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[alsa-devel] Audio support for TLV320AIC23 on Simtec S3C24XX products
by Ben Dooks 07 Aug '09

07 Aug '09

Add support for the TLV320AIC23 on most of the current Simtec S3C24XX with an initial core set of routines for use with the other S3C24XX designs that do not use the TLV320AIC23. Signed-off-by: Ben Dooks <ben(a)simtec.co.uk> --- arch/arm/common/sbcd/s3c24xx/dev-audio.c | 146 ++++++--- arch/arm/plat-s3c/include/plat/audio-simtec.h | 37 ++ arch/arm/plat-s3c24xx/audio-simtec.c | 97 +----- sound/soc/s3c24xx/Kconfig | 12 sound/soc/s3c24xx/Makefile | 4 sound/soc/s3c24xx/s3c24xx_simtec.c | 391 +++++++++++++++++++++++++ sound/soc/s3c24xx/s3c24xx_simtec.h | 22 + sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c | 137 ++++++++ 8 files changed, 728 insertions(+), 118 deletions(-) Index: b/sound/soc/s3c24xx/Kconfig =================================================================== --- a/sound/soc/s3c24xx/Kconfig 2009-08-04 12:24:09.000000000 +0100 +++ b/sound/soc/s3c24xx/Kconfig 2009-08-06 11:53:38.000000000 +0100 @@ -115,3 +115,15 @@ config SND_S3C24XX_SOC_S3C24XX_UDA134X select SND_S3C24XX_SOC_I2S select SND_SOC_L3 select SND_SOC_UDA134X + +config SND_S3C24XX_SOC_SIMTEC + tristate + help + Internal node for common S3C24XX/Simtec suppor + +config SND_S3C24XX_SOC_SIMTEC_TLV320AIC23 + tristate "SoC I2S Audio support for TLV320AIC23 on Simtec boards" + depends on SND_S3C24XX_SOC + select SND_S3C24XX_SOC_I2S + select SND_SOC_TLV320AIC23 + select SND_S3C24XX_SOC_SIMTEC Index: b/sound/soc/s3c24xx/Makefile =================================================================== --- a/sound/soc/s3c24xx/Makefile 2009-08-04 12:24:09.000000000 +0100 +++ b/sound/soc/s3c24xx/Makefile 2009-08-06 11:53:38.000000000 +0100 @@ -23,6 +23,8 @@ snd-soc-smdk2440-wm8991-objs := smdk2440 snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o snd-soc-neo1973-gta02-wm8753-objs := neo1973_gta02_wm8753.o snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o +snd-soc-s3c24xx-simtec-objs := s3c24xx_simtec.o +snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -33,3 +35,5 @@ obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2443_WM obj-$(CONFIG_SND_S3C24XX_SOC_LN2440SBC_ALC650) += snd-soc-ln2440sbc-alc650.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_GTA02_WM8753) += snd-soc-neo1973-gta02-wm8753.o obj-$(CONFIG_SND_S3C24XX_SOC_S3C24XX_UDA134X) += snd-soc-s3c24xx-uda134x.o +obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC) += snd-soc-s3c24xx-simtec.o +obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o Index: b/sound/soc/s3c24xx/s3c24xx_simtec.c =================================================================== --- /dev/null 1970-01-01 00:00:00.000000000 +0000 +++ b/sound/soc/s3c24xx/s3c24xx_simtec.c 2009-08-04 12:29:02.000000000 +0100 @@ -0,0 +1,391 @@ +/* sound/soc/s3c24xx/s3c24xx_simtec.c + * + * Copyright 2009 Simtec Electronics + * + * 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. +*/ + +#define DEBUG + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/platform_device.h> +#include <linux/gpio.h> +#include <linux/clk.h> +#include <linux/i2c.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <plat/audio-simtec.h> + +#include "s3c24xx-pcm.h" +#include "s3c24xx-i2s.h" +#include "s3c24xx_simtec.h" + +static struct s3c24xx_audio_simtec_pdata *pdata; +static struct clk *xtal_clk; + +static int spk_gain; +static int spk_unmute; + +/** + * speaker_gain_get - read the speaker gain setting. + * @kcontrol: The control for the speaker gain. + * @ucontrol: The value that needs to be updated. + * + * Read the value for the AMP gain control. + */ +static int speaker_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = spk_gain; + return 0; +} + +/** + * speaker_gain_set - set the value of the speaker amp gain + * @value: The value to write. + */ +static void speaker_gain_set(int value) +{ + gpio_set_value_cansleep(pdata->amp_gain[0], value & 1); + gpio_set_value_cansleep(pdata->amp_gain[1], value >> 1); +} + +/** + * speaker_gain_put - set the speaker gain setting. + * @kcontrol: The control for the speaker gain. + * @ucontrol: The value that needs to be set. + * + * Set the value of the speaker gain from the specified + * @ucontrol setting. + * + * Note, if the speaker amp is muted, then we do not set a gain value + * as at-least one of the ICs that is fitted will try and power up even + * if the main control is set to off. + */ +static int speaker_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int value = ucontrol->value.integer.value[0]; + + spk_gain = value; + + if (!spk_unmute) + speaker_gain_set(value); + + return 0; +} + +static const struct snd_kcontrol_new amp_gain_controls[] = { + SOC_SINGLE_EXT("Speaker Gain", 0, 0, 3, 0, + speaker_gain_get, speaker_gain_put), +}; + +/** + * spk_unmute_state - set the unmute state of the speaker + * @to: zero to unmute, non-zero to ununmute. + */ +static void spk_unmute_state(int to) +{ + pr_debug("%s: to=%d\n", __func__, to); + + spk_unmute = to; + gpio_set_value(pdata->amp_gpio, to); + + /* if we're umuting, also re-set the gain */ + if (to && pdata->amp_gain[0] > 0) + speaker_gain_set(spk_gain); +} + +/** + * speaker_unmute_get - read the speaker unmute setting. + * @kcontrol: The control for the speaker gain. + * @ucontrol: The value that needs to be updated. + * + * Read the value for the AMP gain control. + */ +static int speaker_unmute_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = spk_unmute; + return 0; +} + +/** + * speaker_unmute_put - set the speaker unmute setting. + * @kcontrol: The control for the speaker gain. + * @ucontrol: The value that needs to be set. + * + * Set the value of the speaker gain from the specified + * @ucontrol setting. + */ +static int speaker_unmute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + spk_unmute_state(ucontrol->value.integer.value[0]); + return 0; +} + +static const struct snd_kcontrol_new amp_unmute_controls[] = { + SOC_SINGLE_EXT("Speaker", 0, 0, 1, 0, + speaker_unmute_get, speaker_unmute_put), +}; + +void simtec_audio_init(struct snd_soc_codec *codec) +{ + if (pdata->amp_gpio > 0) { + pr_debug("%s: adding amp routes\n", __func__); + + snd_soc_add_controls(codec, amp_unmute_controls, + ARRAY_SIZE(amp_unmute_controls)); + } + + if (pdata->amp_gain[0] > 0) { + pr_debug("%s: adding amp controls\n", __func__); + snd_soc_add_controls(codec, amp_gain_controls, + ARRAY_SIZE(amp_gain_controls)); + } +} +EXPORT_SYMBOL_GPL(simtec_audio_init); + +#define CODEC_CLOCK 12000000 + +/** + * simtec_hw_params - update hardware parameters + * @substream: The audio substream instance. + * @params: The parameters requested. + * + * Update the codec data routing and configuration settings + * from the supplied data. + */ +static int simtec_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + /* Set the CODEC as the bus clock master, I2S */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret) { + pr_err("%s: failed set cpu dai format\n", __func__); + return ret; + } + + /* Set the CODEC as the bus clock master */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret) { + pr_err("%s: failed set codec dai format\n", __func__); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, + CODEC_CLOCK, SND_SOC_CLOCK_IN); + if (ret) { + pr_err( "%s: failed setting codec sysclk\n", __func__); + return ret; + } + + if (pdata->use_mpllin) { + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C24XX_CLKSRC_MPLL, + 0, SND_SOC_CLOCK_OUT); + + if (ret) { + pr_err("%s: failed to set MPLLin as clksrc\n", + __func__); + return ret; + } + } + + if (pdata->output_cdclk) { + int cdclk_scale; + + cdclk_scale = clk_get_rate(xtal_clk) / CODEC_CLOCK; + cdclk_scale--; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, + cdclk_scale); + } + + return 0; +} + +static int simtec_call_startup(struct s3c24xx_audio_simtec_pdata *pd) +{ + /* call any board supplied startup code, this currently only + * covers the bast/vr1000 which have a CPLD in the way of the + * LRCLK */ + if (pd->startup) + pd->startup(); + + return 0; +} + +static struct snd_soc_ops simtec_snd_ops = { + .hw_params = simtec_hw_params, +}; + +/** + * attach_gpio_amp - get and configure the necessary gpios + * @dev: The device we're probing. + * @pd: The platform data supplied by the board. + * + * If there is a GPIO based amplifier attached to the board, claim + * the necessary GPIO lines for it, and set default values. + */ +static int attach_gpio_amp(struct device *dev, + struct s3c24xx_audio_simtec_pdata *pd) +{ + int ret; + + /* attach gpio amp gain (if any) */ + if (pdata->amp_gain[0] > 0) { + ret = gpio_request(pd->amp_gain[0], "gpio-amp-gain0"); + if (ret) { + dev_err(dev, "cannot get amp gpio gain0\n"); + return ret; + } + + ret = gpio_request(pd->amp_gain[1], "gpio-amp-gain1"); + if (ret) { + dev_err(dev, "cannot get amp gpio gain1\n"); + gpio_free(pdata->amp_gain[0]); + return ret; + } + + gpio_direction_output(pd->amp_gain[0], 0); + gpio_direction_output(pd->amp_gain[1], 0); + } + + /* note, curently we assume GPA0 isn't valid amp */ + if (pdata->amp_gpio > 0) { + ret = gpio_request(pd->amp_gpio, "gpio-amp"); + if (ret) { + dev_err(dev, "cannot get amp gpio %d (%d)\n", + pd->amp_gpio, ret); + goto err_amp; + } + + /* set the amp off at startup */ + spk_unmute_state(0); + } + + return 0; + +err_amp: + if (pd->amp_gain[0] > 0) { + gpio_free(pd->amp_gain[0]); + gpio_free(pd->amp_gain[1]); + } + + return ret; +} + +static void detach_gpio_amp(struct s3c24xx_audio_simtec_pdata *pd) +{ + if (pd->amp_gain[0] > 0) { + gpio_free(pd->amp_gain[0]); + gpio_free(pd->amp_gain[1]); + } + + if (pd->amp_gpio > 0) + gpio_free(pd->amp_gpio); +} + +#ifdef CONFIG_PM +int simtec_audio_resume(struct device *dev) +{ + simtec_call_startup(pdata); + return 0; +} + +struct dev_pm_ops simtec_audio_pmops = { + .resume = simtec_audio_resume, +}; +EXPORT_SYMBOL_GPL(simtec_audio_pmops); +#endif + +int __devinit simtec_audio_core_probe(struct platform_device *pdev, + struct snd_soc_device *socdev) +{ + struct platform_device *snd_dev; + int ret; + + socdev->card->dai_link->ops = &simtec_snd_ops; + + pdata = pdev->dev.platform_data; + if (!pdata) { + dev_err(&pdev->dev, "no platform data supplied\n"); + return -EINVAL; + } + + simtec_call_startup(pdata); + + xtal_clk = clk_get(&pdev->dev, "xtal"); + if (IS_ERR(xtal_clk)) { + dev_err(&pdev->dev, "could not get clkout0\n"); + return -EINVAL; + } + + dev_info(&pdev->dev, "xtal rate is %ld\n", clk_get_rate(xtal_clk)); + + ret = attach_gpio_amp(&pdev->dev, pdata); + if (ret) + goto err_clk; + + snd_dev = platform_device_alloc("soc-audio", -1); + if (!snd_dev) { + dev_err(&pdev->dev, "failed to alloc soc-audio devicec\n"); + ret = -ENOMEM; + goto err_gpio; + } + + platform_set_drvdata(snd_dev, socdev); + socdev->dev = &snd_dev->dev; + + ret = platform_device_add(snd_dev); + if (ret) { + dev_err(&pdev->dev, "failed to add soc-audio dev\n"); + goto err_pdev; + } + + platform_set_drvdata(pdev, snd_dev); + return 0; + +err_pdev: + platform_device_put(snd_dev); + +err_gpio: + detach_gpio_amp(pdata); + +err_clk: + clk_put(xtal_clk); + return ret; +} + +int __devexit simtec_audio_remove(struct platform_device *pdev) +{ + struct platform_device *snd_dev = platform_get_drvdata(pdev); + + platform_device_unregister(snd_dev); + + detach_gpio_amp(pdata); + clk_put(xtal_clk); + return 0; +} +EXPORT_SYMBOL_GPL(simtec_audio_remove); + +MODULE_AUTHOR("Ben Dooks <ben(a)simtec.co.uk>"); +MODULE_DESCRIPTION("ALSA SoC Simtec Audio common support"); +MODULE_LICENSE("GPL"); Index: b/arch/arm/plat-s3c/include/plat/audio-simtec.h =================================================================== --- /dev/null 1970-01-01 00:00:00.000000000 +0000 +++ b/arch/arm/plat-s3c/include/plat/audio-simtec.h 2009-08-04 12:24:11.000000000 +0100 @@ -0,0 +1,37 @@ +/* arch/arm/plat-s3c/include/plat/audio-simtec.h + * + * Copyright 2008 Simtec Electronics + * http://armlinux.simtec.co.uk/ + * Ben Dooks <ben(a)simtec.co.uk> + * + * 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. + * + * Simtec Audio support. +*/ + +/** + * struct s3c24xx_audio_simtec_pdata - platform data for simtec audio + * @use_mpllin: Select codec clock from MPLLin + * @output_cdclk: Need to output CDCLK to the codec + * @have_mic: Set if we have a MIC socket + * @have_lout: Set if we have a LineOut socket + * @amp_gpio: GPIO pin to enable the AMP + * @amp_gain: Option GPIO to control AMP gain + */ +struct s3c24xx_audio_simtec_pdata { + unsigned int use_mpllin:1; + unsigned int output_cdclk:1; + + unsigned int have_mic:1; + unsigned int have_lout:1; + + int amp_gpio; + int amp_gain[2]; + + void (*startup)(void); +}; + +extern int simtec_audio_add(const char *codec_name, + struct s3c24xx_audio_simtec_pdata *pdata); Index: b/arch/arm/plat-s3c24xx/audio-simtec.c =================================================================== --- a/arch/arm/plat-s3c24xx/audio-simtec.c 2009-08-04 12:05:59.000000000 +0100 +++ b/arch/arm/plat-s3c24xx/audio-simtec.c 2009-08-04 12:24:11.000000000 +0100 @@ -45,7 +45,7 @@ #include <plat/audio.h> #include <plat/regs-iis.h> -#include <asm/plat-s3c24xx/audio-simtec.h> +#include <plat/audio-simtec.h> #include <plat/devs.h> #include <plat/pm.h> @@ -56,7 +56,7 @@ static inline int has_lr_routing(void) return machine_is_bast() || machine_is_vr1000(); } -static int simtec_audio_startup(void *pw) +static void simtec_audio_startup(void) { unsigned int tmp; unsigned long flags; @@ -71,94 +71,40 @@ static int simtec_audio_startup(void *pw } local_irq_restore(flags); - - return 0; -} - -static void simtec_audio_shutdown(void *pw) -{ - unsigned int tmp; - unsigned long flags; - - /* switch off LR clock routing */ - - local_irq_save(flags); - - if (has_lr_routing()) { - tmp = __raw_readb(BAST_VA_CTRL1); - tmp &= ~BAST_CPLD_CTRL1_LRMASK; - tmp |= BAST_CPLD_CTRL1_LRCOFF; - __raw_writeb(tmp, BAST_VA_CTRL1); - } - - local_irq_restore(flags); -} - -static int simtec_audio_matchdev(struct device *dev) -{ - printk(KERN_INFO "%s: dev = %p\n", __func__, dev); - - if (dev->parent == NULL || dev->parent->parent == NULL) - return 0; - - printk(KERN_INFO "%s: %s\n", __func__, dev_name(dev->parent)); - if (strncmp(dev_name(dev->parent), "i2c", strlen("i2c")) != 0) - return 0; - - printk(KERN_INFO "%s: %s\n", __func__, dev_name(dev->parent->parent)); - if (strncmp(dev_name(dev->parent->parent), "s3c24", - strlen("s3c24")) == 0) - return 1; - - return 0; } -/* device registration info */ - -static struct s3c24xx_iis_ops simtec_audio_ops = { - .startup = simtec_audio_startup, - .shutdown = simtec_audio_shutdown, +static struct s3c24xx_audio_simtec_pdata simtec_audio_platdata = { }; -static struct s3c24xx_platdata_iis simtec_audio_platdata = { - .ops = &simtec_audio_ops, - .codec_clk = "clkout0", - .match_dev = simtec_audio_matchdev, -}; +static char our_name[32]; static struct platform_device audio_dev = { - .name = "s3c24xx-tlv320aic23", + .name = our_name, + .id = -1, .dev = { .parent = &s3c_device_iis.dev, .platform_data = &simtec_audio_platdata, }, }; -static struct platform_device audio_dev2 = { - .name = "s3c24xx-tlv320aic33", - .dev = { - .parent = &s3c_device_iis.dev, - .platform_data = &simtec_audio_platdata, - }, -}; - -static void __init simtec_audio_cdclk_mpllin(void) +int __init simtec_audio_add(const char *name, + struct s3c24xx_audio_simtec_pdata *spd) { - simtec_audio_platdata.flags = AUDIO_CDCLK_MPLL | AUDIO_CDCLK_OUTPUT; -} + if (!name) + name = "tlv320aic23"; -int __init simtec_audio_add(unsigned int is_aic33, - unsigned int cdclk_master) -{ - if (cdclk_master) - simtec_audio_cdclk_mpllin(); + snprintf(our_name, sizeof(our_name)-1, "s3c24xx-simtec-%s", name); - platform_device_register(&s3c_device_iis); + if (!spd) { + printk(KERN_ERR "%s: no source platadata\n", __func__); + return -EINVAL; + } - if (is_aic33) - platform_device_register(&audio_dev2); - else - platform_device_register(&audio_dev); + /* copy platform data so the source can be __initdata */ + simtec_audio_platdata = *spd; + + platform_device_register(&s3c_device_iis); + platform_device_register(&audio_dev); return 0; } @@ -172,7 +118,8 @@ static int __init simtec_audio_init(void printk(KERN_INFO "Simtec TLV320AIC23 Audio platform driver\n"); - simtec_audio_shutdown(NULL); + if (has_lr_routing()) + simtec_audio_platdata.startup = simtec_audio_startup; platform_device_register(&s3c_device_iis); return platform_device_register(&audio_dev); Index: b/arch/arm/common/sbcd/s3c24xx/dev-audio.c =================================================================== --- a/arch/arm/common/sbcd/s3c24xx/dev-audio.c 2009-08-04 12:05:59.000000000 +0100 +++ b/arch/arm/common/sbcd/s3c24xx/dev-audio.c 2009-08-04 12:24:11.000000000 +0100 @@ -12,6 +12,9 @@ #include <linux/kernel.h> #include <linux/ioport.h> #include <linux/device.h> +#include <linux/ioport.h> +#include <linux/gpio.h> +#include <linux/io.h> #include "s3c24xx.h" #include <asm/mach/sbcd-utils.h> @@ -24,22 +27,69 @@ #include "../devices/platform.h" #include <plat/devs.h> -#include <asm/plat-s3c24xx/audio-simtec.h> +#include <plat/audio-simtec.h> + +enum amp_type { + AMP_VR2K, + AMP_HERMES, /* GAIN0 = GPD9, GAIN1 = GPD10 */ +}; + +static void __iomem *reg; +static enum amp_type type; + +static void amp_gpio_set(struct gpio_chip *chip, + unsigned offset, int value) +{ + u32 to_write; -static unsigned char is_cdclk_master; + if (type == AMP_VR2K) { + to_write = value ? 0x01 : 0x03; + } else { + to_write = value ? 0x03 : 0x02; + } + + writeb(to_write, reg); +} + +static int amp_gpio_output(struct gpio_chip *chip, + unsigned offset, int value) +{ + amp_gpio_set(chip, offset, value); + return 0; +} + +static struct gpio_chip amp_gpio = { + .label = "amp", + .direction_output = amp_gpio_output, + .set = amp_gpio_set, + .ngpio = 1, + .base = -1, +}; + +static struct s3c24xx_audio_simtec_pdata __initdata pdata = { +}; static __init int s3c24xx_drv_bast_audio_attach(struct sbcd_state *state, struct sbcd_item *parent, struct sbcd_item *drv_it) { - int is_tlv320aic33 = sbcd_board_has_driver("tlv320aic33"); + char *name = NULL; /* some vr2000s do not have the necessary setup in their - * configuration data, so force is_cdclk_master here. */ - if (sbcd_board_has_driver("vr2k-amp")) - is_cdclk_master = 1; + * configuration data, so force cdclk_master here. */ + + if (sbcd_board_has_driver("vr2k-amp") || + sbcd_board_has_driver("hermes-amp")) { + pdata.use_mpllin = 1; + pdata.output_cdclk = 1; + } - return simtec_audio_add(is_tlv320aic33, is_cdclk_master); + if (sbcd_board_has_driver("hermes-amp")) + name = "hermes-snd"; + + pdata.amp_gpio = amp_gpio.base; + + return simtec_audio_add(name, &pdata); } @@ -61,8 +111,10 @@ static __init int s3c24xx_drv_audio_atta sbcd_get_optvals(ptr, &optname, &optval); if (strcmp(optname, "cdclk") == 0 && - strcmp(optval, "master") == 0) - is_cdclk_master = 1; + strcmp(optval, "master") == 0) { + pdata.use_mpllin = 1; + pdata.output_cdclk = 1; + } break; } } @@ -76,49 +128,57 @@ static __init int s3c24xx_drv_audio_atta sbcd_drv_decl2(cpu_audio, "cpu-audio", s3c24xx_drv_audio_attach); sbcd_drv_decl2(bast_audio, "bast-audio", s3c24xx_drv_bast_audio_attach); -static struct resource vr2k_amp_resource[] = { - [0] = { - .start = S3C2410_CS1 + 0xc0, - .end = S3C2410_CS1 + 0xc1, - .flags = IORESOURCE_MEM, - }, -}; - -static struct resource hermes_amp_resource[] = { - [0] = { - .start = S3C2410_CS1 + 0x140, - .end = S3C2410_CS1 + 0x141, - .flags = IORESOURCE_MEM, - }, -}; static __init int s3c24xx_drv_amp_attach(struct sbcd_state *state, - struct sbcd_item *parent, - struct sbcd_item *drv_it) + enum amp_type amp_type) { - struct platform_device *pdev; + unsigned long base; int ret; - ret = sbcd_dev_platform_create(state, parent, drv_it, &pdev); - if (ret) { - printk(KERN_ERR "%s: no platform device\n", __func__); - return -ENODEV; + if (state->pass != SBCD_PASS_GPIO) + return 0; + + type = amp_type; + + if (amp_type == AMP_VR2K) { + base = S3C2410_CS1 + 0xc0; + } else { + base = S3C2410_CS1 + 0x140; + pdata.amp_gain[0] = S3C2410_GPD(9); + pdata.amp_gain[1] = S3C2410_GPD(10); } - if (pdev->num_resources == 0) { - if (strcmp(pdev->name, "vr2k-amp") == 0) { - pdev->resource = vr2k_amp_resource; - pdev->num_resources = ARRAY_SIZE(vr2k_amp_resource); - } else if (strcmp(pdev->name, "hermes-amp") == 0) { - pdev->resource = hermes_amp_resource; - pdev->num_resources = ARRAY_SIZE(hermes_amp_resource); - } + /* note, we don't request the region, other gpio blocks may + * be sharing it. */ + + reg = ioremap(base, 2); + if (!reg) { + printk(KERN_ERR "%s: failed to map regs\n", __func__); + return -ENXIO; + } + + ret = gpiochip_add(&amp_gpio); + if (!ret) { + printk(KERN_ERR "%s: failed to add gpio\n", __func__); + return ret; } - pdev->dev.parent = &s3c_device_iis.dev; - platform_device_register(pdev); return 0; } -sbcd_drv_decl2(vr2k_amp, "vr2k-amp", s3c24xx_drv_amp_attach); -sbcd_drv_decl2(hermes_amp, "hermes-amp", s3c24xx_drv_amp_attach); +static __init int s3c24xx_drv_vr2k_amp_attach(struct sbcd_state *state, + struct sbcd_item *parent, + struct sbcd_item *drv_it) +{ + return s3c24xx_drv_amp_attach(state, AMP_VR2K); +} + +static __init int s3c24xx_drv_hermes_amp_attach(struct sbcd_state *state, + struct sbcd_item *parent, + struct sbcd_item *drv_it) +{ + return s3c24xx_drv_amp_attach(state, AMP_HERMES); +} + +sbcd_drv_decl2_gpio(vr2k_amp, "vr2k-amp", s3c24xx_drv_vr2k_amp_attach); +sbcd_drv_decl2_gpio(hermes_amp, "hermes-amp", s3c24xx_drv_hermes_amp_attach); Index: b/sound/soc/s3c24xx/s3c24xx_simtec.h =================================================================== --- /dev/null 1970-01-01 00:00:00.000000000 +0000 +++ b/sound/soc/s3c24xx/s3c24xx_simtec.h 2009-08-04 12:24:11.000000000 +0100 @@ -0,0 +1,22 @@ +/* sound/soc/s3c24xx/s3c24xx_simtec.h + * + * Copyright 2009 Simtec Electronics + * + * 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. +*/ + +extern void simtec_audio_init(struct snd_soc_codec *codec); + +extern int simtec_audio_core_probe(struct platform_device *pdev, + struct snd_soc_device *socdev); + +extern int simtec_audio_remove(struct platform_device *pdev); + +#ifdef CONFIG_PM +extern struct dev_pm_ops simtec_audio_pmops; +#define simtec_audio_pm &simtec_audio_pmops +#else +#define simtec_audio_pm NULL +#endif Index: b/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c =================================================================== --- /dev/null 1970-01-01 00:00:00.000000000 +0000 +++ b/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c 2009-08-04 12:24:11.000000000 +0100 @@ -0,0 +1,137 @@ +/* sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c + * + * Copyright 2009 Simtec Electronics + * + * 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/clk.h> +#include <linux/platform_device.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <plat/audio-simtec.h> + +#include "s3c24xx-pcm.h" +#include "s3c24xx-i2s.h" +#include "s3c24xx_simtec.h" + +#include "../codecs/tlv320aic23.h" + +/* supported machines: + * + * Machine Connections AMP + * ------- ----------- --- + * BAST MIC, HPOUT, LOUT, LIN TPA2001D1 (HPOUTL,R) (gain hardwired) + * VR1000 HPOUT, LIN None + * VR2000 LIN, LOUT, MIC, HP LM4871 (HPOUTL,R) + * DePicture LIN, LOUT, MIC, HP LM4871 (HPOUTL,R) + * Anubis LIN, LOUT, MIC, HP TPA2001D1 (HPOUTL,R) + */ + +static const struct snd_soc_dapm_widget dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), + SND_SOC_DAPM_LINE("Line Out", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), +}; + +static const struct snd_soc_dapm_route base_map[] = { + { "Headphone Jack", NULL, "LHPOUT"}, + { "Headphone Jack", NULL, "RHPOUT"}, + + { "Line Out", NULL, "LOUT" }, + { "Line Out", NULL, "ROUT" }, + + { "LLINEIN", NULL, "Line In"}, + { "RLINEIN", NULL, "Line In"}, + + { "MICIN", NULL, "Mic Jack"}, +}; + +/** + * simtec_tlv320aic23_init - initialise and add controls + * @codec; The codec instance to attach to. + * + * Attach our controls and configure the necessary codec + * mappings for our sound card instance. +*/ +static int simtec_tlv320aic23_init(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, dapm_widgets, + ARRAY_SIZE(dapm_widgets)); + + snd_soc_dapm_add_routes(codec, base_map, ARRAY_SIZE(base_map)); + + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Line In"); + snd_soc_dapm_enable_pin(codec, "Line Out"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + + simtec_audio_init(codec); + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link simtec_dai_aic23 = { + .name = "tlv320aic23", + .stream_name = "TLV320AIC23", + .cpu_dai = &s3c24xx_i2s_dai, + .codec_dai = &tlv320aic23_dai, + .init = simtec_tlv320aic23_init, +}; + +/* simtec audio machine driver */ +static struct snd_soc_card snd_soc_machine_simtec_aic23 = { + .name = "Simtec", + .platform = &s3c24xx_soc_platform, + .dai_link = &simtec_dai_aic23, + .num_links = 1, +}; + +/* simtec audio subsystem */ +static struct snd_soc_device simtec_snd_devdata_aic23 = { + .card = &snd_soc_machine_simtec_aic23, + .codec_dev = &soc_codec_dev_tlv320aic23, +}; + +static int __devinit simtec_audio_tlv320aic23_probe(struct platform_device *pd) +{ + return simtec_audio_core_probe(pd, &simtec_snd_devdata_aic23); +} + +static struct platform_driver simtec_audio_tlv320aic23_platdrv = { + .driver = { + .owner = THIS_MODULE, + .name = "s3c24xx-simtec-tlv320aic23", + .pm = simtec_audio_pm, + }, + .probe = simtec_audio_tlv320aic23_probe, + .remove = __devexit_p(simtec_audio_remove), +}; + +MODULE_ALIAS("platform:s3c24xx-simtec-tlv320aic23"); + +static int __init simtec_tlv320aic23_modinit(void) +{ + return platform_driver_register(&simtec_audio_tlv320aic23_platdrv); +} + +static void __exit simtec_tlv320aic23_modexit(void) +{ + platform_driver_unregister(&simtec_audio_tlv320aic23_platdrv); +} + +module_init(simtec_tlv320aic23_modinit); +module_exit(simtec_tlv320aic23_modexit); + +MODULE_AUTHOR("Ben Dooks <ben(a)simtec.co.uk>"); +MODULE_DESCRIPTION("ALSA SoC Simtec Audio support"); +MODULE_LICENSE("GPL"); -- Ben (ben(a)fluff.org, http://www.fluff.org/) 'a smiley only costs 4 bytes'

2 1

[alsa-devel] [PATCH][2.6.31-rc5] ARM: OMAP: McBSP: Fix ASoC on OMAP1510 by fixing API of omap_mcbsp_start/stop
by Jarkko Nikula 07 Aug '09

07 Aug '09

Simultaneous audio playback and capture on OMAP1510 can cause that second stream is stalled if there is enough delay between startup of the audio streams. Current implementation of the omap_mcbsp_start is starting both transmitter and receiver at the same time and it is called only for firstly started audio stream from the OMAP McBSP based ASoC DAI driver. Since DMA request lines on OMAP1510 are edge sensitive, the DMA request is missed if there is no DMA transfer set up at that time when the first word after McBSP startup is transmitted. The problem hasn't noted before since later OMAPs are using level sensitive DMA request lines. Fix the problem by changing API of omap_mcbsp_start and omap_mcbsp_stop by allowing to start and stop individually McBSP transmitter and receiver logics. Then call those functions individually for both audio playback and capture streams. This ensures that DMA transfer is setup before transmitter or receiver is started. Thanks to Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> for detailed problem analysis and Peter Ujfalusi <peter.ujfalusi(a)nokia.com> for info about DMA request line behavior differences between the OMAP generations. Reported-and-tested-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Tony Lindgren <tony(a)atomide.com> --- arch/arm/plat-omap/include/mach/mcbsp.h | 4 +- arch/arm/plat-omap/mcbsp.c | 50 +++++++++++++++++++----------- sound/soc/omap/omap-mcbsp.c | 10 +++--- 3 files changed, 39 insertions(+), 25 deletions(-) diff --git a/arch/arm/plat-omap/include/mach/mcbsp.h b/arch/arm/plat-omap/include/mach/mcbsp.h index bb154ea..57249bb 100644 --- a/arch/arm/plat-omap/include/mach/mcbsp.h +++ b/arch/arm/plat-omap/include/mach/mcbsp.h @@ -387,8 +387,8 @@ void omap_mcbsp_register_board_cfg(struct omap_mcbsp_platform_data *config, void omap_mcbsp_config(unsigned int id, const struct omap_mcbsp_reg_cfg * config); int omap_mcbsp_request(unsigned int id); void omap_mcbsp_free(unsigned int id); -void omap_mcbsp_start(unsigned int id); -void omap_mcbsp_stop(unsigned int id); +void omap_mcbsp_start(unsigned int id, int tx, int rx); +void omap_mcbsp_stop(unsigned int id, int tx, int rx); void omap_mcbsp_xmit_word(unsigned int id, u32 word); u32 omap_mcbsp_recv_word(unsigned int id); diff --git a/arch/arm/plat-omap/mcbsp.c b/arch/arm/plat-omap/mcbsp.c index efa0e01..a3d2313 100644 --- a/arch/arm/plat-omap/mcbsp.c +++ b/arch/arm/plat-omap/mcbsp.c @@ -328,14 +328,15 @@ void omap_mcbsp_free(unsigned int id) EXPORT_SYMBOL(omap_mcbsp_free); /* - * Here we start the McBSP, by enabling the sample - * generator, both transmitter and receivers, - * and the frame sync. + * Here we start the McBSP, by enabling transmitter, receiver or both. + * If no transmitter or receiver is active prior calling, then sample-rate + * generator and frame sync are started. */ -void omap_mcbsp_start(unsigned int id) +void omap_mcbsp_start(unsigned int id, int tx, int rx) { struct omap_mcbsp *mcbsp; void __iomem *io_base; + int idle; u16 w; if (!omap_mcbsp_check_valid_id(id)) { @@ -348,32 +349,40 @@ void omap_mcbsp_start(unsigned int id) mcbsp->rx_word_length = (OMAP_MCBSP_READ(io_base, RCR1) >> 5) & 0x7; mcbsp->tx_word_length = (OMAP_MCBSP_READ(io_base, XCR1) >> 5) & 0x7; - /* Start the sample generator */ - w = OMAP_MCBSP_READ(io_base, SPCR2); - OMAP_MCBSP_WRITE(io_base, SPCR2, w | (1 << 6)); + idle = !((OMAP_MCBSP_READ(io_base, SPCR2) | + OMAP_MCBSP_READ(io_base, SPCR1)) & 1); + + if (idle) { + /* Start the sample generator */ + w = OMAP_MCBSP_READ(io_base, SPCR2); + OMAP_MCBSP_WRITE(io_base, SPCR2, w | (1 << 6)); + } /* Enable transmitter and receiver */ w = OMAP_MCBSP_READ(io_base, SPCR2); - OMAP_MCBSP_WRITE(io_base, SPCR2, w | 1); + OMAP_MCBSP_WRITE(io_base, SPCR2, w | (tx & 1)); w = OMAP_MCBSP_READ(io_base, SPCR1); - OMAP_MCBSP_WRITE(io_base, SPCR1, w | 1); + OMAP_MCBSP_WRITE(io_base, SPCR1, w | (rx & 1)); udelay(100); - /* Start frame sync */ - w = OMAP_MCBSP_READ(io_base, SPCR2); - OMAP_MCBSP_WRITE(io_base, SPCR2, w | (1 << 7)); + if (idle) { + /* Start frame sync */ + w = OMAP_MCBSP_READ(io_base, SPCR2); + OMAP_MCBSP_WRITE(io_base, SPCR2, w | (1 << 7)); + } /* Dump McBSP Regs */ omap_mcbsp_dump_reg(id); } EXPORT_SYMBOL(omap_mcbsp_start); -void omap_mcbsp_stop(unsigned int id) +void omap_mcbsp_stop(unsigned int id, int tx, int rx) { struct omap_mcbsp *mcbsp; void __iomem *io_base; + int idle; u16 w; if (!omap_mcbsp_check_valid_id(id)) { @@ -386,15 +395,20 @@ void omap_mcbsp_stop(unsigned int id) /* Reset transmitter */ w = OMAP_MCBSP_READ(io_base, SPCR2); - OMAP_MCBSP_WRITE(io_base, SPCR2, w & ~(1)); + OMAP_MCBSP_WRITE(io_base, SPCR2, w & ~(tx & 1)); /* Reset receiver */ w = OMAP_MCBSP_READ(io_base, SPCR1); - OMAP_MCBSP_WRITE(io_base, SPCR1, w & ~(1)); + OMAP_MCBSP_WRITE(io_base, SPCR1, w & ~(rx & 1)); - /* Reset the sample rate generator */ - w = OMAP_MCBSP_READ(io_base, SPCR2); - OMAP_MCBSP_WRITE(io_base, SPCR2, w & ~(1 << 6)); + idle = !((OMAP_MCBSP_READ(io_base, SPCR2) | + OMAP_MCBSP_READ(io_base, SPCR1)) & 1); + + if (idle) { + /* Reset the sample rate generator */ + w = OMAP_MCBSP_READ(io_base, SPCR2); + OMAP_MCBSP_WRITE(io_base, SPCR2, w & ~(1 << 6)); + } } EXPORT_SYMBOL(omap_mcbsp_stop); diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index a5d46a7..6a837ff 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -183,21 +183,21 @@ static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); - int err = 0; + int err = 0, play = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (!mcbsp_data->active++) - omap_mcbsp_start(mcbsp_data->bus_id); + mcbsp_data->active++; + omap_mcbsp_start(mcbsp_data->bus_id, play, !play); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (!--mcbsp_data->active) - omap_mcbsp_stop(mcbsp_data->bus_id); + omap_mcbsp_stop(mcbsp_data->bus_id, play, !play); + mcbsp_data->active--; break; default: err = -EINVAL; -- 1.6.3.3

4 6

[alsa-devel] tlv320aic3x: Allow driver to use already registered i2c device
by Ben Dooks 07 Aug '09

07 Aug '09

Add the ability for the driver to use an already registered i2c device instead of creating one of them itself. Add the field i2c_already field to the setup data and not set the i2c bus address the driver will then use the device already setup. Signed-off-by: Ben Dooks <ben(a)simtec.co.uk> --- sound/soc/codecs/tlv320aic3x.c | 9 ++++++++- sound/soc/codecs/tlv320aic3x.h | 1 + 2 files changed, 9 insertions(+), 1 deletion(-) Index: b/sound/soc/codecs/tlv320aic3x.c =================================================================== --- a/sound/soc/codecs/tlv320aic3x.c 2009-08-06 11:53:30.000000000 +0100 +++ b/sound/soc/codecs/tlv320aic3x.c 2009-08-06 15:29:45.000000000 +0100 @@ -1310,6 +1310,7 @@ static int aic3x_i2c_remove(struct i2c_c static const struct i2c_device_id aic3x_i2c_id[] = { { "tlv320aic3x", 0 }, + { "tlv320aic33", 0 }, { } }; MODULE_DEVICE_TABLE(i2c, aic3x_i2c_id); @@ -1405,6 +1406,10 @@ static int aic3x_probe(struct platform_d codec->hw_write = (hw_write_t) i2c_master_send; codec->hw_read = (hw_read_t) aic3x_i2c_read; ret = aic3x_add_i2c_device(pdev, setup); + } else if (setup->i2c_already) { + codec->hw_write = (hw_write_t) i2c_master_send; + codec->hw_read = (hw_read_t) aic3x_i2c_read; + i2c_add_driver(&aic3x_i2c_driver); } #else /* Add other interfaces here */ @@ -1421,6 +1426,7 @@ static int aic3x_remove(struct platform_ { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; + struct aic3x_setup_data *setup = socdev->codec_data; /* power down chip */ if (codec->control_data) @@ -1429,7 +1435,8 @@ static int aic3x_remove(struct platform_ snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - i2c_unregister_device(codec->control_data); + if (setup && !setup->i2c_already) + i2c_unregister_device(codec->control_data); i2c_del_driver(&aic3x_i2c_driver); #endif kfree(codec->private_data); Index: b/sound/soc/codecs/tlv320aic3x.h =================================================================== --- a/sound/soc/codecs/tlv320aic3x.h 2009-08-06 11:53:30.000000000 +0100 +++ b/sound/soc/codecs/tlv320aic3x.h 2009-08-06 12:11:03.000000000 +0100 @@ -283,6 +283,7 @@ int aic3x_button_pressed(struct snd_soc_ struct aic3x_setup_data { int i2c_bus; + unsigned char i2c_already; /* i2c device already registered */ unsigned short i2c_address; unsigned int gpio_func[2]; }; -- Ben (ben(a)fluff.org, http://www.fluff.org/) 'a smiley only costs 4 bytes'

2 1

[alsa-devel] usb audio
by bruce 07 Aug '09

07 Aug '09

Hi I have the TASCAM US-122L USB.I am using ubuntu studio 9.04.The usb audio inputs and outputs are no functional cant fined them anywhere I am using jack connection kit on my studio the midi are recognized but not the audio.I have submitted to launchpad this report.Here are my jack start up log this is with tascam us-122L set as input and output.It will start with just tascam input but there is no audio only midi with that start. jackd 0.116.1 Copyright 2001-2005 Paul Davis and others. jackd comes with ABSOLUTELY NO WARRANTY This is free software, and you are welcome to redistribute it under certain conditions; see the file COPYING for details JACK compiled with System V SHM support. server `default' registered 23:53:21.206 JACK was started with PID=8295. registered builtin port type 32 bit float mono audio registered builtin port type 8 bit raw midi clock source = system clock via clock_gettime start poll on 3 fd's loading driver .. new client: alsa_pcm, id = 1 type 1 @ 0x9821d48 fd = -1 apparent rate = 44100 creating alsa driver ... hw:1|hw:1|1024|3|44100|0|0|hwmon|hwmeter|-|32bit control device hw:1 23:53:21.383 ALSA active patchbay scan... 23:53:21.407 jack-imput-output003: 14:0 Midi Through -> 20:0 TASCAM US-122L checked. 23:53:21.416 ALSA connection change. 23:53:21.811 ALSA active patchbay scan... 23:53:21.813 jack-imput-output003: 14:0 Midi Through -> 20:0 TASCAM US-122L checked. ALSA: Cannot open PCM device alsa_pcm for playback. Falling back to capture-only mode cannot load driver module alsa starting server engine shutdown server thread back from poll freeing shared port segments stopping server thread last xrun delay: 0.000 usecs max delay reported by backend: 0.000 usecs freeing engine shared memory max usecs: 0.000, engine deleted cleaning up shared memory cleaning up files unregistering server `default' 23:53:22.586 JACK was stopped successfully. 23:53:22.588 Post-shutdown script... 23:53:22.590 killall jackd jackd: no process killed 23:53:23.015 Post-shutdown script terminated with exit status=256. 23:53:23.727 Could not connect to JACK server as client. - Overall operation failed. - Unable to connect to server. Please check the messages window for more info. thank you for any help you mite give :-) Bruce McCandless

1 0

[alsa-devel] [PATCH 1/3] ASoC: change set_tdm_slot api to allow slot_width override.
by Daniel Ribeiro 06 Aug '09

06 Aug '09

Extend set_tdm_slot to allow the user to arbitrarily set the frame width and active TX/RX slots. Updates magician.c and wm9081.c for the new set_tdm_slot(). wm9081.c still doesn't handle the slot_width override. While being there, correct an incorrect use of SlotsPerFrm(7) use in bitmask on pxa-ssp.c (SSCR0_SlotsPerFrm(x) is (((x) - 1) << 24)) ). (this series is meant for Mark's for-2.6.32 branch) Signed-off-by: Daniel Ribeiro <drwyrm(a)gmail.com> --- include/sound/soc-dai.h | 5 +++-- sound/soc/codecs/wm9081.c | 5 +++-- sound/soc/pxa/magician.c | 2 +- sound/soc/pxa/pxa-ssp.c | 27 ++++++++++++++++++++------- sound/soc/soc-core.c | 9 ++++++--- 5 files changed, 33 insertions(+), 15 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 352d7ee..6692410 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -106,7 +106,7 @@ int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, - unsigned int mask, int slots); + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width); int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); @@ -140,7 +140,8 @@ struct snd_soc_dai_ops { */ int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt); int (*set_tdm_slot)(struct snd_soc_dai *dai, - unsigned int mask, int slots); + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width); int (*set_tristate)(struct snd_soc_dai *dai, int tristate); /* diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 86fc57e..c47d573 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1206,8 +1206,9 @@ static int wm9081_set_sysclk(struct snd_soc_dai *codec_dai, return 0; } +/* FIXME: Needs to handle slot_width */ static int wm9081_set_tdm_slot(struct snd_soc_dai *dai, - unsigned int mask, int slots) + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { struct snd_soc_codec *codec = dai->codec; unsigned int aif1 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_1); @@ -1219,7 +1220,7 @@ static int wm9081_set_tdm_slot(struct snd_soc_dai *dai, aif1 |= (slots - 1) << WM9081_AIFDAC_TDM_MODE_SHIFT; - switch (mask) { + switch (rx_mask) { case 1: break; case 2: diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index c89a3cd..34437d8 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -188,7 +188,7 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 1); + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 0, 1, width); if (ret < 0) return ret; diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index e22c5ce..5b9ed64 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -375,21 +375,34 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, * Set the active slots in TDM/Network mode */ static int pxa_ssp_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, - unsigned int mask, int slots) + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { struct ssp_priv *priv = cpu_dai->private_data; struct ssp_device *ssp = priv->dev.ssp; u32 sscr0; - sscr0 = ssp_read_reg(ssp, SSCR0) & ~SSCR0_SlotsPerFrm(7); + sscr0 = ssp_read_reg(ssp, SSCR0); + sscr0 &= ~(SSCR0_MOD | SSCR0_SlotsPerFrm(8) | SSCR0_EDSS | SSCR0_DSS); + + /* set slot width */ + if (slot_width > 16) + sscr0 |= SSCR0_EDSS | SSCR0_DataSize(slot_width - 16); + else + sscr0 |= SSCR0_DataSize(slot_width); - /* set number of active slots */ - sscr0 |= SSCR0_SlotsPerFrm(slots); + if (slots > 1) { + /* enable network mode */ + sscr0 |= SSCR0_MOD; + + /* set number of active slots */ + sscr0 |= SSCR0_SlotsPerFrm(slots); + + /* set active slot mask */ + ssp_write_reg(ssp, SSTSA, tx_mask); + ssp_write_reg(ssp, SSRSA, rx_mask); + } ssp_write_reg(ssp, SSCR0, sscr0); - /* set active slot mask */ - ssp_write_reg(ssp, SSTSA, mask); - ssp_write_reg(ssp, SSRSA, mask); return 0; } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e1a920c..c72a8e8 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2133,17 +2133,20 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt); /** * snd_soc_dai_set_tdm_slot - configure DAI TDM. * @dai: DAI - * @mask: DAI specific mask representing used slots. + * @tx_mask: bitmask representing active TX slots. + * @rx_mask: bitmask representing active RX slots. * @slots: Number of slots in use. + * @slot_width: Width in bits for each slot. * * Configures a DAI for TDM operation. Both mask and slots are codec and DAI * specific. */ int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, - unsigned int mask, int slots) + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { if (dai->ops && dai->ops->set_tdm_slot) - return dai->ops->set_tdm_slot(dai, mask, slots); + return dai->ops->set_tdm_slot(dai, tx_mask, rx_mask, + slots, slot_width); else return -EINVAL; } -- tg: (c787980..) asoc/set_tdm_slot (depends on: asoc/remove-code-duplication) -- Daniel Ribeiro

6 6

[alsa-devel] [PATCH 2/2] ASoC: AMS_DELTA: Use updated line discipline name
by Janusz Krzysztofik 06 Aug '09

06 Aug '09

This patch updates the tty line discipline name used in the Amstrad Delta machine driver. Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- --- linux-2.6.31-rc5/sound/soc/omap/ams-delta.c.orig 2009-08-05 21:44:35.000000000 +0200 +++ linux-2.6.31-rc5/sound/soc/omap/ams-delta.c 2009-08-05 23:30:53.000000000 +0200 @@ -519,7 +519,7 @@ static int ams_delta_cx20442_init(struct } /* Register optional line discipline for over the modem control */ - ret = tty_register_ldisc(N_AMSDELTA, &cx81801_ops); + ret = tty_register_ldisc(N_V253, &cx81801_ops); if (ret) { dev_warn(card->dev, "Failed to register line discipline, " @@ -644,9 +644,9 @@ static void __exit ams_delta_module_exit } } - if (tty_unregister_ldisc(N_AMSDELTA) != 0) + if (tty_unregister_ldisc(N_V253) != 0) dev_warn(&ams_delta_audio_platform_device->dev, - "failed to unregister AMSDELTA line discipline\n"); + "failed to unregister V253 line discipline\n"); snd_soc_jack_free_gpios(&ams_delta_hook_switch, ARRAY_SIZE(ams_delta_hook_switch_gpios),

1 0

[alsa-devel] [PATCH 1/2] TTY: Rename N_AMSDELTA line discipline to N_V253
by Janusz Krzysztofik 06 Aug '09

06 Aug '09

The patch changes the line discipline name registered in include/linux/tty.h Signed-off-by: Janusz Krzysztofik --- --- linux-2.6.31-rc5/include/linux/tty.h.orig 2009-08-05 21:44:33.000000000 +0200 +++ linux-2.6.31-rc5/include/linux/tty.h 2009-08-05 22:54:12.000000000 +0200 @@ -47,7 +47,7 @@ #define N_SLCAN 17 /* Serial / USB serial CAN Adaptors */ #define N_PPS 18 /* Pulse per Second */ -#define N_AMSDELTA 19 /* codec control over modem, board specific */ +#define N_V253 19 /* Codec control over voice modem */ /* * This character is the same as _POSIX_VDISABLE: it cannot be used as

1 0

[alsa-devel] [PATCH 0/2] ASoC: CX2044: full duplex mode fix
by Janusz Krzysztofik 06 Aug '09

06 Aug '09

This series of patches tries to solve the problem of audio mode changes not working correctly while in full duplex mode. [PATCH 1/2] ASoC: CX20442: add some debugging [PATCH 2/2] ASoC: CX20442: simplify codec controller usage Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl>

2 1

[alsa-devel] [PATCH 2/2] ASoC: CX20442: simplify codec controller usage
by Janusz Krzysztofik 06 Aug '09

06 Aug '09

This patch is a workaround for the problem of several subsequent control statements not being applied correctly to the codec controller (modem). In order to follow the hook switch state change from handset to handsfree while in full duplex mode, two consecutive +VLS control commands were sent to the modem. The first one was M1 (microphone only), the seconds one was M1S1 (both microphone and speaker). As there was no real modem handshaking procedure implemented, neither in the codec nor in the machine driver part of the line discipline, the modem was having the second command missed. Since a possibility to switch to microphone only mode (and speaker only mode as well) seams of no value, I have modified the code to issue single M1S1 command only for any of those cases. Tested on my Amstrad Delta. Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- --- linux-2.6.31-rc5/sound/soc/codecs/cx20442.c.orig 2009-08-04 23:41:03.000000000 +0200 +++ linux-2.6.31-rc5/sound/soc/codecs/cx20442.c 2009-08-05 00:56:30.000000000 +0200 @@ -138,9 +138,7 @@ static int cx20442_pm_to_v253_vls(u8 val case 0: return V253_VLS_T; case (1 << CX20442_SPKOUT): - return V253_VLS_S1; case (1 << CX20442_MIC): - return V253_VLS_M1; case (1 << CX20442_SPKOUT) | (1 << CX20442_MIC): return V253_VLS_M1S1; case (1 << CX20442_TELOUT):

1 0

[alsa-devel] [PATCH 1/2] ASoC: CX20442: add some debugging
by Janusz Krzysztofik 06 Aug '09

06 Aug '09

This patch adds debugging statement that can help in tracing how the driver is trying to control the codec device. Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- --- linux-2.6.31-rc5/sound/soc/codecs/cx20442.c.orig 2009-08-04 23:41:03.000000000 +0200 +++ linux-2.6.31-rc5/sound/soc/codecs/cx20442.c 2009-08-05 00:56:30.000000000 +0200 @@ -203,6 +201,7 @@ static int cx20442_write(struct snd_soc_ if (unlikely(len > (ARRAY_SIZE(buf) - 1))) return -ENOMEM; + dev_dbg(codec->dev, "%s: %s\n", __func__, buf); if (codec->hw_write(codec->control_data, buf, len) != len) return -EIO;

1 0

[alsa-devel] [PATCH] ASoC: Add WM8776 CODEC driver
by Mark Brown 06 Aug '09

06 Aug '09

The WM8776 is a high performance, stereo audio CODEC with five channel input selector. The WM8776 is ideal for surround sound processing applications for home hi-fi, DVD-RW and other audio visual equipment. This driver implements support for most WM8776 features - currently the ADC automatic level control/limiter functionality is omitted. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8776.c | 781 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8776.h | 51 +++ 4 files changed, 838 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/wm8776.c create mode 100644 sound/soc/codecs/wm8776.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index b6c7f7a..8cd24b0 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -33,6 +33,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8731 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8750 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8753 if SND_SOC_I2C_AND_SPI + select SND_SOC_WM8776 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8900 if I2C select SND_SOC_WM8903 if I2C select SND_SOC_WM8971 if I2C @@ -132,6 +133,9 @@ config SND_SOC_WM8750 config SND_SOC_WM8753 tristate +config SND_SOC_WM8776 + tristate + config SND_SOC_WM8900 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f265380..e80f164 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -21,6 +21,7 @@ snd-soc-wm8728-objs := wm8728.o snd-soc-wm8731-objs := wm8731.o snd-soc-wm8750-objs := wm8750.o snd-soc-wm8753-objs := wm8753.o +snd-soc-wm8776-objs := wm8776.o snd-soc-wm8900-objs := wm8900.o snd-soc-wm8903-objs := wm8903.o snd-soc-wm8971-objs := wm8971.o @@ -52,6 +53,7 @@ obj-$(CONFIG_SND_SOC_WM8728) += snd-soc-wm8728.o obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o +obj-$(CONFIG_SND_SOC_WM8776) += snd-soc-wm8776.o obj-$(CONFIG_SND_SOC_WM8900) += snd-soc-wm8900.o obj-$(CONFIG_SND_SOC_WM8903) += snd-soc-wm8903.o obj-$(CONFIG_SND_SOC_WM8971) += snd-soc-wm8971.o diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c new file mode 100644 index 0000000..ee12bf8 --- /dev/null +++ b/sound/soc/codecs/wm8776.c @@ -0,0 +1,781 @@ +/* + * wm8776.c -- WM8776 ALSA SoC Audio driver + * + * Copyright 2009 Wolfson Microelectronics plc + * + * Author: Mark Brown <broonie(a)opensource.wolfsonmicro.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. + * + * TODO: Input ALC/limiter support + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "wm8776.h" + +static struct snd_soc_codec *wm8776_codec; +struct snd_soc_codec_device soc_codec_dev_wm8776; + +/* codec private data */ +struct wm8776_priv { + struct snd_soc_codec codec; + u16 reg_cache[WM8776_CACHEREGNUM]; + int sysclk[2]; +}; + +#ifdef CONFIG_SPI_MASTER +static int wm8776_spi_write(struct spi_device *spi, const char *data, int len); +#endif + +static const u16 wm8776_reg[WM8776_CACHEREGNUM] = { + 0x79, 0x79, 0x79, 0xff, 0xff, /* 4 */ + 0xff, 0x00, 0x90, 0x00, 0x00, /* 9 */ + 0x22, 0x22, 0x22, 0x08, 0xcf, /* 14 */ + 0xcf, 0x7b, 0x00, 0x32, 0x00, /* 19 */ + 0xa6, 0x01, 0x01 +}; + +/* + * read wm8776 register cache + */ +static inline unsigned int wm8776_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg >= WM8776_CACHEREGNUM) + return -1; + return cache[reg]; +} + +/* + * write wm8776 register cache + */ +static inline void wm8776_write_reg_cache(struct snd_soc_codec *codec, + u16 reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + if (reg >= WM8776_CACHEREGNUM) + return; + cache[reg] = value; +} + +/* + * write to the WM8776 register space + */ +static int wm8776_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + + data[0] = (reg << 1) | ((value >> 8) & 0x0001); + data[1] = value & 0x00ff; + + wm8776_write_reg_cache(codec, reg, value); + if (codec->hw_write(codec->control_data, data, 2) == 2) + return 0; + else + return -EIO; +} + +static int wm8776_reset(struct snd_soc_codec *codec) +{ + return wm8776_write(codec, WM8776_RESET, 0); +} + +static const DECLARE_TLV_DB_SCALE(hp_tlv, -12100, 100, 1); +static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1); +static const DECLARE_TLV_DB_SCALE(adc_tlv, -10350, 50, 1); + +static const struct snd_kcontrol_new wm8776_snd_controls[] = { +SOC_DOUBLE_R_TLV("Headphone Playback Volume", WM8776_HPLVOL, WM8776_HPRVOL, + 0, 127, 0, hp_tlv), +SOC_DOUBLE_R_TLV("Digital Playback Volume", WM8776_DACLVOL, WM8776_DACRVOL, + 0, 255, 0, dac_tlv), +SOC_SINGLE("Digital Playback ZC Switch", WM8776_DACCTRL1, 0, 1, 0), + +SOC_SINGLE("Deemphasis Switch", WM8776_DACCTRL2, 0, 1, 0), + +SOC_DOUBLE_R_TLV("Capture Volume", WM8776_ADCLVOL, WM8776_ADCRVOL, + 0, 255, 0, adc_tlv), +SOC_DOUBLE("Capture Switch", WM8776_ADCMUX, 7, 6, 1, 1), +SOC_DOUBLE_R("Capture ZC Switch", WM8776_ADCLVOL, WM8776_ADCRVOL, 8, 1, 0), +SOC_SINGLE("Capture HPF Switch", WM8776_ADCIFCTRL, 8, 1, 1), +}; + +static const struct snd_kcontrol_new inmix_controls[] = { +SOC_DAPM_SINGLE("AIN1 Switch", WM8776_ADCMUX, 0, 1, 0), +SOC_DAPM_SINGLE("AIN2 Switch", WM8776_ADCMUX, 1, 1, 0), +SOC_DAPM_SINGLE("AIN3 Switch", WM8776_ADCMUX, 2, 1, 0), +SOC_DAPM_SINGLE("AIN4 Switch", WM8776_ADCMUX, 3, 1, 0), +SOC_DAPM_SINGLE("AIN5 Switch", WM8776_ADCMUX, 4, 1, 0), +}; + +static const struct snd_kcontrol_new outmix_controls[] = { +SOC_DAPM_SINGLE("DAC Switch", WM8776_OUTMUX, 0, 1, 0), +SOC_DAPM_SINGLE("AUX Switch", WM8776_OUTMUX, 1, 1, 0), +SOC_DAPM_SINGLE("Bypass Switch", WM8776_OUTMUX, 2, 1, 0), +}; + +static const struct snd_soc_dapm_widget wm8776_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("AUX"), +SND_SOC_DAPM_INPUT("AUX"), + +SND_SOC_DAPM_INPUT("AIN1"), +SND_SOC_DAPM_INPUT("AIN2"), +SND_SOC_DAPM_INPUT("AIN3"), +SND_SOC_DAPM_INPUT("AIN4"), +SND_SOC_DAPM_INPUT("AIN5"), + +SND_SOC_DAPM_MIXER("Input Mixer", WM8776_PWRDOWN, 6, 1, + inmix_controls, ARRAY_SIZE(inmix_controls)), + +SND_SOC_DAPM_ADC("ADC", "Capture", WM8776_PWRDOWN, 1, 1), +SND_SOC_DAPM_DAC("DAC", "Playback", WM8776_PWRDOWN, 2, 1), + +SND_SOC_DAPM_MIXER("Output Mixer", SND_SOC_NOPM, 0, 0, + outmix_controls, ARRAY_SIZE(outmix_controls)), + +SND_SOC_DAPM_PGA("Headphone PGA", WM8776_PWRDOWN, 3, 1, NULL, 0), + +SND_SOC_DAPM_OUTPUT("VOUT"), + +SND_SOC_DAPM_OUTPUT("HPOUTL"), +SND_SOC_DAPM_OUTPUT("HPOUTR"), +}; + +static const struct snd_soc_dapm_route routes[] = { + { "Input Mixer", "AIN1 Switch", "AIN1" }, + { "Input Mixer", "AIN2 Switch", "AIN2" }, + { "Input Mixer", "AIN3 Switch", "AIN3" }, + { "Input Mixer", "AIN4 Switch", "AIN4" }, + { "Input Mixer", "AIN5 Switch", "AIN5" }, + + { "ADC", NULL, "Input Mixer" }, + + { "Output Mixer", "DAC Switch", "DAC" }, + { "Output Mixer", "AUX Switch", "AUX" }, + { "Output Mixer", "Bypass Switch", "Input Mixer" }, + + { "VOUT", NULL, "Output Mixer" }, + + { "Headphone PGA", NULL, "Output Mixer" }, + + { "HPOUTL", NULL, "Headphone PGA" }, + { "HPOUTR", NULL, "Headphone PGA" }, +}; + +static int wm8776_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + int reg, iface, master; + + switch (dai->id) { + case WM8776_DAI_DAC: + reg = WM8776_DACIFCTRL; + master = 0x80; + break; + case WM8776_DAI_ADC: + reg = WM8776_ADCIFCTRL; + master = 0x100; + break; + default: + return -EINVAL; + } + + iface = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + break; + case SND_SOC_DAIFMT_CBS_CFS: + master = 0; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= 0x0002; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= 0x0001; + break; + /* FIXME: CHECK A/B */ + case SND_SOC_DAIFMT_DSP_A: + iface |= 0x0003; + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= 0x0007; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= 0x00c; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= 0x008; + break; + case SND_SOC_DAIFMT_NB_IF: + iface |= 0x004; + break; + default: + return -EINVAL; + } + + /* Finally, write out the values */ + snd_soc_update_bits(codec, reg, 0xf, iface); + snd_soc_update_bits(codec, WM8776_MSTRCTRL, 0x180, master); + + return 0; +} + +static int mclk_ratios[] = { + 128, + 192, + 256, + 384, + 512, + 768, +}; + +static int wm8776_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8776_priv *wm8776 = codec->private_data; + int iface_reg, iface; + int ratio_shift, master; + int i; + + iface = 0; + + switch (dai->id) { + case WM8776_DAI_DAC: + iface_reg = WM8776_DACIFCTRL; + master = 0x80; + ratio_shift = 4; + break; + case WM8776_DAI_ADC: + iface_reg = WM8776_ADCIFCTRL; + master = 0x100; + ratio_shift = 0; + break; + default: + return -EINVAL; + } + + + /* Set word length */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface |= 0x10; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface |= 0x20; + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface |= 0x30; + break; + } + + /* Only need to set MCLK/LRCLK ratio if we're master */ + if (snd_soc_read(codec, WM8776_MSTRCTRL) & master) { + for (i = 0; i < ARRAY_SIZE(mclk_ratios); i++) { + if (wm8776->sysclk[dai->id] / params_rate(params) + == mclk_ratios[i]) + break; + } + + if (i == ARRAY_SIZE(mclk_ratios)) { + dev_err(codec->dev, + "Unable to configure MCLK ratio %d/%d\n", + wm8776->sysclk[dai->id], params_rate(params)); + return -EINVAL; + } + + dev_dbg(codec->dev, "MCLK is %dfs\n", mclk_ratios[i]); + + snd_soc_update_bits(codec, WM8776_MSTRCTRL, + 0x7 << ratio_shift, i << ratio_shift); + } else { + dev_dbg(codec->dev, "DAI in slave mode\n"); + } + + snd_soc_update_bits(codec, iface_reg, 0x30, iface); + + return 0; +} + +static int wm8776_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + + return snd_soc_write(codec, WM8776_DACMUTE, !!mute); +} + +static int wm8776_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8776_priv *wm8776 = codec->private_data; + + BUG_ON(dai->id >= ARRAY_SIZE(wm8776->sysclk)); + + wm8776->sysclk[dai->id] = freq; + + return 0; +} + +static int wm8776_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (codec->bias_level == SND_SOC_BIAS_OFF) { + /* Disable the global powerdown; DAPM does the rest */ + snd_soc_update_bits(codec, WM8776_PWRDOWN, 1, 0); + } + + break; + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, WM8776_PWRDOWN, 1, 1); + break; + } + + codec->bias_level = level; + return 0; +} + +#define WM8776_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\ + SNDRV_PCM_RATE_96000) + + +#define WM8776_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops wm8776_dac_ops = { + .digital_mute = wm8776_mute, + .hw_params = wm8776_hw_params, + .set_fmt = wm8776_set_fmt, + .set_sysclk = wm8776_set_sysclk, +}; + +static struct snd_soc_dai_ops wm8776_adc_ops = { + .hw_params = wm8776_hw_params, + .set_fmt = wm8776_set_fmt, + .set_sysclk = wm8776_set_sysclk, +}; + +struct snd_soc_dai wm8776_dai[] = { + { + .name = "WM8776 Playback", + .id = WM8776_DAI_DAC, + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = WM8776_RATES, + .formats = WM8776_FORMATS, + }, + .ops = &wm8776_dac_ops, + }, + { + .name = "WM8776 Capture", + .id = WM8776_DAI_ADC, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = WM8776_RATES, + .formats = WM8776_FORMATS, + }, + .ops = &wm8776_adc_ops, + }, +}; +EXPORT_SYMBOL_GPL(wm8776_dai); + +#ifdef CONFIG_PM +static int wm8776_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; + + wm8776_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int wm8776_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + int i; + u8 data[2]; + u16 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(wm8776_reg); i++) { + data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); + data[1] = cache[i] & 0x00ff; + codec->hw_write(codec->control_data, data, 2); + } + + wm8776_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} +#else +#define wm8776_suspend NULL +#define wm8776_resume NULL +#endif + +static int wm8776_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (wm8776_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = wm8776_codec; + codec = wm8776_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms: %d\n", ret); + goto pcm_err; + } + + snd_soc_add_controls(codec, wm8776_snd_controls, + ARRAY_SIZE(wm8776_snd_controls)); + snd_soc_dapm_new_controls(codec, wm8776_dapm_widgets, + ARRAY_SIZE(wm8776_dapm_widgets)); + snd_soc_dapm_add_routes(codec, routes, ARRAY_SIZE(routes)); + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(codec->dev, "failed to register card: %d\n", ret); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + return ret; +} + +/* power down chip */ +static int wm8776_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8776 = { + .probe = wm8776_probe, + .remove = wm8776_remove, + .suspend = wm8776_suspend, + .resume = wm8776_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8776); + +static int wm8776_register(struct wm8776_priv *wm8776) +{ + int ret, i; + struct snd_soc_codec *codec = &wm8776->codec; + + if (wm8776_codec) { + dev_err(codec->dev, "Another WM8776 is registered\n"); + ret = -EINVAL; + goto err; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = wm8776; + codec->name = "WM8776"; + codec->owner = THIS_MODULE; + codec->read = wm8776_read_reg_cache; + codec->write = wm8776_write; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = wm8776_set_bias_level; + codec->dai = wm8776_dai; + codec->num_dai = ARRAY_SIZE(wm8776_dai); + codec->reg_cache_size = WM8776_CACHEREGNUM; + codec->reg_cache = &wm8776->reg_cache; + + memcpy(codec->reg_cache, wm8776_reg, sizeof(wm8776_reg)); + + for (i = 0; i < ARRAY_SIZE(wm8776_dai); i++) + wm8776_dai[i].dev = codec->dev; + + ret = wm8776_reset(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to issue reset: %d\n", ret); + goto err; + } + + wm8776_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* Latch the update bits; right channel only since we always + * update both. */ + snd_soc_update_bits(codec, WM8776_HPRVOL, 0x100, 0x100); + snd_soc_update_bits(codec, WM8776_DACRVOL, 0x100, 0x100); + + wm8776_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto err; + } + + ret = snd_soc_register_dais(wm8776_dai, ARRAY_SIZE(wm8776_dai)); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAIs: %d\n", ret); + goto err_codec; + } + + return 0; + +err_codec: + snd_soc_unregister_codec(codec); +err: + kfree(wm8776); + return ret; +} + +static void wm8776_unregister(struct wm8776_priv *wm8776) +{ + wm8776_set_bias_level(&wm8776->codec, SND_SOC_BIAS_OFF); + snd_soc_unregister_dais(wm8776_dai, ARRAY_SIZE(wm8776_dai)); + snd_soc_unregister_codec(&wm8776->codec); + kfree(wm8776); + wm8776_codec = NULL; +} + +#if defined(CONFIG_SPI_MASTER) +static int wm8776_spi_write(struct spi_device *spi, const char *data, int len) +{ + struct spi_transfer t; + struct spi_message m; + u8 msg[2]; + + if (len <= 0) + return 0; + + msg[0] = data[0]; + msg[1] = data[1]; + + spi_message_init(&m); + memset(&t, 0, (sizeof t)); + + t.tx_buf = &msg[0]; + t.len = len; + + spi_message_add_tail(&t, &m); + spi_sync(spi, &m); + + return len; +} + +static int __devinit wm8776_spi_probe(struct spi_device *spi) +{ + struct snd_soc_codec *codec; + struct wm8776_priv *wm8776; + + wm8776 = kzalloc(sizeof(struct wm8776_priv), GFP_KERNEL); + if (wm8776 == NULL) + return -ENOMEM; + + codec = &wm8776->codec; + codec->control_data = spi; + codec->hw_write = (hw_write_t)wm8776_spi_write; + codec->dev = &spi->dev; + + dev_set_drvdata(&spi->dev, wm8776); + + return wm8776_register(wm8776); +} + +static int __devexit wm8776_spi_remove(struct spi_device *spi) +{ + struct wm8776_priv *wm8776 = dev_get_drvdata(&spi->dev); + + wm8776_unregister(wm8776); + + return 0; +} + +#ifdef CONFIG_PM +static int wm8776_spi_suspend(struct spi_device *spi, pm_message_t msg) +{ + return snd_soc_suspend_device(&spi->dev); +} + +static int wm8776_spi_resume(struct spi_device *spi) +{ + return snd_soc_resume_device(&spi->dev); +} +#else +#define wm8776_spi_suspend NULL +#define wm8776_spi_resume NULL +#endif + +static struct spi_driver wm8776_spi_driver = { + .driver = { + .name = "wm8776", + .bus = &spi_bus_type, + .owner = THIS_MODULE, + }, + .probe = wm8776_spi_probe, + .suspend = wm8776_spi_suspend, + .resume = wm8776_spi_resume, + .remove = __devexit_p(wm8776_spi_remove), +}; +#endif /* CONFIG_SPI_MASTER */ + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static __devinit int wm8776_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct wm8776_priv *wm8776; + struct snd_soc_codec *codec; + + wm8776 = kzalloc(sizeof(struct wm8776_priv), GFP_KERNEL); + if (wm8776 == NULL) + return -ENOMEM; + + codec = &wm8776->codec; + codec->hw_write = (hw_write_t)i2c_master_send; + + i2c_set_clientdata(i2c, wm8776); + codec->control_data = i2c; + + codec->dev = &i2c->dev; + + return wm8776_register(wm8776); +} + +static __devexit int wm8776_i2c_remove(struct i2c_client *client) +{ + struct wm8776_priv *wm8776 = i2c_get_clientdata(client); + wm8776_unregister(wm8776); + return 0; +} + +#ifdef CONFIG_PM +static int wm8776_i2c_suspend(struct i2c_client *i2c, pm_message_t msg) +{ + return snd_soc_suspend_device(&i2c->dev); +} + +static int wm8776_i2c_resume(struct i2c_client *i2c) +{ + return snd_soc_resume_device(&i2c->dev); +} +#else +#define wm8776_i2c_suspend NULL +#define wm8776_i2c_resume NULL +#endif + +static const struct i2c_device_id wm8776_i2c_id[] = { + { "wm8776", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8776_i2c_id); + +static struct i2c_driver wm8776_i2c_driver = { + .driver = { + .name = "wm8776", + .owner = THIS_MODULE, + }, + .probe = wm8776_i2c_probe, + .remove = __devexit_p(wm8776_i2c_remove), + .suspend = wm8776_i2c_suspend, + .resume = wm8776_i2c_resume, + .id_table = wm8776_i2c_id, +}; +#endif + +static int __init wm8776_modinit(void) +{ + int ret; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&wm8776_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register WM8776 I2C driver: %d\n", + ret); + } +#endif +#if defined(CONFIG_SPI_MASTER) + ret = spi_register_driver(&wm8776_spi_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register WM8776 SPI driver: %d\n", + ret); + } +#endif + return 0; +} +module_init(wm8776_modinit); + +static void __exit wm8776_exit(void) +{ +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&wm8776_i2c_driver); +#endif +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&wm8776_spi_driver); +#endif +} +module_exit(wm8776_exit); + +MODULE_DESCRIPTION("ASoC WM8776 driver"); +MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8776.h b/sound/soc/codecs/wm8776.h new file mode 100644 index 0000000..6606d25 --- /dev/null +++ b/sound/soc/codecs/wm8776.h @@ -0,0 +1,51 @@ +/* + * wm8776.h -- WM8776 ASoC driver + * + * Copyright 2009 Wolfson Microelectronics plc + * + * Author: Mark Brown <broonie(a)opensource.wolfsonmicro.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 _WM8776_H +#define _WM8776_H + +/* Registers */ + +#define WM8776_HPLVOL 0x00 +#define WM8776_HPRVOL 0x01 +#define WM8776_HPMASTER 0x02 +#define WM8776_DACLVOL 0x03 +#define WM8776_DACRVOL 0x04 +#define WM8776_DACMASTER 0x05 +#define WM8776_PHASESWAP 0x06 +#define WM8776_DACCTRL1 0x07 +#define WM8776_DACMUTE 0x08 +#define WM8776_DACCTRL2 0x09 +#define WM8776_DACIFCTRL 0x0a +#define WM8776_ADCIFCTRL 0x0b +#define WM8776_MSTRCTRL 0x0c +#define WM8776_PWRDOWN 0x0d +#define WM8776_ADCLVOL 0x0e +#define WM8776_ADCRVOL 0x0f +#define WM8776_ALCCTRL1 0x10 +#define WM8776_ALCCTRL2 0x11 +#define WM8776_ALCCTRL3 0x12 +#define WM8776_NOISEGATE 0x13 +#define WM8776_LIMITER 0x14 +#define WM8776_ADCMUX 0x15 +#define WM8776_OUTMUX 0x16 +#define WM8776_RESET 0x17 + +#define WM8776_CACHEREGNUM 0x17 + +#define WM8776_DAI_DAC 0 +#define WM8776_DAI_ADC 1 + +extern struct snd_soc_dai wm8776_dai[]; +extern struct snd_soc_codec_device soc_codec_dev_wm8776; + +#endif -- 1.6.3.3

1 0

[alsa-devel] [RFC] [PATCH] ASoC: OMAP: full duplex mode fix
by Janusz Krzysztofik 06 Aug '09

06 Aug '09

This patch tries to correct the problem of full duplex mode not working over a single McBSP based CPU DAI. Created against linux-2.6.31-rc5. Tested on Amstrad Delta. Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- --- linux-2.6.31-rc5/sound/soc/omap/omap-mcbsp.c.orig 2009-08-01 02:40:45.000000000 +0200 +++ linux-2.6.31-rc5/sound/soc/omap/omap-mcbsp.c 2009-08-03 03:28:07.000000000 +0200 @@ -183,6 +183,7 @@ static int omap_mcbsp_dai_trigger(struct struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); + u16 buf; int err = 0; switch (cmd) { @@ -191,6 +192,14 @@ static int omap_mcbsp_dai_trigger(struct case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (!mcbsp_data->active++) omap_mcbsp_start(mcbsp_data->bus_id); + else if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + /* looks like capture already in progress, + * start playback by taking it out of error condition */ + omap_mcbsp_pollwrite(mcbsp_data->bus_id, 0x0); + else + /* looks like playback already in progress, + * start capture by taking it out of error condition */ + omap_mcbsp_pollread(mcbsp_data->bus_id, &buf); break; case SNDRV_PCM_TRIGGER_STOP:

5 15

[alsa-devel] ASoC: Patches for Freescale MXC SoCs (SSI, DMA and machine).
by javier Martin 06 Aug '09

06 Aug '09

Here comes a series of three patches which currently add audio support for MX1x and MX2x platforms. It has been tested in our custom Visstrim_m10 board based on an i.MX27 processor. All patches are based on stable 2.6.30 kernel and each patch applies over the previous one. [alsa-devel] [PATCH 1/3] ASoC: add DMA platform driver for MX1x and MX2x [alsa-devel] [PATCH 2/3] ASoC: add DAI platform ssi driver for MXC [alsa-devel] [PATCH 3/3] ASoC: add machine driver for i.mx27_visstrim_m10 board -- Javier Martin Vista Silicon S.L. Universidad de Cantabria CDTUC - FASE C - Oficina S-345 Avda de los Castros s/n 39005- Santander. Cantabria. Spain +34 942 25 32 60 www.vista-silicon.com

2 3

[alsa-devel] [PATCH] ALSA: ASoC: cs4270: move power management hooks to snd_soc_codec_device
by Daniel Mack 06 Aug '09

06 Aug '09

Power management for the cs4270 codec is currently implemented as part of the i2c_driver struct. The disadvantage of doing it this way is that the callbacks registered in the snd_soc_card struct are called _before_ the codec's callbacks. That doesn't work, because the snd_soc_card callbacks will most likely switch down the codec's power domains or pull the reset GPIOs, and hence make the i2c communication bail out. Fix this by binding the suspend and resume code to the snd_soc_codec_device driver model. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Timur Tabi <timur(a)freescale.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/cs4270.c | 55 ++++++++++++++++++++++----------------------- 1 files changed, 27 insertions(+), 28 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index a32b822..69b794b 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -791,6 +791,22 @@ static struct i2c_device_id cs4270_id[] = { }; MODULE_DEVICE_TABLE(i2c, cs4270_id); +/* + * cs4270_i2c_driver - I2C device identification + * + * This structure tells the I2C subsystem how to identify and support a + * given I2C device type. + */ +static struct i2c_driver cs4270_i2c_driver = { + .driver = { + .name = "cs4270", + .owner = THIS_MODULE, + }, + .id_table = cs4270_id, + .probe = cs4270_i2c_probe, + .remove = cs4270_i2c_remove, +}; + #ifdef CONFIG_PM /* This suspend/resume implementation can handle both - a simple standby @@ -802,19 +818,18 @@ MODULE_DEVICE_TABLE(i2c, cs4270_id); * and all registers are written back to the hardware when resuming. */ -static int cs4270_i2c_suspend(struct i2c_client *client, pm_message_t mesg) +static int cs4270_soc_suspend(struct platform_device *pdev, pm_message_t mesg) { - struct cs4270_private *cs4270 = i2c_get_clientdata(client); - struct snd_soc_codec *codec = &cs4270->codec; + struct snd_soc_codec *codec = cs4270_codec; int reg = snd_soc_read(codec, CS4270_PWRCTL) | CS4270_PWRCTL_PDN_ALL; return snd_soc_write(codec, CS4270_PWRCTL, reg); } -static int cs4270_i2c_resume(struct i2c_client *client) +static int cs4270_soc_resume(struct platform_device *pdev) { - struct cs4270_private *cs4270 = i2c_get_clientdata(client); - struct snd_soc_codec *codec = &cs4270->codec; + struct snd_soc_codec *codec = cs4270_codec; + struct i2c_client *i2c_client = codec->control_data; int reg; /* In case the device was put to hard reset during sleep, we need to @@ -825,7 +840,7 @@ static int cs4270_i2c_resume(struct i2c_client *client) for (reg = CS4270_FIRSTREG; reg <= CS4270_LASTREG; reg++) { u8 val = snd_soc_read(codec, reg); - if (i2c_smbus_write_byte_data(client, reg, val)) { + if (i2c_smbus_write_byte_data(i2c_client, reg, val)) { dev_err(codec->dev, "i2c write failed\n"); return -EIO; } @@ -838,29 +853,11 @@ static int cs4270_i2c_resume(struct i2c_client *client) return snd_soc_write(codec, CS4270_PWRCTL, reg); } #else -#define cs4270_i2c_suspend NULL -#define cs4270_i2c_resume NULL +#define cs4270_soc_suspend NULL +#define cs4270_soc_resume NULL #endif /* CONFIG_PM */ /* - * cs4270_i2c_driver - I2C device identification - * - * This structure tells the I2C subsystem how to identify and support a - * given I2C device type. - */ -static struct i2c_driver cs4270_i2c_driver = { - .driver = { - .name = "cs4270", - .owner = THIS_MODULE, - }, - .id_table = cs4270_id, - .probe = cs4270_i2c_probe, - .remove = cs4270_i2c_remove, - .suspend = cs4270_i2c_suspend, - .resume = cs4270_i2c_resume, -}; - -/* * ASoC codec device structure * * Assign this variable to the codec_dev field of the machine driver's @@ -868,7 +865,9 @@ static struct i2c_driver cs4270_i2c_driver = { */ struct snd_soc_codec_device soc_codec_device_cs4270 = { .probe = cs4270_probe, - .remove = cs4270_remove + .remove = cs4270_remove, + .suspend = cs4270_soc_suspend, + .resume = cs4270_soc_resume, }; EXPORT_SYMBOL_GPL(soc_codec_device_cs4270); -- 1.6.3.3

3 5

[alsa-devel] [PATCH] ASoC: MPC5200: Support for buffer wrap around
by John Bonesio 06 Aug '09

06 Aug '09

Here's another appl_ptr related patch to the alsa soc mpc5200 ac97 driver. This time the patch is to fix a problem with not correctly handling appl_ptr wrapping back around to the beginning of the buffer. I haven't yet seen a symptom related to this. This came up when I was chatting with Jon Smirl about another bug I was examining. Right now there is another problem that is preventing me from playing really long audio streams. Let me know if this is the wrong approach. - John The code in psc_dma_bcom_enqueue_tx() didn't account for the fact that s->runtime->control->appl_ptr can wrap around to the beginning of the buffer. This change fixes this problem. Signed-off-by: John Bonesio <bones(a)secretlab.ca> --- sound/soc/fsl/mpc5200_dma.c | 17 +++++++++++++++++ 1 files changed, 17 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index cfe0ea4..2551c58 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -70,6 +70,23 @@ static void psc_dma_bcom_enqueue_next_buffer(struct psc_dma_stream *s) static void psc_dma_bcom_enqueue_tx(struct psc_dma_stream *s) { + if (s->appl_ptr > s->runtime->control->appl_ptr) { + /* + * In this case s->runtime->control->appl_ptr has wrapped around. + * Play the data to the end of the boundary, then wrap our own + * appl_ptr back around. + */ + while (s->appl_ptr < s->runtime->boundary) { + if (bcom_queue_full(s->bcom_task)) + return; + + s->appl_ptr += s->period_size; + + psc_dma_bcom_enqueue_next_buffer(s); + } + s->appl_ptr -= s->runtime->boundary; + } + while (s->appl_ptr < s->runtime->control->appl_ptr) { if (bcom_queue_full(s->bcom_task))

4 5

[PATCH] ALSA: ASoC: cs4270: move power management hooks to snd_soc_codec_device
by Daniel Mack 05 Aug '09

05 Aug '09

Power management for the cs4270 codec is currently implemented as part of the i2c_driver struct. The disadvantage of doing it this way is that the callbacks registered in the snd_soc_card struct are called _before_ the codec's callbacks. That doesn't work, because the snd_soc_card callbacks will most likely switch down the codec's power domains or pull the reset GPIOs, and hence make the i2c communication bail out. Fix this by binding the suspend and resume code to the snd_soc_codec_device driver model and let the I2C functions only call the SoC core function for resume and suspend, which do nothing currently but will do later. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Timur Tabi <timur(a)freescale.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/cs4270.c | 27 +++++++++++++++++++++++---- 1 files changed, 23 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index a32b822..ca1e24a 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -806,15 +806,30 @@ static int cs4270_i2c_suspend(struct i2c_client *client, pm_message_t mesg) { struct cs4270_private *cs4270 = i2c_get_clientdata(client); struct snd_soc_codec *codec = &cs4270->codec; - int reg = snd_soc_read(codec, CS4270_PWRCTL) | CS4270_PWRCTL_PDN_ALL; - return snd_soc_write(codec, CS4270_PWRCTL, reg); + return snd_soc_suspend_device(codec->dev); } static int cs4270_i2c_resume(struct i2c_client *client) { struct cs4270_private *cs4270 = i2c_get_clientdata(client); struct snd_soc_codec *codec = &cs4270->codec; + + return snd_soc_resume_device(codec->dev); +} + +static int cs4270_soc_suspend(struct platform_device *pdev, pm_message_t mesg) +{ + struct snd_soc_codec *codec = cs4270_codec; + int reg = snd_soc_read(codec, CS4270_PWRCTL) | CS4270_PWRCTL_PDN_ALL; + + return snd_soc_write(codec, CS4270_PWRCTL, reg); +} + +static int cs4270_soc_resume(struct platform_device *pdev) +{ + struct snd_soc_codec *codec = cs4270_codec; + struct i2c_client *i2c_client = codec->control_data; int reg; /* In case the device was put to hard reset during sleep, we need to @@ -825,7 +840,7 @@ static int cs4270_i2c_resume(struct i2c_client *client) for (reg = CS4270_FIRSTREG; reg <= CS4270_LASTREG; reg++) { u8 val = snd_soc_read(codec, reg); - if (i2c_smbus_write_byte_data(client, reg, val)) { + if (i2c_smbus_write_byte_data(i2c_client, reg, val)) { dev_err(codec->dev, "i2c write failed\n"); return -EIO; } @@ -840,6 +855,8 @@ static int cs4270_i2c_resume(struct i2c_client *client) #else #define cs4270_i2c_suspend NULL #define cs4270_i2c_resume NULL +#define cs4270_soc_suspend NULL +#define cs4270_soc_resume NULL #endif /* CONFIG_PM */ /* @@ -868,7 +885,9 @@ static struct i2c_driver cs4270_i2c_driver = { */ struct snd_soc_codec_device soc_codec_device_cs4270 = { .probe = cs4270_probe, - .remove = cs4270_remove + .remove = cs4270_remove, + .suspend = cs4270_soc_suspend, + .resume = cs4270_soc_resume, }; EXPORT_SYMBOL_GPL(soc_codec_device_cs4270); -- 1.6.3.3

1 0

[alsa-devel] [PATCH - amixer] listen to D-Bus for volume-up, volume-down and mute signals.
by Łukasz Stelmach 05 Aug '09

05 Aug '09

The patch introduces D-Bus support in amixer. With --dbus option given amixer works continiously (like with the --stdin) waiting for org.freedesktop.Hal.Device.Condition with the first argument "ButtonPressed" and the second being one of: "volume-up", "volume-down" and "mute". These are emmited by the HAL input helper. A single amixer process can controll one mixer item. There are options to choose a specific item to controll, an input device to listen and a single adjustment step size. Signed-off-by: Łukasz Stelmach <stlman(a)poczta.fm> --- amixer/Makefile.am | 3 +- amixer/amixer.1 | 18 ++++++ amixer/amixer.c | 171 ++++++++++++++++++++++++++++++++++++++++++++++++++++ configure.in | 12 ++++ 4 files changed, 203 insertions(+), 1 deletions(-) diff --git a/amixer/Makefile.am b/amixer/Makefile.am index fcd0e81..fa951a6 100644 --- a/amixer/Makefile.am +++ b/amixer/Makefile.am @@ -1,5 +1,6 @@ INCLUDES = -I$(top_srcdir)/include -LDADD = -lm +LDADD = -lm @DBUS_LIBS@ +CFLAGS += @DBUS_CFLAGS@ # LDFLAGS = -static # CFLAGS += -g -Wall diff --git a/amixer/amixer.1 b/amixer/amixer.1 index b1ac323..bc160be 100644 --- a/amixer/amixer.1 +++ b/amixer/amixer.1 @@ -88,6 +88,24 @@ Select the card number to control. The device name created from this parameter has syntax 'hw:N' where N is specified card number. .TP +\fI\-\-dbus[=\fISCONTROL\fP] + +Connect to system D-Bus and wait for \fIorg.freedesktop.Hal.Device.Condition\fP +signals with ButtonPressed as the first argument and one of: \fIvolume-up\fP, +\fIvolume-down\fP or \fImute\fP as the second one. Control the selected +\fISCONTROL\fP item. The default item is 'Master,0'. + +.TP +\fI\-\-dbus-path=<\fIUDI\fP> + +Choose a particular input device that emits control events. + +.TP +\fI\-\-dbus-step=<\fIN\fP> + +Set an amount by wich to increase or decrease volume upon event. + +.TP \fI\-D\fP device Select the device name to control. The default control name is 'default'. diff --git a/amixer/amixer.c b/amixer/amixer.c index 9620721..31de674 100644 --- a/amixer/amixer.c +++ b/amixer/amixer.c @@ -18,6 +18,8 @@ * */ +#include "aconfig.h" + #include <stdio.h> #include <stdlib.h> #include <string.h> @@ -31,6 +33,10 @@ #include <sys/poll.h> #include "amixer.h" +#ifdef HAVE_DBUS +#include <dbus/dbus.h> +#endif /* HAVE_DBUS */ + #define LEVEL_BASIC (1<<0) #define LEVEL_INACTIVE (1<<1) #define LEVEL_ID (1<<2) @@ -43,6 +49,12 @@ static int ignore_error = 0; static struct snd_mixer_selem_regopt smixer_options; static char card[64] = "default"; +#ifdef HAVE_DBUS +static char dbus_sctl[64] = "Master,0"; +static char *dbus_path = NULL; +static char dbus_step[8] = "1"; +#endif /* HAVE_DBUS */ + static void error(const char *fmt,...) { va_list va; @@ -60,6 +72,11 @@ static int help(void) printf("\nAvailable options:\n"); printf(" -h,--help this help\n"); printf(" -c,--card N select the card\n"); +#ifdef HAVE_DBUS + printf(" --dbus[=sID] Connect to the system D-Bus and wait for volume messages\n"); + printf(" --dbus-path UDI Choose a particular input device (udi)\n"); + printf(" --dbus-step N Set the step size (default: 1)\n"); +#endif /* HAVE_DBUS */ printf(" -D,--device N select the device, default '%s'\n", card); printf(" -d,--debug debug mode\n"); printf(" -n,--nocheck do not perform range checking\n"); @@ -1887,15 +1904,147 @@ static int exec_stdin(void) return 0; } +#ifdef HAVE_DBUS + +#define OPT_DBUS 0x81 +#define OPT_DBUS_PATH 0x82 +#define OPT_DBUS_STEP 0x83 + +void* get_dbus_arg(DBusMessageIter* args, int type, void** out) { + if (type != dbus_message_iter_get_arg_type(args)) + return NULL; + dbus_message_iter_get_basic(args, out); + dbus_message_iter_next(args); + return *out; +} + +int exec_dbus(void) +{ + char *sargv[MAX_ARGS]; + int sargc; + const char* s; + int stepl; + DBusError error; + DBusMessage* msg; + DBusConnection *conn; + DBusMessageIter args; + + stepl = strlen(dbus_step); + dbus_step[stepl+1] = '\0'; + + dbus_error_init (&error); + conn = dbus_bus_get (DBUS_BUS_SYSTEM, &error); + + if(!conn) { + fprintf(stderr, "dbus:%s: %s\n", + error.name, error.message); + return 1; + } + + dbus_bus_add_match(conn, + "type='signal'," + "interface='org.freedesktop.Hal.Device'," + "member='Condition'", + &error); + /* The first message is a NameOwnerChanged signal, ignore it. */ + msg = dbus_connection_pop_message(conn); + dbus_message_unref(msg); + + if(dbus_error_is_set(&error)) { + fprintf(stderr, "dbus:match error:%s\n", + error.message); + return 1; + } + + while (1) { + sargv[0] = dbus_sctl; sargc = 1; + + if (!(msg = dbus_connection_pop_message(conn))) { + dbus_connection_read_write(conn, -1); + continue; + } + + s = dbus_message_get_interface(msg); + if(strncmp("org.freedesktop.Hal.Device", s, 26)) { + if(debugflag) + fprintf(stderr, "dbus:unsupported interface:%s\n", s); + goto badmessage; + } + + if(dbus_path && + (s=dbus_message_get_path(msg)) && + strncmp(dbus_path, s, strlen(s))) { + if(debugflag) + fprintf(stderr, "dbus:not the selected udi:%s\n", s); + goto badmessage; + } + + if(!dbus_message_iter_init(msg, &args)) { + if(debugflag) + fprintf(stderr, "dbus:message has no arguments\n"); + goto badmessage; + } + + if (strncmp(get_dbus_arg(&args, DBUS_TYPE_STRING, (void*)&s), "ButtonPressed", 13)) { + if(debugflag) + fprintf(stderr, "dbus:not a ButtonPressed event\n"); + goto badmessage; + } + + if (!get_dbus_arg(&args, DBUS_TYPE_STRING, (void*)&s)) { + if(debugflag) + fprintf(stderr, "dbus:invalid argument type\n"); + goto badmessage; + } + + /* static int sset(unsigned int argc, char *argv[], int roflag, int keep_handle) */ + if (!strncmp(s, "volume-up", 9)) { + if(debugflag) + fprintf(stderr, "dbus:volume-up\n"); + dbus_step[stepl]='+'; + sargv[1] = dbus_step; + } else if (!strncmp(s, "volume-down", 11)) { + if(debugflag) + fprintf(stderr, "dbus:volume-down\n"); + dbus_step[stepl]='-'; + sargv[1] = dbus_step; + } else if (!strncmp(s, "mute", 4)) { + if(debugflag) + fprintf(stderr, "dbus:mute (toggle)\n"); + sargv[1] = "toggle"; + } else { + if(debugflag) + fprintf(stderr, "dbus:invalid argument value\n"); + goto badmessage; + } + sset(++sargc, sargv, 0, 1); +badmessage: + dbus_message_unref(msg); + } + + if(conn) + dbus_connection_unref(conn); + return 0; +} +#endif /* HAVE_DBUS */ int main(int argc, char *argv[]) { int morehelp, level = 0; int read_stdin = 0; +#ifdef HAVE_DBUS + int read_dbus = 0; + int t; +#endif /* HAVE_DBUS */ static const struct option long_option[] = { {"help", 0, NULL, 'h'}, {"card", 1, NULL, 'c'}, +#ifdef HAVE_DBUS + {"dbus", optional_argument, NULL, OPT_DBUS}, + {"dbus-path", 1, NULL, OPT_DBUS_PATH}, + {"dbus-step", 1, NULL, OPT_DBUS_STEP}, +#endif /* HAVE_DBUS */ {"device", 1, NULL, 'D'}, {"quiet", 0, NULL, 'q'}, {"inactive", 0, NULL, 'i'}, @@ -1964,6 +2113,24 @@ int main(int argc, char *argv[]) case 's': read_stdin = 1; break; +#ifdef HAVE_DBUS + case OPT_DBUS: + if(optarg) { + strncpy(dbus_sctl, optarg, sizeof(dbus_sctl)-1); + dbus_sctl[sizeof(dbus_sctl)-1] = '\0'; + } + read_dbus = 1; + break; + case OPT_DBUS_PATH: + dbus_path=optarg; + break; + case OPT_DBUS_STEP: + t=atoi(optarg); + if(t > 0 && t <= 999999) { + snprintf(dbus_step, sizeof(dbus_step), "%d", atoi(optarg)); + } + break; +#endif /* HAVE_DBUS */ default: fprintf(stderr, "Invalid switch or option needs an argument.\n"); morehelp++; @@ -1978,6 +2145,10 @@ int main(int argc, char *argv[]) if (read_stdin) return exec_stdin(); +#ifdef HAVE_DBUS + if(read_dbus) + return exec_dbus(); +#endif /* HAVE_DBUS */ if (argc - optind <= 0) { return selems(LEVEL_BASIC | level) ? 1 : 0; } diff --git a/configure.in b/configure.in index 1349ff3..717d6c5 100644 --- a/configure.in +++ b/configure.in @@ -7,6 +7,9 @@ AM_INIT_AUTOMAKE(alsa-utils, 1.0.20) AM_GNU_GETTEXT([external]) AM_GNU_GETTEXT_VERSION([0.15]) +dnl required versions of other packages +m4_define([dbus_required_version], [1.2.0]) + dnl Checks for programs. dnl try to gues cross-compiler if not set @@ -74,6 +77,15 @@ AC_ARG_ENABLE(alsamixer, no) alsamixer=false ;; *) AC_MSG_ERROR(bad value ${enableval} for --enable-alsamixer) ;; esac],[alsamixer=true]) +AC_ARG_WITH(dbus, [ --with-dbus Enable D-Bus support in amixer]) + if test "x$with_dbus" != xno; then + PKG_CHECK_MODULES(DBUS, dbus-1 >= dbus_required_version, + have_dbus=yes, + have_dbus=no) + fi + if test "x$have_dbus" = xyes; then + AC_DEFINE(HAVE_DBUS, 1, [Define to 1 if amixer is to support D-Bus]) + fi fi AM_CONDITIONAL(ALSAMIXER, test x$alsamixer = xtrue) -- 1.6.3.3 ---------------------------------------------------------------------- Zostan Dziewczyna lub Chlopakiem lata! 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7 13

[alsa-devel] Question to sophisticated alsa developer
by Andreas Bießmann 05 Aug '09

05 Aug '09

Hi all, I currently test alsa playback on AVR32 board with atmel's abdac driver. My environment is 2.6.30.4 customized kernel (customization for my own board), rootfs and toolchain made of customized buildroot from atmel. I use alsa-lib 1.0.20 with patch [1] applied and alsa-utils with patch [2] applied. The alsa code does compile and work fine (aside from my problem described some lines later). My test case is a sine wave file with 440Hz tone, samplerate is 17089 (cause of samplerate abdac driver offers and to avoid resampling in alsa-lib) playing with aplay. This works fine, I see a 440Hz sine wave on my oscilloscope and there is no effect as described later on. My problem is when I'm enabling dmix layer to run concurrent aplay processes I always get a click at the end of my last played sine wave file. My Oscilloscope shows me that there is sometimes a short part of a single or somtimes multiple sine period(s) moved down (or up) to mid value (I can supply a screenshot of oscilloscope, if this description is not clear enough). Mostly this effect is about 500 us long. The sine wave is missing the information at this position, it is no delay of the analog value at this position. If I overlay a correct sine wave the following part after the problematic position matches exactly the overlayed sine wave. I investigated that this problem always occur somewhere at the end of dmix. When I play a long "null" sample and start repeated play of my short sine wave sample the effect is not there while the null sample is playing. After ending of null sample the described effect appears instantly on each end of my repeated test sample. Here is my asound.conf to enable dmix layer for alsa: ---8<--- pcm.!default { type plug slave.pcm "dmixer" } ctl.mixer0 { type hw card 0 } pcm.!plughw { type plug slave.pcm "dmixer" } pcm.dmixer { type dmix ipc_key 1024 slave { pcm "hw:0" format S16_BE rate 17089 } bindings { 0 0 1 1 } slowptr false } ctl.dmixer { type hw card 0 } --->8--- Maybe there is already a bloomer in. I have tried to change slowptr stuff in pcm.dmixer without any effect to my problem. I also tried to change buffer sizes in dmixer's slave but these changes had no effect. I also tried some changes in kernel driver for ABDAC. I played a bit with periods_min and period_bytes_min without effect to my problem. The buffer_size in "Direct Stream Mixing PCM" is fixed to the value in "hw:0". But if I double the periods_min in Kernel (default: 6 periods_min, 64 periods_max) for "hw:0" the maximum value for buffer_size in "Direct Stream Mixing PCM" is fixed to 8192. The "Route conversion" layer has always the buffer_size value of the next layer. This is 8192 for doubled periods_min in kernel and dmix enabled and 12288 for doubled periods_min and dmix disabled. Any advice to this "feature"? Here is the output of aplay -v for dmix enabled: ---8<--- # aplay -v sin_440_17089_1sec.wav Playing WAVE 'sin_440_17089_1sec.wav' : Signed 16 bit Little Endian, Rate 17089 Hz, Mono Plug PCM: Route conversion PCM (sformat=S16_BE) Transformation table: 0 <- 0 1 <- 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 17089 exact rate : 17089 (17089/1) msbits : 16 buffer_size : 8192 period_size : 1024 period_time : 59921 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 8192 stop_threshold : 8192 silence_threshold: 0 silence_size : 0 boundary : 1073741824 Slave: Direct Stream Mixing PCM Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_BE subformat : STD channels : 2 rate : 17089 exact rate : 17089 (17089/1) msbits : 16 buffer_size : 8192 period_size : 1024 period_time : 59921 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 8192 stop_threshold : 8192 silence_threshold: 0 silence_size : 0 boundary : 1073741824 Hardware PCM card 0 'Atmel ABDAC' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_BE subformat : STD channels : 2 rate : 17089 exact rate : 17089 (17089/1) msbits : 16 buffer_size : 12288 period_size : 1024 period_time : 59921 tstamp_mode : ENABLE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 1 stop_threshold : 1610612736 silence_threshold: 0 silence_size : 1610612736 boundary : 1610612736 appl_ptr : 0 hw_ptr : 0 --->8--- And here the output of aplay -v without dmix: ---8<--- # aplay -v sin_440_17089_1sec.wav Playing WAVE 'sin_440_17089_1sec.wav' : Signed 16 bit Little Endian, Rate 17089 Hz, Mono Plug PCM: Route conversion PCM (sformat=S16_BE) Transformation table: 0 <- 0 1 <- 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 17089 exact rate : 17089 (17089/1) msbits : 16 buffer_size : 12288 period_size : 1024 period_time : 59921 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 12288 stop_threshold : 12288 silence_threshold: 0 silence_size : 0 boundary : 1610612736 Slave: Hardware PCM card 0 'Atmel ABDAC' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_BE subformat : STD channels : 2 rate : 17089 exact rate : 17089 (17089/1) msbits : 16 buffer_size : 12288 period_size : 1024 period_time : 59921 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 12288 stop_threshold : 12288 silence_threshold: 0 silence_size : 0 boundary : 1610612736 appl_ptr : 0 hw_ptr : 0 --->8--- Well, currently I have no more ideas where to dig for my problem. I do need a mixing layer to allow two or three concurrent aplay instances at the same time but the output in this case is (atm) worse. Has anyone some advice where to search for a solution to my problem. Or has anybody some advice to use another feature of alsa to allow concurrent aplay processes? regards Andreas Bießmann [1] http://git.buildroot.net/buildroot/tree/package/multimedia/alsa-lib/alsa-li… [2] http://git.buildroot.net/buildroot/tree/package/multimedia/alsa-utils/alsa-…

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[alsa-devel] pyalsa: synchronizing queue with MIDI clock events
by Christopher Arndt 05 Aug '09

05 Aug '09

Hi everybody, this is my first post on this list, so please excuse me, should it not be the right place to post this. I started exploring programming the ALSA sequencer via the pyalsa Python interface, i.e the alsaseq module. I wrote a small utility that reads MIDI events from one port, filters/processes them and writes it to another. This allows for routing by channel, note range, controller number etc. and to filter out, replace or add MIDI events. Now to synchronize the queue tempo to incoming MIDI clock events, to be able implement synchronized delays or arpeggiators, I use the method in the code shown below. This works ok, but I notice that the measured BPM oscillates +-1 BPM around the value displayed by the clock source (my synth's sequencer) if I measure/average only a few (~10) ticks or do not round the result to an integer value. Is this the right approach? What is a sensible number of ticks to take into account for measurement? Should I use another reference timer than Python's time.time() function? Is pyalsa generally suitable for this kind of application even on older/weaker hardware (e.g. a NSLU2)? Any comments or suggestions for improvement would be very much appreciated! Chris class MidiProcessor(object): def run(self): # This is simplified to only show the general logic while True: events = self.sequencer.receive_events( timeout=RECEIVE_TIMEOUT, maxevents=1) for event in events: if event.type == SEQ_EVENT_CLOCK: self.sync_queue(event) else: # do other MIDI processing / routing def sync_queue(self, event): """Sync queue tempo to incoming MIDI clock events.""" # list to collect the timestamps of the last few ticks lt = self._last_ticks lt.append(time.time()) ltlen = len(lt) if ltlen > 1: # calculate & set bpm: calculate difference between # the times the last few ticks were received and average # all results avg_delta = sum( [y-x for x,y in zip(lt, lt[1:])]) / (ltlen-1) # tick length is a 24th of a quarter note bpm = round(60 / avg_delta / 24) if bpm != self.bpm: self.bpm = bpm self.sequencer.queue_tempo(self.queue, tempo=int(6e7 / self.bpm), ppq=self.ppq) # only remember last 24 received ticks # (length of a quarter note) if ltlen > 24: lt.pop(0)

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[alsa-devel] [PATCH 3/3] ASoC: add machine driver for i.mx27_visstrim_m10 board
by javier Martin 05 Aug '09

05 Aug '09

This adds support for i.mx27_visstrim_sm10 board machine driver which uses an i.mx27 processor plus a wm8974 codec. It has been tested on a visstrim_sm10 board. Signed-off-by: Javier Martin <javier.martin(a)vista-silicon.com> --- sound/soc/imx/Kconfig | 8 + sound/soc/imx/Makefile | 4 + sound/soc/imx/mx27vis_wm8974.c | 317 ++++++++++++++++++++++++++++++++++++++++ 3 files changed, 329 insertions(+), 0 deletions(-) diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index 886dadd..2c6f568 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -9,5 +9,13 @@ config SND_MX1_MX2_SOC config SND_MXC_SOC_SSI tristate +config SND_SOC_MX27VIS_WM8974 + tristate "SoC Audio support for MX27 - WM8974 Visstrim_sm10 board" + depends on SND_MX1_MX2_SOC && MACH_MX27 && MACH_IMX27_VISSTRIM_M10 + select SND_MXC_SOC_SSI + select SND_SOC_WM8974 + help + Say Y if you want to add support for SoC audio on Visstrim SM10 + board with WM8974. diff --git a/sound/soc/imx/Makefile b/sound/soc/imx/Makefile index 6552cb2..c2ffd2c 100644 --- a/sound/soc/imx/Makefile +++ b/sound/soc/imx/Makefile @@ -4,3 +4,7 @@ snd-soc-mxc-ssi-objs := mxc-ssi.o obj-$(CONFIG_SND_MX1_MX2_SOC) += snd-soc-mx1_mx2.o obj-$(CONFIG_SND_MXC_SOC_SSI) += snd-soc-mxc-ssi.o + +# i.MX Machine Support +snd-soc-mx27vis-wm8974-objs := mx27vis_wm8974.o +obj-$(CONFIG_SND_SOC_MX27VIS_WM8974) += snd-soc-mx27vis-wm8974.o diff --git a/sound/soc/imx/mx27vis_wm8974.c b/sound/soc/imx/mx27vis_wm8974.c new file mode 100644 index 0000000..e4dcb53 --- /dev/null +++ b/sound/soc/imx/mx27vis_wm8974.c @@ -0,0 +1,317 @@ +/* + * mx27vis_wm8974.c -- SoC audio for mx27vis + * + * Copyright 2009 Vista Silicon S.L. + * Author: Javier Martin + * javier.martin(a)vista-silicon.com + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/device.h> +#include <linux/i2c.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + + +#include "../codecs/wm8974.h" +#include "mx1_mx2-pcm.h" +#include "mxc-ssi.h" +#include <mach/gpio.h> +#include <mach/iomux.h> + +#define IGNORED_ARG 0 + + +static struct snd_soc_card mx27vis; + +/** + * This function connects SSI1 (HPCR1) as slave to + * SSI1 external signals (PPCR1) + * As slave, HPCR1 must set TFSDIR and TCLKDIR as inputs from + * port 4 + */ +void audmux_connect_1_4(void) +{ + pr_debug("AUDMUX: normal operation mode\n"); + /* Reset HPCR1 and PPCR1 */ + + DAM_HPCR1 = 0x00000000; + DAM_PPCR1 = 0x00000000; + + /* set to synchronous */ + DAM_HPCR1 |= AUDMUX_HPCR_SYN; + DAM_PPCR1 |= AUDMUX_PPCR_SYN; + + + /* set Rx sources 1 <--> 4 */ + DAM_HPCR1 |= AUDMUX_HPCR_RXDSEL(3); /* port 4 */ + DAM_PPCR1 |= AUDMUX_PPCR_RXDSEL(0); /* port 1 */ + + /* set Tx frame and Clock direction and source 4 --> 1 output */ + DAM_HPCR1 |= AUDMUX_HPCR_TFSDIR | AUDMUX_HPCR_TCLKDIR; + DAM_HPCR1 |= AUDMUX_HPCR_TFCSEL(3); /* TxDS and TxCclk from port 4 */ + + return; +} + +static int mx27vis_hifi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + unsigned int pll_out = 0, bclk = 0, fmt = 0, mclk = 0; + int ret = 0; + + /* + * The WM8974 is better at generating accurate audio clocks than the + * MX27 SSI controller, so we will use it as master when we can. + */ + switch (params_rate(params)) { + case 8000: + fmt = SND_SOC_DAIFMT_CBM_CFM; + mclk = WM8974_MCLKDIV_12; + pll_out = 24576000; + break; + case 16000: + fmt = SND_SOC_DAIFMT_CBM_CFM; + pll_out = 12288000; + break; + case 48000: + fmt = SND_SOC_DAIFMT_CBM_CFM; + bclk = WM8974_BCLKDIV_4; + pll_out = 12288000; + break; + case 96000: + fmt = SND_SOC_DAIFMT_CBM_CFM; + bclk = WM8974_BCLKDIV_2; + pll_out = 12288000; + break; + case 11025: + fmt = SND_SOC_DAIFMT_CBM_CFM; + bclk = WM8974_BCLKDIV_16; + pll_out = 11289600; + break; + case 22050: + fmt = SND_SOC_DAIFMT_CBM_CFM; + bclk = WM8974_BCLKDIV_8; + pll_out = 11289600; + break; + case 44100: + fmt = SND_SOC_DAIFMT_CBM_CFM; + bclk = WM8974_BCLKDIV_4; + mclk = WM8974_MCLKDIV_2; + pll_out = 11289600; + break; + case 88200: + fmt = SND_SOC_DAIFMT_CBM_CFM; + bclk = WM8974_BCLKDIV_2; + pll_out = 11289600; + break; + } + + /* set codec DAI configuration */ + ret = codec_dai->ops->set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_SYNC | fmt); + if (ret < 0) { + printk(KERN_ERR "Error from codec DAI configuration\n"); + return ret; + } + + /* set cpu DAI configuration */ + ret = cpu_dai->ops->set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_SYNC | fmt); + if (ret < 0) { + printk(KERN_ERR "Error from cpu DAI configuration\n"); + return ret; + } + + /* Put DC field of STCCR to 1 (not zero) */ + ret = cpu_dai->ops->set_tdm_slot(cpu_dai, 0, 2); + + /* set the SSI system clock as input */ + ret = cpu_dai->ops->set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "Error when setting system SSI clk\n"); + return ret; + } + + /* set codec BCLK division for sample rate */ + ret = codec_dai->ops->set_clkdiv(codec_dai, WM8974_BCLKDIV, bclk); + if (ret < 0) { + printk(KERN_ERR "Error when setting BCLK division\n"); + return ret; + } + + + /* codec PLL input is 25 MHz */ + ret = codec_dai->ops->set_pll(codec_dai, IGNORED_ARG, + 25000000, pll_out); + if (ret < 0) { + printk(KERN_ERR "Error when setting PLL input\n"); + return ret; + } + + /*set codec MCLK division for sample rate */ + ret = codec_dai->ops->set_clkdiv(codec_dai, WM8974_MCLKDIV, mclk); + if (ret < 0) { + printk(KERN_ERR "Error when setting MCLK division\n"); + return ret; + } + + return 0; +} + +static int mx27vis_hifi_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + + /* disable the PLL */ + return codec_dai->ops->set_pll(codec_dai, IGNORED_ARG, 0, 0); +} + +/* + * mx27vis WM8974 HiFi DAI opserations. + */ +static struct snd_soc_ops mx27vis_hifi_ops = { + .hw_params = mx27vis_hifi_hw_params, + .hw_free = mx27vis_hifi_hw_free, +}; + + +static int mx27vis_suspend(struct platform_device *pdev, pm_message_t state) +{ + return 0; +} + +static int mx27vis_resume(struct platform_device *pdev) +{ + return 0; +} + +static int mx27vis_probe(struct platform_device *pdev) +{ + int ret = 0; + + ret = get_ssi_clk(0, &pdev->dev); + + if (ret < 0) { + printk(KERN_ERR "%s: cant get ssi clock\n", __func__); + return ret; + } + + + return 0; +} + +static int mx27vis_remove(struct platform_device *pdev) +{ + put_ssi_clk(0); + return 0; +} + +static struct snd_soc_dai_link mx27vis_dai[] = { +{ /* Hifi Playback*/ + .name = "WM8974", + .stream_name = "WM8974 HiFi", + .cpu_dai = &imx_ssi_pcm_dai[0], + .codec_dai = &wm8974_dai, + .ops = &mx27vis_hifi_ops, +}, +}; + +static struct snd_soc_card mx27vis = { + .name = "mx27vis", + .platform = &mx1_mx2_soc_platform, + .probe = mx27vis_probe, + .remove = mx27vis_remove, + .suspend_pre = mx27vis_suspend, + .resume_post = mx27vis_resume, + .dai_link = mx27vis_dai, + .num_links = ARRAY_SIZE(mx27vis_dai), +}; + +static struct snd_soc_device mx27vis_snd_devdata = { + .card = &mx27vis, + .codec_dev = &soc_codec_dev_wm8974, +}; + +static struct platform_device *mx27vis_snd_device; + +/* Temporal definition of board specific behaviour */ +void gpio_ssi_active(int ssi_num) +{ + int ret = 0; + + unsigned int ssi1_pins[] = { + PC20_PF_SSI1_FS, + PC21_PF_SSI1_RXD, + PC22_PF_SSI1_TXD, + PC23_PF_SSI1_CLK, + }; + unsigned int ssi2_pins[] = { + PC24_PF_SSI2_FS, + PC25_PF_SSI2_RXD, + PC26_PF_SSI2_TXD, + PC27_PF_SSI2_CLK, + }; + if (ssi_num == 0) + ret = mxc_gpio_setup_multiple_pins(ssi1_pins, + ARRAY_SIZE(ssi1_pins), "USB OTG"); + else + ret = mxc_gpio_setup_multiple_pins(ssi2_pins, + ARRAY_SIZE(ssi2_pins), "USB OTG"); + if (ret) + printk(KERN_ERR "Error requesting ssi %x pins\n", ssi_num); +} + + +static int __init mx27vis_init(void) +{ + int ret; + + mx27vis_snd_device = platform_device_alloc("soc-audio", -1); + if (!mx27vis_snd_device) + return -ENOMEM; + + platform_set_drvdata(mx27vis_snd_device, &mx27vis_snd_devdata); + mx27vis_snd_devdata.dev = &mx27vis_snd_device->dev; + ret = platform_device_add(mx27vis_snd_device); + + if (ret) { + printk(KERN_ERR "ASoC: Platform device allocation failed\n"); + platform_device_put(mx27vis_snd_device); + } + + /* WM8974 uses SSI1 (HPCR1) via AUDMUX port 4 for audio (PPCR1) */ + gpio_ssi_active(0); + audmux_connect_1_4(); + + return ret; +} + +static void __exit mx27vis_exit(void) +{ + /* We should call some "ssi_gpio_inactive()" properly */ +} + +module_init(mx27vis_init); +module_exit(mx27vis_exit); + + +MODULE_AUTHOR("Javier Martin, javier.martin(a)vista-silicon.com"); +MODULE_DESCRIPTION("ALSA SoC WM8974 mx27vis"); +MODULE_LICENSE("GPL"); --- -- Javier Martin Vista Silicon S.L. Universidad de Cantabria CDTUC - FASE C - Oficina S-345 Avda de los Castros s/n 39005- Santander. Cantabria. Spain +34 942 25 32 60 www.vista-silicon.com

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[alsa-devel] [PATCH 1/6] pulse: replace manual mainloop by pa_mainloop_iterate()
by Lennart Poettering 05 Aug '09

05 Aug '09

The pa_mainloop_prepare()/_poll()/_dispatch() can be simplified by simply calling pa_mainloop_iterate() which does all this in one call. --- pulse/conf_pulse.c | 10 +--------- 1 files changed, 1 insertions(+), 9 deletions(-) diff --git a/pulse/conf_pulse.c b/pulse/conf_pulse.c index 3f9e071..35dd383 100644 --- a/pulse/conf_pulse.c +++ b/pulse/conf_pulse.c @@ -61,15 +61,7 @@ conf_pulse_hook_load_if_running(snd_config_t * root, snd_config_t * config, goto out; do { - err = pa_mainloop_prepare(loop, -1); - if (err < 0) - goto out; - - err = pa_mainloop_poll(loop); - if (err < 0) - goto out; - - err = pa_mainloop_dispatch(loop); + err = pa_mainloop_iterate(loop, 1, NULL); if (err < 0) goto out; -- 1.6.4 Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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[alsa-devel] [PATCH 6/6] pulse: immediately trigger EIO when connection is dropped
by Lennart Poettering 05 Aug '09

05 Aug '09

When the connection is dropped notify the application immediately instead of waiting until the applications calls into us the next time. This makes "aplay" handle connections shutdown similar to hardware unplugs: an immediate EIO is thrown. --- pulse/pulse.c | 9 ++++++++- 1 files changed, 8 insertions(+), 1 deletions(-) diff --git a/pulse/pulse.c b/pulse/pulse.c index 9c05171..a94caaf 100644 --- a/pulse/pulse.c +++ b/pulse/pulse.c @@ -77,10 +77,17 @@ int pulse_wait_operation(snd_pulse_t * p, pa_operation * o) static void context_state_cb(pa_context * c, void *userdata) { + pa_context_state_t state; snd_pulse_t *p = userdata; assert(c); - switch (pa_context_get_state(c)) { + state = pa_context_get_state(c); + + /* When we get disconnected, tell the process */x + if (!PA_CONTEXT_IS_GOOD(state)) + pulse_poll_activate(p); + + switch (state) { case PA_CONTEXT_READY: case PA_CONTEXT_TERMINATED: case PA_CONTEXT_FAILED: -- 1.6.4 Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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[alsa-devel] [PATCH 5/6] pulse: rework object destruction paths a bit
by Lennart Poettering 05 Aug '09

