March 2010 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 2/2] Quick fix for the audio clocks on SAM9-L9261
by Benjamin Valentin 15 Mar '10

15 Mar '10

Signed-off-by: Dimitar Dimitrov <dimitar(a)olimex.com> From 10a85229b2357c89d5170efde5ac0e1cba4c9f9e Mon Sep 17 00:00:00 2001 From: Dimitar Dimitrov <dimitar(a)olimex.com> Date: Wed, 27 May 2009 08:11:37 +0300 Subject: [PATCH] Quick fix for the audio clocks on SAM9-L9261 This patch together with support from SAM9-L9261 hardware fix an issue with audio clocks. AT73C213 needs MCLK and BCLK with equal phase. The MCLK from AT91SAM9XXX clock output and BCLK from the SSC TK clock output are not phase-synchronized. Thus the SAM9-L9261 board provides external hardware divison by 8 from MCLK to BCLK, ensuring zero phase difference. See: http://www.at91.com/phpbb/viewtopic.php?t=2633 --- sound/spi/at73c213.c | 54 ++++++++++++++++++++++++++++++++++++++----------- 1 files changed, 42 insertions(+), 12 deletions(-) diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c index 4c7b051..325e534 100644 --- a/sound/spi/at73c213.c +++ b/sound/spi/at73c213.c @@ -29,6 +29,8 @@ #include <linux/atmel-ssc.h> +#include <asm/mach-types.h> + #include <linux/spi/spi.h> #include <linux/spi/at73c213.h> @@ -114,7 +116,11 @@ snd_at73c213_write_reg(struct snd_at73c213 *chip, u8 reg, u8 val) static struct snd_pcm_hardware snd_at73c213_playback_hw = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER, +#ifdef __BIG_ENDIAN .formats = SNDRV_PCM_FMTBIT_S16_BE, +#else + .formats = SNDRV_PCM_FMTBIT_S16_LE, +#endif .rates = SNDRV_PCM_RATE_CONTINUOUS, .rate_min = 8000, /* Replaced by chip->bitrate later. */ .rate_max = 50000, /* Replaced by chip->bitrate later. */ @@ -144,7 +150,8 @@ static int snd_at73c213_set_bitrate(struct snd_at73c213 *chip) * as practical to the desired target rate. * * The DAC master clock (MCLK) is programmable, and is either 256 - * or (not here) 384 times the I2S output clock (BCLK). + * or (not here) 384 times the I2S frame clock (LRC), that is + * 8 or 12 times the output bit clock (BCLK). */ /* SSC clock / (bitrate * stereo * 16-bit). */ @@ -747,17 +754,40 @@ cleanup: */ static int __devinit snd_at73c213_ssc_init(struct snd_at73c213 *chip) { - /* - * Continuous clock output. - * Starts on falling TF. - * Delay 1 cycle (1 bit). - * Periode is 16 bit (16 - 1). - */ - ssc_writel(chip->ssc->regs, TCMR, - SSC_BF(TCMR_CKO, 1) - | SSC_BF(TCMR_START, 4) - | SSC_BF(TCMR_STTDLY, 1) - | SSC_BF(TCMR_PERIOD, 16 - 1)); + if (machine_is_sam9_l9261()) { + /* + * dimitar(a)olimex.com : AT73C213 needs MCLK and BCLK with equal + * phase. The MCLK from AT91SAM9XXX clock output and BCLK from + * the SSC TK clock output are not phase-synchronized. Thus + * the SAM9-L9261 board provides external hardware divison by + * 8 from MCLK to BCLK, ensuring zero phase difference. + * See: http://www.at91.com/phpbb/viewtopic.php?t=2633 + * + * TK is clock input. + * Clock source is TK pin. + * Starts on falling TF. + * Delay 1 cycle (1 bit). + * Period is 16 bit (16 - 1). + */ + ssc_writel(chip->ssc->regs, TCMR, + SSC_BF(TCMR_CKS, 2) + | SSC_BF(TCMR_CKO, 0) + | SSC_BF(TCMR_START, 4) + | SSC_BF(TCMR_STTDLY, 1) + | SSC_BF(TCMR_PERIOD, 16 - 1)); + } else { + /* + * Continuous clock output. + * Starts on falling TF. + * Delay 1 cycle (1 bit). + * Period is 16 bit (16 - 1). + */ + ssc_writel(chip->ssc->regs, TCMR, + SSC_BF(TCMR_CKO, 1) + | SSC_BF(TCMR_START, 4) + | SSC_BF(TCMR_STTDLY, 1) + | SSC_BF(TCMR_PERIOD, 16 - 1)); + } /* * Data length is 16 bit (16 - 1). * Transmit MSB first. -- 1.5.6.1

