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Re: [alsa-devel] [REPOST] opti93x: conversion to use cs4231 library
by krzysztof.h1＠poczta.fm 07 Feb '08

07 Feb '08

Rene Herman napisał(a): > On 30-01-08 01:43, Rene Herman wrote: > > > 931 and 933 work and 930 (a non-PnP chip) is being a royal pain in the > > ass again. With the new code, I haven't yet been able to get it to work > > > -- no IRQ is firing. However, I also experienced this _once_ with the > > old driver so I'm not sure what's going on. When I now try to reproduce > > > with the old one, I can't. Maybe a timing difference. > > > > Will need further testing -- I have no time in the next few days. > > Hate it but will need to confirm that this is definitely a regression. > OPTi > 82C930A does not work with the cs4231-lib using driver, does work with > the > current driver. It loads, but gets no interrupts. > Thank you for testing. If it works with the old driver I will analyze what the old driver does for opti930. Thanks again, Krzysztof ---------------------------------------------------------------------- Zmus swojego faceta, zeby to przeczytal Kliknij >>> http://link.interia.pl/f1ceb

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[alsa-devel] [REPOST] opti93x: conversion to use cs4231 library
by Krzysztof Helt 07 Feb '08

07 Feb '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This patch converts the Opti93x driver to use the cs4231 library instead of just duplicating the code. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- This patch requires the patch which adds opti93x support to the cs4231 library. The patch is done against the current alsa hg tree, which means that the optin92x-ad1848.c file must be downloaded from the hg tree (it is newer by one patch than the file from 2.6.24-rc8-mm1 kernel). Regards, Krzysztof diff -urp linux-mm-ref/sound/isa/Kconfig linux-mm/sound/isa/Kconfig --- linux-mm-ref/sound/isa/Kconfig 2008-01-26 07:41:05.000000000 +0100 +++ linux-mm/sound/isa/Kconfig 2008-01-26 09:12:21.561626560 +0100 @@ -321,7 +321,7 @@ config SND_OPTI93X depends on SND select SND_OPL3_LIB select SND_MPU401_UART - select SND_PCM + select SND_CS4231_LIB help Say Y here to include support for soundcards based on Opti 82C93x chips. diff -urp linux-mm-ref/sound/isa/opti9xx/opti92x-ad1848.c linux-mm/sound/isa/opti9xx/opti92x-ad1848.c --- linux-mm-ref/sound/isa/opti9xx/opti92x-ad1848.c 2008-01-26 09:08:40.877305846 +0100 +++ linux-mm/sound/isa/opti9xx/opti92x-ad1848.c 2008-01-26 09:09:14.435180122 +0100 @@ -33,15 +33,10 @@ #include <asm/io.h> #include <asm/dma.h> #include <sound/core.h> -#ifdef CS4231 +#if defined(CS4231) || defined(OPTi93X) #include <sound/cs4231.h> #else -#ifndef OPTi93X #include <sound/ad1848.h> -#else -#include <sound/control.h> -#include <sound/pcm.h> -#endif /* OPTi93X */ #endif /* CS4231 */ #include <sound/mpu401.h> #include <sound/opl3.h> @@ -109,7 +104,6 @@ module_param(dma2, int, 0444); MODULE_PARM_DESC(dma2, "2nd dma # for opti9xx driver."); #endif /* CS4231 || OPTi93X */ -#define OPTi9XX_HW_DETECT 0 #define OPTi9XX_HW_82C928 1 #define OPTi9XX_HW_82C929 2 #define OPTi9XX_HW_82C924 3 @@ -123,105 +117,12 @@ MODULE_PARM_DESC(dma2, "2nd dma # for op #ifdef OPTi93X -#define OPTi93X_INDEX 0x00 -#define OPTi93X_DATA 0x01 #define OPTi93X_STATUS 0x02 -#define OPTi93X_DDATA 0x03 #define OPTi93X_PORT(chip, r) ((chip)->port + OPTi93X_##r) -#define OPTi93X_MIXOUT_LEFT 0x00 -#define OPTi93X_MIXOUT_RIGHT 0x01 -#define OPTi93X_CD_LEFT_INPUT 0x02 -#define OPTi93X_CD_RIGHT_INPUT 0x03 -#define OPTi930_AUX_LEFT_INPUT 0x04 -#define OPTi930_AUX_RIGHT_INPUT 0x05 -#define OPTi931_FM_LEFT_INPUT 0x04 -#define OPTi931_FM_RIGHT_INPUT 0x05 -#define OPTi93X_DAC_LEFT 0x06 -#define OPTi93X_DAC_RIGHT 0x07 -#define OPTi93X_PLAY_FORMAT 0x08 -#define OPTi93X_IFACE_CONF 0x09 -#define OPTi93X_PIN_CTRL 0x0a -#define OPTi93X_ERR_INIT 0x0b -#define OPTi93X_ID 0x0c -#define OPTi93X_PLAY_UPR_CNT 0x0e -#define OPTi93X_PLAY_LWR_CNT 0x0f -#define OPTi931_AUX_LEFT_INPUT 0x10 -#define OPTi931_AUX_RIGHT_INPUT 0x11 -#define OPTi93X_LINE_LEFT_INPUT 0x12 -#define OPTi93X_LINE_RIGHT_INPUT 0x13 -#define OPTi93X_MIC_LEFT_INPUT 0x14 -#define OPTi93X_MIC_RIGHT_INPUT 0x15 -#define OPTi93X_OUT_LEFT 0x16 -#define OPTi93X_OUT_RIGHT 0x17 -#define OPTi93X_CAPT_FORMAT 0x1c -#define OPTi93X_CAPT_UPR_CNT 0x1e -#define OPTi93X_CAPT_LWR_CNT 0x1f - -#define OPTi93X_TRD 0x20 -#define OPTi93X_MCE 0x40 -#define OPTi93X_INIT 0x80 - -#define OPTi93X_MIXOUT_MIC_GAIN 0x20 -#define OPTi93X_MIXOUT_LINE 0x00 -#define OPTi93X_MIXOUT_CD 0x40 -#define OPTi93X_MIXOUT_MIC 0x80 -#define OPTi93X_MIXOUT_MIXER 0xc0 - -#define OPTi93X_STEREO 0x10 -#define OPTi93X_LINEAR_8 0x00 -#define OPTi93X_ULAW_8 0x20 -#define OPTi93X_LINEAR_16_LIT 0x40 -#define OPTi93X_ALAW_8 0x60 -#define OPTi93X_ADPCM_16 0xa0 -#define OPTi93X_LINEAR_16_BIG 0xc0 - -#define OPTi93X_CAPTURE_PIO 0x80 -#define OPTi93X_PLAYBACK_PIO 0x40 -#define OPTi93X_AUTOCALIB 0x08 -#define OPTi93X_SINGLE_DMA 0x04 -#define OPTi93X_CAPTURE_ENABLE 0x02 -#define OPTi93X_PLAYBACK_ENABLE 0x01 - -#define OPTi93X_IRQ_ENABLE 0x02 - -#define OPTi93X_DMA_REQUEST 0x10 -#define OPTi93X_CALIB_IN_PROGRESS 0x20 - #define OPTi93X_IRQ_PLAYBACK 0x04 #define OPTi93X_IRQ_CAPTURE 0x08 - -struct snd_opti93x { - unsigned long port; - struct resource *res_port; - int irq; - int dma1; - int dma2; - - struct snd_opti9xx *chip; - unsigned short hardware; - unsigned char image[32]; - - unsigned char mce_bit; - unsigned short mode; - int mute; - - spinlock_t lock; - - struct snd_card *card; - struct snd_pcm *pcm; - struct snd_pcm_substream *playback_substream; - struct snd_pcm_substream *capture_substream; - unsigned int p_dma_size; - unsigned int c_dma_size; -}; - -#define OPTi93X_MODE_NONE 0x00 -#define OPTi93X_MODE_PLAY 0x01 -#define OPTi93X_MODE_CAPTURE 0x02 -#define OPTi93X_MODE_OPEN (OPTi93X_MODE_PLAY | OPTi93X_MODE_CAPTURE) - #endif /* OPTi93X */ struct snd_opti9xx { @@ -234,6 +135,7 @@ struct snd_opti9xx { unsigned long mc_base_size; #ifdef OPTi93X unsigned long mc_indir_index; + struct snd_cs4231 *codec; #endif /* OPTi93X */ unsigned long pwd_reg; @@ -654,979 +556,23 @@ __skip_mpu: #ifdef OPTi93X -static unsigned char snd_opti93x_default_image[32] = -{ - 0x00, /* 00/00 - l_mixout_outctrl */ - 0x00, /* 01/01 - r_mixout_outctrl */ - 0x88, /* 02/02 - l_cd_inctrl */ - 0x88, /* 