05 Aug '09

Make sure we deal better with partially initialized structs. Don't check for pointer state before calling free() since free() does that anyway. --- pulse/ctl_pulse.c | 7 ++----- pulse/pcm_pulse.c | 19 ++++++++++--------- 2 files changed, 12 insertions(+), 14 deletions(-) diff --git a/pulse/ctl_pulse.c b/pulse/ctl_pulse.c index 879f260..85863ed 100644 --- a/pulse/ctl_pulse.c +++ b/pulse/ctl_pulse.c @@ -606,11 +606,8 @@ static void pulse_close(snd_ctl_ext_t * ext) if (ctl->p) pulse_free(ctl->p); - if (ctl->source) - free(ctl->source); - if (ctl->sink) - free(ctl->sink); - + free(ctl->source); + free(ctl->sink); free(ctl); } diff --git a/pulse/pcm_pulse.c b/pulse/pcm_pulse.c index 98983f8..02a837e 100644 --- a/pulse/pcm_pulse.c +++ b/pulse/pcm_pulse.c @@ -819,21 +819,22 @@ static int pulse_close(snd_pcm_ioplug_t * io) assert(pcm); - pa_threaded_mainloop_lock(pcm->p->mainloop); + if (pcm->p && pcm->p->mainloop) { - if (pcm->stream) { - pa_stream_disconnect(pcm->stream); - pa_stream_unref(pcm->stream); - } + pa_threaded_mainloop_lock(pcm->p->mainloop); - pa_threaded_mainloop_unlock(pcm->p->mainloop); + if (pcm->stream) { + pa_stream_disconnect(pcm->stream); + pa_stream_unref(pcm->stream); + } + + pa_threaded_mainloop_unlock(pcm->p->mainloop); + } if (pcm->p) pulse_free(pcm->p); - if (pcm->device) - free(pcm->device); - + free(pcm->device); free(pcm); return 0; -- 1.6.4 Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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[alsa-devel] [PATCH 4/6] pulse: unify stream/context state checks
by Lennart Poettering 05 Aug '09

05 Aug '09

Unify (and simplify) the paths that check for the validity of a stream/context: always call into check_stream()/pulse_check_connection() when applicable instead of rolling our own checks each time. Also check for validity of mainloop before locking it. --- pulse/ctl_pulse.c | 72 ++++++++++++++++++++++++++++++++---------- pulse/pcm_pulse.c | 90 +++++++++++++++++++++++----------------------------- 2 files changed, 95 insertions(+), 67 deletions(-) diff --git a/pulse/ctl_pulse.c b/pulse/ctl_pulse.c index 2caa29b..879f260 100644 --- a/pulse/ctl_pulse.c +++ b/pulse/ctl_pulse.c @@ -150,9 +150,13 @@ static int pulse_update_volume(snd_ctl_pulse_t * ctl) assert(ctl); - if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + if (!ctl->p) return -EBADFD; + err = pulse_check_connection(ctl->p); + if (err < 0) + return err; + o = pa_context_get_sink_info_by_name(ctl->p->context, ctl->sink, sink_info_cb, ctl); if (o) { @@ -182,17 +186,27 @@ static int pulse_update_volume(snd_ctl_pulse_t * ctl) static int pulse_elem_count(snd_ctl_ext_t * ext) { snd_ctl_pulse_t *ctl = ext->private_data; - int count = 0; + int count = 0, err; assert(ctl); + if (!ctl->p || !ctl->p->mainloop) + return -EBADFD; + pa_threaded_mainloop_lock(ctl->p->mainloop); + err = pulse_check_connection(ctl->p); + if (err < 0) { + count = err; + goto finish; + } + if (ctl->source) count += 2; if (ctl->sink) count += 2; +finish: pa_threaded_mainloop_unlock(ctl->p->mainloop); return count; @@ -202,16 +216,21 @@ static int pulse_elem_list(snd_ctl_ext_t * ext, unsigned int offset, snd_ctl_elem_id_t * id) { snd_ctl_pulse_t *ctl = ext->private_data; + int err; assert(ctl); - if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + if (!ctl->p || !ctl->p->mainloop) return -EBADFD; snd_ctl_elem_id_set_interface(id, SND_CTL_ELEM_IFACE_MIXER); pa_threaded_mainloop_lock(ctl->p->mainloop); + err = pulse_check_connection(ctl->p); + if (err < 0) + goto finish; + if (ctl->source) { if (offset == 0) snd_ctl_elem_id_set_name(id, SOURCE_VOL_NAME); @@ -220,14 +239,19 @@ static int pulse_elem_list(snd_ctl_ext_t * ext, unsigned int offset, } else offset += 2; + err = 0; + +finish: pa_threaded_mainloop_unlock(ctl->p->mainloop); - if (offset == 2) - snd_ctl_elem_id_set_name(id, SINK_VOL_NAME); - else if (offset == 3) - snd_ctl_elem_id_set_name(id, SINK_MUTE_NAME); + if (err >= 0) { + if (offset == 2) + snd_ctl_elem_id_set_name(id, SINK_VOL_NAME); + else if (offset == 3) + snd_ctl_elem_id_set_name(id, SINK_MUTE_NAME); + } - return 0; + return err; } static snd_ctl_ext_key_t pulse_find_elem(snd_ctl_ext_t * ext, @@ -266,7 +290,7 @@ static int pulse_get_attribute(snd_ctl_ext_t * ext, snd_ctl_ext_key_t key, assert(ctl); - if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + if (!ctl->p || !ctl->p->mainloop) return -EBADFD; pa_threaded_mainloop_lock(ctl->p->mainloop); @@ -319,7 +343,7 @@ static int pulse_read_integer(snd_ctl_ext_t * ext, snd_ctl_ext_key_t key, assert(ctl); - if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + if (!ctl->p || !ctl->p->mainloop) return -EBADFD; pa_threaded_mainloop_lock(ctl->p->mainloop); @@ -371,7 +395,7 @@ static int pulse_write_integer(snd_ctl_ext_t * ext, snd_ctl_ext_key_t key, assert(ctl); - if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + if (!ctl->p || !ctl->p->mainloop) return -EBADFD; pa_threaded_mainloop_lock(ctl->p->mainloop); @@ -476,7 +500,7 @@ static void pulse_subscribe_events(snd_ctl_ext_t * ext, int subscribe) assert(ctl); - if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + if (!ctl->p || !ctl->p->mainloop) return; pa_threaded_mainloop_lock(ctl->p->mainloop); @@ -491,17 +515,23 @@ static int pulse_read_event(snd_ctl_ext_t * ext, snd_ctl_elem_id_t * id, { snd_ctl_pulse_t *ctl = ext->private_data; int offset; - int err = -EAGAIN; + int err; assert(ctl); - if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + if (!ctl->p || !ctl->p->mainloop) return -EBADFD; pa_threaded_mainloop_lock(ctl->p->mainloop); - if (!ctl->updated || !ctl->subscribed) + err = pulse_check_connection(ctl->p); + if (err < 0) + goto finish; + + if (!ctl->updated || !ctl->subscribed) { + err = -EAGAIN; goto finish; + } if (ctl->source) offset = 2; @@ -540,20 +570,28 @@ static int pulse_ctl_poll_revents(snd_ctl_ext_t * ext, struct pollfd *pfd, unsigned short *revents) { snd_ctl_pulse_t *ctl = ext->private_data; - int err = 0; + int err; assert(ctl); - if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + if (!ctl->p || !ctl->p->mainloop) return -EBADFD; pa_threaded_mainloop_lock(ctl->p->mainloop); + err = pulse_check_connection(ctl->p); + if (err < 0) + goto finish; + if (ctl->updated) *revents = POLLIN; else *revents = 0; + err = 0; + +finish: + pa_threaded_mainloop_unlock(ctl->p->mainloop); return err; diff --git a/pulse/pcm_pulse.c b/pulse/pcm_pulse.c index a625f55..98983f8 100644 --- a/pulse/pcm_pulse.c +++ b/pulse/pcm_pulse.c @@ -135,11 +135,15 @@ static int update_active(snd_pcm_pulse_t *pcm) { if (!pcm->p) return -EBADFD; - ret = check_active(pcm); + ret = check_stream(pcm); if (ret < 0) - return ret; + goto finish; + + ret = check_active(pcm); - if (ret > 0) +finish: + + if (ret != 0) /* On error signal the caller, too */ pulse_poll_activate(pcm->p); else pulse_poll_deactivate(pcm->p); @@ -199,12 +203,12 @@ static int pulse_start(snd_pcm_ioplug_t * io) assert(pcm); - if (!pcm->p) + if (!pcm->p || !pcm->p->mainloop) return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); - err = pulse_check_connection(pcm->p); + err = check_stream(pcm); if (err < 0) goto finish; @@ -244,12 +248,12 @@ static int pulse_stop(snd_pcm_ioplug_t * io) assert(pcm); - if (!pcm->p) + if (!pcm->p || !pcm->p->mainloop) return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); - err = pulse_check_connection(pcm->p); + err = check_stream(pcm); if (err < 0) goto finish; @@ -291,12 +295,12 @@ static int pulse_drain(snd_pcm_ioplug_t * io) assert(pcm); - if (!pcm->p) + if (!pcm->p || !pcm->p->mainloop) return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); - err = pulse_check_connection(pcm->p); + err = check_stream(pcm); if (err < 0) goto finish; @@ -328,7 +332,7 @@ static snd_pcm_sframes_t pulse_pointer(snd_pcm_ioplug_t * io) assert(pcm); - if (!pcm->p) + if (!pcm->p || !pcm->p->mainloop) return -EBADFD; if (io->state == SND_PCM_STATE_XRUN) @@ -339,12 +343,7 @@ static snd_pcm_sframes_t pulse_pointer(snd_pcm_ioplug_t * io) pa_threaded_mainloop_lock(pcm->p->mainloop); - if (!pcm->stream) { - ret = -EBADFD; - goto finish; - } - - ret = pulse_check_connection(pcm->p); + ret = check_stream(pcm); if (ret < 0) goto finish; @@ -379,18 +378,13 @@ static int pulse_delay(snd_pcm_ioplug_t * io, snd_pcm_sframes_t * delayp) assert(pcm); - if (!pcm->p) + if (!pcm->p || !pcm->p->mainloop) return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); - if (!pcm->stream) { - err = -EBADFD; - goto finish; - } - for (;;) { - err = pulse_check_connection(pcm->p); + err = check_stream(pcm); if (err < 0) goto finish; @@ -433,17 +427,12 @@ static snd_pcm_sframes_t pulse_write(snd_pcm_ioplug_t * io, assert(pcm); - if (!pcm->p) + if (!pcm->p || !pcm->p->mainloop) return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); - if (!pcm->stream) { - ret = -EBADFD; - goto finish; - } - - ret = pulse_check_connection(pcm->p); + ret = check_stream(pcm); if (ret < 0) goto finish; @@ -493,17 +482,12 @@ static snd_pcm_sframes_t pulse_read(snd_pcm_ioplug_t * io, assert(pcm); - if (!pcm->p) + if (!pcm->p || !pcm->p->mainloop) return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); - if (!pcm->stream) { - ret = -EBADFD; - goto finish; - } - - ret = pulse_check_connection(pcm->p); + ret = check_stream(pcm); if (ret < 0) goto finish; @@ -624,13 +608,16 @@ static int pulse_pcm_poll_revents(snd_pcm_ioplug_t * io, assert(pcm); - if (!pcm->p) + if (!pcm->p || !pcm->p->mainloop) return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); - err = check_active(pcm); + err = check_stream(pcm); + if (err < 0) + goto finish; + err = check_active(pcm); if (err < 0) goto finish; @@ -655,7 +642,7 @@ static int pulse_prepare(snd_pcm_ioplug_t * io) assert(pcm); - if (!pcm->p) + if (!pcm->p || !pcm->p->mainloop) return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); @@ -757,7 +744,7 @@ static int pulse_hw_params(snd_pcm_ioplug_t * io, assert(pcm); - if (!pcm->p) + if (!pcm->p || !pcm->p->mainloop) return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); @@ -856,23 +843,26 @@ static int pulse_pause(snd_pcm_ioplug_t * io, int enable) { snd_pcm_pulse_t *pcm = io->private_data; int err = 0; + pa_operation *o; assert (pcm); - if (!pcm->p) + if (!pcm->p || !pcm->p->mainloop) return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); - if (pcm->stream) { - pa_operation *o; - o = pa_stream_cork(pcm->stream, enable, NULL, NULL); - if (o) - pa_operation_unref(o); - else - err = -EIO; - } + err = check_stream(pcm); + if (err < 0) + goto finish; + + o = pa_stream_cork(pcm->stream, enable, NULL, NULL); + if (o) + pa_operation_unref(o); + else + err = -EIO; +finish: pa_threaded_mainloop_unlock(pcm->p->mainloop); return err; -- 1.6.4 Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

1 0

[alsa-devel] [PATCH 3/6] pulse: get rid of redundant state variable
by Lennart Poettering 05 Aug '09

05 Aug '09

snd_pulse_t::state was mostly shadowing the state of pa_context_get_state(snd_pulse_t::context), so get rid of it and use the state of the context directly. --- pulse/pulse.c | 14 +++----------- pulse/pulse.h | 6 ------ 2 files changed, 3 insertions(+), 17 deletions(-) diff --git a/pulse/pulse.c b/pulse/pulse.c index c313182..9c05171 100644 --- a/pulse/pulse.c +++ b/pulse/pulse.c @@ -59,12 +59,6 @@ int pulse_wait_operation(snd_pulse_t * p, pa_operation * o) assert(p); assert(o); - if (p->state != PULSE_STATE_READY) - return -EBADFD; - - if (!p->mainloop) - return -EBADFD; - for (;;) { int err; @@ -124,8 +118,6 @@ snd_pulse_t *pulse_new(void) if (!p) return NULL; - p->state = PULSE_STATE_INIT; - if (pipe(fd)) { free(p); return NULL; @@ -192,13 +184,15 @@ void pulse_free(snd_pulse_t * p) int pulse_connect(snd_pulse_t * p, const char *server) { int err; + pa_context_state_t state; assert(p); if (!p->context || !p->mainloop) return -EBADFD; - if (p->state != PULSE_STATE_INIT) + state = pa_context_get_state(p->context); + if (state != PA_CONTEXT_UNCONNECTED) return -EBADFD; pa_threaded_mainloop_lock(p->mainloop); @@ -221,8 +215,6 @@ int pulse_connect(snd_pulse_t * p, const char *server) pa_threaded_mainloop_unlock(p->mainloop); - p->state = PULSE_STATE_READY; - return 0; error: diff --git a/pulse/pulse.h b/pulse/pulse.h index 51f9a11..e98124f 100644 --- a/pulse/pulse.h +++ b/pulse/pulse.h @@ -31,12 +31,6 @@ typedef struct snd_pulse { pa_context *context; int thread_fd, main_fd; - - enum { - PULSE_STATE_INIT, - PULSE_STATE_READY, - } state; - } snd_pulse_t; int pulse_check_connection(snd_pulse_t * p); -- 1.6.4 Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

1 0

[alsa-devel] [PATCH 2/6] pulse: move a couple of PCM related functions from pulse.c to pcm_pulse.c
by Lennart Poettering 05 Aug '09

05 Aug '09

A number of functions in pulse.c are only relevant for the PCM handling, so let's move them to pcm_pulse.c. This allows us to simplify their argument lists a bit. --- pulse/pcm_pulse.c | 118 ++++++++++++++++++++++++++++++++++++++++++---------- pulse/pulse.c | 55 ------------------------- pulse/pulse.h | 4 -- 3 files changed, 95 insertions(+), 82 deletions(-) diff --git a/pulse/pcm_pulse.c b/pulse/pcm_pulse.c index 24347f9..a625f55 100644 --- a/pulse/pcm_pulse.c +++ b/pulse/pcm_pulse.c @@ -49,6 +49,32 @@ typedef struct snd_pcm_pulse { pa_buffer_attr buffer_attr; } snd_pcm_pulse_t; +static int check_stream(snd_pcm_pulse_t *pcm) +{ + int err; + pa_stream_state_t state; + + assert(pcm); + + if (!pcm->p) + return -EBADFD; + + err = pulse_check_connection(pcm->p); + if (err < 0) + return err; + + if (!pcm->stream) + return -EBADFD; + + state = pa_stream_get_state(pcm->stream); + if (!PA_STREAM_IS_GOOD(state)) + return -EIO; + + err = 0; + + return err; +} + static int update_ptr(snd_pcm_pulse_t *pcm) { size_t size; @@ -118,9 +144,53 @@ static int update_active(snd_pcm_pulse_t *pcm) { else pulse_poll_deactivate(pcm->p); + return ret; +} + +static int wait_stream_state(snd_pcm_pulse_t *pcm, pa_stream_state_t target) +{ + pa_stream_state_t state; + + assert(pcm); + + if (!pcm->p) + return -EBADFD; + + for (;;) { + int err; + + err = pulse_check_connection(pcm->p); + if (err < 0) + return err; + + if (!pcm->stream) + return -EBADFD; + + state = pa_stream_get_state(pcm->stream); + if (state == target) + break; + + if (!PA_STREAM_IS_GOOD(state)) + return -EIO; + + pa_threaded_mainloop_wait(pcm->p->mainloop); + } + return 0; } +static void stream_success_cb(pa_stream * p, int success, void *userdata) +{ + snd_pcm_pulse_t *pcm = userdata; + + assert(pcm); + + if (!pcm->p) + return; + + pa_threaded_mainloop_signal(pcm->p->mainloop, 0); +} + static int pulse_start(snd_pcm_ioplug_t * io) { snd_pcm_pulse_t *pcm = io->private_data; @@ -138,18 +208,13 @@ static int pulse_start(snd_pcm_ioplug_t * io) if (err < 0) goto finish; - if (pcm->stream == NULL) - goto finish; - - o = pa_stream_cork(pcm->stream, 0, pulse_stream_success_cb, - pcm->p); + o = pa_stream_cork(pcm->stream, 0, stream_success_cb, pcm); if (!o) { err = -EIO; goto finish; } - u = pa_stream_trigger(pcm->stream, pulse_stream_success_cb, - pcm->p); + u = pa_stream_trigger(pcm->stream, stream_success_cb, pcm); pcm->underrun = 0; err_o = pulse_wait_operation(pcm->p, o); @@ -165,7 +230,6 @@ static int pulse_start(snd_pcm_ioplug_t * io) goto finish; } - finish: pa_threaded_mainloop_unlock(pcm->p->mainloop); @@ -189,18 +253,13 @@ static int pulse_stop(snd_pcm_ioplug_t * io) if (err < 0) goto finish; - if (pcm->stream == NULL) - goto finish; - - o = pa_stream_cork(pcm->stream, 1, pulse_stream_success_cb, - pcm->p); + o = pa_stream_cork(pcm->stream, 1, stream_success_cb, pcm); if (!o) { err = -EIO; goto finish; } - u = pa_stream_flush(pcm->stream, pulse_stream_success_cb, - pcm->p); + u = pa_stream_flush(pcm->stream, stream_success_cb, pcm); if (!u) { pa_operation_unref(o); err = -EIO; @@ -241,7 +300,7 @@ static int pulse_drain(snd_pcm_ioplug_t * io) if (err < 0) goto finish; - o = pa_stream_drain(pcm->stream, pulse_stream_success_cb, pcm->p); + o = pa_stream_drain(pcm->stream, stream_success_cb, pcm); if (!o) { err = -EIO; goto finish; @@ -504,6 +563,23 @@ finish: return ret; } +static void stream_state_cb(pa_stream * p, void *userdata) +{ + snd_pcm_pulse_t *pcm = userdata; + pa_stream_state_t state; + + assert(pcm); + + if (!pcm->p) + return; + + state = pa_stream_get_state(p); + if (!PA_STREAM_IS_GOOD(state)) + pulse_poll_activate(pcm->p); + + pa_threaded_mainloop_signal(pcm->p->mainloop, 0); +} + static void stream_request_cb(pa_stream * p, size_t length, void *userdata) { snd_pcm_pulse_t *pcm = userdata; @@ -586,8 +662,7 @@ static int pulse_prepare(snd_pcm_ioplug_t * io) if (pcm->stream) { pa_stream_disconnect(pcm->stream); - pulse_wait_stream_state(pcm->p, pcm->stream, - PA_STREAM_TERMINATED); + wait_stream_state(pcm, PA_STREAM_TERMINATED); pa_stream_unref(pcm->stream); pcm->stream = NULL; } @@ -620,9 +695,7 @@ static int pulse_prepare(snd_pcm_ioplug_t * io) goto finish; } - pa_stream_set_state_callback(pcm->stream, pulse_stream_state_cb, - pcm->p); - + pa_stream_set_state_callback(pcm->stream, stream_state_cb, pcm); pa_stream_set_latency_update_callback(pcm->stream, stream_latency_cb, pcm); if (io->stream == SND_PCM_STREAM_PLAYBACK) { @@ -659,8 +732,7 @@ static int pulse_prepare(snd_pcm_ioplug_t * io) goto finish; } - err = - pulse_wait_stream_state(pcm->p, pcm->stream, PA_STREAM_READY); + err = wait_stream_state(pcm, PA_STREAM_READY); if (err < 0) { SNDERR("PulseAudio: Unable to create stream: %s\n", pa_strerror(pa_context_errno(pcm->p->context))); pa_stream_unref(pcm->stream); diff --git a/pulse/pulse.c b/pulse/pulse.c index 6f58a7e..c313182 100644 --- a/pulse/pulse.c +++ b/pulse/pulse.c @@ -44,26 +44,6 @@ int pulse_check_connection(snd_pulse_t * p) return 0; } -void pulse_stream_state_cb(pa_stream * s, void *userdata) -{ - snd_pulse_t *p = userdata; - - assert(s); - assert(p); - - pa_threaded_mainloop_signal(p->mainloop, 0); -} - -void pulse_stream_success_cb(pa_stream * s, int success, void *userdata) -{ - snd_pulse_t *p = userdata; - - assert(s); - assert(p); - - pa_threaded_mainloop_signal(p->mainloop, 0); -} - void pulse_context_success_cb(pa_context * c, int success, void *userdata) { snd_pulse_t *p = userdata; @@ -101,41 +81,6 @@ int pulse_wait_operation(snd_pulse_t * p, pa_operation * o) return 0; } -int pulse_wait_stream_state(snd_pulse_t * p, pa_stream * stream, - pa_stream_state_t target) -{ - pa_stream_state_t state; - - assert(p); - assert(stream); - - if (p->state != PULSE_STATE_READY) - return -EBADFD; - - if (!p->mainloop) - return -EBADFD; - - for (;;) { - int err; - - err = pulse_check_connection(p); - if (err < 0) - return err; - - state = pa_stream_get_state(stream); - - if (state == target) - break; - - if (!PA_STREAM_IS_GOOD(state)) - return -EIO; - - pa_threaded_mainloop_wait(p->mainloop); - } - - return 0; -} - static void context_state_cb(pa_context * c, void *userdata) { snd_pulse_t *p = userdata; diff --git a/pulse/pulse.h b/pulse/pulse.h index 7bf1a5b..51f9a11 100644 --- a/pulse/pulse.h +++ b/pulse/pulse.h @@ -41,13 +41,9 @@ typedef struct snd_pulse { int pulse_check_connection(snd_pulse_t * p); -void pulse_stream_state_cb(pa_stream * s, void *userdata); -void pulse_stream_success_cb(pa_stream * s, int success, void *userdata); void pulse_context_success_cb(pa_context * c, int success, void *userdata); int pulse_wait_operation(snd_pulse_t * p, pa_operation * o); -int pulse_wait_stream_state(snd_pulse_t * p, pa_stream * stream, - pa_stream_state_t target); snd_pulse_t *pulse_new(void); void pulse_free(snd_pulse_t * p); -- 1.6.4 Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

1 0

[alsa-devel] [PATCH 2/3] ASoC: add DAI platform ssi driver for MXC
by javier Martin 04 Aug '09

04 Aug '09

This adds support for DAI platform for the SSI present in MXC platforms. It currently does not support i.MX3, the only thing necessary to do this is to export DMA data for i.MX3 interface which I haven't done because I don't have a i.MX3 based board available. It has been tested on i.MX27 board. Signed-off-by: Javier Martin <javier.martin(a)vista-silicon.com> --- sound/soc/imx/Kconfig | 2 + sound/soc/imx/Makefile | 2 + sound/soc/imx/mxc-ssi.c | 868 +++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/imx/mxc-ssi.h | 238 +++++++++++++ 4 files changed, 1110 insertions(+), 0 deletions(-) diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index a1bf053..886dadd 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -6,6 +6,8 @@ config SND_MX1_MX2_SOC Say Y or M if you want to add support for codecs attached to the MX1 or MX2 SSI interface. +config SND_MXC_SOC_SSI + tristate diff --git a/sound/soc/imx/Makefile b/sound/soc/imx/Makefile index c390f0f..6552cb2 100644 --- a/sound/soc/imx/Makefile +++ b/sound/soc/imx/Makefile @@ -1,4 +1,6 @@ # i.MX Platform Support snd-soc-mx1_mx2-objs := mx1_mx2-pcm.o +snd-soc-mxc-ssi-objs := mxc-ssi.o obj-$(CONFIG_SND_MX1_MX2_SOC) += snd-soc-mx1_mx2.o +obj-$(CONFIG_SND_MXC_SOC_SSI) += snd-soc-mxc-ssi.o diff --git a/sound/soc/imx/mxc-ssi.c b/sound/soc/imx/mxc-ssi.c new file mode 100644 index 0000000..3806ff2 --- /dev/null +++ b/sound/soc/imx/mxc-ssi.c @@ -0,0 +1,868 @@ +/* + * mxc-ssi.c -- SSI driver for Freescale IMX + * + * Copyright 2006 Wolfson Microelectronics PLC. + * Author: Liam Girdwood + * liam.girdwood(a)wolfsonmicro.com or linux(a)wolfsonmicro.com + * + * Based on mxc-alsa-mc13783 (C) 2006 Freescale. + * + * 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. + * + * TODO: + * Need to rework SSI register defs when new defs go into mainline. + * Add support for TDM and FIFO 1. + * Add support for i.mx3x DMA interface. + * + */ + + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/dma-mapping.h> +#include <linux/clk.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <mach/dma-mx1-mx2.h> +#include <asm/mach-types.h> + +#include "mxc-ssi.h" +#include "mx1_mx2-pcm.h" + +#define SSI1_PORT 0 +#define SSI2_PORT 1 + +static int ssi_active[2] = {0, 0}; + +/* DMA information for mx1_mx2 platforms */ +static struct mx1_mx2_pcm_dma_params imx_ssi1_pcm_stereo_out0 = { + .name = "SSI1 PCM Stereo out 0", + .transfer_type = DMA_MODE_WRITE, + .per_address = SSI1_BASE_ADDR + STX0, + .event_id = DMA_REQ_SSI1_TX0, + .watermark_level = TXFIFO_WATERMARK, + .per_config = IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO, + .mem_config = IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR, +}; + +static struct mx1_mx2_pcm_dma_params imx_ssi1_pcm_stereo_out1 = { + .name = "SSI1 PCM Stereo out 1", + .transfer_type = DMA_MODE_WRITE, + .per_address = SSI1_BASE_ADDR + STX1, + .event_id = DMA_REQ_SSI1_TX1, + .watermark_level = TXFIFO_WATERMARK, + .per_config = IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO, + .mem_config = IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR, +}; + +static struct mx1_mx2_pcm_dma_params imx_ssi1_pcm_stereo_in0 = { + .name = "SSI1 PCM Stereo in 0", + .transfer_type = DMA_MODE_READ, + .per_address = SSI1_BASE_ADDR + SRX0, + .event_id = DMA_REQ_SSI1_RX0, + .watermark_level = RXFIFO_WATERMARK, + .per_config = IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO, + .mem_config = IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR, +}; + +static struct mx1_mx2_pcm_dma_params imx_ssi1_pcm_stereo_in1 = { + .name = "SSI1 PCM Stereo in 1", + .transfer_type = DMA_MODE_READ, + .per_address = SSI1_BASE_ADDR + SRX1, + .event_id = DMA_REQ_SSI1_RX1, + .watermark_level = RXFIFO_WATERMARK, + .per_config = IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO, + .mem_config = IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR, +}; + +static struct mx1_mx2_pcm_dma_params imx_ssi2_pcm_stereo_out0 = { + .name = "SSI2 PCM Stereo out 0", + .transfer_type = DMA_MODE_WRITE, + .per_address = SSI2_BASE_ADDR + STX0, + .event_id = DMA_REQ_SSI2_TX0, + .watermark_level = TXFIFO_WATERMARK, + .per_config = IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO, + .mem_config = IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR, +}; + +static struct mx1_mx2_pcm_dma_params imx_ssi2_pcm_stereo_out1 = { + .name = "SSI2 PCM Stereo out 1", + .transfer_type = DMA_MODE_WRITE, + .per_address = SSI2_BASE_ADDR + STX1, + .event_id = DMA_REQ_SSI2_TX1, + .watermark_level = TXFIFO_WATERMARK, + .per_config = IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO, + .mem_config = IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR, +}; + +static struct mx1_mx2_pcm_dma_params imx_ssi2_pcm_stereo_in0 = { + .name = "SSI2 PCM Stereo in 0", + .transfer_type = DMA_MODE_READ, + .per_address = SSI2_BASE_ADDR + SRX0, + .event_id = DMA_REQ_SSI2_RX0, + .watermark_level = RXFIFO_WATERMARK, + .per_config = IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO, + .mem_config = IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR, +}; + +static struct mx1_mx2_pcm_dma_params imx_ssi2_pcm_stereo_in1 = { + .name = "SSI2 PCM Stereo in 1", + .transfer_type = DMA_MODE_READ, + .per_address = SSI2_BASE_ADDR + SRX1, + .event_id = DMA_REQ_SSI2_RX1, + .watermark_level = RXFIFO_WATERMARK, + .per_config = IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO, + .mem_config = IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR, +}; + +static struct clk *ssi_clk0, *ssi_clk1; + +int get_ssi_clk(int ssi, struct device *dev) +{ + switch (ssi) { + case 0: + ssi_clk0 = clk_get(dev, "ssi1"); + if (IS_ERR(ssi_clk0)) + return PTR_ERR(ssi_clk0); + return 0; + case 1: + ssi_clk1 = clk_get(dev, "ssi2"); + if (IS_ERR(ssi_clk1)) + return PTR_ERR(ssi_clk1); + return 0; + default: + return -EINVAL; + } +} +EXPORT_SYMBOL(get_ssi_clk); + +void put_ssi_clk(int ssi) +{ + switch (ssi) { + case 0: + clk_put(ssi_clk0); + ssi_clk0 = NULL; + break; + case 1: + clk_put(ssi_clk1); + ssi_clk1 = NULL; + break; + } +} +EXPORT_SYMBOL(put_ssi_clk); + +/* + * SSI system clock configuration. + * Should only be called when port is inactive (i.e. SSIEN = 0). + */ +static int imx_ssi_set_dai_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + u32 scr; + + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { + scr = SSI1_SCR; + pr_debug("%s: SCR for SSI1 is %x\n", __func__, scr); + } else { + scr = SSI2_SCR; + pr_debug("%s: SCR for SSI2 is %x\n", __func__, scr); + } + + if (scr & SSI_SCR_SSIEN) { + printk(KERN_WARNING "Warning ssi already enabled\n"); + return 0; + } + + switch (clk_id) { + case IMX_SSP_SYS_CLK: + if (dir == SND_SOC_CLOCK_OUT) { + scr |= SSI_SCR_SYS_CLK_EN; + pr_debug("%s: clk of is output\n", __func__); + } else { + scr &= ~SSI_SCR_SYS_CLK_EN; + pr_debug("%s: clk of is input\n", __func__); + } + break; + default: + return -EINVAL; + } + + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { + pr_debug("%s: writeback of SSI1_SCR\n", __func__); + SSI1_SCR = scr; + } else { + pr_debug("%s: writeback of SSI2_SCR\n", __func__); + SSI2_SCR = scr; + } + + return 0; +} + +/* + * SSI Clock dividers + * Should only be called when port is inactive (i.e. SSIEN = 0). + */ +static int imx_ssi_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + u32 stccr, srccr; + + pr_debug("%s\n", __func__); + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { + if (SSI1_SCR & SSI_SCR_SSIEN) + return 0; + srccr = SSI1_STCCR; + stccr = SSI1_STCCR; + } else { + if (SSI2_SCR & SSI_SCR_SSIEN) + return 0; + srccr = SSI2_STCCR; + stccr = SSI2_STCCR; + } + + switch (div_id) { + case IMX_SSI_TX_DIV_2: + stccr &= ~SSI_STCCR_DIV2; + stccr |= div; + break; + case IMX_SSI_TX_DIV_PSR: + stccr &= ~SSI_STCCR_PSR; + stccr |= div; + break; + case IMX_SSI_TX_DIV_PM: + stccr &= ~0xff; + stccr |= SSI_STCCR_PM(div); + break; + case IMX_SSI_RX_DIV_2: + stccr &= ~SSI_STCCR_DIV2; + stccr |= div; + break; + case IMX_SSI_RX_DIV_PSR: + stccr &= ~SSI_STCCR_PSR; + stccr |= div; + break; + case IMX_SSI_RX_DIV_PM: + stccr &= ~0xff; + stccr |= SSI_STCCR_PM(div); + break; + default: + return -EINVAL; + } + + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { + SSI1_STCCR = stccr; + SSI1_SRCCR = srccr; + } else { + SSI2_STCCR = stccr; + SSI2_SRCCR = srccr; + } + return 0; +} + +/* + * SSI Network Mode or TDM slots configuration. + * Should only be called when port is inactive (i.e. SSIEN = 0). + */ +static int imx_ssi_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, + unsigned int mask, int slots) +{ + u32 stmsk, srmsk, stccr; + + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { + if (SSI1_SCR & SSI_SCR_SSIEN) { + printk(KERN_WARNING "Warning ssi already enabled\n"); + return 0; + } + stccr = SSI1_STCCR; + } else { + if (SSI2_SCR & SSI_SCR_SSIEN) { + printk(KERN_WARNING "Warning ssi already enabled\n"); + return 0; + } + stccr = SSI2_STCCR; + } + + stmsk = srmsk = mask; + stccr &= ~SSI_STCCR_DC_MASK; + stccr |= SSI_STCCR_DC(slots - 1); + + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { + SSI1_STMSK = stmsk; + SSI1_SRMSK = srmsk; + SSI1_SRCCR = SSI1_STCCR = stccr; + } else { + SSI2_STMSK = stmsk; + SSI2_SRMSK = srmsk; + SSI2_SRCCR = SSI2_STCCR = stccr; + } + + return 0; +} + +/* + * SSI DAI format configuration. + * Should only be called when port is inactive (i.e. SSIEN = 0). + * Note: We don't use the I2S modes but instead manually configure the + * SSI for I2S. + */ +static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + u32 stcr = 0, srcr = 0, scr; + + /* + * This is done to avoid this function to modify + * previous set values in stcr + */ + stcr = SSI1_STCR; + + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) + scr = SSI1_SCR & ~(SSI_SCR_SYN | SSI_SCR_NET); + else + scr = SSI2_SCR & ~(SSI_SCR_SYN | SSI_SCR_NET); + + if (scr & SSI_SCR_SSIEN) { + printk(KERN_WARNING "Warning ssi already enabled\n"); + return 0; + } + + /* DAI mode */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + /* data on rising edge of bclk, frame low 1clk before data */ + stcr |= SSI_STCR_TFSI | SSI_STCR_TEFS | SSI_STCR_TXBIT0; + srcr |= SSI_SRCR_RFSI | SSI_SRCR_REFS | SSI_SRCR_RXBIT0; + break; + case SND_SOC_DAIFMT_LEFT_J: + /* data on rising edge of bclk, frame high with data */ + stcr |= SSI_STCR_TXBIT0; + srcr |= SSI_SRCR_RXBIT0; + break; + case SND_SOC_DAIFMT_DSP_B: + /* data on rising edge of bclk, frame high with data */ + stcr |= SSI_STCR_TFSL; + srcr |= SSI_SRCR_RFSL; + break; + case SND_SOC_DAIFMT_DSP_A: + /* data on rising edge of bclk, frame high 1clk before data */ + stcr |= SSI_STCR_TFSL | SSI_STCR_TEFS; + srcr |= SSI_SRCR_RFSL | SSI_SRCR_REFS; + break; + } + + /* DAI clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_IF: + stcr |= SSI_STCR_TFSI; + stcr &= ~SSI_STCR_TSCKP; + srcr |= SSI_SRCR_RFSI; + srcr &= ~SSI_SRCR_RSCKP; + break; + case SND_SOC_DAIFMT_IB_NF: + stcr &= ~(SSI_STCR_TSCKP | SSI_STCR_TFSI); + srcr &= ~(SSI_SRCR_RSCKP | SSI_SRCR_RFSI); + break; + case SND_SOC_DAIFMT_NB_IF: + stcr |= SSI_STCR_TFSI | SSI_STCR_TSCKP; + srcr |= SSI_SRCR_RFSI | SSI_SRCR_RSCKP; + break; + case SND_SOC_DAIFMT_NB_NF: + stcr &= ~SSI_STCR_TFSI; + stcr |= SSI_STCR_TSCKP; + srcr &= ~SSI_SRCR_RFSI; + srcr |= SSI_SRCR_RSCKP; + break; + } + + /* DAI clock master masks */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + stcr |= SSI_STCR_TFDIR | SSI_STCR_TXDIR; + srcr |= SSI_SRCR_RFDIR | SSI_SRCR_RXDIR; + break; + case SND_SOC_DAIFMT_CBM_CFS: + stcr |= SSI_STCR_TFDIR; + srcr |= SSI_SRCR_RFDIR; + break; + case SND_SOC_DAIFMT_CBS_CFM: + stcr |= SSI_STCR_TXDIR; + srcr |= SSI_SRCR_RXDIR; + break; + } + + /* sync */ + if (!(fmt & SND_SOC_DAIFMT_ASYNC)) + scr |= SSI_SCR_SYN; + + /* tdm - only for stereo atm */ + if (fmt & SND_SOC_DAIFMT_TDM) + scr |= SSI_SCR_NET; + + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { + SSI1_STCR = stcr; + SSI1_SRCR = srcr; + SSI1_SCR = scr; + } else { + SSI2_STCR = stcr; + SSI2_SRCR = srcr; + SSI2_SCR = scr; + } + + return 0; +} + +static int imx_ssi_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* set up TX DMA params */ + switch (cpu_dai->id) { + case IMX_DAI_SSI0: + cpu_dai->dma_data = &imx_ssi1_pcm_stereo_out0; + break; + case IMX_DAI_SSI1: + cpu_dai->dma_data = &imx_ssi1_pcm_stereo_out1; + break; + case IMX_DAI_SSI2: + cpu_dai->dma_data = &imx_ssi2_pcm_stereo_out0; + break; + case IMX_DAI_SSI3: + cpu_dai->dma_data = &imx_ssi2_pcm_stereo_out1; + } + pr_debug("%s: (playback)\n", __func__); + } else { + /* set up RX DMA params */ + switch (cpu_dai->id) { + case IMX_DAI_SSI0: + cpu_dai->dma_data = &imx_ssi1_pcm_stereo_in0; + break; + case IMX_DAI_SSI1: + cpu_dai->dma_data = &imx_ssi1_pcm_stereo_in1; + break; + case IMX_DAI_SSI2: + cpu_dai->dma_data = &imx_ssi2_pcm_stereo_in0; + break; + case IMX_DAI_SSI3: + cpu_dai->dma_data = &imx_ssi2_pcm_stereo_in1; + } + pr_debug("%s: (capture)\n", __func__); + } + + /* + * we cant really change any SSI values after SSI is enabled + * need to fix in software for max flexibility - lrg + */ + if (cpu_dai->active) { + printk(KERN_WARNING "Warning ssi already enabled\n"); + return 0; + } + + /* reset the SSI port - Sect 45.4.4 */ + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { + + if (!ssi_clk0) + return -EINVAL; + + if (ssi_active[SSI1_PORT]++) { + pr_debug("%s: exit before reset\n", __func__); + return 0; + } + + /* SSI1 Reset */ + SSI1_SCR = 0; + + SSI1_SFCSR = SSI_SFCSR_RFWM1(RXFIFO_WATERMARK) | + SSI_SFCSR_RFWM0(RXFIFO_WATERMARK) | + SSI_SFCSR_TFWM1(TXFIFO_WATERMARK) | + SSI_SFCSR_TFWM0(TXFIFO_WATERMARK); + } else { + + if (!ssi_clk1) + return -EINVAL; + + if (ssi_active[SSI2_PORT]++) { + pr_debug("%s: exit before reset\n", __func__); + return 0; + } + + /* SSI2 Reset */ + SSI2_SCR = 0; + + SSI2_SFCSR = SSI_SFCSR_RFWM1(RXFIFO_WATERMARK) | + SSI_SFCSR_RFWM0(RXFIFO_WATERMARK) | + SSI_SFCSR_TFWM1(TXFIFO_WATERMARK) | + SSI_SFCSR_TFWM0(TXFIFO_WATERMARK); + } + + return 0; +} + +int imx_ssi_hw_tx_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + u32 stccr, stcr, sier; + + pr_debug("%s\n", __func__); + + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { + stccr = SSI1_STCCR & ~SSI_STCCR_WL_MASK; + stcr = SSI1_STCR; + sier = SSI1_SIER; + } else { + stccr = SSI2_STCCR & ~SSI_STCCR_WL_MASK; + stcr = SSI2_STCR; + sier = SSI2_SIER; + } + + /* DAI data (word) size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + stccr |= SSI_STCCR_WL(16); + break; + case SNDRV_PCM_FORMAT_S20_3LE: + stccr |= SSI_STCCR_WL(20); + break; + case SNDRV_PCM_FORMAT_S24_LE: + stccr |= SSI_STCCR_WL(24); + break; + } + + /* enable interrupts */ + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) + stcr |= SSI_STCR_TFEN0; + else + stcr |= SSI_STCR_TFEN1; + sier |= SSI_SIER_TDMAE; + + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { + SSI1_STCR = stcr; + SSI1_STCCR = stccr; + SSI1_SIER = sier; + } else { + SSI2_STCR = stcr; + SSI2_STCCR = stccr; + SSI2_SIER = sier; + } + + return 0; +} + +int imx_ssi_hw_rx_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + u32 srccr, srcr, sier; + + pr_debug("%s\n", __func__); + + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { + srccr = SSI1_SRCCR & ~SSI_SRCCR_WL_MASK; + srcr = SSI1_SRCR; + sier = SSI1_SIER; + } else { + srccr = SSI2_SRCCR & ~SSI_SRCCR_WL_MASK; + srcr = SSI2_SRCR; + sier = SSI2_SIER; + } + + /* DAI data (word) size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + srccr |= SSI_SRCCR_WL(16); + break; + case SNDRV_PCM_FORMAT_S20_3LE: + srccr |= SSI_SRCCR_WL(20); + break; + case SNDRV_PCM_FORMAT_S24_LE: + srccr |= SSI_SRCCR_WL(24); + break; + } + + /* enable interrupts */ + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) + srcr |= SSI_SRCR_RFEN0; + else + srcr |= SSI_SRCR_RFEN1; + sier |= SSI_SIER_RDMAE; + + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { + SSI1_SRCR = srcr; + SSI1_SRCCR = srccr; + SSI1_SIER = sier; + } else { + SSI2_SRCR = srcr; + SSI2_SRCCR = srccr; + SSI2_SIER = sier; + } + + return 0; +} + +/* + * Should only be called when port is inactive (i.e. SSIEN = 0), + * although can be called multiple times by upper layers. + */ +int imx_ssi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + int ret; + + /* cant change any parameters when SSI is running */ + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { + if (SSI1_SCR & SSI_SCR_SSIEN) { + printk(KERN_WARNING "Warning ssi already enabled\n"); + return 0; + } + } else { + if (SSI2_SCR & SSI_SCR_SSIEN) { + printk(KERN_WARNING "Warning ssi already enabled\n"); + return 0; + } + } + + /* + * Configure both tx and rx params with the same settings. This is + * really a harware restriction because SSI must be disabled until + * we can change those values. If there is an active audio stream in + * one direction, enabling the other direction with different + * settings would mean disturbing the running one. + */ + ret = imx_ssi_hw_tx_params(substream, params); + if (ret < 0) + return ret; + return imx_ssi_hw_rx_params(substream, params); +} + +int imx_ssi_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + pr_debug("%s\n", __func__); + + /* Enable clks here to follow SSI recommended init sequence */ + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { + ret = clk_enable(ssi_clk0); + if (ret < 0) + printk(KERN_ERR "Unable to enable ssi_clk0\n"); + } else { + ret = clk_enable(ssi_clk1); + if (ret < 0) + printk(KERN_ERR "Unable to enable ssi_clk1\n"); + } + + return 0; +} + +static int imx_ssi_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + u32 scr; + + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) + scr = SSI1_SCR; + else + scr = SSI2_SCR; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + scr |= SSI_SCR_TE | SSI_SCR_SSIEN; + else + scr |= SSI_SCR_RE | SSI_SCR_SSIEN; + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + scr &= ~SSI_SCR_TE; + else + scr &= ~SSI_SCR_RE; + break; + default: + return -EINVAL; + } + + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) + SSI1_SCR = scr; + else + SSI2_SCR = scr; + + return 0; +} + +static void imx_ssi_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + /* shutdown SSI if neither Tx or Rx is active */ + if (!cpu_dai->active) { + + if (cpu_dai->id == IMX_DAI_SSI0 || + cpu_dai->id == IMX_DAI_SSI2) { + + if (--ssi_active[SSI1_PORT] > 1) + return; + + SSI1_SCR = 0; + clk_disable(ssi_clk0); + } else { + if (--ssi_active[SSI2_PORT]) + return; + SSI2_SCR = 0; + clk_disable(ssi_clk1); + } + } +} + +#ifdef CONFIG_PM +static int imx_ssi_suspend(struct platform_device *dev, + struct snd_soc_dai *dai) +{ + return 0; +} + +static int imx_ssi_resume(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + return 0; +} + +#else +#define imx_ssi_suspend NULL +#define imx_ssi_resume NULL +#endif + +#define IMX_SSI_RATES \ + (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | \ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | \ + SNDRV_PCM_RATE_96000) + +#define IMX_SSI_BITS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops imx_ssi_pcm_dai_ops = { + .startup = imx_ssi_startup, + .shutdown = imx_ssi_shutdown, + .trigger = imx_ssi_trigger, + .prepare = imx_ssi_prepare, + .hw_params = imx_ssi_hw_params, + .set_sysclk = imx_ssi_set_dai_sysclk, + .set_clkdiv = imx_ssi_set_dai_clkdiv, + .set_fmt = imx_ssi_set_dai_fmt, + .set_tdm_slot = imx_ssi_set_dai_tdm_slot, +}; + +struct snd_soc_dai imx_ssi_pcm_dai[] = { +{ + .name = "imx-i2s-1-0", + .id = IMX_DAI_SSI0, + .suspend = imx_ssi_suspend, + .resume = imx_ssi_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .formats = IMX_SSI_BITS, + .rates = IMX_SSI_RATES,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .formats = IMX_SSI_BITS, + .rates = IMX_SSI_RATES,}, + .ops = &imx_ssi_pcm_dai_ops, +}, +{ + .name = "imx-i2s-2-0", + .id = IMX_DAI_SSI1, + .playback = { + .channels_min = 1, + .channels_max = 2, + .formats = IMX_SSI_BITS, + .rates = IMX_SSI_RATES,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .formats = IMX_SSI_BITS, + .rates = IMX_SSI_RATES,}, + .ops = &imx_ssi_pcm_dai_ops, +}, +{ + .name = "imx-i2s-1-1", + .id = IMX_DAI_SSI2, + .suspend = imx_ssi_suspend, + .resume = imx_ssi_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .formats = IMX_SSI_BITS, + .rates = IMX_SSI_RATES,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .formats = IMX_SSI_BITS, + .rates = IMX_SSI_RATES,}, + .ops = &imx_ssi_pcm_dai_ops, +}, +{ + .name = "imx-i2s-2-1", + .id = IMX_DAI_SSI3, + .playback = { + .channels_min = 1, + .channels_max = 2, + .formats = IMX_SSI_BITS, + .rates = IMX_SSI_RATES,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .formats = IMX_SSI_BITS, + .rates = IMX_SSI_RATES,}, + .ops = &imx_ssi_pcm_dai_ops, +}, +}; +EXPORT_SYMBOL_GPL(imx_ssi_pcm_dai); + +static int __init imx_ssi_init(void) +{ + return snd_soc_register_dais(imx_ssi_pcm_dai, + ARRAY_SIZE(imx_ssi_pcm_dai)); +} + +static void __exit imx_ssi_exit(void) +{ + snd_soc_unregister_dais(imx_ssi_pcm_dai, + ARRAY_SIZE(imx_ssi_pcm_dai)); +} + +module_init(imx_ssi_init); +module_exit(imx_ssi_exit); +MODULE_AUTHOR("Liam Girdwood, liam.girdwood(a)wolfsonmicro.com"); +MODULE_DESCRIPTION("i.MX ASoC I2S driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/imx/mxc-ssi.h b/sound/soc/imx/mxc-ssi.h new file mode 100644 index 0000000..12bbdc9 --- /dev/null +++ b/sound/soc/imx/mxc-ssi.h @@ -0,0 +1,238 @@ +/* + * 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 _IMX_SSI_H +#define _IMX_SSI_H + +#include <mach/hardware.h> + +/* SSI regs definition - MOVE to /arch/arm/plat-mxc/include/mach/ when stable */ +#define SSI1_IO_BASE_ADDR IO_ADDRESS(SSI1_BASE_ADDR) +#define SSI2_IO_BASE_ADDR IO_ADDRESS(SSI2_BASE_ADDR) + +#define STX0 0x00 +#define STX1 0x04 +#define SRX0 0x08 +#define SRX1 0x0c +#define SCR 0x10 +#define SISR 0x14 +#define SIER 0x18 +#define STCR 0x1c +#define SRCR 0x20 +#define STCCR 0x24 +#define SRCCR 0x28 +#define SFCSR 0x2c +#define STR 0x30 +#define SOR 0x34 +#define SACNT 0x38 +#define SACADD 0x3c +#define SACDAT 0x40 +#define SATAG 0x44 +#define STMSK 0x48 +#define SRMSK 0x4c + +#define SSI1_STX0 (*((volatile u32 *)(SSI1_IO_BASE_ADDR + STX0))) +#define SSI1_STX1 (*((volatile u32 *)(SSI1_IO_BASE_ADDR + STX1))) +#define SSI1_SRX0 (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SRX0))) +#define SSI1_SRX1 (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SRX1))) +#define SSI1_SCR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SCR))) +#define SSI1_SISR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SISR))) +#define SSI1_SIER (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SIER))) +#define SSI1_STCR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + STCR))) +#define SSI1_SRCR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SRCR))) +#define SSI1_STCCR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + STCCR))) +#define SSI1_SRCCR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SRCCR))) +#define SSI1_SFCSR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SFCSR))) +#define SSI1_STR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + STR))) +#define SSI1_SOR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SOR))) +#define SSI1_SACNT (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SACNT))) +#define SSI1_SACADD (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SACADD))) +#define SSI1_SACDAT (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SACDAT))) +#define SSI1_SATAG (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SATAG))) +#define SSI1_STMSK (*((volatile u32 *)(SSI1_IO_BASE_ADDR + STMSK))) +#define SSI1_SRMSK (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SRMSK))) + + +#define SSI2_STX0 (*((volatile u32 *)(SSI2_IO_BASE_ADDR + STX0))) +#define SSI2_STX1 (*((volatile u32 *)(SSI2_IO_BASE_ADDR + STX1))) +#define SSI2_SRX0 (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SRX0))) +#define SSI2_SRX1 (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SRX1))) +#define SSI2_SCR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SCR))) +#define SSI2_SISR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SISR))) +#define SSI2_SIER (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SIER))) +#define SSI2_STCR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + STCR))) +#define SSI2_SRCR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SRCR))) +#define SSI2_STCCR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + STCCR))) +#define SSI2_SRCCR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SRCCR))) +#define SSI2_SFCSR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SFCSR))) +#define SSI2_STR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + STR))) +#define SSI2_SOR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SOR))) +#define SSI2_SACNT (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SACNT))) +#define SSI2_SACADD (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SACADD))) +#define SSI2_SACDAT (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SACDAT))) +#define SSI2_SATAG (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SATAG))) +#define SSI2_STMSK (*((volatile u32 *)(SSI2_IO_BASE_ADDR + STMSK))) +#define SSI2_SRMSK (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SRMSK))) + +#define SSI_SCR_CLK_IST (1 << 9) +#define SSI_SCR_TCH_EN (1 << 8) +#define SSI_SCR_SYS_CLK_EN (1 << 7) +#define SSI_SCR_I2S_MODE_NORM (0 << 5) +#define SSI_SCR_I2S_MODE_MSTR (1 << 5) +#define SSI_SCR_I2S_MODE_SLAVE (2 << 5) +#define SSI_SCR_SYN (1 << 4) +#define SSI_SCR_NET (1 << 3) +#define SSI_SCR_RE (1 << 2) +#define SSI_SCR_TE (1 << 1) +#define SSI_SCR_SSIEN (1 << 0) + +#define SSI_SISR_CMDAU (1 << 18) +#define SSI_SISR_CMDDU (1 << 17) +#define SSI_SISR_RXT (1 << 16) +#define SSI_SISR_RDR1 (1 << 15) +#define SSI_SISR_RDR0 (1 << 14) +#define SSI_SISR_TDE1 (1 << 13) +#define SSI_SISR_TDE0 (1 << 12) +#define SSI_SISR_ROE1 (1 << 11) +#define SSI_SISR_ROE0 (1 << 10) +#define SSI_SISR_TUE1 (1 << 9) +#define SSI_SISR_TUE0 (1 << 8) +#define SSI_SISR_TFS (1 << 7) +#define SSI_SISR_RFS (1 << 6) +#define SSI_SISR_TLS (1 << 5) +#define SSI_SISR_RLS (1 << 4) +#define SSI_SISR_RFF1 (1 << 3) +#define SSI_SISR_RFF0 (1 << 2) +#define SSI_SISR_TFE1 (1 << 1) +#define SSI_SISR_TFE0 (1 << 0) + +#define SSI_SIER_RDMAE (1 << 22) +#define SSI_SIER_RIE (1 << 21) +#define SSI_SIER_TDMAE (1 << 20) +#define SSI_SIER_TIE (1 << 19) +#define SSI_SIER_CMDAU_EN (1 << 18) +#define SSI_SIER_CMDDU_EN (1 << 17) +#define SSI_SIER_RXT_EN (1 << 16) +#define SSI_SIER_RDR1_EN (1 << 15) +#define SSI_SIER_RDR0_EN (1 << 14) +#define SSI_SIER_TDE1_EN (1 << 13) +#define SSI_SIER_TDE0_EN (1 << 12) +#define SSI_SIER_ROE1_EN (1 << 11) +#define SSI_SIER_ROE0_EN (1 << 10) +#define SSI_SIER_TUE1_EN (1 << 9) +#define SSI_SIER_TUE0_EN (1 << 8) +#define SSI_SIER_TFS_EN (1 << 7) +#define SSI_SIER_RFS_EN (1 << 6) +#define SSI_SIER_TLS_EN (1 << 5) +#define SSI_SIER_RLS_EN (1 << 4) +#define SSI_SIER_RFF1_EN (1 << 3) +#define SSI_SIER_RFF0_EN (1 << 2) +#define SSI_SIER_TFE1_EN (1 << 1) +#define SSI_SIER_TFE0_EN (1 << 0) + +#define SSI_STCR_TXBIT0 (1 << 9) +#define SSI_STCR_TFEN1 (1 << 8) +#define SSI_STCR_TFEN0 (1 << 7) +#define SSI_STCR_TFDIR (1 << 6) +#define SSI_STCR_TXDIR (1 << 5) +#define SSI_STCR_TSHFD (1 << 4) +#define SSI_STCR_TSCKP (1 << 3) +#define SSI_STCR_TFSI (1 << 2) +#define SSI_STCR_TFSL (1 << 1) +#define SSI_STCR_TEFS (1 << 0) + +#define SSI_SRCR_RXBIT0 (1 << 9) +#define SSI_SRCR_RFEN1 (1 << 8) +#define SSI_SRCR_RFEN0 (1 << 7) +#define SSI_SRCR_RFDIR (1 << 6) +#define SSI_SRCR_RXDIR (1 << 5) +#define SSI_SRCR_RSHFD (1 << 4) +#define SSI_SRCR_RSCKP (1 << 3) +#define SSI_SRCR_RFSI (1 << 2) +#define SSI_SRCR_RFSL (1 << 1) +#define SSI_SRCR_REFS (1 << 0) + +#define SSI_STCCR_DIV2 (1 << 18) +#define SSI_STCCR_PSR (1 << 15) +#define SSI_STCCR_WL(x) ((((x) - 2) >> 1) << 13) +#define SSI_STCCR_DC(x) (((x) & 0x1f) << 8) +#define SSI_STCCR_PM(x) (((x) & 0xff) << 0) +#define SSI_STCCR_WL_MASK (0xf << 13) +#define SSI_STCCR_DC_MASK (0x1f << 8) +#define SSI_STCCR_PM_MASK (0xff << 0) + +#define SSI_SRCCR_DIV2 (1 << 18) +#define SSI_SRCCR_PSR (1 << 15) +#define SSI_SRCCR_WL(x) ((((x) - 2) >> 1) << 13) +#define SSI_SRCCR_DC(x) (((x) & 0x1f) << 8) +#define SSI_SRCCR_PM(x) (((x) & 0xff) << 0) +#define SSI_SRCCR_WL_MASK (0xf << 13) +#define SSI_SRCCR_DC_MASK (0x1f << 8) +#define SSI_SRCCR_PM_MASK (0xff << 0) + + +#define SSI_SFCSR_RFCNT1(x) (((x) & 0xf) << 28) +#define SSI_SFCSR_TFCNT1(x) (((x) & 0xf) << 24) +#define SSI_SFCSR_RFWM1(x) (((x) & 0xf) << 20) +#define SSI_SFCSR_TFWM1(x) (((x) & 0xf) << 16) +#define SSI_SFCSR_RFCNT0(x) (((x) & 0xf) << 12) +#define SSI_SFCSR_TFCNT0(x) (((x) & 0xf) << 8) +#define SSI_SFCSR_RFWM0(x) (((x) & 0xf) << 4) +#define SSI_SFCSR_TFWM0(x) (((x) & 0xf) << 0) + +#define SSI_STR_TEST (1 << 15) +#define SSI_STR_RCK2TCK (1 << 14) +#define SSI_STR_RFS2TFS (1 << 13) +#define SSI_STR_RXSTATE(x) (((x) & 0xf) << 8) +#define SSI_STR_TXD2RXD (1 << 7) +#define SSI_STR_TCK2RCK (1 << 6) +#define SSI_STR_TFS2RFS (1 << 5) +#define SSI_STR_TXSTATE(x) (((x) & 0xf) << 0) + +#define SSI_SOR_CLKOFF (1 << 6) +#define SSI_SOR_RX_CLR (1 << 5) +#define SSI_SOR_TX_CLR (1 << 4) +#define SSI_SOR_INIT (1 << 3) +#define SSI_SOR_WAIT(x) (((x) & 0x3) << 1) +#define SSI_SOR_SYNRST (1 << 0) + +#define SSI_SACNT_FRDIV(x) (((x) & 0x3f) << 5) +#define SSI_SACNT_WR (x << 4) +#define SSI_SACNT_RD (x << 3) +#define SSI_SACNT_TIF (x << 2) +#define SSI_SACNT_FV (x << 1) +#define SSI_SACNT_AC97EN (x << 0) + +/* Watermarks for FIFO's */ +#define TXFIFO_WATERMARK 0x4 +#define RXFIFO_WATERMARK 0x4 + +/* i.MX DAI SSP ID's */ +#define IMX_DAI_SSI0 0 /* SSI1 FIFO 0 */ +#define IMX_DAI_SSI1 1 /* SSI1 FIFO 1 */ +#define IMX_DAI_SSI2 2 /* SSI2 FIFO 0 */ +#define IMX_DAI_SSI3 3 /* SSI2 FIFO 1 */ + +/* SSI clock sources */ +#define IMX_SSP_SYS_CLK 0 + +/* SSI audio dividers */ +#define IMX_SSI_TX_DIV_2 0 +#define IMX_SSI_TX_DIV_PSR 1 +#define IMX_SSI_TX_DIV_PM 2 +#define IMX_SSI_RX_DIV_2 3 +#define IMX_SSI_RX_DIV_PSR 4 +#define IMX_SSI_RX_DIV_PM 5 + + +/* SSI Div 2 */ +#define IMX_SSI_DIV_2_OFF (~SSI_STCCR_DIV2) +#define IMX_SSI_DIV_2_ON SSI_STCCR_DIV2 + +extern struct snd_soc_dai imx_ssi_pcm_dai[4]; +extern int get_ssi_clk(int ssi, struct device *dev); +extern void put_ssi_clk(int ssi); +#endif --- -- Javier Martin Vista Silicon S.L. Universidad de Cantabria CDTUC - FASE C - Oficina S-345 Avda de los Castros s/n 39005- Santander. Cantabria. Spain +34 942 25 32 60 www.vista-silicon.com

2 2

[alsa-devel] [PATCH 1/3] ASoC: add DMA platform driver for MX1x and MX2x
by javier Martin 04 Aug '09

04 Aug '09

This adds support for DMA platform valid for i.MX1 and i.MX2 platforms. This is not valid for i.MX3 since it doesn't share the same DMA interface than i.MX1 and i.MX2. It has been tested on i.MX27 board. Signed-off-by: Javier Martin <javier.martin(a)vista-silicon.com> --- sound/soc/Makefile | 1 + sound/soc/imx/Kconfig | 11 + sound/soc/imx/Makefile | 4 + sound/soc/imx/mx1_mx2-pcm.c | 487 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/imx/mx1_mx2-pcm.h | 48 +++++ 5 files changed, 551 insertions(+), 0 deletions(-) diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 0237879..8120b52 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -11,3 +11,4 @@ obj-$(CONFIG_SND_SOC) += omap/ obj-$(CONFIG_SND_SOC) += pxa/ obj-$(CONFIG_SND_SOC) += s3c24xx/ obj-$(CONFIG_SND_SOC) += sh/ +obj-$(CONFIG_SND_SOC) += imx/ diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig new file mode 100644 index 0000000..a1bf053 --- /dev/null +++ b/sound/soc/imx/Kconfig @@ -0,0 +1,11 @@ +config SND_MX1_MX2_SOC + tristate "SoC Audio for Freecale i.MX1x i.MX2x CPUs" + depends on (ARCH_MX2 || ARCH_MX1) && SND + select SND_PCM + help + Say Y or M if you want to add support for codecs attached to + the MX1 or MX2 SSI interface. + + + + diff --git a/sound/soc/imx/Makefile b/sound/soc/imx/Makefile new file mode 100644 index 0000000..c390f0f --- /dev/null +++ b/sound/soc/imx/Makefile @@ -0,0 +1,4 @@ +# i.MX Platform Support +snd-soc-mx1_mx2-objs := mx1_mx2-pcm.o + +obj-$(CONFIG_SND_MX1_MX2_SOC) += snd-soc-mx1_mx2.o diff --git a/sound/soc/imx/mx1_mx2-pcm.c b/sound/soc/imx/mx1_mx2-pcm.c new file mode 100644 index 0000000..94807f8 --- /dev/null +++ b/sound/soc/imx/mx1_mx2-pcm.c @@ -0,0 +1,487 @@ +/* + * mx1_mx2-pcm.c -- ALSA SoC interface for Freescale i.MX1x, i.MX2x CPUs + * + * Copyright 2009 Vista Silicon S.L. + * Author: Javier Martin + * javier.martin(a)vista-silicon.com + * + * Based on mxc-pcm.c by Liam Girdwood. + * + * 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/init.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/dma-mapping.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <asm/dma.h> +#include <mach/hardware.h> +#include <mach/dma-mx1-mx2.h> + +#include "mx1_mx2-pcm.h" + + +static const struct snd_pcm_hardware mx1_mx2_pcm_hardware = { + .info = (SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID), + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .buffer_bytes_max = 32 * 1024, + .period_bytes_min = 64, + .period_bytes_max = 8 * 1024, + .periods_min = 2, + .periods_max = 255, + .fifo_size = 0, +}; + +struct mx1_mx2_runtime_data { + int dma_ch; + int active; + unsigned int period; + unsigned int periods; + int tx_spin; + spinlock_t dma_lock; + struct mx1_mx2_pcm_dma_params *dma_params; +}; + + +/** + * This function stops the current dma transfer for playback + * and clears the dma pointers. + * + * @param substream pointer to the structure of the current stream. + * + */ +static int audio_stop_dma(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct mx1_mx2_runtime_data *prtd = runtime->private_data; + unsigned long flags; + + spin_lock_irqsave(&prtd->dma_lock, flags); + + pr_debug("%s\n", __func__); + + prtd->active = 0; + prtd->period = 0; + prtd->periods = 0; + + /* this stops the dma channel and clears the buffer ptrs */ + + imx_dma_disable(prtd->dma_ch); + + spin_unlock_irqrestore(&prtd->dma_lock, flags); + + return 0; +} + +/** + * This function is called whenever a new audio block needs to be + * transferred to the codec. The function receives the address and the size + * of the new block and start a new DMA transfer. + * + * @param substream pointer to the structure of the current stream. + * + */ +static int dma_new_period(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct mx1_mx2_runtime_data *prtd = runtime->private_data; + unsigned int dma_size; + unsigned int offset; + int ret = 0; + dma_addr_t mem_addr; + unsigned int dev_addr; + + if (prtd->active) { + dma_size = frames_to_bytes(runtime, runtime->period_size); + offset = dma_size * prtd->period; + + pr_debug("%s: period (%d) out of (%d)\n", __func__, + prtd->period, + runtime->periods); + pr_debug("period_size %d frames\n offset %d bytes\n", + (unsigned int)runtime->period_size, + offset); + pr_debug("dma_size %d bytes\n", dma_size); + + snd_BUG_ON(dma_size > mx1_mx2_pcm_hardware.period_bytes_max); + + mem_addr = (dma_addr_t)(runtime->dma_addr + offset); + dev_addr = prtd->dma_params->per_address; + pr_debug("%s: mem_addr is %x\n dev_addr is %x\n", + __func__, mem_addr, dev_addr); + + ret = imx_dma_setup_single(prtd->dma_ch, mem_addr, + dma_size, dev_addr, + prtd->dma_params->transfer_type); + if (ret < 0) { + printk(KERN_ERR "Error configuring DMA\n"); + return ret; + } + imx_dma_enable(prtd->dma_ch); + + pr_debug("%s: transfer enabled\nmem_addr = %x\n", + __func__, (unsigned int) mem_addr); + pr_debug("dev_addr = %x\ndma_size = %d\n", + (unsigned int) dev_addr, dma_size); + + prtd->tx_spin = 1; /* FGA little trick to retrieve DMA pos */ + prtd->period++; + prtd->period %= runtime->periods; + } + return ret; +} + + +/** + * This is a callback which will be called + * when a TX transfer finishes. The call occurs + * in interrupt context. + * + * @param dat pointer to the structure of the current stream. + * + */ +static void audio_dma_irq(int channel, void *data) +{ + struct snd_pcm_substream *substream; + struct snd_pcm_runtime *runtime; + struct mx1_mx2_runtime_data *prtd; + unsigned int dma_size; + unsigned int previous_period; + unsigned int offset; + + substream = data; + runtime = substream->runtime; + prtd = runtime->private_data; + previous_period = prtd->periods; + dma_size = frames_to_bytes(runtime, runtime->period_size); + offset = dma_size * previous_period; + + prtd->tx_spin = 0; + prtd->periods++; + prtd->periods %= runtime->periods; + + pr_debug("%s: irq per %d offset %x\n", __func__, prtd->periods, offset); + + /* + * If we are getting a callback for an active stream then we inform + * the PCM middle layer we've finished a period + */ + if (prtd->active) + snd_pcm_period_elapsed(substream); + + /* + * Trig next DMA transfer + */ + dma_new_period(substream); +} + +/** + * This function configures the hardware to allow audio + * playback operations. It is called by ALSA framework. + * + * @param substream pointer to the structure of the current stream. + * + * @return 0 on success, -1 otherwise. + */ +static int +snd_mx1_mx2_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct mx1_mx2_runtime_data *prtd = runtime->private_data; + + prtd->period = 0; + prtd->periods = 0; + + return 0; +} + +static int mx1_mx2_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + int ret; + + ret = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + if (ret < 0) { + printk(KERN_ERR "%s: failed to malloc pcm pages\n", __func__); + return ret; + } + + pr_debug("%s: snd_imx1_mx2_audio_hw_params runtime->dma_addr 0x(%x)\n", + __func__, (unsigned int)runtime->dma_addr); + pr_debug("%s: snd_imx1_mx2_audio_hw_params runtime->dma_area 0x(%x)\n", + __func__, (unsigned int)runtime->dma_area); + pr_debug("%s: snd_imx1_mx2_audio_hw_params runtime->dma_bytes 0x(%x)\n", + __func__, (unsigned int)runtime->dma_bytes); + + return ret; +} + +static int mx1_mx2_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct mx1_mx2_runtime_data *prtd = runtime->private_data; + + imx_dma_free(prtd->dma_ch); + + snd_pcm_lib_free_pages(substream); + + return 0; +} + +static int mx1_mx2_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct mx1_mx2_runtime_data *prtd = substream->runtime->private_data; + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + prtd->tx_spin = 0; + /* requested stream startup */ + prtd->active = 1; + pr_debug("%s: starting dma_new_period\n", __func__); + ret = dma_new_period(substream); + break; + case SNDRV_PCM_TRIGGER_STOP: + /* requested stream shutdown */ + pr_debug("%s: stopping dma transfer\n", __func__); + ret = audio_stop_dma(substream); + break; + default: + ret = -EINVAL; + break; + } + + return ret; +} + +static snd_pcm_uframes_t +mx1_mx2_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct mx1_mx2_runtime_data *prtd = runtime->private_data; + unsigned int offset = 0; + + /* tx_spin value is used here to check if a transfer is active */ + if (prtd->tx_spin) { + offset = (runtime->period_size * (prtd->periods)) + + (runtime->period_size >> 1); + if (offset >= runtime->buffer_size) + offset = runtime->period_size >> 1; + } else { + offset = (runtime->period_size * (prtd->periods)); + if (offset >= runtime->buffer_size) + offset = 0; + } + pr_debug("%s: pointer offset %x\n", __func__, offset); + + return offset; +} + +static int mx1_mx2_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct mx1_mx2_runtime_data *prtd; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct mx1_mx2_pcm_dma_params *dma_data = rtd->dai->cpu_dai->dma_data; + int ret; + + snd_soc_set_runtime_hwparams(substream, &mx1_mx2_pcm_hardware); + + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + return ret; + + prtd = kzalloc(sizeof(struct mx1_mx2_runtime_data), GFP_KERNEL); + if (prtd == NULL) { + ret = -ENOMEM; + goto out; + } + + runtime->private_data = prtd; + + if (!dma_data) + return -ENODEV; + + prtd->dma_params = dma_data; + + pr_debug("%s: Requesting dma channel (%s)\n", __func__, + prtd->dma_params->name); + prtd->dma_ch = imx_dma_request_by_prio(prtd->dma_params->name, + DMA_PRIO_HIGH); + if (prtd->dma_ch < 0) { + printk(KERN_ERR "Error requesting dma channel\n"); + return ret; + } + imx_dma_config_burstlen(prtd->dma_ch, + prtd->dma_params->watermark_level); + + ret = imx_dma_config_channel(prtd->dma_ch, + prtd->dma_params->per_config, + prtd->dma_params->mem_config, + prtd->dma_params->event_id, 0); + + if (ret) { + pr_debug(KERN_ERR "Error configuring dma channel %d\n", + prtd->dma_ch); + return ret; + } + + pr_debug("%s: Setting tx dma callback function\n", __func__); + ret = imx_dma_setup_handlers(prtd->dma_ch, + audio_dma_irq, NULL, + (void *)substream); + if (ret < 0) { + printk(KERN_ERR "Error setting dma callback function\n"); + return ret; + } + return 0; + + out: + return ret; +} + +static int mx1_mx2_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct mx1_mx2_runtime_data *prtd = runtime->private_data; + + kfree(prtd); + + return 0; +} + +static int mx1_mx2_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + return dma_mmap_writecombine(substream->pcm->card->dev, vma, + runtime->dma_area, + runtime->dma_addr, + runtime->dma_bytes); +} + +struct snd_pcm_ops mx1_mx2_pcm_ops = { + .open = mx1_mx2_pcm_open, + .close = mx1_mx2_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = mx1_mx2_pcm_hw_params, + .hw_free = mx1_mx2_pcm_hw_free, + .prepare = snd_mx1_mx2_prepare, + .trigger = mx1_mx2_pcm_trigger, + .pointer = mx1_mx2_pcm_pointer, + .mmap = mx1_mx2_pcm_mmap, +}; + +static u64 mx1_mx2_pcm_dmamask = 0xffffffff; + +static int mx1_mx2_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = mx1_mx2_pcm_hardware.buffer_bytes_max; + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + + /* Reserve uncached-buffered memory area for DMA */ + buf->area = dma_alloc_writecombine(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + + pr_debug("%s: preallocate_dma_buffer: area=%p, addr=%p, size=%d\n", + __func__, (void *) buf->area, (void *) buf->addr, size); + + if (!buf->area) + return -ENOMEM; + + buf->bytes = size; + return 0; +} + +static void mx1_mx2_pcm_free_dma_buffers(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + + dma_free_writecombine(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + +int mx1_mx2_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, + struct snd_pcm *pcm) +{ + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &mx1_mx2_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = 0xffffffff; + + if (dai->playback.channels_min) { + ret = mx1_mx2_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + pr_debug("%s: preallocate playback buffer\n", __func__); + if (ret) + goto out; + } + + if (dai->capture.channels_min) { + ret = mx1_mx2_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + pr_debug("%s: preallocate capture buffer\n", __func__); + if (ret) + goto out; + } + out: + return ret; +} + +struct snd_soc_platform mx1_mx2_soc_platform = { + .name = "mx1_mx2-audio", + .pcm_ops = &mx1_mx2_pcm_ops, + .pcm_new = mx1_mx2_pcm_new, + .pcm_free = mx1_mx2_pcm_free_dma_buffers, +}; +EXPORT_SYMBOL_GPL(mx1_mx2_soc_platform); + +static int __init mx1_mx2_soc_platform_init(void) +{ + return snd_soc_register_platform(&mx1_mx2_soc_platform); +} +module_init(mx1_mx2_soc_platform_init); + +static void __exit mx1_mx2_soc_platform_exit(void) +{ + snd_soc_unregister_platform(&mx1_mx2_soc_platform); +} +module_exit(mx1_mx2_soc_platform_exit); + +MODULE_AUTHOR("Javier Martin, javier.martin(a)vista-silicon.com"); +MODULE_DESCRIPTION("Freescale i.MX2x, i.MX1x PCM DMA module"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/imx/mx1_mx2-pcm.h b/sound/soc/imx/mx1_mx2-pcm.h new file mode 100644 index 0000000..e1e3a3f --- /dev/null +++ b/sound/soc/imx/mx1_mx2-pcm.h @@ -0,0 +1,48 @@ +/* + * mx1_mx2-pcm.h :- ASoC platform header for Freescale i.MX1x, i.MX2x + * + * 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 _MXC_PCM_H +#define _MXC_PCM_H + +/* AUDMUX register definitions */ +#define AUDMUX_IO_BASE_ADDR IO_ADDRESS(AUDMUX_BASE_ADDR) + +#define DAM_HPCR1 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x00))) +#define DAM_HPCR2 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x04))) +#define DAM_HPCR3 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x08))) +#define DAM_PPCR1 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x10))) +#define DAM_PPCR2 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x14))) +#define DAM_PPCR3 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x1C))) + +#define AUDMUX_HPCR_TFSDIR (1 << 31) +#define AUDMUX_HPCR_TCLKDIR (1 << 30) +#define AUDMUX_HPCR_TFCSEL(x) (((x) & 0xff) << 26) +#define AUDMUX_HPCR_RXDSEL(x) (((x) & 0x7) << 13) +#define AUDMUX_HPCR_SYN (1 << 12) + +#define AUDMUX_PPCR_TFSDIR (1 << 31) +#define AUDMUX_PPCR_TCLKDIR (1 << 30) +#define AUDMUX_PPCR_TFCSEL(x) (((x) & 0xff) << 26) +#define AUDMUX_PPCR_RXDSEL(x) (((x) & 0x7) << 13) +#define AUDMUX_PPCR_SYN (1 << 12) + +/* DMA information for mx1_mx2 platforms */ +struct mx1_mx2_pcm_dma_params { + char *name; /* stream identifier */ + unsigned int transfer_type; /* READ or WRITE DMA transfer */ + dma_addr_t per_address; /* physical address of SSI fifo */ + int event_id; /* fixed DMA number for SSI fifo */ + int watermark_level; /* SSI fifo watermark level */ + int per_config; /* DMA Config flags for peripheral */ + int mem_config; /* DMA Config flags for RAM */ + }; + +/* platform data */ +extern struct snd_soc_platform mx1_mx2_soc_platform; + +#endif -- -- Javier Martin Vista Silicon S.L. Universidad de Cantabria CDTUC - FASE C - Oficina S-345 Avda de los Castros s/n 39005- Santander. Cantabria. Spain +34 942 25 32 60 www.vista-silicon.com

2 1

[alsa-devel] [PATCH - BT and FM audio for zoom2 1/2] Add clock-only codec to provide McBSP clock source.
by sean.mcneil＠ti.com 04 Aug '09

04 Aug '09

From: Sean McNeil <sean.mcneil(a)ti.com> Signed-off-by: Sean McNeil <sean.mcneil(a)ti.com> diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 818fb37..c343ceb 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -56,7 +56,7 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0x00, /* REG_AVTXL2PGA (0xC) */ 0x00, /* REG_AVTXR2PGA (0xD) */ 0x01, /* REG_AUDIO_IF (0xE) */ - 0x00, /* REG_VOICE_IF (0xF) */ + 0x04, /* REG_VOICE_IF (0xF) */ 0x00, /* REG_ARXR1PGA (0x10) */ 0x00, /* REG_ARXL1PGA (0x11) */ 0x6c, /* REG_ARXR2PGA (0x12) */ @@ -118,25 +118,32 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0x00, /* REG_SW_SHADOW (0x4A) - Shadow, non HW register */ }; +struct substream_item { + struct list_head started; + struct list_head configured; + struct snd_pcm_substream *substream; + int use256FS; +}; + /* codec private data */ struct twl4030_priv { + struct mutex mutex; + + unsigned int extClock; unsigned int bypass_state; unsigned int codec_powered; unsigned int codec_muted; - struct snd_pcm_substream *master_substream; - struct snd_pcm_substream *slave_substream; - - unsigned int configured; - unsigned int rate; - unsigned int sample_bits; - unsigned int channels; + struct list_head started_list; + struct list_head config_list; unsigned int sysclk; /* Headset output state handling */ unsigned int hsl_enabled; unsigned int hsr_enabled; + + struct snd_pcm_hw_params params; }; /* @@ -966,7 +973,7 @@ static int snd_soc_put_twl4030_opmode_enum_double(struct snd_kcontrol *kcontrol, unsigned short val; unsigned short mask, bitmask; - if (twl4030->configured) { + if (!list_empty(&twl4030->config_list)) { printk(KERN_ERR "twl4030 operation mode cannot be " "changed on-the-fly\n"); return -EBUSY; @@ -1534,34 +1541,68 @@ static int twl4030_set_bias_level(struct snd_soc_codec *codec, return 0; } -static void twl4030_constraints(struct twl4030_priv *twl4030, - struct snd_pcm_substream *mst_substream) +static unsigned int twl4030_rate_min(struct substream_item *item, + unsigned int rate) { - struct snd_pcm_substream *slv_substream; - - /* Pick the stream, which need to be constrained */ - if (mst_substream == twl4030->master_substream) - slv_substream = twl4030->slave_substream; - else if (mst_substream == twl4030->slave_substream) - slv_substream = twl4030->master_substream; - else /* This should not happen.. */ - return; + static const unsigned int table[] = { + 8000, 11025, 12000, 16000, 22050, + 24000, 32000, 44100, 48000, 96000}; + unsigned int value = rate; + + if (item->use256FS) { + int i; + rate *= 256; + for (i = 0; i < ARRAY_SIZE(table); i++) + if (rate % table[i] == 0) { + value = table[i]; + break; + } + } + return value; +} - /* Set the constraints according to the already configured stream */ - snd_pcm_hw_constraint_minmax(slv_substream->runtime, - SNDRV_PCM_HW_PARAM_RATE, - twl4030->rate, - twl4030->rate); - - snd_pcm_hw_constraint_minmax(slv_substream->runtime, - SNDRV_PCM_HW_PARAM_SAMPLE_BITS, - twl4030->sample_bits, - twl4030->sample_bits); - - snd_pcm_hw_constraint_minmax(slv_substream->runtime, - SNDRV_PCM_HW_PARAM_CHANNELS, - twl4030->channels, - twl4030->channels); +static unsigned int twl4030_rate_max(struct substream_item *item, + unsigned int rate) +{ + static const unsigned int table[] = { + 96000, 48000, 44100, 32000, 24000, + 22050, 16000, 12000, 11025, 8000}; + unsigned int value = rate; + + if (item->use256FS) { + int i; + rate *= 256; + for (i = 0; i < ARRAY_SIZE(table); i++) + if (rate % table[i] == 0) { + value = table[i]; + break; + } + } + return value; +} + +static void twl4030_constraints(struct twl4030_priv *twl4030) +{ + struct substream_item *item; + unsigned int value; + + list_for_each_entry(item, &twl4030->started_list, started) { + + /* Set constraints according to the already configured stream */ + value = params_rate(&twl4030->params); + if (value) + snd_pcm_hw_constraint_minmax(item->substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + twl4030_rate_min(item, value), + twl4030_rate_max(item, value)); + + value = hw_param_interval(&twl4030->params, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS)->min; + if (value && !item->use256FS) + snd_pcm_hw_constraint_minmax(item->substream->runtime, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + value, value); + } } /* In case of 4 channel mode, the RX1 L/R for playback and the TX2 L/R for @@ -1586,6 +1627,53 @@ static void twl4030_tdm_enable(struct snd_soc_codec *codec, int direction, twl4030_write(codec, TWL4030_REG_OPTION, reg); } +static int twl4030_new_substream(struct twl4030_priv *twl4030, + struct snd_pcm_substream *substream, int use256FS) +{ + struct substream_item *item; + + item = kzalloc(sizeof(struct snd_pcm_substream), GFP_KERNEL); + if (!item) + return -ENOMEM; + + item->substream = substream; + item->use256FS = use256FS; + + mutex_lock(&twl4030->mutex); + list_add_tail(&item->started, &twl4030->started_list); + twl4030->extClock += item->use256FS; + mutex_unlock(&twl4030->mutex); + + return 0; +} + +static void twl4030_del_substream(struct twl4030_priv *twl4030, + struct snd_pcm_substream *substream) +{ + struct substream_item *item; + + mutex_lock(&twl4030->mutex); + + list_for_each_entry(item, &twl4030->config_list, configured) { + if (item->substream == substream) { + printk(KERN_ERR "TWL4030 deleted substream still configured!\n"); + list_del(&item->configured); + break; + } + } + + list_for_each_entry(item, &twl4030->started_list, started) { + if (item->substream == substream) { + list_del(&item->started); + twl4030->extClock -= item->use256FS; + kfree(item); + break; + } + } + + mutex_unlock(&twl4030->mutex); +} + static int twl4030_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -1594,82 +1682,45 @@ static int twl4030_startup(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = socdev->card->codec; struct twl4030_priv *twl4030 = codec->private_data; - if (twl4030->master_substream) { - twl4030->slave_substream = substream; - /* The DAI has one configuration for playback and capture, so - * if the DAI has been already configured then constrain this - * substream to match it. */ - if (twl4030->configured) - twl4030_constraints(twl4030, twl4030->master_substream); - } else { - if (!(twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE) & - TWL4030_OPTION_1)) { - /* In option2 4 channel is not supported, set the - * constraint for the first stream for channels, the - * second stream will 'inherit' this cosntraint */ - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_CHANNELS, - 2, 2); - } - twl4030->master_substream = substream; + if (!(twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE) & + TWL4030_OPTION_1)) { + /* In option2 4 channel is not supported, set the + * constraint for the first stream for channels, the + * second stream will 'inherit' this cosntraint */ + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_CHANNELS, + 2, 2); } - return 0; + return twl4030_new_substream(twl4030, substream, 0); } static void twl4030_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; struct twl4030_priv *twl4030 = codec->private_data; - if (twl4030->master_substream == substream) - twl4030->master_substream = twl4030->slave_substream; - - twl4030->slave_substream = NULL; - - /* If all streams are closed, or the remaining stream has not yet - * been configured than set the DAI as not configured. */ - if (!twl4030->master_substream) - twl4030->configured = 0; - else if (!twl4030->master_substream->runtime->channels) - twl4030->configured = 0; + twl4030_del_substream(twl4030, substream); /* If the closing substream had 4 channel, do the necessary cleanup */ if (substream->runtime->channels == 4) twl4030_tdm_enable(codec, substream->stream, 0); } -static int twl4030_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) +int twl4030_set_rate(struct snd_soc_codec *codec, + struct snd_pcm_hw_params *params) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->card->codec; struct twl4030_priv *twl4030 = codec->private_data; - u8 mode, old_mode, format, old_format; + u8 mode, old_mode; - /* If the substream has 4 channel, do the necessary setup */ - if (params_channels(params) == 4) { - format = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF); - mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE); - - /* Safety check: are we in the correct operating mode and - * the interface is in TDM mode? */ - if ((mode & TWL4030_OPTION_1) && - ((format & TWL4030_AIF_FORMAT) == TWL4030_AIF_FORMAT_TDM)) - twl4030_tdm_enable(codec, substream->stream, 1); - else - return -EINVAL; + if (params_rate(&twl4030->params) && + params_rate(&twl4030->params) != params_rate(params)) { + return -EBUSY; } - if (twl4030->configured) - /* Ignoring hw_params for already configured DAI */ - return 0; - /* bit rate */ old_mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE) & ~TWL4030_CODECPDZ; @@ -1712,6 +1763,8 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + params_rate(&twl4030->params) = params_rate(params); + if (mode != old_mode) { /* change rate and set CODECPDZ */ twl4030_codec_enable(codec, 0); @@ -1719,10 +1772,51 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, twl4030_codec_enable(codec, 1); } + return 0; +} +EXPORT_SYMBOL_GPL(twl4030_set_rate); + +int twl4030_get_clock_divisor(struct snd_soc_codec *codec, + struct snd_pcm_hw_params *params) +{ + struct twl4030_priv *twl4030 = codec->private_data; + int clock, divisor; + + clock = params_rate(&twl4030->params) * 256; + divisor = clock / params_rate(params); + divisor /= params_channels(params); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_U8: + case SNDRV_PCM_FORMAT_S8: + divisor /= 8; + break; + case SNDRV_PCM_FORMAT_S16_LE: + divisor /= 16; + break; + case SNDRV_PCM_FORMAT_S24_LE: + divisor /= 24; + break; + default: + printk(KERN_ERR "TWL4030 get_clock_divisor: unknown format %d\n", + params_format(params)); + return -EINVAL; + } + + return divisor; +} +EXPORT_SYMBOL_GPL(twl4030_get_clock_divisor); + +static int twl4030_set_format(struct snd_soc_codec *codec, + struct snd_pcm_hw_params *params) +{ + struct twl4030_priv *twl4030 = codec->private_data; + u8 format, old_format; + /* sample size */ old_format = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF); - format = old_format; - format &= ~TWL4030_DATA_WIDTH; + format = old_format & ~TWL4030_DATA_WIDTH; + switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: format |= TWL4030_DATA_WIDTH_16S_16W; @@ -1736,31 +1830,96 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - if (format != old_format) { + if (format == old_format) + return 0; - /* clear CODECPDZ before changing format (codec requirement) */ - twl4030_codec_enable(codec, 0); + if (params_format(&twl4030->params) && + params_format(&twl4030->params) != params_format(params)) + return -EBUSY; - /* change format */ - twl4030_write(codec, TWL4030_REG_AUDIO_IF, format); + *hw_param_mask(&twl4030->params, SNDRV_PCM_HW_PARAM_FORMAT) = + *hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - /* set CODECPDZ afterwards */ - twl4030_codec_enable(codec, 1); + /* clear CODECPDZ before changing format (codec requirement) */ + twl4030_codec_enable(codec, 0); + + /* change format */ + twl4030_write(codec, TWL4030_REG_AUDIO_IF, format); + + /* set CODECPDZ afterwards */ + twl4030_codec_enable(codec, 1); + + return 0; +} + +static int twl4030_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct twl4030_priv *twl4030 = codec->private_data; + int rval; + + mutex_lock(&twl4030->mutex); + + /* If the substream has 4 channel, do the necessary setup */ + if (params_channels(params) == 4) { + /* Safety check: are we in the correct operating mode? */ + if ((twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE) & + TWL4030_OPTION_1)) { + twl4030_tdm_enable(codec, substream->stream, 1); + } else { + mutex_unlock(&twl4030->mutex); + return -EINVAL; + } } - /* Store the important parameters for the DAI configuration and set - * the DAI as configured */ - twl4030->configured = 1; - twl4030->rate = params_rate(params); - twl4030->sample_bits = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_SAMPLE_BITS)->min; - twl4030->channels = params_channels(params); + rval = twl4030_set_rate(codec, params); + if (rval < 0) { + mutex_unlock(&twl4030->mutex); + return rval; + } - /* If both playback and capture streams are open, and one of them + rval = twl4030_set_format(codec, params); + if (rval < 0) { + mutex_unlock(&twl4030->mutex); + return rval; + } + + /* If any other streams are currently open, and one of them * is setting the hw parameters right now (since we are here), set - * constraints to the other stream to match the current one. */ - if (twl4030->slave_substream) - twl4030_constraints(twl4030, substream); + * constraints to the other stream(s) to match the current one. */ + twl4030_constraints(twl4030); + + mutex_unlock(&twl4030->mutex); + + return 0; +} + +static int twl4030_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct twl4030_priv *twl4030 = codec->private_data; + struct substream_item *item; + + mutex_lock(&twl4030->mutex); + + list_for_each_entry(item, &twl4030->config_list, configured) { + if (item->substream == substream) { + list_del(&item->configured); + break; + } + } + + if (list_empty(&twl4030->config_list)) + memset(&twl4030->params, 0, sizeof(twl4030->params)); + + mutex_unlock(&twl4030->mutex); return 0; } @@ -1797,10 +1956,39 @@ static int twl4030_set_dai_sysclk(struct snd_soc_dai *codec_dai, return 0; } +static int twl4030_set_ext_clock(struct snd_soc_codec *codec, int enable) +{ + u8 old_format, format; + + /* get format */ + old_format = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF); + + if (enable) + format = old_format | TWL4030_CLK256FS_EN; + else + format = old_format & ~TWL4030_CLK256FS_EN; + + if (format != old_format) { + + /* clear CODECPDZ before changing format (codec requirement) */ + twl4030_codec_enable(codec, 0); + + /* change format */ + twl4030_write(codec, TWL4030_REG_AUDIO_IF, format); + + /* set CODECPDZ afterwards */ + twl4030_codec_enable(codec, 1); + } + + return 0; +} + static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; + struct twl4030_priv *twl4030 = codec->private_data; + int use256FS = 0; u8 old_format, format; /* get format */ @@ -1811,11 +1999,10 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: format &= ~(TWL4030_AIF_SLAVE_EN); - format &= ~(TWL4030_CLK256FS_EN); break; case SND_SOC_DAIFMT_CBS_CFS: format |= TWL4030_AIF_SLAVE_EN; - format |= TWL4030_CLK256FS_EN; + use256FS = 1; break; default: return -EINVAL; @@ -1846,7 +2033,7 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, twl4030_codec_enable(codec, 1); } - return 0; + return twl4030_set_ext_clock(codec, use256FS | twl4030->extClock); } static int twl4030_set_tristate(struct snd_soc_dai *dai, int tristate) @@ -1890,6 +2077,7 @@ static int twl4030_voice_startup(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; + struct twl4030_priv *twl4030 = codec->private_data; u8 infreq; u8 mode; @@ -1917,7 +2105,7 @@ static int twl4030_voice_startup(struct snd_pcm_substream *substream, return -EINVAL; } - return 0; + return twl4030_new_substream(twl4030, substream, 1); } static void twl4030_voice_shutdown(struct snd_pcm_substream *substream, @@ -1926,6 +2114,9 @@ static void twl4030_voice_shutdown(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; + struct twl4030_priv *twl4030 = codec->private_data; + + twl4030_del_substream(twl4030, substream); /* Enable voice digital filters */ twl4030_voice_enable(codec, substream->stream, 0); @@ -1937,15 +2128,18 @@ static int twl4030_voice_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; + struct twl4030_priv *twl4030 = codec->private_data; u8 old_mode, mode; /* Enable voice digital filters */ twl4030_voice_enable(codec, substream->stream, 1); + mutex_lock(&twl4030->mutex); + /* bit rate */ - old_mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE) - & ~(TWL4030_CODECPDZ); - mode = old_mode; + old_mode = twl4030_read_reg_cache(codec, + TWL4030_REG_CODEC_MODE) & ~TWL4030_CODECPDZ; + mode = old_mode & ~TWL4030_APLL_RATE; switch (params_rate(params)) { case 8000: @@ -1955,6 +2149,7 @@ static int twl4030_voice_hw_params(struct snd_pcm_substream *substream, mode |= TWL4030_SEL_16K; break; default: + mutex_unlock(&twl4030->mutex); printk(KERN_ERR "TWL4030 voice hw params: unknown rate %d\n", params_rate(params)); return -EINVAL; @@ -1967,6 +2162,7 @@ static int twl4030_voice_hw_params(struct snd_pcm_substream *substream, twl4030_codec_enable(codec, 1); } + mutex_unlock(&twl4030->mutex); return 0; } @@ -1996,19 +2192,22 @@ static int twl4030_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; + struct twl4030_priv *twl4030 = codec->private_data; + int use256FS = 0; u8 old_format, format; /* get format */ old_format = twl4030_read_reg_cache(codec, TWL4030_REG_VOICE_IF); - format = old_format; + format = old_format & ~TWL4030_VIF_TRI_EN; /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBS_CFM: format &= ~(TWL4030_VIF_SLAVE_EN); break; case SND_SOC_DAIFMT_CBS_CFS: format |= TWL4030_VIF_SLAVE_EN; + use256FS = 1; break; default: return -EINVAL; @@ -2033,7 +2232,59 @@ static int twl4030_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, twl4030_codec_enable(codec, 1); } - return 0; + return twl4030_set_ext_clock(codec, use256FS | twl4030->extClock); +} + +static int twl4030_clock_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct twl4030_priv *twl4030 = codec->private_data; + + return twl4030_new_substream(twl4030, substream, 1); +} + +static int twl4030_clock_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct twl4030_priv *twl4030 = codec->private_data; + int rval; + + mutex_lock(&twl4030->mutex); + + rval = twl4030_set_rate(codec, params); + + /* See if we are a multiple of the current FS. If so, then still OK. */ + if (rval) { + int divisor = twl4030_get_clock_divisor(codec, params); + int clock = params_rate(&twl4030->params) * 256; + int remainder = clock % params_rate(params); + + if (remainder == 0 && divisor <= 256) + rval = 0; + } + + /* If any other streams are currently open, and one of them + * is setting the hw parameters right now (since we are here), set + * constraints to the other stream(s) to match the current one. */ + twl4030_constraints(twl4030); + + mutex_unlock(&twl4030->mutex); + + return rval; +} + +static int twl4030_clock_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + + return twl4030_set_ext_clock(codec, 1); } static int twl4030_voice_set_tristate(struct snd_soc_dai *dai, int tristate) @@ -2056,6 +2307,7 @@ static struct snd_soc_dai_ops twl4030_dai_ops = { .startup = twl4030_startup, .shutdown = twl4030_shutdown, .hw_params = twl4030_hw_params, + .hw_free = twl4030_hw_free, .set_sysclk = twl4030_set_dai_sysclk, .set_fmt = twl4030_set_dai_fmt, .set_tristate = twl4030_set_tristate, @@ -2065,26 +2317,38 @@ static struct snd_soc_dai_ops twl4030_dai_voice_ops = { .startup = twl4030_voice_startup, .shutdown = twl4030_voice_shutdown, .hw_params = twl4030_voice_hw_params, + .hw_free = twl4030_hw_free, .set_sysclk = twl4030_voice_set_dai_sysclk, .set_fmt = twl4030_voice_set_dai_fmt, .set_tristate = twl4030_voice_set_tristate, }; +static struct snd_soc_dai_ops twl4030_dai_clock_ops = { + .startup = twl4030_clock_startup, + .shutdown = twl4030_shutdown, + .hw_params = twl4030_clock_hw_params, + .hw_free = twl4030_hw_free, + .set_sysclk = twl4030_set_dai_sysclk, + .set_fmt = twl4030_clock_set_dai_fmt, +}; + struct snd_soc_dai twl4030_dai[] = { { .name = "twl4030", .playback = { .stream_name = "HiFi Playback", - .channels_min = 2, + .channels_min = 1, .channels_max = 4, .rates = TWL4030_RATES | SNDRV_PCM_RATE_96000, - .formats = TWL4030_FORMATS,}, + .formats = TWL4030_FORMATS, + }, .capture = { .stream_name = "Capture", - .channels_min = 2, + .channels_min = 1, .channels_max = 4, .rates = TWL4030_RATES, - .formats = TWL4030_FORMATS,}, + .formats = TWL4030_FORMATS, + }, .ops = &twl4030_dai_ops, }, { @@ -2092,17 +2356,37 @@ struct snd_soc_dai twl4030_dai[] = { .playback = { .stream_name = "Voice Playback", .channels_min = 1, - .channels_max = 1, + .channels_max = 2, .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, .capture = { .stream_name = "Capture", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, .ops = &twl4030_dai_voice_ops, }, +{ + .name = "twl4030 Clock", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = TWL4030_RATES, + .formats = SNDRV_PCM_FMTBIT_U8 | TWL4030_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = TWL4030_RATES, + .formats = SNDRV_PCM_FMTBIT_U8 | TWL4030_FORMATS, + }, + .ops = &twl4030_dai_clock_ops, +}, }; EXPORT_SYMBOL_GPL(twl4030_dai); @@ -2220,6 +2504,9 @@ static int twl4030_probe(struct platform_device *pdev) return -ENOMEM; } + mutex_init(&twl4030->mutex); + INIT_LIST_HEAD(&twl4030->started_list); + INIT_LIST_HEAD(&twl4030->config_list); codec->private_data = twl4030; socdev->card->codec = codec; mutex_init(&codec->mutex); @@ -2257,13 +2544,13 @@ EXPORT_SYMBOL_GPL(soc_codec_dev_twl4030); static int __init twl4030_modinit(void) { - return snd_soc_register_dais(&twl4030_dai[0], ARRAY_SIZE(twl4030_dai)); + return snd_soc_register_dais(twl4030_dai, ARRAY_SIZE(twl4030_dai)); } module_init(twl4030_modinit); static void __exit twl4030_exit(void) { - snd_soc_unregister_dais(&twl4030_dai[0], ARRAY_SIZE(twl4030_dai)); + snd_soc_unregister_dais(twl4030_dai, ARRAY_SIZE(twl4030_dai)); } module_exit(twl4030_exit); diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index 2b4bfa2..fa23613 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -267,10 +267,15 @@ #define TWL4030_DAI_HIFI 0 #define TWL4030_DAI_VOICE 1 +#define TWL4030_DAI_CLOCK 2 -extern struct snd_soc_dai twl4030_dai[2]; +extern struct snd_soc_dai twl4030_dai[]; extern struct snd_soc_codec_device soc_codec_dev_twl4030; +extern int twl4030_set_rate(struct snd_soc_codec *, struct snd_pcm_hw_params *); +extern int twl4030_get_clock_divisor(struct snd_soc_codec *, + struct snd_pcm_hw_params *); + struct twl4030_setup_data { unsigned int ramp_delay_value; unsigned int sysclk; -- 1.6.0.4

3 3

[alsa-devel] [PATCH 1/8] ASoC: Begin to factor out register cache I/O functions
by Mark Brown 03 Aug '09

