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

[alsa-devel] New asoc-tools repository
by Dimitris Papastamos 07 Mar '11

07 Mar '11

Hi folks, I've set up a new asoc-tools repository for various tools/scripts specific to ASoC. So far I've got two scripts, one for visualizing DAPM graphs and rendering them to jpeg and another tool for beautifying the output of the tracepoint hooks of ASoC so that it is more verbose. Information on how to use these scripts is self-contained in the code. The repo is at git://opensource.wolfsonmicro.com/asoc-tools.git Try them out, any feedback is highly appreciated. Thanks, Dimitris

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[alsa-devel] [PATCH] ASoC: sound:soc:soc-core fix one small indentation issue
by Lu Guanqun 07 Mar '11

07 Mar '11

Signed-off-by: Lu Guanqun <guanqun.lu(a)intel.com> --- sound/soc/soc-core.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 64befac..f6ba502 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3438,7 +3438,7 @@ int snd_soc_register_platform(struct device *dev, platform = kzalloc(sizeof(struct snd_soc_platform), GFP_KERNEL); if (platform == NULL) - return -ENOMEM; + return -ENOMEM; /* create platform component name */ platform->name = fmt_single_name(dev, &platform->id); -- 1.7.2.3

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[alsa-devel] [RFC][PATCH] ASoC: Move symmetric rate check for substreams to hw_params
by Vaibhav Bedia 07 Mar '11

07 Mar '11

Currently the symmetric rate check is present in the snd_pcm_open() call. However this causes snd_pcm_open() to fail for the 2nd substream is the 2nd call to snd_pcm_open() is made before the 1st substream has set a non-zero rate by calling snd_pcm_hw_params(). Fix this by moving the call to snd_pcm_apply_symmetry() to soc_pcm_hw_params(). Signed-off-by: Afzal Mohammed <afzal(a)ti.com> Signed-off-by: Vaibhav Bedia <vaibhav.bedia(a)ti.com> --- This patch is one way of fixing the issues reported here http://comments.gmane.org/gmane.linux.alsa.devel/75489 and http://thread.gmane.org/gmane.linux.alsa.devel/81055/focus=21653 sound/soc/soc-core.c | 25 +++++++++++++++++-------- 1 files changed, 17 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 64befac..edec9af 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -638,13 +638,6 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) goto config_err; } - /* Symmetry only applies if we've already got an active stream. */ - if (cpu_dai->active || codec_dai->active) { - ret = soc_pcm_apply_symmetry(substream); - if (ret != 0) - goto config_err; - } - pr_debug("asoc: %s <-> %s info:\n", codec_dai->name, cpu_dai->name); pr_debug("asoc: rate mask 0x%x\n", runtime->hw.rates); @@ -742,6 +735,9 @@ static int soc_codec_close(struct snd_pcm_substream *substream) codec_dai->active--; codec->active--; + if(!codec_dai->active && !cpu_dai->active) + rtd->rate = 0; + /* Muting the DAC suppresses artifacts caused during digital * shutdown, for example from stopping clocks. */ @@ -898,12 +894,25 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, } } - rtd->rate = params_rate(params); + /* Symmetry only applies if we've already got an active stream, + a non zero rtd->rate implies that stream was already active and + rtd->rate has been set previously */ + if (rtd->rate) { + ret = soc_pcm_apply_symmetry(substream); + if (ret < 0) + goto symmetry_err; + } else { + rtd->rate = params_rate(params); + } out: mutex_unlock(&pcm_mutex); return ret; +symmetry_err: + if(platform->driver->ops->hw_free) + platform->driver->ops->hw_free(substream); + platform_err: if (cpu_dai->driver->ops->hw_free) cpu_dai->driver->ops->hw_free(substream, cpu_dai); -- 1.6.2.4

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[alsa-devel] MIDI, snd-serial-u16550 and dropped notes
by r10kindsofpeople 07 Mar '11

