type: text/plain; charset=us-ascii On Sat, Dec 19, 2009 at 09:48:16PM -0700, Brian Welch wrote: >> When I use kbuild to build my custom sound hardware driver against the >> SDK >> in the kernel, it always trys to pick up the older ALSA header files that >> are in the >> SDK, but I want to use the newer versions that are in the directory with >> the newer >> version of ALSA. What the best way to make sure I get the correct >> headers? >> Is what I'm trying "dangerous" and should I be going about this a >> different way? > The best way of integrating with the toolchain you're using will usually > depend on the toolchain - you'll generally be able to get better advice > about how to work with the toolchain by asking people who work with the > toolchain. I guess I don't understand exactly what you mean by "toolchain." I've already contacted the people that package up the SDK and they said I should solve it myself. I originally hoped that I could just re-build the kernel without Alsa built-in, then add my new version of Alsa, but they claim that causes other issues (i.e. no sound at all).

2 3

[alsa-devel] ASoC updates for 2.6.33
by Mark Brown 22 Dec '09

22 Dec '09

The following changes since commit 48c03ce72f2665f79a3fe54fc6d71b8cc3d30803: Guennadi Liakhovetski (1): ASoC: wm8974: fix a wrong bit definition are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.33 Guennadi Liakhovetski (1): ASoC: add missing parameter to mx27vis_hifi_hw_free() Uwe Kleine-König (1): ASoC: sh: FSI:: don't check platform_get_irq's return value against zero sound/soc/imx/mx27vis_wm8974.c | 3 ++- sound/soc/sh/fsi.c | 2 +- 2 files changed, 3 insertions(+), 2 deletions(-)

2 1

Re: [alsa-devel] [2.6.33-rc1] [SOUND] HDA: high cpu usage, noise
by Takashi Iwai 22 Dec '09

22 Dec '09

At Tue, 22 Dec 2009 16:15:57 +0100, Éric Piel wrote: > > Op 22-12-09 10:17, Takashi Iwai schreef: > : > >>> It shouldn't. The polling mode doesn't give any behavioral change, > >>> and the CPU usage can't be that high. > >>> > >>> Try is to pass bdl_pos_adj=0 option. This should reduce the CPU hog, > >>> at least. But it doesn't mean it's the right fix... > >> I'll try > Just tried: it doesn't have any effect, as enable_msi=0. Still 25% of > the CPU used, as before. > $ cat bdl_pos_adj > 0,-1,-1,-1,-1,-1,-1,-1 > > Could it be coming from the beeper? Also, just in case it can help, > attached is my .config . Do you have multiple HD-audio devices? See /proc/asound/cards. If there are multiple, you need to pass a value for each. Takashi

1 0

[alsa-devel] [PATCH 7/7] ASoC: sh: FSI:: don't check platform_get_irq's return value against zero
by Uwe Kleine-König 22 Dec '09

22 Dec '09

platform_get_irq returns -ENXIO on failure, so !irq was probably always true. Better use (int)irq <= 0. Note that a return value of zero is still handled as error even though this could mean irq0. This is a followup to 305b3228f9ff4d59f49e6d34a7034d44ee8ce2f0 that changed the return value of platform_get_irq from 0 to -ENXIO on error. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> Cc: David Vrabel <dvrabel(a)arcom.com> Cc: Greg Kroah-Hartman <gregkh(a)suse.de> Cc: Liam Girdwood <lrg(a)slimlogic.co.uk> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> Cc: Paul Mundt <lethal(a)linux-sh.org> Cc: alsa-devel(a)alsa-project.org --- sound/soc/sh/fsi.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 9c49c11..42813b8 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -876,7 +876,7 @@ static int fsi_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); irq = platform_get_irq(pdev, 0); - if (!res || !irq) { + if (!res || (int)irq <= 0) { dev_err(&pdev->dev, "Not enough FSI platform resources.\n"); ret = -ENODEV; goto exit; -- 1.6.5.2

3 3

Re: [alsa-devel] [PATCH] ASoC: add missing parameter to mx27vis_hifi_hw_free()
by Mark Brown 22 Dec '09

22 Dec '09

On Tue, Dec 22, 2009 at 09:26:10AM +0100, Guennadi Liakhovetski wrote: > Commit 2ccafed4 added an extra parameter to the DAI .set_pll() method, but > it missed this call in sound/soc/imx/mx27vis_wm8974.c. Appled, thanks.

1 0

[alsa-devel] io plugin: polling on unix socket, socket buffer sizes ?
by Stefan Schoenleitner 22 Dec '09

22 Dec '09

Hi, I'm currently writing an alsa io plugin that should exchange PCM samples with another piece of software over a unix socket. More precisely exactly 160 PCM samples with 16bit each should be exchanged at a time. AFAIK the alsa plugin framework works by utilizing poll(). Thus if the plugin is used for playback it would use POLLOUT on the socket to see if the next chunk of data can be written to the socket without blocking. By reading quite a lot of alsa code (including pcm_bluetooth.c from bluez) it seems that I need to set the socket buffer size to a value small enough, right ? Thus if I would like to send one period consisting of 160 samples (with 16bits each) at a time, I would need to set the unix send buffer size to 320 bytes. This way some kind of handshaking between my application and the alsa plugin would be established: Each time the application reads a full period of data from the socket, the alsa plugin will start to send the next period as poll() will return the POLLOUT event. (Thus the transfer callback function would be called in the plugin code.) Is this correct so far ? To test the behavior I wrote two small code snippets: One is unix socket reader that reads a chunk of 10 bytes at a time and then sleeps for 10 seconds. The socket recv buffer size SO_RCVBUF has been set to 10 bytes with setsockopt(). The other snippet is the unix socket writer: It sets the socket send buffer size SO_SNDBUF to 10 bytes so that after writing 10 bytes to the socket, poll with POLLOUT would block until the other end (the reader) reads some bytes and the send buffer is drained. The reader polls on the socket with a timeout of 1 sec and in case poll returns the POLLOUT event, it writes a chunk of 10 bytes to the socket. With these small code snippets I would expect the following behavior:

1 0

Re: [alsa-devel] [PATCH] ASoC: add missing parameter to mx27vis_hifi_hw_free()
by Liam Girdwood 22 Dec '09

22 Dec '09

On Tue, 2009-12-22 at 09:26 +0100, Guennadi Liakhovetski wrote: > Commit 2ccafed4 added an extra parameter to the DAI .set_pll() method, but > it missed this call in sound/soc/imx/mx27vis_wm8974.c. > > Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> > --- > sound/soc/imx/mx27vis_wm8974.c | 3 ++- > 1 files changed, 2 insertions(+), 1 deletions(-) > > diff --git a/sound/soc/imx/mx27vis_wm8974.c b/sound/soc/imx/mx27vis_wm8974.c > index 0267d2d..07d2a24 100644 > --- a/sound/soc/imx/mx27vis_wm8974.c > +++ b/sound/soc/imx/mx27vis_wm8974.c > @@ -180,7 +180,8 @@ static int mx27vis_hifi_hw_free(struct snd_pcm_substream *substream) > struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; > > /* disable the PLL */ > - return codec_dai->ops->set_pll(codec_dai, IGNORED_ARG, 0, 0); > + return codec_dai->ops->set_pll(codec_dai, IGNORED_ARG, IGNORED_ARG, > + 0, 0); > } > > /* Acked-by: Liam Girdwood <lrg(a)slimlogic.co.uk>

1 0

Re: [alsa-devel] [2.6.33-rc1] [SOUND] HDA: high cpu usage, noise
by Takashi Iwai 22 Dec '09

22 Dec '09

At Tue, 22 Dec 2009 10:15:20 +0100, Éric Piel wrote: > > Op 22-12-09 08:04, Takashi Iwai schreef: > > At Mon, 21 Dec 2009 20:08:34 +0100, > > Éric Piel wrote: > : > >> Could it be coming from this timeout? > > > > It shouldn't. The polling mode doesn't give any behavioral change, > > and the CPU usage can't be that high. > > > > Try is to pass bdl_pos_adj=0 option. This should reduce the CPU hog, > > at least. But it doesn't mean it's the right fix... > I'll try > > > > > Another thing to try is to replace the whole HD-audio stack with the > > last working one (was it 2.6.32?), and check whether it works with > > 2.6.32 core. > Could you give more details on how to do this? > Does it consist in taking the sound/ directory of 2.6.32 and copying it > over 2.6.33-rc1 ? Or less files? I'd try first only sound/pci/hda/* files. Takashi

1 0

[alsa-devel] [PATCH 1/2] ALSA: pcm - Ensure wakeup at each periodic interrupt
by jassisinghbrar＠gmail.com 22 Dec '09

22 Dec '09

From: Jassi Brar <jassi.brar(a)samsung.com> The check for at least 'avail_min' available data before calling wake_up doesn't always hold good as it does not guarantee callbacks at each periodic interrupt. An example of such situation is snd_pcm_lib_read1/write1 consuming some space of the period and going to sleep from wait_for_avail_min upon syncing with the DMA pointer. Clearly just the remainder of period size is needed, but wake_up is called only after _two_ periodic interrupts from that point. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/core/pcm_lib.c | 10 ++++++++-- 1 files changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 30f4108..ee2d7b9 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -189,7 +189,7 @@ snd_pcm_update_hw_ptr_pos(struct snd_pcm_substream *substream, static int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime) { - snd_pcm_uframes_t avail; + snd_pcm_uframes_t avail, prd_short; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) avail = snd_pcm_playback_avail(runtime); @@ -208,8 +208,14 @@ static int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream, return -EPIPE; } } - if (avail >= runtime->control->avail_min) + + prd_short = runtime->period_size + - runtime->status->hw_ptr % runtime->period_size; + + if (avail >= runtime->control->avail_min + || avail >= prd_short) wake_up(&runtime->sleep); + return 0; } -- 1.6.2.5

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[alsa-devel] [PATCH] refine rate selection in snd_interval_ratnum()
by Krzysztof Helt 22 Dec '09

22 Dec '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Refine the rate selection by choosing the rate closer to the requested one in case of selecting single frequency. Previously, the higher rate was always selected. Also, fix problem with the best_diff unsigned int value wrapping (turning negative). Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- sound/core/pcm_lib.c | 20 +++++++++++++++++--- 1 files changed, 17 insertions(+), 3 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index a27545b..b07cc36 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -745,10 +745,13 @@ int snd_interval_ratnum(struct snd_interval *i, unsigned int rats_count, struct snd_ratnum *rats, unsigned int *nump, unsigned int *denp) { - unsigned int best_num, best_diff, best_den; + unsigned int best_num, best_den; + int best_diff; unsigned int k; struct snd_interval t; int err; + unsigned int result_num, result_den; + int result_diff; best_num = best_den = best_diff = 0; for (k = 0; k < rats_count; ++k) { @@ -770,6 +773,8 @@ int snd_interval_ratnum(struct snd_interval *i, den -= r; } diff = num - q * den; + if (diff < 0) + diff = -diff; if (best_num == 0 || diff * best_den < best_diff * den) { best_diff = diff; @@ -784,6 +789,9 @@ int snd_interval_ratnum(struct snd_interval *i, t.min = div_down(best_num, best_den); t.openmin = !!(best_num % best_den); + result_num = best_num; + result_diff = best_diff; + result_den = best_den; best_num = best_den = best_diff = 0; for (k = 0; k < rats_count; ++k) { unsigned int num = rats[k].num; @@ -806,6 +814,8 @@ int snd_interval_ratnum(struct snd_interval *i, den += rats[k].den_step - r; } diff = q * den - num; + if (diff < 0) + diff = -diff; if (best_num == 0 || diff * best_den < best_diff * den) { best_diff = diff; @@ -825,10 +835,14 @@ int snd_interval_ratnum(struct snd_interval *i, return err; if (snd_interval_single(i)) { + if (best_diff * result_den < result_diff * best_den) { + result_num = best_num; + result_den = best_den; + } if (nump) - *nump = best_num; + *nump = result_num; if (denp) - *denp = best_den; + *denp = result_den; } return err; } -- 1.6.4 ---------------------------------------------------------------------- Milosc, praca, pieniadze. Sprawdz swoj horoskop na dzis >> http://link.interia.pl/f2531

2 1

[alsa-devel] Problem with the new BEEP framework and HP dv6
by Ozan Çağlayan 22 Dec '09

22 Dec '09

Hi, Before the new CONFIG_SND_HDA_INPUT_BEEP_MODE is introduced, I was patching the alsa-driver package with a very trivial patch which was setting the default of enable_beep to 0. I created another patch to completely disable beep stuff for the new BEEP_MODE stuff. It seemed to work fine on my computer but triggered NULL pointer dereference on an HP dv6: Index: alsa-driver/configure =================================================================== --- alsa-driver.orig/configure +++ alsa-driver/configure @@ -15945,7 +15945,7 @@ alsa_check_kconfig_option () { if ( test "$CONFIG_SND_PCI" = "y" -o "$CONFIG_SND_PCI" = "m" ) && ( test "$CONFIG_SND_HDA_INTEL" = "y" -o "$CONFIG_SND_HDA_INTEL" = "m" ) && ( test "$CONFIG_SND_HDA_INPUT_BEEP" = "y" -o "$CONFIG_SND_HDA_INPUT_BEEP" = "m" ); then - CONFIG_SND_HDA_INPUT_BEEP_MODE="1" + CONFIG_SND_HDA_INPUT_BEEP_MODE="0" fi if alsa_check_kconfig_option "hda-input-jack"; then if ( test "$CONFIG_SND_PCI" = "y" -o "$CONFIG_SND_PCI" = "m" ) && @@ -18366,7 +18366,7 @@ _ACEOF fi if test -n "$CONFIG_SND_HDA_INPUT_BEEP_MODE"; then cat >>confdefs.h <<_ACEOF -#define CONFIG_SND_HDA_INPUT_BEEP_MODE 1 +#define CONFIG_SND_HDA_INPUT_BEEP_MODE 0 _ACEOF fi Index: alsa-driver/sound/pci/hda/Kconfig =================================================================== --- alsa-driver.orig/sound/pci/hda/Kconfig +++ alsa-driver/sound/pci/hda/Kconfig @@ -41,7 +41,7 @@ config SND_HDA_INPUT_BEEP config SND_HDA_INPUT_BEEP_MODE int "Digital beep registration mode (0=off, 1=on, 2=mute sw on/off)" depends on SND_HDA_INPUT_BEEP=y - default "1" + default "0" range 0 2 help Set 0 to disable the digital beep interface for HD-audio by default. Index: alsa-driver/sound/pci/hda/hda_intel.c =================================================================== --- alsa-driver.orig/sound/pci/hda/hda_intel.c +++ alsa-driver/sound/pci/hda/hda_intel.c @@ -98,7 +98,7 @@ MODULE_PARM_DESC(patch, "Patch file for #ifdef CONFIG_SND_HDA_INPUT_BEEP module_param_array(beep_mode, int, NULL, 0444); MODULE_PARM_DESC(beep_mode, "Select HDA Beep registration mode " - "(0=off, 1=on, 2=mute switch on/off) (default=1)."); + "(0=off, 1=on, 2=mute switch on/off) (default=0)."); #endif #ifdef CONFIG_SND_HDA_POWER_SAVE I really don't know if I mistakenly modified the code path with those modified #defines, but I'm sure that the patch correctly disables beep device on my Realtek ALC883 codec. Here's the stack trace of the patched alsa-driver: [ 14.516122] HDA Intel 0000:00:1b.0: PCI INT A -> GSI 22 (level, low) -> IRQ 22 [ 14.516178] HDA Intel 0000:00:1b.0: setting latency timer to 64 [ 14.569087] BUG: unable to handle kernel NULL pointer dereference at 00000032 [ 14.569527] IP: [<f8f35b2b>] stac92xx_parse_auto_config+0x616/0xd5f [snd_hda_codec_idt] [ 14.570007] *pde = 00000000 [ 14.570014] Oops: 0002 [#1] SMP [ 14.570014] last sysfs file: /sys/module/snd_hda_codec/initstate [ 14.570014] Modules linked in: snd_hda_codec_idt snd_hda_intel(+) snd_hda_codec snd_seq_dummy snd_hwdep snd_seq_oss arc4 snd_seq_midi_event ecb snd_seq snd_seq_device iwlagn iwlcore snd_pcm_oss snd_mixer_oss snd_pcm intel_agp rfkill agpgart btusb uvcvideo snd_timer mac80211 snd jmb38x_ms r8169 video iTCO_wdt videodev memstick v4l1_compat iTCO_vendor_support soundcore mii output pcspkr joydev snd_page_alloc wmi cfg80211 hp_accel bluetooth lis3lv02d brd usb_storage firewire_ohci firewire_core sdhci_pci sdhci mmc_core led_class ohci1394 ieee1394 ahci libata [ 14.570014] [ 14.570014] Pid: 175, comm: modprobe Not tainted (2.6.30.9-128 #1) HP Pavilion dv6 Notebook PC [ 14.570014] EIP: 0060:[<f8f35b2b>] EFLAGS: 00010246 CPU: 0 [ 14.570014] EIP is at stac92xx_parse_auto_config+0x616/0xd5f [snd_hda_codec_idt] [ 14.570014] EAX: 00000000 EBX: 00000026 ECX: 00000001 EDX: 00000026 [ 14.570014] ESI: f6a20000 EDI: f6a23800 EBP: f6ae9de8 ESP: f6ae9d7c [ 14.570014] DS: 007b ES: 007b FS: 00d8 GS: 0033 SS: 0068 [ 14.570014] Process modprobe (pid: 175, ti=f6ae8000 task=f694be30 task.ti=f6ae8000) [ 14.570014] Stack: [ 14.570014] 00000400 f6ac11a4 00210000 f6ac1000 f6a23a74 f6a23800 00000000 f6a23800 [ 14.570014] f6a23800 00000007 0007070d 0000000d f6ae9dcc f8ecb558 00000000 00000000 [ 14.570014] 2063694d 6b63614a 646f4d20 00000065 f6ae9de0 f8ecc3ff f6a23800 00000009 [ 14.570014] Call Trace: [ 14.578754] [<f8ecb558>] ? get_alloc_hash+0x41/0x7c [snd_hda_codec] [ 14.578754] [<f8ecc3ff>] ? snd_hda_codec_write_cache+0x5c/0x66 [snd_hda_codec] [ 14.578754] [<f8f37b14>] ? patch_stac92hd71bxx+0x5ce/0x624 [snd_hda_codec_idt] [ 14.578754] [<f8ece8b6>] ? snd_hda_codec_configure+0x253/0x2fe [snd_hda_codec] [ 14.578754] [<f8f0530f>] ? azx_probe+0x802/0x8ca [snd_hda_intel] [ 14.578754] [<f8f0496c>] ? azx_send_cmd+0x0/0x136 [snd_hda_intel] [ 14.578754] [<f8f047a9>] ? azx_get_response+0x0/0x1c3 [snd_hda_intel] [ 14.578754] [<f8f03fe6>] ? azx_attach_pcm_stream+0x0/0x148 [snd_hda_intel] [ 14.578754] [<f8f03c99>] ? azx_bus_reset+0x0/0x56 [snd_hda_intel] [ 14.578754] [<f8f03b50>] ? azx_power_notify+0x0/0x57 [snd_hda_intel] [ 14.578754] [<c029ae28>] ? local_pci_probe+0xe/0x10 [ 14.578754] [<c029b842>] ? pci_device_probe+0x43/0x66 [ 14.578754] [<c030e0fb>] ? really_probe+0x6a/0xf5 [ 14.578754] [<c030e1a2>] ? driver_probe_device+0x1c/0x1f [ 14.578754] [<c030e1e8>] ? __driver_attach+0x43/0x5f [ 14.578754] [<c030db8a>] ? bus_for_each_dev+0x3d/0x67 [ 14.578754] [<c030dfe3>] ? driver_attach+0x14/0x16 [ 14.578754] [<c030e1a5>] ? __driver_attach+0x0/0x5f [ 14.578754] [<c030d61a>] ? bus_add_driver+0xa2/0x1c9 [ 14.578754] [<c030e42b>] ? driver_register+0x8b/0xeb [ 14.578754] [<c029bb6d>] ? __pci_register_driver+0x38/0x93 [ 14.578754] [<c01453d1>] ? __blocking_notifier_call_chain+0x40/0x4c [ 14.578754] [<f8f0b000>] ? alsa_card_azx_init+0x0/0x19 [snd_hda_intel] [ 14.578754] [<f8f0b017>] ? alsa_card_azx_init+0x17/0x19 [snd_hda_intel] [ 14.578754] [<c0101139>] ? do_one_initcall+0x4c/0x13a [ 14.578754] [<c01531f9>] ? sys_init_module+0x87/0x18b [ 14.578754] [<c01031d4>] ? sysenter_do_call+0x12/0x28 [ 14.578754] Code: 06 f5 ff ff 85 c0 0f 88 b6 06 00 00 8b 45 a0 89 da e8 c2 c9 f9 ff 85 c0 0f 88 a4 06 00 00 8b 4d a0 89 da 8b 41 68 b9 01 00 00 00 <80> 48 32 04 8b 45 a0 e8 fc 79 f9 ff 8b 5d a0 83 7b 68 00 74 2f [ 14.578754] EIP: [<f8f35b2b>] stac92xx_parse_auto_config+0x616/0xd5f [snd_hda_codec_idt] SS:ESP 0068:f6ae9d7c [ 14.578754] CR2: 0000000000000032 [ 14.810381] ---[ end trace af42daf6416d16f0 ]--- --- Note that I just tried 1.0.22 without this patch on that dv6 and I wasn't able to get any sound even the modules were correctly loaded. I checked the mixer levels they were OK. I'll try to dig into this with alsa-info but the main question now is how should I cleanly disable BEEP device? Thanks, Ozan

