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Re: [alsa-devel] [alsa-announce] ALSA 1.0.21 release
by David Favor 01 Sep '09

01 Sep '09

To -> Kernel + ALSA Guru. I'm trying to understand exactly how ALSA updates work. Specifically I'm using Ubuntu Karmic (Kernel is 2.6.31-9) and I'm interested in testing all the X-Fi card driver fixes which are listed in the ALSA 1.0.21 release notes. Someone clarify the process of how ALSA and Kernel updates work. From the 1.0.20 Upgrade Guide it appears ALSA drivers are deposited in /lib/modules/... as part of the upgrade process. Can someone verify this for me. Thanks. -- Love feeling your best ever, all day, every day? Click http://RadicalHealth.com for the easy way!

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[alsa-devel] ALC262 models
by Jure Koren 01 Sep '09

01 Sep '09

Hi! I've received a Vaio P netbook recently, which contains an ALC262 of an unknown model. Playback generally works out of the box, but I haven't yet managed to raise the capture level for the internal microphone (though it's easy to rev up the passthrough channel which makes a nice white noise without any feedback). What I'd like to know is how difficult would it be for me to actually get test out more channels than are available with any particular model= parameter setting. Some other chipsets include model=test or model=debug, but not this one. Also, is there a tool that could help me probe around the chipset to properly identify available mixer channels? If poking the code is the only way, that's ok, but I guess it'll have to wait until I have some time available. Thanks for any pointers in advance. Best regards, -- Jure Koren, unix developer

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[alsa-devel] ALSA driver - make error
by arnuld uttre 01 Sep '09

01 Sep '09

I tried to compile latest alsa drivers anf I got make error. all the relevant information is included here. I think there is a bug in source code. [root@dune alsa-driver-1.0.21]# ./configure --with-oss=yes --with-pcm-oss=yes --with-sequencer=yes --with-cards=all --with-card-options=all --prefix=/usr ................... .................... make[2]: Leaving directory `/usr/local/src/alsa/alsa-driver-1.0.21/misc' make[1]: Leaving directory `/usr/local/src/alsa/alsa-driver-1.0.21' make -C /lib/modules/2.6.18-128.4.1.el5/source SUBDIRS=/usr/local/src/alsa/alsa-driver-1.0.21 CPP="gcc -E" CC="gcc" modules make[1]: Entering directory `/usr/src/kernels/2.6.18-128.4.1.el5-i686' CC [M] /usr/local/src/alsa/alsa-driver-1.0.21/acore/hwdep.o In file included from /usr/local/src/alsa/alsa-driver-1.0.21/acore/hwdep.c:1: /usr/local/src/alsa/alsa-driver-1.0.21/include/adriver.h:1786: error: redefinition of typedef ‘bool’ include/linux/types.h:36: error: previous declaration of ‘bool’ was here make[3]: *** [/usr/local/src/alsa/alsa-driver-1.0.21/acore/hwdep.o] Error 1 make[2]: *** [/usr/local/src/alsa/alsa-driver-1.0.21/acore] Error 2 make[1]: *** [_module_/usr/local/src/alsa/alsa-driver-1.0.21] Error 2 make[1]: Leaving directory `/usr/src/kernels/2.6.18-128.4.1.el5-i686' make: *** [compile] Error 2 [root@dune alsa-driver-1.0.21]# gcc --version gcc (GCC) 4.1.2 20080704 (Red Hat 4.1.2-44) Copyright (C) 2006 Free Software Foundation, Inc. This is free software; see the source for copying conditions. There is NO warranty; not even for MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. [root@dune alsa-driver-1.0.21]# uname -a Linux dune 2.6.18-128.4.1.el5 #1 SMP Tue Aug 4 20:23:34 EDT 2009 i686 athlon i386 GNU/Linux [root@dune alsa-driver-1.0.21]# cat /etc/issue CentOS release 5.3 (Final) Kernel \r on an \m [root@dune alsa-driver-1.0.21]# -- http://uttre.wordpress.com/2008/05/14/the-lost-love-of-mine/