03/03 - r_cd_inctrl */ - 0x88, /* 04/04 - l_a1/fm_inctrl */ - 0x88, /* 05/05 - r_a1/fm_inctrl */ - 0x80, /* 06/06 - l_dac_inctrl */ - 0x80, /* 07/07 - r_dac_inctrl */ - 0x00, /* 08/08 - ply_dataform_reg */ - 0x00, /* 09/09 - if_conf */ - 0x00, /* 0a/10 - pin_ctrl */ - 0x00, /* 0b/11 - err_init_reg */ - 0x0a, /* 0c/12 - id_reg */ - 0x00, /* 0d/13 - reserved */ - 0x00, /* 0e/14 - ply_upcount_reg */ - 0x00, /* 0f/15 - ply_lowcount_reg */ - 0x88, /* 10/16 - reserved/l_a1_inctrl */ - 0x88, /* 11/17 - reserved/r_a1_inctrl */ - 0x88, /* 12/18 - l_line_inctrl */ - 0x88, /* 13/19 - r_line_inctrl */ - 0x88, /* 14/20 - l_mic_inctrl */ - 0x88, /* 15/21 - r_mic_inctrl */ - 0x80, /* 16/22 - l_out_outctrl */ - 0x80, /* 17/23 - r_out_outctrl */ - 0x00, /* 18/24 - reserved */ - 0x00, /* 19/25 - reserved */ - 0x00, /* 1a/26 - reserved */ - 0x00, /* 1b/27 - reserved */ - 0x00, /* 1c/28 - cap_dataform_reg */ - 0x00, /* 1d/29 - reserved */ - 0x00, /* 1e/30 - cap_upcount_reg */ - 0x00 /* 1f/31 - cap_lowcount_reg */ -}; - - -static int snd_opti93x_busy_wait(struct snd_opti93x *chip) -{ - int timeout; - - for (timeout = 250; timeout-- > 0; udelay(10)) - if (!(inb(OPTi93X_PORT(chip, INDEX)) & OPTi93X_INIT)) - return 0; - - snd_printk("chip still busy.\n"); - return -EBUSY; -} - -static unsigned char snd_opti93x_in(struct snd_opti93x *chip, unsigned char reg) -{ - snd_opti93x_busy_wait(chip); - outb(chip->mce_bit | (reg & 0x1f), OPTi93X_PORT(chip, INDEX)); - return inb(OPTi93X_PORT(chip, DATA)); -} - -static void snd_opti93x_out(struct snd_opti93x *chip, unsigned char reg, - unsigned char value) -{ - snd_opti93x_busy_wait(chip); - outb(chip->mce_bit | (reg & 0x1f), OPTi93X_PORT(chip, INDEX)); - outb(value, OPTi93X_PORT(chip, DATA)); -} - -static void snd_opti93x_out_image(struct snd_opti93x *chip, unsigned char reg, - unsigned char value) -{ - snd_opti93x_out(chip, reg, chip->image[reg] = value); -} - -static void snd_opti93x_out_mask(struct snd_opti93x *chip, unsigned char reg, - unsigned char mask, unsigned char value) -{ - snd_opti93x_out_image(chip, reg, - (chip->image[reg] & ~mask) | (value & mask)); -} - - -static void snd_opti93x_mce_up(struct snd_opti93x *chip) -{ - snd_opti93x_busy_wait(chip); - - chip->mce_bit = OPTi93X_MCE; - if (!(inb(OPTi93X_PORT(chip, INDEX)) & OPTi93X_MCE)) - outb(chip->mce_bit, OPTi93X_PORT(chip, INDEX)); -} - -static void snd_opti93x_mce_down(struct snd_opti93x *chip) -{ - snd_opti93x_busy_wait(chip); - - chip->mce_bit = 0; - if (inb(OPTi93X_PORT(chip, INDEX)) & OPTi93X_MCE) - outb(chip->mce_bit, OPTi93X_PORT(chip, INDEX)); -} - -#define snd_opti93x_mute_reg(chip, reg, mute) \ - snd_opti93x_out(chip, reg, mute ? 0x80 : chip->image[reg]); - -static void snd_opti93x_mute(struct snd_opti93x *chip, int mute) -{ - mute = mute ? 1 : 0; - if (chip->mute == mute) - return; - - chip->mute = mute; - - snd_opti93x_mute_reg(chip, OPTi93X_CD_LEFT_INPUT, mute); - snd_opti93x_mute_reg(chip, OPTi93X_CD_RIGHT_INPUT, mute); - switch (chip->hardware) { - case OPTi9XX_HW_82C930: - snd_opti93x_mute_reg(chip, OPTi930_AUX_LEFT_INPUT, mute); - snd_opti93x_mute_reg(chip, OPTi930_AUX_RIGHT_INPUT, mute); - break; - case OPTi9XX_HW_82C931: - case OPTi9XX_HW_82C933: - snd_opti93x_mute_reg(chip, OPTi931_FM_LEFT_INPUT, mute); - snd_opti93x_mute_reg(chip, OPTi931_FM_RIGHT_INPUT, mute); - snd_opti93x_mute_reg(chip, OPTi931_AUX_LEFT_INPUT, mute); - snd_opti93x_mute_reg(chip, OPTi931_AUX_RIGHT_INPUT, mute); - } - snd_opti93x_mute_reg(chip, OPTi93X_DAC_LEFT, mute); - snd_opti93x_mute_reg(chip, OPTi93X_DAC_RIGHT, mute); - snd_opti93x_mute_reg(chip, OPTi93X_LINE_LEFT_INPUT, mute); - snd_opti93x_mute_reg(chip, OPTi93X_LINE_RIGHT_INPUT, mute); - snd_opti93x_mute_reg(chip, OPTi93X_MIC_LEFT_INPUT, mute); - snd_opti93x_mute_reg(chip, OPTi93X_MIC_RIGHT_INPUT, mute); - snd_opti93x_mute_reg(chip, OPTi93X_OUT_LEFT, mute); - snd_opti93x_mute_reg(chip, OPTi93X_OUT_RIGHT, mute); -} - - -static unsigned int snd_opti93x_get_count(unsigned char format, - unsigned int size) -{ - switch (format & 0xe0) { - case OPTi93X_LINEAR_16_LIT: - case OPTi93X_LINEAR_16_BIG: - size >>= 1; - break; - case OPTi93X_ADPCM_16: - return size >> 2; - } - return (format & OPTi93X_STEREO) ? (size >> 1) : size; -} - -static unsigned int rates[] = { 5512, 6615, 8000, 9600, 11025, 16000, - 18900, 22050, 27428, 32000, 33075, 37800, - 44100, 48000 }; -#define RATES ARRAY_SIZE(rates) - -static struct snd_pcm_hw_constraint_list hw_constraints_rates = { - .count = RATES, - .list = rates, - .mask = 0, -}; - -static unsigned char bits[] = { 0x01, 0x0f, 0x00, 0x0e, 0x03, 0x02, - 0x05, 0x07, 0x04, 0x06, 0x0d, 0x09, - 0x0b, 0x0c}; - -static unsigned char snd_opti93x_get_freq(unsigned int rate) -{ - unsigned int i; - - for (i = 0; i < RATES; i++) { - if (rate == rates[i]) - return bits[i]; - } - snd_BUG(); - return bits[RATES-1]; -} - -static unsigned char snd_opti93x_get_format(struct snd_opti93x *chip, - unsigned int format, int channels) -{ - unsigned char retval = OPTi93X_LINEAR_8; - - switch (format) { - case SNDRV_PCM_FORMAT_MU_LAW: - retval = OPTi93X_ULAW_8; - break; - case SNDRV_PCM_FORMAT_A_LAW: - retval = OPTi93X_ALAW_8; - break; - case SNDRV_PCM_FORMAT_S16_LE: - retval = OPTi93X_LINEAR_16_LIT; - break; - case SNDRV_PCM_FORMAT_S16_BE: - retval = OPTi93X_LINEAR_16_BIG; - break; - case SNDRV_PCM_FORMAT_IMA_ADPCM: - retval = OPTi93X_ADPCM_16; - } - return (channels > 1) ? (retval | OPTi93X_STEREO) : retval; -} - - -static void snd_opti93x_playback_format(struct snd_opti93x *chip, unsigned char fmt) -{ - unsigned char mask; - - snd_opti93x_mute(chip, 1); - - snd_opti93x_mce_up(chip); - mask = (chip->mode & OPTi93X_MODE_CAPTURE) ? 0xf0 : 0xff; - snd_opti93x_out_mask(chip, OPTi93X_PLAY_FORMAT, mask, fmt); - snd_opti93x_mce_down(chip); - - snd_opti93x_mute(chip, 0); -} - -static void snd_opti93x_capture_format(struct snd_opti93x *chip, unsigned char fmt) -{ - snd_opti93x_mute(chip, 1); - - snd_opti93x_mce_up(chip); - if (!(chip->mode & OPTi93X_MODE_PLAY)) - snd_opti93x_out_mask(chip, OPTi93X_PLAY_FORMAT, 0x0f, fmt); - else - fmt = chip->image[OPTi93X_PLAY_FORMAT] & 0xf0; - snd_opti93x_out_image(chip, OPTi93X_CAPT_FORMAT, fmt); - snd_opti93x_mce_down(chip); - - snd_opti93x_mute(chip, 0); -} - - -static int snd_opti93x_open(struct snd_opti93x *chip, unsigned int mode) -{ - unsigned long flags; - - spin_lock_irqsave(&chip->lock, flags); - - if (chip->mode & mode) { - spin_unlock_irqrestore(&chip->lock, flags); - return -EAGAIN; - } - - if (!