May 2009 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] Increasing maximum sampling frequency?
by Cameron Stone 29 May '09

29 May '09

The A/D chip is an Analog Devices 7928. It can do 2 mega-samples per second, multiplexed across 8 inputs. The USB audio interface is provided by my firmware code, assisted by the AVR AT90USB647 micro which has some low-level USB support. HTH. Cameron. Alex Austin wrote: > By Codec, I think he's asking about the A/D chip. For example, the TI > PCM2902E is a stereo USB audio codec. > > - Alex Austin > (651) 238-9273 > > "...and then I visited Wikipedia ...and the next 8 hours are a blur." > > > On Thu, May 28, 2009 at 8:30 AM, Cameron Stone > <camerons.lists(a)cse.unsw.edu.au <mailto:camerons.lists@cse.unsw.edu.au>> > wrote: > > Daniel Mack wrote: > > On Thu, May 28, 2009 at 11:47:34AM +1000, Cameron Stone wrote: > >> I've got a USB microphone application that I want to sample at > 256000 8 > >> bit samples per second so I can listen to bats. I know that my device > >> can get close to that rate, but alsa seems to restrict sampling > rates to > >> 192kHz. > > > > Just out of curiosity - which microphone and what kind of codec > are you > > using for such frequencies? > > I don't know which microphone but I can find out if you're interested. I > do know that its sensitivity starts at 10kHz, if that helps. > > There is no codec used. I'm using Type I (uncompressed PCM) because it's > being powered by an AVR running at 16MHz and there's not enough clock > cycles to encode the samples. > > Cameron. > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org <mailto:Alsa-devel@alsa-project.org> > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > >

1 0

[alsa-devel] [PATCH V3 0/4] mpc5200 audio rework for AC97
by Jon Smirl 28 May '09

28 May '09

The following series implements audio support for the mpc5200. It adds an AC97 driver and STAC9766 codec driver. Board support for the Efika and Phytec pcm030 are also included. I've tried to implement the feedback received on the previous two versions. based on commit 0bc53a67ac831ec84f730a657dbcadd80a589ef5 on broonie/for-2.6.31 at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git --- Jon Smirl (4): Fabric bindings for STAC9766 on the Efika Support for AC97 on Phytec pmc030 base board. AC97 driver for mpc5200 Main rewite of the mpc5200 audio DMA code sound/soc/fsl/Kconfig | 27 ++ sound/soc/fsl/Makefile | 5 sound/soc/fsl/efika-audio-fabric.c | 95 +++++++ sound/soc/fsl/mpc5200_dma.c | 504 +++++++++++++++++++++++------------ sound/soc/fsl/mpc5200_dma.h | 33 +- sound/soc/fsl/mpc5200_psc_ac97.c | 392 +++++++++++++++++++++++++++ sound/soc/fsl/mpc5200_psc_ac97.h | 15 + sound/soc/fsl/mpc5200_psc_i2s.c | 247 +++-------------- sound/soc/fsl/mpc5200_psc_i2s.h | 12 + sound/soc/fsl/pcm030-audio-fabric.c | 95 +++++++ 10 files changed, 1036 insertions(+), 389 deletions(-) create mode 100644 sound/soc/fsl/efika-audio-fabric.c create mode 100644 sound/soc/fsl/mpc5200_psc_ac97.c create mode 100644 sound/soc/fsl/mpc5200_psc_ac97.h create mode 100644 sound/soc/fsl/mpc5200_psc_i2s.h create mode 100644 sound/soc/fsl/pcm030-audio-fabric.c -- Jon Smirl jonsmirl(a)gmail.com