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[alsa-devel] 2.6.34-rc1: regression: ^Z no longer stops sound
by Pavel Machek 15 Mar '10

15 Mar '10

Hi! In 2.6.33, hitting ^Z stopped mpg123 in a way you'd expect (both on thinkpad and geode notebook). On 2.6.34-rc1, hit ^Z and sound starts looping. Pavel -- (english) http://www.livejournal.com/~pavelmachek (cesky, pictures) http://atrey.karlin.mff.cuni.cz/~pavel/picture/horses/blog.html

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Re: [alsa-devel] [2.6.34-rc1 REGRESSION] alsa: no mixer controls (and no sound) with 2.6.24-rc1+ on Realtek ALC260
by Takashi Iwai 15 Mar '10

15 Mar '10

At Mon, 15 Mar 2010 01:06:17 +0100, Alessandro Guido wrote: > > With 2.6.34-rc1+ my laptop (Sony Vaio VGN-FS215S) can't reproduce any > sound. All controls are missing from alsamixer and this message is found > in dmesg: > > hda_codec: cannot build controlsfor #0 (error -22) > > Kernel config is attached, and the output from alsa-info.sh follows: Does the patch below fix? Takashi --- diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3a83719..ba45868 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6477,7 +6477,7 @@ static struct alc_config_preset alc260_presets[] = { .num_dacs = ARRAY_SIZE(alc260_dac_nids), .dac_nids = alc260_dac_nids, .num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids), - .adc_nids = alc260_adc_nids, + .adc_nids = alc260_dual_adc_nids, .num_channel_mode = ARRAY_SIZE(alc260_modes), .channel_mode = alc260_modes, .input_mux = &alc260_capture_source,

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Re: [alsa-devel] [PATCH]Support MCP89 and GT21x hdmi audio
by Jaroslav Kysela 14 Mar '10

14 Mar '10

On Sat, 13 Mar 2010, VDR User wrote: > [pid 9629] 1268511516.215224 ioctl(18, 0x400c4150, 0xb43631a4) = 0 ^^^^ here xine feeds new samples to the ALSA driver (0x400c4150 == SNDRV_PCM_IOCTL_WRITEI_FRAMES) > [pid 9629] 1268511516.215275 ioctl(18, 0x80044121, 0xb43632cc) = 0 > [pid 9627] 1268511520.700109 ioctl(10, 0xc0104652, 0xb5443f00) = 0 > [pid 9629] 1268511520.700340 ioctl(18, 0x80044121, 0xb43632cc) = -1 > EPIPE (Broken pipe) ^^^^ next iteration (0x80044121 == SNDRV_PCM_IOCTL_DELAY) - see time difference - 4.485 seconds from last WRITEI It seems like a xine bug or the system scheduler bug or any kernel bug affecting the system scheduler. The ALSA subsystem is not reponsible for this behaviour - the ring buffer in your case is for about 0.372 seconds of samples. This limit must not be crossed for the click-free playback. Ask the xine developers for more help or try another player. Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

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[alsa-devel] ASoC updates for 2.6.35
by Mark Brown 13 Mar '10