(chip->mode & OPTi93X_MODE_OPEN)) { - outb(0x00, OPTi93X_PORT(chip, STATUS)); - snd_opti93x_out_mask(chip, OPTi93X_PIN_CTRL, - OPTi93X_IRQ_ENABLE, OPTi93X_IRQ_ENABLE); - chip->mode = mode; - } - else - chip->mode |= mode; - - spin_unlock_irqrestore(&chip->lock, flags); - return 0; -} - -static void snd_opti93x_close(struct snd_opti93x *chip, unsigned int mode) -{ - unsigned long flags; - - spin_lock_irqsave(&chip->lock, flags); - - chip->mode &= ~mode; - if (chip->mode & OPTi93X_MODE_OPEN) { - spin_unlock_irqrestore(&chip->lock, flags); - return; - } - - snd_opti93x_mute(chip, 1); - - outb(0, OPTi93X_PORT(chip, STATUS)); - snd_opti93x_out_mask(chip, OPTi93X_PIN_CTRL, OPTi93X_IRQ_ENABLE, - ~OPTi93X_IRQ_ENABLE); - - snd_opti93x_mce_up(chip); - snd_opti93x_out_image(chip, OPTi93X_IFACE_CONF, 0x00); - snd_opti93x_mce_down(chip); - chip->mode = 0; - - snd_opti93x_mute(chip, 0); - spin_unlock_irqrestore(&chip->lock, flags); -} - -static int snd_opti93x_trigger(struct snd_pcm_substream *substream, - unsigned char what, int cmd) -{ - struct snd_opti93x *chip = snd_pcm_substream_chip(substream); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_STOP: - { - unsigned int what = 0; - struct snd_pcm_substream *s; - snd_pcm_group_for_each_entry(s, substream) { - if (s == chip->playback_substream) { - what |= OPTi93X_PLAYBACK_ENABLE; - snd_pcm_trigger_done(s, substream); - } else if (s == chip->capture_substream) { - what |= OPTi93X_CAPTURE_ENABLE; - snd_pcm_trigger_done(s, substream); - } - } - spin_lock(&chip->lock); - if (cmd == SNDRV_PCM_TRIGGER_START) { - snd_opti93x_out_mask(chip, OPTi93X_IFACE_CONF, what, what); - if (what & OPTi93X_CAPTURE_ENABLE) - udelay(50); - } else - snd_opti93x_out_mask(chip, OPTi93X_IFACE_CONF, what, 0x00); - spin_unlock(&chip->lock); - break; - } - default: - return -EINVAL; - } - return 0; -} - -static int snd_opti93x_playback_trigger(struct snd_pcm_substream *substream, int cmd) -{ - return snd_opti93x_trigger(substream, - OPTi93X_PLAYBACK_ENABLE, cmd); -} - -static int snd_opti93x_capture_trigger(struct snd_pcm_substream *substream, int cmd) -{ - return snd_opti93x_trigger(substream, - OPTi93X_CAPTURE_ENABLE, cmd); -} - -static int snd_opti93x_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params) -{ - return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); -} - - -static int snd_opti93x_hw_free(struct snd_pcm_substream *substream) -{ - snd_pcm_lib_free_pages(substream); - return 0; -} - - -static int snd_opti93x_playback_prepare(struct snd_pcm_substream *substream) -{ - struct snd_opti93x *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - unsigned long flags; - unsigned char format; - unsigned int count = snd_pcm_lib_period_bytes(substream); - unsigned int size = snd_pcm_lib_buffer_bytes(substream); - - spin_lock_irqsave(&chip->lock, flags); - - chip->p_dma_size = size; - snd_opti93x_out_mask(chip, OPTi93X_IFACE_CONF, - OPTi93X_PLAYBACK_ENABLE | OPTi93X_PLAYBACK_PIO, - ~(OPTi93X_PLAYBACK_ENABLE | OPTi93X_PLAYBACK_PIO)); - - snd_dma_program(chip->dma1, runtime->dma_addr, size, - DMA_MODE_WRITE | DMA_AUTOINIT); - - format = snd_opti93x_get_freq(runtime->rate); - format |= snd_opti93x_get_format(chip, runtime->format, - runtime->channels); - snd_opti93x_playback_format(chip, format); - format = chip->image[OPTi93X_PLAY_FORMAT]; - - count = snd_opti93x_get_count(format, count) - 1; - snd_opti93x_out_image(chip, OPTi93X_PLAY_LWR_CNT, count); - snd_opti93x_out_image(chip, OPTi93X_PLAY_UPR_CNT, count >> 8); - - spin_unlock_irqrestore(&chip->lock, flags); - return 0; -} - -static int snd_opti93x_capture_prepare(struct snd_pcm_substream *substream) -{ - struct snd_opti93x *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - unsigned long flags; - unsigned char format; - unsigned int count = snd_pcm_lib_period_bytes(substream); - unsigned int size = snd_pcm_lib_buffer_bytes(substream); - - spin_lock_irqsave(&chip->lock, flags); - - chip->c_dma_size = size; - snd_opti93x_out_mask(chip, OPTi93X_IFACE_CONF, - OPTi93X_CAPTURE_ENABLE | OPTi93X_CAPTURE_PIO, 0); - - snd_dma_program(chip->dma2, runtime->dma_addr, size, - DMA_MODE_READ | DMA_AUTOINIT); - - format = snd_opti93x_get_freq(runtime->rate); - format |= snd_opti93x_get_format(chip, runtime->format, - runtime->channels); - snd_opti93x_capture_format(chip, format); - format = chip->image[OPTi93X_CAPT_FORMAT]; - - count = snd_opti93x_get_count(format, count) - 1; - snd_opti93x_out_image(chip, OPTi93X_CAPT_LWR_CNT, count); - snd_opti93x_out_image(chip, OPTi93X_CAPT_UPR_CNT, count >> 8); - - spin_unlock_irqrestore(&chip->lock, flags); - return 0; -} - -static snd_pcm_uframes_t snd_opti93x_playback_pointer(struct snd_pcm_substream *substream) -{ - struct snd_opti93x *chip = snd_pcm_substream_chip(substream); - size_t ptr; - - if (!(chip->image[OPTi93X_IFACE_CONF] & OPTi93X_PLAYBACK_ENABLE)) - return 0; - - ptr = snd_dma_pointer(chip->dma1, chip->p_dma_size); - return bytes_to_frames(substream->runtime, ptr); -} - -static snd_pcm_uframes_t snd_opti93x_capture_pointer(struct snd_pcm_substream *substream) -{ - struct snd_opti93x *chip = snd_pcm_substream_chip(substream); - size_t ptr; - - if (!(chip->image[OPTi93X_IFACE_CONF] & OPTi93X_CAPTURE_ENABLE)) - return 0; - - ptr = snd_dma_pointer(chip->dma2, chip->c_dma_size); - return bytes_to_frames(substream->runtime, ptr); -} - - -static void snd_opti93x_overrange(struct snd_opti93x *chip) -{ - unsigned long flags; - - spin_lock_irqsave(&chip->lock, flags); - - if (snd_opti93x_in(chip, OPTi93X_ERR_INIT) & (0x08 | 0x02)) - chip->capture_substream->runtime->overrange++; - - spin_unlock_irqrestore(&chip->lock, flags); -} - static irqreturn_t snd_opti93x_interrupt(int irq, void *dev_id) { - struct snd_opti93x *codec = dev_id; + struct snd_cs4231 *codec = dev_id; + struct snd_opti9xx *chip = codec->card->private_data; unsigned char status; - status = snd_opti9xx_read(codec->chip, OPTi9XX_MC_REG(11)); + status = snd_opti9xx_read(chip, OPTi9XX_MC_REG(11)); if ((status & OPTi93X_IRQ_PLAYBACK) && codec->playback_substream) snd_pcm_period_elapsed(codec->playback_substream); if ((status & OPTi93X_IRQ_CAPTURE) && codec->capture_substream) { - snd_opti93x_overrange(codec); + snd_cs4231_overrange(codec); snd_pcm_period_elapsed(codec->capture_substream); } outb(0x00, OPTi93X_PORT(codec, STATUS)); return IRQ_HANDLED; } - -static struct snd_pcm_hardware snd_opti93x_playback = { - .