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[alsa-devel] [PATCH 1/2] ASoC: DaVinci: Voice Codec Support
by miguel.aguilar＠ridgerun.com 18 Jan '10

18 Jan '10

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> There are two clients for the MFD driver: * Voice Codec Interface. * Voice Codec itself, CQ0093. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- drivers/mfd/Kconfig | 5 + drivers/mfd/Makefile | 1 + drivers/mfd/davinci_voicecodec.c | 216 ++++++++++++++++++++ include/linux/mfd/davinci_voicecodec.h | 113 +++++++++++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cq93vc.c | 342 ++++++++++++++++++++++++++++++++ sound/soc/codecs/cq93vc.h | 29 +++ sound/soc/davinci/Kconfig | 10 +- sound/soc/davinci/Makefile | 2 + sound/soc/davinci/davinci-vcif.h | 28 +++ 11 files changed, 750 insertions(+), 2 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.h diff --git a/drivers/mfd/Kconfig b/drivers/mfd/Kconfig index 8782978..efeba26 100644 --- a/drivers/mfd/Kconfig +++ b/drivers/mfd/Kconfig @@ -43,6 +43,11 @@ config MFD_SH_MOBILE_SDHI This driver supports the SDHI hardware block found in many SuperH Mobile SoCs. +config MFD_DAVINCI_VOICECODEC + bool "DaVinci Voice Codec" + help + This driver supports the DaVinci voice codec and its interface. + 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..94dbeaa --- /dev/null +++ b/drivers/mfd/davinci_voicecodec.c @@ -0,0 +1,216 @@ +/* + * 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/core.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 davinci_vc_client_dev_register(struct davinci_vc *davinci_vc, + const char *name, + struct platform_device **pdev) +{ + int ret; + + *pdev = platform_device_alloc(name, -1); + if (pdev == NULL) { + dev_err(davinci_vc->dev, "failed to allocate %s\n", name); + return -ENODEV; + } + + (*pdev)->dev.parent = davinci_vc->dev; + platform_set_drvdata(*pdev, davinci_vc); + ret = platform_device_add(*pdev); + if (ret != 0) { + dev_err(davinci_vc->dev, "failed to register %s: %d\n", name, + ret); + platform_device_put(*pdev); + *pdev = NULL; + return ret; + } + + return 0; +} + +static int __init davinci_vc_probe(struct platform_device *pdev) +{ + struct davinci_vc *davinci_vc; + struct resource *res, *mem; + 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; + + /* Register voice codec interface */ + ret = davinci_vc_client_dev_register(davinci_vc, "davinci_vcif", + &(davinci_vc->davinci_vcif.pdev)); + if (ret != 0) { + dev_err(&pdev->dev, "fail to register client device\n"); + goto fail4; + } + + /* Register voice codec */ + ret = davinci_vc_client_dev_register(davinci_vc, "cq93vc", + &(davinci_vc->cq93vc.pdev)); + if (ret != 0) { + dev_err(&pdev->dev, "fail to register client device\n"); + goto fail5; + } + + return 0; + +fail5: + platform_device_unregister(davinci_vc->davinci_vcif.pdev); +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); + + platform_device_unregister(davinci_vc->cq93vc.pdev); + platform_device_unregister(davinci_vc->davinci_vcif.pdev); + + 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..6a1757a --- /dev/null +++ b/include/linux/mfd/davinci_voicecodec.h @@ -0,0 +1,113 @@ +/* + * 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 <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 + +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 clk *clk; + + /* Memory resources */ + void __iomem *base; + resource_size_t pbase; + size_t base_size; + + /* Client devices */ + struct davinci_vcif davinci_vcif; + struct cq93vc cq93vc; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 52b005f..a759d06 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -21,6 +21,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4535 if I2C select SND_SOC_AK4642 if I2C select SND_SOC_AK4671 if I2C + select SND_SOC_CQ0093VC select SND_SOC_CS4270 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 @@ -108,6 +109,9 @@ config SND_SOC_AK4642 config SND_SOC_AK4671 tristate +config SND_SOC_CQ0093VC + tristate + # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index dbaecb1..674db53 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -8,6 +8,7 @@ snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o +snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs4270-objs := cs4270.o snd-soc-cx20442-objs := cx20442.o snd-soc-l3-objs := l3.o @@ -64,6 +65,7 @@ obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o +obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c new file mode 100644 index 0000000..f420e01 --- /dev/null +++ b/sound/soc/codecs/cq93vc.c @@ -0,0 +1,342 @@ +/* + * ALSA SoC CQ0093 Voice Codec Driver for DaVinci 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/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/io.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/device.h> +#include <linux/clk.h> +#include <linux/mfd/davinci_voicecodec.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include <mach/dm365.h> + +#include "cq93vc.h" + +static inline unsigned int cq93vc_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + struct davinci_vc *davinci_vc = codec->control_data; + + return readl(davinci_vc->base + reg); +} + +static inline int cq93vc_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + struct davinci_vc *davinci_vc = codec->control_data; + + writel(value, davinci_vc->base + reg); + + return 0; +} + +static const struct snd_kcontrol_new cq93vc_snd_controls[] = { + SOC_SINGLE("PGA Capture Volume", DAVINCI_VC_REG05, 0, 0x03, 0), + SOC_SINGLE("Mono DAC Playback Volume", DAVINCI_VC_REG09, 0, 0x3f, 0), +}; + +static int cq93vc_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u8 reg = cq93vc_read(codec, DAVINCI_VC_REG09) & ~DAVINCI_VC_REG09_MUTE; + + if (mute) + cq93vc_write(codec, DAVINCI_VC_REG09, + reg | DAVINCI_VC_REG09_MUTE); + else + cq93vc_write(codec, DAVINCI_VC_REG09, reg); + + return 0; +} + +static int cq93vc_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(cq93vc_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&cq93vc_snd_controls[i], + codec, NULL)); + if (err < 0) + return err; + } + + return 0; +} + +static int cq93vc_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct davinci_vc *davinci_vc = codec->control_data; + + switch (freq) { + case 22579200: + case 27000000: + case 33868800: + davinci_vc->cq93vc.sysclk = freq; + return 0; + } + + return -EINVAL; +} + +static int cq93vc_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + cq93vc_write(codec, DAVINCI_VC_REG12, + DAVINCI_VC_REG12_POWER_ALL_ON); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + cq93vc_write(codec, DAVINCI_VC_REG12, + DAVINCI_VC_REG12_POWER_ALL_OFF); + break; + case SND_SOC_BIAS_OFF: + /* force all power off */ + cq93vc_write(codec, DAVINCI_VC_REG12, + DAVINCI_VC_REG12_POWER_ALL_OFF); + break; + } + codec->bias_level = level; + + return 0; +} + +#define CQ93VC_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000) +#define CQ93VC_FORMATS (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE) + +static struct snd_soc_dai_ops cq93vc_dai_ops = { + .digital_mute = cq93vc_mute, + .set_sysclk = cq93vc_set_dai_sysclk, +}; + +struct snd_soc_dai cq93vc_dai = { + .name = "CQ93VC", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CQ93VC_RATES, + .formats = CQ93VC_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CQ93VC_RATES, + .formats = CQ93VC_FORMATS,}, + .ops = &cq93vc_dai_ops, +}; +EXPORT_SYMBOL_GPL(cq93vc_dai); + +static int cq93vc_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + cq93vc_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int cq93vc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + cq93vc_set_bias_level(codec, codec->suspend_bias_level); + + return 0; +} + +static struct snd_soc_codec *cq93vc_codec; + +static int cq93vc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct device *dev = &pdev->dev; + struct snd_soc_codec *codec; + int ret; + + socdev->card->codec = cq93vc_codec; + codec = socdev->card->codec; + + /* Set the PGA Gain to 18 dB */ + cq93vc_write(codec, DAVINCI_VC_REG05, DAVINCI_VC_REG05_PGA_GAIN); + + /* Set the DAC digital attenuation to 0 dB */ + cq93vc_write(codec, DAVINCI_VC_REG09, DAVINCI_VC_REG09_DIG_ATTEN); + + /* Register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(dev, "%s: failed to create pcms\n", pdev->name); + return ret; + } + + /* Set controls */ + cq93vc_add_controls(codec); + + /* Off, with power on */ + cq93vc_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +static int cq93vc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + /* Power down chip */ + if (codec->control_data) + cq93vc_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_cq93vc = { + .probe = cq93vc_probe, + .remove = cq93vc_remove, + .suspend = cq93vc_suspend, + .resume = cq93vc_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_cq93vc); + +static __init int cq93vc_codec_probe(struct platform_device *pdev) +{ + struct davinci_vc *davinci_vc = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret; + + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) { + dev_dbg(davinci_vc->dev, + "could not allocate memory for codec data\n"); + return -ENOMEM; + } + + davinci_vc->cq93vc.codec = codec; + + cq93vc_dai.dev = &pdev->dev; + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + codec->dev = &pdev->dev; + codec->name = "CQ93VC"; + codec->owner = THIS_MODULE; + codec->read = cq93vc_read; + codec->write = cq93vc_write; + codec->set_bias_level = cq93vc_set_bias_level; + codec->dai = &cq93vc_dai; + codec->num_dai = 1; + codec->control_data = davinci_vc; + + cq93vc_codec = codec; + + ret = snd_soc_register_dai(&cq93vc_dai); + if (ret != 0) { + dev_err(davinci_vc->dev, "could register dai\n"); + goto fail1; + } + return 0; + +fail1: + kfree(codec); + cq93vc_codec = NULL; + + return ret; +} + +static int __devexit cq93vc_codec_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + snd_soc_unregister_dai(&cq93vc_dai); + + kfree(codec); + cq93vc_codec = NULL; + + return 0; +} + +#ifdef CONFIG_PM +static int cq93vc_codec_suspend(struct platform_device *pdev, pm_message_t m) +{ + return snd_soc_suspend_device(&pdev->dev); +} + +static int cq93vc_codec_resume(struct platform_device *pdev) +{ + return snd_soc_resume_device(&pdev->dev); +} +#else +#define cq93vc_codec_suspend NULL +#define cq93vc_codec_resume NULL +#endif + +static struct platform_driver cq93vc_codec_driver = { + .driver = { + .name = "cq93vc", + .owner = THIS_MODULE, + }, + .probe = cq93vc_codec_probe, + .remove = __devexit_p(cq93vc_codec_remove), + .suspend = cq93vc_codec_suspend, + .resume = cq93vc_codec_resume, +}; + +static __init int cq93vc_init(void) +{ + return platform_driver_probe(&cq93vc_codec_driver, cq93vc_codec_probe); +} +module_init(cq93vc_init); + +static __exit void cq93vc_exit(void) +{ + platform_driver_unregister(&cq93vc_codec_driver); +} +module_exit(cq93vc_exit); + +MODULE_DESCRIPTION("Texas Instruments DaVinci ASoC CQ0093 Voice Codec Driver"); +MODULE_AUTHOR("Miguel Aguilar"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cq93vc.h b/sound/soc/codecs/cq93vc.h new file mode 100644 index 0000000..845b196 --- /dev/null +++ b/sound/soc/codecs/cq93vc.h @@ -0,0 +1,29 @@ +/* + * ALSA SoC CQ0093 Voice Codec Driver for DaVinci 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 _CQ93VC_H +#define _CQ93VC_H + +extern struct snd_soc_dai cq93vc_dai; +extern struct snd_soc_codec_device soc_codec_dev_cq93vc; + +#endif diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 047ee39..3aa04e6 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -12,15 +12,21 @@ config SND_DAVINCI_SOC_I2S config SND_DAVINCI_SOC_MCASP tristate +config SND_DAVINCI_SOC_VCIF + tristate + config SND_DAVINCI_SOC_EVM tristate "SoC Audio support for DaVinci DM6446, DM355 or DM365 EVM" depends on SND_DAVINCI_SOC - depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM || MACH_DAVINCI_DM365_EVM + depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM || MACH_DAVINCI_DM365_EVM select SND_DAVINCI_SOC_I2S select SND_SOC_TLV320AIC3X + select MFD_DAVINCI_VOICECODEC + select SND_DAVINCI_SOC_VCIF + select SND_SOC_CQ0093VC help Say Y if you want to add support for SoC audio on TI - DaVinci DM6446 or DM355 EVM platforms. + DaVinci DM6446, DM355 or DM365 EVM platforms. config SND_DM6467_SOC_EVM tristate "SoC Audio support for DaVinci DM6467 EVM" diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index a6939d7..a93679d 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -2,10 +2,12 @@ snd-soc-davinci-objs := davinci-pcm.o snd-soc-davinci-i2s-objs := davinci-i2s.o snd-soc-davinci-mcasp-objs:= davinci-mcasp.o +snd-soc-davinci-vcif-objs:= davinci-vcif.o obj-$(CONFIG_SND_DAVINCI_SOC) += snd-soc-davinci.o obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o obj-$(CONFIG_SND_DAVINCI_SOC_MCASP) += snd-soc-davinci-mcasp.o +obj-$(CONFIG_SND_DAVINCI_SOC_VCIF) += snd-soc-davinci-vcif.o # DAVINCI Machine Support snd-soc-evm-objs := davinci-evm.o diff --git a/sound/soc/davinci/davinci-vcif.h b/sound/soc/davinci/davinci-vcif.h new file mode 100644 index 0000000..571c994 --- /dev/null +++ b/sound/soc/davinci/davinci-vcif.h @@ -0,0 +1,28 @@ +/* + * ALSA SoC Voice Codec Interface for TI DAVINCI processor + * + * Copyright (C) 2010 Texas Instruments. + * + * 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 _DAVINCI_VCIF_H +#define _DAVINCI_VCIF_H + +extern struct snd_soc_dai davinci_vcif_dai; + +#endif -- 1.6.0.4

