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

Re: [alsa-devel] Plugging in headphones doesn't mute speakers, internal microphone does not work
by Takashi Iwai 04 Mar '09

04 Mar '09

At Tue, 3 Mar 2009 15:18:13 -0200, Emilio López wrote: > > Hi, > > 'Side' is the subwoofer on my laptop (my laptop has an integrated subwoofer, > believe it or not. Hmm, then it's a wrong assignment. "Side" shouldn't be LFE. > Acer calls it "Tuba cinebass" or something like that). > Apart from that, the PC has a headphones/SPDIF jack, so I believe it supports > 7.1? No, it's totally irrelevant. (SPDIF is 2-channel only, BTW. Multi-channel is done via compressed formats, but the physical channels are only two.) Takashi > > 2009/3/3 Takashi Iwai <tiwai(a)suse.de> > > At Mon, 2 Mar 2009 21:46:04 -0200, > Emilio López wrote: > > > > 2009/3/2 Takashi Iwai <tiwai(a)suse.de> > > > > At Sat, 28 Feb 2009 14:38:10 -0200, > > Emilio López wrote: > > > > > > Hello, > > > > > > This is my first mail to the list, I hope you can help me. I have > been > > > talking with people on #alsa, but they couldn't help me, so that's > why > > > I'm writing to you. > > > > > > Well, as lspci says, my Acer Aspire 6930 laptop has an "Intel > > > Corporation 82801I (ICH9 Family) HD Audio Controller (rev 03)". > With > > > this card, and without adding any mode=... to > > > /etc/modprobe.d/alsa-base, playback works perfectly, but when I > plug > > > in headphones, the integrated speakers do not mute. I can mute the > > > speakers muting "front" and "side" and use the headphones, but it > > > should be automatic. > > > > > > Setting mode=auto makes the headphones, when plugged, mute the > > > speakers. But the problem with this is, that when using the > speakers, > > > I only get sound from the speaker that was "front" in the other > mode. > > > The microphone does not work in this mode either. > > > > > > I also tried other modes, including (but not limited to) acer, > > > acer-aspire. I ran alsa-info.sh, here is the URL: > > > http://www.alsa-project.org/db/?f= > > 7910da439e8e190a8c2a6f18167d4879c74881c9 > > > > > > And before I forget, regarding the microphone, if I plug an > external > > > one on the microphone plug, I can record. And I tried the internal > one > > > on windows, and it works, so I believe it's an alsa problem. > > > > > > Looking forward to hearing from you! > > > > Could you check the very latest sound git tree or alsa-driver > snapshot > > below, first without model option? > > git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git > > ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/ > > alsa-driver-snapshot.tar.gz > > > > Hi, > > > > I tried with the alsa-driver snapshot without any model option, and now > the > > headphones mute the speakers. But the internal microphone doesn't still > work, > > and 'Side' doesn't output sound like it did before, even if it's unmuted > and > > with volume at 100% (with the old version and without any model option, > it > > worked) > > What "Side" volume would do exactly? It's for 7.1 output. > Does the machine have 7.1 output? > > Takashi > >

2 4

[alsa-devel] S3C64XX: Add s3c64xx-i2s support
by Ben Dooks 04 Mar '09

04 Mar '09

From: Ben Dooks <ben(a)simtec.co.uk> Add the initial code to support the S3C64XX I2S hardware using the s3c-i2s-v2 core code. Signed-off-by: Ben Dooks <ben(a)simtec.co.uk> Index: linux.git/sound/soc/s3c24xx/Makefile =================================================================== --- linux.git.orig/sound/soc/s3c24xx/Makefile 2009-03-04 00:44:28.000000000 +0000 +++ linux.git/sound/soc/s3c24xx/Makefile 2009-03-04 00:56:40.000000000 +0000 @@ -2,6 +2,7 @@ snd-soc-s3c24xx-objs := s3c24xx-pcm.o snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o +snd-soc-s3c64xx-i2s-objs := s3c64xx-i2s.o snd-soc-s3c2443-ac97-objs := s3c2443-ac97.o snd-soc-s3c-i2s-v2-objs := s3c-i2s-v2.o @@ -9,6 +10,7 @@ obj-$(CONFIG_SND_S3C24XX_SOC) += snd-soc obj-$(CONFIG_SND_S3C24XX_SOC_I2S) += snd-soc-s3c24xx-i2s.o obj-$(CONFIG_SND_S3C2443_SOC_AC97) += snd-soc-s3c2443-ac97.o obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o +obj-$(CONFIG_SND_S3C64XX_SOC_I2S) += snd-soc-s3c64xx-i2s.o obj-$(CONFIG_SND_S3C_I2SV2_SOC) += snd-soc-s3c-i2s-v2.o # S3C24XX Machine Support Index: linux.git/sound/soc/s3c24xx/s3c64xx-i2s.c =================================================================== --- /dev/null 1970-01-01 00:00:00.000000000 +0000 +++ linux.git/sound/soc/s3c24xx/s3c64xx-i2s.c 2009-03-04 00:58:09.000000000 +0000 @@ -0,0 +1,223 @@ +/* sound/soc/s3c24xx/s3c64xx-i2s.c + * + * ALSA SoC Audio Layer - S3C64XX I2S driver + * + * Copyright 2008 Openmoko, Inc. + * Copyright 2008 Simtec Electronics + * Ben Dooks <ben(a)simtec.co.uk> + * http://armlinux.simtec.co.uk/ + * + * Based on s3c24xx-i2s.c and s3c-i2c-v2.c + * (c) 2006 Wolfson Microelectronics PLC. + * + * 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/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/clk.h> +#include <linux/kernel.h> +#include <linux/gpio.h> +#include <linux/io.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include <plat/regs-s3c2412-iis.h> +#include <plat/gpio-bank-d.h> +#include <plat/gpio-bank-e.h> +#include <plat/gpio-cfg.h> +#include <plat/audio.h> + +#include <mach/map.h> +#include <mach/dma.h> + +#include "s3c24xx-pcm.h" +#include "s3c64xx-i2s.h" + +static struct s3c2410_dma_client s3c64xx_dma_client_out = { + .name = "I2S PCM Stereo out" +}; + +static struct s3c2410_dma_client s3c64xx_dma_client_in = { + .name = "I2S PCM Stereo in" +}; + +static struct s3c24xx_pcm_dma_params s3c64xx_i2s_pcm_stereo_out[2] = { + [0] = { + .channel = DMACH_I2S0_OUT, + .client = &s3c64xx_dma_client_out, + .dma_addr = S3C64XX_PA_IIS0 + S3C2412_IISTXD, + .dma_size = 4, + }, + [1] = { + .channel = DMACH_I2S1_OUT, + .client = &s3c64xx_dma_client_out, + .dma_addr = S3C64XX_PA_IIS1 + S3C2412_IISTXD, + .dma_size = 4, + }, +}; + +static struct s3c24xx_pcm_dma_params s3c64xx_i2s_pcm_stereo_in[2] = { + [0] = { + .channel = DMACH_I2S0_IN, + .client = &s3c64xx_dma_client_in, + .dma_addr = S3C64XX_PA_IIS0 + S3C2412_IISRXD, + .dma_size = 4, + }, + [1] = { + .channel = DMACH_I2S1_IN, + .client = &s3c64xx_dma_client_in, + .dma_addr = S3C64XX_PA_IIS1 + S3C2412_IISRXD, + .dma_size = 4, + }, +}; + +static struct s3c_i2sv2_info s3c64xx_i2s[2]; + +static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai) +{ + return cpu_dai->private_data; +} + +static int s3c64xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + struct s3c_i2sv2_info *i2s = to_info(cpu_dai); + u32 iismod = readl(i2s->regs + S3C2412_IISMOD); + + switch (clk_id) { + case S3C64XX_CLKSRC_PCLK: + iismod &= ~S3C64XX_IISMOD_IMS_SYSMUX; + break; + + case S3C64XX_CLKSRC_MUX: + iismod |= S3C64XX_IISMOD_IMS_SYSMUX; + break; + + default: + return -EINVAL; + } + + writel(iismod, i2s->regs + S3C2412_IISMOD); + + return 0; +} + + +unsigned long s3c64xx_i2s_get_clockrate(struct snd_soc_dai *dai) +{ + struct s3c_i2sv2_info *i2s = to_info(dai); + + return clk_get_rate(i2s->iis_cclk); +} +EXPORT_SYMBOL_GPL(s3c64xx_i2s_get_clockrate); + +static int s3c64xx_i2s_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + struct device *dev = &pdev->dev; + struct s3c_i2sv2_info *i2s; + int ret; + + dev_dbg(dev, "%s: probing dai %d\n", __func__, pdev->id); + + if (pdev->id < 0 || pdev->id > ARRAY_SIZE(s3c64xx_i2s)) { + dev_err(dev, "id %d out of range\n", pdev->id); + return -EINVAL; + } + + i2s = &s3c64xx_i2s[pdev->id]; + + ret = s3c_i2sv2_probe(pdev, dai, i2s, + pdev->id ? S3C64XX_PA_IIS1 : S3C64XX_PA_IIS0); + if (ret) + return ret; + + i2s->dma_capture = &s3c64xx_i2s_pcm_stereo_in[pdev->id]; + i2s->dma_playback = &s3c64xx_i2s_pcm_stereo_out[pdev->id]; + + i2s->iis_cclk = clk_get(dev, "audio-bus"); + if (IS_ERR(i2s->iis_cclk)) { + dev_err(dev, "failed to get audio-bus"); + iounmap(i2s->regs); + return -ENODEV; + } + + /* configure GPIO for i2s port */ + switch (pdev->id) { + case 0: + s3c_gpio_cfgpin(S3C64XX_GPD(0), S3C64XX_GPD0_I2S0_CLK); + s3c_gpio_cfgpin(S3C64XX_GPD(1), S3C64XX_GPD1_I2S0_CDCLK); + s3c_gpio_cfgpin(S3C64XX_GPD(2), S3C64XX_GPD2_I2S0_LRCLK); + s3c_gpio_cfgpin(S3C64XX_GPD(3), S3C64XX_GPD3_I2S0_DI); + s3c_gpio_cfgpin(S3C64XX_GPD(4), S3C64XX_GPD4_I2S0_D0); + break; + case 1: + s3c_gpio_cfgpin(S3C64XX_GPE(0), S3C64XX_GPE0_I2S1_CLK); + s3c_gpio_cfgpin(S3C64XX_GPE(1), S3C64XX_GPE1_I2S1_CDCLK); + s3c_gpio_cfgpin(S3C64XX_GPE(2), S3C64XX_GPE2_I2S1_LRCLK); + s3c_gpio_cfgpin(S3C64XX_GPE(3), S3C64XX_GPE3_I2S1_DI); + s3c_gpio_cfgpin(S3C64XX_GPE(4), S3C64XX_GPE4_I2S1_D0); + } + + return 0; +} + + +#define S3C64XX_I2S_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 S3C64XX_I2S_FMTS \ + (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE) + +struct snd_soc_dai s3c64xx_i2s_dai = { + .name = "s3c64xx-i2s", + .id = 0, + .probe = s3c64xx_i2s_probe, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = S3C64XX_I2S_RATES, + .formats = S3C64XX_I2S_FMTS, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = S3C64XX_I2S_RATES, + .formats = S3C64XX_I2S_FMTS, + }, + .ops = { + .set_sysclk = s3c64xx_i2s_set_sysclk, + }, +}; +EXPORT_SYMBOL_GPL(s3c64xx_i2s_dai); + +static int __init s3c64xx_i2s_init(void) +{ + return s3c_i2sv2_register_dai(&s3c64xx_i2s_dai); +} +module_init(s3c64xx_i2s_init); + +static void __exit s3c64xx_i2s_exit(void) +{ + snd_soc_unregister_dai(&s3c64xx_i2s_dai); +} +module_exit(s3c64xx_i2s_exit); + +/* Module information */ +MODULE_AUTHOR("Ben Dooks, <ben(a)simtec.co.uk>"); +MODULE_DESCRIPTION("S3C64XX I2S SoC Interface"); +MODULE_LICENSE("GPL"); + + + Index: linux.git/sound/soc/s3c24xx/s3c64xx-i2s.h =================================================================== --- /dev/null 1970-01-01 00:00:00.000000000 +0000 +++ linux.git/sound/soc/s3c24xx/s3c64xx-i2s.h 2009-03-04 00:44:31.000000000 +0000 @@ -0,0 +1,31 @@ +/* sound/soc/s3c24xx/s3c64xx-i2s.h + * + * ALSA SoC Audio Layer - S3C64XX I2S driver + * + * Copyright 2008 Openmoko, Inc. + * Copyright 2008 Simtec Electronics + * Ben Dooks <ben(a)simtec.co.uk> + * http://armlinux.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. + */ + +#ifndef __SND_SOC_S3C24XX_S3C64XX_I2S_H +#define __SND_SOC_S3C24XX_S3C64XX_I2S_H __FILE__ + +#include "s3c-i2s-v2.h" + +#define S3C64XX_DIV_BCLK S3C_I2SV2_DIV_BCLK +#define S3C64XX_DIV_RCLK S3C_I2SV2_DIV_RCLK +#define S3C64XX_DIV_PRESCALER S3C_I2SV2_DIV_PRESCALER + +#define S3C64XX_CLKSRC_PCLK (0) +#define S3C64XX_CLKSRC_MUX (1) + +extern struct snd_soc_dai s3c64xx_i2s_dai; + +extern unsigned long s3c64xx_i2s_get_clockrate(struct snd_soc_dai *cpu_dai); + +#endif /* __SND_SOC_S3C24XX_S3C64XX_I2S_H */ Index: linux.git/sound/soc/s3c24xx/Kconfig =================================================================== --- linux.git.orig/sound/soc/s3c24xx/Kconfig 2009-03-04 00:44:28.000000000 +0000 +++ linux.git/sound/soc/s3c24xx/Kconfig 2009-03-04 00:56:40.000000000 +0000 @@ -1,6 +1,6 @@ config SND_S3C24XX_SOC tristate "SoC Audio for the Samsung S3C24XX chips" - depends on ARCH_S3C2410 + depends on ARCH_S3C2410 || ARCH_S3C64XX help Say Y or M if you want to add support for codecs attached to the S3C24XX AC97, I2S or SSP interface. You will also need @@ -16,6 +16,10 @@ config SND_S3C2412_SOC_I2S tristate select SND_S3C_I2SV2_SOC +config SND_S3C64XX_SOC_I2S + tristate + select SND_S3C_I2SV2_SOC + config SND_S3C2443_SOC_AC97 tristate select AC97_BUS Index: linux.git/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h =================================================================== --- linux.git.orig/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h 2009-03-04 00:44:10.000000000 +0000 +++ linux.git/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h 2009-03-04 00:44:31.000000000 +0000 @@ -33,6 +33,9 @@ #define S3C2412_IISCON_RXDMA_ACTIVE (1 << 1) #define S3C2412_IISCON_IIS_ACTIVE (1 << 0) +#define S3C64XX_IISMOD_IMS_PCLK (0 << 10) +#define S3C64XX_IISMOD_IMS_SYSMUX (1 << 10) + #define S3C2412_IISMOD_MASTER_INTERNAL (0 << 10) #define S3C2412_IISMOD_MASTER_EXTERNAL (1 << 10) #define S3C2412_IISMOD_SLAVE (2 << 10) -- Ben (ben(a)fluff.org, http://www.fluff.org/) 'a smiley only costs 4 bytes'