4 21

[alsa-devel] ASoC updates for 2.6.31
by Mark Brown 28 May '09

28 May '09

The following changes since commit 08d15f034e94251606479d7ca9070994c2e2fcf0: Mark Brown (1): ASoC: Switch FSL SSI DAI over to symmetric_rates are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31 Chaithrika U S (1): ASoC: Add dummy S/PDIF codec support Jon Smirl (1): ASoC: Modify mpc5200 AC97 driver to use V9 of spin_event_timeout() Mark Brown (1): ASoC: Initialise dev for the dummy S/PDIF DAI Roel Kluin (1): ASoC: correct print specifiers for unsigneds sound/soc/atmel/playpaq_wm8510.c | 2 +- sound/soc/codecs/Kconfig | 4 ++ sound/soc/codecs/Makefile | 2 + sound/soc/codecs/spdif_transciever.c | 71 ++++++++++++++++++++++++++++++++++ sound/soc/codecs/spdif_transciever.h | 17 ++++++++ sound/soc/codecs/tlv320aic23.c | 4 +- sound/soc/codecs/uda134x.c | 4 +- sound/soc/codecs/wm8350.c | 2 +- sound/soc/codecs/wm8400.c | 4 +- sound/soc/codecs/wm8510.c | 2 +- sound/soc/codecs/wm8580.c | 4 +- sound/soc/codecs/wm8753.c | 2 +- sound/soc/codecs/wm8900.c | 6 +- sound/soc/codecs/wm8990.c | 2 +- sound/soc/codecs/wm9713.c | 2 +- sound/soc/fsl/mpc5200_psc_ac97.c | 30 +++++++-------- sound/soc/pxa/pxa-ssp.c | 4 +- sound/soc/s3c24xx/s3c-i2s-v2.c | 4 +- sound/soc/sh/ssi.c | 2 +- 19 files changed, 130 insertions(+), 38 deletions(-) create mode 100644 sound/soc/codecs/spdif_transciever.c create mode 100644 sound/soc/codecs/spdif_transciever.h

2 1

[alsa-devel] [PATCH v3 4/6] ARM: DaVinci: ASoC: Add dummy codec support
by Chaithrika U S 28 May '09