13 Mar '10

The following changes since commit fad837c16cdd856c68ce2e1335ad0fe836ed8ecd: Mark Brown (1): Merge commit 'v2.6.34-rc1' into for-2.6.35 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.35 Jassi Brar (3): ASoC: S3C64XX: I2S: Make BCLK independent of sample size ASoC: S3C: I2Sv2: Segregate hw_params callback ASoC: PCM_RATE: Check for KNOT and CONTINUOUS flags Kuninori Morimoto (1): ASoC: da7210: Add 11025/22050/44100/88200 rate support Miguel Aguilar (4): MFD: DaVinci Voice Codec ASoC: DaVinci: Voice Codec Interface ASoC: DaVinci: CQ93VC Voice Codec DaVinci: DM365: Voice Codec support for the DM365 EVM Peter Ujfalusi (4): ASoC: tlv320dac33: Start/stop sequence change ASoC: tlv320dac33: Add option for keeping the BCLK running OMAP3: McBSP: Add interface for FIFO caused delay query ASoC: OMAP3: Report delay caused by the internal FIFO arch/arm/mach-davinci/board-dm365-evm.c | 4 + arch/arm/plat-omap/include/plat/mcbsp.h | 6 + arch/arm/plat-omap/mcbsp.c | 55 ++++++ drivers/mfd/Kconfig | 4 + drivers/mfd/Makefile | 1 + drivers/mfd/davinci_voicecodec.c | 189 +++++++++++++++++++ include/linux/mfd/davinci_voicecodec.h | 126 +++++++++++++ include/sound/tlv320dac33-plat.h | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cq93vc.c | 298 +++++++++++++++++++++++++++++++ sound/soc/codecs/cq93vc.h | 29 +++ sound/soc/codecs/da7210.c | 90 +++++++++- sound/soc/codecs/tlv320dac33.c | 38 ++--- sound/soc/davinci/Kconfig | 27 +++- sound/soc/davinci/Makefile | 2 + sound/soc/davinci/davinci-evm.c | 45 +++++- sound/soc/davinci/davinci-vcif.c | 272 ++++++++++++++++++++++++++++ sound/soc/davinci/davinci-vcif.h | 28 +++ sound/soc/omap/omap-mcbsp.c | 26 +++ sound/soc/s3c24xx/s3c-i2s-v2.c | 27 +--- sound/soc/s3c24xx/s3c2412-i2s.c | 37 ++++ sound/soc/soc-core.c | 12 ++ 23 files changed, 1270 insertions(+), 53 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.c create mode 100644 sound/soc/davinci/davinci-vcif.h

2 1

[alsa-devel] [PATCHv5 0/2] TWL6040 audio codec initial support
by Olaya, Margarita 13 Mar '10

13 Mar '10

Following patch series add initial support for TWL6040 codec driver. The patch set is based on the series sent from Misael Lopez in October 2009. Changes from v4: - Rename twl6030_codec as twl6040_codec - Rebase the 6 codec patches into a single patch - Rename twl6040_read as twl6040_read_reg_volatile - Use msleep instead of large mdelays - Add more trigger sources - Restructure set_bias_level function to avoid transitions like PREPARE->ON Regards, Margarita --- Misael Lopez Cruz (1): ASoC: TWL6040: Add twl6040 codec driver Margarita Olaya Cabrera (1): OMAP4: PMIC: Rename twl6030_codec as twl6040_codec drivers/mfd/twl-core.c | 4 +- include/linux/i2c/twl.h | 6 +- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/twl6040.c | 1223 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/twl6040.h | 141 +++++ 6 files changed, 1375 insertions(+), 5 deletions(-) create mode 100644 sound/soc/codecs/twl6040.c create mode 100644 sound/soc/codecs/twl6040.h

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[alsa-devel] [PATCH] ASoC: PCM_RATE: Check for KNOT and CONTINUOUS flags
by Jassi Brar 12 Mar '10

