April 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] ALC269VB config for recording on multiple pcm streams.
by Shrirang Bagul 26 Apr '11

26 Apr '11

Hi, I need to record on multiple (four) channels for a beam-forming application. The sound card is Intel-HDA controller with Realtek ALC269VB codec chip.I am able to create two pcm sub-streams (stereo) for capturing; however, the recorded audio is always mono. The linux kernel version is 2.6.38-rc7. The changes to the patch_realtek.c are attached along with alsa-info.sh output. The $arecord -l command output is: **** List of CAPTURE Hardware Devices ****card 0: Intel [HDA Intel], device 0: ALC269VB Analog [ALC269VB Analog] Subdevices: 1/1 Subdevice #0: subdevice #0card 0: Intel [HDA Intel], device 2: ALC269VB Analog [ALC269VB Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 Can anyone help me resolve this problem? Regards,Shrirang.

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[alsa-devel] M-audio Fast Track Pro
by Andreas Brudin 26 Apr '11

26 Apr '11

Hi all, I was wondering how one would go about trying to get 24-bit support for the M-audio Fast Track Pro merged into the kernel? A (by now rather old) patch exists to enable 24-bit support, and Joe Giampaoli has written an excellent howto at http://www.joegiampaoli.com/blog/?p=462 but it is still hard for many users to build a custom kernel. It is a nice low-to-mid-range audio interface for hobby musicians, and it is still available in the stores, and I guess it is quite common out there, so it would be great if it became supported out of the box. Regards, Andreas Berheim Brudin

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[alsa-devel] [PATCH] ASoC: Move WM8962 FLL configuration to CODEC
by Mark Brown 26 Apr '11

26 Apr '11

There's only one DAI anyway. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 5 ++--- 1 files changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 7d6ace3..bc84a1a 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3172,10 +3172,9 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref, return 0; } -static int wm8962_set_fll(struct snd_soc_dai *dai, int fll_id, int source, +static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source, unsigned int Fref, unsigned int Fout) { - struct snd_soc_codec *codec = dai->codec; struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); struct _fll_div fll_div; unsigned long timeout; @@ -3280,7 +3279,6 @@ static struct snd_soc_dai_ops wm8962_dai_ops = { .hw_params = wm8962_hw_params, .set_sysclk = wm8962_set_dai_sysclk, .set_fmt = wm8962_set_dai_fmt, - .set_pll = wm8962_set_fll, .digital_mute = wm8962_mute, }; @@ -3931,6 +3929,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8962 = { .reg_cache_default = wm8962_reg, .volatile_register = wm8962_volatile_register, .readable_register = wm8962_readable_register, + .set_pll = wm8962_set_fll, }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) -- 1.7.4.1

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[alsa-devel] [PATCH] ASoC: Don't warn if the WM8962 SYSCLK FLL setting doesn't match reality
by Mark Brown 26 Apr '11

26 Apr '11

When bringing up audio low power modes boards may configure SYSCLK before they actually start the FLL as we do much of the clocking setup prior to the power up sequence. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index d6b78b2..7d6ace3 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2991,7 +2991,6 @@ static int wm8962_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, case WM8962_SYSCLK_FLL: wm8962->sysclk = WM8962_SYSCLK_FLL; src = 1 << WM8962_SYSCLK_SRC_SHIFT; - WARN_ON(freq != wm8962->fll_fout); break; default: return -EINVAL; -- 1.7.4.1

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[alsa-devel] [PATCH] ASoC: Support FLL lock interrupt on WM8962
by Mark Brown 26 Apr '11