1 0

[alsa-devel] [PATCH] ASoC: Add GPIO support for jack reporting interface
by Lopez Cruz, Misael 03 Mar '09

03 Mar '09

Add GPIO support to jack reporting framework in ASoC using gpiolib calls. The gpio support exports two new functions: snd_soc_jack_add_gpios and snd_soc_jack_free_gpios. Client drivers using gpio feature must pass an array of jack_gpio pins belonging to a specific jack to the snd_soc_jack_add_gpios function. The framework will request the gpio, set the data direction and request irq. The framework will update power status of related jack_pins when an event on the gpio pins comes according to the reporting bits defined for each gpio. All gpio resources allocated when adding jack_gpio pins for a particular jack can be released using snd_soc_jack_free_gpios function. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- include/sound/soc.h | 27 +++++++++++ sound/soc/soc-jack.c | 123 ++++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 150 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 68d8149..ad0466b 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -16,6 +16,8 @@ #include <linux/platform_device.h> #include <linux/types.h> #include <linux/workqueue.h> +#include <linux/interrupt.h> +#include <linux/kernel.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/control.h> @@ -168,6 +170,7 @@ struct soc_enum; struct snd_soc_ac97_ops; struct snd_soc_jack; struct snd_soc_jack_pin; +struct snd_soc_jack_gpio; typedef int (*hw_write_t)(void *,const char* ,int); typedef int (*hw_read_t)(void *,char* ,int); @@ -194,6 +197,9 @@ int snd_soc_jack_new(struct snd_soc_card *card, const char *id, int type, void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask); int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count, struct snd_soc_jack_pin *pins); +int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, + struct snd_soc_jack_gpio *gpios); +void snd_soc_jack_free_gpios(struct snd_soc_jack *jack); /* codec IO */ #define snd_soc_read(codec, reg) codec->read(codec, reg) @@ -262,10 +268,31 @@ struct snd_soc_jack_pin { bool invert; }; +/** + * struct snd_soc_jack_gpio - Describes a gpio pin for jack detection + * + * @gpio: gpio number + * @name: gpio name + * @report: value to report when jack detected + * @invert: report presence in low state + * @debouce_time: debouce time in ms + */ +struct snd_soc_jack_gpio { + unsigned int gpio; + const char *name; + int report; + int invert; + int debounce_time; + struct snd_soc_jack *jack; + struct work_struct work; +}; + struct snd_soc_jack { struct snd_jack *jack; struct snd_soc_card *card; struct list_head pins; + struct snd_soc_jack_gpio *gpios; + int gpio_count; int status; }; diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 8cc00c3..0e9ee28 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -14,6 +14,10 @@ #include <sound/jack.h> #include <sound/soc.h> #include <sound/soc-dapm.h> +#include <linux/gpio.h> +#include <linux/interrupt.h> +#include <linux/workqueue.h> +#include <linux/delay.h> /** * snd_soc_jack_new - Create a new jack @@ -136,3 +140,122 @@ int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count, return 0; } EXPORT_SYMBOL_GPL(snd_soc_jack_add_pins); + +/* gpio detect */ +void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio) +{ + struct snd_soc_jack *jack = gpio->jack; + int enable; + int report; + + if (gpio->debounce_time > 0) + mdelay(gpio->debounce_time); + + enable = gpio_get_value(gpio->gpio); + if (gpio->invert) + enable != enable; + + if (enable) + report = gpio->report; + else + report = 0; + + snd_soc_jack_report(jack, report, gpio->report); +} + +/* irq handler for gpio pin */ +static irqreturn_t gpio_handler(int irq, void *data) +{ + struct snd_soc_jack_gpio *gpio = data; + + return IRQ_RETVAL(schedule_work(&gpio->work)); +} + +/* gpio work */ +static void gpio_work(struct work_struct *work) +{ + struct snd_soc_jack_gpio *gpio; + + gpio = container_of(work, struct snd_soc_jack_gpio, work); + snd_soc_jack_gpio_detect(gpio); +} + +/** + * snd_soc_jack_add_gpios - Associate GPIO pins with an ASoC jack + * + * @jack: ASoC jack + * @count: number of pins + * @gpios: array of gpio pins + * + * This function will request gpio, set data direction and request irq + * for each gpio in the array. Valid gpio pins references will be saved. + */ +int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, + struct snd_soc_jack_gpio *gpios) +{ + int i, ret = 0; + + /* Link gpio array with the soc_jack */ + if (count > 0) + jack->gpios = gpios; + + for (i = 0; i < count; i++) { + if (!gpio_is_valid(gpios[i].gpio)) { + printk(KERN_ERR "Invalid gpio %d\n", + gpios[i].gpio); + return -EINVAL; + } + if (!gpios[i].name) { + printk(KERN_ERR "No name for gpio %d\n", + gpios[i].gpio); + return -EINVAL; + } + + ret = gpio_request(gpios[i].gpio, gpios[i].name); + if (ret) + return ret; + + ret = gpio_direction_input(gpios[i].gpio); + if (ret) + break; + + ret = request_irq(gpio_to_irq(gpios[i].gpio), + gpio_handler, + IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING, + jack->card->dev->driver->name, + &gpios[i]); + if (ret) + break; + + INIT_WORK(&gpios[i].work, gpio_work); + gpios[i].jack = jack; + } + + if (ret) + gpio_free(gpios[i].gpio); + + jack->gpio_count = i; /* gpios allocated properly */ + + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_jack_add_gpios); + +/** + * snd_soc_jack_free_gpios - Release GPIO pins' resources of an ASoC jack + * + * @jack: ASoC jack + * @count: number of pins + * + * Release gpio and irq resources for gpio pins associated with an ASoC jack. + */ +void snd_soc_jack_free_gpios(struct snd_soc_jack *jack) +{ + struct snd_soc_jack_gpio *gpios = jack->gpios; + int i; + + for (i = 0; i < jack->gpio_count; i++) { + free_irq(gpio_to_irq(gpios[i].gpio), &gpios[i]); + gpio_free(gpios[i].gpio); + } +} +EXPORT_SYMBOL_GPL(snd_soc_jack_free_gpios); -- 1.5.4.3

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[alsa-devel] PCM File plugin - minor fix
by Pavel Hofman 03 Mar '09

03 Mar '09

Hi, After some consideration I decided to ask for changing the meaning of the %b key from bytes to bits as it provides more convenience. Thanks for commiting the patch. Pavel Hofman.