3 3

[alsa-devel] Echo Indigo card on Sabayon Gentoo
by S. Aguinaga 18 Jan '10

18 Jan '10

Hello Fellows, Just downloaded the latest (released) alsa-lib, alsa-driver, and alsa-utils, but my Indigo card does not work. It has been tested on the same system using the same drivers and on the same system using Fedora 10, that works fine. Do you know what I might have to do to get it working on Gentoo (sabayon)? When I type the following, this is what I get: ~/Downloads/alsa-utils-1.0.22/alsactl $ ./alsactl init 2 Unknown hardware: "Echo_Indigo IO" "" "" "0xecc0" "0x00a0" Hardware is initialized using a guess method Using aplay to list the device: card 2: IO [Indigo IO], device 0: PCM [Indigo IO] Subdevices: 8/8 .... etc. Using aplay to test the card here is the command: aplay -D plughw:2,0 <wav file> And this does not work, I have checked that things are ok with alsamixer, but still! Any help will be appreciated or let me know if this is best addressed by submitting a bug. // Salvador

2 2

[alsa-devel] WARNING: at sound/core/sound_oss.c:96 snd_oss_kernel_minor+0xdf/0x120 [snd]()
by John Kacur 18 Jan '10

18 Jan '10

The following was in my 2.6.33-rc3 dmesg (of course that doesn't mean that the problem didn't exist earlier too) WARNING: at /home/jkacur/rt.linux.git/sound/core/sound_oss.c:96 snd_oss_kernel_minor+0xdf/0x120 [snd]() Hardware name: 2241B48 BUG? (minor < 0 || minor >= 128) Modules linked in: snd_mixer_oss thinkpad_acpi(+) sdhci hwmon i2c_i801 firewire_ohci snd_pcm mmc_core snd_timer cfg80211 firewire_core btusb joydev iTCO_wdt bluetooth ata_generic e1000e i2c_core ppdev snd battery ac iTCO_vendor_support parport_pc ricoh_mmc intel_agp pata_acpi pcspkr parport tpm_tis video wmi output soundcore button tpm rfkill sr_mod tpm_bios snd_page_alloc crc_itu_t cdrom sg ahci libata sd_mod scsi_mod crc_t10dif xfs exportfs uhci_hcd ohci_hcd ehci_hcd [last unloaded: scsi_wait_scan] Pid: 1317, comm: modprobe Not tainted 2.6.33-rc3 #1 Call Trace: [<ffffffffa027bb25>] ? snd_oss_kernel_minor+0xdf/0x120 [snd] [<ffffffff8104338c>] warn_slowpath_common+0x7c/0xa9 [<ffffffff81043410>] warn_slowpath_fmt+0x41/0x43 [<ffffffffa027bb25>] snd_oss_kernel_minor+0xdf/0x120 [snd] [<ffffffffa027bc7e>] snd_register_oss_device+0x2c/0x21a [snd] [<ffffffffa0271e1e>] snd_mixer_oss_notify_handler+0xa5/0x2e5 [snd_mixer_oss] [<ffffffff81375e00>] ? __mutex_unlock_slowpath+0x5c/0x141 [<ffffffff810710d5>] ? trace_hardirqs_on_caller+0x11f/0x14a [<ffffffff8107110d>] ? trace_hardirqs_on+0xd/0xf sdhci-pci 0000:15:00.2: SDHCI controller found [1180:0822] (rev 21) sdhci-pci 0000:15:00.2: PCI INT C -> GSI 18 (level, low) -> IRQ 18 [<ffffffff81375ed3>] ? __mutex_unlock_slowpath+0x12f/0x141 sdhci-pci 0000:15:00.2: Will use DMA mode even though HW doesn't fully claim to support it. Registered led device: mmc0:: mmc0: SDHCI controller on PCI [0000:15:00.2] using DMA [<ffffffffa027628a>] snd_card_register+0x135/0x14b [snd] [<ffffffffa0370c88>] volume_init+0x335/0x3d2 [thinkpad_acpi] [<ffffffffa037289b>] thinkpad_acpi_module_init+0x61f/0xa4f [thinkpad_acpi] [<ffffffffa037227c>] ? thinkpad_acpi_module_init+0x0/0xa4f [thinkpad_acpi] [<ffffffff810001fa>] do_one_initcall+0x5f/0x154 [<ffffffff8107ff34>] sys_init_module+0xd7/0x234 [<ffffffff81002d5b>] system_call_fastpath+0x16/0x1b ---[ end trace a6d53dc318f09cb5 ]---