2 3

Re: [alsa-devel] [2.6.33-rc1] [SOUND] HDA: high cpu usage, noise
by Takashi Iwai 22 Dec '09

22 Dec '09

At Mon, 21 Dec 2009 20:08:34 +0100, Éric Piel wrote: > > Op 21-12-09 19:12, Maciej Rutecki schreef: > > 2009/12/21 Takashi Iwai <tiwai(a)suse.de>: > >> At Sun, 20 Dec 2009 18:27:45 +0100, > >> Maciej Rutecki wrote: > >>> > >>> Hi > >>> > >>> Affected kernel: 2.6.33-rc1 > >>> Last known good: 2.6.32 > >> > >> The only fundamental change, AFAIK, is the use of MSI as default. > >> Could you try to add enable_msi=0 option to snd-hda-intel? > >> > > > > root@gumis:/# cat /sys/module/snd_hda_intel/parameters/enable_msi > > 0 > > > > Still the same: > > ps aux: > > > > root 1117 27.4 0.0 0 0 ? S 19:00 2:05 [hd-audio0] > > > > I also notice short sound during booting system, when sound modules is > > loading. I can try bisection, but in next week; already I don't have > > to much time. > Hello, > I've got the same problem here (high hd-audio0 usage + clicks in the > output sound). > > Hardware is identical: > /proc/asound/pcm > 00-00: AD198x Analog : AD198x Analog : playback 1 : capture 1 > > I haven't tried the enable_msi parameter but, just to give more info, > here is the log ("dmesg | grep -i hda") when the bug happens: > Dec 18 10:39:10 dutifh kernel: HDA Intel 0000:00:1b.0: power state > changed by ACPI to D0 > Dec 18 10:39:10 dutifh kernel: HDA Intel 0000:00:1b.0: PCI INT A -> GSI > 17 (level, low) -> IRQ 17 > Dec 18 10:39:10 dutifh kernel: input: HDA Digital PCBeep as > /devices/pci0000:00/0000:00:1b.0/input/input12 > Dec 18 10:39:10 dutifh kernel: hda-intel: azx_get_response timeout, > switching to polling mode: last cmd=0x006f0900 > > And here is with 2.6.32, working fine > Dec 18 11:46:03 dutifh kernel: HDA Intel 0000:00:1b.0: power state > changed by ACPI to D0 > Dec 18 11:46:03 dutifh kernel: HDA Intel 0000:00:1b.0: PCI INT A -> GSI > 17 (level, low) -> IRQ 17 > Dec 18 11:46:03 dutifh kernel: input: HDA Digital PCBeep as > /devices/pci0000:00/0000:00:1b.0/input/input11 > > Could it be coming from this timeout? It shouldn't. The polling mode doesn't give any behavioral change, and the CPU usage can't be that high. Try is to pass bdl_pos_adj=0 option. This should reduce the CPU hog, at least. But it doesn't mean it's the right fix... Another thing to try is to replace the whole HD-audio stack with the last working one (was it 2.6.32?), and check whether it works with 2.6.32 core. thanks, Takashi

1 0

[alsa-devel] [patch] Add support for the new 27 inch IMacs
by Rafael Espíndola 22 Dec '09

22 Dec '09

With the attached patch I am able to use the sound on a new IMac 27. What works: *) Internal speakers *) Internal microphone *) Headphone I don't have an external mic or a SPDIF device to test the rest. Cheers, Rafael

2 3

[PATCH] ALSA: hda - Fix NULL dereference with enable_beep=0 option
by Takashi Iwai 22 Dec '09

22 Dec '09

Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_sigmatel.c | 19 ++++++++++--------- 1 files changed, 10 insertions(+), 9 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 3d59f83..417fb22 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -3779,15 +3779,16 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out err = snd_hda_attach_beep_device(codec, nid); if (err < 0) return err; - /* IDT/STAC codecs have linear beep tone parameter */ - codec->beep->linear_tone = 1; - /* if no beep switch is available, make its own one */ - caps = query_amp_caps(codec, nid, HDA_OUTPUT); - if (codec->beep && - !((caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT)) { - err = stac92xx_beep_switch_ctl(codec); - if (err < 0) - return err; + if (codec->beep) { + /* IDT/STAC codecs have linear beep tone parameter */ + codec->beep->linear_tone = 1; + /* if no beep switch is available, make its own one */ + caps = query_amp_caps(codec, nid, HDA_OUTPUT); + if (!(caps & AC_AMPCAP_MUTE)) { + err = stac92xx_beep_switch_ctl(codec); + if (err < 0) + return err; + } } } #endif -- 1.6.5.7

1 0

[alsa-devel] [PATCH 2/3] ASoC: OMAP4: Add support for McPDM
by Candelaria Villareal, Jorge 22 Dec '09