07 Mar '11

Sorry for cross-posting, I've been writing about this problem on alsa-user, but it seems to me that this probably belongs over here now (if it hasn't all along). A brief recap: Using the snd-serial-u16550 driver I have two MIDI streams coming in that are near duplicates of each other. Every now and then (after 5-45 minutes of playing) I get a dropped or corrupted message. I've captured this multiple times using both a logic analyzer on the serial data and software logging of the ALSA ports. Sometimes whole events get dropped, sometimes the tail end of one message and the beginning of the next gets dropped to form a completely different event. Aside from the fact that both ports are getting hit simultaneously, the ports are not being hit very hard...normal playing. This has occurred under: Ubuntu 10.04 netbook Ubuntu 10.10 netbook Ubuntu 10.10 with Natty lowlatency kernel It does not appear to occur under: Ubuntu 10.10 server Ubuntu 10.04 netbook with linux-rt kernel I've not eliminated the possibility that it's being caused by my ALSA application - quite the opposite - but what could I be doing that corrupts the behavior of the driver? I've tried several methods of reading the data to no effect. With great respect to the authors of the driver, I don't think it has changed in many years and kernel has certainly changed underneath it. Is there any sense that it could be improved to eliminate the problem and the apparent requirement for the -rt kernel? Or should I be looking at my ALSA application more closely? John

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[alsa-devel] [PATCH] ASoC: Add LM4857 to SND_SOC_ALL_CODECS
by Mark Brown 07 Mar '11

07 Mar '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/Kconfig | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 212859c..10f11dd 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -31,6 +31,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_DA7210 if I2C select SND_SOC_DFBMCS320 select SND_SOC_JZ4740_CODEC if SOC_JZ4740 + select SND_SOC_LM4857 if I2C select SND_SOC_MAX98088 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 -- 1.7.2.3

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[alsa-devel] [PATCH] ASoC: Add machine driver for Visstrim_M10 board.
by Javier Martin 07 Mar '11