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_SYNC_START), - .formats = (SNDRV_PCM_FMTBIT_MU_LAW | SNDRV_PCM_FMTBIT_A_LAW | SNDRV_PCM_FMTBIT_IMA_ADPCM | - SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE), - .rates = SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_8000_48000, - .rate_min = 5512, - .rate_max = 48000, - .channels_min = 1, - .channels_max = 2, - .buffer_bytes_max = (128*1024), - .period_bytes_min = 64, - .period_bytes_max = (128*1024), - .periods_min = 1, - .periods_max = 1024, - .fifo_size = 0, -}; - -static struct snd_pcm_hardware snd_opti93x_capture = { - .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_SYNC_START), - .formats = (SNDRV_PCM_FMTBIT_MU_LAW | SNDRV_PCM_FMTBIT_A_LAW | SNDRV_PCM_FMTBIT_IMA_ADPCM | - SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE), - .rates = SNDRV_PCM_RATE_8000_48000, - .rate_min = 5512, - .rate_max = 48000, - .channels_min = 1, - .channels_max = 2, - .buffer_bytes_max = (128*1024), - .period_bytes_min = 64, - .period_bytes_max = (128*1024), - .periods_min = 1, - .periods_max = 1024, - .fifo_size = 0, -}; - -static int snd_opti93x_playback_open(struct snd_pcm_substream *substream) -{ - int error; - struct snd_opti93x *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - - if ((error = snd_opti93x_open(chip, OPTi93X_MODE_PLAY)) < 0) - return error; - snd_pcm_set_sync(substream); - chip->playback_substream = substream; - runtime->hw = snd_opti93x_playback; - snd_pcm_limit_isa_dma_size(chip->dma1, &runtime->hw.buffer_bytes_max); - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); - return error; -} - -static int snd_opti93x_capture_open(struct snd_pcm_substream *substream) -{ - int error; - struct snd_opti93x *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - - if ((error = snd_opti93x_open(chip, OPTi93X_MODE_CAPTURE)) < 0) - return error; - runtime->hw = snd_opti93x_capture; - snd_pcm_set_sync(substream); - chip->capture_substream = substream; - snd_pcm_limit_isa_dma_size(chip->dma2, &runtime->hw.buffer_bytes_max); - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); - return error; -} - -static int snd_opti93x_playback_close(struct snd_pcm_substream *substream) -{ - struct snd_opti93x *chip = snd_pcm_substream_chip(substream); - - chip->playback_substream = NULL; - snd_opti93x_close(chip, OPTi93X_MODE_PLAY); - return 0; -} - -static int snd_opti93x_capture_close(struct snd_pcm_substream *substream) -{ - struct snd_opti93x *chip = snd_pcm_substream_chip(substream); - - chip->capture_substream = NULL; - snd_opti93x_close(chip, OPTi93X_MODE_CAPTURE); - return 0; -} - - -static void snd_opti93x_init(struct snd_opti93x *chip) -{ - unsigned long flags; - int i; - - spin_lock_irqsave(&chip->lock, flags); - snd_opti93x_mce_up(chip); - - for (i = 0; i < 32; i++) - snd_opti93x_out_image(chip, i, snd_opti93x_default_image[i]); - - snd_opti93x_mce_down(chip); - spin_unlock_irqrestore(&chip->lock, flags); -} - -static int snd_opti93x_probe(struct snd_opti93x *chip) -{ - unsigned long flags; - unsigned char val; - - spin_lock_irqsave(&chip->lock, flags); - val = snd_opti93x_in(chip, OPTi93X_ID) & 0x0f; - spin_unlock_irqrestore(&chip->lock, flags); - - return (val == 0x0a) ? 0 : -ENODEV; -} - -static int snd_opti93x_free(struct snd_opti93x *chip) -{ - release_and_free_resource(chip->res_port); - if (chip->dma1 >= 0) { - disable_dma(chip->dma1); - free_dma(chip->dma1); - } - if (chip->dma2 >= 0) { - disable_dma(chip->dma2); - free_dma(chip->dma2); - } - if (chip->irq >= 0) { - free_irq(chip->irq, chip); - } - kfree(chip); - return 0; -} - -static int snd_opti93x_dev_free(struct snd_device *device) -{ - struct snd_opti93x *chip = device->device_data; - return snd_opti93x_free(chip); -} - -static const char *snd_opti93x_chip_id(struct snd_opti93x *codec) -{ - switch (codec->hardware) { - case OPTi9XX_HW_82C930: return "82C930"; - case OPTi9XX_HW_82C931: return "82C931"; - case OPTi9XX_HW_82C933: return "82C933"; - default: return "???"; - } -} - -static int snd_opti93x_create(struct snd_card *card, struct snd_opti9xx *chip, - int dma1, int dma2, - struct snd_opti93x **rcodec) -{ - static struct snd_device_ops ops = { - .dev_free = snd_opti93x_dev_free, - }; - int error; - struct snd_opti93x *codec; - - *rcodec = NULL; - codec = kzalloc(sizeof(*codec), GFP_KERNEL); - if (codec == NULL) - return -ENOMEM; - codec->irq = -1; - codec->dma1 = -1; - codec->dma2 = -1; - - if ((codec->res_port = request_region(chip->wss_base + 4, 4, "OPTI93x CODEC")) == NULL) { - snd_printk(KERN_ERR "opti9xx: can't grab port 0x%lx\n", chip->wss_base + 4); - snd_opti93x_free(codec); - return -EBUSY; - } - if (request_dma(dma1, "OPTI93x - 1")) { - snd_printk(KERN_ERR "opti9xx: can't grab DMA1 %d\n", dma1); - snd_opti93x_free(codec); - return -EBUSY; - } - codec->dma1 = chip->dma1; - if (request_dma(dma2, "OPTI93x - 2")) { - snd_printk(KERN_ERR "opti9xx: can't grab DMA2 %d\n", dma2); - snd_opti93x_free(codec); - return -EBUSY; - } - codec->dma2 = chip->dma2; - - if (request_irq(chip->irq, snd_opti93x_interrupt, IRQF_DISABLED, DEV_NAME" - WSS", codec)) { - snd_printk(KERN_ERR "opti9xx: can't grab IRQ %d\n", chip->irq); - snd_opti93x_free(codec); - return -EBUSY; - } - - codec->card = card; - codec->port = chip->wss_base + 4; - codec->irq = chip->irq; - - spin_lock_init(&codec->lock); - codec->hardware = chip->hardware; - codec->chip = chip; - - if ((error = snd_opti93x_probe(codec))) { - snd_opti93x_free(codec); - return error; - } - - snd_opti93x_init(codec); - - /* Register device */ - if ((error = snd_device_new(card, SNDRV_DEV_LOWLEVEL, codec, &ops)) < 0) { - snd_opti93x_free(codec); - return error; - } - - *rcodec = codec; - return 0; -} - -static struct snd_pcm_ops snd_opti93x_playback_ops = { - .open = snd_opti93x_playback_open, - .close = snd_opti93x_playback_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_opti93x_hw_params, - .hw_free = snd_opti93x_hw_free, - .prepare = snd_opti93x_playback_prepare, - .trigger = snd_opti93x_playback_trigger, - .pointer = snd_opti93x_playback_pointer, -}; - -static struct snd_pcm_ops snd_opti93x_capture_ops = { - .open = snd_opti93x_capture_open, - .close = snd_opti93x_capture_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_opti93x_hw_params, - .hw_free = snd_opti93x_hw_free, - .prepare = snd_opti93x_capture_prepare, - .trigger = snd_opti93x_capture_trigger, - .pointer = snd_opti93x_capture_pointer, -}; - -static int snd_opti93x_pcm(struct snd_opti93x *codec, int device, struct snd_pcm **rpcm) -{ - int error; - struct snd_pcm *pcm; - - if ((error = snd_pcm_new(codec->card, "OPTi 82C93X", device, 1, 1, &pcm)) < 0) - return error; - - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_opti93x_playback_ops); - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_opti93x_capture_ops); - - pcm->private_data = codec; - pcm->info_flags = SNDRV_PCM_INFO_JOINT_DUPLEX; - - strcpy(pcm->name, snd_opti93x_chip_id(codec)); - - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_isa_data(), - 64*1024, codec->dma1 > 3 || codec->dma2 > 3 ? 