03 Aug '09

A lot of CODECs share the same register data formats and therefore replicate the code to manage access to and caching of the register map. In order to reduce code duplication centralised versions of this code will be introduced with drivers able to configure the use of the common code by calling the new snd_soc_codec_set_cache_io() API call during startup. As an initial user the 7 bit address/9 bit data format used by many Wolfson devices is supported for write only CODECs and the drivers with straightforward register cache implementations are converted to use it. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 2 + sound/soc/Makefile | 2 +- sound/soc/codecs/wm8510.c | 141 +++++++++++--------------------- sound/soc/codecs/wm8728.c | 78 +++++------------- sound/soc/codecs/wm8731.c | 108 +++++++------------------ sound/soc/codecs/wm8750.c | 121 ++++++++++------------------ sound/soc/codecs/wm8960.c | 194 +++++++++++++++++---------------------------- sound/soc/codecs/wm8971.c | 121 ++++++++++------------------ sound/soc/codecs/wm8988.c | 105 ++++++++----------------- sound/soc/soc-cache.c | 105 ++++++++++++++++++++++++ 10 files changed, 400 insertions(+), 577 deletions(-) create mode 100644 sound/soc/soc-cache.c diff --git a/include/sound/soc.h b/include/sound/soc.h index 94fcc65..27409dd 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -192,6 +192,8 @@ void snd_soc_unregister_platform(struct snd_soc_platform *platform); int snd_soc_register_codec(struct snd_soc_codec *codec); void snd_soc_unregister_codec(struct snd_soc_codec *codec); int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, int reg); +int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, + int addr_bits, int data_bits); #ifdef CONFIG_PM int snd_soc_suspend_device(struct device *dev); diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 6f1e28d..4eaf48a 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,4 +1,4 @@ -snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o +snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o obj-$(CONFIG_SND_SOC) += codecs/ diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index c8b8dba..7a169bf 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -58,55 +58,7 @@ static const u16 wm8510_reg[WM8510_CACHEREGNUM] = { #define WM8510_POWER1_BIASEN 0x08 #define WM8510_POWER1_BUFIOEN 0x10 -/* - * read wm8510 register cache - */ -static inline unsigned int wm8510_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u16 *cache = codec->reg_cache; - if (reg == WM8510_RESET) - return 0; - if (reg >= WM8510_CACHEREGNUM) - return -1; - return cache[reg]; -} - -/* - * write wm8510 register cache - */ -static inline void wm8510_write_reg_cache(struct snd_soc_codec *codec, - u16 reg, unsigned int value) -{ - u16 *cache = codec->reg_cache; - if (reg >= WM8510_CACHEREGNUM) - return; - cache[reg] = value; -} - -/* - * write to the WM8510 register space - */ -static int wm8510_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[2]; - - /* data is - * D15..D9 WM8510 register offset - * D8...D0 register data - */ - data[0] = (reg << 1) | ((value >> 8) & 0x0001); - data[1] = value & 0x00ff; - - wm8510_write_reg_cache(codec, reg, value); - if (codec->hw_write(codec->control_data, data, 2) == 2) - return 0; - else - return -EIO; -} - -#define wm8510_reset(c) wm8510_write(c, WM8510_RESET, 0) +#define wm8510_reset(c) snd_soc_write(c, WM8510_RESET, 0) static const char *wm8510_companding[] = { "Off", "NC", "u-law", "A-law" }; static const char *wm8510_deemp[] = { "None", "32kHz", "44.1kHz", "48kHz" }; @@ -327,27 +279,27 @@ static int wm8510_set_dai_pll(struct snd_soc_dai *codec_dai, if (freq_in == 0 || freq_out == 0) { /* Clock CODEC directly from MCLK */ - reg = wm8510_read_reg_cache(codec, WM8510_CLOCK); - wm8510_write(codec, WM8510_CLOCK, reg & 0x0ff); + reg = snd_soc_read(codec, WM8510_CLOCK); + snd_soc_write(codec, WM8510_CLOCK, reg & 0x0ff); /* Turn off PLL */ - reg = wm8510_read_reg_cache(codec, WM8510_POWER1); - wm8510_write(codec, WM8510_POWER1, reg & 0x1df); + reg = snd_soc_read(codec, WM8510_POWER1); + snd_soc_write(codec, WM8510_POWER1, reg & 0x1df); return 0; } pll_factors(freq_out*4, freq_in); - wm8510_write(codec, WM8510_PLLN, (pll_div.pre_div << 4) | pll_div.n); - wm8510_write(codec, WM8510_PLLK1, pll_div.k >> 18); - wm8510_write(codec, WM8510_PLLK2, (pll_div.k >> 9) & 0x1ff); - wm8510_write(codec, WM8510_PLLK3, pll_div.k & 0x1ff); - reg = wm8510_read_reg_cache(codec, WM8510_POWER1); - wm8510_write(codec, WM8510_POWER1, reg | 0x020); + snd_soc_write(codec, WM8510_PLLN, (pll_div.pre_div << 4) | pll_div.n); + snd_soc_write(codec, WM8510_PLLK1, pll_div.k >> 18); + snd_soc_write(codec, WM8510_PLLK2, (pll_div.k >> 9) & 0x1ff); + snd_soc_write(codec, WM8510_PLLK3, pll_div.k & 0x1ff); + reg = snd_soc_read(codec, WM8510_POWER1); + snd_soc_write(codec, WM8510_POWER1, reg | 0x020); /* Run CODEC from PLL instead of MCLK */ - reg = wm8510_read_reg_cache(codec, WM8510_CLOCK); - wm8510_write(codec, WM8510_CLOCK, reg | 0x100); + reg = snd_soc_read(codec, WM8510_CLOCK); + snd_soc_write(codec, WM8510_CLOCK, reg | 0x100); return 0; } @@ -363,24 +315,24 @@ static int wm8510_set_dai_clkdiv(struct snd_soc_dai *codec_dai, switch (div_id) { case WM8510_OPCLKDIV: - reg = wm8510_read_reg_cache(codec, WM8510_GPIO) & 0x1cf; - wm8510_write(codec, WM8510_GPIO, reg | div); + reg = snd_soc_read(codec, WM8510_GPIO) & 0x1cf; + snd_soc_write(codec, WM8510_GPIO, reg | div); break; case WM8510_MCLKDIV: - reg = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x11f; - wm8510_write(codec, WM8510_CLOCK, reg | div); + reg = snd_soc_read(codec, WM8510_CLOCK) & 0x11f; + snd_soc_write(codec, WM8510_CLOCK, reg | div); break; case WM8510_ADCCLK: - reg = wm8510_read_reg_cache(codec, WM8510_ADC) & 0x1f7; - wm8510_write(codec, WM8510_ADC, reg | div); + reg = snd_soc_read(codec, WM8510_ADC) & 0x1f7; + snd_soc_write(codec, WM8510_ADC, reg | div); break; case WM8510_DACCLK: - reg = wm8510_read_reg_cache(codec, WM8510_DAC) & 0x1f7; - wm8510_write(codec, WM8510_DAC, reg | div); + reg = snd_soc_read(codec, WM8510_DAC) & 0x1f7; + snd_soc_write(codec, WM8510_DAC, reg | div); break; case WM8510_BCLKDIV: - reg = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x1e3; - wm8510_write(codec, WM8510_CLOCK, reg | div); + reg = snd_soc_read(codec, WM8510_CLOCK) & 0x1e3; + snd_soc_write(codec, WM8510_CLOCK, reg | div); break; default: return -EINVAL; @@ -394,7 +346,7 @@ static int wm8510_set_dai_fmt(struct snd_soc_dai *codec_dai, { struct snd_soc_codec *codec = codec_dai->codec; u16 iface = 0; - u16 clk = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x1fe; + u16 clk = snd_soc_read(codec, WM8510_CLOCK) & 0x1fe; /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { @@ -441,8 +393,8 @@ static int wm8510_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - wm8510_write(codec, WM8510_IFACE, iface); - wm8510_write(codec, WM8510_CLOCK, clk); + snd_soc_write(codec, WM8510_IFACE, iface); + snd_soc_write(codec, WM8510_CLOCK, clk); return 0; } @@ -453,8 +405,8 @@ static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; - u16 iface = wm8510_read_reg_cache(codec, WM8510_IFACE) & 0x19f; - u16 adn = wm8510_read_reg_cache(codec, WM8510_ADD) & 0x1f1; + u16 iface = snd_soc_read(codec, WM8510_IFACE) & 0x19f; + u16 adn = snd_soc_read(codec, WM8510_ADD) & 0x1f1; /* bit size */ switch (params_format(params)) { @@ -493,20 +445,20 @@ static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream, break; } - wm8510_write(codec, WM8510_IFACE, iface); - wm8510_write(codec, WM8510_ADD, adn); + snd_soc_write(codec, WM8510_IFACE, iface); + snd_soc_write(codec, WM8510_ADD, adn); return 0; } static int wm8510_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - u16 mute_reg = wm8510_read_reg_cache(codec, WM8510_DAC) & 0xffbf; + u16 mute_reg = snd_soc_read(codec, WM8510_DAC) & 0xffbf; if (mute) - wm8510_write(codec, WM8510_DAC, mute_reg | 0x40); + snd_soc_write(codec, WM8510_DAC, mute_reg | 0x40); else - wm8510_write(codec, WM8510_DAC, mute_reg); + snd_soc_write(codec, WM8510_DAC, mute_reg); return 0; } @@ -514,13 +466,13 @@ static int wm8510_mute(struct snd_soc_dai *dai, int mute) static int wm8510_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 power1 = wm8510_read_reg_cache(codec, WM8510_POWER1) & ~0x3; + u16 power1 = snd_soc_read(codec, WM8510_POWER1) & ~0x3; switch (level) { case SND_SOC_BIAS_ON: case SND_SOC_BIAS_PREPARE: power1 |= 0x1; /* VMID 50k */ - wm8510_write(codec, WM8510_POWER1, power1); + snd_soc_write(codec, WM8510_POWER1, power1); break; case SND_SOC_BIAS_STANDBY: @@ -528,18 +480,18 @@ static int wm8510_set_bias_level(struct snd_soc_codec *codec, if (codec->bias_level == SND_SOC_BIAS_OFF) { /* Initial cap charge at VMID 5k */ - wm8510_write(codec, WM8510_POWER1, power1 | 0x3); + snd_soc_write(codec, WM8510_POWER1, power1 | 0x3); mdelay(100); } power1 |= 0x2; /* VMID 500k */ - wm8510_write(codec, WM8510_POWER1, power1); + snd_soc_write(codec, WM8510_POWER1, power1); break; case SND_SOC_BIAS_OFF: - wm8510_write(codec, WM8510_POWER1, 0); - wm8510_write(codec, WM8510_POWER2, 0); - wm8510_write(codec, WM8510_POWER3, 0); + snd_soc_write(codec, WM8510_POWER1, 0); + snd_soc_write(codec, WM8510_POWER2, 0); + snd_soc_write(codec, WM8510_POWER3, 0); break; } @@ -619,8 +571,6 @@ static int wm8510_init(struct snd_soc_device *socdev) codec->name = "WM8510"; codec->owner = THIS_MODULE; - codec->read = wm8510_read_reg_cache; - codec->write = wm8510_write; codec->set_bias_level = wm8510_set_bias_level; codec->dai = &wm8510_dai; codec->num_dai = 1; @@ -630,13 +580,20 @@ static int wm8510_init(struct snd_soc_device *socdev) if (codec->reg_cache == NULL) return -ENOMEM; + ret = snd_soc_codec_set_cache_io(codec, 7, 9); + if (ret < 0) { + printk(KERN_ERR "wm8510: failed to set cache I/O: %d\n", + ret); + goto err; + } + wm8510_reset(codec); /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) { printk(KERN_ERR "wm8510: failed to create pcms\n"); - goto pcm_err; + goto err; } /* power on device */ @@ -655,7 +612,7 @@ static int wm8510_init(struct snd_soc_device *socdev) card_err: snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); -pcm_err: +err: kfree(codec->reg_cache); return ret; } diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index e7ff212..66da44b 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -43,45 +43,6 @@ static const u16 wm8728_reg_defaults[] = { 0x100, }; -static inline unsigned int wm8728_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u16 *cache = codec->reg_cache; - BUG_ON(reg >= ARRAY_SIZE(wm8728_reg_defaults)); - return cache[reg]; -} - -static inline void wm8728_write_reg_cache(struct snd_soc_codec *codec, - u16 reg, unsigned int value) -{ - u16 *cache = codec->reg_cache; - BUG_ON(reg >= ARRAY_SIZE(wm8728_reg_defaults)); - cache[reg] = value; -} - -/* - * write to the WM8728 register space - */ -static int wm8728_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[2]; - - /* data is - * D15..D9 WM8728 register offset - * D8...D0 register data - */ - data[0] = (reg << 1) | ((value >> 8) & 0x0001); - data[1] = value & 0x00ff; - - wm8728_write_reg_cache(codec, reg, value); - - if (codec->hw_write(codec->control_data, data, 2) == 2) - return 0; - else - return -EIO; -} - static const DECLARE_TLV_DB_SCALE(wm8728_tlv, -12750, 50, 1); static const struct snd_kcontrol_new wm8728_snd_controls[] = { @@ -121,12 +82,12 @@ static int wm8728_add_widgets(struct snd_soc_codec *codec) static int wm8728_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - u16 mute_reg = wm8728_read_reg_cache(codec, WM8728_DACCTL); + u16 mute_reg = snd_soc_read(codec, WM8728_DACCTL); if (mute) - wm8728_write(codec, WM8728_DACCTL, mute_reg | 1); + snd_soc_write(codec, WM8728_DACCTL, mute_reg | 1); else - wm8728_write(codec, WM8728_DACCTL, mute_reg & ~1); + snd_soc_write(codec, WM8728_DACCTL, mute_reg & ~1); return 0; } @@ -138,7 +99,7 @@ static int wm8728_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; - u16 dac = wm8728_read_reg_cache(codec, WM8728_DACCTL); + u16 dac = snd_soc_read(codec, WM8728_DACCTL); dac &= ~0x18; @@ -155,7 +116,7 @@ static int wm8728_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - wm8728_write(codec, WM8728_DACCTL, dac); + snd_soc_write(codec, WM8728_DACCTL, dac); return 0; } @@ -164,7 +125,7 @@ static int wm8728_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; - u16 iface = wm8728_read_reg_cache(codec, WM8728_IFCTL); + u16 iface = snd_soc_read(codec, WM8728_IFCTL); /* Currently only I2S is supported by the driver, though the * hardware is more flexible. @@ -204,7 +165,7 @@ static int wm8728_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - wm8728_write(codec, WM8728_IFCTL, iface); + snd_soc_write(codec, WM8728_IFCTL, iface); return 0; } @@ -220,19 +181,19 @@ static int wm8728_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->bias_level == SND_SOC_BIAS_OFF) { /* Power everything up... */ - reg = wm8728_read_reg_cache(codec, WM8728_DACCTL); - wm8728_write(codec, WM8728_DACCTL, reg & ~0x4); + reg = snd_soc_read(codec, WM8728_DACCTL); + snd_soc_write(codec, WM8728_DACCTL, reg & ~0x4); /* ..then sync in the register cache. */ for (i = 0; i < ARRAY_SIZE(wm8728_reg_defaults); i++) - wm8728_write(codec, i, - wm8728_read_reg_cache(codec, i)); + snd_soc_write(codec, i, + snd_soc_read(codec, i)); } break; case SND_SOC_BIAS_OFF: - reg = wm8728_read_reg_cache(codec, WM8728_DACCTL); - wm8728_write(codec, WM8728_DACCTL, reg | 0x4); + reg = snd_soc_read(codec, WM8728_DACCTL); + snd_soc_write(codec, WM8728_DACCTL, reg | 0x4); break; } codec->bias_level = level; @@ -294,8 +255,6 @@ static int wm8728_init(struct snd_soc_device *socdev) codec->name = "WM8728"; codec->owner = THIS_MODULE; - codec->read = wm8728_read_reg_cache; - codec->write = wm8728_write; codec->set_bias_level = wm8728_set_bias_level; codec->dai = &wm8728_dai; codec->num_dai = 1; @@ -307,11 +266,18 @@ static int wm8728_init(struct snd_soc_device *socdev) if (codec->reg_cache == NULL) return -ENOMEM; + ret = snd_soc_codec_set_cache_io(codec, 7, 9); + if (ret < 0) { + printk(KERN_ERR "wm8728: failed to configure cache I/O: %d\n", + ret); + goto err; + } + /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) { printk(KERN_ERR "wm8728: failed to create pcms\n"); - goto pcm_err; + goto err; } /* power on device */ @@ -331,7 +297,7 @@ static int wm8728_init(struct snd_soc_device *socdev) card_err: snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); -pcm_err: +err: kfree(codec->reg_cache); return ret; } diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index dfbc1bb..4eb84ff 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -50,60 +50,12 @@ static int wm8731_spi_write(struct spi_device *spi, const char *data, int len); * There is no point in caching the reset register */ static const u16 wm8731_reg[WM8731_CACHEREGNUM] = { - 0x0097, 0x0097, 0x0079, 0x0079, - 0x000a, 0x0008, 0x009f, 0x000a, - 0x0000, 0x0000 + 0x0097, 0x0097, 0x0079, 0x0079, + 0x000a, 0x0008, 0x009f, 0x000a, + 0x0000, 0x0000 }; -/* - * read wm8731 register cache - */ -static inline unsigned int wm8731_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u16 *cache = codec->reg_cache; - if (reg == WM8731_RESET) - return 0; - if (reg >= WM8731_CACHEREGNUM) - return -1; - return cache[reg]; -} - -/* - * write wm8731 register cache - */ -static inline void wm8731_write_reg_cache(struct snd_soc_codec *codec, - u16 reg, unsigned int value) -{ - u16 *cache = codec->reg_cache; - if (reg >= WM8731_CACHEREGNUM) - return; - cache[reg] = value; -} - -/* - * write to the WM8731 register space - */ -static int wm8731_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[2]; - - /* data is - * D15..D9 WM8731 register offset - * D8...D0 register data - */ - data[0] = (reg << 1) | ((value >> 8) & 0x0001); - data[1] = value & 0x00ff; - - wm8731_write_reg_cache(codec, reg, value); - if (codec->hw_write(codec->control_data, data, 2) == 2) - return 0; - else - return -EIO; -} - -#define wm8731_reset(c) wm8731_write(c, WM8731_RESET, 0) +#define wm8731_reset(c) snd_soc_write(c, WM8731_RESET, 0) static const char *wm8731_input_select[] = {"Line In", "Mic"}; static const char *wm8731_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"}; @@ -260,12 +212,12 @@ static int wm8731_hw_params(struct snd_pcm_substream *substream, struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; struct wm8731_priv *wm8731 = codec->private_data; - u16 iface = wm8731_read_reg_cache(codec, WM8731_IFACE) & 0xfff3; + u16 iface = snd_soc_read(codec, WM8731_IFACE) & 0xfff3; int i = get_coeff(wm8731->sysclk, params_rate(params)); u16 srate = (coeff_div[i].sr << 2) | (coeff_div[i].bosr << 1) | coeff_div[i].usb; - wm8731_write(codec, WM8731_SRATE, srate); + snd_soc_write(codec, WM8731_SRATE, srate); /* bit size */ switch (params_format(params)) { @@ -279,7 +231,7 @@ static int wm8731_hw_params(struct snd_pcm_substream *substream, break; } - wm8731_write(codec, WM8731_IFACE, iface); + snd_soc_write(codec, WM8731_IFACE, iface); return 0; } @@ -291,7 +243,7 @@ static int wm8731_pcm_prepare(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = socdev->card->codec; /* set active */ - wm8731_write(codec, WM8731_ACTIVE, 0x0001); + snd_soc_write(codec, WM8731_ACTIVE, 0x0001); return 0; } @@ -306,19 +258,19 @@ static void wm8731_shutdown(struct snd_pcm_substream *substream, /* deactivate */ if (!codec->active) { udelay(50); - wm8731_write(codec, WM8731_ACTIVE, 0x0); + snd_soc_write(codec, WM8731_ACTIVE, 0x0); } } static int wm8731_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - u16 mute_reg = wm8731_read_reg_cache(codec, WM8731_APDIGI) & 0xfff7; + u16 mute_reg = snd_soc_read(codec, WM8731_APDIGI) & 0xfff7; if (mute) - wm8731_write(codec, WM8731_APDIGI, mute_reg | 0x8); + snd_soc_write(codec, WM8731_APDIGI, mute_reg | 0x8); else - wm8731_write(codec, WM8731_APDIGI, mute_reg); + snd_soc_write(codec, WM8731_APDIGI, mute_reg); return 0; } @@ -396,7 +348,7 @@ static int wm8731_set_dai_fmt(struct snd_soc_dai *codec_dai, } /* set iface */ - wm8731_write(codec, WM8731_IFACE, iface); + snd_soc_write(codec, WM8731_IFACE, iface); return 0; } @@ -412,12 +364,12 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: /* Clear PWROFF, gate CLKOUT, everything else as-is */ - reg = wm8731_read_reg_cache(codec, WM8731_PWR) & 0xff7f; - wm8731_write(codec, WM8731_PWR, reg | 0x0040); + reg = snd_soc_read(codec, WM8731_PWR) & 0xff7f; + snd_soc_write(codec, WM8731_PWR, reg | 0x0040); break; case SND_SOC_BIAS_OFF: - wm8731_write(codec, WM8731_ACTIVE, 0x0); - wm8731_write(codec, WM8731_PWR, 0xffff); + snd_soc_write(codec, WM8731_ACTIVE, 0x0); + snd_soc_write(codec, WM8731_PWR, 0xffff); break; } codec->bias_level = level; @@ -466,7 +418,7 @@ static int wm8731_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; - wm8731_write(codec, WM8731_ACTIVE, 0x0); + snd_soc_write(codec, WM8731_ACTIVE, 0x0); wm8731_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } @@ -556,7 +508,6 @@ static int wm8731_register(struct wm8731_priv *wm8731) { int ret; struct snd_soc_codec *codec = &wm8731->codec; - u16 reg; if (wm8731_codec) { dev_err(codec->dev, "Another WM8731 is registered\n"); @@ -571,8 +522,6 @@ static int wm8731_register(struct wm8731_priv *wm8731) codec->private_data = wm8731; codec->name = "WM8731"; codec->owner = THIS_MODULE; - codec->read = wm8731_read_reg_cache; - codec->write = wm8731_write; codec->bias_level = SND_SOC_BIAS_OFF; codec->set_bias_level = wm8731_set_bias_level; codec->dai = &wm8731_dai; @@ -582,6 +531,12 @@ static int wm8731_register(struct wm8731_priv *wm8731) memcpy(codec->reg_cache, wm8731_reg, sizeof(wm8731_reg)); + ret = snd_soc_codec_set_cache_io(codec, 7, 9); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + goto err; + } + ret = wm8731_reset(codec); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset: %d\n", ret); @@ -593,18 +548,13 @@ static int wm8731_register(struct wm8731_priv *wm8731) wm8731_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Latch the update bits */ - reg = wm8731_read_reg_cache(codec, WM8731_LOUT1V); - wm8731_write(codec, WM8731_LOUT1V, reg & ~0x0100); - reg = wm8731_read_reg_cache(codec, WM8731_ROUT1V); - wm8731_write(codec, WM8731_ROUT1V, reg & ~0x0100); - reg = wm8731_read_reg_cache(codec, WM8731_LINVOL); - wm8731_write(codec, WM8731_LINVOL, reg & ~0x0100); - reg = wm8731_read_reg_cache(codec, WM8731_RINVOL); - wm8731_write(codec, WM8731_RINVOL, reg & ~0x0100); + snd_soc_update_bits(codec, WM8731_LOUT1V, 0x100, 0); + snd_soc_update_bits(codec, WM8731_ROUT1V, 0x100, 0); + snd_soc_update_bits(codec, WM8731_LINVOL, 0x100, 0); + snd_soc_update_bits(codec, WM8731_RINVOL, 0x100, 0); /* Disable bypass path by default */ - reg = wm8731_read_reg_cache(codec, WM8731_APANA); - wm8731_write(codec, WM8731_APANA, reg & ~0x4); + snd_soc_update_bits(codec, WM8731_APANA, 0x4, 0); wm8731_codec = codec; diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index b64509b..ed09043 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -55,50 +55,7 @@ static const u16 wm8750_reg[] = { 0x0079, 0x0079, 0x0079, /* 40 */ }; -/* - * read wm8750 register cache - */ -static inline unsigned int wm8750_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u16 *cache = codec->reg_cache; - if (reg > WM8750_CACHE_REGNUM) - return -1; - return cache[reg]; -} - -/* - * write wm8750 register cache - */ -static inline void wm8750_write_reg_cache(struct snd_soc_codec *codec, - unsigned int reg, unsigned int value) -{ - u16 *cache = codec->reg_cache; - if (reg > WM8750_CACHE_REGNUM) - return; - cache[reg] = value; -} - -static int wm8750_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[2]; - - /* data is - * D15..D9 WM8753 register offset - * D8...D0 register data - */ - data[0] = (reg << 1) | ((value >> 8) & 0x0001); - data[1] = value & 0x00ff; - - wm8750_write_reg_cache(codec, reg, value); - if (codec->hw_write(codec->control_data, data, 2) == 2) - return 0; - else - return -EIO; -} - -#define wm8750_reset(c) wm8750_write(c, WM8750_RESET, 0) +#define wm8750_reset(c) snd_soc_write(c, WM8750_RESET, 0) /* * WM8750 Controls @@ -594,7 +551,7 @@ static int wm8750_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - wm8750_write(codec, WM8750_IFACE, iface); + snd_soc_write(codec, WM8750_IFACE, iface); return 0; } @@ -606,8 +563,8 @@ static int wm8750_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; struct wm8750_priv *wm8750 = codec->private_data; - u16 iface = wm8750_read_reg_cache(codec, WM8750_IFACE) & 0x1f3; - u16 srate = wm8750_read_reg_cache(codec, WM8750_SRATE) & 0x1c0; + u16 iface = snd_soc_read(codec, WM8750_IFACE) & 0x1f3; + u16 srate = snd_soc_read(codec, WM8750_SRATE) & 0x1c0; int coeff = get_coeff(wm8750->sysclk, params_rate(params)); /* bit size */ @@ -626,9 +583,9 @@ static int wm8750_pcm_hw_params(struct snd_pcm_substream *substream, } /* set iface & srate */ - wm8750_write(codec, WM8750_IFACE, iface); + snd_soc_write(codec, WM8750_IFACE, iface); if (coeff >= 0) - wm8750_write(codec, WM8750_SRATE, srate | + snd_soc_write(codec, WM8750_SRATE, srate | (coeff_div[coeff].sr << 1) | coeff_div[coeff].usb); return 0; @@ -637,35 +594,35 @@ static int wm8750_pcm_hw_params(struct snd_pcm_substream *substream, static int wm8750_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - u16 mute_reg = wm8750_read_reg_cache(codec, WM8750_ADCDAC) & 0xfff7; + u16 mute_reg = snd_soc_read(codec, WM8750_ADCDAC) & 0xfff7; if (mute) - wm8750_write(codec, WM8750_ADCDAC, mute_reg | 0x8); + snd_soc_write(codec, WM8750_ADCDAC, mute_reg | 0x8); else - wm8750_write(codec, WM8750_ADCDAC, mute_reg); + snd_soc_write(codec, WM8750_ADCDAC, mute_reg); return 0; } static int wm8750_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 pwr_reg = wm8750_read_reg_cache(codec, WM8750_PWR1) & 0xfe3e; + u16 pwr_reg = snd_soc_read(codec, WM8750_PWR1) & 0xfe3e; switch (level) { case SND_SOC_BIAS_ON: /* set vmid to 50k and unmute dac */ - wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x00c0); + snd_soc_write(codec, WM8750_PWR1, pwr_reg | 0x00c0); break; case SND_SOC_BIAS_PREPARE: /* set vmid to 5k for quick power up */ - wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x01c1); + snd_soc_write(codec, WM8750_PWR1, pwr_reg | 0x01c1); break; case SND_SOC_BIAS_STANDBY: /* mute dac and set vmid to 500k, enable VREF */ - wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x0141); + snd_soc_write(codec, WM8750_PWR1, pwr_reg | 0x0141); break; case SND_SOC_BIAS_OFF: - wm8750_write(codec, WM8750_PWR1, 0x0001); + snd_soc_write(codec, WM8750_PWR1, 0x0001); break; } codec->bias_level = level; @@ -761,8 +718,6 @@ static int wm8750_init(struct snd_soc_device *socdev) codec->name = "WM8750"; codec->owner = THIS_MODULE; - codec->read = wm8750_read_reg_cache; - codec->write = wm8750_write; codec->set_bias_level = wm8750_set_bias_level; codec->dai = &wm8750_dai; codec->num_dai = 1; @@ -771,13 +726,23 @@ static int wm8750_init(struct snd_soc_device *socdev) if (codec->reg_cache == NULL) return -ENOMEM; - wm8750_reset(codec); + ret = snd_soc_codec_set_cache_io(codec, 7, 9); + if (ret < 0) { + printk(KERN_ERR "wm8750: failed to set cache I/O: %d\n", ret); + goto err; + } + + ret = wm8750_reset(codec); + if (ret < 0) { + printk(KERN_ERR "wm8750: failed to reset: %d\n", ret); + goto err; + } /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) { printk(KERN_ERR "wm8750: failed to create pcms\n"); - goto pcm_err; + goto err; } /* charge output caps */ @@ -786,22 +751,22 @@ static int wm8750_init(struct snd_soc_device *socdev) schedule_delayed_work(&codec->delayed_work, msecs_to_jiffies(1000)); /* set the update bits */ - reg = wm8750_read_reg_cache(codec, WM8750_LDAC); - wm8750_write(codec, WM8750_LDAC, reg | 0x0100); - reg = wm8750_read_reg_cache(codec, WM8750_RDAC); - wm8750_write(codec, WM8750_RDAC, reg | 0x0100); - reg = wm8750_read_reg_cache(codec, WM8750_LOUT1V); - wm8750_write(codec, WM8750_LOUT1V, reg | 0x0100); - reg = wm8750_read_reg_cache(codec, WM8750_ROUT1V); - wm8750_write(codec, WM8750_ROUT1V, reg | 0x0100); - reg = wm8750_read_reg_cache(codec, WM8750_LOUT2V); - wm8750_write(codec, WM8750_LOUT2V, reg | 0x0100); - reg = wm8750_read_reg_cache(codec, WM8750_ROUT2V); - wm8750_write(codec, WM8750_ROUT2V, reg | 0x0100); - reg = wm8750_read_reg_cache(codec, WM8750_LINVOL); - wm8750_write(codec, WM8750_LINVOL, reg | 0x0100); - reg = wm8750_read_reg_cache(codec, WM8750_RINVOL); - wm8750_write(codec, WM8750_RINVOL, reg | 0x0100); + reg = snd_soc_read(codec, WM8750_LDAC); + snd_soc_write(codec, WM8750_LDAC, reg | 0x0100); + reg = snd_soc_read(codec, WM8750_RDAC); + snd_soc_write(codec, WM8750_RDAC, reg | 0x0100); + reg = snd_soc_read(codec, WM8750_LOUT1V); + snd_soc_write(codec, WM8750_LOUT1V, reg | 0x0100); + reg = snd_soc_read(codec, WM8750_ROUT1V); + snd_soc_write(codec, WM8750_ROUT1V, reg | 0x0100); + reg = snd_soc_read(codec, WM8750_LOUT2V); + snd_soc_write(codec, WM8750_LOUT2V, reg | 0x0100); + reg = snd_soc_read(codec, WM8750_ROUT2V); + snd_soc_write(codec, WM8750_ROUT2V, reg | 0x0100); + reg = snd_soc_read(codec, WM8750_LINVOL); + snd_soc_write(codec, WM8750_LINVOL, reg | 0x0100); + reg = snd_soc_read(codec, WM8750_RINVOL); + snd_soc_write(codec, WM8750_RINVOL, reg | 0x0100); snd_soc_add_controls(codec, wm8750_snd_controls, ARRAY_SIZE(wm8750_snd_controls)); @@ -816,7 +781,7 @@ static int wm8750_init(struct snd_soc_device *socdev) card_err: snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); -pcm_err: +err: kfree(codec->reg_cache); return ret; } diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index d1769e6..c529ffc 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -69,61 +69,7 @@ struct wm8960_priv { struct snd_soc_codec codec; }; -/* - * read wm8960 register cache - */ -static inline unsigned int wm8960_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u16 *cache = codec->reg_cache; - if (reg == WM8960_RESET) - return 0; - if (reg >= WM8960_CACHEREGNUM) - return -1; - return cache[reg]; -} - -/* - * write wm8960 register cache - */ -static inline void wm8960_write_reg_cache(struct snd_soc_codec *codec, - u16 reg, unsigned int value) -{ - u16 *cache = codec->reg_cache; - if (reg >= WM8960_CACHEREGNUM) - return; - cache[reg] = value; -} - -static inline unsigned int wm8960_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - return wm8960_read_reg_cache(codec, reg); -} - -/* - * write to the WM8960 register space - */ -static int wm8960_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[2]; - - /* data is - * D15..D9 WM8960 register offset - * D8...D0 register data - */ - data[0] = (reg << 1) | ((value >> 8) & 0x0001); - data[1] = value & 0x00ff; - - wm8960_write_reg_cache(codec, reg, value); - if (codec->hw_write(codec->control_data, data, 2) == 2) - return 0; - else - return -EIO; -} - -#define wm8960_reset(c) wm8960_write(c, WM8960_RESET, 0) +#define wm8960_reset(c) snd_soc_write(c, WM8960_RESET, 0) /* enumerated controls */ static const char *wm8960_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"}; @@ -420,7 +366,7 @@ static int wm8960_set_dai_fmt(struct snd_soc_dai *codec_dai, } /* set iface */ - wm8960_write(codec, WM8960_IFACE1, iface); + snd_soc_write(codec, WM8960_IFACE1, iface); return 0; } @@ -431,7 +377,7 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; - u16 iface = wm8960_read(codec, WM8960_IFACE1) & 0xfff3; + u16 iface = snd_soc_read(codec, WM8960_IFACE1) & 0xfff3; /* bit size */ switch (params_format(params)) { @@ -446,19 +392,19 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, } /* set iface */ - wm8960_write(codec, WM8960_IFACE1, iface); + snd_soc_write(codec, WM8960_IFACE1, iface); return 0; } static int wm8960_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - u16 mute_reg = wm8960_read(codec, WM8960_DACCTL1) & 0xfff7; + u16 mute_reg = snd_soc_read(codec, WM8960_DACCTL1) & 0xfff7; if (mute) - wm8960_write(codec, WM8960_DACCTL1, mute_reg | 0x8); + snd_soc_write(codec, WM8960_DACCTL1, mute_reg | 0x8); else - wm8960_write(codec, WM8960_DACCTL1, mute_reg); + snd_soc_write(codec, WM8960_DACCTL1, mute_reg); return 0; } @@ -474,16 +420,16 @@ static int wm8960_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: /* Set VMID to 2x50k */ - reg = wm8960_read(codec, WM8960_POWER1); + reg = snd_soc_read(codec, WM8960_POWER1); reg &= ~0x180; reg |= 0x80; - wm8960_write(codec, WM8960_POWER1, reg); + snd_soc_write(codec, WM8960_POWER1, reg); break; case SND_SOC_BIAS_STANDBY: if (codec->bias_level == SND_SOC_BIAS_OFF) { /* Enable anti-pop features */ - wm8960_write(codec, WM8960_APOP1, + snd_soc_write(codec, WM8960_APOP1, WM8960_POBCTRL | WM8960_SOFT_ST | WM8960_BUFDCOPEN | WM8960_BUFIOEN); @@ -491,43 +437,43 @@ static int wm8960_set_bias_level(struct snd_soc_codec *codec, reg = WM8960_DISOP; if (pdata) reg |= pdata->dres << 4; - wm8960_write(codec, WM8960_APOP2, reg); + snd_soc_write(codec, WM8960_APOP2, reg); msleep(400); - wm8960_write(codec, WM8960_APOP2, 0); + snd_soc_write(codec, WM8960_APOP2, 0); /* Enable & ramp VMID at 2x50k */ - reg = wm8960_read(codec, WM8960_POWER1); + reg = snd_soc_read(codec, WM8960_POWER1); reg |= 0x80; - wm8960_write(codec, WM8960_POWER1, reg); + snd_soc_write(codec, WM8960_POWER1, reg); msleep(100); /* Enable VREF */ - wm8960_write(codec, WM8960_POWER1, reg | WM8960_VREF); + snd_soc_write(codec, WM8960_POWER1, reg | WM8960_VREF); /* Disable anti-pop features */ - wm8960_write(codec, WM8960_APOP1, WM8960_BUFIOEN); + snd_soc_write(codec, WM8960_APOP1, WM8960_BUFIOEN); } /* Set VMID to 2x250k */ - reg = wm8960_read(codec, WM8960_POWER1); + reg = snd_soc_read(codec, WM8960_POWER1); reg &= ~0x180; reg |= 0x100; - wm8960_write(codec, WM8960_POWER1, reg); + snd_soc_write(codec, WM8960_POWER1, reg); break; case SND_SOC_BIAS_OFF: /* Enable anti-pop features */ - wm8960_write(codec, WM8960_APOP1, + snd_soc_write(codec, WM8960_APOP1, WM8960_POBCTRL | WM8960_SOFT_ST | WM8960_BUFDCOPEN | WM8960_BUFIOEN); /* Disable VMID and VREF, let them discharge */ - wm8960_write(codec, WM8960_POWER1, 0); + snd_soc_write(codec, WM8960_POWER1, 0); msleep(600); - wm8960_write(codec, WM8960_APOP1, 0); + snd_soc_write(codec, WM8960_APOP1, 0); break; } @@ -610,33 +556,33 @@ static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, /* Disable the PLL: even if we are changing the frequency the * PLL needs to be disabled while we do so. */ - wm8960_write(codec, WM8960_CLOCK1, - wm8960_read(codec, WM8960_CLOCK1) & ~1); - wm8960_write(codec, WM8960_POWER2, - wm8960_read(codec, WM8960_POWER2) & ~1); + snd_soc_write(codec, WM8960_CLOCK1, + snd_soc_read(codec, WM8960_CLOCK1) & ~1); + snd_soc_write(codec, WM8960_POWER2, + snd_soc_read(codec, WM8960_POWER2) & ~1); if (!freq_in || !freq_out) return 0; - reg = wm8960_read(codec, WM8960_PLL1) & ~0x3f; + reg = snd_soc_read(codec, WM8960_PLL1) & ~0x3f; reg |= pll_div.pre_div << 4; reg |= pll_div.n; if (pll_div.k) { reg |= 0x20; - wm8960_write(codec, WM8960_PLL2, (pll_div.k >> 18) & 0x3f); - wm8960_write(codec, WM8960_PLL3, (pll_div.k >> 9) & 0x1ff); - wm8960_write(codec, WM8960_PLL4, pll_div.k & 0x1ff); + snd_soc_write(codec, WM8960_PLL2, (pll_div.k >> 18) & 0x3f); + snd_soc_write(codec, WM8960_PLL3, (pll_div.k >> 9) & 0x1ff); + snd_soc_write(codec, WM8960_PLL4, pll_div.k & 0x1ff); } - wm8960_write(codec, WM8960_PLL1, reg); + snd_soc_write(codec, WM8960_PLL1, reg); /* Turn it on */ - wm8960_write(codec, WM8960_POWER2, - wm8960_read(codec, WM8960_POWER2) | 1); + snd_soc_write(codec, WM8960_POWER2, + snd_soc_read(codec, WM8960_POWER2) | 1); msleep(250); - wm8960_write(codec, WM8960_CLOCK1, - wm8960_read(codec, WM8960_CLOCK1) | 1); + snd_soc_write(codec, WM8960_CLOCK1, + snd_soc_read(codec, WM8960_CLOCK1) | 1); return 0; } @@ -649,28 +595,28 @@ static int wm8960_set_dai_clkdiv(struct snd_soc_dai *codec_dai, switch (div_id) { case WM8960_SYSCLKSEL: - reg = wm8960_read(codec, WM8960_CLOCK1) & 0x1fe; - wm8960_write(codec, WM8960_CLOCK1, reg | div); + reg = snd_soc_read(codec, WM8960_CLOCK1) & 0x1fe; + snd_soc_write(codec, WM8960_CLOCK1, reg | div); break; case WM8960_SYSCLKDIV: - reg = wm8960_read(codec, WM8960_CLOCK1) & 0x1f9; - wm8960_write(codec, WM8960_CLOCK1, reg | div); + reg = snd_soc_read(codec, WM8960_CLOCK1) & 0x1f9; + snd_soc_write(codec, WM8960_CLOCK1, reg | div); break; case WM8960_DACDIV: - reg = wm8960_read(codec, WM8960_CLOCK1) & 0x1c7; - wm8960_write(codec, WM8960_CLOCK1, reg | div); + reg = snd_soc_read(codec, WM8960_CLOCK1) & 0x1c7; + snd_soc_write(codec, WM8960_CLOCK1, reg | div); break; case WM8960_OPCLKDIV: - reg = wm8960_read(codec, WM8960_PLL1) & 0x03f; - wm8960_write(codec, WM8960_PLL1, reg | div); + reg = snd_soc_read(codec, WM8960_PLL1) & 0x03f; + snd_soc_write(codec, WM8960_PLL1, reg | div); break; case WM8960_DCLKDIV: - reg = wm8960_read(codec, WM8960_CLOCK2) & 0x03f; - wm8960_write(codec, WM8960_CLOCK2, reg | div); + reg = snd_soc_read(codec, WM8960_CLOCK2) & 0x03f; + snd_soc_write(codec, WM8960_CLOCK2, reg | div); break; case WM8960_TOCLKSEL: - reg = wm8960_read(codec, WM8960_ADDCTL1) & 0x1fd; - wm8960_write(codec, WM8960_ADDCTL1, reg | div); + reg = snd_soc_read(codec, WM8960_ADDCTL1) & 0x1fd; + snd_soc_write(codec, WM8960_ADDCTL1, reg | div); break; default: return -EINVAL; @@ -830,8 +776,6 @@ static int wm8960_register(struct wm8960_priv *wm8960) codec->private_data = wm8960; codec->name = "WM8960"; codec->owner = THIS_MODULE; - codec->read = wm8960_read_reg_cache; - codec->write = wm8960_write; codec->bias_level = SND_SOC_BIAS_OFF; codec->set_bias_level = wm8960_set_bias_level; codec->dai = &wm8960_dai; @@ -841,6 +785,12 @@ static int wm8960_register(struct wm8960_priv *wm8960) memcpy(codec->reg_cache, wm8960_reg, sizeof(wm8960_reg)); + ret = snd_soc_codec_set_cache_io(codec, 7, 9); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + goto err; + } + ret = wm8960_reset(codec); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset\n"); @@ -852,26 +802,26 @@ static int wm8960_register(struct wm8960_priv *wm8960) wm8960_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Latch the update bits */ - reg = wm8960_read(codec, WM8960_LINVOL); - wm8960_write(codec, WM8960_LINVOL, reg | 0x100); - reg = wm8960_read(codec, WM8960_RINVOL); - wm8960_write(codec, WM8960_RINVOL, reg | 0x100); - reg = wm8960_read(codec, WM8960_LADC); - wm8960_write(codec, WM8960_LADC, reg | 0x100); - reg = wm8960_read(codec, WM8960_RADC); - wm8960_write(codec, WM8960_RADC, reg | 0x100); - reg = wm8960_read(codec, WM8960_LDAC); - wm8960_write(codec, WM8960_LDAC, reg | 0x100); - reg = wm8960_read(codec, WM8960_RDAC); - wm8960_write(codec, WM8960_RDAC, reg | 0x100); - reg = wm8960_read(codec, WM8960_LOUT1); - wm8960_write(codec, WM8960_LOUT1, reg | 0x100); - reg = wm8960_read(codec, WM8960_ROUT1); - wm8960_write(codec, WM8960_ROUT1, reg | 0x100); - reg = wm8960_read(codec, WM8960_LOUT2); - wm8960_write(codec, WM8960_LOUT2, reg | 0x100); - reg = wm8960_read(codec, WM8960_ROUT2); - wm8960_write(codec, WM8960_ROUT2, reg | 0x100); + reg = snd_soc_read(codec, WM8960_LINVOL); + snd_soc_write(codec, WM8960_LINVOL, reg | 0x100); + reg = snd_soc_read(codec, WM8960_RINVOL); + snd_soc_write(codec, WM8960_RINVOL, reg | 0x100); + reg = snd_soc_read(codec, WM8960_LADC); + snd_soc_write(codec, WM8960_LADC, reg | 0x100); + reg = snd_soc_read(codec, WM8960_RADC); + snd_soc_write(codec, WM8960_RADC, reg | 0x100); + reg = snd_soc_read(codec, WM8960_LDAC); + snd_soc_write(codec, WM8960_LDAC, reg | 0x100); + reg = snd_soc_read(codec, WM8960_RDAC); + snd_soc_write(codec, WM8960_RDAC, reg | 0x100); + reg = snd_soc_read(codec, WM8960_LOUT1); + snd_soc_write(codec, WM8960_LOUT1, reg | 0x100); + reg = snd_soc_read(codec, WM8960_ROUT1); + snd_soc_write(codec, WM8960_ROUT1, reg | 0x100); + reg = snd_soc_read(codec, WM8960_LOUT2); + snd_soc_write(codec, WM8960_LOUT2, reg | 0x100); + reg = snd_soc_read(codec, WM8960_ROUT2); + snd_soc_write(codec, WM8960_ROUT2, reg | 0x100); wm8960_codec = codec; diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 032dca2..53f3bc9 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -59,44 +59,7 @@ static const u16 wm8971_reg[] = { 0x0079, 0x0079, 0x0079, /* 40 */ }; -static inline unsigned int wm8971_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u16 *cache = codec->reg_cache; - if (reg < WM8971_REG_COUNT) - return cache[reg]; - - return -1; -} - -static inline void wm8971_write_reg_cache(struct snd_soc_codec *codec, - unsigned int reg, unsigned int value) -{ - u16 *cache = codec->reg_cache; - if (reg < WM8971_REG_COUNT) - cache[reg] = value; -} - -static int wm8971_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[2]; - - /* data is - * D15..D9 WM8753 register offset - * D8...D0 register data - */ - data[0] = (reg << 1) | ((value >> 8) & 0x0001); - data[1] = value & 0x00ff; - - wm8971_write_reg_cache (codec, reg, value); - if (codec->hw_write(codec->control_data, data, 2) == 2) - return 0; - else - return -EIO; -} - -#define wm8971_reset(c) wm8971_write(c, WM8971_RESET, 0) +#define wm8971_reset(c) snd_soc_write(c, WM8971_RESET, 0) /* WM8971 Controls */ static const char *wm8971_bass[] = { "Linear Control", "Adaptive Boost" }; @@ -521,7 +484,7 @@ static int wm8971_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - wm8971_write(codec, WM8971_IFACE, iface); + snd_soc_write(codec, WM8971_IFACE, iface); return 0; } @@ -533,8 +496,8 @@ static int wm8971_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; struct wm8971_priv *wm8971 = codec->private_data; - u16 iface = wm8971_read_reg_cache(codec, WM8971_IFACE) & 0x1f3; - u16 srate = wm8971_read_reg_cache(codec, WM8971_SRATE) & 0x1c0; + u16 iface = snd_soc_read(codec, WM8971_IFACE) & 0x1f3; + u16 srate = snd_soc_read(codec, WM8971_SRATE) & 0x1c0; int coeff = get_coeff(wm8971->sysclk, params_rate(params)); /* bit size */ @@ -553,9 +516,9 @@ static int wm8971_pcm_hw_params(struct snd_pcm_substream *substream, } /* set iface & srate */ - wm8971_write(codec, WM8971_IFACE, iface); + snd_soc_write(codec, WM8971_IFACE, iface); if (coeff >= 0) - wm8971_write(codec, WM8971_SRATE, srate | + snd_soc_write(codec, WM8971_SRATE, srate | (coeff_div[coeff].sr << 1) | coeff_div[coeff].usb); return 0; @@ -564,33 +527,33 @@ static int wm8971_pcm_hw_params(struct snd_pcm_substream *substream, static int wm8971_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - u16 mute_reg = wm8971_read_reg_cache(codec, WM8971_ADCDAC) & 0xfff7; + u16 mute_reg = snd_soc_read(codec, WM8971_ADCDAC) & 0xfff7; if (mute) - wm8971_write(codec, WM8971_ADCDAC, mute_reg | 0x8); + snd_soc_write(codec, WM8971_ADCDAC, mute_reg | 0x8); else - wm8971_write(codec, WM8971_ADCDAC, mute_reg); + snd_soc_write(codec, WM8971_ADCDAC, mute_reg); return 0; } static int wm8971_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 pwr_reg = wm8971_read_reg_cache(codec, WM8971_PWR1) & 0xfe3e; + u16 pwr_reg = snd_soc_read(codec, WM8971_PWR1) & 0xfe3e; switch (level) { case SND_SOC_BIAS_ON: /* set vmid to 50k and unmute dac */ - wm8971_write(codec, WM8971_PWR1, pwr_reg | 0x00c1); + snd_soc_write(codec, WM8971_PWR1, pwr_reg | 0x00c1); break; case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: /* mute dac and set vmid to 500k, enable VREF */ - wm8971_write(codec, WM8971_PWR1, pwr_reg | 0x0140); + snd_soc_write(codec, WM8971_PWR1, pwr_reg | 0x0140); break; case SND_SOC_BIAS_OFF: - wm8971_write(codec, WM8971_PWR1, 0x0001); + snd_soc_write(codec, WM8971_PWR1, 0x0001); break; } codec->bias_level = level; @@ -667,8 +630,8 @@ static int wm8971_resume(struct platform_device *pdev) /* charge wm8971 caps */ if (codec->suspend_bias_level == SND_SOC_BIAS_ON) { - reg = wm8971_read_reg_cache(codec, WM8971_PWR1) & 0xfe3e; - wm8971_write(codec, WM8971_PWR1, reg | 0x01c0); + reg = snd_soc_read(codec, WM8971_PWR1) & 0xfe3e; + snd_soc_write(codec, WM8971_PWR1, reg | 0x01c0); codec->bias_level = SND_SOC_BIAS_ON; queue_delayed_work(wm8971_workq, &codec->delayed_work, msecs_to_jiffies(1000)); @@ -684,8 +647,6 @@ static int wm8971_init(struct snd_soc_device *socdev) codec->name = "WM8971"; codec->owner = THIS_MODULE; - codec->read = wm8971_read_reg_cache; - codec->write = wm8971_write; codec->set_bias_level = wm8971_set_bias_level; codec->dai = &wm8971_dai; codec->reg_cache_size = ARRAY_SIZE(wm8971_reg); @@ -695,42 +656,48 @@ static int wm8971_init(struct snd_soc_device *socdev) if (codec->reg_cache == NULL) return -ENOMEM; + ret = snd_soc_codec_set_cache_io(codec, 7, 9); + if (ret < 0) { + printk(KERN_ERR "wm8971: failed to set cache I/O: %d\n", ret); + goto err; + } + wm8971_reset(codec); /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) { printk(KERN_ERR "wm8971: failed to create pcms\n"); - goto pcm_err; + goto err; } /* charge output caps - set vmid to 5k for quick power up */ - reg = wm8971_read_reg_cache(codec, WM8971_PWR1) & 0xfe3e; - wm8971_write(codec, WM8971_PWR1, reg | 0x01c0); + reg = snd_soc_read(codec, WM8971_PWR1) & 0xfe3e; + snd_soc_write(codec, WM8971_PWR1, reg | 0x01c0); codec->bias_level = SND_SOC_BIAS_STANDBY; queue_delayed_work(wm8971_workq, &codec->delayed_work, msecs_to_jiffies(1000)); /* set the update bits */ - reg = wm8971_read_reg_cache(codec, WM8971_LDAC); - wm8971_write(codec, WM8971_LDAC, reg | 0x0100); - reg = wm8971_read_reg_cache(codec, WM8971_RDAC); - wm8971_write(codec, WM8971_RDAC, reg | 0x0100); - - reg = wm8971_read_reg_cache(codec, WM8971_LOUT1V); - wm8971_write(codec, WM8971_LOUT1V, reg | 0x0100); - reg = wm8971_read_reg_cache(codec, WM8971_ROUT1V); - wm8971_write(codec, WM8971_ROUT1V, reg | 0x0100); - - reg = wm8971_read_reg_cache(codec, WM8971_LOUT2V); - wm8971_write(codec, WM8971_LOUT2V, reg | 0x0100); - reg = wm8971_read_reg_cache(codec, WM8971_ROUT2V); - wm8971_write(codec, WM8971_ROUT2V, reg | 0x0100); - - reg = wm8971_read_reg_cache(codec, WM8971_LINVOL); - wm8971_write(codec, WM8971_LINVOL, reg | 0x0100); - reg = wm8971_read_reg_cache(codec, WM8971_RINVOL); - wm8971_write(codec, WM8971_RINVOL, reg | 0x0100); + reg = snd_soc_read(codec, WM8971_LDAC); + snd_soc_write(codec, WM8971_LDAC, reg | 0x0100); + reg = snd_soc_read(codec, WM8971_RDAC); + snd_soc_write(codec, WM8971_RDAC, reg | 0x0100); + + reg = snd_soc_read(codec, WM8971_LOUT1V); + snd_soc_write(codec, WM8971_LOUT1V, reg | 0x0100); + reg = snd_soc_read(codec, WM8971_ROUT1V); + snd_soc_write(codec, WM8971_ROUT1V, reg | 0x0100); + + reg = snd_soc_read(codec, WM8971_LOUT2V); + snd_soc_write(codec, WM8971_LOUT2V, reg | 0x0100); + reg = snd_soc_read(codec, WM8971_ROUT2V); + snd_soc_write(codec, WM8971_ROUT2V, reg | 0x0100); + + reg = snd_soc_read(codec, WM8971_LINVOL); + snd_soc_write(codec, WM8971_LINVOL, reg | 0x0100); + reg = snd_soc_read(codec, WM8971_RINVOL); + snd_soc_write(codec, WM8971_RINVOL, reg | 0x0100); snd_soc_add_controls(codec, wm8971_snd_controls, ARRAY_SIZE(wm8971_snd_controls)); @@ -745,7 +712,7 @@ static int wm8971_init(struct snd_soc_device *socdev) card_err: snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); -pcm_err: +err: kfree(codec->reg_cache); return ret; } diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index 6f15acd..7d5b807 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -57,50 +57,7 @@ struct wm8988_priv { }; -/* - * read wm8988 register cache - */ -static inline unsigned int wm8988_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u16 *cache = codec->reg_cache; - if (reg > WM8988_NUM_REG) - return -1; - return cache[reg]; -} - -/* - * write wm8988 register cache - */ -static inline void wm8988_write_reg_cache(struct snd_soc_codec *codec, - unsigned int reg, unsigned int value) -{ - u16 *cache = codec->reg_cache; - if (reg > WM8988_NUM_REG) - return; - cache[reg] = value; -} - -static int wm8988_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[2]; - - /* data is - * D15..D9 WM8753 register offset - * D8...D0 register data - */ - data[0] = (reg << 1) | ((value >> 8) & 0x0001); - data[1] = value & 0x00ff; - - wm8988_write_reg_cache(codec, reg, value); - if (codec->hw_write(codec->control_data, data, 2) == 2) - return 0; - else - return -EIO; -} - -#define wm8988_reset(c) wm8988_write(c, WM8988_RESET, 0) +#define wm8988_reset(c) snd_soc_write(c, WM8988_RESET, 0) /* * WM8988 Controls @@ -226,15 +183,15 @@ static int wm8988_lrc_control(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; - u16 adctl2 = wm8988_read_reg_cache(codec, WM8988_ADCTL2); + u16 adctl2 = snd_soc_read(codec, WM8988_ADCTL2); /* Use the DAC to gate LRC if active, otherwise use ADC */ - if (wm8988_read_reg_cache(codec, WM8988_PWR2) & 0x180) + if (snd_soc_read(codec, WM8988_PWR2) & 0x180) adctl2 &= ~0x4; else adctl2 |= 0x4; - return wm8988_write(codec, WM8988_ADCTL2, adctl2); + return snd_soc_write(codec, WM8988_ADCTL2, adctl2); } static const char *wm8988_line_texts[] = { @@ -619,7 +576,7 @@ static int wm8988_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - wm8988_write(codec, WM8988_IFACE, iface); + snd_soc_write(codec, WM8988_IFACE, iface); return 0; } @@ -653,8 +610,8 @@ static int wm8988_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; struct wm8988_priv *wm8988 = codec->private_data; - u16 iface = wm8988_read_reg_cache(codec, WM8988_IFACE) & 0x1f3; - u16 srate = wm8988_read_reg_cache(codec, WM8988_SRATE) & 0x180; + u16 iface = snd_soc_read(codec, WM8988_IFACE) & 0x1f3; + u16 srate = snd_soc_read(codec, WM8988_SRATE) & 0x180; int coeff; coeff = get_coeff(wm8988->sysclk, params_rate(params)); @@ -685,9 +642,9 @@ static int wm8988_pcm_hw_params(struct snd_pcm_substream *substream, } /* set iface & srate */ - wm8988_write(codec, WM8988_IFACE, iface); + snd_soc_write(codec, WM8988_IFACE, iface); if (coeff >= 0) - wm8988_write(codec, WM8988_SRATE, srate | + snd_soc_write(codec, WM8988_SRATE, srate | (coeff_div[coeff].sr << 1) | coeff_div[coeff].usb); return 0; @@ -696,19 +653,19 @@ static int wm8988_pcm_hw_params(struct snd_pcm_substream *substream, static int wm8988_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - u16 mute_reg = wm8988_read_reg_cache(codec, WM8988_ADCDAC) & 0xfff7; + u16 mute_reg = snd_soc_read(codec, WM8988_ADCDAC) & 0xfff7; if (mute) - wm8988_write(codec, WM8988_ADCDAC, mute_reg | 0x8); + snd_soc_write(codec, WM8988_ADCDAC, mute_reg | 0x8); else - wm8988_write(codec, WM8988_ADCDAC, mute_reg); + snd_soc_write(codec, WM8988_ADCDAC, mute_reg); return 0; } static int wm8988_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 pwr_reg = wm8988_read_reg_cache(codec, WM8988_PWR1) & ~0x1c1; + u16 pwr_reg = snd_soc_read(codec, WM8988_PWR1) & ~0x1c1; switch (level) { case SND_SOC_BIAS_ON: @@ -716,24 +673,24 @@ static int wm8988_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: /* VREF, VMID=2x50k, digital enabled */ - wm8988_write(codec, WM8988_PWR1, pwr_reg | 0x00c0); + snd_soc_write(codec, WM8988_PWR1, pwr_reg | 0x00c0); break; case SND_SOC_BIAS_STANDBY: if (codec->bias_level == SND_SOC_BIAS_OFF) { /* VREF, VMID=2x5k */ - wm8988_write(codec, WM8988_PWR1, pwr_reg | 0x1c1); + snd_soc_write(codec, WM8988_PWR1, pwr_reg | 0x1c1); /* Charge caps */ msleep(100); } /* VREF, VMID=2*500k, digital stopped */ - wm8988_write(codec, WM8988_PWR1, pwr_reg | 0x0141); + snd_soc_write(codec, WM8988_PWR1, pwr_reg | 0x0141); break; case SND_SOC_BIAS_OFF: - wm8988_write(codec, WM8988_PWR1, 0x0000); + snd_soc_write(codec, WM8988_PWR1, 0x0000); break; } codec->bias_level = level; @@ -887,8 +844,6 @@ static int wm8988_register(struct wm8988_priv *wm8988) codec->private_data = wm8988; codec->name = "WM8988"; codec->owner = THIS_MODULE; - codec->read = wm8988_read_reg_cache; - codec->write = wm8988_write; codec->dai = &wm8988_dai; codec->num_dai = 1; codec->reg_cache_size = ARRAY_SIZE(wm8988->reg_cache); @@ -899,6 +854,12 @@ static int wm8988_register(struct wm8988_priv *wm8988) memcpy(codec->reg_cache, wm8988_reg, sizeof(wm8988_reg)); + ret = snd_soc_codec_set_cache_io(codec, 7, 9); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + goto err; + } + ret = wm8988_reset(codec); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset\n"); @@ -906,16 +867,16 @@ static int wm8988_register(struct wm8988_priv *wm8988) } /* set the update bits (we always update left then right) */ - reg = wm8988_read_reg_cache(codec, WM8988_RADC); - wm8988_write(codec, WM8988_RADC, reg | 0x100); - reg = wm8988_read_reg_cache(codec, WM8988_RDAC); - wm8988_write(codec, WM8988_RDAC, reg | 0x0100); - reg = wm8988_read_reg_cache(codec, WM8988_ROUT1V); - wm8988_write(codec, WM8988_ROUT1V, reg | 0x0100); - reg = wm8988_read_reg_cache(codec, WM8988_ROUT2V); - wm8988_write(codec, WM8988_ROUT2V, reg | 0x0100); - reg = wm8988_read_reg_cache(codec, WM8988_RINVOL); - wm8988_write(codec, WM8988_RINVOL, reg | 0x0100); + reg = snd_soc_read(codec, WM8988_RADC); + snd_soc_write(codec, WM8988_RADC, reg | 0x100); + reg = snd_soc_read(codec, WM8988_RDAC); + snd_soc_write(codec, WM8988_RDAC, reg | 0x0100); + reg = snd_soc_read(codec, WM8988_ROUT1V); + snd_soc_write(codec, WM8988_ROUT1V, reg | 0x0100); + reg = snd_soc_read(codec, WM8988_ROUT2V); + snd_soc_write(codec, WM8988_ROUT2V, reg | 0x0100); + reg = snd_soc_read(codec, WM8988_RINVOL); + snd_soc_write(codec, WM8988_RINVOL, reg | 0x0100); wm8988_set_bias_level(&wm8988->codec, SND_SOC_BIAS_STANDBY); diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c new file mode 100644 index 0000000..4eb4333 --- /dev/null +++ b/sound/soc/soc-cache.c @@ -0,0 +1,105 @@ +/* + * soc-cache.c -- ASoC register cache helpers + * + * Copyright 2009 Wolfson Microelectronics PLC. + * + * Author: Mark Brown <broonie(a)opensource.wolfsonmicro.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. + */ + +#include <sound/soc.h> + +static unsigned int snd_soc_7_9_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg >= codec->reg_cache_size) + return -1; + return cache[reg]; +} + +static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u16 *cache = codec->reg_cache; + u8 data[2]; + int ret; + + BUG_ON(codec->volatile_register); + + data[0] = (reg << 1) | ((value >> 8) & 0x0001); + data[1] = value & 0x00ff; + + if (reg < codec->reg_cache_size) + cache[reg] = value; + ret = codec->hw_write(codec->control_data, data, 2); + if (ret == 2) + return 0; + if (ret < 0) + return ret; + else + return -EIO; +} + + +static struct { + int addr_bits; + int data_bits; + int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); + unsigned int (*read)(struct snd_soc_codec *, unsigned int); +} io_types[] = { + { 7, 9, snd_soc_7_9_write, snd_soc_7_9_read }, +}; + +/** + * snd_soc_codec_set_cache_io: Set up standard I/O functions. + * + * @codec: CODEC to configure. + * @type: Type of cache. + * @addr_bits: Number of bits of register address data. + * @data_bits: Number of bits of data per register. + * + * Register formats are frequently shared between many I2C and SPI + * devices. In order to promote code reuse the ASoC core provides + * some standard implementations of CODEC read and write operations + * which can be set up using this function. + * + * The caller is responsible for allocating and initialising the + * actual cache. + * + * Note that at present this code cannot be used by CODECs with + * volatile registers. + */ +int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, + int addr_bits, int data_bits) +{ + int i; + + /* We don't support volatile registers yet - refactoring of + * the hw_read operation will be required to do so. */ + if (codec->volatile_register) { + printk(KERN_ERR "Volatile registers not yet supported\n"); + return -EINVAL; + } + + for (i = 0; i < ARRAY_SIZE(io_types); i++) + if (io_types[i].addr_bits == addr_bits && + io_types[i].data_bits == data_bits) + break; + if (i == ARRAY_SIZE(io_types)) { + printk(KERN_ERR + "No I/O functions for %d bit address %d bit data\n", + addr_bits, data_bits); + return -EINVAL; + } + + codec->write = io_types[i].write; + codec->read = io_types[i].read; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_codec_set_cache_io); -- 1.6.3.3

1 7

[alsa-devel] [PATCH 1/4] pulse: get rid of a number of assert()s
by Lennart Poettering 03 Aug '09

03 Aug '09

Instead of hitting an assert when any of the plugin functions is called in an invalid context we should return a clean error to make sure programs are not unnecessarily aborted. This should fix issues such as http://pulseaudio.org/ticket/595 --- pulse/ctl_pulse.c | 35 +++++++++++++++++------ pulse/pcm_pulse.c | 79 +++++++++++++++++++++++++++++++++++++++++------------ pulse/pulse.c | 34 ++++++++++++++++------- 3 files changed, 111 insertions(+), 37 deletions(-) diff --git a/pulse/ctl_pulse.c b/pulse/ctl_pulse.c index c6cf9e2..1b057ef 100644 --- a/pulse/ctl_pulse.c +++ b/pulse/ctl_pulse.c @@ -125,8 +125,9 @@ static void event_cb(pa_context * c, pa_subscription_event_type_t t, pa_operation *o; assert(ctl); - assert(ctl->p); - assert(ctl->p->context); + + if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + return; o = pa_context_get_sink_info_by_name(ctl->p->context, ctl->sink, sink_info_cb, ctl); @@ -148,8 +149,9 @@ static int pulse_update_volume(snd_ctl_pulse_t * ctl) pa_operation *o; assert(ctl); - assert(ctl->p); - assert(ctl->p->context); + + if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + return -EBADFD; o = pa_context_get_sink_info_by_name(ctl->p->context, ctl->sink, sink_info_cb, ctl); @@ -203,6 +205,9 @@ static int pulse_elem_list(snd_ctl_ext_t * ext, unsigned int offset, assert(ctl); + if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + return -EBADFD; + snd_ctl_elem_id_set_interface(id, SND_CTL_ELEM_IFACE_MIXER); pa_threaded_mainloop_lock(ctl->p->mainloop); @@ -260,7 +265,9 @@ static int pulse_get_attribute(snd_ctl_ext_t * ext, snd_ctl_ext_key_t key, return -EINVAL; assert(ctl); - assert(ctl->p); + + if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + return -EBADFD; pa_threaded_mainloop_lock(ctl->p->mainloop); @@ -311,7 +318,9 @@ static int pulse_read_integer(snd_ctl_ext_t * ext, snd_ctl_ext_key_t key, pa_cvolume *vol = NULL; assert(ctl); - assert(ctl->p); + + if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + return -EBADFD; pa_threaded_mainloop_lock(ctl->p->mainloop); @@ -361,7 +370,9 @@ static int pulse_write_integer(snd_ctl_ext_t * ext, snd_ctl_ext_key_t key, pa_cvolume *vol = NULL; assert(ctl); - assert(ctl->p && ctl->p->context); + + if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + return -EBADFD; pa_threaded_mainloop_lock(ctl->p->mainloop); @@ -465,6 +476,9 @@ static void pulse_subscribe_events(snd_ctl_ext_t * ext, int subscribe) assert(ctl); + if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + return; + pa_threaded_mainloop_lock(ctl->p->mainloop); ctl->subscribed = !!(subscribe & SND_CTL_EVENT_MASK_VALUE); @@ -481,6 +495,9 @@ static int pulse_read_event(snd_ctl_ext_t * ext, snd_ctl_elem_id_t * id, assert(ctl); + if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + return -EBADFD; + pa_threaded_mainloop_lock(ctl->p->mainloop); if (!ctl->updated || !ctl->subscribed) @@ -525,8 +542,8 @@ static int pulse_ctl_poll_revents(snd_ctl_ext_t * ext, struct pollfd *pfd, snd_ctl_pulse_t *ctl = ext->private_data; int err = 0; - assert(ctl); - assert(ctl->p); + if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + return -EBADFD; pa_threaded_mainloop_lock(ctl->p->mainloop); diff --git a/pulse/pcm_pulse.c b/pulse/pcm_pulse.c index c276839..24347f9 100644 --- a/pulse/pcm_pulse.c +++ b/pulse/pcm_pulse.c @@ -106,6 +106,9 @@ static int update_active(snd_pcm_pulse_t *pcm) { assert(pcm); + if (!pcm->p) + return -EBADFD; + ret = check_active(pcm); if (ret < 0) return ret; @@ -125,7 +128,9 @@ static int pulse_start(snd_pcm_ioplug_t * io) int err = 0, err_o = 0, err_u = 0; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); @@ -174,7 +179,9 @@ static int pulse_stop(snd_pcm_ioplug_t * io) int err = 0, err_o = 0, err_u = 0; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); @@ -224,7 +231,9 @@ static int pulse_drain(snd_pcm_ioplug_t * io) int err = 0; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); @@ -259,7 +268,9 @@ static snd_pcm_sframes_t pulse_pointer(snd_pcm_ioplug_t * io) snd_pcm_sframes_t ret = 0; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return -EBADFD; if (io->state == SND_PCM_STATE_XRUN) return -EPIPE; @@ -269,7 +280,10 @@ static snd_pcm_sframes_t pulse_pointer(snd_pcm_ioplug_t * io) pa_threaded_mainloop_lock(pcm->p->mainloop); - assert(pcm->stream); + if (!pcm->stream) { + ret = -EBADFD; + goto finish; + } ret = pulse_check_connection(pcm->p); if (ret < 0) @@ -305,11 +319,16 @@ static int pulse_delay(snd_pcm_ioplug_t * io, snd_pcm_sframes_t * delayp) pa_usec_t lat = 0; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); - assert(pcm->stream); + if (!pcm->stream) { + err = -EBADFD; + goto finish; + } for (;;) { err = pulse_check_connection(pcm->p); @@ -354,11 +373,16 @@ static snd_pcm_sframes_t pulse_write(snd_pcm_ioplug_t * io, snd_pcm_sframes_t ret = 0; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); - assert(pcm->stream); + if (!pcm->stream) { + ret = -EBADFD; + goto finish; + } ret = pulse_check_connection(pcm->p); if (ret < 0) @@ -409,11 +433,16 @@ static snd_pcm_sframes_t pulse_read(snd_pcm_ioplug_t * io, snd_pcm_sframes_t ret = 0; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); - assert(pcm->stream); + if (!pcm->stream) { + ret = -EBADFD; + goto finish; + } ret = pulse_check_connection(pcm->p); if (ret < 0) @@ -480,7 +509,9 @@ static void stream_request_cb(pa_stream * p, size_t length, void *userdata) snd_pcm_pulse_t *pcm = userdata; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return; update_active(pcm); } @@ -490,7 +521,9 @@ static void stream_underrun_cb(pa_stream * p, void *userdata) snd_pcm_pulse_t *pcm = userdata; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return; pcm->underrun = 1; } @@ -499,7 +532,9 @@ static void stream_latency_cb(pa_stream *p, void *userdata) { snd_pcm_pulse_t *pcm = userdata; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return; pa_threaded_mainloop_signal(pcm->p->mainloop, 0); } @@ -512,7 +547,9 @@ static int pulse_pcm_poll_revents(snd_pcm_ioplug_t * io, snd_pcm_pulse_t *pcm = io->private_data; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); @@ -541,7 +578,9 @@ static int pulse_prepare(snd_pcm_ioplug_t * io) unsigned c, d; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); @@ -645,7 +684,9 @@ static int pulse_hw_params(snd_pcm_ioplug_t * io, int err = 0; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); @@ -745,7 +786,9 @@ static int pulse_pause(snd_pcm_ioplug_t * io, int enable) int err = 0; assert (pcm); - assert (pcm->p); + + if (!pcm->p) + return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); diff --git a/pulse/pulse.c b/pulse/pulse.c index 3940238..95d8dde 100644 --- a/pulse/pulse.c +++ b/pulse/pulse.c @@ -32,8 +32,9 @@ int pulse_check_connection(snd_pulse_t * p) pa_context_state_t state; assert(p); - assert(p->context); - assert(p->mainloop); + + if (!p->context || !p->mainloop) + return -EBADFD; state = pa_context_get_state(p->context); @@ -77,8 +78,12 @@ int pulse_wait_operation(snd_pulse_t * p, pa_operation * o) { assert(p); assert(o); - assert(p->state == PULSE_STATE_READY); - assert(p->mainloop); + + if (p->state != PULSE_STATE_READY) + return -EBADFD; + + if (!p->mainloop) + return -EBADFD; for (;;) { int err; @@ -103,8 +108,12 @@ int pulse_wait_stream_state(snd_pulse_t * p, pa_stream * stream, assert(p); assert(stream); - assert(p->state == PULSE_STATE_READY); - assert(p->mainloop); + + if (p->state != PULSE_STATE_READY) + return -EBADFD; + + if (!p->mainloop) + return -EBADFD; for (;;) { int err; @@ -197,7 +206,9 @@ snd_pulse_t *pulse_new(void) p->context = pa_context_new(pa_threaded_mainloop_get_api(p->mainloop), buf); - assert(p->context); + + if (!p->context) + goto fail; pa_context_set_state_callback(p->context, context_state_cb, p); @@ -246,9 +257,12 @@ int pulse_connect(snd_pulse_t * p, const char *server) int err; assert(p); - assert(p->context); - assert(p->mainloop); - assert(p->state == PULSE_STATE_INIT); + + if (!p->context || !p->mainloop) + return -EBADFD; + + if (p->state != PULSE_STATE_INIT) + return -EBADFD; pa_threaded_mainloop_lock(p->mainloop); -- 1.6.3.3 Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

3 4

[alsa-devel] ASoC: DMA and SSI drivers for i.MX27
by javier Martin 03 Aug '09

03 Aug '09

As I described in one of my previous mails, I have been trying to port an old ASoC driver for kernel 2.6.22 to latest stable 2.6.30 kernel. I am currently facing a big problem which make some repeated samples appear with a regular pattern: - frequency of bad samples = 2 * period_size - bad samples in audio stream are taken from "buffer_size" samples back in the original file. You can reproduce this with: aplay -t raw -f S16_LE -r8000 8k16bitpcm.wav (http://en.wikipedia.org/wiki/WAV to get this file) Here I submit two patches: - One which adds ssi and dma drivers plus a custom machine driver (imx27asoc_list.patch). - Another patch which adds wm8974 codec support taken from ASoC wm8974-upstream branch (wm8974_list.patch). Note that the machine driver attached uses wm8974 codec, thus second patch is needed. Please, note also that patches are currently quite dirty, and in an early development state. The purpose of this mail is not to even consider to include them in any branch right now, but to gather some help to solve the problem. As soon as I have it solved I will clean all the stuff and try to prepare it for mainline. -- Javier Martin Vista Silicon S.L. Universidad de Cantabria CDTUC - FASE C - Oficina S-345 Avda de los Castros s/n 39005- Santander. Cantabria. Spain +34 942 25 32 60 www.vista-silicon.com

1 1

[alsa-devel] ALSA -LADSPA buffer
by arif setiawan 03 Aug '09

03 Aug '09

Hi all, I have some questions about ALSA - LADSPA data exchange in ALSA plugin. As I understand it, LADSPA only use 32 bit floating point, while ALSA can use many types of buffer, S8, S16_LE and so on. My question is how this buffer converted? As I work with OMAP processor I want to use DSP for signal processing. We want the LADSPA plugin connect to the DSP and get plugin advantage. But DSP only work on fixed point arithmatic, so we need to convert the 32 bit float in LADSPA to an integer type buffer. I think I can achieve it by following ALSA-LADSPA buffer conversion. How I can do this? Thank you for your help. Best regards, Arif

1 0

[alsa-devel] omap34xx mixer elements
by Venkam Bala krishna 03 Aug '09

03 Aug '09

Hi all Thanks in advance, I am newbie to alsa Currently working on porting android to omap34xx Right now porting alsa. This is a general question. Please check the below log and specify any suggestions, why is it not listing any of my capture devices. What should be done to recognise my capture devices? Do i have to do it in kernel level or user level?(since my amixer is listing all my elements, given below the log is my amixer output for verification). Any suggestions are welcome thanks once agiain, log ======================== D/AudioHardwareALSA( 799): Mixer: element name: 'Carkit' D/AudioHardwareALSA( 799): Mixer: element name: 'DAC Voice Analog Downlink' D/AudioHardwareALSA( 799): Mixer: element name: 'DAC Voice Digital Downlink' D/AudioHardwareALSA( 799): Mixer: element name: 'DAC1 Analog' D/AudioHardwareALSA( 799): Mixer: element name: 'DAC1 Digital Coarse' D/AudioHardwareALSA( 799): Mixer: element name: 'DAC1 Digital Fine' D/AudioHardwareALSA( 799): Mixer: element name: 'DAC2 Analog' D/AudioHardwareALSA( 799): Mixer: element name: 'DAC2 Digital Coarse' D/AudioHardwareALSA( 799): Mixer: element name: 'DAC2 Digital Fine' D/AudioHardwareALSA( 799): Mixer: element name: 'Earpiece' D/AudioHardwareALSA( 799): Mixer: element name: 'Headset' D/AudioHardwareALSA( 799): Mixer: element name: 'Left Digital Loopback' D/AudioHardwareALSA( 799): Mixer: element name: 'PreDriv' D/AudioHardwareALSA( 799): Mixer: element name: 'Right Digital Loopback' D/AudioHardwareALSA( 799): Mixer: element name: 'Voice Digital Loopback' D/AudioHardwareALSA( 799): Mixer: master 'PCM' not found. D/AudioHardwareALSA( 799): Mixer: route 'Earpiece' found. D/AudioHardwareALSA( 799): Mixer: route 'Speaker' not found. D/AudioHardwareALSA( 799): Mixer: route 'Bluetooth' not found. D/AudioHardwareALSA( 799): Mixer: route 'Headphone' not found. D/AudioHardwareALSA( 799): Mixer: route 'Bluetooth A2DP' not found. D/AudioHardwareALSA( 799): Mixer: element name: 'Analog' D/AudioHardwareALSA( 799): Mixer: element name: 'DAC Voice Analog Downlink' D/AudioHardwareALSA( 799): Mixer: element name: 'DAC Voice Digital Downlink' D/AudioHardwareALSA( 799): Mixer: element name: 'Left Digital Loopback' D/AudioHardwareALSA( 799): Mixer: element name: 'Right Digital Loopback' D/AudioHardwareALSA( 799): Mixer: element name: 'TX1 Digital' D/AudioHardwareALSA( 799): Mixer: element name: 'TX2 Digital' D/AudioHardwareALSA( 799): Mixer: element name: 'Voice Digital Loopback' D/AudioHardwareALSA( 799): Mixer: master 'Capture' not found. D/AudioHardwareALSA( 799): Mixer: route 'Capture' not found. D/AudioHardwareALSA( 799): Mixer: route '' not found. D/AudioHardwareALSA( 799): Mixer: route 'Bluetooth Capture' not found. D/AudioHardwareALSA( 799): Mixer: route 'Capture' not found. D/AudioHardwareALSA( 799): Mixer: route 'Bluetooth A2DP Capture' not found. D/AudioHardwareALSA( 799): mixer initialized. =========================== amixer output ================= numid=18,iface=MIXER,name='Analog Capture Volume' numid=23,iface=MIXER,name='Analog Left Capture Route' numid=22,iface=MIXER,name='Analog Right Capture Route' numid=14,iface=MIXER,name='Carkit Playback Volume' numid=34,iface=MIXER,name='CarkitL Mixer AudioL1' numid=35,iface=MIXER,name='CarkitL Mixer AudioL2' numid=33,iface=MIXER,name='CarkitL Mixer Voice' numid=31,iface=MIXER,name='CarkitR Mixer AudioR1' numid=32,iface=MIXER,name='CarkitR Mixer AudioR2' numid=30,iface=MIXER,name='CarkitR Mixer Voice' numid=11,iface=MIXER,name='DAC Voice Analog Downlink Switch' numid=10,iface=MIXER,name='DAC Voice Analog Downlink Volume' numid=9,iface=MIXER,name='DAC Voice Digital Downlink Volume' numid=7,iface=MIXER,name='DAC1 Analog Playback Switch' numid=5,iface=MIXER,name='DAC1 Analog Playback Volume' numid=3,iface=MIXER,name='DAC1 Digital Coarse Playback Volume' numid=1,iface=MIXER,name='DAC1 Digital Fine Playback Volume' numid=8,iface=MIXER,name='DAC2 Analog Playback Switch' numid=6,iface=MIXER,name='DAC2 Analog Playback Volume' numid=4,iface=MIXER,name='DAC2 Digital Coarse Playback Volume' numid=2,iface=MIXER,name='DAC2 Digital Fine Playback Volume' numid=51,iface=MIXER,name='Earpiece Mixer AudioL1' numid=52,iface=MIXER,name='Earpiece Mixer AudioL2' numid=53,iface=MIXER,name='Earpiece Mixer AudioR1' numid=50,iface=MIXER,name='Earpiece Mixer Voice' numid=15,iface=MIXER,name='Earpiece Playback Volume' numid=19,iface=MIXER,name='HS ramp delay' numid=29,iface=MIXER,name='HandsfreeL Mux' numid=28,iface=MIXER,name='HandsfreeR Mux' numid=13,iface=MIXER,name='Headset Playback Volume' numid=40,iface=MIXER,name='HeadsetL Mixer AudioL1' numid=41,iface=MIXER,name='HeadsetL Mixer AudioL2' numid=39,iface=MIXER,name='HeadsetL Mixer Voice' numid=37,iface=MIXER,name='HeadsetR Mixer AudioR1' numid=38,iface=MIXER,name='HeadsetR Mixer AudioR2' numid=36,iface=MIXER,name='HeadsetR Mixer Voice' numid=56,iface=MIXER,name='Left Digital Loopback Volume' numid=60,iface=MIXER,name='Left1 Analog Loopback Switch' numid=58,iface=MIXER,name='Left2 Analog Loopback Switch' numid=12,iface=MIXER,name='PreDriv Playback Volume' numid=47,iface=MIXER,name='PredriveL Mixer AudioL1' numid=48,iface=MIXER,name='PredriveL Mixer AudioL2' numid=49,iface=MIXER,name='PredriveL Mixer AudioR2' numid=46,iface=MIXER,name='PredriveL Mixer Voice' numid=45,iface=MIXER,name='PredriveR Mixer AudioL2' numid=43,iface=MIXER,name='PredriveR Mixer AudioR1' numid=44,iface=MIXER,name='PredriveR Mixer AudioR2' numid=42,iface=MIXER,name='PredriveR Mixer Voice' numid=55,iface=MIXER,name='Right Digital Loopback Volume' numid=61,iface=MIXER,name='Right1 Analog Loopback Switch' numid=59,iface=MIXER,name='Right2 Analog Loopback Switch' numid=25,iface=MIXER,name='TX1 Capture Route' numid=16,iface=MIXER,name='TX1 Digital Capture Volume' numid=24,iface=MIXER,name='TX2 Capture Route' numid=17,iface=MIXER,name='TX2 Digital Capture Volume' numid=21,iface=MIXER,name='Vibra H-bridge direction' numid=20,iface=MIXER,name='Vibra H-bridge mode' numid=27,iface=MIXER,name='Vibra Mux' numid=26,iface=MIXER,name='Vibra Route' numid=57,iface=MIXER,name='Voice Analog Loopback Switch' numid=54,iface=MIXER,name='Voice Digital Loopback Volume' =================== -- Regards Venkam Balakrishna

2 1

[alsa-devel] [PATCH] hda: add IbexPeak/Clarkdale HDMI model with static cvt/pin number
by Wu Fengguang 03 Aug '09

03 Aug '09

The new IbexPeak HDMI codec has 3 pin nodes and 2 converter nodes. Here we assume only the first ones will be used. Signed-off-by: Wu Fengguang <fengguang.wu(a)intel.com> --- sound/pci/hda/patch_intelhdmi.c | 104 +++++++++++++++--------------- 1 file changed, 52 insertions(+), 52 deletions(-) --- sound-2.6.orig/sound/pci/hda/patch_intelhdmi.c +++ sound-2.6/sound/pci/hda/patch_intelhdmi.c @@ -33,8 +33,8 @@ #include "hda_codec.h" #include "hda_local.h" -#define CVT_NID 0x02 /* audio converter */ -#define PIN_NID 0x03 /* HDMI output pin */ +static hda_nid_t cvt_nid; /* audio converter */ +static hda_nid_t pin_nid; /* HDMI output pin */ #define INTEL_HDMI_EVENT_TAG 0x08 @@ -44,30 +44,6 @@ struct intel_hdmi_spec { struct hdmi_eld sink_eld; }; -static struct hda_verb pinout_enable_verb[] = { - {PIN_NID, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {} /* terminator */ -}; - -static struct hda_verb unsolicited_response_verb[] = { - {PIN_NID, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | - INTEL_HDMI_EVENT_TAG}, - {} -}; - -static struct hda_verb def_chan_map[] = { - {CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x00}, - {CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x11}, - {CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x22}, - {CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x33}, - {CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x44}, - {CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x55}, - {CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x66}, - {CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x77}, - {} -}; - - struct hdmi_audio_infoframe { u8 type; /* 0x84 */ u8 ver; /* 0x01 */ @@ -244,11 +220,12 @@ static void hdmi_write_dip_byte(struct h static void hdmi_enable_output(struct hda_codec *codec) { /* Unmute */ - if (get_wcaps(codec, PIN_NID) & AC_WCAP_OUT_AMP) - snd_hda_codec_write(codec, PIN_NID, 0, + if (get_wcaps(codec, pin_nid) & AC_WCAP_OUT_AMP) + snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); /* Enable pin out */ - snd_hda_sequence_write(codec, pinout_enable_verb); + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); } /* @@ -256,8 +233,8 @@ static void hdmi_enable_output(struct hd */ static void hdmi_start_infoframe_trans(struct hda_codec *codec) { - hdmi_set_dip_index(codec, PIN_NID, 0x0, 0x0); - snd_hda_codec_write(codec, PIN_NID, 0, AC_VERB_SET_HDMI_DIP_XMIT, + hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); + snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_XMIT, AC_DIPXMIT_BEST); } @@ -266,20 +243,20 @@ static void hdmi_start_infoframe_trans(s */ static void hdmi_stop_infoframe_trans(struct hda_codec *codec) { - hdmi_set_dip_index(codec, PIN_NID, 0x0, 0x0); - snd_hda_codec_write(codec, PIN_NID, 0, AC_VERB_SET_HDMI_DIP_XMIT, + hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); + snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_XMIT, AC_DIPXMIT_DISABLE); } static int hdmi_get_channel_count(struct hda_codec *codec) { - return 1 + snd_hda_codec_read(codec, CVT_NID, 0, + return 1 + snd_hda_codec_read(codec, cvt_nid, 0, AC_VERB_GET_CVT_CHAN_COUNT, 0); } static void hdmi_set_channel_count(struct hda_codec *codec, int chs) { - snd_hda_codec_write(codec, CVT_NID, 0, + snd_hda_codec_write(codec, cvt_nid, 0, AC_VERB_SET_CVT_CHAN_COUNT, chs - 1); if (chs != hdmi_get_channel_count(codec)) @@ -294,7 +271,7 @@ static void hdmi_debug_channel_mapping(s int slot; for (i = 0; i < 8; i++) { - slot = snd_hda_codec_read(codec, CVT_NID, 0, + slot = snd_hda_codec_read(codec, cvt_nid, 0, AC_VERB_GET_HDMI_CHAN_SLOT, i); printk(KERN_DEBUG "HDMI: ASP channel %d => slot %d\n", slot >> 4, slot & 0x7); @@ -307,7 +284,7 @@ static void hdmi_parse_eld(struct hda_co struct intel_hdmi_spec *spec = codec->spec; struct hdmi_eld *eld = &spec->sink_eld; - if (!snd_hdmi_get_eld(eld, codec, PIN_NID)) + if (!snd_hdmi_get_eld(eld, codec, pin_nid)) snd_hdmi_show_eld(eld); } @@ -322,11 +299,11 @@ static void hdmi_debug_dip_size(struct h int i; int size; - size = snd_hdmi_get_eld_size(codec, PIN_NID); + size = snd_hdmi_get_eld_size(codec, pin_nid); printk(KERN_DEBUG "HDMI: ELD buf size is %d\n", size); for (i = 0; i < 8; i++) { - size = snd_hda_codec_read(codec, PIN_NID, 0, + size = snd_hda_codec_read(codec, pin_nid, 0, AC_VERB_GET_HDMI_DIP_SIZE, i); printk(KERN_DEBUG "HDMI: DIP GP[%d] buf size is %d\n", i, size); } @@ -340,15 +317,15 @@ static void hdmi_clear_dip_buffers(struc int size; int pi, bi; for (i = 0; i < 8; i++) { - size = snd_hda_codec_read(codec, PIN_NID, 0, + size = snd_hda_codec_read(codec, pin_nid, 0, AC_VERB_GET_HDMI_DIP_SIZE, i); if (size == 0) continue; - hdmi_set_dip_index(codec, PIN_NID, i, 0x0); + hdmi_set_dip_index(codec, pin_nid, i, 0x0); for (j = 1; j < 1000; j++) { - hdmi_write_dip_byte(codec, PIN_NID, 0x0); - hdmi_get_dip_index(codec, PIN_NID, &pi, &bi); + hdmi_write_dip_byte(codec, pin_nid, 0x0); + hdmi_get_dip_index(codec, pin_nid, &pi, &bi); if (pi != i) snd_printd(KERN_INFO "dip index %d: %d != %d\n", bi, pi, i); @@ -376,9 +353,9 @@ static void hdmi_fill_audio_infoframe(st sum += params[i]; ai->checksum = - sum; - hdmi_set_dip_index(codec, PIN_NID, 0x0, 0x0); + hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); for (i = 0; i < sizeof(ai); i++) - hdmi_write_dip_byte(codec, PIN_NID, params[i]); + hdmi_write_dip_byte(codec, pin_nid, params[i]); } /* @@ -465,6 +442,8 @@ static int hdmi_setup_channel_allocation static void hdmi_setup_channel_mapping(struct hda_codec *codec, struct hdmi_audio_infoframe *ai) { + int i; + if (!ai->CA) return; @@ -473,7 +452,11 @@ static void hdmi_setup_channel_mapping(s * ALSA sequence is front/surr/clfe/side? */ - snd_hda_sequence_write(codec, def_chan_map); + for (i = 0; i < 8; i++) + snd_hda_codec_write(codec, cvt_nid, 0, + AC_VERB_SET_HDMI_CHAN_SLOT, + (i << 4) | i); + hdmi_debug_channel_mapping(codec); } @@ -597,7 +580,6 @@ static struct hda_pcm_stream intel_hdmi_ .substreams = 1, .channels_min = 2, .channels_max = 8, - .nid = CVT_NID, /* NID to query formats and rates and setup streams */ .ops = { .open = intel_hdmi_playback_pcm_open, .close = intel_hdmi_playback_pcm_close, @@ -613,6 +595,9 @@ static int intel_hdmi_build_pcms(struct codec->num_pcms = 1; codec->pcm_info = info; + /* NID to query formats and rates and setup streams */ + intel_hdmi_pcm_playback.nid = cvt_nid; + info->name = "INTEL HDMI"; info->pcm_type = HDA_PCM_TYPE_HDMI; info->stream[SNDRV_PCM_STREAM_PLAYBACK] = intel_hdmi_pcm_playback; @@ -636,8 +621,9 @@ static int intel_hdmi_init(struct hda_co { hdmi_enable_output(codec); - snd_hda_sequence_write(codec, unsolicited_response_verb); - + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | INTEL_HDMI_EVENT_TAG); return 0; } @@ -657,7 +643,7 @@ static struct hda_codec_ops intel_hdmi_p .unsol_event = intel_hdmi_unsol_event, }; -static int patch_intel_hdmi(struct hda_codec *codec) +static int do_patch_intel_hdmi(struct hda_codec *codec) { struct intel_hdmi_spec *spec; @@ -667,7 +653,7 @@ static int patch_intel_hdmi(struct hda_c spec->multiout.num_dacs = 0; /* no analog */ spec->multiout.max_channels = 8; - spec->multiout.dig_out_nid = CVT_NID; + spec->multiout.dig_out_nid = cvt_nid; codec->spec = spec; codec->patch_ops = intel_hdmi_patch_ops; @@ -679,13 +665,27 @@ static int patch_intel_hdmi(struct hda_c return 0; } +static int patch_intel_hdmi(struct hda_codec *codec) +{ + cvt_nid = 0x02; + pin_nid = 0x03; + return do_patch_intel_hdmi(codec); +} + +static int patch_intel_hdmi_ibexpeak(struct hda_codec *codec) +{ + cvt_nid = 0x02; + pin_nid = 0x04; + return do_patch_intel_hdmi(codec); +} + static struct hda_codec_preset snd_hda_preset_intelhdmi[] = { { .id = 0x808629fb, .name = "G45 DEVCL", .patch = patch_intel_hdmi }, { .id = 0x80862801, .name = "G45 DEVBLC", .patch = patch_intel_hdmi }, { .id = 0x80862802, .name = "G45 DEVCTG", .patch = patch_intel_hdmi }, { .id = 0x80862803, .name = "G45 DEVELK", .patch = patch_intel_hdmi }, { .id = 0x80862804, .name = "G45 DEVIBX", .patch = patch_intel_hdmi }, - { .id = 0x80860054, .name = "G45 DEVIBX", .patch = patch_intel_hdmi }, + { .id = 0x80860054, .name = "P55 DEVIBX", .patch = patch_intel_hdmi_ibexpeak }, { .id = 0x10951392, .name = "SiI1392 HDMI", .patch = patch_intel_hdmi }, {} /* terminator */ };

3 6

[alsa-devel] [PATCH] Tell user that stream to be rewound is suspended
by Lubomir Rintel 03 Aug '09

03 Aug '09

Return STRPIPE instead of EBADF when userspace attempts to rewind of forward a stream that was suspended in meanwhile, so that it can be recovered by snd_pcm_recover(). This was causing Pulseaudio to unload the ALSA sink module under a race condition when it attempted to rewind the stream right after resume from suspend, before writing to the stream which would cause it to revive the stream otherwise. Tested to work with Pulseaudio patched to attempt to snd_pcm_recover() upon receiving an error from snd_pcm_rewind(). Signed-off-by: Lubomir Rintel <lkundrak(a)v3.sk> --- sound/core/pcm_native.c | 12 ++++++++++++ 1 files changed, 12 insertions(+), 0 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index ac2150e..d89c816 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2208,6 +2208,9 @@ static snd_pcm_sframes_t snd_pcm_playback_rewind(struct snd_pcm_substream *subst case SNDRV_PCM_STATE_XRUN: ret = -EPIPE; goto __end; + case SNDRV_PCM_STATE_SUSPENDED: + ret = -ESTRPIPE; + goto __end; default: ret = -EBADFD; goto __end; @@ -2253,6 +2256,9 @@ static snd_pcm_sframes_t snd_pcm_capture_rewind(struct snd_pcm_substream *substr case SNDRV_PCM_STATE_XRUN: ret = -EPIPE; goto __end; + case SNDRV_PCM_STATE_SUSPENDED: + ret = -ESTRPIPE; + goto __end; default: ret = -EBADFD; goto __end; @@ -2299,6 +2305,9 @@ static snd_pcm_sframes_t snd_pcm_playback_forward(struct snd_pcm_substream *subs case SNDRV_PCM_STATE_XRUN: ret = -EPIPE; goto __end; + case SNDRV_PCM_STATE_SUSPENDED: + ret = -ESTRPIPE; + goto __end; default: ret = -EBADFD; goto __end; @@ -2345,6 +2354,9 @@ static snd_pcm_sframes_t snd_pcm_capture_forward(struct snd_pcm_substream *subst case SNDRV_PCM_STATE_XRUN: ret = -EPIPE; goto __end; + case SNDRV_PCM_STATE_SUSPENDED: + ret = -ESTRPIPE; + goto __end; default: ret = -EBADFD; goto __end; -- 1.6.2.5

2 2

[alsa-devel] [PATCH] ALSA: Read buffer overflow
by Roel Kluin 03 Aug '09

03 Aug '09

Check whether index is within bounds before testing the element. Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8c8b273..45e0ee0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -559,7 +559,7 @@ static int alc_pin_mode_get(struct snd_kcontrol *kcontrol, /* Find enumerated value for current pinctl setting */ i = alc_pin_mode_min(dir); - while (alc_pin_mode_values[i] != pinctl && i <= alc_pin_mode_max(dir)) + while (i <= alc_pin_mode_max(dir) && alc_pin_mode_values[i] != pinctl) i++; *valp = i <= alc_pin_mode_max(dir) ? i: alc_pin_mode_min(dir); return 0;

2 1

[alsa-devel] [PATCH] [ALSA] hda: Correct EAPD for Dell Inspiron 1525
by Chengu Wang 03 Aug '09

03 Aug '09

2 1

[alsa-devel] [RFC][PATCH] hda: track CIRB/CORB command/response states for each codec
by Wu Fengguang 03 Aug '09

03 Aug '09

Recently we hit a bug in our dev board, whose HDMI codec#3 may emit redundant/spurious responses, which were then taken as responses to command for another onboard Realtek codec#2, and mess up both codecs. Extend the azx_rb.cmds and azx_rb.res to array and track each codec's commands/responses separately. This helps keep good codec safe from broken ones. Signed-off-by: Wu Fengguang <fengguang.wu(a)intel.com> --- sound/pci/hda/hda_codec.c | 2 sound/pci/hda/hda_codec.h | 2 sound/pci/hda/hda_intel.c | 76 +++++++++++++++++++++++++----------- 3 files changed, 56 insertions(+), 24 deletions(-) --- sound-2.6.orig/sound/pci/hda/hda_codec.c +++ sound-2.6/sound/pci/hda/hda_codec.c @@ -185,7 +185,7 @@ static int codec_exec_verb(struct hda_co mutex_lock(&bus->cmd_mutex); err = bus->ops.command(bus, cmd); if (!err && res) - *res = bus->ops.get_response(bus); + *res = bus->ops.get_response(bus, codec->addr); mutex_unlock(&bus->cmd_mutex); snd_hda_power_down(codec); if (res && *res == -1 && bus->rirb_error) { --- sound-2.6.orig/sound/pci/hda/hda_codec.h +++ sound-2.6/sound/pci/hda/hda_codec.h @@ -568,7 +568,7 @@ struct hda_bus_ops { /* send a single command */ int (*command)(struct hda_bus *bus, unsigned int cmd); /* get a response from the last command */ - unsigned int (*get_response)(struct hda_bus *bus); + unsigned int (*get_response)(struct hda_bus *bus, unsigned int addr); /* free the private data */ void (*private_free)(struct hda_bus *); /* attach a PCM stream */ --- sound-2.6.orig/sound/pci/hda/hda_intel.c +++ sound-2.6/sound/pci/hda/hda_intel.c @@ -260,7 +260,7 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO /* STATESTS int mask: S3,SD2,SD1,SD0 */ #define AZX_MAX_CODECS 4 -#define STATESTS_INT_MASK 0x0f +#define STATESTS_INT_MASK ((1 << AZX_MAX_CODECS) - 1) /* SD_CTL bits */ #define SD_CTL_STREAM_RESET 0x01 /* stream reset bit */ @@ -368,8 +368,8 @@ struct azx_rb { dma_addr_t addr; /* physical address of CORB/RIRB buffer */ /* for RIRB */ unsigned short rp, wp; /* read/write pointers */ - int cmds; /* number of pending requests */ - u32 res; /* last read value */ + int cmds[AZX_MAX_CODECS]; /* number of pending requests */ + u32 res[AZX_MAX_CODECS]; /* last read value */ }; struct azx { @@ -538,7 +538,8 @@ static void azx_init_cmd_io(struct azx * /* RIRB set up */ chip->rirb.addr = chip->rb.addr + 2048; chip->rirb.buf = (u32 *)(chip->rb.area + 2048); - chip->rirb.wp = chip->rirb.rp = chip->rirb.cmds = 0; + chip->rirb.wp = chip->rirb.rp = 0; + memset(chip->rirb.cmds, 0, sizeof(chip->rirb.cmds)); azx_writel(chip, RIRBLBASE, (u32)chip->rirb.addr); azx_writel(chip, RIRBUBASE, upper_32_bits(chip->rirb.addr)); @@ -559,10 +560,35 @@ static void azx_free_cmd_io(struct azx * azx_writeb(chip, CORBCTL, 0); } +static unsigned int azx_command_addr(u32 cmd) +{ + unsigned int addr = cmd >> 28; + + if (addr >= AZX_MAX_CODECS) { + snd_BUG(); + addr = 0; + } + + return addr; +} + +static unsigned int azx_response_addr(u32 res) +{ + unsigned int addr = res & 0xf; + + if (addr >= AZX_MAX_CODECS) { + snd_BUG(); + addr = 0; + } + + return addr; +} + /* send a command */ static int azx_corb_send_cmd(struct hda_bus *bus, u32 val) { struct azx *chip = bus->private_data; + unsigned int addr = azx_command_addr(val); unsigned int wp; /* add command to corb */ @@ -571,7 +597,7 @@ static int azx_corb_send_cmd(struct hda_ wp %= ICH6_MAX_CORB_ENTRIES; spin_lock_irq(&chip->reg_lock); - chip->rirb.cmds++; + chip->rirb.cmds[addr]++; chip->corb.buf[wp] = cpu_to_le32(val); azx_writel(chip, CORBWP, wp); spin_unlock_irq(&chip->reg_lock); @@ -585,13 +611,14 @@ static int azx_corb_send_cmd(struct hda_ static void azx_update_rirb(struct azx *chip) { unsigned int rp, wp; + unsigned int addr; u32 res, res_ex; wp = azx_readb(chip, RIRBWP); if (wp == chip->rirb.wp) return; chip->rirb.wp = wp; - + while (chip->rirb.rp != wp) { chip->rirb.rp++; chip->rirb.rp %= ICH6_MAX_RIRB_ENTRIES; @@ -599,18 +626,20 @@ static void azx_update_rirb(struct azx * rp = chip->rirb.rp << 1; /* an RIRB entry is 8-bytes */ res_ex = le32_to_cpu(chip->rirb.buf[rp + 1]); res = le32_to_cpu(chip->rirb.buf[rp]); + addr = azx_response_addr(res_ex); if (res_ex & ICH6_RIRB_EX_UNSOL_EV) snd_hda_queue_unsol_event(chip->bus, res, res_ex); - else if (chip->rirb.cmds) { - chip->rirb.res = res; + else if (chip->rirb.cmds[addr]) { + chip->rirb.res[addr] = res; smp_wmb(); - chip->rirb.cmds--; + chip->rirb.cmds[addr]--; } } } /* receive a response */ -static unsigned int azx_rirb_get_response(struct hda_bus *bus) +static unsigned int azx_rirb_get_response(struct hda_bus *bus, + unsigned int addr) { struct azx *chip = bus->private_data; unsigned long timeout; @@ -623,10 +652,10 @@ static unsigned int azx_rirb_get_respons azx_update_rirb(chip); spin_unlock_irq(&chip->reg_lock); } - if (!chip->rirb.cmds) { + if (!chip->rirb.cmds[addr]) { smp_rmb(); bus->rirb_error = 0; - return chip->rirb.res; /* the last value */ + return chip->rirb.res[addr]; /* the last value */ } if (time_after(jiffies, timeout)) break; @@ -699,7 +728,7 @@ static unsigned int azx_rirb_get_respons */ /* receive a response */ -static int azx_single_wait_for_response(struct azx *chip) +static int azx_single_wait_for_response(struct azx *chip, unsigned int addr) { int timeout = 50; @@ -707,7 +736,7 @@ static int azx_single_wait_for_response( /* check IRV busy bit */ if (azx_readw(chip, IRS) & ICH6_IRS_VALID) { /* reuse rirb.res as the response return value */ - chip->rirb.res = azx_readl(chip, IR); + chip->rirb.res[addr] = azx_readl(chip, IR); return 0; } udelay(1); @@ -715,7 +744,7 @@ static int azx_single_wait_for_response( if (printk_ratelimit()) snd_printd(SFX "get_response timeout: IRS=0x%x\n", azx_readw(chip, IRS)); - chip->rirb.res = -1; + chip->rirb.res[addr] = -1; return -EIO; } @@ -723,6 +752,7 @@ static int azx_single_wait_for_response( static int azx_single_send_cmd(struct hda_bus *bus, u32 val) { struct azx *chip = bus->private_data; + unsigned int addr = azx_command_addr(val); int timeout = 50; bus->rirb_error = 0; @@ -735,7 +765,7 @@ static int azx_single_send_cmd(struct hd azx_writel(chip, IC, val); azx_writew(chip, IRS, azx_readw(chip, IRS) | ICH6_IRS_BUSY); - return azx_single_wait_for_response(chip); + return azx_single_wait_for_response(chip, addr); } udelay(1); } @@ -746,10 +776,11 @@ static int azx_single_send_cmd(struct hd } /* receive a response */ -static unsigned int azx_single_get_response(struct hda_bus *bus) +static unsigned int azx_single_get_response(struct hda_bus *bus, + unsigned int addr) { struct azx *chip = bus->private_data; - return chip->rirb.res; + return chip->rirb.res[addr]; } /* @@ -772,13 +803,14 @@ static int azx_send_cmd(struct hda_bus * } /* get a response */ -static unsigned int azx_get_response(struct hda_bus *bus) +static unsigned int azx_get_response(struct hda_bus *bus, + unsigned int addr) { struct azx *chip = bus->private_data; if (chip->single_cmd) - return azx_single_get_response(bus); + return azx_single_get_response(bus, addr); else - return azx_rirb_get_response(bus); + return azx_rirb_get_response(bus, addr); } #ifdef CONFIG_SND_HDA_POWER_SAVE @@ -1252,7 +1284,7 @@ static int probe_codec(struct azx *chip, chip->probing = 1; azx_send_cmd(chip->bus, cmd); - res = azx_get_response(chip->bus); + res = azx_get_response(chip->bus, addr); chip->probing = 0; if (res == -1) return -EIO;

2 6

[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 03 Aug '09

03 Aug '09

The following changes since commit c8489c3ed319677f23e5387fec09c844365566e8: Barry Song (1): ASoC: board driver to connect bf5xx with ad1938 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Barry Song (1): ASoC: Fix checkpatch issues and typos of ad1938 codec and bf5xx-tdm dai Janusz Krzysztofik (3): TTY: Add definition of a new line discipline required by Amstrad E3 (Delta) ASoC driver ASoC: CX20442: push down machine independent line discipline bits ASoC: add support for Amstrad E3 (Delta) machine Lars-Peter Clausen (3): ASoC: neo1973_gta02_wm8753: Replace snd_soc_cnew with snd_soc_add_controls. ASoC: neo1973_gta02_wm8753: Replace deprecated s3c_gpio calls with gpiolib ASoC: jack: Fix race in snd_soc_jack_add_gpios Mark Brown (3): ASoC: Add Openmoko Neo FreeRunner (GTA02) audio driver Merge branch 'gta02-audio' into for-2.6.32 ASoC: Allow CODECs to flag invalid registers include/linux/tty.h | 4 +- include/sound/soc.h | 1 + sound/soc/blackfin/bf5xx-tdm-pcm.c | 4 +- sound/soc/blackfin/bf5xx-tdm-pcm.h | 2 +- sound/soc/blackfin/bf5xx-tdm.h | 2 +- sound/soc/codecs/ad1938.c | 50 ++- sound/soc/codecs/cx20442.c | 116 +++++- sound/soc/codecs/cx20442.h | 1 + sound/soc/omap/Kconfig | 8 + sound/soc/omap/Makefile | 2 + sound/soc/omap/ams-delta.c | 646 ++++++++++++++++++++++++++++++ sound/soc/s3c24xx/Kconfig | 9 + sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/neo1973_gta02_wm8753.c | 498 +++++++++++++++++++++++ sound/soc/soc-core.c | 3 + sound/soc/soc-jack.c | 6 +- 16 files changed, 1323 insertions(+), 31 deletions(-) create mode 100644 sound/soc/omap/ams-delta.c create mode 100644 sound/soc/s3c24xx/neo1973_gta02_wm8753.c

2 1

Re: [alsa-devel] ploting waves!
by Guilherme Longo 03 Aug '09

03 Aug '09

1 0

Re: [alsa-devel] Fwd: HDA Intel IDT 92HD71B7X - HP Pavilion DV7-1299EF - Subwoofer not Working
by Takashi Iwai 02 Aug '09