26 Apr '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 16 +++++++++++++++- 1 files changed, 15 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 500011e..d6b78b2 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -58,6 +58,7 @@ struct wm8962_priv { int bclk; /* Desired BCLK */ int lrclk; + struct completion fll_lock; int fll_src; int fll_fref; int fll_fout; @@ -3178,6 +3179,7 @@ static int wm8962_set_fll(struct snd_soc_dai *dai, int fll_id, int source, struct snd_soc_codec *codec = dai->codec; struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); struct _fll_div fll_div; + unsigned long timeout; int ret; int fll1 = snd_soc_read(codec, WM8962_FLL_CONTROL_1) & WM8962_FLL_ENA; @@ -3244,6 +3246,11 @@ static int wm8962_set_fll(struct snd_soc_dai *dai, int fll_id, int source, dev_dbg(codec->dev, "FLL configured for %dHz->%dHz\n", Fref, Fout); + /* This should be a massive overestimate */ + timeout = msecs_to_jiffies(1); + + wait_for_completion_timeout(&wm8962->fll_lock, timeout); + wm8962->fll_fref = Fref; wm8962->fll_fout = Fout; wm8962->fll_src = source; @@ -3340,6 +3347,11 @@ static irqreturn_t wm8962_irq(int irq, void *data) active = snd_soc_read(codec, WM8962_INTERRUPT_STATUS_2); active &= ~mask; + if (active & WM8962_FLL_LOCK_EINT) { + dev_dbg(codec->dev, "FLL locked\n"); + complete(&wm8962->fll_lock); + } + if (active & WM8962_FIFOS_ERR_EINT) dev_err(codec->dev, "FIFO error\n"); @@ -3712,6 +3724,7 @@ static int wm8962_probe(struct snd_soc_codec *codec) wm8962->codec = codec; INIT_DELAYED_WORK(&wm8962->mic_work, wm8962_mic_work); + init_completion(&wm8962->fll_lock); codec->cache_sync = 1; codec->dapm.idle_bias_off = 1; @@ -3868,9 +3881,10 @@ static int wm8962_probe(struct snd_soc_codec *codec) i2c->irq, ret); /* Non-fatal */ } else { - /* Enable error reporting IRQs by default */ + /* Enable some IRQs by default */ snd_soc_update_bits(codec, WM8962_INTERRUPT_STATUS_2_MASK, + WM8962_FLL_LOCK_EINT | WM8962_TEMP_SHUT_EINT | WM8962_FIFOS_ERR_EINT, 0); } -- 1.7.4.1

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[alsa-devel] [PATCH v2] ASoC: tegra: TrimSlice machine support
by Mike Rapoport 26 Apr '11