128*1024 : 64*1024); - - codec->pcm = pcm; - if (rpcm) - *rpcm = pcm; - return 0; -} - -/* - * MIXER part - */ - -static int snd_opti93x_info_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - static char *texts[4] = { - "Line1", "Aux", "Mic", "Mix" - }; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 2; - uinfo->value.enumerated.items = 4; - if (uinfo->value.enumerated.item > 3) - uinfo->value.enumerated.item = 3; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; -} - -static int snd_opti93x_get_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_opti93x *chip = snd_kcontrol_chip(kcontrol); - unsigned long flags; - - spin_lock_irqsave(&chip->lock, flags); - ucontrol->value.enumerated.item[0] = (chip->image[OPTi93X_MIXOUT_LEFT] & OPTi93X_MIXOUT_MIXER) >> 6; - ucontrol->value.enumerated.item[1] = (chip->image[OPTi93X_MIXOUT_RIGHT] & OPTi93X_MIXOUT_MIXER) >> 6; - spin_unlock_irqrestore(&chip->lock, flags); - return 0; -} - -static int snd_opti93x_put_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_opti93x *chip = snd_kcontrol_chip(kcontrol); - unsigned long flags; - unsigned short left, right; - int change; - - if (ucontrol->value.enumerated.item[0] > 3 || - ucontrol->value.enumerated.item[1] > 3) - return -EINVAL; - left = ucontrol->value.enumerated.item[0] << 6; - right = ucontrol->value.enumerated.item[1] << 6; - spin_lock_irqsave(&chip->lock, flags); - left = (chip->image[OPTi93X_MIXOUT_LEFT] & ~OPTi93X_MIXOUT_MIXER) | left; - right = (chip->image[OPTi93X_MIXOUT_RIGHT] & ~OPTi93X_MIXOUT_MIXER) | right; - change = left != chip->image[OPTi93X_MIXOUT_LEFT] || - right != chip->image[OPTi93X_MIXOUT_RIGHT]; - snd_opti93x_out_image(chip, OPTi93X_MIXOUT_LEFT, left); - snd_opti93x_out_image(chip, OPTi93X_MIXOUT_RIGHT, right); - spin_unlock_irqrestore(&chip->lock, flags); - return change; -} - -#if 0 - -#define OPTi93X_SINGLE(xname, xindex, reg, shift, mask, invert) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ - .info = snd_opti93x_info_single, \ - .get = snd_opti93x_get_single, .put = snd_opti93x_put_single, \ - .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24) } - -static int snd_opti93x_info_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - int mask = (kcontrol->private_value >> 16) & 0xff; - - uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = mask; - return 0; -} - -static int snd_opti93x_get_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_opti93x *chip = snd_kcontrol_chip(kcontrol); - unsigned long flags; - int reg = kcontrol->private_value & 0xff; - int shift = (kcontrol->private_value >> 8) & 0xff; - int mask = (kcontrol->private_value >> 16) & 0xff; - int invert = (kcontrol->private_value >> 24) & 0xff; - - spin_lock_irqsave(&chip->lock, flags); - ucontrol->value.integer.value[0] = (chip->image[reg] >> shift) & mask; - spin_unlock_irqrestore(&chip->lock, flags); - if (invert) - ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0]; - return 0; -} - -static int snd_opti93x_put_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_opti93x *chip = snd_kcontrol_chip(kcontrol); - unsigned long flags; - int reg = kcontrol->private_value & 0xff; - int shift = (kcontrol->private_value >> 8) & 0xff; - int mask = (kcontrol->private_value >> 16) & 0xff; - int invert = (kcontrol->private_value >> 24) & 0xff; - int change; - unsigned short val; - - val = (ucontrol->value.integer.value[0] & mask); - if (invert) - val = mask - val; - val <<= shift; - spin_lock_irqsave(&chip->lock, flags); - val = (chip->image[reg] & ~(mask << shift)) | val; - change = val != chip->image[reg]; - snd_opti93x_out(chip, reg, val); - spin_unlock_irqrestore(&chip->lock, flags); - return change; -} - -#endif /* single */ - -#define OPTi93X_DOUBLE(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ - .info = snd_opti93x_info_double, \ - .get = snd_opti93x_get_double, .put = snd_opti93x_put_double, \ - .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | (shift_right << 19) | (mask << 24) | (invert << 22) } - -#define OPTi93X_DOUBLE_INVERT_INVERT(xctl) \ - do { xctl.private_value ^= 22; } while (0) -#define OPTi93X_DOUBLE_CHANGE_REGS(xctl, left_reg, right_reg) \ - do { xctl.private_value &= ~0x0000ffff; \ - xctl.private_value |= left_reg | (right_reg << 8); } while (0) - -static int snd_opti93x_info_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - int mask = (kcontrol->private_value >> 24) & 0xff; - - uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 2; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = mask; - return 0; -} - -static int snd_opti93x_get_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_opti93x *chip = snd_kcontrol_chip(kcontrol); - unsigned long flags; - int left_reg = kcontrol->private_value & 0xff; - int right_reg = (kcontrol->private_value >> 8) & 0xff; - int shift_left = (kcontrol->private_value >> 16) & 0x07; - int shift_right = (kcontrol->private_value >> 19) & 0x07; - int mask = (kcontrol->private_value >> 24) & 0xff; - int invert = (kcontrol->private_value >> 22) & 1; - - spin_lock_irqsave(&chip->lock, flags); - ucontrol->value.integer.value[0] = (chip->image[left_reg] >> shift_left) & mask; - ucontrol->value.integer.value[1] = (chip->image[right_reg] >> shift_right) & mask; - spin_unlock_irqrestore(&chip->lock, flags); - if (invert) { - ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0]; - ucontrol->value.integer.value[1] = mask - ucontrol->value.integer.value[1]; - } - return 0; -} - -static