22 Dec '09

McPDM is the interface between Phoenix audio codec and the OMAP4430 processor. It enables data to be transfered to/from Phoenix at sample rates of 88.4 or 96 KHz. Signed-off-by: Jorge Eduardo Candelaria <x0107209(a)ti.com> Signed-off-by: Margarita Olaya <x0080101(a)ti.com> --- sound/soc/omap/Kconfig | 3 + sound/soc/omap/Makefile | 2 + sound/soc/omap/mcpdm.c | 505 +++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/omap/mcpdm.h | 156 +++++++++++++++ 4 files changed, 666 insertions(+), 0 deletions(-) create mode 100644 sound/soc/omap/mcpdm.c create mode 100644 sound/soc/omap/mcpdm.h diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 61952aa..b96e9d8 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -6,6 +6,9 @@ config SND_OMAP_SOC_MCBSP tristate select OMAP_MCBSP +config SND_OMAP_SOC_MCPDM + tristate + config SND_OMAP_SOC_N810 tristate "SoC Audio support for Nokia N810" depends on SND_OMAP_SOC && MACH_NOKIA_N810 && I2C diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 3db8a6c..bf8b388 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -1,9 +1,11 @@ # OMAP Platform Support snd-soc-omap-objs := omap-pcm.o snd-soc-omap-mcbsp-objs := omap-mcbsp.o +snd-soc-omap-mcpdm-objs := mcpdm.o obj-$(CONFIG_SND_OMAP_SOC) += snd-soc-omap.o obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o +obj-$(CONFIG_SND_OMAP_SOC_MCPDM) += snd-soc-omap-mcpdm.o # OMAP Machine Support snd-soc-n810-objs := n810.o diff --git a/sound/soc/omap/mcpdm.c b/sound/soc/omap/mcpdm.c new file mode 100644 index 0000000..e162dd1 --- /dev/null +++ b/sound/soc/omap/mcpdm.c @@ -0,0 +1,505 @@ +/* + * mcpdm.c -- McPDM interface driver + * + * Author: Jorge Eduardo Candelaria <x0107209(a)ti.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/device.h> +#include <linux/platform_device.h> +#include <linux/wait.h> +#include <linux/interrupt.h> +#include <linux/err.h> +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/irq.h> + +#include "mcpdm.h" + +static struct omap_mcpdm *mcpdm; + +static void omap_mcpdm_write(u16 reg, u32 val) +{ + __raw_writel(val, mcpdm->io_base + reg); +} + +static int omap_mcpdm_read(u16 reg) +{ + return __raw_readl(mcpdm->io_base + reg); +} + +void omap_mcpdm_reg_dump(void) +{ + dev_dbg(mcpdm->dev, "***********************\n"); + dev_dbg(mcpdm->dev, "IRQSTATUS_RAW: 0x%04x\n", + omap_mcpdm_read(MCPDM_IRQSTATUS_RAW)); + dev_dbg(mcpdm->dev, "IRQSTATUS: 0x%04x\n", + omap_mcpdm_read(MCPDM_IRQSTATUS)); + dev_dbg(mcpdm->dev, "IRQENABLE_SET: 0x%04x\n", + omap_mcpdm_read(MCPDM_IRQENABLE_SET)); + dev_dbg(mcpdm->dev, "IRQENABLE_CLR: 0x%04x\n", + omap_mcpdm_read(MCPDM_IRQENABLE_CLR)); + dev_dbg(mcpdm->dev, "IRQWAKE_EN: 0x%04x\n", + omap_mcpdm_read(MCPDM_IRQWAKE_EN)); + dev_dbg(mcpdm->dev, "DMAENABLE_SET: 0x%04x\n", + omap_mcpdm_read(MCPDM_DMAENABLE_SET)); + dev_dbg(mcpdm->dev, "DMAENABLE_CLR: 0x%04x\n", + omap_mcpdm_read(MCPDM_DMAENABLE_CLR)); + dev_dbg(mcpdm->dev, "DMAWAKEEN: 0x%04x\n", + omap_mcpdm_read(MCPDM_DMAWAKEEN)); + dev_dbg(mcpdm->dev, "CTRL: 0x%04x\n", + omap_mcpdm_read(MCPDM_CTRL)); + dev_dbg(mcpdm->dev, "DN_DATA: 0x%04x\n", + omap_mcpdm_read(MCPDM_DN_DATA)); + dev_dbg(mcpdm->dev, "UP_DATA: 0x%04x\n", + omap_mcpdm_read(MCPDM_UP_DATA)); + dev_dbg(mcpdm->dev, "FIFO_CTRL_DN: 0x%04x\n", + omap_mcpdm_read(MCPDM_FIFO_CTRL_DN)); + dev_dbg(mcpdm->dev, "FIFO_CTRL_UP: 0x%04x\n", + omap_mcpdm_read(MCPDM_FIFO_CTRL_UP)); + dev_dbg(mcpdm->dev, "DN_OFFSET: 0x%04x\n", + omap_mcpdm_read(MCPDM_DN_OFFSET)); + dev_dbg(mcpdm->dev, "***********************\n"); +} +EXPORT_SYMBOL(omap_mcpdm_reg_dump); + +/* + * Takes the McPDM module in and out of reset state. + * Uplink and downlink can be reset individually. + */ +void omap_mcpdm_reset(int links, int reset) +{ + int ctrl = omap_mcpdm_read(MCPDM_CTRL); + + if (links & MCPDM_UPLINK) { + if (reset) + ctrl |= SW_UP_RST; + else + ctrl &= ~SW_UP_RST; + } + + if (links & MCPDM_DOWNLINK) { + if (reset) + ctrl |= SW_DN_RST; + else + ctrl &= ~SW_DN_RST; + } + + omap_mcpdm_write(MCPDM_CTRL, ctrl); +} +EXPORT_SYMBOL(omap_mcpdm_reset); + +/* + * Enables the transfer through the PDM interface to/from the Phoenix + * codec by enabling the corresponding UP or DN channels. + */ +void omap_mcpdm_start(int stream) +{ + int ctrl = omap_mcpdm_read(MCPDM_CTRL); + + if (stream) + ctrl |= mcpdm->up_channels; + else + ctrl |= mcpdm->dn_channels; + + omap_mcpdm_write(MCPDM_CTRL, ctrl); +} +EXPORT_SYMBOL(omap_mcpdm_start); + +/* + * Disables the transfer through the PDM interface to/from the Phoenix + * codec by disabling the corresponding UP or DN channels. + */ +void omap_mcpdm_stop(int stream) +{ + int ctrl = omap_mcpdm_read(MCPDM_CTRL); + + if (stream) + ctrl &= ~mcpdm->up_channels; + else + ctrl &= ~mcpdm->dn_channels; + + omap_mcpdm_write(MCPDM_CTRL, ctrl); +} +EXPORT_SYMBOL(omap_mcpdm_stop); + +/* + * Configures McPDM uplink for audio recording. + * This function should be called before omap_mcpdm_start. + */ +int omap_mcpdm_set_uplink(struct omap_mcpdm_link *uplink) +{ + int irq_mask = 0; + int ctrl; + + if (!uplink) + return -EINVAL; + + mcpdm->uplink = uplink; + + /* Enable irq request generation */ + irq_mask |= uplink->irq_mask & UPLINK_IRQ_MASK; + omap_mcpdm_write(MCPDM_IRQENABLE_SET, irq_mask); + + /* Configure uplink threshold */ + if (uplink->threshold > UP_THRES_MAX) + uplink->threshold = UP_THRES_MAX; + + omap_mcpdm_write(MCPDM_FIFO_CTRL_UP, uplink->threshold); + + /* Configure DMA controller */ + omap_mcpdm_write(MCPDM_DMAENABLE_SET, DMA_UP_ENABLE); + + /* Set pdm out format */ + ctrl = omap_mcpdm_read(MCPDM_CTRL); + ctrl &= ~PDMOUTFORMAT; + ctrl |= uplink->format & PDMOUTFORMAT; + + /* Uplink channels */ + mcpdm->up_channels = uplink->channels & (PDM_UP_MASK | PDM_STATUS_MASK); + + omap_mcpdm_write(MCPDM_CTRL, ctrl); + + return 0; +} +EXPORT_SYMBOL(omap_mcpdm_set_uplink); + +/* + * Configures McPDM downlink for audio playback. + * This function should be called before omap_mcpdm_start. + */ +int omap_mcpdm_set_downlink(struct omap_mcpdm_link *downlink) +{ + int irq_mask = 0; + int ctrl; + + if (!downlink) + return -EINVAL; + + mcpdm->downlink = downlink; + + /* Enable irq request generation */ + irq_mask |= downlink->irq_mask & DOWNLINK_IRQ_MASK; + omap_mcpdm_write(MCPDM_IRQENABLE_SET, irq_mask); + + /* Configure uplink threshold */ + if (downlink->threshold > DN_THRES_MAX) + downlink->threshold = DN_THRES_MAX; + + omap_mcpdm_write(MCPDM_FIFO_CTRL_DN, downlink->threshold); + + /* Enable DMA request generation */ + omap_mcpdm_write(MCPDM_DMAENABLE_SET, DMA_DN_ENABLE); + + /* Set pdm out format */ + ctrl = omap_mcpdm_read(MCPDM_CTRL); + ctrl &= ~PDMOUTFORMAT; + ctrl |= downlink->format & PDMOUTFORMAT; + + /* Downlink channels */ + mcpdm->dn_channels = downlink->channels & (PDM_DN_MASK | PDM_CMD_MASK); + + omap_mcpdm_write(MCPDM_CTRL, ctrl); + + return 0; +} +EXPORT_SYMBOL(omap_mcpdm_set_downlink); + +/* + * Cleans McPDM uplink configuration. + * This function should be called when the stream is closed. + */ +int omap_mcpdm_clr_uplink(struct omap_mcpdm_link *uplink) +{ + int irq_mask = 0; + + if (!uplink) + return -EINVAL; + + /* Disable irq request generation */ + irq_mask |= uplink->irq_mask & UPLINK_IRQ_MASK; + omap_mcpdm_write(MCPDM_IRQENABLE_CLR, irq_mask); + + /* Disable DMA request generation */ + omap_mcpdm_write(MCPDM_DMAENABLE_CLR, DMA_UP_ENABLE); + + /* Clear Downlink channels */ + mcpdm->up_channels = 0; + + mcpdm->uplink = NULL; + + return 0; +} +EXPORT_SYMBOL(omap_mcpdm_clr_uplink); + +/* + * Cleans McPDM downlink configuration. + * This function should be called when the stream is closed. + */ +int omap_mcpdm_clr_downlink(struct omap_mcpdm_link *downlink) +{ + int irq_mask = 0; + + if (!downlink) + return -EINVAL; + + /* Disable irq request generation */ + irq_mask |= downlink->irq_mask & DOWNLINK_IRQ_MASK; + omap_mcpdm_write(MCPDM_IRQENABLE_CLR, irq_mask); + + /* Disable DMA request generation */ + omap_mcpdm_write(MCPDM_DMAENABLE_CLR, DMA_DN_ENABLE); + + /* clear Downlink channels */ + mcpdm->dn_channels = 0; + + mcpdm->downlink = NULL; + + return 0; +} +EXPORT_SYMBOL(omap_mcpdm_clr_downlink); + +static irqreturn_t omap_mcpdm_irq_handler(int irq, void *dev_id) +{ + struct omap_mcpdm *mcpdm_irq = dev_id; + int irq_status; + + irq_status = omap_mcpdm_read(MCPDM_IRQSTATUS); + + /* Acknowledge irq event */ + omap_mcpdm_write(MCPDM_IRQSTATUS, irq_status); + + switch (irq_status) { + case DN_IRQ_FULL: + case DN_IRQ_EMTPY: + dev_err(mcpdm_irq->dev, "DN FIFO error %x\n", irq_status); + omap_mcpdm_reset(MCPDM_DOWNLINK, 1); + omap_mcpdm_set_downlink(mcpdm_irq->downlink); + omap_mcpdm_reset(MCPDM_DOWNLINK, 0); + break; + case DN_IRQ: + dev_dbg(mcpdm_irq->dev, "DN write request\n"); + break; + case UP_IRQ_FULL: + case UP_IRQ_EMPTY: + dev_err(mcpdm_irq->dev, "UP FIFO error %x\n", irq_status); + omap_mcpdm_reset(MCPDM_UPLINK, 1); + omap_mcpdm_set_uplink(mcpdm_irq->uplink); + omap_mcpdm_reset(MCPDM_UPLINK, 0); + break; + case UP_IRQ: + dev_dbg(mcpdm_irq->dev, "UP write request\n"); + break; + } + + return IRQ_HANDLED; +} + +int omap_mcpdm_request(void) +{ + int ret; + + clk_enable(mcpdm->clk); + + spin_lock(&mcpdm->lock); + + if (!mcpdm->free) { + dev_err(mcpdm->dev, "McPDM interface is in use\n"); + spin_unlock(&mcpdm->lock); + return -EBUSY; + } + mcpdm->free = 0; + + spin_unlock(&mcpdm->lock); + + /* Disable lines while request is ongoing */ + omap_mcpdm_write(MCPDM_CTRL, 0x00); + + ret = request_irq(mcpdm->irq, omap_mcpdm_irq_handler, + 0, "McPDM", (void *)mcpdm); + if (ret) { + dev_err(mcpdm->dev, "Request for McPDM IRQ failed\n"); + return ret; + } + + return 0; +} +EXPORT_SYMBOL(omap_mcpdm_request); + +void omap_mcpdm_free(void) +{ + clk_disable(mcpdm->clk); + + spin_lock(&mcpdm->lock); + if (mcpdm->free) { + dev_err(mcpdm->dev, "McPDM interface is already free\n"); + spin_unlock(&mcpdm->lock); + return; + } + mcpdm->free = 1; + spin_unlock(&mcpdm->lock); + + free_irq(mcpdm->irq, (void *)mcpdm); +} +EXPORT_SYMBOL(omap_mcpdm_free); + +int omap_mcpdm_set_offset(int offset1, int offset2) +{ + int offset; + + if ((offset1 > DN_OFST_MAX) || (offset2 > DN_OFST_MAX)) + return -EINVAL; + + offset = (offset1 << DN_OFST_RX1) | (offset2 << DN_OFST_RX2); + + /* Enable/disable offset cancellation for downlink channel 1 */ + if (offset1) + offset |= DN_OFST_RX1_EN; + else + offset &= ~DN_OFST_RX1_EN; + + /* Enable/disable offset cancellation for downlink channel 2 */ + if (offset2) + offset |= DN_OFST_RX2_EN; + else + offset &= ~DN_OFST_RX2_EN; + + omap_mcpdm_write(MCPDM_DN_OFFSET, offset); + + return 0; +} +EXPORT_SYMBOL(omap_mcpdm_set_offset); + +static int __devinit omap_mcpdm_probe(struct platform_device *pdev) +{ + struct omap_mcpdm_platform_data *pdata = pdev->dev.platform_data; + int ret = 0; + + if (!pdata) { + dev_err(&pdev->dev, "McPDM device initialized without " + "platform data\n"); + ret = -EINVAL; + goto exit; + } + dev_dbg(&pdev->dev, "Initializing OMAP McPDM driver \n"); + + mcpdm = kzalloc(sizeof(struct omap_mcpdm), GFP_KERNEL); + if (!mcpdm) { + ret = -ENOMEM; + goto exit; + } + + spin_lock_init(&mcpdm->lock); + mcpdm->free = 1; + mcpdm->phys_base = pdata->phys_base; + mcpdm->io_base = ioremap(mcpdm->phys_base, SZ_4K); + if (!mcpdm->io_base) { + ret = -ENOMEM; + goto err_ioremap; + } + + mcpdm->irq = pdata->irq; + + /* FIXME: Enable this ones correct clk nodes available */ + if (!cpu_is_omap44xx()) { + mcpdm->clk = clk_get(&pdev->dev, "fclk"); + if (IS_ERR(mcpdm->clk)) { + ret = PTR_ERR(mcpdm->clk); + dev_err(&pdev->dev, "unable to get fclk: %d\n", ret); + goto err_clk; + } + } + + mcpdm->pdata = pdata; + mcpdm->dev = &pdev->dev; + platform_set_drvdata(pdev, mcpdm); + + return 0; + +err_clk: + iounmap(mcpdm->io_base); +err_ioremap: + kfree(mcpdm); +exit: + return ret; +} + +static int __devexit omap_mcpdm_remove(struct platform_device *pdev) +{ + struct omap_mcpdm *mcpdm_ptr = platform_get_drvdata(pdev); + + platform_set_drvdata(pdev, NULL); + + if (mcpdm_ptr) { + clk_disable(mcpdm_ptr->clk); + clk_put(mcpdm_ptr->clk); + + iounmap(mcpdm_ptr->io_base); + + mcpdm_ptr->clk = NULL; + mcpdm_ptr->free = 0; + mcpdm_ptr->dev = NULL; + } + + printk(KERN_INFO "McPDM driver removed \n"); + + return 0; +} + +static struct platform_driver omap_mcpdm_driver = { + .probe = omap_mcpdm_probe, + .remove = omap_mcpdm_remove, + .driver = { + .name = "omap-mcpdm", + }, +}; + +static struct platform_device *omap_mcpdm_device; + +static struct omap_mcpdm_platform_data mcpdm_pdata = { + .phys_base = OMAP44XX_MCPDM_BASE, + .irq = INT_44XX_MCPDM_IRQ, +}; + +static int __init omap_mcpdm_init(void) +{ + int ret; + struct platform_device *device; + + device = platform_device_alloc("omap-mcpdm", -1); + if (!device) { + printk(KERN_ERR "McPDM platform device allocation failed\n"); + return -ENOMEM; + } + device->dev.platform_data = &mcpdm_pdata; + + omap_mcpdm_device = device; + (void) platform_device_add(omap_mcpdm_device); + + ret = platform_driver_register(&omap_mcpdm_driver); + if (ret) + goto error; + return 0; + +error: + printk(KERN_ERR "OMAP McPDM initialization error\n"); + return ret; +} +arch_initcall(omap_mcpdm_init); diff --git a/sound/soc/omap/mcpdm.h b/sound/soc/omap/mcpdm.h new file mode 100644 index 0000000..c953e95 --- /dev/null +++ b/sound/soc/omap/mcpdm.h @@ -0,0 +1,156 @@ +/* + * mcpdm.h -- Defines for McPDM driver + * + * Author: Jorge Eduardo Candelaria <x0107209(a)ti.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#define OMAP44XX_MCPDM_BASE 0x40132000 +#define OMAP44XX_MCPDM_L3_BASE 0x49032000 + +/* McPDM registers */ + +#define MCPDM_REVISION 0x00 +#define MCPDM_SYSCONFIG 0x10 +#define MCPDM_IRQSTATUS_RAW 0x24 +#define MCPDM_IRQSTATUS 0x28 +#define MCPDM_IRQENABLE_SET 0x2C +#define MCPDM_IRQENABLE_CLR 0x30 +#define MCPDM_IRQWAKE_EN 0x34 +#define MCPDM_DMAENABLE_SET 0x38 +#define MCPDM_DMAENABLE_CLR 0x3C +#define MCPDM_DMAWAKEEN 0x40 +#define MCPDM_CTRL 0x44 +#define MCPDM_DN_DATA 0x48 +#define MCPDM_UP_DATA 0x4C +#define MCPDM_FIFO_CTRL_DN 0x50 +#define MCPDM_FIFO_CTRL_UP 0x54 +#define MCPDM_DN_OFFSET 0x58 + +/* + * MCPDM_IRQ bit fields + * IRQSTATUS_RAW, IRQSTATUS, IRQENABLE_SET, IRQENABLE_CLR + */ + +#define DN_IRQ (1 << 0) +#define DN_IRQ_EMTPY (1 << 1) +#define DN_IRQ_ALMST_EMPTY (1 << 2) +#define DN_IRQ_FULL (1 << 3) + +#define UP_IRQ (1 << 8) +#define UP_IRQ_EMPTY (1 << 9) +#define UP_IRQ_ALMST_FULL (1 << 10) +#define UP_IRQ_FULL (1 << 11) + +#define DOWNLINK_IRQ_MASK 0x00F +#define UPLINK_IRQ_MASK 0xF00 + +/* + * MCPDM_DMAENABLE bit fields + */ + +#define DMA_DN_ENABLE 0x1 +#define DMA_UP_ENABLE 0x2 + +/* + * MCPDM_CTRL bit fields + */ + +#define PDM_UP1_EN 0x0001 +#define PDM_UP2_EN 0x0002 +#define PDM_UP3_EN 0x0004 +#define PDM_DN1_EN 0x0008 +#define PDM_DN2_EN 0x0010 +#define PDM_DN3_EN 0x0020 +#define PDM_DN4_EN 0x0040 +#define PDM_DN5_EN 0x0080 +#define PDMOUTFORMAT 0x0100 +#define CMD_INT 0x0200 +#define STATUS_INT 0x0400 +#define SW_UP_RST 0x0800 +#define SW_DN_RST 0x1000 +#define PDM_UP_MASK 0x007 +#define PDM_DN_MASK 0x0F8 +#define PDM_CMD_MASK 0x200 +#define PDM_STATUS_MASK 0x400 + + +#define PDMOUTFORMAT_LJUST (0 << 8) +#define PDMOUTFORMAT_RJUST (1 << 8) + +/* + * MCPDM_FIFO_CTRL bit fields + */ + +#define UP_THRES_MAX 0xF +#define DN_THRES_MAX 0xF + +/* + * MCPDM_DN_OFFSET bit fields + */ + +#define DN_OFST_RX1_EN 0x0001 +#define DN_OFST_RX2_EN 0x0100 + +#define DN_OFST_RX1 1 +#define DN_OFST_RX2 9 +#define DN_OFST_MAX 0x1F + +#define MCPDM_UPLINK 1 +#define MCPDM_DOWNLINK 2 + +struct omap_mcpdm_link { + int irq_mask; + int threshold; + int format; + int channels; +}; + +struct omap_mcpdm_platform_data { + unsigned long phys_base; + u16 irq; +}; + +struct omap_mcpdm { + struct device *dev; + unsigned long phys_base; + void __iomem *io_base; + u8 free; + int irq; + + spinlock_t lock; + struct omap_mcpdm_platform_data *pdata; + struct clk *clk; + struct omap_mcpdm_link *downlink; + struct omap_mcpdm_link *uplink; + struct completion irq_completion; + + int dn_channels; + int up_channels; +}; + +void omap_mcpdm_reg_dump(void); +void omap_mcpdm_reset(int links, int reset); +void omap_mcpdm_start(int stream); +void omap_mcpdm_stop(int stream); +int omap_mcpdm_set_uplink(struct omap_mcpdm_link *uplink); +int omap_mcpdm_set_downlink(struct omap_mcpdm_link *downlink); +int omap_mcpdm_clr_uplink(struct omap_mcpdm_link *uplink); +int omap_mcpdm_clr_downlink(struct omap_mcpdm_link *downlink); +int omap_mcpdm_request(void); +void omap_mcpdm_free(void); +int omap_mcpdm_set_offset(int offset1, int offset2); -- 1.6.0.4

3 10

[alsa-devel] [PATCH 1/1] alsa-lib softvol S24_3LE distrotion fix
by Zerg Cannibal 21 Dec '09

21 Dec '09

Summary: Fix the sound distortions for S24_3LE stream in alsa-lib softvol. This patch fixes sound distortions in alsa-lib "softvol" for S24_3LE sound stream, when softvol slider isn't at 0.0dB position. Signed-off-by: CannibalZerg <cnb_zerg(a)yahoo.com> --- src/pcm/pcm_softvol.c | 22 ++++++++++++++++++++-- 1 files changed, 20 insertions(+), 2 deletions(-) diff --git a/src/pcm/pcm_softvol.c b/src/pcm/pcm_softvol.c index 637e5cb..2c7c006 100644 --- a/src/pcm/pcm_softvol.c +++ b/src/pcm/pcm_softvol.c @@ -107,7 +107,8 @@ static inline int MULTI_DIV_32x16(int a, unsigned short b) v.i = a; y.i = 0; #if __BYTE_ORDER == __LITTLE_ENDIAN - x.i = (unsigned int)v.s[0] * b; + x.i = (unsigned short)v.s[0]; + x.i *= b; y.s[0] = x.s[1]; y.i += (int)v.s[1] * b; #else @@ -135,6 +136,23 @@ static inline int MULTI_DIV_int(int a, unsigned int b, int swap) return swap ? (int)bswap_32(fraction) : fraction; } +/* always little endian */ +static inline int MULTI_DIV_24(int a, unsigned int b) +{ + unsigned int gain = b >> VOL_SCALE_SHIFT; + int fraction; + fraction = MULTI_DIV_32x16(a, b & VOL_SCALE_MASK); + if (gain) { + long long amp = (long long)a * gain + fraction; + if (amp > (int)0x7fffff) + amp = (int)0x7fffff; + else if (amp < (int)0x800000) + amp = (int)0x800000; + return (int)amp; + } + return fraction; +} + static inline short MULTI_DIV_short(short a, unsigned int b, int swap) { unsigned int gain = b >> VOL_SCALE_SHIFT; @@ -223,7 +241,7 @@ static inline short MULTI_DIV_short(short a, unsigned int b, int swap) tmp = src[0] | \ (src[1] << 8) | \ (((signed char *) src)[2] << 16); \ - tmp = MULTI_DIV_int(tmp, vol_scale, 0); \ + tmp = MULTI_DIV_24(tmp, vol_scale); \ dst[0] = tmp; \ dst[1] = tmp >> 8; \ dst[2] = tmp >> 16; \ -- 1.6.5.6

2 2

Re: [alsa-devel] [PATCH] jazz16: Add support for Media Vision Jazz16 chipset
by krzysztof.h1＠poczta.fm 21 Dec '09