02 Aug '09

At Tue, 28 Jul 2009 20:07:29 +0200, Wael Nasreddine (a.k.a eMxyzptlk) wrote: > > On Mon, Jul 27, 2009 at 3:15 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Sat, 25 Jul 2009 21:44:46 +0200, > Wael Nasreddine (a.k.a eMxyzptlk) wrote: > > > > I still got no solution for this problem, > > Which problem? > It'd be helpful if you give a problem summary and what you've done > until now (and don't do top-posting)... > > thanks, > > Takashi > > Hi Takashi, > > I'm sorry I didn't notice that the problem wasn't quoted in my last message, > anyway the whole topic is included in the forwarded message below, I tried > using hda-verb and hda-analyzer, like you suggested in one of your emails, but > since I don't even know what I'm looking for nor what to try, I did not got > too far! > > What else do you suggest? Are you aware of anyone having a similar issue I can > query to help me figure it out ? First off, try the very latest alsa-driver snapshot. ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver-sna… The problem is that BIOS sets only one stereo speaker in the pin configuration, so the driver cannot know what to do with another speaker. In short, it's a BIOS bug. To overcome that, you need to figure out which pin corresponds to the speaker. The rest available (analog) output pins are 0x0f and 0x14. Set these as output pins, and connect to DAC. For example, # hda-verb /dev/snd/hwC0D0 0x0f SET_PIN_WID 0x40 # hda-verb /dev/snd/hwC0D0 0x0f SET_CONN 0 which enables the output of the pin 0x0f, then connects to the first DAC (= 0x10). Similarly, # hda-verb /dev/snd/hwC0D0 0x14 SET_PIN_WID 0x40 # hda-verb /dev/snd/hwC0D0 0x15 SET_CONN 0 which enables the output of the pin 0x14, then changes the connection of its mux (0x15) to the first DAC (0x10). If these don't help, it's possible that some GPIO bits play any role. Play with GPIO bits, which can be also changed via hda-verb. Takashi > > Regards, > Wael > > Forwarded conversation > Subject: HDA Intel IDT 92HD71B7X - HP Pavilion DV7-1299EF - Subwoofer not > Working > ------------------------ > > From: Wael Nasreddine (a.k.a eMxyzptlk) <mla(a)nasreddine.com> > Date: Sun, Jun 7, 2009 at 4:52 PM > To: alsa-devel(a)alsa-project.org > > Hello, > > Last night I bought myself a new laptop, the HP Pavilion DV7-1299EF. > This laptop has 2 frontal speakers and a subwoofer in the back, on > Windows, when I play a sound the 3 speakers works perfectly, and since > it has a subwoofer the spearkers hardly has any bass capabilities, On > linux the subwoofer is not working, only the 2 speakers are, hence the > sound doesn't have any bass and it's loud and annoying, very > annoying... > > I'm using alsa-drivers compiled from today's git checkout.. > > Here's the snd-hda-intel module options I tried > > options snd-hda-intel enable=1 probe_mask=1 enable_msi=1 single_cmd=0 > power_save_controller=0 power_save=0 model=hp-m4 > > options snd-hda-intel enable=1 probe_mask=1 enable_msi=1 single_cmd=0 > power_save_controller=0 power_save=0 model=hp-dv5 > > options snd-hda-intel enable=1 probe_mask=1 enable_msi=1 single_cmd=0 > power_save_controller=0 power_save=0 model=hp-hdx > > options snd-hda-intel enable=1 probe_mask=1 enable_msi=1 single_cmd=0 > power_save_controller=0 power_save=0 model=dell-m4-1 > > options snd-hda-intel enable=1 probe_mask=1 enable_msi=1 single_cmd=0 > power_save_controller=0 power_save=0 model=dell-m4-2 > > options snd-hda-intel enable=1 probe_mask=1 enable_msi=1 single_cmd=0 > power_save_controller=0 power_save=0 model=dell-m4-3 > > Whatever the model option I try, I always end up with only 2 speakers... > > Here's the alsa-info.sh output -> > http://www.alsa-project.org/db/?f=0fd7cb5749b12dd4322879bb3828f9391bfee487 > I also attached some more informative files.. > > Thanks for any help, it's really appreciated > > -- > Wael Nasreddine > > Blog : http://wael.nasreddine.com > E-mail : wael.nasreddine(a)gmail.com > gTalk : wael.nasreddine(a)gmail.com > Tel : +33.6.32.94.70.13 > Skype : eMxyzptlk > Twitter : @eMxyzptlk > > Sabayon Linux Chief Development Officer - http://www.sabayonlinux.org > > PGP: 1024D/C8DD18A2 06F6 1622 4BC8 4CEB D724 DE12 5565 3945 C8DD 18A2 > > .: An infinite number of monkeys typing into GNU emacs, > would never make a good program. (L. Torvalds 1995) :. > > ---------- > From: Wael Nasreddine (a.k.a eMxyzptlk) <mla(a)nasreddine.com> > Date: Sun, Jun 7, 2009 at 4:57 PM > To: alsa-devel <alsa-devel(a)alsa-project.org> > > ---------- > From: Takashi Iwai <tiwai(a)suse.de> > Date: Mon, Jun 8, 2009 at 7:48 AM > To: "Wael Nasreddine (a.k.a eMxyzptlk)" <mla(a)nasreddine.com> > Cc: alsa-devel <alsa-devel(a)alsa-project.org> > > At Sun, 7 Jun 2009 16:57:34 +0200, > Judging from your alsa-info.sh output, BIOS doesn't give any multiple > speaker configuration, so the driver cannot know it. > > What you need is to figure out which pin corresponds to which I/O. > Try to adjust the codec routing and the pin / amp setup manually via > hda-verb or hda-analyzer. Then try to give the default pin via sysfs > and call reconfigure to confirm that it can be detected. See > $LINUX/Documentation/sound/alsa/HD-Audio.txt for details. > > thanks, > > Takashi > > ---------- > From: Wael Nasreddine (a.k.a eMxyzptlk) <mla(a)nasreddine.com> > Date: Mon, Jun 8, 2009 at 2:54 PM > To: Takashi Iwai <tiwai(a)suse.de> > > Thank you for your reply, I'll do that once I get back home as I'm at work > ATM. > > ---------- > From: Wael Nasreddine (a.k.a eMxyzptlk) <mla(a)nasreddine.com> > Date: Tue, Jun 9, 2009 at 2:32 AM > To: Takashi Iwai <tiwai(a)suse.de> > > I took a look at hda-verb and hda-analyser and I'm afraid I can't > figure out what to do exactly, did anyone do any tutorial on how to > play with them? or should I maybe have a chroot so someone can SSH in > and try and figure it out for me?? > > Thanks > > ---------- > From: Gustavo Vieira <gustavo(a)sagui.org> > Date: Thu, Jun 11, 2009 at 8:12 PM > To: "Wael Nasreddine (a.k.a eMxyzptlk)" <mla(a)nasreddine.com> > Cc: alsa-devel <alsa-devel(a)alsa-project.org> > > Em Dom, 2009-06-07 às 16:57 +0200, Wael Nasreddine (a.k.a eMxyzptlk) > escreveu: > Sorry for hijacking your thread, but does this laptop mute the speaker > if you plug in the headphones? > > I'm experiencing problems with jack sensing in a Pavilion DV4 with the > same codec and what to know if this problem is general to the codec or > specific to my model. > > Abraços, > Gustavo > > ---------- > From: Wael Nasreddine (a.k.a eMxyzptlk) <mla(a)nasreddine.com> > Date: Thu, Jun 11, 2009 at 8:31 PM > To: Gustavo Vieira <gustavo(a)sagui.org> > > Yes it does mute the speakers once I plug in headphones.. > > ---------- > From: Wael Nasreddine (a.k.a eMxyzptlk) <mla(a)nasreddine.com> > Date: Sat, Jul 25, 2009 at 9:44 PM > To: Gustavo Vieira <gustavo(a)sagui.org> > Cc: alsa-devel <alsa-devel(a)alsa-project.org>, Takashi Iwai <tiwai(a)suse.de> > > I still got no solution for this problem, guys please HELP!! I removed > completely Windows, and I can't even enjoy music nor a movie!! > > ---------- > From: Takashi Iwai <tiwai(a)suse.de> > Date: Mon, Jul 27, 2009 at 3:15 PM > To: "Wael Nasreddine (a.k.a eMxyzptlk)" <mla(a)nasreddine.com> > Cc: Gustavo Vieira <gustavo(a)sagui.org>, alsa-devel < > alsa-devel(a)alsa-project.org> > > At Sat, 25 Jul 2009 21:44:46 +0200, > Which problem? > It'd be helpful if you give a problem summary and what you've done > until now (and don't do top-posting)... > > thanks, > > Takashi > > -- > Wael Nasreddine > > Blog : http://wael.nasreddine.com > E-mail : wael.nasreddine(a)gmail.com > gTalk : wael.nasreddine(a)gmail.com > Tel : +33.6.32.94.70.13 > Skype : eMxyzptlk > Twitter : @eMxyzptlk > > Sabayon Linux Chief Development Officer - http://www.sabayonlinux.org > > PGP: 1024D/C8DD18A2 06F6 1622 4BC8 4CEB D724 DE12 5565 3945 C8DD 18A2 > > .: An infinite number of monkeys typing into GNU emacs, > would never make a good program. (L. Torvalds 1995) :. > > > [2 aplay-l.txt <text/plain; US-ASCII (base64)>] > **** List of PLAYBACK Hardware Devices **** > card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: Intel [HDA Intel], device 1: STAC92xx Digital [STAC92xx Digital] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > [3 card0-codec0.txt <text/plain; US-ASCII (base64)>] > Codec: IDT 92HD71B7X > Address: 0 > Function Id: 0x1 > Vendor Id: 0x111d76b2 > Subsystem Id: 0x103c30f6 > Revision Id: 0x100302 > No Modem Function Group found > Default PCM: > rates [0x7e0]: 44100 48000 88200 96000 176400 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Default Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Default Amp-Out caps: ofs=0x7f, nsteps=0x7f, stepsize=0x02, mute=1 > GPIO: io=8, o=0, i=0, unsolicited=1, wake=1 > IO[0]: enable=1, dir=1, wake=0, sticky=0, data=1, unsol=0 > IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > IO[4]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > IO[5]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > IO[6]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > IO[7]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > Power-Map: 0x01 > Analog Loopback: 0x00 > Node 0x0a [Pin Complex] wcaps 0x400181: Stereo > Pincap 0x0000001c: OUT HP Detect > Pin Default 0x0221401f: [Jack] HP Out at Ext Front > Conn = 1/8, Color = Green > DefAssociation = 0x1, Sequence = 0xf > Pin-ctls: 0x00: > Unsolicited: tag=01, enabled=1 > Connection: 3 > 0x10 0x11* 0x17 > Node 0x0b [Pin Complex] wcaps 0x400081: Stereo > Pincap 0x00001724: IN Detect > Vref caps: HIZ 50 GRD 80 > Pin Default 0x02a19020: [Jack] Mic at Ext Front > Conn = 1/8, Color = Pink > DefAssociation = 0x2, Sequence = 0x0 > Pin-ctls: 0x24: IN VREF_80 > Unsolicited: tag=02, enabled=1 > Node 0x0c [Pin Complex] wcaps 0x400081: Stereo > Pincap 0x00001724: IN Detect > Vref caps: HIZ 50 GRD 80 > Pin Default 0x40f100f0: [N/A] Other at Ext N/A > Conn = 1/8, Color = Unknown > DefAssociation = 0xf, Sequence = 0x0 > Pin-ctls: 0x00: VREF_HIZ > Unsolicited: tag=00, enabled=0 > Node 0x0d [Pin Complex] wcaps 0x400181: Stereo > Pincap 0x00000014: OUT Detect > Pin Default 0x90170010: [Fixed] Speaker at Int N/A > Conn = Analog, Color = Unknown > DefAssociation = 0x1, Sequence = 0x0 > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Connection: 3 > 0x10* 0x11 0x17 > Node 0x0e [Pin Complex] wcaps 0x400081: Stereo > Pincap 0x00001724: IN Detect > Vref caps: HIZ 50 GRD 80 > Pin Default 0x40f100f2: [N/A] Other at Ext N/A > Conn = 1/8, Color = Unknown > DefAssociation = 0xf, Sequence = 0x2 > Pin-ctls: 0x00: VREF_HIZ > Unsolicited: tag=00, enabled=0 > Node 0x0f [Pin Complex] wcaps 0x400181: Stereo > Pincap 0x00000014: OUT Detect > Pin Default 0x40f100f7: [N/A] Other at Ext N/A > Conn = 1/8, Color = Unknown > DefAssociation = 0xf, Sequence = 0x7 > Pin-ctls: 0x00: > Unsolicited: tag=00, enabled=0 > Connection: 3 > 0x10 0x11 0x17* > Node 0x10 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L > Amp-Out caps: N/A > Amp-Out vals: [0x7f 0x7f] > Converter: stream=0, channel=0 > Power: setting=D0, actual=D0 > Delay: 13 samples > Node 0x11 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L > Amp-Out caps: N/A > Amp-Out vals: [0x7f 0x7f] > Converter: stream=0, channel=0 > Power: setting=D0, actual=D0 > Delay: 13 samples > Node 0x12 [Audio Input] wcaps 0x1d0541: Stereo > Converter: stream=0, channel=0 > SDI-Select: 0 > Power: setting=D3, actual=D3 > Delay: 13 samples > Connection: 1 > 0x1c > Processing caps: benign=0, ncoeff=0 > Node 0x13 [Audio Input] wcaps 0x1d0541: Stereo > Converter: stream=0, channel=0 > SDI-Select: 0 > Power: setting=D3, actual=D3 > Delay: 13 samples > Connection: 1 > 0x1d > Processing caps: benign=0, ncoeff=0 > Node 0x14 [Pin Complex] wcaps 0x400100: Mono > Pincap 0x00000010: OUT > Pin Default 0x40f100f3: [N/A] Other at Ext N/A > Conn = 1/8, Color = Unknown > DefAssociation = 0xf, Sequence = 0x3 > Pin-ctls: 0x00: > Connection: 1 > 0x16 > Node 0x15 [Audio Selector] wcaps 0x300101: Stereo > Connection: 3 > 0x10* 0x11 0x17 > Node 0x16 [Audio Mixer] wcaps 0x200100: Mono > Connection: 1 > 0x15 > Node 0x17 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 > Amp-In vals: [0x97 0x97] [0x97 0x97] [0x97 0x97] [0x97 0x97] [0x97 0x97] > Connection: 5 > 0x10 0x11 0x27 0x1a 0x1b > Node 0x18 [Pin Complex] wcaps 0x40000d: Stereo Amp-Out > Amp-Out caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-Out vals: [0x00 0x00] > Pincap 0x00000020: IN > Pin Default 0x85a61150: [Fixed] Mic at Ext Top > Conn = Digital, Color = Black > DefAssociation = 0x5, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x20: IN > Node 0x19 [Pin Complex] wcaps 0x40000d: Stereo Amp-Out > Amp-Out caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-Out vals: [0x00 0x00] > Pincap 0x00000020: IN > Pin Default 0x40f100f4: [N/A] Other at Ext N/A > Conn = 1/8, Color = Unknown > DefAssociation = 0xf, Sequence = 0x4 > Pin-ctls: 0x00: > Node 0x1a [Audio Selector] wcaps 0x30010d: Stereo Amp-Out > Amp-Out caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-Out vals: [0x00 0x00] > Connection: 3 > 0x0b* 0x0c 0x0e > Node 0x1b [Audio Selector] wcaps 0x30010d: Stereo Amp-Out > Amp-Out caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-Out vals: [0x00 0x00] > Connection: 3 > 0x0b* 0x0c 0x0e > Node 0x1c [Audio Selector] wcaps 0x30090d: Stereo Amp-Out R/L > Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 > Amp-Out vals: [0x0c 0x0c] > Connection: 4 > 0x1a* 0x17 0x18 0x19 > Node 0x1d [Audio Selector] wcaps 0x30090d: Stereo Amp-Out R/L > Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 > Amp-Out vals: [0x80 0x80] > Connection: 4 > 0x1b* 0x17 0x18 0x19 > Node 0x1e [Pin Complex] wcaps 0x400301: Stereo Digital > Pincap 0x00000010: OUT > Pin Default 0x014411a0: [Jack] SPDIF Out at Ext Rear > Conn = RCA, Color = Black > DefAssociation = 0xa, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Connection: 1 > 0x24 > Node 0x1f [Pin Complex] wcaps 0x400701: Stereo Digital > Pincap 0x00010010: OUT EAPD > EAPD 0x0: > Pin Default 0x40f100f5: [N/A] Other at Ext N/A > Conn = 1/8, Color = Unknown > DefAssociation = 0xf, Sequence = 0x5 > Pin-ctls: 0x00: > Power: setting=D0, actual=D0 > Connection: 2 > 0x24* 0x25 > Node 0x20 [Pin Complex] wcaps 0x400301: Stereo Digital > Pincap 0x00000010: OUT > Pin Default 0x40f100f6: [N/A] Other at Ext N/A > Conn = 1/8, Color = Unknown > DefAssociation = 0xf, Sequence = 0x6 > Pin-ctls: 0x00: > Connection: 1 > 0x25 > Node 0x21 [Audio Output] wcaps 0x40211: Stereo Digital > Converter: stream=0, channel=0 > Digital: Enabled > Digital category: 0x0 > PCM: > rates [0x7e0]: 44100 48000 88200 96000 176400 192000 > bits [0xe]: 16 20 24 > formats [0x5]: PCM AC3 > Delay: 4 samples > Node 0x22 [Audio Output] wcaps 0x40211: Stereo Digital > Converter: stream=0, channel=0 > Digital: Enabled > Digital category: 0x0 > PCM: > rates [0x7e0]: 44100 48000 88200 96000 176400 192000 > bits [0xe]: 16 20 24 > formats [0x5]: PCM AC3 > Delay: 4 samples > Node 0x23 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x24 [Audio Selector] wcaps 0x300101: Stereo > Connection: 3 > 0x21* 0x1c 0x1d > Node 0x25 [Audio Selector] wcaps 0x300101: Stereo > Connection: 3 > 0x22* 0x1c 0x1d > Node 0x26 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out > Amp-Out caps: ofs=0x03, nsteps=0x03, stepsize=0x17, mute=1 > Amp-Out vals: [0x00] > Node 0x27 [Pin Complex] wcaps 0x400000: Mono > Pincap 0x00000020: IN > Pin Default 0x40f100f8: [N/A] Other at Ext N/A > Conn = 1/8, Color = Unknown > DefAssociation = 0xf, Sequence = 0x8 > Pin-ctls: 0x00: > Node 0x28 [Volume Knob Widget] wcaps 0x600000: Mono > Volume-Knob: delta=1, steps=127, direct=1, val=127 > Connection: 2 > 0x10 0x11 > [4 kernel-config.txt <text/plain; US-ASCII (base64)>] > # > # Automatically generated make config: don't edit > # Linux kernel version: 2.6.29-gentoo-r4 > # Sun Jun 7 17:04:50 2009 > # > CONFIG_64BIT=y > # CONFIG_X86_32 is not set > CONFIG_X86_64=y > CONFIG_X86=y > CONFIG_ARCH_DEFCONFIG="arch/x86/configs/x86_64_defconfig" > CONFIG_GENERIC_TIME=y > CONFIG_GENERIC_CMOS_UPDATE=y > CONFIG_CLOCKSOURCE_WATCHDOG=y > CONFIG_GENERIC_CLOCKEVENTS=y > CONFIG_GENERIC_CLOCKEVENTS_BROADCAST=y > CONFIG_LOCKDEP_SUPPORT=y > CONFIG_STACKTRACE_SUPPORT=y > CONFIG_HAVE_LATENCYTOP_SUPPORT=y > CONFIG_FAST_CMPXCHG_LOCAL=y > CONFIG_MMU=y > CONFIG_ZONE_DMA=y > CONFIG_GENERIC_ISA_DMA=y > CONFIG_GENERIC_IOMAP=y > CONFIG_GENERIC_BUG=y > CONFIG_GENERIC_BUG_RELATIVE_POINTERS=y > CONFIG_GENERIC_HWEIGHT=y > CONFIG_ARCH_MAY_HAVE_PC_FDC=y > CONFIG_RWSEM_GENERIC_SPINLOCK=y > # CONFIG_RWSEM_XCHGADD_ALGORITHM is not set > CONFIG_ARCH_HAS_CPU_IDLE_WAIT=y > CONFIG_GENERIC_CALIBRATE_DELAY=y > CONFIG_GENERIC_TIME_VSYSCALL=y > CONFIG_ARCH_HAS_CPU_RELAX=y > CONFIG_ARCH_HAS_DEFAULT_IDLE=y > CONFIG_ARCH_HAS_CACHE_LINE_SIZE=y > CONFIG_HAVE_SETUP_PER_CPU_AREA=y > CONFIG_HAVE_CPUMASK_OF_CPU_MAP=y > CONFIG_ARCH_HIBERNATION_POSSIBLE=y > CONFIG_ARCH_SUSPEND_POSSIBLE=y > CONFIG_ZONE_DMA32=y > CONFIG_ARCH_POPULATES_NODE_MAP=y > CONFIG_AUDIT_ARCH=y > CONFIG_ARCH_SUPPORTS_OPTIMIZED_INLINING=y > CONFIG_GENERIC_HARDIRQS=y > CONFIG_GENERIC_IRQ_PROBE=y > CONFIG_GENERIC_PENDING_IRQ=y > CONFIG_X86_SMP=y > CONFIG_USE_GENERIC_SMP_HELPERS=y > CONFIG_X86_64_SMP=y > CONFIG_X86_HT=y > CONFIG_X86_BIOS_REBOOT=y > CONFIG_X86_TRAMPOLINE=y > # CONFIG_KTIME_SCALAR is not set > CONFIG_DEFCONFIG_LIST="/lib/modules/$UNAME_RELEASE/.config" > > # > # General setup > # > CONFIG_EXPERIMENTAL=y > CONFIG_LOCK_KERNEL=y > CONFIG_INIT_ENV_ARG_LIMIT=32 > CONFIG_LOCALVERSION="" > # CONFIG_LOCALVERSION_AUTO is not set > CONFIG_SWAP=y > CONFIG_SYSVIPC=y > CONFIG_SYSVIPC_SYSCTL=y > # CONFIG_POSIX_MQUEUE is not set > CONFIG_BSD_PROCESS_ACCT=y > CONFIG_BSD_PROCESS_ACCT_V3=y > CONFIG_TASKSTATS=y > # CONFIG_TASK_DELAY_ACCT is not set > CONFIG_TASK_XACCT=y > CONFIG_TASK_IO_ACCOUNTING=y > CONFIG_AUDIT=y > CONFIG_AUDITSYSCALL=y > CONFIG_AUDIT_TREE=y > > # > # RCU Subsystem > # > CONFIG_CLASSIC_RCU=y > # CONFIG_TREE_RCU is not set > # CONFIG_PREEMPT_RCU is not set > # CONFIG_TREE_RCU_TRACE is not set > # CONFIG_PREEMPT_RCU_TRACE is not set > CONFIG_IKCONFIG=y > CONFIG_IKCONFIG_PROC=y > CONFIG_LOG_BUF_SHIFT=15 > CONFIG_HAVE_UNSTABLE_SCHED_CLOCK=y > CONFIG_GROUP_SCHED=y > CONFIG_FAIR_GROUP_SCHED=y > # CONFIG_RT_GROUP_SCHED is not set > CONFIG_USER_SCHED=y > # CONFIG_CGROUP_SCHED is not set > # CONFIG_CGROUPS is not set > CONFIG_SYSFS_DEPRECATED=y > CONFIG_SYSFS_DEPRECATED_V2=y > # CONFIG_RELAY is not set > # CONFIG_NAMESPACES is not set > CONFIG_BLK_DEV_INITRD=y > CONFIG_INITRAMFS_SOURCE="/usr/share/v86d/initramfs" > CONFIG_INITRAMFS_ROOT_UID=0 > CONFIG_INITRAMFS_ROOT_GID=0 > CONFIG_CC_OPTIMIZE_FOR_SIZE=y > CONFIG_SYSCTL=y > CONFIG_ANON_INODES=y > CONFIG_EMBEDDED=y > CONFIG_UID16=y > CONFIG_SYSCTL_SYSCALL=y > # CONFIG_KALLSYMS is not set > CONFIG_HOTPLUG=y > CONFIG_PRINTK=y > CONFIG_BUG=y > CONFIG_ELF_CORE=y > CONFIG_PCSPKR_PLATFORM=y > CONFIG_BASE_FULL=y > CONFIG_FUTEX=y > CONFIG_EPOLL=y > CONFIG_SIGNALFD=y > CONFIG_TIMERFD=y > CONFIG_EVENTFD=y > CONFIG_SHMEM=y > CONFIG_AIO=y > CONFIG_VM_EVENT_COUNTERS=y > CONFIG_PCI_QUIRKS=y > CONFIG_COMPAT_BRK=y > CONFIG_SLAB=y > # CONFIG_SLUB is not set > # CONFIG_SLOB is not set > # CONFIG_PROFILING is not set > CONFIG_HAVE_OPROFILE=y > CONFIG_HAVE_EFFICIENT_UNALIGNED_ACCESS=y > CONFIG_HAVE_IOREMAP_PROT=y > CONFIG_HAVE_KPROBES=y > CONFIG_HAVE_KRETPROBES=y > CONFIG_HAVE_ARCH_TRACEHOOK=y > # CONFIG_HAVE_GENERIC_DMA_COHERENT is not set > CONFIG_SLABINFO=y > CONFIG_RT_MUTEXES=y > CONFIG_BASE_SMALL=0 > CONFIG_MODULES=y > # CONFIG_MODULE_FORCE_LOAD is not set > CONFIG_MODULE_UNLOAD=y > CONFIG_MODULE_FORCE_UNLOAD=y > CONFIG_MODVERSIONS=y > # CONFIG_MODULE_SRCVERSION_ALL is not set > CONFIG_STOP_MACHINE=y > CONFIG_BLOCK=y > # CONFIG_BLK_DEV_IO_TRACE is not set > # CONFIG_BLK_DEV_BSG is not set > # CONFIG_BLK_DEV_INTEGRITY is not set > CONFIG_BLOCK_COMPAT=y > > # > # IO Schedulers > # > CONFIG_IOSCHED_NOOP=y > CONFIG_IOSCHED_AS=y > CONFIG_IOSCHED_DEADLINE=y > CONFIG_IOSCHED_CFQ=y > # CONFIG_DEFAULT_AS is not set > # CONFIG_DEFAULT_DEADLINE is not set > CONFIG_DEFAULT_CFQ=y > # CONFIG_DEFAULT_NOOP is not set > CONFIG_DEFAULT_IOSCHED="cfq" > CONFIG_FREEZER=y > > # > # Processor type and features > # > # CONFIG_NO_HZ is not set > # CONFIG_HIGH_RES_TIMERS is not set > CONFIG_GENERIC_CLOCKEVENTS_BUILD=y > CONFIG_SMP=y > # CONFIG_SPARSE_IRQ is not set > CONFIG_X86_FIND_SMP_CONFIG=y > CONFIG_X86_MPPARSE=y > CONFIG_X86_PC=y > # CONFIG_X86_ELAN is not set > # CONFIG_X86_VOYAGER is not set > # CONFIG_X86_GENERICARCH is not set > # CONFIG_X86_VSMP is not set > CONFIG_SCHED_OMIT_FRAME_POINTER=y > # CONFIG_PARAVIRT_GUEST is not set > CONFIG_MEMTEST=y > # CONFIG_M386 is not set > # CONFIG_M486 is not set > # CONFIG_M586 is not set > # CONFIG_M586TSC is not set > # CONFIG_M586MMX is not set > # CONFIG_M686 is not set > # CONFIG_MPENTIUMII is not set > # CONFIG_MPENTIUMIII is not set > # CONFIG_MPENTIUMM is not set > # CONFIG_MPENTIUM4 is not set > # CONFIG_MK6 is not set > # CONFIG_MK7 is not set > # CONFIG_MK8 is not set > # CONFIG_MCRUSOE is not set > # CONFIG_MEFFICEON is not set > # CONFIG_MWINCHIPC6 is not set > # CONFIG_MWINCHIP3D is not set > # CONFIG_MGEODEGX1 is not set > # CONFIG_MGEODE_LX is not set > # CONFIG_MCYRIXIII is not set > # CONFIG_MVIAC3_2 is not set > # CONFIG_MVIAC7 is not set > # CONFIG_MPSC is not set > CONFIG_MCORE2=y > # CONFIG_GENERIC_CPU is not set > CONFIG_X86_CPU=y > CONFIG_X86_L1_CACHE_BYTES=64 > CONFIG_X86_INTERNODE_CACHE_BYTES=64 > CONFIG_X86_CMPXCHG=y > CONFIG_X86_L1_CACHE_SHIFT=6 > CONFIG_X86_WP_WORKS_OK=y > CONFIG_X86_INTEL_USERCOPY=y > CONFIG_X86_USE_PPRO_CHECKSUM=y > CONFIG_X86_P6_NOP=y > CONFIG_X86_TSC=y > CONFIG_X86_CMPXCHG64=y > CONFIG_X86_CMOV=y > CONFIG_X86_MINIMUM_CPU_FAMILY=64 > CONFIG_X86_DEBUGCTLMSR=y > # CONFIG_PROCESSOR_SELECT is not set > CONFIG_CPU_SUP_INTEL=y > CONFIG_CPU_SUP_AMD=y > CONFIG_CPU_SUP_CENTAUR_64=y > # CONFIG_X86_DS is not set > CONFIG_HPET_TIMER=y > CONFIG_HPET_EMULATE_RTC=y > CONFIG_DMI=y > CONFIG_GART_IOMMU=y > # CONFIG_CALGARY_IOMMU is not set > # CONFIG_AMD_IOMMU is not set > CONFIG_SWIOTLB=y > CONFIG_IOMMU_HELPER=y > # CONFIG_IOMMU_API is not set > CONFIG_NR_CPUS=4 > CONFIG_SCHED_SMT=y > CONFIG_SCHED_MC=y > # CONFIG_PREEMPT_NONE is not set > CONFIG_PREEMPT_VOLUNTARY=y > # CONFIG_PREEMPT is not set > CONFIG_X86_LOCAL_APIC=y > CONFIG_X86_IO_APIC=y > # CONFIG_X86_REROUTE_FOR_BROKEN_BOOT_IRQS is not set > CONFIG_X86_MCE=y > CONFIG_X86_MCE_INTEL=y > CONFIG_X86_MCE_AMD=y > # CONFIG_I8K is not set > CONFIG_MICROCODE=m > CONFIG_MICROCODE_INTEL=y > # CONFIG_MICROCODE_AMD is not set > CONFIG_MICROCODE_OLD_INTERFACE=y > CONFIG_X86_MSR=m > CONFIG_X86_CPUID=m > CONFIG_ARCH_PHYS_ADDR_T_64BIT=y > CONFIG_DIRECT_GBPAGES=y > # CONFIG_NUMA is not set > CONFIG_ARCH_SPARSEMEM_DEFAULT=y > CONFIG_ARCH_SPARSEMEM_ENABLE=y > CONFIG_ARCH_SELECT_MEMORY_MODEL=y > CONFIG_SELECT_MEMORY_MODEL=y > # CONFIG_FLATMEM_MANUAL is not set > # CONFIG_DISCONTIGMEM_MANUAL is not set > CONFIG_SPARSEMEM_MANUAL=y > CONFIG_SPARSEMEM=y > CONFIG_HAVE_MEMORY_PRESENT=y > CONFIG_SPARSEMEM_EXTREME=y > CONFIG_SPARSEMEM_VMEMMAP_ENABLE=y > CONFIG_SPARSEMEM_VMEMMAP=y > > # > # Memory hotplug is currently incompatible with Software Suspend > # > CONFIG_PAGEFLAGS_EXTENDED=y > CONFIG_SPLIT_PTLOCK_CPUS=4 > CONFIG_PHYS_ADDR_T_64BIT=y > CONFIG_ZONE_DMA_FLAG=1 > CONFIG_BOUNCE=y > CONFIG_VIRT_TO_BUS=y > CONFIG_UNEVICTABLE_LRU=y > # CONFIG_X86_CHECK_BIOS_CORRUPTION is not set > CONFIG_X86_RESERVE_LOW_64K=y > CONFIG_MTRR=y > CONFIG_MTRR_SANITIZER=y > CONFIG_MTRR_SANITIZER_ENABLE_DEFAULT=0 > CONFIG_MTRR_SANITIZER_SPARE_REG_NR_DEFAULT=1 > CONFIG_X86_PAT=y > # CONFIG_EFI is not set > CONFIG_SECCOMP=y > # CONFIG_HZ_100 is not set > # CONFIG_HZ_250 is not set > # CONFIG_HZ_300 is not set > CONFIG_HZ_1000=y > CONFIG_HZ=1000 > # CONFIG_SCHED_HRTICK is not set > CONFIG_KEXEC=y > # CONFIG_CRASH_DUMP is not set > CONFIG_PHYSICAL_START=0x200000 > # CONFIG_RELOCATABLE is not set > CONFIG_PHYSICAL_ALIGN=0x200000 > CONFIG_HOTPLUG_CPU=y > CONFIG_COMPAT_VDSO=y > # CONFIG_CMDLINE_BOOL is not set > CONFIG_ARCH_ENABLE_MEMORY_HOTPLUG=y > > # > # Power management and ACPI options > # > CONFIG_ARCH_HIBERNATION_HEADER=y > CONFIG_PM=y > # CONFIG_PM_DEBUG is not set > CONFIG_PM_SLEEP_SMP=y > CONFIG_PM_SLEEP=y > CONFIG_SUSPEND=y > CONFIG_SUSPEND_FREEZER=y > CONFIG_HIBERNATION=y > CONFIG_PM_STD_PARTITION="/dev/system/suspend-swap" > CONFIG_ACPI=y > CONFIG_ACPI_SLEEP=y > CONFIG_ACPI_PROCFS=y > CONFIG_ACPI_PROCFS_POWER=y > CONFIG_ACPI_SYSFS_POWER=y > CONFIG_ACPI_PROC_EVENT=y > CONFIG_ACPI_AC=y > CONFIG_ACPI_BATTERY=y > CONFIG_ACPI_BUTTON=y > CONFIG_ACPI_VIDEO=m > CONFIG_ACPI_FAN=y > CONFIG_ACPI_DOCK=y > CONFIG_ACPI_PROCESSOR=y > CONFIG_ACPI_HOTPLUG_CPU=y > CONFIG_ACPI_THERMAL=y > # CONFIG_ACPI_CUSTOM_DSDT is not set > CONFIG_ACPI_BLACKLIST_YEAR=0 > # CONFIG_ACPI_DEBUG is not set > # CONFIG_ACPI_PCI_SLOT is not set > CONFIG_X86_PM_TIMER=y > CONFIG_ACPI_CONTAINER=y > # CONFIG_ACPI_SBS is not set > > # > # CPU Frequency scaling > # > CONFIG_CPU_FREQ=y > CONFIG_CPU_FREQ_TABLE=y > # CONFIG_CPU_FREQ_DEBUG is not set > CONFIG_CPU_FREQ_STAT=y > CONFIG_CPU_FREQ_STAT_DETAILS=y > # CONFIG_CPU_FREQ_DEFAULT_GOV_PERFORMANCE is not set > # CONFIG_CPU_FREQ_DEFAULT_GOV_POWERSAVE is not set > # CONFIG_CPU_FREQ_DEFAULT_GOV_USERSPACE is not set > CONFIG_CPU_FREQ_DEFAULT_GOV_ONDEMAND=y > # CONFIG_CPU_FREQ_DEFAULT_GOV_CONSERVATIVE is not set > CONFIG_CPU_FREQ_GOV_PERFORMANCE=y > CONFIG_CPU_FREQ_GOV_POWERSAVE=m > CONFIG_CPU_FREQ_GOV_USERSPACE=m > CONFIG_CPU_FREQ_GOV_ONDEMAND=y > CONFIG_CPU_FREQ_GOV_CONSERVATIVE=m > > # > # CPUFreq processor drivers > # > CONFIG_X86_ACPI_CPUFREQ=m > # CONFIG_X86_POWERNOW_K8 is not set > # CONFIG_X86_SPEEDSTEP_CENTRINO is not set > # CONFIG_X86_P4_CLOCKMOD is not set > > # > # shared options > # > # CONFIG_X86_SPEEDSTEP_LIB is not set > CONFIG_CPU_IDLE=y > CONFIG_CPU_IDLE_GOV_LADDER=y > > # > # Memory power savings > # > # CONFIG_I7300_IDLE is not set > > # > # Bus options (PCI etc.) > # > CONFIG_PCI=y > CONFIG_PCI_DIRECT=y > CONFIG_PCI_MMCONFIG=y > CONFIG_PCI_DOMAINS=y > # CONFIG_PCIEPORTBUS is not set > CONFIG_ARCH_SUPPORTS_MSI=y > # CONFIG_PCI_MSI is not set > CONFIG_PCI_LEGACY=y > # CONFIG_PCI_STUB is not set > CONFIG_HT_IRQ=y > CONFIG_ISA_DMA_API=y > CONFIG_K8_NB=y > CONFIG_PCCARD=m > # CONFIG_PCMCIA_DEBUG is not set > # CONFIG_PCMCIA is not set > CONFIG_CARDBUS=y > > # > # PC-card bridges > # > CONFIG_YENTA=m > # CONFIG_YENTA_O2 is not set > # CONFIG_YENTA_RICOH is not set > CONFIG_YENTA_TI=y > CONFIG_YENTA_ENE_TUNE=y > # CONFIG_YENTA_TOSHIBA is not set > CONFIG_PCCARD_NONSTATIC=m > # CONFIG_HOTPLUG_PCI is not set > > # > # Executable file formats / Emulations > # > CONFIG_BINFMT_ELF=y > CONFIG_COMPAT_BINFMT_ELF=y > # CONFIG_CORE_DUMP_DEFAULT_ELF_HEADERS is not set > # CONFIG_HAVE_AOUT is not set > CONFIG_BINFMT_MISC=m > CONFIG_IA32_EMULATION=y > CONFIG_IA32_AOUT=m > CONFIG_COMPAT=y > CONFIG_COMPAT_FOR_U64_ALIGNMENT=y > CONFIG_SYSVIPC_COMPAT=y > CONFIG_NET=y > > # > # Networking options > # > CONFIG_COMPAT_NET_DEV_OPS=y > CONFIG_PACKET=y > # CONFIG_PACKET_MMAP is not set > CONFIG_UNIX=y > CONFIG_XFRM=y > # CONFIG_XFRM_USER is not set > # CONFIG_XFRM_SUB_POLICY is not set > # CONFIG_XFRM_MIGRATE is not set > # CONFIG_XFRM_STATISTICS is not set > CONFIG_XFRM_IPCOMP=m > # CONFIG_NET_KEY is not set > CONFIG_INET=y > CONFIG_IP_MULTICAST=y > # CONFIG_IP_ADVANCED_ROUTER is not set > CONFIG_IP_FIB_HASH=y > # CONFIG_IP_PNP is not set > CONFIG_NET_IPIP=m > # CONFIG_NET_IPGRE is not set > CONFIG_IP_MROUTE=y > CONFIG_IP_PIMSM_V1=y > CONFIG_IP_PIMSM_V2=y > # CONFIG_ARPD is not set > # CONFIG_SYN_COOKIES is not set > # CONFIG_INET_AH is not set > # CONFIG_INET_ESP is not set > # CONFIG_INET_IPCOMP is not set > # CONFIG_INET_XFRM_TUNNEL is not set > CONFIG_INET_TUNNEL=m > CONFIG_INET_XFRM_MODE_TRANSPORT=y > CONFIG_INET_XFRM_MODE_TUNNEL=y > CONFIG_INET_XFRM_MODE_BEET=y > # CONFIG_INET_LRO is not set > CONFIG_INET_DIAG=y > CONFIG_INET_TCP_DIAG=y > # CONFIG_TCP_CONG_ADVANCED is not set > CONFIG_TCP_CONG_CUBIC=y > CONFIG_DEFAULT_TCP_CONG="cubic" > # CONFIG_TCP_MD5SIG is not set > CONFIG_IPV6=m > CONFIG_IPV6_PRIVACY=y > # CONFIG_IPV6_ROUTER_PREF is not set > # CONFIG_IPV6_OPTIMISTIC_DAD is not set > CONFIG_INET6_AH=m > CONFIG_INET6_ESP=m > CONFIG_INET6_IPCOMP=m > # CONFIG_IPV6_MIP6 is not set > CONFIG_INET6_XFRM_TUNNEL=m > CONFIG_INET6_TUNNEL=m > CONFIG_INET6_XFRM_MODE_TRANSPORT=m > CONFIG_INET6_XFRM_MODE_TUNNEL=m > CONFIG_INET6_XFRM_MODE_BEET=m > # CONFIG_INET6_XFRM_MODE_ROUTEOPTIMIZATION is not set > CONFIG_IPV6_SIT=m > CONFIG_IPV6_NDISC_NODETYPE=y > CONFIG_IPV6_TUNNEL=m > # CONFIG_IPV6_MULTIPLE_TABLES is not set > # CONFIG_IPV6_MROUTE is not set > # CONFIG_NETWORK_SECMARK is not set > CONFIG_NETFILTER=y > # CONFIG_NETFILTER_DEBUG is not set > CONFIG_NETFILTER_ADVANCED=y > CONFIG_BRIDGE_NETFILTER=y > > # > # Core Netfilter Configuration > # > # CONFIG_NETFILTER_NETLINK_QUEUE is not set > # CONFIG_NETFILTER_NETLINK_LOG is not set > CONFIG_NF_CONNTRACK=m > CONFIG_NF_CT_ACCT=y > CONFIG_NF_CONNTRACK_MARK=y > CONFIG_NF_CONNTRACK_EVENTS=y > # CONFIG_NF_CT_PROTO_DCCP is not set > # CONFIG_NF_CT_PROTO_SCTP is not set > # CONFIG_NF_CT_PROTO_UDPLITE is not set > # CONFIG_NF_CONNTRACK_AMANDA is not set > # CONFIG_NF_CONNTRACK_FTP is not set > # CONFIG_NF_CONNTRACK_H323 is not set > # CONFIG_NF_CONNTRACK_IRC is not set > # CONFIG_NF_CONNTRACK_NETBIOS_NS is not set > # CONFIG_NF_CONNTRACK_PPTP is not set > # CONFIG_NF_CONNTRACK_SANE is not set > # CONFIG_NF_CONNTRACK_SIP is not set > # CONFIG_NF_CONNTRACK_TFTP is not set > # CONFIG_NF_CT_NETLINK is not set > # CONFIG_NETFILTER_TPROXY is not set > CONFIG_NETFILTER_XTABLES=y > # CONFIG_NETFILTER_XT_TARGET_CLASSIFY is not set > # CONFIG_NETFILTER_XT_TARGET_CONNMARK is not set > # CONFIG_NETFILTER_XT_TARGET_DSCP is not set > # CONFIG_NETFILTER_XT_TARGET_MARK is not set > # CONFIG_NETFILTER_XT_TARGET_NFLOG is not set > # CONFIG_NETFILTER_XT_TARGET_NFQUEUE is not set > # CONFIG_NETFILTER_XT_TARGET_NOTRACK is not set > # CONFIG_NETFILTER_XT_TARGET_RATEEST is not set > # CONFIG_NETFILTER_XT_TARGET_TRACE is not set > # CONFIG_NETFILTER_XT_TARGET_TCPMSS is not set > # CONFIG_NETFILTER_XT_TARGET_TCPOPTSTRIP is not set > # CONFIG_NETFILTER_XT_MATCH_COMMENT is not set > # CONFIG_NETFILTER_XT_MATCH_CONNBYTES is not set > # CONFIG_NETFILTER_XT_MATCH_CONNLIMIT is not set > # CONFIG_NETFILTER_XT_MATCH_CONNMARK is not set > # CONFIG_NETFILTER_XT_MATCH_CONNTRACK is not set > # CONFIG_NETFILTER_XT_MATCH_DCCP is not set > # CONFIG_NETFILTER_XT_MATCH_DSCP is not set > # CONFIG_NETFILTER_XT_MATCH_ESP is not set > # CONFIG_NETFILTER_XT_MATCH_HASHLIMIT is not set > # CONFIG_NETFILTER_XT_MATCH_HELPER is not set > # CONFIG_NETFILTER_XT_MATCH_IPRANGE is not set > # CONFIG_NETFILTER_XT_MATCH_LENGTH is not set > # CONFIG_NETFILTER_XT_MATCH_LIMIT is not set > # CONFIG_NETFILTER_XT_MATCH_MAC is not set > # CONFIG_NETFILTER_XT_MATCH_MARK is not set > # CONFIG_NETFILTER_XT_MATCH_MULTIPORT is not set > # CONFIG_NETFILTER_XT_MATCH_OWNER is not set > # CONFIG_NETFILTER_XT_MATCH_POLICY is not set > # CONFIG_NETFILTER_XT_MATCH_PHYSDEV is not set > # CONFIG_NETFILTER_XT_MATCH_PKTTYPE is not set > # CONFIG_NETFILTER_XT_MATCH_QUOTA is not set > # CONFIG_NETFILTER_XT_MATCH_RATEEST is not set > # CONFIG_NETFILTER_XT_MATCH_REALM is not set > # CONFIG_NETFILTER_XT_MATCH_RECENT is not set > # CONFIG_NETFILTER_XT_MATCH_SCTP is not set > # CONFIG_NETFILTER_XT_MATCH_STATE is not set > # CONFIG_NETFILTER_XT_MATCH_STATISTIC is not set > # CONFIG_NETFILTER_XT_MATCH_STRING is not set > # CONFIG_NETFILTER_XT_MATCH_TCPMSS is not set > # CONFIG_NETFILTER_XT_MATCH_TIME is not set > # CONFIG_NETFILTER_XT_MATCH_U32 is not set > # CONFIG_IP_VS is not set > > # > # IP: Netfilter Configuration > # > CONFIG_NF_DEFRAG_IPV4=m > CONFIG_NF_CONNTRACK_IPV4=m > CONFIG_NF_CONNTRACK_PROC_COMPAT=y > # CONFIG_IP_NF_QUEUE is not set > CONFIG_IP_NF_IPTABLES=m > CONFIG_IP_NF_MATCH_ADDRTYPE=m > CONFIG_IP_NF_MATCH_AH=m > CONFIG_IP_NF_MATCH_ECN=m > CONFIG_IP_NF_MATCH_TTL=m > CONFIG_IP_NF_FILTER=m > CONFIG_IP_NF_TARGET_REJECT=m > CONFIG_IP_NF_TARGET_LOG=m > CONFIG_IP_NF_TARGET_ULOG=m > CONFIG_NF_NAT=m > CONFIG_NF_NAT_NEEDED=y > CONFIG_IP_NF_TARGET_MASQUERADE=m > CONFIG_IP_NF_TARGET_NETMAP=m > CONFIG_IP_NF_TARGET_REDIRECT=m > CONFIG_NF_NAT_SNMP_BASIC=m > # CONFIG_NF_NAT_FTP is not set > # CONFIG_NF_NAT_IRC is not set > # CONFIG_NF_NAT_TFTP is not set > # CONFIG_NF_NAT_AMANDA is not set > # CONFIG_NF_NAT_PPTP is not set > # CONFIG_NF_NAT_H323 is not set > # CONFIG_NF_NAT_SIP is not set > CONFIG_IP_NF_MANGLE=m > # CONFIG_IP_NF_TARGET_CLUSTERIP is not set > CONFIG_IP_NF_TARGET_ECN=m > CONFIG_IP_NF_TARGET_TTL=m > CONFIG_IP_NF_RAW=m > # CONFIG_IP_NF_ARPTABLES is not set > > # > # IPv6: Netfilter Configuration > # > # CONFIG_NF_CONNTRACK_IPV6 is not set > # CONFIG_IP6_NF_QUEUE is not set > # CONFIG_IP6_NF_IPTABLES is not set > # CONFIG_BRIDGE_NF_EBTABLES is not set > # CONFIG_IP_DCCP is not set > # CONFIG_IP_SCTP is not set > # CONFIG_TIPC is not set > # CONFIG_ATM is not set > CONFIG_STP=m > CONFIG_BRIDGE=m > # CONFIG_NET_DSA is not set > CONFIG_VLAN_8021Q=m > # CONFIG_VLAN_8021Q_GVRP is not set > # CONFIG_DECNET is not set > CONFIG_LLC=m > # CONFIG_LLC2 is not set > # CONFIG_IPX is not set > # CONFIG_ATALK is not set > # CONFIG_X25 is not set > # CONFIG_LAPB is not set > # CONFIG_ECONET is not set > # CONFIG_WAN_ROUTER is not set > # CONFIG_NET_SCHED is not set > # CONFIG_DCB is not set > > # > # Network testing > # > # CONFIG_NET_PKTGEN is not set > # CONFIG_HAMRADIO is not set > # CONFIG_CAN is not set > # CONFIG_IRDA is not set > CONFIG_BT=m > CONFIG_BT_L2CAP=m > CONFIG_BT_SCO=m > CONFIG_BT_RFCOMM=m > CONFIG_BT_RFCOMM_TTY=y > CONFIG_BT_BNEP=m > CONFIG_BT_BNEP_MC_FILTER=y > CONFIG_BT_BNEP_PROTO_FILTER=y > CONFIG_BT_HIDP=m > > # > # Bluetooth device drivers > # > # CONFIG_BT_HCIBTUSB is not set > # CONFIG_BT_HCIBTSDIO is not set > CONFIG_BT_HCIUART=m > CONFIG_BT_HCIUART_H4=y > CONFIG_BT_HCIUART_BCSP=y > # CONFIG_BT_HCIUART_LL is not set > CONFIG_BT_HCIBCM203X=m > CONFIG_BT_HCIBPA10X=m > CONFIG_BT_HCIBFUSB=m > CONFIG_BT_HCIVHCI=m > # CONFIG_AF_RXRPC is not set > # CONFIG_PHONET is not set > CONFIG_WIRELESS=y > CONFIG_CFG80211=m > # CONFIG_CFG80211_REG_DEBUG is not set > # CONFIG_NL80211 is not set > CONFIG_WIRELESS_OLD_REGULATORY=y > CONFIG_WIRELESS_EXT=y > CONFIG_WIRELESS_EXT_SYSFS=y > CONFIG_LIB80211=m > # CONFIG_LIB80211_DEBUG is not set > CONFIG_MAC80211=m > > # > # Rate control algorithm selection > # > CONFIG_MAC80211_RC_PID=y > CONFIG_MAC80211_RC_MINSTREL=y > CONFIG_MAC80211_RC_DEFAULT_PID=y > # CONFIG_MAC80211_RC_DEFAULT_MINSTREL is not set > CONFIG_MAC80211_RC_DEFAULT="pid" > # CONFIG_MAC80211_MESH is not set > CONFIG_MAC80211_LEDS=y > # CONFIG_MAC80211_DEBUG_MENU is not set > # CONFIG_WIMAX is not set > CONFIG_RFKILL=y > CONFIG_RFKILL_INPUT=m > CONFIG_RFKILL_LEDS=y > # CONFIG_NET_9P is not set > > # > # Device Drivers > # > > # > # Generic Driver Options > # > CONFIG_UEVENT_HELPER_PATH="/sbin/hotplug" > CONFIG_STANDALONE=y > CONFIG_PREVENT_FIRMWARE_BUILD=y > CONFIG_FW_LOADER=m > CONFIG_FIRMWARE_IN_KERNEL=y > CONFIG_EXTRA_FIRMWARE="" > # CONFIG_SYS_HYPERVISOR is not set > CONFIG_CONNECTOR=y > CONFIG_PROC_EVENTS=y > # CONFIG_MTD is not set > # CONFIG_PARPORT is not set > CONFIG_PNP=y > CONFIG_PNP_DEBUG_MESSAGES=y > > # > # Protocols > # > CONFIG_PNPACPI=y > CONFIG_BLK_DEV=y > # CONFIG_BLK_DEV_FD is not set > # CONFIG_BLK_CPQ_DA is not set > # CONFIG_BLK_CPQ_CISS_DA is not set > # CONFIG_BLK_DEV_DAC960 is not set > # CONFIG_BLK_DEV_UMEM is not set > # CONFIG_BLK_DEV_COW_COMMON is not set > CONFIG_BLK_DEV_LOOP=y > # CONFIG_BLK_DEV_CRYPTOLOOP is not set > CONFIG_BLK_DEV_NBD=m > # CONFIG_BLK_DEV_SX8 is not set > # CONFIG_BLK_DEV_UB is not set > CONFIG_BLK_DEV_RAM=y > CONFIG_BLK_DEV_RAM_COUNT=16 > CONFIG_BLK_DEV_RAM_SIZE=4096 > # CONFIG_BLK_DEV_XIP is not set > # CONFIG_CDROM_PKTCDVD is not set > # CONFIG_ATA_OVER_ETH is not set > # CONFIG_BLK_DEV_HD is not set > CONFIG_MISC_DEVICES=y > # CONFIG_IBM_ASM is not set > # CONFIG_PHANTOM is not set > # CONFIG_SGI_IOC4 is not set > CONFIG_TIFM_CORE=m > CONFIG_TIFM_7XX1=m > # CONFIG_ICS932S401 is not set > # CONFIG_ENCLOSURE_SERVICES is not set > # CONFIG_SGI_XP is not set > # CONFIG_HP_ILO is not set > # CONFIG_SGI_GRU is not set > # CONFIG_C2PORT is not set > > # > # EEPROM support > # > # CONFIG_EEPROM_AT24 is not set > # CONFIG_EEPROM_LEGACY is not set > # CONFIG_EEPROM_93CX6 is not set > CONFIG_HAVE_IDE=y > # CONFIG_IDE is not set > > # > # SCSI device support > # > CONFIG_RAID_ATTRS=m > CONFIG_SCSI=y > CONFIG_SCSI_DMA=y > # CONFIG_SCSI_TGT is not set > CONFIG_SCSI_NETLINK=y > CONFIG_SCSI_PROC_FS=y > > # > # SCSI support type (disk, tape, CD-ROM) > # > CONFIG_BLK_DEV_SD=m > # CONFIG_CHR_DEV_ST is not set > # CONFIG_CHR_DEV_OSST is not set > CONFIG_BLK_DEV_SR=m > CONFIG_BLK_DEV_SR_VENDOR=y > CONFIG_CHR_DEV_SG=m > # CONFIG_CHR_DEV_SCH is not set > > # > # Some SCSI devices (e.g. CD jukebox) support multiple LUNs > # > CONFIG_SCSI_MULTI_LUN=y > # CONFIG_SCSI_CONSTANTS is not set > # CONFIG_SCSI_LOGGING is not set > # CONFIG_SCSI_SCAN_ASYNC is not set > CONFIG_SCSI_WAIT_SCAN=m > > # > # SCSI Transports > # > CONFIG_SCSI_SPI_ATTRS=m > CONFIG_SCSI_FC_ATTRS=m > # CONFIG_SCSI_ISCSI_ATTRS is not set > # CONFIG_SCSI_SAS_LIBSAS is not set > # CONFIG_SCSI_SRP_ATTRS is not set > # CONFIG_SCSI_LOWLEVEL is not set > # CONFIG_SCSI_DH is not set > CONFIG_ATA=y > # CONFIG_ATA_NONSTANDARD is not set > # CONFIG_ATA_ACPI is not set > # CONFIG_SATA_PMP is not set > CONFIG_SATA_AHCI=m > # CONFIG_SATA_SIL24 is not set > CONFIG_ATA_SFF=y > # CONFIG_SATA_SVW is not set > CONFIG_ATA_PIIX=m > # CONFIG_SATA_MV is not set > # CONFIG_SATA_NV is not set > # CONFIG_PDC_ADMA is not set > # CONFIG_SATA_QSTOR is not set > # CONFIG_SATA_PROMISE is not set > # CONFIG_SATA_SX4 is not set > # CONFIG_SATA_SIL is not set > # CONFIG_SATA_SIS is not set > # CONFIG_SATA_ULI is not set > # CONFIG_SATA_VIA is not set > # CONFIG_SATA_VITESSE is not set > # CONFIG_SATA_INIC162X is not set > # CONFIG_PATA_ALI is not set > # CONFIG_PATA_AMD is not set > # CONFIG_PATA_ARTOP is not set > # CONFIG_PATA_ATIIXP is not set > # CONFIG_PATA_CMD640_PCI is not set > # CONFIG_PATA_CMD64X is not set > # CONFIG_PATA_CS5520 is not set > # CONFIG_PATA_CS5530 is not set > # CONFIG_PATA_CYPRESS is not set > # CONFIG_PATA_EFAR is not set > # CONFIG_ATA_GENERIC is not set > # CONFIG_PATA_HPT366 is not set > # CONFIG_PATA_HPT37X is not set > # CONFIG_PATA_HPT3X2N is not set > # CONFIG_PATA_HPT3X3 is not set > # CONFIG_PATA_IT821X is not set > # CONFIG_PATA_IT8213 is not set > # CONFIG_PATA_JMICRON is not set > # CONFIG_PATA_TRIFLEX is not set > # CONFIG_PATA_MARVELL is not set > # CONFIG_PATA_MPIIX is not set > # CONFIG_PATA_OLDPIIX is not set > # CONFIG_PATA_NETCELL is not set > # CONFIG_PATA_NINJA32 is not set > # CONFIG_PATA_NS87410 is not set > # CONFIG_PATA_NS87415 is not set > # CONFIG_PATA_OPTI is not set > # CONFIG_PATA_OPTIDMA is not set > # CONFIG_PATA_PDC_OLD is not set > # CONFIG_PATA_RADISYS is not set > # CONFIG_PATA_RZ1000 is not set > # CONFIG_PATA_SC1200 is not set > # CONFIG_PATA_SERVERWORKS is not set > # CONFIG_PATA_PDC2027X is not set > # CONFIG_PATA_SIL680 is not set > # CONFIG_PATA_SIS is not set > # CONFIG_PATA_VIA is not set > # CONFIG_PATA_WINBOND is not set > # CONFIG_PATA_PLATFORM is not set > # CONFIG_PATA_SCH is not set > CONFIG_MD=y > # CONFIG_BLK_DEV_MD is not set > CONFIG_BLK_DEV_DM=y > # CONFIG_DM_DEBUG is not set > CONFIG_DM_CRYPT=y > CONFIG_DM_SNAPSHOT=m > CONFIG_DM_MIRROR=m > # CONFIG_DM_ZERO is not set > # CONFIG_DM_MULTIPATH is not set > # CONFIG_DM_DELAY is not set > # CONFIG_DM_UEVENT is not set > # CONFIG_BLK_DEV_DM_BBR is not set > # CONFIG_FUSION is not set > > # > # IEEE 1394 (FireWire) support > # > > # > # Enable only one of the two stacks, unless you know what you are doing > # > # CONFIG_FIREWIRE is not set > CONFIG_IEEE1394=m > CONFIG_IEEE1394_OHCI1394=m > # CONFIG_IEEE1394_PCILYNX is not set > # CONFIG_IEEE1394_SBP2 is not set > # CONFIG_IEEE1394_ETH1394_ROM_ENTRY is not set > # CONFIG_IEEE1394_ETH1394 is not set > # CONFIG_IEEE1394_RAWIO is not set > # CONFIG_IEEE1394_VIDEO1394 is not set > # CONFIG_IEEE1394_DV1394 is not set > # CONFIG_IEEE1394_VERBOSEDEBUG is not set > # CONFIG_I2O is not set > # CONFIG_MACINTOSH_DRIVERS is not set > CONFIG_NETDEVICES=y > # CONFIG_DUMMY is not set > # CONFIG_BONDING is not set > # CONFIG_MACVLAN is not set > # CONFIG_EQUALIZER is not set > # CONFIG_TUN is not set > # CONFIG_VETH is not set > # CONFIG_NET_SB1000 is not set > # CONFIG_ARCNET is not set > # CONFIG_NET_ETHERNET is not set > CONFIG_MII=m > CONFIG_NETDEV_1000=y > # CONFIG_ACENIC is not set > # CONFIG_DL2K is not set > # CONFIG_E1000 is not set > # CONFIG_E1000E is not set > # CONFIG_IP1000 is not set > # CONFIG_IGB is not set > # CONFIG_NS83820 is not set > # CONFIG_HAMACHI is not set > # CONFIG_YELLOWFIN is not set > CONFIG_R8169=m > CONFIG_R8169_VLAN=y > # CONFIG_SIS190 is not set > # CONFIG_SKGE is not set > # CONFIG_SKY2 is not set > # CONFIG_VIA_VELOCITY is not set > # CONFIG_TIGON3 is not set > # CONFIG_BNX2 is not set > # CONFIG_QLA3XXX is not set > # CONFIG_ATL1 is not set > # CONFIG_ATL1E is not set > # CONFIG_ATL1C is not set > # CONFIG_JME is not set > # CONFIG_NETDEV_10000 is not set > # CONFIG_TR is not set > > # > # Wireless LAN > # > # CONFIG_WLAN_PRE80211 is not set > CONFIG_WLAN_80211=y > # CONFIG_LIBERTAS is not set > # CONFIG_LIBERTAS_THINFIRM is not set > # CONFIG_AIRO is not set > # CONFIG_HERMES is not set > # CONFIG_ATMEL is not set > # CONFIG_PRISM54 is not set > # CONFIG_USB_ZD1201 is not set > # CONFIG_USB_NET_RNDIS_WLAN is not set > # CONFIG_RTL8180 is not set > # CONFIG_RTL8187 is not set > # CONFIG_ADM8211 is not set > # CONFIG_MAC80211_HWSIM is not set > # CONFIG_P54_COMMON is not set > # CONFIG_ATH5K is not set > # CONFIG_ATH9K is not set > # CONFIG_IPW2100 is not set > # CONFIG_IPW2200 is not set > CONFIG_IWLWIFI=m > CONFIG_IWLCORE=m > CONFIG_IWLWIFI_LEDS=y > CONFIG_IWLWIFI_RFKILL=y > # CONFIG_IWLWIFI_DEBUG is not set > CONFIG_IWLAGN=m > CONFIG_IWLAGN_SPECTRUM_MEASUREMENT=y > CONFIG_IWLAGN_LEDS=y > # CONFIG_IWL4965 is not set > CONFIG_IWL5000=y > # CONFIG_IWL3945 is not set > # CONFIG_HOSTAP is not set > # CONFIG_B43 is not set > # CONFIG_B43LEGACY is not set > # CONFIG_ZD1211RW is not set > # CONFIG_RT2X00 is not set > > # > # Enable WiMAX (Networking options) to see the WiMAX drivers > # > > # > # USB Network Adapters > # > # CONFIG_USB_CATC is not set > # CONFIG_USB_KAWETH is not set > # CONFIG_USB_PEGASUS is not set > # CONFIG_USB_RTL8150 is not set > # CONFIG_USB_USBNET is not set > # CONFIG_USB_HSO is not set > # CONFIG_WAN is not set > # CONFIG_FDDI is not set > # CONFIG_HIPPI is not set > CONFIG_PPP=m > # CONFIG_PPP_MULTILINK is not set > CONFIG_PPP_FILTER=y > CONFIG_PPP_ASYNC=m > # CONFIG_PPP_SYNC_TTY is not set > CONFIG_PPP_DEFLATE=m > CONFIG_PPP_BSDCOMP=m > # CONFIG_PPP_MPPE is not set > CONFIG_PPPOE=m > # CONFIG_PPPOL2TP is not set > CONFIG_SLIP=m > CONFIG_SLIP_COMPRESSED=y > CONFIG_SLHC=m > CONFIG_SLIP_SMART=y > CONFIG_SLIP_MODE_SLIP6=y > # CONFIG_NET_FC is not set > # CONFIG_NETCONSOLE is not set > # CONFIG_NETPOLL is not set > # CONFIG_NET_POLL_CONTROLLER is not set > # CONFIG_ISDN is not set > # CONFIG_PHONE is not set > > # > # Input device support > # > CONFIG_INPUT=y > # CONFIG_INPUT_FF_MEMLESS is not set > CONFIG_INPUT_POLLDEV=m > > # > # Userland interfaces > # > CONFIG_INPUT_MOUSEDEV=y > CONFIG_INPUT_MOUSEDEV_PSAUX=y > CONFIG_INPUT_MOUSEDEV_SCREEN_X=1280 > CONFIG_INPUT_MOUSEDEV_SCREEN_Y=800 > # CONFIG_INPUT_JOYDEV is not set > CONFIG_INPUT_EVDEV=m > # CONFIG_INPUT_EVBUG is not set > > # > # Input Device Drivers > # > CONFIG_INPUT_KEYBOARD=y > CONFIG_KEYBOARD_ATKBD=y > CONFIG_KEYBOARD_SUNKBD=m > CONFIG_KEYBOARD_LKKBD=m > CONFIG_KEYBOARD_XTKBD=m > CONFIG_KEYBOARD_NEWTON=m > # CONFIG_KEYBOARD_STOWAWAY is not set > CONFIG_INPUT_MOUSE=y > CONFIG_MOUSE_PS2=m > # CONFIG_MOUSE_PS2_ALPS is not set > CONFIG_MOUSE_PS2_LOGIPS2PP=y > CONFIG_MOUSE_PS2_SYNAPTICS=y > # CONFIG_MOUSE_PS2_LIFEBOOK is not set > # CONFIG_MOUSE_PS2_TRACKPOINT is not set > # CONFIG_MOUSE_PS2_ELANTECH is not set > # CONFIG_MOUSE_PS2_TOUCHKIT is not set > # CONFIG_MOUSE_SERIAL is not set > # CONFIG_MOUSE_APPLETOUCH is not set > # CONFIG_MOUSE_BCM5974 is not set > # CONFIG_MOUSE_VSXXXAA is not set > # CONFIG_INPUT_JOYSTICK is not set > # CONFIG_INPUT_TABLET is not set > # CONFIG_INPUT_TOUCHSCREEN is not set > CONFIG_INPUT_MISC=y > CONFIG_INPUT_PCSPKR=m > # CONFIG_INPUT_APANEL is not set > # CONFIG_INPUT_ATLAS_BTNS is not set > # CONFIG_INPUT_ATI_REMOTE is not set > # CONFIG_INPUT_ATI_REMOTE2 is not set > # CONFIG_INPUT_KEYSPAN_REMOTE is not set > # CONFIG_INPUT_POWERMATE is not set > # CONFIG_INPUT_YEALINK is not set > # CONFIG_INPUT_CM109 is not set > CONFIG_INPUT_UINPUT=m > > # > # Hardware I/O ports > # > CONFIG_SERIO=y > CONFIG_SERIO_I8042=y > CONFIG_SERIO_SERPORT=m > CONFIG_SERIO_CT82C710=m > # CONFIG_SERIO_PCIPS2 is not set > CONFIG_SERIO_LIBPS2=y > # CONFIG_SERIO_RAW is not set > # CONFIG_GAMEPORT is not set > > # > # Character devices > # > CONFIG_VT=y > CONFIG_CONSOLE_TRANSLATIONS=y > CONFIG_VT_CONSOLE=y > CONFIG_HW_CONSOLE=y > # CONFIG_VT_HW_CONSOLE_BINDING is not set > CONFIG_DEVKMEM=y > # CONFIG_SERIAL_NONSTANDARD is not set > # CONFIG_NOZOMI is not set > > # > # Serial drivers > # > CONFIG_SERIAL_8250=y > CONFIG_SERIAL_8250_CONSOLE=y > CONFIG_FIX_EARLYCON_MEM=y > CONFIG_SERIAL_8250_PCI=y > CONFIG_SERIAL_8250_PNP=y > CONFIG_SERIAL_8250_NR_UARTS=4 > CONFIG_SERIAL_8250_RUNTIME_UARTS=4 > CONFIG_SERIAL_8250_EXTENDED=y > CONFIG_SERIAL_8250_MANY_PORTS=y > CONFIG_SERIAL_8250_SHARE_IRQ=y > # CONFIG_SERIAL_8250_DETECT_IRQ is not set > CONFIG_SERIAL_8250_RSA=y > > # > # Non-8250 serial port support > # > CONFIG_SERIAL_CORE=y > CONFIG_SERIAL_CORE_CONSOLE=y > # CONFIG_SERIAL_JSM is not set > CONFIG_UNIX98_PTYS=y > # CONFIG_DEVPTS_MULTIPLE_INSTANCES is not set > # CONFIG_LEGACY_PTYS is not set > # CONFIG_IPMI_HANDLER is not set > CONFIG_HW_RANDOM=m > CONFIG_HW_RANDOM_INTEL=m > # CONFIG_HW_RANDOM_AMD is not set > CONFIG_NVRAM=m > # CONFIG_R3964 is not set > # CONFIG_APPLICOM is not set > # CONFIG_MWAVE is not set > # CONFIG_PC8736x_GPIO is not set > # CONFIG_RAW_DRIVER is not set > # CONFIG_HPET is not set > # CONFIG_HANGCHECK_TIMER is not set > # CONFIG_TCG_TPM is not set > # CONFIG_TELCLOCK is not set > CONFIG_DEVPORT=y > CONFIG_I2C=y > CONFIG_I2C_BOARDINFO=y > CONFIG_I2C_CHARDEV=m > CONFIG_I2C_HELPER_AUTO=y > CONFIG_I2C_ALGOBIT=m > > # > # I2C Hardware Bus support > # > > # > # PC SMBus host controller drivers > # > # CONFIG_I2C_ALI1535 is not set > # CONFIG_I2C_ALI1563 is not set > # CONFIG_I2C_ALI15X3 is not set > # CONFIG_I2C_AMD756 is not set > # CONFIG_I2C_AMD8111 is not set > CONFIG_I2C_I801=m > # CONFIG_I2C_ISCH is not set > # CONFIG_I2C_PIIX4 is not set > # CONFIG_I2C_NFORCE2 is not set > # CONFIG_I2C_SIS5595 is not set > # CONFIG_I2C_SIS630 is not set > # CONFIG_I2C_SIS96X is not set > # CONFIG_I2C_VIA is not set > # CONFIG_I2C_VIAPRO is not set > > # > # I2C system bus drivers (mostly embedded / system-on-chip) > # > # CONFIG_I2C_OCORES is not set > # CONFIG_I2C_SIMTEC is not set > > # > # External I2C/SMBus adapter drivers > # > # CONFIG_I2C_PARPORT_LIGHT is not set > # CONFIG_I2C_TAOS_EVM is not set > # CONFIG_I2C_TINY_USB is not set > > # > # Graphics adapter I2C/DDC channel drivers > # > # CONFIG_I2C_VOODOO3 is not set > > # > # Other I2C/SMBus bus drivers > # > # CONFIG_I2C_PCA_PLATFORM is not set > # CONFIG_I2C_STUB is not set > > # > # Miscellaneous I2C Chip support > # > # CONFIG_DS1682 is not set > # CONFIG_SENSORS_PCF8574 is not set > # CONFIG_PCF8575 is not set > # CONFIG_SENSORS_PCA9539 is not set > # CONFIG_SENSORS_PCF8591 is not set > # CONFIG_SENSORS_MAX6875 is not set > # CONFIG_SENSORS_TSL2550 is not set > # CONFIG_I2C_DEBUG_CORE is not set > # CONFIG_I2C_DEBUG_ALGO is not set > # CONFIG_I2C_DEBUG_BUS is not set > # CONFIG_I2C_DEBUG_CHIP is not set > # CONFIG_SPI is not set > CONFIG_ARCH_WANT_OPTIONAL_GPIOLIB=y > # CONFIG_GPIOLIB is not set > # CONFIG_W1 is not set > CONFIG_POWER_SUPPLY=y > # CONFIG_POWER_SUPPLY_DEBUG is not set > # CONFIG_PDA_POWER is not set > # CONFIG_BATTERY_DS2760 is not set > # CONFIG_BATTERY_BQ27x00 is not set > CONFIG_HWMON=m > # CONFIG_HWMON_VID is not set > # CONFIG_SENSORS_ABITUGURU is not set > # CONFIG_SENSORS_ABITUGURU3 is not set > # CONFIG_SENSORS_AD7414 is not set > # CONFIG_SENSORS_AD7418 is not set > # CONFIG_SENSORS_ADM1021 is not set > # CONFIG_SENSORS_ADM1025 is not set > # CONFIG_SENSORS_ADM1026 is not set > # CONFIG_SENSORS_ADM1029 is not set > # CONFIG_SENSORS_ADM1031 is not set > # CONFIG_SENSORS_ADM9240 is not set > # CONFIG_SENSORS_ADT7462 is not set > # CONFIG_SENSORS_ADT7470 is not set > # CONFIG_SENSORS_ADT7473 is not set > # CONFIG_SENSORS_ADT7475 is not set > # CONFIG_SENSORS_K8TEMP is not set > # CONFIG_SENSORS_ASB100 is not set > # CONFIG_SENSORS_ATXP1 is not set > # CONFIG_SENSORS_DS1621 is not set > # CONFIG_SENSORS_I5K_AMB is not set > # CONFIG_SENSORS_F71805F is not set > # CONFIG_SENSORS_F71882FG is not set > # CONFIG_SENSORS_F75375S is not set > # CONFIG_SENSORS_FSCHER is not set > # CONFIG_SENSORS_FSCPOS is not set > # CONFIG_SENSORS_FSCHMD is not set > # CONFIG_SENSORS_GL518SM is not set > # CONFIG_SENSORS_GL520SM is not set > CONFIG_SENSORS_CORETEMP=m > # CONFIG_SENSORS_IT87 is not set > # CONFIG_SENSORS_LM63 is not set > # CONFIG_SENSORS_LM75 is not set > # CONFIG_SENSORS_LM77 is not set > # CONFIG_SENSORS_LM78 is not set > # CONFIG_SENSORS_LM80 is not set > # CONFIG_SENSORS_LM83 is not set > # CONFIG_SENSORS_LM85 is not set > # CONFIG_SENSORS_LM87 is not set > # CONFIG_SENSORS_LM90 is not set > # CONFIG_SENSORS_LM92 is not set > # CONFIG_SENSORS_LM93 is not set > # CONFIG_SENSORS_LTC4245 is not set > # CONFIG_SENSORS_MAX1619 is not set > # CONFIG_SENSORS_MAX6650 is not set > # CONFIG_SENSORS_PC87360 is not set > # CONFIG_SENSORS_PC87427 is not set > # CONFIG_SENSORS_SIS5595 is not set > # CONFIG_SENSORS_DME1737 is not set > # CONFIG_SENSORS_SMSC47M1 is not set > # CONFIG_SENSORS_SMSC47M192 is not set > # CONFIG_SENSORS_SMSC47B397 is not set > # CONFIG_SENSORS_ADS7828 is not set > # CONFIG_SENSORS_THMC50 is not set > # CONFIG_SENSORS_VIA686A is not set > # CONFIG_SENSORS_VT1211 is not set > # CONFIG_SENSORS_VT8231 is not set > # CONFIG_SENSORS_W83781D is not set > # CONFIG_SENSORS_W83791D is not set > # CONFIG_SENSORS_W83792D is not set > # CONFIG_SENSORS_W83793 is not set > # CONFIG_SENSORS_W83L785TS is not set > # CONFIG_SENSORS_W83L786NG is not set > # CONFIG_SENSORS_W83627HF is not set > # CONFIG_SENSORS_W83627EHF is not set > # CONFIG_SENSORS_HDAPS is not set > # CONFIG_SENSORS_LIS3LV02D is not set > # CONFIG_SENSORS_APPLESMC is not set > # CONFIG_HWMON_DEBUG_CHIP is not set > CONFIG_THERMAL=y > # CONFIG_WATCHDOG is not set > CONFIG_SSB_POSSIBLE=y > > # > # Sonics Silicon Backplane > # > # CONFIG_SSB is not set > > # > # Multifunction device drivers > # > # CONFIG_MFD_CORE is not set > # CONFIG_MFD_SM501 is not set > # CONFIG_HTC_PASIC3 is not set > # CONFIG_TWL4030_CORE is not set > # CONFIG_MFD_TMIO is not set > # CONFIG_PMIC_DA903X is not set > # CONFIG_MFD_WM8400 is not set > # CONFIG_MFD_WM8350_I2C is not set > # CONFIG_MFD_PCF50633 is not set > # CONFIG_REGULATOR is not set > > # > # Multimedia devices > # > > # > # Multimedia core support > # > CONFIG_VIDEO_DEV=m > CONFIG_VIDEO_V4L2_COMMON=m > # CONFIG_VIDEO_ALLOW_V4L1 is not set > CONFIG_VIDEO_V4L1_COMPAT=y > # CONFIG_DVB_CORE is not set > CONFIG_VIDEO_MEDIA=m > > # > # Multimedia drivers > # > # CONFIG_MEDIA_ATTACH is not set > CONFIG_MEDIA_TUNER=m > # CONFIG_MEDIA_TUNER_CUSTOMIZE is not set > CONFIG_MEDIA_TUNER_SIMPLE=m > CONFIG_MEDIA_TUNER_TDA8290=m > CONFIG_MEDIA_TUNER_TDA9887=m > CONFIG_MEDIA_TUNER_TEA5761=m > CONFIG_MEDIA_TUNER_TEA5767=m > CONFIG_MEDIA_TUNER_MT20XX=m > CONFIG_MEDIA_TUNER_XC2028=m > CONFIG_MEDIA_TUNER_XC5000=m > CONFIG_VIDEO_V4L2=m > CONFIG_VIDEO_CAPTURE_DRIVERS=y > # CONFIG_VIDEO_ADV_DEBUG is not set > # CONFIG_VIDEO_FIXED_MINOR_RANGES is not set > CONFIG_VIDEO_HELPER_CHIPS_AUTO=y > # CONFIG_VIDEO_VIVI is not set > # CONFIG_VIDEO_BT848 is not set > # CONFIG_VIDEO_SAA5246A is not set > # CONFIG_VIDEO_SAA5249 is not set > # CONFIG_VIDEO_SAA7134 is not set > # CONFIG_VIDEO_HEXIUM_ORION is not set > # CONFIG_VIDEO_HEXIUM_GEMINI is not set > # CONFIG_VIDEO_CX88 is not set > # CONFIG_VIDEO_IVTV is not set > # CONFIG_VIDEO_CAFE_CCIC is not set > # CONFIG_SOC_CAMERA is not set > CONFIG_V4L_USB_DRIVERS=y > CONFIG_USB_VIDEO_CLASS=m > CONFIG_USB_VIDEO_CLASS_INPUT_EVDEV=y > # CONFIG_USB_GSPCA is not set > # CONFIG_VIDEO_PVRUSB2 is not set > # CONFIG_VIDEO_EM28XX is not set > # CONFIG_VIDEO_USBVISION is not set > # CONFIG_USB_ET61X251 is not set > # CONFIG_USB_SN9C102 is not set > # CONFIG_USB_ZC0301 is not set > # CONFIG_USB_ZR364XX is not set > # CONFIG_USB_STKWEBCAM is not set > # CONFIG_USB_S2255 is not set > # CONFIG_RADIO_ADAPTERS is not set > # CONFIG_DAB is not set > > # > # Graphics support > # > CONFIG_AGP=y > CONFIG_AGP_AMD64=y > CONFIG_AGP_INTEL=m > # CONFIG_AGP_SIS is not set > # CONFIG_AGP_VIA is not set > CONFIG_DRM=m > # CONFIG_DRM_TDFX is not set > # CONFIG_DRM_R128 is not set > # CONFIG_DRM_RADEON is not set > # CONFIG_DRM_I810 is not set > # CONFIG_DRM_I830 is not set > CONFIG_DRM_I915=m > # CONFIG_DRM_I915_KMS is not set > # CONFIG_DRM_MGA is not set > # CONFIG_DRM_SIS is not set > # CONFIG_DRM_VIA is not set > # CONFIG_DRM_SAVAGE is not set > # CONFIG_VGASTATE is not set > CONFIG_VIDEO_OUTPUT_CONTROL=m > CONFIG_FB=y > CONFIG_FIRMWARE_EDID=y > # CONFIG_FB_DDC is not set > # CONFIG_FB_BOOT_VESA_SUPPORT is not set > CONFIG_FB_CFB_FILLRECT=y > CONFIG_FB_CFB_COPYAREA=y > CONFIG_FB_CFB_IMAGEBLIT=y > # CONFIG_FB_CFB_REV_PIXELS_IN_BYTE is not set > # CONFIG_FB_SYS_FILLRECT is not set > # CONFIG_FB_SYS_COPYAREA is not set > # CONFIG_FB_SYS_IMAGEBLIT is not set > # CONFIG_FB_FOREIGN_ENDIAN is not set > # CONFIG_FB_SYS_FOPS is not set > # CONFIG_FB_SVGALIB is not set > # CONFIG_FB_MACMODES is not set > # CONFIG_FB_BACKLIGHT is not set > CONFIG_FB_MODE_HELPERS=y > # CONFIG_FB_TILEBLITTING is not set > > # > # Frame buffer hardware drivers > # > # CONFIG_FB_CIRRUS is not set > # CONFIG_FB_PM2 is not set > # CONFIG_FB_CYBER2000 is not set > # CONFIG_FB_ARC is not set > # CONFIG_FB_ASILIANT is not set > # CONFIG_FB_IMSTT is not set > # CONFIG_FB_VGA16 is not set > CONFIG_FB_UVESA=y > # CONFIG_FB_VESA is not set > # CONFIG_FB_N411 is not set > # CONFIG_FB_HGA is not set > # CONFIG_FB_S1D13XXX is not set > # CONFIG_FB_NVIDIA is not set > # CONFIG_FB_RIVA is not set > # CONFIG_FB_LE80578 is not set > # CONFIG_FB_INTEL is not set > # CONFIG_FB_MATROX is not set > # CONFIG_FB_RADEON is not set > # CONFIG_FB_ATY128 is not set > # CONFIG_FB_ATY is not set > # CONFIG_FB_S3 is not set > # CONFIG_FB_SAVAGE is not set > # CONFIG_FB_SIS is not set > # CONFIG_FB_VIA is not set > # CONFIG_FB_NEOMAGIC is not set > # CONFIG_FB_KYRO is not set > # CONFIG_FB_3DFX is not set > # CONFIG_FB_VOODOO1 is not set > # CONFIG_FB_VT8623 is not set > # CONFIG_FB_TRIDENT is not set > # CONFIG_FB_ARK is not set > # CONFIG_FB_PM3 is not set > # CONFIG_FB_CARMINE is not set > # CONFIG_FB_GEODE is not set > # CONFIG_FB_VIRTUAL is not set > # CONFIG_FB_METRONOME is not set > # CONFIG_FB_MB862XX is not set > CONFIG_BACKLIGHT_LCD_SUPPORT=y > CONFIG_LCD_CLASS_DEVICE=y > # CONFIG_LCD_ILI9320 is not set > # CONFIG_LCD_PLATFORM is not set > CONFIG_BACKLIGHT_CLASS_DEVICE=y > CONFIG_BACKLIGHT_GENERIC=y > # CONFIG_BACKLIGHT_PROGEAR is not set > # CONFIG_BACKLIGHT_MBP_NVIDIA is not set > # CONFIG_BACKLIGHT_SAHARA is not set > > # > # Display device support > # > CONFIG_DISPLAY_SUPPORT=m > > # > # Display hardware drivers > # > > # > # Console display driver support > # > CONFIG_VGA_CONSOLE=y > # CONFIG_VGACON_SOFT_SCROLLBACK is not set > CONFIG_DUMMY_CONSOLE=y > CONFIG_FRAMEBUFFER_CONSOLE=y > # CONFIG_FRAMEBUFFER_CONSOLE_DETECT_PRIMARY is not set > # CONFIG_FRAMEBUFFER_CONSOLE_ROTATION is not set > CONFIG_FB_CON_DECOR=y > # CONFIG_FONTS is not set > CONFIG_FONT_8x8=y > CONFIG_FONT_8x16=y > # CONFIG_LOGO is not set > CONFIG_SOUND=y > # CONFIG_SOUND_OSS_CORE is not set > # CONFIG_SND is not set > # CONFIG_SOUND_PRIME is not set > CONFIG_HID_SUPPORT=y > CONFIG_HID=y > # CONFIG_HID_DEBUG is not set > # CONFIG_HIDRAW is not set > > # > # USB Input Devices > # > CONFIG_USB_HID=m > # CONFIG_HID_PID is not set > CONFIG_USB_HIDDEV=y > > # > # USB HID Boot Protocol drivers > # > # CONFIG_USB_KBD is not set > # CONFIG_USB_MOUSE is not set > > # > # Special HID drivers > # > CONFIG_HID_COMPAT=y > # CONFIG_HID_A4TECH is not set > # CONFIG_HID_APPLE is not set > # CONFIG_HID_BELKIN is not set > # CONFIG_HID_CHERRY is not set > # CONFIG_HID_CHICONY is not set > # CONFIG_HID_CYPRESS is not set > # CONFIG_HID_EZKEY is not set > # CONFIG_HID_GYRATION is not set > # CONFIG_HID_LOGITECH is not set > # CONFIG_HID_MICROSOFT is not set > # CONFIG_HID_MONTEREY is not set > # CONFIG_HID_NTRIG is not set > # CONFIG_HID_PANTHERLORD is not set > # CONFIG_HID_PETALYNX is not set > # CONFIG_HID_SAMSUNG is not set > # CONFIG_HID_SONY is not set > # CONFIG_HID_SUNPLUS is not set > # CONFIG_GREENASIA_FF is not set > # CONFIG_HID_TOPSEED is not set > # CONFIG_THRUSTMASTER_FF is not set > # CONFIG_ZEROPLUS_FF is not set > CONFIG_USB_SUPPORT=y > CONFIG_USB_ARCH_HAS_HCD=y > CONFIG_USB_ARCH_HAS_OHCI=y > CONFIG_USB_ARCH_HAS_EHCI=y > CONFIG_USB=m > # CONFIG_USB_DEBUG is not set > # CONFIG_USB_ANNOUNCE_NEW_DEVICES is not set > > # > # Miscellaneous USB options > # > CONFIG_USB_DEVICEFS=y > CONFIG_USB_DEVICE_CLASS=y > # CONFIG_USB_DYNAMIC_MINORS is not set > # CONFIG_USB_SUSPEND is not set > # CONFIG_USB_OTG is not set > # CONFIG_USB_OTG_WHITELIST is not set > # CONFIG_USB_OTG_BLACKLIST_HUB is not set > # CONFIG_USB_MON is not set > # CONFIG_USB_WUSB is not set > # CONFIG_USB_WUSB_CBAF is not set > > # > # USB Host Controller Drivers > # > # CONFIG_USB_C67X00_HCD is not set > CONFIG_USB_EHCI_HCD=m > # CONFIG_USB_EHCI_ROOT_HUB_TT is not set > # CONFIG_USB_EHCI_TT_NEWSCHED is not set > # CONFIG_USB_OXU210HP_HCD is not set > CONFIG_USB_ISP116X_HCD=m > # CONFIG_USB_ISP1760_HCD is not set > CONFIG_USB_OHCI_HCD=m > # CONFIG_USB_OHCI_BIG_ENDIAN_DESC is not set > # CONFIG_USB_OHCI_BIG_ENDIAN_MMIO is not set > CONFIG_USB_OHCI_LITTLE_ENDIAN=y > CONFIG_USB_UHCI_HCD=m > CONFIG_USB_SL811_HCD=m > # CONFIG_USB_R8A66597_HCD is not set > # CONFIG_USB_WHCI_HCD is not set > # CONFIG_USB_HWA_HCD is not set > > # > # Enable Host or Gadget support to see Inventra options > # > > # > # USB Device Class drivers > # > # CONFIG_USB_ACM is not set > # CONFIG_USB_PRINTER is not set > # CONFIG_USB_WDM is not set > # CONFIG_USB_TMC is not set > > # > # NOTE: USB_STORAGE depends on SCSI but BLK_DEV_SD may also be needed; > # > > # > # see USB_STORAGE Help for more information > # > CONFIG_USB_STORAGE=m > # CONFIG_USB_STORAGE_DEBUG is not set > # CONFIG_USB_STORAGE_DATAFAB is not set > CONFIG_USB_STORAGE_FREECOM=y > # CONFIG_USB_STORAGE_ISD200 is not set > # CONFIG_USB_STORAGE_USBAT is not set > # CONFIG_USB_STORAGE_SDDR09 is not set > # CONFIG_USB_STORAGE_SDDR55 is not set > # CONFIG_USB_STORAGE_JUMPSHOT is not set > # CONFIG_USB_STORAGE_ALAUDA is not set > # CONFIG_USB_STORAGE_ONETOUCH is not set > # CONFIG_USB_STORAGE_KARMA is not set > # CONFIG_USB_STORAGE_CYPRESS_ATACB is not set > # CONFIG_USB_LIBUSUAL is not set > > # > # USB Imaging devices > # > # CONFIG_USB_MDC800 is not set > # CONFIG_USB_MICROTEK is not set > > # > # USB port drivers > # > CONFIG_USB_SERIAL=m > CONFIG_USB_EZUSB=y > CONFIG_USB_SERIAL_GENERIC=y > # CONFIG_USB_SERIAL_AIRCABLE is not set > # CONFIG_USB_SERIAL_ARK3116 is not set > # CONFIG_USB_SERIAL_BELKIN is not set > # CONFIG_USB_SERIAL_CH341 is not set > CONFIG_USB_SERIAL_WHITEHEAT=m > # CONFIG_USB_SERIAL_DIGI_ACCELEPORT is not set > # CONFIG_USB_SERIAL_CP2101 is not set > # CONFIG_USB_SERIAL_CYPRESS_M8 is not set > # CONFIG_USB_SERIAL_EMPEG is not set > # CONFIG_USB_SERIAL_FTDI_SIO is not set > CONFIG_USB_SERIAL_FUNSOFT=m > # CONFIG_USB_SERIAL_VISOR is not set > # CONFIG_USB_SERIAL_IPAQ is not set > # CONFIG_USB_SERIAL_IR is not set > # CONFIG_USB_SERIAL_EDGEPORT is not set > # CONFIG_USB_SERIAL_EDGEPORT_TI is not set > # CONFIG_USB_SERIAL_GARMIN is not set > # CONFIG_USB_SERIAL_IPW is not set > # CONFIG_USB_SERIAL_IUU is not set > # CONFIG_USB_SERIAL_KEYSPAN_PDA is not set > # CONFIG_USB_SERIAL_KEYSPAN is not set > # CONFIG_USB_SERIAL_KLSI is not set > # CONFIG_USB_SERIAL_KOBIL_SCT is not set > # CONFIG_USB_SERIAL_MCT_U232 is not set > # CONFIG_USB_SERIAL_MOS7720 is not set > # CONFIG_USB_SERIAL_MOS7840 is not set > # CONFIG_USB_SERIAL_MOTOROLA is not set > CONFIG_USB_SERIAL_NAVMAN=m > # CONFIG_USB_SERIAL_PL2303 is not set > # CONFIG_USB_SERIAL_OTI6858 is not set > # CONFIG_USB_SERIAL_SPCP8X5 is not set > # CONFIG_USB_SERIAL_HP4X is not set > # CONFIG_USB_SERIAL_SAFE is not set > # CONFIG_USB_SERIAL_SIEMENS_MPI is not set > # CONFIG_USB_SERIAL_SIERRAWIRELESS is not set > # CONFIG_USB_SERIAL_TI is not set > # CONFIG_USB_SERIAL_CYBERJACK is not set > # CONFIG_USB_SERIAL_XIRCOM is not set > CONFIG_USB_SERIAL_OPTION=m > # CONFIG_USB_SERIAL_OMNINET is not set > # CONFIG_USB_SERIAL_OPTICON is not set > # CONFIG_USB_SERIAL_DEBUG is not set > > # > # USB Miscellaneous drivers > # > # CONFIG_USB_EMI62 is not set > # CONFIG_USB_EMI26 is not set > # CONFIG_USB_ADUTUX is not set > # CONFIG_USB_SEVSEG is not set > # CONFIG_USB_RIO500 is not set > # CONFIG_USB_LEGOTOWER is not set > # CONFIG_USB_LCD is not set > CONFIG_USB_BERRY_CHARGE=m > # CONFIG_USB_LED is not set > # CONFIG_USB_CYPRESS_CY7C63 is not set > # CONFIG_USB_CYTHERM is not set > # CONFIG_USB_PHIDGET is not set > # CONFIG_USB_IDMOUSE is not set > # CONFIG_USB_FTDI_ELAN is not set > # CONFIG_USB_APPLEDISPLAY is not set > # CONFIG_USB_SISUSBVGA is not set > # CONFIG_USB_LD is not set > # CONFIG_USB_TRANCEVIBRATOR is not set > # CONFIG_USB_IOWARRIOR is not set > # CONFIG_USB_TEST is not set > # CONFIG_USB_ISIGHTFW is not set > # CONFIG_USB_VST is not set > # CONFIG_USB_GADGET is not set > > # > # OTG and related infrastructure > # > # CONFIG_UWB is not set > CONFIG_MMC=m > # CONFIG_MMC_DEBUG is not set > # CONFIG_MMC_UNSAFE_RESUME is not set > > # > # MMC/SD/SDIO Card Drivers > # > CONFIG_MMC_BLOCK=m > CONFIG_MMC_BLOCK_BOUNCE=y > # CONFIG_SDIO_UART is not set > # CONFIG_MMC_TEST is not set > > # > # MMC/SD/SDIO Host Controller Drivers > # > CONFIG_MMC_SDHCI=m > # CONFIG_MMC_SDHCI_PCI is not set > # CONFIG_MMC_WBSD is not set > CONFIG_MMC_TIFM_SD=m > # CONFIG_MEMSTICK is not set > CONFIG_NEW_LEDS=y > CONFIG_LEDS_CLASS=y > > # > # LED drivers > # > # CONFIG_LEDS_ALIX2 is not set > # CONFIG_LEDS_PCA9532 is not set > # CONFIG_LEDS_CLEVO_MAIL is not set > # CONFIG_LEDS_PCA955X is not set > > # > # LED Triggers > # > CONFIG_LEDS_TRIGGERS=y > CONFIG_LEDS_TRIGGER_TIMER=y > CONFIG_LEDS_TRIGGER_HEARTBEAT=y > # CONFIG_LEDS_TRIGGER_BACKLIGHT is not set > # CONFIG_LEDS_TRIGGER_DEFAULT_ON is not set > # CONFIG_ACCESSIBILITY is not set > # CONFIG_INFINIBAND is not set > # CONFIG_EDAC is not set > CONFIG_RTC_LIB=m > CONFIG_RTC_CLASS=m > > # > # RTC interfaces > # > CONFIG_RTC_INTF_SYSFS=y > CONFIG_RTC_INTF_PROC=y > # CONFIG_RTC_INTF_DEV is not set > # CONFIG_RTC_DRV_TEST is not set > > # > # I2C RTC drivers > # > # CONFIG_RTC_DRV_DS1307 is not set > # CONFIG_RTC_DRV_DS1374 is not set > # CONFIG_RTC_DRV_DS1672 is not set > # CONFIG_RTC_DRV_MAX6900 is not set > # CONFIG_RTC_DRV_RS5C372 is not set > # CONFIG_RTC_DRV_ISL1208 is not set > # CONFIG_RTC_DRV_X1205 is not set > # CONFIG_RTC_DRV_PCF8563 is not set > # CONFIG_RTC_DRV_PCF8583 is not set > # CONFIG_RTC_DRV_M41T80 is not set > # CONFIG_RTC_DRV_S35390A is not set > # CONFIG_RTC_DRV_FM3130 is not set > # CONFIG_RTC_DRV_RX8581 is not set > > # > # SPI RTC drivers > # > > # > # Platform RTC drivers > # > CONFIG_RTC_DRV_CMOS=m > # CONFIG_RTC_DRV_DS1286 is not set > # CONFIG_RTC_DRV_DS1511 is not set > # CONFIG_RTC_DRV_DS1553 is not set > # CONFIG_RTC_DRV_DS1742 is not set > # CONFIG_RTC_DRV_STK17TA8 is not set > # CONFIG_RTC_DRV_M48T86 is not set > # CONFIG_RTC_DRV_M48T35 is not set > # CONFIG_RTC_DRV_M48T59 is not set > # CONFIG_RTC_DRV_BQ4802 is not set > # CONFIG_RTC_DRV_V3020 is not set > > # > # on-CPU RTC drivers > # > # CONFIG_DMADEVICES is not set > # CONFIG_UIO is not set > # CONFIG_STAGING is not set > # CONFIG_X86_PLATFORM_DEVICES is not set > > # > # Firmware Drivers > # > # CONFIG_EDD is not set > CONFIG_FIRMWARE_MEMMAP=y > # CONFIG_DELL_RBU is not set > # CONFIG_DCDBAS is not set > CONFIG_DMIID=y > # CONFIG_ISCSI_IBFT_FIND is not set > > # > # File systems > # > CONFIG_EXT2_FS=m > CONFIG_EXT2_FS_XATTR=y > CONFIG_EXT2_FS_POSIX_ACL=y > CONFIG_EXT2_FS_SECURITY=y > CONFIG_EXT2_FS_XIP=y > CONFIG_EXT3_FS=m > CONFIG_EXT3_FS_XATTR=y > CONFIG_EXT3_FS_POSIX_ACL=y > CONFIG_EXT3_FS_SECURITY=y > CONFIG_EXT4_FS=y > CONFIG_EXT4DEV_COMPAT=y > CONFIG_EXT4_FS_XATTR=y > CONFIG_EXT4_FS_POSIX_ACL=y > CONFIG_EXT4_FS_SECURITY=y > CONFIG_FS_XIP=y > CONFIG_JBD=m > CONFIG_JBD2=y > CONFIG_FS_MBCACHE=y > CONFIG_REISERFS_FS=m > # CONFIG_REISERFS_CHECK is not set > # CONFIG_REISERFS_PROC_INFO is not set > CONFIG_REISERFS_FS_XATTR=y > CONFIG_REISERFS_FS_POSIX_ACL=y > CONFIG_REISERFS_FS_SECURITY=y > CONFIG_JFS_FS=m > CONFIG_JFS_POSIX_ACL=y > CONFIG_JFS_SECURITY=y > # CONFIG_JFS_DEBUG is not set > # CONFIG_JFS_STATISTICS is not set > CONFIG_FS_POSIX_ACL=y > CONFIG_FILE_LOCKING=y > CONFIG_XFS_FS=m > CONFIG_XFS_QUOTA=y > CONFIG_XFS_POSIX_ACL=y > CONFIG_XFS_RT=y > # CONFIG_XFS_DEBUG is not set > # CONFIG_GFS2_FS is not set > # CONFIG_OCFS2_FS is not set > # CONFIG_BTRFS_FS is not set > CONFIG_DNOTIFY=y > CONFIG_INOTIFY=y > CONFIG_INOTIFY_USER=y > CONFIG_QUOTA=y > CONFIG_QUOTA_NETLINK_INTERFACE=y > # CONFIG_PRINT_QUOTA_WARNING is not set > CONFIG_QUOTA_TREE=y > # CONFIG_QFMT_V1 is not set > CONFIG_QFMT_V2=y > CONFIG_QUOTACTL=y > # CONFIG_AUTOFS_FS is not set > # CONFIG_AUTOFS4_FS is not set > CONFIG_FUSE_FS=m > > # > # CD-ROM/DVD Filesystems > # > CONFIG_ISO9660_FS=m > CONFIG_JOLIET=y > CONFIG_ZISOFS=y > CONFIG_UDF_FS=m > CONFIG_UDF_NLS=y > > # > # DOS/FAT/NT Filesystems > # > CONFIG_FAT_FS=m > CONFIG_MSDOS_FS=m > CONFIG_VFAT_FS=m > CONFIG_FAT_DEFAULT_CODEPAGE=437 > CONFIG_FAT_DEFAULT_IOCHARSET="iso8859-1" > CONFIG_NTFS_FS=m > # CONFIG_NTFS_DEBUG is not set > # CONFIG_NTFS_RW is not set > > # > # Pseudo filesystems > # > CONFIG_PROC_FS=y > CONFIG_PROC_KCORE=y > CONFIG_PROC_SYSCTL=y > CONFIG_PROC_PAGE_MONITOR=y > CONFIG_SYSFS=y > CONFIG_TMPFS=y > # CONFIG_TMPFS_POSIX_ACL is not set > # CONFIG_HUGETLBFS is not set > # CONFIG_HUGETLB_PAGE is not set > # CONFIG_CONFIGFS_FS is not set > CONFIG_MISC_FILESYSTEMS=y > # CONFIG_ADFS_FS is not set > # CONFIG_AFFS_FS is not set > CONFIG_HFS_FS=m > CONFIG_HFSPLUS_FS=y > # CONFIG_BEFS_FS is not set > # CONFIG_BFS_FS is not set > # CONFIG_EFS_FS is not set > # CONFIG_CRAMFS is not set > CONFIG_SQUASHFS=m > CONFIG_SQUASHFS_EMBEDDED=y > CONFIG_SQUASHFS_FRAGMENT_CACHE_SIZE=3 > # CONFIG_VXFS_FS is not set > # CONFIG_MINIX_FS is not set > # CONFIG_OMFS_FS is not set > # CONFIG_HPFS_FS is not set > # CONFIG_QNX4FS_FS is not set > # CONFIG_ROMFS_FS is not set > # CONFIG_SYSV_FS is not set > CONFIG_UFS_FS=m > # CONFIG_UFS_FS_WRITE is not set > # CONFIG_UFS_DEBUG is not set > CONFIG_NETWORK_FILESYSTEMS=y > CONFIG_NFS_FS=m > CONFIG_NFS_V3=y > # CONFIG_NFS_V3_ACL is not set > CONFIG_NFS_V4=y > CONFIG_NFSD=m > CONFIG_NFSD_V3=y > # CONFIG_NFSD_V3_ACL is not set > # CONFIG_NFSD_V4 is not set > CONFIG_LOCKD=m > CONFIG_LOCKD_V4=y > CONFIG_EXPORTFS=m > CONFIG_NFS_COMMON=y > CONFIG_SUNRPC=m > CONFIG_SUNRPC_GSS=m > # CONFIG_SUNRPC_REGISTER_V4 is not set > CONFIG_RPCSEC_GSS_KRB5=m > # CONFIG_RPCSEC_GSS_SPKM3 is not set > CONFIG_SMB_FS=m > CONFIG_SMB_NLS_DEFAULT=y > CONFIG_SMB_NLS_REMOTE="cp437" > CONFIG_CIFS=m > # CONFIG_CIFS_STATS is not set > # CONFIG_CIFS_WEAK_PW_HASH is not set > CONFIG_CIFS_XATTR=y > CONFIG_CIFS_POSIX=y > # CONFIG_CIFS_DEBUG2 is not set > # CONFIG_CIFS_EXPERIMENTAL is not set > # CONFIG_NCP_FS is not set > # CONFIG_CODA_FS is not set > # CONFIG_AFS_FS is not set > > # > # Partition Types > # > CONFIG_PARTITION_ADVANCED=y > # CONFIG_ACORN_PARTITION is not set > # CONFIG_OSF_PARTITION is not set > # CONFIG_AMIGA_PARTITION is not set > # CONFIG_ATARI_PARTITION is not set > CONFIG_MAC_PARTITION=y > CONFIG_MSDOS_PARTITION=y > CONFIG_BSD_DISKLABEL=y > CONFIG_MINIX_SUBPARTITION=y > CONFIG_SOLARIS_X86_PARTITION=y > CONFIG_UNIXWARE_DISKLABEL=y > CONFIG_LDM_PARTITION=y > # CONFIG_LDM_DEBUG is not set > # CONFIG_SGI_PARTITION is not set > # CONFIG_ULTRIX_PARTITION is not set > # CONFIG_SUN_PARTITION is not set > CONFIG_KARMA_PARTITION=y > CONFIG_EFI_PARTITION=y > # CONFIG_SYSV68_PARTITION is not set > CONFIG_NLS=y > CONFIG_NLS_DEFAULT="utf8" > CONFIG_NLS_CODEPAGE_437=y > # CONFIG_NLS_CODEPAGE_737 is not set > # CONFIG_NLS_CODEPAGE_775 is not set > CONFIG_NLS_CODEPAGE_850=y > CONFIG_NLS_CODEPAGE_852=y > # CONFIG_NLS_CODEPAGE_855 is not set > # CONFIG_NLS_CODEPAGE_857 is not set > # CONFIG_NLS_CODEPAGE_860 is not set > # CONFIG_NLS_CODEPAGE_861 is not set > # CONFIG_NLS_CODEPAGE_862 is not set > # CONFIG_NLS_CODEPAGE_863 is not set > CONFIG_NLS_CODEPAGE_864=y > # CONFIG_NLS_CODEPAGE_865 is not set > # CONFIG_NLS_CODEPAGE_866 is not set > # CONFIG_NLS_CODEPAGE_869 is not set > # CONFIG_NLS_CODEPAGE_936 is not set > # CONFIG_NLS_CODEPAGE_950 is not set > # CONFIG_NLS_CODEPAGE_932 is not set > # CONFIG_NLS_CODEPAGE_949 is not set > # CONFIG_NLS_CODEPAGE_874 is not set > # CONFIG_NLS_ISO8859_8 is not set > # CONFIG_NLS_CODEPAGE_1250 is not set > # CONFIG_NLS_CODEPAGE_1251 is not set > # CONFIG_NLS_ASCII is not set > CONFIG_NLS_ISO8859_1=y > CONFIG_NLS_ISO8859_2=y > # CONFIG_NLS_ISO8859_3 is not set > # CONFIG_NLS_ISO8859_4 is not set > # CONFIG_NLS_ISO8859_5 is not set > CONFIG_NLS_ISO8859_6=y > # CONFIG_NLS_ISO8859_7 is not set > # CONFIG_NLS_ISO8859_9 is not set > # CONFIG_NLS_ISO8859_13 is not set > # CONFIG_NLS_ISO8859_14 is not set > # CONFIG_NLS_ISO8859_15 is not set > # CONFIG_NLS_KOI8_R is not set > # CONFIG_NLS_KOI8_U is not set > CONFIG_NLS_UTF8=y > # CONFIG_DLM is not set > > # > # Kernel hacking > # > CONFIG_TRACE_IRQFLAGS_SUPPORT=y > # CONFIG_PRINTK_TIME is not set > # CONFIG_ENABLE_WARN_DEPRECATED is not set > # CONFIG_ENABLE_MUST_CHECK is not set > CONFIG_FRAME_WARN=1024 > CONFIG_MAGIC_SYSRQ=y > # CONFIG_UNUSED_SYMBOLS is not set > # CONFIG_DEBUG_FS is not set > # CONFIG_HEADERS_CHECK is not set > # CONFIG_DEBUG_KERNEL is not set > # CONFIG_DEBUG_BUGVERBOSE is not set > # CONFIG_DEBUG_MEMORY_INIT is not set > CONFIG_ARCH_WANT_FRAME_POINTERS=y > CONFIG_FRAME_POINTER=y > # CONFIG_RCU_CPU_STALL_DETECTOR is not set > # CONFIG_LATENCYTOP is not set > CONFIG_SYSCTL_SYSCALL_CHECK=y > CONFIG_USER_STACKTRACE_SUPPORT=y > CONFIG_HAVE_FUNCTION_TRACER=y > CONFIG_HAVE_FUNCTION_GRAPH_TRACER=y > CONFIG_HAVE_FUNCTION_TRACE_MCOUNT_TEST=y > CONFIG_HAVE_DYNAMIC_FTRACE=y > CONFIG_HAVE_FTRACE_MCOUNT_RECORD=y > > # > # Tracers > # > # CONFIG_SYSPROF_TRACER is not set > # CONFIG_PROVIDE_OHCI1394_DMA_INIT is not set > # CONFIG_DYNAMIC_PRINTK_DEBUG is not set > # CONFIG_SAMPLES is not set > CONFIG_HAVE_ARCH_KGDB=y > # CONFIG_STRICT_DEVMEM is not set > CONFIG_X86_VERBOSE_BOOTUP=y > CONFIG_EARLY_PRINTK=y > # CONFIG_EARLY_PRINTK_DBGP is not set > CONFIG_HAVE_MMIOTRACE_SUPPORT=y > CONFIG_IO_DELAY_TYPE_0X80=0 > CONFIG_IO_DELAY_TYPE_0XED=1 > CONFIG_IO_DELAY_TYPE_UDELAY=2 > CONFIG_IO_DELAY_TYPE_NONE=3 > CONFIG_IO_DELAY_0X80=y > # CONFIG_IO_DELAY_0XED is not set > # CONFIG_IO_DELAY_UDELAY is not set > # CONFIG_IO_DELAY_NONE is not set > CONFIG_DEFAULT_IO_DELAY_TYPE=0 > # CONFIG_OPTIMIZE_INLINING is not set > > # > # Security options > # > # CONFIG_KEYS is not set > # CONFIG_SECURITY is not set > # CONFIG_SECURITYFS is not set > # CONFIG_SECURITY_FILE_CAPABILITIES is not set > CONFIG_CRYPTO=y > > # > # Crypto core or helper > # > # CONFIG_CRYPTO_FIPS is not set > CONFIG_CRYPTO_ALGAPI=y > CONFIG_CRYPTO_ALGAPI2=y > CONFIG_CRYPTO_AEAD=m > CONFIG_CRYPTO_AEAD2=y > CONFIG_CRYPTO_BLKCIPHER=y > CONFIG_CRYPTO_BLKCIPHER2=y > CONFIG_CRYPTO_HASH=y > CONFIG_CRYPTO_HASH2=y > CONFIG_CRYPTO_RNG2=y > CONFIG_CRYPTO_MANAGER=y > CONFIG_CRYPTO_MANAGER2=y > # CONFIG_CRYPTO_GF128MUL is not set > CONFIG_CRYPTO_NULL=m > CONFIG_CRYPTO_CRYPTD=m > CONFIG_CRYPTO_AUTHENC=m > CONFIG_CRYPTO_TEST=m > > # > # Authenticated Encryption with Associated Data > # > # CONFIG_CRYPTO_CCM is not set > # CONFIG_CRYPTO_GCM is not set > # CONFIG_CRYPTO_SEQIV is not set > > # > # Block modes > # > CONFIG_CRYPTO_CBC=y > # CONFIG_CRYPTO_CTR is not set > # CONFIG_CRYPTO_CTS is not set > CONFIG_CRYPTO_ECB=m > # CONFIG_CRYPTO_LRW is not set > CONFIG_CRYPTO_PCBC=y > # CONFIG_CRYPTO_XTS is not set > > # > # Hash modes > # > CONFIG_CRYPTO_HMAC=m > CONFIG_CRYPTO_XCBC=m > > # > # Digest > # > CONFIG_CRYPTO_CRC32C=y > # CONFIG_CRYPTO_CRC32C_INTEL is not set > CONFIG_CRYPTO_MD4=y > CONFIG_CRYPTO_MD5=y > CONFIG_CRYPTO_MICHAEL_MIC=m > # CONFIG_CRYPTO_RMD128 is not set > # CONFIG_CRYPTO_RMD160 is not set > # CONFIG_CRYPTO_RMD256 is not set > # CONFIG_CRYPTO_RMD320 is not set > CONFIG_CRYPTO_SHA1=y > CONFIG_CRYPTO_SHA256=y > CONFIG_CRYPTO_SHA512=y > CONFIG_CRYPTO_TGR192=m > CONFIG_CRYPTO_WP512=m > > # > # Ciphers > # > CONFIG_CRYPTO_AES=y > # CONFIG_CRYPTO_AES_X86_64 is not set > CONFIG_CRYPTO_ANUBIS=m > CONFIG_CRYPTO_ARC4=m > CONFIG_CRYPTO_BLOWFISH=m > CONFIG_CRYPTO_CAMELLIA=m > CONFIG_CRYPTO_CAST5=m > CONFIG_CRYPTO_CAST6=m > CONFIG_CRYPTO_DES=m > CONFIG_CRYPTO_FCRYPT=m > CONFIG_CRYPTO_KHAZAD=m > # CONFIG_CRYPTO_SALSA20 is not set > # CONFIG_CRYPTO_SALSA20_X86_64 is not set > # CONFIG_CRYPTO_SEED is not set > CONFIG_CRYPTO_SERPENT=m > CONFIG_CRYPTO_TEA=m > CONFIG_CRYPTO_TWOFISH=m > CONFIG_CRYPTO_TWOFISH_COMMON=m > # CONFIG_CRYPTO_TWOFISH_X86_64 is not set > > # > # Compression > # > CONFIG_CRYPTO_DEFLATE=m > # CONFIG_CRYPTO_LZO is not set > > # > # Random Number Generation > # > # CONFIG_CRYPTO_ANSI_CPRNG is not set > CONFIG_CRYPTO_HW=y > # CONFIG_CRYPTO_DEV_HIFN_795X is not set > CONFIG_HAVE_KVM=y > CONFIG_HAVE_KVM_IRQCHIP=y > CONFIG_VIRTUALIZATION=y > # CONFIG_KVM is not set > # CONFIG_VIRTIO_PCI is not set > # CONFIG_VIRTIO_BALLOON is not set > > # > # Library routines > # > CONFIG_BITREVERSE=y > CONFIG_GENERIC_FIND_FIRST_BIT=y > CONFIG_GENERIC_FIND_NEXT_BIT=y > CONFIG_GENERIC_FIND_LAST_BIT=y > CONFIG_CRC_CCITT=m > CONFIG_CRC16=y > # CONFIG_CRC_T10DIF is not set > CONFIG_CRC_ITU_T=m > CONFIG_CRC32=y > # CONFIG_CRC7 is not set > CONFIG_LIBCRC32C=y > CONFIG_ZLIB_INFLATE=m > CONFIG_ZLIB_DEFLATE=m > CONFIG_PLIST=y > CONFIG_HAS_IOMEM=y > CONFIG_HAS_IOPORT=y > CONFIG_HAS_DMA=y > [5 lspci-vvv.txt <text/plain; US-ASCII (base64)>] > 00:1b.0 Audio device: Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 03) > Subsystem: Hewlett-Packard Company Device 30f4 > Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- > Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- > Latency: 0, Cache Line Size: 64 bytes > Interrupt: pin B routed to IRQ 22 > Region 0: Memory at df000000 (64-bit, non-prefetchable) [size=16K] > Capabilities: [50] Power Management version 2 > Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,D3cold+) > Status: D0 PME-Enable- DSel=0 DScale=0 PME- > Capabilities: [60] MSI: Mask- 64bit+ Count=1/1 Enable+ > Address: 0000000000000000 Data: 0000 > Capabilities: [70] Express (v1) Root Complex Integrated Endpoint, MSI 00 > DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s <64ns, L1 <1us > ExtTag- RBE- FLReset+ > DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- > RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ > MaxPayload 128 bytes, MaxReadReq 128 bytes > DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr+ TransPend- > LnkCap: Port #0, Speed unknown, Width x0, ASPM unknown, Latency L0 <64ns, L1 <1us > ClockPM- Surprise- LLActRep- BwNot- > LnkCtl: ASPM Disabled; Disabled- Retrain- CommClk- > ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- > LnkSta: Speed unknown, Width x0, TrErr- Train- SlotClk- DLActive- BWMgmt- ABWMgmt- > Capabilities: [100] Virtual Channel <?> > Capabilities: [130] Root Complex Link <?> > Kernel driver in use: HDA Intel > Kernel modules: snd-hda-intel

2 2

[PATCH] ASoC: WM9712 Codec: Workaround an unmute problem
by John Bonesio 01 Aug '09

01 Aug '09

When setting the PCM mixer (unuting), the main Headphone mixer setting gets set back to mute. At this time it appears this is occuring in hardware. Signed-off-by: John Bonesio <bones(a)secretlab.ca> --- sound/soc/codecs/wm9712.c | 21 +++++++++++++++++++-- 1 files changed, 19 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index b57c817..6f8d164 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -28,6 +28,7 @@ static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg); static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int val); +static int ac97_flush(struct snd_soc_codec *codec, unsigned int reg); /* * WM9712 register cache @@ -177,9 +178,14 @@ static int mixer_event(struct snd_soc_dapm_widget *w, else ac97_write(w->codec, AC97_VIDEO, mic | 0x8000); - if (l & 0x2 || r & 0x2) + if (l & 0x2 || r & 0x2) { ac97_write(w->codec, AC97_PCM, pcm & 0x7fff); - else + /* + * Workaround an apparent bug where the headphone mute setting + * is modified when the PCM mute setting is enabled. + */ + ac97_flush(w->codec, AC97_HEADPHONE); + } else ac97_write(w->codec, AC97_PCM, pcm | 0x8000); if (l & 0x4 || r & 0x4) @@ -472,6 +478,17 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, return 0; } +static int ac97_flush(struct snd_soc_codec *codec, unsigned int reg) +{ + unsigned int val; + u16 *cache = codec->reg_cache; + + if ((reg >> 1) < (ARRAY_SIZE(wm9712_reg))) { + val = cache[reg >> 1]; + soc_ac97_ops.write(codec->ac97, reg, val); + } +} + static int ac97_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { -- 1.6.0.4 --=-3OuCfmlE7wntN9J+RSz+--

1 0

[alsa-devel] [RFC] [PATCH 3/3 v2] ASoC: add support for Amstrad E3 (Delta) machine
by Janusz Krzysztofik 01 Aug '09

01 Aug '09

This patch adds machine support for Amstrad E3 (Delta) videophone to ASoC. Created and tested against linux-2.6.31-rc3. Applies and works with linux-omap-2.6 commit 7c5cb7862d32cb344be7831d466535d5255e35ac as well. Depends on: 1) latest version of the CX20442 codec driver that exposes v253_ops structure[1], 2) patch 2/3 form this series: TTY: Add definition of a new line discipline required by Amstrad E3 (Delta) ASoC driver[2]. CPU DAI parameters best matching the codec DAI has been selected out empirically for best user experience. Board specific audio function control (with related DAPM widgets) has been modeled after empirically discovered codec capabilities. Unlike other ASoC machine drivers, this one makes use of a codec provided line discipline that is required for talking to a modem chip that can control the codec behavoiur. As the line discipline operations must call board specific bits as well, the machine driver registers its own line discipline ops, not the codec provided, and then calls those codec provided from inside its own callbacks. If some kind of a glue, like a bus over a tty, exsited that could help in runtime detection of a modem (bus adapter) over a more generic line discipline (bus driver)[3], the line discipline code could be probably designed in a more generic way. In order to work at all, this driver requires a working McBSP1. On OMAP1510 based machines (not sure if other OMAP1 variants as well), where McBSP1 is a DSP public peripheral, that means the kernel must provide basic DSP support, ie. omap_dsp_init(), in order to power up the DSP. This used to be included in linux-omap-2.6 tree up to commit 2512fd29db4eb09e82d182596304c7aaf76d2c5c. Without that, the driver would not work, ie. not shift in/out any bits over the CPU DAI[4]. This limitation is not board, but CPU specific, and may apply to other code that makes use of McBSP1/McBSP3 on affected machines. I provide an extra patch (4/3) as a temporary solution. To work correctly in playback mode, this driver requires my prevoiusly submitted patch that corrects pcm pointer calculation for OMAP1510 based machines[5] (already included in linux-2.6.31-rc3). To support codec controls, this driver requires my previously submitted patch that adds support for modem found on Amstrad Delta[6]. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2009-July/019780.html [2] http://www.spinics.net/lists/linux-serial/msg01862.html [3] http://www.spinics.net/lists/linux-serial/msg01856.html [4] http://www.spinics.net/lists/linux-omap/msg15114.html [5] http://mailman.alsa-project.org/pipermail/alsa-devel/2009-June/018950.html [6] http://www.spinics.net/lists/linux-omap/msg15432.html Credits to: Mark Underwood - for his initial, omap-alsa based sound driver for this machine, Mark Brown - for his help, patience and excellent subsytem maintainer support. Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- --- linux-2.6.31-rc3/sound/soc/omap/Kconfig.orig 2009-07-04 16:40:00.000000000 +0200 +++ linux-2.6.31-rc3/sound/soc/omap/Kconfig 2009-07-22 01:06:51.000000000 +0200 @@ -15,6 +15,14 @@ config SND_OMAP_SOC_N810 help Say Y if you want to add support for SoC audio on Nokia N810. +config SND_OMAP_SOC_AMS_DELTA + tristate "SoC Audio support for Amstrad E3 (Delta) videophone" + depends on SND_OMAP_SOC && MACH_AMS_DELTA + select SND_OMAP_SOC_MCBSP + select SND_SOC_CX20442 + help + Say Y if you want to add support for SoC audio on Amstrad Delta. + config SND_OMAP_SOC_OSK5912 tristate "SoC Audio support for omap osk5912" depends on SND_OMAP_SOC && MACH_OMAP_OSK && I2C --- linux-2.6.31-rc3/sound/soc/omap/Makefile.orig 2009-07-04 16:40:00.000000000 +0200 +++ linux-2.6.31-rc3/sound/soc/omap/Makefile 2009-07-22 01:06:51.000000000 +0200 @@ -7,6 +7,7 @@ obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd- # OMAP Machine Support snd-soc-n810-objs := n810.o +snd-soc-ams-delta-objs := ams-delta.o snd-soc-osk5912-objs := osk5912.o snd-soc-overo-objs := overo.o snd-soc-omap2evm-objs := omap2evm.o @@ -16,6 +17,7 @@ snd-soc-omap3pandora-objs := omap3pandor snd-soc-omap3beagle-objs := omap3beagle.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o +obj-$(CONFIG_SND_OMAP_SOC_AMS_DELTA) += snd-soc-ams-delta.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o --- /dev/null 2009-07-27 16:01:44.916962115 +0200 +++ linux-2.6.31-rc3/sound/soc/omap/ams-delta.c 2009-07-29 12:11:57.000000000 +0200 @@ -0,0 +1,646 @@ +/* + * ams-delta.c -- SoC audio for Amstrad E3 (Delta) videophone + * + * Copyright (C) 2009 Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> + * + * Initially based on sound/soc/omap/osk5912.x + * Copyright (C) 2008 Mistral Solutions + * + * 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. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/gpio.h> +#include <linux/spinlock.h> +#include <linux/tty.h> + +#include <sound/soc-dapm.h> +#include <sound/jack.h> + +#include <asm/mach-types.h> + +#include <mach/board-ams-delta.h> +#include <mach/mcbsp.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/cx20442.h" + + +/* Board specific DAPM widgets */ + const struct snd_soc_dapm_widget ams_delta_dapm_widgets[] = { + /* Handset */ + SND_SOC_DAPM_MIC("Mouthpiece", NULL), + SND_SOC_DAPM_HP("Earpiece", NULL), + /* Handsfree/Speakerphone */ + SND_SOC_DAPM_MIC("Microphone", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), +}; + +/* How they are connected to codec pins */ +static const struct snd_soc_dapm_route ams_delta_audio_map[] = { + {"TELIN", NULL, "Mouthpiece"}, + {"Earpiece", NULL, "TELOUT"}, + + {"MIC", NULL, "Microphone"}, + {"Speaker", NULL, "SPKOUT"}, +}; + +/* + * Controls, functional after the modem line discipline is activated. + */ + +/* Virtual switch: audio input/output constellations */ +static const char *ams_delta_audio_mode[] = + {"Mixed", "Handset", "Handsfree", "Speakerphone"}; + +/* Selection <-> pin translation */ +#define AMS_DELTA_MOUTHPIECE 0 +#define AMS_DELTA_EARPIECE 1 +#define AMS_DELTA_MICROPHONE 2 +#define AMS_DELTA_SPEAKER 3 +#define AMS_DELTA_AGC 4 + +#define AMS_DELTA_MIXED ((1 << AMS_DELTA_EARPIECE) | \ + (1 << AMS_DELTA_MICROPHONE)) +#define AMS_DELTA_HANDSET ((1 << AMS_DELTA_MOUTHPIECE) | \ + (1 << AMS_DELTA_EARPIECE)) +#define AMS_DELTA_HANDSFREE ((1 << AMS_DELTA_MICROPHONE) | \ + (1 << AMS_DELTA_SPEAKER)) +#define AMS_DELTA_SPEAKERPHONE (AMS_DELTA_HANDSFREE | (1 << AMS_DELTA_AGC)) + +unsigned short ams_delta_audio_mode_pins[] = { + AMS_DELTA_MIXED, + AMS_DELTA_HANDSET, + AMS_DELTA_HANDSFREE, + AMS_DELTA_SPEAKERPHONE, +}; + +static unsigned short ams_delta_audio_agc; + +static int ams_delta_set_audio_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct soc_enum *control = (struct soc_enum *)kcontrol->private_value; + unsigned short pins; + int pin, changed = 0; + + /* Refuse any mode changes if we are not able to control the codec. */ + if (!codec->control_data) + return -EUNATCH; + + if (ucontrol->value.enumerated.item[0] >= control->max) + return -EINVAL; + + mutex_lock(&codec->mutex); + + /* Translate selection to bitmap */ + pins = ams_delta_audio_mode_pins[ucontrol->value.enumerated.item[0]]; + + /* Setup pins after corresponding bits if changed */ + pin = !!(pins & (1 << AMS_DELTA_MOUTHPIECE)); + if (pin != snd_soc_dapm_get_pin_status(codec, "Mouthpiece")) { + changed = 1; + if (pin) + snd_soc_dapm_enable_pin(codec, "Mouthpiece"); + else + snd_soc_dapm_disable_pin(codec, "Mouthpiece"); + } + pin = !!(pins & (1 << AMS_DELTA_EARPIECE)); + if (pin != snd_soc_dapm_get_pin_status(codec, "Earpiece")) { + changed = 1; + if (pin) + snd_soc_dapm_enable_pin(codec, "Earpiece"); + else + snd_soc_dapm_disable_pin(codec, "Earpiece"); + } + pin = !!(pins & (1 << AMS_DELTA_MICROPHONE)); + if (pin != snd_soc_dapm_get_pin_status(codec, "Microphone")) { + changed = 1; + if (pin) + snd_soc_dapm_enable_pin(codec, "Microphone"); + else + snd_soc_dapm_disable_pin(codec, "Microphone"); + } + pin = !!(pins & (1 << AMS_DELTA_SPEAKER)); + if (pin != snd_soc_dapm_get_pin_status(codec, "Speaker")) { + changed = 1; + if (pin) + snd_soc_dapm_enable_pin(codec, "Speaker"); + else + snd_soc_dapm_disable_pin(codec, "Speaker"); + } + pin = !!(pins & (1 << AMS_DELTA_AGC)); + if (pin != ams_delta_audio_agc) { + ams_delta_audio_agc = pin; + changed = 1; + if (pin) + snd_soc_dapm_enable_pin(codec, "AGCIN"); + else + snd_soc_dapm_disable_pin(codec, "AGCIN"); + } + if (changed) + snd_soc_dapm_sync(codec); + + mutex_unlock(&codec->mutex); + + return changed; +} + +static int ams_delta_get_audio_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned short pins, mode; + + pins = ((snd_soc_dapm_get_pin_status(codec, "Mouthpiece") << + AMS_DELTA_MOUTHPIECE) | + (snd_soc_dapm_get_pin_status(codec, "Earpiece") << + AMS_DELTA_EARPIECE)); + if (pins) + pins |= (snd_soc_dapm_get_pin_status(codec, "Microphone") << + AMS_DELTA_MICROPHONE); + else + pins = ((snd_soc_dapm_get_pin_status(codec, "Microphone") << + AMS_DELTA_MICROPHONE) | + (snd_soc_dapm_get_pin_status(codec, "Speaker") << + AMS_DELTA_SPEAKER) | + (ams_delta_audio_agc << AMS_DELTA_AGC)); + + for (mode = 0; mode < ARRAY_SIZE(ams_delta_audio_mode); mode++) + if (pins == ams_delta_audio_mode_pins[mode]) + break; + + if (mode >= ARRAY_SIZE(ams_delta_audio_mode)) + return -EINVAL; + + ucontrol->value.enumerated.item[0] = mode; + + return 0; +} + +static const struct soc_enum ams_delta_audio_enum[] = { + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(ams_delta_audio_mode), + ams_delta_audio_mode), +}; + +static const struct snd_kcontrol_new ams_delta_audio_controls[] = { + SOC_ENUM_EXT("Audio Mode", ams_delta_audio_enum[0], + ams_delta_get_audio_mode, ams_delta_set_audio_mode), +}; + +/* Hook switch */ +static struct snd_soc_jack ams_delta_hook_switch; +static struct snd_soc_jack_gpio ams_delta_hook_switch_gpios[] = { + { + .gpio = 4, + .name = "hook_switch", + .report = SND_JACK_HEADSET, + .invert = 1, + .debounce_time = 150, + } +}; + +/* After we are able to control the codec over the modem, + * the hook switch can be used for dynamic DAPM reconfiguration. */ +static struct snd_soc_jack_pin ams_delta_hook_switch_pins[] = { + /* Handset */ + { + .pin = "Mouthpiece", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Earpiece", + .mask = SND_JACK_HEADPHONE, + }, + /* Handsfree */ + { + .pin = "Microphone", + .mask = SND_JACK_MICROPHONE, + .invert = 1, + }, + { + .pin = "Speaker", + .mask = SND_JACK_HEADPHONE, + .invert = 1, + }, +}; + + +/* + * Modem line discipline, required for making above controls functional. + * Activated from userspace with ldattach, possibly invoked from udev rule. + */ + +/* To actually apply any modem controlled configuration changes to the codec, + * we must connect codec DAI pins to the modem for a moment. Be carefull not + * to interfere with our digital mute function that shares the same hardware. */ +static struct timer_list cx81801_timer; +static bool cx81801_cmd_pending; +static bool ams_delta_muted; +static spinlock_t ams_delta_lock = SPIN_LOCK_UNLOCKED; + +static void cx81801_timeout(unsigned long data) +{ + int muted; + + spin_lock(&ams_delta_lock); + cx81801_cmd_pending = 0; + muted = ams_delta_muted; + spin_unlock(&ams_delta_lock); + + /* Reconnect the codec DAI back from the modem to the CPU DAI + * only if digital mute still off */ + if (!muted) + ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC, 0); +} + +/* Line discipline .open() */ +static int cx81801_open(struct tty_struct *tty) +{ + return v253_ops.open(tty); +} + +/* Line discipline .close() */ +static void cx81801_close(struct tty_struct *tty) +{ + struct snd_soc_codec *codec = tty->disc_data; + + del_timer_sync(&cx81801_timer); + + v253_ops.close(tty); + + /* Prevent the hook switch from further changing the DAPM pins */ + INIT_LIST_HEAD(&ams_delta_hook_switch.pins); + + /* Revert back to default audio input/output constellation */ + snd_soc_dapm_disable_pin(codec, "Mouthpiece"); + snd_soc_dapm_enable_pin(codec, "Earpiece"); + snd_soc_dapm_enable_pin(codec, "Microphone"); + snd_soc_dapm_disable_pin(codec, "Speaker"); + snd_soc_dapm_disable_pin(codec, "AGCIN"); + snd_soc_dapm_sync(codec); +} + +/* Line discipline .hangup() */ +static int cx81801_hangup(struct tty_struct *tty) +{ + cx81801_close(tty); + return 0; +} + +/* Line discipline .recieve_buf() */ +static void cx81801_receive(struct tty_struct *tty, + const unsigned char *cp, char *fp, int count) +{ + struct snd_soc_codec *codec = tty->disc_data; + const unsigned char *c; + int apply, ret; + + if (!codec->control_data) { + /* First modem response, complete setup procedure */ + + /* Initialize timer used for config pulse generation */ + setup_timer(&cx81801_timer, cx81801_timeout, 0); + + v253_ops.receive_buf(tty, cp, fp, count); + + /* Link hook switch to DAPM pins */ + ret = snd_soc_jack_add_pins(&ams_delta_hook_switch, + ARRAY_SIZE(ams_delta_hook_switch_pins), + ams_delta_hook_switch_pins); + if (ret) + dev_warn(codec->socdev->card->dev, + "Failed to link hook switch to DAPM pins, " + "will continue with hook switch unlinked.\n"); + + return; + } + + v253_ops.receive_buf(tty, cp, fp, count); + + for (c = &cp[count - 1]; c >= cp; c--) { + if (*c != '\r') + continue; + /* Complete modem response received, apply config to codec */ + + spin_lock_bh(&ams_delta_lock); + mod_timer(&cx81801_timer, jiffies + msecs_to_jiffies(150)); + apply = !ams_delta_muted && !cx81801_cmd_pending; + cx81801_cmd_pending = 1; + spin_unlock_bh(&ams_delta_lock); + + /* Apply config pulse by connecting the codec to the modem + * if not already done */ + if (apply) + ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC, + AMS_DELTA_LATCH2_MODEM_CODEC); + break; + } +} + +/* Line discipline .write_wakeup() */ +static void cx81801_wakeup(struct tty_struct *tty) +{ + v253_ops.write_wakeup(tty); +} + +static struct tty_ldisc_ops cx81801_ops = { + .magic = TTY_LDISC_MAGIC, + .name = "cx81801", + .owner = THIS_MODULE, + .open = cx81801_open, + .close = cx81801_close, + .hangup = cx81801_hangup, + .receive_buf = cx81801_receive, + .write_wakeup = cx81801_wakeup, +}; + + +/* + * Even if not very usefull, the sound card can still work without any of the + * above functonality activated. You can still control its audio input/output + * constellation and speakerphone gain from userspace by issueing AT commands + * over the modem port. + */ + +static int ams_delta_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + /* Set cpu DAI configuration */ + return snd_soc_dai_set_fmt(rtd->dai->cpu_dai, + SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); +} + +static struct snd_soc_ops ams_delta_ops = { + .hw_params = ams_delta_hw_params, +}; + + +/* Board specific codec bias level control */ +static int ams_delta_set_bias_level(struct snd_soc_card *card, + enum snd_soc_bias_level level) +{ + struct snd_soc_codec *codec = card->codec; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + case SND_SOC_BIAS_STANDBY: + if (codec->bias_level == SND_SOC_BIAS_OFF) + ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET, + AMS_DELTA_LATCH2_MODEM_NRESET); + break; + case SND_SOC_BIAS_OFF: + if (codec->bias_level != SND_SOC_BIAS_OFF) + ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET, + 0); + } + codec->bias_level = level; + + return 0; +} + +/* Digital mute implemented using modem/CPU multiplexer. + * Shares hardware with codec config pulse generation */ +static bool ams_delta_muted = 1; + +static int ams_delta_digital_mute(struct snd_soc_dai *dai, int mute) +{ + int apply; + + if (ams_delta_muted == mute) + return 0; + + spin_lock_bh(&ams_delta_lock); + ams_delta_muted = mute; + apply = !cx81801_cmd_pending; + spin_unlock_bh(&ams_delta_lock); + + if (apply) + ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC, + mute ? AMS_DELTA_LATCH2_MODEM_CODEC : 0); + return 0; +} + +/* Our codec DAI probably doesn't have its own .ops structure */ +static struct snd_soc_dai_ops ams_delta_dai_ops = { + .digital_mute = ams_delta_digital_mute, +}; + +/* Will be used if the codec ever has its own digital_mute function */ +static int ams_delta_startup(struct snd_pcm_substream *substream) +{ + return ams_delta_digital_mute(NULL, 0); +} + +static void ams_delta_shutdown(struct snd_pcm_substream *substream) +{ + ams_delta_digital_mute(NULL, 1); +} + + +/* + * Card initialization + */ + +static int ams_delta_cx20442_init(struct snd_soc_codec *codec) +{ + struct snd_soc_dai *codec_dai = codec->dai; + struct snd_soc_card *card = codec->socdev->card; + int ret; + /* Codec is ready, now add/activate board specific controls */ + + /* Set up digital mute if not provided by the codec */ + if (!codec_dai->ops) { + codec_dai->ops = &ams_delta_dai_ops; + } else if (!codec_dai->ops->digital_mute) { + codec_dai->ops->digital_mute = ams_delta_digital_mute; + } else { + ams_delta_ops.startup = ams_delta_startup; + ams_delta_ops.shutdown = ams_delta_shutdown; + } + + /* Set codec bias level */ + ams_delta_set_bias_level(card, SND_SOC_BIAS_STANDBY); + + /* Add hook switch - can be used to control the codec from userspace + * even if line discipline fails */ + ret = snd_soc_jack_new(card, "hook_switch", + SND_JACK_HEADSET, &ams_delta_hook_switch); + if (ret) + dev_warn(card->dev, + "Failed to allocate resources for hook switch, " + "will continue without one.\n"); + else { + ret = snd_soc_jack_add_gpios(&ams_delta_hook_switch, + ARRAY_SIZE(ams_delta_hook_switch_gpios), + ams_delta_hook_switch_gpios); + if (ret) + dev_warn(card->dev, + "Failed to set up hook switch GPIO line, " + "will continue with hook switch inactive.\n"); + } + + /* Register optional line discipline for over the modem control */ + ret = tty_register_ldisc(N_AMSDELTA, &cx81801_ops); + if (ret) { + dev_warn(card->dev, + "Failed to register line discipline, " + "will continue without any controls.\n"); + return 0; + } + + /* Add board specific DAPM widgets and routes */ + ret = snd_soc_dapm_new_controls(codec, ams_delta_dapm_widgets, + ARRAY_SIZE(ams_delta_dapm_widgets)); + if (ret) { + dev_warn(card->dev, + "Failed to register DAPM controls, " + "will continue without any.\n"); + return 0; + } + + ret = snd_soc_dapm_add_routes(codec, ams_delta_audio_map, + ARRAY_SIZE(ams_delta_audio_map)); + if (ret) { + dev_warn(card->dev, + "Failed to set up DAPM routes, " + "will continue with codec default map.\n"); + return 0; + } + + /* Set up initial pin constellation */ + snd_soc_dapm_disable_pin(codec, "Mouthpiece"); + snd_soc_dapm_enable_pin(codec, "Earpiece"); + snd_soc_dapm_enable_pin(codec, "Microphone"); + snd_soc_dapm_disable_pin(codec, "Speaker"); + snd_soc_dapm_disable_pin(codec, "AGCIN"); + snd_soc_dapm_disable_pin(codec, "AGCOUT"); + snd_soc_dapm_sync(codec); + + /* Add virtual switch */ + ret = snd_soc_add_controls(codec, ams_delta_audio_controls, + ARRAY_SIZE(ams_delta_audio_controls)); + if (ret) + dev_warn(card->dev, + "Failed to register audio mode control, " + "will continue without it.\n"); + + return 0; +} + +/* DAI glue - connects codec <--> CPU */ +static struct snd_soc_dai_link ams_delta_dai_link = { + .name = "CX20442", + .stream_name = "CX20442", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &cx20442_dai, + .init = ams_delta_cx20442_init, + .ops = &ams_delta_ops, +}; + +/* Audio card driver */ +static struct snd_soc_card ams_delta_audio_card = { + .name = "AMS_DELTA", + .platform = &omap_soc_platform, + .dai_link = &ams_delta_dai_link, + .num_links = 1, + .set_bias_level = ams_delta_set_bias_level, +}; + +/* Audio subsystem */ +static struct snd_soc_device ams_delta_snd_soc_device = { + .card = &ams_delta_audio_card, + .codec_dev = &cx20442_codec_dev, +}; + +/* Module init/exit */ +static struct platform_device *ams_delta_audio_platform_device; +static struct platform_device *cx20442_platform_device; + +static int __init ams_delta_module_init(void) +{ + int ret; + + if (!(machine_is_ams_delta())) + return -ENODEV; + + ams_delta_audio_platform_device = + platform_device_alloc("soc-audio", -1); + if (!ams_delta_audio_platform_device) + return -ENOMEM; + + platform_set_drvdata(ams_delta_audio_platform_device, + &ams_delta_snd_soc_device); + ams_delta_snd_soc_device.dev = &ams_delta_audio_platform_device->dev; + *(unsigned int *)ams_delta_dai_link.cpu_dai->private_data = OMAP_MCBSP1; + + ret = platform_device_add(ams_delta_audio_platform_device); + if (ret) + goto err; + + /* + * Codec platform device could be registered from elsewhere (board?), + * but I do it here as it makes sense only if used with the card. + */ + cx20442_platform_device = platform_device_register_simple("cx20442", + -1, NULL, 0); + return 0; +err: + platform_device_put(ams_delta_audio_platform_device); + return ret; +} +module_init(ams_delta_module_init); + +static void __exit ams_delta_module_exit(void) +{ + struct snd_soc_codec *codec; + struct tty_struct *tty; + + if (ams_delta_audio_card.codec) { + codec = ams_delta_audio_card.codec; + + if (codec->control_data) { + tty = codec->control_data; + + tty_hangup(tty); + } + } + + if (tty_unregister_ldisc(N_AMSDELTA) != 0) + dev_warn(&ams_delta_audio_platform_device->dev, + "failed to unregister AMSDELTA line discipline\n"); + + snd_soc_jack_free_gpios(&ams_delta_hook_switch, + ARRAY_SIZE(ams_delta_hook_switch_gpios), + ams_delta_hook_switch_gpios); + + /* Keep modem power on */ + ams_delta_set_bias_level(&ams_delta_audio_card, SND_SOC_BIAS_STANDBY); + + platform_device_unregister(cx20442_platform_device); + platform_device_unregister(ams_delta_audio_platform_device); +} +module_exit(ams_delta_module_exit); + +MODULE_AUTHOR("Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl>"); +MODULE_DESCRIPTION("ALSA SoC driver for Amstrad E3 (Delta) videophone"); +MODULE_LICENSE("GPL");

3 6

[alsa-devel] [PATCH] ASoC: jack: Fix race in snd_soc_jack_add_gpios
by Lars-Peter Clausen 01 Aug '09

01 Aug '09

The irq can fire as soon as it has been requested, thus all fields accessed from within the irq handler must be initialized prior to requesting the irq. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- The patch is against alsa-kernel/master but I guess something similar should also go into 2.6.31. sound/soc/soc-jack.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 4aa7d8f..1d455ab 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -221,6 +221,9 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, if (ret) goto err; + INIT_WORK(&gpios[i].work, gpio_work); + gpios[i].jack = jack; + ret = request_irq(gpio_to_irq(gpios[i].gpio), gpio_handler, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING, @@ -234,9 +237,6 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, gpio_export(gpios[i].gpio, false); #endif - INIT_WORK(&gpios[i].work, gpio_work); - gpios[i].jack = jack; - /* Update initial jack status */ snd_soc_jack_gpio_detect(&gpios[i]); } -- 1.5.6.5

2 1

[alsa-devel] [PATCH 4/4] pulse: call pa_threaded_mainloop_wait() to handle spurious wakeups
by Lennart Poettering 31 Jul '09

31 Jul '09

pa_threaded_mainloop_wait() can wake up for no reason, according to the specs of the underlying POSIX ptrhead_cond_wait() docs, so we need to call it in a loop here which should be cleaner anyway. --- pulse/pulse.c | 13 ++++++++++--- 1 files changed, 10 insertions(+), 3 deletions(-) diff --git a/pulse/pulse.c b/pulse/pulse.c index ae66b0c..6f58a7e 100644 --- a/pulse/pulse.c +++ b/pulse/pulse.c @@ -262,10 +262,17 @@ int pulse_connect(snd_pulse_t * p, const char *server) if (err < 0) goto error; - pa_threaded_mainloop_wait(p->mainloop); + for (;;) { + pa_context_state_t state = pa_context_get_state(p->context); - if (pa_context_get_state(p->context) != PA_CONTEXT_READY) - goto error; + if (!PA_CONTEXT_IS_GOOD(state)) + goto error; + + if (state == PA_CONTEXT_READY) + break; + + pa_threaded_mainloop_wait(p->mainloop); + } pa_threaded_mainloop_unlock(p->mainloop); -- 1.6.3.3 Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

1 0

[alsa-devel] [PATCH 3/4] pulse: unify destruction of snd_pulse_t
by Lennart Poettering 31 Jul '09

31 Jul '09

--- pulse/pulse.c | 31 ++++++++++++------------------- 1 files changed, 12 insertions(+), 19 deletions(-) diff --git a/pulse/pulse.c b/pulse/pulse.c index dd17384..ae66b0c 100644 --- a/pulse/pulse.c +++ b/pulse/pulse.c @@ -217,36 +217,29 @@ snd_pulse_t *pulse_new(void) return p; fail: + pulse_free(p); + + return NULL; +} + +void pulse_free(snd_pulse_t * p) +{ if (p->mainloop) pa_threaded_mainloop_stop(p->mainloop); - if (p->context) + if (p->context) { + pa_context_disconnect(p->context); pa_context_unref(p->context); + } if (p->mainloop) pa_threaded_mainloop_free(p->mainloop); - if (p->main_fd >= 0) - close(p->main_fd); - if (p->thread_fd >= 0) close(p->thread_fd); - free(p); - - return NULL; -} - -void pulse_free(snd_pulse_t * p) -{ - pa_threaded_mainloop_stop(p->mainloop); - - pa_context_disconnect(p->context); - pa_context_unref(p->context); - pa_threaded_mainloop_free(p->mainloop); - - close(p->thread_fd); - close(p->main_fd); + if (p->main_fd >= 0) + close(p->main_fd); free(p); } -- 1.6.3.3 Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

1 0

[alsa-devel] [PATCH 2/4] pulse: use PA_CONTEXT_IS_GOOD where applicable
by Lennart Poettering 31 Jul '09

31 Jul '09

PA_CONTEXT_IS_GOOD is a safer way to check whether a context is still valid. This patch also bumps the version requirement of libpulse to 0.9.11. --- configure.in | 2 +- pulse/pulse.c | 5 ++--- 2 files changed, 3 insertions(+), 4 deletions(-) diff --git a/configure.in b/configure.in index 36740e9..f8bc669 100644 --- a/configure.in +++ b/configure.in @@ -30,7 +30,7 @@ AC_ARG_ENABLE([pulseaudio], AS_HELP_STRING([--disable-pulseaudio], [Disable building of pulseaudio plugin])) if test "x$enable_pulseaudio" != "xno"; then - PKG_CHECK_MODULES(pulseaudio, [libpulse >= 0.9.2], [HAVE_PULSE=yes], [HAVE_PULSE=no]) + PKG_CHECK_MODULES(pulseaudio, [libpulse >= 0.9.11], [HAVE_PULSE=yes], [HAVE_PULSE=no]) fi AM_CONDITIONAL(HAVE_PULSE, test x$HAVE_PULSE = xyes) diff --git a/pulse/pulse.c b/pulse/pulse.c index 95d8dde..dd17384 100644 --- a/pulse/pulse.c +++ b/pulse/pulse.c @@ -38,7 +38,7 @@ int pulse_check_connection(snd_pulse_t * p) state = pa_context_get_state(p->context); - if (state != PA_CONTEXT_READY) + if (!PA_CONTEXT_IS_GOOD(state)) return -EIO; return 0; @@ -127,8 +127,7 @@ int pulse_wait_stream_state(snd_pulse_t * p, pa_stream * stream, if (state == target) break; - if (state == PA_STREAM_FAILED || - state == PA_STREAM_TERMINATED) + if (!PA_STREAM_IS_GOOD(state)) return -EIO; pa_threaded_mainloop_wait(p->mainloop); -- 1.6.3.3 Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