28 May '09

McASP on DM646x can operate in DIT (S/PDIF) where no codec is needed. This patch provides stub codec that can be used in these configurations. On DM646x EVM the McASP1 is connected to the S/PDIF out. Signed-off-by: Steve Chen <schen(a)mvista.com> Signed-off-by: Pavel Kiryukhin <pkiryukhin(a)ru.mvista.com> Signed-off-by: Naresh Medisetty <naresh(a)ti.com> Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- This patch applies against ALSA mainline - that's the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git sound/soc/codecs/Kconfig | 4 ++ sound/soc/codecs/Makefile | 2 + sound/soc/codecs/spdif_transciever.c | 69 ++++++++++++++++++++++++++++++++++ sound/soc/codecs/spdif_transciever.h | 17 ++++++++ 4 files changed, 92 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/spdif_transciever.c create mode 100644 sound/soc/codecs/spdif_transciever.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index cb07d9b..bbc97fd 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -18,6 +18,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4535 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_PCM3008 + select SND_SOC_SPDIF select SND_SOC_SSM2602 if I2C select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TLV320AIC23 if I2C @@ -91,6 +92,9 @@ config SND_SOC_L3 config SND_SOC_PCM3008 tristate +config SND_SOC_SPDIF + tristate + config SND_SOC_SSM2602 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 46c007c..8b75305 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -6,6 +6,7 @@ snd-soc-ak4535-objs := ak4535.o snd-soc-cs4270-objs := cs4270.o snd-soc-l3-objs := l3.o snd-soc-pcm3008-objs := pcm3008.o +snd-soc-spdif-objs := spdif_transciever.o snd-soc-ssm2602-objs := ssm2602.o snd-soc-stac9766-objs := stac9766.o snd-soc-tlv320aic23-objs := tlv320aic23.o @@ -42,6 +43,7 @@ obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o +obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o diff --git a/sound/soc/codecs/spdif_transciever.c b/sound/soc/codecs/spdif_transciever.c new file mode 100644 index 0000000..118e976 --- /dev/null +++ b/sound/soc/codecs/spdif_transciever.c @@ -0,0 +1,69 @@ +/* + * ALSA SoC SPDIF DIT driver + * + * This driver is used by controllers which can operate in DIT (SPDI/F) where + * no codec is needed. This file provides stub codec that can be used + * in these configurations. TI DaVinci Audio controller uses this driver. + * + * Author: Steve Chen, <schen(a)mvista.com> + * Copyright: (C) 2009 MontaVista Software, Inc., <source(a)mvista.com> + * Copyright: (C) 2009 Texas Instruments, India + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <sound/soc.h> +#include <sound/pcm.h> + +#define STUB_RATES SNDRV_PCM_RATE_8000_96000 +#define STUB_FORMATS SNDRV_PCM_FMTBIT_S16_LE + + +struct snd_soc_dai dit_stub_dai = { + .name = "DIT", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 384, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, +}; + +static int spdif_dit_probe(struct platform_device *pdev) +{ + return snd_soc_register_dai(&dit_stub_dai); +} + +static int spdif_dit_remove(struct platform_device *pdev) +{ + snd_soc_unregister_dai(&dit_stub_dai); + return 0; +} + +static struct platform_driver spdif_dit_driver = { + .probe = spdif_dit_probe, + .remove = spdif_dit_remove, + .driver = { + .name = "spdif-dit", + .owner = THIS_MODULE, + }, +}; + +static int __init dit_modinit(void) +{ + return platform_driver_register(&spdif_dit_driver); +} + +static void __exit dit_exit(void) +{ + platform_driver_unregister(&spdif_dit_driver); +} + +module_init(dit_modinit); +module_exit(dit_exit); + diff --git a/sound/soc/codecs/spdif_transciever.h b/sound/soc/codecs/spdif_transciever.h new file mode 100644 index 0000000..296f2eb --- /dev/null +++ b/sound/soc/codecs/spdif_transciever.h @@ -0,0 +1,17 @@ +/* + * ALSA SoC DIT/DIR driver header + * + * Author: Steve Chen, <schen(a)mvista.com> + * Copyright: (C) 2008 MontaVista Software, Inc., <source(a)mvista.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. + */ + +#ifndef CODEC_STUBS_H +#define CODEC_STUBS_H + +extern struct snd_soc_dai dit_stub_dai; + +#endif /* CODEC_STUBS_H */ -- 1.5.6

2 1

[alsa-devel] [PATCH v3 6/6] ARM: DaVinci: ASoC: Add machine driver support for DM646x
by Chaithrika U S 28 May '09