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[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 01 Sep '09

01 Sep '09

The following changes since commit d2c0bdaa9362c4b2ab7416420d034a0a2d1ec979: Jarkko Nikula (1): ASoC: OMAP: Add functionality to set CLKR and FSR sources in McBSP DAI are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Barry Song (1): ASoC: new board driver to connect bfin-5xx with ad1836 codec jassi brar (1): ASoC: Debugged improper setting of PLL fields in WM8580 driver sound/soc/blackfin/Kconfig | 8 ++ sound/soc/blackfin/Makefile | 2 + sound/soc/blackfin/bf5xx-ad1836.c | 128 +++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8580.c | 4 +- 4 files changed, 140 insertions(+), 2 deletions(-) create mode 100644 sound/soc/blackfin/bf5xx-ad1836.c

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[alsa-devel] Debugged improper setting of PLL fields in WM8580 driver
by jassi brar 01 Sep '09

01 Sep '09

Bug was caught while trying to use WM8580 as I2S master on SMDK. Symptoms were lesser LRCLK read by CRO(41.02 instead of 44.1 KHz) Solved by referring to WM8580A manual and setting mask value correctly and making the code to not touch 'reserved' bits of PLL4 register. Signed-off-by: Jassi <jassi.brar(a)samsung.com> diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index d547347..6bded8c 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -458,12 +458,12 @@ static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai, return 0; snd_soc_write(codec, WM8580_PLLA1 + offset, pll_div.k & 0x1ff); - snd_soc_write(codec, WM8580_PLLA2 + offset, (pll_div.k >> 9) & 0xff); + snd_soc_write(codec, WM8580_PLLA2 + offset, (pll_div.k >> 9) & 0x1ff); snd_soc_write(codec, WM8580_PLLA3 + offset, (pll_div.k >> 18 & 0xf) | (pll_div.n << 4)); reg = snd_soc_read(codec, WM8580_PLLA4 + offset); - reg &= ~0x3f; + reg &= ~0x1b; reg |= pll_div.prescale | pll_div.postscale << 1 | pll_div.freqmode << 3;

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[alsa-devel] [PATCH] new board driver to connect bfin-5xx with ad1836 codec
by Barry Song 01 Sep '09