3 8

[alsa-devel] issue with sound stopping and not being able to restart
by Andrew Kohlsmith (Mailing List Account) 18 Jan '10

18 Jan '10

Good morning, I am new to ALSA development, but experienced in C. I wrote up a little test application which simply opens one playback and one capture device and spits noise into the playback and recording it on the capture device. I cross- correlate the data streams to determine the "depth" or latency of the pipe, end-to-end. I'm using playback and capture callbacks to handle the audio. The program seems to work reasonably well, but at seemingly random times either the playback or capture stream will "hiccup" (xrun) -- I am handling this case, but it's fairly often that the stream does NOT resume and I'm stuck with either no playback or no capture. I've looked at the example programs and the documentation and I do not see what it is I'm doing wrong. I'm posting my code here, hopefully someone can show me the error of my way. Capture/playback setup (error checking removed): snd_pcm_open(handle, fname, stream, SND_PCM_NONBLOCK); snd_pcm_hw_params_malloc(hwptr); snd_pcm_hw_params_any(h, hwp); snd_pcm_hw_params_set_access(h, hwp, SND_PCM_ACCESS_RW_INTERLEAVED); snd_pcm_hw_params_set_format(h, hwp, SND_PCM_FORMAT_S16_LE); snd_pcm_hw_params_set_rate(h, hwp, 8000, 0); snd_pcm_hw_params_set_channels(h, hwp, 2); snd_pcm_hw_params_set_buffer_size(h, hwp, BUF_SIZE); snd_pcm_hw_params(h, hwp); At this point I have the capture or playback device set up for 16 bit signed linear data in interleaved format and two channels. I've set up the buffers to be 160 frames, or 20ms. Each of these calls is checked for errors, and I do verify that the rate, buffer size and buffer time are what I expect. There is nothing unusual going on with these calls that I can determine. I set up the playback and capture callbacks thusly (again checking for errors) snd_async_add_pcm_handler(&ahp, hp, playback_callback, &pb_data); snd_async_add_pcm_handler(&ahc, hc, capture_callback, &cap_data); At this point I believe I'm ready to go, all I have to do is start playback and capture: if((fcount = snd_pcm_writei(h[PLAYBACK], pb_data.buf, BUF_SIZE)) != BUF_SIZE) { if(fcount < 0) { if((e = snd_pcm_recover(h[PLAYBACK], fcount, 0)) < 0) fprintf(stderr, "initial: can't writei: %s\n", snd_strerror(e)); } else { fprintf(stderr, "initial: short write: %d instead of %d\n", fcount, BUF_SIZE); } } if((e = snd_pcm_start(h[CAPTURE])) < 0) { fprintf(stderr, "capture prepare error: %s\n", snd_strerror(e)); return 1; } Now the playback and capture handlers are called by ALSA as required. The capture handler looks like this: /* capture callback. Called by ALSA whenever a period time is up */ void capture_callback(snd_async_handler_t *ah) { snd_pcm_t *h = snd_async_handler_get_pcm(ah); struct async_data *d = snd_async_handler_get_callback_private(ah); snd_pcm_sframes_t fa; short *p; int i, fc, e, faloops; struct timeval tv; static int c=0; /* read in whatever is available */ fa = snd_pcm_avail_update(h); faloops = 0; while(fa > 0) { if(fa > BUF_SIZE) fa = BUF_SIZE; if((fc = snd_pcm_readi(h, d->buf, fa)) != fa) { if(fc < 0) { if((e = snd_pcm_recover(h, e, 0)) < 0) { fprintf(stderr, "callback: can't readi: %s\n", snd_strerror(e)); } fc = 0; } else { fprintf(stderr, "callback: short read: %d instead of %ld\n", fc, fa); } } for(i=0,p=&d->buf[0]; i<fc; i++) { *d->ptr++ = *p++; p++; if(d->ptr > &in[MAX_BUFS * (BUF_SIZE - 1)]) { d->ptr = (short *)&in[0]; } } c += fc; if(c >= BUF_SIZE) { gettimeofday(&tv, NULL); c -= BUF_SIZE; in_step++; if(in_step >= MAX_BUFS) in_step = 0; } frames_cap += fc; fa = snd_pcm_avail_update(h); faloops++; }; cap_faloops=faloops; } The playback callback is very similar: /* playback callback. Called by ALSA whenever a period time is up */ void playback_callback(snd_async_handler_t *ah) { snd_pcm_t *h = snd_async_handler_get_pcm(ah); struct async_data *d = snd_async_handler_get_callback_private(ah); snd_pcm_sframes_t fa; int i, fc, e, faloops; static int c=0; struct timeval tv; fa = snd_pcm_avail_update(h); faloops = 0; while(fa > 0) { if(fa > BUF_SIZE) fa = BUF_SIZE; for(i=0; i<fa; i++) { d->buf[i*2] = awgn(&noise_source); d->buf[i*2+1] = awgn(&noise_source); } if((fc = snd_pcm_writei(h, d->buf, fa)) != fa) { if(fc < 0) { if((e = snd_pcm_recover(h, e, 0)) < 0) { fprintf(stderr, "callback: can't writei: error %x %s\n", e, snd_strerror(e)); } fc = 0; } else { fprintf(stderr, "callback: short write: %d instead of %ld\n", fc, fa); } } for(i=0; i<fc; i++) { *d->ptr++ = d->buf[i*2]; if(d->ptr > &out[MAX_BUFS * (BUF_SIZE - 1)]) { d->ptr = (short *)&out[0]; } } c += fc; if(c >= BUF_SIZE) { gettimeofday(&tv, NULL); c -= BUF_SIZE; out_step++; if(out_step >= MAX_BUFS) out_step = 0; } frames_pb += fc; fa = snd_pcm_avail_update(h); faloops++; }; pb_faloops = faloops; } That's really it; the main loop keeps track of the in_step and out_step variables and does the cross-correlation as the data comes in. Now it seems that it's the capture playback that is getting xruns, but I when capture xruns occur my playback callback seems to stop, and sometimes the entire program just hangs as if it is waiting for ALSA to call one of the callbacks. Can anyone shed any light on this? I'm certain it's something very simple and braindead, but I can't find it. Regards, Andrew