28 May '09

This patch does the following: (1) Add support for the DM646x machine (2)Modifications required to introduce the platform driver model to get platform data for all the machines including dm355 and dm644x. Signed-off-by: Steve Chen <schen(a)mvista.com> Signed-off-by: Pavel Kiryukhin <pkiryukhin(a)ru.mvista.com> Signed-off-by: Naresh Medisetty <naresh(a)ti.com> Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- This patch applies against ALSA mainline - the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git sound/soc/davinci/davinci-evm.c | 103 ++++++++++++++++++-------------------- 1 files changed, 49 insertions(+), 54 deletions(-) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 58fd1cb..d66fbd3 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -25,11 +25,13 @@ #include <mach/asp.h> #include <mach/edma.h> #include <mach/mux.h> +#include <mach/cpu.h> #include "../codecs/tlv320aic3x.h" +#include "../codecs/spdif_transciever.h" #include "davinci-pcm.h" #include "davinci-i2s.h" - +#include "davinci-i2s-mcasp.h" #define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \ SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF) @@ -43,7 +45,7 @@ static int evm_hw_params(struct snd_pcm_substream *substream, unsigned sysclk; /* ASP1 on DM355 EVM is clocked by an external oscillator */ - if (machine_is_davinci_dm355_evm()) + if (machine_is_davinci_dm355_evm() || machine_is_davinci_dm6467_evm()) sysclk = 27000000; /* ASP0 in DM6446 EVM is clocked by U55, as configured by @@ -144,6 +146,24 @@ static struct snd_soc_dai_link evm_dai = { .ops = &evm_ops, }; +static struct snd_soc_dai_link dm6467_evm_dai[] = { + { + .name = "TLV320AIC3X", + .stream_name = "AIC3X", + .cpu_dai = &davinci_iis_mcasp_dai[DAVINCI_MCASP_I2S_DAI], + .codec_dai = &aic3x_dai, + .init = evm_aic3x_init, + .ops = &evm_ops, + }, + { + .name = "McASP", + .stream_name = "spdif", + .cpu_dai = &davinci_iis_mcasp_dai[DAVINCI_MCASP_DIT_DAI], + .codec_dai = &dit_stub_dai, + .ops = &evm_ops, + }, +}; + /* davinci-evm audio machine driver */ static struct snd_soc_card snd_soc_card_evm = { .name = "DaVinci EVM", @@ -152,12 +172,26 @@ static struct snd_soc_card snd_soc_card_evm = { .num_links = 1, }; +/* davinci dm6467 evm audio machine driver */ +static struct snd_soc_card dm6467_snd_soc_card_evm = { + .name = "DaVinci DM6467 EVM", + .platform = &davinci_soc_platform, + .dai_link = dm6467_evm_dai, + .num_links = ARRAY_SIZE(dm6467_evm_dai), +}; + /* evm audio private data */ static struct aic3x_setup_data evm_aic3x_setup = { .i2c_bus = 1, .i2c_address = 0x1b, }; +/* dm6467 evm audio private data */ +static struct aic3x_setup_data dm6467_evm_aic3x_setup = { + .i2c_bus = 1, + .i2c_address = 0x18, +}; + /* evm audio subsystem */ static struct snd_soc_device evm_snd_devdata = { .card = &snd_soc_card_evm, @@ -165,60 +199,30 @@ static struct snd_soc_device evm_snd_devdata = { .codec_data = &evm_aic3x_setup, }; -/* DM6446 EVM uses ASP0; line-out is a pair of RCA jacks */ -static struct resource evm_snd_resources[] = { - { - .start = DAVINCI_ASP0_BASE, - .end = DAVINCI_ASP0_BASE + SZ_8K - 1, - .flags = IORESOURCE_MEM, - }, -}; - -static struct evm_snd_platform_data evm_snd_data = { - .tx_dma_ch = DAVINCI_DMA_ASP0_TX, - .rx_dma_ch = DAVINCI_DMA_ASP0_RX, -}; - -/* DM335 EVM uses ASP1; line-out is a stereo mini-jack */ -static struct resource dm335evm_snd_resources[] = { - { - .start = DAVINCI_ASP1_BASE, - .end = DAVINCI_ASP1_BASE + SZ_8K - 1, - .flags = IORESOURCE_MEM, - }, -}; - -static struct evm_snd_platform_data dm335evm_snd_data = { - .tx_dma_ch = DAVINCI_DMA_ASP1_TX, - .rx_dma_ch = DAVINCI_DMA_ASP1_RX, +/* evm audio subsystem */ +static struct snd_soc_device dm6467_evm_snd_devdata = { + .card = &dm6467_snd_soc_card_evm, + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &dm6467_evm_aic3x_setup, }; static struct platform_device *evm_snd_device; static int __init evm_init(void) { - struct resource *resources; - unsigned num_resources; - struct evm_snd_platform_data *data; + struct snd_soc_device *evm_snd_dev_data; int index; int ret; if (machine_is_davinci_evm()) { - davinci_cfg_reg(DM644X_MCBSP); - - resources = evm_snd_resources; - num_resources = ARRAY_SIZE(evm_snd_resources); - data = &evm_snd_data; + evm_snd_dev_data = &evm_snd_devdata; index = 0; } else if (machine_is_davinci_dm355_evm()) { - /* we don't use ASP1 IRQs, or we'd need to mux them ... */ - davinci_cfg_reg(DM355_EVT8_ASP1_TX); - davinci_cfg_reg(DM355_EVT9_ASP1_RX); - - resources = dm335evm_snd_resources; - num_resources = ARRAY_SIZE(dm335evm_snd_resources); - data = &dm335evm_snd_data; + evm_snd_dev_data = &evm_snd_devdata; index = 1; + } else if (machine_is_davinci_dm6467_evm()) { + evm_snd_dev_data = &dm6467_evm_snd_devdata; + index = 0; } else return -EINVAL; @@ -226,17 +230,8 @@ static int __init evm_init(void) if (!evm_snd_device) return -ENOMEM; - platform_set_drvdata(evm_snd_device, &evm_snd_devdata); - evm_snd_devdata.dev = &evm_snd_device->dev; - platform_device_add_data(evm_snd_device, data, sizeof(*data)); - - ret = platform_device_add_resources(evm_snd_device, resources, - num_resources); - if (ret) { - platform_device_put(evm_snd_device); - return ret; - } - + platform_set_drvdata(evm_snd_device, evm_snd_dev_data); + evm_snd_dev_data->dev = &evm_snd_device->dev; ret = platform_device_add(evm_snd_device); if (ret) platform_device_put(evm_snd_device); -- 1.5.6