2 1

[PATCH] ALSA: Rewrite hw_ptr updaters
by Takashi Iwai 03 Mar '09

03 Mar '09

Clean up and improve snd_pcm_update_hw_ptr*() functions. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_lib.c | 125 +++++++++++++++++++++++++++++++++----------------- 1 files changed, 82 insertions(+), 43 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 9216910..7d13eee 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -125,19 +125,23 @@ void snd_pcm_playback_silence(struct snd_pcm_substream *substream, snd_pcm_ufram } } +#ifdef CONFIG_SND_PCM_XRUN_DEBUG +#define xrun_debug(substream) ((substream)->pstr->xrun_debug) +#else +#define xrun_debug(substream) 0 +#endif + static void xrun(struct snd_pcm_substream *substream) { snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); -#ifdef CONFIG_SND_PCM_XRUN_DEBUG - if (substream->pstr->xrun_debug) { + if (xrun_debug(substream)) { snd_printd(KERN_DEBUG "XRUN: pcmC%dD%d%c\n", substream->pcm->card->number, substream->pcm->device, substream->stream ? 'c' : 'p'); - if (substream->pstr->xrun_debug > 1) + if (xrun_debug(substream) > 1) dump_stack(); } -#endif } static inline snd_pcm_uframes_t snd_pcm_update_hw_ptr_pos(struct snd_pcm_substream *substream, @@ -182,11 +186,26 @@ static inline int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream return 0; } +static void hw_ptr_error(struct snd_pcm_substream *substream, + const char *fmt, ...) +{ + if (xrun_debug(substream)) { + if (printk_ratelimit()) { + va_list args; + va_start(args, fmt); + vprintk(fmt, args); + va_end(args); + if (xrun_debug(substream) > 1) + dump_stack(); + } + } +} + static inline int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_uframes_t pos; - snd_pcm_uframes_t new_hw_ptr, hw_ptr_interrupt; + snd_pcm_uframes_t new_hw_ptr, hw_ptr_interrupt, hw_base; snd_pcm_sframes_t delta; pos = snd_pcm_update_hw_ptr_pos(substream, runtime); @@ -194,36 +213,47 @@ static inline int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *subs xrun(substream); return -EPIPE; } - if (runtime->period_size == runtime->buffer_size) - goto __next_buf; - new_hw_ptr = runtime->hw_ptr_base + pos; + hw_base = runtime->hw_ptr_base; + new_hw_ptr = hw_base + pos; hw_ptr_interrupt = runtime->hw_ptr_interrupt + runtime->period_size; - - delta = hw_ptr_interrupt - new_hw_ptr; - if (delta > 0) { - if ((snd_pcm_uframes_t)delta < runtime->buffer_size / 2) { -#ifdef CONFIG_SND_PCM_XRUN_DEBUG - if (runtime->periods > 1 && substream->pstr->xrun_debug) { - snd_printd(KERN_ERR "Unexpected hw_pointer value [1] (stream = %i, delta: -%ld, max jitter = %ld): wrong interrupt acknowledge?\n", substream->stream, (long) delta, runtime->buffer_size / 2); - if (substream->pstr->xrun_debug > 1) - dump_stack(); - } -#endif - return 0; + delta = new_hw_ptr - hw_ptr_interrupt; + if (hw_ptr_interrupt == runtime->boundary) + hw_ptr_interrupt = 0; + if (delta < 0) { + 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); + /* rebase to interrupt position */ + hw_base = new_hw_ptr = hw_ptr_interrupt; + delta = 0; + } else { + hw_base += runtime->buffer_size; + if (hw_base == runtime->boundary) + hw_base = 0; + new_hw_ptr = hw_base + pos; } - __next_buf: - runtime->hw_ptr_base += runtime->buffer_size; - if (runtime->hw_ptr_base == runtime->boundary) - runtime->hw_ptr_base = 0; - new_hw_ptr = runtime->hw_ptr_base + pos; } - + if (delta > runtime->period_size) { + hw_ptr_error(substream, + "Lost interrupts? " + "(stream=%i, delta=%ld, intr_ptr=%ld)\n", + substream->stream, (long)delta, + (long)hw_ptr_interrupt); + /* rebase hw_ptr_interrupt */ + hw_ptr_interrupt = + new_hw_ptr - new_hw_ptr % runtime->period_size; + } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) snd_pcm_playback_silence(substream, new_hw_ptr); + runtime->hw_ptr_base = hw_base; runtime->status->hw_ptr = new_hw_ptr; - runtime->hw_ptr_interrupt = new_hw_ptr - new_hw_ptr % runtime->period_size; + runtime->hw_ptr_interrupt = hw_ptr_interrupt; return snd_pcm_update_hw_ptr_post(substream, runtime); } @@ -233,7 +263,7 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_uframes_t pos; - snd_pcm_uframes_t old_hw_ptr, new_hw_ptr; + snd_pcm_uframes_t old_hw_ptr, new_hw_ptr, hw_base; snd_pcm_sframes_t delta; old_hw_ptr = runtime->status->hw_ptr; @@ -242,29 +272,38 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream) xrun(substream); return -EPIPE; } - new_hw_ptr = runtime->hw_ptr_base + pos; - - delta = old_hw_ptr - new_hw_ptr; - if (delta > 0) { - if ((snd_pcm_uframes_t)delta < runtime->buffer_size / 2) { -#ifdef CONFIG_SND_PCM_XRUN_DEBUG - if (runtime->periods > 2 && substream->pstr->xrun_debug) { - snd_printd(KERN_ERR "Unexpected hw_pointer value [2] (stream = %i, delta: -%ld, max jitter = %ld): wrong interrupt acknowledge?\n", substream->stream, (long) delta, runtime->buffer_size / 2); - if (substream->pstr->xrun_debug > 1) - dump_stack(); - } -#endif + hw_base = runtime->hw_ptr_base; + new_hw_ptr = hw_base + pos; + + delta = new_hw_ptr - old_hw_ptr; + if (delta < 0) { + delta += runtime->buffer_size; + if (delta < 0) { + hw_ptr_error(substream, + "Unexpected hw_pointer value [2] " + "(stream=%i, pos=%ld, old_ptr=%ld)\n", + substream->stream, (long)pos, + (long)old_hw_ptr); return 0; } - runtime->hw_ptr_base += runtime->buffer_size; - if (runtime->hw_ptr_base == runtime->boundary) - runtime->hw_ptr_base = 0; - new_hw_ptr = runtime->hw_ptr_base + pos; + hw_base += runtime->buffer_size; + if (hw_base == runtime->boundary) + hw_base = 0; + new_hw_ptr = hw_base + pos; + } + if (delta > runtime->period_size && runtime->periods > 1) { + hw_ptr_error(substream, + "hw_ptr skipping! " + "(pos=%ld, delta=%ld, period=%ld)\n", + (long)pos, (long)delta, + (long)runtime->period_size); + return 0; } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) snd_pcm_playback_silence(substream, new_hw_ptr); + runtime->hw_ptr_base = hw_base; runtime->status->hw_ptr = new_hw_ptr; return snd_pcm_update_hw_ptr_post(substream, runtime); -- 1.6.1.3

1 0

Re: [alsa-devel] Plugging in headphones doesn't mute speakers, internal microphone does not work
by Takashi Iwai 03 Mar '09

03 Mar '09

At Mon, 2 Mar 2009 21:46:04 -0200, Emilio López wrote: > > 2009/3/2 Takashi Iwai <tiwai(a)suse.de> > > At Sat, 28 Feb 2009 14:38:10 -0200, > Emilio López wrote: > > > > Hello, > > > > This is my first mail to the list, I hope you can help me. I have been > > talking with people on #alsa, but they couldn't help me, so that's why > > I'm writing to you. > > > > Well, as lspci says, my Acer Aspire 6930 laptop has an "Intel > > Corporation 82801I (ICH9 Family) HD Audio Controller (rev 03)". With > > this card, and without adding any mode=... to > > /etc/modprobe.d/alsa-base, playback works perfectly, but when I plug > > in headphones, the integrated speakers do not mute. I can mute the > > speakers muting "front" and "side" and use the headphones, but it > > should be automatic. > > > > Setting mode=auto makes the headphones, when plugged, mute the > > speakers. But the problem with this is, that when using the speakers, > > I only get sound from the speaker that was "front" in the other mode. > > The microphone does not work in this mode either. > > > > I also tried other modes, including (but not limited to) acer, > > acer-aspire. I ran alsa-info.sh, here is the URL: > > http://www.alsa-project.org/db/?f= > 7910da439e8e190a8c2a6f18167d4879c74881c9 > > > > And before I forget, regarding the microphone, if I plug an external > > one on the microphone plug, I can record. And I tried the internal one > > on windows, and it works, so I believe it's an alsa problem. > > > > Looking forward to hearing from you! > > Could you check the very latest sound git tree or alsa-driver snapshot > below, first without model option? > git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git > ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/ > alsa-driver-snapshot.tar.gz > > Hi, > > I tried with the alsa-driver snapshot without any model option, and now the > headphones mute the speakers. But the internal microphone doesn't still work, > and 'Side' doesn't output sound like it did before, even if it's unmuted and > with volume at 100% (with the old version and without any model option, it > worked) What "Side" volume would do exactly? It's for 7.1 output. Does the machine have 7.1 output? Takashi

1 0

[alsa-devel] [PATCH] ASoC: make ops a pointer in 'struct snd_soc_dai'
by Eric Miao 03 Mar '09