int snd_opti93x_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_opti93x *chip = snd_kcontrol_chip(kcontrol); - unsigned long flags; - int left_reg = kcontrol->private_value & 0xff; - int right_reg = (kcontrol->private_value >> 8) & 0xff; - int shift_left = (kcontrol->private_value >> 16) & 0x07; - int shift_right = (kcontrol->private_value >> 19) & 0x07; - int mask = (kcontrol->private_value >> 24) & 0xff; - int invert = (kcontrol->private_value >> 22) & 1; - int change; - unsigned short val1, val2; - - val1 = ucontrol->value.integer.value[0] & mask; - val2 = ucontrol->value.integer.value[1] & mask; - if (invert) { - val1 = mask - val1; - val2 = mask - val2; - } - val1 <<= shift_left; - val2 <<= shift_right; - spin_lock_irqsave(&chip->lock, flags); - val1 = (chip->image[left_reg] & ~(mask << shift_left)) | val1; - val2 = (chip->image[right_reg] & ~(mask << shift_right)) | val2; - change = val1 != chip->image[left_reg] || val2 != chip->image[right_reg]; - snd_opti93x_out_image(chip, left_reg, val1); - snd_opti93x_out_image(chip, right_reg, val2); - spin_unlock_irqrestore(&chip->lock, flags); - return change; -} - -static struct snd_kcontrol_new snd_opti93x_controls[] __devinitdata = { -OPTi93X_DOUBLE("Master Playback Switch", 0, OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 7, 7, 1, 1), -OPTi93X_DOUBLE("Master Playback Volume", 0, OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 1, 1, 31, 1), -OPTi93X_DOUBLE("PCM Playback Switch", 0, OPTi93X_DAC_LEFT, OPTi93X_DAC_RIGHT, 7, 7, 1, 1), -OPTi93X_DOUBLE("PCM Playback Volume", 0, OPTi93X_DAC_LEFT, OPTi93X_DAC_RIGHT, 0, 0, 31, 1), -OPTi93X_DOUBLE("FM Playback Switch", 0, OPTi931_FM_LEFT_INPUT, OPTi931_FM_RIGHT_INPUT, 7, 7, 1, 1), -OPTi93X_DOUBLE("FM Playback Volume", 0, OPTi931_FM_LEFT_INPUT, OPTi931_FM_RIGHT_INPUT, 1, 1, 15, 1), -OPTi93X_DOUBLE("Line Playback Switch", 0, OPTi93X_LINE_LEFT_INPUT, OPTi93X_LINE_RIGHT_INPUT, 7, 7, 1, 1), -OPTi93X_DOUBLE("Line Playback Volume", 0, OPTi93X_LINE_LEFT_INPUT, OPTi93X_LINE_RIGHT_INPUT, 1, 1, 15, 1), -OPTi93X_DOUBLE("Mic Playback Switch", 0, OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 7, 7, 1, 1), -OPTi93X_DOUBLE("Mic Playback Volume", 0, OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 1, 1, 15, 1), -OPTi93X_DOUBLE("Mic Boost", 0, OPTi93X_MIXOUT_LEFT, OPTi93X_MIXOUT_RIGHT, 5, 5, 1, 1), -OPTi93X_DOUBLE("CD Playback Switch", 0, OPTi93X_CD_LEFT_INPUT, OPTi93X_CD_RIGHT_INPUT, 7, 7, 1, 1), -OPTi93X_DOUBLE("CD Playback Volume", 0, OPTi93X_CD_LEFT_INPUT, OPTi93X_CD_RIGHT_INPUT, 1, 1, 15, 1), -OPTi93X_DOUBLE("Aux Playback Switch", 0, OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 7, 7, 1, 1), -OPTi93X_DOUBLE("Aux Playback Volume", 0, OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 1, 1, 15, 1), -OPTi93X_DOUBLE("Capture Volume", 0, OPTi93X_MIXOUT_LEFT, OPTi93X_MIXOUT_RIGHT, 0, 0, 15, 0), -{ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = snd_opti93x_info_mux, - .get = snd_opti93x_get_mux, - .put = snd_opti93x_put_mux, -} -}; - -static int __devinit snd_opti93x_mixer(struct snd_opti93x *chip) -{ - struct snd_card *card; - struct snd_kcontrol_new knew; - int err; - unsigned int idx; - - snd_assert(chip != NULL && chip->card != NULL, return -EINVAL); - - card = chip->card; - - strcpy(card->mixername, snd_opti93x_chip_id(chip)); - - for (idx = 0; idx < ARRAY_SIZE(snd_opti93x_controls); idx++) { - knew = snd_opti93x_controls[idx]; - if (chip->hardware == OPTi9XX_HW_82C930) { - if (strstr(knew.name, "FM")) /* skip FM controls */ - continue; - else if (strcmp(knew.name, "Mic Playback Volume")) - OPTi93X_DOUBLE_INVERT_INVERT(knew); - else if (strstr(knew.name, "Aux")) - OPTi93X_DOUBLE_CHANGE_REGS(knew, OPTi930_AUX_LEFT_INPUT, OPTi930_AUX_RIGHT_INPUT); - else if (strcmp(knew.name, "PCM Playback Volume")) - OPTi93X_DOUBLE_INVERT_INVERT(knew); - else if (strcmp(knew.name, "Master Playback Volume")) - OPTi93X_DOUBLE_INVERT_INVERT(knew); - } - if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_opti93x_controls[idx], chip))) < 0) - return err; - } - return 0; -} - #endif /* OPTi93X */ static int __devinit snd_card_opti9xx_detect(struct snd_card *card, @@ -1739,8 +685,16 @@ static void snd_card_opti9xx_free(struct { struct snd_opti9xx *chip = card->private_data; - if (chip) + if (chip) { +#ifdef OPTi93X + struct snd_cs4231 *codec = chip->codec; + if (codec->irq > 0) { + disable_irq(codec->irq); + free_irq(codec->irq, codec); + } +#endif release_and_free_resource(chip->res_mc_base); + } } static int __devinit snd_opti9xx_probe(struct snd_card *card) @@ -1748,11 +702,11 @@ static int __devinit snd_opti9xx_probe(s static long possible_ports[] = {0x530, 0xe80, 0xf40, 0x604, -1}; int error; struct snd_opti9xx *chip = card->private_data; -#if defined(OPTi93X) - struct snd_opti93x *codec; -#elif defined(CS4231) +#if defined(CS4231) || defined(OPTi93X) struct snd_cs4231 *codec; +#ifdef CS4231 struct snd_timer *timer; +#endif #else struct snd_ad1848 *codec; #endif @@ -1784,26 +738,31 @@ static int __devinit snd_opti9xx_probe(s if ((error = snd_opti9xx_configure(chip))) return error; -#if defined(OPTi93X) - if ((error = snd_opti93x_create(card, chip, chip->dma1, chip->dma2, &codec))) - return error; - if ((error = snd_opti93x_pcm(codec, 0, &pcm)) < 0) - return error; - if ((error = snd_opti93x_mixer(codec)) < 0) - return error; -#elif defined(CS4231) +#if defined(CS4231) || defined(OPTi93X) if ((error = snd_cs4231_create(card, chip->wss_base + 4, -1, chip->irq, chip->dma1, chip->dma2, - CS4231_HW_DETECT, - 0, +#ifdef CS4231 + CS4231_HW_DETECT, 0, +#else /* OPTi93x */ + CS4231_HW_OPTI93X, CS4231_HWSHARE_IRQ, +#endif &codec)) < 0) return error; + chip->codec = codec; if ((error = snd_cs4231_pcm(codec, 0, &pcm)) < 0) return error; if ((error = snd_cs4231_mixer(codec)) < 0) return error; +#ifdef CS4231 if ((error = snd_cs4231_timer(codec, 0, &timer)) < 0) return error; +#else /* OPTI93X */ + error = request_irq(chip->irq, snd_opti93x_interrupt, IRQF_DISABLED, DEV_NAME" - WSS", codec); + if (error < 0) { + snd_printk(KERN_ERR "opti9xx: can't grab IRQ %d\n", chip->irq); + return error; + } +#endif #else if ((error = snd_ad1848_create(card, chip->wss_base + 4, chip->irq, chip->dma1, ---------------------------------------------------------------------- Nadchodzi galaktyczna wojna! Sprawdz >> http://link.interia.pl/f1ce0