1 0

Re: [alsa-devel] ALSA DMIX support without MMAP ?
by ANISH KUMAR 18 Jan '10

18 Jan '10

> Actually i am working on soc platform(android-S3C6410) and wanted to > use DMIX and some good resampler for my software mixing instead of > current software mixing and resampling in upper layer(Above > alsa-lib).DMIX is working fine but this resampling is causing some > jitter.During rate conversion it looks for > libasound_module_rate_speexrate.so which i don't have so its using > default one causing the jitter. >>Have you considered using PulseAudio for this? Mark thanks for the suggestion and i checked out PulseAudio for my requirements.I found out that i don't need all the things in PulseAudio a. I just need good resampling code.Can i just use just the resampler code of pulseaudio(mine is mobile enviroment)? b. Just want a good mixer at ALSA-LIB level.As of now this is provided by android framework but i want this to be done by ALSA-LIB so that i can replace with some good mixer later on(if i find that some other mixer is good).DMIX is providing that but it's not that good. c.Jaroslav suggested to use sox resamplers but for that to work i have to get the sox resampler code(proper license) and need to build it for my system and link it with corresponding functions of ALSA-LIB so that they can call relevant functions for resampling.As i am working on mobile domain i think i need to do the integration of sox resampler code with ALSA-LIB for that to work?? or Can i just use the integrated sox library(some .so i guess) and change the asound.conf for using that library??Will this work?? d. I saw some check-ins related to Scenario Manager.As of now scenario manager functionality is also given in Android(we open a differnet device and the corresponding hooks for changing the routes dynamically) but i think if it given in alsa-lib switching will be much faster and this will increase the system performance.

2 1

[alsa-devel] [PATCH] add atomic operations for Blackfin parts
by Mike Frysinger 18 Jan '10

18 Jan '10

Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- include/iatomic.h | 23 +++++++++++++++++++++++ 1 files changed, 23 insertions(+), 0 deletions(-) diff --git a/include/iatomic.h b/include/iatomic.h index 8f6ec22..e92dbfd 100644 --- a/include/iatomic.h +++ b/include/iatomic.h @@ -1079,6 +1079,29 @@ static __inline__ int atomic_sub_return(int i, volatile atomic_t *v) #endif /* __sh__ */ +#ifdef __bfin__ + +#include <bfin_fixed_code.h> + +typedef struct { volatile int counter; } atomic_t; + +#define ATOMIC_INIT(i) { (i) } + +#define atomic_read(v) ((v)->counter) +#define atomic_set(v,i) (((v)->counter) = (i)) +#define atomic_add(i,v) bfin_atomic_add32(&(v)->counter, i) +#define atomic_sub(i,v) bfin_atomic_sub32(&(v)->counter, i) +#define atomic_inc(v) bfin_atomic_inc32(&(v)->counter); +#define atomic_dec(v) bfin_atomic_dec32(&(v)->counter); + +#define mb() __asm__ __volatile__ ("" : : : "memory") +#define rmb() mb() +#define wmb() mb() + +#define IATOMIC_DEFINED 1 + +#endif /* __bfin__ */ + #ifndef IATOMIC_DEFINED /* * non supported architecture. -- 1.6.6

3 4

[alsa-devel] ASoC updates for 2.6.34
by Mark Brown 18 Jan '10

18 Jan '10

The following changes since commit 735fe4cfbc3cedea41bd0ed31955054dae6beb46: Mark Brown (1): ASoC: Add missing __devexit and __devinit annotations are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.34 Daniel Mack (3): ASoC: ak4104: allow more sample rates ASoC: cs4270: allow passing freq=0 in set_dai_sysclk() ASoC: support more sample rates on raumfeld devices Mark Brown (6): ASoC: Fix i.MX audio build for i.MX3x ASoC: Convert new i.MX SSI driver to use static DAI array ASoC: i.MX SSI driver does not yet support master mode ASoC: Remove old i.MX driver code ASoC: Mark new i.MX drivers as BROKEN until arch/arm merged Merge branch 'mxc-audio' into for-2.6.34 Peter Ujfalusi (1): ASoC: TWL4030: Replace comma with semicolon in probe function Sascha Hauer (1): ASoC: Add a new imx-ssi sound driver Thomas Weber (1): ASoC: TWL4030: Fix typo in comment in header file arch/arm/plat-mxc/Makefile | 4 + arch/arm/plat-mxc/ssi-fiq-ksym.c | 20 + arch/arm/plat-mxc/ssi-fiq.S | 134 ++++++ sound/soc/codecs/ak4104.c | 4 +- sound/soc/codecs/cs4270.c | 38 ++- sound/soc/codecs/twl4030.c | 2 +- sound/soc/codecs/twl4030.h | 2 +- sound/soc/imx/Kconfig | 20 +- sound/soc/imx/Makefile | 12 +- sound/soc/imx/imx-pcm-dma-mx2.c | 313 ++++++++++++++ sound/soc/imx/imx-pcm-fiq.c | 277 ++++++++++++ sound/soc/imx/imx-ssi.c | 758 +++++++++++++++++++++++++++++++++ sound/soc/imx/imx-ssi.h | 237 +++++++++++ sound/soc/imx/mx1_mx2-pcm.c | 488 --------------------- sound/soc/imx/mx1_mx2-pcm.h | 26 -- sound/soc/imx/mx27vis_wm8974.c | 318 -------------- sound/soc/imx/mxc-ssi.c | 860 -------------------------------------- sound/soc/imx/mxc-ssi.h | 238 ----------- sound/soc/pxa/raumfeld.c | 61 ++- 19 files changed, 1823 insertions(+), 1989 deletions(-) create mode 100644 arch/arm/plat-mxc/ssi-fiq-ksym.c create mode 100644 arch/arm/plat-mxc/ssi-fiq.S create mode 100644 sound/soc/imx/imx-pcm-dma-mx2.c create mode 100644 sound/soc/imx/imx-pcm-fiq.c create mode 100644 sound/soc/imx/imx-ssi.c create mode 100644 sound/soc/imx/imx-ssi.h delete mode 100644 sound/soc/imx/mx1_mx2-pcm.c delete mode 100644 sound/soc/imx/mx1_mx2-pcm.h delete mode 100644 sound/soc/imx/mx27vis_wm8974.c delete mode 100644 sound/soc/imx/mxc-ssi.c delete mode 100644 sound/soc/imx/mxc-ssi.h