1 0

[PATCH] pulse: get rid of a number of assert()s
by Lennart Poettering 31 Jul '09

31 Jul '09

Instead of hitting an assert when any of the plugin functions is called in an invalid context we should return a clean error to make sure programs are not unnecessarily aborted. This should fix issues such as http://pulseaudio.org/ticket/595 --- pulse/ctl_pulse.c | 35 ++++++++++++++++++----- pulse/pcm_pulse.c | 79 +++++++++++++++++++++++++++++++++++++++++------------ pulse/pulse.c | 34 ++++++++++++++++------- 3 files changed, 112 insertions(+), 36 deletions(-) diff --git a/pulse/ctl_pulse.c b/pulse/ctl_pulse.c index c6cf9e2..2caa29b 100644 --- a/pulse/ctl_pulse.c +++ b/pulse/ctl_pulse.c @@ -125,8 +125,9 @@ static void event_cb(pa_context * c, pa_subscription_event_type_t t, pa_operation *o; assert(ctl); - assert(ctl->p); - assert(ctl->p->context); + + if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + return; o = pa_context_get_sink_info_by_name(ctl->p->context, ctl->sink, sink_info_cb, ctl); @@ -148,8 +149,9 @@ static int pulse_update_volume(snd_ctl_pulse_t * ctl) pa_operation *o; assert(ctl); - assert(ctl->p); - assert(ctl->p->context); + + if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + return -EBADFD; o = pa_context_get_sink_info_by_name(ctl->p->context, ctl->sink, sink_info_cb, ctl); @@ -203,6 +205,9 @@ static int pulse_elem_list(snd_ctl_ext_t * ext, unsigned int offset, assert(ctl); + if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + return -EBADFD; + snd_ctl_elem_id_set_interface(id, SND_CTL_ELEM_IFACE_MIXER); pa_threaded_mainloop_lock(ctl->p->mainloop); @@ -260,7 +265,9 @@ static int pulse_get_attribute(snd_ctl_ext_t * ext, snd_ctl_ext_key_t key, return -EINVAL; assert(ctl); - assert(ctl->p); + + if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + return -EBADFD; pa_threaded_mainloop_lock(ctl->p->mainloop); @@ -311,7 +318,9 @@ static int pulse_read_integer(snd_ctl_ext_t * ext, snd_ctl_ext_key_t key, pa_cvolume *vol = NULL; assert(ctl); - assert(ctl->p); + + if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + return -EBADFD; pa_threaded_mainloop_lock(ctl->p->mainloop); @@ -361,7 +370,9 @@ static int pulse_write_integer(snd_ctl_ext_t * ext, snd_ctl_ext_key_t key, pa_cvolume *vol = NULL; assert(ctl); - assert(ctl->p && ctl->p->context); + + if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + return -EBADFD; pa_threaded_mainloop_lock(ctl->p->mainloop); @@ -465,6 +476,9 @@ static void pulse_subscribe_events(snd_ctl_ext_t * ext, int subscribe) assert(ctl); + if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + return; + pa_threaded_mainloop_lock(ctl->p->mainloop); ctl->subscribed = !!(subscribe & SND_CTL_EVENT_MASK_VALUE); @@ -481,6 +495,9 @@ static int pulse_read_event(snd_ctl_ext_t * ext, snd_ctl_elem_id_t * id, assert(ctl); + if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + return -EBADFD; + pa_threaded_mainloop_lock(ctl->p->mainloop); if (!ctl->updated || !ctl->subscribed) @@ -526,7 +543,9 @@ static int pulse_ctl_poll_revents(snd_ctl_ext_t * ext, struct pollfd *pfd, int err = 0; assert(ctl); - assert(ctl->p); + + if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + return -EBADFD; pa_threaded_mainloop_lock(ctl->p->mainloop); diff --git a/pulse/pcm_pulse.c b/pulse/pcm_pulse.c index c276839..24347f9 100644 --- a/pulse/pcm_pulse.c +++ b/pulse/pcm_pulse.c @@ -106,6 +106,9 @@ static int update_active(snd_pcm_pulse_t *pcm) { assert(pcm); + if (!pcm->p) + return -EBADFD; + ret = check_active(pcm); if (ret < 0) return ret; @@ -125,7 +128,9 @@ static int pulse_start(snd_pcm_ioplug_t * io) int err = 0, err_o = 0, err_u = 0; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); @@ -174,7 +179,9 @@ static int pulse_stop(snd_pcm_ioplug_t * io) int err = 0, err_o = 0, err_u = 0; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); @@ -224,7 +231,9 @@ static int pulse_drain(snd_pcm_ioplug_t * io) int err = 0; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); @@ -259,7 +268,9 @@ static snd_pcm_sframes_t pulse_pointer(snd_pcm_ioplug_t * io) snd_pcm_sframes_t ret = 0; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return -EBADFD; if (io->state == SND_PCM_STATE_XRUN) return -EPIPE; @@ -269,7 +280,10 @@ static snd_pcm_sframes_t pulse_pointer(snd_pcm_ioplug_t * io) pa_threaded_mainloop_lock(pcm->p->mainloop); - assert(pcm->stream); + if (!pcm->stream) { + ret = -EBADFD; + goto finish; + } ret = pulse_check_connection(pcm->p); if (ret < 0) @@ -305,11 +319,16 @@ static int pulse_delay(snd_pcm_ioplug_t * io, snd_pcm_sframes_t * delayp) pa_usec_t lat = 0; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); - assert(pcm->stream); + if (!pcm->stream) { + err = -EBADFD; + goto finish; + } for (;;) { err = pulse_check_connection(pcm->p); @@ -354,11 +373,16 @@ static snd_pcm_sframes_t pulse_write(snd_pcm_ioplug_t * io, snd_pcm_sframes_t ret = 0; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); - assert(pcm->stream); + if (!pcm->stream) { + ret = -EBADFD; + goto finish; + } ret = pulse_check_connection(pcm->p); if (ret < 0) @@ -409,11 +433,16 @@ static snd_pcm_sframes_t pulse_read(snd_pcm_ioplug_t * io, snd_pcm_sframes_t ret = 0; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); - assert(pcm->stream); + if (!pcm->stream) { + ret = -EBADFD; + goto finish; + } ret = pulse_check_connection(pcm->p); if (ret < 0) @@ -480,7 +509,9 @@ static void stream_request_cb(pa_stream * p, size_t length, void *userdata) snd_pcm_pulse_t *pcm = userdata; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return; update_active(pcm); } @@ -490,7 +521,9 @@ static void stream_underrun_cb(pa_stream * p, void *userdata) snd_pcm_pulse_t *pcm = userdata; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return; pcm->underrun = 1; } @@ -499,7 +532,9 @@ static void stream_latency_cb(pa_stream *p, void *userdata) { snd_pcm_pulse_t *pcm = userdata; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return; pa_threaded_mainloop_signal(pcm->p->mainloop, 0); } @@ -512,7 +547,9 @@ static int pulse_pcm_poll_revents(snd_pcm_ioplug_t * io, snd_pcm_pulse_t *pcm = io->private_data; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); @@ -541,7 +578,9 @@ static int pulse_prepare(snd_pcm_ioplug_t * io) unsigned c, d; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); @@ -645,7 +684,9 @@ static int pulse_hw_params(snd_pcm_ioplug_t * io, int err = 0; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); @@ -745,7 +786,9 @@ static int pulse_pause(snd_pcm_ioplug_t * io, int enable) int err = 0; assert (pcm); - assert (pcm->p); + + if (!pcm->p) + return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); diff --git a/pulse/pulse.c b/pulse/pulse.c index 3940238..95d8dde 100644 --- a/pulse/pulse.c +++ b/pulse/pulse.c @@ -32,8 +32,9 @@ int pulse_check_connection(snd_pulse_t * p) pa_context_state_t state; assert(p); - assert(p->context); - assert(p->mainloop); + + if (!p->context || !p->mainloop) + return -EBADFD; state = pa_context_get_state(p->context); @@ -77,8 +78,12 @@ int pulse_wait_operation(snd_pulse_t * p, pa_operation * o) { assert(p); assert(o); - assert(p->state == PULSE_STATE_READY); - assert(p->mainloop); + + if (p->state != PULSE_STATE_READY) + return -EBADFD; + + if (!p->mainloop) + return -EBADFD; for (;;) { int err; @@ -103,8 +108,12 @@ int pulse_wait_stream_state(snd_pulse_t * p, pa_stream * stream, assert(p); assert(stream); - assert(p->state == PULSE_STATE_READY); - assert(p->mainloop); + + if (p->state != PULSE_STATE_READY) + return -EBADFD; + + if (!p->mainloop) + return -EBADFD; for (;;) { int err; @@ -197,7 +206,9 @@ snd_pulse_t *pulse_new(void) p->context = pa_context_new(pa_threaded_mainloop_get_api(p->mainloop), buf); - assert(p->context); + + if (!p->context) + goto fail; pa_context_set_state_callback(p->context, context_state_cb, p); @@ -246,9 +257,12 @@ int pulse_connect(snd_pulse_t * p, const char *server) int err; assert(p); - assert(p->context); - assert(p->mainloop); - assert(p->state == PULSE_STATE_INIT); + + if (!p->context || !p->mainloop) + return -EBADFD; + + if (p->state != PULSE_STATE_INIT) + return -EBADFD; pa_threaded_mainloop_lock(p->mainloop); -- 1.6.3.3 --ikeVEW9yuYc//A+q--

1 0

[alsa-devel] DMA HW bug in MX2x platforms?
by javier Martin 31 Jul '09

31 Jul '09

I am currently working on an ASoC driver for mx2x platforms. I am porting an old working ASoC driver for 2.6.22 + Freescale BSP kernel to current 2.6.30 kernel. Among other things, I had to translate old Freescale's DMA api to the one provided in 2.6.30 for i.mx2x platforms. Here I will describe the problem that I am having: in "aplay" I am using (buffer_size/period_size=4), each (2*period_size) samples I have some erroneous ones that should come (buffer_size) samples later, and this relation is valid for several values with (buffer_size/period_size=4). I have a trace log with all details for DMA transfers that I have compared for both 2.6.22 and new 2.6.30 and they are identical. Some people has also told me that there could be a problem with DMA in mx2x platforms. Do you have knowledge of this kind of problem? Is there a new DMA version for mx2x in any git repository? -- Javier Martin Vista Silicon S.L. Universidad de Cantabria CDTUC - FASE C - Oficina S-345 Avda de los Castros s/n 39005- Santander. Cantabria. Spain +34 942 25 32 60 www.vista-silicon.com

3 2

[alsa-devel] [PATCH 5/5] sound/aoa: Add kmalloc NULL tests
by Julia Lawall 31 Jul '09

31 Jul '09

From: Julia Lawall <julia(a)diku.dk> Check that the result of kzalloc is not NULL before a dereference. The semantic match that finds this problem is as follows: (http://www.emn.fr/x-info/coccinelle/) // <smpl> @@ expression *x; identifier f; constant char *C; @@ x = $kmalloc\|kcalloc\|kzalloc$(...); ... when != x == NULL when != x != NULL when != (x || ...) ( kfree(x) | f(...,C,...,x,...) | *f(...,x,...) | *x->f ) // </smpl> Signed-off-by: Julia Lawall <julia(a)diku.dk> --- sound/aoa/core/gpio-pmf.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/aoa/core/gpio-pmf.c b/sound/aoa/core/gpio-pmf.c index 5ca2220..b4439ce 100644 --- a/sound/aoa/core/gpio-pmf.c +++ b/sound/aoa/core/gpio-pmf.c @@ -182,6 +182,12 @@ static int pmf_set_notify(struct gpio_runtime *rt, if (!old && notify) { irq_client = kzalloc(sizeof(struct pmf_irq_client), GFP_KERNEL); + if (!irq_client) { + err = -ENOMEM; + printk(KERN_ERR "snd-aoa: gpio layer failed to" + " register %s irq (%d)\n", name, err); + goto out_unlock; + } irq_client->data = notif; irq_client->handler = pmf_handle_notify_irq; irq_client->owner = THIS_MODULE;

3 6

[alsa-devel] Huawei K3715 UMTS stick and alsa
by Artem Makhutov 31 Jul '09

31 Jul '09

Hello, i am wondering if it is possible to write a quirk in usbquirks.h for a Huawei K3715 UMTS stick to make the audio interfaces available in linux. This two interfaces are used for audio (I have sniffed in Windows): bEndpointAddress 0x83 EP 3 IN bEndpointAddress 0x02 EP 2 OUT I have attached a lsusb -vv. The payload of this interfaces is 320 bytes. Here is an example payload data. What kind of audio could this be? 93 06 1D 01 BD F6 17 EA 94 E3 DA DB 1C E2 E1 FA 23 0C 13 0B FD 08 78 11 51 17 02 13 C4 04 F5 F2 CA EC 03 F9 5F 09 1E 15 4D 1B B2 0A B7 F5 3D EE A4 E2 62 E5 D2 FA 23 07 26 05 B2 09 6A 20 D8 22 97 0B 77 F7 3F EE C0 E8 48 E1 11 DB 55 E7 E9 05 AC 24 57 28 FF 1B D5 0E 45 07 DC 06 26 07 FE FC 17 ED C8 E0 3F D4 3D E0 8B 03 75 13 0F 04 D3 F2 FF F4 31 08 A2 09 A6 01 FF FA 14 01 49 10 D3 1A 63 25 27 1F C5 00 4C E8 15 E0 0A D8 E3 DE 53 F0 D5 F2 BD F9 F5 10 8B 26 F1 20 55 0C 3E 00 56 FA C8 F5 E8 F0 39 ED F5 F1 F7 00 66 13 7F 1A 67 0F 70 01 D1 FC 71 01 63 09 61 05 0B F7 7F EB 6E E8 CA F6 D2 07 BF 0B 75 FC 70 F0 10 F5 44 06 ED 08 0B 05 92 FE 2F FD 95 0B 40 16 05 22 A7 16 A5 FA 08 E9 52 E5 B5 D9 1F E3 F7 FB 9C 02 6E 03 EF 09 76 1C E1 19 50 06 02 F6 20 F0 8A F0 99 EE 7E E6 42 EB EF 01 AA 1C FF 20 73 14 F0 06 D4 01 41 05 F4 07 EF 00 2E EF D8 E4 12 E2 55 F5 DE 03 C3 09 EF 04 A4 FB 93 F9 77 FE F9 03 CC 07 14 07 BF 0B CF 16 26 1A 6B 16 CC 05 CC F7 FA ED D9 E1 DA D5 Thanks, Artem

2 3

[alsa-devel] [PATCH] Enable HP output on nVidia Corporation MCP79 HDA (rev b1) (Macbook Pro 5, 5)
by Stelian Pop 30 Jul '09

30 Jul '09

Hi, The patch below, to be applied on the latest sound-unstable-2.6.git, enables headphones output on my MacBookPro 5,5, together with the automuting feature. Here is the exact soundcard id: Vendor Id: 0x10134206 Subsystem Id: 0x106b4d00 Revision Id: 0x100301 Thanks for the previous efforts to make this work ! Stelian. diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index f552738..09be633 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -832,6 +832,12 @@ static void cs_automute(struct hda_codec *codec) AC_VERB_SET_PIN_WIDGET_CONTROL, hp_present ? 0 : PIN_OUT); } + if (spec->board_config == CS420X_MBP55) { + if (hp_present) + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 0x2); + else + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 0x8); + } } static void cs_automic(struct hda_codec *codec) @@ -1133,10 +1139,10 @@ static int patch_cs420x(struct hda_codec *codec) switch (spec->board_config) { case CS420X_MBP55: - /* GPIO3 = EAPD? */ - spec->gpio_mask = 0x08; - spec->gpio_dir = 0x08; - spec->gpio_data = 0x08; + /* GPIO1 = headphones */ + /* GPIO3 = speakers */ + spec->gpio_mask = 0x0a; + spec->gpio_dir = 0x0a; break; } -- Stelian Pop <stelian(a)popies.net>

2 4

[PATCH] [ALSA] hda: Correct EAPD for Dell Inspiron 1525
by Chengu Wang 30 Jul '09

30 Jul '09

The commit 24918b61b55c21e09a3e07cd82e1b3a8154782dc statically changes the model from dell-bios to dell-3stack to solve the sound decreasing regression (http://lkml.org/lkml/2008/9/12/203), however it leads to another problem that the 2nd headphone jack doesn't work (https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3987). So I think the commit 249**2dc is just a workaround. I would like to give a true solution here. The datasheet for STAC9228 says, GPIO2 is the same pin as VOL DOWN, and the EAPD pin is GPIO0. This is why the sound decreases if we set EAPD as GPIO2. This patch changes EAPD to GPIO0 to solve the problem. Signed-off-by: Chengu Wang <wangchengu(a)gmail.com> --- sound/pci/hda/patch_sigmatel.c | 9 ++++++++- 1 files changed, 8 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 8c48f86..3857add 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2264,7 +2264,7 @@ static struct snd_pci_quirk stac927x_cfg_tbl[] = { SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f3, "Dell Inspiron 1420", STAC_DELL_BIOS), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0227, "Dell Vostro 1400 ", STAC_DELL_BIOS), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x022e, "Dell ", STAC_DELL_BIOS), - SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x022f, "Dell Inspiron 1525", STAC_DELL_3ST), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x022f, "Dell Inspiron 1525", STAC_DELL_BIOS), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0242, "Dell ", STAC_DELL_BIOS), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0243, "Dell ", STAC_DELL_BIOS), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02ff, "Dell ", STAC_DELL_BIOS), @@ -5659,6 +5659,13 @@ static int patch_stac927x(struct hda_codec *codec) /* GPIO2 High = Enable EAPD */ spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x04; spec->gpio_data = 0x04; + switch (codec->subsystem_id) { + case 0x1028022f: + /* correct EAPD to be GPIO0 */ + spec->eapd_mask = spec->gpio_mask = 0x01; + spec->gpio_dir = spec->gpio_data = 0x01; + break; + }; spec->dmic_nids = stac927x_dmic_nids; spec->num_dmics = STAC927X_NUM_DMICS; -- 1.6.0.4 --001636457ed0b1bcc70470129999 Content-Type: text/x-patch; charset=US-ASCII; name="0001--ALSA-hda-Correct-EAPD-for-Dell-Inspiron-1525.patch" Content-Disposition: attachment; filename="0001--ALSA-hda-Correct-EAPD-for-Dell-Inspiron-1525.patch" Content-Transfer-Encoding: base64 X-Attachment-Id: f_fxu8s8l10 RnJvbSAzZGZiNTQ2NjZkMDk0MGJmZjQ3NGJlOWRiNzkxNjAzZjgxYzQ2MzAzIE1vbiBTZXAgMTcg MDA6MDA6MDAgMjAwMQpGcm9tOiBDaGVuZ3UgV2FuZyA8d2FuZ2NoZW5ndUBnbWFpbC5jb20+CkRh dGU6IFRodSwgMzAgSnVsIDIwMDkgMTk6NDM6NTUgKzA4MDAKU3ViamVjdDogW1BBVENIXSBbQUxT QV0gaGRhOiBDb3JyZWN0IEVBUEQgZm9yIERlbGwgSW5zcGlyb24gMTUyNQoKVGhlIGNvbW1pdCAy NDkxOGI2MWI1NWMyMWUwOWEzZTA3Y2Q4MmUxYjNhODE1NDc4MmRjIHN0YXRpY2FsbHkgY2hhbmdl cwp0aGUgbW9kZWwgZnJvbSBkZWxsLWJpb3MgdG8gZGVsbC0zc3RhY2sgdG8gc29sdmUgdGhlIHNv dW5kIGRlY3JlYXNpbmcKcmVncmVzc2lvbiAoaHR0cDovL2xrbWwub3JnL2xrbWwvMjAwOC85LzEy LzIwMyksIGhvd2V2ZXIgaXQgbGVhZHMgdG8gYW5vdGhlcgpwcm9ibGVtIHRoYXQgdGhlIDJuZCBo ZWFkcGhvbmUgamFjayBkb2Vzbid0IHdvcmsKKGh0dHBzOi8vYnVndHJhY2suYWxzYS1wcm9qZWN0 Lm9yZy9hbHNhLWJ1Zy92aWV3LnBocD9pZD0zOTg3KS4gU28gSSB0aGluawp0aGUgY29tbWl0IDI0 OSoqMmRjIGlzIGp1c3QgYSB3b3JrYXJvdW5kLiBJIHdvdWxkIGxpa2UgdG8gZ2l2ZSBhIHRydWUg c29sdXRpb24KaGVyZS4KClRoZSBkYXRhc2hlZXQgZm9yIFNUQUM5MjI4IHNheXMsIEdQSU8yIGlz IHRoZSBzYW1lIHBpbiBhcyBWT0wgRE9XTiwgYW5kCnRoZSBFQVBEIHBpbiBpcyBHUElPMC4gVGhp cyBpcyB3aHkgdGhlIHNvdW5kIGRlY3JlYXNlcyBpZiB3ZSBzZXQgRUFQRCBhcwpHUElPMi4gVGhp cyBwYXRjaCBjaGFuZ2VzIEVBUEQgdG8gR1BJTzAgdG8gc29sdmUgdGhlIHByb2JsZW0uCgpTaWdu ZWQtb2ZmLWJ5OiBDaGVuZ3UgV2FuZyA8d2FuZ2NoZW5ndUBnbWFpbC5jb20+Ci0tLQogc291bmQv cGNpL2hkYS9wYXRjaF9zaWdtYXRlbC5jIHwgICAgOSArKysrKysrKy0KIDEgZmlsZXMgY2hhbmdl ZCwgOCBpbnNlcnRpb25zKCspLCAxIGRlbGV0aW9ucygtKQoKZGlmZiAtLWdpdCBhL3NvdW5kL3Bj aS9oZGEvcGF0Y2hfc2lnbWF0ZWwuYyBiL3NvdW5kL3BjaS9oZGEvcGF0Y2hfc2lnbWF0ZWwuYwpp bmRleCA4YzQ4Zjg2Li4zODU3YWRkIDEwMDY0NAotLS0gYS9zb3VuZC9wY2kvaGRhL3BhdGNoX3Np Z21hdGVsLmMKKysrIGIvc291bmQvcGNpL2hkYS9wYXRjaF9zaWdtYXRlbC5jCkBAIC0yMjY0LDcg KzIyNjQsNyBAQCBzdGF0aWMgc3RydWN0IHNuZF9wY2lfcXVpcmsgc3RhYzkyN3hfY2ZnX3RibFtd ID0gewogCVNORF9QQ0lfUVVJUksoUENJX1ZFTkRPUl9JRF9ERUxMLCAgMHgwMWYzLCAiRGVsbCBJ bnNwaXJvbiAxNDIwIiwgU1RBQ19ERUxMX0JJT1MpLAogCVNORF9QQ0lfUVVJUksoUENJX1ZFTkRP Ul9JRF9ERUxMLCAgMHgwMjI3LCAiRGVsbCBWb3N0cm8gMTQwMCAgIiwgU1RBQ19ERUxMX0JJT1Mp LAogCVNORF9QQ0lfUVVJUksoUENJX1ZFTkRPUl9JRF9ERUxMLCAgMHgwMjJlLCAiRGVsbCAgICAg IiwgU1RBQ19ERUxMX0JJT1MpLAotCVNORF9QQ0lfUVVJUksoUENJX1ZFTkRPUl9JRF9ERUxMLCAg MHgwMjJmLCAiRGVsbCBJbnNwaXJvbiAxNTI1IiwgU1RBQ19ERUxMXzNTVCksCisJU05EX1BDSV9R VUlSSyhQQ0lfVkVORE9SX0lEX0RFTEwsICAweDAyMmYsICJEZWxsIEluc3Bpcm9uIDE1MjUiLCBT VEFDX0RFTExfQklPUyksCiAJU05EX1BDSV9RVUlSSyhQQ0lfVkVORE9SX0lEX0RFTEwsICAweDAy NDIsICJEZWxsICAgICAiLCBTVEFDX0RFTExfQklPUyksCiAJU05EX1BDSV9RVUlSSyhQQ0lfVkVO RE9SX0lEX0RFTEwsICAweDAyNDMsICJEZWxsICAgICAiLCBTVEFDX0RFTExfQklPUyksCiAJU05E X1BDSV9RVUlSSyhQQ0lfVkVORE9SX0lEX0RFTEwsICAweDAyZmYsICJEZWxsICAgICAiLCBTVEFD X0RFTExfQklPUyksCkBAIC01NjU5LDYgKzU2NTksMTMgQEAgc3RhdGljIGludCBwYXRjaF9zdGFj OTI3eChzdHJ1Y3QgaGRhX2NvZGVjICpjb2RlYykKIAkJLyogR1BJTzIgSGlnaCA9IEVuYWJsZSBF QVBEICovCiAJCXNwZWMtPmVhcGRfbWFzayA9IHNwZWMtPmdwaW9fbWFzayA9IHNwZWMtPmdwaW9f ZGlyID0gMHgwNDsKIAkJc3BlYy0+Z3Bpb19kYXRhID0gMHgwNDsKKwkJc3dpdGNoIChjb2RlYy0+ c3Vic3lzdGVtX2lkKSB7CisJCWNhc2UgMHgxMDI4MDIyZjoKKwkJCS8qIGNvcnJlY3QgRUFQRCB0 byBiZSBHUElPMCAqLworCQkJc3BlYy0+ZWFwZF9tYXNrID0gc3BlYy0+Z3Bpb19tYXNrID0gMHgw MTsKKwkJCXNwZWMtPmdwaW9fZGlyID0gc3BlYy0+Z3Bpb19kYXRhID0gMHgwMTsKKwkJCWJyZWFr OworCQl9OwogCQlzcGVjLT5kbWljX25pZHMgPSBzdGFjOTI3eF9kbWljX25pZHM7CiAJCXNwZWMt Pm51bV9kbWljcyA9IFNUQUM5MjdYX05VTV9ETUlDUzsKIAotLSAKMS42LjAuNAoK --001636457ed0b1bcc70470129999--

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[alsa-devel] [PATCH] ALSA: Write outside array bounds
by Roel Kluin 30 Jul '09

30 Jul '09

e->sad[] is declared with size ELD_MAX_SAD (16), but the guard allows the range 0-31 Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- Found with Parfait, http://research.sun.com/projects/parfait/ diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index fcad5ec..ec04e58 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -539,7 +539,7 @@ static void hdmi_write_eld_info(struct snd_info_entry *entry, sname++; n = 10 * n + name[4] - '0'; } - if (n < 0 || n > 31) /* double the CEA limit */ + if (n < 0 || n > ELD_MAX_SAD) continue; if (!strcmp(sname, "_coding_type")) e->sad[n].format = val;

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[alsa-devel] [PATCH] hda: add HP automute support to Intel ALC889/ALC889A models
by Wu Fengguang 30 Jul '09

30 Jul '09

It auto mutes all 8-channel outputs at rear panel when the front panel headphone is connected. Signed-off-by: Wu Fengguang <fengguang.wu(a)intel.com> --- sound/pci/hda/patch_realtek.c | 42 ++++++++++++++++++++++++-------- 1 file changed, 32 insertions(+), 10 deletions(-) --- sound-2.6.orig/sound/pci/hda/patch_realtek.c +++ sound-2.6/sound/pci/hda/patch_realtek.c @@ -1436,6 +1436,25 @@ static void alc_automute_amp_unsol_event alc_automute_amp(codec); } +static void alc889_automute_init(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x14; + spec->autocfg.speaker_pins[1] = 0x16; + spec->autocfg.speaker_pins[2] = 0x17; + spec->autocfg.speaker_pins[3] = 0x19; + spec->autocfg.speaker_pins[4] = 0x1a; + alc_automute_amp(codec); +} + +static void alc889_intel_init_hook(struct hda_codec *codec) +{ + alc889_coef_init(codec); + alc889_automute_init(codec); +} + static void alc888_fujitsu_xa3530_init_hook(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -6994,6 +7013,11 @@ static struct hda_verb alc889_eapd_verbs { } }; +static struct hda_verb alc262_hippo_unsol_verbs[] = { + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {} +}; static struct hda_verb alc885_init_verbs[] = { /* Front mixer: unmute input/output amp left and right (volume = 0) */ @@ -7019,7 +7043,7 @@ static struct hda_verb alc885_init_verbs {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* Front HP Pin: output 0 (0x0c) */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front Pin: output 0 (0x0c) */ @@ -8861,7 +8885,8 @@ static struct alc_config_preset alc882_p }, [ALC889A_INTEL] = { .mixers = { alc885_8ch_intel_mixer, alc883_chmode_mixer }, - .init_verbs = { alc885_init_verbs, alc885_init_input_verbs }, + .init_verbs = { alc885_init_verbs, alc885_init_input_verbs, + alc262_hippo_unsol_verbs }, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .num_adc_nids = ARRAY_SIZE(alc889_adc_nids), @@ -8873,12 +8898,14 @@ static struct alc_config_preset alc882_p .channel_mode = alc889_8ch_intel_modes, .capsrc_nids = alc889_capsrc_nids, .input_mux = &alc889_capture_source, + .init_hook = alc889_automute_init, + .unsol_event = alc_automute_amp_unsol_event, .need_dac_fix = 1, }, [ALC889_INTEL] = { .mixers = { alc885_8ch_intel_mixer, alc883_chmode_mixer }, .init_verbs = { alc885_init_verbs, alc889_init_input_verbs, - alc889_eapd_verbs }, + alc889_eapd_verbs, alc262_hippo_unsol_verbs}, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .num_adc_nids = ARRAY_SIZE(alc889_adc_nids), @@ -8890,7 +8917,8 @@ static struct alc_config_preset alc882_p .channel_mode = alc889_8ch_intel_modes, .capsrc_nids = alc889_capsrc_nids, .input_mux = &alc889_capture_source, - .init_hook = alc889_coef_init, + .init_hook = alc889_intel_init_hook, + .unsol_event = alc_automute_amp_unsol_event, .need_dac_fix = 1, }, [ALC883_6ST_DIG] = { @@ -10123,12 +10151,6 @@ static struct hda_verb alc262_eapd_verbs { } }; -static struct hda_verb alc262_hippo_unsol_verbs[] = { - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {} -}; - static struct hda_verb alc262_hippo1_unsol_verbs[] = { {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0}, {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},

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[alsa-devel] [PATCH] hda: add 2-channel mode to Intel ALC889/ALC889A models
by Wu Fengguang 30 Jul '09

30 Jul '09

This 2-channel mode is useful in that it will broadcast a 2-channel audio stream to all front/side/... ports. Signed-off-by: Wu Fengguang <fengguang.wu(a)intel.com> --- sound/pci/hda/patch_realtek.c | 44 ++++++++++++++++++-------------- 1 file changed, 25 insertions(+), 19 deletions(-) --- sound-2.6.orig/sound/pci/hda/patch_realtek.c +++ sound-2.6/sound/pci/hda/patch_realtek.c @@ -6682,18 +6682,27 @@ static struct hda_channel_mode alc883_3S }; /* + * 2ch mode + */ +static struct hda_verb alc889_ch2_intel_init[] = { + { 0x14, AC_VERB_SET_CONNECT_SEL, 0x00 }, + { 0x19, AC_VERB_SET_CONNECT_SEL, 0x00 }, + { 0x16, AC_VERB_SET_CONNECT_SEL, 0x00 }, + { 0x17, AC_VERB_SET_CONNECT_SEL, 0x00 }, + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { } /* end */ +}; + +/* * 6ch mode */ static struct hda_verb alc889_ch6_intel_init[] = { - { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, + { 0x14, AC_VERB_SET_CONNECT_SEL, 0x00 }, + { 0x19, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { 0x16, AC_VERB_SET_CONNECT_SEL, 0x02 }, + { 0x17, AC_VERB_SET_CONNECT_SEL, 0x03 }, + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, { } /* end */ }; @@ -6702,21 +6711,18 @@ static struct hda_verb alc889_ch6_intel_ * 8ch mode */ static struct hda_verb alc889_ch8_intel_init[] = { - { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x14, AC_VERB_SET_CONNECT_SEL, 0x00 }, + { 0x19, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { 0x16, AC_VERB_SET_CONNECT_SEL, 0x02 }, + { 0x17, AC_VERB_SET_CONNECT_SEL, 0x03 }, + { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x03 }, { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x03 }, { } /* end */ }; -static struct hda_channel_mode alc889_8ch_intel_modes[2] = { +static struct hda_channel_mode alc889_8ch_intel_modes[3] = { + { 2, alc889_ch2_intel_init }, { 6, alc889_ch6_intel_init }, { 8, alc889_ch8_intel_init }, };

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[alsa-devel] Chat 50 at usb-0000:00:1a.7-4.3 audio issues
by Jason Bell 30 Jul '09

30 Jul '09

G'day ASLA Developers I am hoping that someone might be able to assist me with an audio issue I am having. Recently I have begun using the Dell T3500 precisions as an Video Conferencing machine (using the Access Grid and EVO Software). Information on hardware specs, etc can be found at http://www.alsa-project.org/db/?f=7f33770c89802b0ab3132f21914cc6ca7f6626 37 Unfortunately, I have issues when using either a clearone Chat50 or Chat150 (these are essentially speaker phone devices [See http://www.clearone.com/products/product.php?cat=9&prod=94 for device description]) with the speaker audio being distorted and static, cracks and popping type sounds being introduced. The microphone audio quality is fine. These issues were seen using Fedora 10 and Fedora 11. I tried upgrading to the latest ALSA drivers and "testing" kernels without any success (on a previous Fedora 10 install). It was found (still in the initial stages of testings) that if I plugged the Chat50 device into a USB hub, rather than directly into the Machine, the audio issues disappeared (at least for some of the software). It should also be noted that I have a number of Dell T3400 series machines that don't experience these issues. Additionally is should be mentioned that the T3400 uses the ICH9 chipset family whereas the T3500 uses the ICH10 chipset family which includes the USB controller. Just wondering if there is anything I can do to resolve this issue? Many thanks in advance, Jason. -------------------------------------------- Jason Bell, B.I.T. (Honours) Research Systems Support Officer Information Technology Division CQ University Australia Australian Research Collaboration Service http://www.arcs.org.au/ E-mail : j.bell(a)cqu.edu.au jason.bell(a)arcs.org.au Work : +61 7 4930 9229 Mobile : 0409 630897 Postal : Building 19 Central Queensland University Bruce Highway Rockhampton, Queensland, Australia, 4702 -------------------------------------------- Patience is a virtue. But if I wanted Patience, I would have become a Doctor. --------------------------------------------

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[alsa-devel] [PATCH] Fix checkpatch issues and typos of ad1938 codec and bf5xx-tdm dai
by Barry Song 29 Jul '09

29 Jul '09

1. fix "line over 80 characters" checkpatch warnings 2. ‘DMA_nnBIT_MASK’ is deprecated, use DMA_BIT_MASK instead 3. fix typos Signed-off-by: Barry Song <21cnbao(a)gmail.com> --- sound/soc/blackfin/bf5xx-tdm-pcm.c | 4 +- sound/soc/blackfin/bf5xx-tdm-pcm.h | 2 +- sound/soc/blackfin/bf5xx-tdm.h | 2 +- sound/soc/codecs/ad1938.c | 50 +++++++++++++++++++++++------------ 4 files changed, 37 insertions(+), 21 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.c b/sound/soc/blackfin/bf5xx-tdm-pcm.c index af7fa8d..ccb5e82 100644 --- a/sound/soc/blackfin/bf5xx-tdm-pcm.c +++ b/sound/soc/blackfin/bf5xx-tdm-pcm.c @@ -276,7 +276,7 @@ static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm) sport_done(sport_handle); } -static u64 bf5xx_pcm_dmamask = DMA_32BIT_MASK; +static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32); static int bf5xx_pcm_tdm_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) @@ -286,7 +286,7 @@ static int bf5xx_pcm_tdm_new(struct snd_card *card, struct snd_soc_dai *dai, if (!card->dev->dma_mask) card->dev->dma_mask = &bf5xx_pcm_dmamask; if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = DMA_32BIT_MASK; + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); if (dai->playback.channels_min) { ret = bf5xx_pcm_preallocate_dma_buffer(pcm, diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.h b/sound/soc/blackfin/bf5xx-tdm-pcm.h index 9e8473d..ddc5047 100644 --- a/sound/soc/blackfin/bf5xx-tdm-pcm.h +++ b/sound/soc/blackfin/bf5xx-tdm-pcm.h @@ -1,5 +1,5 @@ /* - * linux/sound/arm/bf5xx-tdm-pcm.h -- ALSA PCM interface for the Blackfin + * sound/soc/blackfin/bf5xx-tdm-pcm.h -- ALSA PCM interface for the Blackfin * * Copyright 2009 Analog Device Inc. * diff --git a/sound/soc/blackfin/bf5xx-tdm.h b/sound/soc/blackfin/bf5xx-tdm.h index e5157cd..618ec3d 100644 --- a/sound/soc/blackfin/bf5xx-tdm.h +++ b/sound/soc/blackfin/bf5xx-tdm.h @@ -1,5 +1,5 @@ /* - * linux/sound/arm/bf5xx-i2s.h + * sound/soc/blackfin/bf5xx-tdm.h * * 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 diff --git a/sound/soc/codecs/ad1938.c b/sound/soc/codecs/ad1938.c index ffc97c9..aafda74 100644 --- a/sound/soc/codecs/ad1938.c +++ b/sound/soc/codecs/ad1938.c @@ -72,14 +72,20 @@ static const struct snd_kcontrol_new ad1938_snd_controls[] = { AD1938_DAC_R4_VOL, 0, 0xFF, 1), /* ADC switch control */ - SOC_DOUBLE("ADC1 Switch", AD1938_ADC_CTRL0, AD1938_ADCL1_MUTE, AD1938_ADCR1_MUTE, 1, 1), - SOC_DOUBLE("ADC2 Switch", AD1938_ADC_CTRL0, AD1938_ADCL2_MUTE, AD1938_ADCR2_MUTE, 1, 1), + SOC_DOUBLE("ADC1 Switch", AD1938_ADC_CTRL0, AD1938_ADCL1_MUTE, + AD1938_ADCR1_MUTE, 1, 1), + SOC_DOUBLE("ADC2 Switch", AD1938_ADC_CTRL0, AD1938_ADCL2_MUTE, + AD1938_ADCR2_MUTE, 1, 1), /* DAC switch control */ - SOC_DOUBLE("DAC1 Switch", AD1938_DAC_CHNL_MUTE, AD1938_DACL1_MUTE, AD1938_DACR1_MUTE, 1, 1), - SOC_DOUBLE("DAC2 Switch", AD1938_DAC_CHNL_MUTE, AD1938_DACL2_MUTE, AD1938_DACR2_MUTE, 1, 1), - SOC_DOUBLE("DAC3 Switch", AD1938_DAC_CHNL_MUTE, AD1938_DACL3_MUTE, AD1938_DACR3_MUTE, 1, 1), - SOC_DOUBLE("DAC4 Switch", AD1938_DAC_CHNL_MUTE, AD1938_DACL4_MUTE, AD1938_DACR4_MUTE, 1, 1), + SOC_DOUBLE("DAC1 Switch", AD1938_DAC_CHNL_MUTE, AD1938_DACL1_MUTE, + AD1938_DACR1_MUTE, 1, 1), + SOC_DOUBLE("DAC2 Switch", AD1938_DAC_CHNL_MUTE, AD1938_DACL2_MUTE, + AD1938_DACR2_MUTE, 1, 1), + SOC_DOUBLE("DAC3 Switch", AD1938_DAC_CHNL_MUTE, AD1938_DACL3_MUTE, + AD1938_DACR3_MUTE, 1, 1), + SOC_DOUBLE("DAC4 Switch", AD1938_DAC_CHNL_MUTE, AD1938_DACL4_MUTE, + AD1938_DACR4_MUTE, 1, 1), /* ADC high-pass filter */ SOC_SINGLE("ADC High Pass Filter Switch", AD1938_ADC_CTRL0, @@ -110,7 +116,8 @@ static int ad1938_mute(struct snd_soc_dai *dai, int mute) int reg; reg = codec->read(codec, AD1938_DAC_CTRL2); - reg = (mute > 0) ? reg | AD1938_DAC_MASTER_MUTE : reg & (~AD1938_DAC_MASTER_MUTE); + reg = (mute > 0) ? reg | AD1938_DAC_MASTER_MUTE : reg & + (~AD1938_DAC_MASTER_MUTE); codec->write(codec, AD1938_DAC_CTRL2, reg); return 0; @@ -119,7 +126,8 @@ static int ad1938_mute(struct snd_soc_dai *dai, int mute) static inline int ad1938_pll_powerctrl(struct snd_soc_codec *codec, int cmd) { int reg = codec->read(codec, AD1938_PLL_CLK_CTRL0); - reg = (cmd > 0) ? reg & (~AD1938_PLL_POWERDOWN) : reg | AD1938_PLL_POWERDOWN; + reg = (cmd > 0) ? reg & (~AD1938_PLL_POWERDOWN) : reg | + AD1938_PLL_POWERDOWN; codec->write(codec, AD1938_PLL_CLK_CTRL0, reg); return 0; @@ -171,8 +179,8 @@ static int ad1938_set_dai_fmt(struct snd_soc_dai *codec_dai, adc_reg = codec->read(codec, AD1938_ADC_CTRL2); dac_reg = codec->read(codec, AD1938_DAC_CTRL1); - /* At present, the driver only support AUX ADC mode(SND_SOC_DAIFMT_I2S with TDM) - * and ADC&DAC TDM mode(SND_SOC_DAIFMT_DSP_A) + /* At present, the driver only support AUX ADC mode(SND_SOC_DAIFMT_I2S + * with TDM) and ADC&DAC TDM mode(SND_SOC_DAIFMT_DSP_A) */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: @@ -363,7 +371,8 @@ static unsigned int ad1938_read_reg_cache(struct snd_soc_codec *codec, * read from the ad1938 register space */ -static unsigned int ad1938_read_reg(struct snd_soc_codec *codec, unsigned int reg) +static unsigned int ad1938_read_reg(struct snd_soc_codec *codec, + unsigned int reg) { char w_buf[AD1938_SPI_BUFLEN]; char r_buf[AD1938_SPI_BUFLEN]; @@ -502,12 +511,19 @@ static int ad1938_register(struct ad1938_priv *ad1938) ad1938_codec = codec; /* default setting for ad1938 */ - codec->write(codec, AD1938_DAC_CHNL_MUTE, 0x0); /* unmute dac channels */ - codec->write(codec, AD1938_DAC_CTRL2, 0x1A); /* de-emphasis: 48kHz, powedown dac */ - codec->write(codec, AD1938_DAC_CTRL0, 0x21); /* powerdown dac, dac tdm mode */ - codec->write(codec, AD1938_ADC_CTRL0, 0x3); /* high-pass filter enable */ - codec->write(codec, AD1938_ADC_CTRL1, 0x43); /* sata delay=1, adc aux mode */ - codec->write(codec, AD1938_PLL_CLK_CTRL0, 0x9D); /* pll input:mclki/xi */ + + /* unmute dac channels */ + codec->write(codec, AD1938_DAC_CHNL_MUTE, 0x0); + /* de-emphasis: 48kHz, powedown dac */ + codec->write(codec, AD1938_DAC_CTRL2, 0x1A); + /* powerdown dac, dac in tdm mode */ + codec->write(codec, AD1938_DAC_CTRL0, 0x41); + /* high-pass filter enable */ + codec->write(codec, AD1938_ADC_CTRL0, 0x3); + /* sata delay=1, adc aux mode */ + codec->write(codec, AD1938_ADC_CTRL1, 0x43); + /* pll input: mclki/xi */ + codec->write(codec, AD1938_PLL_CLK_CTRL0, 0x9D); codec->write(codec, AD1938_PLL_CLK_CTRL1, 0x04); ad1938_fill_cache(codec); -- 1.5.6.3

2 1

[alsa-devel] how to configure .asoundrc to make arecord use a default device
by Jelle de Jong 29 Jul '09

29 Jul '09

Hello everybody, I am having some issues making arecord use the default device for recording. Could somebody help me get a working .asoundrc configuration? I attached all information in the attachment and also made temporary hyperlink: http://debian.pastebin.com/m52601230 It works with: arecord -v -f S16_LE -r 44100 -D audiodevice0 ~/record0.wav but it does not work with: arecord -v -f S16_LE -r 44100 ~/record0.wav background: http://mailman.alsa-project.org/pipermail/alsa-devel/2008-November/012366.h… Thanks in advance, Best regards, Jelle de Jong

2 6

Re: [alsa-devel] clicking noise with Sigmatel STAC9202
by Christian Steffen 29 Jul '09

29 Jul '09

Here I send the two also-info files as attachments. Initial problem was: Under kernel 2.6.30, Sound is working, but has constant clicking noise in the background, also if the sound is completely muted. First I thought this sound would come from some hdd or whatever, but the device has no mechanical parts. But this is what it sounds like - Irregular clicking noise. This noise immediately starts during boot up as soon as the module is loaded and before X11 has started. I already tried options like "position_fix=1 or 2" and different "model="-options with no success. "position_fix" didn't do anything at all and different model-options have muted the device completely. Kernel 2.6.24 with the same files has no issues. >> Then give as attachments later (and don't forget Cc to alsa-devel ML) >> these two files, not URL. It's important because ML is archived >> while pastebin can be removed at any time. Thanks, Christian Steffen -----Ursprüngliche Nachricht----- Von: Takashi Iwai [mailto:tiwai@suse.de] Gesendet: Mittwoch, 29. Juli 2009 17:32 An: Christian Steffen Cc: 'Takashi Iwai' Betreff: Re: AW: [alsa-devel] clicking noise with Sigmatel STAC9202 At Wed, 29 Jul 2009 17:11:18 +0200, Christian Steffen wrote: > > Thanks, > > >> Well, then at least, the contents of /proc/asound/card0/codec#*, > >> /sys/class/sound/hwC0D*/*_pin_configs files (if present), and the file > >> generated via "alsactl -f somefile store". These are the most > >> important information. > > There is no pinconfig file and there is also no alsactl tool present. > I could obtain the output from the codec stuff: > > http://pastebin.com/m1b81f47d (old kernel 2.6.24, only the codec info) > http://pastebin.com/m5baeca9a (new 2.6.30) > > that's all I could get for the older kernel as I only have access using > telnet. Then give as attachments later (and don't forget Cc to alsa-devel ML) these two files, not URL. It's important because ML is archived while pastebin can be removed at any time. Takashi

1 0

[alsa-devel] backward/forward binary compatibility checker
by Andrey Ponomarenko 29 Jul '09

29 Jul '09

Colleagues, I'm software engineer from Institute for System Programing of Russian Academy of Sciences and we are developing a free lightweight tool for checking backward/forward binary compatibility of shared C/C++ libraries in OS Linux. It checks interface signatures and data type definitions in two library versions (headers and shared objects) and searches differences that may lead to incompatibility according to ABI standards. We have released beta version of this tool and we'd like you to consider its usefulness for your project. The package with the latest version of tool is in the attachment. The wiki-page with the latest release of binary compatibility checker is http://ispras.linux-foundation.org/index.php/ABI_compliance_checker Andrey Ponomarenko <susanin(a)ispras.ru>

1 0

[alsa-devel] Master channel is only mono
by Sune Mølgaard 29 Jul '09

29 Jul '09

As per gnubien's advice on #alsa on irc.freenode.net, I have run alsa-info.sh - the URL is http://www.alsa-project.org/db/?f=fbe93f301ca214007e66c521cd6c41a17a604c79 . My problem is that the master channel for my card is shown to be mono only, which I don't think seems right... Also, sound in general is rather low. My system is a very new Dell Studio laptop - I have loaded snd-hda-intel with mode=dell-m6 as per the kernel documentation. Best regards, Sune Mølgaard

2 2

[alsa-devel] pulseaudio workaround for alsa-driver bug
by jack craig 29 Jul '09

29 Jul '09

Hi alsa-developers, I wanted to report my recent issues with audio on fc11. after an update from fc10, my audio started failing. not the failure to year any audio, but instead, it would start off fine and then tail spin. i am using mplayer to play an internet asf stream to the local audio. the stream processing failure is described in https://bugzilla.redhat.com/show_bug.cgi?id=514213 i was able to get a workaround from the #pulseaudio guys, but! they encouraged me to report my experience and its fix to you to encourage the fix of this snd_pcm_delay() defect. if you get a driver fix for this, i am happy to try it out when its ready. thx, jackc... -- jack craig jackc(a)LinuxLightHouse.com 831-684-1375 (Office) 831-596-6924 (cell) IM: jackcraigaptos (AIM) _________________________________ This email has been ClamScanned ! www.LinuxLightHouse.com

3 3

Re: [alsa-devel] clicking noise with Sigmatel STAC9202
by Takashi Iwai 29 Jul '09

29 Jul '09

At Wed, 29 Jul 2009 16:46:34 +0200, Christian Steffen wrote: > > >>>Is the older kernel as is even with the same user-space stuff? > > Probaly yes, I mean I can start the on-device linux with the newer kernel > and also get clicking noise. > But this way I can't run alsa-info.sh as this system has not all the apps it > needs. > > So I started another ubuntu linux on the device to run alsa-info.sh on the > 2.6.30 kernel. > > I've uploaded the result to: > http://pastebin.com/m5baeca9a Please run it with --no-upload on both kernels, and attach both generated files. > Does this already help? > It's difficult to install apps to the linux which is on the device as it is > very very limited. Well, then at least, the contents of /proc/asound/card0/codec#*, /sys/class/sound/hwC0D*/*_pin_configs files (if present), and the file generated via "alsactl -f somefile store". These are the most important information. Takashi

1 0

[alsa-devel] clicking noise with Sigmatel STAC9202
by Christian Steffen 29 Jul '09

29 Jul '09

Hello to all subscribers, I'm new to this list and will start directly with my problem: I'm running some ubuntu linux on a device called "o2 Joggler", which is a digital picture frame with touchscreen control. Its chipsets are from intel, the cpu is an intel atom, and the other hardware devices are controlled by an Intel System Controller Hub "Poulsbo". I'm running kernel 2.6.30 and the newest alsa version 1.0.20. This is lscpi output: >>00:1b.0 Audio Device: Intel Corporation System Controller Hub (SCH Poulsbo) HD Audio Controller (rev 07) cat proc/asound/cards: >>0 [MID ]: HDA-Intel - HDA Intel MID >> HDA Intel MID at 0xd01c000 irq 16 The Codec is a >>Sigmatel STAC9202 My problem is, that sound is working, but has constant clicking noise in the background, also if the sound is completely muted. First I thought this sound would come from some hdd or whatever, but the device has no mechanical parts. But this is what it sounds like - Irregular clicking noise. This noise immediately starts during boot up as soon as the module is loaded and before X11 has started. I already tried options like "position_fix=1 or 2" and different "model="-options with no success. "position_fix" didn't do anything at all and different model-options have muted the device completely. Has someone any idea to get this working? The device comes with an own (hardly) closed linux which runs a 2.6.24 kernel - here sound is working with no problems at all. If you need further info, let me know. Thanks in advance, best! Christian

2 1

[alsa-devel] [PATCH] ASoC: MPC5200: Changed how appl_ptr is initialized
by John Bonesio 29 Jul '09

29 Jul '09

Hello, I've been working in the ALSA SOC area for the MPC5200. I've noticed that in the routine psc_dma_trigger() in sound/soc/fsl/mpc5200_dma.c there are times when the computed value for s->appl_ptr ends up wrapping negative. This occurs when s->runtime->control->appl_ptr is less than runtime->period_size * runtime->periods It seemed like this would occur at then of the data stream or when the data stream is small (smaller than runtime->period_size * runtime->periods). As far as I can tell, s->appl_ptr is supposed to be the previous value of s->runtime->control->appl_ptr, so the driver can send out the right number of bytes (from previous appl_ptr to current appl_ptr) to avoid pops and clicks at the end of the audio. So it appears the assumption in this code is wrong, that the previous s->runtime->control->appl_ptr is always rumtime->period_size * runtime->periods bytes prior to the current s->runtime->control->appl_ptr. I took out this code and instead added code to intialize s->appl_ptr to 0 in psc_dma_open(). Does this fix seem right? - John The computation for appl_ptr in the trigger() routine made a bad assumption when the stream is at the end of the file and there is not (periods * period_size) bytes remaining in the data. This causes playback issues when the total stream size is less than this amount, as the s->appl_ptr wrapps negative and the data is never played. Signed-off-by: John Bonesio <bones(a)secretlab.ca> --- sound/soc/fsl/mpc5200_dma.c | 3 +-- 1 files changed, 1 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index b2eba27..cfe0ea4 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -181,8 +181,6 @@ static int psc_dma_trigger(struct snd_pcm_substream *substream, int cmd) * end of stream and not over running it. */ s->runtime = runtime; - s->appl_ptr = s->runtime->control->appl_ptr - - (runtime->period_size * runtime->periods); /* Fill up the bestcomm bd queue and enable DMA. * This will begin filling the PSC's fifo. @@ -287,6 +285,7 @@ static int psc_dma_open(struct snd_pcm_substream *substream) } s->stream = substream; + s->appl_ptr = 0; return 0; }

2 1

[alsa-devel] [PATCH] aedsp16: Buffer overflow
by Roel Kluin 29 Jul '09

29 Jul '09

DSPVersion is declared as char[3], but the sprintf writes at least 4 bytes including terminating null. Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- Found with Parfait, http://research.sun.com/projects/parfait/ on line 498: static char DSPVersion[CARDVERLEN + 1] __initdata = {0, }; diff --git a/sound/oss/aedsp16.c b/sound/oss/aedsp16.c index 3ee9900..35b5912 100644 --- a/sound/oss/aedsp16.c +++ b/sound/oss/aedsp16.c @@ -325,8 +325,9 @@ /* * Size of character arrays that store name and version of sound card */ -#define CARDNAMELEN 15 /* Size of the card's name in chars */ -#define CARDVERLEN 2 /* Size of the card's version in chars */ +#define CARDNAMELEN 15 /* Size of the card's name in chars */ +#define CARDVERLEN 10 /* Size of the card's version in chars */ +#define CARDVERDIGITS 2 /* Number of digits in the version */ #if defined(CONFIG_SC6600) /* @@ -410,7 +411,7 @@ static int soft_cfg __initdata = 0; /* bitmapped config */ static int soft_cfg_mss __initdata = 0; /* bitmapped mss config */ -static int ver[CARDVERLEN] __initdata = {0, 0}; /* DSP Ver: +static int ver[CARDVERDIGITS] __initdata = {0, 0}; /* DSP Ver: hi->ver[0] lo->ver[1] */ #if defined(CONFIG_SC6600) @@ -957,7 +958,7 @@ static int __init aedsp16_dsp_version(int port) * string is finished. */ ver[len++] = ret; - } while (len < CARDVERLEN); + } while (len < CARDVERDIGITS); sprintf(DSPVersion, "%d.%d", ver[0], ver[1]); DBG(("success.\n"));

2 1

[alsa-devel] [PATCH] mpu401.c: Buffer overflow
by Roel Kluin 29 Jul '09

29 Jul '09

mpu_synth_info[m].name is a char[30], and the minimum length of the data written by sprintf is 31 bytes including terminating null. Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- Found with Parfait, http://research.sun.com/projects/parfait/ diff --git a/sound/oss/mpu401.c b/sound/oss/mpu401.c index 1b2316f..734b8f9 100644 --- a/sound/oss/mpu401.c +++ b/sound/oss/mpu401.c @@ -1074,7 +1074,7 @@ int attach_mpu401(struct address_info *hw_config, struct module *owner) sprintf(mpu_synth_info[m].name, "%s (MPU401)", hw_config->name); else sprintf(mpu_synth_info[m].name, - "MPU-401 %d.%d%c Midi interface #%d", + "MPU-401 %d.%d%c MIDI #%d", (int) (devc->version & 0xf0) >> 4, devc->version & 0x0f, revision_char,

2 1

[alsa-devel] my udev rules are breaking my dmixer setup why?
by Jelle de Jong 29 Jul '09

29 Jul '09

Hello everybody, I am trying to use udev to get fixed device names depending on the location of the device on the usb bus. The udev rules seems to be working fine, however i cant get dmix to work with the udev rules using only hw dies work. If i remove the udev rules the dmixer works fine. Can somebody look at the attachment, and maybe do some testing, i would highly appreciate it. I am also on chat.freenode.org and #alsa as tuxcrafter Thanks in advance, Jelle ls -hal /dev/snd/ total 0 drwxr-xr-x 2 root root 320 2008-11-09 15:44 . drwxr-xr-x 12 root root 4.8K 2008-11-09 15:50 .. crw-rw---- 1 root user0 116, 0 2008-11-09 15:44 controlC0 crw-rw---- 1 root user1 116, 32 2008-11-09 15:44 controlC1 crw-rw---- 1 root user2 116, 64 2008-11-09 15:44 controlC2 crw-rw---- 1 root user3 116, 96 2008-11-09 15:44 controlC3 crw-rw---- 1 root user0 116, 24 2008-11-09 15:44 pcmC0D0c crw-rw---- 1 root user0 116, 16 2008-11-09 15:49 pcmC0D0p crw-rw---- 1 root user1 116, 56 2008-11-09 15:44 pcmC1D0c crw-rw---- 1 root user1 116, 48 2008-11-09 15:44 pcmC1D0p crw-rw---- 1 root user2 116, 88 2008-11-09 15:44 pcmC2D0c crw-rw---- 1 root user2 116, 80 2008-11-09 15:44 pcmC2D0p crw-rw---- 1 root user3 116, 120 2008-11-09 15:44 pcmC3D0c crw-rw---- 1 root user3 116, 112 2008-11-09 15:44 pcmC3D0p crw-rw---- 1 root audio 116, 1 2008-11-09 15:44 seq crw-rw---- 1 root audio 116, 33 2008-11-09 15:44 timer ------------------------------------------------------------------------ udevinfo info --attribute-walk --name=/dev/snd/controlC0 ------------------------------------------------------------------------ echo 'SUBSYSTEM=="sound", KERNELS=="3-1", KERNEL=="controlC[0-9]*", NAME="snd/controlC4" SUBSYSTEM=="sound", KERNELS=="3-2", KERNEL=="controlC[0-9]", NAME="snd/controlC5" SUBSYSTEM=="sound", KERNELS=="4-1", KERNEL=="controlC[0-9]", NAME="snd/controlC6" SUBSYSTEM=="sound", KERNELS=="4-2", KERNEL=="controlC[0-9]", NAME="snd/controlC7" SUBSYSTEM=="sound", KERNELS=="3-1", KERNEL=="hwC[0-9]D0c", NAME="snd/hwC4D0c" SUBSYSTEM=="sound", KERNELS=="3-2", KERNEL=="hwC[0-9]D0c", NAME="snd/hwC5D0c" SUBSYSTEM=="sound", KERNELS=="4-1", KERNEL=="hwC[0-9]D0c", NAME="snd/hwC6D0c" SUBSYSTEM=="sound", KERNELS=="4-2", KERNEL=="hwC[0-9]D0c", NAME="snd/hwC7D0c" SUBSYSTEM=="sound", KERNELS=="3-1", KERNEL=="pcmC[0-9]D0c", NAME="snd/pcmC4D0c" SUBSYSTEM=="sound", KERNELS=="3-2", KERNEL=="pcmC[0-9]D0c", NAME="snd/pcmC5D0c" SUBSYSTEM=="sound", KERNELS=="4-1", KERNEL=="pcmC[0-9]D0c", NAME="snd/pcmC6D0c" SUBSYSTEM=="sound", KERNELS=="4-2", KERNEL=="pcmC[0-9]D0c", NAME="snd/pcmC7D0c" SUBSYSTEM=="sound", KERNELS=="3-1", KERNEL=="pcmC[0-9]D0p", NAME="snd/pcmC4D0p" SUBSYSTEM=="sound", KERNELS=="3-2", KERNEL=="pcmC[0-9]D0p", NAME="snd/pcmC5D0p" SUBSYSTEM=="sound", KERNELS=="4-1", KERNEL=="pcmC[0-9]D0p", NAME="snd/pcmC6D0p" SUBSYSTEM=="sound", KERNELS=="4-2", KERNEL=="pcmC[0-9]D0p", NAME="snd/pcmC7D0p"' | tee /etc/udev/rules.d/10-persistent-sound.rules chmod 600 /etc/udev/rules.d/10-persistent-sound.rules ------------------------------------------------------------------------ # without dmixer # this works .... echo 'pcm.softvol { type softvol slave.pcm hw:1 control.name Master control.card 1 } pcm.!default { type plug slave.pcm softvol } ctl.!default { type hw card 1 }' | tee /home/user1/.asoundrc chmod 600 /home/user1/.asoundrc chown user1:user1 /home/user1/.asoundrc ------------------------------------------------------------------------ # with dmixer # this does not work echo 'pcm.softvol { type softvol slave.pcm dmix:1 control.name Master control.card 1 } pcm.!default { type plug slave.pcm softvol } ctl.!default { type hw card 1 }' | tee /home/user1/.asoundrc chmod 600 /home/user1/.asoundrc chown user1:user1 /home/user1/.asoundrc ------------------------------------------------------------------------ # with dmixer # this does not work echo 'pcm.!default { type plug slave.pcm dmixer } pcm.dmixer { type dmix ipc_key 2048 slave.pcm hw:1 } ctl.dmixer { type hw card 1 } pcm.dsp { type plug slave.pcm dmixer } ctl.mixer { type hw card 1 }' | tee /home/user1/.asoundrc chmod 600 /home/user1/.asoundrc chown user1:user1 /home/user1/.asoundrc ------------------------------------------------------------------------

3 13

[alsa-devel] [RFC] [PATCH v2] ASoC: CX20442: push down machine independent line discipline bits
by Janusz Krzysztofik 29 Jul '09

29 Jul '09

This corrected patch adds machine independent line discipline code, prevoiusly exsiting inside my Amstrad Delta ASoC machine dirver, to the Conexant CX20442 codec driver. The code can be used as a standalone line discipline, or as a set of codec specific functions called from machine's line discipline callbacks. Anyway, the line discipline itself must be registered by a machine driver. Applies on top of the followup to my initial driver version: http://mailman.alsa-project.org/pipermail/alsa-devel/2009-July/019757.html Suggested by ASoC manintainer Mark Brown <broonie(a)opensource.wolfsonmicro.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- Corrections: sound/soc/codecs/cx20442.c line 300: -static struct tty_ldisc_ops v253_ops = { +struct tty_ldisc_ops v253_ops = { line 420: - dev_err(&codec->dev, "Failed to register codec: %d\n", ret); + dev_err(codec->dev, "Failed to register codec: %d\n", ret); line 425: - dev_err(&codec->dev, "Failed to register DAI: %d\n", ret); + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); Corrected patch: --- linux-2.6.31-rc3/sound/soc/codecs/cx20442.h.orig 2009-07-22 04:56:45.000000000 +0200 +++ linux-2.6.31-rc3/sound/soc/codecs/cx20442.h 2009-07-28 21:23:17.000000000 +0200 @@ -15,5 +15,6 @@ extern struct snd_soc_dai cx20442_dai; extern struct snd_soc_codec_device cx20442_codec_dev; +extern struct tty_ldisc_ops v253_ops; #endif --- linux-2.6.31-rc3/sound/soc/codecs/cx20442.c.orig 2009-07-28 20:26:18.000000000 +0200 +++ linux-2.6.31-rc3/sound/soc/codecs/cx20442.c 2009-07-29 02:16:05.000000000 +0200 @@ -13,6 +13,8 @@ * option) any later version. */ +#include <linux/tty.h> + #include <sound/core.h> #include <sound/initval.h> #include <sound/soc-dapm.h> @@ -172,8 +174,7 @@ static int cx20442_write(struct snd_soc_ return -EINVAL; /* hw_write and control_data pointers required for talking to the modem - * are expected to be set by the machine driver's line discipline - * initialization code */ + * are expected to be set by the line discipline initialization code */ if (!codec->hw_write || !codec->control_data) return -EIO; @@ -208,6 +209,111 @@ static int cx20442_write(struct snd_soc_ return 0; } + +/* Moved up here as line discipline referres it during initialization */ +static struct snd_soc_codec *cx20442_codec; + + +/* + * Line discpline related code + * + * Any of the callback functions below can be used in two ways: + * 1) registerd by a machine driver as one of line discipline operations, + * 2) called from a machine's provided line discipline callback function + * in case when extra machine specific code must be run as well. + */ + +/* Modem init: echo off, digital speaker off, quiet off, voice mode */ +static const char *v253_init = "ate0m0q0+fclass=8\r"; + +/* Line discipline .open() */ +static int v253_open(struct tty_struct *tty) +{ + struct snd_soc_codec *codec = cx20442_codec; + int ret, len = strlen(v253_init); + + /* Doesn't make sense without write callback */ + if (!tty->ops->write) + return -EINVAL; + + /* Pass the codec structure address for use by other ldisc callbacks */ + tty->disc_data = codec; + + if (tty->ops->write(tty, v253_init, len) != len) { + ret = -EIO; + goto err; + } + /* Actual setup will be performed after the modem responds. */ + return 0; +err: + tty->disc_data = NULL; + return ret; +} + +/* Line discipline .close() */ +static void v253_close(struct tty_struct *tty) +{ + struct snd_soc_codec *codec = tty->disc_data; + + tty->disc_data = NULL; + + if (!codec) + return; + + /* Prevent the codec driver from further accessing the modem */ + codec->hw_write = NULL; + codec->control_data = NULL; + codec->pop_time = 0; +} + +/* Line discipline .hangup() */ +static int v253_hangup(struct tty_struct *tty) +{ + v253_close(tty); + return 0; +} + +/* Line discipline .receive_buf() */ +static void v253_receive(struct tty_struct *tty, + const unsigned char *cp, char *fp, int count) +{ + struct snd_soc_codec *codec = tty->disc_data; + + if (!codec) + return; + + if (!codec->control_data) { + /* First modem response, complete setup procedure */ + + /* Set up codec driver access to modem controls */ + codec->control_data = tty; + codec->hw_write = (hw_write_t)tty->ops->write; + codec->pop_time = 1; + } +} + +/* Line discipline .write_wakeup() */ +static void v253_wakeup(struct tty_struct *tty) +{ +} + +struct tty_ldisc_ops v253_ops = { + .magic = TTY_LDISC_MAGIC, + .name = "cx20442", + .owner = THIS_MODULE, + .open = v253_open, + .close = v253_close, + .hangup = v253_hangup, + .receive_buf = v253_receive, + .write_wakeup = v253_wakeup, +}; +EXPORT_SYMBOL_GPL(v253_ops); + + +/* + * Codec DAI + */ + struct snd_soc_dai cx20442_dai = { .name = "CX20442", .playback = { @@ -227,8 +333,6 @@ struct snd_soc_dai cx20442_dai = { }; EXPORT_SYMBOL_GPL(cx20442_dai); -static struct snd_soc_codec *cx20442_codec; - static int cx20442_codec_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); @@ -313,13 +417,13 @@ static int cx20442_register(struct cx204 ret = snd_soc_register_codec(codec); if (ret != 0) { - dev_err(&codec->dev, "Failed to register codec: %d\n", ret); + dev_err(codec->dev, "Failed to register codec: %d\n", ret); goto err; } ret = snd_soc_register_dai(&cx20442_dai); if (ret != 0) { - dev_err(&codec->dev, "Failed to register DAI: %d\n", ret); + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); goto err_codec; }

1 0

[alsa-devel] Strange behaviour with varying frame sizes on read
by Metathronius Galabant 29 Jul '09

29 Jul '09

Hi all, please consider the following: Period size is 80 frames, buffer size is 320 frames. For all 3 tests I have 240 frames ready to read. snd_pcm_avail_update() confirms this. Outcome of my tests: 1) I want to read 80 frames, and get 80 frames. - OK, this is really basic. 2) I want to read 81 frames, and get 80 frames. - FAIL. 3) I want to read 80 frames, and get 80 frames. Then I read 1 frame, and get 1 frame. - HUH? I've both tested snd_pcm_readi() and snd_pcm_mmap_readi() - both have the same semantic. Is there any explanation for this kind of illogical behaviour (I have version 1.0.14, which might be a bit old, but still)? I should be able to read rather arbitrary amounts from a software buffer, don't I? Cheers, M.

1 0

[alsa-devel] Using Alsa for AD1981b on 9263
by Shilpa Kedar Walvekar 29 Jul '09

29 Jul '09

Hi, Addition to my earlier post: We are using AT91SAM9263 evaluation kit with on-board AD1981b codec. We are using Linux kernel version 2.6.27 with patches available on http://linux4sam.com. We found from the forum that we need to use ALSA for playing and recording audio using this codec. So we downloaded Alsa-lib-1.0.20 Alsa-oss-1.0.20 Alsa-utils-1.0.20 We installed above using: ./configure && make && make install We then downloaded alsa-driver-1.0.20 and installed it using below: ./configure --with-cards=atmel-ac97c --with-kernel=../../linux-2.6.27 --with-build=../../linux-2.6.27 && make ARCH=arm && make install-modules Driver was compiled and installed successfully. We found following: # cat /proc/asound/cards 0 [AC97 ]: ac97c - Atmel AC97 Atmel AC97 Controller at 0xfffa0000, irq 18 # lsmod Module Size Used by Not tainted snd_at91_ac97 8420 0 snd_ac97_codec 99904 1 snd_at91_ac97 snd_soc_atmel_pcm 4484 0 snd_soc_core 30852 1 snd_soc_atmel_pcm snd_pcm 60772 4 snd_at91_ac97,snd_ac97_codec,snd_soc_atmel_pcm,s nd_soc_core snd_page_alloc 6056 1 snd_pcm snd_timer 19588 1 snd_pcm snd 47700 5 snd_at91_ac97,snd_ac97_codec,snd_soc_core,snd_pc m,snd_timer soundcore 6052 1 snd ac97_bus 1568 1 snd_ac97_codec # cat /home/.asoundrc pcm.!default { type hw card 0 } ctl.!default { type hw card 0 } We have following devices in /dev/snd # cat /dev/snd/ /dev/snd/controlC0 /dev/snd/controlC3 /dev/snd/hwC0D2 /dev/snd/timer /dev/snd/controlC1 /dev/snd/hwC0D0 /dev/snd/hwC0D3 /dev/snd/controlC2 /dev/snd/hwC0D1 /dev/snd/seq # aplay -l **** List of PLAYBACK Hardware Devices **** ALSA lib conf.c:2700:(snd_config_hooks_call) Cannot open shared library (null) ALSA lib conf.c:3079:(snd_config_update_r) hooks failed, removing configuration aplay: device_list:232: control open (0): No such file or directory We are not able to get if module is getting loaded, then why aplay doesn't list the device? Please help us to get the audio codec working. Regards, --Shilpa.

3 3

[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 29 Jul '09

29 Jul '09

The following changes since commit 4ce2f2fe61002ab35c9333d52c95b5ff9e949f50: Marek Vasut (1): ASoC: Switch palm27x-asoc to jack detection api are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Barry Song (2): ASoC: blackfin I2S(TDM mode) CPU DAI driver ASoC: board driver to connect bf5xx with ad1938 Janusz Krzysztofik (1): ASoC: CX20442: fix issues pointed out by subsystem maintainer sound/soc/blackfin/Kconfig | 23 +++- sound/soc/blackfin/Makefile | 6 + sound/soc/blackfin/bf5xx-ad1938.c | 142 +++++++++++++++ sound/soc/blackfin/bf5xx-tdm-pcm.c | 330 ++++++++++++++++++++++++++++++++++ sound/soc/blackfin/bf5xx-tdm-pcm.h | 21 +++ sound/soc/blackfin/bf5xx-tdm.c | 343 ++++++++++++++++++++++++++++++++++++ sound/soc/blackfin/bf5xx-tdm.h | 14 ++ sound/soc/codecs/cx20442.c | 17 +- 8 files changed, 888 insertions(+), 8 deletions(-) create mode 100644 sound/soc/blackfin/bf5xx-ad1938.c create mode 100644 sound/soc/blackfin/bf5xx-tdm-pcm.c create mode 100644 sound/soc/blackfin/bf5xx-tdm-pcm.h create mode 100644 sound/soc/blackfin/bf5xx-tdm.c create mode 100644 sound/soc/blackfin/bf5xx-tdm.h

2 1

Re: [alsa-devel] Testers wanted: New SB X-Fi driver
by Takashi Iwai 29 Jul '09

29 Jul '09

[Please don't do top-posting] At Tue, 28 Jul 2009 14:50:10 -0500, Ted T. Logan wrote: > > Just got the latest as of today, > > click > click > click > click > click > > :( > mplayer yesterday.mp3 For testing this kind of problem, use aplay and try different --period-size and --buffer-size options. mplayer isn't suitable for primary testing. If the click period doesn't change regardless of the period size option value, it implies that the noise doesn't come from the buffer set up but rather from other routes. If the noise period changes with different period size options, the click noise is a buffer setup issue. In addition, it'd be better to use aplay with -Dplughw to avoid the automatic routing to PA. Also, your description was unclear. It it only click without other tone, or repeated sound, or the click noise appear on the normal playback, or what? The repeated sound bug with emu20k2 (it didn't appear with emu20k1) was fixed as I mentioned. But, if it's click noise on the normal playback, then I don't know. Better to ask Creative guys... thanks, Takashi

1 0

[alsa-devel] Assert when attempting to use pulseaudio git master with the a52 plugin.
by Luke Yelavich 29 Jul '09

29 Jul '09

Hi all I would like to bring to your attention a bug in the a52 plugin when working with pulseaudio. The full details are found in the ALSA bug tracker: https://bugtrack.alsa-project.org/alsa-bug/bug_view_advanced_page.php?bug_i… I am happy for follow ups to be in the bug or on this list. Thanks Luke

1 0

[PATCH] ASoC: MPC5200: Support for buffer wrap around
by John Bonesio 29 Jul '09

29 Jul '09

The code in psc_dma_bcom_enqueue_tx() didn't account for the fact that s->runtime->control->appl_ptr can wrap around to the beginning of the buffer. This change fixes this problem. Signed-off-by: John Bonesio <bones(a)secretlab.ca> --- sound/soc/fsl/mpc5200_dma.c | 17 +++++++++++++++++ 1 files changed, 17 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index cfe0ea4..2551c58 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -70,6 +70,23 @@ static void psc_dma_bcom_enqueue_next_buffer(struct psc_dma_stream *s) static void psc_dma_bcom_enqueue_tx(struct psc_dma_stream *s) { + if (s->appl_ptr > s->runtime->control->appl_ptr) { + /* + * In this case s->runtime->control->appl_ptr has wrapped around. + * Play the data to the end of the boundary, then wrap our own + * appl_ptr back around. + */ + while (s->appl_ptr < s->runtime->boundary) { + if (bcom_queue_full(s->bcom_task)) + return; + + s->appl_ptr += s->period_size; + + psc_dma_bcom_enqueue_next_buffer(s); + } + s->appl_ptr -= s->runtime->boundary; + } + while (s->appl_ptr < s->runtime->control->appl_ptr) { if (bcom_queue_full(s->bcom_task)) -- 1.6.0.4 --=-D7TcNx246dz2g/fqLr/C--

1 0

[alsa-devel] [PATCH] ASoC: CX20442: fix issues pointed out by subsystem maintainer
by Janusz Krzysztofik 29 Jul '09

29 Jul '09

The patch fixes some checkpatch identified issues and adds a comment about line discipline interaction to my driver code, as requested by Mark on my inital submission (thank you Mark for applying my imperfect patch anyway). It also fixes MODULE_ALIAS mismatch as used in my machine driver. Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- In my next attempt, I'll try to satisfy Mark's request for pushing down the line discipline code, now existing in my machine driver, to the codec driver. --- linux-2.6.31-rc3/sound/soc/codecs/cx20442.c.orig 2009-07-22 04:56:45.000000000 +0200 +++ linux-2.6.31-rc3/sound/soc/codecs/cx20442.c 2009-07-22 21:46:41.000000000 +0200 @@ -132,7 +132,7 @@ enum v253_vls { static int cx20442_pm_to_v253_vls(u8 value) { - switch(value & ~(1 << CX20442_AGC)) { + switch (value & ~(1 << CX20442_AGC)) { case 0: return V253_VLS_T; case (1 << CX20442_SPKOUT): @@ -152,7 +152,7 @@ static int cx20442_pm_to_v253_vls(u8 val } static int cx20442_pm_to_v253_vsp(u8 value) { - switch(value & ~(1 << CX20442_AGC)) { + switch (value & ~(1 << CX20442_AGC)) { case (1 << CX20442_SPKOUT): case (1 << CX20442_MIC): case (1 << CX20442_SPKOUT) | (1 << CX20442_MIC): @@ -171,6 +171,9 @@ static int cx20442_write(struct snd_soc_ if (reg >= codec->reg_cache_size) return -EINVAL; + /* hw_write and control_data pointers required for talking to the modem + * are expected to be set by the machine driver's line discipline + * initialization code */ if (!codec->hw_write || !codec->control_data) return -EIO; @@ -182,7 +185,7 @@ static int cx20442_write(struct snd_soc_ return vls; vsp = cx20442_pm_to_v253_vsp(value); - if (vsp < 0 ) + if (vsp < 0) return vsp; if ((vls == V253_VLS_T) || @@ -232,7 +235,7 @@ static int cx20442_codec_probe(struct pl struct snd_soc_codec *codec; int ret; - if(!cx20442_codec) { + if (!cx20442_codec) { dev_err(&pdev->dev, "cx20442 not yet discovered\n"); return -ENODEV; } @@ -310,13 +313,13 @@ static int cx20442_register(struct cx204 ret = snd_soc_register_codec(codec); if (ret != 0) { - //dev_err(&dev->dev, "Failed to register codec: %d\n", ret); + dev_err(&codec->dev, "Failed to register codec: %d\n", ret); goto err; } ret = snd_soc_register_dai(&cx20442_dai); if (ret != 0) { - //dev_err(&dev->dev, "Failed to register DAI: %d\n", ret); + dev_err(&codec->dev, "Failed to register DAI: %d\n", ret); goto err_codec; } @@ -392,4 +395,4 @@ module_exit(cx20442_exit); MODULE_DESCRIPTION("ASoC CX20442-11 voice modem codec driver"); MODULE_AUTHOR("Janusz Krzysztofik"); MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:cx20442-codec"); +MODULE_ALIAS("platform:cx20442");

2 1

[alsa-devel] [RFC] [PATCH] ASoC: CX20442: push down machine independent line discipline bits
by Janusz Krzysztofik 29 Jul '09

29 Jul '09

This patch adds machine independent line discipline code, prevoiusly exsiting inside my Amstrad Delta ASoC machine dirver, to the Conexant CX20442 codec driver. The code can be used as a standalone line discipline, or as a set of codec specific functions called from machine's line discipline callbacks. Anyway, the line discipline itself must be registered by a machine driver. Suggested by ASoC manintainer Mark Brown <broonie(a)opensource.wolfsonmicro.com> (Hardly understood, imperfectly implemented and) Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- Mark, does it follow your idea closely enough? --- linux-2.6.31-rc3/sound/soc/codecs/cx20442.h.orig 2009-07-22 04:56:45.000000000 +0200 +++ linux-2.6.31-rc3/sound/soc/codecs/cx20442.h 2009-07-28 21:23:17.000000000 +0200 @@ -15,5 +15,6 @@ extern struct snd_soc_dai cx20442_dai; extern struct snd_soc_codec_device cx20442_codec_dev; +extern struct tty_ldisc_ops v253_ops; #endif --- linux-2.6.31-rc3/sound/soc/codecs/cx20442.c.orig 2009-07-28 20:26:18.000000000 +0200 +++ linux-2.6.31-rc3/sound/soc/codecs/cx20442.c 2009-07-28 21:19:43.000000000 +0200 @@ -13,6 +13,8 @@ * option) any later version. */ +#include <linux/tty.h> + #include <sound/core.h> #include <sound/initval.h> #include <sound/soc-dapm.h> @@ -172,8 +174,7 @@ static int cx20442_write(struct snd_soc_ return -EINVAL; /* hw_write and control_data pointers required for talking to the modem - * are expected to be set by the machine driver's line discipline - * initialization code */ + * are expected to be set by a line discipline initialization code */ if (!codec->hw_write || !codec->control_data) return -EIO; @@ -208,6 +209,111 @@ static int cx20442_write(struct snd_soc_ return 0; } + +/* Moved up here as line discipline referres it during initialization */ +static struct snd_soc_codec *cx20442_codec; + + +/* + * Line discpline related code + * + * Any of the callback functions below can be used in two ways: + * 1) registerd by a machine driver as one of line discipline operations, + * 2) called from inside a machine specific line discipline callback function + * in case when extra machine specific code must be run as well. + */ + +/* Modem init: echo off, digital speaker off, quiet off, voice mode */ +static const char *v253_init = "ate0m0q0+fclass=8\r"; + +/* Line discipline .open() */ +static int v253_open(struct tty_struct *tty) +{ + struct snd_soc_codec *codec = cx20442_codec; + int ret, len = strlen(v253_init); + + /* Doesn't make sense without write callback */ + if (!tty->ops->write) + return -EINVAL; + + /* Pass the codec structure address for use by other ldisc callbacks */ + tty->disc_data = codec; + + if (tty->ops->write(tty, v253_init, len) != len) { + ret = -EIO; + goto err; + } + /* Actual setup will be performed after the modem responds. */ + return 0; +err: + tty->disc_data = NULL; + return ret; +} + +/* Line discipline .close() */ +static void v253_close(struct tty_struct *tty) +{ + struct snd_soc_codec *codec = tty->disc_data; + + tty->disc_data = NULL; + + if (!codec) + return; + + /* Prevent the codec driver from further accessing the modem */ + codec->hw_write = NULL; + codec->control_data = NULL; + codec->pop_time = 0; +} + +/* Line discipline .hangup() */ +static int v253_hangup(struct tty_struct *tty) +{ + v253_close(tty); + return 0; +} + +/* Line discipline .receive_buf() */ +static void v253_receive(struct tty_struct *tty, + const unsigned char *cp, char *fp, int count) +{ + struct snd_soc_codec *codec = tty->disc_data; + + if (!codec) + return; + + if (!codec->control_data) { + /* First modem response, complete setup procedure */ + + /* Set up codec driver access to modem controls */ + codec->control_data = tty; + codec->hw_write = (hw_write_t)tty->ops->write; + codec->pop_time = 1; + } +} + +/* Line discipline .write_wakeup() */ +static void v253_wakeup(struct tty_struct *tty) +{ +} + +static struct tty_ldisc_ops v253_ops = { + .magic = TTY_LDISC_MAGIC, + .name = "cx20442", + .owner = THIS_MODULE, + .open = v253_open, + .close = v253_close, + .hangup = v253_hangup, + .receive_buf = v253_receive, + .write_wakeup = v253_wakeup, +}; +EXPORT_SYMBOL_GPL(v253_ops); + + +/* + * Codec DAI + */ + struct snd_soc_dai cx20442_dai = { .name = "CX20442", .playback = { @@ -227,8 +333,6 @@ struct snd_soc_dai cx20442_dai = { }; EXPORT_SYMBOL_GPL(cx20442_dai); -static struct snd_soc_codec *cx20442_codec; - static int cx20442_codec_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev);

2 1

[alsa-devel] [PATCH 0/2]blackfin TDM dai and blackfin-ad1938 machine driver
by Barry Song 29 Jul '09

29 Jul '09

Hi Mark, The following two patches are the 2nd version for the "new support for ad1938 codec" patch sequence. Since you have applied the codec driver patch([PATCH 1/3] new ad1938 codec driver based on asoc), here I only send the DAI and Board driver patches again according to your feedback. Patch 1: blackfin I2S(TDM mode) CPU DAI driver Patch 2: board driver to connect bf5xx with ad1938 Detailed description about every patch and changes is included in the corresponding patch. Thanks Barry

2 9

[alsa-devel] [PATCH 1/2] blackfin I2S(TDM mode) CPU DAI driver
by Barry Song 29 Jul '09

29 Jul '09

The I2S DAI driver for blackfin SPORT, but works in TDM mode. I2S is not a special case of TDM with only left and right two slots for SPORT interface. I2S coordinates with TDM in SPORT, but not a part of TDM. TDM require different hardware configuration with I2S, not only different slot number. One is "Stereo Serial Operation" mode of SPORT, the other one is "Multichannel Operation" mode. They are incompatible at the same time. Hardware and DMA description and data transfer flow are much different for I2S and TDM. Merging them as a whole will be very ugly and difficult to maintain. So we don't define a new DAI type, but give two DAI instances for standard I2S and TDM, both in I2S-family DAI type. The TDM instance still uses the I2S-family DAI type. Signed-off-by: Barry Song <21cnbao(a)gmail.com> --- sound/soc/blackfin/Kconfig | 15 ++- sound/soc/blackfin/Makefile | 4 + sound/soc/blackfin/bf5xx-tdm-pcm.c | 330 ++++++++++++++++++++++++++++++++++ sound/soc/blackfin/bf5xx-tdm-pcm.h | 21 +++ sound/soc/blackfin/bf5xx-tdm.c | 343 ++++++++++++++++++++++++++++++++++++ sound/soc/blackfin/bf5xx-tdm.h | 14 ++ 6 files changed, 726 insertions(+), 1 deletions(-) create mode 100644 sound/soc/blackfin/bf5xx-tdm-pcm.c create mode 100644 sound/soc/blackfin/bf5xx-tdm-pcm.h create mode 100644 sound/soc/blackfin/bf5xx-tdm.c create mode 100644 sound/soc/blackfin/bf5xx-tdm.h diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index 811596f..2ab6d2f 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -7,6 +7,15 @@ config SND_BF5XX_I2S mode (supports single stereo In/Out). You will also need to select the audio interfaces to support below. +config SND_BF5XX_TDM + tristate "SoC I2S(TDM mode) Audio for the ADI BF5xx chip" + depends on (BLACKFIN && SND_SOC) + help + Say Y or M if you want to add support for codecs attached to + the Blackfin SPORT (synchronous serial ports) interface in TDM + mode. + You will also need to select the audio interfaces to support below. + config SND_BF5XX_SOC_SSM2602 tristate "SoC SSM2602 Audio support for BF52x ezkit" depends on SND_BF5XX_I2S @@ -69,6 +78,10 @@ config SND_BF5XX_SOC_I2S tristate select SND_BF5XX_SOC_SPORT +config SND_BF5XX_SOC_TDM + tristate + select SND_BF5XX_SOC_SPORT + config SND_BF5XX_SOC_AC97 tristate select AC97_BUS @@ -85,7 +98,7 @@ config SND_BF5XX_SOC_AD1980 config SND_BF5XX_SPORT_NUM int "Set a SPORT for Sound chip" - depends on (SND_BF5XX_I2S || SND_BF5XX_AC97) + depends on (SND_BF5XX_I2S || SND_BF5XX_AC97 || SND_BF5XX_TDM) range 0 3 if BF54x range 0 1 if !BF54x default 0 diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile index 97bb37a..f861926 100644 --- a/sound/soc/blackfin/Makefile +++ b/sound/soc/blackfin/Makefile @@ -1,15 +1,19 @@ # Blackfin Platform Support snd-bf5xx-ac97-objs := bf5xx-ac97-pcm.o snd-bf5xx-i2s-objs := bf5xx-i2s-pcm.o +snd-bf5xx-tdm-objs := bf5xx-tdm-pcm.o snd-soc-bf5xx-sport-objs := bf5xx-sport.o snd-soc-bf5xx-ac97-objs := bf5xx-ac97.o snd-soc-bf5xx-i2s-objs := bf5xx-i2s.o +snd-soc-bf5xx-tdm-objs := bf5xx-tdm.o obj-$(CONFIG_SND_BF5XX_AC97) += snd-bf5xx-ac97.o obj-$(CONFIG_SND_BF5XX_I2S) += snd-bf5xx-i2s.o +obj-$(CONFIG_SND_BF5XX_TDM) += snd-bf5xx-tdm.o obj-$(CONFIG_SND_BF5XX_SOC_SPORT) += snd-soc-bf5xx-sport.o obj-$(CONFIG_SND_BF5XX_SOC_AC97) += snd-soc-bf5xx-ac97.o obj-$(CONFIG_SND_BF5XX_SOC_I2S) += snd-soc-bf5xx-i2s.o +obj-$(CONFIG_SND_BF5XX_SOC_TDM) += snd-soc-bf5xx-tdm.o # Blackfin Machine Support snd-ad1980-objs := bf5xx-ad1980.o diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.c b/sound/soc/blackfin/bf5xx-tdm-pcm.c new file mode 100644 index 0000000..af7fa8d --- /dev/null +++ b/sound/soc/blackfin/bf5xx-tdm-pcm.c @@ -0,0 +1,330 @@ +/* + * File: sound/soc/blackfin/bf5xx-tdm-pcm.c + * Author: Barry Song <Barry.Song(a)analog.com> + * + * Created: Tue June 06 2009 + * Description: DMA driver for tdm codec + * + * Modified: + * Copyright 2009 Analog Devices Inc. + * + * Bugs: Enter bugs at http://blackfin.uclinux.org/ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * 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, see the file COPYING, or write + * to the Free Software Foundation, Inc., + * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/dma-mapping.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <asm/dma.h> + +#include "bf5xx-tdm-pcm.h" +#include "bf5xx-tdm.h" +#include "bf5xx-sport.h" + +#define PCM_BUFFER_MAX 0x10000 +#define FRAGMENT_SIZE_MIN (4*1024) +#define FRAGMENTS_MIN 2 +#define FRAGMENTS_MAX 32 + +static void bf5xx_dma_irq(void *data) +{ + struct snd_pcm_substream *pcm = data; + snd_pcm_period_elapsed(pcm); +} + +static const struct snd_pcm_hardware bf5xx_pcm_hardware = { + .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_RESUME), + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .rates = SNDRV_PCM_RATE_48000, + .channels_min = 2, + .channels_max = 8, + .buffer_bytes_max = PCM_BUFFER_MAX, + .period_bytes_min = FRAGMENT_SIZE_MIN, + .period_bytes_max = PCM_BUFFER_MAX/2, + .periods_min = FRAGMENTS_MIN, + .periods_max = FRAGMENTS_MAX, +}; + +static int bf5xx_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + size_t size = bf5xx_pcm_hardware.buffer_bytes_max; + snd_pcm_lib_malloc_pages(substream, size * 4); + + return 0; +} + +static int bf5xx_pcm_hw_free(struct snd_pcm_substream *substream) +{ + snd_pcm_lib_free_pages(substream); + + return 0; +} + +static int bf5xx_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct sport_device *sport = runtime->private_data; + int fragsize_bytes = frames_to_bytes(runtime, runtime->period_size); + + fragsize_bytes /= runtime->channels; + /* inflate the fragsize to match the dma width of SPORT */ + fragsize_bytes *= 8; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + sport_set_tx_callback(sport, bf5xx_dma_irq, substream); + sport_config_tx_dma(sport, runtime->dma_area, + runtime->periods, fragsize_bytes); + } else { + sport_set_rx_callback(sport, bf5xx_dma_irq, substream); + sport_config_rx_dma(sport, runtime->dma_area, + runtime->periods, fragsize_bytes); + } + + return 0; +} + +static int bf5xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct sport_device *sport = runtime->private_data; + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + sport_tx_start(sport); + else + sport_rx_start(sport); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + sport_tx_stop(sport); + else + sport_rx_stop(sport); + break; + default: + ret = -EINVAL; + } + + return ret; +} + +static snd_pcm_uframes_t bf5xx_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct sport_device *sport = runtime->private_data; + unsigned int diff; + snd_pcm_uframes_t frames; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + diff = sport_curr_offset_tx(sport); + frames = diff / (8*4); /* 32 bytes per frame */ + } else { + diff = sport_curr_offset_rx(sport); + frames = diff / (8*4); + } + return frames; +} + +static int bf5xx_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + int ret = 0; + + snd_soc_set_runtime_hwparams(substream, &bf5xx_pcm_hardware); + + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + goto out; + + if (sport_handle != NULL) + runtime->private_data = sport_handle; + else { + pr_err("sport_handle is NULL\n"); + ret = -ENODEV; + } +out: + return ret; +} + +static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, + snd_pcm_uframes_t pos, void *buf, snd_pcm_uframes_t count) +{ + unsigned int *src; + unsigned int *dst; + int i; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + src = buf; + dst = (unsigned int *)substream->runtime->dma_area; + + dst += pos * 8; + while (count--) { + for (i = 0; i < substream->runtime->channels; i++) + *(dst + i) = *src++; + dst += 8; + } + } else { + src = (unsigned int *)substream->runtime->dma_area; + dst = buf; + + src += pos * 8; + while (count--) { + for (i = 0; i < substream->runtime->channels; i++) + *dst++ = *(src+i); + src += 8; + } + } + + return 0; +} + +static int bf5xx_pcm_silence(struct snd_pcm_substream *substream, + int channel, snd_pcm_uframes_t pos, snd_pcm_uframes_t count) +{ + unsigned char *buf = substream->runtime->dma_area; + buf += pos * 8 * 4; + memset(buf, '\0', count * 8 * 4); + + return 0; +} + + +struct snd_pcm_ops bf5xx_pcm_tdm_ops = { + .open = bf5xx_pcm_open, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = bf5xx_pcm_hw_params, + .hw_free = bf5xx_pcm_hw_free, + .prepare = bf5xx_pcm_prepare, + .trigger = bf5xx_pcm_trigger, + .pointer = bf5xx_pcm_pointer, + .copy = bf5xx_pcm_copy, + .silence = bf5xx_pcm_silence, +}; + +static int bf5xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = bf5xx_pcm_hardware.buffer_bytes_max; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->area = dma_alloc_coherent(pcm->card->dev, size * 4, + &buf->addr, GFP_KERNEL); + if (!buf->area) { + pr_err("Failed to allocate dma memory \ + Please increase uncached DMA memory region\n"); + return -ENOMEM; + } + buf->bytes = size; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + sport_handle->tx_buf = buf->area; + else + sport_handle->rx_buf = buf->area; + + return 0; +} + +static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + dma_free_coherent(NULL, buf->bytes, buf->area, 0); + buf->area = NULL; + } + if (sport_handle) + sport_done(sport_handle); +} + +static u64 bf5xx_pcm_dmamask = DMA_32BIT_MASK; + +static int bf5xx_pcm_tdm_new(struct snd_card *card, struct snd_soc_dai *dai, + struct snd_pcm *pcm) +{ + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &bf5xx_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = DMA_32BIT_MASK; + + if (dai->playback.channels_min) { + ret = bf5xx_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + } + + if (dai->capture.channels_min) { + ret = bf5xx_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + } +out: + return ret; +} + +struct snd_soc_platform bf5xx_tdm_soc_platform = { + .name = "bf5xx-audio", + .pcm_ops = &bf5xx_pcm_tdm_ops, + .pcm_new = bf5xx_pcm_tdm_new, + .pcm_free = bf5xx_pcm_free_dma_buffers, +}; +EXPORT_SYMBOL_GPL(bf5xx_tdm_soc_platform); + +static int __init bfin_pcm_tdm_init(void) +{ + return snd_soc_register_platform(&bf5xx_tdm_soc_platform); +} +module_init(bfin_pcm_tdm_init); + +static void __exit bfin_pcm_tdm_exit(void) +{ + snd_soc_unregister_platform(&bf5xx_tdm_soc_platform); +} +module_exit(bfin_pcm_tdm_exit); + +MODULE_AUTHOR("Barry Song"); +MODULE_DESCRIPTION("ADI Blackfin TDM PCM DMA module"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.h b/sound/soc/blackfin/bf5xx-tdm-pcm.h new file mode 100644 index 0000000..9e8473d --- /dev/null +++ b/sound/soc/blackfin/bf5xx-tdm-pcm.h @@ -0,0 +1,21 @@ +/* + * linux/sound/arm/bf5xx-tdm-pcm.h -- ALSA PCM interface for the Blackfin + * + * Copyright 2009 Analog Device Inc. + * + * 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 _BF5XX_TDM_PCM_H +#define _BF5XX_TDM_PCM_H + +struct bf5xx_pcm_dma_params { + char *name; /* stream identifier */ +}; + +/* platform data */ +extern struct snd_soc_platform bf5xx_tdm_soc_platform; + +#endif diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c new file mode 100644 index 0000000..3096bad --- /dev/null +++ b/sound/soc/blackfin/bf5xx-tdm.c @@ -0,0 +1,343 @@ +/* + * File: sound/soc/blackfin/bf5xx-tdm.c + * Author: Barry Song <Barry.Song(a)analog.com> + * + * Created: Thurs June 04 2009 + * Description: Blackfin I2S(TDM) CPU DAI driver + * Even though TDM mode can be as part of I2S DAI, but there + * are so much difference in configuration and data flow, + * it's very ugly to integrate I2S and TDM into a module + * + * Modified: + * Copyright 2009 Analog Devices Inc. + * + * Bugs: Enter bugs at http://blackfin.uclinux.org/ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * 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, see the file COPYING, or write + * to the Free Software Foundation, Inc., + * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include <asm/irq.h> +#include <asm/portmux.h> +#include <linux/mutex.h> +#include <linux/gpio.h> + +#include "bf5xx-sport.h" +#include "bf5xx-tdm.h" + +struct bf5xx_tdm_port { + u16 tcr1; + u16 rcr1; + u16 tcr2; + u16 rcr2; + int configured; +}; + +static struct bf5xx_tdm_port bf5xx_tdm; +static int sport_num = CONFIG_SND_BF5XX_SPORT_NUM; + +static struct sport_param sport_params[2] = { + { + .dma_rx_chan = CH_SPORT0_RX, + .dma_tx_chan = CH_SPORT0_TX, + .err_irq = IRQ_SPORT0_ERROR, + .regs = (struct sport_register *)SPORT0_TCR1, + }, + { + .dma_rx_chan = CH_SPORT1_RX, + .dma_tx_chan = CH_SPORT1_TX, + .err_irq = IRQ_SPORT1_ERROR, + .regs = (struct sport_register *)SPORT1_TCR1, + } +}; + +/* + * Setting the TFS pin selector for SPORT 0 based on whether the selected + * port id F or G. If the port is F then no conflict should exist for the + * TFS. When Port G is selected and EMAC then there is a conflict between + * the PHY interrupt line and TFS. Current settings prevent the conflict + * by ignoring the TFS pin when Port G is selected. This allows both + * ssm2602 using Port G and EMAC concurrently. + */ +#ifdef CONFIG_BF527_SPORT0_PORTF +#define LOCAL_SPORT0_TFS (P_SPORT0_TFS) +#else +#define LOCAL_SPORT0_TFS (0) +#endif + +static u16 sport_req[][7] = { {P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS, + P_SPORT0_DRPRI, P_SPORT0_RSCLK, LOCAL_SPORT0_TFS, 0}, + {P_SPORT1_DTPRI, P_SPORT1_TSCLK, P_SPORT1_RFS, P_SPORT1_DRPRI, + P_SPORT1_RSCLK, P_SPORT1_TFS, 0} }; + +static int bf5xx_tdm_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + int ret = 0; + + /* interface format:support TDM,slave mode */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + break; + default: + printk(KERN_ERR "%s: Unknown DAI format type\n", __func__); + ret = -EINVAL; + break; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + break; + case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBS_CFM: + ret = -EINVAL; + break; + default: + printk(KERN_ERR "%s: Unknown DAI master type\n", __func__); + ret = -EINVAL; + break; + } + + return ret; +} + +static int bf5xx_tdm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + int ret = 0; + + bf5xx_tdm.tcr2 &= ~0x1f; + bf5xx_tdm.rcr2 &= ~0x1f; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S32_LE: + bf5xx_tdm.tcr2 |= 31; + bf5xx_tdm.rcr2 |= 31; + sport_handle->wdsize = 4; + break; + /* at present, we only support 32bit transfer */ + default: + pr_err("not supported PCM format yet\n"); + return -EINVAL; + break; + } + + if (!bf5xx_tdm.configured) { + /* + * TX and RX are not independent,they are enabled at the + * same time, even if only one side is running. So, we + * need to configure both of them at the time when the first + * stream is opened. + * + * CPU DAI:slave mode. + */ + ret = sport_config_rx(sport_handle, bf5xx_tdm.rcr1, + bf5xx_tdm.rcr2, 0, 0); + if (ret) { + pr_err("SPORT is busy!\n"); + return -EBUSY; + } + + ret = sport_config_tx(sport_handle, bf5xx_tdm.tcr1, + bf5xx_tdm.tcr2, 0, 0); + if (ret) { + pr_err("SPORT is busy!\n"); + return -EBUSY; + } + + bf5xx_tdm.configured = 1; + } + + return 0; +} + +static void bf5xx_tdm_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + /* No active stream, SPORT is allowed to be configured again. */ + if (!dai->active) + bf5xx_tdm.configured = 0; +} + +#ifdef CONFIG_PM +static int bf5xx_tdm_suspend(struct snd_soc_dai *dai) +{ + struct sport_device *sport = + (struct sport_device *)dai->private_data; + + if (!dai->active) + return 0; + if (dai->capture.active) + sport_rx_stop(sport); + if (dai->playback.active) + sport_tx_stop(sport); + return 0; +} + +static int bf5xx_tdm_resume(struct snd_soc_dai *dai) +{ + int ret; + struct sport_device *sport = + (struct sport_device *)dai->private_data; + + if (!dai->active) + return 0; + + ret = sport_set_multichannel(sport, 8, 0xFF, 1); + if (ret) { + pr_err("SPORT is busy!\n"); + ret = -EBUSY; + } + + ret = sport_config_rx(sport, IRFS, 0x1F, 0, 0); + if (ret) { + pr_err("SPORT is busy!\n"); + ret = -EBUSY; + } + + ret = sport_config_tx(sport, ITFS, 0x1F, 0, 0); + if (ret) { + pr_err("SPORT is busy!\n"); + ret = -EBUSY; + } + + return 0; +} + +#else +#define bf5xx_tdm_suspend NULL +#define bf5xx_tdm_resume NULL +#endif + +static struct snd_soc_dai_ops bf5xx_tdm_dai_ops = { + .hw_params = bf5xx_tdm_hw_params, + .set_fmt = bf5xx_tdm_set_dai_fmt, + .shutdown = bf5xx_tdm_shutdown, +}; + +struct snd_soc_dai bf5xx_tdm_dai = { + .name = "bf5xx-tdm", + .id = 0, + .suspend = bf5xx_tdm_suspend, + .resume = bf5xx_tdm_resume, + .playback = { + .channels_min = 2, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S32_LE,}, + .capture = { + .channels_min = 2, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S32_LE,}, + .ops = &bf5xx_tdm_dai_ops, +}; +EXPORT_SYMBOL_GPL(bf5xx_tdm_dai); + +static int __devinit bfin_tdm_probe(struct platform_device *pdev) +{ + int ret = 0; + + if (peripheral_request_list(&sport_req[sport_num][0], "soc-audio")) { + pr_err("Requesting Peripherals failed\n"); + return -EFAULT; + } + + /* request DMA for SPORT */ + sport_handle = sport_init(&sport_params[sport_num], 4, \ + 8 * sizeof(u32), NULL); + if (!sport_handle) { + peripheral_free_list(&sport_req[sport_num][0]); + return -ENODEV; + } + + /* SPORT works in TDM mode */ + ret = sport_set_multichannel(sport_handle, 8, 0xFF, 1); + if (ret) { + pr_err("SPORT is busy!\n"); + ret = -EBUSY; + goto sport_config_err; + } + + ret = sport_config_rx(sport_handle, IRFS, 0x1F, 0, 0); + if (ret) { + pr_err("SPORT is busy!\n"); + ret = -EBUSY; + goto sport_config_err; + } + + ret = sport_config_tx(sport_handle, ITFS, 0x1F, 0, 0); + if (ret) { + pr_err("SPORT is busy!\n"); + ret = -EBUSY; + goto sport_config_err; + } + + ret = snd_soc_register_dai(&bf5xx_tdm_dai); + if (ret) { + pr_err("Failed to register DAI: %d\n", ret); + goto sport_config_err; + } + return 0; + +sport_config_err: + peripheral_free_list(&sport_req[sport_num][0]); + return ret; +} + +static int __devexit bfin_tdm_remove(struct platform_device *pdev) +{ + peripheral_free_list(&sport_req[sport_num][0]); + snd_soc_unregister_dai(&bf5xx_tdm_dai); + + return 0; +} + +static struct platform_driver bfin_tdm_driver = { + .probe = bfin_tdm_probe, + .remove = __devexit_p(bfin_tdm_remove), + .driver = { + .name = "bfin-tdm", + .owner = THIS_MODULE, + }, +}; + +static int __init bfin_tdm_init(void) +{ + return platform_driver_register(&bfin_tdm_driver); +} +module_init(bfin_tdm_init); + +static void __exit bfin_tdm_exit(void) +{ + platform_driver_unregister(&bfin_tdm_driver); +} +module_exit(bfin_tdm_exit); + +/* Module information */ +MODULE_AUTHOR("Barry Song"); +MODULE_DESCRIPTION("TDM driver for ADI Blackfin"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/blackfin/bf5xx-tdm.h b/sound/soc/blackfin/bf5xx-tdm.h new file mode 100644 index 0000000..e5157cd --- /dev/null +++ b/sound/soc/blackfin/bf5xx-tdm.h @@ -0,0 +1,14 @@ +/* + * linux/sound/arm/bf5xx-i2s.h + * + * 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 _BF5XX_TDM_H +#define _BF5XX_TDM_H + +extern struct snd_soc_dai bf5xx_tdm_dai; + +#endif -- 1.5.6.3

2 1

[alsa-devel] [RFC] [PATCH 1/3] ASoC: Add support for Conexant CX20442-11 voice modem codec
by Janusz Krzysztofik 28 Jul '09

28 Jul '09

This patch adds support for Conexant CX20442-11 voice modem codec, suitable for use by the ASoC board driver for Amstrad E3 (Delta) videophone. Related sound card driver will follow. Created and tested against linux-2.6.31-rc3. Applies and works with linux-omap-2.6 commit 7c5cb7862d32cb344be7831d466535d5255e35ac as well. Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- This codec is an optional part of the Conexant SmartV three chip modem design. As such, documentation for its proprietary digital audio interface is not available. However, on Amstrad Delta board, thanks to Mark Underwood who created an initial, omap-alsa based sound driver a few years ago[1], the codec has been discovered to be accessible not only from the modem side, but also over the OMAP McBSP based CPU DAI. Thus, the driver can be used by any sound card that can access the codec DAI directly. The DAI configuration parameters (sample rate and format, number of channels) has been selected out empirically for best user experience. The codec analogue interface consists of two pairs of analogue I/O pins: speakerphone interface or telephone handset/headset interface. Furthermore, it seams to provide two operation modes for speakerphone I/O: standard and advanced, with automatic gain control and echo cancelation. Even if the codec control interface is unknown and not available, all those interfaces and modes can be selected over the modem chip using V.253 commands. The driver is able to issue necessary commands over a suitable hw_write function if provided by a sound card driver. Otherwise, the codec can be controlled over the modem from userspace while inactive. Even if nothig is known about the codec internal power management capabilities, DAPM widgets has been used to model the codec audio map. Automatically performed powering up/down of those virtual widgets results in corresponding V.253 commands being issued. Some driver features/oddities may be board specific, but I have no way to verify that with any board other than Amstrad Delta. [1] http://www.earth.li/pipermail/e3-hacking/2006-April/000481.html --- linux-2.6.31-rc1/sound/soc/codecs/Kconfig.orig 2009-07-04 16:40:00.000000000 +0200 +++ linux-2.6.31-rc1/sound/soc/codecs/Kconfig 2009-07-22 01:01:08.000000000 +0200 @@ -17,6 +17,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C select SND_SOC_CS4270 if I2C + select SND_SOC_CX20442 select SND_SOC_PCM3008 select SND_SOC_SPDIF select SND_SOC_SSM2602 if I2C @@ -86,6 +87,9 @@ config SND_SOC_CS4270_VD33_ERRATA bool depends on SND_SOC_CS4270 +config SND_SOC_CX20442 + tristate + config SND_SOC_L3 tristate --- linux-2.6.31-rc1/sound/soc/codecs/Makefile.orig 2009-07-04 16:40:00.000000000 +0200 +++ linux-2.6.31-rc1/sound/soc/codecs/Makefile 2009-07-22 01:01:08.000000000 +0200 @@ -4,6 +4,7 @@ snd-soc-ad73311-objs := ad73311.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o snd-soc-cs4270-objs := cs4270.o +snd-soc-cx20442-objs := cx20442.o snd-soc-l3-objs := l3.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-spdif-objs := spdif_transciever.o @@ -41,6 +42,7 @@ obj-$(CONFIG_SND_SOC_AD73311) += snd-soc obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(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 obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o --- /dev/null 2009-06-24 10:33:03.437003511 +0200 +++ linux-2.6.31-rc1/sound/soc/codecs/cx20442.h 2009-07-22 01:01:08.000000000 +0200 @@ -0,0 +1,19 @@ +/* + * cx20442.h -- audio driver for CX20442 + * + * Copyright 2009 Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> + * + * 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. + * + */ + +#ifndef _CX20442_CODEC_H +#define _CX20442_CODEC_H + +extern struct snd_soc_dai cx20442_dai; +extern struct snd_soc_codec_device cx20442_codec_dev; + +#endif --- /dev/null 2009-06-24 10:33:03.437003511 +0200 +++ linux-2.6.31-rc1/sound/soc/codecs/cx20442.c 2009-07-22 01:01:08.000000000 +0200 @@ -0,0 +1,395 @@ +/* + * cx20442.c -- CX20442 ALSA Soc Audio driver + * + * Copyright 2009 Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> + * + * Initially based on sound/soc/codecs/wm8400.c + * Copyright 2008, 2009 Wolfson Microelectronics PLC. + * Author: Mark Brown <broonie(a)opensource.wolfsonmicro.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. + */ + +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/soc-dapm.h> + +#include "cx20442.h" + + +struct cx20442_priv { + struct snd_soc_codec codec; + u8 reg_cache[1]; +}; + +#define CX20442_PM 0x0 + +#define CX20442_TELIN 0 +#define CX20442_TELOUT 1 +#define CX20442_MIC 2 +#define CX20442_SPKOUT 3 +#define CX20442_AGC 4 + +static const struct snd_soc_dapm_widget cx20442_dapm_widgets[] = { + SND_SOC_DAPM_OUTPUT("TELOUT"), + SND_SOC_DAPM_OUTPUT("SPKOUT"), + SND_SOC_DAPM_OUTPUT("AGCOUT"), + + SND_SOC_DAPM_MIXER("SPKOUT Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_PGA("TELOUT Amp", CX20442_PM, CX20442_TELOUT, 0, NULL, 0), + SND_SOC_DAPM_PGA("SPKOUT Amp", CX20442_PM, CX20442_SPKOUT, 0, NULL, 0), + SND_SOC_DAPM_PGA("SPKOUT AGC", CX20442_PM, CX20442_AGC, 0, NULL, 0), + + SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_MIXER("Input Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_MICBIAS("TELIN Bias", CX20442_PM, CX20442_TELIN, 0), + SND_SOC_DAPM_MICBIAS("MIC Bias", CX20442_PM, CX20442_MIC, 0), + + SND_SOC_DAPM_PGA("MIC AGC", CX20442_PM, CX20442_AGC, 0, NULL, 0), + + SND_SOC_DAPM_INPUT("TELIN"), + SND_SOC_DAPM_INPUT("MIC"), + SND_SOC_DAPM_INPUT("AGCIN"), +}; + +static const struct snd_soc_dapm_route cx20442_audio_map[] = { + {"TELOUT", NULL, "TELOUT Amp"}, + + {"SPKOUT", NULL, "SPKOUT Mixer"}, + {"SPKOUT Mixer", NULL, "SPKOUT Amp"}, + + {"TELOUT Amp", NULL, "DAC"}, + {"SPKOUT Amp", NULL, "DAC"}, + + {"SPKOUT Mixer", NULL, "SPKOUT AGC"}, + {"SPKOUT AGC", NULL, "AGCIN"}, + + {"AGCOUT", NULL, "MIC AGC"}, + {"MIC AGC", NULL, "MIC"}, + + {"MIC Bias", NULL, "MIC"}, + {"Input Mixer", NULL, "MIC Bias"}, + + {"TELIN Bias", NULL, "TELIN"}, + {"Input Mixer", NULL, "TELIN Bias"}, + + {"ADC", NULL, "Input Mixer"}, +}; + +static int cx20442_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, cx20442_dapm_widgets, + ARRAY_SIZE(cx20442_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, cx20442_audio_map, + ARRAY_SIZE(cx20442_audio_map)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static unsigned int cx20442_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 *reg_cache = codec->reg_cache; + + if (reg >= codec->reg_cache_size) + return -EINVAL; + + return reg_cache[reg]; +} + +enum v253_vls { + V253_VLS_NONE = 0, + V253_VLS_T, + V253_VLS_L, + V253_VLS_LT, + V253_VLS_S, + V253_VLS_ST, + V253_VLS_M, + V253_VLS_MST, + V253_VLS_S1, + V253_VLS_S1T, + V253_VLS_MS1T, + V253_VLS_M1, + V253_VLS_M1ST, + V253_VLS_M1S1T, + V253_VLS_H, + V253_VLS_HT, + V253_VLS_MS, + V253_VLS_MS1, + V253_VLS_M1S, + V253_VLS_M1S1, + V253_VLS_TEST, +}; + +static int cx20442_pm_to_v253_vls(u8 value) +{ + switch(value & ~(1 << CX20442_AGC)) { + case 0: + return V253_VLS_T; + case (1 << CX20442_SPKOUT): + return V253_VLS_S1; + case (1 << CX20442_MIC): + return V253_VLS_M1; + case (1 << CX20442_SPKOUT) | (1 << CX20442_MIC): + return V253_VLS_M1S1; + case (1 << CX20442_TELOUT): + case (1 << CX20442_TELIN): + case (1 << CX20442_TELOUT) | (1 << CX20442_TELIN): + return V253_VLS_L; + case (1 << CX20442_TELOUT) | (1 << CX20442_MIC): + return V253_VLS_NONE; + } + return -EINVAL; +} +static int cx20442_pm_to_v253_vsp(u8 value) +{ + switch(value & ~(1 << CX20442_AGC)) { + case (1 << CX20442_SPKOUT): + case (1 << CX20442_MIC): + case (1 << CX20442_SPKOUT) | (1 << CX20442_MIC): + return (bool)(value & (1 << CX20442_AGC)); + } + return (value & (1 << CX20442_AGC)) ? -EINVAL : 0; +} + +static int cx20442_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 *reg_cache = codec->reg_cache; + int vls, vsp, old, len; + char buf[18]; + + if (reg >= codec->reg_cache_size) + return -EINVAL; + + if (!codec->hw_write || !codec->control_data) + return -EIO; + + old = reg_cache[reg]; + reg_cache[reg] = value; + + vls = cx20442_pm_to_v253_vls(value); + if (vls < 0) + return vls; + + vsp = cx20442_pm_to_v253_vsp(value); + if (vsp < 0 ) + return vsp; + + if ((vls == V253_VLS_T) || + (vls == cx20442_pm_to_v253_vls(old))) { + if (vsp == cx20442_pm_to_v253_vsp(old)) + return 0; + len = snprintf(buf, ARRAY_SIZE(buf), "at+vsp=%d\r", vsp); + } else if (vsp == cx20442_pm_to_v253_vsp(old)) + len = snprintf(buf, ARRAY_SIZE(buf), "at+vls=%d\r", vls); + else + len = snprintf(buf, ARRAY_SIZE(buf), + "at+vls=%d;+vsp=%d\r", vls, vsp); + + if (unlikely(len > (ARRAY_SIZE(buf) - 1))) + return -ENOMEM; + + if (codec->hw_write(codec->control_data, buf, len) != len) + return -EIO; + + return 0; +} + +struct snd_soc_dai cx20442_dai = { + .name = "CX20442", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}; +EXPORT_SYMBOL_GPL(cx20442_dai); + +static struct snd_soc_codec *cx20442_codec; + +static int cx20442_codec_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret; + + if(!cx20442_codec) { + dev_err(&pdev->dev, "cx20442 not yet discovered\n"); + return -ENODEV; + } + codec = cx20442_codec; + + socdev->card->codec = codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(&pdev->dev, "failed to create pcms\n"); + goto pcm_err; + } + + cx20442_add_widgets(codec); + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(&pdev->dev, "failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + return ret; +} + +/* power down chip */ +static int cx20442_codec_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device cx20442_codec_dev = { + .probe = cx20442_codec_probe, + .remove = cx20442_codec_remove, +}; +EXPORT_SYMBOL_GPL(cx20442_codec_dev); + +static int cx20442_register(struct cx20442_priv *cx20442) +{ + struct snd_soc_codec *codec = &cx20442->codec; + int ret; + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->name = "CX20442"; + codec->owner = THIS_MODULE; + codec->private_data = cx20442; + + codec->dai = &cx20442_dai; + codec->num_dai = 1; + + codec->reg_cache = &cx20442->reg_cache; + codec->reg_cache_size = ARRAY_SIZE(cx20442->reg_cache); + codec->read = cx20442_read_reg_cache; + codec->write = cx20442_write; + + codec->bias_level = SND_SOC_BIAS_OFF; + + cx20442_dai.dev = codec->dev; + + cx20442_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + //dev_err(&dev->dev, "Failed to register codec: %d\n", ret); + goto err; + } + + ret = snd_soc_register_dai(&cx20442_dai); + if (ret != 0) { + //dev_err(&dev->dev, "Failed to register DAI: %d\n", ret); + goto err_codec; + } + + return 0; + +err_codec: + snd_soc_unregister_codec(codec); +err: + cx20442_codec = NULL; + kfree(cx20442); + return ret; +} + +static void cx20442_unregister(struct cx20442_priv *cx20442) +{ + snd_soc_unregister_dai(&cx20442_dai); + snd_soc_unregister_codec(&cx20442->codec); + + cx20442_codec = NULL; + kfree(cx20442); +} + +static int cx20442_platform_probe(struct platform_device *pdev) +{ + struct cx20442_priv *cx20442; + struct snd_soc_codec *codec; + + cx20442 = kzalloc(sizeof(struct cx20442_priv), GFP_KERNEL); + if (cx20442 == NULL) + return -ENOMEM; + + codec = &cx20442->codec; + + codec->control_data = NULL; + codec->hw_write = NULL; + codec->pop_time = 0; + + codec->dev = &pdev->dev; + platform_set_drvdata(pdev, cx20442); + + return cx20442_register(cx20442); +} + +static int __exit cx20442_platform_remove(struct platform_device *pdev) +{ + struct cx20442_priv *cx20442 = platform_get_drvdata(pdev); + + cx20442_unregister(cx20442); + return 0; +} + +static struct platform_driver cx20442_platform_driver = { + .driver = { + .name = "cx20442", + .owner = THIS_MODULE, + }, + .probe = cx20442_platform_probe, + .remove = __exit_p(cx20442_platform_remove), +}; + +static int __init cx20442_init(void) +{ + return platform_driver_register(&cx20442_platform_driver); +} +module_init(cx20442_init); + +static void __exit cx20442_exit(void) +{ + platform_driver_unregister(&cx20442_platform_driver); +} +module_exit(cx20442_exit); + +MODULE_DESCRIPTION("ASoC CX20442-11 voice modem codec driver"); +MODULE_AUTHOR("Janusz Krzysztofik"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:cx20442-codec");

2 8

[alsa-devel] Using Alsa for AD1981b on 9263
by Shilpa Kedar Walvekar 28 Jul '09

28 Jul '09

Hi, I am using AT91SAM9263 evaluation kit with AD1981b codec. I want to use alsa to play and record audio on this kit. I tried to install the alsa library, alsa-oss, alsa-util. Alsa driver was part of kernel (2.6.27). I did, ./configure && make && make install for library, oss and util. Now when I try to do aplay -l following happens: aplay: device_list:223: no soundcards found... Don't know what is happening. Am I missing any step? What else should be done? Please help me asap. Regards, --Shilpa.

2 1

Re: [alsa-devel] get two mono channel from stereo line
by Clemens Ladisch 28 Jul '09

28 Jul '09

arif setiawan wrote: > It worked now. thank you. Apparently we changed something in your > plugin sample. Sorry, that was my error; I wrote a wrong number for the right dshare plugin. > The ipc_key for dshare in both left and right is supposed to be same > right?, also for dsnoop. Yes; the two processes that are to share the device use this to communicate with each other. Best regards, Clemens

1 0

Re: [alsa-devel] HDA Intel IDT 92HD71B7X - HP Pavilion DV7-1299EF - Subwoofer not Working
by Wael Nasreddine (a.k.a eMxyzptlk) 27 Jul '09

27 Jul '09

I still got no solution for this problem, guys please HELP!! I removed completely Windows, and I can't even enjoy music nor a movie!! On Thu, Jun 11, 2009 at 8:31 PM, Wael Nasreddine (a.k.a eMxyzptlk) < mla(a)nasreddine.com> wrote: > Yes it does mute the speakers once I plug in headphones.. > > > On Thu, Jun 11, 2009 at 8:12 PM, Gustavo Vieira <gustavo(a)sagui.org> wrote: > >> Em Dom, 2009-06-07 às 16:57 +0200, Wael Nasreddine (a.k.a eMxyzptlk) >> escreveu: >> > Last night I bought myself a new laptop, the HP Pavilion DV7-1299EF. >> > >> >> Sorry for hijacking your thread, but does this laptop mute the speaker >> if you plug in the headphones? >> >> I'm experiencing problems with jack sensing in a Pavilion DV4 with the >> same codec and what to know if this problem is general to the codec or >> specific to my model. >> >> Abraços, >> Gustavo >> >> >> > > > -- > Wael Nasreddine > > Blog : http://wael.nasreddine.com > E-mail : wael.nasreddine(a)gmail.com > gTalk : wael.nasreddine(a)gmail.com > Tel : +33.6.32.94.70.13 > Skype : eMxyzptlk > Twitter : @eMxyzptlk > > Sabayon Linux Chief Development Officer - http://www.sabayonlinux.org > > PGP: 1024D/C8DD18A2 06F6 1622 4BC8 4CEB D724 DE12 5565 3945 C8DD 18A2 > > .: An infinite number of monkeys typing into GNU emacs, > would never make a good program. (L. Torvalds 1995) :. > > -- Wael Nasreddine Blog : http://wael.nasreddine.com E-mail : wael.nasreddine(a)gmail.com gTalk : wael.nasreddine(a)gmail.com Tel : +33.6.32.94.70.13 Skype : eMxyzptlk Twitter : @eMxyzptlk Sabayon Linux Chief Development Officer - http://www.sabayonlinux.org PGP: 1024D/C8DD18A2 06F6 1622 4BC8 4CEB D724 DE12 5565 3945 C8DD 18A2 .: An infinite number of monkeys typing into GNU emacs, would never make a good program. (L. Torvalds 1995) :.