07 Mar '11

Visstrim_M10 boards have an external tlcv320aic3205 codec attached to SSI1. This driver glues together both interfaces. External amplifier is not supported in this first version. Signed-off-by: Javier Martin <javier.martin(a)vista-silicon.com> --- sound/soc/imx/Kconfig | 10 +++ sound/soc/imx/Makefile | 2 + sound/soc/imx/mx27vis-aic32x4.c | 137 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 149 insertions(+), 0 deletions(-) create mode 100644 sound/soc/imx/mx27vis-aic32x4.c diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index 642270a..15e7c04 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -30,6 +30,16 @@ config SND_MXC_SOC_WM1133_EV1 Enable support for audio on the i.MX31ADS with the WM1133-EV1 PMIC board with WM8835x fitted. +config SND_SOC_MX27VIS_AIC32X4 + tristate "SoC audio support for Visstrim M10 boards" + depends on MACH_IMX27_VISSTRIM_M10 + select SND_SOC_TVL320AIC32X4 + select SND_MXC_SOC_SSI + select SND_MXC_SOC_MX2 + help + Say Y if you want to add support for SoC audio on Visstrim SM10 + board with TLV320AIC32X4 codec. + config SND_SOC_PHYCORE_AC97 tristate "SoC Audio support for Phytec phyCORE (and phyCARD) boards" depends on MACH_PCM043 || MACH_PCA100 diff --git a/sound/soc/imx/Makefile b/sound/soc/imx/Makefile index b67fc02..d6d609b 100644 --- a/sound/soc/imx/Makefile +++ b/sound/soc/imx/Makefile @@ -10,8 +10,10 @@ obj-$(CONFIG_SND_MXC_SOC_MX2) += snd-soc-imx-mx2.o # i.MX Machine Support snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o snd-soc-phycore-ac97-objs := phycore-ac97.o +snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o +obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o diff --git a/sound/soc/imx/mx27vis-aic32x4.c b/sound/soc/imx/mx27vis-aic32x4.c new file mode 100644 index 0000000..054110b --- /dev/null +++ b/sound/soc/imx/mx27vis-aic32x4.c @@ -0,0 +1,137 @@ +/* + * mx27vis-aic32x4.c + * + * Copyright 2011 Vista Silicon S.L. + * + * Author: Javier Martin <javier.martin(a)vista-silicon.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., 51 Franklin Street, Fifth Floor, Boston, + * MA 02110-1301, USA. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/device.h> +#include <linux/i2c.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <asm/mach-types.h> +#include <mach/audmux.h> + +#include "../codecs/tlv320aic32x4.h" +#include "imx-ssi.h" + +static int mx27vis_aic32x4_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; + int ret; + u32 dai_format; + + dai_format = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM; + + /* set codec DAI configuration */ + snd_soc_dai_set_fmt(codec_dai, dai_format); + + /* set cpu DAI configuration */ + snd_soc_dai_set_fmt(cpu_dai, dai_format); + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, + 25000000, SND_SOC_CLOCK_OUT); + if (ret) { + pr_err("%s: failed setting codec sysclk\n", __func__); + return ret; + } + + ret = snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0, + SND_SOC_CLOCK_IN); + if (ret) { + pr_err("can't set CPU system clock IMX_SSP_SYS_CLK\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops mx27vis_aic32x4_snd_ops = { + .hw_params = mx27vis_aic32x4_hw_params, +}; + +static struct snd_soc_dai_link mx27vis_aic32x4_dai = { + .name = "tlv320aic32x4", + .stream_name = "TLV320AIC32X4", + .codec_dai_name = "tlv320aic32x4-hifi", + .platform_name = "imx-pcm-audio.0", + .codec_name = "tlv320aic32x4.0-0018", + .cpu_dai_name = "imx-ssi.0", + .ops = &mx27vis_aic32x4_snd_ops, +}; + +static struct snd_soc_card mx27vis_aic32x4 = { + .name = "visstrim_m10-audio", + .dai_link = &mx27vis_aic32x4_dai, + .num_links = 1, +}; + +static struct platform_device *mx27vis_aic32x4_snd_device; + +static int __init mx27vis_aic32x4_init(void) +{ + int ret; + + mx27vis_aic32x4_snd_device = platform_device_alloc("soc-audio", -1); + if (!mx27vis_aic32x4_snd_device) + return -ENOMEM; + + platform_set_drvdata(mx27vis_aic32x4_snd_device, &mx27vis_aic32x4); + ret = platform_device_add(mx27vis_aic32x4_snd_device); + + if (ret) { + printk(KERN_ERR "ASoC: Platform device allocation failed\n"); + platform_device_put(mx27vis_aic32x4_snd_device); + } + + /* Connect SSI0 as clock slave to SSI1 external pins */ + mxc_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0, + MXC_AUDMUX_V1_PCR_SYN | + MXC_AUDMUX_V1_PCR_TFSDIR | + MXC_AUDMUX_V1_PCR_TCLKDIR | + MXC_AUDMUX_V1_PCR_TFCSEL(MX27_AUDMUX_PPCR1_SSI_PINS_1) | + MXC_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_PPCR1_SSI_PINS_1) + ); + mxc_audmux_v1_configure_port(MX27_AUDMUX_PPCR1_SSI_PINS_1, + MXC_AUDMUX_V1_PCR_SYN | + MXC_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR1_SSI0) + ); + + return ret; +} + +static void __exit mx27vis_aic32x4_exit(void) +{ + platform_device_unregister(mx27vis_aic32x4_snd_device); +} + +module_init(mx27vis_aic32x4_init); +module_exit(mx27vis_aic32x4_exit); + +MODULE_AUTHOR("Javier Martin <javier.martin(a)vista-silicon.com>"); +MODULE_DESCRIPTION("ALSA SoC AIC32X4 mx27 visstrim"); +MODULE_LICENSE("GPL"); -- 1.7.0.4