4 6

[alsa-devel] Hg update problems
by Eliot Blennerhassett 07 Feb '08

07 Feb '08

Hi Takashi, Wierd, I can see your checkin in a browser at http://hg.alsa-project.org/alsa-driver/ 14 hours ago: asihpi version 3.09.09 changeset 3506: e600ecb8fbd7 tag: tip author: tiwai Also I see it in a clean clone of the repo. but hg pull on my original repo still gets nothing same as before, head is out of date - details: eliot@zaphod:~/src/alsahg/alsa-driver$ hg -v --debug heads changeset: 3507:86adcd0d6d849d5f8d41448d3cd4d1a8660d6214 tag: tip description: Added tag v1.0.16rc2 for changeset 1dc44ca35555 Tracing (strace) further I see that Hg queries the repository like this http://hg.alsa-project.org/alsa-driver/?cmd=heads and gets back 86adcd0d6d849d5f8d41448d3cd4d1a8660d6214 So the remote repo is telling me that my local is up to date, even though the web interface shows more recent tip. Very wierd... Anyone else see this or shed any light on it? -- Eliot

3 3

[alsa-devel] problems with es1688 isa card on etch and lenny (ok with sarge)
by Norayr Chilingaryan 06 Feb '08

06 Feb '08

Hello, I have an old notebook with ISA sound card which is detected by alsaconf as es18xx, or es1688 or sb16 and works. The problem is appeared when I replaced Debian Sarge with Debian Etch, and when I upgraded Etch to Lenny (i. e. with alsa versions in that distros). So, alsaconf detects and configures a sound driver, it works, but after restart to have a sound I must run alsaconf again. I think something is changed in the recent alsa distributions which probably deals with debian configuration of isa cards. Thank you Norayr --- norayr.livejournal.com

3 2

[alsa-devel] ESI Maya44 initial patch
by Rainer Zimmermann 06 Feb '08

06 Feb '08

Hello, this is the first testing version of the driver for the ESI Maya44 card, which I developed on the initiative of Piotr Makowski. The driver is actually an addition to the existing ice1724 code. More information can be found in doc/README.maya44 . We would appreciate your testing and feedback. Also, since I am new to ALSA driver development, I would appreciate feedback on whether the way I integrated this into the ALSA tree is correct. Thanks for your feedback and best regards, Rainer Zimmermann -- Lightshed IT Services Löfflerstr. 27, 22765 Hamburg, Germany * mail(a)lightshed.de * www.lightshed.de

4 4

[alsa-devel] 1.0.16 final release
by Jaroslav Kysela 06 Feb '08

06 Feb '08

Hello all, I am happy to announce ALSA 1.0.16 final release. All packages except alsa-oss (no changes) are available for download. To see changes, please, follow this URL: http://www.alsa-project.org/main/index.php/Changes_v1.0.15_v1.0.16 Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

2 1

[alsa-devel] [PATCH] aplay/arecord - Add support for IEEE float 32-bit WAV files
by Andrew Paprocki 06 Feb '08

06 Feb '08

This patch modifies aplay/arecord to support playing/capturing IEEE float 32-bit WAV files. Tested on HDA hardware in both stereo and multi-channel modes. Added the WAV file constant for Dolby AC-3 S/PDIF passthrough to formats.h for future use when AC-3 passthrough is better supported. Signed-off-by: Andrew Paprocki <andrew(a)ishiboo.com> diff -r 675baf17b7c9 -r 711f7227eb13 aplay/aplay.c --- a/aplay/aplay.c Mon Jan 21 10:42:48 2008 +0100 +++ b/aplay/aplay.c Sat Feb 02 17:28:32 2008 -0500 @@ -745,8 +745,9 @@ static ssize_t test_wavefile(int fd, u_c check_wavefile_space(buffer, len, blimit); test_wavefile_read(fd, buffer, &size, len, __LINE__); f = (WaveFmtBody*) buffer; - if (LE_SHORT(f->format) != WAV_PCM_CODE) { - error(_("can't play not PCM-coded WAVE-files")); + if (LE_SHORT(f->format) != WAV_FMT_PCM && + LE_SHORT(f->format) != WAV_FMT_IEEE_FLOAT) { + error(_("can't play WAVE-file format 0x%04x which is not PCM or FLOAT encoded"), LE_SHORT(f->format)); exit(EXIT_FAILURE); } if (LE_SHORT(f->modus) < 1) { @@ -788,7 +789,10 @@ static ssize_t test_wavefile(int fd, u_c } break; case 32: - hwparams.format = SND_PCM_FORMAT_S32_LE; + if (LE_SHORT(f->format) == WAV_FMT_PCM) + hwparams.format = SND_PCM_FORMAT_S32_LE; + else if (LE_SHORT(f->format) == WAV_FMT_IEEE_FLOAT) + hwparams.format = SND_PCM_FORMAT_FLOAT_LE; break; default: error(_(" can't play WAVE-files with sample %d bits wide"), @@ -1778,6 +1782,7 @@ static void begin_wave(int fd, size_t cn bits = 16; break; case SND_PCM_FORMAT_S32_LE: + case SND_PCM_FORMAT_FLOAT_LE: bits = 32; break; case SND_PCM_FORMAT_S24_LE: @@ -1796,7 +1801,10 @@ static void begin_wave(int fd, size_t cn cf.type = WAV_FMT; cf.length = LE_INT(16); - f.format = LE_SHORT(WAV_PCM_CODE); + if (hwparams.format == SND_PCM_FORMAT_FLOAT_LE) + f.format = LE_SHORT(WAV_FMT_IEEE_FLOAT); + else + f.format = LE_SHORT(WAV_FMT_PCM); f.modus = LE_SHORT(hwparams.channels); f.sample_fq = LE_INT(hwparams.rate); #if 0 diff -r 675baf17b7c9 -r 711f7227eb13 aplay/formats.h --- a/aplay/formats.h Mon Jan 21 10:42:48 2008 +0100 +++ b/aplay/formats.h Sat Feb 02 17:28:32 2008 -0500 @@ -64,7 +64,11 @@ typedef struct voc_ext_block { #define WAV_WAVE COMPOSE_ID('W','A','V','E') #define WAV_FMT COMPOSE_ID('f','m','t',' ') #define WAV_DATA COMPOSE_ID('d','a','t','a') -#define WAV_PCM_CODE 1 + +/* WAVE fmt block constants from Microsoft mmreg.h header */ +#define WAV_FMT_PCM 0x0001 +#define WAV_FMT_IEEE_FLOAT 0x0003 +#define WAV_FMT_DOLBY_AC3_SPDIF 0x0092 /* it's in chunks like .voc and AMIGA iff, but my source say there are in only in this combination, so I combined them in one header;

4 3

[alsa-devel] DSP2000 Midi In - Not a correct fix??
by Alan Horstmann 06 Feb '08

06 Feb '08

I think it is time to post my investigations so far... re: Hoontech DSP2000 Midi In non-functional see : https://bugtrack.alsa-project.org/alsa-bug/view.php?id=55 https://bugtrack.alsa-project.org/alsa-bug/view.php?id=1161 https://bugtrack.alsa-project.org/alsa-bug/view.php?id=1846 I missed a fix patch in March 07 for this as it was put on a later duplicate of the bug report I had contributed to. However I noticed JK closing -> resolved these in Jan 08, so tried the patch: http://hg.alsa-project.org/alsa-kernel/rev/319688a93b6e From my testing, it seems the only part necessary is:- - for (box = 0; box < 4; box++) { + for (box = 0; box < 1; box++) { as it is successful without the rest. However I feel this is not really a correct fix, preventing use of multiple boxes in code that originally allowed it. The code is a little strange and has taken me a while to approximately grasp. What seems to be going wrong is that when a second box is configured (ie box = 1) the function snd_ice1712_stdsp24_box_midi(...) switches the midi input off box[0]. If box[1] has ICE1712_STDSP24_BOX_MIDI1 set in it's config, then midi in and out can work on box[1] instead of box[0], but otherwise it is just stopped on box[0]. After many hours/days of tests I still can't propose a definite way to fix this function to provide correct configuration to all 4 allowed boxes, as I really don't understand why the code does all the gpio writes that it does. Unfortunately I do not have any circuit information either. In some way it seems to be latching some of the configuration data, but not all. Clemens, can you remember anything about this change: http://hg.alsa-project.org/alsa-kernel/rev/a42043505e46 -was it someone else's patch or did you work on this? I am hoping that someone can contribute enough of an idea about this to help formulate a proper fix. Unfortunately, as the bugs are closed, I can not continue the discussions there. Alan