1 0

[alsa-devel] [PATCH v3 3/6] ARM: DaVinci: ASoC: Introduce platform driver model for dm644x, dm355
by Chaithrika U S 28 May '09

28 May '09

Introduce the platform driver model to get platform data for dm355 and dm644x. Register platform driver and acquire the resources in the probe function Since the platform specific code had been moved from machine driver to dm<soc>.c Signed-off-by: Naresh Medisetty <naresh(a)ti.com> Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- This patch applies against ALSA mainline - that's the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git sound/soc/davinci/davinci-i2s.c | 115 ++++++++++++++++++++++----------------- 1 files changed, 66 insertions(+), 49 deletions(-) diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index b1ea52f..dec7611 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -436,15 +436,39 @@ static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } -static int davinci_i2s_probe(struct platform_device *pdev, - struct snd_soc_dai *dai) +#define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 + +static struct snd_soc_dai_ops davinci_i2s_dai_ops = { + .startup = davinci_i2s_startup, + .trigger = davinci_i2s_trigger, + .hw_params = davinci_i2s_hw_params, + .set_fmt = davinci_i2s_set_dai_fmt, + +}; + +struct snd_soc_dai davinci_i2s_dai = { + .name = "davinci-i2s", + .id = 0, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = &davinci_i2s_dai_ops, + +}; +EXPORT_SYMBOL_GPL(davinci_i2s_dai); + +static int davinci_i2s_probe(struct platform_device *pdev) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_card *card = socdev->card; - struct snd_soc_dai *cpu_dai = card->dai_link->cpu_dai; + struct evm_snd_platform_data *pdata = pdev->dev.platform_data;; struct davinci_mcbsp_dev *dev; - struct resource *mem, *ioarea; - struct evm_snd_platform_data *pdata; + struct resource *mem, *ioarea, *res; int ret; mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); @@ -466,9 +490,7 @@ static int davinci_i2s_probe(struct platform_device *pdev, goto err_release_region; } - cpu_dai->private_data = dev; - - dev->clk = clk_get(&pdev->dev, NULL); + dev->clk = clk_get(&pdev->dev, pdata->clk_name); if (IS_ERR(dev->clk)) { ret = -ENODEV; goto err_free_mem; @@ -476,18 +498,35 @@ static int davinci_i2s_probe(struct platform_device *pdev, clk_enable(dev->clk); dev->base = (void __iomem *)IO_ADDRESS(mem->start); - pdata = pdev->dev.platform_data; dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK] = &davinci_i2s_pcm_out; - dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->channel = pdata->tx_dma_ch; dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->dma_addr = (dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DXR_REG); dev->dma_params[SNDRV_PCM_STREAM_CAPTURE] = &davinci_i2s_pcm_in; - dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->channel = pdata->rx_dma_ch; dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->dma_addr = (dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DRR_REG); + /* first TX, then RX */ + res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!res) { + dev_err(&pdev->dev, "no DMA resource\n"); + goto err_free_mem; + } + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->channel = res->start; + + res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!res) { + dev_err(&pdev->dev, "no DMA resource\n"); + goto err_free_mem; + } + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->channel = res->start; + + davinci_i2s_dai.private_data = dev; + ret = snd_soc_register_dai(&davinci_i2s_dai); + if (ret != 0) + goto err_free_mem; + return 0; err_free_mem: @@ -498,62 +537,40 @@ err_release_region: return ret; } -static void davinci_i2s_remove(struct platform_device *pdev, - struct snd_soc_dai *dai) +static int davinci_i2s_remove(struct platform_device *pdev) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_card *card = socdev->card; - struct snd_soc_dai *cpu_dai = card->dai_link->cpu_dai; - struct davinci_mcbsp_dev *dev = cpu_dai->private_data; + struct davinci_mcbsp_dev *dev = davinci_i2s_dai.private_data; struct resource *mem; + snd_soc_unregister_dai(&davinci_i2s_dai); clk_disable(dev->clk); clk_put(dev->clk); dev->clk = NULL; - kfree(dev); - mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); release_mem_region(mem->start, (mem->end - mem->start) + 1); -} -#define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 + return 0; +} -static struct snd_soc_dai_ops davinci_i2s_dai_ops = { - .startup = davinci_i2s_startup, - .trigger = davinci_i2s_trigger, - .hw_params = davinci_i2s_hw_params, - .set_fmt = davinci_i2s_set_dai_fmt, +static struct platform_driver davinci_mcbsp_driver = { + .probe = davinci_i2s_probe, + .remove = davinci_i2s_remove, + .driver = { + .name = "davinci-asp", + .owner = THIS_MODULE, + }, }; -struct snd_soc_dai davinci_i2s_dai = { - .name = "davinci-i2s", - .id = 0, - .probe = davinci_i2s_probe, - .remove = davinci_i2s_remove, - .playback = { - .channels_min = 2, - .channels_max = 2, - .rates = DAVINCI_I2S_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .capture = { - .channels_min = 2, - .channels_max = 2, - .rates = DAVINCI_I2S_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = &davinci_i2s_dai_ops, -}; -EXPORT_SYMBOL_GPL(davinci_i2s_dai); - static int __init davinci_i2s_init(void) { - return snd_soc_register_dai(&davinci_i2s_dai); + return platform_driver_register(&davinci_mcbsp_driver); } module_init(davinci_i2s_init); static void __exit davinci_i2s_exit(void) { - snd_soc_unregister_dai(&davinci_i2s_dai); + platform_driver_unregister(&davinci_mcbsp_driver); } module_exit(davinci_i2s_exit); -- 1.5.6