21 Dec '09

"Takashi Iwai" <tiwai(a)suse.de> pisze: > At Sun, 20 Dec 2009 19:01:50 +0100, > Krzysztof Helt wrote: > > > > From: Krzysztof Helt <krzysztof.h1(a)wp.pl> > > > > This is one of Sound Blaster Pro compatible chipsets which is supported > > by Linux OSS driver and was missing native supoort for ALSA. > > > > The Jazz16 audio codec is Crystal CS4216 which is capable > > of playback and recording up to 48 kHz stereo. > > > > Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> > > Applied. Could you add the entry to ALSA-Configuration.txt, too? > Yes. There is also one more thing to test - if only one DMA channel is specified (the same for both DMAs). Regards, Krzysztof ---------------------------------------------------------------------- Sypie, sypie, ale czy sypac bedzie nadal? Sprawdz pogode >> http://link.interia.pl/f2530

1 0

[alsa-devel] [PATCH] sbawe: fix memory detection
by Krzysztof Helt 21 Dec '09

21 Dec '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Memory amount is increased before a successful write-read sequence is done. Thus, 512 kB of onboard memory is detected on memoryless cards like SB32. Move the increasing of memory counter after successful read is done. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- sound/isa/sb/emu8000.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/isa/sb/emu8000.c b/sound/isa/sb/emu8000.c index 96678d5..751762f 100644 --- a/sound/isa/sb/emu8000.c +++ b/sound/isa/sb/emu8000.c @@ -393,8 +393,6 @@ size_dram(struct snd_emu8000 *emu) while (size < EMU8000_MAX_DRAM) { - size += 512 * 1024; /* increment 512kbytes */ - /* Write a unique data on the test address. * if the address is out of range, the data is written on * 0x200000(=EMU8000_DRAM_OFFSET). Then the id word is @@ -414,7 +412,9 @@ size_dram(struct snd_emu8000 *emu) /*snd_emu8000_read_wait(emu);*/ EMU8000_SMLD_READ(emu); /* discard stale data */ if (EMU8000_SMLD_READ(emu) != UNIQUE_ID2) - break; /* we must have wrapped around */ + break; /* no memory at this address */ + + size += 512 * 1024; /* increment 512kbytes */ snd_emu8000_read_wait(emu); -- 1.6.4 ---------------------------------------------------------------------- Sypie, sypie, ale czy sypac bedzie nadal? Sprawdz pogode >> http://link.interia.pl/f2530

2 1

[alsa-devel] [PATCH] jazz16: Add support for Media Vision Jazz16 chipset
by Krzysztof Helt 21 Dec '09