2 1

[alsa-devel] my driver can't reproduce continuously
by Kuninori Morimoto 27 Jul '09

27 Jul '09

Hi Now I'm creating sound driver. But can't reproduce continuously. So, I would like to ask. I use module of SH to play sound. Player is aplay, sound data is wav file. Q1) When I play sound, aplay said underrun!!! (at least 0.000 ms long) underrun!!! (at least 0.000 ms long) underrun!!! (at least 0.000 ms long) underrun!!! (at least 0.000 ms long) I checked source of aplay, but can not understand about snd_pcm_status_get_state. why snd_pcm_status_get_state return SND_PCM_STATE_XRUN ? What should my driver do (or where should I check) ? Q2) The reason that it can not reproduce continuously seems soc-core.c :: soc_pcm_trigger timing. Call timing of soc_pcm_trigger is slow. What should my driver do to make call timing more fast ? Best regards -- Kuninori Morimoto

3 11

[alsa-devel] get two mono channel from stereo line
by arif setiawan 27 Jul '09

27 Jul '09

Thank you. It works as we expected, however we found that we can't run two lines (left-capture - left-playback) and (right-capture - right-playback) simulatenously. The error message is : ALSA lib pcm_dshare.c:706:(snd_pcm_dshare_open) unable to open slave cannot open output audio device right: Device or resource busy Is there any workaround for this?

1 0

Re: [alsa-devel] Getting around initial sample loss with intel HDMI
by Takashi Iwai 27 Jul '09

27 Jul '09

At Sun, 26 Jul 2009 23:21:50 -0700, Shane W wrote: > > On Mon, Jul 27, 2009 at 08:07:40AM +0200, Takashi Iwai wrote: > > At Sun, 26 Jul 2009 10:32:10 -0700, > > Shane W wrote: > > > I did some digging in the drivers to try and find the > > > source of this problem. Note that I am really not familiar > > > with this stuff. It appears to be here: > > > hda_codec.c > > > void snd_hda_codec_setup_stream(...) > > > ... > > > snd_hda_codec_write(codec, nid, 0, > > > AC_VERB_SET_CHANNEL_STREAMID, > > > > > > Introducing a delay after this point fixes the issue but > > > it's around a full second so that's obviously not a > > > solution. Could we not cache the current format and only do > > > this when the format changes rather than on every device > > > open? > > > > The format itself is kept. But the stream tag and channel assignment > > are assigned dynamically at each open. Thus, this call has to be > > reset at each open, at least. > > > > > For testing, I tried doing this though and disabling > > > the set_streamid, 0 call in snd_codec_cleanup() but it > > > still missed those first samples. Is there somewhere else I > > > need to be looking? > > > > Does it happen without dmix, too? > > Yeah, with and without dmix it still misses the first bit. OK. I just thought of a possibility of dmix bug. Good to know that it's not :) > The only way it doesn't is if I have another program open > the device and sleep, then have the samples play. Actually, > an open isn't sufficient, it has to go and set the playback > parameters as well. It's logical since the stream and format ids are assigned dynamically first after the stream is set up. One another problem is that we currently call azx_stream_reset() in each prepare callback in hda_intel.c. This was newly added because keeping the format id without resetting caused some other serious errors. Maybe this can be optimized for specific hardwares, e.g. skipping the unneeded reset for HDMI/SPDIF. Takashi

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Re: [alsa-devel] Getting around initial sample loss with intel HDMI
by Takashi Iwai 27 Jul '09

27 Jul '09

At Sun, 26 Jul 2009 10:32:10 -0700, Shane W wrote: > > On Sun, Jul 26, 2009 at 10:52:01AM +0200, Takashi Iwai wrote: > > > It's a known issue that Intel HDMI audio loses the first > > > half second or so of audio when opening the device. Using > > > the dmix interface, I've been able to get around that by > > > writing a quick app to hold the device open so the HDMI > > > stays live. However, with the recent asound.conf posted > > > here which uses the route plugin to fix the multichannel > > > order, I can't do that as dmix isn't in the picture. I'd > > > rather not use dmix anyway since it can bugger with the > > > sample rate so I am wondering if there's another way to get > > > around this. > > > > It's possible to combine with dmix. > > You set up dmix with 8 channels, put route plugin over it. > > This has a few problems. > > It doesn't seem to handle varying sample rates. IE if I > play a 96khz stream at 6 channels, sound is terrible. This is rather a problem of Nvidia HDMI codec driver... > Also, > streaming DTS or AC3 no longer works since the streams are > being upmixed to 6ch. This has to be sent anyway exclusively without dmix, so you'll need another PCM definition. > I did some digging in the drivers to try and find the > source of this problem. Note that I am really not familiar > with this stuff. It appears to be here: > hda_codec.c > void snd_hda_codec_setup_stream(...) > ... > snd_hda_codec_write(codec, nid, 0, > AC_VERB_SET_CHANNEL_STREAMID, > > Introducing a delay after this point fixes the issue but > it's around a full second so that's obviously not a > solution. Could we not cache the current format and only do > this when the format changes rather than on every device > open? The format itself is kept. But the stream tag and channel assignment are assigned dynamically at each open. Thus, this call has to be reset at each open, at least. > For testing, I tried doing this though and disabling > the set_streamid, 0 call in snd_codec_cleanup() but it > still missed those first samples. Is there somewhere else I > need to be looking? Does it happen without dmix, too? The reset part can be a bit more optimized. But, this is really a sensitive part, which can easily break the I/O status without knowing... Takashi

1 0

Re: [alsa-devel] Getting around initial sample loss with intel HDMI
by Takashi Iwai 26 Jul '09

26 Jul '09

At Sat, 25 Jul 2009 13:35:18 -0700, Shane W wrote: > > Hi, > > It's a known issue that Intel HDMI audio loses the first > half second or so of audio when opening the device. Using > the dmix interface, I've been able to get around that by > writing a quick app to hold the device open so the HDMI > stays live. However, with the recent asound.conf posted > here which uses the route plugin to fix the multichannel > order, I can't do that as dmix isn't in the picture. I'd > rather not use dmix anyway since it can bugger with the > sample rate so I am wondering if there's another way to get > around this. It's possible to combine with dmix. You set up dmix with 8 channels, put route plugin over it. Takashi

1 0

[alsa-devel] [snd_via82xx] EAGAIN on snd_pcm_prepare after EPIPE on mmap_commit
by rom1dep 25 Jul '09

25 Jul '09

Hello ! I am encountering some issues with pulseaudio : after some time and for no evident reason PA stops routing the sound. That's what I get with $pulseaudio -vvv : http://pastebin.mandriva.com/13648 (got from a 2.6.29.6_mdv kernel) running $pacmd list-sinks when stuck gives : http://pastebin.mandriva.com/13649 Our pulseaudio integrator Colin Guthrie concluded : "OK, so what is happening is that some alsa level problem is occurring (due to mmap). Pulse tries to recover from this but cannot. The alsa sink (and the whole card) is subsequently unloaded due to this fatal error." And now.. a quick paste from irc : "[00:44:20] <coling> Jul 22 14:04:29 <mezcalero> coling: driver issue i am pretty sure [00:44:20] <coling> Jul 22 14:04:33 <mezcalero> i saw that before once [00:44:20] <coling> Jul 22 14:04:55 <mezcalero> coling: i.e. EAGAIN on snd_pcm_prepare after EPIPE on mmap_commit [00:44:23] <coling> Jul 22 14:05:08 <mezcalero> coling: needs to be brought to attention of alsa folks i guess [00:44:30] <coling> That's what I was told." I'm now on 2.6.30 and I can reproduce that bug, then I'm bringing that to you ! :) Thanks !

3 4

[alsa-devel] Fw: ALSA Volume control with Creative X
by yes work 25 Jul '09

25 Jul '09

----- Forwarded Message ---- From: yes work <yeswork(a)yahoo.com> To: clemens(a)ladisch.de Sent: Wednesday, July 1, 2009 13:03:22 Subject: ALSA Volume control with Creative X Hi, I read in one of your posts about an interest info to improve ALSA. I would like to provide some information that might help improve/augment ALSA for me and others using a creative external usb sound "card." I am running Fedora 11, command line (no windowing environment) and I have a Creative Xmod external USB sound card http://us.creative.com/products/product.asp?category=209&subcategory=668&pr… This Creative device outputs sound OK through ALSA, but the Volume control/Mute knob DO NOT CHANGE the audio. Creative does not provide Linux drivers (the knob works to control volume and mute without installing any drivers on a MS Windows setup). In Fedora, turning the volume knob causes "escape characters" to be displayed on the command line. I have included a copy of an cropped lsusb -v log, below. I hope this might be a step towards getting the volume/mute knob to work with ALSA/Fedora. Let me know if I can be of more help. Thanks, Keith [vortexbox.localdomain ~]# lsusb -v Bus 002 Device 002: ID 041e:30d0 Creative Technology, Ltd Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 1.10 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 64 idVendor 0x041e Creative Technology, Ltd idProduct 0x30d0 bcdDevice 1.00 iManufacturer 1 Creative Technology Ltd iProduct 2 Creative Xmod iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 315 bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x80 (Bus Powered) MaxPower 98mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 159 bInCollection 2 baInterfaceNr( 0) 1 baInterfaceNr( 1) 2 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bNrChannels 6 wChannelConfig 0x003f Left Front (L) Right Front (R) Center Front (C) Low Freqency Enhancement (LFE) Left Surround (LS) Right Surround (RS) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 12 wTerminalType 0x0603 Line Connector bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 4 wTerminalType 0x0603 Line Connector bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 7 wTerminalType 0x0201 Microphone bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 17 bDescriptorType 36 bDescriptorSubtype 4 (MIXER_UNIT) bUnitID 11 bNrInPins 2 baSourceID( 0) 1 baSourceID( 1) 13 bNrChannels 6 wChannelConfig 0x003f Left Front (L) Right Front (R) Center Front (C) Low Freqency Enhancement (LFE) Left Surround (LS) Right Surround (RS) iChannelNames 0 bmControls 0x00 bmControls 0x00 iMixer 0 junk at descriptor end: 00 00 00 AudioControl Interface Descriptor: bLength 18 bDescriptorType 36 bDescriptorSubtype 7 (PROCESSING_UNIT) bUnitID 10 wProcessType 1 bNrPins 1 baSourceID( 0) 11 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 bControlSize 1 bmControls( 0) 0x01 Enable Processing iProcessing 0 Process-Specific 01 03 00 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 5 bSourceID 10 bControlSize 1 bmaControls( 0) 0x01 Mute bmaControls( 1) 0x02 Volume bmaControls( 2) 0x02 Volume iFeature 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 6 bSourceID 4 bControlSize 1 bmaControls( 0) 0x00 bmaControls( 1) 0x02 Volume bmaControls( 2) 0x02 Volume iFeature 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 9 bSourceID 7 bControlSize 1 bmaControls( 0) 0x40 Automatic Gain bmaControls( 1) 0x02 Volume bmaControls( 2) 0x02 Volume iFeature 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 13 bSourceID 12 bControlSize 1 bmaControls( 0) 0x01 Mute bmaControls( 1) 0x02 Volume bmaControls( 2) 0x02 Volume iFeature 0 AudioControl Interface Descriptor: bLength 8 bDescriptorType 36 bDescriptorSubtype 5 (SELECTOR_UNIT) bUnitID 8 bNrInPins 2 baSource( 0) 6 baSource( 1) 9 iSelector 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 2 wTerminalType 0x0301 Speaker bAssocTerminal 0 bSourceID 5 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 3 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 8 iTerminal 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 1 bDelay 0 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 11 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 6 bSubframeSize 2 bBitResolution 16 bSamFreqType 1 Discrete tSamFreq[ 0] 44100 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x021c 1x 540 bytes bInterval 1 bRefresh 0 bSynchAddress 129 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 1 Transfer Type Isochronous Synch Type None Usage Type Data wMaxPacketSize 0x0003 1x 3 bytes bInterval 1 bRefresh 5 bSynchAddress 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 3 bDelay 0 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 11 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 1 Discrete tSamFreq[ 0] 44100 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 1 Transfer Type Isochronous Synch Type None Usage Type Data wMaxPacketSize 0x00b4 1x 180 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 3 Human Interface Device bInterfaceSubClass 0 No Subclass bInterfaceProtocol 0 None iInterface 0 HID Device Descriptor: bLength 9 bDescriptorType 33 bcdHID 1.11 bCountryCode 0 Not supported bNumDescriptors 1 bDescriptorType 34 Report wDescriptorLength 45 Report Descriptors: ** UNAVAILABLE ** Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x83 EP 3 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0001 1x 1 bytes bInterval 10 Device Status: 0x0000 (Bus Powered) [vortexbox.localdomain ~]# New Email names for you! Get the Email name you've always wanted on the new @ymail and @rocketmail. Hurry before someone else does! http://mail.promotions.yahoo.com/newdomains/aa/ Get your preferred Email name! Now you can @ymail.com and @rocketmail.com. http://mail.promotions.yahoo.com/newdomains/aa/

1 0

[alsa-devel] Help with multichannel High Speed USB audio device
by Laurent . 25 Jul '09

25 Jul '09

Dear list, I made an embedded device that has a CODEC with 6 inputs connected to an ARM9 micro-controller that has an integrated high-speed USB controller. Now, I need to write the USB code on that micro-controller so that i can capture those channels in real-time at 24bits / 48 KHz with low latency (ie. sub 10ms), on a computer. I only need my device to work with Linux, and if possible, I would like to avoid writing a device driver on the Linux-side. Is there already such a USB audio driver available that I could nearly use "as is" and to which i would conform ? Would you have some general guideline on how to write the high-speed USB device code ? Would you have any comment on the following : For simplicity, my device is going to be the "clock master" of the system since it is already has the exact clock that drives the CODEC. I will maintain a circular buffer of samples (well, frames of samples) coming from the CODEC. Each time a new frame arrives, i put the samples in the buffer and increase a counter so i know how many frames of samples I have in the buffer, ready to be sent. Each USB micro-frame (125ms/8000Hz), I send all the bytes corresponding to the frames that I have in the circular buffer. In average, 6*3*48000/8000 = 108, but I presume that could jitter slightly, which should not be a problem at all. Thanks in advance, Laurent _________________________________________________________________ Windows Live™ Hotmail®: Celebrate the moment with your favorite sports pics. Check it out. http://www.windowslive.com/Online/Hotmail/Campaign/QuickAdd?ocid=TXT_TAGLM_…

2 2

[alsa-devel] ich10 usb_host vs snd_usb_audio Hercules - irq broadcast.
by Alexey Fisher 24 Jul '09

24 Jul '09

Hallo, i found some time to investigate the issue with my "Hercules USB 5.1 Cannel Audio Adapter". The issue is pretty tricky and depend to host hardware, in my situation this is Intel DG45ID board. Symptoms are: if i use this sound adapter after some time i'll get: irq 16: nobody cared ... Disabling IRQ #16. After closer look i found, it depend on usb port. And if i use one "wrong" usb port on IRQ 18 it will broadcast interrupts to two IRQs 18 and 16, until IRQ 16 will die. The problem is... this happens only with this sound card. So this is not easy to locate bug. And it's working fine on Win Vista and on Mac OS on same hardware.. so it should be software issue. Here is a copy of my post on http://bugzilla.kernel.org/show_bug.cgi?id=13816 If use any usb connector located on irq 18 for usb soundcard it will cause "irq 16: nobody cared", so every thing connected (usb mouse, keyboard ) to irq 16 will be broken. If i connect devices with less traffic (like mouse or keyboard) to irq 18, and usb soundcard to irq 16 or 21 ... every thing will be fine. irqpoll do not do any difference. / irq16|uhci-3 (mouse, keyboard) irq18|ehci-1- irq21|uhci-4 (sound work fine normal) \ irq18|uhci-5 (sound will brake uhci-3) / irq23|uhci-6 (not used) irq23|ehci-2- irq19|uhci-7 (not used) \ irq18|uhci-8 (not used) ========================================================================= [ 315.056793] irq 16: nobody cared (try booting with the "irqpoll" option) [ 315.056799] Pid: 0, comm: swapper Not tainted 2.6.31-rc3-00216-gea11951-dirty #98 [ 315.056802] Call Trace: [ 315.056804] <IRQ> [<ffffffff8109d666>] __report_bad_irq+0x26/0xa0 [ 315.056814] [<ffffffff8109d86c>] note_interrupt+0x18c/0x1d0 [ 315.056819] [<ffffffff8106a373>] ? sched_clock_tick+0x73/0xb0 [ 315.056823] [<ffffffff8109deb5>] handle_fasteoi_irq+0xb5/0xe0 [ 315.056827] [<ffffffff8100dffd>] handle_irq+0x1d/0x30 [ 315.056830] [<ffffffff8100d527>] do_IRQ+0x67/0xe0 [ 315.056834] [<ffffffff8100c013>] ret_from_intr+0x0/0xa [ 315.056837] <EOI> [<ffffffff812822e1>] ? acpi_idle_enter_bm+0x256/0x288 [ 315.056844] [<ffffffff812822de>] ? acpi_idle_enter_bm+0x253/0x288 [ 315.056849] [<ffffffff813a480b>] ? cpuidle_idle_call+0x9b/0xf0 [ 315.056853] [<ffffffff8100a59c>] ? cpu_idle+0xac/0x100 [ 315.056858] [<ffffffff81496825>] ? rest_init+0x65/0x70 [ 315.056862] [<ffffffff8172ed35>] ? start_kernel+0x3ba/0x3c5 [ 315.056866] [<ffffffff8172e315>] ? x86_64_start_reservations+0x125/0x129 [ 315.056870] [<ffffffff8172e3fd>] ? x86_64_start_kernel+0xe4/0xeb [ 315.056872] handlers: [ 315.056874] [<ffffffff8134bdb0>] (usb_hcd_irq+0x0/0x70) [ 315.056878] Disabling IRQ #16 ================================================================== I did smole script to test haw interrupts are used: $for i in $(seq 1 1000); do cat /proc/interrupts >> test; sleep 10s; done $grep usb[3,4,5] test there is no connections on usb3. on usb4 are keyboard an mouse. on usb5 is the soundcard. ====================================================================== -------after boot. there is too match interrupts on 16, -------there is nothing connected. 16: 4276 4520 IO-APIC-fasteoi uhci_hcd:usb3 18: 2732 2598 IO-APIC-fasteoi ehci_hcd:usb1, uhci_hcd:usb5, uhci_hcd:usb8 21: 856 816 IO-APIC-fasteoi uhci_hcd:usb4 -------here i begin to play to usb sound card on usb5. 16: 12162 10987 IO-APIC-fasteoi uhci_hcd:usb3 18: 8070 7134 IO-APIC-fasteoi ehci_hcd:usb1, uhci_hcd:usb5, uhci_hcd:usb8 21: 887 875 IO-APIC-fasteoi uhci_hcd:usb4 16: 20097 17784 IO-APIC-fasteoi uhci_hcd:usb3 18: 13539 11674 IO-APIC-fasteoi ehci_hcd:usb1, uhci_hcd:usb5, uhci_hcd:usb8 21: 1003 958 IO-APIC-fasteoi uhci_hcd:usb4 ------- count of interrupts on irq 16 is almost double of irq18 16: 28083 24881 IO-APIC-fasteoi uhci_hcd:usb3 18: 19091 16130 IO-APIC-fasteoi ehci_hcd:usb1, uhci_hcd:usb5, uhci_hcd:usb8 21: 1005 968 IO-APIC-fasteoi uhci_hcd:usb4 ...... ...... 16: 94758 81082 IO-APIC-fasteoi uhci_hcd:usb3 18: 61430 53863 IO-APIC-fasteoi ehci_hcd:usb1, uhci_hcd:usb5, uhci_hcd:usb8 21: 1389 1539 IO-APIC-fasteoi uhci_hcd:usb4 16: 103141 87919 IO-APIC-fasteoi uhci_hcd:usb3 18: 66678 58625 IO-APIC-fasteoi ehci_hcd:usb1, uhci_hcd:usb5, uhci_hcd:usb8 21: 1389 1539 IO-APIC-fasteoi uhci_hcd:usb4 ------- irq 16: nobody cared, Disabling IRQ #16. 16: 107840 92161 IO-APIC-fasteoi uhci_hcd:usb3 18: 71854 63458 IO-APIC-fasteoi ehci_hcd:usb1, uhci_hcd:usb5, uhci_hcd:usb8 21: 1389 1539 IO-APIC-fasteoi uhci_hcd:usb4 16: 107840 92161 IO-APIC-fasteoi uhci_hcd:usb3 18: 76498 68823 IO-APIC-fasteoi ehci_hcd:usb1, uhci_hcd:usb5, uhci_hcd:usb8 21: 1389 1539 IO-APIC-fasteoi uhci_hcd:usb4

2 2

[alsa-devel] snd_pcm_hw_params_set_period_time_near
by Guilherme Raymo Longo 24 Jul '09

24 Jul '09

Mates, I can´t find the real meaning of this function. snd_pcm_hw_params_set_period_time_near At first I thought that it was used to set the time between 2 consecutive samples, but it is not. Does anyone could gimme some help with that? Tks

1 0

[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 24 Jul '09

24 Jul '09

The following changes since commit 416c8fe3cdcd2d46a6ca12d8ec6143c62d8569d9: Takashi Iwai (1): ASoC: Kill direct accesses to driver_data are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Candelaria Villareal, Jorge (1): ASoC: SDP3430: Add support for EXTMUTE using TWL GPIO6 Chaithrika U S (1): ASoC: tlv320aic3x: Enable PLL when not bypassed Janusz Krzysztofik (2): ASoC: Add support for Conexant CX20442-11 voice modem codec ASoC: Jack handling enhancements as suggested by subsystem maintainer John Bonesio (1): ASoC: MPC5200: Increase the delay time between resets Joonyoung Shim (2): ASoC: MAX9877: separate callback functions ASoC: MAX9877: fix write operation for register Lopez Cruz, Misael (1): ASoC: OMAP: Staticise pcm creation function of omap-pcm Marek Vasut (2): ALSA: Allow passing platform_data to devices attached to AC97 bus ASoC: Switch palm27x-asoc to jack detection api Mark Brown (4): ASoC: Bodge around GCC 4.4.0 flow analysis bug in GCC 4.4.0 ASoC: Fix FLL reference clock division setup in WM8993 ASoC: Fix checkpatch issues in AD1938 Merge branch 'for-2.6.31' into for-2.6.32 include/sound/ac97_codec.h | 6 + include/sound/soc-dai.h | 1 + sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ad1938.c | 156 ++++++++-------- sound/soc/codecs/cx20442.c | 395 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cx20442.h | 19 ++ sound/soc/codecs/max9877.c | 110 +++++++---- sound/soc/codecs/tlv320aic3x.c | 11 +- sound/soc/codecs/wm8400.c | 9 +- sound/soc/codecs/wm8993.c | 2 + sound/soc/fsl/mpc5200_psc_ac97.c | 3 +- sound/soc/omap/omap-pcm.c | 2 +- sound/soc/omap/sdp3430.c | 9 + sound/soc/pxa/palm27x.c | 204 +++++++------------- sound/soc/soc-core.c | 6 +- sound/soc/soc-jack.c | 20 ++- 17 files changed, 703 insertions(+), 255 deletions(-) create mode 100644 sound/soc/codecs/cx20442.c create mode 100644 sound/soc/codecs/cx20442.h

2 1

[alsa-devel] AC97 on pcm030 board with WM9712 codec
by John Bonesio 24 Jul '09

24 Jul '09

Hello, I've been working with Grant Likely exploring an existing issue with resetting the AC97 system on the pcm030 board. The easiest way to produce the problem is by power cycling the board and seeing if the AC97 and wm9712 codec show up in /sys/bus/ac97. I wanted to share with you folks to make sure there's not something else going on that I should be looking at, and if the results I'm seeing seem to make sense. A bit of background for those not familiar with this area of the code: When the board powers up, a reset() message is sent to the wm9712 codec driver which then calls the AC97 driver to perform a cold reset. The AC97 driver performs the cold reset by yanking the reset line on the AC97 link. The AC97 driver also calls warm_reset() after the cold reset is complete. Earlier the warm_reset() call was added in the AC97 cold_reset() routine to attempt to solve this bug, but it appears to not completely solve the issue. After the reset the wm9712 driver attempts to read a register. If the register value is verified the driver assumes the reset was a success, otherwise it flags an error. The code for the routines we're using is included here for reference. ------- code snippets ------- static int wm9712_reset(struct snd_soc_codec *codec, int try_warm) { if (try_warm && soc_ac97_ops.warm_reset) { soc_ac97_ops.warm_reset(codec->ac97); if (ac97_read(codec, 0) == wm9712_reg[0]) return 1; } soc_ac97_ops.reset(codec->ac97); if (ac97_read(codec, 0) != wm9712_reg[0]) goto err; return 0; err: printk(KERN_ERR "WM9712 AC97 reset failed\n"); return -EIO; } static void psc_ac97_cold_reset(struct snd_ac97 *ac97) { struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; /* Do a cold reset */ out_8(&regs->op1, MPC52xx_PSC_OP_RES); udelay(10); out_8(&regs->op0, MPC52xx_PSC_OP_RES); udelay(50); psc_ac97_warm_reset(ac97); } static void psc_ac97_warm_reset(struct snd_ac97 *ac97) { struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; out_be32(&regs->sicr, psc_dma->sicr | MPC52xx_PSC_SICR_AWR); udelay(3); out_be32(&regs->sicr, psc_dma->sicr); } ------- end code snippets ------- I've explored the following possibilities: 1. SDATAOUT or SYNC lines were non-zero during reset, putting the codec chip into a test mode. 2. Perhaps the cold_reset is not necessary and the warm_reset is all that is needed 3. The delay between the cold reset and the warm reset was not long enough. Possibility #1 I added some test code to force the pcm030 board to use gpio instead of the ac97 to perform the reset. This way the SDATAOUT and the SYNC signals can be forced to 0 while resetting. What I discovered is that this didn't seem to make a difference. I don't know if these signals might contain uninitialized values on power up. Should I still consider this as a potential problem, or is this pretty much a non-issue? Possibility #2 I tried taking out the cold reset code in the AC97 driver in hopes that perhaps the hardware would come up in a good state on its own. This didn't work at all. I put the cold reset code back in. Possibility #3 I stumbled upon this when I added a printk to make sure I correctly understood the flow of control. With the printk in place, the problem would go away. So instead I increased the length of time in the udelay() call right before the call to warm_reset(). I've empirically found that delaying 1ms (udelay(50) -> udelay(1000)) seems to avoid the problem. I'm not sure why this delay is necessary, or if it makes sense for the hardware to need this much time to come out of the cold reset. Thanks for your help. Cheers, - John

3 4

[alsa-devel] 2.6.31 - snd_intel8x0: fails to initialize NVidia CK804 sound chip
by Frederik Himpe 24 Jul '09

24 Jul '09

With Linux 2.6.31-rc3-git4 with included Alsa and Linux 2.6.31-rc4 with an Alsa 1.0.20+ snapshot from 13 July, sound on my NVidia CK804 (NForce 4 chipset) is not working). With 2.6.30 everything was working fine. dmesg shows this error: cannot find the slot for index 0 (range 0-1), error: -16 Intel ICH: probe of 0000:00:04.0 failed with error -16 lspci: 00:04.0 Multimedia audio controller [0401]: nVidia Corporation CK804 AC'97 Audio Controller [10de:0059] (rev a2) Subsystem: ASUSTeK Computer Inc. K8N4-E or A8N-E Mainboard [1043:812a] Flags: bus master, 66MHz, fast devsel, latency 0, IRQ 12 I/O ports at dc00 [size=256] I/O ports at e000 [size=256] Memory at d2003000 (32-bit, non-prefetchable) [size=4K] Capabilities: [44] Power Management version 2 Kernel modules: snd-intel8x0 Full dmesg: Linux version 2.6.31-tmb-desktop-0.rc4.1mdv (mandrake(a)celeste.mandriva.com) (gcc version 4.4.0 (GCC) ) #1 SMP PREEMPT Thu Jul 23 08:39:00 EDT 2009 Command line: BOOT_IMAGE=2.6.31-tmb-desktoprc4-1 root=UUID=30bb01dd-1789-400b-955b-d9b15228ff9f resume=UUID=ace6dc48-3103-4a3a-8ee2-6f9eb76d716b splash=silent vga=788 KERNEL supported cpus: Intel GenuineIntel AMD AuthenticAMD Centaur CentaurHauls BIOS-provided physical RAM map: BIOS-e820: 0000000000000000 - 000000000009f800 (usable) BIOS-e820: 000000000009f800 - 00000000000a0000 (reserved) BIOS-e820: 00000000000f0000 - 0000000000100000 (reserved) BIOS-e820: 0000000000100000 - 00000000bfff0000 (usable) BIOS-e820: 00000000bfff0000 - 00000000bfff3000 (ACPI NVS) BIOS-e820: 00000000bfff3000 - 00000000c0000000 (ACPI data) BIOS-e820: 00000000e0000000 - 00000000f0000000 (reserved) BIOS-e820: 00000000fec00000 - 0000000100000000 (reserved) DMI 2.3 present. Phoenix BIOS detected: BIOS may corrupt low RAM, working around it. e820 update range: 0000000000000000 - 0000000000010000 (usable) ==> (reserved) last_pfn = 0xbfff0 max_arch_pfn = 0x400000000 MTRR default type: uncachable MTRR fixed ranges enabled: 00000-9FFFF write-back A0000-BFFFF uncachable C0000-C7FFF write-protect C8000-FFFFF uncachable MTRR variable ranges enabled: 0 base 0000000000 mask FF80000000 write-back 1 base 0080000000 mask FFC0000000 write-back 2 disabled 3 disabled 4 disabled 5 disabled 6 disabled 7 disabled x86 PAT enabled: cpu 0, old 0x7040600070406, new 0x7010600070106 initial memory mapped : 0 - 20000000 init_memory_mapping: 0000000000000000-00000000bfff0000 0000000000 - 00bfe00000 page 2M 00bfe00000 - 00bfff0000 page 4k kernel direct mapping tables up to bfff0000 @ 10000-15000 RAMDISK: 37b71000 - 37fefde6 ACPI: RSDP 00000000000f7c80 00014 (v00 Nvidia) ACPI: RSDT 00000000bfff3040 00038 (v01 Nvidia AWRDACPI 42302E31 AWRD 00000000) ACPI: FACP 00000000bfff30c0 00074 (v01 Nvidia AWRDACPI 42302E31 AWRD 00000000) ACPI: DSDT 00000000bfff3180 06677 (v01 NVIDIA AWRDACPI 00001000 MSFT 0100000E) ACPI: FACS 00000000bfff0000 00040 ACPI: SSDT 00000000bfff9900 000F7 (v01 PTLTD POWERNOW 00000001 LTP 00000001) ACPI: SRAT 00000000bfff9a40 00090 (v01 AMD HAMMER 00000001 AMD 00000001) ACPI: MCFG 00000000bfff9b40 0003C (v01 Nvidia AWRDACPI 42302E31 AWRD 00000000) ACPI: APIC 00000000bfff9840 0007C (v01 Nvidia AWRDACPI 42302E31 AWRD 00000000) ACPI: Local APIC address 0xfee00000 (7 early reservations) ==> bootmem [0000000000 - 00bfff0000] #0 [0000000000 - 0000001000] BIOS data page ==> [0000000000 - 0000001000] #1 [0000006000 - 0000008000] TRAMPOLINE ==> [0000006000 - 0000008000] #2 [0001000000 - 000158983c] TEXT DATA BSS ==> [0001000000 - 000158983c] #3 [0037b71000 - 0037fefde6] RAMDISK ==> [0037b71000 - 0037fefde6] #4 [000009f800 - 0000100000] BIOS reserved ==> [000009f800 - 0000100000] #5 [000158a000 - 000158a145] BRK ==> [000158a000 - 000158a145] #6 [0000010000 - 0000013000] PGTABLE ==> [0000010000 - 0000013000] found SMP MP-table at [ffff8800000f58d0] f58d0 [ffffea0000000000-ffffea00029fffff] PMD -> [ffff880001a00000-ffff8800043fffff] on node 0 Zone PFN ranges: DMA 0x00000010 -> 0x00001000 DMA32 0x00001000 -> 0x00100000 Normal 0x00100000 -> 0x00100000 Movable zone start PFN for each node early_node_map[2] active PFN ranges 0: 0x00000010 -> 0x0000009f 0: 0x00000100 -> 0x000bfff0 On node 0 totalpages: 786303 DMA zone: 56 pages used for memmap DMA zone: 102 pages reserved DMA zone: 3825 pages, LIFO batch:0 DMA32 zone: 10696 pages used for memmap DMA32 zone: 771624 pages, LIFO batch:31 Nvidia board detected. Ignoring ACPI timer override. If you got timer trouble try acpi_use_timer_override ACPI: PM-Timer IO Port: 0x4008 ACPI: Local APIC address 0xfee00000 ACPI: LAPIC (acpi_id[0x00] lapic_id[0x00] enabled) ACPI: LAPIC (acpi_id[0x01] lapic_id[0x01] disabled) ACPI: LAPIC_NMI (acpi_id[0x00] high edge lint[0x1]) ACPI: LAPIC_NMI (acpi_id[0x01] high edge lint[0x1]) ACPI: IOAPIC (id[0x02] address[0xfec00000] gsi_base[0]) IOAPIC[0]: apic_id 2, version 17, address 0xfec00000, GSI 0-23 ACPI: INT_SRC_OVR (bus 0 bus_irq 0 global_irq 2 dfl dfl) ACPI: BIOS IRQ0 pin2 override ignored. ACPI: INT_SRC_OVR (bus 0 bus_irq 9 global_irq 9 high level) ACPI: INT_SRC_OVR (bus 0 bus_irq 14 global_irq 14 high edge) ACPI: INT_SRC_OVR (bus 0 bus_irq 15 global_irq 15 high edge) ACPI: IRQ9 used by override. ACPI: IRQ14 used by override. ACPI: IRQ15 used by override. Using ACPI (MADT) for SMP configuration information SMP: Allowing 2 CPUs, 1 hotplug CPUs nr_irqs_gsi: 24 PM: Registered nosave memory: 000000000009f000 - 00000000000a0000 PM: Registered nosave memory: 00000000000a0000 - 00000000000f0000 PM: Registered nosave memory: 00000000000f0000 - 0000000000100000 Allocating PCI resources starting at c0000000 (gap: c0000000:20000000) NR_CPUS:16 nr_cpumask_bits:16 nr_cpu_ids:2 nr_node_ids:1 PERCPU: Embedded 25 pages at ffff8800015a7000, static data 71136 bytes Built 1 zonelists in Zone order, mobility grouping on. Total pages: 775449 Kernel command line: BOOT_IMAGE=2.6.31-tmb-desktoprc4-1 root=UUID=30bb01dd-1789-400b-955b-d9b15228ff9f resume=UUID=ace6dc48-3103-4a3a-8ee2-6f9eb76d716b splash=silent vga=788 PID hash table entries: 4096 (order: 12, 32768 bytes) Dentry cache hash table entries: 524288 (order: 10, 4194304 bytes) Inode-cache hash table entries: 262144 (order: 9, 2097152 bytes) Initializing CPU#0 Checking aperture... No AGP bridge found Node 0: aperture @ 1c2000000 size 32 MB Aperture beyond 4GB. Ignoring. Memory: 3084700k/3145664k available (3039k kernel code, 452k absent, 59820k reserved, 1609k data, 416k init) SLUB: Genslabs=13, HWalign=64, Order=0-3, MinObjects=0, CPUs=2, Nodes=1 NR_IRQS:768 Fast TSC calibration using PIT Detected 2211.234 MHz processor. spurious 8259A interrupt: IRQ7. Console: colour dummy device 80x25 console [tty0] enabled Calibrating delay loop (skipped), value calculated using timer frequency.. 4422.46 BogoMIPS (lpj=2211234) Mount-cache hash table entries: 256 CPU: L1 I Cache: 64K (64 bytes/line), D cache 64K (64 bytes/line) CPU: L2 Cache: 512K (64 bytes/line) tseg: 0000000000 mce: CPU supports 5 MCE banks SMP alternatives: switching to UP code ACPI: Core revision 20090521 Setting APIC routing to flat ..TIMER: vector=0x30 apic1=0 pin1=0 apic2=-1 pin2=-1 CPU0: AMD Athlon(tm) 64 Processor 3500+ stepping 0a Brought up 1 CPUs Total of 1 processors activated (4422.46 BogoMIPS). regulator: core version 0.5 NET: Registered protocol family 16 node 0 link 0: io port [1000, fffff] TOM: 00000000c0000000 aka 3072M node 0 link 0: mmio [e0000000, efffffff] node 0 link 0: mmio [feb00000, fec0ffff] node 0 link 0: mmio [a0000, bffff] node 0 link 0: mmio [c0000000, fed3ffff] bus: [00,ff] on node 0 link 0 bus: 00 index 0 io port: [0, ffff] bus: 00 index 1 mmio: [c0000000, fcffffffff] bus: 00 index 2 mmio: [feb00000, fec0ffff] bus: 00 index 3 mmio: [a0000, bffff] ACPI: bus type pci registered PCI: MCFG configuration 0: base e0000000 segment 0 buses 0 - 255 PCI: MCFG area at e0000000 reserved in E820 PCI: Using MMCONFIG at e0000000 - efffffff PCI: Using configuration type 1 for base access bio: create slab <bio-0> at 0 ACPI: EC: Look up EC in DSDT ACPI: Interpreter enabled ACPI: (supports S0 S1 S3 S4 S5) ACPI: Using IOAPIC for interrupt routing ACPI: No dock devices found. ACPI: PCI Root Bridge [PCI0] (0000:00) HPET not enabled in BIOS. You might try hpet=force boot option pci 0000:00:01.1: reg 10 io port: [0xe400-0xe41f] pci 0000:00:01.1: reg 20 io port: [0x4c00-0x4c3f] pci 0000:00:01.1: reg 24 io port: [0x4c40-0x4c7f] pci 0000:00:01.1: PME# supported from D3hot D3cold pci 0000:00:01.1: PME# disabled pci 0000:00:02.0: reg 10 32bit mmio: [0xd2004000-0xd2004fff] pci 0000:00:02.0: supports D1 D2 pci 0000:00:02.0: PME# supported from D0 D1 D2 D3hot D3cold pci 0000:00:02.0: PME# disabled pci 0000:00:02.1: reg 10 32bit mmio: [0xfeb00000-0xfeb000ff] pci 0000:00:02.1: supports D1 D2 pci 0000:00:02.1: PME# supported from D0 D1 D2 D3hot D3cold pci 0000:00:02.1: PME# disabled pci 0000:00:04.0: reg 10 io port: [0xdc00-0xdcff] pci 0000:00:04.0: reg 14 io port: [0xe000-0xe0ff] pci 0000:00:04.0: reg 18 32bit mmio: [0xd2003000-0xd2003fff] pci 0000:00:04.0: supports D1 D2 pci 0000:00:06.0: reg 20 io port: [0xf000-0xf00f] pci 0000:00:07.0: reg 10 io port: [0x9f0-0x9f7] pci 0000:00:07.0: reg 14 io port: [0xbf0-0xbf3] pci 0000:00:07.0: reg 18 io port: [0x970-0x977] pci 0000:00:07.0: reg 1c io port: [0xb70-0xb73] pci 0000:00:07.0: reg 20 io port: [0xd800-0xd80f] pci 0000:00:07.0: reg 24 32bit mmio: [0xd2002000-0xd2002fff] pci 0000:00:08.0: reg 10 io port: [0x9e0-0x9e7] pci 0000:00:08.0: reg 14 io port: [0xbe0-0xbe3] pci 0000:00:08.0: reg 18 io port: [0x960-0x967] pci 0000:00:08.0: reg 1c io port: [0xb60-0xb63] pci 0000:00:08.0: reg 20 io port: [0xc400-0xc40f] pci 0000:00:08.0: reg 24 32bit mmio: [0xd2001000-0xd2001fff] pci 0000:00:0a.0: reg 10 32bit mmio: [0xd2000000-0xd2000fff] pci 0000:00:0a.0: reg 14 io port: [0xb000-0xb007] pci 0000:00:0a.0: supports D1 D2 pci 0000:00:0a.0: PME# supported from D0 D1 D2 D3hot D3cold pci 0000:00:0a.0: PME# disabled pci 0000:00:0b.0: PME# supported from D0 D1 D2 D3hot D3cold pci 0000:00:0b.0: PME# disabled pci 0000:00:0c.0: PME# supported from D0 D1 D2 D3hot D3cold pci 0000:00:0c.0: PME# disabled pci 0000:00:0d.0: PME# supported from D0 D1 D2 D3hot D3cold pci 0000:00:0d.0: PME# disabled pci 0000:00:0e.0: PME# supported from D0 D1 D2 D3hot D3cold pci 0000:00:0e.0: PME# disabled pci 0000:05:0a.0: reg 10 io port: [0x9000-0x9007] pci 0000:05:0a.0: reg 14 io port: [0x9400-0x9403] pci 0000:05:0a.0: reg 18 io port: [0x9800-0x9807] pci 0000:05:0a.0: reg 1c io port: [0x9c00-0x9c03] pci 0000:05:0a.0: reg 20 io port: [0xa000-0xa00f] pci 0000:05:0a.0: reg 24 32bit mmio: [0xd1008000-0xd10083ff] pci 0000:05:0a.0: reg 30 32bit mmio: [0xd0000000-0xd007ffff] pci 0000:05:0a.0: supports D1 D2 pci 0000:05:0b.0: reg 10 32bit mmio: [0xd1009000-0xd10097ff] pci 0000:05:0b.0: reg 14 32bit mmio: [0xd1000000-0xd1003fff] pci 0000:05:0b.0: supports D1 D2 pci 0000:05:0b.0: PME# supported from D0 D1 D2 D3hot pci 0000:05:0b.0: PME# disabled pci 0000:05:0c.0: reg 10 32bit mmio: [0xd1004000-0xd1007fff] pci 0000:05:0c.0: reg 14 io port: [0xa400-0xa4ff] pci 0000:05:0c.0: reg 30 32bit mmio: [0xd0000000-0xd001ffff] pci 0000:05:0c.0: supports D1 D2 pci 0000:05:0c.0: PME# supported from D0 D1 D2 D3hot D3cold pci 0000:05:0c.0: PME# disabled pci 0000:00:09.0: transparent bridge pci 0000:00:09.0: bridge io port: [0x9000-0xafff] pci 0000:00:09.0: bridge 32bit mmio: [0xd0000000-0xd1ffffff] pci 0000:01:00.0: reg 10 32bit mmio: [0xc8000000-0xcbffffff] pci 0000:01:00.0: reg 14 64bit mmio: [0xc0000000-0xc7ffffff] pci 0000:01:00.0: reg 1c 64bit mmio: [0xcc000000-0xccffffff] pci 0000:01:00.0: reg 30 32bit mmio: [0xcd000000-0xcd01ffff] pci 0000:01:00.0: disabling ASPM on pre-1.1 PCIe device. You can enable it with 'pcie_aspm=force' pci 0000:00:0e.0: bridge 32bit mmio: [0xc8000000-0xcfffffff] pci 0000:00:0e.0: bridge 64bit mmio pref: [0xc0000000-0xc7ffffff] pci_bus 0000:00: on NUMA node 0 ACPI: PCI Interrupt Routing Table [\_SB_.PCI0._PRT] ACPI: PCI Interrupt Routing Table [\_SB_.PCI0.HUB0._PRT] ACPI: PCI Interrupt Link [LNK1] (IRQs 3 4 5 7 9 10 11 12 14 15) *0, disabled. ACPI: PCI Interrupt Link [LNK2] (IRQs *3 4 5 7 9 10 11 12 14 15) ACPI: PCI Interrupt Link [LNK3] (IRQs 3 4 5 7 9 10 *11 12 14 15) ACPI: PCI Interrupt Link [LNK4] (IRQs 3 4 *5 7 9 10 11 12 14 15) ACPI: PCI Interrupt Link [LNK5] (IRQs 3 4 5 7 9 10 11 12 14 15) *0, disabled. ACPI: PCI Interrupt Link [LUBA] (IRQs 3 4 5 7 9 10 11 12 14 15) *0, disabled. ACPI: PCI Interrupt Link [LUBB] (IRQs 3 4 5 7 9 10 11 12 14 15) *0, disabled. ACPI: PCI Interrupt Link [LMAC] (IRQs 3 4 *5 7 9 10 11 12 14 15) ACPI: PCI Interrupt Link [LACI] (IRQs 3 4 5 7 9 10 11 *12 14 15) ACPI: PCI Interrupt Link [LMCI] (IRQs 3 4 5 7 9 10 11 12 14 15) *0, disabled. ACPI: PCI Interrupt Link [LSMB] (IRQs 3 4 5 7 9 10 11 12 14 15) *0, disabled. ACPI: PCI Interrupt Link [LUB2] (IRQs 3 4 5 7 9 10 11 12 14 15) *0, disabled. ACPI: PCI Interrupt Link [LIDE] (IRQs 3 4 5 7 9 10 11 12 14 15) *0, disabled. ACPI: PCI Interrupt Link [LSID] (IRQs 3 4 5 7 9 10 *11 12 14 15) ACPI: PCI Interrupt Link [LFID] (IRQs 3 4 *5 7 9 10 11 12 14 15) ACPI: PCI Interrupt Link [LPCA] (IRQs 3 4 5 7 9 10 11 12 14 15) *0, disabled. ACPI: PCI Interrupt Link [APC1] (IRQs 16) *0, disabled. ACPI: PCI Interrupt Link [APC2] (IRQs 17) *0 ACPI: PCI Interrupt Link [APC3] (IRQs 18) *0 ACPI: PCI Interrupt Link [APC4] (IRQs 19) *0 ACPI: PCI Interrupt Link [APC5] (IRQs *16), disabled. ACPI: PCI Interrupt Link [APCF] (IRQs 20 21 22 23) *0, disabled. ACPI: PCI Interrupt Link [APCG] (IRQs 20 21 22 23) *0, disabled. ACPI: PCI Interrupt Link [APCH] (IRQs 20 21 22 23) *0 ACPI: PCI Interrupt Link [APCJ] (IRQs 20 21 22 23) *0 ACPI: PCI Interrupt Link [APCK] (IRQs 20 21 22 23) *0, disabled. ACPI: PCI Interrupt Link [APCS] (IRQs 20 21 22 23) *0, disabled. ACPI: PCI Interrupt Link [APCL] (IRQs 20 21 22 23) *0, disabled. ACPI: PCI Interrupt Link [APCZ] (IRQs 20 21 22 23) *0, disabled. ACPI: PCI Interrupt Link [APSI] (IRQs 20 21 22 23) *0 ACPI: PCI Interrupt Link [APSJ] (IRQs 20 21 22 23) *0 ACPI: PCI Interrupt Link [APCP] (IRQs 20 21 22 23) *0, disabled. PCI: Using ACPI for IRQ routing pnp: PnP ACPI init ACPI: bus type pnp registered pnp: PnP ACPI: found 15 devices ACPI: ACPI bus type pnp unregistered system 00:01: ioport range 0x4000-0x407f has been reserved system 00:01: ioport range 0x4080-0x40ff has been reserved system 00:01: ioport range 0x4400-0x447f has been reserved system 00:01: ioport range 0x4480-0x44ff has been reserved system 00:01: ioport range 0x4800-0x487f has been reserved system 00:01: ioport range 0x4880-0x48ff has been reserved system 00:02: ioport range 0x4d0-0x4d1 has been reserved system 00:02: ioport range 0x800-0x87f has been reserved system 00:02: ioport range 0x290-0x297 has been reserved system 00:0d: iomem range 0xe0000000-0xefffffff has been reserved system 00:0e: iomem range 0xf0000-0xf3fff could not be reserved system 00:0e: iomem range 0xf4000-0xf7fff could not be reserved system 00:0e: iomem range 0xf8000-0xfbfff could not be reserved system 00:0e: iomem range 0xfc000-0xfffff could not be reserved system 00:0e: iomem range 0xbfff0000-0xbfffffff could not be reserved system 00:0e: iomem range 0xffff0000-0xffffffff has been reserved system 00:0e: iomem range 0x0-0x9ffff could not be reserved system 00:0e: iomem range 0x100000-0xbffeffff could not be reserved system 00:0e: iomem range 0xfec00000-0xfec00fff could not be reserved system 00:0e: iomem range 0xfee00000-0xfeefffff has been reserved system 00:0e: iomem range 0xfefff000-0xfeffffff has been reserved system 00:0e: iomem range 0xfff80000-0xfff80fff has been reserved system 00:0e: iomem range 0xfff90000-0xfffbffff has been reserved system 00:0e: iomem range 0xfffed000-0xfffeffff has been reserved pci 0000:00:09.0: PCI bridge, secondary bus 0000:05 pci 0000:00:09.0: IO window: 0x9000-0xafff pci 0000:00:09.0: MEM window: 0xd0000000-0xd1ffffff pci 0000:00:09.0: PREFETCH window: disabled pci 0000:00:0b.0: PCI bridge, secondary bus 0000:04 pci 0000:00:0b.0: IO window: disabled pci 0000:00:0b.0: MEM window: disabled pci 0000:00:0b.0: PREFETCH window: disabled pci 0000:00:0c.0: PCI bridge, secondary bus 0000:03 pci 0000:00:0c.0: IO window: disabled pci 0000:00:0c.0: MEM window: disabled pci 0000:00:0c.0: PREFETCH window: disabled pci 0000:00:0d.0: PCI bridge, secondary bus 0000:02 pci 0000:00:0d.0: IO window: disabled pci 0000:00:0d.0: MEM window: disabled pci 0000:00:0d.0: PREFETCH window: disabled pci 0000:00:0e.0: PCI bridge, secondary bus 0000:01 pci 0000:00:0e.0: IO window: disabled pci 0000:00:0e.0: MEM window: 0xc8000000-0xcfffffff pci 0000:00:0e.0: PREFETCH window: 0x000000c0000000-0x000000c7ffffff pci 0000:00:09.0: setting latency timer to 64 pci 0000:00:0b.0: setting latency timer to 64 pci 0000:00:0c.0: setting latency timer to 64 pci 0000:00:0d.0: setting latency timer to 64 pci 0000:00:0e.0: setting latency timer to 64 pci_bus 0000:00: resource 0 io: [0x00-0xffff] pci_bus 0000:00: resource 1 mem: [0x000000-0xffffffffffffffff] pci_bus 0000:05: resource 0 io: [0x9000-0xafff] pci_bus 0000:05: resource 1 mem: [0xd0000000-0xd1ffffff] pci_bus 0000:05: resource 3 io: [0x00-0xffff] pci_bus 0000:05: resource 4 mem: [0x000000-0xffffffffffffffff] pci_bus 0000:01: resource 1 mem: [0xc8000000-0xcfffffff] pci_bus 0000:01: resource 2 pref mem [0xc0000000-0xc7ffffff] NET: Registered protocol family 2 IP route cache hash table entries: 131072 (order: 8, 1048576 bytes) Switched to high resolution mode on CPU 0 TCP established hash table entries: 262144 (order: 10, 4194304 bytes) TCP bind hash table entries: 65536 (order: 8, 1048576 bytes) TCP: Hash tables configured (established 262144 bind 65536) TCP reno registered NET: Registered protocol family 1 Trying to unpack rootfs image as initramfs... Freeing initrd memory: 4603k freed audit: initializing netlink socket (disabled) type=2000 audit(1248431174.234:1): initialized HugeTLB registered 2 MB page size, pre-allocated 0 pages VFS: Disk quotas dquot_6.5.2 Dquot-cache hash table entries: 512 (order 0, 4096 bytes) msgmni has been set to 6035 alg: No test for stdrng (krng) Block layer SCSI generic (bsg) driver version 0.4 loaded (major 253) io scheduler noop registered io scheduler anticipatory registered io scheduler deadline registered io scheduler cfq registered (default) pci 0000:00:00.0: Found enabled HT MSI Mapping pci 0000:00:0b.0: Found disabled HT MSI Mapping pci 0000:00:00.0: Found enabled HT MSI Mapping pci 0000:00:0b.0: Linking AER extended capability pci 0000:00:00.0: Found enabled HT MSI Mapping pci 0000:00:0c.0: Found disabled HT MSI Mapping pci 0000:00:00.0: Found enabled HT MSI Mapping pci 0000:00:0c.0: Linking AER extended capability pci 0000:00:00.0: Found enabled HT MSI Mapping pci 0000:00:0d.0: Found disabled HT MSI Mapping pci 0000:00:00.0: Found enabled HT MSI Mapping pci 0000:00:0d.0: Linking AER extended capability pci 0000:00:00.0: Found enabled HT MSI Mapping pci 0000:00:0e.0: Found disabled HT MSI Mapping pci 0000:00:00.0: Found enabled HT MSI Mapping pci 0000:00:0e.0: Linking AER extended capability pci 0000:01:00.0: Boot video device pcieport-driver 0000:00:0b.0: irq 24 for MSI/MSI-X pcieport-driver 0000:00:0b.0: setting latency timer to 64 pcieport-driver 0000:00:0c.0: irq 25 for MSI/MSI-X pcieport-driver 0000:00:0c.0: setting latency timer to 64 pcieport-driver 0000:00:0d.0: irq 26 for MSI/MSI-X pcieport-driver 0000:00:0d.0: setting latency timer to 64 pcieport-driver 0000:00:0e.0: irq 27 for MSI/MSI-X pcieport-driver 0000:00:0e.0: setting latency timer to 64 vesafb: framebuffer at 0xc0000000, mapped to 0xffffc90010100000, using 1875k, total 131072k vesafb: mode is 800x600x16, linelength=1600, pages=2 vesafb: scrolling: redraw vesafb: Truecolor: size=0:5:6:5, shift=0:11:5:0 Console: switching to colour frame buffer device 100x37 fb0: VESA VGA frame buffer device Linux agpgart interface v0.103 Serial: 8250/16550 driver, 4 ports, IRQ sharing enabled serial8250: ttyS0 at I/O 0x3f8 (irq = 4) is a 16550A Platform driver 'serial8250' needs updating - please use dev_pm_ops 00:08: ttyS0 at I/O 0x3f8 (irq = 4) is a 16550A brd: module loaded input: Macintosh mouse button emulation as /devices/virtual/input/input0 PNP: PS/2 Controller [PNP0303:PS2K] at 0x60,0x64 irq 1 PNP: PS/2 appears to have AUX port disabled, if this is incorrect please boot with i8042.nopnp Platform driver 'i8042' needs updating - please use dev_pm_ops serio: i8042 KBD port at 0x60,0x64 irq 1 mice: PS/2 mouse device common for all mice cpuidle: using governor ladder cpuidle: using governor menu TCP cubic registered PM: Checking image partition UUID=ace6dc48-3103-4a3a-8ee2-6f9eb76d716b input: AT Translated Set 2 keyboard as /devices/platform/i8042/serio0/input/input1 PM: Resume from disk failed. registered taskstats version 1 BIOS EDD facility v0.16 2004-Jun-25, 1 devices found Freeing unused kernel memory: 416k freed device-mapper: uevent: version 1.0.3 device-mapper: ioctl: 4.15.0-ioctl (2009-04-01) initialised: dm-devel(a)redhat.com SCSI subsystem initialized libata version 3.00 loaded. sata_nv 0000:00:07.0: version 3.5 ACPI: PCI Interrupt Link [APSI] enabled at IRQ 23 sata_nv 0000:00:07.0: PCI INT A -> Link[APSI] -> GSI 23 (level, low) -> IRQ 23 sata_nv 0000:00:07.0: setting latency timer to 64 scsi0 : sata_nv scsi1 : sata_nv ata1: SATA max UDMA/133 cmd 0x9f0 ctl 0xbf0 bmdma 0xd800 irq 23 ata2: SATA max UDMA/133 cmd 0x970 ctl 0xb70 bmdma 0xd808 irq 23 ACPI: PCI Interrupt Link [APSJ] enabled at IRQ 22 sata_nv 0000:00:08.0: PCI INT A -> Link[APSJ] -> GSI 22 (level, low) -> IRQ 22 sata_nv 0000:00:08.0: setting latency timer to 64 ata1: SATA link down (SStatus 0 SControl 300) scsi2 : sata_nv scsi3 : sata_nv ata3: SATA max UDMA/133 cmd 0x9e0 ctl 0xbe0 bmdma 0xc400 irq 22 ata4: SATA max UDMA/133 cmd 0x960 ctl 0xb60 bmdma 0xc408 irq 22 ata3: SATA link down (SStatus 0 SControl 300) ata2: SATA link up 1.5 Gbps (SStatus 113 SControl 300) ata2.00: ATA-6: ST3200822AS, 3.01, max UDMA/133 ata2.00: 390721968 sectors, multi 1: LBA48 ata2.00: configured for UDMA/133 scsi 1:0:0:0: Direct-Access ATA ST3200822AS 3.01 PQ: 0 ANSI: 5 sd 1:0:0:0: [sda] 390721968 512-byte logical blocks: (200 GB/186 GiB) sd 1:0:0:0: [sda] Write Protect is off sd 1:0:0:0: [sda] Mode Sense: 00 3a 00 00 sd 1:0:0:0: [sda] Write cache: enabled, read cache: enabled, doesn't support DPO or FUA sda: ata4: SATA link down (SStatus 0 SControl 300) sda1 sda2 < sda5 sda6 sda7 sda8 sda9 > sda3 sd 1:0:0:0: [sda] Attached SCSI disk pata_acpi 0000:00:06.0: setting latency timer to 64 Uniform Multi-Platform E-IDE driver ide-gd driver 1.18 sata_sil 0000:05:0a.0: version 2.4 ACPI: PCI Interrupt Link [APC4] enabled at IRQ 19 sata_sil 0000:05:0a.0: PCI INT A -> Link[APC4] -> GSI 19 (level, low) -> IRQ 19 sata_sil 0000:05:0a.0: Applying R_ERR on DMA activate FIS errata fix scsi4 : sata_sil scsi5 : sata_sil scsi6 : sata_sil scsi7 : sata_sil ata5: SATA max UDMA/100 mmio m1024@0xd1008000 tf 0xd1008080 irq 19 ata6: SATA max UDMA/100 mmio m1024@0xd1008000 tf 0xd10080c0 irq 19 ata7: SATA max UDMA/100 mmio m1024@0xd1008000 tf 0xd1008280 irq 19 ata8: SATA max UDMA/100 mmio m1024@0xd1008000 tf 0xd10082c0 irq 19 ata5: SATA link down (SStatus 0 SControl 310) ata6: SATA link down (SStatus 0 SControl 310) ata7: SATA link down (SStatus 0 SControl 310) ata8: SATA link down (SStatus 0 SControl 310) pata_amd 0000:00:06.0: version 0.4.1 pata_amd 0000:00:06.0: setting latency timer to 64 scsi8 : pata_amd scsi9 : pata_amd ata9: PATA max UDMA/133 cmd 0x1f0 ctl 0x3f6 bmdma 0xf000 irq 14 ata10: PATA max UDMA/133 cmd 0x170 ctl 0x376 bmdma 0xf008 irq 15 ata9.00: ATAPI: LITE-ON DVD SOHD-167T, 9IS1, max UDMA/33 ata9: nv_mode_filter: 0x739f&0x701f->0x701f, BIOS=0x7000 (0xc000c000) ACPI=0x701f (60:600:0x13) ata9.00: configured for UDMA/33 scsi 8:0:0:0: CD-ROM LITE-ON DVD SOHD-167T 9IS1 PQ: 0 ANSI: 5 ata10.00: ATAPI: _NEC DVD_RW ND-3520AW, 3.06, max UDMA/33 ata10: nv_mode_filter: 0x739f&0x701f->0x701f, BIOS=0x7000 (0xc000c000) ACPI=0x701f (60:600:0x13) ata10.00: configured for UDMA/33 scsi 9:0:0:0: CD-ROM _NEC DVD_RW ND-3520AW 3.06 PQ: 0 ANSI: 5 ide_generic: please use "probe_mask=0x3f" module parameter for probing all legacy ISA IDE ports EXT4-fs (dm-2): barriers enabled kjournald2 starting: pid 449, dev dm-2:8, commit interval 5 seconds EXT4-fs (dm-2): delayed allocation enabled EXT4-fs: file extents enabled EXT4-fs: mballoc enabled EXT4-fs (dm-2): mounted filesystem with ordered data mode usbcore: registered new interface driver usbfs usbcore: registered new interface driver hub usbcore: registered new device driver usb udev: starting version 145 input: Power Button as /devices/LNXSYSTM:00/LNXPWRBN:00/input/input2 ACPI: Power Button [PWRF] input: Power Button as /devices/LNXSYSTM:00/device:00/PNP0C0C:00/input/input3 ACPI: Power Button [PWRB] processor LNXCPU:00: registered as cooling_device0 thermal LNXTHERM:01: registered as thermal_zone0 ACPI: Thermal Zone [THRM] (40 C) forcedeth: Reverse Engineered nForce ethernet driver. Version 0.64. ACPI: PCI Interrupt Link [APCH] enabled at IRQ 21 forcedeth 0000:00:0a.0: PCI INT A -> Link[APCH] -> GSI 21 (level, low) -> IRQ 21 forcedeth 0000:00:0a.0: setting latency timer to 64 nv_probe: set workaround bit for reversed mac addr gameport: NS558 PnP Gameport is pnp00:0c/gameport0, io 0x201, speed 922kHz fan PNP0C0B:00: registered as cooling_device1 ACPI: Fan [FAN] (on) ehci_hcd: USB 2.0 'Enhanced' Host Controller (EHCI) Driver ohci_hcd: USB 1.1 'Open' Host Controller (OHCI) Driver sd 1:0:0:0: Attached scsi generic sg0 type 0 scsi 8:0:0:0: Attached scsi generic sg1 type 5 scsi 9:0:0:0: Attached scsi generic sg2 type 5 Platform driver 'floppy' needs updating - please use dev_pm_ops Floppy drive(s): fd0 is 1.44M EDAC MC: Ver: 2.1.0 Jul 23 2009 FDC 0 is a post-1991 82077 parport_pc 00:09: reported by Plug and Play ACPI parport0: PC-style at 0x378 (0x778), irq 7, dma 3 [PCSPP,TRISTATE,COMPAT,ECP,DMA] EDAC amd64_edac: Ver: 3.2.0 Jul 23 2009 ACPI: PCI Interrupt Link [APC2] enabled at IRQ 17 skge 0000:05:0c.0: PCI INT A -> Link[APC2] -> GSI 17 (level, low) -> IRQ 17 skge 1.13 addr 0xd1004000 irq 17 chip Yukon-Lite rev 9 skge eth0: addr 00:11:d8:98:22:aa sr0: scsi3-mmc drive: 48x/48x cd/rw xa/form2 cdda tray Uniform CD-ROM driver Revision: 3.20 sr 8:0:0:0: Attached scsi CD-ROM sr0 sr1: scsi3-mmc drive: 48x/48x writer cd/rw xa/form2 cdda tray sr 9:0:0:0: Attached scsi CD-ROM sr1 udev: renamed network interface eth0 to eth1 rtc_cmos 00:04: RTC can wake from S4 rtc_cmos 00:04: rtc core: registered rtc_cmos as rtc0 rtc0: alarms up to one year, y3k, 242 bytes nvram ACPI: PCI Interrupt Link [APC1] enabled at IRQ 16 ohci1394 0000:05:0b.0: PCI INT A -> Link[APC1] -> GSI 16 (level, low) -> IRQ 16 ohci1394: fw-host0: OHCI-1394 1.1 (PCI): IRQ=[16] MMIO=[d1009000-d10097ff] Max Packet=[2048] IR/IT contexts=[4/8] ppdev: user-space parallel port driver forcedeth 0000:00:0a.0: ifname eth0, PHY OUI 0x5043 @ 9, addr 00:11:d8:95:74:ec forcedeth 0000:00:0a.0: highdma csum gbit lnktim desc-v3 i2c-adapter i2c-0: nForce2 SMBus adapter at 0x4c00 i2c-adapter i2c-1: nForce2 SMBus adapter at 0x4c40 ACPI: PCI Interrupt Link [APCL] enabled at IRQ 20 ehci_hcd 0000:00:02.1: PCI INT B -> Link[APCL] -> GSI 20 (level, low) -> IRQ 20 ehci_hcd 0000:00:02.1: setting latency timer to 64 ehci_hcd 0000:00:02.1: EHCI Host Controller ehci_hcd 0000:00:02.1: new USB bus registered, assigned bus number 1 ehci_hcd 0000:00:02.1: debug port 1 ehci_hcd 0000:00:02.1: cache line size of 64 is not supported ehci_hcd 0000:00:02.1: irq 20, io mem 0xfeb00000 ehci_hcd 0000:00:02.1: USB 2.0 started, EHCI 1.00 usb usb1: New USB device found, idVendor=1d6b, idProduct=0002 usb usb1: New USB device strings: Mfr=3, Product=2, SerialNumber=1 usb usb1: Product: EHCI Host Controller usb usb1: Manufacturer: Linux 2.6.31-tmb-desktop-0.rc4.1mdv ehci_hcd usb usb1: SerialNumber: 0000:00:02.1 usb usb1: configuration #1 chosen from 1 choice hub 1-0:1.0: USB hub found hub 1-0:1.0: 10 ports detected ACPI: PCI Interrupt Link [APCF] enabled at IRQ 23 ohci_hcd 0000:00:02.0: PCI INT A -> Link[APCF] -> GSI 23 (level, low) -> IRQ 23 ohci_hcd 0000:00:02.0: setting latency timer to 64 ohci_hcd 0000:00:02.0: OHCI Host Controller ohci_hcd 0000:00:02.0: new USB bus registered, assigned bus number 2 ohci_hcd 0000:00:02.0: irq 23, io mem 0xd2004000 usb usb2: New USB device found, idVendor=1d6b, idProduct=0001 usb usb2: New USB device strings: Mfr=3, Product=2, SerialNumber=1 usb usb2: Product: OHCI Host Controller usb usb2: Manufacturer: Linux 2.6.31-tmb-desktop-0.rc4.1mdv ohci_hcd usb usb2: SerialNumber: 0000:00:02.0 usb usb2: configuration #1 chosen from 1 choice hub 2-0:1.0: USB hub found hub 2-0:1.0: 10 ports detected EDAC amd64: This node reports that Memory ECC is currently disabled. EDAC amd64: bit 0x400000 in register F3x44 of the MISC_CONTROL device (0000:00:18.3) should be enabled EDAC amd64: WARNING: ECC is NOT currently enabled by the BIOS. Module will NOT be loaded. Either Enable ECC in the BIOS, or use the 'ecc_enable_override' parameter. Might be a BIOS bug, if BIOS says ECC is enabled Use of the override can cause unknown side effects. amd64_edac: probe of 0000:00:18.2 failed with error -22 Platform driver 'pcspkr' needs updating - please use dev_pm_ops input: PC Speaker as /devices/platform/pcspkr/input/input4 Error: Driver 'pcspkr' is already registered, aborting... cannot find the slot for index 0 (range 0-1), error: -16 Intel ICH: probe of 0000:00:04.0 failed with error -16 usb 2-2: new low speed USB device using ohci_hcd and address 2 usb 2-2: New USB device found, idVendor=046d, idProduct=c01d usb 2-2: New USB device strings: Mfr=1, Product=2, SerialNumber=0 usb 2-2: Product: USB-PS/2 Optical Mouse usb 2-2: Manufacturer: Logitech usb 2-2: configuration #1 chosen from 1 choice powernow-k8: Found 1 AMD Athlon(tm) 64 Processor 3500+ processors (1 cpu cores) (version 2.20.00) powernow-k8: 0 : fid 0xe (2200 MHz), vid 0x2 powernow-k8: 1 : fid 0xc (2000 MHz), vid 0x6 powernow-k8: 2 : fid 0xa (1800 MHz), vid 0xa powernow-k8: 3 : fid 0x2 (1000 MHz), vid 0x12 Error: Driver 'pcspkr' is already registered, aborting... usbcore: registered new interface driver hiddev input: Logitech USB-PS/2 Optical Mouse as /devices/pci0000:00/0000:00:02.0/usb2/2-2/2-2:1.0/input/input5 generic-usb 0003:046D:C01D.0001: input,hidraw0: USB HID v1.10 Mouse [Logitech USB-PS/2 Optical Mouse] on usb-0000:00:02.0-2/input0 usbcore: registered new interface driver usbhid usbhid: v2.6:USB HID core driver usb 2-3: new full speed USB device using ohci_hcd and address 3 ieee1394: Host added: ID:BUS[0-00:1023] GUID[0011d80000132b85] usb 2-3: New USB device found, idVendor=03f0, idProduct=3404 usb 2-3: New USB device strings: Mfr=1, Product=2, SerialNumber=3 usb 2-3: Product: deskjet 6122 usb 2-3: Manufacturer: hp usb 2-3: SerialNumber: MY2881B2HY64 usb 2-3: configuration #1 chosen from 1 choice usblp0: USB Bidirectional printer dev 3 if 0 alt 0 proto 2 vid 0x03F0 pid 0x3404 usbcore: registered new interface driver usblp EXT4-fs (dm-2): internal journal on dm-2:8 kjournald starting. Commit interval 5 seconds EXT3 FS on sda6, internal journal EXT3-fs: mounted filesystem with writeback data mode. SGI XFS with ACLs, security attributes, realtime, large block/inode numbers, no debug enabled SGI XFS Quota Management subsystem XFS mounting filesystem dm-4 Ending clean XFS mount for filesystem: dm-4 XFS mounting filesystem dm-1 Ending clean XFS mount for filesystem: dm-1 XFS mounting filesystem dm-0 Ending clean XFS mount for filesystem: dm-0 fuse init (API version 7.12) EXT4-fs (dm-3): barriers enabled kjournald2 starting: pid 953, dev dm-3:8, commit interval 5 seconds EXT4-fs (dm-3): internal journal on dm-3:8 EXT4-fs (dm-3): delayed allocation enabled EXT4-fs: file extents enabled EXT4-fs: mballoc enabled EXT4-fs (dm-3): mounted filesystem with ordered data mode loop: module loaded Adding 4883720k swap on /dev/sda8. Priority:-1 extents:1 across:4883720k $ cat /proc/interrupts CPU0 0: 127 IO-APIC-edge timer 1: 7417 IO-APIC-edge i8042 4: 1 IO-APIC-edge 6: 11 IO-APIC-edge floppy 7: 1 IO-APIC-edge parport0 8: 1 IO-APIC-edge rtc0 9: 0 IO-APIC-fasteoi acpi 10: 0 IO-APIC-edge MPU401 UART 14: 56298 IO-APIC-edge pata_amd 15: 75410 IO-APIC-edge pata_amd 16: 3 IO-APIC-fasteoi ohci1394 17: 0 IO-APIC-fasteoi eth1 18: 640221 IO-APIC-fasteoi nvidia 19: 0 IO-APIC-fasteoi sata_sil 20: 35875 IO-APIC-fasteoi ehci_hcd:usb1 21: 215300 IO-APIC-fasteoi eth0 22: 0 IO-APIC-fasteoi sata_nv 23: 327934 IO-APIC-fasteoi sata_nv, ohci_hcd:usb2 NMI: 0 Non-maskable interrupts LOC: 25790888 Local timer interrupts SPU: 0 Spurious interrupts CNT: 0 Performance counter interrupts PND: 0 Performance pending work RES: 0 Rescheduling interrupts CAL: 0 Function call interrupts TLB: 0 TLB shootdowns TRM: 0 Thermal event interrupts THR: 0 Threshold APIC interrupts MCE: 0 Machine check exceptions MCP: 35 Machine check polls ERR: 1 MIS: 0 $ lsmod | grep snd snd_intel8x0 35368 0 snd_ac97_codec 122808 1 snd_intel8x0 ac97_bus 1952 1 snd_ac97_codec snd_mpu401 8072 0 snd_mpu401_uart 7936 1 snd_mpu401 snd_seq_dummy 3172 0 snd_pcsp 11404 0 snd_rawmidi 23712 1 snd_mpu401_uart snd_seq_oss 31456 0 snd_seq_midi_event 7296 1 snd_seq_oss snd_seq 56864 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi_event snd_seq_device 7188 4 snd_seq_dummy,snd_rawmidi,snd_seq_oss,snd_seq snd_pcm_oss 42752 0 snd_pcm 80968 4 snd_intel8x0,snd_ac97_codec,snd_pcsp,snd_pcm_oss snd_timer 22736 2 snd_seq,snd_pcm snd_mixer_oss 16736 1 snd_pcm_oss snd 65288 13 snd_intel8x0,snd_ac97_codec,snd_mpu401,snd_mpu401_uart,snd_pcsp,snd_rawmidi,snd_seq_oss,snd_seq,snd_seq_device,snd_pcm_oss,snd_pcm,snd_timer,snd_mixer_oss soundcore 7904 1 snd snd_page_alloc 9552 2 snd_intel8x0,snd_pcm

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[alsa-devel] [PATCH] Switch palm27x-asoc to jack detection api
by Marek Vasut 24 Jul '09

24 Jul '09

This patch removes the old method of jack detection from palm27x-asoc driver and adds jack detection api. It also removes some other (now) useless stuff from the driver and corrects pin configuration for the codec. Please consider applying.

2 1

[alsa-devel] [PATCH] ASoC: Jack handling enhancements as suggested by subsystem maintainer
by Janusz Krzysztofik 24 Jul '09

24 Jul '09

The patch adds a few small enhancements to the ASoC jack handling, as suggested by Mark in his comments to my Amstrad Delta driver, and a few fixes for related bugs found while learning Mark's code and testing results. Enhancements: 1. Update status of an ASoC jack while associating it with new gpios. 2. Really update DAPM pins while associating them with an ASoC jack. 3. Export ASoC jack gpios over gpiolib sysfs for diagnostic purposes. Fixes: 1. Apply mask on jack status report before using it, just for case. 2. While updating jack associated DAPM pins, use full resulting jack status, not the status report passed as an argument. Created and tested on linux-2.6.31-rc3 Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- --- linux-2.6.31-rc3/sound/soc/soc-jack.c.orig 2009-07-14 03:18:52.000000000 +0200 +++ linux-2.6.31-rc3/sound/soc/soc-jack.c 2009-07-24 02:47:06.000000000 +0200 @@ -73,14 +73,15 @@ void snd_soc_jack_report(struct snd_soc_ oldstatus = jack->status; jack->status &= ~mask; - jack->status |= status; + jack->status |= status & mask; - /* The DAPM sync is expensive enough to be worth skipping */ - if (jack->status == oldstatus) + /* The DAPM sync is expensive enough to be worth skipping. + * However, empty mask means pin synchronization is desired. */ + if (mask && (jack->status == oldstatus)) goto out; list_for_each_entry(pin, &jack->pins, list) { - enable = pin->mask & status; + enable = pin->mask & jack->status; if (pin->invert) enable = !enable; @@ -228,8 +229,16 @@ int snd_soc_jack_add_gpios(struct snd_so if (ret) goto err; +#ifdef CONFIG_GPIO_SYSFS + /* Expose GPIO value over sysfs for diagnostic purposes */ + gpio_export(gpios[i].gpio, false); +#endif + INIT_WORK(&gpios[i].work, gpio_work); gpios[i].jack = jack; + + /* Update initial jack status */ + snd_soc_jack_gpio_detect(&gpios[i]); } return 0; @@ -258,6 +267,9 @@ void snd_soc_jack_free_gpios(struct snd_ int i; for (i = 0; i < count; i++) { +#ifdef CONFIG_GPIO_SYSFS + gpio_unexport(gpios[i].gpio); +#endif free_irq(gpio_to_irq(gpios[i].gpio), &gpios[i]); gpio_free(gpios[i].gpio); gpios[i].jack = NULL;

2 1

[alsa-devel] ASoC: about the array to cache registers
by Joonyoung Shim 24 Jul '09

24 Jul '09

Hi, Many ASoC codec drivers have an array to cache registers. The array is not used any longer after memcpy to codec->reg_cache. I think that memcpy is unnecessary, and it a waste of the memory. How about continue using the array instead of memcpy? For example, i modified such following patch in wm8731.c diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 1560028..43f8de3 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -36,7 +36,6 @@ struct snd_soc_codec_device soc_codec_dev_wm8731; /* codec private data */ struct wm8731_priv { struct snd_soc_codec codec; - u16 reg_cache[WM8731_CACHEREGNUM]; unsigned int sysclk; }; @@ -50,7 +49,7 @@ static int wm8731_spi_write(struct spi_device *spi, const char *data, int len); * using 2 wire for device control, so we cache them instead. * There is no point in caching the reset register */ -static const u16 wm8731_reg[WM8731_CACHEREGNUM] = { +static u16 wm8731_reg[WM8731_CACHEREGNUM] = { 0x0097, 0x0097, 0x0079, 0x0079, 0x000a, 0x0008, 0x009f, 0x000a, 0x0000, 0x0000 @@ -586,9 +585,7 @@ static int wm8731_register(struct wm8731_priv *wm8731) codec->dai = &wm8731_dai; codec->num_dai = 1; codec->reg_cache_size = WM8731_CACHEREGNUM; - codec->reg_cache = &wm8731->reg_cache; - - memcpy(codec->reg_cache, wm8731_reg, sizeof(wm8731_reg)); + codec->reg_cache = &wm8731_reg; ret = wm8731_reset(codec); if (ret < 0) {

3 2

[alsa-devel] get two mono channel from stereo line
by arif setiawan 24 Jul '09

24 Jul '09

Hi, We have a beagleboard with one audio line-in and one audio line-out. From the documentation both are stereo line. Now we want to use four mono audio line for our system. two input from radio and microphone, two output to radio and speaker. We use ALSA and we would like to know how we can achieve this either by plugin or programming alsa. Thank you for your help Arif

2 1

[alsa-devel] Is snd_pcm_hw_params_set_period_time_near the same as sampling period?
by Guilherme Longo 24 Jul '09

24 Jul '09

Hi mates. I am trying to understand a piece of a code I am studing . What does exactly the functions snd_pcm_hw_params_set_period_time_near sets? In my understand period time should be the time between 2 consecutive samples, but the following code does not appears to be calculating that. buffer_time = 500000 // What I believe that means 0.5s as US is the unit used. if (period_time == 0 && period_frames == 0) { if (buffer_time > 0) period_time = buffer_time / 4; else period_frames = buffer_frames / 4; } Is this period time the same as period describled in here: http://www.fon.hum.uva.nl/praat/manual/sampling_period.html Thanks!

2 1

[alsa-devel] get two mono channel from stereo line
by arif setiawan 24 Jul '09

24 Jul '09

Hi, We have a beagleboard with one audio line-in and one audio line-out. From the documentation both are stereo line. Now we want to use four mono audio line for our system. two input from radio and microphone, two output to radio and speaker. We use ALSA and we would like to know how we can achieve this either by plugin or programming alsa. Thank you for your help Arif

1 0

[alsa-devel] multi-channel playback regression
by Wu Fengguang 24 Jul '09

24 Jul '09

Hi Takashi, When doing multi-channel playback tests on IbexPeak, I found that the following patch makes the playback enter an infinite loop, repeatedly playing a range of ~0.5s audio content. (Seems that some buffer pointer can never advance.) Thanks, Fengguang --- 79452f0a28aa5a40522c487b42a5fc423647ad98 Author: Takashi Iwai <tiwai(a)suse.de> Date: Wed Jul 22 12:51:51 2009 +0200 ALSA: pcm - Fix regressions with VMware VMware tends to report PCM positions and period updates at utterly wrong timing. This screws up the recent PCM core code that tries to correct the position based on the irq timing. Now, when a backward irq position is detected, skip the update instead of rebasing. (This is almost the old behavior before 2.6.30.) Signed-off-by: Takashi Iwai <tiwai(a)suse.de> diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 333e4dd..3b673e2 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -244,18 +244,27 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) delta = new_hw_ptr - hw_ptr_interrupt; } if (delta < 0) { - delta += runtime->buffer_size; + if (runtime->periods == 1) + delta += runtime->buffer_size; if (delta < 0) { hw_ptr_error(substream, "Unexpected hw_pointer value " "(stream=%i, pos=%ld, intr_ptr=%ld)\n", substream->stream, (long)pos, (long)hw_ptr_interrupt); +#if 1 + /* simply skipping the hwptr update seems more + * robust in some cases, e.g. on VMware with + * inaccurate timer source + */ + return 0; /* skip this update */ +#else /* rebase to interrupt position */ hw_base = new_hw_ptr = hw_ptr_interrupt; /* align hw_base to buffer_size */ hw_base -= hw_base % runtime->buffer_size;

3 21

[alsa-devel] snd_pcm_hw_params _set_period_time_near
by Guilherme Longo 24 Jul '09

24 Jul '09

Mates, I can´t find the real meaning of this function. At first I thought that it was used to set the time between 2 consecutive samples, but it is not. Does anyone could gimme some help with that? Tks

1 0

[alsa-devel] SoC codec read callback definition
by Lopez Cruz, Misael 23 Jul '09

23 Jul '09

read callback (of snd_soc_codec struct) is defined as a function that returns unsigned int: struct snd_soc_codec { ... unsigned int (*read)(struct snd_soc_codec *, unsigned int); int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); ... } However, actual implementation of that callback in many CODEC drivers may return a negative error value (i.e. when a register is out of cache reg size). Should the callback definition be changed to return _int_ instead of _unsigned int_? Some drivers (like wm8900, wm9081) handle invalid registers with BUG_ON, in that case it's not necessary to return the negative error code. What should be the best way to handle this? Or is it ok as currently implemented? Well, it was just a thought. -Misa

2 1

[alsa-devel] [PATCH] ALSA: snd_usb_caiaq: add support for Audio2DJ
by Daniel Mack 23 Jul '09

23 Jul '09

This adds support for Native Instrument's freshly announced Audio2DJ sound device hardware. Version number bumped to 1.3.19. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/Kconfig | 1 + sound/usb/caiaq/audio.c | 1 + sound/usb/caiaq/device.c | 8 +++++++- sound/usb/caiaq/device.h | 1 + 4 files changed, 10 insertions(+), 1 deletions(-) diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index 523aec1..73525c0 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -48,6 +48,7 @@ config SND_USB_CAIAQ * Native Instruments Kore Controller * Native Instruments Kore Controller 2 * Native Instruments Audio Kontrol 1 + * Native Instruments Audio 2 DJ * Native Instruments Audio 4 DJ * Native Instruments Audio 8 DJ * Native Instruments Guitar Rig Session I/O diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index 8f9b60c..121af06 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -646,6 +646,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_GUITARRIGMOBILE): dev->samplerates |= SNDRV_PCM_RATE_192000; /* fall thru */ + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO2DJ): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ): dev->samplerates |= SNDRV_PCM_RATE_88200; diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index de38108..83e6c13 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -35,13 +35,14 @@ #include "input.h" MODULE_AUTHOR("Daniel Mack <daniel(a)caiaq.de>"); -MODULE_DESCRIPTION("caiaq USB audio, version 1.3.18"); +MODULE_DESCRIPTION("caiaq USB audio, version 1.3.19"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, RigKontrol3}," "{Native Instruments, Kore Controller}," "{Native Instruments, Kore Controller 2}," "{Native Instruments, Audio Kontrol 1}," + "{Native Instruments, Audio 2 DJ}," "{Native Instruments, Audio 4 DJ}," "{Native Instruments, Audio 8 DJ}," "{Native Instruments, Session I/O}," @@ -121,6 +122,11 @@ static struct usb_device_id snd_usb_id_table[] = { .idVendor = USB_VID_NATIVEINSTRUMENTS, .idProduct = USB_PID_AUDIO4DJ }, + { + .match_flags = USB_DEVICE_ID_MATCH_DEVICE, + .idVendor = USB_VID_NATIVEINSTRUMENTS, + .idProduct = USB_PID_AUDIO2DJ + }, { /* terminator */ } }; diff --git a/sound/usb/caiaq/device.h b/sound/usb/caiaq/device.h index ece7351..44e3edf 100644 --- a/sound/usb/caiaq/device.h +++ b/sound/usb/caiaq/device.h @@ -10,6 +10,7 @@ #define USB_PID_KORECONTROLLER 0x4711 #define USB_PID_KORECONTROLLER2 0x4712 #define USB_PID_AK1 0x0815 +#define USB_PID_AUDIO2DJ 0x041c #define USB_PID_AUDIO4DJ 0x0839 #define USB_PID_AUDIO8DJ 0x1978 #define USB_PID_SESSIONIO 0x1915 -- 1.6.3.1

2 2

[alsa-devel] [PATCH 4/4] Let wm97xx-battery use irq to detect charger state
by Marek Vasut 23 Jul '09

23 Jul '09

Hi, I split the "[PATCH 3/4] Convert WM97xx driver to use platform_data" patch into three smaller as Mark requested. These three will be: [PATCH 2/4] Make wm97xx-core pass platform_data to battery, ts [PATCH 3/4] Convert WM97xx battery to use platform_data [PATCH 4/4] Let wm97xx-battery use irq to detect charger state

1 0

[alsa-devel] [PATCH 3/4] Convert WM97xx battery to use platform_data
by Marek Vasut 23 Jul '09

23 Jul '09

Hi, I split the "[PATCH 3/4] Convert WM97xx driver to use platform_data" patch into three smaller as Mark requested. These three will be: [PATCH 2/4] Make wm97xx-core pass platform_data to battery, ts [PATCH 3/4] Convert WM97xx battery to use platform_data [PATCH 4/4] Let wm97xx-battery use irq to detect charger state

1 0

[alsa-devel] [PATCH 2/4] Make wm97xx-core pass platform_data to battery, ts
by Marek Vasut 23 Jul '09

23 Jul '09

Hi, I split the "[PATCH 3/4] Convert WM97xx driver to use platform_data" patch into three smaller as Mark requested. These three will be: [PATCH 2/4] Make wm97xx-core pass platform_data to battery, ts [PATCH 3/4] Convert WM97xx battery to use platform_data [PATCH 4/4] Let wm97xx-battery use irq to detect charger state

1 0

[alsa-devel] ASoC update for 2.6.31
by Mark Brown 23 Jul '09

23 Jul '09

The following changes since commit b7d4de7ff03085fda8310b2983b907166dd40f43: Daniel Mack (1): ASoC: Fix NULL pointer dereference in __pxa2xx_pcm_hw_free 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: tlv320aic3x: Enable PLL when not bypassed sound/soc/codecs/tlv320aic3x.c | 11 ++++++++++- 1 files changed, 10 insertions(+), 1 deletions(-)

2 1

[alsa-devel] ASoC: MAX9877: fix write operation for register
by Joonyoung Shim 23 Jul '09

23 Jul '09

The MAX9877 needs an address of start register when we write values to registers through i2c_master_send(), but the code for this was missed in max9877_write_regs(). If the value of control is 0 in the max9877_set_out_mode(), the value is not increased to 1, but actually the value to write to the register should be 1. And the register bits for out_mode and osc_mode should be cleared before writing. Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- sound/soc/codecs/max9877.c | 14 +++++++++++--- 1 files changed, 11 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/max9877.c b/sound/soc/codecs/max9877.c index a1f1119..9e7e964 100644 --- a/sound/soc/codecs/max9877.c +++ b/sound/soc/codecs/max9877.c @@ -25,7 +25,14 @@ static u8 max9877_regs[5] = { 0x40, 0x00, 0x00, 0x00, 0x49 }; static void max9877_write_regs(void) { - if (i2c_master_send(i2c, max9877_regs, 5) != 5) + unsigned int i; + u8 data[6]; + + data[0] = MAX9877_INPUT_MODE; + for (i = 0; i < ARRAY_SIZE(max9877_regs); i++) + data[i + 1] = max9877_regs[i]; + + if (i2c_master_send(i2c, data, 6) != 6) dev_err(&i2c->dev, "i2c write failed\n"); } @@ -135,12 +142,12 @@ static int max9877_set_out_mode(struct snd_kcontrol *kcontrol, { u8 value = ucontrol->value.integer.value[0]; - if (value) - value += 1; + value += 1; if ((max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OUTMODE_MASK) == value) return 0; + max9877_regs[MAX9877_OUTPUT_MODE] &= ~MAX9877_OUTMODE_MASK; max9877_regs[MAX9877_OUTPUT_MODE] |= value; max9877_write_regs(); return 1; @@ -166,6 +173,7 @@ static int max9877_set_osc_mode(struct snd_kcontrol *kcontrol, if ((max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OSC_MASK) == value) return 0; + max9877_regs[MAX9877_OUTPUT_MODE] &= ~MAX9877_OSC_MASK; max9877_regs[MAX9877_OUTPUT_MODE] |= value; max9877_write_regs(); return 1; -- 1.6.0.4

2 1

[alsa-devel] [RFC] [PATCH 3/3] ASoC: add support for Amstrad E3 (Delta) machine
by Janusz Krzysztofik 23 Jul '09

23 Jul '09

This patch adds machine support for Amstrad E3 (Delta) videophone to ASoC. Created and tested against linux-2.6.31-rc3. Applies and works with linux-omap-2.6 commit 7c5cb7862d32cb344be7831d466535d5255e35ac as well. Depends on patch 2 form this series: TTY: Add definition of a new line discipline required by Amstrad E3 (Delta) ASoC driver. Credits to Mark Underwood for his initial, omap-alsa based sound driver for this machine. Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- CPU DAI parameters best matching the codec DAI has been selected out empirically for best user experience. Board specific audio function control (with related DAPM widgets) has been modeled after empirically discovered codec capabilities. Unlike other ASoC machine drivers, this one contains a line discipline. This is required for providig the codec driver with support for talking to the modem chip that can control the codec behavoiur. As the line discipline code contains board specific bits, I decided to make it a part of the ASoC machine driver, not the codec, and not a separate source file. Otherwise, some kind of a glue, like a bus over a tty, could help in linking the board specific part (bus device) to a more generic line discipline (bus adapter)[1]. In order to work at all, this driver requires a working McBSP1. On OMAP1510 based machines (not sure if other OMAP1 variants as well), where McBSP1 is a DSP public peripheral, that means the kernel must provide basic DSP support, ie. omap_dsp_init(), in order to power up the DSP. This used to be included in linux-omap-2.6 tree up to commit 2512fd29db4eb09e82d182596304c7aaf76d2c5c. Without that, the driver would not work, ie. not shift in/out any bits over the CPU DAI[2]. This limitation is not board, but CPU specific, and may apply to other code that makes use of McBSP1/McBSP3 on affected machines. I provide an extra patch [4/3] as a temporary solution. To work correctly in playback mode, this driver requires my prevoiusly submitted patch that corrects pcm pointer calculation for OMAP1510 based machines[3] (already included in linux-2.6.31-rc3). To support codec controls, this driver requires my previously submitted patch that adds support for modem found on Amstrad Delta[4]. [1] http://www.spinics.net/lists/linux-serial/msg01856.html [2] http://www.spinics.net/lists/linux-omap/msg15114.html [3] http://mailman.alsa-project.org/pipermail/alsa-devel/2009-June/018950.html [4] http://www.spinics.net/lists/linux-omap/msg15432.html --- linux-2.6.31-rc1/sound/soc/omap/Kconfig.orig 2009-07-04 16:40:00.000000000 +0200 +++ linux-2.6.31-rc1/sound/soc/omap/Kconfig 2009-07-22 01:06:51.000000000 +0200 @@ -15,6 +15,14 @@ config SND_OMAP_SOC_N810 help Say Y if you want to add support for SoC audio on Nokia N810. +config SND_OMAP_SOC_AMS_DELTA + tristate "SoC Audio support for Amstrad E3 (Delta) videophone" + depends on SND_OMAP_SOC && MACH_AMS_DELTA + select SND_OMAP_SOC_MCBSP + select SND_SOC_CX20442 + help + Say Y if you want to add support for SoC audio on Amstrad Delta. + config SND_OMAP_SOC_OSK5912 tristate "SoC Audio support for omap osk5912" depends on SND_OMAP_SOC && MACH_OMAP_OSK && I2C --- linux-2.6.31-rc1/sound/soc/omap/Makefile.orig 2009-07-04 16:40:00.000000000 +0200 +++ linux-2.6.31-rc1/sound/soc/omap/Makefile 2009-07-22 01:06:51.000000000 +0200 @@ -7,6 +7,7 @@ obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd- # OMAP Machine Support snd-soc-n810-objs := n810.o +snd-soc-ams-delta-objs := ams-delta.o snd-soc-osk5912-objs := osk5912.o snd-soc-overo-objs := overo.o snd-soc-omap2evm-objs := omap2evm.o @@ -16,6 +17,7 @@ snd-soc-omap3pandora-objs := omap3pandor snd-soc-omap3beagle-objs := omap3beagle.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o +obj-$(CONFIG_SND_OMAP_SOC_AMS_DELTA) += snd-soc-ams-delta.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o --- /dev/null 2009-06-24 10:33:03.437003511 +0200 +++ linux-2.6.31-rc1/sound/soc/omap/ams-delta.c 2009-07-22 01:06:51.000000000 +0200 @@ -0,0 +1,618 @@ +/* + * ams-delta.c -- SoC audio for Amstrad E3 (Delta) videophone + * + * Copyright (C) 2009 Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> + * + * 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. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/gpio.h> +#include <linux/tty.h> + +#include <sound/soc-dapm.h> +#include <sound/jack.h> + +#include <asm/mach-types.h> + +#include <mach/board-ams-delta.h> +#include <mach/mcbsp.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/cx20442.h" + + +/* + * Controls available after the modem line discipline has been activated. + */ + +/* Virtual switch: audio input/output constellations */ +static const char *ams_delta_audio_mode[] = + {"Mixed", "Handset", "HandsFree", "SpeakerPhone"}; + +/* Selection <-> pin translation */ +#define AMS_DELTA_MOUTHPIECE 0 +#define AMS_DELTA_EARPHONE 1 +#define AMS_DELTA_MICROPHONE 2 +#define AMS_DELTA_SPEAKER 3 +#define AMS_DELTA_AGC 4 + +#define AMS_DELTA_MIXED ((1 << AMS_DELTA_EARPHONE) | \ + (1 << AMS_DELTA_MICROPHONE)) +#define AMS_DELTA_HANDSET ((1 << AMS_DELTA_MOUTHPIECE) | \ + (1 << AMS_DELTA_EARPHONE)) +#define AMS_DELTA_HANDSFREE ((1 << AMS_DELTA_MICROPHONE) | \ + (1 << AMS_DELTA_SPEAKER)) +#define AMS_DELTA_SPEAKERPHONE (AMS_DELTA_HANDSFREE | (1 << AMS_DELTA_AGC)) + +unsigned short ams_delta_audio_mode_pins[] = { + AMS_DELTA_MIXED, + AMS_DELTA_HANDSET, + AMS_DELTA_HANDSFREE, + AMS_DELTA_SPEAKERPHONE, +}; + +static bool ams_delta_audio_agc = 0; + +static int ams_delta_set_audio_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct soc_enum *control = (struct soc_enum *)kcontrol->private_value; + unsigned short function; + bool changed = 0; + + if (ucontrol->value.enumerated.item[0] >= control->max) + return -EINVAL; + + mutex_lock(&codec->mutex); + + /* Translate selection to bitmap */ + function = ams_delta_audio_mode_pins[ucontrol->value.enumerated.item[0]]; + + /* Setup pins after corresponding bits if changed */ + if ((bool)snd_soc_dapm_get_pin_status(codec, "Speaker") != + (bool)(function & (1 << AMS_DELTA_SPEAKER))) { + changed = 1; + if (function & (1 << AMS_DELTA_SPEAKER)) + snd_soc_dapm_enable_pin(codec, "Speaker"); + else + snd_soc_dapm_disable_pin(codec, "Speaker"); + } + if ((bool)snd_soc_dapm_get_pin_status(codec, "Microphone") != + (bool)(function & (1 << AMS_DELTA_MICROPHONE))) { + changed = 1; + if (function & (1 << AMS_DELTA_MICROPHONE)) + snd_soc_dapm_enable_pin(codec, "Microphone"); + else + snd_soc_dapm_disable_pin(codec, "Microphone"); + } + if ((bool)snd_soc_dapm_get_pin_status(codec, "Earphone") != + (bool)(function & (1 << AMS_DELTA_EARPHONE))) { + changed = 1; + if (function & (1 << AMS_DELTA_EARPHONE)) + snd_soc_dapm_enable_pin(codec, "Earphone"); + else + snd_soc_dapm_disable_pin(codec, "Earphone"); + } + if ((bool)snd_soc_dapm_get_pin_status(codec, "Mouthpiece") != + (bool)(function & (1 << AMS_DELTA_MOUTHPIECE))) { + changed = 1; + if (function & (1 << AMS_DELTA_MOUTHPIECE)) + snd_soc_dapm_enable_pin(codec, "Mouthpiece"); + else + snd_soc_dapm_disable_pin(codec, "Mouthpiece"); + } + if (ams_delta_audio_agc != (bool)(function & (1 << AMS_DELTA_AGC))) { + ams_delta_audio_agc = (bool)(function & (1 << AMS_DELTA_AGC)); + changed = 1; + if (ams_delta_audio_agc) + snd_soc_dapm_enable_pin(codec, "AGCIN"); + else + snd_soc_dapm_disable_pin(codec, "AGCIN"); + } + if (changed) + snd_soc_dapm_sync(codec); + + mutex_unlock(&codec->mutex); + + return changed; +} + +static int ams_delta_get_audio_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned short value, mux; + + value = ((snd_soc_dapm_get_pin_status(codec, "Mouthpiece") << + AMS_DELTA_MOUTHPIECE) | + (snd_soc_dapm_get_pin_status(codec, "Earphone") << + AMS_DELTA_EARPHONE)); + if (value) + value |= (snd_soc_dapm_get_pin_status(codec, "Microphone") << + AMS_DELTA_MICROPHONE); + else + value = ((snd_soc_dapm_get_pin_status(codec, "Microphone") << + AMS_DELTA_MICROPHONE) | + (snd_soc_dapm_get_pin_status(codec, "Speaker") << + AMS_DELTA_SPEAKER) | + ((unsigned short)ams_delta_audio_agc << AMS_DELTA_AGC)); + + for (mux = 0; mux < ARRAY_SIZE(ams_delta_audio_mode); mux++) + if (value == ams_delta_audio_mode_pins[mux]) + break; + + if (mux >= ARRAY_SIZE(ams_delta_audio_mode)) + return -EINVAL; + + ucontrol->value.enumerated.item[0] = mux; + + return 0; +} + +static const struct soc_enum ams_delta_audio_enum[] = { + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(ams_delta_audio_mode), + ams_delta_audio_mode), +}; + +static const struct snd_kcontrol_new ams_delta_audio_controls[] = { + SOC_ENUM_EXT("Audio Function", ams_delta_audio_enum[0], + ams_delta_get_audio_mode, ams_delta_set_audio_mode), +}; + +/* Now that we are able to control the codec over the modem, + * our hook switch can be used for dynamic DAPM reconfiguration */ +static struct snd_soc_jack ams_delta_hook_switch; +static struct snd_soc_jack_gpio ams_delta_hook_switch_gpios[]; +static struct snd_soc_jack_pin ams_delta_hook_switch_pins[] = { + { + .pin = "Mouthpiece", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Earphone", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Speaker", + .mask = SND_JACK_HEADPHONE, + .invert = 1, + }, + { + .pin = "Microphone", + .mask = SND_JACK_MICROPHONE, + .invert = 1, + }, +}; + +/* To actually apply any modem controlled configuration changes to the codec, + * we must connect codec DAI pins to the modem for a moment. Be carefull + * not to interfere with digital mute function that shares the same hardware. */ +static struct timer_list cx81801_timer; +static bool cx81801_cmd_pending = 0; +static bool ams_delta_muted; + +static void cx81801_timeout(unsigned long data) +{ + /* REVISIT - locking? */ + if(!ams_delta_muted) + /* Reconnect the codec DAI back from the modem to the CPU DAI + * only if digital mute still off */ + ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC, 0); + cx81801_cmd_pending = 0; +} + +/* Modem init: echo off, digital speaker off, quiet off, voice mode */ +static const char *cx81801_init = "ate0m0q0+fclass=8\r"; + + +/* + * Modem line discipline required for making above controls functional. + * Activated from userspace with ldattach, possibly invoked from udev rule. + */ + +static struct snd_soc_card ams_delta_audio_card; + +/* Line discipline setup */ +static int cx81801_open(struct tty_struct *tty) +{ + struct snd_soc_codec *codec = ams_delta_audio_card.codec; + int ret, len = strlen(cx81801_init); + + /* Doesn't make sense without write callback */ + if (!tty->ops->write) + return -EINVAL; + + /* Pass codec structure address to other ldisc callbacks */ + tty->disc_data = codec; + + if (tty->ops->write(tty, cx81801_init, len) != len) { + ret = -EIO; + goto err; + } + /* Actual setup will be performed after the modem responds. */ + return 0; +err: + tty->disc_data = NULL; + return ret; +} + +/* Can this empty callback be omitted? */ +static void cx81801_wakeup(struct tty_struct *tty) +{ +} + +/* Modem response callback */ +static void cx81801_receive(struct tty_struct *tty, + const unsigned char *cp, char *fp, int count) +{ + struct snd_soc_codec *codec = tty->disc_data; + const unsigned char *c; + + if (!codec->control_data) { + /* First modem response, complete setup procedure */ + + /* Initialize timer used for config pulse generation */ + setup_timer(&cx81801_timer, cx81801_timeout, 0); + + /* Set up codec access to modem controls */ + codec->control_data = tty; + codec->hw_write = (hw_write_t)tty->ops->write; + codec->pop_time = 1; + + /* Link hook switch to DAPM pins */ + snd_soc_jack_add_pins(&ams_delta_hook_switch, + ARRAY_SIZE(ams_delta_hook_switch_pins), + ams_delta_hook_switch_pins); + + /* Add virtual switches */ + snd_soc_add_controls(codec, ams_delta_audio_controls, + ARRAY_SIZE(ams_delta_audio_controls)); + + /* Set DAPM pins after hook switch present state */ +#if 0 + /* Fails for switch state matching initial gpio->state = 0 */ + snd_soc_jack_report(&ams_delta_ams_delta_hook_switch, + gpio_get_value(ams_delta_hook_switch_gpios[0].gpio) ? + 0 : SND_JACK_HEADSET, SND_JACK_HEADSET); +#else + if (gpio_get_value(ams_delta_hook_switch_gpios[0].gpio)) { + snd_soc_dapm_enable_pin(codec, "Speaker"); + snd_soc_dapm_enable_pin(codec, "Microphone"); + snd_soc_dapm_disable_pin(codec, "Earphone"); + snd_soc_dapm_disable_pin(codec, "Mouthpiece"); + } else { + snd_soc_dapm_disable_pin(codec, "Speaker"); + snd_soc_dapm_disable_pin(codec, "Microphone"); + snd_soc_dapm_enable_pin(codec, "Earphone"); + snd_soc_dapm_enable_pin(codec, "Mouthpiece"); + } + snd_soc_dapm_sync(codec); +#endif + return; + } + for (c = &cp[count - 1]; c >= cp; c--) { + if (*c != '\r') + continue; + /* Complete modem response received, apply config to codec */ + + /* REVISIT - locking? */ + /* Reconnect the codec to the modem if not already done */ + if (!ams_delta_muted && !cx81801_cmd_pending) + ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC, + AMS_DELTA_LATCH2_MODEM_CODEC); + + /* Keep config pulse long enough for proper config activation */ + cx81801_cmd_pending = 1; + mod_timer(&cx81801_timer, jiffies + msecs_to_jiffies(200)); + break; + } +} + +static void cx81801_close(struct tty_struct *tty) +{ + struct snd_soc_codec *codec = tty->disc_data; + + del_timer_sync(&cx81801_timer); + + /* Revert back to default audio input/output constellation */ + snd_soc_dapm_disable_pin(codec, "AGCIN"); + snd_soc_dapm_disable_pin(codec, "AGCOUT"); + snd_soc_jack_report(&ams_delta_hook_switch, + SND_JACK_HEADPHONE, SND_JACK_HEADSET); + + /* REVISIT - Don't know how to do that */ + /* + * Remove controls that we expose when over the modem control available: + * - virtual audio mode switch, + * - hook switch to DAPM pins links + */ + + /* Prevent the codec driver from further accessing the modem */ + codec->hw_write = NULL; + codec->control_data = NULL; + codec->pop_time = 0; + + tty->disc_data = NULL; +} + +static struct tty_ldisc_ops cx81801_ops = { + .magic = TTY_LDISC_MAGIC, + .name = "cx81801", + .owner = THIS_MODULE, + .open = cx81801_open, + .close = cx81801_close, + .receive_buf = cx81801_receive, + .write_wakeup = cx81801_wakeup, +}; + + +/* + * Even if not very usefull, the sound card can still work without any of the + * above functonality activated. You can still control its audio input/output + * constellation and speakerphone gain from userspace by issueing AT commands + * over the modem port. + */ + +static int ams_delta_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + /* Set cpu DAI configuration */ + return snd_soc_dai_set_fmt(rtd->dai->cpu_dai, + SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); +} + +static struct snd_soc_ops ams_delta_ops = { + .hw_params = ams_delta_hw_params, +}; + + +/* + * Board specific DAPM widgets + */ + +static const struct snd_soc_dapm_widget ams_delta_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Microphone", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_HP("Earphone", NULL), + SND_SOC_DAPM_MIC("Mouthpiece", NULL), +}; + +static const struct snd_soc_dapm_route ams_delta_audio_map[] = { + {"TELIN", NULL, "Mouthpiece"}, + {"Earphone", NULL, "TELOUT"}, + + {"MIC", NULL, "Microphone"}, + {"Speaker", NULL, "SPKOUT"}, +}; + +/* Hook switch */ +static struct snd_soc_jack ams_delta_hook_switch; +static struct snd_soc_jack_gpio ams_delta_hook_switch_gpios[] = { + { + .gpio = 4, + .name = "hook_switch", + .report = SND_JACK_HEADSET, + .invert = 1, + .debounce_time = 100, + } +}; + +/* Board specific codec bias level control */ +static int ams_delta_set_bias_level(struct snd_soc_card *card, + enum snd_soc_bias_level level) +{ + struct snd_soc_codec *codec = card->codec; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + case SND_SOC_BIAS_STANDBY: + if (codec->bias_level == SND_SOC_BIAS_OFF) + ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET, + AMS_DELTA_LATCH2_MODEM_NRESET); + break; + case SND_SOC_BIAS_OFF: + if (codec->bias_level != SND_SOC_BIAS_OFF) + ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET, + 0); + } + codec->bias_level = level; + + return 0; +} + +/* Digital mute implemented using modem/CPU multiplexer. + * Shares hardware with codec config pulse generation */ +static bool ams_delta_muted = 1; + +static int ams_delta_digital_mute(struct snd_soc_dai *dai, int mute) +{ + /* REVISIT - locking? */ + if (ams_delta_muted == mute) + return 0; + ams_delta_muted = mute; + if (cx81801_cmd_pending) + return 0; + + ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC, + mute ? AMS_DELTA_LATCH2_MODEM_CODEC : 0); + + return 0; +} + +/* Our codec DAI probably doesn't have its own .ops structure */ +static struct snd_soc_dai_ops ams_delta_dai_ops = { + .digital_mute = ams_delta_digital_mute, +}; + +/* Will be used if the codec ever has its own digital_mute function */ +static int ams_delta_startup(struct snd_pcm_substream *substream) +{ + return ams_delta_digital_mute(NULL, 0); +} + +static void ams_delta_shutdown(struct snd_pcm_substream *substream) +{ + ams_delta_digital_mute(NULL, 1); +} + + +/* + * Card initialization + */ + +static int ams_delta_cx20442_init(struct snd_soc_codec *codec) +{ + struct snd_soc_dai *codec_dai = codec->dai; + /* Codec is ready, now add/activate board specific controls */ + + /* Set up digital mute if not provided by codec */ + if (!codec_dai->ops) { + codec_dai->ops = &ams_delta_dai_ops; + } else if (!codec_dai->ops->digital_mute) { + codec_dai->ops->digital_mute = ams_delta_digital_mute; + } else { + ams_delta_ops.startup = ams_delta_startup; + ams_delta_ops.shutdown = ams_delta_shutdown; + } + + /* Set codec bias level */ + ams_delta_set_bias_level(&ams_delta_audio_card, SND_SOC_BIAS_STANDBY); + + /* Add board specific DAPM controls */ + if (!snd_soc_dapm_new_controls(codec, ams_delta_dapm_widgets, + ARRAY_SIZE(ams_delta_dapm_widgets))) { + if (!snd_soc_dapm_add_routes(codec, ams_delta_audio_map, + ARRAY_SIZE(ams_delta_audio_map))) { + /* Actual pin constellation if no modem control */ + snd_soc_dapm_disable_pin(codec, "Mouthpiece"); + snd_soc_dapm_enable_pin(codec, "Earphone"); + snd_soc_dapm_enable_pin(codec, "Microphone"); + snd_soc_dapm_disable_pin(codec, "Speaker"); + snd_soc_dapm_disable_pin(codec, "AGCIN"); + snd_soc_dapm_disable_pin(codec, "AGCOUT"); + + snd_soc_dapm_sync(codec); + } + } + /* Add hook switch */ + if (!snd_soc_jack_new(&ams_delta_audio_card, "hook_switch", + SND_JACK_HEADSET, &ams_delta_hook_switch)) { + if (!snd_soc_jack_add_gpios(&ams_delta_hook_switch, + ARRAY_SIZE(ams_delta_hook_switch_gpios), + ams_delta_hook_switch_gpios)) { +#ifdef CONFIG_GPIO_SYSFS + /* Expose hook switch over sysfs if configured */ + gpio_export(ams_delta_hook_switch_gpios[0].gpio, false); +#endif + } + } + /* Register optional line discipline for over the modem control */ + tty_register_ldisc(N_AMSDELTA, &cx81801_ops); + + return 0; +} + +/* DAI glue - connects codec <--> CPU */ +static struct snd_soc_dai_link ams_delta_dai_link = { + .name = "CX20442", + .stream_name = "CX20442", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &cx20442_dai, + .init = ams_delta_cx20442_init, + .ops = &ams_delta_ops, +}; + +/* Audio card driver */ +static struct snd_soc_card ams_delta_audio_card = { + .name = "AMS_DELTA", + .platform = &omap_soc_platform, + .dai_link = &ams_delta_dai_link, + .num_links = 1, + .set_bias_level = ams_delta_set_bias_level, +}; + +/* Audio subsystem */ +static struct snd_soc_device ams_delta_snd_soc_device = { + .card = &ams_delta_audio_card, + .codec_dev = &cx20442_codec_dev, +}; + +/* Module init/exit */ +static struct platform_device *ams_delta_audio_platform_device; +static struct platform_device *cx20442_platform_device; + +static int __init ams_delta_module_init(void) +{ + int ret; + + if (!(machine_is_ams_delta())) + return -ENODEV; + + ams_delta_audio_platform_device = platform_device_alloc("soc-audio", -1); + if (!ams_delta_audio_platform_device) + return -ENOMEM; + + platform_set_drvdata(ams_delta_audio_platform_device, + &ams_delta_snd_soc_device); + ams_delta_snd_soc_device.dev = &ams_delta_audio_platform_device->dev; + *(unsigned int *)ams_delta_dai_link.cpu_dai->private_data = OMAP_MCBSP1; + + ret = platform_device_add(ams_delta_audio_platform_device); + if (ret) + goto err; + + /* + * Codec platform device could be registered from elsewhere (board?), + * but I do it here as it makes sense only if used with the card. + */ + cx20442_platform_device = platform_device_register_simple("cx20442", + -1, NULL, 0); + return 0; +err: + platform_device_put(ams_delta_audio_platform_device); + return ret; +} +module_init(ams_delta_module_init); + +static void __exit ams_delta_module_exit(void) +{ + tty_unregister_ldisc(N_AMSDELTA); + +#ifdef CONFIG_GPIO_SYSFS + gpio_unexport(ams_delta_hook_switch_gpios[0].gpio); +#endif + snd_soc_jack_free_gpios(&ams_delta_hook_switch, + ARRAY_SIZE(ams_delta_hook_switch_gpios), + ams_delta_hook_switch_gpios); + + /* Keep modem power on */ + ams_delta_set_bias_level(&ams_delta_audio_card, SND_SOC_BIAS_STANDBY); + + platform_device_unregister(cx20442_platform_device); + platform_device_unregister(ams_delta_audio_platform_device); +} +module_exit(ams_delta_module_exit); + +MODULE_AUTHOR("Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl>"); +MODULE_DESCRIPTION("ALSA SoC driver for Amstrad E3 (Delta) videophone"); +MODULE_LICENSE("GPL");

3 8

[alsa-devel] [PATCH] ASoC: OMAP: Staticise pcm creation function of omap-pcm
by Lopez Cruz, Misael 23 Jul '09

23 Jul '09

Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/omap/omap-pcm.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 84a1950..c3c931d 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -330,7 +330,7 @@ static void omap_pcm_free_dma_buffers(struct snd_pcm *pcm) } } -int omap_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, +static int omap_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) { int ret = 0; -- 1.5.4.3

2 1

Re: [alsa-devel] distinguish events based on input ports
by Lieven Moors 22 Jul '09

22 Jul '09

Clemens Ladisch wrote: > Lieven Moors wrote: > >> I had a question regarding the use of input(read) ports. >> What I want to do is distinguish events based on the input port >> they came through. Since the destination of the event is mostly >> set to 'broadcast', I cannot extract this information from the event >> itself. >> > > Yes you can; when delivering broadcast or to-all-subscribers events, > ALSA sets the event's destination field to the actual destination. > > > HTH > Clemens > > Thanks for confirming that. it was my own silly mistake, because i had used snd_seq_ev_set_subs before trying to filter... Sorry for that... greets, Lieven

1 0

[alsa-devel] [PATCH] Mixer quirk for QuickCam E 3500
by Alexey Fisher 22 Jul '09

22 Jul '09

1. E3500 report cval->max more than it actually can handel, so if you set 95% capture level it will be muted. 2. betwen cval->min and cval-max(real) is 2940 control units, but real are only 7. 3. we found only 7 real control units, with cval->res = 384, but alsa can't handel less than 10 controls, so make it more and set cval->res = 192. Signed-off-by: Alexey Fisher <bug-track(a)fisher-privat.net> --- sound/usb/usbmixer.c | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-) diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index 4bd3a7a..72562ee 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -995,6 +995,13 @@ static void build_feature_ctl(struct mixer_build *state, unsigned char *desc, /* is not very clear from datasheets */ /* I hope that the min value is -15360 for newer firmware --jk */ switch (state->chip->usb_id) { + case USB_ID(0x046d, 0x09a4): + if (!strcmp(kctl->id.name, "Mic Capture Volume")) { + snd_printk(KERN_INFO "using volume control quirk for the QuickCam E3500\n"); + cval->min = 6080; + cval->max = 8768; + cval->res = 192; + } case USB_ID(0x0471, 0x0101): case USB_ID(0x0471, 0x0104): case USB_ID(0x0471, 0x0105): -- 1.6.3.3

2 17

[alsa-devel] [PATCH] ASoC: tlv320aic3x: Enable PLL when not bypassed
by Chaithrika U S 22 Jul '09

22 Jul '09

PLL was not being enabled when it was not bypassed. This patch enables the PLL when it is used. Additionally, it disables the PLL when it is bypassed. Without this patch, the audio on TI DM646x EVM and DM355 EVM does not work properly. The bit clocks and the frame sync signals from the codec are not correct and hence the playback/record are faster than usual for most sample rates. The reason for this was that the PLL was not enabled when it was not bypassed. Tested on DM6467 EVM, playback tested on DM355 EVM. Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- Applies to master branch of ALSA GIT tree at http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=summary sound/soc/codecs/tlv320aic3x.c | 11 ++++++++++- 1 files changed, 10 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index ab099f4..cb0d1bf 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -767,6 +767,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, int codec_clk = 0, bypass_pll = 0, fsref, last_clk = 0; u8 data, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1; u16 pll_d = 1; + u8 reg; /* select data word length */ data = @@ -801,8 +802,16 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, pll_q &= 0xf; aic3x_write(codec, AIC3X_PLL_PROGA_REG, pll_q << PLLQ_SHIFT); aic3x_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_CLKDIV); - } else + /* disable PLL if it is bypassed */ + reg = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG); + aic3x_write(codec, AIC3X_PLL_PROGA_REG, reg & ~PLL_ENABLE); + + } else { aic3x_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_PLLDIV); + /* enable PLL when it is used */ + reg = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG); + aic3x_write(codec, AIC3X_PLL_PROGA_REG, reg | PLL_ENABLE); + } /* Route Left DAC to left channel input and * right DAC to right channel input */ -- 1.5.6

4 5

[alsa-devel] Blocked in snd_pcm_open
by Nicolas Krieger 22 Jul '09

22 Jul '09

Hello, I don't know if I am in the right place to expose my problem, if not, excuse me. I have a problem with snd_pcm_open function. The program seems to be blocked and gdb output shows this is happening while using snd_pcm_open. I am upgrading a program that reads videos so as to use alsa now. On my computer, there is no problem. But the problem appears on test computer. My questions are : - is it only a configuration problem (.asoundrc) ? - if not, what else could it be ? _rc = snd_pcm_open(&_handle, "defaultTEST", SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK); <= this is line *42 in ../../src/CAudio.cpp* in gdb output This a part of gdb when the progam is blocked : *0x009877a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2 (gdb) bt #0 0x009877a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2 #1 0x00bdbd4b in __open_nocancel () from /lib/tls/libpthread.so.0 #2 0xf6fedb9d in open () from /usr/lib/libesddsp.so.0 #3 0x04e7e5f2 in snd_pcm_hw_open () from /lib/libasound.so.2 #4 0x04e7ea90 in _snd_pcm_hw_open () from /lib/libasound.so.2 #5 0x04e70893 in snd_pcm_close () from /lib/libasound.so.2 #6 0x04e70e78 in snd_pcm_close () from /lib/libasound.so.2 #7 0x04e710c9 in snd_pcm_open_slave () from /lib/libasound.so.2 #8 0x04ea4541 in snd_pcm_dmix_open () from /lib/libasound.so.2 #9 0x04ea4aea in _snd_pcm_dmix_open () from /lib/libasound.so.2 #10 0x04e70893 in snd_pcm_close () from /lib/libasound.so.2 #11 0x04e70e78 in snd_pcm_close () from /lib/libasound.so.2 #12 0x04e710c9 in snd_pcm_open_slave () from /lib/libasound.so.2 #13 0x04ea8186 in _snd_pcm_asym_open () from /lib/libasound.so.2 #14 0x04e70893 in snd_pcm_close () from /lib/libasound.so.2 #15 0x04e70e78 in snd_pcm_close () from /lib/libasound.so.2 #16 0x04e710c9 in snd_pcm_open_slave () from /lib/libasound.so.2 #17 0x04e8e1de in _snd_pcm_plug_open () from /lib/libasound.so.2 #18 0x04e70893 in snd_pcm_close () from /lib/libasound.so.2 #19 0x04e70e78 in snd_pcm_close () from /lib/libasound.so.2 #20 0x08065543 in CAudio::openDevice (this=0x87795e0, sampleRate=@0x8884558<https://messagerie.si.c-s.fr/horde/imp/message.php?index=988#>, nbChannels=@0x888455c<https://messagerie.si.c-s.fr/horde/imp/message.php?index=988#>, devicename=@0xfeee8670<https://messagerie.si.c-s.fr/horde/imp/message.php?index=988#>) at ../../src/CAudio.cpp:42* .asoundrc I am using : *pcm.card0 { type hw card 0 device 0 } pcm.card1 { type hw card 1 device 0 } pcm.dmixerDEF { type dmix ipc_key 1025 slave { pcm card0 period_time 0 period_size 2048 buffer_size 32768 rate 44100 } bindings { 0 0 1 1 } } pcm.dmixerTEST { type dmix ipc_key 1027 slave { pcm card1 format S16_LE period_time 0 period_size 2048 buffer_size 32768 rate 44100 } #bindings { # 0 0 # 1 1 #} } pcm.DEF { type asym playback.pcm "dmixerDEF" capture.pcm "card0" } pcm.TEST { type asym playback.pcm "dmixerTEST" capture.pcm "card1" } pcm.!default { type plug slave.pcm "DEF" } pcm.defaultTEST { type plug slave.pcm "TEST" }* Nicolas Krieger