03 Mar '09

Considering the fact that most cpu_dai or codec_dai are using a same 'snd_soc_dai_ops' for several similar interfaces, 'ops' would be better made a pointer instead, to make sharing easier and code a bit cleaner. The patch below is rather preliminary since the asoc tree is being actively developed, and this touches almost every piece of code, (and possibly many others in development need to be changed as well). Building of all codecs are OK, yet to every SoC, I didn't test that. Signed-off-by: Eric Miao <eric.miao(a)marvell.com> diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 24247f7..1367647 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -203,7 +203,7 @@ struct snd_soc_dai { int (*resume)(struct snd_soc_dai *dai); /* ops */ - struct snd_soc_dai_ops ops; + struct snd_soc_dai_ops *ops; /* DAI capabilities */ struct snd_soc_pcm_stream capture; diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index ff0054b..e588e63 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -697,6 +697,15 @@ static int atmel_ssc_resume(struct snd_soc_dai *cpu_dai) #define ATMEL_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) +static struct snd_soc_dai_ops atmel_ssc_dai_ops = { + .startup = atmel_ssc_startup, + .shutdown = atmel_ssc_shutdown, + .prepare = atmel_ssc_prepare, + .hw_params = atmel_ssc_hw_params, + .set_fmt = atmel_ssc_set_dai_fmt, + .set_clkdiv = atmel_ssc_set_dai_clkdiv, +}; + struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = { { .name = "atmel-ssc0", .id = 0, @@ -712,13 +721,7 @@ struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = { .channels_max = 2, .rates = ATMEL_SSC_RATES, .formats = ATMEL_SSC_FORMATS,}, - .ops = { - .startup = atmel_ssc_startup, - .shutdown = atmel_ssc_shutdown, - .prepare = atmel_ssc_prepare, - .hw_params = atmel_ssc_hw_params, - .set_fmt = atmel_ssc_set_dai_fmt, - .set_clkdiv = atmel_ssc_set_dai_clkdiv,}, + .ops = &atmel_ssc_dai_ops, .private_data = &ssc_info[0], }, #if NUM_SSC_DEVICES == 3 @@ -736,13 +739,7 @@ struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = { .channels_max = 2, .rates = ATMEL_SSC_RATES, .formats = ATMEL_SSC_FORMATS,}, - .ops = { - .startup = atmel_ssc_startup, - .shutdown = atmel_ssc_shutdown, - .prepare = atmel_ssc_prepare, - .hw_params = atmel_ssc_hw_params, - .set_fmt = atmel_ssc_set_dai_fmt, - .set_clkdiv = atmel_ssc_set_dai_clkdiv,}, + .ops = &atmel_ssc_dai_ops, .private_data = &ssc_info[1], }, { .name = "atmel-ssc2", @@ -759,13 +756,7 @@ struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = { .channels_max = 2, .rates = ATMEL_SSC_RATES, .formats = ATMEL_SSC_FORMATS,}, - .ops = { - .startup = atmel_ssc_startup, - .shutdown = atmel_ssc_shutdown, - .prepare = atmel_ssc_prepare, - .hw_params = atmel_ssc_hw_params, - .set_fmt = atmel_ssc_set_dai_fmt, - .set_clkdiv = atmel_ssc_set_dai_clkdiv,}, + .ops = &atmel_ssc_dai_ops, .private_data = &ssc_info[2], }, #endif diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index f0e30ae..479d7bd 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -342,6 +342,11 @@ static int au1xpsc_ac97_resume(struct snd_soc_dai *dai) return 0; } +static struct snd_soc_dai_ops au1xpsc_ac97_dai_ops = { + .trigger = au1xpsc_ac97_trigger, + .hw_params = au1xpsc_ac97_hw_params, +}; + struct snd_soc_dai au1xpsc_ac97_dai = { .name = "au1xpsc_ac97", .ac97_control = 1, @@ -361,10 +366,7 @@ struct snd_soc_dai au1xpsc_ac97_dai = { .channels_min = 2, .channels_max = 2, }, - .ops = { - .trigger = au1xpsc_ac97_trigger, - .hw_params = au1xpsc_ac97_hw_params, - }, + .ops = &au1xpsc_ac97_dai_ops, }; EXPORT_SYMBOL_GPL(au1xpsc_ac97_dai); diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index f916de4..bb58932 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -367,6 +367,12 @@ static int au1xpsc_i2s_resume(struct snd_soc_dai *cpu_dai) return 0; } +static struct snd_soc_dai_ops au1xpsc_i2s_dai_ops = { + .trigger = au1xpsc_i2s_trigger, + .hw_params = au1xpsc_i2s_hw_params, + .set_fmt = au1xpsc_i2s_set_fmt, +}; + struct snd_soc_dai au1xpsc_i2s_dai = { .name = "au1xpsc_i2s", .probe = au1xpsc_i2s_probe, @@ -385,11 +391,7 @@ struct snd_soc_dai au1xpsc_i2s_dai = { .channels_min = 2, .channels_max = 8, /* 2 without external help */ }, - .ops = { - .trigger = au1xpsc_i2s_trigger, - .hw_params = au1xpsc_i2s_hw_params, - .set_fmt = au1xpsc_i2s_set_fmt, - }, + .ops = &au1xpsc_i2s_dai_ops, }; EXPORT_SYMBOL(au1xpsc_i2s_dai); diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index d1d95d2..9648244 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -287,6 +287,13 @@ static int bf5xx_i2s_resume(struct platform_device *pdev, #define BF5XX_I2S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |\ SNDRV_PCM_FMTBIT_S32_LE) +static struct snd_soc_dai_ops bf5xx_i2s_dai_ops = { + .startup = bf5xx_i2s_startup, + .shutdown = bf5xx_i2s_shutdown, + .hw_params = bf5xx_i2s_hw_params, + .set_fmt = bf5xx_i2s_set_dai_fmt, +}; + struct snd_soc_dai bf5xx_i2s_dai = { .name = "bf5xx-i2s", .id = 0, @@ -304,12 +311,7 @@ struct snd_soc_dai bf5xx_i2s_dai = { .channels_max = 2, .rates = BF5XX_I2S_RATES, .formats = BF5XX_I2S_FORMATS,}, - .ops = { - .startup = bf5xx_i2s_startup, - .shutdown = bf5xx_i2s_shutdown, - .hw_params = bf5xx_i2s_hw_params, - .set_fmt = bf5xx_i2s_set_dai_fmt, - }, + .ops = &bf5xx_i2s_dai_ops, }; EXPORT_SYMBOL_GPL(bf5xx_i2s_dai); diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index fb53e65..7c0f412 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -41,6 +41,10 @@ static int ac97_prepare(struct snd_pcm_substream *substream, SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\ SNDRV_PCM_RATE_48000) +static struct snd_soc_dai_ops ac97_dai_ops = { + .prepare = ac97_prepare, +}; + struct snd_soc_dai ac97_dai = { .name = "AC97 HiFi", .ac97_control = 1, @@ -56,8 +60,7 @@ struct snd_soc_dai ac97_dai = { .channels_max = 2, .rates = STD_AC97_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .prepare = ac97_prepare,}, + .ops = &ac97_dai_ops, }; EXPORT_SYMBOL_GPL(ac97_dai); diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 81300d8..e81526a 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -436,6 +436,13 @@ static int ak4535_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +static struct snd_soc_dai_ops ak4535_dai_ops = { + .hw_params = ak4535_hw_params, + .set_fmt = ak4535_set_dai_fmt, + .digital_mute = ak4535_mute, + .set_sysclk = ak4535_set_dai_sysclk, +}; + struct snd_soc_dai ak4535_dai = { .name = "AK4535", .playback = { @@ -450,12 +457,7 @@ struct snd_soc_dai ak4535_dai = { .channels_max = 2, .rates = AK4535_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .hw_params = ak4535_hw_params, - .set_fmt = ak4535_set_dai_fmt, - .digital_mute = ak4535_mute, - .set_sysclk = ak4535_set_dai_sysclk, - }, + .ops = &ak4535_dai_ops, }; EXPORT_SYMBOL_GPL(ak4535_dai); diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index f1aa0c3..c6e155c 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -635,6 +635,17 @@ error: #endif /* USE_I2C*/ +#ifdef USE_I2C +static struct snd_soc_dai_ops cs4270_dai_ops = { + .hw_params = cs4270_hw_params, + .set_sysclk = cs4270_set_dai_sysclk, + .set_fmt = cs4270_set_dai_fmt, +#ifdef CONFIG_SND_SOC_CS4270_HWMUTE + .digital_mute = cs4270_mute, +#endif +}; +#endif + struct snd_soc_dai cs4270_dai = { .name = "CS4270", .playback = { @@ -708,15 +719,10 @@ static int cs4270_probe(struct platform_device *pdev) } /* Did we find a CS4270 on the I2C bus? */ - if (codec->control_data) { + if (codec->control_data) /* Initialize codec ops */ - cs4270_dai.ops.hw_params = cs4270_hw_params; - cs4270_dai.ops.set_sysclk = cs4270_set_dai_sysclk; - cs4270_dai.ops.set_fmt = cs4270_set_dai_fmt; -#ifdef CONFIG_SND_SOC_CS4270_HWMUTE - cs4270_dai.ops.digital_mute = cs4270_mute; -#endif - } else + cs4270_dai.ops = &cs4270_dai_ops; + else printk(KERN_INFO "cs4270: no I2C device found, " "using stand-alone mode\n"); #else diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index cac3736..a577ea1 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -521,6 +521,16 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec, #define SSM2602_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 ssm2602_dai_ops = { + .startup = ssm2602_startup, + .prepare = ssm2602_pcm_prepare, + .hw_params = ssm2602_hw_params, + .shutdown = ssm2602_shutdown, + .digital_mute = ssm2602_mute, + .set_sysclk = ssm2602_set_dai_sysclk, + .set_fmt = ssm2602_set_dai_fmt, +}; + struct snd_soc_dai ssm2602_dai = { .name = "SSM2602", .playback = { @@ -535,15 +545,7 @@ struct snd_soc_dai ssm2602_dai = { .channels_max = 2, .rates = SSM2602_RATES, .formats = SSM2602_FORMATS,}, - .ops = { - .startup = ssm2602_startup, - .prepare = ssm2602_pcm_prepare, - .hw_params = ssm2602_hw_params, - .shutdown = ssm2602_shutdown, - .digital_mute = ssm2602_mute, - .set_sysclk = ssm2602_set_dai_sysclk, - .set_fmt = ssm2602_set_dai_fmt, - } + .ops = &ssm2602_dai_ops, }; EXPORT_SYMBOL_GPL(ssm2602_dai); diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index cfdea00..daa5b88 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -598,6 +598,15 @@ static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec, #define AIC23_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) +static struct snd_soc_dai_ops tlv320aic23_dai_ops = { + .prepare = tlv320aic23_pcm_prepare, + .hw_params = tlv320aic23_hw_params, + .shutdown = tlv320aic23_shutdown, + .digital_mute = tlv320aic23_mute, + .set_fmt = tlv320aic23_set_dai_fmt, + .set_sysclk = tlv320aic23_set_dai_sysclk, +}; + struct snd_soc_dai tlv320aic23_dai = { .name = "tlv320aic23", .playback = { @@ -612,14 +621,7 @@ struct snd_soc_dai tlv320aic23_dai = { .channels_max = 2, .rates = AIC23_RATES, .formats = AIC23_FORMATS,}, - .ops = { - .prepare = tlv320aic23_pcm_prepare, - .hw_params = tlv320aic23_hw_params, - .shutdown = tlv320aic23_shutdown, - .digital_mute = tlv320aic23_mute, - .set_fmt = tlv320aic23_set_dai_fmt, - .set_sysclk = tlv320aic23_set_dai_sysclk, - } + .ops = &tlv320aic23_dai_ops, }; EXPORT_SYMBOL_GPL(tlv320aic23_dai); diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index 29f2f1a..8c6b15b 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -270,6 +270,13 @@ static int aic26_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) #define AIC26_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE |\ SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE) +static struct snd_soc_dai_ops aic26_dai_ops = { + .hw_params = aic26_hw_params, + .digital_mute = aic26_mute, + .set_sysclk = aic26_set_sysclk, + .set_fmt = aic26_set_fmt, +}; + struct snd_soc_dai aic26_dai = { .name = "tlv320aic26", .playback = { @@ -286,12 +293,7 @@ struct snd_soc_dai aic26_dai = { .rates = AIC26_RATES, .formats = AIC26_FORMATS, }, - .ops = { - .hw_params = aic26_hw_params, - .digital_mute = aic26_mute, - .set_sysclk = aic26_set_sysclk, - .set_fmt = aic26_set_fmt, - }, + .ops = &aic26_dai_ops, }; EXPORT_SYMBOL_GPL(aic26_dai); diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index b47a749..e24170a 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1069,6 +1069,13 @@ EXPORT_SYMBOL_GPL(aic3x_button_pressed); #define AIC3X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) +static struct snd_soc_dai_ops aic3x_dai_ops = { + .hw_params = aic3x_hw_params, + .digital_mute = aic3x_mute, + .set_sysclk = aic3x_set_dai_sysclk, + .set_fmt = aic3x_set_dai_fmt, +}; + struct snd_soc_dai aic3x_dai = { .name = "tlv320aic3x", .playback = { @@ -1083,12 +1090,7 @@ struct snd_soc_dai aic3x_dai = { .channels_max = 2, .rates = AIC3X_RATES, .formats = AIC3X_FORMATS,}, - .ops = { - .hw_params = aic3x_hw_params, - .digital_mute = aic3x_mute, - .set_sysclk = aic3x_set_dai_sysclk, - .set_fmt = aic3x_set_dai_fmt, - } + .ops = &aic3x_dai_ops, }; EXPORT_SYMBOL_GPL(aic3x_dai); diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index a2c5064..e75a2f9 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -464,6 +464,15 @@ static int uda134x_add_controls(struct snd_soc_codec *codec) return 0; } +static struct snd_soc_dai_ops uda134x_dai_ops = { + .startup = uda134x_startup, + .shutdown = uda134x_shutdown, + .hw_params = uda134x_hw_params, + .digital_mute = uda134x_mute, + .set_sysclk = uda134x_set_dai_sysclk, + .set_fmt = uda134x_set_dai_fmt, +}; + struct snd_soc_dai uda134x_dai = { .name = "UDA134X", /* playback capabilities */ @@ -483,14 +492,7 @@ struct snd_soc_dai uda134x_dai = { .formats = UDA134X_FORMATS, }, /* pcm operations */ - .ops = { - .startup = uda134x_startup, - .shutdown = uda134x_shutdown, - .hw_params = uda134x_hw_params, - .digital_mute = uda134x_mute, - .set_sysclk = uda134x_set_dai_sysclk, - .set_fmt = uda134x_set_dai_fmt, - } + .ops = &uda134x_dai_ops, }; EXPORT_SYMBOL(uda134x_dai); diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index e6bf084..bdd830f 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -544,6 +544,21 @@ static int uda1380_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +static struct snd_soc_dai_ops uda1380_dai_ops = { + .hw_params = uda1380_pcm_hw_params, + .shutdown = uda1380_pcm_shutdown, + .prepare = uda1380_pcm_prepare, + .digital_mute = uda1380_mute, + .set_fmt = uda1380_set_dai_fmt, +}; + +static