01 Sep '09

Hi Mark, As discussed, the patch uses the original TDM order without rewriting. For the match between TDM slot number and audio channel number, a new API need be added. Thanks Barry Signed-off-by: Barry Song <21cnbao(a)gmail.com> --- sound/soc/blackfin/Kconfig | 8 ++ sound/soc/blackfin/Makefile | 2 + sound/soc/blackfin/bf5xx-ad1836.c | 128 +++++++++++++++++++++++++++++++++++++ 3 files changed, 138 insertions(+), 0 deletions(-) create mode 100644 sound/soc/blackfin/bf5xx-ad1836.c diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index 8a4de4d..ac927ff 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -88,6 +88,14 @@ config SND_BF5XX_SOC_AC97 select SND_SOC_AC97_BUS select SND_BF5XX_SOC_SPORT +config SND_BF5XX_SOC_AD1836 + tristate "SoC AD1836 Audio support for BF5xx" + depends on SND_BF5XX_TDM + select SND_BF5XX_SOC_TDM + select SND_SOC_AD1836 + help + Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT. + config SND_BF5XX_SOC_AD1980 tristate "SoC AD1980/1 Audio support for BF5xx" depends on SND_BF5XX_AC97 diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile index f4d7607..87e3042 100644 --- a/sound/soc/blackfin/Makefile +++ b/sound/soc/blackfin/Makefile @@ -16,11 +16,13 @@ obj-$(CONFIG_SND_BF5XX_SOC_I2S) += snd-soc-bf5xx-i2s.o obj-$(CONFIG_SND_BF5XX_SOC_TDM) += snd-soc-bf5xx-tdm.o # Blackfin Machine Support +snd-ad1836-objs := bf5xx-ad1836.o snd-ad1980-objs := bf5xx-ad1980.o snd-ssm2602-objs := bf5xx-ssm2602.o snd-ad73311-objs := bf5xx-ad73311.o snd-ad1938-objs := bf5xx-ad1938.o +obj-$(CONFIG_SND_BF5XX_SOC_AD1836) += snd-ad1836.o obj-$(CONFIG_SND_BF5XX_SOC_AD1980) += snd-ad1980.o obj-$(CONFIG_SND_BF5XX_SOC_SSM2602) += snd-ssm2602.o obj-$(CONFIG_SND_BF5XX_SOC_AD73311) += snd-ad73311.o diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c new file mode 100644 index 0000000..cd361e3 --- /dev/null +++ b/sound/soc/blackfin/bf5xx-ad1836.c @@ -0,0 +1,128 @@ +/* + * File: sound/soc/blackfin/bf5xx-ad1836.c + * Author: Barry Song <Barry.Song(a)analog.com> + * + * Created: Aug 4 2009 + * Description: Board driver for ad1836 sound chip + * + * Bugs: Enter bugs at http://blackfin.uclinux.org/ + * + * 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/module.h> +#include <linux/moduleparam.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/pcm_params.h> + +#include <asm/blackfin.h> +#include <asm/cacheflush.h> +#include <asm/irq.h> +#include <asm/dma.h> +#include <asm/portmux.h> + +#include "../codecs/ad1836.h" +#include "bf5xx-sport.h" + +#include "bf5xx-tdm-pcm.h" +#include "bf5xx-tdm.h" + +static struct snd_soc_card bf5xx_ad1836; + +static int bf5xx_ad1836_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + cpu_dai->private_data = sport_handle; + return 0; +} + +static int bf5xx_ad1836_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 *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + int ret = 0; + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops bf5xx_ad1836_ops = { + .startup = bf5xx_ad1836_startup, + .hw_params = bf5xx_ad1836_hw_params, +}; + +static struct snd_soc_dai_link bf5xx_ad1836_dai = { + .name = "ad1836", + .stream_name = "AD1836", + .cpu_dai = &bf5xx_tdm_dai, + .codec_dai = &ad1836_dai, + .ops = &bf5xx_ad1836_ops, +}; + +static struct snd_soc_card bf5xx_ad1836 = { + .name = "bf5xx_ad1836", + .platform = &bf5xx_tdm_soc_platform, + .dai_link = &bf5xx_ad1836_dai, + .num_links = 1, +}; + +static struct snd_soc_device bf5xx_ad1836_snd_devdata = { + .card = &bf5xx_ad1836, + .codec_dev = &soc_codec_dev_ad1836, +}; + +static struct platform_device *bfxx_ad1836_snd_device; + +static int __init bf5xx_ad1836_init(void) +{ + int ret; + + bfxx_ad1836_snd_device = platform_device_alloc("soc-audio", -1); + if (!bfxx_ad1836_snd_device) + return -ENOMEM; + + platform_set_drvdata(bfxx_ad1836_snd_device, &bf5xx_ad1836_snd_devdata); + bf5xx_ad1836_snd_devdata.dev = &bfxx_ad1836_snd_device->dev; + ret = platform_device_add(bfxx_ad1836_snd_device); + + if (ret) + platform_device_put(bfxx_ad1836_snd_device); + + return ret; +} + +static void __exit bf5xx_ad1836_exit(void) +{ + platform_device_unregister(bfxx_ad1836_snd_device); +} + +module_init(bf5xx_ad1836_init); +module_exit(bf5xx_ad1836_exit); + +/* Module information */ +MODULE_AUTHOR("Barry Song"); +MODULE_DESCRIPTION("ALSA SoC AD1836 board driver"); +MODULE_LICENSE("GPL"); + -- 1.5.6.3

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[alsa-devel] How to support multiple audio codecs on a single board
by Aggarwal, Anuj 01 Sep '09

01 Sep '09

Hi, I have 3 instances of AIC23 on my board and all are connected to different I2C/McBSP and hence can be used independently. How should I use/modify the current AIC23 driver and write the asoc board-specific file to enable all the AIC23 instances? Is there any similar implementation in the audio subsystem which I can look as a reference? Regards, Anuj Aggarwal