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[alsa-devel] [PATCH v3 0/6] ARM: DaVinci: Audio support for DM646x
by Chaithrika U S 28 May '09

28 May '09

From: Chaithrika U S <chaithrika(a)ti.com> Add audio support for DM646x Signed-off-by: Naresh Medisetty <naresh(a)ti.com> Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- This patch set adds audio support for DM646x This updated version has included most of the review comments received for the previous version of the patch set. This series is dependent on the following patches submitted to the DaVinci mailing list: [PATCH v2 1/2] ARM: DaVinci: EDMA: Move queue related mappings to dm<soc>.c [PATCH v2 2/2] ARM: DaVinci: EDMA: Enable EDMA support for dm646x Patch 1: Adds McASP clock support for the two instances of mcasp(mcasp0,mcasp1) This patch applies to the DaVinci git tree available at http://git.kernel.org/?p=linux/kernel/git/khilman/linux-davinci.git Patch 2: Registers the platform devices for ASP on dm355, dm644x and dm646x This patch applies to the DaVinci git tree available at http://git.kernel.org/?p=linux/kernel/git/khilman/linux-davinci.git Patch 3: Introduces platform driver model for dm644x and dm355 Applies against ALSA mainline - that's the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git Patch 4: Add dummy codec support Applies against ALSA mainline - that's the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git Patch 5: Adds driver support for the two instances of McASP on TI's DM646x Applies against ALSA mainline - that's the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git Patch 6: Add machine driver support for DM646x Applies against ALSA mainline - that's the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git These patches are based on the work done by Steve Chen <schen(a)mvista.com> and Pavel Kiryukhin <pkiryukhin(a)ru.mvista.com> on Monstavista 2.6.18 kernel.