2 1

[alsa-devel] MSM DSP
by GNUtoo 17 Jan '10

17 Jan '10

hi, I've bought an HTC dream,before buying I looked at the feasibility of porting GNU/Linux on it(so not Android). There is a patch(for maing the driver standard) or a standard(everything is non-standard in android) driver for most of the components. The biggest problem seems the sound. So knowing that codeaurora( https://www.codeaurora.org/gitweb/quic/le/?p=kernel/msm.git ) had alsa in its kenrel I first tried to compile that kernel...but due to numerous bugs(It didn't compile and needed too much fixing) in it android-msm-2.6.29 branch I decided to import alsa in the linuxtogo kernel,and I pushed the sources at a temporary htc-alsa branch at git://gitorious.org/replicant/gnulinuxkernel.git (it's called gnulinux in order to differentiate from an android kernel,I didn't find a good name for it) from the commit message available here : http://gitorious.org/replicant/gnulinuxkernel/commit/92945ccd0921e940fc2675… I had to change snd_rpc_ids.vers = 0x00020001 into snd_rpc_ids.vers = 0xaa2b1a44 ,and to make minor build system changes and I had aplay working rapidely... but only aplay... mplayer was waiting indefinitely for the following system call: ioctl(6, 0x400c4150 ls -l /proc/$(pido mplayer)/fd/6 gave /dev/snd/pcmC0D0p and 0x400c4150 gave that: SNDRV_PCM_IOCTL_WRITEI_FRAMES = _IOW('A', 0x50, struct snd_xferi) Then with printks(I had no serial cable(I've looked for a craddle/docking station with the special HTC connector for easy soldering...but I didn't found one) so I couldn't use kgdb) I went until that point: rc = wait_event_interruptible(the_locks.write_wait, (frame->used == 0)|| (prtd->stopped)); That's in the alsa_send_buffer function in /sound/soc/msm/msm-pcm.c Then I continued the tracing puting printks everywhere...and I found that: with aplay: msm_pcm_playback_copy in the /sound/soc/msm/msm7k-pcm.c was called rc = alsa_send_buffer(prtd, buf, fbytes, NULL); was called once inside alsa_send_buffer (in /sound/soc/msm/msm-pcm.c) the count loop was only executed once because count was exhausted on the first run then copy_count is incremented and the audio is configured then at second call of the alsa_send_buffer the audio is configured and works. here's an output of the aplay parameters: $ aplay -v 05\ -\ Jet\ Set.wav Playing WAVE '05 - Jet Set.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Plug PCM: Hardware PCM card 0 'msm-audio' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 4800 period_size : 1200 period_time : 27210 tick_time : 0 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 1200 xfer_align : 1200 start_threshold : 4800 stop_threshold : 4800 silence_threshold: 0 silence_size : 0 boundary : 1258291200 note that frame = prtd->out + prtd->out_tail; makes frame->used have a bad value the first time it is run but as the count is only run once it doesn't import. the next runs prtd->out + prtd->out_tail have a good value with mplayer the story was quite different,even if the beginning was the same : msm_pcm_playback_copy in the /sound/soc/msm/msm7k-pcm.c was also called,rc = alsa_send_buffer(prtd, buf, fbytes, NULL); was also called inside alsa_send_buffer (in /sound/soc/msm/msm-pcm.c) but once in alsa_send_buffer,more precisely under the count loop the story changes: count is not exhausted the first time,because in "xfer = count > frame->size ? frame->size : count;" the smallest between count and frame->size is assigned,and so after it does "count -= xfer" and so count is not 0,because the buffer size detected by mplayer is 19200 insterad of 4800 so here what it does in detail: it passes 19200 as count here: fbytes = frames_to_bytes(runtime, frames); rc = alsa_send_buffer(prtd, buf, fbytes, NULL); that goes into that: ssize_t alsa_send_buffer(struct msm_audio *prtd, const char __user *buf, size_t count, loff_t *pos) and then the count variable is 19200 in alsa_send_buffer then there is xfer = count > frame->size ? frame->size : count; so it assigns the minimum