12 Mar '10

For ASoC, if either CPU or CODEC driver has set the flag, the MACHINE driver should be given a chance to figure out if the dai, that set the flag, can accomodate a rate that it does not explicitly specify but is specified by the dai at the other end of the link. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/soc-core.c | 12 ++++++++++++ 1 files changed, 12 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 06c38d1..eb73aab 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -404,6 +404,12 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) codec_dai->playback.formats & cpu_dai->playback.formats; runtime->hw.rates = codec_dai->playback.rates & cpu_dai->playback.rates; + if (codec_dai->playback.rates + & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) + runtime->hw.rates |= cpu_dai->playback.rates; + if (cpu_dai->playback.rates + & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) + runtime->hw.rates |= codec_dai->playback.rates; } else { runtime->hw.rate_min = max(codec_dai->capture.rate_min, @@ -421,6 +427,12 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) codec_dai->capture.formats & cpu_dai->capture.formats; runtime->hw.rates = codec_dai->capture.rates & cpu_dai->capture.rates; + if (codec_dai->capture.rates + & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) + runtime->hw.rates |= cpu_dai->capture.rates; + if (cpu_dai->capture.rates + & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) + runtime->hw.rates |= codec_dai->capture.rates; } snd_pcm_limit_hw_rates(runtime); -- 1.6.2.5

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[alsa-devel] [PATCH v2 1/5] MFD: DaVinci Voice Codec
by miguel.aguilar＠ridgerun.com 12 Mar '10

12 Mar '10

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> This is the MFD driver for the DaVinci Voice codec, it has two clients: * Voice codec interface * Voice codec CQ93VC Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- drivers/mfd/Kconfig | 4 + drivers/mfd/Makefile | 1 + drivers/mfd/davinci_voicecodec.c | 189 ++++++++++++++++++++++++++++++++ include/linux/mfd/davinci_voicecodec.h | 126 +++++++++++++++++++++ 4 files changed, 320 insertions(+), 0 deletions(-) create mode 100644 drivers/mfd/davinci_voicecodec.c create mode 100644 include/linux/mfd/davinci_voicecodec.h diff --git a/drivers/mfd/Kconfig b/drivers/mfd/Kconfig index 8782978..6e93c0b 100644 --- a/drivers/mfd/Kconfig +++ b/drivers/mfd/Kconfig @@ -43,6 +43,10 @@ config MFD_SH_MOBILE_SDHI This driver supports the SDHI hardware block found in many SuperH Mobile SoCs. +config MFD_DAVINCI_VOICECODEC + tristate + select MFD_CORE + 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..9886aa8 --- /dev/null +++ b/drivers/mfd/davinci_voicecodec.c @@ -0,0 +1,189 @@ +/* + * 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/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 __init davinci_vc_probe(struct platform_device *pdev) +{ + struct davinci_vc *davinci_vc; + struct resource *res, *mem; + struct mfd_cell *cell = NULL; + 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; + davinci_vc->pdev = pdev; + + /* Voice codec interface client */ + cell = &davinci_vc->cells[DAVINCI_VC_VCIF_CELL]; + cell->name = "davinci_vcif"; + cell->driver_data = davinci_vc; + + /* Voice codec CQ93VC client */ + cell = &davinci_vc->cells[DAVINCI_VC_CQ93VC_CELL]; + cell->name = "cq93vc"; + cell->driver_data = davinci_vc; + + ret = mfd_add_devices(&pdev->dev, pdev->id, davinci_vc->cells, + DAVINCI_VC_CELLS, NULL, 0); + if (ret != 0) { + dev_err(&pdev->dev, "fail to register client devices\n"); + goto fail4; + } + + return 0; + +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); + + mfd_remove_devices(&pdev->dev); + + 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..0ab6132 --- /dev/null +++ b/include/linux/mfd/davinci_voicecodec.h @@ -0,0 +1,126 @@ +/* + * 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 <linux/mfd/core.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 + +#define DAVINCI_VC_CELLS 2 + +enum davinci_vc_cells { + DAVINCI_VC_VCIF_CELL, + DAVINCI_VC_CQ93VC_CELL, +}; + +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 platform_device *pdev; + struct clk *clk; + + /* Memory resources */ + void __iomem *base; + resource_size_t pbase; + size_t base_size; + + /* MFD cells */ + struct mfd_cell cells[DAVINCI_VC_CELLS]; + + /* Client devices */ + struct davinci_vcif davinci_vcif; + struct cq93vc cq93vc; +}; + +#endif -- 1.6.0.4