21 Dec '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This is one of Sound Blaster Pro compatible chipsets which is supported by Linux OSS driver and was missing native supoort for ALSA. The Jazz16 audio codec is Crystal CS4216 which is capable of playback and recording up to 48 kHz stereo. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- include/sound/sb.h | 1 + sound/isa/Kconfig | 16 ++ sound/isa/sb/Makefile | 2 + sound/isa/sb/jazz16.c | 385 ++++++++++++++++++++++++++++++++++++++++++++++ sound/isa/sb/sb8_main.c | 117 ++++++++++++-- sound/isa/sb/sb_common.c | 3 + sound/isa/sb/sb_mixer.c | 3 + 7 files changed, 511 insertions(+), 16 deletions(-) create mode 100644 sound/isa/sb/jazz16.c diff --git a/include/sound/sb.h b/include/sound/sb.h index 4e62ee1..9535354 100644 --- a/include/sound/sb.h +++ b/include/sound/sb.h @@ -33,6 +33,7 @@ enum sb_hw_type { SB_HW_20, SB_HW_201, SB_HW_PRO, + SB_HW_JAZZ16, /* Media Vision Jazz16 */ SB_HW_16, SB_HW_16CSP, /* SB16 with CSP chip */ SB_HW_ALS100, /* Avance Logic ALS100 chip */ diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 8ebd070..2024d3e 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -239,6 +239,22 @@ config SND_INTERWAVE_STB To compile this driver as a module, choose M here: the module will be called snd-interwave-stb. +config SND_JAZZ16 + tristate "Media Vision Jazz16 card and compatibles" + select SND_OPL3_LIB + select SND_MPU401_UART + select SND_SB8_DSP + help + Say Y here to include support for soundcards based on the + Media Vision Jazz16 chipset: digital chip MVD1216 (Jazz16), + codec MVA416 (CS4216) and mixer MVA514 (ICS2514). + Media Vision's Jazz16 cards were sold under names Pro Sonic 16, + Premium 3-D and Pro 3-D. There were also OEMs cards with the + Jazz16 chipset. + + To compile this driver as a module, choose M here: the module + will be called snd-jazz16. + config SND_OPL3SA2 tristate "Yamaha OPL3-SA2/SA3" select SND_OPL3_LIB diff --git a/sound/isa/sb/Makefile b/sound/isa/sb/Makefile index faeffce..af36696 100644 --- a/sound/isa/sb/Makefile +++ b/sound/isa/sb/Makefile @@ -12,6 +12,7 @@ snd-sb16-objs := sb16.o snd-sbawe-objs := sbawe.o emu8000.o snd-emu8000-synth-objs := emu8000_synth.o emu8000_callback.o emu8000_patch.o emu8000_pcm.o snd-es968-objs := es968.o +snd-jazz16-objs := jazz16.o # Toplevel Module Dependency obj-$(CONFIG_SND_SB_COMMON) += snd-sb-common.o @@ -21,6 +22,7 @@ obj-$(CONFIG_SND_SB8) += snd-sb8.o obj-$(CONFIG_SND_SB16) += snd-sb16.o obj-$(CONFIG_SND_SBAWE) += snd-sbawe.o obj-$(CONFIG_SND_ES968) += snd-es968.o +obj-$(CONFIG_SND_JAZZ16) += snd-jazz16.o ifeq ($(CONFIG_SND_SB16_CSP),y) obj-$(CONFIG_SND_SB16) += snd-sb16-csp.o obj-$(CONFIG_SND_SBAWE) += snd-sb16-csp.o diff --git a/sound/isa/sb/jazz16.c b/sound/isa/sb/jazz16.c new file mode 100644 index 0000000..d52966b --- /dev/null +++ b/sound/isa/sb/jazz16.c @@ -0,0 +1,385 @@ + +/* + * jazz16.c - driver for Media Vision Jazz16 based soundcards. + * Copyright (C) 2009 Krzysztof Helt <krzysztof.h1(a)wp.pl> + * Based on patches posted by Rask Ingemann Lambertsen and Rene Herman. + * Based on OSS Sound Blaster driver. + * + * This file is subject to the terms and conditions of the GNU General Public + * License. See the file COPYING in the main directory of this archive for + * more details. + * + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/io.h> +#include <asm/dma.h> +#include <linux/isa.h> +#include <sound/core.h> +#include <sound/mpu401.h> +#include <sound/opl3.h> +#include <sound/sb.h> +#define SNDRV_LEGACY_FIND_FREE_IRQ +#define SNDRV_LEGACY_FIND_FREE_DMA +#include <sound/initval.h> + +#define PFX "jazz16: " + +MODULE_DESCRIPTION("Media Vision Jazz16"); +MODULE_SUPPORTED_DEVICE("{{Media Vision ??? }," + "{RTL,RTL3000}}"); + +MODULE_AUTHOR("Krzysztof Helt <krzysztof.h1(a)wp.pl>"); +MODULE_LICENSE("GPL"); + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ +static unsigned long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +static unsigned long mpu_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; +static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; +static int dma8[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; +static int dma16[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for Media Vision Jazz16 based soundcard."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for Media Vision Jazz16 based soundcard."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable Media Vision Jazz16 based soundcard."); +module_param_array(port, long, NULL, 0444); +MODULE_PARM_DESC(port, "Port # for jazz16 driver."); +module_param_array(mpu_port, long, NULL, 0444); +MODULE_PARM_DESC(mpu_port, "MPU-401 port # for jazz16 driver."); +module_param_array(irq, int, NULL, 0444); +MODULE_PARM_DESC(irq, "IRQ # for jazz16 driver."); +module_param_array(mpu_irq, int, NULL, 0444); +MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for jazz16 driver."); +module_param_array(dma8, int, NULL, 0444); +MODULE_PARM_DESC(dma8, "DMA8 # for jazz16 driver."); +module_param_array(dma16, int, NULL, 0444); +MODULE_PARM_DESC(dma16, "DMA16 # for jazz16 driver."); + +#define SB_JAZZ16_WAKEUP 0xaf +#define SB_JAZZ16_SET_PORTS 0x50 +#define SB_DSP_GET_JAZZ_BRD_REV 0xfa +#define SB_JAZZ16_SET_DMAINTR 0xfb +#define SB_DSP_GET_JAZZ_MODEL 0xfe + +struct snd_card_jazz16 { + struct snd_sb *chip; +}; + +static irqreturn_t jazz16_interrupt(int irq, void *chip) +{ + return snd_sb8dsp_interrupt(chip); +} + +static int __devinit jazz16_configure_ports(unsigned long port, + unsigned long mpu_port, int idx) +{ + unsigned char val; + + if (!request_region(0x201, 1, "jazz16 config")) { + snd_printk(KERN_ERR "config port region is already in use.\n"); + return -EBUSY; + } + outb(SB_JAZZ16_WAKEUP - idx, 0x201); + udelay(100); + outb(SB_JAZZ16_SET_PORTS + idx, 0x201); + udelay(100); + val = port & 0x70; + val |= (mpu_port & 0x30) >> 4; + outb(val, 0x201); + + release_region(0x201, 1); + return 0; +} + +static int __devinit jazz16_detect_board(unsigned long port, + unsigned long mpu_port) +{ + int err; + int val; + struct snd_sb chip; + + if (!request_region(port, 0x10, "jazz16")) { + snd_printk(KERN_ERR "I/O port region is already in use.\n"); + return -EBUSY; + } + /* just to call snd_sbdsp_command/reset/get_byte() */ + chip.port = port; + + err = snd_sbdsp_reset(&chip); + if (err < 0) + for (val = 0; val < 4; val++) { + err = jazz16_configure_ports(port, mpu_port, val); + if (err < 0) + break; + + err = snd_sbdsp_reset(&chip); + if (!err) + break; + } + if (err < 0) { + err = -ENODEV; + goto err_unmap; + } + if (!snd_sbdsp_command(&chip, SB_DSP_GET_JAZZ_BRD_REV)) { + err = -EBUSY; + goto err_unmap; + } + val = snd_sbdsp_get_byte(&chip); + if (val >= 0x30) + snd_sbdsp_get_byte(&chip); + + if ((val & 0xf0) != 0x10) { + err = -ENODEV; + goto err_unmap; + } + if (!snd_sbdsp_command(&chip, SB_DSP_GET_JAZZ_MODEL)) { + err = -EBUSY; + goto err_unmap; + } + snd_sbdsp_get_byte(&chip); + err = snd_sbdsp_get_byte(&chip); + snd_printd("Media Vision Jazz16 board detected: rev 0x%x, model 0x%x\n", + val, err); + + err = 0; + +err_unmap: + release_region(port, 0x10); + return err; +} + +static int __devinit jazz16_configure_board(struct snd_sb *chip, int mpu_irq) +{ + static unsigned char jazz_irq_bits[] = { 0, 0, 2, 3, 0, 1, 0, 4, + 0, 2, 5, 0, 0, 0, 0, 6 }; + static unsigned char jazz_dma_bits[] = { 0, 1, 0, 2, 0, 3, 0, 4 }; + + if (jazz_dma_bits[chip->dma8] == 0 || + jazz_dma_bits[chip->dma16] == 0 || + jazz_irq_bits[chip->irq] == 0) + return -EINVAL; + + if (!snd_sbdsp_command(chip, SB_JAZZ16_SET_DMAINTR)) + return -EBUSY; + + if (!snd_sbdsp_command(chip, + jazz_dma_bits[chip->dma8] | + (jazz_dma_bits[chip->dma16] << 4))) + return -EBUSY; + + if (!snd_sbdsp_command(chip, + jazz_irq_bits[chip->irq] | + (jazz_irq_bits[mpu_irq] << 4))) + return -EBUSY; + + return 0; +} + +static int __devinit snd_jazz16_match(struct device *devptr, unsigned int dev) +{ + if (!enable[dev]) + return 0; + if (port[dev] == SNDRV_AUTO_PORT) { + snd_printk(KERN_ERR "please specify port\n"); + return 0; + } + if (dma16[dev] != SNDRV_AUTO_DMA && + dma16[dev] != 5 && dma16[dev] != 7) { + snd_printk(KERN_ERR "dma16 must be 5 or 7"); + return 0; + } + return 1; +} + +static int __devinit snd_jazz16_probe(struct device *devptr, unsigned int dev) +{ + struct snd_card *card; + struct snd_card_jazz16 *jazz16; + struct snd_sb *chip; + struct snd_opl3 *opl3; + static int possible_irqs[] = {2, 3, 5, 7, 9, 10, 15, -1}; + static int possible_dmas8[] = {1, 3, -1}; + static int possible_dmas16[] = {5, 7, -1}; + int err, xirq, xdma8, xdma16, xmpu_port, xmpu_irq; + + err = snd_card_create(index[dev], id[dev], THIS_MODULE, + sizeof(struct snd_card_jazz16), &card); + if (err < 0) + return err; + + jazz16 = card->private_data; + + xirq = irq[dev]; + if (xirq == SNDRV_AUTO_IRQ) { + xirq = snd_legacy_find_free_irq(possible_irqs); + if (xirq < 0) { + snd_printk(KERN_ERR "unable to find a free IRQ\n"); + err = -EBUSY; + goto err_free; + } + } + xdma8 = dma8[dev]; + if (xdma8 == SNDRV_AUTO_DMA) { + xdma8 = snd_legacy_find_free_dma(possible_dmas8); + if (xdma8 < 0) { + snd_printk(KERN_ERR "unable to find a free DMA8\n"); + err = -EBUSY; + goto err_free; + } + } + xdma16 = dma16[dev]; + if (xdma16 == SNDRV_AUTO_DMA) { + xdma16 = snd_legacy_find_free_dma(possible_dmas16); + if (xdma16 < 0) { + snd_printk(KERN_ERR "unable to find a free DMA16\n"); + err = -EBUSY; + goto err_free; + } + } + + xmpu_port = mpu_port[dev]; + if (xmpu_port == SNDRV_AUTO_PORT) + xmpu_port = 0; + err = jazz16_detect_board(port[dev], xmpu_port); + if (err < 0) { + printk(KERN_ERR "Media Vision Jazz16 board not detected\n"); + goto err_free; + } + err = snd_sbdsp_create(card, port[dev], irq[dev], + jazz16_interrupt, + dma8[dev], dma16[dev], + SB_HW_JAZZ16, + &chip); + if (err < 0) + goto err_free; + + xmpu_irq = mpu_irq[dev]; + if (xmpu_irq == SNDRV_AUTO_IRQ || mpu_port[dev] == SNDRV_AUTO_PORT) + xmpu_irq = 0; + err = jazz16_configure_board(chip, xmpu_irq); + if (err < 0) { + printk(KERN_ERR "Media Vision Jazz16 configuration failed\n"); + goto err_free; + } + + jazz16->chip = chip; + + strcpy(card->driver, "jazz16"); + strcpy(card->shortname, "Media Vision Jazz16"); + sprintf(card->longname, + "Media Vision Jazz16 at 0x%lx, irq %d, dma8 %d, dma16 %d", + port[dev], xirq, xdma8, xdma16); + + err = snd_sb8dsp_pcm(chip, 0, NULL); + if (err < 0) + goto err_free; + err = snd_sbmixer_new(chip); + if (err < 0) + goto err_free; + + err = snd_opl3_create(card, chip->port, chip->port + 2, + OPL3_HW_AUTO, 1, &opl3); + if (err < 0) + snd_printk(KERN_WARNING "no OPL device at 0x%lx-0x%lx\n", + chip->port, chip->port + 2); + else { + err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); + if (err < 0) + goto err_free; + } + if (mpu_port[dev] > 0 && mpu_port[dev] != SNDRV_AUTO_PORT) { + if (mpu_irq[dev] == SNDRV_AUTO_IRQ) + mpu_irq[dev] = -1; + + if (snd_mpu401_uart_new(card, 0, + MPU401_HW_MPU401, + mpu_port[dev], 0, + mpu_irq[dev], + mpu_irq[dev] >= 0 ? IRQF_DISABLED : 0, + NULL) < 0) + snd_printk(KERN_ERR "no MPU-401 device at 0x%lx\n", + mpu_port[dev]); + } + + snd_card_set_dev(card, devptr); + + err = snd_card_register(card); + if (err < 0) + goto err_free; + + dev_set_drvdata(devptr, card); + return 0; + +err_free: + snd_card_free(card); + return err; +} + +static int __devexit snd_jazz16_remove(struct device *devptr, unsigned int dev) +{ + struct snd_card *card = dev_get_drvdata(devptr); + + dev_set_drvdata(devptr, NULL); + snd_card_free(card); + return 0; +} + +#ifdef CONFIG_PM +static int snd_jazz16_suspend(struct device *pdev, unsigned int n, + pm_message_t state) +{ + struct snd_card *card = dev_get_drvdata(pdev); + struct snd_card_jazz16 *acard = card->private_data; + struct snd_sb *chip = acard->chip; + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + snd_pcm_suspend_all(chip->pcm); + snd_sbmixer_suspend(chip); + return 0; +} + +static int snd_jazz16_resume(struct device *pdev, unsigned int n) +{ + struct snd_card *card = dev_get_drvdata(pdev); + struct snd_card_jazz16 *acard = card->private_data; + struct snd_sb *chip = acard->chip; + + snd_sbdsp_reset(chip); + snd_sbmixer_resume(chip); + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + return 0; +} +#endif + +static struct isa_driver snd_jazz16_driver = { + .match = snd_jazz16_match, + .probe = snd_jazz16_probe, + .remove = __devexit_p(snd_jazz16_remove), +#ifdef CONFIG_PM + .suspend = snd_jazz16_suspend, + .resume = snd_jazz16_resume, +#endif + .driver = { + .name = "jazz16" + }, +}; + +static int __init alsa_card_jazz16_init(void) +{ + return isa_register_driver(&snd_jazz16_driver, SNDRV_CARDS); +} + +static void __exit alsa_card_jazz16_exit(void) +{ + isa_unregister_driver(&snd_jazz16_driver); +} + +module_init(alsa_card_jazz16_init) +module_exit(alsa_card_jazz16_exit) diff --git a/sound/isa/sb/sb8_main.c b/sound/isa/sb/sb8_main.c index 658d557..3222aed 100644 --- a/sound/isa/sb/sb8_main.c +++ b/sound/isa/sb/sb8_main.c @@ -106,9 +106,21 @@ static int snd_sb8_playback_prepare(struct snd_pcm_substream *substream) struct snd_sb *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; unsigned int mixreg, rate, size, count; + unsigned char format; + unsigned char stereo = runtime->channels > 1; + int dma; rate = runtime->rate; switch (chip->hardware) { + case SB_HW_JAZZ16: + if (runtime->format == SNDRV_PCM_FORMAT_S16_LE) { + if (chip->mode & SB_MODE_CAPTURE_16) + return -EBUSY; + else + chip->mode |= SB_MODE_PLAYBACK_16; + } + chip->playback_format = SB_DSP_LO_OUTPUT_AUTO; + break; case SB_HW_PRO: if (runtime->channels > 1) { if (snd_BUG_ON(rate != SB8_RATE(11025) && @@ -133,11 +145,21 @@ static int snd_sb8_playback_prepare(struct snd_pcm_substream *substream) default: return -EINVAL; } + if (chip->mode & SB_MODE_PLAYBACK_16) { + format = stereo ? SB_DSP_STEREO_16BIT : SB_DSP_MONO_16BIT; + dma = chip->dma16; + } else { + format = stereo ? SB_DSP_STEREO_8BIT : SB_DSP_MONO_8BIT; + chip->mode |= SB_MODE_PLAYBACK_8; + dma = chip->dma8; + } size = chip->p_dma_size = snd_pcm_lib_buffer_bytes(substream); count = chip->p_period_size = snd_pcm_lib_period_bytes(substream); spin_lock_irqsave(&chip->reg_lock, flags); snd_sbdsp_command(chip, SB_DSP_SPEAKER_ON); - if (runtime->channels > 1) { + if (chip->hardware == SB_HW_JAZZ16) + snd_sbdsp_command(chip, format); + else if (stereo) { /* set playback stereo mode */ spin_lock(&chip->mixer_lock); mixreg = snd_sbmixer_read(chip, SB_DSP_STEREO_SW); @@ -147,15 +169,14 @@ static int snd_sb8_playback_prepare(struct snd_pcm_substream *substream) /* Soundblaster hardware programming reference guide, 3-23 */ snd_sbdsp_command(chip, SB_DSP_DMA8_EXIT); runtime->dma_area[0] = 0x80; - snd_dma_program(chip->dma8, runtime->dma_addr, 1, DMA_MODE_WRITE); + snd_dma_program(dma, runtime->dma_addr, 1, DMA_MODE_WRITE); /* force interrupt */ - chip->mode = SB_MODE_HALT; snd_sbdsp_command(chip, SB_DSP_OUTPUT); snd_sbdsp_command(chip, 0); snd_sbdsp_command(chip, 0); } snd_sbdsp_command(chip, SB_DSP_SAMPLE_RATE); - if (runtime->channels > 1) { + if (stereo) { snd_sbdsp_command(chip, 256 - runtime->rate_den / 2); spin_lock(&chip->mixer_lock); /* save output filter status and turn it off */ @@ -168,13 +189,15 @@ static int snd_sb8_playback_prepare(struct snd_pcm_substream *substream) snd_sbdsp_command(chip, 256 - runtime->rate_den); } if (chip->playback_format != SB_DSP_OUTPUT) { + if (chip->mode & SB_MODE_PLAYBACK_16) + count /= 2; count--; snd_sbdsp_command(chip, SB_DSP_BLOCK_SIZE); snd_sbdsp_command(chip, count & 0xff); snd_sbdsp_command(chip, count >> 8); } spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_dma_program(chip->dma8, runtime->dma_addr, + snd_dma_program(dma, runtime->dma_addr, size, DMA_MODE_WRITE | DMA_AUTOINIT); return 0; } @@ -212,7 +235,6 @@ static int snd_sb8_playback_trigger(struct snd_pcm_substream *substream, snd_sbdsp_command(chip, SB_DSP_SPEAKER_OFF); } spin_unlock_irqrestore(&chip->reg_lock, flags); - chip->mode = (cmd == SNDRV_PCM_TRIGGER_START) ? SB_MODE_PLAYBACK_8 : SB_MODE_HALT; return 0; } @@ -234,9 +256,21 @@ static int snd_sb8_capture_prepare(struct snd_pcm_substream *substream) struct snd_sb *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; unsigned int mixreg, rate, size, count; + unsigned char format; + unsigned char stereo = runtime->channels > 1; + int dma; rate = runtime->rate; switch (chip->hardware) { + case SB_HW_JAZZ16: + if (runtime->format == SNDRV_PCM_FORMAT_S16_LE) { + if (chip->mode & SB_MODE_PLAYBACK_16) + return -EBUSY; + else + chip->mode |= SB_MODE_CAPTURE_16; + } + chip->capture_format = SB_DSP_LO_INPUT_AUTO; + break; case SB_HW_PRO: if (runtime->channels > 1) { if (snd_BUG_ON(rate != SB8_RATE(11025) && @@ -262,14 +296,24 @@ static int snd_sb8_capture_prepare(struct snd_pcm_substream *substream) default: return -EINVAL; } + if (chip->mode & SB_MODE_CAPTURE_16) { + format = stereo ? SB_DSP_STEREO_16BIT : SB_DSP_MONO_16BIT; + dma = chip->dma16; + } else { + format = stereo ? SB_DSP_STEREO_8BIT : SB_DSP_MONO_8BIT; + chip->mode |= SB_MODE_CAPTURE_8; + dma = chip->dma8; + } size = chip->c_dma_size = snd_pcm_lib_buffer_bytes(substream); count = chip->c_period_size = snd_pcm_lib_period_bytes(substream); spin_lock_irqsave(&chip->reg_lock, flags); snd_sbdsp_command(chip, SB_DSP_SPEAKER_OFF); - if (runtime->channels > 1) + if (chip->hardware == SB_HW_JAZZ16) + snd_sbdsp_command(chip, format); + else if (stereo) snd_sbdsp_command(chip, SB_DSP_STEREO_8BIT); snd_sbdsp_command(chip, SB_DSP_SAMPLE_RATE); - if (runtime->channels > 1) { + if (stereo) { snd_sbdsp_command(chip, 256 - runtime->rate_den / 2); spin_lock(&chip->mixer_lock); /* save input filter status and turn it off */ @@ -282,13 +326,15 @@ static int snd_sb8_capture_prepare(struct snd_pcm_substream *substream) snd_sbdsp_command(chip, 256 - runtime->rate_den); } if (chip->capture_format != SB_DSP_INPUT) { + if (chip->mode & SB_MODE_PLAYBACK_16) + count /= 2; count--; snd_sbdsp_command(chip, SB_DSP_BLOCK_SIZE); snd_sbdsp_command(chip, count & 0xff); snd_sbdsp_command(chip, count >> 8); } spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_dma_program(chip->dma8, runtime->dma_addr, + snd_dma_program(dma, runtime->dma_addr, size, DMA_MODE_READ | DMA_AUTOINIT); return 0; } @@ -328,7 +374,6 @@ static int snd_sb8_capture_trigger(struct snd_pcm_substream *substream, snd_sbdsp_command(chip, SB_DSP_SPEAKER_OFF); } spin_unlock_irqrestore(&chip->reg_lock, flags); - chip->mode = (cmd == SNDRV_PCM_TRIGGER_START) ? SB_MODE_CAPTURE_8 : SB_MODE_HALT; return 0; } @@ -339,13 +384,21 @@ irqreturn_t snd_sb8dsp_interrupt(struct snd_sb *chip) snd_sb_ack_8bit(chip); switch (chip->mode) { - case SB_MODE_PLAYBACK_8: /* ok.. playback is active */ + case SB_MODE_PLAYBACK_16: /* ok.. playback is active */ + if (chip->hardware != SB_HW_JAZZ16) + break; + /* fallthru */ + case SB_MODE_PLAYBACK_8: substream = chip->playback_substream; runtime = substream->runtime; if (chip->playback_format == SB_DSP_OUTPUT) snd_sb8_playback_trigger(substream, SNDRV_PCM_TRIGGER_START); snd_pcm_period_elapsed(substream); break; + case SB_MODE_CAPTURE_16: + if (chip->hardware != SB_HW_JAZZ16) + break; + /* fallthru */ case SB_MODE_CAPTURE_8: substream = chip->capture_substream; runtime = substream->runtime; @@ -361,10 +414,15 @@ static snd_pcm_uframes_t snd_sb8_playback_pointer(struct snd_pcm_substream *subs { struct snd_sb *chip = snd_pcm_substream_chip(substream); size_t ptr; + int dma; - if (chip->mode != SB_MODE_PLAYBACK_8) + if (chip->mode & SB_MODE_PLAYBACK_8) + dma = chip->dma8; + else if (chip->mode & SB_MODE_PLAYBACK_16) + dma = chip->dma16; + else return 0; - ptr = snd_dma_pointer(chip->dma8, chip->p_dma_size); + ptr = snd_dma_pointer(dma, chip->p_dma_size); return bytes_to_frames(substream->runtime, ptr); } @@ -372,10 +430,15 @@ static snd_pcm_uframes_t snd_sb8_capture_pointer(struct snd_pcm_substream *subst { struct snd_sb *chip = snd_pcm_substream_chip(substream); size_t ptr; + int dma; - if (chip->mode != SB_MODE_CAPTURE_8) + if (chip->mode & SB_MODE_CAPTURE_8) + dma = chip->dma8; + else if (chip->mode & SB_MODE_CAPTURE_16) + dma = chip->dma16; + else return 0; - ptr = snd_dma_pointer(chip->dma8, chip->c_dma_size); + ptr = snd_dma_pointer(dma, chip->c_dma_size); return bytes_to_frames(substream->runtime, ptr); } @@ -446,6 +509,13 @@ static int snd_sb8_open(struct snd_pcm_substream *substream) runtime->hw = snd_sb8_capture; } switch (chip->hardware) { + case SB_HW_JAZZ16: + runtime->hw.formats |= SNDRV_PCM_FMTBIT_S16_LE; + runtime->hw.rates |= SNDRV_PCM_RATE_8000_48000; + runtime->hw.rate_min = 4000; + runtime->hw.rate_max = 50000; + runtime->hw.channels_max = 2; + break; case SB_HW_PRO: runtime->hw.rate_max = 44100; runtime->hw.channels_max = 2; @@ -468,6 +538,14 @@ static int snd_sb8_open(struct snd_pcm_substream *substream) } snd_pcm_hw_constraint_ratnums(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_clock); + if (chip->dma8 > 3 || chip->dma16 >= 0) { + snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 2); + snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 2); + runtime->hw.buffer_bytes_max = 128 * 1024 * 1024; + runtime->hw.period_bytes_max = 128 * 1024 * 1024; + } return 0; } @@ -480,6 +558,10 @@ static int snd_sb8_close(struct snd_pcm_substream *substream) chip->capture_substream = NULL; spin_lock_irqsave(&chip->open_lock, flags); chip->open &= ~SB_OPEN_PCM; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + chip->mode &= ~SB_MODE_PLAYBACK; + else + chip->mode &= ~SB_MODE_CAPTURE; spin_unlock_irqrestore(&chip->open_lock, flags); return 0; } @@ -515,6 +597,7 @@ int snd_sb8dsp_pcm(struct snd_sb *chip, int device, struct snd_pcm ** rpcm) struct snd_card *card = chip->card; struct snd_pcm *pcm; int err; + size_t max_prealloc = 64 * 1024; if (rpcm) *rpcm = NULL; @@ -527,9 +610,11 @@ int snd_sb8dsp_pcm(struct snd_sb *chip, int device, struct snd_pcm ** rpcm) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_sb8_playback_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_sb8_capture_ops); + if (chip->dma8 > 3 || chip->dma16 >= 0) + max_prealloc = 128 * 1024; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_isa_data(), - 64*1024, 64*1024); + 64*1024, max_prealloc); if (rpcm) *rpcm = pcm; diff --git a/sound/isa/sb/sb_common.c b/sound/isa/sb/sb_common.c index 27a6515..eae6c1c 100644 --- a/sound/isa/sb/sb_common.c +++ b/sound/isa/sb/sb_common.c @@ -170,6 +170,9 @@ static int snd_sbdsp_probe(struct snd_sb * chip) case SB_HW_CS5530: str = "16 (CS5530)"; break; + case SB_HW_JAZZ16: + str = "Pro (Jazz16)"; + break; default: return -ENODEV; } diff --git a/sound/isa/sb/sb_mixer.c b/sound/isa/sb/sb_mixer.c index 8cfc41f..6496822 100644 --- a/sound/isa/sb/sb_mixer.c +++ b/sound/isa/sb/sb_mixer.c @@ -779,6 +779,7 @@ int snd_sbmixer_new(struct snd_sb *chip) return err; break; case SB_HW_PRO: + case SB_HW_JAZZ16: if ((err = snd_sbmixer_init(chip, snd_sbpro_controls, ARRAY_SIZE(snd_sbpro_controls), @@ -929,6 +930,7 @@ void snd_sbmixer_suspend(struct snd_sb *chip) save_mixer(chip, sb20_saved_regs, ARRAY_SIZE(sb20_saved_regs)); break; case SB_HW_PRO: + case SB_HW_JAZZ16: save_mixer(chip, sbpro_saved_regs, ARRAY_SIZE(sbpro_saved_regs)); break; case SB_HW_16: @@ -955,6 +957,7 @@ void snd_sbmixer_resume(struct snd_sb *chip) restore_mixer(chip, sb20_saved_regs, ARRAY_SIZE(sb20_saved_regs)); break; case SB_HW_PRO: + case SB_HW_JAZZ16: restore_mixer(chip, sbpro_saved_regs, ARRAY_SIZE(sbpro_saved_regs)); break; case SB_HW_16: -- 1.6.4 ---------------------------------------------------------------------- Sprawdz sekrety swojego biorytmu. Kliknij >> http://link.interia.pl/f2534

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[alsa-devel] [PATCH] fix incorrect rounding direction in snd_interval_ratnum()
by Krzysztof Helt 21 Dec '09

21 Dec '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> The direction of rounding is incorrect in the snd_interval_ratnum() It was detected with following parameters (sb8 driver playing 8kHz stereo file): - num is always 1000000 - requested frequency rate is from 7999 to 7999 (single frequency) The first loop calculates div_down(num, freq->min) which is 125. Thus, a frequency range's minimum value is 1000000 / 125 = 8000 Hz. The second loop calculates div_up(num, freq->max) which is 126 The frequency range's maximum value is 1000000 / 126 = 7936 Hz. The range maximum is lower than the range minimum so the function fails due to empty result range. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- sound/core/pcm_lib.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 30f4108..a27545b 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -758,7 +758,7 @@ int snd_interval_ratnum(struct snd_interval *i, int diff; if (q == 0) q = 1; - den = div_down(num, q); + den = div_up(num, q); if (den < rats[k].den_min) continue; if (den > rats[k].den_max) @@ -794,7 +794,7 @@ int snd_interval_ratnum(struct snd_interval *i, i->empty = 1; return -EINVAL; } - den = div_up(num, q); + den = div_down(num, q); if (den > rats[k].den_max) continue; if (den < rats[k].den_min) -- 1.6.4 ---------------------------------------------------------------------- Co Ci sie dzisiaj snilo? Czy wiesz, co oznacza Twoj sen? Sprawdz w senniku >> http://link.interia.pl/f2532

2 1

[alsa-devel] [PATCH] HDA: add powersaving hook for Realtek
by Hector Martin 21 Dec '09