between count and frame->size to xfer count is still 19200 frame->size is 4800 so xfer becomes 4800 then count -= xfer decrement count but only of 4800 so count is not exhausted and runs another time then : frame = prtd->out + prtd->out_tail; makes frame->used have a bad value,and then it blocks here: rc = wait_event_interruptible(the_locks.write_wait,(frame->used == 0) || (prtd->stopped)); because frame->used is not 0. here an mplayer output: Forced audio codec: mad Opening audio decoder: [pcm] Uncompressed PCM audio decoder dec_audio: Allocating 2048 + 65536 = 67584 bytes for output buffer. AUDIO: 44100 Hz, 2 ch, s16le, 1411.2 kbit/100.00% (ratio: 176400->176400) Selected audio codec: [pcm] afm: pcm (Uncompressed PCM) ========================================================================== Building audio filter chain for 44100Hz/2ch/s16le -> 0Hz/0ch/??... [libaf] Adding filter dummy [dummy] Was reinitialized: 44100Hz/2ch/s16le [dummy] Was reinitialized: 44100Hz/2ch/s16le Trying preferred audio driver 'alsa', options '[none]' alsa-init: requested format: 44100 Hz, 2 channels, 9 alsa-init: using ALSA 1.0.15 alsa-init: setup for 1/2 channel(s) alsa-init: using device default alsa-init: pcm opened in blocking mode alsa-init: got buffersize=19200 alsa-init: got period size 300 alsa: 44100 Hz/2 channels/4 bpf/19200 bytes buffer/Signed 16 bit Little Endian AO: [alsa] 44100Hz 2ch s16le (2 bytes per sample) AO: Description: ALSA-0.9.x-1.x audio output AO: Author: Alex Beregszaszi, Zsolt Barat <joy(a)streamminister.de> AO: Comment: under developement Building audio filter chain for 44100Hz/2ch/s16le -> 44100Hz/2ch/s16le... [dummy] Was reinitialized: 44100Hz/2ch/s16le [dummy] Was reinitialized: 44100Hz/2ch/s16le Video: no video Freeing 0 unused video chunks. Starting playback... Increasing filtered audio buffer size from 0 to 21248 so the quick solution was to force mplayer to use 4800 as buffer size with something like that: diff --git a/sound/soc/msm/msm7k-pcm.c b/sound/soc/msm/msm7k-pcm.c index 38e8283..bfbc66d 100644 --- a/sound/soc/msm/msm7k-pcm.c +++ b/sound/soc/msm/msm7k-pcm.c @@ -115,7 +115,7 @@ static struct snd_pcm_hardware msm_pcm_playback_hardware = { .rate_max = USE_RATE_MAX, .channels_min = USE_CHANNELS_MIN, .channels_max = USE_CHANNELS_MAX, - .buffer_bytes_max = MAX_BUFFER_PLAYBACK_SIZE, + .buffer_bytes_max = 4800, .period_bytes_min = 64, .period_bytes_max = MAX_PERIOD_SIZE, .periods_min = USE_PERIODS_MIN, but then mplayer seem to loop on an ioctl: mplayer is frozen here and doesn't respond to other input than ctrl+c: 0.0 (00.0) of 206.0 (03:26.0) ??,?% strace shows a lot of : nanosleep followed by SNDRV_PCM_IOCTL_STATUS (ioctl(6, 0x806c4120, 0xbe8bc9c8) = 0) an analyse of the status shows a constant : status->delay is 1200,and status->avail is 0,they don't vary over time and are printed just after that code in snd_pcm_status in sound/core/pcm_native.c if (runtime->status->state == SNDRV_PCM_STATE_RUNNING || runtime->status->state == SNDRV_PCM_STATE_DRAINING) status->delay = runtime->buffer_size - status->avail; else status->delay = 0; I was told in #mplayerdev that it was because the DSP didn't play the frames...indeed I hear no sound.... So how should I fix that mess? *did the quick and dirty fix cause the second problem? *should I use the DSP directly and make a pulseaudio plugin that talks to the android driver, instead(the driver seems very buggy) The good would be support for mp3 and similar things within the DSP,and that it should be more stable. The bad is that I already spent *a lot* of time with the alsa kernel driver...and that writing that will also require some userspaces changes such as: *modification of the phone routing in the "FSO" ( http://www.freesmartphone.org/index.php/Main_Page ) dbus framework to use pulseaudio *would require changes in the build system for using pulseaudio(openembedded(http://wiki.openembedded.net/index.php/Main_Page )) *may be slow...openmoko stopped to use pulseaudio because of speed issues. Denis