26 Apr '11

Signed-off-by: Mike Rapoport <mike(a)compulab.co.il> --- v2 changes: * address Stephen and Mark comments * rebase against Mark's for-next branch * switch to data based specification of widgets and routes * explicitly disable unused codec pins sound/soc/tegra/Kconfig | 9 ++ sound/soc/tegra/Makefile | 2 + sound/soc/tegra/trimslice.c | 228 +++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 239 insertions(+), 0 deletions(-) create mode 100644 sound/soc/tegra/trimslice.c diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 14f7119..183b118 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -26,3 +26,12 @@ config SND_SOC_TEGRA_WM8903 boards using the WM8093 codec. Currently, the supported boards are Harmony, Ventana, Seaboard, Kaen, and Aebl. +config SND_TEGRA_SOC_TRIMSLICE + tristate "SoC Audio support for TrimSlice board" + depends on SND_SOC_TEGRA && MACH_TRIMSLICE && I2C + default m + select SND_TEGRA_SOC_I2S + select SND_SOC_TLV320AIC23 + help + Say Y or M here if you want to add support for SoC audio on the + TrimSlice platform. diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile index 13bef8d..0a3d981 100644 --- a/sound/soc/tegra/Makefile +++ b/sound/soc/tegra/Makefile @@ -11,5 +11,7 @@ obj-$(CONFIG_SND_SOC_TEGRA_I2S) += snd-soc-tegra-i2s.o # Tegra machine Support snd-soc-tegra-wm8903-objs := tegra_wm8903.o +snd-soc-tegra-trimslice-objs := trimslice.o obj-$(CONFIG_SND_SOC_TEGRA_WM8903) += snd-soc-tegra-wm8903.o +obj-$(CONFIG_SND_TEGRA_SOC_TRIMSLICE) += snd-soc-tegra-trimslice.o diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c new file mode 100644 index 0000000..8fc07e9 --- /dev/null +++ b/sound/soc/tegra/trimslice.c @@ -0,0 +1,228 @@ +/* + * trimslice.c - TrimSlice machine ASoC driver + * + * Copyright (C) 2011 - CompuLab, Ltd. + * Author: Mike Rapoport <mike(a)compulab.co.il> + * + * Based on code copyright/by: + * Author: Stephen Warren <swarren(a)nvidia.com> + * Copyright (C) 2010-2011 - NVIDIA, Inc. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <asm/mach-types.h> + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> + +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "../codecs/tlv320aic23.h" + +#include "tegra_das.h" +#include "tegra_i2s.h" +#include "tegra_pcm.h" +#include "tegra_asoc_utils.h" + +#define DRV_NAME "tegra-snd-trimslice" + +struct tegra_trimslice { + struct tegra_asoc_utils_data util_data; +}; + +static int trimslice_asoc_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_card *card = codec->card; + struct tegra_trimslice *trimslice = snd_soc_card_get_drvdata(card); + int srate, mclk; + int err; + + srate = params_rate(params); + mclk = 128 * srate; + + err = tegra_asoc_utils_set_rate(&trimslice->util_data, srate, mclk); + if (err < 0) { + dev_err(card->dev, "Can't configure clocks\n"); + return err; + } + + err = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); + if (err < 0) { + dev_err(card->dev, "codec_dai fmt not set\n"); + return err; + } + + err = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); + if (err < 0) { + dev_err(card->dev, "cpu_dai fmt not set\n"); + return err; + } + + err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, + SND_SOC_CLOCK_IN); + if (err < 0) { + dev_err(card->dev, "codec_dai clock not set\n"); + return err; + } + + return 0; +} + +static struct snd_soc_ops trimslice_asoc_ops = { + .hw_params = trimslice_asoc_hw_params, +}; + +static const struct snd_soc_dapm_widget trimslice_dapm_widgets[] = { + SND_SOC_DAPM_HP("Line Out", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), +}; + +static const struct snd_soc_dapm_route trimslice_audio_map[] = { + {"Line Out", NULL, "LOUT"}, + {"Line Out", NULL, "ROUT"}, + + {"LLINEIN", NULL, "Line In"}, + {"RLINEIN", NULL, "Line In"}, +}; + +static int trimslice_asoc_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + + snd_soc_dapm_nc_pin(dapm, "LHPOUT"); + snd_soc_dapm_nc_pin(dapm, "RHPOUT"); + snd_soc_dapm_nc_pin(dapm, "MICIN"); + + snd_soc_dapm_sync(dapm); + + return 0; +} + +static struct snd_soc_dai_link trimslice_tlv320aic23_dai = { + .name = "TLV320AIC23", + .stream_name = "AIC23", + .codec_name = "tlv320aic23-codec.2-001a", + .platform_name = "tegra-pcm-audio", + .cpu_dai_name = "tegra-i2s.0", + .codec_dai_name = "tlv320aic23-hifi", + .init = trimslice_asoc_init, + .ops = &trimslice_asoc_ops, +}; + +static struct snd_soc_card snd_soc_trimslice = { + .name = "tegra-trimslice", + .dai_link = &trimslice_tlv320aic23_dai, + .num_links = 1, + + .dapm_widgets = trimslice_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(trimslice_dapm_widgets), + .dapm_routes = trimslice_audio_map, + .num_dapm_routes = ARRAY_SIZE(trimslice_audio_map), +}; + +static __devinit int tegra_snd_trimslice_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &snd_soc_trimslice; + struct tegra_trimslice *trimslice; + int ret; + + trimslice = kzalloc(sizeof(struct tegra_trimslice), GFP_KERNEL); + if (!trimslice) { + dev_err(&pdev->dev, "Can't allocate tegra_trimslice\n"); + return -ENOMEM; + } + + ret = tegra_asoc_utils_init(&trimslice->util_data, &pdev->dev); + if (ret) + goto err_free_trimslice; + + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + snd_soc_card_set_drvdata(card, trimslice); + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", + ret); + goto err_fini_utils; + } + + return 0; + +err_fini_utils: + tegra_asoc_utils_fini(&trimslice->util_data); +err_free_trimslice: + kfree(trimslice); + return ret; +} + +static int __devexit tegra_snd_trimslice_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct tegra_trimslice *trimslice = snd_soc_card_get_drvdata(card); + + snd_soc_unregister_card(card); + + tegra_asoc_utils_fini(&trimslice->util_data); + + kfree(trimslice); + + return 0; +} + +static struct platform_driver tegra_snd_trimslice_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + }, + .probe = tegra_snd_trimslice_probe, + .remove = __devexit_p(tegra_snd_trimslice_remove), +}; + +static int __init snd_tegra_trimslice_init(void) +{ + return platform_driver_register(&tegra_snd_trimslice_driver); +} +module_init(snd_tegra_trimslice_init); + +static void __exit snd_tegra_trimslice_exit(void) +{ + platform_driver_unregister(&tegra_snd_trimslice_driver); +} +module_exit(snd_tegra_trimslice_exit); + +MODULE_AUTHOR("Mike Rapoport <mike(a)compulab.co.il>"); +MODULE_DESCRIPTION("Trimslice machine ASoC driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); -- 1.7.3.1

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[alsa-devel] (no subject)
by Sangbeom Kim 26 Apr '11

26 Apr '11

[PATCH 0/2] Add WM8994 PCM Machine driver for Exynos4 Hi, This is patchset for WM8994 pcm machine driver that supports PCM audio on SMDKV310, SMDKC210 and test is done. Based on these patches WM8994 pcm machine driver supports followings: o Playback supports 8kHz sampling rates. o Capture supports 8kHz sampling rates. This patchset contains followings o To Kukjin Kim and Ben Dooks, [PATCH 1/2] ARM: EXYNOS4: Add PCM audio support for WM8994 o To Jassi Brar, Mark Brown and Liam Girdwood, [PATCH 2/2] ASoC: SAMSUNG: Add WM8580 PCM Machine driver Best Regards, SB Kim (Sangbeom Kim)