21 Dec '09

The current Realtek code makes no specific provision for turning stuff off. The codec chip is placed into low-power mode generically, but this doesn't turn off any external hardware connected to it, in particular external amplifiers. This patch creates a hook function that is called by the codec suspend/resume functions. It ought to disable any external hardware in a device-specific way. I've implemented a generic ALC889 function that sets the EAPD pin properly, and used it for the Acer Aspire 8930G which can benefit from this feature. On my laptop, this results in ~0.5W extra savings. -- Hector Martin (hector(a)marcansoft.com) Public Key: http://www.marcansoft.com/marcan.asc Signed-off-by: Hector Martin <hector(a)marcansoft.com> --- linux-2.6.32/sound/pci/hda/patch_realtek_orig.c 2009-12-20 19:31:08.940900751 +0100 +++ linux-2.6.32/sound/pci/hda/patch_realtek.c 2009-12-20 22:14:11.852025021 +0100 @@ -334,6 +334,9 @@ /* hooks */ void (*init_hook)(struct hda_codec *codec); void (*unsol_event)(struct hda_codec *codec, unsigned int res); +#ifdef CONFIG_SND_HDA_POWER_SAVE + void (*power_hook)(struct hda_codec *codec, int power); +#endif /* for pin sensing */ unsigned int sense_updated: 1; @@ -385,6 +388,7 @@ void (*init_hook)(struct hda_codec *); #ifdef CONFIG_SND_HDA_POWER_SAVE struct hda_amp_list *loopbacks; + void (*power_hook)(struct hda_codec *codec, int power); #endif }; @@ -897,6 +901,7 @@ spec->unsol_event = preset->unsol_event; spec->init_hook = preset->init_hook; #ifdef CONFIG_SND_HDA_POWER_SAVE + spec->power_hook = preset->power_hook; spec->loopback.amplist = preset->loopbacks; #endif @@ -1824,6 +1845,16 @@ spec->autocfg.speaker_pins[2] = 0x1b; } +#ifdef CONFIG_SND_HDA_POWER_SAVE +static void alc889_power_eapd(struct hda_codec *codec, int power) +{ + snd_hda_codec_write(codec, 0x14, 0, + AC_VERB_SET_EAPD_BTLENABLE, power ? 2 : 0); + snd_hda_codec_write(codec, 0x15, 0, + AC_VERB_SET_EAPD_BTLENABLE, power ? 2 : 0); +} +#endif + /* * ALC880 3-stack model * @@ -3611,12 +3642,25 @@ snd_hda_detach_beep_device(codec); } +#ifdef CONFIG_SND_HDA_POWER_SAVE +static int alc_suspend(struct hda_codec *codec, pm_message_t state) +{ + struct alc_spec *spec = codec->spec; + if (spec && spec->power_hook) + spec->power_hook(codec, 0); + return 0; +} +#endif + #ifdef SND_HDA_NEEDS_RESUME static int alc_resume(struct hda_codec *codec) { + struct alc_spec *spec = codec->spec; codec->patch_ops.init(codec); snd_hda_codec_resume_amp(codec); snd_hda_codec_resume_cache(codec); + if (spec && spec->power_hook) + spec->power_hook(codec, 1); return 0; } #endif @@ -3633,6 +3677,7 @@ .resume = alc_resume, #endif #ifdef CONFIG_SND_HDA_POWER_SAVE + .suspend = alc_suspend, .check_power_status = alc_check_power_status, #endif }; @@ -9320,6 +9365,9 @@ .unsol_event = alc_automute_amp_unsol_event, .setup = alc889_acer_aspire_8930g_setup, .init_hook = alc_automute_amp, +#ifdef CONFIG_SND_HDA_POWER_SAVE + .power_hook = alc889_power_eapd, +#endif }, [ALC888_ACER_ASPIRE_7730G] = { .mixers = { alc883_3ST_6ch_mixer,

2 2

[alsa-devel] [PATCH] HDA: remove useless mixers on Aspire 8930G
by Hector Martin 21 Dec '09

21 Dec '09

This patch removes some extra mixers that do nothing on the Acer Aspire 8930G. The CD mixer is useless because the SATA DVD/Blu-Ray drive has no analog audio output, and the Side mixer is useless because we max out at 6ch anyway. -- Hector Martin (hector(a)marcansoft.com) Public Key: http://www.marcansoft.com/marcan.asc Signed-off-by: Hector Martin <hector(a)marcansoft.com> --- linux-2.6.32/sound/pci/hda/patch_realtek_orig.c 2009-12-20 19:31:08.940900751 +0100 +++ linux-2.6.32/sound/pci/hda/patch_realtek.c 2009-12-20 22:14:11.852025021 +0100 @@ -1780,6 +1782,25 @@ { } /* end */ }; +static struct snd_kcontrol_new alc889_acer_aspire_8930g_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, + HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("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 void alc888_acer_aspire_4930g_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -9283,7 +9283,7 @@ .init_hook = alc_automute_amp, }, [ALC888_ACER_ASPIRE_8930G] = { - .mixers = { alc888_base_mixer, + .mixers = { alc889_acer_aspire_8930g_mixer, alc883_chmode_mixer }, .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs, alc889_acer_aspire_8930g_verbs },

2 1

[alsa-devel] [PATCH] HDA: simplify Aspire 8930G verb array
by Hector Martin 21 Dec '09

21 Dec '09

This patch just simplifies the 8930G verb array a bit. Just use the common ALC889 EAPD verb array to make things more consistent. The file is already huge enough already. -- Hector Martin (hector(a)marcansoft.com) Public Key: http://www.marcansoft.com/marcan.asc Signed-off-by: Hector Martin <hector(a)marcansoft.com> --- linux-2.6.32/sound/pci/hda/patch_realtek_orig.c 2009-12-20 19:31:08.940900751 +0100 +++ linux-2.6.32/sound/pci/hda/patch_realtek.c 2009-12-20 22:14:11.852025021 +0100 @@ -1665,9 +1670,6 @@ /* some bit here disables the other DACs. Init=0x4900 */ {0x20, AC_VERB_SET_COEF_INDEX, 0x08}, {0x20, AC_VERB_SET_PROC_COEF, 0x0000}, -/* Enable amplifiers */ - {0x14, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, - {0x15, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, /* DMIC fix * This laptop has a stereo digital microphone. The mics are only 1cm apart * which makes the stereo useless. However, either the mic or the ALC889 @@ -9305,7 +9305,8 @@ .mixers = { alc889_acer_aspire_8930g_mixer, alc883_chmode_mixer }, .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs, - alc889_acer_aspire_8930g_verbs }, + alc889_acer_aspire_8930g_verbs, + alc889_eapd_verbs}, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .num_adc_nids = ARRAY_SIZE(alc889_adc_nids),

2 1

Re: [alsa-devel] [2.6.33-rc1] [SOUND] HDA: high cpu usage, noise
by Takashi Iwai 21 Dec '09

21 Dec '09

At Sun, 20 Dec 2009 18:27:45 +0100, Maciej Rutecki wrote: > > Hi > > Affected kernel: 2.6.33-rc1 > Last known good: 2.6.32 The only fundamental change, AFAIK, is the use of MSI as default. Could you try to add enable_msi=0 option to snd-hda-intel? thanks, Takashi

1 0

[alsa-devel] [PATCH] ALSA: hda: Set Front Mic to input vref 50% for Lenovo 3000 Y410
by Daniel Chen 21 Dec '09

21 Dec '09

Hi, Attached is a patch against master HEAD setting input vref 50% for the ALC262_LENOVO_3000 model. Thanks, Dan

2 1

[alsa-devel] [PATCH] ALSA: pcm: Properly close timer when freeing slaves
by Daniel Chen 21 Dec '09

21 Dec '09

Hi, Here's a patch originally from Dan McCombs against alsa-lib.git HEAD fixing a fairly nasty fd leak for timers. Thanks, Dan

2 1

[alsa-devel] No device for DAI omap-eac-dai error
by Belisko Marek 21 Dec '09

21 Dec '09

Hi, I'm writing new omap dai (for EAC codec base omap's). My code is based on existing omap-mcbsp.c code basically. After loadingof module I got error which appear in init function when call snd_soc_register_dai(). I browse for code but can't figure out where dai->dev should be set to don't get error message and init dai properly. Thanks for any suggestions Marek

3 2

[alsa-devel] Installing a "different" version of Alsa
by Brian Welch 20 Dec '09

20 Dec '09

Hello, It's been a year since I posted to this list, but I'm running into a problem and can't figure out the best way to solve it, so I'm hoping someone here can help. I've got an SDK for an embedded platform that includes a specific version of Linux with ALSA built-in. On my platform, I want to use a different (newer) version of ALSA than the built-in version, so I want to build and load the binaries from the newer version and, more importantly, I want to compile my custom sound hardware driver against the newer alsa version and the SDK's Linux version. I don't really want to touch the kernel files that are in the kerenl/include/sound directory in the SDK. When I use kbuild to build my custom sound hardware driver against the SDK in the kernel, it always trys to pick up the older ALSA header files that are in the SDK, but I want to use the newer versions that are in the directory with the newer version of ALSA. What the best way to make sure I get the correct headers? Is what I'm trying "dangerous" and should I be going about this a different way? Thanks for any advice, Brian

2 1

[alsa-devel] ASoC updates for 2.6.33
by Mark Brown 19 Dec '09

19 Dec '09

The following changes since commit 1cf86f6f9b000e98c1b7f866f99633ae67464e2f: Kuninori Morimoto (1): ASoC: ak4642: Add default return value in ak4642_modinit are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.33 Guennadi Liakhovetski (1): ASoC: wm8974: fix a wrong bit definition Jon Smirl (1): ASoC: Fix disable of SPDIF on STAC9766 codec sound/soc/codecs/stac9766.c | 18 +----------------- sound/soc/codecs/wm8974.c | 2 +- 2 files changed, 2 insertions(+), 18 deletions(-)

2 3

[alsa-devel] [PATCH 0/3] McPDM driver patch series
by Candelaria Villareal, Jorge 19 Dec '09

19 Dec '09

McPDM module is the interface between the OMAP and TWL6030 audio codec. It transfers data between the two chips using a Pulse Density Modulation protocol, which allows sound to be played or recorded at speeds of up to 96000 Hz. Besides data, McPDM can transfer commands and read status to/from the codec using the frame synchronization line. The interface supports the following: * Up to 5 downlink channels and 3 uplink channels * 88.2 and 96 KHz * 32 bit sample size Currently, driver is set to support up to 4 downlink channels. If 2 channels are selected, output will go to the headset, and if 4 channels are selected, output will go to the headset and speakers. Functionality to select the output at runtime will be added afterwards. Command and status mode is not supported in current patches. -- Jorge Eduardo Candelaria (2): OMAP4: IRQ: Add McPDM IRQ definition. ASoC: OMAP4: Add support for McPDM Misael Lopez Cruz (1): ASoC: OMAP4: Add McPDM platform driver arch/arm/plat-omap/include/plat/irqs.h | 2 +- sound/soc/omap/Kconfig | 3 + sound/soc/omap/Makefile | 2 + sound/soc/omap/mcpdm.c | 505 ++++++++++++++++++++++++++++++++ sound/soc/omap/mcpdm.h | 156 ++++++++++ sound/soc/omap/omap-mcpdm.c | 248 ++++++++++++++++ sound/soc/omap/omap-mcpdm.h | 29 ++ 7 files changed, 944 insertions(+), 1 deletions(-) create mode 100644 sound/soc/omap/mcpdm.c create mode 100644 sound/soc/omap/mcpdm.h create mode 100644 sound/soc/omap/omap-mcpdm.c create mode 100644 sound/soc/omap/omap-mcpdm.h

2 1

[alsa-devel] [PATCH 7/7] sound: Use kzalloc for allocating only one thing
by Julia Lawall 19 Dec '09

19 Dec '09

From: Julia Lawall <julia(a)diku.dk> Use kzalloc rather than kcalloc(1,...) The semantic patch that makes this change is as follows: (http://coccinelle.lip6.fr/) // <smpl> @@ @@ - kcalloc(1, + kzalloc( ...) // </smpl> Signed-off-by: Julia Lawall <julia(a)diku.dk> --- sound/isa/msnd/msnd_midi.c | 2 +- sound/pci/hda/patch_realtek.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff -u -p a/sound/isa/msnd/msnd_midi.c b/sound/isa/msnd/msnd_midi.c --- a/sound/isa/msnd/msnd_midi.c 2009-03-28 23:34:55.000000000 +0100 +++ b/sound/isa/msnd/msnd_midi.c 2009-12-19 07:52:27.000000000 +0100 @@ -162,7 +162,7 @@ int snd_msndmidi_new(struct snd_card *ca err = snd_rawmidi_new(card, "MSND-MIDI", device, 1, 1, &rmidi); if (err < 0) return err; - mpu = kcalloc(1, sizeof(*mpu), GFP_KERNEL); + mpu = kzalloc(sizeof(*mpu), GFP_KERNEL); if (mpu == NULL) { snd_device_free(card, rmidi); return -ENOMEM; diff -u -p a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c --- a/sound/pci/hda/patch_realtek.c 2009-12-17 08:30:15.000000000 +0100 +++ b/sound/pci/hda/patch_realtek.c 2009-12-19 07:52:50.000000000 +0100 @@ -12857,7 +12857,7 @@ static int patch_alc268(struct hda_codec int board_config; int i, has_beep, err; - spec = kcalloc(1, sizeof(*spec), GFP_KERNEL); + spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) return -ENOMEM;

2 1

[alsa-devel] [PATCH 0/5] AACI: cleanups
by Russell King - ARM Linux 18 Dec '09

18 Dec '09

This patch series is in preparation of adding DMA support to the AACI driver. I'd appreciate it if ALSA people could do me the curtesy of copying me on patches they dream up on ALSA drivers maintained by me, rather than running rough-shod over work that they may have queued - especially when they say "you can also do this" "yes I know, I have a patch" and then they do it anyway. If this is how they want to operate, I'll avoid working on any ALSA stuff in the future. Its just not worth me wasting my time.

2 6

[alsa-devel] Help needed to develop quirks for USB Sound device for Roland Edirol PC-80 MIDI controller
by Alain Lauzon 18 Dec '09

18 Dec '09

Heloo I just bought a Roland Edirol PC-80 MIDI controller and found out that it was not supported on LINUX. I looked on the web and found the /usr/src/alsa/alsa-driver-xxx/usbquirks.h file to be the place to change some code to support it. I have tried many things but I don't enough info to get it right once and for all. Here is what I did : { /* PC-80 Wave */ USB_DEVICE(0x0582, 0x0092), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { /* .vendor_name = "EDIROL", */ /* .product_name = "PC-80", */ .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 0, .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = 1, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { .format = SNDRV_PCM_FORMAT_S24_3LE, .channels = 2, .iface = 0, .altsetting = 1, .altset_idx = 1, .attributes = 0, .endpoint = 0x81, .ep_attr = 0x01, .rates = SNDRV_PCM_RATE_CONTINUOUS, .rate_min = 48000, .rate_max = 48000, //(96?) } }, { .ifnum = 2, .type = QUIRK_MIDI_FIXED_ENDPOINT, .data = & (const struct snd_usb_midi_endpoint_info) { .out_cables = 0x0007, .in_cables = 0x0007 } }, { .ifnum = -1 } } } What I got working is the audio out but cannot have the audio working. Is there some resources somewhere that I could read to understand what I need to do? Thank you, Alain Lauzon -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

2 5

[alsa-devel] ASoC updates for 2.6.34
by Mark Brown 18 Dec '09

18 Dec '09

The following changes since commit bc2580061e42c323d7777029f01318f395edac0d: Julia Lawall (1): ASoC: Correct code taking the size of a pointer are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.34 Nothing terribly exciting here, almost all of it is a bunch of new drivers. Chaithrika U S (1): ASoC: DaVinci: Update suspend/resume support for McASP driver Daniel Mack (1): ASoC: Add regulator support to CS4270 codec driver Guennadi Liakhovetski (1): ASoC: wm8974: fix a wrong bit definition Ilkka Koskinen (2): ASoC: tlv320dac33: Add support for regulator framework ASoC: tpa6130a2: Add support for regulator framework Jassi Brar (2): ASoC: S3C64XX: Remove unnecessary header includes ASoC: S3C64XX: Compress and generalize the CPU driver Jon Smirl (1): ASoC: Fix disable of SPDIF on STAC9766 codec Kuninori Morimoto (4): ASoC: sh_fsi: avoid using global variable ASoC: ak4642: Add default return value in ak4642_modinit ASoC: Add DA7210 codec device support for ALSA ASoC: Add FSI-DA7210 sound support for SuperH Mark Brown (12): Merge branch 'for-2.6.33' into for-2.6.34 ASoC: Export snd_soc_update_bits_unlocked() ASoC: Display the power register in DAPM widget debugfs ASoC: Initial WM8904 CODEC driver ASoC: Push registers out of mixer power decision ASoC: Sort DAPM sequences by CODEC as well ASoC: Register the CODEC in WM8727 ASoC: Fix sorting of codecs Makefile entries ASoC: Fix build of DA7210 ASoC: Add initial WM8955 CODEC driver ASoC: Set AIF word length for WM8904 Merge branch 'for-2.6.33' into for-2.6.34 Peter Meerwald (1): ASoC: PLL computation in TLV320AIC3x SoC driver include/sound/soc.h | 3 + include/sound/wm8904.h | 57 + include/sound/wm8955.h | 26 + sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 10 +- sound/soc/codecs/ak4642.c | 2 +- sound/soc/codecs/cs4270.c | 43 +- sound/soc/codecs/da7210.c | 589 +++++++++ sound/soc/codecs/da7210.h | 24 + sound/soc/codecs/stac9766.c | 18 +- sound/soc/codecs/tlv320aic3x.c | 75 +- sound/soc/codecs/tlv320dac33.c | 92 ++- sound/soc/codecs/tpa6130a2.c | 87 +- sound/soc/codecs/wm8727.c | 66 +- sound/soc/codecs/wm8904.c | 2555 +++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8904.h | 1681 ++++++++++++++++++++++++ sound/soc/codecs/wm8955.c | 1151 +++++++++++++++++ sound/soc/codecs/wm8955.h | 489 +++++++ sound/soc/codecs/wm8974.c | 2 +- sound/soc/davinci/davinci-mcasp.c | 18 +- sound/soc/davinci/davinci-mcasp.h | 1 + sound/soc/davinci/davinci-pcm.c | 2 +- sound/soc/s3c24xx/s3c64xx-i2s.c | 120 +-- sound/soc/sh/Kconfig | 8 + sound/soc/sh/Makefile | 2 + sound/soc/sh/fsi-da7210.c | 83 ++ sound/soc/sh/fsi.c | 115 +- sound/soc/soc-core.c | 7 +- sound/soc/soc-dapm.c | 37 +- 29 files changed, 7127 insertions(+), 248 deletions(-) create mode 100644 include/sound/wm8904.h create mode 100644 include/sound/wm8955.h create mode 100644 sound/soc/codecs/da7210.c create mode 100644 sound/soc/codecs/da7210.h create mode 100644 sound/soc/codecs/wm8904.c create mode 100644 sound/soc/codecs/wm8904.h create mode 100644 sound/soc/codecs/wm8955.c create mode 100644 sound/soc/codecs/wm8955.h create mode 100644 sound/soc/sh/fsi-da7210.c