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[alsa-devel] Audio buffer mmap-ing on sh arch vs. cache aliasing problem
by Pawel MOLL 16 Jan '10

16 Jan '10

Hi Guys, Recently I was investigating an issue with capturing audio using USB Audio Class device on a sh4-based board. "Bad voice quality" was reported... Finally I have traced the problem to something which is (unfortunately) well known to sh developers as a D-cache aliasing (or synonym) problem. Briefly speaking: due to some MMU design decisions, one can have two different virtual address pointing to the same physical location, which is fine, but going via different cache slots! So if there was a value of "0" in the memory and user "A" will write "1" there, user "B" will still read "0"... The solution is to ensure all TLB entries (so virtual memory areas) are beginning from a 16kB-aligned virtual address. Otherwise it is necessary to flush the cache between accesses from "A" and "B" sides. And now. The USB Audio Class driver (sound/usb/usbaudio.c) is allocating the sound buffer like this... static int snd_pcm_alloc_vmalloc_buffer(struct snd_pcm_substream *subs, size_t size) { [...] runtime->dma_area = vmalloc(size); [...] } ... and vmalloc will return a page(4k)-aligned pointer, possibly not 16k-aligned one. This is the source of all evil ;-) When using RW transfers everything is fine, as data is memcopied between two independent buffers. In a case of MMAP mode we will end up with such a situation: 1. The driver will memcpy data from an URB to the buffer. It will populate several cache lines. Let's call it a "kernel cache". 2. As the library has mapped the non-16k-aligned address via different cache line ("user cache"), it will read some rubbish from the physical memory, populating the "user cache" by the way. 3. Some time later the "kernel cache" is flushed, writing the data to the memory. 4. Some new data from URB is entering "kernel cache". 5. The library will access mmap-ed area again, going via the "user cache", which may or may not reflect the correct data (depending on the fact was the "user cache" flushed or not) etc. Of course this cycle is completely not deterministic, so some of the "kernel cache" lines will be flushed before being accessed from user space, other not... The final effect is... hmm... bizarre :-) First, of course, you will get a (probably loud) glitch (rubbish from buffer's underlying memory, before the first valid data is written back), and then something that could be described as an "echo" ;-) I mean - you are capturing "1 2 3 4 5 6 7..." and the result is (G stands for Glitch ;-) "G 1 23 3 4 4 6 67..." As a quick-and-dirty work-around I have modified snd_pcm_alloc_vmalloc_buffer() to allocate always 12kB more and then use 16k-aligned pointer: static int snd_pcm_alloc_vmalloc_buffer(struct snd_pcm_substream *subs, size_t size) { struct snd_pcm_runtime *runtime = subs->runtime; if (runtime->dma_addr) { if (runtime->dma_bytes >= size) return 0; /* already large enough */ vfree((void *)runtime->dma_addr); } runtime->dma_addr = (unsigned long)vmalloc(size + 12 * 1024); if (!runtime->dma_addr) return -ENOMEM; runtime->dma_area = (void *)ALIGN(runtime->dma_addr, 16 * 1024); runtime->dma_bytes = size; return 0; } Of course it cannot be regarded as anything more as a hack. And definitely not as a solution... So my question is: Any idea how the "proper solution" should look like? I see no obvious method to get an arch-independent "mmap-compliant" buffer. This problem was found on a sh arch, using an USB Audio Class device, however out there some other architectures are suffering from this MMU "feature" as well (ie. see http://docs.hp.com/en/B3906-90006/ch07s09.html) and possibly other drivers could behave "wrongly" (quoted as the driver is actually innocent...) To be honest I just have absolutely no idea what to do with this all! :-O I hope I was clear enough in the description... Any feedback, advise, idea etc. will be more than appreciated. Cheers Paweł

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[alsa-devel] [PATCH] - fix speaker output for toshiba nb205 netbooks
by Nickolas 16 Jan '10

16 Jan '10

this is a quick fix i wrote up a couple days ago to fix speaker output on the Realtek ALC272 codec present in toshiba nb205 netbooks. presently, speaker output doesn't work at all on this model, regardless of settings in the mixer. this is because the current driver thinks the speaker is at nid 0x0c instead of the correct 0x17. i'm not too familiar with this hardware, so i'm not sure that this patch won't have any negative effects for other hardware. it also seems like a messy way of fixing the problem. any comments or suggestions are welcome, as is testing for those who have this board. diffed against the latest git. Thanks, Nick -- Signed-off-by: Nickolas Lloyd <ultrageek.lloyd(a)gmail.com> --- alsa-driver/sound/pci/hda/patch_realtek.c 2010-01-12 08:03:28.000000000 -0500 +++ alsa-driver-b/sound/pci/hda/patch_realtek.c 2010-01-12 16:27:48.000000000 -0500 @@ -192,6 +192,7 @@ enum { ALC663_ASUS_MODE8, ALC272_DELL, ALC272_DELL_ZM1, + ALC272_TOSHIBA_N200, ALC272_SAMSUNG_NC10, ALC662_AUTO, ALC662_MODEL_LAST, @@ -16725,6 +16726,11 @@ static struct hda_verb alc272_dell_init_ {} }; +static struct hda_verb alc272_toshiba_init_verbs[] = { + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {} +}; + static struct hda_verb alc663_mode7_init_verbs[] = { {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, @@ -17255,6 +17261,36 @@ static void alc663_g50v_inithook(struct alc_mic_automute(codec); } +static void alc272_toshiba_speaker_automute(struct hda_codec *codec) +{ + unsigned int present; + unsigned char bits; + + present = snd_hda_jack_detect(codec, 0x21); + bits = present ? HDA_AMP_MUTE : 0; + snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); +} + +static void alc272_toshiba_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case ALC880_HP_EVENT: + alc272_toshiba_speaker_automute(codec); + break; + case ALC880_MIC_EVENT: + alc_mic_automute(codec); + break; + } +} + +static void alc272_toshiba_init_hook(struct hda_codec *codec) +{ + alc272_toshiba_speaker_automute(codec); + alc_mic_automute(codec); +} + static struct snd_kcontrol_new alc662_ecs_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x02, 0x0, HDA_OUTPUT), ALC262_HIPPO_MASTER_SWITCH, @@ -17391,7 +17427,7 @@ static struct snd_pci_quirk alc662_cfg_t SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS), SND_PCI_QUIRK(0x105b, 0x0d47, "Foxconn 45CMX/45GMX/45CMX-K", ALC662_3ST_6ch_DIG), - SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB200", ALC663_ASUS_MODE4), + SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB200", ALC272_TOSHIBA_N200), SND_PCI_QUIRK(0x144d, 0xca00, "Samsung NC10", ALC272_SAMSUNG_NC10), SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L", ALC662_3ST_6ch_DIG), @@ -17696,6 +17732,22 @@ static struct alc_config_preset alc662_p .setup = alc663_m51va_setup, .init_hook = alc663_m51va_inithook, }, + [ALC272_TOSHIBA_N200] = { + .mixers = { alc663_asus_21jd_clfe_mixer }, + .cap_mixer = alc662_auto_capture_mixer, + .init_verbs = { alc662_init_verbs, + alc663_21jd_amic_init_verbs, + alc272_toshiba_init_verbs }, + .num_dacs = ARRAY_SIZE(alc662_dac_nids), + .hp_nid = 0x03, + .dac_nids = alc662_dac_nids, + .dig_out_nid = ALC662_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), + .channel_mode = alc662_3ST_2ch_modes, + .unsol_event = alc272_toshiba_unsol_event, + .setup = alc663_mode4_setup, + .init_hook = alc272_toshiba_init_hook, + }, [ALC272_SAMSUNG_NC10] = { .mixers = { alc272_nc10_mixer }, .init_verbs = { alc662_init_verbs,

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