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[alsa-devel] [PATCH] ALSA: allocation may fail in snd_pcm_oss_change_params()
by Roel Kluin 31 Aug '09

31 Aug '09

Allocation may fail, show if it did. Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index dbe406b..ea91a0a 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -1047,6 +1047,10 @@ static int snd_pcm_oss_change_params(struct snd_pcm_substream *substream) runtime->oss.prepare = 1; vfree(runtime->oss.buffer); runtime->oss.buffer = vmalloc(runtime->oss.period_bytes); + if (runtime->oss.buffer == NULL) { + err = -ENOMEM; + goto failure; + } runtime->oss.buffer_used = 0; if (runtime->dma_area) snd_pcm_format_set_silence(runtime->format, runtime->dma_area, bytes_to_samples(runtime, runtime->dma_bytes));

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[alsa-devel] [PATCH V2 2/3] add set_tdm_slot in tdm dai to define the relationship between audio channel and tdm slot
by Barry Song 31 Aug '09

31 Aug '09

For ad1938, the audio channel n just uses slot n, but for ad1836, it's different, channel 0 uses slot 0, channel 1 uses slot 4, channels 2 uses slot1, ... So add set_tdm_slot entry and use the mask field to define the relationship between audio channel and slot number. Signed-off-by: Barry Song <21cnbao(a)gmail.com> --- sound/soc/blackfin/bf5xx-ad1938.c | 7 ++++++- sound/soc/blackfin/bf5xx-tdm-pcm.c | 12 +++++++++--- sound/soc/blackfin/bf5xx-tdm.c | 29 +++++++++++++++++++++-------- sound/soc/blackfin/bf5xx-tdm.h | 10 ++++++++++ 4 files changed, 46 insertions(+), 12 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ad1938.c b/sound/soc/blackfin/bf5xx-ad1938.c index 08269e9..05163ea 100644 --- a/sound/soc/blackfin/bf5xx-ad1938.c +++ b/sound/soc/blackfin/bf5xx-ad1938.c @@ -74,8 +74,13 @@ static int bf5xx_ad1938_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; + /* set cpu DAI slots, 8 channels, mask is defined as slot seq */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x76543210, 8, 32); + if (ret < 0) + return ret; + /* set codec DAI slots, 8 channels, all channels are enabled */ - ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xFF, 8); + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xFF, 8, 32); if (ret < 0) return ret; diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.c b/sound/soc/blackfin/bf5xx-tdm-pcm.c index ccb5e82..044ef45 100644 --- a/sound/soc/blackfin/bf5xx-tdm-pcm.c +++ b/sound/soc/blackfin/bf5xx-tdm-pcm.c @@ -43,7 +43,7 @@ #include "bf5xx-tdm.h" #include "bf5xx-sport.h" -#define PCM_BUFFER_MAX 0x10000 +#define PCM_BUFFER_MAX 0x8000 #define FRAGMENT_SIZE_MIN (4*1024) #define FRAGMENTS_MIN 2 #define FRAGMENTS_MAX 32 @@ -177,6 +177,9 @@ out: static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, snd_pcm_uframes_t pos, void *buf, snd_pcm_uframes_t count) { + struct snd_pcm_runtime *runtime = substream->runtime; + struct sport_device *sport = runtime->private_data; + struct bf5xx_tdm_port *tdm_port = sport->private_data; unsigned int *src; unsigned int *dst; int i; @@ -186,9 +189,11 @@ static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, dst = (unsigned int *)substream->runtime->dma_area; dst += pos * 8; + while (count--) { for (i = 0; i < substream->runtime->channels; i++) - *(dst + i) = *src++; + *(dst + (((tdm_port->slot_seq >> + (4 * i)) & 0xF))) = *src++; dst += 8; } } else { @@ -198,7 +203,8 @@ static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, src += pos * 8; while (count--) { for (i = 0; i < substream->runtime->channels; i++) - *dst++ = *(src+i); + *dst++ = *(src + (((tdm_port->slot_seq >> + (4 * i)) & 0xF))); src += 8; } } diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c index 3096bad..0f4ee15 100644 --- a/sound/soc/blackfin/bf5xx-tdm.c +++ b/sound/soc/blackfin/bf5xx-tdm.c @@ -46,14 +46,6 @@ #include "bf5xx-sport.h" #include "bf5xx-tdm.h" -struct bf5xx_tdm_port { - u16 tcr1; - u16 rcr1; - u16 tcr2; - u16 rcr2; - int configured; -}; - static struct bf5xx_tdm_port bf5xx_tdm; static int sport_num = CONFIG_SND_BF5XX_SPORT_NUM; @@ -181,6 +173,24 @@ static void bf5xx_tdm_shutdown(struct snd_pcm_substream *substream, bf5xx_tdm.configured = 0; } +static int bf5xx_tdm_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int mask, int slots, int width) +{ + /* + * For ad1938, the audio channel n just uses slot n, but for + * ad1836, it's different, channel 0 uses slot 0, channel 1 + * uses slot 4, channels 2 uses slot1, ... So add set_tdm_slot + * entry and use the mask field to define the relationship + * between audio channel and slot number. + * + * bit[0..3] means the slot channel 0 will use + * bit[4..7] means the slot channel 1 will use + * ... + */ + bf5xx_tdm.slot_seq = mask; + return 0; +} + #ifdef CONFIG_PM static int bf5xx_tdm_suspend(struct snd_soc_dai *dai) { @@ -235,6 +245,7 @@ static struct snd_soc_dai_ops bf5xx_tdm_dai_ops = { .hw_params = bf5xx_tdm_hw_params, .set_fmt = bf5xx_tdm_set_dai_fmt, .shutdown = bf5xx_tdm_shutdown, + .set_tdm_slot = bf5xx_tdm_set_tdm_slot, }; struct snd_soc_dai bf5xx_tdm_dai = { @@ -300,6 +311,8 @@ static int __devinit bfin_tdm_probe(struct platform_device *pdev) pr_err("Failed to register DAI: %d\n", ret); goto sport_config_err; } + + sport_handle->private_data = &bf5xx_tdm; return 0; sport_config_err: diff --git a/sound/soc/blackfin/bf5xx-tdm.h b/sound/soc/blackfin/bf5xx-tdm.h index 618ec3d..701c769 100644 --- a/sound/soc/blackfin/bf5xx-tdm.h +++ b/sound/soc/blackfin/bf5xx-tdm.h @@ -9,6 +9,16 @@ #ifndef _BF5XX_TDM_H #define _BF5XX_TDM_H +struct bf5xx_tdm_port { + u16 tcr1; + u16 rcr1; + u16 tcr2; + u16 rcr2; + /* which slot used for the corresponding audio channel? */ + int slot_seq; + int configured; +}; + extern struct snd_soc_dai bf5xx_tdm_dai; #endif -- 1.5.6.3

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Re: [alsa-devel] [alsa-cvslog] alsa-tools: ALSA tools repository branch, master now at v1.0.20-5-ga539d8b
by Takashi Iwai 31 Aug '09

31 Aug '09

At Mon, 31 Aug 2009 12:32:40 +0200 (CEST), noreply-git(a)alsa-project.org wrote: > > commit a539d8b261699856095154a65e17b3b3b73155b6 > Author: Jaroslav Kysela <perex(a)perex.cz> > AuthorDate: Mon Aug 31 12:16:48 2009 +0200 > Commit: Jaroslav Kysela <perex(a)perex.cz> > CommitDate: Mon Aug 31 12:32:35 2009 +0200 > > qlo10k1: remove acinlude.m4 (QT configuration hell) - use PKG_CHECK_MODULES This breaks the build with Qt installed in the non-standard paths like /opt. Qt provides no pkgconfig files, so it can't be fixed in that way (and this program isn't for Gtk!) Please revert it. Takashi

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