struct snd_soc_dai_ops uda1380_dai_ops_capture = { + .hw_params = uda1380_pcm_hw_params, + .shutdown = uda1380_pcm_shutdown, + .prepare = uda1380_pcm_prepare, + .set_fmt = uda1380_set_dai_fmt, +}; + struct snd_soc_dai uda1380_dai[] = { { .name = "UDA1380", @@ -559,13 +574,7 @@ struct snd_soc_dai uda1380_dai[] = { .channels_max = 2, .rates = UDA1380_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .hw_params = uda1380_pcm_hw_params, - .shutdown = uda1380_pcm_shutdown, - .prepare = uda1380_pcm_prepare, - .digital_mute = uda1380_mute, - .set_fmt = uda1380_set_dai_fmt, - }, + .ops = &uda1380_dai_ops, }, { /* playback only - dual interface */ .name = "UDA1380", @@ -576,13 +585,7 @@ struct snd_soc_dai uda1380_dai[] = { .rates = UDA1380_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, - .ops = { - .hw_params = uda1380_pcm_hw_params, - .shutdown = uda1380_pcm_shutdown, - .prepare = uda1380_pcm_prepare, - .digital_mute = uda1380_mute, - .set_fmt = uda1380_set_dai_fmt, - }, + .ops = &uda1380_dai_ops, }, { /* capture only - dual interface*/ .name = "UDA1380", @@ -593,12 +596,7 @@ struct snd_soc_dai uda1380_dai[] = { .rates = UDA1380_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, - .ops = { - .hw_params = uda1380_pcm_hw_params, - .shutdown = uda1380_pcm_shutdown, - .prepare = uda1380_pcm_prepare, - .set_fmt = uda1380_set_dai_fmt, - }, + .ops = &uda1380_dai_ops_capture, }, }; EXPORT_SYMBOL_GPL(uda1380_dai); diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 35d9975..8b9e64b 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1436,6 +1436,16 @@ static int wm8350_remove(struct platform_device *pdev) SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) +static struct snd_soc_dai_ops wm8350_dai_ops = { + .hw_params = wm8350_pcm_hw_params, + .digital_mute = wm8350_mute, + .trigger = wm8350_pcm_trigger, + .set_fmt = wm8350_set_dai_fmt, + .set_sysclk = wm8350_set_dai_sysclk, + .set_pll = wm8350_set_fll, + .set_clkdiv = wm8350_set_clkdiv, +}; + struct snd_soc_dai wm8350_dai = { .name = "WM8350", .playback = { @@ -1452,15 +1462,7 @@ struct snd_soc_dai wm8350_dai = { .rates = WM8350_RATES, .formats = WM8350_FORMATS, }, - .ops = { - .hw_params = wm8350_pcm_hw_params, - .digital_mute = wm8350_mute, - .trigger = wm8350_pcm_trigger, - .set_fmt = wm8350_set_dai_fmt, - .set_sysclk = wm8350_set_dai_sysclk, - .set_pll = wm8350_set_fll, - .set_clkdiv = wm8350_set_clkdiv, - }, + .ops = &wm8350_dai_ops, }; EXPORT_SYMBOL_GPL(wm8350_dai); diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 40f8238..85e7dd7 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -570,6 +570,14 @@ static int wm8510_set_bias_level(struct snd_soc_codec *codec, #define WM8510_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 wm8510_dai_ops = { + .hw_params = wm8510_pcm_hw_params, + .digital_mute = wm8510_mute, + .set_fmt = wm8510_set_dai_fmt, + .set_clkdiv = wm8510_set_dai_clkdiv, + .set_pll = wm8510_set_dai_pll, +}; + struct snd_soc_dai wm8510_dai = { .name = "WM8510 HiFi", .playback = { @@ -584,13 +592,7 @@ struct snd_soc_dai wm8510_dai = { .channels_max = 2, .rates = WM8510_RATES, .formats = WM8510_FORMATS,}, - .ops = { - .hw_params = wm8510_pcm_hw_params, - .digital_mute = wm8510_mute, - .set_fmt = wm8510_set_dai_fmt, - .set_clkdiv = wm8510_set_dai_clkdiv, - .set_pll = wm8510_set_dai_pll, - }, + .ops = &wm8510_dai_ops, }; EXPORT_SYMBOL_GPL(wm8510_dai); diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index d004e58..4e343db 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -785,6 +785,21 @@ static int wm8580_set_bias_level(struct snd_soc_codec *codec, #define WM8580_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 wm8580_dai_ops_playback = { + .hw_params = wm8580_paif_hw_params, + .set_fmt = wm8580_set_paif_dai_fmt, + .set_clkdiv = wm8580_set_dai_clkdiv, + .set_pll = wm8580_set_dai_pll, + .digital_mute = wm8580_digital_mute, +}; + +static struct snd_soc_dai_ops wm8580_dai_ops_capture = { + .hw_params = wm8580_paif_hw_params, + .set_fmt = wm8580_set_paif_dai_fmt, + .set_clkdiv = wm8580_set_dai_clkdiv, + .set_pll = wm8580_set_dai_pll, +}; + struct snd_soc_dai wm8580_dai[] = { { .name = "WM8580 PAIFRX", @@ -796,13 +811,7 @@ struct snd_soc_dai wm8580_dai[] = { .rates = SNDRV_PCM_RATE_8000_192000, .formats = WM8580_FORMATS, }, - .ops = { - .hw_params = wm8580_paif_hw_params, - .set_fmt = wm8580_set_paif_dai_fmt, - .set_clkdiv = wm8580_set_dai_clkdiv, - .set_pll = wm8580_set_dai_pll, - .digital_mute = wm8580_digital_mute, - }, + .ops = &wm8580_dai_ops_playback, }, { .name = "WM8580 PAIFTX", @@ -814,12 +823,7 @@ struct snd_soc_dai wm8580_dai[] = { .rates = SNDRV_PCM_RATE_8000_192000, .formats = WM8580_FORMATS, }, - .ops = { - .hw_params = wm8580_paif_hw_params, - .set_fmt = wm8580_set_paif_dai_fmt, - .set_clkdiv = wm8580_set_dai_clkdiv, - .set_pll = wm8580_set_dai_pll, - }, + .ops = &wm8580_dai_ops_capture, }, }; EXPORT_SYMBOL_GPL(wm8580_dai); diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index 80b1198..d22ee46 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -259,6 +259,12 @@ static int wm8728_set_bias_level(struct snd_soc_codec *codec, #define WM8728_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) +static struct snd_soc_dai_ops wm8728_dai_ops = { + .hw_params = wm8728_hw_params, + .digital_mute = wm8728_mute, + .set_fmt = wm8728_set_dai_fmt, +}; + struct snd_soc_dai wm8728_dai = { .name = "WM8728", .playback = { @@ -268,11 +274,7 @@ struct snd_soc_dai wm8728_dai = { .rates = WM8728_RATES, .formats = WM8728_FORMATS, }, - .ops = { - .hw_params = wm8728_hw_params, - .digital_mute = wm8728_mute, - .set_fmt = wm8728_set_dai_fmt, - } + .ops = &wm8728_dai_ops, }; EXPORT_SYMBOL_GPL(wm8728_dai); diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index c444b9f..97f0851 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -446,6 +446,15 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, #define WM8731_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) +static struct snd_soc_dai_ops wm8731_dai_ops = { + .prepare = wm8731_pcm_prepare, + .hw_params = wm8731_hw_params, + .shutdown = wm8731_shutdown, + .digital_mute = wm8731_mute, + .set_sysclk = wm8731_set_dai_sysclk, + .set_fmt = wm8731_set_dai_fmt, +}; + struct snd_soc_dai wm8731_dai = { .name = "WM8731", .playback = { @@ -460,14 +469,7 @@ struct snd_soc_dai wm8731_dai = { .channels_max = 2, .rates = WM8731_RATES, .formats = WM8731_FORMATS,}, - .ops = { - .prepare = wm8731_pcm_prepare, - .hw_params = wm8731_hw_params, - .shutdown = wm8731_shutdown, - .digital_mute = wm8731_mute, - .set_sysclk = wm8731_set_dai_sysclk, - .set_fmt = wm8731_set_dai_fmt, - } + .ops = &wm8731_dai_ops, }; EXPORT_SYMBOL_GPL(wm8731_dai); diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 5997fa6..35de8e6 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -694,6 +694,13 @@ static int wm8750_set_bias_level(struct snd_soc_codec *codec, #define WM8750_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) +static struct snd_soc_dai_ops wm8750_dai_ops = { + .hw_params = wm8750_pcm_hw_params, + .digital_mute = wm8750_mute, + .set_fmt = wm8750_set_dai_fmt, + .set_sysclk = wm8750_set_dai_sysclk, +}; + struct snd_soc_dai wm8750_dai = { .name = "WM8750", .playback = { @@ -708,12 +715,7 @@ struct snd_soc_dai wm8750_dai = { .channels_max = 2, .rates = WM8750_RATES, .formats = WM8750_FORMATS,}, - .ops = { - .hw_params = wm8750_pcm_hw_params, - .digital_mute = wm8750_mute, - .set_fmt = wm8750_set_dai_fmt, - .set_sysclk = wm8750_set_dai_sysclk, - }, + .ops = &wm8750_dai_ops, }; EXPORT_SYMBOL_GPL(wm8750_dai); diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 77620ab..2f291c5 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1316,6 +1316,51 @@ static int wm8753_set_bias_level(struct snd_soc_codec *codec, * 3. Voice disabled - HIFI over HIFI * 4. Voice disabled - HIFI over HIFI, uses voice DAI LRC for capture */ +static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode1 = { + .hw_params = wm8753_i2s_hw_params, + .digital_mute = wm8753_mute, + .set_fmt = wm8753_mode1h_set_dai_fmt, + .set_clkdiv = wm8753_set_dai_clkdiv, + .set_pll = wm8753_set_dai_pll, + .set_sysclk = wm8753_set_dai_sysclk, +}; + +static struct snd_soc_dai_ops wm8753_dai_ops_voice_mode1 = { + .hw_params = wm8753_pcm_hw_params, + .digital_mute = wm8753_mute, + .set_fmt = wm8753_mode1v_set_dai_fmt, + .set_clkdiv = wm8753_set_dai_clkdiv, + .set_pll = wm8753_set_dai_pll, + .set_sysclk = wm8753_set_dai_sysclk, +}; + +static struct snd_soc_dai_ops wm8753_dai_ops_voice_mode2 = { + .hw_params = wm8753_pcm_hw_params, + .digital_mute = wm8753_mute, + .set_fmt = wm8753_mode2_set_dai_fmt, + .set_clkdiv = wm8753_set_dai_clkdiv, + .set_pll = wm8753_set_dai_pll, + .set_sysclk = wm8753_set_dai_sysclk, +}; + +static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode3 = { + .hw_params = wm8753_i2s_hw_params, + .digital_mute = wm8753_mute, + .set_fmt = wm8753_mode3_4_set_dai_fmt, + .set_clkdiv = wm8753_set_dai_clkdiv, + .set_pll = wm8753_set_dai_pll, + .set_sysclk = wm8753_set_dai_sysclk, +}; + +static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode4 = { + .hw_params = wm8753_i2s_hw_params, + .digital_mute = wm8753_mute, + .set_fmt = wm8753_mode3_4_set_dai_fmt, + .set_clkdiv = wm8753_set_dai_clkdiv, + .set_pll = wm8753_set_dai_pll, + .set_sysclk = wm8753_set_dai_sysclk, +}; + static const struct snd_soc_dai wm8753_all_dai[] = { /* DAI HiFi mode 1 */ { .name = "WM8753 HiFi", @@ -1332,14 +1377,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = { .channels_max = 2, .rates = WM8753_RATES, .formats = WM8753_FORMATS}, - .ops = { - .hw_params = wm8753_i2s_hw_params, - .digital_mute = wm8753_mute, - .set_fmt = wm8753_mode1h_set_dai_fmt, - .set_clkdiv = wm8753_set_dai_clkdiv, - .set_pll = wm8753_set_dai_pll, - .set_sysclk = wm8753_set_dai_sysclk, - }, + .ops = &wm8753_dai_ops_hifi_mode1, }, /* DAI Voice mode 1 */ { .name = "WM8753 Voice", @@ -1356,14 +1394,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = { .channels_max = 2, .rates = WM8753_RATES, .formats = WM8753_FORMATS,}, - .ops = { - .hw_params = wm8753_pcm_hw_params, - .digital_mute = wm8753_mute, - .set_fmt = wm8753_mode1v_set_dai_fmt, - .set_clkdiv = wm8753_set_dai_clkdiv, - .set_pll = wm8753_set_dai_pll, - .set_sysclk = wm8753_set_dai_sysclk, - }, + .ops = &wm8753_dai_ops_voice_mode1, }, /* DAI HiFi mode 2 - dummy */ { .name = "WM8753 HiFi", @@ -1384,14 +1415,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = { .channels_max = 2, .rates = WM8753_RATES, .formats = WM8753_FORMATS,}, - .ops = { - .hw_params = wm8753_pcm_hw_params, - .digital_mute = wm8753_mute, - .set_fmt = wm8753_mode2_set_dai_fmt, - .set_clkdiv = wm8753_set_dai_clkdiv, - .set_pll = wm8753_set_dai_pll, - .set_sysclk = wm8753_set_dai_sysclk, - }, + .ops = &wm8753_dai_ops_voice_mode2, }, /* DAI HiFi mode 3 */ { .name = "WM8753 HiFi", @@ -1408,14 +1432,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = { .channels_max = 2, .rates = WM8753_RATES, .formats = WM8753_FORMATS,}, - .ops = { - .hw_params = wm8753_i2s_hw_params, - .digital_mute = wm8753_mute, - .set_fmt = wm8753_mode3_4_set_dai_fmt, - .set_clkdiv = wm8753_set_dai_clkdiv, - .set_pll = wm8753_set_dai_pll, - .set_sysclk = wm8753_set_dai_sysclk, - }, + .ops = &wm8753_dai_ops_hifi_mode3, }, /* DAI Voice mode 3 - dummy */ { .name = "WM8753 Voice", @@ -1436,14 +1453,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = { .channels_max = 2, .rates = WM8753_RATES, .formats = WM8753_FORMATS,}, - .ops = { - .hw_params = wm8753_i2s_hw_params, - .digital_mute = wm8753_mute, - .set_fmt = wm8753_mode3_4_set_dai_fmt, - .set_clkdiv = wm8753_set_dai_clkdiv, - .set_pll = wm8753_set_dai_pll, - .set_sysclk = wm8753_set_dai_sysclk, - }, + .ops = &wm8753_dai_ops_hifi_mode4, }, /* DAI Voice mode 4 - dummy */ { .name = "WM8753 Voice", diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 6767de1..589924d 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -1104,6 +1104,14 @@ static int wm8900_digital_mute(struct snd_soc_dai *codec_dai, int mute) (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \ SNDRV_PCM_FORMAT_S24_LE) +static struct snd_soc_dai_ops wm8900_dai_ops = { + .hw_params = wm8900_hw_params, + .set_clkdiv = wm8900_set_dai_clkdiv, + .set_pll = wm8900_set_dai_pll, + .set_fmt = wm8900_set_dai_fmt, + .digital_mute = wm8900_digital_mute, +}; + struct snd_soc_dai wm8900_dai = { .name = "WM8900 HiFi", .playback = { @@ -1120,13 +1128,7 @@ struct snd_soc_dai wm8900_dai = { .rates = WM8900_RATES, .formats = WM8900_PCM_FORMATS, }, - .ops = { - .hw_params = wm8900_hw_params, - .set_clkdiv = wm8900_set_dai_clkdiv, - .set_pll = wm8900_set_dai_pll, - .set_fmt = wm8900_set_dai_fmt, - .digital_mute = wm8900_digital_mute, - }, + .ops = &wm8900_dai_ops, }; EXPORT_SYMBOL_GPL(wm8900_dai); diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index bde7454..f8f19d8 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1512,6 +1512,15 @@ static int wm8903_hw_params(struct snd_pcm_substream *substream, SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) +static struct snd_soc_dai_ops wm8903_dai_ops = { + .startup = wm8903_startup, + .shutdown = wm8903_shutdown, + .hw_params = wm8903_hw_params, + .digital_mute = wm8903_digital_mute, + .set_fmt = wm8903_set_dai_fmt, + .set_sysclk = wm8903_set_dai_sysclk, +}; + struct snd_soc_dai wm8903_dai = { .name = "WM8903", .playback = { @@ -1528,14 +1537,7 @@ struct snd_soc_dai wm8903_dai = { .rates = WM8903_CAPTURE_RATES, .formats = WM8903_FORMATS, }, - .ops = { - .startup = wm8903_startup, - .shutdown = wm8903_shutdown, - .hw_params = wm8903_hw_params, - .digital_mute = wm8903_digital_mute, - .set_fmt = wm8903_set_dai_fmt, - .set_sysclk = wm8903_set_dai_sysclk - } + .ops = &wm8903_dai_ops, }; EXPORT_SYMBOL_GPL(wm8903_dai); diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 88ead7f..826d1c9 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -619,6 +619,13 @@ static int wm8971_set_bias_level(struct snd_soc_codec *codec, #define WM8971_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) +static struct snd_soc_dai_ops wm8971_dai_ops = { + .hw_params = wm8971_pcm_hw_params, + .digital_mute = wm8971_mute, + .set_fmt = wm8971_set_dai_fmt, + .set_sysclk = wm8971_set_dai_sysclk, +}; + struct snd_soc_dai wm8971_dai = { .name = "WM8971", .playback = { @@ -633,12 +640,7 @@ struct snd_soc_dai wm8971_dai = { .channels_max = 2, .rates = WM8971_RATES, .formats = WM8971_FORMATS,}, - .ops = { - .hw_params = wm8971_pcm_hw_params, - .digital_mute = wm8971_mute, - .set_fmt = wm8971_set_dai_fmt, - .set_sysclk = wm8971_set_dai_sysclk, - }, + .ops = &wm8971_dai_ops, }; EXPORT_SYMBOL_GPL(wm8971_dai); diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 1cbb7b9..e0141d8 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1345,6 +1345,15 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, * 1. ADC/DAC on Primary Interface * 2. ADC on Primary Interface/DAC on secondary */ +static struct snd_soc_dai_ops wm8990_dai_ops = { + .hw_params = wm8990_hw_params, + .digital_mute = wm8990_mute, + .set_fmt = wm8990_set_dai_fmt, + .set_clkdiv = wm8990_set_dai_clkdiv, + .set_pll = wm8990_set_dai_pll, + .set_sysclk = wm8990_set_dai_sysclk, +}; + struct snd_soc_dai wm8990_dai = { /* ADC/DAC on primary */ .name = "WM8990 ADC/DAC Primary", @@ -1361,14 +1370,7 @@ struct snd_soc_dai wm8990_dai = { .channels_max = 2, .rates = WM8990_RATES, .formats = WM8990_FORMATS,}, - .ops = { - .hw_params = wm8990_hw_params, - .digital_mute = wm8990_mute, - .set_fmt = wm8990_set_dai_fmt, - .set_clkdiv = wm8990_set_dai_clkdiv, - .set_pll = wm8990_set_dai_pll, - .set_sysclk = wm8990_set_dai_sysclk, - }, + .ops = &wm8990_dai_ops, }; EXPORT_SYMBOL_GPL(wm8990_dai); diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index af83d62..684e94e 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -532,6 +532,14 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream, SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\ SNDRV_PCM_RATE_48000) +static struct snd_soc_dai_ops wm9712_dai_ops_hifi = { + .prepare = ac97_prepare, +}; + +static struct snd_soc_dai_ops wm9712_dai_ops_aux = { + .prepare = ac97_aux_prepare, +}; + struct snd_soc_dai wm9712_dai[] = { { .name = "AC97 HiFi", @@ -548,8 +556,7 @@ struct snd_soc_dai wm9712_dai[] = { .channels_max = 2, .rates = WM9712_AC97_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .prepare = ac97_prepare,}, + .ops = &wm9712_dai_ops_hifi, }, { .name = "AC97 Aux", @@ -559,8 +566,7 @@ struct snd_soc_dai wm9712_dai[] = { .channels_max = 1, .rates = WM9712_AC97_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .prepare = ac97_aux_prepare,}, + .ops = &wm9712_dai_ops_aux, } }; EXPORT_SYMBOL_GPL(wm9712_dai); diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index f3ca8aa..ec8f303 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1021,6 +1021,27 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream, (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \ SNDRV_PCM_FORMAT_S24_LE) +static struct snd_soc_dai_ops wm9713_dai_ops_hifi = { + .prepare = ac97_hifi_prepare, + .set_clkdiv = wm9713_set_dai_clkdiv, + .set_pll = wm9713_set_dai_pll, +}; + +static struct snd_soc_dai_ops wm9713_dai_ops_aux = { + .prepare = ac97_aux_prepare, + .set_clkdiv = wm9713_set_dai_clkdiv, + .set_pll = wm9713_set_dai_pll, +}; + +static struct snd_soc_dai_ops wm9713_dai_ops_voice = { + .hw_params = wm9713_pcm_hw_params, + .shutdown = wm9713_voiceshutdown, + .set_clkdiv = wm9713_set_dai_clkdiv, + .set_pll = wm9713_set_dai_pll, + .set_fmt = wm9713_set_dai_fmt, + .set_tristate = wm9713_set_dai_tristate, +}; + struct snd_soc_dai wm9713_dai[] = { { .name = "AC97 HiFi", @@ -1037,10 +1058,7 @@ struct snd_soc_dai wm9713_dai[] = { .channels_max = 2, .rates = WM9713_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .prepare = ac97_hifi_prepare, - .set_clkdiv = wm9713_set_dai_clkdiv, - .set_pll = wm9713_set_dai_pll,}, + .ops = &wm9713_dai_ops_hifi, }, { .name = "AC97 Aux", @@ -1050,10 +1068,7 @@ struct snd_soc_dai wm9713_dai[] = { .channels_max = 1, .rates = WM9713_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .prepare = ac97_aux_prepare, - .set_clkdiv = wm9713_set_dai_clkdiv, - .set_pll = wm9713_set_dai_pll,}, + .ops = &wm9713_dai_ops_aux, }, { .name = "WM9713 Voice", @@ -1069,14 +1084,7 @@ struct snd_soc_dai wm9713_dai[] = { .channels_max = 2, .rates = WM9713_PCM_RATES, .formats = WM9713_PCM_FORMATS,}, - .ops = { - .hw_params = wm9713_pcm_hw_params, - .shutdown = wm9713_voiceshutdown, - .set_clkdiv = wm9713_set_dai_clkdiv, - .set_pll = wm9713_set_dai_pll, - .set_fmt = wm9713_set_dai_fmt, - .set_tristate = wm9713_set_dai_tristate, - }, + .ops = &wm9713_dai_ops_voice, }, }; EXPORT_SYMBOL_GPL(wm9713_dai); diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 0fee779..ffdb943 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -499,6 +499,13 @@ static void davinci_i2s_remove(struct platform_device *pdev, #define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 +static struct snd_soc_dai_ops davinci_i2s_dai_ops = { + .startup = davinci_i2s_startup, + .trigger = davinci_i2s_trigger, + .hw_params = davinci_i2s_hw_params, + .set_fmt = davinci_i2s_set_dai_fmt, +}; + struct snd_soc_dai davinci_i2s_dai = { .name = "davinci-i2s", .id = 0, @@ -514,12 +521,7 @@ struct snd_soc_dai davinci_i2s_dai = { .channels_max = 2, .rates = DAVINCI_I2S_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .startup = davinci_i2s_startup, - .trigger = davinci_i2s_trigger, - .hw_params = davinci_i2s_hw_params, - .set_fmt = davinci_i2s_set_dai_fmt, - }, + .ops = &davinci_i2s_dai_ops, }; EXPORT_SYMBOL_GPL(davinci_i2s_dai); diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index c6d6eb7..104d5f9 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -563,6 +563,15 @@ static int fsl_ssi_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format) /** * fsl_ssi_dai_template: template CPU DAI for the SSI */ +static struct snd_soc_dai_ops fsl_ssi_dai_ops = { + .startup = fsl_ssi_startup, + .prepare = fsl_ssi_prepare, + .shutdown = fsl_ssi_shutdown, + .trigger = fsl_ssi_trigger, + .set_sysclk = fsl_ssi_set_sysclk, + .set_fmt = fsl_ssi_set_fmt, +}; + static struct snd_soc_dai fsl_ssi_dai_template = { .playback = { /* The SSI does not support monaural audio. */ @@ -577,14 +586,7 @@ static struct snd_soc_dai fsl_ssi_dai_template = { .rates = FSLSSI_I2S_RATES, .formats = FSLSSI_I2S_FORMATS, }, - .ops = { - .startup = fsl_ssi_startup, - .prepare = fsl_ssi_prepare, - .shutdown = fsl_ssi_shutdown, - .trigger = fsl_ssi_trigger, - .set_sysclk = fsl_ssi_set_sysclk, - .set_fmt = fsl_ssi_set_fmt, - }, + .ops = &fsl_ssi_dai_ops, }; /** diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 05dd5ab..d6882be 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -461,6 +461,16 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, return err; } +static struct snd_soc_dai_ops omap_mcbsp_dai_ops = { + .startup = omap_mcbsp_dai_startup, + .shutdown = omap_mcbsp_dai_shutdown, + .trigger = omap_mcbsp_dai_trigger, + .hw_params = omap_mcbsp_dai_hw_params, + .set_fmt = omap_mcbsp_dai_set_dai_fmt, + .set_clkdiv = omap_mcbsp_dai_set_clkdiv, + .set_sysclk = omap_mcbsp_dai_set_dai_sysclk, +}; + #define OMAP_MCBSP_DAI_BUILDER(link_id) \ { \ .name = "omap-mcbsp-dai-"#link_id, \ @@ -477,15 +487,7 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, .rates = OMAP_MCBSP_RATES, \ .formats = SNDRV_PCM_FMTBIT_S16_LE, \ }, \ - .ops = { \ - .startup = omap_mcbsp_dai_startup, \ - .shutdown = omap_mcbsp_dai_shutdown, \ - .trigger = omap_mcbsp_dai_trigger, \ - .hw_params = omap_mcbsp_dai_hw_params, \ - .set_fmt = omap_mcbsp_dai_set_dai_fmt, \ - .set_clkdiv = omap_mcbsp_dai_set_clkdiv, \ - .set_sysclk = omap_mcbsp_dai_set_dai_sysclk, \ - }, \ + .ops = &omap_mcbsp_dai_ops, \ .private_data = &mcbsp_data[(link_id)].bus_id, \ } diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 724e8fb..1242d68 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -766,6 +766,19 @@ static void pxa_ssp_remove(struct platform_device *pdev, SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_S32_LE) +static struct snd_soc_dai_ops pxa_ssp_dai_ops = { + .startup = pxa_ssp_startup, + .shutdown = pxa_ssp_shutdown, + .trigger = pxa_ssp_trigger, + .hw_params = pxa_ssp_hw_params, + .set_sysclk = pxa_ssp_set_dai_sysclk, + .set_clkdiv = pxa_ssp_set_dai_clkdiv, + .set_pll = pxa_ssp_set_dai_pll, + .set_fmt = pxa_ssp_set_dai_fmt, + .set_tdm_slot = pxa_ssp_set_dai_tdm_slot, + .set_tristate = pxa_ssp_set_dai_tristate, +}; + struct snd_soc_dai pxa_ssp_dai[] = { { .name = "pxa2xx-ssp1", @@ -786,18 +799,7 @@ struct snd_soc_dai pxa_ssp_dai[] = { .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, - .ops = { - .startup = pxa_ssp_startup, - .shutdown = pxa_ssp_shutdown, - .trigger = pxa_ssp_trigger, - .hw_params = pxa_ssp_hw_params, - .set_sysclk = pxa_ssp_set_dai_sysclk, - .set_clkdiv = pxa_ssp_set_dai_clkdiv, - .set_pll = pxa_ssp_set_dai_pll, - .set_fmt = pxa_ssp_set_dai_fmt, - .set_tdm_slot = pxa_ssp_set_dai_tdm_slot, - .set_tristate = pxa_ssp_set_dai_tristate, - }, + .ops = &pxa_ssp_dai_ops, }, { .name = "pxa2xx-ssp2", .id = 1, @@ -817,18 +819,7 @@ struct snd_soc_dai pxa_ssp_dai[] = { .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, - .ops = { - .startup = pxa_ssp_startup, - .shutdown = pxa_ssp_shutdown, - .trigger = pxa_ssp_trigger, - .hw_params = pxa_ssp_hw_params, - .set_sysclk = pxa_ssp_set_dai_sysclk, - .set_clkdiv = pxa_ssp_set_dai_clkdiv, - .set_pll = pxa_ssp_set_dai_pll, - .set_fmt = pxa_ssp_set_dai_fmt, - .set_tdm_slot = pxa_ssp_set_dai_tdm_slot, - .set_tristate = pxa_ssp_set_dai_tristate, - }, + .ops = &pxa_ssp_dai_ops, }, { .name = "pxa2xx-ssp3", @@ -849,18 +840,7 @@ struct snd_soc_dai pxa_ssp_dai[] = { .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, - .ops = { - .startup = pxa_ssp_startup, - .shutdown = pxa_ssp_shutdown, - .trigger = pxa_ssp_trigger, - .hw_params = pxa_ssp_hw_params, - .set_sysclk = pxa_ssp_set_dai_sysclk, - .set_clkdiv = pxa_ssp_set_dai_clkdiv, - .set_pll = pxa_ssp_set_dai_pll, - .set_fmt = pxa_ssp_set_dai_fmt, - .set_tdm_slot = pxa_ssp_set_dai_tdm_slot, - .set_tristate = pxa_ssp_set_dai_tristate, - }, + .ops = &pxa_ssp_dai_ops, }, { .name = "pxa2xx-ssp4", @@ -881,18 +861,7 @@ struct snd_soc_dai pxa_ssp_dai[] = { .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, - .ops = { - .startup = pxa_ssp_startup, - .shutdown = pxa_ssp_shutdown, - .trigger = pxa_ssp_trigger, - .hw_params = pxa_ssp_hw_params, - .set_sysclk = pxa_ssp_set_dai_sysclk, - .set_clkdiv = pxa_ssp_set_dai_clkdiv, - .set_pll = pxa_ssp_set_dai_pll, - .set_fmt = pxa_ssp_set_dai_fmt, - .set_tdm_slot = pxa_ssp_set_dai_tdm_slot, - .set_tristate = pxa_ssp_set_dai_tristate, - }, + .ops = &pxa_ssp_dai_ops, }, }; EXPORT_SYMBOL_GPL(pxa_ssp_dai); diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index a4a655f..a7b8a3f 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -164,6 +164,10 @@ static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream, SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000) +static struct snd_soc_dai_ops pxa_ac97_dai_ops = { + .hw_params = pxa2xx_ac97_hw_params, +}; + /* * There is only 1 physical AC97 interface for pxa2xx, but it * has extra fifo's that can be used for aux DACs and ADCs. @@ -189,8 +193,7 @@ struct snd_soc_dai pxa_ac97_dai[] = { .channels_max = 2, .rates = PXA2XX_AC97_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .hw_params = pxa2xx_ac97_hw_params,}, + .ops = &pxa_ac97_dai_ops, }, { .name = "pxa2xx-ac97-aux", @@ -208,8 +211,7 @@ struct snd_soc_dai pxa_ac97_dai[] = { .channels_max = 1, .rates = PXA2XX_AC97_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .hw_params = pxa2xx_ac97_hw_aux_params,}, + .ops = &pxa_ac97_dai_ops, }, { .name = "pxa2xx-ac97-mic", @@ -221,8 +223,7 @@ struct snd_soc_dai pxa_ac97_dai[] = { .channels_max = 1, .rates = PXA2XX_AC97_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .hw_params = pxa2xx_ac97_hw_mic_params,}, + .ops = &pxa_ac97_dai_ops, }, }; diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 223de89..bd63a86 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -335,6 +335,15 @@ static int pxa2xx_i2s_resume(struct snd_soc_dai *dai) SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000) +static struct snd_soc_dai_ops pxa_i2s_dai_ops = { + .startup = pxa2xx_i2s_startup, + .shutdown = pxa2xx_i2s_shutdown, + .trigger = pxa2xx_i2s_trigger, + .hw_params = pxa2xx_i2s_hw_params, + .set_fmt = pxa2xx_i2s_set_dai_fmt, + .set_sysclk = pxa2xx_i2s_set_dai_sysclk, +}; + struct snd_soc_dai pxa_i2s_dai = { .name = "pxa2xx-i2s", .id = 0, @@ -350,14 +359,7 @@ struct snd_soc_dai pxa_i2s_dai = { .channels_max = 2, .rates = PXA2XX_I2S_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .startup = pxa2xx_i2s_startup, - .shutdown = pxa2xx_i2s_shutdown, - .trigger = pxa2xx_i2s_trigger, - .hw_params = pxa2xx_i2s_hw_params, - .set_fmt = pxa2xx_i2s_set_dai_fmt, - .set_sysclk = pxa2xx_i2s_set_dai_sysclk, - }, + .ops = &pxa_i2s_dai_ops, }; EXPORT_SYMBOL_GPL(pxa_i2s_dai); diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index f3fc0ab..382d7ee 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -708,6 +708,14 @@ static int s3c2412_i2s_resume(struct snd_soc_dai *dai) SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) +static struct snd_soc_dai_ops s3c2412_i2s_dai_ops = { + .trigger = s3c2412_i2s_trigger, + .hw_params = s3c2412_i2s_hw_params, + .set_fmt = s3c2412_i2s_set_fmt, + .set_clkdiv = s3c2412_i2s_set_clkdiv, + .set_sysclk = s3c2412_i2s_set_sysclk, +}; + struct snd_soc_dai s3c2412_i2s_dai = { .name = "s3c2412-i2s", .id = 0, @@ -726,13 +734,7 @@ struct snd_soc_dai s3c2412_i2s_dai = { .rates = S3C2412_I2S_RATES, .