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[alsa-devel] [PATCH] ASoC: MAX9877: separate callback functions
by Joonyoung Shim 22 Jul '09

22 Jul '09

The callback function to control register was used by whole controls in MAX9877 driver, but this causes using many if statement for double register control or invert. So, the callback function for double register control is separate differently, and the code for invert is added in the callback function. Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- sound/soc/codecs/max9877.c | 96 +++++++++++++++++++++++++++++--------------- 1 files changed, 63 insertions(+), 33 deletions(-) diff --git a/sound/soc/codecs/max9877.c b/sound/soc/codecs/max9877.c index 7df9a7c..a1f1119 100644 --- a/sound/soc/codecs/max9877.c +++ b/sound/soc/codecs/max9877.c @@ -34,16 +34,16 @@ static int max9877_get_reg(struct snd_kcontrol *kcontrol, { struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; - int reg = mc->reg; - int reg2 = mc->rreg; - int shift = mc->shift; - int mask = mc->max; + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + unsigned int mask = mc->max; + unsigned int invert = mc->invert; ucontrol->value.integer.value[0] = (max9877_regs[reg] >> shift) & mask; - if (reg2) - ucontrol->value.integer.value[1] = - (max9877_regs[reg2] >> shift) & mask; + if (invert) + ucontrol->value.integer.value[0] = + mask - ucontrol->value.integer.value[0]; return 0; } @@ -53,39 +53,69 @@ static int max9877_set_reg(struct snd_kcontrol *kcontrol, { struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; - int reg = mc->reg; - int reg2 = mc->rreg; - int shift = mc->shift; - int mask = mc->max; - int change = 1; - int change2 = 1; - int ret = 0; - - if (((max9877_regs[reg] >> shift) & mask) == - ucontrol->value.integer.value[0]) + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + unsigned int mask = mc->max; + unsigned int invert = mc->invert; + unsigned int val = (ucontrol->value.integer.value[0] & mask); + + if (invert) + val = mask - val; + + if (((max9877_regs[reg] >> shift) & mask) == val) + return 0; + + max9877_regs[reg] &= ~(mask << shift); + max9877_regs[reg] |= val << shift; + max9877_write_regs(); + + return 1; +} + +static int max9877_get_2reg(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; + unsigned int shift = mc->shift; + unsigned int mask = mc->max; + + ucontrol->value.integer.value[0] = (max9877_regs[reg] >> shift) & mask; + ucontrol->value.integer.value[1] = (max9877_regs[reg2] >> shift) & mask; + + return 0; +} + +static int max9877_set_2reg(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; + unsigned int shift = mc->shift; + unsigned int mask = mc->max; + unsigned int val = (ucontrol->value.integer.value[0] & mask); + unsigned int val2 = (ucontrol->value.integer.value[1] & mask); + unsigned int change = 1; + + if (((max9877_regs[reg] >> shift) & mask) == val) change = 0; - if (reg2) - if (((max9877_regs[reg2] >> shift) & mask) == - ucontrol->value.integer.value[1]) - change2 = 0; + if (((max9877_regs[reg2] >> shift) & mask) == val2) + change = 0; if (change) { max9877_regs[reg] &= ~(mask << shift); - max9877_regs[reg] |= ucontrol->value.integer.value[0] << shift; - ret = change; - } - - if (reg2 && change2) { + max9877_regs[reg] |= val << shift; max9877_regs[reg2] &= ~(mask << shift); - max9877_regs[reg2] |= ucontrol->value.integer.value[1] << shift; - ret = change2; - } - - if (ret) + max9877_regs[reg2] |= val2 << shift; max9877_write_regs(); + } - return ret; + return change; } static int max9877_get_out_mode(struct snd_kcontrol *kcontrol, @@ -190,7 +220,7 @@ static const struct snd_kcontrol_new max9877_controls[] = { max9877_get_reg, max9877_set_reg, max9877_output_tlv), SOC_DOUBLE_R_EXT_TLV("MAX9877 Amp HP Playback Volume", MAX9877_HPL_VOLUME, MAX9877_HPR_VOLUME, 0, 31, 0, - max9877_get_reg, max9877_set_reg, max9877_output_tlv), + max9877_get_2reg, max9877_set_2reg, max9877_output_tlv), SOC_SINGLE_EXT("MAX9877 INB Stereo Switch", MAX9877_INPUT_MODE, 4, 1, 1, max9877_get_reg, max9877_set_reg), -- 1.6.0.4

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[alsa-devel] [RFC] [PATCH 0/3] Add support for Amstrad E3 (Delta) audio to ASoC
by Janusz Krzysztofik 22 Jul '09

22 Jul '09

This series of patches adds support for audio device found on Amstrad E3 (Delta) videophone to ASoC. 1/3: ASoC: Add support for Conexant CX20442-11 voice modem codec 2/3: TTY: Add definition of a new line discipline required by Amstrad E3 (Delta) ASoC driver 3/3: ASoC: add support for Amstrad E3 (Delta) machine 4/3: OMAP1: Readd basic support for DSP initialization Created and tested against linux-2.5.31-rc3. Applies and works with linux-omap-2.6 commit 7c5cb7862d32cb344be7831d466535d5255e35ac as well Patch 4/3 is a temporary solution that should help interested parties in testing the driver or even making use of it before an official DSP support gets back to the omap or mainline tree. Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl>

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[alsa-devel] [RFC] [PATCH 4/3] OMAP1: Readd basic support for DSP initialization
by Janusz Krzysztofik 22 Jul '09

22 Jul '09

This patch adds back support for powering up the DSP. This is required on OMAP1510 and posiibly other OMAP1 CPU based machines for making back McBSP1/MsBSP3 working. I provide this patch as a temporary solution that should enable interested people to use my ASoC driver for Amstrad Delta before an official DSP support is added back to the omap or mainline tree. Created and tested against linux-omap-2.6 commit 7c5cb7862d32cb344be7831d466535d5255e35ac. Applies and works with linux-2.6.31-rc3 as well. Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- diff -Npru a/arch/arm/plat-omap/Makefile b/arch/arm/plat-omap/Makefile --- a/arch/arm/plat-omap/Makefile 2009-05-12 21:13:59.000000000 +0200 +++ b/arch/arm/plat-omap/Makefile 2009-06-01 22:31:15.000000000 +0200 @@ -25,3 +25,5 @@ # OMAP mailbox framework obj-$(CONFIG_OMAP_MBOX_FWK) += mailbox.o +obj-y += dsp/ + diff -Npru a/arch/arm/plat-omap/dsp/Makefile b/arch/arm/plat-omap/dsp/Makefile --- a/arch/arm/plat-omap/dsp/Makefile 2009-05-12 21:13:59.000000000 +0200 +++ b/arch/arm/plat-omap/dsp/Makefile 2009-06-01 22:31:15.000000000 +0200 @@ -0,0 +1,14 @@ +# +# Makefile for the OMAP DSP driver. +# + +# The target object and module list name. + +obj-y := dsp_common.o + +obj-$(CONFIG_OMAP_DSP) += dsp.o + +# Declare multi-part drivers + +dsp-objs := dsp_core.o ipbuf.o mblog.o task.o \ + dsp_ctl_core.o dsp_ctl.o taskwatch.o error.o dsp_mem.o \ + uaccess_dsp.o diff -Npru a/arch/arm/plat-omap/dsp/dsp_common.c b/arch/arm/plat-omap/dsp/dsp_common.c --- a/arch/arm/plat-omap/dsp/dsp_common.c 2009-05-12 21:13:59.000000000 +0200 +++ b/arch/arm/plat-omap/dsp/dsp_common.c 2009-06-01 22:31:15.000000000 +0200 @@ -0,0 +1,639 @@ +/* + * This file is part of OMAP DSP driver (DSP Gateway version 3.3.1) + * + * Copyright (C) 2002-2006 Nokia Corporation. All rights reserved. + * + * Contact: Toshihiro Kobayashi <toshihiro.kobayashi(a)nokia.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. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/sched.h> +#include <linux/delay.h> +#include <linux/mm.h> +#include <linux/err.h> +#include <linux/clk.h> +#include <linux/mutex.h> +#include <linux/interrupt.h> +#include <asm/io.h> +#include <asm/tlbflush.h> +#include <asm/irq.h> +#include <mach/dsp_common.h> +#include "dsp.h" + +#ifdef CONFIG_ARCH_OMAP1 +#include <mach/tc.h> +#endif + +#if defined(CONFIG_ARCH_OMAP1) +#define dsp_boot_config(mode) omap_writew((mode), MPUI_DSP_BOOT_CONFIG) +#endif +#if defined(CONFIG_ARCH_OMAP2) || defined(CONFIG_ARCH_OMAP3) +#define dsp_boot_config(mode) writel((mode), DSP_IPI_DSPBOOTCONFIG) +#endif + +struct omap_dsp *omap_dsp; + +#if defined(CONFIG_ARCH_OMAP1) +struct clk *dsp_ck_handle; +struct clk *api_ck_handle; +#elif defined(CONFIG_ARCH_OMAP2) +struct clk *dsp_fck_handle; +struct clk *dsp_ick_handle; +#endif +dsp_long_t dspmem_base, dspmem_size, + daram_base, daram_size, + saram_base, saram_size; + +static struct cpustat { + struct mutex lock; + enum cpustat_e stat; + enum cpustat_e req; + u16 icrmask; +#ifdef CONFIG_ARCH_OMAP1 + struct { + int mpui; + int mem; + int mem_delayed; + } usecount; + int (*mem_req_cb)(void); + void (*mem_rel_cb)(void); +#endif +} cpustat = { + .stat = CPUSTAT_RESET, + .icrmask = 0xffff, +}; + +int dsp_set_rstvect(dsp_long_t adr) +{ + unsigned long *dst_adr; + + if (adr >= DSPSPACE_SIZE) + return -EINVAL; + + dst_adr = dspbyte_to_virt(DSP_BOOT_ADR_DIRECT); + /* word swap */ + *dst_adr = ((adr & 0xffff) << 16) | (adr >> 16); + /* fill 8 bytes! */ + *(dst_adr + 1) = 0; + /* direct boot */ + dsp_boot_config(DSP_BOOT_CONFIG_DIRECT); + + return 0; +} + +dsp_long_t dsp_get_rstvect(void) +{ + unsigned long *dst_adr; + + dst_adr = dspbyte_to_virt(DSP_BOOT_ADR_DIRECT); + return ((*dst_adr & 0xffff) << 16) | (*dst_adr >> 16); +} + +#ifdef CONFIG_ARCH_OMAP1 +static void simple_load_code(unsigned char *src_c, u16 *dst, int len) +{ + int i; + u16 *src = (u16 *)src_c; + int len_w; + + /* len must be multiple of 2. */ + if (len & 1) + BUG(); + + len_w = len / 2; + for (i = 0; i < len_w; i++) { + /* byte swap copy */ + *dst = ((*src & 0x00ff) << 8) | + ((*src & 0xff00) >> 8); + src++; + dst++; + } +} + +/* program size must be multiple of 2 */ +#define GBL_IDLE_TEXT_SIZE 52 +#define GBL_IDLE_TEXT_INIT { \ + /* SAM */ \ + 0x3c, 0x4a, /* 0x3c4a: MOV 0x4, AR2 */ \ + 0xf4, 0x41, 0xfc, 0xff, /* 0xf441fcff: AND 0xfcff, *AR2 */ \ + /* disable WDT */ \ + 0x76, 0x34, 0x04, 0xb8, /* 0x763404b8: MOV 0x3404, AR3 */ \ + 0xfb, 0x61, 0x00, 0xf5, /* 0xfb6100f5: MOV 0x00f5, *AR3 */ \ + 0x9a, /* 0x9a: PORT */ \ + 0xfb, 0x61, 0x00, 0xa0, /* 0xfb6100a0: MOV 0x00a0, *AR3 */ \ + 0x9a, /* 0x9a: PORT */ \ + /* *IER0 = 0, *IER1 = 0 */ \ + 0x3c, 0x0b, /* 0x3c0b: MOV 0x0, AR3 */ \ + 0xe6, 0x61, 0x00, /* 0xe66100: MOV 0, *AR3 */ \ + 0x76, 0x00, 0x45, 0xb8, /* 0x76004508: MOV 0x45, AR3 */ \ + 0xe6, 0x61, 0x00, /* 0xe66100: MOV 0, *AR3 */ \ + /* *ICR = 0xffff */ \ + 0x3c, 0x1b, /* 0x3c1b: MOV 0x1, AR3 */ \ + 0xfb, 0x61, 0xff, 0xff, /* 0xfb61ffff: MOV 0xffff, *AR3 */ \ + 0x9a, /* 0x9a: PORT */ \ + /* HOM */ \ + 0xf5, 0x41, 0x03, 0x00, /* 0xf5410300: OR 0x0300, *AR2 */ \ + /* idle and loop forever */ \ + 0x7a, 0x00, 0x00, 0x0c, /* 0x7a00000c: IDLE */ \ + 0x4a, 0x7a, /* 0x4a7a: B -6 (infinite loop) */ \ + 0x20, 0x20, 0x20, /* 0x20: NOP */ \ +} + +/* program size must be multiple of 2 */ +#define CPU_IDLE_TEXT_SIZE 48 +#define CPU_IDLE_TEXT_INIT(icrh, icrl) { \ + /* SAM */ \ + 0x3c, 0x4b, /* 0x3c4b: MOV 0x4, AR3 */ \ + 0xf4, 0x61, 0xfc, 0xff, /* 0xf461fcff: AND 0xfcff, *AR3 */ \ + /* disable WDT */ \ + 0x76, 0x34, 0x04, 0xb8, /* 0x763404b8: MOV 0x3404, AR3 */ \ + 0xfb, 0x61, 0x00, 0xf5, /* 0xfb6100f5: MOV 0x00f5, *AR3 */ \ + 0x9a, /* 0x9a: PORT */ \ + 0xfb, 0x61, 0x00, 0xa0, /* 0xfb6100a0: MOV 0x00a0, *AR3 */ \ + 0x9a, /* 0x9a: PORT */ \ + /* *IER0 = 0, *IER1 = 0 */ \ + 0x3c, 0x0b, /* 0x3c0b: MOV 0x0, AR3 */ \ + 0xe6, 0x61, 0x00, /* 0xe66100: MOV 0, *AR3 */ \ + 0x76, 0x00, 0x45, 0xb8, /* 0x76004508: MOV 0x45, AR3 */ \ + 0xe6, 0x61, 0x00, /* 0xe66100: MOV 0, *AR3 */ \ + /* set ICR = icr */ \ + 0x3c, 0x1b, /* 0x3c1b: MOV AR3 0x1 */ \ + 0xfb, 0x61, (icrh), (icrl), /* 0xfb61****: MOV *AR3, icr */ \ + 0x9a, /* 0x9a: PORT */ \ + /* idle and loop forever */ \ + 0x7a, 0x00, 0x00, 0x0c, /* 0x7a00000c: IDLE */ \ + 0x4a, 0x7a, /* 0x4a7a: B -6 (infinite loop) */ \ + 0x20, 0x20, 0x20 /* 0x20: nop */ \ +} + +/* + * idle_boot base: + * Initialized with DSP_BOOT_ADR_MPUI (=0x010000). + * This value is used before DSP Gateway driver is initialized. + * DSP Gateway driver will overwrite this value with other value, + * to avoid confliction with the user program. + */ +static dsp_long_t idle_boot_base = DSP_BOOT_ADR_MPUI; + +static void dsp_gbl_idle(void) +{ + unsigned char idle_text[GBL_IDLE_TEXT_SIZE] = GBL_IDLE_TEXT_INIT; + + __dsp_reset(); + clk_enable(api_ck_handle); + +#if 0 + dsp_boot_config(DSP_BOOT_CONFIG_IDLE); +#endif + simple_load_code(idle_text, dspbyte_to_virt(idle_boot_base), + GBL_IDLE_TEXT_SIZE); + if (idle_boot_base == DSP_BOOT_ADR_MPUI) + dsp_boot_config(DSP_BOOT_CONFIG_MPUI); + else + dsp_set_rstvect(idle_boot_base); + + __dsp_run(); + udelay(100); /* to make things stable */ + clk_disable(api_ck_handle); +} + +static void dsp_cpu_idle(void) +{ + u16 icr_tmp; + unsigned char icrh, icrl; + + __dsp_reset(); + clk_enable(api_ck_handle); + + /* + * icr settings: + * DMA should not sleep for DARAM/SARAM access + * DPLL should not sleep while any other domain is active + */ + icr_tmp = cpustat.icrmask & ~(DSPREG_ICR_DMA | DSPREG_ICR_DPLL); + icrh = icr_tmp >> 8; + icrl = icr_tmp & 0xff; + { + unsigned char idle_text[CPU_IDLE_TEXT_SIZE] = CPU_IDLE_TEXT_INIT(icrh, icrl); + simple_load_code(idle_text, dspbyte_to_virt(idle_boot_base), + CPU_IDLE_TEXT_SIZE); + } + if (idle_boot_base == DSP_BOOT_ADR_MPUI) + dsp_boot_config(DSP_BOOT_CONFIG_MPUI); + else + dsp_set_rstvect(idle_boot_base); + __dsp_run(); + udelay(100); /* to make things stable */ + clk_disable(api_ck_handle); +} + +void dsp_set_idle_boot_base(dsp_long_t adr, size_t size) +{ + if (adr == idle_boot_base) + return; + idle_boot_base = adr; + if ((size < GBL_IDLE_TEXT_SIZE) || + (size < CPU_IDLE_TEXT_SIZE)) { + printk(KERN_ERR + "omapdsp: size for idle program is not enough!\n"); + BUG(); + } + + /* restart idle program with new base address */ + if (cpustat.stat == CPUSTAT_GBL_IDLE) + dsp_gbl_idle(); + if (cpustat.stat == CPUSTAT_CPU_IDLE) + dsp_cpu_idle(); +} + +void dsp_reset_idle_boot_base(void) +{ + idle_boot_base = DSP_BOOT_ADR_MPUI; +} +#else +void dsp_reset_idle_boot_base(void) { } +#endif /* CONFIG_ARCH_OMAP1 */ + +static int init_done; + +static int omap_dsp_init(void) +{ + mutex_init(&cpustat.lock); + + dspmem_size = 0; +#ifdef CONFIG_ARCH_OMAP15XX + if (cpu_is_omap15xx()) { + dspmem_base = OMAP1510_DSP_BASE; + dspmem_size = OMAP1510_DSP_SIZE; + daram_base = OMAP1510_DARAM_BASE; + daram_size = OMAP1510_DARAM_SIZE; + saram_base = OMAP1510_SARAM_BASE; + saram_size = OMAP1510_SARAM_SIZE; + } +#endif +#ifdef CONFIG_ARCH_OMAP16XX + if (cpu_is_omap16xx()) { + dspmem_base = OMAP16XX_DSP_BASE; + dspmem_size = OMAP16XX_DSP_SIZE; + daram_base = OMAP16XX_DARAM_BASE; + daram_size = OMAP16XX_DARAM_SIZE; + saram_base = OMAP16XX_SARAM_BASE; + saram_size = OMAP16XX_SARAM_SIZE; + } +#endif +#ifdef CONFIG_ARCH_OMAP24XX + if (cpu_is_omap24xx()) { + dspmem_base = DSP_MEM_24XX_VIRT; + dspmem_size = DSP_MEM_24XX_SIZE; + daram_base = OMAP24XX_DARAM_BASE; + daram_size = OMAP24XX_DARAM_SIZE; + saram_base = OMAP24XX_SARAM_BASE; + saram_size = OMAP24XX_SARAM_SIZE; + } +#endif +#ifdef CONFIG_ARCH_OMAP34XX + /* To be Revisited for 3430 */ + if (cpu_is_omap34xx()) { + return -ENODEV; + } +#endif + if (dspmem_size == 0) { + printk(KERN_ERR "omapdsp: unsupported omap architecture.\n"); + return -ENODEV; + } + +#if defined(CONFIG_ARCH_OMAP1) + dsp_ck_handle = clk_get(NULL, "dsp_ck"); + if (IS_ERR(dsp_ck_handle)) { + printk(KERN_ERR "omapdsp: could not acquire dsp_ck handle.\n"); + return PTR_ERR(dsp_ck_handle); + } + + api_ck_handle = clk_get(NULL, "api_ck"); + if (IS_ERR(api_ck_handle)) { + printk(KERN_ERR "omapdsp: could not acquire api_ck handle.\n"); + if (dsp_ck_handle != NULL) + clk_put(dsp_ck_handle); + return PTR_ERR(api_ck_handle); + } + + /* This is needed for McBSP init, released in late_initcall */ + clk_enable(api_ck_handle); + + __dsp_enable(); + mpui_byteswap_off(); + mpui_wordswap_on(); + tc_wordswap(); +#elif defined(CONFIG_ARCH_OMAP2) + dsp_fck_handle = clk_get(NULL, "dsp_fck"); + if (IS_ERR(dsp_fck_handle)) { + printk(KERN_ERR "omapdsp: could not acquire dsp_fck handle.\n"); + return PTR_ERR(dsp_fck_handle); + } + +# if defined(CONFIG_ARCH_OMAP2420) + dsp_ick_handle = clk_get(NULL, "dsp_ick"); +# elif defined(CONFIG_ARCH_OMAP2430) + /* + * 2430 has no separate switch for DSP ICLK, but this at least + * involves the minimal change to the rest of the code. + */ + dsp_ick_handle = clk_get(NULL, "iva2_1_ick"); +# endif + if (IS_ERR(dsp_ick_handle)) { + printk(KERN_ERR "omapdsp: could not acquire dsp_ick handle.\n"); + if (dsp_fck_handle != NULL) + clk_put(dsp_fck_handle); + return PTR_ERR(dsp_ick_handle); + } +#endif + + init_done = 1; + pr_info("omap_dsp_init() done\n"); + return 0; +} + +#if defined(CONFIG_ARCH_OMAP1) +static int __dsp_late_init(void) +{ + clk_disable(api_ck_handle); + return 0; +} +late_initcall(__dsp_late_init); +#endif + +static void dsp_cpustat_update(void) +{ + if (!init_done) + omap_dsp_init(); + + if (cpustat.req == CPUSTAT_RUN) { + if (cpustat.stat < CPUSTAT_RUN) { +#if defined(CONFIG_ARCH_OMAP1) + __dsp_reset(); + clk_enable(api_ck_handle); + udelay(10); + __dsp_run(); +#elif defined(CONFIG_ARCH_OMAP2) + __dsp_core_disable(); + udelay(10); + __dsp_core_enable(); +#endif + cpustat.stat = CPUSTAT_RUN; + } + return; + } + + /* cpustat.req < CPUSTAT_RUN */ + + if (cpustat.stat == CPUSTAT_RUN) { +#ifdef CONFIG_ARCH_OMAP1 + clk_disable(api_ck_handle); +#endif + } + +#ifdef CONFIG_ARCH_OMAP1 + /* + * (1) when ARM wants DARAM access, MPUI should be SAM and + * DSP needs to be on. + * (2) if any bits of icr is masked, we can not enter global idle. + */ + if ((cpustat.req == CPUSTAT_CPU_IDLE) || + (cpustat.usecount.mem > 0) || + (cpustat.usecount.mem_delayed > 0) || + ((cpustat.usecount.mpui > 0) && (cpustat.icrmask != 0xffff))) { + if (cpustat.stat != CPUSTAT_CPU_IDLE) { + dsp_cpu_idle(); + cpustat.stat = CPUSTAT_CPU_IDLE; + } + return; + } + + /* + * when ARM only needs MPUI access, MPUI can be HOM and + * DSP can be idling. + */ + if ((cpustat.req == CPUSTAT_GBL_IDLE) || + (cpustat.usecount.mpui > 0)) { + if (cpustat.stat != CPUSTAT_GBL_IDLE) { + dsp_gbl_idle(); + cpustat.stat = CPUSTAT_GBL_IDLE; + } + return; + } +#endif /* CONFIG_ARCH_OMAP1 */ + + /* + * no user, no request + */ + if (cpustat.stat != CPUSTAT_RESET) { +#if defined(CONFIG_ARCH_OMAP1) + __dsp_reset(); +#elif defined(CONFIG_ARCH_OMAP2) + __dsp_core_disable(); +#endif + cpustat.stat = CPUSTAT_RESET; + } +} + +void dsp_cpustat_request(enum cpustat_e req) +{ + mutex_lock(&cpustat.lock); + cpustat.req = req; + dsp_cpustat_update(); + mutex_unlock(&cpustat.lock); +} + +enum cpustat_e dsp_cpustat_get_stat(void) +{ + return cpustat.stat; +} + +u16 dsp_cpustat_get_icrmask(void) +{ + return cpustat.icrmask; +} + +void dsp_cpustat_set_icrmask(u16 mask) +{ + mutex_lock(&cpustat.lock); + cpustat.icrmask = mask; + dsp_cpustat_update(); + mutex_unlock(&cpustat.lock); +} + +#ifdef CONFIG_ARCH_OMAP1 +void omap_dsp_request_mpui(void) +{ + mutex_lock(&cpustat.lock); + if (cpustat.usecount.mpui++ == 0) + dsp_cpustat_update(); + mutex_unlock(&cpustat.lock); +} + +void omap_dsp_release_mpui(void) +{ + mutex_lock(&cpustat.lock); + if (cpustat.usecount.mpui-- == 0) { + printk(KERN_ERR + "omapdsp: unbalanced mpui request/release detected.\n" + " cpustat.usecount.mpui is going to be " + "less than zero! ... fixed to be zero.\n"); + cpustat.usecount.mpui = 0; + } + if (cpustat.usecount.mpui == 0) + dsp_cpustat_update(); + mutex_unlock(&cpustat.lock); +} + +#if defined(CONFIG_ARCH_OMAP1) && defined(CONFIG_OMAP_MMU_FWK) +int omap_dsp_request_mem(void) +{ + int ret = 0; + + mutex_lock(&cpustat.lock); + if ((cpustat.usecount.mem++ == 0) && + (cpustat.usecount.mem_delayed == 0)) { + if (cpustat.mem_req_cb) { + if ((ret = cpustat.mem_req_cb()) < 0) { + cpustat.usecount.mem--; + goto out; + } + } + dsp_cpustat_update(); + } +out: + mutex_unlock(&cpustat.lock); + + return ret; +} + +/* + * release_mem will be delayed. + */ +static void do_release_mem(struct work_struct *dummy) +{ + mutex_lock(&cpustat.lock); + cpustat.usecount.mem_delayed = 0; + if (cpustat.usecount.mem == 0) { + dsp_cpustat_update(); + if (cpustat.mem_rel_cb) + cpustat.mem_rel_cb(); + } + mutex_unlock(&cpustat.lock); +} + +static DECLARE_DELAYED_WORK(mem_rel_work, do_release_mem); + +int omap_dsp_release_mem(void) +{ + mutex_lock(&cpustat.lock); + + /* cancel previous release work */ + cancel_delayed_work(&mem_rel_work); + cpustat.usecount.mem_delayed = 0; + + if (cpustat.usecount.mem-- == 0) { + printk(KERN_ERR + "omapdsp: unbalanced memory request/release detected.\n" + " cpustat.usecount.mem is going to be " + "less than zero! ... fixed to be zero.\n"); + cpustat.usecount.mem = 0; + } + if (cpustat.usecount.mem == 0) { + cpustat.usecount.mem_delayed = 1; + schedule_delayed_work(&mem_rel_work, HZ); + } + + mutex_unlock(&cpustat.lock); + + return 0; +} +#endif + +void dsp_register_mem_cb(int (*req_cb)(void), void (*rel_cb)(void)) +{ + mutex_lock(&cpustat.lock); + + cpustat.mem_req_cb = req_cb; + cpustat.mem_rel_cb = rel_cb; + + /* + * This function must be called while mem is enabled! + */ + BUG_ON(cpustat.usecount.mem == 0); + + mutex_unlock(&cpustat.lock); +} + +void dsp_unregister_mem_cb(void) +{ + mutex_lock(&cpustat.lock); + cpustat.mem_req_cb = NULL; + cpustat.mem_rel_cb = NULL; + mutex_unlock(&cpustat.lock); +} +#else +void dsp_register_mem_cb(int (*req_cb)(void), void (*rel_cb)(void)) { } +void dsp_unregister_mem_cb(void) { } +#endif /* CONFIG_ARCH_OMAP1 */ + +arch_initcall(omap_dsp_init); + +#ifdef CONFIG_ARCH_OMAP1 +EXPORT_SYMBOL(omap_dsp_request_mpui); +EXPORT_SYMBOL(omap_dsp_release_mpui); +#if defined(CONFIG_ARCH_OMAP1) && defined(CONFIG_OMAP_MMU_FWK) +EXPORT_SYMBOL(omap_dsp_request_mem); +EXPORT_SYMBOL(omap_dsp_release_mem); +#endif +#endif /* CONFIG_ARCH_OMAP1 */ + +#ifdef CONFIG_OMAP_DSP_MODULE +#if defined(CONFIG_ARCH_OMAP1) +EXPORT_SYMBOL(dsp_ck_handle); +EXPORT_SYMBOL(api_ck_handle); +#elif defined(CONFIG_ARCH_OMAP2) +EXPORT_SYMBOL(dsp_fck_handle); +EXPORT_SYMBOL(dsp_ick_handle); +#endif +EXPORT_SYMBOL(omap_dsp); +EXPORT_SYMBOL(dspmem_base); +EXPORT_SYMBOL(dspmem_size); +EXPORT_SYMBOL(daram_base); +EXPORT_SYMBOL(daram_size); +EXPORT_SYMBOL(saram_base); +EXPORT_SYMBOL(saram_size); +EXPORT_SYMBOL(dsp_set_rstvect); +EXPORT_SYMBOL(dsp_get_rstvect); +#ifdef CONFIG_ARCH_OMAP1 +EXPORT_SYMBOL(dsp_set_idle_boot_base); +EXPORT_SYMBOL(dsp_reset_idle_boot_base); +#endif /* CONFIG_ARCH_OMAP1 */ +EXPORT_SYMBOL(dsp_cpustat_request); +EXPORT_SYMBOL(dsp_cpustat_get_stat); +EXPORT_SYMBOL(dsp_cpustat_get_icrmask); +EXPORT_SYMBOL(dsp_cpustat_set_icrmask); +EXPORT_SYMBOL(dsp_register_mem_cb); +EXPORT_SYMBOL(dsp_unregister_mem_cb); + +EXPORT_SYMBOL(__cpu_flush_kern_tlb_range); +EXPORT_SYMBOL(cpu_architecture); +EXPORT_SYMBOL(pmd_clear_bad); +#endif diff -Npru a/arch/arm/plat-omap/dsp/dsp.h b/arch/arm/plat-omap/dsp/dsp.h --- a/arch/arm/plat-omap/dsp/dsp.h 2009-05-12 21:13:59.000000000 +0200 +++ b/arch/arm/plat-omap/dsp/dsp.h 2009-06-01 22:31:15.000000000 +0200 @@ -0,0 +1,391 @@ +/* + * This file is part of OMAP DSP driver (DSP Gateway version 3.3.1) + * + * Copyright (C) 2002-2006 Nokia Corporation. All rights reserved. + * + * Contact: Toshihiro Kobayashi <toshihiro.kobayashi(a)nokia.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. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __PLAT_OMAP_DSP_DSP_H +#define __PLAT_OMAP_DSP_DSP_H + +#include "hardware_dsp.h" +#include <mach/dsp_common.h> +#include <mach/mmu.h> + + +#ifdef CONFIG_ARCH_OMAP2 +#include "../../../arch/arm/mach-omap2/prm.h" +#include "../../../arch/arm/mach-omap2/prm-regbits-24xx.h" +#include "../../../arch/arm/mach-omap2/cm.h" +#include "../../../arch/arm/mach-omap2/cm-regbits-24xx.h" +#endif + +/* + * MAJOR device number: !! allocated arbitrary !! + */ +#define OMAP_DSP_CTL_MAJOR 96 +#define OMAP_DSP_TASK_MAJOR 97 + +#define OLD_BINARY_SUPPORT y + +#ifdef OLD_BINARY_SUPPORT +#define MBREV_3_0 0x0017 +#define MBREV_3_2 0x0018 +#endif + +#define DSP_INIT_PAGE 0xfff000 + +#ifdef CONFIG_ARCH_OMAP1 +/* idle program will be placed at IDLEPG_BASE. */ +#define IDLEPG_BASE 0xfffe00 +#define IDLEPG_SIZE 0x100 +#endif /* CONFIG_ARCH_OMAP1 */ + +/* timeout value for DSP response */ +#define DSP_TIMEOUT (10 * HZ) + +enum dsp_mem_type_e { + MEM_TYPE_CROSSING = -1, + MEM_TYPE_NONE = 0, + MEM_TYPE_DARAM, + MEM_TYPE_SARAM, + MEM_TYPE_EXTERN, +}; + + +typedef int __bitwise arm_dsp_dir_t; +#define DIR_A2D ((__force arm_dsp_dir_t) 1) +#define DIR_D2A ((__force arm_dsp_dir_t) 2) + +enum cfgstat_e { + CFGSTAT_CLEAN = 0, + CFGSTAT_READY, + CFGSTAT_SUSPEND, + CFGSTAT_RESUME, /* request only */ + CFGSTAT_MAX +}; + +enum errcode_e { + ERRCODE_WDT = 0, + ERRCODE_MMU, + ERRCODE_MAX +}; + +/* keep 2 entries for TID_FREE and TID_ANON */ +#define TASKDEV_MAX 254 + +#define MK32(uw,lw) (((u32)(uw)) << 16 | (lw)) +#define MKLONG(uw,lw) (((unsigned long)(uw)) << 16 | (lw)) +#define MKVIRT(uw,lw) dspword_to_virt(MKLONG((uw), (lw))); + +struct sync_seq { + u16 da_dsp; + u16 da_arm; + u16 ad_dsp; + u16 ad_arm; +}; + +struct mem_sync_struct { + struct sync_seq *DARAM; + struct sync_seq *SARAM; + struct sync_seq *SDRAM; +}; + +/* struct mbcmd and union mbcmd_hw must be compatible */ +struct mbcmd { + u32 data:16; + u32 cmd_l:8; + u32 cmd_h:7; + u32 seq:1; +}; + +#define MBCMD_INIT(h, l, d) { \ + .cmd_h = (h), \ + .cmd_l = (l), \ + .data = (d), \ + } + +struct mb_exarg { + u8 tid; + int argc; + u16 *argv; +}; + +typedef u32 dsp_long_t; /* must have ability to carry TADD_ABORTADR */ + +extern void dsp_mbox_start(void); +extern void dsp_mbox_stop(void); +extern int dsp_mbox_config(void *p); +extern int sync_with_dsp(u16 *syncwd, u16 tid, int try_cnt); +extern int __dsp_mbcmd_send_exarg(struct mbcmd *mb, struct mb_exarg *arg, + int recovery_flag); +#define dsp_mbcmd_send(mb) __dsp_mbcmd_send_exarg((mb), NULL, 0) +#define dsp_mbcmd_send_exarg(mb, arg) __dsp_mbcmd_send_exarg((mb), (arg), 0) +extern int dsp_mbcmd_send_and_wait_exarg(struct mbcmd *mb, struct mb_exarg *arg, + wait_queue_head_t *q); +#define dsp_mbcmd_send_and_wait(mb, q) \ + dsp_mbcmd_send_and_wait_exarg((mb), NULL, (q)) + +static inline int __mbcompose_send_exarg(u8 cmd_h, u8 cmd_l, u16 data, + struct mb_exarg *arg, + int recovery_flag) +{ + struct mbcmd mb = MBCMD_INIT(cmd_h, cmd_l, data); + return __dsp_mbcmd_send_exarg(&mb, arg, recovery_flag); +} +#define mbcompose_send(cmd_h, cmd_l, data) \ + __mbcompose_send_exarg(MBOX_CMD_DSP_##cmd_h, (cmd_l), (data), NULL, 0) +#define mbcompose_send_exarg(cmd_h, cmd_l, data, arg) \ + __mbcompose_send_exarg(MBOX_CMD_DSP_##cmd_h, (cmd_l), (data), arg, 0) +#define mbcompose_send_recovery(cmd_h, cmd_l, data) \ + __mbcompose_send_exarg(MBOX_CMD_DSP_##cmd_h, (cmd_l), (data), NULL, 1) + +static inline int __mbcompose_send_and_wait_exarg(u8 cmd_h, u8 cmd_l, + u16 data, + struct mb_exarg *arg, + wait_queue_head_t *q) +{ + struct mbcmd mb = MBCMD_INIT(cmd_h, cmd_l, data); + return dsp_mbcmd_send_and_wait_exarg(&mb, arg, q); +} +#define mbcompose_send_and_wait(cmd_h, cmd_l, data, q) \ + __mbcompose_send_and_wait_exarg(MBOX_CMD_DSP_##cmd_h, (cmd_l), (data), \ + NULL, (q)) +#define mbcompose_send_and_wait_exarg(cmd_h, cmd_l, data, arg, q) \ + __mbcompose_send_and_wait_exarg(MBOX_CMD_DSP_##cmd_h, (cmd_l), (data), \ + (arg), (q)) + +extern struct ipbuf_head *bid_to_ipbuf(u16 bid); +extern void ipbuf_start(void); +extern void ipbuf_stop(void); +extern int ipbuf_config(u16 ln, u16 lsz, void *base); +extern int ipbuf_sys_config(void *p, arm_dsp_dir_t dir); +extern int ipbuf_p_validate(void *p, arm_dsp_dir_t dir); +extern struct ipbuf_head *get_free_ipbuf(u8 tid); +extern void release_ipbuf(struct ipbuf_head *ipb_h); +extern void balance_ipbuf(void); +extern void unuse_ipbuf(struct ipbuf_head *ipb_h); +extern void unuse_ipbuf_nowait(struct ipbuf_head *ipb_h); + +#define release_ipbuf_pvt(ipbuf_pvt) \ + do { \ + (ipbuf_pvt)->s = TID_FREE; \ + } while(0) + +extern int mbox_revision; + +extern int dsp_cfgstat_request(enum cfgstat_e st); +extern enum cfgstat_e dsp_cfgstat_get_stat(void); +extern int dsp_set_runlevel(u8 level); + +extern int dsp_task_config_all(u8 n); +extern void dsp_task_unconfig_all(void); +extern u8 dsp_task_count(void); +extern int dsp_taskmod_busy(void); +extern int dsp_mkdev(char *name); +extern int dsp_rmdev(char *name); +extern int dsp_tadd_minor(unsigned char minor, dsp_long_t adr); +extern int dsp_tdel_minor(unsigned char minor); +extern int dsp_tkill_minor(unsigned char minor); +extern long taskdev_state_stale(unsigned char minor); +extern int dsp_dbg_config(u16 *buf, u16 sz, u16 lsz); +extern void dsp_dbg_stop(void); + +extern int ipbuf_is_held(u8 tid, u16 bid); + +extern int dsp_mem_sync_inc(void); +extern int dsp_mem_sync_config(struct mem_sync_struct *sync); +extern enum dsp_mem_type_e dsp_mem_type(void *vadr, size_t len); +extern int dsp_address_validate(void *p, size_t len, char *fmt, ...); +#ifdef CONFIG_ARCH_OMAP1 +extern void dsp_mem_usecount_clear(void); +#endif +extern void exmap_use(void *vadr, size_t len); +extern void exmap_unuse(void *vadr, size_t len); +extern unsigned long dsp_virt_to_phys(void *vadr, size_t *len); +extern void dsp_mem_start(void); +extern void dsp_mem_stop(void); + +extern void dsp_twch_start(void); +extern void dsp_twch_stop(void); +extern void dsp_twch_touch(void); + +extern void dsp_err_start(void); +extern void dsp_err_stop(void); +extern void dsp_err_set(enum errcode_e code, unsigned long arg); +extern void dsp_err_clear(enum errcode_e code); +extern int dsp_err_isset(enum errcode_e code); + +enum cmd_l_type_e { + CMD_L_TYPE_NULL, + CMD_L_TYPE_TID, + CMD_L_TYPE_SUBCMD, +}; + +struct cmdinfo { + char *name; + enum cmd_l_type_e cmd_l_type; + void (*handler)(struct mbcmd *mb); +}; + +extern const struct cmdinfo *cmdinfo[]; + +#define cmd_name(mb) (cmdinfo[(mb).cmd_h]->name) +extern char *subcmd_name(struct mbcmd *mb); + +extern void mblog_add(struct mbcmd *mb, arm_dsp_dir_t dir); + +extern struct omap_mmu dsp_mmu; + +#define dsp_mem_enable(addr) omap_mmu_mem_enable(&dsp_mmu, (addr)) +#define dsp_mem_disable(addr) omap_mmu_mem_disable(&dsp_mmu, (addr)) + +#define DSPSPACE_SIZE 0x1000000 + +#define omap_set_bit_regw(b,r) \ + do { omap_writew(omap_readw(r) | (b), (r)); } while(0) +#define omap_clr_bit_regw(b,r) \ + do { omap_writew(omap_readw(r) & ~(b), (r)); } while(0) +#define omap_set_bit_regl(b,r) \ + do { omap_writel(omap_readl(r) | (b), (r)); } while(0) +#define omap_clr_bit_regl(b,r) \ + do { omap_writel(omap_readl(r) & ~(b), (r)); } while(0) +#define omap_set_bits_regl(val,mask,r) \ + do { omap_writel((omap_readl(r) & ~(mask)) | (val), (r)); } while(0) + +#define dspword_to_virt(dw) ((void *)(dspmem_base + ((dw) << 1))) +#define dspbyte_to_virt(db) ((void *)(dspmem_base + (db))) +#define virt_to_dspword(va) \ + ((dsp_long_t)(((unsigned long)(va) - dspmem_base) >> 1)) +#define virt_to_dspbyte(va) \ + ((dsp_long_t)((unsigned long)(va) - dspmem_base)) +#define is_dsp_internal_mem(va) \ + (((unsigned long)(va) >= dspmem_base) && \ + ((unsigned long)(va) < dspmem_base + dspmem_size)) +#define is_dspbyte_internal_mem(db) ((db) < dspmem_size) +#define is_dspword_internal_mem(dw) (((dw) << 1) < dspmem_size) + +#ifdef CONFIG_ARCH_OMAP1 +/* + * MPUI byteswap/wordswap on/off + * default setting: wordswap = all, byteswap = APIMEM only + */ +#define mpui_wordswap_on() \ + omap_set_bits_regl(MPUI_CTRL_WORDSWAP_ALL, MPUI_CTRL_WORDSWAP_MASK, \ + MPUI_CTRL) + +#define mpui_wordswap_off() \ + omap_set_bits_regl(MPUI_CTRL_WORDSWAP_NONE, MPUI_CTRL_WORDSWAP_MASK, \ + MPUI_CTRL) + +#define mpui_byteswap_on() \ + omap_set_bits_regl(MPUI_CTRL_BYTESWAP_API, MPUI_CTRL_BYTESWAP_MASK, \ + MPUI_CTRL) + +#define mpui_byteswap_off() \ + omap_set_bits_regl(MPUI_CTRL_BYTESWAP_NONE, MPUI_CTRL_BYTESWAP_MASK, \ + MPUI_CTRL) + +/* + * TC wordswap on / off + */ +#define tc_wordswap() \ + do { \ + omap_writel(TC_ENDIANISM_SWAP_WORD | TC_ENDIANISM_EN, \ + TC_ENDIANISM); \ + } while(0) + +#define tc_noswap() omap_clr_bit_regl(TC_ENDIANISM_EN, TC_ENDIANISM) + +/* + * enable priority registers, EMIF, MPUI control logic + */ +#define __dsp_enable() omap_set_bit_regw(ARM_RSTCT1_DSP_RST, ARM_RSTCT1) +#define __dsp_disable() omap_clr_bit_regw(ARM_RSTCT1_DSP_RST, ARM_RSTCT1) +#define __dsp_run() omap_set_bit_regw(ARM_RSTCT1_DSP_EN, ARM_RSTCT1) +#define __dsp_reset() omap_clr_bit_regw(ARM_RSTCT1_DSP_EN, ARM_RSTCT1) +#endif /* CONFIG_ARCH_OMAP1 */ + +#ifdef CONFIG_ARCH_OMAP2 +/* + * PRCM / IPI control logic + * + * REVISIT: these macros should probably be static inline functions + */ +#define __dsp_core_enable() \ + do { prm_write_mod_reg(prm_read_mod_reg(OMAP24XX_DSP_MOD, RM_RSTCTRL) \ + & ~OMAP24XX_RST1_DSP, OMAP24XX_DSP_MOD, RM_RSTCTRL); } while (0) +#define __dsp_core_disable() \ + do { prm_write_mod_reg(prm_read_mod_reg(OMAP24XX_DSP_MOD, RM_RSTCTRL) \ + | OMAP24XX_RST1_DSP, OMAP24XX_DSP_MOD, RM_RSTCTRL); } while (0) +#define __dsp_per_enable() \ + do { prm_write_mod_reg(prm_read_mod_reg(OMAP24XX_DSP_MOD, RM_RSTCTRL) \ + & ~OMAP24XX_RST2_DSP, OMAP24XX_DSP_MOD, RM_RSTCTRL); } while (0) +#define __dsp_per_disable() \ + do { prm_write_mod_reg(prm_read_mod_reg(OMAP24XX_DSP_MOD, RM_RSTCTRL) \ + | OMAP24XX_RST2_DSP, OMAP24XX_DSP_MOD, RM_RSTCTRL); } while (0) +#endif /* CONFIG_ARCH_OMAP2 */ + +#if defined(CONFIG_ARCH_OMAP1) +extern struct clk *dsp_ck_handle; +extern struct clk *api_ck_handle; +#elif defined(CONFIG_ARCH_OMAP2) +extern struct clk *dsp_fck_handle; +extern struct clk *dsp_ick_handle; +#endif +extern dsp_long_t dspmem_base, dspmem_size, + daram_base, daram_size, + saram_base, saram_size; + +enum cpustat_e { + CPUSTAT_RESET = 0, +#ifdef CONFIG_ARCH_OMAP1 + CPUSTAT_GBL_IDLE, + CPUSTAT_CPU_IDLE, +#endif + CPUSTAT_RUN, + CPUSTAT_MAX +}; + +int dsp_set_rstvect(dsp_long_t adr); +dsp_long_t dsp_get_rstvect(void); +void dsp_set_idle_boot_base(dsp_long_t adr, size_t size); +void dsp_reset_idle_boot_base(void); +void dsp_cpustat_request(enum cpustat_e req); +enum cpustat_e dsp_cpustat_get_stat(void); +u16 dsp_cpustat_get_icrmask(void); +void dsp_cpustat_set_icrmask(u16 mask); +void dsp_register_mem_cb(int (*req_cb)(void), void (*rel_cb)(void)); +void dsp_unregister_mem_cb(void); + +#if defined(CONFIG_ARCH_OMAP1) +#define command_dvfs_stop(m) (0) +#define command_dvfs_start(m) (0) +#elif defined(CONFIG_ARCH_OMAP2) +#define command_dvfs_stop(m) \ + (((m)->cmd_l == KFUNC_POWER) && ((m)->data == DVFS_STOP)) +#define command_dvfs_start(m) \ + (((m)->cmd_l == KFUNC_POWER) && ((m)->data == DVFS_START)) +#endif + +extern struct omap_dsp *omap_dsp; + +extern int dsp_late_init(void); + +#endif /* __PLAT_OMAP_DSP_DSP_H */ diff -Npru a/arch/arm/plat-omap/dsp/hardware_dsp.h b/arch/arm/plat-omap/dsp/hardware_dsp.h --- a/arch/arm/plat-omap/dsp/hardware_dsp.h 2009-05-12 21:13:59.000000000 +0200 +++ b/arch/arm/plat-omap/dsp/hardware_dsp.h 2009-06-01 22:31:15.000000000 +0200 @@ -0,0 +1,34 @@ +/* + * This file is part of OMAP DSP driver (DSP Gateway version 3.3.1) + * + * Copyright (C) 2002-2006 Nokia Corporation. All rights reserved. + * + * Contact: Toshihiro Kobayashi <toshihiro.kobayashi(a)nokia.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. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __OMAP_DSP_HARDWARE_DSP_H +#define __OMAP_DSP_HARDWARE_DSP_H + +#ifdef CONFIG_ARCH_OMAP1 +#include "omap1_dsp.h" +#endif +#if defined(CONFIG_ARCH_OMAP2) || defined(CONFIG_ARCH_OMAP3430) +#include "omap2_dsp.h" +#endif + +#endif /* __OMAP_DSP_HARDWARE_DSP_H */ diff -Npru a/arch/arm/plat-omap/dsp/mmu.h b/arch/arm/plat-omap/dsp/mmu.h --- a/arch/arm/plat-omap/dsp/mmu.h 2009-05-12 21:13:59.000000000 +0200 +++ b/arch/arm/plat-omap/dsp/mmu.h 2009-06-01 22:31:15.000000000 +0200 @@ -0,0 +1,140 @@ +#ifndef __PLAT_OMAP_DSP_MMU_H +#define __PLAT_OMAP_DSP_MMU_H + +#ifdef CONFIG_ARCH_OMAP1 + +#ifdef CONFIG_ARCH_OMAP15XX +struct omap_mmu dsp_mmu = { + .name = "mmu:dsp", + .type = OMAP_MMU_DSP, + .base = IO_ADDRESS(OMAP1510_DSP_MMU_BASE), + .membase = OMAP1510_DSP_BASE, + .memsize = OMAP1510_DSP_SIZE, + .nr_tlb_entries = 32, + .addrspace = 24, + .irq = INT_1510_DSP_MMU, + .ops = &omap1_mmu_ops, +}; +#endif +#ifdef CONFIG_ARCH_OMAP16XX +struct omap_mmu dsp_mmu = { + .name = "mmu:dsp", + .type = OMAP_MMU_DSP, + .base = IO_ADDRESS(OMAP16XX_DSP_MMU_BASE), + .membase = OMAP16XX_DSP_BASE, + .memsize = OMAP16XX_DSP_SIZE, + .nr_tlb_entries = 32, + .addrspace = 24, + .irq = INT_1610_DSP_MMU, + .ops = &omap1_mmu_ops, +}; +#endif +#else /* OMAP2 */ +struct omap_mmu dsp_mmu = { + .name = "mmu:dsp", + .type = OMAP_MMU_DSP, + .base = DSP_MMU_24XX_VIRT, + .membase = DSP_MEM_24XX_VIRT, + .memsize = DSP_MEM_24XX_SIZE, + .nr_tlb_entries = 32, + .addrspace = 24, + .irq = INT_24XX_DSP_MMU, + .ops = &omap2_mmu_ops, +}; + +#define IOMAP_VAL 0x3f +#endif + +#ifdef CONFIG_FB_OMAP_LCDC_EXTERNAL +static struct omapfb_notifier_block *omapfb_nb; +static int omapfb_ready; +#endif + +/* + * OMAP1 EMIFF access + */ +#ifdef CONFIG_ARCH_OMAP1 +#define EMIF_PRIO_LB_MASK 0x0000f000 +#define EMIF_PRIO_LB_SHIFT 12 +#define EMIF_PRIO_DMA_MASK 0x00000f00 +#define EMIF_PRIO_DMA_SHIFT 8 +#define EMIF_PRIO_DSP_MASK 0x00000070 +#define EMIF_PRIO_DSP_SHIFT 4 +#define EMIF_PRIO_MPU_MASK 0x00000007 +#define EMIF_PRIO_MPU_SHIFT 0 +#define set_emiff_dma_prio(prio) \ + do { \ + omap_writel((omap_readl(OMAP_TC_OCPT1_PRIOR) & \ + ~EMIF_PRIO_DMA_MASK) | \ + ((prio) << EMIF_PRIO_DMA_SHIFT), \ + OMAP_TC_OCPT1_PRIOR); \ + } while(0) +#else +#define set_emiff_dma_prio(prio) do { } while (0) +#endif /* CONFIG_ARCH_OMAP1 */ + +#ifdef CONFIG_ARCH_OMAP1 +static int dsp_mmu_itack(void) +{ + unsigned long dspadr; + + pr_info("omapdsp: sending DSP MMU interrupt ack.\n"); + if (!dsp_err_isset(ERRCODE_MMU)) { + printk(KERN_ERR "omapdsp: DSP MMU error has not been set.\n"); + return -EINVAL; + } + dspadr = dsp_mmu.fault_address & ~(SZ_4K-1); + /* FIXME: reserve TLB entry for this */ + omap_mmu_exmap(&dsp_mmu, dspadr, 0, SZ_4K, EXMAP_TYPE_MEM); + pr_info("omapdsp: falling into recovery runlevel...\n"); + dsp_set_runlevel(RUNLEVEL_RECOVERY); + omap_mmu_itack(&dsp_mmu); + udelay(100); + omap_mmu_exunmap(&dsp_mmu, dspadr); + dsp_err_clear(ERRCODE_MMU); + return 0; +} + +/* + * intmem_enable() / disable(): + * if the address is in DSP internal memories, + * we send PM mailbox commands so that DSP DMA domain won't go in idle + * when ARM is accessing to those memories. + */ +static int intmem_enable(void) +{ + int ret = 0; + + if (dsp_cfgstat_get_stat() == CFGSTAT_READY) + ret = mbcompose_send(PM, PM_ENABLE, DSPREG_ICR_DMA); + + return ret; +} + +static void intmem_disable(void) { + if (dsp_cfgstat_get_stat() == CFGSTAT_READY) + mbcompose_send(PM, PM_DISABLE, DSPREG_ICR_DMA); +} +#else +static int intmem_enable(void) { return 0; } +static void intmem_disable(void) { } +static int dsp_mmu_itack(void) { return 0; } +#endif + +#ifdef CONFIG_ARCH_OMAP2 +static inline void dsp_mem_ipi_init(void) +{ + int i, dspmem_pg_count; + dspmem_pg_count = dspmem_size >> 12; + for (i = 0; i < dspmem_pg_count; i++) { + writel(i, DSP_IPI_INDEX); + writel(DSP_IPI_ENTRY_ELMSIZEVALUE_16, DSP_IPI_ENTRY); + } + writel(1, DSP_IPI_ENABLE); + writel(IOMAP_VAL, DSP_IPI_IOMAP); +} +#else +static inline void dsp_mem_ipi_init(void) { } +#endif + +#endif /* __PLAT_OMAP_DSP_MMU_H */ diff -Npru a/arch/arm/plat-omap/include/mach/mmu.h b/arch/arm/plat-omap/include/mach/mmu.h --- a/arch/arm/plat-omap/include/mach/mmu.h 2009-05-12 21:13:59.000000000 +0200 +++ b/arch/arm/plat-omap/include/mach/mmu.h 2009-06-01 22:31:15.000000000 +0200 @@ -0,0 +1,211 @@ +#ifndef __ARCH_OMAP_MMU_H +#define __ARCH_OMAP_MMU_H + +#include <linux/device.h> +#include <linux/workqueue.h> + +enum exmap_type { + EXMAP_TYPE_MEM, + EXMAP_TYPE_FB +}; + +enum omap_mmu_type { + OMAP_MMU_DSP, + OMAP_MMU_IVA1, + OMAP_MMU_CAMERA, +}; + +struct exmap_tbl { + unsigned int valid:1; + unsigned int prsvd:1; + int usecount; /* reference count by mmap */ + enum exmap_type type; + void *buf; /* virtual address of the buffer, + * i.e. 0xc0000000 - */ + void *vadr; /* DSP shadow space, + * i.e. 0xe0000000 - 0xe0ffffff */ + unsigned int order; + struct { + int prev; + int next; + } link; /* grouping */ +}; + +struct cam_ram_regset { + union { + struct { + u16 cam_l; + u16 cam_h; + }; + + u32 cam; + }; + + union { + struct { + u16 ram_l; + u16 ram_h; + }; + + u32 ram; + }; +}; + +struct omap_mmu_tlb_lock { + int base; + int victim; +}; + +struct omap_mmu; +struct omap_mmu_tlb_entry; + +#ifdef CONFIG_ARCH_OMAP1 +extern struct omap_mmu_ops omap1_mmu_ops; +extern void omap_mmu_itack(struct omap_mmu *mmu); +#elif defined(CONFIG_ARCH_OMAP2) +extern struct omap_mmu_ops omap2_mmu_ops; +static inline void omap_mmu_itack(struct omap_mmu *mmu) +{ +} +#endif + +struct omap_mmu_ops { + int (*startup)(struct omap_mmu *mmu); + void (*shutdown)(struct omap_mmu *mmu); + + /* TLB operations */ + void (*read_tlb)(struct omap_mmu *, struct cam_ram_regset *); + void (*load_tlb)(struct omap_mmu *, struct cam_ram_regset *); + ssize_t (*show)(struct omap_mmu *, char *, struct omap_mmu_tlb_lock *); + + /* CAM / RAM operations */ + struct cam_ram_regset *(*cam_ram_alloc)(struct omap_mmu *, + struct omap_mmu_tlb_entry *); + int (*cam_ram_valid)(struct cam_ram_regset *); + unsigned long (*cam_va)(struct cam_ram_regset *); + + /* Memory operations */ + int (*mem_enable)(struct omap_mmu *, void *); + int (*mem_disable)(struct omap_mmu *, void *); + + void (*interrupt)(struct omap_mmu *); + + /* PTE attribute operations */ + pgprot_t (*pte_get_attr)(struct omap_mmu_tlb_entry *); +}; + +struct omap_mmu { + const char *name; + unsigned long base; + struct clk *clk; + + unsigned long membase, memsize; + struct clk *memclk; + + enum omap_mmu_type type; + + struct device *dev; + + struct rw_semaphore exmap_sem; + struct exmap_tbl *exmap_tbl; + + unsigned int nr_tlb_entries; + unsigned int nr_exmap_preserved; + + struct mm_struct *twl_mm; + + /* Size of virtual address space, in bits */ + unsigned int addrspace; + + /* Interrupt */ + unsigned int irq; + unsigned long fault_address; + struct work_struct irq_work; + + struct omap_mmu_ops *ops; +}; + +#define omap_mmu_internal_memory(mmu, addr) \ + (likely(mmu->membase) && (((unsigned long)(addr) >= mmu->membase) && \ + ((unsigned long)(addr) < mmu->membase + mmu->memsize))) + +#define INIT_EXMAP_TBL_ENTRY(ent, b, v, typ, od) \ +do { \ + (ent)->buf = (b); \ + (ent)->vadr = (v); \ + (ent)->valid = 1; \ + (ent)->prsvd = 0; \ + (ent)->usecount = 0; \ + (ent)->type = (typ); \ + (ent)->order = (od); \ + (ent)->link.next = -1; \ + (ent)->link.prev = -1; \ +} while (0) + +#define INIT_EXMAP_TBL_ENTRY_4KB_PRESERVED(ent, b, v) \ +do { \ + (ent)->buf = (b); \ + (ent)->vadr = (v); \ + (ent)->valid = 1; \ + (ent)->prsvd = 1; \ + (ent)->usecount = 0; \ + (ent)->type = EXMAP_TYPE_MEM; \ + (ent)->order = 0; \ + (ent)->link.next = -1; \ + (ent)->link.prev = -1; \ +} while (0) + +#define omap_mmu_to_virt(mmu, db) ((void *)((mmu)->membase + (db))) +#define virt_to_omap_mmu(mmu, va) \ + (((unsigned long)(va) - (mmu)->membase)) + +/* arch/arm/plat-omap/mmu.c */ +int omap_mmu_register(struct omap_mmu *mmu); +void omap_mmu_unregister(struct omap_mmu *mmu); + +void omap_mmu_enable(struct omap_mmu *mmu, int reset); +void omap_mmu_disable(struct omap_mmu *mmu); + +int omap_mmu_mem_enable(struct omap_mmu *mmu, void *addr); +void omap_mmu_mem_disable(struct omap_mmu *mmu, void *addr); + +void omap_mmu_read_tlb(struct omap_mmu *mmu, struct omap_mmu_tlb_lock *lock, + struct cam_ram_regset *cr); + +int omap_mmu_load_tlb_entry(struct omap_mmu *, struct omap_mmu_tlb_entry *); +int omap_mmu_clear_tlb_entry(struct omap_mmu *, unsigned long vadr); + +int omap_mmu_load_pte_entry(struct omap_mmu *mmu, + struct omap_mmu_tlb_entry *entry); +int omap_mmu_clear_pte_entry(struct omap_mmu *mmu, unsigned long vadr); + +int omap_mmu_kmem_reserve(struct omap_mmu *mmu, unsigned long size); +void omap_mmu_kmem_release(void); + +unsigned long omap_mmu_virt_to_phys(struct omap_mmu *mmu, void *vadr, + size_t *len); + +int omap_mmu_exmap(struct omap_mmu *mmu, unsigned long dspadr, + unsigned long padr, unsigned long size, + enum exmap_type type); +int omap_mmu_exunmap(struct omap_mmu *mmu, unsigned long dspadr); +void omap_mmu_exmap_flush(struct omap_mmu *mmu); +void omap_mmu_exmap_use(struct omap_mmu *mmu, void *vadr, size_t len); +void omap_mmu_exmap_unuse(struct omap_mmu *mmu, void *vadr, size_t len); + +int exmap_set_armmmu(struct omap_mmu *mmu, unsigned long virt, + unsigned long phys, unsigned long size); +void exmap_clear_armmmu(struct omap_mmu *mmu, unsigned long virt, + unsigned long size); +void exmap_setup_preserved_mem_page(struct omap_mmu *mmu, void *buf, + unsigned long dspadr, int index); +void exmap_clear_mem_page(struct omap_mmu *mmu, unsigned long dspadr); +int exmap_valid(struct omap_mmu *mmu, void *vadr, size_t len); + +/* To be obsolete for backward compatibility */ +ssize_t __omap_mmu_mem_read(struct omap_mmu *mmu, struct bin_attribute *, + char *buf, loff_t offset, size_t count); +ssize_t __omap_mmu_mem_write(struct omap_mmu *mmu, struct bin_attribute *, + char *buf, loff_t offset, size_t count); + +#endif /* __ARCH_OMAP_MMU_H */ diff -Npru a/arch/arm/mach-omap1/mmu.h b/arch/arm/mach-omap1/mmu.h --- a/arch/arm/mach-omap1/mmu.h 2009-05-12 21:13:59.000000000 +0200 +++ b/arch/arm/mach-omap1/mmu.h 2009-06-01 22:31:15.000000000 +0200 @@ -0,0 +1,119 @@ +#ifndef __MACH_OMAP1_MMU_H +#define __MACH_OMAP1_MMU_H + +#include <linux/io.h> +#include <mach/mmu.h> + +#define MMU_LOCK_BASE_MASK (0x3f << 10) +#define MMU_LOCK_VICTIM_MASK (0x3f << 4) + +#define OMAP_MMU_PREFETCH 0x00 +#define OMAP_MMU_WALKING_ST 0x04 +#define OMAP_MMU_CNTL 0x08 +#define OMAP_MMU_FAULT_AD_H 0x0c +#define OMAP_MMU_FAULT_AD_L 0x10 +#define OMAP_MMU_FAULT_ST 0x14 +#define OMAP_MMU_IT_ACK 0x18 +#define OMAP_MMU_TTB_H 0x1c +#define OMAP_MMU_TTB_L 0x20 +#define OMAP_MMU_LOCK 0x24 +#define OMAP_MMU_LD_TLB 0x28 +#define OMAP_MMU_CAM_H 0x2c +#define OMAP_MMU_CAM_L 0x30 +#define OMAP_MMU_RAM_H 0x34 +#define OMAP_MMU_RAM_L 0x38 +#define OMAP_MMU_GFLUSH 0x3c +#define OMAP_MMU_FLUSH_ENTRY 0x40 +#define OMAP_MMU_READ_CAM_H 0x44 +#define OMAP_MMU_READ_CAM_L 0x48 +#define OMAP_MMU_READ_RAM_H 0x4c +#define OMAP_MMU_READ_RAM_L 0x50 + +#define OMAP_MMU_CNTL_BURST_16MNGT_EN 0x0020 +#define OMAP_MMU_CNTL_WTL_EN 0x0004 +#define OMAP_MMU_CNTL_MMU_EN 0x0002 +#define OMAP_MMU_CNTL_RESET_SW 0x0001 + +#define OMAP_MMU_FAULT_AD_H_DP 0x0100 +#define OMAP_MMU_FAULT_AD_H_ADR_MASK 0x00ff + +#define OMAP_MMU_FAULT_ST_PREF 0x0008 +#define OMAP_MMU_FAULT_ST_PERM 0x0004 +#define OMAP_MMU_FAULT_ST_TLB_MISS 0x0002 +#define OMAP_MMU_FAULT_ST_TRANS 0x0001 + +#define OMAP_MMU_IT_ACK_IT_ACK 0x0001 + +#define OMAP_MMU_CAM_H_VA_TAG_H_MASK 0x0003 + +#define OMAP_MMU_CAM_L_VA_TAG_L1_MASK 0xc000 +#define OMAP_MMU_CAM_L_VA_TAG_L2_MASK_1MB 0x0000 +#define OMAP_MMU_CAM_L_VA_TAG_L2_MASK_64KB 0x3c00 +#define OMAP_MMU_CAM_L_VA_TAG_L2_MASK_4KB 0x3fc0 +#define OMAP_MMU_CAM_L_VA_TAG_L2_MASK_1KB 0x3ff0 +#define OMAP_MMU_CAM_L_P 0x0008 +#define OMAP_MMU_CAM_L_V 0x0004 +#define OMAP_MMU_CAM_L_PAGESIZE_MASK 0x0003 +#define OMAP_MMU_CAM_L_PAGESIZE_1MB 0x0000 +#define OMAP_MMU_CAM_L_PAGESIZE_64KB 0x0001 +#define OMAP_MMU_CAM_L_PAGESIZE_4KB 0x0002 +#define OMAP_MMU_CAM_L_PAGESIZE_1KB 0x0003 + +#define OMAP_MMU_CAM_P OMAP_MMU_CAM_L_P +#define OMAP_MMU_CAM_V OMAP_MMU_CAM_L_V +#define OMAP_MMU_CAM_PAGESIZE_MASK OMAP_MMU_CAM_L_PAGESIZE_MASK +#define OMAP_MMU_CAM_PAGESIZE_1MB OMAP_MMU_CAM_L_PAGESIZE_1MB +#define OMAP_MMU_CAM_PAGESIZE_64KB OMAP_MMU_CAM_L_PAGESIZE_64KB +#define OMAP_MMU_CAM_PAGESIZE_4KB OMAP_MMU_CAM_L_PAGESIZE_4KB +#define OMAP_MMU_CAM_PAGESIZE_1KB OMAP_MMU_CAM_L_PAGESIZE_1KB +#define OMAP_MMU_CAM_PAGESIZE_16MB -1 /* unused in omap1 */ + +#define OMAP_MMU_RAM_L_RAM_LSB_MASK 0xfc00 +#define OMAP_MMU_RAM_L_AP_MASK 0x0300 +#define OMAP_MMU_RAM_L_AP_NA 0x0000 +#define OMAP_MMU_RAM_L_AP_RO 0x0200 +#define OMAP_MMU_RAM_L_AP_FA 0x0300 + +#define OMAP_MMU_LD_TLB_RD 0x0002 + +#define INIT_TLB_ENTRY(ent, v, p, ps) \ +do { \ + (ent)->va = (v); \ + (ent)->pa = (p); \ + (ent)->pgsz = (ps); \ + (ent)->prsvd = 0; \ + (ent)->ap = OMAP_MMU_RAM_L_AP_FA; \ + (ent)->tlb = 1; \ +} while (0) + +#define INIT_TLB_ENTRY_4KB_PRESERVED(ent, v, p) \ +do { \ + (ent)->va = (v); \ + (ent)->pa = (p); \ + (ent)->pgsz = OMAP_MMU_CAM_PAGESIZE_4KB; \ + (ent)->prsvd = OMAP_MMU_CAM_P; \ + (ent)->ap = OMAP_MMU_RAM_L_AP_FA; \ +} while (0) + +struct omap_mmu_tlb_entry { + unsigned long va; + unsigned long pa; + unsigned int pgsz, prsvd, valid; + + u16 ap; + unsigned int tlb; +}; + +static inline unsigned short +omap_mmu_read_reg(struct omap_mmu *mmu, unsigned long reg) +{ + return __raw_readw(mmu->base + reg); +} + +static inline void omap_mmu_write_reg(struct omap_mmu *mmu, + unsigned short val, unsigned long reg) +{ + __raw_writew(val, mmu->base + reg); +} + +#endif /* __MACH_OMAP1_MMU_H */ diff -Npru a/arch/arm/plat-omap/dsp/omap1_dsp.h b/arch/arm/plat-omap/dsp/omap1_dsp.h --- a/arch/arm/plat-omap/dsp/omap1_dsp.h 2009-05-12 21:13:59.000000000 +0200 +++ b/arch/arm/plat-omap/dsp/omap1_dsp.h 2009-06-01 22:31:15.000000000 +0200 @@ -0,0 +1,114 @@ +/* + * This file is part of OMAP DSP driver (DSP Gateway version 3.3.1) + * + * Copyright (C) 2002-2006 Nokia Corporation. All rights reserved. + * + * Contact: Toshihiro Kobayashi <toshihiro.kobayashi(a)nokia.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. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __OMAP_DSP_OMAP1_DSP_H +#define __OMAP_DSP_OMAP1_DSP_H + +#ifdef CONFIG_ARCH_OMAP15XX +#define OMAP1510_DARAM_BASE (OMAP1510_DSP_BASE + 0x0) +#define OMAP1510_DARAM_SIZE 0x10000 +#define OMAP1510_SARAM_BASE (OMAP1510_DSP_BASE + 0x10000) +#define OMAP1510_SARAM_SIZE 0x18000 +#endif + +#ifdef CONFIG_ARCH_OMAP16XX +#define OMAP16XX_DARAM_BASE (OMAP16XX_DSP_BASE + 0x0) +#define OMAP16XX_DARAM_SIZE 0x10000 +#define OMAP16XX_SARAM_BASE (OMAP16XX_DSP_BASE + 0x10000) +#define OMAP16XX_SARAM_SIZE 0x18000 +#endif + +/* + * Reset Control + */ +#define ARM_RSTCT1_SW_RST 0x0008 +#define ARM_RSTCT1_DSP_RST 0x0004 +#define ARM_RSTCT1_DSP_EN 0x0002 +#define ARM_RSTCT1_ARM_RST 0x0001 + +/* + * MPUI + */ +#define MPUI_CTRL_WORDSWAP_MASK 0x00600000 +#define MPUI_CTRL_WORDSWAP_ALL 0x00000000 +#define MPUI_CTRL_WORDSWAP_NONAPI 0x00200000 +#define MPUI_CTRL_WORDSWAP_API 0x00400000 +#define MPUI_CTRL_WORDSWAP_NONE 0x00600000 +#define MPUI_CTRL_AP_MASK 0x001c0000 +#define MPUI_CTRL_AP_MDH 0x00000000 +#define MPUI_CTRL_AP_MHD 0x00040000 +#define MPUI_CTRL_AP_DMH 0x00080000 +#define MPUI_CTRL_AP_HMD 0x000c0000 +#define MPUI_CTRL_AP_DHM 0x00100000 +#define MPUI_CTRL_AP_HDM 0x00140000 +#define MPUI_CTRL_BYTESWAP_MASK 0x00030000 +#define MPUI_CTRL_BYTESWAP_NONE 0x00000000 +#define MPUI_CTRL_BYTESWAP_NONAPI 0x00010000 +#define MPUI_CTRL_BYTESWAP_ALL 0x00020000 +#define MPUI_CTRL_BYTESWAP_API 0x00030000 +#define MPUI_CTRL_TIMEOUT_MASK 0x0000ff00 +#define MPUI_CTRL_APIF_HNSTB_DIV_MASK 0x000000f0 +#define MPUI_CTRL_S_NABORT_GL 0x00000008 +#define MPUI_CTRL_S_NABORT_32BIT 0x00000004 +#define MPUI_CTRL_EN_TIMEOUT 0x00000002 +#define MPUI_CTRL_HF_MCUCLK 0x00000001 +#define DSP_BOOT_CONFIG_DIRECT 0x00000000 +#define DSP_BOOT_CONFIG_PSD_DIRECT 0x00000001 +#define DSP_BOOT_CONFIG_IDLE 0x00000002 +#define DSP_BOOT_CONFIG_DL16 0x00000003 +#define DSP_BOOT_CONFIG_DL32 0x00000004 +#define DSP_BOOT_CONFIG_MPUI 0x00000005 +#define DSP_BOOT_CONFIG_INTERNAL 0x00000006 + +/* + * DSP boot mode + * direct: 0xffff00 + * pseudo direct: 0x080000 + * MPUI: branch 0x010000 + * internel: branch 0x024000 + */ +#define DSP_BOOT_ADR_DIRECT 0xffff00 +#define DSP_BOOT_ADR_PSD_DIRECT 0x080000 +#define DSP_BOOT_ADR_MPUI 0x010000 +#define DSP_BOOT_ADR_INTERNAL 0x024000 + +/* + * TC + */ +#define TC_ENDIANISM_SWAP 0x00000002 +#define TC_ENDIANISM_SWAP_WORD 0x00000002 +#define TC_ENDIANISM_SWAP_BYTE 0x00000000 +#define TC_ENDIANISM_EN 0x00000001 + +/* + * DSP ICR + */ +#define DSPREG_ICR_RESERVED_BITS 0xffc0 +#define DSPREG_ICR_EMIF 0x0020 +#define DSPREG_ICR_DPLL 0x0010 +#define DSPREG_ICR_PER 0x0008 +#define DSPREG_ICR_CACHE 0x0004 +#define DSPREG_ICR_DMA 0x0002 +#define DSPREG_ICR_CPU 0x0001 + +#endif /* __OMAP_DSP_OMAP1_DSP_H */

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[alsa-devel] [RFC] [PATCH 2/3] TTY: Add definition of a new line discipline required by Amstrad E3 (Delta) ASoC driver
by Janusz Krzysztofik 22 Jul '09

22 Jul '09

This patch adds new line discipline name an number to include/linux/tty.h. The line discipline will be used by the Amstrad E3 (Delta) sound driver that will come next in this series of patches. Created against linux-2.6.31-rc3. Applies to linux-omap-2.6 commit 7c5cb7862d32cb344be7831d466535d5255e35ac as well. Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- The line discipline is named after the board name as its code will be board specific. --- linux-2.6.31-rc1/include/linux/tty.h.orig 2009-07-04 16:39:56.000000000 +0200 +++ linux-2.6.31-rc1/include/linux/tty.h 2009-07-06 16:14:31.000000000 +0200 @@ -23,7 +23,7 @@ */ #define NR_UNIX98_PTY_DEFAULT 4096 /* Default maximum for Unix98 ptys */ #define NR_UNIX98_PTY_MAX (1 << MINORBITS) /* Absolute limit */ -#define NR_LDISCS 19 +#define NR_LDISCS 20 /* line disciplines */ #define N_TTY 0 @@ -47,6 +47,8 @@ #define N_SLCAN 17 /* Serial / USB serial CAN Adaptors */ #define N_PPS 18 /* Pulse per Second */ +#define N_AMSDELTA 19 /* codec control over modem, board specific */ + /* * This character is the same as _POSIX_VDISABLE: it cannot be used as * a c_cc[] character, but indicates that a particular special character

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[alsa-devel] ASoC: Correct a bug with "ADC Inversion Switch" in wm8974 codec.
by javier Martin 22 Jul '09

22 Jul '09

This corrects a bug with ADC Inversion Switch in wm8974 codec. Signed-off-by: Javier Martin <javier.martin(a)vista-silicon.com> --- diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index f26c8d0..38a0c44 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -162,7 +162,7 @@ SOC_SINGLE_TLV("PCM Volume", WM8974_DACVOL, 0, 255, 0, digital_tlv), SOC_SINGLE("High Pass Filter Switch", WM8974_ADC, 8, 1, 0), SOC_SINGLE("High Pass Cut Off", WM8974_ADC, 4, 7, 0), -SOC_SINGLE("ADC Inversion Switch", WM8974_COMP, 0, 1, 0), +SOC_SINGLE("ADC Inversion Switch", WM8974_ADC, 0, 1, 0), SOC_SINGLE_TLV("Capture Volume", WM8974_ADCVOL, 0, 255, 0, digital_tlv), -- Javier Martin Vista Silicon S.L. Universidad de Cantabria CDTUC - FASE C - Oficina S-345 Avda de los Castros s/n 39005- Santander. Cantabria. Spain +34 942 25 32 60 www.vista-silicon.com

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[alsa-devel] how to quirk max capture level?
by Alexey Fisher 21 Jul '09

21 Jul '09

hallo all, i have Logitech QuickCam E3500 ... with little issue. Setting capture volume to more then 94% will mute this device... no capture will be possible. This device has same issue with windows vista. So best way to handle it, is usbquirks.h , but i do not know any way to set max level. amixer report this device like: ========================================= Simple mixer control 'Mic',0 Capabilities: cvolume cvolume-joined cswitch cswitch-joined Capture channels: Mono Limits: Capture 0 - 3072 Mono: Capture 2950 [96%] [23.00dB] [on] ========================================== here is alsa-info.sh output: http://www.alsa-project.org/db/?f=ecd6ea8c7a37c7aa036c88026885f324ef2ab067 As you can see it report it's limits 0-3072 .. but real limit is 0-2943. Please if any one has some idea, meail me. Regards, Alexey

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[alsa-devel] Frame-based access (via mmap) and cross-device synchronisation
by Metathronius Galabant 21 Jul '09

21 Jul '09

Hi all, I could not find any examples, plus the alsa documentation and the pcm.c example on mmap-style access is rather slim. Consider the following setting: I have 2 seperate HW devices that are NOT clock sync'ed; their period is set to 10ms, and the total SW buffer is set to 20ms (= classic double buffering) or 40ms. Now I need to read from the 1st device and write to the 2nd device - and read from the 2nd device and write to the 1st device. Now, to synchronise - because their clocks do drift - I'd like to have the following semantics: At any given time x0, x1, ... xn I'd like to take some kind of "snapshot" of both devices, ask them "how many frame do you have?". This MUST NOT be restricted to the period, but it should be like device1 has 80 frames, device2 only has 78 frames. In short I would need pure frame-based access at any given time. I will then manually discard or duplicate frames to keep both streams happy. How do I achieve this kind of access method via alsa? Any help is welcome. Thanks! M.

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[alsa-devel] distinguish events based on input ports
by Lieven Moors 21 Jul '09

21 Jul '09

Hi everybody, I've just started writing my first little program with alsa, and I had a question regarding the use of input(read) ports. What I want to do is distinguish events based on the input port they came through. Since the destination of the event is mostly set to 'broadcast', I cannot extract this information from the event itself. I would like to know if there is another way to do this, or if it is simply impossible. Thanks for your help, Greetings, Lieven

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[alsa-devel] Testers wanted: New SB X-Fi driver
by Matthias Diehl 21 Jul '09

21 Jul '09

> At Tue, 30 Jun 2009 16:10:45 +0200, > Matthias Diehl wrote: > > > > hi, > > > > i guess you dont speak german, do you? well, i'll try it in > english ;-) > > > > i tried to install your newest x-fi "hack" in ubuntu 9.04 64bit. > it's an x-fi extreme music. i downloaded this snapshot: >> > ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver-uns… > > Use alsa-driver-snapshot.tar.gz instead of the unstable one. > > > while the installation process there are some warnings. after > replacing the demon.conf and reboot there was no sound. > > > > then i noticed the sentence "Note that the ctxfi driver might > conflict with the HD-audio driver" in the description. i think my > problem is the hdmi soundchip from my graphics adapter (sapphire > radeon hd 4870) which was automatically installed at the > ubuntu-installation > ("hda intel sound"). > > You are likely looking at irrelevant things. > > There are several variants of X-Fi boards. Many of them are equipped > with emu20k1 or emu20k2 chip. Some of these models can behave as > HD-audio compatible, too. These are supported by snd-ctxfi driver. > > In addition, there is X-Fi models that are without emu20k1/emu20k2 > chips. Some of them (CA0110-IBG) behave only as a HD-audio device, > and some others are with CA0106 chip. The former is supported by > snd-hda-intel driver and the latter by snd-ca0106 driver. > > So, they cannot be deactivated. The only question is whether you have > emu20k1/emu20k2 chip on your board. And this can be checked when you > load snd-ctxfi driver. If the driver is properly loaded and the > card instance is created (see /proc/asound/cards), then it's good. > Otherwise, your board is no such one. You need other drivers. > > Please ask further questions on alsa-devel ML (you can add me to Cc, > though) so that other people with your board can advice better. > > > HTH, > > Takashi hi there, i now tried to install the stable version, but i have still no sound. here's what lspci | grep audio says: 01:00.1 Audio device: ATI Technologies Inc HD48x0 audio 05:00.0 Multimedia audio controller: Creative Labs SB X-Fi lsmod | grep snd says: snd_hda_intel 557492 1 snd_pcm_oss 52352 0 snd_mixer_oss 24960 2 snd_pcm_oss snd_pcm 99336 2 snd_hda_intel,snd_pcm_oss snd_seq_dummy 11524 0 snd_seq_oss 41984 0 snd_seq_midi 15744 0 snd_rawmidi 33920 1 snd_seq_midi snd_seq_midi_event 16512 2 snd_seq_oss,snd_seq_midi snd_seq 66272 6 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_seq_midi_event snd_timer 34064 2 snd_pcm,snd_seq snd_seq_device 16276 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_rawmidi,snd_seq snd 78792 9 snd_hda_intel,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_seq_oss,snd_rawmidi,snd_seq,snd_timer,snd_seq_device soundcore 16800 2 snd snd_page_alloc 18704 2 snd_hda_intel,snd_pcm cat /proc/asound/cards says: 0 [HDMI ]: HDA-Intel - HDA ATI HDMI HDA ATI HDMI at 0xf7dfc000 irq 17 i hope, someone can help me ... greetings from germany, matthias

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[alsa-devel] Two Codecs in the machine driver
by Olaya, Margarita 21 Jul '09

21 Jul '09

Hi all, I have a hardware platform which has two codecs and I want to create a SoC machine driver to connect them to the processor, but it's not clear to me how to do that. Do you have any suggestions? Is there any driver that I can see as an example? Thanks & Regards, Margarita

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Re: [alsa-devel] Testers wanted: New SB X-Fi driver
by Takashi Iwai 21 Jul '09

21 Jul '09

At Thu, 16 Jul 2009 12:19:15 +0200, Frashman wrote: > > Takashi Iwai schrieb: > > At Wed, 15 Jul 2009 12:55:23 +0200, > > Frashman wrote: > > > >> Hi Takashi, > >> > >> first of all, thank you for developing this driver. > >> I tested it today. It works, so far. > >> > >> But two annoying things. As you mentioned, I got problems, if the > >> onboard sound modules are loaded. Then i get no sound out auf my xfi. > >> > > > > It's likely the collision of PCI device ID. > > Some X-Fi models support both HD-audio and emu20kx-native modes. The > > modes are switched by some magical sequence. > > > > Anyway, please run alsa-info.sh with --no-upload option, and attach > > the generated file. > > > I let the kernel auto probe all sound modules and generated then the > alsa-info.txt > It's attached. The PCI ID looks different from what snd-hda-intel driver supports. When compiled together with snd-ctxfi, snd-hda-intel only probes PCI ID 1002:0009. Are you sure that it's hda-intel that interferes the xfi? Any relevant kernel messages or asound entry when this happens? Takashi

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Re: [alsa-devel] [PATCH 2/2] Add possibility to control the GPIO_STATUS shift
by Mark Brown 21 Jul '09

21 Jul '09

On Fri, Jul 17, 2009 at 04:45:21PM +0200, Marek Vasut wrote: > From 1ea286d088993e05b5b374a08ee6f933242d3561 Mon Sep 17 00:00:00 2001 > From: Marek Vasut <marek.vasut(a)gmail.com> > Date: Thu, 4 Jun 2009 21:56:59 +0200 > Subject: [PATCH 2/2] Add possibility to control the GPIO_STATUS shift > > This patch allows tweaking the behaviour of GPIO_STATUS register > shift quirk that's in wm97xx-core. The problem with GPIO_STATUS > register being shifted by one doesn't appear on all hardware it > seems and causes problems with accelerated touchscreen drivers on > Palm hardware. Therefore an accelerated touchscreen driver can select > if the shift is/isn't happening on the hardware. Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> > > Signed-off-by: Marek Vasut <marek.vasut(a)gmail.com> > --- > drivers/input/touchscreen/mainstone-wm97xx.c | 3 +++ > drivers/input/touchscreen/wm97xx-core.c | 6 ++++-- > include/linux/wm97xx.h | 7 +++++++ > 3 files changed, 14 insertions(+), 2 deletions(-) > > diff --git a/drivers/input/touchscreen/mainstone-wm97xx.c b/drivers/input/touchscreen/mainstone-wm97xx.c > index de02f87..e000ab3 100644 > --- a/drivers/input/touchscreen/mainstone-wm97xx.c > +++ b/drivers/input/touchscreen/mainstone-wm97xx.c > @@ -198,6 +198,9 @@ static int wm97xx_acc_startup(struct wm97xx *wm) > if (machine_is_palmt5() || machine_is_palmtx() || machine_is_palmld()) { > pen_int = 1; > irq = 27; > + /* There is some obscure mutant of WM9712 interbred with WM9713 > + * used on Palm HW */ > + wm->variant = WM97xx_WM1613; > } else if (machine_is_mainstone() && pen_int) > irq = 4; > > diff --git a/drivers/input/touchscreen/wm97xx-core.c b/drivers/input/touchscreen/wm97xx-core.c > index 2957d48..252eb11 100644 > --- a/drivers/input/touchscreen/wm97xx-core.c > +++ b/drivers/input/touchscreen/wm97xx-core.c > @@ -204,7 +204,7 @@ void wm97xx_set_gpio(struct wm97xx *wm, u32 gpio, > else > reg &= ~gpio; > > - if (wm->id == WM9712_ID2) > + if (wm->id == WM9712_ID2 && wm->variant != WM97xx_WM1613) > wm97xx_reg_write(wm, AC97_GPIO_STATUS, reg << 1); > else > wm97xx_reg_write(wm, AC97_GPIO_STATUS, reg); > @@ -307,7 +307,7 @@ static void wm97xx_pen_irq_worker(struct work_struct *work) > WM97XX_GPIO_13); > } > > - if (wm->id == WM9712_ID2) > + if (wm->id == WM9712_ID2 && wm->variant != WM97xx_WM1613) > wm97xx_reg_write(wm, AC97_GPIO_STATUS, (status & > ~WM97XX_GPIO_13) << 1); > else > @@ -582,6 +582,8 @@ static int wm97xx_probe(struct device *dev) > > wm->id = wm97xx_reg_read(wm, AC97_VENDOR_ID2); > > + wm->variant = WM97xx_GENERIC; > + > dev_info(wm->dev, "detected a wm97%02x codec\n", wm->id & 0xff); > > switch (wm->id & 0xff) { > diff --git a/include/linux/wm97xx.h b/include/linux/wm97xx.h > index 6f69968..0c98781 100644 > --- a/include/linux/wm97xx.h > +++ b/include/linux/wm97xx.h > @@ -16,6 +16,12 @@ > #include <linux/platform_device.h> > > /* > + * WM97xx variants > + */ > +#define WM97xx_GENERIC 0x0000 > +#define WM97xx_WM1613 0x1613 > + > +/* > * WM97xx AC97 Touchscreen registers > */ > #define AC97_WM97XX_DIGITISER1 0x76 > @@ -283,6 +289,7 @@ struct wm97xx { > unsigned pen_is_down:1; /* Pen is down */ > unsigned aux_waiting:1; /* aux measurement waiting */ > unsigned pen_probably_down:1; /* used in polling mode */ > + u16 variant; /* WM97xx chip variant */ > u16 suspend_mode; /* PRP in suspend mode */ > }; > > -- > 1.6.3.3 >

2 1

[alsa-devel] [PATCH 0/2]ad1938 dai and board driver clean-up according to feedback
by Barry Song 20 Jul '09

20 Jul '09

Hi Mark, Update bf5xx-tdm dai(patch 1) and bf5xx-ad1938 machine(patch 2) drivers according to your feedback. Thanks Barry

3 4

[alsa-devel] [PATCH] ALSA: hda - Add quirk for Gateway T6834c laptop
by 宋浩 20 Jul '09

20 Jul '09

From: Hao Song <baritono.tux(a)gmail.com> Gateway T6834c laptops need EAPD always on while the default behavior for the STAC9205 reference board is to turn it off upon every HP plug. By using the special "eapd" model, which is first introduced for Gateway T1616 laptops for this same reason, this peculiarity can be properly handled. Signed-off-by: Hao Song <baritono.tux(a)gmail.com> --- sound/pci/hda/patch_sigmatel.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 41b5b3a..d9efe5c 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2379,6 +2379,7 @@ static struct snd_pci_quirk stac9205_cfg_tbl[] = { "Dell Vostro 1500", STAC_9205_DELL_M42), /* Gateway */ SND_PCI_QUIRK(0x107b, 0x0565, "Gateway T1616", STAC_9205_EAPD), + SND_PCI_QUIRK(0x107b, 0x0560, "Gateway T6834c", STAC_9205_EAPD), {} /* terminator */ }; -- 1.6.0.4 -- 宋浩 Song, Hao Institute of Theoretical Computer Science and Communications Room 117, Ho Sin Hang Engineering Building Chinese University of Hong Kong, Hong Kong, P.R.China E-mail: baritono.tux(a)gmail.com Office Tel: (+8600852)-31634255

2 1

Re: [alsa-devel] New Audio Driver Design
by Daniel Mack 20 Jul '09

20 Jul '09

On Thu, Jul 16, 2009 at 09:13:28AM +0900, jassi brar wrote: > On Thu, Jul 16, 2009 at 5:22 AM, Daniel Mack <daniel(a)caiaq.de> wrote: > > Are these decoder instruction something you need to do at runtime? Or > > would it suffice to provide them once when the driver is probed? > > The decoder insn, params atleast, depend upon the mp3 file i want the dsp to > decode and play. > So, let us assume, i have to load decoder insns and params into the dsp for > every mp3 file playback. Then the alsa control API seems appropriate. Unless there is any other interface for that specific purpose I'm not aware of. > Also, by what interface name(like I2S/PCM/AC97) shud my driver get encoded > mp3 data from my mp3-player and then decode that mp3 data by dsp and play > the output pcm stream? > I am afraid i must bypass/hack ALSA stack to directly get mp3 data from > user-space. Am i right? Have a look at include/sound/pcm.h - there are more formats defined than just linear PCM, and I think SNDRV_PCM_FMTBIT_MPEG could be something you can use. But I'm not sure and never used something like that before, so maybe anyone else can elaborate more? Daniel

1 0

[alsa-devel] CS42L52 codec
by Anna White 20 Jul '09

20 Jul '09

Hello, I am working with CS42L52 codec under Linux. I have CS42L52 driver code, but I need ALSA configuration files. Does somebody have ALSA configuration files ? Or recomendation how to write them ? Thanks for responces. P.S. It may be related to thread: http://mailman.alsa-project.org/pipermail/alsa-devel/2008-February/006130.h…

2 1

Re: [alsa-devel] [PATCH] hda: add bounds checking for the codec command fields v2
by Wu Fengguang 20 Jul '09

20 Jul '09

On Fri, Jul 17, 2009 at 06:46:01PM +0800, Takashi Iwai wrote: > At Fri, 17 Jul 2009 18:41:05 +0800, > Guo, Chaohong wrote: > > > > Although it does address this issue, I am not comfortable with this fixing. > > It seems more like a workaround than fix. > > No, it's rather for catching a bug. This is definitely neither > "workaround" nor "fix". Yes, I wrote this mainly for catching unknown bugs. > > Moreover, if long format node > > ID is used in the future, the code will cause little trouble for maintaining. > > Well, the current code doesn't support the long id (as the restriction > of HD-audio controller side), so we'd need major changes in anyway. > Thus this check is no big issue for maintenance, at least to me :) Does the HDA spec define cmd format that accept long form NID? If so, can you point me to the specific location please? Thanks. > > what I want is to fix it during parsing connection list, and verify the node > > id is valid there . > > Heh, this was already fixed :) AFAIK, snd_hda_get_connections() won't return NIDs bigger than 0x7f(short) or 0x7ffff(long). Thanks, Fengguang

2 1

[alsa-devel] [RFC] tty (or char) bus?
by Janusz Krzysztofik 19 Jul '09

19 Jul '09

Hi, In my attempt to add support for contols to a voice modem codec sound device driver, I found that in order to talk to the modem, it would be convenient if I can get access to a tty device from inside the kernel in a way similiar to that available form userspace. AFAICS, even if tty lowlevel write() could be used unmodified, a convenient way of reading characters from a tty is missing and should be implemented in a line discipline. Please correct me if I am wrong. OTOH, I found that some kind of abstraction layer for acccessing devices over a tty could be convenient. Instead of allocating a new line discipline for each specific device, sometimes found on a specific board only, why not just create a new bus type? Implemented as a line discipline, activated (hot-plugged) from userspace with ldattach, that new bus adapter would give kernel level access to an arbitrary device hanging off a tty. As the bus can be assumed point-to-point, a single generic device could be registered automatically in order to trigger that bus registerd drivers' probes (vendor/model id queries, for example). If not enough, an ioctl and a ldattach replacement could be provided for setting up a device id that would match a driver id (something like inputattach does for N_MOUSE). Once detected by a voice modem codec driver tty bus probe(), a codec subdevice could then be registered with a sound card. Please let me know if you like the idea. If not, please push me in a better direction. Many of the new code could be probably based on currently existing drivers and line discplines. As I'm not familiar enough with the linux kernel code, unlike most of you, so please give me some hints on what specific drivers should I look at to get examples best matching my idea. Thanks, Janusz

3 3

[alsa-devel] [PATCH] alsa-info.sh: include 1 line of dmesg context
by Wu Fengguang 19 Jul '09

19 Jul '09

So as to include possible ALSA messages without the common keywords, as well as be more confident on the completeness/coverage of the report. It also shows nice '--' separators between the dmesg ranges. Signed-off-by: Wu Fengguang <fengguang.wu(a)intel.com> --- utils/alsa-info.sh | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- alsa-driver.orig/utils/alsa-info.sh +++ alsa-driver/utils/alsa-info.sh @@ -247,7 +247,7 @@ withdmesg() { echo "!!ALSA/HDA dmesg" >> $FILE echo "!!------------------" >> $FILE echo "" >> $FILE - dmesg | grep -E 'ALSA|HDA|HDMI|sound|hda.codec|hda.intel' >> $FILE + dmesg | grep -C1 -E 'ALSA|HDA|HDMI|sound|hda.codec|hda.intel' >> $FILE echo "" >> $FILE echo "" >> $FILE }

2 1

[alsa-devel] [PATCH 6/6] sound/isa: convert nested spin_lock_irqsave to spin_lock
by Julia Lawall 19 Jul '09

19 Jul '09

From: Julia Lawall <julia(a)diku.dk> If spin_lock_irqsave is called twice in a row with the same second argument, the interrupt state at the point of the second call overwrites the value saved by the first call. Indeed, the second call does not need to save the interrupt state, so it is changed to a simple spin_lock. The semantic match that finds this problem is as follows: (http://www.emn.fr/x-info/coccinelle/) // <smpl> @@ expression lock1,lock2; expression flags; @@ *spin_lock_irqsave(lock1,flags) ... when != flags *spin_lock_irqsave(lock2,flags) // </smpl> Signed-off-by: Julia Lawall <julia(a)diku.dk> --- sound/isa/gus/gus_pcm.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/isa/gus/gus_pcm.c b/sound/isa/gus/gus_pcm.c index edb11ee..2dcf45b 100644 --- a/sound/isa/gus/gus_pcm.c +++ b/sound/isa/gus/gus_pcm.c @@ -795,13 +795,13 @@ static int snd_gf1_pcm_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ if (!(pcmp->flags & SNDRV_GF1_PCM_PFLG_ACTIVE)) continue; /* load real volume - better precision */ - spin_lock_irqsave(&gus->reg_lock, flags); + spin_lock(&gus->reg_lock); snd_gf1_select_voice(gus, pvoice->number); snd_gf1_ctrl_stop(gus, SNDRV_GF1_VB_VOLUME_CONTROL); vol = pvoice == pcmp->pvoices[0] ? gus->gf1.pcm_volume_level_left : gus->gf1.pcm_volume_level_right; snd_gf1_write16(gus, SNDRV_GF1_VW_VOLUME, vol); pcmp->final_volume = 1; - spin_unlock_irqrestore(&gus->reg_lock, flags); + spin_unlock(&gus->reg_lock); } spin_unlock_irqrestore(&gus->voice_alloc, flags); return change;

2 1

[alsa-devel] [PATCH 0/3] new support for ad1938 codec
by Barry Song 18 Jul '09

18 Jul '09

Hi Liam&Mark, The following series add support for ad1938 codec *PATCH 1: ad1938 codec driver *PATCH 2: blackfin I2S(in TDM mode) DAI driver Even though TDM mode can be as part of I2S DAI, but there are so much difference in configuration and data flow, it's very ugly to integrate I2S and TDM into a module. So we use the DAI type, but split normal I2S and TDM into individual files. *PATCH 3: board driver to connect bf5xx with ad1938 -- Thanks, Barry

2 7

[alsa-devel] snd_pcm_avail_update returns bogus values with the OSS PCM plugin
by Sindre Aamås 17 Jul '09

17 Jul '09

Hi. For the attached test case snd_pcm_avail_update returns values higher than the buffer size when run with the OSS PCM plugin. Output of alsa-info.sh is at http://www.alsa-project.org/db/?f=77a873e55d5d6ca50d67a035a9d6b7e10cca179d . --- alsa_oss_pcm_avail_bug.c --- /* * Public domain test case to illustrate a bug in the ALSA OSS PCM plugin * where snd_pcm_avail_update returns bogus values. */ #include <alsa/asoundlib.h> #include <stdio.h> #include <stdlib.h> #include <string.h> static short buf[803 * 2]; static snd_pcm_t *pcm_handle = NULL; void close_pcm(void) { if (pcm_handle) snd_pcm_close(pcm_handle); pcm_handle = NULL; } static int set_pcm_parameters(void) { snd_pcm_hw_params_t *hwparams; snd_pcm_hw_params_alloca(&hwparams); if (snd_pcm_hw_params_any(pcm_handle, hwparams) < 0) { fprintf(stderr, "snd_pcm_hw_params_any failed.\n"); return -1; } if (snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED) < 0) { fprintf(stderr, "snd_pcm_hw_params_set_access failed.\n"); return -1; } if (snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16) < 0) { fprintf(stderr, "snd_pcm_hw_params_set_format failed.\n"); return -1; } { unsigned rate = 48000; if (snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &rate, 0) < 0) { fprintf(stderr, "snd_pcm_hw_params_set_rate_near failed.\n"); return -1; } } if (snd_pcm_hw_params_set_channels(pcm_handle, hwparams, 2) < 0) { fprintf(stderr, "snd_pcm_hw_params_set_channels failed.\n"); return -1; } { unsigned ulatency = 100000; if (snd_pcm_hw_params_set_buffer_time_near(pcm_handle, hwparams, &ulatency, 0) < 0) { fprintf(stderr, "snd_pcm_hw_params_set_buffer_time_near failed.\n"); return -1; } } if (snd_pcm_hw_params(pcm_handle, hwparams) < 0) { fprintf(stderr, "snd_pcm_hw_params failed.\n"); return -1; } { snd_pcm_uframes_t bSize = 0; if (snd_pcm_hw_params_get_buffer_size(hwparams, &bSize) < 0) { fprintf(stderr, "snd_pcm_hw_params_get_buffer_size failed\n"); return -1; } printf("buffer_size: %u\n", (unsigned int) bSize); } return 0; } static void test_loop(int n) { while (n--) { snd_pcm_sframes_t sframes = snd_pcm_avail_update(pcm_handle); printf("snd_pcm_avail: %d\n", (int) sframes); printf("writing %d frames\n", (int) sizeof(buf) / 4); sframes = snd_pcm_writei(pcm_handle, buf, sizeof(buf) / 4); if (sframes != sizeof(buf) / 4) { if (sframes < 0) snd_pcm_prepare(pcm_handle); printf("snd_pcm_writei returned %d\n", (int) sframes); } } } int main(int argc, char **argv) { atexit(close_pcm); if (argc < 2) { printf("usage: %s <ALSA pcm device>\n", argv[0]); return 0; } if (snd_pcm_open(&pcm_handle, argv[1], SND_PCM_STREAM_PLAYBACK, 0) < 0) { fprintf(stderr, "Error opening PCM device %s\n", argv[1]); pcm_handle = NULL; return -1; } if (set_pcm_parameters() < 0) return -1; memset(buf, 0, sizeof(buf)); test_loop(500); return 0; } -- Sindre Aamås

2 2

[alsa-devel] ALSA porting to h/w having kernel version 2.6.18.5
by Rakesh Kumar 17 Jul '09

17 Jul '09

Hi I am new to ALSA and going to start porting of ALSA driver on out H/W box which is having 2.6.18.5 kernel (ARM), soc scenario. ALSA version supported on 2.6.18.5 is 1.0.12. I have following questions: 1. Shall I go ahead with ALSA 1.0.12 (which is already part of my kernel)? 2. or shall I download latest ALSA version 1.0.20? I can go with either with 1.0.12 or 1.0.20 versions, I want to know what is general trend followed in this scenario. I tried building my kernel with latest ALSA (as a part of kernel) but facing some compilation errors. Thanks Regards, Rakesh _________________________________________________________________ Stay updated! Add Facebook, LinkedIn, MySpace & Hi5 friends to your Windows Live network instantly. Add Now! http://profile.live.com/webactivities/?mkt=en-in

2 1

Re: [alsa-devel] [PATCH 2/2] Add possibility to control the GPIO_STATUS shift
by Mark Brown 17 Jul '09

17 Jul '09

On Fri, Jul 17, 2009 at 10:01:16AM +0200, Marek Vasut wrote: > This patch allows tweaking the behaviour of GPIO_STATUS register > shift quirk that's in wm97xx-core. The problem with GPIO_STATUS > register being shifted by one doesn't appear on all hardware it > seems and causes problems with accelerated touchscreen drivers on > Palm hardware. Therefore an accelerated touchscreen driver can select > if the shift is/isn't happening on the hardware. Again, this isn't an ALSA patch. I've done some checking internally and I'm very suspicous that what's going on here is actually the masking of some other bug elsewhere in the system. Which CPU are the systems you're observing this on using? There is at least one special case in the current code for GPIO register readbacks in the PXA AC97 driver code and there have been a number of interoperabilty issues in that area...

2 6

[alsa-devel] [PATCH 1/2] Add Palm support to Mainstone accelerated touch
by Marek Vasut 17 Jul '09

17 Jul '09

From f32fa9ef6476be0cf8106bc57a98724c48d4baa1 Mon Sep 17 00:00:00 2001 From: Marek Vasut <marek.vasut(a)gmail.com> Date: Fri, 17 Jul 2009 09:54:13 +0200 Subject: [PATCH 1/2] Add Palm support to Mainstone accelerated touch This patch refactors the Mainstone accelerated touch code a little and adds support for interrupt driven touchscreen on Palm LifeDrive, TX and Tungsten T5. Signed-off-by: Marek Vasut <marek.vasut(a)gmail.com> --- drivers/input/touchscreen/Kconfig | 4 +- drivers/input/touchscreen/mainstone-wm97xx.c | 52 +++++++++++++++++-------- 2 files changed, 37 insertions(+), 19 deletions(-) diff --git a/drivers/input/touchscreen/Kconfig b/drivers/input/touchscreen/Kconfig index 72e2712..0fdae1a 100644 --- a/drivers/input/touchscreen/Kconfig +++ b/drivers/input/touchscreen/Kconfig @@ -366,11 +366,11 @@ config TOUCHSCREEN_WM97XX_ATMEL be called atmel-wm97xx. config TOUCHSCREEN_WM97XX_MAINSTONE - tristate "WM97xx Mainstone accelerated touch" + tristate "WM97xx Mainstone/Palm accelerated touch" depends on TOUCHSCREEN_WM97XX && ARCH_PXA help Say Y here for support for streaming mode with WM97xx touchscreens - on Mainstone systems. + on Mainstone, Palm Tungsten T5, TX and LifeDrive systems. If unsure, say N. diff --git a/drivers/input/touchscreen/mainstone-wm97xx.c b/drivers/input/touchscreen/mainstone-wm97xx.c index 4cc047a..60bda9a 100644 --- a/drivers/input/touchscreen/mainstone-wm97xx.c +++ b/drivers/input/touchscreen/mainstone-wm97xx.c @@ -31,9 +31,13 @@ #include <linux/interrupt.h> #include <linux/wm97xx.h> #include <linux/io.h> +#include <linux/gpio.h> + #include <mach/regs-ac97.h> -#define VERSION "0.13" +#include <asm/mach-types.h> + +#define VERSION "0.14" struct continuous { u16 id; /* codec id */ @@ -62,6 +66,7 @@ static const struct continuous cinfo[] = { /* continuous speed index */ static int sp_idx; static u16 last, tries; +static int irq; /* * Pen sampling frequency (Hz) in continuous mode. @@ -171,7 +176,7 @@ up: static int wm97xx_acc_startup(struct wm97xx *wm) { - int idx = 0; + int idx = 0, ret = 0; /* check we have a codec */ if (wm->ac97 == NULL) @@ -191,18 +196,37 @@ static int wm97xx_acc_startup(struct wm97xx *wm) "mainstone accelerated touchscreen driver, %d samples/sec\n", cinfo[sp_idx].speed); + /* IRQ driven touchscreen is used on Palm hardware */ + if (machine_is_palmt5() || machine_is_palmtx() || machine_is_palmld()) { + pen_int = 1; + irq = 27; + } else if (machine_is_mainstone() && pen_int) + irq = 4; + + if (irq) { + ret = gpio_request(irq, "Touchscreen IRQ"); + if (ret) + goto out; + + ret = gpio_direction_input(irq); + if (ret) { + gpio_free(irq); + goto out; + } + + wm->pen_irq = gpio_to_irq(irq); + set_irq_type(wm->pen_irq, IRQ_TYPE_EDGE_BOTH); + } else /* pen irq not supported */ + pen_int = 0; + /* codec specific irq config */ if (pen_int) { switch (wm->id) { case WM9705_ID2: - wm->pen_irq = IRQ_GPIO(4); - set_irq_type(IRQ_GPIO(4), IRQ_TYPE_EDGE_BOTH); break; case WM9712_ID2: case WM9713_ID2: - /* enable pen down interrupt */ /* use PEN_DOWN GPIO 13 to assert IRQ on GPIO line 2 */ - wm->pen_irq = MAINSTONE_AC97_IRQ; wm97xx_config_gpio(wm, WM97XX_GPIO_13, WM97XX_GPIO_IN, WM97XX_GPIO_POL_HIGH, WM97XX_GPIO_STICKY, @@ -220,23 +244,17 @@ static int wm97xx_acc_startup(struct wm97xx *wm) } } - return 0; +out: + return ret; } static void wm97xx_acc_shutdown(struct wm97xx *wm) { /* codec specific deconfig */ if (pen_int) { - switch (wm->id & 0xffff) { - case WM9705_ID2: - wm->pen_irq = 0; - break; - case WM9712_ID2: - case WM9713_ID2: - /* disable interrupt */ - wm->pen_irq = 0; - break; - } + if (irq) + gpio_free(irq); + wm->pen_irq = 0; } } -- 1.6.3.3

2 3

Re: [alsa-devel] [PATCH] hda: add bounds checking for the codec command fields v2
by Takashi Iwai 17 Jul '09

17 Jul '09

At Fri, 17 Jul 2009 12:46:01 +0200, I wrote: > > At Fri, 17 Jul 2009 18:41:05 +0800, > Guo, Chaohong wrote: > > > > Although it does address this issue, I am not comfortable with this fixing. > > It seems more like a workaround than fix. > > No, it's rather for catching a bug. This is definitely neither > "workaround" nor "fix". BTW, I fixed the patch again as below. Takashi --- From 82e1b804b03defe46fb69ffd2c8b19e6649bcb0d Mon Sep 17 00:00:00 2001 From: Takashi Iwai <tiwai(a)suse.de> Date: Fri, 17 Jul 2009 12:47:34 +0200 Subject: [PATCH] ALSA: hda - Fix the previous sanity check in make_codec_cmd() The newly added sanity-check for a codec verb can be better written with logical ORs. Also, the parameter can be more than 8bit. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index d9d3262..35f0f22 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -150,8 +150,8 @@ make_codec_cmd(struct hda_codec *codec, hda_nid_t nid, int direct, { u32 val; - if ((codec->addr & ~0xf) | (direct & ~1) | (nid & ~0x7f) | - (verb & ~0xfff) | (parm & ~0xff)) { + if ((codec->addr & ~0xf) || (direct & ~1) || (nid & ~0x7f) || + (verb & ~0xfff) || (parm & ~0xffff)) { printk(KERN_ERR "hda-codec: out of range cmd %x:%x:%x:%x:%x\n", codec->addr, direct, nid, verb, parm); return ~0; -- 1.6.3.2

2 1

[PATCH] oss - Add missing initialization of fragments
by Takashi Iwai 17 Jul '09

17 Jul '09

The periods calculation was missing for initializing OSS fragments. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- oss/pcm_oss.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/oss/pcm_oss.c b/oss/pcm_oss.c index 8a2a672..d43b44c 100644 --- a/oss/pcm_oss.c +++ b/oss/pcm_oss.c @@ -181,6 +181,7 @@ static int oss_hw_params(snd_pcm_ioplug_t *io, fprintf(stderr, "*** OSS: invalid period size %d\n", (int)io->period_size); return -EINVAL; } + oss->periods = io->buffer_size / io->period_size; _retry: tmp = oss->period_shift | (oss->periods << 16); -- 1.6.3.2

1 0

[alsa-devel] [PATCH] hda: add bounds checking for the codec command fields
by Wu Fengguang 17 Jul '09

17 Jul '09

A recent bug involves passing auto detected >0x7f NID to codec command, creating an invalid codec addr field, and finally lead to cmd timeout and fall back into single command mode. Jaroslav fixed that bug in alc880_parse_auto_config(). It would be safer to further check the bounds of all cmd fields. Cc: Jaroslav Kysela <perex(a)perex.cz> Signed-off-by: Wu Fengguang <fengguang.wu(a)intel.com> --- diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 462e2ce..7d09650 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -150,6 +150,16 @@ make_codec_cmd(struct hda_codec *codec, hda_nid_t nid, int direct, { u32 val; + if ((direct & ~1) || (nid & ~0x7f) || + (verb & ~0xfff) || (parm & ~0xff)) { + printk(KERN_ERR "hda-codec: out of range cmd %x:%x:%x:%x:%x\n", + codec->addr, direct, nid, verb, parm); + direct &= 1; + nid &= 0x7f; + verb &= 0xfff; + parm &= 0xff; + } + val = (u32)(codec->addr & 0x0f) << 28; val |= (u32)direct << 27; val |= (u32)nid << 20;

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Re: [alsa-devel] [PATCH]HDA: Add support for new AMD HD audio devices
by Takashi Iwai 17 Jul '09

17 Jul '09

At Fri, 17 Jul 2009 11:32:32 +0800, Andiry Brienza wrote: > > Add support for new AMD HD audio devices. Use generic driver to detect HD audio > devices with Vendor ID AMD. > > Signed-off-by: Andiry Xu <andiry.xu(a)amd.com> Thanks, applied now. Takashi > diff -ruN a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c > --- a/sound/pci/hda/hda_intel.c 2009-07-17 10:46:24.000000000 +0800 > +++ b/sound/pci/hda/hda_intel.c 2009-07-17 11:04:45.000000000 +0800 > @@ -2604,11 +2604,15 @@ > /* this entry seems still valid -- i.e. without emu20kx chip */ > { PCI_DEVICE(0x1102, 0x0009), .driver_data = AZX_DRIVER_GENERIC }, > #endif > - /* AMD Generic, PCI class code and Vendor ID for HD Audio */ > + /* AMD/ATI Generic, PCI class code and Vendor ID for HD Audio */ > { PCI_DEVICE(PCI_VENDOR_ID_ATI, PCI_ANY_ID), > .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8, > .class_mask = 0xffffff, > .driver_data = AZX_DRIVER_GENERIC }, > + { PCI_DEVICE(PCI_VENDOR_ID_AMD, PCI_ANY_ID), > + .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8, > + .class_mask = 0xffffff, > + .driver_data = AZX_DRIVER_GENERIC }, > { 0, } > }; > MODULE_DEVICE_TABLE(pci, azx_ids); > >

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[alsa-devel] Callbacks from opening or closing alsa streams
by Baek Chang 17 Jul '09

17 Jul '09

Hi, I was wondering if it was possible to get callbacks from opening and closing of alsa streams with the alsa sink name that was opened. So if I have an asound.conf configured like so: pcm.music { type hw card 0 } Is it possible to get a callback saying that a stream on the "music" sink was opened? Thanks Baek -- Baek Chang baeksan(a)ccrma.stanford.edu

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[alsa-devel] ANYONE have spdif working with the ADI AD1986A chipset? (Asus M2NPV-VM mainboard)
by VDR User 17 Jul '09

17 Jul '09

This mainboard isn't that new but from what I've read spdif worked in alsa some time in the past. I get the IEC958/IEC958 D options in alsamixer but can not get any sound. Analog works fine. If anyone has spdif working on this mainboard then PLEASE post your exact setup & settings! The only other option would be to buy new hardware and that's not an option right now. Thanks for any/all help! -Derek

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[alsa-devel] While recording from mono codec twice as much data is received
by Louis Lee 16 Jul '09

16 Jul '09

Hi everyone, I am able to do playback from 8 kHz samples to 48 kHz samples properly. However, when recording using the exact same setup for the data interface I am not getting favorable results. For instance, if I record for 5 seconds I get output in a file for 10 seconds of audio (arecord or sox). This results in my recording being stretched. I change the DAI to stereo capture and record two channels and the recorded output comes back correctly (5 seconds recorded is 5 seconds). I'm using a wolfsom WM8974. I'm developing using ASoC framework (can I call it that?) reusing the WM8974 codec from the Linux GIT repository. The development is using a i.MX27. Data interface is a SSI bus in DSP mode. I believe my PCM code is correct since it works for playback. I'm stuck now. From my understanding the clocks are setup the same way for both playback and capture. Has anyone ran into this problem before? The WM8974 is a mono codec so I should not have to set the DAI to stereo in order to capture data properly. Any help is appreciated. Best regards,Louis _________________________________________________________________ Attention all humans. We are your photos. Free us. http://go.microsoft.com/?linkid=9666046

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[alsa-devel] [PATCH] ALSA: hda - Add CX20582 and OLPC XO-1.5 support
by Daniel Drake 16 Jul '09

16 Jul '09

This adds support for the Conexant CX20582 codec, based on code from http://www.linuxant.com/alsa-driver/alsa-driver-linuxant-1.0.19ppch12-1.noa… This is the codec to be shipped in the OLPC XO-1.5, so this patch also includes an XO-specific profile. Resultant configuration: http://dev.laptop.org/~dsd/20090713/codec0.txt http://dev.laptop.org/~dsd/20090713/codec0.svg As the Linuxant code is structured differently to the other codecs, I was unable to cleanly reimplement everything in the generic and Dell profiles as more info is needed (e.g. codec graphs). I simplified those profiles so that hopefully it will not break anyone's audio. If it does, it may be worth returning -ENODEV from patch_cx5066 on non-OLPC systems, and then fixing snd_hda_codec_configure() to fall back on the generic parser, at least until support for other systems is figured out. --- sound/pci/hda/patch_conexant.c | 479 ++++++++++++++++++++++++++++++++++++++++ 1 files changed, 479 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 4fcbe21..77b4f28 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -108,6 +108,8 @@ struct conexant_spec { struct hda_input_mux private_imux; hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS]; + unsigned int dell_automute; + unsigned int port_d_mode; }; static int conexant_playback_pcm_open(struct hda_pcm_stream *hinfo, @@ -1908,6 +1910,480 @@ static int patch_cxt5051(struct hda_codec *codec) return 0; } +/* Conexant 5066 specific */ + +static hda_nid_t cxt5066_dac_nids[1] = { 0x10 }; +static hda_nid_t cxt5066_adc_nids[3] = { 0x14, 0x15, 0x16 }; +static hda_nid_t cxt5066_capsrc_nids[1] = { 0x17 }; +#define CXT5066_SPDIF_OUT 0x21 + +static struct hda_channel_mode cxt5066_modes[1] = { + { 2, NULL }, +}; + +static void cxt5066_update_speaker(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + unsigned int pinctl; + + snd_printdd("CXT5066: update speaker, hp_present=%d\n", + spec->hp_present); + + /* Port A (HP) */ + pinctl = ((spec->hp_present & 1) && spec->cur_eapd) ? PIN_HP : 0; + snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + pinctl); + + /* Port D (HP/LO) */ + pinctl = ((spec->hp_present & 2) && spec->cur_eapd) + ? spec->port_d_mode : 0; + snd_hda_codec_write(codec, 0x1c, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + pinctl); + + /* CLASS_D AMP */ + pinctl = (!spec->hp_present && spec->cur_eapd) ? PIN_OUT : 0; + snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + pinctl); + + if (spec->dell_automute) { + /* DELL AIO Port Rule: PortA > PortD > IntSpk */ + pinctl = (!(spec->hp_present & 1) && spec->cur_eapd) + ? PIN_OUT : 0; + snd_hda_codec_write(codec, 0x1c, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl); + } +} + +/* turn on/off EAPD (+ mute HP) as a master switch */ +static int cxt5066_hp_master_sw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + + if (!cxt_eapd_put(kcontrol, ucontrol)) + return 0; + + cxt5066_update_speaker(codec); + return 1; +} + +/* toggle input of built-in and mic jack appropriately */ +static void cxt5066_automic(struct hda_codec *codec) +{ + static struct hda_verb ext_mic_present[] = { + /* enable external mic, port B */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + + /* switch to external mic input */ + {0x17, AC_VERB_SET_CONNECT_SEL, 0}, + + /* disable internal mic, port C */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {} + }; + static struct hda_verb ext_mic_absent[] = { + /* enable internal mic, port C */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + + /* switch to internal mic input */ + {0x17, AC_VERB_SET_CONNECT_SEL, 1}, + + /* disable external mic, port B */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {} + }; + unsigned int present; + + present = snd_hda_codec_read(codec, 0x1a, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + if (present) { + snd_printdd("CXT5066: external microphone detected\n"); + snd_hda_sequence_write(codec, ext_mic_present); + } else { + snd_printdd("CXT5066: external microphone absent\n"); + snd_hda_sequence_write(codec, ext_mic_absent); + } +} + +/* mute internal speaker if HP is plugged */ +static void cxt5066_hp_automute(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + unsigned int portA, portD; + + /* Port A */ + portA = snd_hda_codec_read(codec, 0x19, 0, AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; + + /* Port D */ + portD = (snd_hda_codec_read(codec, 0x1c, 0, AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE) << 1; + + spec->hp_present = !!(portA | portD); + snd_printdd("CXT5066: hp automute portA=%x portD=%x present=%d\n", + portA, portD, spec->hp_present); + cxt5066_update_speaker(codec); +} + +/* unsolicited event for jack sensing */ +static void cxt5066_unsol_event(struct hda_codec *codec, unsigned int res) +{ + snd_printdd("CXT5066: unsol event %x (%x)\n", res, res >> 26); + switch (res >> 26) { + case CONEXANT_HP_EVENT: + cxt5066_hp_automute(codec); + break; + case CONEXANT_MIC_EVENT: + cxt5066_automic(codec); + break; + } +} + +static const struct hda_input_mux cxt5066_analog_mic_boost = { + .num_items = 5, + .items = { + { "0dB", 0 }, + { "10dB", 1 }, + { "20dB", 2 }, + { "30dB", 3 }, + { "40dB", 4 }, + }, +}; + +static int cxt5066_mic_boost_mux_enum_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + return snd_hda_input_mux_info(&cxt5066_analog_mic_boost, uinfo); +} + +static int cxt5066_mic_boost_mux_enum_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + int val; + + val = snd_hda_codec_read(codec, 0x17, 0, + AC_VERB_GET_AMP_GAIN_MUTE, AC_AMP_GET_OUTPUT); + + ucontrol->value.enumerated.item[0] = val & AC_AMP_GAIN; + return 0; +} + +static int cxt5066_mic_boost_mux_enum_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + const struct hda_input_mux *imux = &cxt5066_analog_mic_boost; + unsigned int idx; + + if (!imux->num_items) + return 0; + idx = ucontrol->value.enumerated.item[0]; + if (idx >= imux->num_items) + idx = imux->num_items - 1; + + snd_hda_codec_write_cache(codec, 0x17, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | AC_AMP_SET_OUTPUT | + imux->items[idx].index); + + return 1; +} + +static struct hda_input_mux cxt5066_capture_source = { + .num_items = 4, + .items = { + { "Mic B", 0 }, + { "Mic C", 1 }, + { "Mic E", 2 }, + { "Mic F", 3 }, + }, +}; + +static struct hda_bind_ctls cxt5066_bind_capture_vol_others = { + .ops = &snd_hda_bind_vol, + .values = { + HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_INPUT), + HDA_COMPOSE_AMP_VAL(0x14, 3, 2, HDA_INPUT), + 0 + }, +}; + +static struct hda_bind_ctls cxt5066_bind_capture_sw_others = { + .ops = &snd_hda_bind_sw, + .values = { + HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_INPUT), + HDA_COMPOSE_AMP_VAL(0x14, 3, 2, HDA_INPUT), + 0 + }, +}; + +static struct snd_kcontrol_new cxt5066_mixer_master[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT), + {} +}; + +static struct snd_kcontrol_new cxt5066_mixer_master_olpc[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ | + SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, + .info = snd_hda_mixer_amp_volume_info, + .get = snd_hda_mixer_amp_volume_get, + .put = snd_hda_mixer_amp_volume_put, + .tlv = { .c = snd_hda_mixer_amp_tlv }, + /* offset by 28 volume steps to limit minimum gain to -46dB */ + .private_value = + HDA_COMPOSE_AMP_VAL_OFS(0x10, 3, 0, HDA_OUTPUT, 28), + }, + {} +}; + +static struct snd_kcontrol_new cxt5066_mixers[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = cxt_eapd_info, + .get = cxt_eapd_get, + .put = cxt5066_hp_master_sw_put, + .private_value = 0x1d, + }, + + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Mic Boost Capture Enum", + .info = cxt5066_mic_boost_mux_enum_info, + .get = cxt5066_mic_boost_mux_enum_get, + .put = cxt5066_mic_boost_mux_enum_put, + }, + + HDA_BIND_VOL("Capture Volume", &cxt5066_bind_capture_vol_others), + HDA_BIND_SW("Capture Switch", &cxt5066_bind_capture_sw_others), + {} +}; + +static struct hda_verb cxt5066_init_verbs[] = { + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port B */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port C */ + {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port F */ + {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port E */ + + /* Speakers */ + {0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1f, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ + + /* HP, Amp */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ + + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x1c, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ + + /* DAC1 */ + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + + /* Node 14 connections: 0x17 0x18 0x23 0x24 0x27 */ + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x50}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2) | 0x50}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + + /* no digital microphone support yet */ + {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* Audio input selector */ + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x3}, + + /* SPDIF route: PCM */ + {0x20, AC_VERB_SET_CONNECT_SEL, 0x0}, + {0x22, AC_VERB_SET_CONNECT_SEL, 0x0}, + + {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x22, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + + /* EAPD */ + {0x1d, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ + + /* not handling these yet */ + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, + {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, + {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, + {0x1d, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, + {0x1e, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, + {0x20, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, + {0x22, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, + { } /* end */ +}; + +static struct hda_verb cxt5066_init_verbs_olpc[] = { + /* Port A: headphones */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ + + /* Port B: external microphone */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + + /* Port C: internal microphone */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + + /* Port D: unused */ + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* Port E: unused, but has primary EAPD */ + {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x1d, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ + + /* Port F: unused */ + {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* Port G: internal speakers */ + {0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1f, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ + + /* DAC1 */ + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + + /* DAC2: unused */ + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x50}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + + /* Disable digital microphone port */ + {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* Audio input selectors */ + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x3}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + + /* Disable SPDIF */ + {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x22, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* enable unsolicited events for Port A and B */ + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, + {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, + { } /* end */ +}; + +static struct hda_verb cxt5066_init_verbs_portd_lo[] = { + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + { } /* end */ +}; + +/* initialize jack-sensing, too */ +static int cxt5066_init(struct hda_codec *codec) +{ + snd_printdd("CXT5066: init\n"); + conexant_init(codec); + if (codec->patch_ops.unsol_event) { + cxt5066_hp_automute(codec); + cxt5066_automic(codec); + } + return 0; +} + +enum { + CXT5066_LAPTOP, /* Laptops w/ EAPD support */ + CXT5066_DELL_LAPTOP, /* Dell Laptop */ + CXT5066_OLPC_XO_1_5, /* OLPC XO 1.5 */ + CXT5066_MODELS +}; + +static const char *cxt5066_models[CXT5066_MODELS] = { + [CXT5066_LAPTOP] = "laptop", + [CXT5066_DELL_LAPTOP] = "dell-laptop", + [CXT5066_OLPC_XO_1_5] = "olpc-xo-1_5", +}; + +static struct snd_pci_quirk cxt5066_cfg_tbl[] = { + SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board", + CXT5066_LAPTOP), + SND_PCI_QUIRK(0x1028, 0x02f5, "Dell", + CXT5066_DELL_LAPTOP), + {} +}; + +static int patch_cxt5066(struct hda_codec *codec) +{ + struct conexant_spec *spec; + int board_config; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + codec->spec = spec; + + codec->patch_ops = conexant_patch_ops; + codec->patch_ops.init = cxt5066_init; + + spec->dell_automute = 0; + spec->multiout.max_channels = 2; + spec->multiout.num_dacs = ARRAY_SIZE(cxt5066_dac_nids); + spec->multiout.dac_nids = cxt5066_dac_nids; + spec->multiout.dig_out_nid = CXT5066_SPDIF_OUT; + spec->num_adc_nids = 1; + spec->adc_nids = cxt5066_adc_nids; + spec->capsrc_nids = cxt5066_capsrc_nids; + spec->input_mux = &cxt5066_capture_source; + + spec->port_d_mode = PIN_HP; + + spec->num_init_verbs = 1; + spec->init_verbs[0] = cxt5066_init_verbs; + spec->num_channel_mode = ARRAY_SIZE(cxt5066_modes); + spec->channel_mode = cxt5066_modes; + spec->cur_adc = 0; + spec->cur_adc_idx = 0; + + board_config = snd_hda_check_board_config(codec, CXT5066_MODELS, + cxt5066_models, cxt5066_cfg_tbl); + switch (board_config) { + default: + case CXT5066_LAPTOP: + spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; + spec->mixers[spec->num_mixers++] = cxt5066_mixers; + break; + case CXT5066_DELL_LAPTOP: + spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; + spec->mixers[spec->num_mixers++] = cxt5066_mixers; + + spec->port_d_mode = PIN_OUT; + spec->init_verbs[spec->num_init_verbs] = cxt5066_init_verbs_portd_lo; + spec->num_init_verbs++; + spec->dell_automute = 1; + break; + case CXT5066_OLPC_XO_1_5: + codec->patch_ops.unsol_event = cxt5066_unsol_event; + spec->init_verbs[0] = cxt5066_init_verbs_olpc; + spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc; + spec->mixers[spec->num_mixers++] = cxt5066_mixers; + spec->port_d_mode = 0; + + /* no S/PDIF out */ + spec->multiout.dig_out_nid = 0; + + /* input source automatically selected */ + spec->input_mux = NULL; + break; + } + + return 0; +} /* */ @@ -1919,12 +2395,15 @@ static struct hda_codec_preset snd_hda_preset_conexant[] = { .patch = patch_cxt5047 }, { .id = 0x14f15051, .name = "CX20561 (Hermosa)", .patch = patch_cxt5051 }, + { .id = 0x14f15066, .name = "CX20582 (Pebble)", + .patch = patch_cxt5066 }, {} /* terminator */ }; MODULE_ALIAS("snd-hda-codec-id:14f15045"); MODULE_ALIAS("snd-hda-codec-id:14f15047"); MODULE_ALIAS("snd-hda-codec-id:14f15051"); +MODULE_ALIAS("snd-hda-codec-id:14f15066"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Conexant HD-audio codec"); -- 1.6.2.5

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Re: [alsa-devel] [PATCH 01/06] Fix compilation warning for sound/soc/codecs/wm8400.c
by Mark Brown 16 Jul '09

16 Jul '09

On Thu, Jul 16, 2009 at 06:40:01PM +0530, Subrata Modak wrote: > How about the following brutal shutdown ? No, this is the patch you originally submitted. You are missing the point here - your patches are all just papering over the warning and won't help at all if there is an actual problem. > 'factors' will get initialized here, as, 'freq_out' will probably never have > a '0' value. So, 'fll_factors()' will actually overwrite values in 'factors' > even after the initial brutal assignment: > "struct fll_factors factors = {}" This is pretty much the point of the warning when it's valid - it's trying to catch situations where there is a code path where the variable is used without being initialised. If you just blindly initialise the variable as you are doing then the warning goes away but any problematic code remains so the situation is worse. In this case I suspect that whatever compiler you are using (none of those I've tried with seem to be doing this) isn't able to figure out that if we skip initialising the variable then we use exactly the same condition to return from the function before we try to use the variable. For something like this the warning can normally be worked around by changing the control flow so that the compiler is able to figure out that things are safe.