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[alsa-devel] [PATCH 0/2] ASoC: tlv320dac33 fixes
by Peter Ujfalusi 12 Mar '10

12 Mar '10

Hello, Revisiting the start/stop sequence used by the tlv320dac33 codec in order to avoid race conditions, which can occur, when the codec is in FIFO modes. It has been found that OMAP3 McBSP needs the BCLK continuously running on the serial interface when it is configured to be slave. DAC33 can be configured in a way, that it keeps the BCLK ticking, and only enable the FS when it is needed. Option for the platform data to select between continuous BCLK or non continuous BCLK. --- Peter Ujfalusi (2): ASoC: tlv320dac33: Start/stop sequence change ASoC: tlv320dac33: Add option for keeping the BCLK running include/sound/tlv320dac33-plat.h | 1 + sound/soc/codecs/tlv320dac33.c | 38 ++++++++++++++------------------------ 2 files changed, 15 insertions(+), 24 deletions(-)

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[alsa-devel] [PATCHv2 0/5] FIFO caused playback delay (latency) handling in soc/omap
by Peter Ujfalusi 12 Mar '10

12 Mar '10

Hello, Changes since the RFC round on the alsa-devel list: - core is not limiting the query for the delay to the playback stream. It is no the dai and platform driver's responsibility handle that - delay interface added for platform driver - Capture path delay calculation is added to OMAP McBSP. - omap_mcbsp_get_tx_buffstat renamed as omap_mcbsp_get_tx_delay to reflect what it is actually doing. Tony: is it still possible to get this taken for 2.6.34? It would be nice to avoid cross tree problems during the 34 cycle... The initial RFC patch description: There has been discussion in alsa-devel in this subject several times, but no actual patches has been sent (it was not soc related, but anyway it was discussed). Based on the available information the latency caused by the HW buffer on some systems can be handled by updating the runtime->delay. It has been discussed, that the runtime->delay can also be updated dynamically to show more accurate delay. To further complicate things, in ASoC we could have more buffer in the chain. To handle this we need soc level support. This series tries to do that in soc by: - introducing a pcm_pointer wrapper - in this wrapper we call the original pcm_pointer functions to get the DMA pointer - introducing a new interface in dai_ops and for platfrom drivers to ask the delay - adding the cpu_dai, codec_dai and platform driver returned delay to form the actual delay - update the runtime->delay with this value. With this approach none of the existing drivers need change, but they can add support for specifying the FIFO caused delay. In this series on top of the core changes the omap(3) code is updated to take this delay reporting into use. I have not added the support to the tlv320dac33 codec driver, since it needs a bit more work, but along the same line it can be done, and if the tlv320dac33 is hooked to omap McBSP than applications can know the whole delay/latency on that path. The series is based on linux-omap master branch (with regcache and sidetone). However the first 3 patch for soc core applies on top of sound-2.6 topic/asoc cleanly as well. Obviously the last patch depends on the OMAP platform patch. --- Peter Ujfalusi (5): ASoC: core: fix tailing whitespace in soc_pcm_apply_symmetry ASoC: core: soc level wrapper for pcm_pointer callback ASoC: core: Add delay operation to snd_soc_dai_ops OMAP3: McBSP: Add interface for FIFO caused delay query ASoC: OMAP3: Report delay caused by the internal FIFO arch/arm/plat-omap/include/plat/mcbsp.h | 6 +++ arch/arm/plat-omap/mcbsp.c | 55 +++++++++++++++++++++++++++++++ include/sound/soc-dai.h | 6 +++ include/sound/soc.h | 7 ++++ sound/soc/omap/omap-mcbsp.c | 26 ++++++++++++++ sound/soc/soc-core.c | 39 +++++++++++++++++++++- 6 files changed, 137 insertions(+), 2 deletions(-)

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