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE, }, - .ops = { - .trigger = s3c2412_i2s_trigger, - .hw_params = s3c2412_i2s_hw_params, - .set_fmt = s3c2412_i2s_set_fmt, - .set_clkdiv = s3c2412_i2s_set_clkdiv, - .set_sysclk = s3c2412_i2s_set_sysclk, - }, + .ops = &s3c2412_i2s_dai_ops, }; EXPORT_SYMBOL_GPL(s3c2412_i2s_dai); diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c index 5822d2d..83ea623 100644 --- a/sound/soc/s3c24xx/s3c2443-ac97.c +++ b/sound/soc/s3c24xx/s3c2443-ac97.c @@ -355,6 +355,11 @@ static int s3c2443_ac97_mic_trigger(struct snd_pcm_substream *substream, SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +static struct snd_soc_dai_ops s3c2443_ac97_dai_ops = { + .hw_params = s3c2443_ac97_hw_params, + .trigger = s3c2443_ac97_trigger, +}; + struct snd_soc_dai s3c2443_ac97_dai[] = { { .name = "s3c2443-ac97", @@ -374,9 +379,7 @@ struct snd_soc_dai s3c2443_ac97_dai[] = { .channels_max = 2, .rates = s3c2443_AC97_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .hw_params = s3c2443_ac97_hw_params, - .trigger = s3c2443_ac97_trigger}, + .ops = &s3c2443_ac97_dai_ops, }, { .name = "pxa2xx-ac97-mic", @@ -388,9 +391,7 @@ struct snd_soc_dai s3c2443_ac97_dai[] = { .channels_max = 1, .rates = s3c2443_AC97_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .hw_params = s3c2443_ac97_hw_mic_params, - .trigger = s3c2443_ac97_mic_trigger,}, + .ops = &s3c2443_ac97_dai_ops, }, }; EXPORT_SYMBOL_GPL(s3c2443_ac97_dai); diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 6f4d439..82d8c6e 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -456,6 +456,14 @@ static int s3c24xx_i2s_resume(struct snd_soc_dai *cpu_dai) SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) +static struct snd_soc_dai_ops s3c24xx_i2s_dai_ops = { + .trigger = s3c24xx_i2s_trigger, + .hw_params = s3c24xx_i2s_hw_params, + .set_fmt = s3c24xx_i2s_set_fmt, + .set_clkdiv = s3c24xx_i2s_set_clkdiv, + .set_sysclk = s3c24xx_i2s_set_sysclk, +}; + struct snd_soc_dai s3c24xx_i2s_dai = { .name = "s3c24xx-i2s", .id = 0, @@ -472,13 +480,7 @@ struct snd_soc_dai s3c24xx_i2s_dai = { .channels_max = 2, .rates = S3C24XX_I2S_RATES, .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .trigger = s3c24xx_i2s_trigger, - .hw_params = s3c24xx_i2s_hw_params, - .set_fmt = s3c24xx_i2s_set_fmt, - .set_clkdiv = s3c24xx_i2s_set_clkdiv, - .set_sysclk = s3c24xx_i2s_set_sysclk, - }, + .ops = &s3c24xx_i2s_dai_ops, }; EXPORT_SYMBOL_GPL(s3c24xx_i2s_dai); diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c index d1e5390..56fa087 100644 --- a/sound/soc/sh/ssi.c +++ b/sound/soc/sh/ssi.c @@ -336,6 +336,16 @@ static int ssi_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3LE | \ SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE) +static struct snd_soc_dai_ops ssi_dai_ops = { + .startup = ssi_startup, + .shutdown = ssi_shutdown, + .trigger = ssi_trigger, + .hw_params = ssi_hw_params, + .set_sysclk = ssi_set_sysclk, + .set_clkdiv = ssi_set_clkdiv, + .set_fmt = ssi_set_fmt, +}; + struct snd_soc_dai sh4_ssi_dai[] = { { .name = "SSI0", @@ -352,15 +362,7 @@ struct snd_soc_dai sh4_ssi_dai[] = { .channels_min = 2, .channels_max = 8, }, - .ops = { - .startup = ssi_startup, - .shutdown = ssi_shutdown, - .trigger = ssi_trigger, - .hw_params = ssi_hw_params, - .set_sysclk = ssi_set_sysclk, - .set_clkdiv = ssi_set_clkdiv, - .set_fmt = ssi_set_fmt, - }, + .ops = &ssi_dai_ops, }, #ifdef CONFIG_CPU_SUBTYPE_SH7760 { @@ -378,15 +380,7 @@ struct snd_soc_dai sh4_ssi_dai[] = { .channels_min = 2, .channels_max = 8, }, - .ops = { - .startup = ssi_startup, - .shutdown = ssi_shutdown, - .trigger = ssi_trigger, - .hw_params = ssi_hw_params, - .set_sysclk = ssi_set_sysclk, - .set_clkdiv = ssi_set_clkdiv, - .set_fmt = ssi_set_fmt, - }, + .ops = &ssi_dai_ops, }, #endif }; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 55fdb4a..a0113f1 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -133,8 +133,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) mutex_lock(&pcm_mutex); /* startup the audio subsystem */ - if (cpu_dai->ops.startup) { - ret = cpu_dai->ops.startup(substream, cpu_dai); + if (cpu_dai->ops->startup) { + ret = cpu_dai->ops->startup(substream, cpu_dai); if (ret < 0) { printk(KERN_ERR "asoc: can't open interface %s\n", cpu_dai->name); @@ -150,8 +150,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) } } - if (codec_dai->ops.startup) { - ret = codec_dai->ops.startup(substream, codec_dai); + if (codec_dai->ops->startup) { + ret = codec_dai->ops->startup(substream, codec_dai); if (ret < 0) { printk(KERN_ERR "asoc: can't open codec %s\n", codec_dai->name); @@ -247,8 +247,8 @@ codec_dai_err: platform->pcm_ops->close(substream); platform_err: - if (cpu_dai->ops.shutdown) - cpu_dai->ops.shutdown(substream, cpu_dai); + if (cpu_dai->ops->shutdown) + cpu_dai->ops->shutdown(substream, cpu_dai); out: mutex_unlock(&pcm_mutex); return ret; @@ -340,11 +340,11 @@ static int soc_codec_close(struct snd_pcm_substream *substream) if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) snd_soc_dai_digital_mute(codec_dai, 1); - if (cpu_dai->ops.shutdown) - cpu_dai->ops.shutdown(substream, cpu_dai); + if (cpu_dai->ops->shutdown) + cpu_dai->ops->shutdown(substream, cpu_dai); - if (codec_dai->ops.shutdown) - codec_dai->ops.shutdown(substream, codec_dai); + if (codec_dai->ops->shutdown) + codec_dai->ops->shutdown(substream, codec_dai); if (machine->ops && machine->ops->shutdown) machine->ops->shutdown(substream); @@ -408,16 +408,16 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) } } - if (codec_dai->ops.prepare) { - ret = codec_dai->ops.prepare(substream, codec_dai); + if (codec_dai->ops->prepare) { + ret = codec_dai->ops->prepare(substream, codec_dai); if (ret < 0) { printk(KERN_ERR "asoc: codec DAI prepare error\n"); goto out; } } - if (cpu_dai->ops.prepare) { - ret = cpu_dai->ops.prepare(substream, cpu_dai); + if (cpu_dai->ops->prepare) { + ret = cpu_dai->ops->prepare(substream, cpu_dai); if (ret < 0) { printk(KERN_ERR "asoc: cpu DAI prepare error\n"); goto out; @@ -494,8 +494,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, } } - if (codec_dai->ops.hw_params) { - ret = codec_dai->ops.hw_params(substream, params, codec_dai); + if (codec_dai->ops->hw_params) { + ret = codec_dai->ops->hw_params(substream, params, codec_dai); if (ret < 0) { printk(KERN_ERR "asoc: can't set codec %s hw params\n", codec_dai->name); @@ -503,8 +503,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, } } - if (cpu_dai->ops.hw_params) { - ret = cpu_dai->ops.hw_params(substream, params, cpu_dai); + if (cpu_dai->ops->hw_params) { + ret = cpu_dai->ops->hw_params(substream, params, cpu_dai); if (ret < 0) { printk(KERN_ERR "asoc: interface %s hw params failed\n", cpu_dai->name); @@ -526,12 +526,12 @@ out: return ret; platform_err: - if (cpu_dai->ops.hw_free) - cpu_dai->ops.hw_free(substream, cpu_dai); + if (cpu_dai->ops->hw_free) + cpu_dai->ops->hw_free(substream, cpu_dai); interface_err: - if (codec_dai->ops.hw_free) - codec_dai->ops.hw_free(substream, codec_dai); + if (codec_dai->ops->hw_free) + codec_dai->ops->hw_free(substream, codec_dai); codec_err: if (machine->ops && machine->ops->hw_free) @@ -570,11 +570,11 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) platform->pcm_ops->hw_free(substream); /* now free hw params for the DAI's */ - if (codec_dai->ops.hw_free) - codec_dai->ops.hw_free(substream, codec_dai); + if (codec_dai->ops->hw_free) + codec_dai->ops->hw_free(substream, codec_dai); - if (cpu_dai->ops.hw_free) - cpu_dai->ops.hw_free(substream, cpu_dai); + if (cpu_dai->ops->hw_free) + cpu_dai->ops->hw_free(substream, cpu_dai); mutex_unlock(&pcm_mutex); return 0; @@ -591,8 +591,8 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_soc_dai *codec_dai = machine->codec_dai; int ret; - if (codec_dai->ops.trigger) { - ret = codec_dai->ops.trigger(substream, cmd, codec_dai); + if (codec_dai->ops->trigger) { + ret = codec_dai->ops->trigger(substream, cmd, codec_dai); if (ret < 0) return ret; } @@ -603,8 +603,8 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return ret; } - if (cpu_dai->ops.trigger) { - ret = cpu_dai->ops.trigger(substream, cmd, cpu_dai); + if (cpu_dai->ops->trigger) { + ret = cpu_dai->ops->trigger(substream, cmd, cpu_dai); if (ret < 0) return ret; } @@ -645,8 +645,8 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) /* mute any active DAC's */ for (i = 0; i < card->num_links; i++) { struct snd_soc_dai *dai = card->dai_link[i].codec_dai; - if (dai->ops.digital_mute && dai->playback.active) - dai->ops.digital_mute(dai, 1); + if (dai->ops->digital_mute && dai->playback.active) + dai->ops->digital_mute(dai, 1); } /* suspend all pcms */ @@ -741,8 +741,8 @@ static void soc_resume_deferred(struct work_struct *work) /* unmute any active DACs */ for (i = 0; i < card->num_links; i++) { struct snd_soc_dai *dai = card->dai_link[i].codec_dai; - if (dai->ops.digital_mute && dai->playback.active) - dai->ops.digital_mute(dai, 0); + if (dai->ops->digital_mute && dai->playback.active) + dai->ops->digital_mute(dai, 0); } for (i = 0; i < card->num_links; i++) { @@ -2020,8 +2020,8 @@ EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8); int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int freq, int dir) { - if (dai->ops.set_sysclk) - return dai->ops.set_sysclk(dai, clk_id, freq, dir); + if (dai->ops->set_sysclk) + return dai->ops->set_sysclk(dai, clk_id, freq, dir); else return -EINVAL; } @@ -2040,8 +2040,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk); int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) { - if (dai->ops.set_clkdiv) - return dai->ops.set_clkdiv(dai, div_id, div); + if (dai->ops->set_clkdiv) + return dai->ops->set_clkdiv(dai, div_id, div); else return -EINVAL; } @@ -2059,8 +2059,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_clkdiv); int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int pll_id, unsigned int freq_in, unsigned int freq_out) { - if (dai->ops.set_pll) - return dai->ops.set_pll(dai, pll_id, freq_in, freq_out); + if (dai->ops->set_pll) + return dai->ops->set_pll(dai, pll_id, freq_in, freq_out); else return -EINVAL; } @@ -2075,8 +2075,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll); */ int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { - if (dai->ops.set_fmt) - return dai->ops.set_fmt(dai, fmt); + if (dai->ops->set_fmt) + return dai->ops->set_fmt(dai, fmt); else return -EINVAL; } @@ -2094,8 +2094,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt); int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, unsigned int mask, int slots) { - if (dai->ops.set_sysclk) - return dai->ops.set_tdm_slot(dai, mask, slots); + if (dai->ops->set_sysclk) + return dai->ops->set_tdm_slot(dai, mask, slots); else return -EINVAL; } @@ -2110,8 +2110,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot); */ int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate) { - if (dai->ops.set_sysclk) - return dai->ops.set_tristate(dai, tristate); + if (dai->ops->set_sysclk) + return dai->ops->set_tristate(dai, tristate); else return -EINVAL; } @@ -2126,8 +2126,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate); */ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute) { - if (dai->ops.digital_mute) - return dai->ops.digital_mute(dai, mute); + if (dai->ops->digital_mute) + return dai->ops->digital_mute(dai, mute); else return -EINVAL; } @@ -2180,6 +2180,9 @@ static int snd_soc_unregister_card(struct snd_soc_card *card) return 0; } +static struct snd_soc_dai_ops null_dai_ops = { +}; + /** * snd_soc_register_dai - Register a DAI with the ASoC core * @@ -2194,6 +2197,9 @@ int snd_soc_register_dai(struct snd_soc_dai *dai) if (!dai->dev) printk(KERN_WARNING "No device for DAI %s\n", dai->name); + if (!dai->ops) + dai->ops = &null_dai_ops; + INIT_LIST_HEAD(&dai->list); mutex_lock(&client_mutex);