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Re: [alsa-devel] No sound with nVidia Corporation MCP79 High Definition Audio (rev b1) (Macbook Pro 5, 5)
by Takashi Iwai 16 Jul '09

16 Jul '09

At Wed, 8 Jul 2009 18:11:29 +0200, Andreas Nüßlein wrote: > > On Wednesday 08 July 2009 16:49:34 you wrote: > > At Tue, 07 Jul 2009 12:53:48 +0200, > > > > I wrote: > > > At Tue, 7 Jul 2009 11:49:57 +0200, > > > > > > Andreas Nüßlein wrote: > > > > On Tuesday 07 July 2009 08:00:19 you wrote: > > > > > At Mon, 6 Jul 2009 22:14:56 +0200, > > > > > > > > > > Andreas Nüßlein wrote: > > > > > > On Monday 06 July 2009 21:26:18 you wrote: > > > > > > > At Mon, 06 Jul 2009 11:16:35 -0600, > > > > > > > > > > > > > > Sean Burke wrote: > > > > > > > > Scríobh Takashi Iwai: > > > > > > > > > At Mon, 6 Jul 2009 17:53:46 +0200, > > > > > > > > > > > > > > > > > > Andreas Nüßlein wrote: > > > > > > > > >>> The missing pin configuration initialization was already > > > > > > > > >>> fixed by the driver overriding it after checking PCI SSID > > > > > > > > >>> (which is different from the codec SSID). So, this should > > > > > > > > >>> be no problem. > > > > > > > > >>> > > > > > > > > >>> However, the reason why the analog output doesn't work > > > > > > > > >>> might be different from that. There might be something > > > > > > > > >>> else missing, but I don't know. > > > > > > > > >>> > > > > > > > > >>> > > > > > > > > >>> Takashi > > > > > > > > >> > > > > > > > > >> oh =( > > > > > > > > >> > > > > > > > > >> > > > > > > > > >> hmm.. anything i can do? would it help if i tried changing > > > > > > > > >> values randomly with hda-analyzer.py? > > > > > > > > > > > > > > > > > > Well, did the driver without my change work more or less with > > > > > > > > > the analog audio, or have you never gotten the analog output? > > > > > > > > > You can use the generic parser (i.e. the state without cirrus > > > > > > > > > patch) by passing model=generic option to snd-hda-intel. > > > > > > > > > > > > > > > > For my part, nothing worked with the generic driver. I can't > > > > > > > > confirm digital out, but I can confirm that the kfree error is > > > > > > > > gone. What options are open for figuring out what remains? > > > > > > > > > > > > > > Easy things to test are GPIO bits. Run hda-verb like > > > > > > > > > > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_MASK 0x0f > > > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x0f > > > > > > > or > > > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIR 0x0f > > > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x0f > > > > > > > > > > > > > > etc. CS4206 seems to have 4 GPIO lines, and each bit (0-3) > > > > > > > corresponds to each GPIO. In many case, GPIO0 or GPIO1 > > > > > > > corresponds to the amplifier (EAPD) bit. > > > > > > > Define the GPIO direction of each GPIO bit by SET_GPIO_DIR, and > > > > > > > turn on/off the GPIO bits by SET_GPIO_DATA. Running > > > > > > > hda-verb /dev/snd/hwC0D0 0x01 GET_GPIO_DATA 0 > > > > > > > will show the current GPIO data bits. Or you can check it in > > > > > > > codec#* proc file. > > > > > > > > > > > > > > > > > > > > > Takashi > > > > > > > > > > > > w000000000000000000000000000000000t! =) > > > > > > > > > > > > takashi, thank you _so_ much! > > > > > > > > > > > > after running all 4 of those: > > > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_MASK 0x0f > > > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x0f > > > > > > > or > > > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIR 0x0f > > > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x0f > > > > > > > > > > > > i suddendly had sound!! :D :D :D :D :D :D :D > > > > > > only via speakers though - there is no sound via headphones right > > > > > > now. > > > > > > > > > > > > mixer channels: > > > > > > - Master (with Mutebutton), PCM and Front (also with Mute) all work > > > > > > =) - i don't know what surround would do (or it's extra switch) - > > > > > > headphones-volumes and mute button don't affect the speakers, which > > > > > > is good =) > > > > > > > > > > > > > > > > > > is there a way to reset what i did with hda-verb, so that i can > > > > > > figure out which combination it was exactly? > > > > > > > > > > You can just change the value 0x0f to a different value. > > > > > At least, you can try commands like > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x01 > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x02 > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x04 > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x08 > > > > > and check the speaker output at each time. > > > > > Also, check the GPIO direction, > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIR 0x01 > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x01 > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIR 0x02 > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x02 > > > > > ... > > > > > > > > > > Regarding the headphone: is the speaker muted when you plug in the > > > > > headphone? If not, it's likely an issue of the jack detection. If > > > > > the speaker is muted but no headphone output, it's a missing > > > > > initialization (or wrong GPIO setup). > > > > > > > > > > > > > > > Takashi > > > > > > > > so here are my findings so far: > > > > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x08 > > > > > > > > > > > > is sufficient to turn on the right functionality for the speakers; > > > > meaning: - 'front speaker'-channel actually controls the two front > > > > speakers within the macbookpro - even controlling left and right > > > > channel seperately works =) - 'surround' mute-switch and > > > > volume-control (why are those not within on thing btw?) also work and > > > > control a third speaker in the mac, which really seems responsibly for > > > > surround :) > > > > > > > > headphones do not yet work, however i made a diff of > > > > "/proc/asound/card0/codec#0" with and without something plugged in: > > > > > > OK, so actually the driver works as expected, but the hardware doesn't > > > behave as expected. For muting the speaker, something else (like > > > toggling that GPIO) is needed. > > > > > > The mystery about the silent HP output still remains, though. > > > > One another thing we can try is to use only one PCM stream instead of > > multi streams. For example, try the patch below. This will enable > > only the pin for the headphone, thus only one DAC is used. > > > > > > Takashi > > > > --- > > diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c > > index 57251d7..d7f52b1 100644 > > --- a/sound/pci/hda/patch_cirrus.c > > +++ b/sound/pci/hda/patch_cirrus.c > > @@ -1086,13 +1086,17 @@ struct cs_pincfg { > > > > static struct cs_pincfg mbp55_pincfgs[] = { > > { 0x09, 0x012b4030 }, > > - { 0x0a, 0x90100121 }, > > - { 0x0b, 0x90100120 }, > > + // { 0x0a, 0x90100121 }, > > + { 0x0a, 0x400000f0 }, > > + // { 0x0b, 0x90100120 }, > > + { 0x0b, 0x400000f0 }, > > { 0x0c, 0x400000f0 }, > > - { 0x0d, 0x90a00110 }, > > + // { 0x0d, 0x90a00110 }, > > + { 0x0d, 0x400000f0 }, > > { 0x0e, 0x400000f0 }, > > { 0x0f, 0x400000f0 }, > > - { 0x10, 0x014be040 }, > > + // { 0x10, 0x014be040 }, > > + { 0x10, 0x400000f0 }, > > { 0x12, 0x400000f0 }, > > { 0x15, 0x400000f0 }, > > {} /* terminator */ > > > =( no - that doesn't work either > > (there is obivously no sound via built-in speakers either anymore) Hmm, then I have really no clue. At best, you can try again GPIO pins and reverting COEF setup by my earlier patch, and try some more adjustments of widgets amp or pinctl via hda-verb or hda-analyzer... Takashi

2 3

[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 16 Jul '09

16 Jul '09

The following changes since commit 4b75e9476737dffef44a66818a4a481edd2512a1: Mark Brown (1): ASoC: Error out if we can't determine a suitable WM9081 sysclk are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Barry Song (1): ASoC: new ad1938 codec driver based on asoc Joonyoung Shim (3): ASoC: add SOC_DOUBLE_EXT_TLV control type ASoC: add SOC_DOUBLE_R_EXT_TLV control type ASoC: MAX9877: add MAX9877 amp driver Kevin Hilman (1): ASoC: davinci: don't use clock names include/sound/soc.h | 22 ++ sound/soc/codecs/Kconfig | 9 + sound/soc/codecs/Makefile | 8 + sound/soc/codecs/ad1938.c | 652 +++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ad1938.h | 100 ++++++ sound/soc/codecs/max9877.c | 270 +++++++++++++++ sound/soc/codecs/max9877.h | 37 ++ sound/soc/davinci/davinci-i2s.c | 2 +- sound/soc/davinci/davinci-mcasp.c | 2 +- 9 files changed, 1100 insertions(+), 2 deletions(-) create mode 100644 sound/soc/codecs/ad1938.c create mode 100644 sound/soc/codecs/ad1938.h create mode 100644 sound/soc/codecs/max9877.c create mode 100644 sound/soc/codecs/max9877.h

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[alsa-devel] [PATCH 0/4] alsa-info.sh updates
by Wu Fengguang 16 Jul '09

16 Jul '09

Hi Takashi, This patchset mainly disables the auto-upload of alsa info by default. Comments are welcome. Thanks, Fengguang --

4 11

[alsa-devel] New Audio Driver Design
by jassi brar 15 Jul '09

15 Jul '09

Hello, I am to write an ALSA driver for my SOC which has a dedicated DSP subsystem which can be loaded with decoder instructions and parameteres, and then we load it with encoded data. The dsp then decodes the data and sends it to the PCM controller. Could someone please refer me to some similar driver/codec which already does that? If not, how should i design the driver? :- Should I load the encoder inst/params using firmware API? Then, which API should I use to submit encoded data to the DSP? Any pointers please? Thanks.

2 1

[alsa-devel] [PATCH 3/3] ASoC: MAX9877: add MAX9877 amp driver
by Joonyoung Shim 15 Jul '09

15 Jul '09

The MAX9877 combines a high-efficiency Class D audio power amplifier with a stereo Class AB capacitor-less DirectDrive headphone amplifier. The max9877_add_controls() is called to register the MAX9877 specific controls on machine specific init() of the machine driver. The datasheet for the MAX9877 can find at the following url: http://datasheets.maxim-ic.com/en/ds/MAX9877.pdf Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- This patch is version 2. I updated from review of version 1. Change log: v2: - add SND_SOC_MAX9877 to SND_SOC_ALL_CODECS on Kconfig. - use SOC_DOUBLE_R_EXT_TLV instead of SOC_SINGLE_EXT_TLV for HP volume control. - modify max9877_get_reg() and max9877_set_reg for double reg controls. - add EXPORT_SYMBOL_GPL(max9877_add_controls). sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/max9877.c | 270 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/max9877.h | 37 ++++++ 4 files changed, 318 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/max9877.c create mode 100644 sound/soc/codecs/max9877.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index b674d3a..16a710f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -49,6 +49,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM9705 if SND_SOC_AC97_BUS select SND_SOC_WM9712 if SND_SOC_AC97_BUS select SND_SOC_WM9713 if SND_SOC_AC97_BUS + select SND_SOC_MAX9877 if I2C help Normally ASoC codec drivers are only built if a machine driver which uses them is also built since they are only usable with a machine @@ -188,3 +189,7 @@ config SND_SOC_WM9712 config SND_SOC_WM9713 tristate + +# Amp +config SND_SOC_MAX9877 + tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 301eb08..cf111c9 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -38,6 +38,9 @@ snd-soc-wm9705-objs := wm9705.o snd-soc-wm9712-objs := wm9712.o snd-soc-wm9713-objs := wm9713.o +# Amp +snd-soc-max9877-objs := max9877.o + obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o @@ -77,3 +80,6 @@ obj-$(CONFIG_SND_SOC_WM9081) += snd-soc-wm9081.o obj-$(CONFIG_SND_SOC_WM9705) += snd-soc-wm9705.o obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o + +# Amp +obj-$(CONFIG_SND_SOC_MAX9877) += snd-soc-max9877.o diff --git a/sound/soc/codecs/max9877.c b/sound/soc/codecs/max9877.c new file mode 100644 index 0000000..7df9a7c --- /dev/null +++ b/sound/soc/codecs/max9877.c @@ -0,0 +1,270 @@ +/* + * max9877.c -- amp driver for max9877 + * + * Copyright (C) 2009 Samsung Electronics Co.Ltd + * Author: Joonyoung Shim <jy0922.shim(a)samsung.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. + * + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/i2c.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include "max9877.h" + +static struct i2c_client *i2c; + +static u8 max9877_regs[5] = { 0x40, 0x00, 0x00, 0x00, 0x49 }; + +static void max9877_write_regs(void) +{ + if (i2c_master_send(i2c, max9877_regs, 5) != 5) + dev_err(&i2c->dev, "i2c write failed\n"); +} + +static int max9877_get_reg(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int reg = mc->reg; + int reg2 = mc->rreg; + int shift = mc->shift; + int mask = mc->max; + + ucontrol->value.integer.value[0] = (max9877_regs[reg] >> shift) & mask; + + if (reg2) + ucontrol->value.integer.value[1] = + (max9877_regs[reg2] >> shift) & mask; + + return 0; +} + +static int max9877_set_reg(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int reg = mc->reg; + int reg2 = mc->rreg; + int shift = mc->shift; + int mask = mc->max; + int change = 1; + int change2 = 1; + int ret = 0; + + if (((max9877_regs[reg] >> shift) & mask) == + ucontrol->value.integer.value[0]) + change = 0; + + if (reg2) + if (((max9877_regs[reg2] >> shift) & mask) == + ucontrol->value.integer.value[1]) + change2 = 0; + + if (change) { + max9877_regs[reg] &= ~(mask << shift); + max9877_regs[reg] |= ucontrol->value.integer.value[0] << shift; + ret = change; + } + + if (reg2 && change2) { + max9877_regs[reg2] &= ~(mask << shift); + max9877_regs[reg2] |= ucontrol->value.integer.value[1] << shift; + ret = change2; + } + + if (ret) + max9877_write_regs(); + + return ret; +} + +static int max9877_get_out_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u8 value = max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OUTMODE_MASK; + + if (value) + value -= 1; + + ucontrol->value.integer.value[0] = value; + return 0; +} + +static int max9877_set_out_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u8 value = ucontrol->value.integer.value[0]; + + if (value) + value += 1; + + if ((max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OUTMODE_MASK) == value) + return 0; + + max9877_regs[MAX9877_OUTPUT_MODE] |= value; + max9877_write_regs(); + return 1; +} + +static int max9877_get_osc_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u8 value = (max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OSC_MASK); + + value = value >> MAX9877_OSC_OFFSET; + + ucontrol->value.integer.value[0] = value; + return 0; +} + +static int max9877_set_osc_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u8 value = ucontrol->value.integer.value[0]; + + value = value << MAX9877_OSC_OFFSET; + if ((max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OSC_MASK) == value) + return 0; + + max9877_regs[MAX9877_OUTPUT_MODE] |= value; + max9877_write_regs(); + return 1; +} + +static const unsigned int max9877_pgain_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 1, TLV_DB_SCALE_ITEM(0, 900, 0), + 2, 2, TLV_DB_SCALE_ITEM(2000, 0, 0), +}; + +static const unsigned int max9877_output_tlv[] = { + TLV_DB_RANGE_HEAD(4), + 0, 7, TLV_DB_SCALE_ITEM(-7900, 400, 1), + 8, 15, TLV_DB_SCALE_ITEM(-4700, 300, 0), + 16, 23, TLV_DB_SCALE_ITEM(-2300, 200, 0), + 24, 31, TLV_DB_SCALE_ITEM(-700, 100, 0), +}; + +static const char *max9877_out_mode[] = { + "INA -> SPK", + "INA -> HP", + "INA -> SPK and HP", + "INB -> SPK", + "INB -> HP", + "INB -> SPK and HP", + "INA + INB -> SPK", + "INA + INB -> HP", + "INA + INB -> SPK and HP", +}; + +static const char *max9877_osc_mode[] = { + "1176KHz", + "1100KHz", + "700KHz", +}; + +static const struct soc_enum max9877_enum[] = { + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(max9877_out_mode), max9877_out_mode), + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(max9877_osc_mode), max9877_osc_mode), +}; + +static const struct snd_kcontrol_new max9877_controls[] = { + SOC_SINGLE_EXT_TLV("MAX9877 PGAINA Playback Volume", + MAX9877_INPUT_MODE, 0, 2, 0, + max9877_get_reg, max9877_set_reg, max9877_pgain_tlv), + SOC_SINGLE_EXT_TLV("MAX9877 PGAINB Playback Volume", + MAX9877_INPUT_MODE, 2, 2, 0, + max9877_get_reg, max9877_set_reg, max9877_pgain_tlv), + SOC_SINGLE_EXT_TLV("MAX9877 Amp Speaker Playback Volume", + MAX9877_SPK_VOLUME, 0, 31, 0, + max9877_get_reg, max9877_set_reg, max9877_output_tlv), + SOC_DOUBLE_R_EXT_TLV("MAX9877 Amp HP Playback Volume", + MAX9877_HPL_VOLUME, MAX9877_HPR_VOLUME, 0, 31, 0, + max9877_get_reg, max9877_set_reg, max9877_output_tlv), + SOC_SINGLE_EXT("MAX9877 INB Stereo Switch", + MAX9877_INPUT_MODE, 4, 1, 1, + max9877_get_reg, max9877_set_reg), + SOC_SINGLE_EXT("MAX9877 INA Stereo Switch", + MAX9877_INPUT_MODE, 5, 1, 1, + max9877_get_reg, max9877_set_reg), + SOC_SINGLE_EXT("MAX9877 Zero-crossing detection Switch", + MAX9877_INPUT_MODE, 6, 1, 0, + max9877_get_reg, max9877_set_reg), + SOC_SINGLE_EXT("MAX9877 Bypass Mode Switch", + MAX9877_OUTPUT_MODE, 6, 1, 0, + max9877_get_reg, max9877_set_reg), + SOC_SINGLE_EXT("MAX9877 Shutdown Mode Switch", + MAX9877_OUTPUT_MODE, 7, 1, 1, + max9877_get_reg, max9877_set_reg), + SOC_ENUM_EXT("MAX9877 Output Mode", max9877_enum[0], + max9877_get_out_mode, max9877_set_out_mode), + SOC_ENUM_EXT("MAX9877 Oscillator Mode", max9877_enum[1], + max9877_get_osc_mode, max9877_set_osc_mode), +}; + +/* This function is called from ASoC machine driver */ +int max9877_add_controls(struct snd_soc_codec *codec) +{ + return snd_soc_add_controls(codec, max9877_controls, + ARRAY_SIZE(max9877_controls)); +} +EXPORT_SYMBOL_GPL(max9877_add_controls); + +static int __devinit max9877_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + i2c = client; + + max9877_write_regs(); + + return 0; +} + +static __devexit int max9877_i2c_remove(struct i2c_client *client) +{ + i2c = NULL; + + return 0; +} + +static const struct i2c_device_id max9877_i2c_id[] = { + { "max9877", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, max9877_i2c_id); + +static struct i2c_driver max9877_i2c_driver = { + .driver = { + .name = "max9877", + .owner = THIS_MODULE, + }, + .probe = max9877_i2c_probe, + .remove = __devexit_p(max9877_i2c_remove), + .id_table = max9877_i2c_id, +}; + +static int __init max9877_init(void) +{ + return i2c_add_driver(&max9877_i2c_driver); +} +module_init(max9877_init); + +static void __exit max9877_exit(void) +{ + i2c_del_driver(&max9877_i2c_driver); +} +module_exit(max9877_exit); + +MODULE_DESCRIPTION("ASoC MAX9877 amp driver"); +MODULE_AUTHOR("Joonyoung Shim <jy0922.shim(a)samsung.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/max9877.h b/sound/soc/codecs/max9877.h new file mode 100644 index 0000000..6da7229 --- /dev/null +++ b/sound/soc/codecs/max9877.h @@ -0,0 +1,37 @@ +/* + * max9877.h -- amp driver for max9877 + * + * Copyright (C) 2009 Samsung Electronics Co.Ltd + * Author: Joonyoung Shim <jy0922.shim(a)samsung.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. + * + */ + +#ifndef _MAX9877_H +#define _MAX9877_H + +#define MAX9877_INPUT_MODE 0x00 +#define MAX9877_SPK_VOLUME 0x01 +#define MAX9877_HPL_VOLUME 0x02 +#define MAX9877_HPR_VOLUME 0x03 +#define MAX9877_OUTPUT_MODE 0x04 + +/* MAX9877_INPUT_MODE */ +#define MAX9877_INB (1 << 4) +#define MAX9877_INA (1 << 5) +#define MAX9877_ZCD (1 << 6) + +/* MAX9877_OUTPUT_MODE */ +#define MAX9877_OUTMODE_MASK (15 << 0) +#define MAX9877_OSC_MASK (3 << 4) +#define MAX9877_OSC_OFFSET 4 +#define MAX9877_BYPASS (1 << 6) +#define MAX9877_SHDN (1 << 7) + +extern int max9877_add_controls(struct snd_soc_codec *codec); + +#endif -- 1.6.0.4

2 1

[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 15 Jul '09

15 Jul '09

The following changes since commit ba3b64b976f91067710ef1a87b9b445406910e8f: Mark Brown (1): Merge branch 'for-2.6.31' into for-2.6.32 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Cliff Cai (2): ASoC: Blackfin AC97: fix resume handling ASoC: Blackfin I2S: fix resume handling Mark Brown (2): ASoC: Fix sample rate lookup in WM8993 ASoC: Error out if we can't determine a suitable WM9081 sysclk sound/soc/blackfin/bf5xx-ac97.c | 10 +++------- sound/soc/blackfin/bf5xx-i2s.c | 8 ++------ sound/soc/codecs/wm8993.c | 3 ++- sound/soc/codecs/wm9081.c | 8 ++++++++ 4 files changed, 15 insertions(+), 14 deletions(-)

2 1

[alsa-devel] [PATCH 2/3] ASoC: add SOC_DOUBLE_R_EXT_TLV control type
by Joonyoung Shim 15 Jul '09

15 Jul '09

This is a macro for double controls with special callback function and TLV. The SOC_DOUBLE_R_EXT_TLV needs two registers and one shift for double controls. Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- include/sound/soc.h | 11 +++++++++++ 1 files changed, 11 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 795a6b4..756fb59 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -146,6 +146,17 @@ .private_value = (unsigned long)&(struct soc_mixer_control) \ {.reg = xreg, .shift = shift_left, .rshift = shift_right, \ .max = xmax, .invert = xinvert} } +#define SOC_DOUBLE_R_EXT_TLV(xname, reg_left, reg_right, xshift, xmax, xinvert,\ + xhandler_get, xhandler_put, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw_2r, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = reg_left, .rreg = reg_right, .shift = xshift, \ + .max = xmax, .invert = xinvert} } #define SOC_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_bool_ext, \ -- 1.6.0.4

1 0

[alsa-devel] [PATCH 1/3] ASoC: add SOC_DOUBLE_EXT_TLV control type
by Joonyoung Shim 15 Jul '09

15 Jul '09

This is a macro for double controls with special callback function and TLV. The SOC_DOUBLE_EXT_TLV needs one register and two shifts for double controls. Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- include/sound/soc.h | 11 +++++++++++ 1 files changed, 11 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 94fcc65..795a6b4 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -135,6 +135,17 @@ .info = snd_soc_info_volsw, \ .get = xhandler_get, .put = xhandler_put, \ .private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) } +#define SOC_DOUBLE_EXT_TLV(xname, xreg, shift_left, shift_right, xmax, xinvert,\ + xhandler_get, xhandler_put, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .shift = shift_left, .rshift = shift_right, \ + .max = xmax, .invert = xinvert} } #define SOC_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_bool_ext, \ -- 1.6.0.4

1 0

[alsa-devel] [PATCH] ASoC: MAX9877: add MAX9877 amp driver
by Joonyoung Shim 15 Jul '09

15 Jul '09

The MAX9877 combines a high-efficiency Class D audio power amplifier with a stereo Class AB capacitor-less DirectDrive headphone amplifier. The max9877_add_controls() is called to register the MAX9877 specific controls on machine specific init() of the machine driver. The datasheet for the MAX9877 can find at the following url: http://datasheets.maxim-ic.com/en/ds/MAX9877.pdf Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- Hi, I added amp driver in codec directory of ASoC. This is because of mail about amp driver the past. http://www.spinics.net/lists/alsa-devel/msg24876.html Please review, thank you. sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/max9877.c | 246 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/max9877.h | 37 +++++++ 4 files changed, 293 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/max9877.c create mode 100644 sound/soc/codecs/max9877.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index b674d3a..8930557 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -188,3 +188,7 @@ config SND_SOC_WM9712 config SND_SOC_WM9713 tristate + +# Amp +config SND_SOC_MAX9877 + tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 301eb08..cf111c9 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -38,6 +38,9 @@ snd-soc-wm9705-objs := wm9705.o snd-soc-wm9712-objs := wm9712.o snd-soc-wm9713-objs := wm9713.o +# Amp +snd-soc-max9877-objs := max9877.o + obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o @@ -77,3 +80,6 @@ obj-$(CONFIG_SND_SOC_WM9081) += snd-soc-wm9081.o obj-$(CONFIG_SND_SOC_WM9705) += snd-soc-wm9705.o obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o + +# Amp +obj-$(CONFIG_SND_SOC_MAX9877) += snd-soc-max9877.o diff --git a/sound/soc/codecs/max9877.c b/sound/soc/codecs/max9877.c new file mode 100644 index 0000000..cb79fe8 --- /dev/null +++ b/sound/soc/codecs/max9877.c @@ -0,0 +1,246 @@ +/* + * max9877.c -- amp driver for max9877 + * + * Copyright (C) 2009 Samsung Electronics Co.Ltd + * Author: Joonyoung Shim <jy0922.shim(a)samsung.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. + * + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/i2c.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include "max9877.h" + +static struct i2c_client *i2c; + +static u8 max9877_regs[5] = { 0x40, 0x00, 0x00, 0x00, 0x49 }; + +static void max9877_write_regs(void) +{ + if (i2c_master_send(i2c, max9877_regs, 5) != 5) + dev_err(&i2c->dev, "i2c write failed\n"); +} + +static int max9877_get_reg(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int reg = mc->reg; + int shift = mc->shift; + int mask = mc->max; + + ucontrol->value.integer.value[0] = (max9877_regs[reg] >> shift) & mask; + + return 0; +} + +static int max9877_set_reg(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int reg = mc->reg; + int shift = mc->shift; + int mask = mc->max; + + if (((max9877_regs[reg] >> shift) & mask) == + ucontrol->value.integer.value[0]) + return 0; + + max9877_regs[reg] &= ~(mask << shift); + max9877_regs[reg] |= ucontrol->value.integer.value[0] << shift; + max9877_write_regs(); + return 1; +} + +static int max9877_get_out_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u8 value = max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OUTMODE_MASK; + + if (value) + value -= 1; + + ucontrol->value.integer.value[0] = value; + return 0; +} + +static int max9877_set_out_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u8 value = ucontrol->value.integer.value[0]; + + if (value) + value += 1; + + if ((max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OUTMODE_MASK) == value) + return 0; + + max9877_regs[MAX9877_OUTPUT_MODE] |= value; + max9877_write_regs(); + return 1; +} + +static int max9877_get_osc_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u8 value = (max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OSC_MASK); + + value = value >> MAX9877_OSC_OFFSET; + + ucontrol->value.integer.value[0] = value; + return 0; +} + +static int max9877_set_osc_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u8 value = ucontrol->value.integer.value[0]; + + value = value << MAX9877_OSC_OFFSET; + if ((max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OSC_MASK) == value) + return 0; + + max9877_regs[MAX9877_OUTPUT_MODE] |= value; + max9877_write_regs(); + return 1; +} + +static const unsigned int max9877_pgain_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 1, TLV_DB_SCALE_ITEM(0, 900, 0), + 2, 2, TLV_DB_SCALE_ITEM(2000, 0, 0), +}; + +static const unsigned int max9877_output_tlv[] = { + TLV_DB_RANGE_HEAD(4), + 0, 7, TLV_DB_SCALE_ITEM(-7900, 400, 1), + 8, 15, TLV_DB_SCALE_ITEM(-4700, 300, 0), + 16, 23, TLV_DB_SCALE_ITEM(-2300, 200, 0), + 24, 31, TLV_DB_SCALE_ITEM(-700, 100, 0), +}; + +static const char *max9877_out_mode[] = { + "INA -> SPK", + "INA -> HP", + "INA -> SPK and HP", + "INB -> SPK", + "INB -> HP", + "INB -> SPK and HP", + "INA + INB -> SPK", + "INA + INB -> HP", + "INA + INB -> SPK and HP", +}; + +static const char *max9877_osc_mode[] = { + "1176KHz", + "1100KHz", + "700KHz", +}; + +static const struct soc_enum max9877_enum[] = { + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(max9877_out_mode), max9877_out_mode), + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(max9877_osc_mode), max9877_osc_mode), +}; + +static const struct snd_kcontrol_new max9877_controls[] = { + SOC_SINGLE_EXT_TLV("MAX9877 PGAINA Playback Volume", + MAX9877_INPUT_MODE, 0, 2, 0, + max9877_get_reg, max9877_set_reg, max9877_pgain_tlv), + SOC_SINGLE_EXT_TLV("MAX9877 PGAINB Playback Volume", + MAX9877_INPUT_MODE, 2, 2, 0, + max9877_get_reg, max9877_set_reg, max9877_pgain_tlv), + SOC_SINGLE_EXT_TLV("MAX9877 Amp Speaker Playback Volume", + MAX9877_SPK_VOLUME, 0, 31, 0, + max9877_get_reg, max9877_set_reg, max9877_output_tlv), + SOC_SINGLE_EXT_TLV("MAX9877 Amp HP Left Playback Volume", + MAX9877_HPL_VOLUME, 0, 31, 0, + max9877_get_reg, max9877_set_reg, max9877_output_tlv), + SOC_SINGLE_EXT_TLV("MAX9877 Amp HP Right Playback Volume", + MAX9877_HPR_VOLUME, 0, 31, 0, + max9877_get_reg, max9877_set_reg, max9877_output_tlv), + SOC_SINGLE_EXT("MAX9877 INB Stereo Switch", + MAX9877_INPUT_MODE, 4, 1, 1, + max9877_get_reg, max9877_set_reg), + SOC_SINGLE_EXT("MAX9877 INA Stereo Switch", + MAX9877_INPUT_MODE, 5, 1, 1, + max9877_get_reg, max9877_set_reg), + SOC_SINGLE_EXT("MAX9877 Zero-crossing detection Switch", + MAX9877_INPUT_MODE, 6, 1, 0, + max9877_get_reg, max9877_set_reg), + SOC_SINGLE_EXT("MAX9877 Bypass Mode Switch", + MAX9877_OUTPUT_MODE, 6, 1, 0, + max9877_get_reg, max9877_set_reg), + SOC_SINGLE_EXT("MAX9877 Shutdown Mode Switch", + MAX9877_OUTPUT_MODE, 7, 1, 1, + max9877_get_reg, max9877_set_reg), + SOC_ENUM_EXT("MAX9877 Output Mode", max9877_enum[0], + max9877_get_out_mode, max9877_set_out_mode), + SOC_ENUM_EXT("MAX9877 Oscillator Mode", max9877_enum[1], + max9877_get_osc_mode, max9877_set_osc_mode), +}; + +/* This function is called from ASoC machine driver */ +int max9877_add_controls(struct snd_soc_codec *codec) +{ + return snd_soc_add_controls(codec, max9877_controls, + ARRAY_SIZE(max9877_controls)); +} + +static int __devinit max9877_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + i2c = client; + + max9877_write_regs(); + + return 0; +} + +static __devexit int max9877_i2c_remove(struct i2c_client *client) +{ + i2c = NULL; + + return 0; +} + +static const struct i2c_device_id max9877_i2c_id[] = { + { "max9877", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, max9877_i2c_id); + +static struct i2c_driver max9877_i2c_driver = { + .driver = { + .name = "max9877", + .owner = THIS_MODULE, + }, + .probe = max9877_i2c_probe, + .remove = __devexit_p(max9877_i2c_remove), + .id_table = max9877_i2c_id, +}; + +static int __init max9877_init(void) +{ + return i2c_add_driver(&max9877_i2c_driver); +} +module_init(max9877_init); + +static void __exit max9877_exit(void) +{ + i2c_del_driver(&max9877_i2c_driver); +} +module_exit(max9877_exit); + +MODULE_DESCRIPTION("ASoC MAX9877 amp driver"); +MODULE_AUTHOR("Joonyoung Shim <jy0922.shim(a)samsung.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/max9877.h b/sound/soc/codecs/max9877.h new file mode 100644 index 0000000..6da7229 --- /dev/null +++ b/sound/soc/codecs/max9877.h @@ -0,0 +1,37 @@ +/* + * max9877.h -- amp driver for max9877 + * + * Copyright (C) 2009 Samsung Electronics Co.Ltd + * Author: Joonyoung Shim <jy0922.shim(a)samsung.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. + * + */ + +#ifndef _MAX9877_H +#define _MAX9877_H + +#define MAX9877_INPUT_MODE 0x00 +#define MAX9877_SPK_VOLUME 0x01 +#define MAX9877_HPL_VOLUME 0x02 +#define MAX9877_HPR_VOLUME 0x03 +#define MAX9877_OUTPUT_MODE 0x04 + +/* MAX9877_INPUT_MODE */ +#define MAX9877_INB (1 << 4) +#define MAX9877_INA (1 << 5) +#define MAX9877_ZCD (1 << 6) + +/* MAX9877_OUTPUT_MODE */ +#define MAX9877_OUTMODE_MASK (15 << 0) +#define MAX9877_OSC_MASK (3 << 4) +#define MAX9877_OSC_OFFSET 4 +#define MAX9877_BYPASS (1 << 6) +#define MAX9877_SHDN (1 << 7) + +extern int max9877_add_controls(struct snd_soc_codec *codec); + +#endif -- 1.6.0.4

2 3

Re: [alsa-devel] [PATCH 01/06] Fix compilation warning for sound/soc/codecs/wm8400.c
by Mark Brown 15 Jul '09

15 Jul '09

On Wed, Jul 15, 2009 at 10:13:38AM +0530, Jaswinder Singh Rajput wrote: > Hello Subrata, > This kind of warnings born due to not handling the variables properly. > Please try to understand the root cause of this warning, just shutting > this warning is of no use. ...as I said last time you submitted rougly the same patch. Please also remember that sometimes these problems are false positives caused by issues with the compiler flow analysis (as appears to be the case here).

1 0

[alsa-devel] [PATCH 06/06] Fix compilation warning for sound/drivers/serial/u16550.c
by Subrata Modak 15 Jul '09

15 Jul '09

Following fix is inspired by David Howells fix few days back: http://lkml.org/lkml/2009/7/9/109, Signed-off-by: Subrata Modak<subrata(a)linux.vnet.ibm.com>, --- --- a/sound/drivers/serial-u16550.c 2009-05-20 10:48:23.000000000 +0530 +++ b/sound/drivers/serial-u16550.c 2009-07-15 06:46:48.000000000 +0530 @@ -902,7 +902,7 @@ static int __devinit snd_uart16550_rmidi static int __devinit snd_serial_probe(struct platform_device *devptr) { struct snd_card *card; - struct snd_uart16550 *uart; + struct snd_uart16550 *uninitialized_var(uart); int err; int dev = devptr->id; --- Regards-- Subrata

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[alsa-devel] [PATCH 01/06] Fix compilation warning for sound/soc/codecs/wm8400.c
by Subrata Modak 15 Jul '09

15 Jul '09

Following fix is inspired by David Howells fix few days back: http://lkml.org/lkml/2009/7/9/109, Signed-off-by: Subrata Modak<subrata(a)linux.vnet.ibm.com>, --- --- a/sound/soc/codecs/wm8400.c 2009-06-15 07:52:31.000000000 +0530 +++ b/sound/soc/codecs/wm8400.c 2009-07-15 05:35:00.000000000 +0530 @@ -1015,7 +1015,7 @@ static int wm8400_set_dai_pll(struct snd { struct snd_soc_codec *codec = codec_dai->codec; struct wm8400_priv *wm8400 = codec->private_data; - struct fll_factors factors; + struct fll_factors uninitialized_var(factors); int ret; u16 reg; --- Regards-- Subrata

1 0

[alsa-devel] Alsa buffer
by arif setiawan 14 Jul '09

14 Jul '09

Hi, I'm trying to get meaningful voice from ALSA buffer. I read tutorial on http://www.suse.de/~mana/alsa090_howto.html<http://www.suse.de/%7Emana/alsa090_howto.html>and realize that if alsa use 16 bit format value then I have to byte shift to store it in 8 bit unsigned char buffer. I see this in sine wave generator in tutorial. To confirm it, I try to save the output into file and open it in Matlab. I want to know if this is correct, because when I try saving alsa buffer on a from recording I still get noise not the voice. Thank you. Best regards, Arif This is my implementation : int main() { int nchannel = 1; int periodsize = 8192; int frame_size = nchannel * (16 / 8); int num_frames = periodsize / frame_size; unsigned char *data; int l2; short s; data = (unsigned char *)malloc(periodsize); // fill the buffer for(l2 = 0; l2 < num_frames; l2++) { s1 = (l2 % 128) * 100 - 5000; data[2*l2] = (unsigned char)s1; data[2*l2+1] = s1 >> 8; } // directly save FILE *fp1 = fopen("sineout1", "w"); for (int i = 0; i < periodsize; i++) { fprintf_s(fp1, "%d\n", data[i]); } fclose(fp1); // with bit shifting from char to short FILE *fp2 = fopen("sineout2", "w"); for (int i = 0; i < num_frames; i++) { short newval = data[2*i+1] << 8; newval += data[2*i]; fprintf_s(fp2, "%d\n", newval); } fclose(fp2); return 0; }

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[alsa-devel] [PATCH 1/2] ASoC: Blackfin AC97: fix resume handling
by Mike Frysinger 14 Jul '09

14 Jul '09

From: Cliff Cai <cliff.cai(a)analog.com> There is no need to manually start playback/capture ourselves as the PCM driver will handle things for us. Signed-off-by: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- sound/soc/blackfin/bf5xx-ac97.c | 10 +++------- 1 files changed, 3 insertions(+), 7 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index b1ed423..2758b90 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -277,28 +277,24 @@ static int bf5xx_ac97_resume(struct snd_soc_dai *dai) if (!dai->active) return 0; - ret = sport_set_multichannel(sport_handle, 16, 0x1F, 1); + ret = sport_set_multichannel(sport, 16, 0x1F, 1); if (ret) { pr_err("SPORT is busy!\n"); return -EBUSY; } - ret = sport_config_rx(sport_handle, IRFS, 0xF, 0, (16*16-1)); + ret = sport_config_rx(sport, IRFS, 0xF, 0, (16*16-1)); if (ret) { pr_err("SPORT is busy!\n"); return -EBUSY; } - ret = sport_config_tx(sport_handle, ITFS, 0xF, 0, (16*16-1)); + ret = sport_config_tx(sport, ITFS, 0xF, 0, (16*16-1)); if (ret) { pr_err("SPORT is busy!\n"); return -EBUSY; } - if (dai->capture.active) - sport_rx_start(sport); - if (dai->playback.active) - sport_tx_start(sport); return 0; } -- 1.6.3.3

2 2

[alsa-devel] Testers wanted: New SB X-Fi driver
by Duke Grable 14 Jul '09

14 Jul '09

Hello! I stumbled upon these X-Fi drivers recently and ended up giving them a try. So far they work flawlessly, except for microphone input... I'm wondering if this is either just unimplemented, or if it's just an issue of configuration? My sound card is an "X-Fi XtremeGamer" and my distro is the latest version of openSuSe (2.6.27.23-0.1-default ). Let me know if I'm missing anything! Thanks, Duke

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[alsa-devel] ASoC updates for 2.6.31
by Mark Brown 14 Jul '09

14 Jul '09

The following changes since commit 637a935aaba2f05e2178c9d1b714d7a2c36c8b44: Lars-Peter Clausen (1): ASoC: Fix wm8753 register cache size and initialization 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 Daniel Mack (1): ASoC: Fix NULL pointer dereference in __pxa2xx_pcm_hw_free sound/arm/pxa2xx-pcm-lib.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-)

1 1

[alsa-devel] Microphone bias
by Daniel Drake 13 Jul '09

13 Jul '09

Hi, I'm working on CX20582 HDA support for OLPC. We have a choice of 50% or 80% microphone bias on the 2 microphone ports (one inbuilt, one socket for external devices). Any suggestions on which one we should pick or how to decide? We can't hear any difference between the 2 settings..should we choose 50% for potential power savings? Daniel

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[alsa-devel] [PATCH] ASoC: Keep index within stac9766_reg[]
by Roel Kluin 13 Jul '09

13 Jul '09

Keep index within stac9766_reg[] Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- please ignore the patch lacking the second hunk diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index 8ad4b7b..befc648 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -149,7 +149,7 @@ static int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg, stac9766_ac97_write(codec, AC97_INT_PAGING, 1); return 0; } - if (reg / 2 > ARRAY_SIZE(stac9766_reg)) + if (reg / 2 >= ARRAY_SIZE(stac9766_reg)) return -EIO; soc_ac97_ops.write(codec->ac97, reg, val); @@ -168,7 +168,7 @@ static unsigned int stac9766_ac97_read(struct snd_soc_codec *codec, stac9766_ac97_write(codec, AC97_INT_PAGING, 1); return val; } - if (reg / 2 > ARRAY_SIZE(stac9766_reg)) + if (reg / 2 >= ARRAY_SIZE(stac9766_reg)) return -EIO; if (reg == AC97_RESET || reg == AC97_GPIO_STATUS ||

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Re: [alsa-devel] [PATCH] New ASoC Drivers for ADI AD1938 codec
by Mark Brown 13 Jul '09

13 Jul '09

On Mon, Jul 13, 2009 at 04:12:14PM +0800, 宋宝华 wrote: > I want to use ADC/DAC widgets. > static const struct snd_soc_dapm_widget ad1938_dapm_widgets[] = { > SND_SOC_DAPM_DAC("DAC", "HiFi Playback", AD1938_DAC_CTRL0, 0, 1), > SND_SOC_DAPM_ADC("ADC", "HiFi Capture", AD1938_ADC_CTRL0, 0, 1), > }; > But for this AD1938 codec, DAC's work depends on ADC is powered on in > hardware. I think there is no any mechanism to handle this kind of > strange depending now. So is there a generic way to handle this? Make the ADC power a SND_SOC_DAPM_SUPPLY() supplying both the DAC and the ADC, with the ADC widget marged as having no power management.

1 0

[alsa-devel] [PATCH] ASoC: Keep index within stac9766_reg[]
by Roel Kluin 13 Jul '09

13 Jul '09

Keep index within stac9766_reg[] Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index 8ad4b7b..11f467e 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -149,7 +149,7 @@ static int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg, stac9766_ac97_write(codec, AC97_INT_PAGING, 1); return 0; } - if (reg / 2 > ARRAY_SIZE(stac9766_reg)) + if (reg / 2 >= ARRAY_SIZE(stac9766_reg)) return -EIO; soc_ac97_ops.write(codec->ac97, reg, val);

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[alsa-devel] Testers wanted: New SB X-Fi driver
by Christian Esken 12 Jul '09

12 Jul '09

Hello, unfortunately I cannot playback anything (PCM) with the new SB X-Fi driver. Driver: ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa- driver-snapshot.tar.bz2 Distribution: openSuSE 11.1 Kernel: 2.6.27.23-0.1-xen My card is a "X-Fi Titanium" (PCI device code: 1102:000b). The card is recognized by the driver, and any mixer (alsamixer, kmix) also recognizes the card. But playing back isn't possible. I tried speaker-test, audacity, amarok and aplay. Most applications hang, some also produce an error in the /var/log/messages (see below). I then used "./configure --with-debug=full --with-cards=ctxfi" due to the possible hd-audio conflict, but it didn't make a difference. Is there anything else I could try? Christian --- Console output from speaker-test --- bash # LANG=en speaker-test speaker-test 1.0.18 Playback device is default Stream parameters are 48000Hz, S16_LE, 1 channels Using 16 octaves of pink noise Rate set to 48000Hz (requested 48000Hz) Buffer size range from 512 to 65536 Period size range from 32 to 32768 Using max buffer size 65536 Periods = 4 was set period_size = 16384 was set buffer_size = 65536 0 - Front Left Write error: -5,Input/output error xrun_recovery failed: -5,Input/output error Transfer failed: Operation not permitted ------------------------------------------------------------------------ --- output in /var/log/messages when running speaker-test --- Jun 27 12:52:53 firefly kernel: ALSA /home/chris/Desktop/alsa- driver/acore/../alsa-kernel/core/pcm_lib.c:1693: playback write error (DMA or IRQ trouble?) ------------------------------------------------------------------------ ---output in /var/log/messages when running amarok with pulseudio --- 20 11:11:00 firefly pulseaudio[5649]: module-alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write! Most likely this is an ALSA driver bug. ------------------------------------------------------------------------ --- Hardware information --- firefly:~ # lspci -nn | grep -i audio 00:07.0 Audio device [0403]: nVidia Corporation MCP65 High Definition Audio [10de:044a] (rev a1) 01:07.0 Multimedia audio controller [0401]: Cirrus Logic CS 4614/22/24/30 [CrystalClear SoundFusion Audio Accelerator] [1013:6003] (rev 01) 05:00.0 Audio device [0403]: Creative Labs Device [1102:000b] (rev 03) --- Hardware information from lshal --- udi = '/org/freedesktop/Hal/devices/pci_1102_b_sound_card_0' info.capabilities = {'sound'} (string list) info.category = 'sound' (string) info.parent = '/org/freedesktop/Hal/devices/pci_1102_b' (string) info.product = 'Creative X-Fi 20K2 SB0880 Sound Card' (string) info.subsystem = 'sound' (string) info.udi = '/org/freedesktop/Hal/devices/pci_1102_b_sound_card_0' (string) linux.hotplug_type = 2 (0x2) (int) linux.subsystem = 'sound' (string) linux.sysfs_path = '/sys/devices/pci0000:00/0000:00:0e.0/0000:05:00.0/sound/card0' (string) sound.card = 0 (0x0) (int) sound.card_id = 'Creative X-Fi 20K2 SB0880' (string) sound.originating_device = '/org/freedesktop/Hal/devices/pci_1102_b' (string)

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[alsa-devel] Testers wanted: New SB X-Fi driver
by Christian Paredes 12 Jul '09

12 Jul '09

Hi everyone, So I have an Auzentech X-Fi Forte PCI-E card, and I got it to output sound through the side and center/LFE channels. When I output sound through the center channel, it is correctly routed to the center channel, but when sound is output to the LFE channel, it is output to both of my front speakers. I am running on kernel 2.6.28, fresh install of Ubuntu 9.04. This also happens on kernel 2.6.30 on Arch Linux. Let me know what sort of information you guys need! Thanks, Christian

1 1

[alsa-devel] Let's discuss the state of Audio on Linux - OSS and ALSA people
by apriori＠dkc-clan.de 11 Jul '09

11 Jul '09

Hi all. I would like to encourage all ALSA devs and users to talk about the current state of audio on linux together with the Open Sound System devs at: http://www.4front-tech.com/forum/viewtopic.php?t=3199 I recommend to take this offer since we all and the linux development would definetly benefit if both core audio system developer team would finally have a talk with each other. For the sake of information and knowledge please don't fill up the thread with off-topic comments or bashing comments to either sound system if it doesn't clearly state (in detail) 1) what is wrong and 2) what could be done better and 3) how it could be done better. Criticism should be done in a constructive way. Looking forward to see you in the dicussion apriori

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[alsa-devel] [PATCH for-2.6.32] sound dapm: fix checking for external widgets bug
by Rongrong Cao 11 Jul '09

11 Jul '09

From: Rongrong Cao rrcao(a)ambarella.com This patch is based on for-2.6.32 In SOC DAPM layer of SOUND subsystem, when add signal route (in the function snd_soc_dapm_add_route() ), the original code has wrong logic when dapm layer check each widget whether an external one. --------------------------------------------------------------------------------------------------------- --- for-2.6.32/sound/soc/soc-dapm.c.orig 2009-07-08 13:33:41.000000000 +0800 +++ for-2.6.32/sound/soc/soc-dapm.c 2009-07-08 13:34:50.000000000 +0800 @@ -1290,8 +1290,8 @@ static int snd_soc_dapm_add_route(struct if (wsink->id == snd_soc_dapm_input) { if (wsource->id == snd_soc_dapm_micbias || wsource->id == snd_soc_dapm_mic || - wsink->id == snd_soc_dapm_line || - wsink->id == snd_soc_dapm_output) + wsource->id == snd_soc_dapm_line || + wsource->id == snd_soc_dapm_output) wsink->ext = 1; } if (wsource->id == snd_soc_dapm_output) { --------------------------------------------------------------------------------------------------------- Signed-off-by: Rongrong Cao <rrcao(a)ambarella.com> ******************** Best regards! Allen (Rongrong) Cao Ambarella Shanghai Ltd. Tel: (86-21) 50942311 Fax: (86-21) 50942321

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[alsa-devel] codec drivers for Cirrus CS8406 or Texas Instruments PCM1748
by just4emmett 10 Jul '09

10 Jul '09

Hello all. Is anyone working on codec drivers for these devices, or know where to find them if they exist already? Checked http://opensource.wolfsonmicro.com/cgi-bin/gitweb.cgi?p=linux-2.6-asoc.git;… and didn't see anything. Thanks!

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Re: [alsa-devel] [Jack-Devel] Says: server is not connected, even though it has been!
by Guilherme Longo 10 Jul '09

10 Jul '09

Yes Patrick.... same user for both Really strange, isn't it! Patrick Shirkey wrote: > > On 07/10/2009 05:20 PM, Guilherme Longo wrote: >> Hi all! >> >> I couldn't find where this problem comes from. > > Hi, > > Sorry, to ask the obvious but, are you running as the same user for > the server and the simple_client? > > > Patrick Shirkey > Boost Hardware Ltd > > > >> I am connecting jack with the command: >> >> jackd -d alsa -d default >> >> and appears right below: >> >> bash-3.1# jackd -d alsa -d default >> jackdmp 1.9.2 >> Copyright 2001-2005 Paul Davis and others. >> Copyright 2004-2008 Grame. >> jackdmp comes with ABSOLUTELY NO WARRANTY >> This is free software, and you are welcome to redistribute it >> under certain conditions; see the file COPYING for details >> JACK server starting in non-realtime mode >> creating alsa driver ... >> default|default|1024|2|48000|0|0|nomon|swmeter|-|32bit >> Using ALSA driver EMU10K1 running on SB Live [Unknown] (rev.10, >> serial:0x80671102) at 0xd000, irq 18 >> configuring for 48000Hz, period = 1024 frames (21.3 ms), buffer = 2 >> periods >> ALSA: final selected sample format for capture: 32bit float >> little-endian >> You appear to be using the ALSA software "plug" layer, probably >> a result of using the "default" ALSA device. This is less >> efficient than it could be. Consider using a hardware device >> instead rather than using the plug layer. Usually the name of the >> hardware device that corresponds to the first soun >> ALSA: use 2 periods for capture >> ALSA: final selected sample format for playback: 32bit float >> little-endian >> You appear to be using the ALSA software "plug" layer, probably >> a result of using the "default" ALSA device. This is less >> efficient than it could be. Consider using a hardware device >> instead rather than using the plug layer. Usually the name of the >> hardware device that corresponds to the first soun >> ALSA: use 2 periods for playback >> >> >> In another terminal, I am running a simple_cliente, and when I run it >> with the code: >> >> quart$: simple_client audiofile >> jack server not running? >> >> It is resulting of: >> >> if ((client = jack_client_open(argv[1], JackNoStartServer, NULL)) == >> 0) { >> fprintf (stderr, "jack server not running?\n"); >> return 1; >> } >> >> >> Plz, can I get som help here. >> Tks. >> _______________________________________________ >> Jack-Devel mailing list >> Jack-Devel(a)lists.jackaudio.org >> http://lists.jackaudio.org/listinfo.cgi/jack-devel-jackaudio.org >

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[alsa-devel] Bug report: Random MIDI output data drop-outs/corruption
by Sean M. Pappalardo 10 Jul '09

10 Jul '09

Hello there. I just wanted to call your attention to a bug report I recently filed, because I'm not sure I put it in the right place: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4606 In a nutshell, ALSA seems to be dropping or corrupting output MIDI messages (both "short" and SysEx) at random as evidenced by the devices not lighting some LEDs or changing modes (sysex) on occasion. The problem is positively affected, but not completely mitigated, by the use of a real-time kernel. aseqdump -d shows everything correctly even when the device response is incorrect. All details are in that bug report including links to alsa-info output in the notes. Thanks for your time. Let me know if I can provide any further information. Sincerely, Sean M. Pappalardo "D.J. Pegasus" <<--------------------------------------------------------------------------------->> This E-Mail message has been scanned for viruses and cleared by >>SmartMail<< from Smarter Technology, Inc. <<--------------------------------------------------------------------------------->>

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[alsa-devel] problems on AT91 with SSC not generating frame sync on RF0 pin
by Stefan Schoenleitner 10 Jul '09

10 Jul '09

Hi, I wrote an ALSA machine driver for the WM8731 codec on the AT91SAM9260 platform (for code see end of mail). I'm using the latest ALSA sound-2.6 tree from git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git The audio codec is configured as I2S slave, so that the SSC is responsible to create the BCLK and LRC frame sync signals. my problem is that while playing audio works (and a LRC signal is created on the TF0 pin), recording does not work since there is no LRC signal on the RF0 pin. The connection between SSC and wm8731 is the following: codec SSC <-- BCLK/TK0 -- <-- ADCLRC/RF0 -- <-- DACLRC/TF0 -- <-- DACDAT/TD0 -- -- ADCDAT/RD0 --> In my board file I'm initializing the SSC with the following function call: ------------------------------------------------------------------------ static void __init ek_board_init(void) { .... at91_add_device_ssc(AT91SAM9260_ID_SSC, ATMEL_SSC_TK | ATMEL_SSC_RF | ATMEL_SSC_TF | ATMEL_SSC_TD | ATMEL_SSC_RD); .... } ------------------------------------------------------------------------ In arch/arm/mach-at91/at91sam9260_devices.c this leads to the following calls: ------------------------------------------------------------------------ static inline void configure_ssc_pins(unsigned pins) { if (pins & ATMEL_SSC_TF) at91_set_A_periph(AT91_PIN_PB17, 1); if (pins & ATMEL_SSC_TK) at91_set_A_periph(AT91_PIN_PB16, 1); if (pins & ATMEL_SSC_TD) at91_set_A_periph(AT91_PIN_PB18, 1); if (pins & ATMEL_SSC_RD) at91_set_A_periph(AT91_PIN_PB19, 1); if (pins & ATMEL_SSC_RK) at91_set_A_periph(AT91_PIN_PB20, 1); if (pins & ATMEL_SSC_RF) at91_set_A_periph(AT91_PIN_PB21, 1); } ------------------------------------------------------------------------ Then in the machine driver I'm initializing both codec and cpu DAI with SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS. For this reason the SSC is I2S master and has to generate the clock and frame sync signals. By looking at atmel_ssc_dai.c (and putting some printk() debug code there) I verified that the SSC registers are correctly set up so that the RF0 pin is configured as output. As mentioned playing audio works just fine. Thus by attaching a scope to the TF0 pin (which carries the transmitter frame sync signal) I can see that the signal is alright. However, if I record something with arecord, the SSC receiver is enabled, but no frame sync pulses can be seen on the RF0 pin. Therefore also audio recording does not work since there is no data being transfered from the codec to the SSC. The kernel output (with some debug code added) looks good: ------------------------------------------------------------------------ Jul 9 22:30:52 devboard kernel: atmel_ssc_startup: SSC_SR=0x12 Jul 9 22:30:52 devboard kernel: sam9l9260_startup() Jul 9 22:30:52 devboard kernel: asoc: WM8731 <-> atmel-ssc0 info: Jul 9 22:30:52 devboard kernel: asoc: rate mask 0x6fe Jul 9 22:30:52 devboard kernel: asoc: min ch 1 max ch 2 Jul 9 22:30:52 devboard kernel: asoc: min rate 8000 max rate 96000 Jul 9 22:30:52 devboard kernel: sam9l9260_hw_params() Jul 9 22:30:52 devboard kernel: sam9l9260_hw_params: sample rate is 44100 Hz Jul 9 22:30:52 devboard kernel: sam9l9260_hw_params: substream->stream is SNDRV_PCM_STREAM_CAPTURE Jul 9 22:30:52 devboard kernel: sam9l9260_hw_params: snd_soc_dai_set_clkdiv() --> I'm receiver ! Jul 9 22:30:52 devboard kernel: atmel_ssc_hw_params: setting up for SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS Jul 9 22:30:52 devboard kernel: atmel_ssc_hw_params: RCMR=21010420 RFMR=001f018f TCMR=00010404 TFMR=001f018f Jul 9 22:30:52 devboard kernel: atmel_ssc_dai: Starting clock Jul 9 22:30:52 devboard kernel: atmel_ssc_dai,hw_params: SSC initialized Jul 9 22:30:52 devboard kernel: atmel-pcm: hw_params: DMA for SSC0 PCM in initialized (dma_bytes=32768, period_size=8192) Jul 9 22:30:52 devboard kernel: receive enabled SSC_SR=0x0002040c Jul 9 22:30:53 devboard kernel: atmel-pcm:buffer_size = 8192,dma_area = ffc1b000, dma_bytes = 32768 Jul 9 22:30:53 devboard kernel: atmel-pcm: trigger: period_ptr=23a9a000, xpr=598310912, xcr=4096, xnpr=598319104, xncr=4096 Jul 9 22:30:53 devboard kernel: sr=133164 imr=0 atmel-pcm:buffer_size = 8192,dma_area = ffc1b000, dma_bytes = 32768 atmel_ssc_shutdown: receive disabled SSC_SR=0x0002083c atmel_ssc_dau: Stopping clock Jul 9 22:36:04 devboard kernel: atmel-pcm:buffer_size = 8192,dma_area = ffc1b000, dma_bytes = 32768 Jul 9 22:36:04 devboard kernel: atmel_ssc_shutdown: receive disabled SSC_SR=0x0002083c Jul 9 22:36:04 devboard kernel: atmel_ssc_dau: Stopping clock Jul 9 22:36:04 devboard kernel: sam9l9260_shutdown() Unregistered DAI 'WM8731' Unregistered codec 'WM8731' Unregistered DAI 'atmel-ssc0' Unregistered platform 'atmel-audio' Jul 9 22:36:18 devboard kernel: Unregistered DAI 'WM8731' Jul 9 22:36:18 devboard kernel: Unregistered codec 'WM8731' Jul 9 22:36:18 devboard kernel: Unregistered DAI 'atmel-ssc0' Jul 9 22:36:18 devboard kernel: Unregistered platform 'atmel-audio' ------------------------------------------------------------------------ As you can see above, the receiver is enabled (SNDRV_PCM_STREAM_CAPTURE), the clock is set for the receiver and the SSC starts to generate the BCLK (which can be measured with tthe scope). Also the SSC registers should be configured correctly (thus the RF0 pin should be output for frame sync), but it still does not work. I also checked for hardware problems by completely disconnecting the SSC from the codec and directly attaching the lines to a scope. The result it the same: While playing sound produces the correct signals, recording does not do so since there is no frame sync on the RF0 pin. I also checked for bad solder junctions and I did do the measurements directly on the tiny AT91SAM9260 controller pins as well (with the same result). And of course I also looked at the schematic and verified that the pins are not connected to anything else. Also, the pins are not anywhere else in my kernel sources as for the SSC (I verified this by looking at the code and searching for the relevant pin names). I read the relevant sections in the at91sam9260 and wm8731 datasheets very carefully and also verified that the setup is according to specification as well. I'm pretty sure that it is a software problem somewhere, but after so much time trying to fix the problem I no longer have any ideas what could be wrong. For my understanding the pins need to be multiplexed for use with the SSC. I verified this and the pins are indeed multiplexed with calls to at91_set_A_periph() on the according pins (see code snippet above). Once this has been done the pins should be ready for use with the SSC. Thus once the SSC registers are configured (which they verifiably are in atmel_ssc_dai.c) the pins should behave according to the SSC register configuration. And since the SSC registers are configured in a way that the RF0 pin is an ouput with with the LR frame sync signal for the receive on it, it should also work (which it doesn't). If you have any ideas why it is not working I would be very grateful. sincerely, Stefan machine driver code: ------------------------------------------------------------------------ #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/kernel.h> #include <linux/timer.h> #include <linux/interrupt.h> #include <linux/platform_device.h> #include <linux/i2c.h> #include <linux/atmel-ssc.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/soc-dapm.h> #include <asm/mach-types.h> #include <mach/hardware.h> #include <mach/gpio.h> #include "../codecs/wm8731.h" #include "atmel-pcm.h" #include "atmel_ssc_dai.h" #define MCLK_RATE 12288000 static int sam9l9260_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; int ret; printk(KERN_INFO "sam9l9260_startup()\n"); ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, MCLK_RATE, SND_SOC_CLOCK_IN); if (ret < 0) { return ret; } return 0; } static void sam9l9260_shutdown(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); printk(KERN_INFO "sam9l9260_shutdown()\n"); dev_dbg(rtd->socdev->dev, "shutdown"); } static int sam9l9260_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct atmel_ssc_info *ssc_p = cpu_dai->private_data; struct ssc_device *ssc = ssc_p->ssc; int ret; unsigned int rate; int cmr_div, period; printk(KERN_INFO "sam9l9260_hw_params()\n"); if (ssc == NULL) { printk(KERN_INFO "sam9l9260_hw_params: ssc is NULL!\n"); return -EINVAL; } /* set codec DAI configuration */ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; /* set cpu DAI configuration */ /* * atmel_ssc_dai.c * SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS * --> I2S format, SSC provides BCLK and LRC clocks. * * SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM: * --> I2S format, CODEC supplies BCLK and LRC clocks. */ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; /* * The SSC clock dividers depend on the sample rate. The CMR.DIV * field divides the system master clock MCK to drive the SSC TK * signal which provides the codec BCLK. The TCMR.PERIOD and * RCMR.PERIOD fields further divide the BCLK signal to drive * the SSC TF and RF signals which provide the codec DACLRC and * ADCLRC clocks. * * The dividers were determined through trial and error, where a * CMR.DIV value is chosen such that the resulting BCLK value is * divisible, or almost divisible, by (2 * sample rate), and then * the TCMR.PERIOD or RCMR.PERIOD is BCLK / (2 * sample rate) - 1. * * AT91SAM9260 datasheet: * CMR - clock mode register - p. 492 * CMR.DIV is 12-bit value, the divided clock equals * the master clock devided by 2 times DIV * * TCMR - transmit clock mode register - p. 497 * TCMR.PERIOD is 8-bit value, divider for frame sync signal (LRCLK). * (Also called: WM8731 --> DACLRC, TLV320AIC23 --> LRCIN) * A period signal is generated each 2*(PERIOD+1) transmit clock * * RCMR - receive clock mode register - p. 493 * RCMR.PERIOD is 8-bit value, divider for frame sync signal (LRCLK). * (Also called: WM8731 --> ADCLRC, TLV320AIC23 --> LRCOUT) * A period signal is generated each 2*(PERIOD+1) receive clock * * The LRCLK sync signals needs to be as close to the sampling rate as possible. * * The following register values have been calculated with my * wm8731_calculate_ssc_dividers.pl script. * * According to the wm8731 datasheet, the minimum BCLK cycle time is 50 ns. * That would be f = 1/T = 1/0,050us = 20 MHz */ rate = params_rate(params); printk(KERN_INFO "sam9l9260_hw_params: sample rate is %i Hz\n", rate); switch (rate) { case 8000: cmr_div=29; /* BCLK = 90 MHz / (2*29) ~= 1.551 MHz */ period=96; /* LRC = BCLK/(2*(96+1)) ~= 7998.6 Hz */ break; case 11025: cmr_div=13; /* BCLK ~= 3.461 Mhz */ period=156; /* LRC ~= 11024.0 Hz */ break; case 16000: cmr_div=38; /* BCLK ~= 1.184 */ period=36; /* LRC ~= 16002.8 */ break; case 22050: cmr_div=17; /* BCLK ~= 2.647 */ period=59; /* LRC ~= 22058.8 */ break; case 32000: cmr_div=19; /* BCLK ~= 2.368 */ period=36; /* LRC ~= 32005.6 */ break; case 44100: cmr_div=15; /* BCLK ~= 3.000 */ period=33; /* LRC ~= 44117.6 */ break; case 48000: cmr_div=7; /* BCLK ~= 6.428 */ period=66; /* LRC ~= 47974.4 */ break; case 88200: cmr_div=5; /* BCLK ~= 9.000 */ period=50; /* LRC ~= 88235.3 */ break; case 96000: cmr_div=6; /* BCLK ~= 7.500 */ period=38; /* LRC ~= 96153.8 */ break; default: printk(KERN_WARNING "unsupported rate %d" " on at91sam9g20ek board\n", rate); return -EINVAL; } /* set the MCK divider for BCLK */ ret = snd_soc_dai_set_clkdiv(cpu_dai, ATMEL_SSC_CMR_DIV, cmr_div); if (ret < 0) return ret; switch (substream->stream) { case SNDRV_PCM_STREAM_PLAYBACK: printk(KERN_DEBUG "%s: substream->stream is SNDRV_PCM_STREAM_PLAYBACK\n", __func__); break; case SNDRV_PCM_STREAM_CAPTURE: printk(KERN_DEBUG "%s: substream->stream is SNDRV_PCM_STREAM_CAPTURE\n", __func__); break; default: printk(KERN_DEBUG "%s: substream->stream is INVALID !!!\n", __func__); break; } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { /* set the BCLK divider for DACLRC */ ret = snd_soc_dai_set_clkdiv(cpu_dai, ATMEL_SSC_TCMR_PERIOD, period); printk(KERN_DEBUG "%s: snd_soc_dai_set_clkdiv() --> I'm transmitter !\n", __func__); } else { /* set the BCLK divider for ADCLRC */ ret = snd_soc_dai_set_clkdiv(cpu_dai, ATMEL_SSC_RCMR_PERIOD, period); printk(KERN_DEBUG "%s: snd_soc_dai_set_clkdiv() --> I'm receiver !\n", __func__); } if (ret < 0) return ret; return 0; } static struct snd_soc_ops sam9l9260_ops = { .startup = sam9l9260_startup, .hw_params = sam9l9260_hw_params, .shutdown = sam9l9260_shutdown, }; static const struct snd_soc_dapm_widget sam9l9260_dapm_widgets[] = { SND_SOC_DAPM_MIC("Mic", NULL), SND_SOC_DAPM_SPK("Phones", NULL), }; static const struct snd_soc_dapm_route intercon[] = { /* speaker connected to LHPOUT */ {"Phones", NULL, "LHPOUT"}, /* mic is connected to Mic Jack, with WM8731 Mic Bias */ {"MICIN", NULL, "Mic Bias"}, {"Mic Bias", NULL, "Mic"}, }; /* * Logic for a wm8731 as connected on a at91sam9g20ek board. */ static int sam9l9260_wm8731_init(struct snd_soc_codec *codec) { printk(KERN_DEBUG "sam9l9260_wm8731 " ": sam9l9260_wm8731_init() called\n"); printk(KERN_INFO "sam9l9260_wm8731_init()\n"); /* Add specific widgets */ snd_soc_dapm_new_controls(codec, sam9l9260_dapm_widgets, ARRAY_SIZE(sam9l9260_dapm_widgets)); /* Set up specific audio path interconnects */ snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); /* not connected */ snd_soc_dapm_nc_pin(codec, "RLINEIN"); snd_soc_dapm_nc_pin(codec, "LLINEIN"); snd_soc_dapm_nc_pin(codec, "ROUT"); snd_soc_dapm_nc_pin(codec, "LOUT"); /* always connected */ snd_soc_dapm_enable_pin(codec, "Mic"); snd_soc_dapm_enable_pin(codec, "Phones"); snd_soc_dapm_sync(codec); return 0; } static struct snd_soc_dai_link sam9l9260_dai = { .name = "WM8731", .stream_name = "WM8731 PCM", .cpu_dai = &atmel_ssc_dai[0], .codec_dai = &wm8731_dai, .init = sam9l9260_wm8731_init, .ops = &sam9l9260_ops, }; static struct snd_soc_card snd_soc_at91sam9g20ek = { .name = "SAM9-L9260", .platform = &atmel_soc_platform, .dai_link = &sam9l9260_dai, .num_links = 1, }; static struct snd_soc_device sam9l9260_snd_devdata = { .card = &snd_soc_at91sam9g20ek, .codec_dev = &soc_codec_dev_wm8731, }; static struct platform_device *sam9l9260_snd_device; static int __init sam9l9260_init(void) { struct atmel_ssc_info *ssc_p = sam9l9260_dai.cpu_dai->private_data; struct ssc_device *ssc = NULL; int ret; printk(KERN_INFO "sam9l9260_init()\n"); /* Olimex SAM9-L9260 board */ if (!(machine_is_sam9_l9260())) return -ENODEV; /* * Request SSC device */ ssc = ssc_request(0); if (IS_ERR(ssc)) { printk(KERN_ERR "ASoC: Failed to request SSC 0\n"); ret = PTR_ERR(ssc); ssc = NULL; goto err_ssc; } ssc_p->ssc = ssc; sam9l9260_snd_device = platform_device_alloc("soc-audio", -1); if (!sam9l9260_snd_device) { printk(KERN_ERR "ASoC: Platform device allocation failed\n"); ret = -ENOMEM; } platform_set_drvdata(sam9l9260_snd_device, &sam9l9260_snd_devdata); sam9l9260_snd_devdata.dev = &sam9l9260_snd_device->dev; ret = platform_device_add(sam9l9260_snd_device); if (ret) { printk(KERN_ERR "ASoC: Platform device allocation failed\n"); platform_device_put(sam9l9260_snd_device); } return ret; err_ssc: ssc_free(ssc); ssc_p->ssc = NULL; return ret; } static void __exit sam9l9260_exit(void) { struct atmel_ssc_info *ssc_p = sam9l9260_dai.cpu_dai->private_data; struct ssc_device *ssc; if (ssc_p != NULL) { ssc = ssc_p->ssc; if (ssc != NULL) ssc_free(ssc); ssc_p->ssc = NULL; } platform_device_unregister(sam9l9260_snd_device); sam9l9260_snd_device = NULL; } module_init(sam9l9260_init); module_exit(sam9l9260_exit); MODULE_DESCRIPTION("ALSA SoC sam9l9260_WM8731"); MODULE_LICENSE("GPL"); ------------------------------------------------------------------------

1 0

[alsa-devel] save alsa buffer
by arif setiawan 10 Jul '09

10 Jul '09

Hi, I am trying to save content of alsa buffer to file for playing it in Matlab.I choose Matlab for prototyping. I run full duplex ALSA based on code from http://www.bel.fi/~alankila/blog/2005/08/21/Full Duplex ALSA.html<http://www.bel.fi/%7Ealankila/blog/2005/08/21/Full%20Duplex%20ALSA.html> I run alsa on Beagleboard. I want to save buffer content to a file, but it failed with error: Segmentation fault when reach fclose(fp). I want to know what kind of data that ALSA processes so I can implement my LPC vocoder. Thank you. Best regards, Arif

1 0

[alsa-devel] runtime->boundary
by Harsha, Priya 10 Jul '09

10 Jul '09

Hi, Can anyone help explain what is runtime->boundary and what is the value that is set for this? It seems to be a pretty huge value when I try and print it. Also in snd_pcm_indiret_capture_transfer, I see this code snippet; If (diff < runtime->boundary /2) diff += runtime->boundary what is the reason for the same? If its for handling ring buffer wrapping, should I be setting my runtime->boundary to rumtime->buffer_size in the driver code? Thanks, Harsha

2 1

[alsa-devel] No incoming sound indigo iox
by Susan Cragin 10 Jul '09

10 Jul '09

I just purchased the indigoaudio iox. Outgoing shows up on Volume Control, incoming does not. In Audacity, incoming shows up as: ALSA: Indigo IOx (hw:1,0) but recording does not work. Here's the catch. My problem may be due to my kernel. I am running Ubuntu-Studio and have the 29.5-2-rt experimental kernel, which is the 5-1 recompiled with alsadriver support for 30 kernels. So you may just tell me to go away until the kernel's better, but I don't think the kernel is the problem. FYI winecfg says: fixme:mixer:ALSA_MixerInit No master control found on Indigo IOx, disabling mixer

2 1

[alsa-devel] targa and targa-2ch fix for 2.6.31
by okias 09 Jul '09

09 Jul '09

Simplify ALC882_TARGA and return gpio3 to ALC883_TARGA_DIG and ALC883_TARGA_2ch_DIG, which I accidentally removed in commit id 64a8be74357477558183b43156c5536b642de134 Signed-off-by: David Heidelberger <d.okias(a)gmail.com> --- --- sound/pci/hda/patch_realtek.c 2009-07-09 17:36:37.593323363 +0200 +++ sound/pci/hda/patch_realtek.c 2009-07-09 17:57:31.014574807 +0200 @@ -6895,9 +6895,6 @@ static struct hda_verb alc882_targa_verb {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, - {0x01, AC_VERB_SET_GPIO_MASK, 0x03}, - {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x03}, - {0x01, AC_VERB_SET_GPIO_DATA, 0x03}, { } /* end */ }; @@ -7217,7 +7214,8 @@ static struct alc_config_preset alc882_p }, [ALC882_TARGA] = { .mixers = { alc882_targa_mixer, alc882_chmode_mixer }, - .init_verbs = { alc882_init_verbs, alc882_targa_verbs}, + .init_verbs = { alc882_init_verbs, alc880_gpio3_init_verbs, + alc882_targa_verbs}, .num_dacs = ARRAY_SIZE(alc882_dac_nids), .dac_nids = alc882_dac_nids, .dig_out_nid = ALC882_DIGOUT_NID, @@ -9212,7 +9210,8 @@ static struct alc_config_preset alc883_p }, [ALC883_TARGA_DIG] = { .mixers = { alc883_targa_mixer, alc883_chmode_mixer }, - .init_verbs = { alc883_init_verbs, alc883_targa_verbs}, + .init_verbs = { alc883_init_verbs, alc880_gpio3_init_verbs, + alc883_targa_verbs}, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .dig_out_nid = ALC883_DIGOUT_NID, @@ -9225,7 +9224,8 @@ static struct alc_config_preset alc883_p }, [ALC883_TARGA_2ch_DIG] = { .mixers = { alc883_targa_2ch_mixer}, - .init_verbs = { alc883_init_verbs, alc883_targa_verbs}, + .init_verbs = { alc883_init_verbs, alc880_gpio3_init_verbs, + alc883_targa_verbs}, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .adc_nids = alc883_adc_nids_alt, -- Jabber/XMPP: okias(a)isgeek.info SIP VoIP: sip:17474537254@proxy01.sipphone.com

2 1

[alsa-devel] [PATCH] snd-hda-intel: fix beep tone calculation for IDT/STAC codecs
by Paul Vojta 09 Jul '09

09 Jul '09

In the beep tone calculation for IDT/STAC codecs, lower numbers correspond to higher frequencies and vice versa. The current code has this backwards, resulting in beep frequencies which are way too high (and sound bad on tinny laptop speakers, resulting in complaints). Signed-off-by: Paul Vojta <vojta(a)math.berkeley.edu> diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 08fe659..d559a7a 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -51,19 +51,20 @@ static void snd_hda_generate_beep(struct work_struct *work) * The tone frequency of beep generator on IDT/STAC codecs is * defined from the 8bit tone parameter, in Hz, * freq = 48000 * (257 - tone) / 1024 - * that is from 12kHz to 93.75kHz in step of 46.875 hz + * that is from 12kHz to 93.75Hz in steps of 46.875 Hz */ static int beep_linear_tone(struct hda_beep *beep, int hz) { hz *= 1000; /* fixed point */ - hz = hz - DIGBEEP_HZ_MIN; + hz = hz - DIGBEEP_HZ_MIN + + DIGBEEP_HZ_STEP / 2; /* round to nearest step */ if (hz < 0) hz = 0; /* turn off PC beep*/ else if (hz >= (DIGBEEP_HZ_MAX - DIGBEEP_HZ_MIN)) - hz = 0xff; + hz = 1; /* max frequency */ else { hz /= DIGBEEP_HZ_STEP; - hz++; + hz = 255 - hz; } return hz; }

2 1

[alsa-devel] [PATCH] [resend] cmi8330: find OPL3 port automatically
by Ondrej Zary 09 Jul '09

09 Jul '09

My CMI8329 had OPL3 port specified in SB16 resources. But now I found out that it was my modification of the card's PnP EEPROM a couple of years ago (can be done using C9SETROM.EXE utility). I did it because the OPL3 port was completely missing from PnP data. It seems to be hardwired to 0x388 on CMI8329. Find OPL3 port automatically by searching in WSS and SB16 resources. If not found, assume that it's hardwired to 0x388. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- linux-test/sound/isa/cmi8330.c 2009-07-06 21:18:06.000000000 +0200 +++ linux-2.6.30-pentium/sound/isa/cmi8330.c 2009-07-06 21:45:29.000000000 +0200 @@ -362,7 +362,7 @@ wssport[dev] = pnp_port_start(pdev, 0); wssdma[dev] = pnp_dma(pdev, 0); wssirq[dev] = pnp_irq(pdev, 0); - if (acard->type == CMI8330) + if (pnp_port_start(pdev, 1)) fmport[dev] = pnp_port_start(pdev, 1); /* allocate SB16 resources */ @@ -377,8 +377,13 @@ sbdma8[dev] = pnp_dma(pdev, 0); sbdma16[dev] = pnp_dma(pdev, 1); sbirq[dev] = pnp_irq(pdev, 0); - if (acard->type == CMI8329) - fmport[dev] = pnp_port_start(pdev, 1); + /* On CMI8239, the OPL3 port might be present in SB16 PnP resources */ + if (fmport[dev] == SNDRV_AUTO_PORT) { + if (pnp_port_start(pdev, 1)) + fmport[dev] = pnp_port_start(pdev, 1); + else + fmport[dev] = 0x388; /* Or hardwired */ + } /* allocate MPU-401 resources */ pdev = acard->mpu; -- Ondrej Zary

2 2

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

09 Jul '09

The following changes since commit 4ec5c9693b46ab34976511cd43bf75c3a0fc704d: Mark Brown (1): Merge branch 'for-2.6.31' into for-2.6.32 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Chaithrika U S (4): ASoC: Introduce platform driver model for dm644x, dm355 ASoC: Add mcasp support for DM646x ASoC: Add machine driver support for DM646x ASoC: Minor fixes to DaVinci I2S probe function Mark Brown (5): Merge branch 'davinci' into for-2.6.32 ASoC: Add TLV information for WM8731 ASoC: Correct WM8731 Mic Capture Switch control name ASoC: DaVinci I2S needs mach/asp.h ASoC: Add WM8993 CODEC driver Troy Kisky (1): ASoC: DaVinci: pcm, don't play 1st sound period twice include/sound/wm8993.h | 44 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8731.c | 17 +- sound/soc/codecs/wm8993.c | 2203 +++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8993.h | 2132 +++++++++++++++++++++++++++++++++++ sound/soc/davinci/Kconfig | 13 + sound/soc/davinci/Makefile | 3 + sound/soc/davinci/davinci-evm.c | 102 +- sound/soc/davinci/davinci-i2s.c | 123 ++- sound/soc/davinci/davinci-mcasp.c | 874 +++++++++++++++ sound/soc/davinci/davinci-mcasp.h | 55 + sound/soc/davinci/davinci-pcm.c | 1 + sound/soc/davinci/davinci-pcm.h | 18 +- 14 files changed, 5473 insertions(+), 118 deletions(-) create mode 100644 include/sound/wm8993.h create mode 100644 sound/soc/codecs/wm8993.c create mode 100644 sound/soc/codecs/wm8993.h create mode 100644 sound/soc/davinci/davinci-mcasp.c create mode 100644 sound/soc/davinci/davinci-mcasp.h

2 1

[alsa-devel] Choosing correct nid and xindex for hda_intel realtek patch for MacBook 5.1
by Chris Lasher 08 Jul '09

08 Jul '09

I'm trying to fix support for my MacBook 5.1. A patch by Kacper is getting me part of the way there but it seems my pin configuration is different than his model of MacBook. I'm trying to wire up my pins to his correctly. I am trying to figure out the bits of snd_kcontrol_new (in alsa-driver-1.0.19/alsa-kernel/pci/hda/patch_realtek.c) for the HDA_CODEC_VOLUME and HDA_BIND_MUTE. There is: string_description, nid , xindex, HDA_INPUT/HDA_OUTPUT I'm trying to figure out how to choose the correct values for nid and xindex. Actually, I'm really wondering how to identify the right xindex for a given nid, as I realize the nid's come from the Node #s from the output of /proc/asound/card0/codec#0. I also need some assistance with regards to the hda_verb settings for modifying the alc885_mb5_init_verbs section; if anyone can offer pointers there, I'd appreciate it. This is directly related to Bug #337314 in Launchpad https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/337314?comments=… and ALSA bug 4432: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4432 Kacper's patch is linked to in both bug reports. The output of my soundcard's codec#0 is attached to the Launchpad report https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/337314/comments/… The output from a soundcard that does seem to have full support by Kacper's patch is available at https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/337314/comments/… Any help would be much appreciated. Thanks, Chris

1 0

[alsa-devel] [PATCH] ALSA: ASoC: Fix NULL pointer dereference in __pxa2xx_pcm_hw_free
by Daniel Mack 08 Jul '09

08 Jul '09

Check for rtd->params->drcmr != NULL before accessing it. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Eric Miao <eric.miao(a)marvell.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/arm/pxa2xx-pcm-lib.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index 108b643..6205f37 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -75,7 +75,7 @@ int __pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream) { struct pxa2xx_runtime_data *rtd = substream->runtime->private_data; - if (rtd && rtd->params) + if (rtd && rtd->params && rtd->params->drcmr) *rtd->params->drcmr = 0; snd_pcm_set_runtime_buffer(substream, NULL); -- 1.6.3.1

2 1

[alsa-devel] [PATCH] ASoC: DaVinci: pcm, don't play 1st sound period twice
by Troy Kisky 08 Jul '09

08 Jul '09

Update the dma link with correct data as soon as the master channel has copied it. Otherwise, the 1st period will play twice. Signed-off-by: Troy Kisky <troy.kisky(a)boundarydevices.com> --- sound/soc/davinci/davinci-pcm.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index a059965..8fd0c3c 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -206,6 +206,7 @@ static int davinci_pcm_prepare(struct snd_pcm_substream *substream) /* Copy self-linked parameter RAM entry into master channel */ edma_read_slot(prtd->slave_lch, &temp); edma_write_slot(prtd->master_lch, &temp); + davinci_pcm_enqueue_dma(substream); return 0; } -- 1.5.6.3

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Re: [alsa-devel] No sound with nVidia Corporation MCP79 High Definition Audio (rev b1) (Macbook Pro 5, 5)
by Takashi Iwai 08 Jul '09

08 Jul '09

At Tue, 7 Jul 2009 11:49:57 +0200, Andreas Nüßlein wrote: > > On Tuesday 07 July 2009 08:00:19 you wrote: > > At Mon, 6 Jul 2009 22:14:56 +0200, > > > > Andreas Nüßlein wrote: > > > On Monday 06 July 2009 21:26:18 you wrote: > > > > At Mon, 06 Jul 2009 11:16:35 -0600, > > > > > > > > Sean Burke wrote: > > > > > Scríobh Takashi Iwai: > > > > > > At Mon, 6 Jul 2009 17:53:46 +0200, > > > > > > > > > > > > Andreas Nüßlein wrote: > > > > > >>> The missing pin configuration initialization was already fixed by > > > > > >>> the driver overriding it after checking PCI SSID (which is > > > > > >>> different from the codec SSID). So, this should be no problem. > > > > > >>> > > > > > >>> However, the reason why the analog output doesn't work might be > > > > > >>> different from that. There might be something else missing, but > > > > > >>> I don't know. > > > > > >>> > > > > > >>> > > > > > >>> Takashi > > > > > >> > > > > > >> oh =( > > > > > >> > > > > > >> > > > > > >> hmm.. anything i can do? would it help if i tried changing values > > > > > >> randomly with hda-analyzer.py? > > > > > > > > > > > > Well, did the driver without my change work more or less with > > > > > > the analog audio, or have you never gotten the analog output? > > > > > > You can use the generic parser (i.e. the state without cirrus > > > > > > patch) by passing model=generic option to snd-hda-intel. > > > > > > > > > > For my part, nothing worked with the generic driver. I can't confirm > > > > > digital out, but I can confirm that the kfree error is gone. What > > > > > options are open for figuring out what remains? > > > > > > > > Easy things to test are GPIO bits. Run hda-verb like > > > > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_MASK 0x0f > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x0f > > > > or > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIR 0x0f > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x0f > > > > > > > > etc. CS4206 seems to have 4 GPIO lines, and each bit (0-3) > > > > corresponds to each GPIO. In many case, GPIO0 or GPIO1 corresponds to > > > > the amplifier (EAPD) bit. > > > > Define the GPIO direction of each GPIO bit by SET_GPIO_DIR, and > > > > turn on/off the GPIO bits by SET_GPIO_DATA. Running > > > > hda-verb /dev/snd/hwC0D0 0x01 GET_GPIO_DATA 0 > > > > will show the current GPIO data bits. Or you can check it in codec#* > > > > proc file. > > > > > > > > > > > > Takashi > > > > > > w000000000000000000000000000000000t! =) > > > > > > takashi, thank you _so_ much! > > > > > > after running all 4 of those: > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_MASK 0x0f > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x0f > > > > or > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIR 0x0f > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x0f > > > > > > i suddendly had sound!! :D :D :D :D :D :D :D > > > only via speakers though - there is no sound via headphones right now. > > > > > > mixer channels: > > > - Master (with Mutebutton), PCM and Front (also with Mute) all work =) > > > - i don't know what surround would do (or it's extra switch) > > > - headphones-volumes and mute button don't affect the speakers, which is > > > good =) > > > > > > > > > is there a way to reset what i did with hda-verb, so that i can figure > > > out which combination it was exactly? > > > > You can just change the value 0x0f to a different value. > > At least, you can try commands like > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x01 > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x02 > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x04 > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x08 > > and check the speaker output at each time. > > Also, check the GPIO direction, > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIR 0x01 > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x01 > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIR 0x02 > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x02 > > ... > > > > Regarding the headphone: is the speaker muted when you plug in the > > headphone? If not, it's likely an issue of the jack detection. If > > the speaker is muted but no headphone output, it's a missing > > initialization (or wrong GPIO setup). > > > > > > Takashi > > > so here are my findings so far: > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x08 > > > is sufficient to turn on the right functionality for the speakers; meaning: > - 'front speaker'-channel actually controls the two front speakers within the > macbookpro - even controlling left and right channel seperately works =) > - 'surround' mute-switch and volume-control (why are those not within on > thing btw?) also work and control a third speaker in the mac, which > really seems responsibly for surround :) > > headphones do not yet work, however i made a diff of > "/proc/asound/card0/codec#0" with and without something plugged in: OK, so actually the driver works as expected, but the hardware doesn't behave as expected. For muting the speaker, something else (like toggling that GPIO) is needed. The mystery about the silent HP output still remains, though. Takashi

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[alsa-devel] [PATCH 2.6.31-rc2] Missing volume controls for Intel HDA (ALC269/EeePC)
by Darren Salt 08 Jul '09

08 Jul '09

There is a regression, introduced in aa202455eec51699e44f658530728162cefa1307 (in alsa-kernel) which I noticed when trying to use the headphone socket on my EeeCPC 901: the output was *very* quiet, practically silent. This patch corrects the control types to that which was obviously intended in the referenced commit. Signed-Off-By: Darren Salt <linux(a)youmustbejoking.demon.co.uk> --- a/sound/pci/hda/patch_realtek.c 2009-07-05 17:30:41.000000000 +0100 +++ b/sound/pci/hda/patch_realtek.c 2009-07-08 14:50:38.000000000 +0100 @@ -12878,9 +12878,9 @@ static struct snd_kcontrol_new alc269_eeepc_mixer[] = { HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), { } /* end */ }; -- | Darren Salt | linux at youmustbejoking | nr. Ashington, | Doon | using Debian GNU/Linux | or ds ,demon,co,uk | Northumberland | Army | <URL:http://www.youmustbejoking.demon.co.uk/progs.packages.html> I have one illusion, and that's that I have no illusions.

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[alsa-devel] Problems with snd-hda-intel and Asus M2NPV-VM. Please help!
by VDR User 08 Jul '09

08 Jul '09

I posted this to the alsa-user mailing list but since then I've read that the problem may actually be a bug in alsa so I've decided to forward the posting here as well. Thanks for your consideration. Best regards, Derek ---------- Forwarded message ---------- From: VDR User <user.vdr(a)gmail.com> Date: Wed, Jun 24, 2009 at 11:08 AM Subject: Problems with snd-hda-intel and Asus M2NPV-VM. Please help! To: "mailing list: alsa" <alsa-user(a)lists.sourceforge.net> Greetings. I've been trying for several days now to get the internal spdif working and am posting here as a last resort in hopes someone will know what's wrong. I'll post system details at the end. My mainboard has 3 1/8" plugs on the back and an internal spdif_out connector. If I plug say headphones into the green connector on the back, I hear sound in them. I don't actually want to use analog sound at all. Instead I've plugged the internal spdif_out+ground into my video cards spdif input. Then I have the dvi output from the video card going to an hdmi input on my tv. Note, I have the same setup (except for different mainboard) in another box and have used it to test the video card, which worked fine. The problem is spdif doesn't seem to be working at all. I've eliminated everything else down to either a problem in alsa somewhere, or possibly my xine settings (though I doubt it since I used the same settings from my working box). For alsa I've tried loading the snd_hda_intel driver with "model=3stack", "model=3stack position_fix=1", and "model=3stack position_fix=1 enable_msi=1". None of these worked, I still get nothing on spdif out. I've also tried (un)muting tons of combinations in alsamixer. Could the problem be my kernel version? Alsa version? libasound? Any ideas at all guys?? Any and all help is GREATLY appreciated! Best regards, Derek System details: Please note there is no desktop/windows manager. There is also no keyboard/mouse plugged in. This box is a dedicated htpc running VDR and connected directly to a tv. I'm using PuTTy SSH from my Windows desktop for maintenance. debian testing kernel 2.6.30 asus m2npv-vm mainboard (tried both firmware 0901 and 1401) - uses mcp51 chipset asus EN8400GS Silent 512MB pci-e video card (with spdif input connector) vdr-1.7.8 xine-0.9.3 xine-vdpau r273 ii alsa-base 1.0.20+dfsg-1 ALSA driver configuration files ii alsa-utils 1.0.20-1 ALSA utilities ii libesd-alsa0 0.2.41-4 Enlightened Sound Daemon (ALSA) - Shared lib ii libsdl1.2debian-alsa 1.2.13-4+b1 Simple DirectMedia Layer (with X11 and ALSA htpc:~$ sudo alsactl init Unknown hardware: "HDA-Intel" "Analog Devices AD1986A" "HDA:11d41986,104381b3,00100500" "0x1043" "0x81cb" Hardware is initialized using a guess method htpc:~$ cat /proc/asound/cards 0 [NVidia ]: HDA-Intel - HDA NVidia HDA NVidia at 0xfe024000 irq 22 htpc:~$ cat /proc/asound/devices 0: [ 0] : control 16: [ 0- 0]: digital audio playback 17: [ 0- 1]: digital audio playback 24: [ 0- 0]: digital audio capture 33: : timer alsamixer shows: Card: HDA NVidia Chip: Analog Devices AD1986A Master Headphon PCM Front Surround Center LFE Line CD Mic Mic Boos IEC958 IEC958 D Aux Mono Beep Channel Stereo D (everything is unmuted except stereo downmix) htpc:~$ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: NVidia [HDA NVidia], device 0: AD198x Analog [AD198x Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 card 0: NVidia [HDA NVidia], device 1: AD198x Digital [AD198x Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 htpc:~$ aplay -L default:CARD=NVidia HDA NVidia, AD198x Analog Default Audio Device front:CARD=NVidia,DEV=0 HDA NVidia, AD198x Analog Front speakers surround40:CARD=NVidia,DEV=0 HDA NVidia, AD198x Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=NVidia,DEV=0 HDA NVidia, AD198x Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=NVidia,DEV=0 HDA NVidia, AD198x Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=NVidia,DEV=0 HDA NVidia, AD198x Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=NVidia,DEV=0 HDA NVidia, AD198x Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=NVidia,DEV=0 HDA NVidia, AD198x Digital IEC958 (S/PDIF) Digital Audio Output null Discard all samples (playback) or generate zero samples (capture) htpc:~$ cat /proc/asound/card0/codec#0 Codec: Analog Devices AD1986A Address: 0 Function Id: 0x1 Vendor Id: 0x11d41986 Subsystem Id: 0x104381b3 Revision Id: 0x100500 No Modem Function Group found Default PCM: rates [0x7f]: 8000 11025 16000 22050 32000 44100 48000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Default Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 GPIO: io=0, o=1, i=0, unsolicited=0, wake=0 Node 0x02 [Audio Output] wcaps 0x30311: Stereo Digital Converter: stream=5, channel=0 Digital: Enabled GenLevel Digital category: 0x2 PCM: rates [0x60]: 44100 48000 bits [0x2]: 16 formats [0x5]: PCM AC3 Delay: 3 samples Connection: 2 0x01* 0x06 Node 0x03 [Audio Output] wcaps 0x44d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x17 0x17] Converter: stream=5, channel=0 Power: setting=D0, actual=D0 Processing caps: benign=1, ncoeff=70 Node 0x04 [Audio Output] wcaps 0x40d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x17 0x17] Converter: stream=0, channel=0 Power: setting=D0, actual=D0 Node 0x05 [Audio Output] wcaps 0x40d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x17 0x17] Converter: stream=12, channel=0 Power: setting=D0, actual=D0 Node 0x06 [Audio Input] wcaps 0x100511: Stereo Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x7f]: 8000 11025 16000 22050 32000 44100 48000 bits [0x6]: 16 20 formats [0x1]: PCM Power: setting=D0, actual=D0 Connection: 1 0x12 Node 0x07 [Audio Mixer] wcaps 0x200101: Stereo Connection: 8 0x03 0x09 0x13 0x14 0x15 0x16 0x17 0x18 Node 0x08 [Audio Mixer] wcaps 0x200100: Mono Connection: 1 0x07 Node 0x09 [Audio Mixer] wcaps 0x20010e: Mono Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80] [0x80] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80] Connection: 2 0x04 0x05 Node 0x0a [Audio Selector] wcaps 0x300101: Stereo Connection: 3 0x07* 0x04 0x05 Node 0x0b [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x07* 0x04 Node 0x0c [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x04* 0x07 Node 0x0d [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x05* 0x08 Node 0x0e [Audio Selector] wcaps 0x300100: Mono Connection: 2 0x08* 0x11 Node 0x0f [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-Out vals: [0x00 0x00] Connection: 8 0x1f 0x20 0x1d 0x1d 0x27* 0x28 0x29 0x2a Node 0x10 [Audio Selector] wcaps 0x300101: Stereo Connection: 3 0x20 0x1c* 0x1f Node 0x11 [Audio Selector] wcaps 0x300941: Stereo R/L Connection: 2 0x0f* 0x2b Processing caps: benign=1, ncoeff=0 Node 0x12 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 Amp-Out vals: [0x08 0x08] Connection: 8 0x11* 0x22 0x00 0x21 0x10 0x07 0x08 0x23 Node 0x13 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x80 0x80] Connection: 1 0x11 Node 0x14 [Audio Selector] wcaps 0x30010c: Mono Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x80] Connection: 1 0x23 Node 0x15 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x17 0x17] Connection: 1 0x22 Node 0x16 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x80 0x80] Connection: 1 0x21 Node 0x17 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x80 0x80] Connection: 1 0x10 Node 0x18 [Audio Selector] wcaps 0x30010c: Mono Amp-Out Amp-Out caps: ofs=0x0f, nsteps=0x0f, stepsize=0x0b, mute=1 Amp-Out vals: [0x80] Connection: 2 0x19* 0x24 Node 0x19 [Beep Generator Widget] wcaps 0x700000: Mono Node 0x1a [Pin Complex] wcaps 0x400185: Stereo Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x11 0x11] Pincap 0x0000001f: OUT HP Detect Trigger ImpSense Pin Default 0x02214021: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x2, Sequence = 0x1 Pin-ctls: 0xc0: OUT HP Unsolicited: tag=00, enabled=0 Connection: 1 0x0a Node 0x1b [Pin Complex] wcaps 0x400185: Stereo Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x11 0x11] Pincap 0x0001001f: OUT HP EAPD Detect Trigger ImpSense EAPD 0x2: EAPD Pin Default 0x01014011: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x1 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x0b Node 0x1c [Pin Complex] wcaps 0x400185: Stereo Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x91 0x91] Pincap 0x00000037: IN OUT Detect Trigger ImpSense Pin Default 0x01013012: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x1, Sequence = 0x2 Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Connection: 1 0x0c Node 0x1d [Pin Complex] wcaps 0x400985: Stereo Amp-Out R/L Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x11 0x11] Pincap 0x00001737: IN OUT Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x01019015: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x1, Sequence = 0x5 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Connection: 1 0x0d Node 0x1e [Pin Complex] wcaps 0x400104: Mono Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x03] Pincap 0x00000010: OUT Pin Default 0x501700f0: [N/A] Speaker at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Connection: 1 0x0e Node 0x1f [Pin Complex] wcaps 0x400081: Stereo Pincap 0x00001727: IN Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x02a190f0: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Node 0x20 [Pin Complex] wcaps 0x400081: Stereo Pincap 0x00001727: IN Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x018130f0: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Node 0x21 [Pin Complex] wcaps 0x400081: Stereo Pincap 0x00000027: IN Detect Trigger ImpSense Pin Default 0x509700f0: [N/A] Aux at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Node 0x22 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x993310f0: [Fixed] CD at Int ATAPI Conn = ATAPI, Color = Black DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x20: IN Node 0x23 [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0x50b700f0: [N/A] Telephony at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x20: IN Node 0x24 [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0x90f700f0: [Fixed] Other at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x20: IN Node 0x25 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x014510f0: [Jack] SPDIF Out at Ext Rear Conn = Optical, Color = Black DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Connection: 1 0x02 Node 0x26 [Power Widget] wcaps 0x500500: Mono Power: setting=D0, actual=D0 Connection: 8 0x07 0x08 0x13 0x14 0x15 0x16 0x17 0x18 Node 0x27 [Audio Mixer] wcaps 0x200101: Stereo Connection: 2 0x1f 0x1d Node 0x28 [Audio Mixer] wcaps 0x200101: Stereo Connection: 2 0x1f 0x20 Node 0x29 [Audio Mixer] wcaps 0x200101: Stereo Connection: 2 0x1d 0x20 Node 0x2a [Audio Mixer] wcaps 0x200101: Stereo Connection: 3 0x1f 0x1d 0x20 Node 0x2b [Audio Mixer] wcaps 0x200100: Mono Connection: 1 0x0f

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[alsa-devel] [Patch] hda-intel: Add quirks for RTL888 & RV630/M76 based MSI GX710
by William Weston 08 Jul '09

08 Jul '09

Hello, This patch is for yet another MSI laptop that needs the quirk to init the autosense on the headphone jack properly. Best Regards, --ww diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index bf4b78a..4301680 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9111,6 +9111,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1462, 0x3fc3, "MSI", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x3fcc, "MSI", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x3fdf, "MSI", ALC883_TARGA_DIG), + SND_PCI_QUIRK(0x1462, 0x42cd, "MSI GX710", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x4314, "MSI", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x4319, "MSI", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x4324, "MSI", ALC883_TARGA_DIG), @@ -9124,6 +9125,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1462, 0x7327, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7350, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0xa422, "MSI", ALC883_TARGA_2ch_DIG), + SND_PCI_QUIRK(0x1462, 0xaa08, "MSI GX710", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x147b, 0x1083, "Abit IP35-PRO", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1558, 0x0721, "Clevo laptop M720R", ALC883_CLEVO_M720), SND_PCI_QUIRK(0x1558, 0x0722, "Clevo laptop M720SR", ALC883_CLEVO_M720),

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[alsa-devel] [PATCH]HDA: Disable AMD SB600 64bit address support only
by Andiry Brienza 08 Jul '09

08 Jul '09

HDA driver disabled HD audio 64bit address support for all AMD SB600/SB700/SB800 platforms with commit 09240cf429505891d6123ce14a29f58f2a60121e due to one SB600 issue reported by community, but we do not see the similar issue on SB700/SB800 platforms. This patch is to refine the workaround for SB600 only. Signed-off-by: Andiry Xu <andiry.xu(a)amd.com> diff -ruN a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c --- a/sound/pci/hda/hda_intel.c 2009-07-01 17:14:23.000000000 +0800 +++ b/sound/pci/hda/hda_intel.c 2009-07-08 10:24:35.000000000 +0800 @@ -2322,9 +2322,19 @@ gcap = azx_readw(chip, GCAP); snd_printdd(SFX "chipset global capabilities = 0x%x\n", gcap); - /* ATI chips seems buggy about 64bit DMA addresses */ - if (chip->driver_type == AZX_DRIVER_ATI) - gcap &= ~ICH6_GCAP_64OK; + /* disable SB600 64bit support for safety */ + if ((chip->driver_type == AZX_DRIVER_ATI) || + (chip->driver_type == AZX_DRIVER_ATIHDMI)) { + struct pci_dev *p_smbus; + p_smbus = pci_get_device(PCI_VENDOR_ID_ATI, + PCI_DEVICE_ID_ATI_SBX00_SMBUS, + NULL); + if (p_smbus) { + if (p_smbus->revision < 0x30) + gcap &= ~ICH6_GCAP_64OK; + pci_dev_put(p_smbus); + } + } /* allow 64bit DMA address if supported by H/W */ if ((gcap & ICH6_GCAP_64OK) && !pci_set_dma_mask(pci, DMA_BIT_MASK(64)))

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[alsa-devel] [PATCH] sound dapm: fix check for external widgets bug (based on linux-2.6.30.1)
by Rongrong Cao 07 Jul '09

07 Jul '09

From: Rongrong Cao rrcao(a)ambarella.com This patch is based on linux-2.6.30.1 In SOC DAPM layer of SOUND subsystem, when add signal route (in the function snd_soc_dapm_add_route() ), the original code has wrong logic when dapm layer check each widget whether an external one. --------------------------------------------------------------------------------------------------------- --- linux-2.6.30.1/sound/soc/soc-dapm.c.orig 2009-07-06 19:48:43.000000000 +0800 +++ linux-2.6.30.1/sound/soc/soc-dapm.c 2009-07-06 19:03:15.000000000 +0800 @@ -983,8 +983,8 @@ static int snd_soc_dapm_add_route(struct if (wsink->id == snd_soc_dapm_input) { if (wsource->id == snd_soc_dapm_micbias || wsource->id == snd_soc_dapm_mic || - wsink->id == snd_soc_dapm_line || - wsink->id == snd_soc_dapm_output) + wsource->id == snd_soc_dapm_line || + wsource->id == snd_soc_dapm_output) wsink->ext = 1; } if (wsource->id == snd_soc_dapm_output) { --------------------------------------------------------------------------------------------------------- Signed-off-by: Rongrong Cao rrcao(a)ambarella.com ******************** Best regards! Allen (Rongrong) Cao Ambarella Shanghai Ltd. Tel: (86-21) 50942311 Fax: (86-21) 50942321 ---------------------------------------------------------------------------------------------------------------------------------- This email message (including attachments) contains Ambarella Proprietary and/or Confidential Information and is intended only for the addressee. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply email immediately and destroy all copies of the original message. ----------------------------------------------------------------------------------------------------------------------------------

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[alsa-devel] ALSA SoC driver playback twice as fast as desired
by Louis Lee 07 Jul '09

07 Jul '09

I've been developing an ALSA SoC driver for the past 3 weeks. The codec is a wolfsom WM8973 on a Freescale i.MX27 development platform. I've added SPI support to the existing (using i.MX27 platform SPI API -- not exactly portable) codec in the GIT repository. In addition I wrote up the DAI (I2S over SSI bus) and the PCM portion of the driver which handles DMA transfers. I written the driver to reduce the number of factors to reduce the number of factors that would be causing any problems. The files being played are all encoded for 48kHz. The PLLs, MCLK and BCLK settings are all set to assume a sample rate of 48KHz. Even the file is a mono file. However, when attempting to play the file back using SoX or aplay a self generated sine file made to play for 5 seconds always plays twice as fast. The windows chimes file for instance (which is a 2 x 16 bit audio file) when played as a single channel file -- essentially doubling the play time (0.6 to 1.2 seconds) -- will end up playing properly. The timer displayed by SoX is correct for that file. However, when the file is played back it is obviously being pumped through too fast. I'm out of ideas as to where to look. The DAI interface is set as a I2S slave. The clock is mastered by the Codec (WM8974). I've set up min_channels and max_channels to be 1 whereever possible. On a scope the clocking seems to be at the right speed. Why is it sound is playing twice as fast? Any ideas? Any help is appreciated. WM8974 is a mono codec. The SSI bus on the i.MX27 is set to I2S slave mode with a 16 bit word length. The word length on the codec is set to 16 bits. The audio interface on the codec is set to I2S, sample rate coded to do only 48kHz. THANK YOU _________________________________________________________________ We are your photos. Share us now with Windows Live Photos. http://go.microsoft.com/?linkid=9666047

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[alsa-devel] [PATCH] cmi8330: Allow MPU-401-less operation
by Ondrej Zary 07 Jul '09

07 Jul '09

Adding MPU-401 support to cmi8330 driver could cause a regression (non-working sound) on a system where there is no free IRQ for the MPU-401 device (which is not very uncommon as this card requires two separate IRQs plus a third one for MPU-401). When MPU-401 PnP configuration fails (mostly because of unavailable IRQ), just ignore MPU-401 and continue without it. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- linux-test/sound/isa/cmi8330.c 2009-07-06 22:39:24.000000000 +0200 +++ linux-2.6.30-pentium/sound/isa/cmi8330.c 2009-07-06 22:39:09.000000000 +0200 @@ -389,12 +389,12 @@ pdev = acard->mpu; err = pnp_activate_dev(pdev); - if (err < 0) { - snd_printk(KERN_ERR "MPU-401 PnP configure failure\n"); - return -EBUSY; + if (err < 0) + snd_printk(KERN_ERR "MPU-401 PnP configure failure: will be disabled\n"); + else { + mpuport[dev] = pnp_port_start(pdev, 0); + mpuirq[dev] = pnp_irq(pdev, 0); } - mpuport[dev] = pnp_port_start(pdev, 0); - mpuirq[dev] = pnp_irq(pdev, 0); return 0; } #endif -- Ondrej Zary

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Re: [alsa-devel] Intel HDA using VIA vt1708s
by Takashi Iwai 07 Jul '09

07 Jul '09

At Tue, 7 Jul 2009 15:49:30 +0200, Chrissi k wrote: > > Nope, no luck recording anything. changing the input source doesn't seem to > change anything at all. When I do "arecord..." using "stereo mix", I get a > peak from the external mic-in, same goes for "mic" and "front mic". > I also noticed, the external mic-in does not monitor, e.g. I can't directly > hear the signal, although it should play, as "mic" and "front" are both turned > on in the playback tab. > > here's an alsa-info, taken during recording. Thanks. Looks like the driver doesn't work well for selecting the capture source. Will check tomorrow. (Meanwhile you can change the connection selection of the widget 0x17 via hda-verb to choose the capture source manually.) Takashi

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