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[alsa-devel] [PATCH] Compaq Presario CQ60 patching for Conexant
by Tony Vroon 28 May '09

28 May '09

A docking mic control is shown by default. The Compaq Presario CQ60 laptop has no docking connector, so designate it as a CXT5051_HP model. This makes the phantom mixer slider disappear. Signed-off-by: Tony Vroon <tony(a)linx.net> diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 56ce19e..4fcbe21 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1848,6 +1848,7 @@ static const char *cxt5051_models[CXT5051_MODELS] = { static struct snd_pci_quirk cxt5051_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x30cf, "HP DV6736", CXT5051_HP_DV6736), + SND_PCI_QUIRK(0x103c, 0x360b, "Compaq Presario CQ60", CXT5051_HP), SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board", CXT5051_LAPTOP), SND_PCI_QUIRK(0x14f1, 0x5051, "HP Spartan 1.1", CXT5051_HP),

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[alsa-devel] Testers wanted: New SB X-Fi driver
by jimi o'beirne 28 May '09

28 May '09

Takashi Thanks so much for making this driver! Unfortunately, I'm a bit of a newbie, how would I actually install the driver? I am using Ubuntu Jaunty with a SB X-Fi card. Thanks for your help, Jim

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Re: [alsa-devel] [Pkg-alsa-devel] Bug#529940: lib32asound2: symbol versioning changed in an incompatible way
by Elimar Riesebieter 28 May '09

28 May '09

tags 529940 pending forwarded 529940 alsa-devel(a)alsa-project.org thanks * Adeodato Simó [090527 13:53 +0200] > severity 529940 serious > retitle 529940 lib32asound2: symbol versioning changed in an incompatible way (ALSA_0.9.0rc4/8 dropped) > thanks > > Hello, > > it seems that between 1.0.19-1 and 1.0.20-1 the list of symbols in > lib32asound2 changed. More precisely, the list of symbols remained but > their associated versioning changed: all symbols associated with > ALSA_0.9.0rc4 and ALSA_0.9.0rc8 were changed to live under ALSA_0.9 > instead, making applications compiled against <= 1.0.19 fail to start > due to missing symbols. > > Note that this problem affects the 32-bit library in amd64, but not the > native libraries nor in i386 nor amd64. Don't we have to verify it at ppc64 as well? CC'd to debian-ppc. > > Also note that we've started having applications like WINE 1.1.19 in > experimental rebuilt against 1.0.20, and which don't work against > alsa-lib 1.0.19 either. And we'll have no easy way to detect these, > since shlibs are not bumped. > > Will you be able to upload fixed packages in a timely manner? According to [0] I created svn version 2249 on alioth [1]. I've tested it on amd64, were rc4 symbols appeared by checking via readelf. But maybe alsa-devel rechecks the buildsystem of alsa-libs to avoid that dirty hacks? --with-versioned appended to ./configure opts didn't help. [0] http://mailman.alsa-project.org/pipermail/alsa-devel/2009-April/016083.html [1] http://svn.debian.org/wsvn/pkg-alsa/?op=comp&compare[]=%2F@2248&compare[]=%… Jordi, please recheck and upload. Index: debian/rules =================================================================== --- debian/rules (Revision 2248) +++ debian/rules (Arbeitskopie) @@ -39,7 +39,7 @@ endif endif -CFLAGS = -g -Wall +CFLAGS = -g -Wall -DUSE_VERSIONED_SYMBOLS ifneq (,$(findstring noopt,$(DEB_BUILD_OPTIONS))) CFLAGS += -O0 -- "Talking much about oneself can also be a means to conceal oneself." -Friedrich Nietzsche

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