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[alsa-devel] [PATCH 1/2] ASoC: tlv320dac33: add MODULE_DEVICE_TABLE
by Axel Lin 07 Mar '11

07 Mar '11

The device table is required to load modules based on modaliases. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/tlv320dac33.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 71d7be8..00b6d87 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -1615,6 +1615,7 @@ static const struct i2c_device_id tlv320dac33_i2c_id[] = { }, { }, }; +MODULE_DEVICE_TABLE(i2c, tlv320dac33_i2c_id); static struct i2c_driver tlv320dac33_i2c_driver = { .driver = { -- 1.7.2

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Re: [alsa-devel] Process of sumit the Audio driver code to ALSA
by Reddy, MR Swami 07 Mar '11

07 Mar '11

Thanks for quick response! Yes, it ASOC based and currently, the audio drivers supported for Samsung 6410 target. Thanks Swami -----Original Message----- From: Takashi Iwai [mailto:tiwai@suse.de] Sent: Wednesday, January 19, 2011 12:39 PM To: Reddy, MR Swami Subject: Re: Process of sumit the Audio driver code to ALSA At Tue, 18 Jan 2011 20:58:28 -0800, Reddy, MR Swami wrote: > > Hello, > We (National Semi) have developed ALSA based audio drivers for our Audio subsystem. Could you please let me know the process to submit and include as part of ALSA project. > > The ALSA audio driver version info as follows: > - Linux - 2.6.24 base > - Andorid- 2.1 base Is this based on ASoC? If not, I recommend you to port the code to ASoC, thus it's the standard for embedded devices. If you have any questions, just post to alsa-devel ML. thanks, Takashi

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[alsa-devel] [PATCH 1/2] ASoC: Fix outdated API usage in tlv320aic32x4
by Mark Brown 07 Mar '11

07 Mar '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/tlv320aic32x4.c | 10 +++++----- 1 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index ee82e38..e93b9d1 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -340,13 +340,13 @@ static inline int aic32x4_get_divs(int mclk, int rate) static int aic32x4_add_widgets(struct snd_soc_codec *codec) { - snd_soc_dapm_new_controls(codec, aic32x4_dapm_widgets, - ARRAY_SIZE(aic32x4_dapm_widgets)); + snd_soc_dapm_new_controls(&codec->dapm, aic32x4_dapm_widgets, + ARRAY_SIZE(aic32x4_dapm_widgets)); - snd_soc_dapm_add_routes(codec, aic32x4_dapm_routes, + snd_soc_dapm_add_routes(&codec->dapm, aic32x4_dapm_routes, ARRAY_SIZE(aic32x4_dapm_routes)); - snd_soc_dapm_new_widgets(codec); + snd_soc_dapm_new_widgets(&codec->dapm); return 0; } @@ -602,7 +602,7 @@ static int aic32x4_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_OFF: break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } -- 1.7.2.3

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[alsa-devel] [PATCH] ASoC: Extend range of supported sample rates for CS4271 CODEC.
by Alexander Sverdlin 07 Mar '11

07 Mar '11

From: Alexander Sverdlin <subaparts(a)yandex.ru> Extend range of supported sample rates for CS4271 CODEC. Signed-off-by: Alexander Sverdlin <subaparts(a)yandex.ru> --- sound/soc/codecs/cs4271.c | 5 +++-- 1 files changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 1791796..e5cc4da 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -33,6 +33,7 @@ #define CS4271_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_S32_LE) +#define CS4271_PCM_RATES SNDRV_PCM_RATE_8000_192000 /* * CS4271 registers @@ -392,14 +393,14 @@ static struct snd_soc_dai_driver cs4271_dai = { .stream_name = "Playback", .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_96000, + .rates = CS4271_PCM_RATES, .formats = CS4271_PCM_FORMATS, }, .capture = { .stream_name = "Capture", .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_96000, + .rates = CS4271_PCM_RATES, .formats = CS4271_PCM_FORMATS, }, .ops = &cs4271_dai_ops,

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