5 6

[alsa-devel] [PATCH 1/3] ASoC: Add GPIO support for jack reporting interface
by Lopez Cruz, Misael 03 Mar '09

03 Mar '09

Add GPIO support for jack reporting framework in ASoC, by using gpiolib calls. It's only required to append GPIO information (gpio number, irq handler and flags) to standard jack_pin declaration. GPIO request, data direction configuration and irq request are handled by the utility. The minimal GPIO information that can be provided is the gpio number, in that case default handler/flags will be used. Default handler queues a work that reads the current value in the gpio pin and informs to the jack framework. If the GPIO support is not required, the "gpio" field ot jack_pin structure must be set to NO_JACK_PIN_GPIO. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- include/sound/soc.h | 15 ++++++++++ sound/soc/soc-jack.c | 70 ++++++++++++++++++++++++++++++++++++++++++++++++- 2 files changed, 83 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 68d8149..846e2c1 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -16,6 +16,8 @@ #include <linux/platform_device.h> #include <linux/types.h> #include <linux/workqueue.h> +#include <linux/interrupt.h> +#include <linux/kernel.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/control.h> @@ -254,14 +256,27 @@ int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, * @pin: name of the pin to update * @mask: bits to check for in reported jack status * @invert: if non-zero then pin is enabled when status is not reported + * @gpio: gpio number associated to the pin (gpiolib calls will be used) + * @irqflags IRQ flags + * @handler: handler for servicing interrupt events on gpio pin */ struct snd_soc_jack_pin { + struct snd_soc_jack *jack; + struct snd_soc_jack_gpio *gpio_pin; struct list_head list; const char *pin; int mask; bool invert; + /* GPIO */ + unsigned int gpio; + unsigned int irq; + unsigned long irqflags; + irq_handler_t handler; + struct work_struct work; }; +#define NO_JACK_PIN_GPIO UINT_MAX + struct snd_soc_jack { struct snd_jack *jack; struct snd_soc_card *card; diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 8cc00c3..0d048b2 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -14,6 +14,9 @@ #include <sound/jack.h> #include <sound/soc.h> #include <sound/soc-dapm.h> +#include <linux/gpio.h> +#include <linux/interrupt.h> +#include <linux/workqueue.h> /** * snd_soc_jack_new - Create a new jack @@ -96,6 +99,32 @@ out: } EXPORT_SYMBOL_GPL(snd_soc_jack_report); +/* Default IRQ handler for a GPIO jack pin, it will queue a + * work that reads current value in GPIO pin and reports it + * to the jack framework. + */ +static irqreturn_t gpio_interrupt(int irq, void *data) +{ + struct snd_soc_jack_pin *pin = data; + + return IRQ_RETVAL(schedule_work(&pin->work)); +} + +static void gpio_work(struct work_struct *work) +{ + struct snd_soc_jack_pin *pin; + int report; + + pin = container_of(work, struct snd_soc_jack_pin, work); + report = pin->jack->status & pin->mask; + if (gpio_get_value(pin->gpio)) + report |= pin->mask; + else + report &= ~pin->mask; + + snd_soc_jack_report(pin->jack, report, pin->jack->jack->type); +} + /** * snd_soc_jack_add_pins - Associate DAPM pins with an ASoC jack * @@ -106,11 +135,18 @@ EXPORT_SYMBOL_GPL(snd_soc_jack_report); * After this function has been called the DAPM pins specified in the * pins array will have their status updated to reflect the current * state of the jack whenever the jack status is updated. + * + * A GPIO pin (using gpiolib) can be used to detect events. It requieres + * an IRQ handler and flags to be set in jack_pin structure; if they are + * not provided, default handler/flags will be used instead. If this + * feature is not desired, "gpio" field of jack_pin structure must be + * set to NO_JACK_PIN_GPIO. */ int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count, struct snd_soc_jack_pin *pins) { - int i; + unsigned int gpio = 0; + int i, ret = 0; for (i = 0; i < count; i++) { if (!pins[i].pin) { @@ -123,6 +159,32 @@ int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count, return -EINVAL; } + if (pins[i].gpio != NO_JACK_PIN_GPIO) { + pins[i].jack = jack; + gpio = pins[i].gpio; + ret = gpio_request(gpio, pins[i].pin); + if (ret) + return ret; + + ret = gpio_direction_input(gpio); + if (ret) + goto out; + pins[i].irq = gpio_to_irq(gpio); + /* If none set, use the default handler */ + if (!pins[i].handler) { + pins[i].handler = gpio_interrupt; + pins[i].irqflags = IRQF_TRIGGER_RISING | + IRQF_TRIGGER_FALLING; + INIT_WORK(&pins[i].work, gpio_work); + } + ret = request_irq(pins[i].irq, + pins[i].handler, + pins[i].irqflags, + jack->card->dev->driver->name, + &pins[i]); + if (ret) + goto out; + } INIT_LIST_HEAD(&pins[i].list); list_add(&(pins[i].list), &jack->pins); } @@ -133,6 +195,10 @@ int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count, */ snd_soc_jack_report(jack, 0, 0); - return 0; +out: + if (ret) + gpio_free(gpio); + + return ret; } EXPORT_SYMBOL_GPL(snd_soc_jack_add_pins); -- 1.5.4.3

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[alsa-devel] Plugging in headphones doesn't mute speakers, internal microphone does not work
by Emilio López 03 Mar '09

03 Mar '09

Hello, This is my first mail to the list, I hope you can help me. I have been talking with people on #alsa, but they couldn't help me, so that's why I'm writing to you. Well, as lspci says, my Acer Aspire 6930 laptop has an "Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 03)". With this card, and without adding any mode=... to /etc/modprobe.d/alsa-base, playback works perfectly, but when I plug in headphones, the integrated speakers do not mute. I can mute the speakers muting "front" and "side" and use the headphones, but it should be automatic. Setting mode=auto makes the headphones, when plugged, mute the speakers. But the problem with this is, that when using the speakers, I only get sound from the speaker that was "front" in the other mode. The microphone does not work in this mode either. I also tried other modes, including (but not limited to) acer, acer-aspire. I ran alsa-info.sh, here is the URL: http://www.alsa-project.org/db/?f=7910da439e8e190a8c2a6f18167d4879c74881c9 And before I forget, regarding the microphone, if I plug an external one on the microphone plug, I can record. And I tried the internal one on windows, and it works, so I believe it's an alsa problem. Looking forward to hearing from you! Emilio

2 2

Re: [alsa-devel] [PATCH] ASoC: make ops a pointer in 'struct snd_soc_dai'
by Timur Tabi 02 Mar '09

02 Mar '09

Eric Miao wrote: > sound/soc/fsl/fsl_ssi.c | 18 ++++--- I think you forgot sound/soc/fsl/mpc5200_psc_i2s.c -- Timur Tabi Linux kernel developer at Freescale

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