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[alsa-devel] [PATCH] ASoC: Support 24.576MHz MCLKs in WM8915
by Mark Brown 26 Apr '11

26 Apr '11

We can safely divide these down to within the supported SYSCLK range. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8915.c | 7 +++++-- 1 files changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8915.c b/sound/soc/codecs/wm8915.c index 0836094..4a3c5cc 100644 --- a/sound/soc/codecs/wm8915.c +++ b/sound/soc/codecs/wm8915.c @@ -1831,6 +1831,7 @@ static int wm8915_set_sysclk(struct snd_soc_dai *dai, struct snd_soc_codec *codec = dai->codec; struct wm8915_priv *wm8915 = snd_soc_codec_get_drvdata(codec); int lfclk = 0; + int ratediv = 0; int src; int old; @@ -1862,6 +1863,8 @@ static int wm8915_set_sysclk(struct snd_soc_dai *dai, snd_soc_update_bits(codec, WM8915_AIF_RATE, WM8915_SYSCLK_RATE, 0); break; + case 24576000: + ratediv = WM8915_SYSCLK_DIV; case 12288000: snd_soc_update_bits(codec, WM8915_AIF_RATE, WM8915_SYSCLK_RATE, WM8915_SYSCLK_RATE); @@ -1877,8 +1880,8 @@ static int wm8915_set_sysclk(struct snd_soc_dai *dai, } snd_soc_update_bits(codec, WM8915_AIF_CLOCKING_1, - WM8915_SYSCLK_SRC_MASK, - src << WM8915_SYSCLK_SRC_SHIFT); + WM8915_SYSCLK_SRC_MASK | WM8915_SYSCLK_DIV_MASK, + src << WM8915_SYSCLK_SRC_SHIFT | ratediv); snd_soc_update_bits(codec, WM8915_CLOCKING_1, WM8915_LFCLK_ENA, lfclk); snd_soc_update_bits(codec, WM8915_AIF_CLOCKING_1, WM8915_SYSCLK_ENA, old); -- 1.7.4.1

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[alsa-devel] snd_pcm_writei BLOCKED
by Irfan Shaikh 26 Apr '11

26 Apr '11

Hello, I have created two threads 1)Thread 1 : reads audio data from USB audio class hw:1,0 (Capture device : open in BLOCKING Mode ) 2)Thread2 : writes audio data to playback device hw:0,0 (Playback device : open in BLOCKING Mode ) First snd_pcm_writei() is succesfull in Thread 2 Thread2 gets stuck/blocked in second snd_pcm_writei() and does not returns. Please can some one please tell me Why i am facing this issue. (Note : This happens with only with 1 specific capture device, when i connect different model i.e capture device my implementation works fine) Regards Irfan SASKEN BUSINESS DISCLAIMER: This message may contain confidential, proprietary or legally privileged information. In case you are not the original intended Recipient of the message, you must not, directly or indirectly, use, disclose, distribute, print, or copy any part of this message and you are requested to delete it and inform the sender. Any views expressed in this message are those of the individual sender unless otherwise stated. Nothing contained in this message shall be construed as an offer or acceptance of any offer by Sasken Communication Technologies Limited ("Sasken") unless sent with that express intent and with due authority of Sasken. Sasken has taken enough precautions to prevent the spread of viruses. However the company accepts no liability for any damage caused by any virus transmitted by this email. Read Disclaimer at http://www.sasken.com/extras/mail_disclaimer.html

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[alsa-devel] [PATCH 0/3 v2] ASoC: sh: fsi: Add module/port control
by Kuninori Morimoto 26 Apr '11

26 Apr '11

Dear Mark, Liam Cc Magnus, Simon, yoshii-san These patches are - resend of "ASoC: fsi: add dev_pm_ops :: suspend/resume" - v2 of "control FSI module/port power" When I sent v1 of "control FSI module/port power" I explained that it are depend on "ASoC: fsi: add dev_pm_ops :: suspend/resume". But Mark seems didn't check it. This time I sent all of them. I changed patch order. Kuninori Morimoto: ASoC: sh: fsi: add fsi_is_clk_master function ASoC: sh: fsi: add dev_pm_ops :: suspend/resume ASoC: sh: fsi: Add module/port control function v1 -> v2 - change patch order - add more text for git-log

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