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[alsa-devel] Play multiple sounds concurrently
by Markus Luttenberger 18 Dec '09

18 Dec '09

Hey, I want to play multiple sounds concurrently with ALSA. The audio data is mono and G.711 a-law encoded. I can successfully play a single source of audio data but the sound is distorted when I try to play two sources. Basically, this is what I'm doing: -----8<----------------- char buf1[1024]; // filled with audio data from source 1 char buf2[1024]; // filled with audio data from source 2 char wr_buf[1024]; for (int count = 0; count < 1024; count++) wr_buf[count] = (char) (0.5 * buf1[count]) + (char) (0.5 * buf2[count]); snd_pcm_writei(dev_handle, wr_buf, 1024); -----8<----------------- I think I have to use a scale value to correct the values or else clipping occures. However, this is the problem. How do I have to modify the audio bytes in order to add them together and write them into one buffer? greetings.. ..Markus

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[alsa-devel] alsa device drain power
by Michael Trimarchi 18 Dec '09

18 Dec '09

Hi all, I have a simple question is the general behavior that an alsa device start to drain power when it starts to play sound or start to recording. What happen if there is an external amplifier connect to sound device? Michael

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[alsa-devel] [PATCH] wm8974: fix a wrong bit definition
by Guennadi Liakhovetski 18 Dec '09

18 Dec '09

The wm8974 datasheet defines BUFIOEN as bit 2. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> --- no idea if this would break anything, so, please test. Maybe the correct fix would be to actually use a correct name for bit 4, but who wants to enable Mic bias when putting bias into standby?... diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index eff2933..fb99433 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -47,7 +47,7 @@ static const u16 wm8974_reg[WM8974_CACHEREGNUM] = { }; #define WM8974_POWER1_BIASEN 0x08 -#define WM8974_POWER1_BUFIOEN 0x10 +#define WM8974_POWER1_BUFIOEN 0x04 struct wm8974_priv { struct snd_soc_codec codec;

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[alsa-devel] 2.6.32-gitX: oops in ac97 codec registration
by Manuel Lauss 18 Dec '09

18 Dec '09

Hi Mark, current 2.6.32-git spits out the following oops when I load my machine driver with the generic AC97 codec: No device for DAI au1xpsc_ac97 AC97 SoC Audio Codec 0.6 asoc: AC97 HiFi <-> au1xpsc_ac97 mapping ok CPU 0 Unable to handle kernel paging request at virtual address 000001f0, epc == 80366320, ra == 803662c0 Oops[#1]: Cpu 0 $ 0 : 00000000 10003c00 00000000 00000000 $ 4 : 00000000 00004b39 00000001 00004b39 $ 8 : 00000000 00000002 8041a7a0 8d140af0 $12 : 00000004 ad55ad55 00000003 00000001 $16 : 00000000 c1179a94 8d1f2520 c007b420 $20 : 8d49cfe0 8d1f2520 804fc03c 804fc130 $24 : 00000000 0000ad55 $28 : 8d590000 8d591af0 00000000 803662c0 Hi : 00000000 Lo : 00000004 epc : 80366320 snd_soc_new_pcms+0x300/0x358 Not tainted ra : 803662c0 snd_soc_new_pcms+0x2a0/0x358 Status: 10003c03 KERNEL EXL IE Cause : 0080800c BadVA : 000001f0 PrId : 04030202 (Au1250) Modules linked in: snd_soc_exm32cexx(+) snd_soc_cs4251x snd_soc_ac97 snd_ac97_codec snd_soc_au1xpsc_ac97 snd_soc_au1xpsc_i2s Process modprobe (pid: 673, threadinfo=8d590000, task=8d430938, tls=2aacf2d0) Stack : 8d591b30 c007b3c4 c0074afc 00010000 00000001 8d591b08 8d17d1c0 37394341 46694820 43412069 48203739 2d694669 8d170030 804fc16c 8d17d2c8 0000001c 10003c01 8d17d2c8 804fc16c 8d17d2c8 10003c01 8d17d2d0 00000024 80500000 c0074b58 00000001 c11799f4 00000000 8d1f2528 00000001 fffffff4 8d1f2520 c11798f0 00000001 8d17d0c0 00000001 804fc130 00000000 804f4980 c007b288 ... Call Trace: [<80366320>] snd_soc_new_pcms+0x300/0x358 [<c007b288>] ac97_soc_probe+0x108/0x1d0 [snd_soc_ac97] [<80365108>] snd_soc_instantiate_cards+0x22c/0x694 [<803655ec>] soc_probe+0x7c/0xc4 [<802c73b8>] driver_probe_device+0xbc/0x194 [<802c6748>] bus_for_each_drv+0x60/0xb0 [<802c7638>] device_attach+0x60/0x88 [<802c6524>] bus_probe_device+0x30/0x54 [<802c4d74>] device_add+0x3a4/0x53c [<802c8e48>] platform_device_add+0x14c/0x1b8 [<c11794ec>] cexx_audio_probe+0x310/0x374 [snd_soc_exm32cexx] [<802c73b8>] driver_probe_device+0xbc/0x194 [<802c74fc>] __driver_attach+0x6c/0xa4 [<802c6afc>] bus_for_each_dev+0x60/0xb0 [<802c6264>] bus_add_driver+0xc8/0x268 [<802c78ac>] driver_register+0xdc/0x194 [<8010a010>] do_one_initcall+0x68/0x1f4 [<80154268>] sys_init_module+0xec/0x23c [<801022a0>] stack_done+0x20/0x3c Oops is inside snd_ac97_dev_add_pdata() call: seems ac97->dev is NULL. I'm not entirely sure that this isn't somehow the fault of my machine driver; has the procedure to register the asoc device changed since 2.6.32? Thanks! Manuel Lauss

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[alsa-devel] [PATCH 3/3] ASoC: OMAP4: Add McPDM platform driver
by Candelaria Villareal, Jorge 17 Dec '09

17 Dec '09

McPDM platform driver is configured to use sDMA in order to transfer to/from memory. Support for interfacing with ABE will be added later. McPDM dai currently supports up to 4 downlink channels and 2 uplink channels simultaneously, as well as 88.2 and 96 KHz, and a sample size of 32 bits. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> Signed-off-by: Margarita Olaya <x0080101(a)ti.com> Signed-off-by: Jorge Eduardo Candelaria <x0107209(a)ti.com> --- sound/soc/omap/Makefile | 2 +- sound/soc/omap/omap-mcpdm.c | 248 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/omap/omap-mcpdm.h | 29 +++++ 3 files changed, 278 insertions(+), 1 deletions(-) create mode 100644 sound/soc/omap/omap-mcpdm.c create mode 100644 sound/soc/omap/omap-mcpdm.h diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index bf8b388..d6382fa 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -1,7 +1,7 @@ # OMAP Platform Support snd-soc-omap-objs := omap-pcm.o snd-soc-omap-mcbsp-objs := omap-mcbsp.o -snd-soc-omap-mcpdm-objs := mcpdm.o +snd-soc-omap-mcpdm-objs := mcpdm.o omap-mcpdm.o obj-$(CONFIG_SND_OMAP_SOC) += snd-soc-omap.o obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c new file mode 100644 index 0000000..9dc0ff5 --- /dev/null +++ b/sound/soc/omap/omap-mcpdm.c @@ -0,0 +1,248 @@ +/* + * omap-mcpdm.c -- OMAP ALSA SoC DAI driver using McPDM port + * + * Copyright (C) 2009 Texas Instruments + * + * Author: Misael Lopez Cruz <x0052729(a)ti.com> + * Contact: Jorge Eduardo Candelaria <x0107209(a)ti.com> + * Margarita Olaya <magi.olaya(a)ti.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include <mach/control.h> +#include <mach/dma.h> +#include <mach/mcbsp.h> +#include "mcpdm.h" +#include "omap-mcpdm.h" +#include "omap-pcm.h" + +struct omap_mcpdm_data { + struct omap_mcpdm_link *links; + int active; +}; + +static struct omap_mcpdm_link omap_mcpdm_links[] = { + /* downlink */ + { + .irq_mask = DN_IRQ_EMTPY | DN_IRQ_FULL, + .threshold = 1, + .format = PDMOUTFORMAT_LJUST, + }, + /* uplink */ + { + .irq_mask = UP_IRQ_EMPTY | UP_IRQ_FULL, + .threshold = 1, + .format = PDMOUTFORMAT_LJUST, + }, +}; + +static struct omap_mcpdm_data mcpdm_data = { + .links = omap_mcpdm_links, + .active = 0, +}; + +/* + * Stream DMA parameters + */ +static struct omap_pcm_dma_data omap_mcpdm_dai_dma_params[] = { + { + .name = "Audio downlink", + .dma_req = OMAP44XX_DMA_MCPDM_DL, + .data_type = OMAP_DMA_DATA_TYPE_S32, + .sync_mode = OMAP_DMA_SYNC_ELEMENT, + .port_addr = OMAP44XX_MCPDM_L3_BASE + MCPDM_DN_DATA, + }, + { + .name = "Audio uplink", + .dma_req = OMAP44XX_DMA_MCPDM_UP, + .data_type = OMAP_DMA_DATA_TYPE_S32, + .sync_mode = OMAP_DMA_SYNC_ELEMENT, + .port_addr = OMAP44XX_MCPDM_BASE + MCPDM_UP_DATA, + }, +}; + +static int omap_mcpdm_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int err = 0; + + if (!cpu_dai->active) + err = omap_mcpdm_request(); + + return err; +} + +static void omap_mcpdm_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + if (!cpu_dai->active) + omap_mcpdm_free(); +} + +static int omap_mcpdm_dai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct omap_mcpdm_data *mcpdm_priv = cpu_dai->private_data; + int stream = substream->stream; + int err = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (!mcpdm_priv->active++) + omap_mcpdm_start(stream); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (!--mcpdm_priv->active) + omap_mcpdm_stop(stream); + break; + default: + err = -EINVAL; + } + + return err; +} + +static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct omap_mcpdm_data *mcpdm_priv = cpu_dai->private_data; + struct omap_mcpdm_link *mcpdm_links = mcpdm_priv->links; + int stream = substream->stream; + int channels, err, link_mask = 0; + + cpu_dai->dma_data = &omap_mcpdm_dai_dma_params[stream]; + + channels = params_channels(params); + switch (channels) { + case 4: + if (stream) + /* up to 2 channels for uplink */ + return -EINVAL; + link_mask |= 1 << 3; + case 3: + if (stream) + /* up to 2 channels for uplink */ + return -EINVAL; + link_mask |= 1 << 2; + case 2: + link_mask |= 1 << 1; + case 1: + link_mask |= 1 << 0; + break; + default: + /* unsupported number of channels */ + return -EINVAL; + } + + if (stream) { + mcpdm_links[stream].channels = link_mask << 0; + err = omap_mcpdm_set_uplink(&mcpdm_links[stream]); + } else { + mcpdm_links[stream].channels = link_mask << 3; + err = omap_mcpdm_set_downlink(&mcpdm_links[stream]); + } + + return err; +} + +static int omap_mcpdm_dai_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct omap_mcpdm_data *mcpdm_priv = cpu_dai->private_data; + struct omap_mcpdm_link *mcpdm_links = mcpdm_priv->links; + int err; + + if (substream->stream) + err = omap_mcpdm_clr_uplink(&mcpdm_links[substream->stream]); + else + err = omap_mcpdm_clr_downlink(&mcpdm_links[substream->stream]); + + return err; +} + +static struct snd_soc_dai_ops omap_mcpdm_dai_ops = { + .startup = omap_mcpdm_dai_startup, + .shutdown = omap_mcpdm_dai_shutdown, + .trigger = omap_mcpdm_dai_trigger, + .hw_params = omap_mcpdm_dai_hw_params, + .hw_free = omap_mcpdm_dai_hw_free, +}; + +#define OMAP_MCPDM_RATES (SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) +#define OMAP_MCPDM_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_dai omap_mcpdm_dai = { + .name = "omap-mcpdm", + .id = -1, + .playback = { + .channels_min = 1, + .channels_max = 4, + .rates = OMAP_MCPDM_RATES, + .formats = OMAP_MCPDM_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = OMAP_MCPDM_RATES, + .formats = OMAP_MCPDM_FORMATS, + }, + .ops = &omap_mcpdm_dai_ops, + .private_data = &mcpdm_data, +}; +EXPORT_SYMBOL_GPL(omap_mcpdm_dai); + +static int __init snd_omap_mcpdm_init(void) +{ + return snd_soc_register_dai(&omap_mcpdm_dai); +} +module_init(snd_omap_mcpdm_init); + +static void __exit snd_omap_mcpdm_exit(void) +{ + snd_soc_unregister_dai(&omap_mcpdm_dai); +} +module_exit(snd_omap_mcpdm_exit); + +MODULE_AUTHOR("Misael Lopez Cruz <x0052729(a)ti.com>"); +MODULE_DESCRIPTION("OMAP PDM SoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap-mcpdm.h b/sound/soc/omap/omap-mcpdm.h new file mode 100644 index 0000000..73b80d5 --- /dev/null +++ b/sound/soc/omap/omap-mcpdm.h @@ -0,0 +1,29 @@ +/* + * omap-mcpdm.h + * + * Copyright (C) 2009 Texas Instruments + * + * Contact: Misael Lopez Cruz <x0052729(a)ti.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __OMAP_MCPDM_H__ +#define __OMAP_MCPDM_H__ + +extern struct snd_soc_dai omap_mcpdm_dai; + +#endif /* End of __OMAP_MCPDM_H__ */ -- 1.6.0.4

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[alsa-devel] [PATCH 1/3] OMAP4: IRQ: Add McPDM IRQ definition.
by Candelaria Villareal, Jorge 17 Dec '09

17 Dec '09

Support for McPDM IRQs in OMAP4 Signed-off-by: Jorge Eduardo Candelaria <x0107209(a)ti.com> --- arch/arm/plat-omap/include/plat/irqs.h | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/arch/arm/plat-omap/include/plat/irqs.h b/arch/arm/plat-omap/include/plat/irqs.h index 97d6c50..7fb2464 100644 --- a/arch/arm/plat-omap/include/plat/irqs.h +++ b/arch/arm/plat-omap/include/plat/irqs.h @@ -432,7 +432,7 @@ #define INT_44XX_PARTHASH_IRQ (79 + IRQ_GIC_START) #define INT_44XX_MMC3_IRQ (94 + IRQ_GIC_START) #define INT_44XX_MMC4_IRQ (96 + IRQ_GIC_START) - +#define INT_44XX_MCPDM_IRQ (112 + IRQ_GIC_START) /* Max. 128 level 2 IRQs (OMAP1610), 192 GPIOs (OMAP730/850) and * 16 MPUIO lines */ -- 1.6.0.4

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Re: [alsa-devel] [alsa-cvslog] alsa-driver: ALSA driver repository - compatibility code, new drivers and standalone building branch, master now at v1.0.22-1-g79c453f
by Takashi Iwai 17 Dec '09

17 Dec '09

At Thu, 17 Dec 2009 17:41:37 +0100 (CET), noreply-git(a)alsa-project.org wrote: > > Also, missing isa/dt019x.c is added. The driver was removed in the recent version. Takashi

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[alsa-devel] PLL computation in TLV320AIC3x SoC driver
by Peter Meerwald 17 Dec '09

17 Dec '09

Hello, I'm trying to use the SoC TLV320AIC3x codec driver with sysclk 16384000 and ran into some problems with setting PLL; below is a patch against linux-2.6-asoc note that the original code uses variables pll_r and pll_p instead of the loop variable r and p to compute tmp, this seems broken further, the original code does not respect the constraints on j (>= 4, <= 55 for d==0) according to the codec's datasheet, and similarly for d!=0 I've tested the code with a number of reasonable sysclk values and got sane PLL values; please apply if acceptable thanks, regards, p. diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 3395cf9..e84e473 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -766,9 +766,10 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = socdev->card->codec; struct aic3x_priv *aic3x = codec->private_data; int codec_clk = 0, bypass_pll = 0, fsref, last_clk = 0; - u8 data, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1; - u16 pll_d = 1; + u8 data, j, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1; + u16 d, pll_d = 1; u8 reg; + int clk; /* select data word length */ data = @@ -835,47 +836,62 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, return 0; /* Use PLL - * find an apropriate setup for j, d, r and p by iterating over - * p and r - j and d are calculated for each fraction. - * Up to 128 values are probed, the closest one wins the game. + * find an appropriate setup for j, d, r and p by iterating over + * p, r and j first, then trying to compute the fraction d. + * Up to 6528 values are probed, the closest one wins the game. * The sysclk is divided by 1000 to prevent integer overflows. */ codec_clk = (2048 * fsref) / (aic3x->sysclk / 1000); - for (r = 1; r <= 16; r++) - for (p = 1; p <= 8; p++) { - int clk, tmp = (codec_clk * pll_r * 10) / pll_p; - u8 j = tmp / 10000; - u16 d = tmp % 10000; - - if (j > 63) - continue; - - if (d != 0 && aic3x->sysclk < 10000000) - continue; - - /* This is actually 1000 * ((j + (d/10000)) * r) / p - * The term had to be converted to get rid of the - * division by 10000 */ - clk = ((10000 * j * r) + (d * r)) / (10 * p); - - /* check whether this values get closer than the best - * ones we had before */ - if (abs(codec_clk - clk) < abs(codec_clk - last_clk)) { - pll_j = j; pll_d = d; pll_r = r; pll_p = p; - last_clk = clk; - } - - /* Early exit for exact matches */ - if (clk == codec_clk) - break; - } + for (r = 1; r <= 16; r++) + for (p = 1; p <= 8; p++) { + for (j = 4; j <= 55; j++) { + /* This is actually 1000 * ((j + (d/10000)) * r) / p + * The term had to be converted to get rid of the + * division by 10000; d = 0 here */ + int clk = (1000 * j * r) / p; + + /* check whether this values get closer than the best + * ones we had before */ + if (abs(codec_clk - clk) < abs(codec_clk - last_clk)) { + pll_j = j; pll_d = 0; pll_r = r; pll_p = p; + last_clk = clk; + } + + /* Early exit for exact matches */ + if (clk == codec_clk) + goto found; + } + } + + /* try with d != 0 */ + for (p = 1; p <= 8; p++) { + j = codec_clk * p / 1000; + + if (j < 4 || j > 11) continue; + + /* do not use codec_clk here since we'd loose precision */ + d = ((2048 * fsref * 10) / (aic3x->sysclk / 1000)) % 10000; + clk = (10000 * j + d) / (10 * p); + + /* check whether this values get closer than the best + * ones we had before */ + if (abs(codec_clk - clk) < abs(codec_clk - last_clk)) { + pll_j = j; pll_d = d; pll_r = 1; pll_p = 1; + last_clk = clk; + } + + /* Early exit for exact matches */ + if (clk == codec_clk) + goto found; + } if (last_clk == 0) { printk(KERN_ERR "%s(): unable to setup PLL\n", __func__); return -EINVAL; } +found: data = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG); aic3x_write(codec, AIC3X_PLL_PROGA_REG, data | (pll_p << PLLP_SHIFT)); aic3x_write(codec, AIC3X_OVRF_STATUS_AND_PLLR_REG, pll_r << PLLR_SHIFT); -- Peter Meerwald Kaigasse 3 / 8 A-5020 Salzburg / AUSTRIA +43-664-2444418 (mobile)

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[alsa-devel] [PATCH] sound/oss/pss: Fix test of unsigned in pss_reset_dsp() and pss_download_boot()
by Roel Kluin 17 Dec '09

17 Dec '09

limit and jiffies are unsigned so the test does not work. Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- Found using coccinelle: http://coccinelle.lip6.fr/ diff --git a/sound/oss/pss.c b/sound/oss/pss.c index 83f5ee2..9325f4d 100644 --- a/sound/oss/pss.c +++ b/sound/oss/pss.c @@ -269,7 +269,7 @@ static int pss_reset_dsp(pss_confdata * devc) unsigned long i, limit = jiffies + HZ/10; outw(0x2000, REG(PSS_CONTROL)); - for (i = 0; i < 32768 && (limit-jiffies >= 0); i++) + for (i = 0; i < 32768 && (limit >= jiffies); i++) inw(REG(PSS_CONTROL)); outw(0x0000, REG(PSS_CONTROL)); return 1; @@ -369,11 +369,11 @@ static int pss_download_boot(pss_confdata * devc, unsigned char *block, int size outw(0, REG(PSS_DATA)); limit = jiffies + HZ/10; - for (i = 0; i < 32768 && (limit - jiffies >= 0); i++) + for (i = 0; i < 32768 && (limit >= jiffies); i++) val = inw(REG(PSS_STATUS)); limit = jiffies + HZ/10; - for (i = 0; i < 32768 && (limit-jiffies >= 0); i++) + for (i = 0; i < 32768 && (limit >= jiffies); i++) { val = inw(REG(PSS_STATUS)); if (val & 0x4000)

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[alsa-devel] Two patches for Dell Vostro 1015 (CX20583)
by Einar Rünkaru 17 Dec '09

17 Dec '09

Hi. Attached two patches: 1. Corrected initialization of Conexant CX20583-10Z - forgot proper initialization of internal mic. Int mic was muted until external mic was plugged/unplugged. Also added control for internal mic boost. I renamed 'analog mic boost' to 'ext mic boost'. Alternative was to use names analog-digital, which seemed very confusing. If the renaming breaks something, it can be dropped. 2. Made use of the chips beep device. Now it's possible to change volume of beep. (It was too loud) To make use of it in Ubuntu 9.04 pcspkr must be blacklisted. Einar Rünkaru

2 1

[alsa-devel] [PATCH] ASoC: Fix build of DA7210
by Mark Brown 17 Dec '09

17 Dec '09

From: Mark Brown <broonie(a)finisterre.wolfsonmicro.main> DAC_VOICE_EN was not defined - looks to have been overly enthusiastically deleted from a previous revision of the patch, pull the value from v1. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/da7210.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 14f5f34..fbf3ab4 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -81,6 +81,9 @@ #define DA7210_ADC_L_EN (1 << 3) #define DA7210_ADC_R_EN (1 << 7) +/* DAC_HPF fields */ +#define DA7210_DAC_VOICE_EN (1 << 7) + /* DAC_SEL bit fields */ #define DA7210_DAC_L_SRC_DAI_L (4 << 0) #define DA7210_DAC_L_EN (1 << 3) -- 1.6.5.4

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[alsa-devel] ALSA 1.0.22 release
by Jaroslav Kysela 17 Dec '09

17 Dec '09

Hello all, the ALSA 1.0.22 packages (except alsa-oss and alsa-firmware) are available for download at http://www.alsa-project.org . The changes are listed in this URL: http://www.alsa-project.org/main/index.php/Changes_v1.0.21_v1.0.22 Jaroslav Kysela ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

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[alsa-devel] [PATCH 2/2] ALSA: pcm - specify amount of data needed in wait_for_avail_min
by jassisinghbrar＠gmail.com 17 Dec '09

17 Dec '09

From: Jassi Brar <jassi.brar(a)samsung.com> For better accuracy, make the waiting time more specific. That helps avoiding users of wait_for_avail_min(i.e, snd_pcm_lib_write1/read1) to wait until atleast avail_min amount is available, whereas all that might be needed is just 1 frame(for example). Since the waiting is not for avail_min, the name has been changed to reflect that. The users of the new function are also updated to pass exact minimum data needed. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/core/pcm_lib.c | 12 +++++++----- 1 files changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index ee2d7b9..4a92804 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1664,12 +1664,12 @@ void snd_pcm_period_elapsed(struct snd_pcm_substream *substream) EXPORT_SYMBOL(snd_pcm_period_elapsed); /* - * Wait until avail_min data becomes available + * Wait until atleast the requested amount(*availp) of data becomes available * Returns a negative error code if any error occurs during operation. * The available space is stored on availp. When err = 0 and avail = 0 * on the capture stream, it indicates the stream is in DRAINING state. */ -static int wait_for_avail_min(struct snd_pcm_substream *substream, +static int wait_for_min(struct snd_pcm_substream *substream, snd_pcm_uframes_t *availp) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -1719,7 +1719,7 @@ static int wait_for_avail_min(struct snd_pcm_substream *substream, avail = snd_pcm_playback_avail(runtime); else avail = snd_pcm_capture_avail(runtime); - if (avail >= runtime->control->avail_min) + if (avail >= *availp) break; } _endloop: @@ -1794,7 +1794,8 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, err = -EAGAIN; goto _end_unlock; } - err = wait_for_avail_min(substream, &avail); + avail = size; + err = wait_for_min(substream, &avail); if (err < 0) goto _end_unlock; } @@ -2016,7 +2017,8 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, err = -EAGAIN; goto _end_unlock; } - err = wait_for_avail_min(substream, &avail); + avail = size; + err = wait_for_min(substream, &avail); if (err < 0) goto _end_unlock; if (!avail) -- 1.6.2.5

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[alsa-devel] build alsa-tools using static alsa lib? linux2.6.28+marvell littleton
by 王剑 17 Dec '09

17 Dec '09

alsa-develHello everybody: i couldnot build alsa-tools, see as follows, could anybody help me, thanks! question : a@a-desktop:~/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools$ make arm-linux-gcc -I/home/a/NMS/kernel/2009WW21/pxalinux/../../2009WW21_build//src/preview-kit/linux/include -I/home/a/NMS/kernel/2009WW21/pxalinux/../../2009WW21_build//rel/host/include -L/home/a/NMS/kernel/2009WW21/pxalinux/../../2009WW21_build//rel/host/lib -L/home/a/NMS/kernel/2009WW21/pxalinux/../../2009WW21_build//rel/target/rootfs/lib -g -Waggregate-return -Wmissing-noreturn -W -Wall -mcpu=iwmmxt -mtune=iwmmxt -mabi=aapcs-linux -I/home/a/NMS/kernel/2009WW21/pxalinux/../../2009WW21_build//rel/host/include -static -L/home/a/NMS/kernel/2009WW21/pxalinux/../../2009WW21_build//rel/host/include -lasound control.c -o control control.c: In function 'main': control.c:72: warning: function call has aggregate value /tmp/ccIFwS0o.o: In function `main': /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:15: undefined reference to `snd_ctl_card_info_sizeof' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:15: undefined reference to `snd_ctl_card_info_sizeof' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:16: undefined reference to `snd_pcm_info_sizeof' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:16: undefined reference to `snd_pcm_info_sizeof' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:17: undefined reference to `snd_rawmidi_info_sizeof' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:17: undefined reference to `snd_rawmidi_info_sizeof' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:21: undefined reference to `snd_card_next' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:22: undefined reference to `snd_strerror' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:28: undefined reference to `snd_ctl_open' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:29: undefined reference to `snd_strerror' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:32: undefined reference to `snd_ctl_card_info' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:33: undefined reference to `snd_strerror' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:37: undefined reference to `snd_ctl_card_info_get_card' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:38: undefined reference to `snd_ctl_card_info_get_id' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:39: undefined reference to `snd_ctl_card_info_get_driver' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:40: undefined reference to `snd_ctl_card_info_get_name' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:41: undefined reference to `snd_ctl_card_info_get_longname' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:42: undefined reference to `snd_ctl_card_info_get_mixername' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:43: undefined reference to `snd_ctl_card_info_get_components' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:47: undefined reference to `snd_ctl_pcm_next_device' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:48: undefined reference to `snd_strerror' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:53: undefined reference to `snd_pcm_info_set_device' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:54: undefined reference to `snd_pcm_info_set_subdevice' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:55: undefined reference to `snd_pcm_info_set_stream' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:56: undefined reference to `snd_ctl_pcm_info' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:57: undefined reference to `snd_strerror' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:61: undefined reference to `snd_pcm_info_get_device' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:62: undefined reference to `snd_pcm_info_get_subdevice' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:63: undefined reference to `snd_pcm_info_get_stream' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:64: undefined reference to `snd_pcm_info_get_card' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:65: undefined reference to `snd_pcm_info_get_id' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:66: undefined reference to `snd_pcm_info_get_name' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:67: undefined reference to `snd_pcm_info_get_subdevice_name' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:68: undefined reference to `snd_pcm_info_get_class' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:69: undefined reference to `snd_pcm_info_get_subclass' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:70: undefined reference to `snd_pcm_info_get_subdevices_count' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:71: undefined reference to `snd_pcm_info_get_subdevices_avail' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:72: undefined reference to `snd_pcm_info_get_sync' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:77: undefined reference to `snd_ctl_rawmidi_next_device' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:78: undefined reference to `snd_strerror' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:83: undefined reference to `snd_rawmidi_info_set_device' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:84: undefined reference to `snd_rawmidi_info_set_subdevice' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:85: undefined reference to `snd_rawmidi_info_set_stream' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:86: undefined reference to `snd_ctl_rawmidi_info' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:87: undefined reference to `snd_strerror' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:91: undefined reference to `snd_rawmidi_info_get_device' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:92: undefined reference to `snd_rawmidi_info_get_subdevice' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:93: undefined reference to `snd_rawmidi_info_get_stream' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:94: undefined reference to `snd_rawmidi_info_get_card' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:95: undefined reference to `snd_rawmidi_info_get_flags' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:96: undefined reference to `snd_rawmidi_info_get_id' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:97: undefined reference to `snd_rawmidi_info_get_name' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:98: undefined reference to `snd_rawmidi_info_get_subdevice_name' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:99: undefined reference to `snd_rawmidi_info_get_subdevices_count' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:100: undefined reference to `snd_rawmidi_info_get_subdevices_avail' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:102: undefined reference to `snd_ctl_close' /home/a/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools/control.c:105: undefined reference to `snd_config_update_free_global' collect2: ld returned 1 exit status make: *** [control] 错误 1 a@a-desktop:~/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools$ conditions: 1.Had added one line in Makefile of alsa-tools directory: a@a-desktop:~/NMS/kernel/2009WW21_build/src/preview-kit/alsa-tools$ vim Makefile LDFLAGS=-static 2.Had modified Makefile of directory: a@a-desktop:~/NMS/kernel/2009WW21_build/src/preview-kit/alsa-lib$ vim Makefile ./configure --prefix=$(PXA_HOST_DIR) --enable-static=yes 3.after make alsa-lib, generated libasound.a a@a-desktop:~/NMS/kernel/2009WW21_build/src/preview-kit/alsa-lib$ make a@a-desktop:~/NMS/kernel/2009WW21_build/rel/host/lib$ ls libasound.a and so on. Best regards! Wangjian --------------------------------------------------------------------------------------------------- Confidentiality Notice: The information contained in this e-mail and any accompanying attachment(s) is intended only for the use of the intended recipient and may be confidential and/or privileged of Neusoft Corporation, its subsidiaries and/or its affiliates. If any reader of this communication is not the intended recipient, unauthorized use, forwarding, printing, storing, disclosure or copying is strictly prohibited, and may be unlawful.If you have received this communication in error,please immediately notify the sender by return e-mail, and delete the original message and all copies from your system. Thank you. ---------------------------------------------------------------------------------------------------

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[alsa-devel] CELF Project Proposal - ALSA scenario/use case support
by Liam Girdwood 17 Dec '09

17 Dec '09

Summary: Complete ALSA hardware scenario/use case support in alsa-lib and salsa-lib Proposer: Liam Girdwood Description: Modern mobile devices are increasingly required to provide a rich set of audio functionality required by today's applications. ALSA currently lacks a portable and high level API to configure audio hardware function and signal routing. e.g. there is no portable method atm to configure the audio hardware to make a "handset gsm phone call" or "handset voip phone call", etc. This purpose of this project would be to complete the ALSA scenario manager API and get it accepted upstream in alsa-lib and salsa-lib. The code is nearly ready for upstream, with some work required on it's file format as the main piece of outstanding work. More Info: ** http://www.slimlogic.co.uk/?p=40 API header: http://git.alsa-project.org/?p=alsa-lib.git;a=blob;f=include/ascenario.h;h=… Development branch now hosted as alsa-lib branch here :- ** http://git.alsa-project.org/?p=alsa-lib.git;a=shortlog;h=ascenario Scope: This project will probably take between 2 - 4 weeks. Thanks Liam

2 1

[alsa-devel] [PATCH] Fix disable of SPDIF on STAC9766 codec
by Jon Smirl 16 Dec '09

16 Dec '09

Change code so that switching to playing music through the analog output disables SPDIF out instead of disabling it when stream ends. Signed-off-by: Jon Smirl <jonsmirl(a)gmail.com> --- sound/soc/codecs/stac9766.c | 18 +----------------- 1 files changed, 1 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index bbc72c2..81b8c9d 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -191,6 +191,7 @@ static int ac97_analog_prepare(struct snd_pcm_substream *substream, vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); vra |= 0x1; /* enable variable rate audio */ + vra &= ~0x4; /* disable SPDIF output */ stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra); @@ -221,22 +222,6 @@ static int ac97_digital_prepare(struct snd_pcm_substream *substream, return stac9766_ac97_write(codec, reg, runtime->rate); } -static int ac97_digital_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *dai) -{ - struct snd_soc_codec *codec = dai->codec; - unsigned short vra; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_STOP: - vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); - vra &= !0x04; - stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra); - break; - } - return 0; -} - static int stac9766_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -315,7 +300,6 @@ static struct snd_soc_dai_ops stac9766_dai_ops_analog = { static struct snd_soc_dai_ops stac9766_dai_ops_digital = { .prepare = ac97_digital_prepare, - .trigger = ac97_digital_trigger, }; struct snd_soc_dai stac9766_dai[] = {

2 1