4 5

[alsa-devel] ALC268 (no sound)
by Marco Schuster 06 Feb '08

06 Feb '08

Hi all, maybe this topic is a bit outdated, but still, when using the current hg code of the kernel module together with 2.6.24 (stable), I can't get any sound :( Do I have also to upgrade the ALSA libs and programs? Marco

3 7

[alsa-devel] [PATCH][sound/soc/fsl/mpc8610_hpcd.c] duplicate strcasecmp test for "rj-master" in mpc8610_hpcd_probe()
by Roel Kluin 06 Feb '08

06 Feb '08

In linus' git tree I found this problem. Is it also in the alsa tree? please confirm it's the right fix. The patch was not yet tested. -- duplicate test for "rj-master" Signed-off-by: Roel Kluin <12o3l(a)tiscali.nl> --- diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index f26c4b2..a00aac7 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -314,9 +314,9 @@ static int mpc8610_hpcd_probe(struct of_device *ofdev, } else if (strcasecmp(sprop, "lj-master") == 0) { machine_data->dai_format = SND_SOC_DAIFMT_LEFT_J; machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; - } else if (strcasecmp(sprop, "rj-master") == 0) { + } else if (strcasecmp(sprop, "rj-slave") == 0) { machine_data->dai_format = SND_SOC_DAIFMT_RIGHT_J; machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; } else if (strcasecmp(sprop, "rj-master") == 0) {

4 4

[alsa-devel] How to change the sound volume?
by Cosmin Humeniuc 06 Feb '08

06 Feb '08

Hi, How would a simple alsa-based sound player change the playback volume? I've been looking into the mixer interface, but there's no connection to snd_pcm_t*, so I'm not sure this is what I need to use. I only want to change the sound volume of my player, without altering the sound streams of the other applications. Thanks, Cosmin

2 1

Re: [alsa-devel] [alsa-cvslog] alsa-tools: hdspmixer - Fix Digiface channel map for ADAT speed mode 1
by Thierry Vignaud 05 Feb '08

05 Feb '08

Takashi Iwai <tiwai(a)suse.de> writes: > changeset: 365:9d1e48f8dd5d > tag: tip > user: tiwai > date: Tue Feb 05 13:30:45 2008 +0100 > files: hdspmixer/src/HDSPMixerCard.cxx > description: > hdspmixer - Fix Digiface channel map for ADAT speed mode 1 > > Fixed the channel map for ADAT speed mode 1 (rate > 48000). > ALSA bug#3732 > https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3732 Will you move the 1.0.16 tag so that it got included in final release? > diff -r 7f30f325ab54 -r 9d1e48f8dd5d hdspmixer/src/HDSPMixerCard.cxx > --- a/hdspmixer/src/HDSPMixerCard.cxx Tue Feb 05 10:08:12 2008 +0100 > +++ b/hdspmixer/src/HDSPMixerCard.cxx Tue Feb 05 13:30:45 2008 +0100 > @@ -299,7 +299,7 @@ void HDSPMixerCard::adjustSettings() { > break; > case 1: > channels = 14; > - channel_map = channel_map_ds; > + channel_map = meter_map_ds; > dest_map = dest_map_ds; > meter_map = meter_map_ds; > lineouts = 2;

2 1

[alsa-devel] Salsa-lib memleak
by Matthieu CASTET 05 Feb '08

05 Feb '08

Hi, it seems Salsa-lib leak memory in case of errors. For snd_pcm_open, snd_pcm_t is allocated, but not free in case of snd_pcm_hw_mmap_status error. The same things happen in _snd_pcm_mmap, there error return in middle of the code, without freeing the allocated memory. Thanks Matthieu

2 1

Re: [alsa-devel] M-Delta Audiophile 192 (ice1724) Recording
by Computer AG im FSK 04 Feb '08

04 Feb '08

Hi, unfortunately it didn't solve the issue. Same problem: no audio signal can be recorded. I provided all the information at http://pastebin.ca/886740 If we can't solve the issue this way, maybe we could lend you one of our cards ? Greeting, Toast

3 5

[alsa-devel] HDA - Add support for the OQO Model 2
by Tobin Davis 04 Feb '08

04 Feb '08

This patch adds support for the OQO Model 2 Ultra Mobile PC. Signed off by Tobin Davis <tdavis(a)dsl-only.net>

3 11

[alsa-devel] alsa-lib-1.0.15 : --disable-hwdep/--disable-rawmidi broken
by Matthieu CASTET 04 Feb '08

04 Feb '08

hi, If I compile alsa-lib-1.0.15 with --disable-hwdep or --disable-rawmidi, I get failure in include/alsa/control.h [1]. Matthieu [1] In file included from ../../include/alsa/asoundlib.h:51, from simple_none.c:39: ../../include/alsa/control.h:246: error: expected declaration specifiers or '...' before 'snd_hwdep_info_t' ../../include/alsa/control.h:251: error: expected declaration specifiers or '...' before 'snd_rawmidi_info_t' make[3]: *** [simple_none.lo] Erreur 1 make[3]: *** Attente des tâches non terminées.... make[2]: *** [all-recursive] Erreur 1 make[1]: *** [all-recursive] Erreur 1 make: *** [alsa] Erreur 2

2 1

[alsa-devel] Intel ICH Modem init fails for laptop built-in SiS modem
by Richard Kennedy 03 Feb '08

03 Feb '08

My JVC mininote laptop has an inbuilt SiS modem 00:01.6 Modem [0703]: Silicon Integrated Systems [SiS] AC'97 Modem Controller [1039:7013] (rev a0) which is detected by the snd-intel8x0m driver but it always fails to initialise and spams the log file :- kernel: codec_semaphore: semaphore is not ready [0x1][0x1300] kernel:last message repeated 2 times kernel: codec_semaphore: semaphore is not ready [0x1][0x300] kernel:last message repeated 431 times kernel: AC'97 1 does not respond - RESET kernel: codec_semaphore: semaphore is not ready [0x1][0x300] kernel: codec_semaphore: semaphore is not ready [0x1][0x300] kernel: AC'97 1 access is not valid [0xffffffff], removing mixer. kernel: Unable to initialize codec #1 kernel: ACPI: PCI interrupt for device 0000:00:01.6 disabled kernel: Intel ICH Modem: probe of 0000:00:01.6 failed with error -5 I never use the modem so I blacklisted the snd-intel8x0m module in modprobe.conf just to make the messages go away. I'm currently running the fedora 2.6.23 kernel but it has been broken for a long time. Is this fixable? or shall I just go on ignoring it ;) ? cheers Richard

2 4

[alsa-devel] HDA - enable snoop on SCH
by Tobin Davis 03 Feb '08

03 Feb '08

This patch enables snoop on Intel SCH chipset, eliminating static during playback. Signed off by Tobin Davis <tdavis(a)dsl-only.net> P.S. I know this is real close to the final release, but it is critically needed for product enabling. The patch has no effect on other components, and has been tested on production desktop and laptop systems, along with Intel's Mobile Internet Device. This is the last patch needed for release. P.P.S Forgot to include the patch in the last email. Doh.

2 1

[alsa-devel] Sound jack isolations
by Richard Marzan 03 Feb '08

03 Feb '08

On Thu, 2008-01-31 at 20:33 -0500, Richard Marzan wrote: > > On Wed, 2008-01-30 at 23:03 -0800, Tobin Davis wrote: > > > Try downloading alsa-driver-1.0.16rc2. It has some new patches in it. > > > > > > If that doesn't work, please report back with the link generated from > > > running http://bulletproof.servebeer.com/alsa/scripts/alsa-info.sh > > > > > > Tobin > > > > > > > > > On Wed, 2008-01-30 at 23:49 -0500, Richard Marzan wrote: > > > > I tried googling around but no helpful content so far. I have a laptop > > > > with built-in sound. It uses the HDA_intel kernel module. My problem is > > > > that I can't silence the onboard/built-in speakers when I plug in the > > > > headphones to the machine. Muting sound would lead to all jacks and > > > > audio ports to be silenced -- not just the built-in speakers, which I > > > > intend to mute solely leaving sound alive on the headphone jacks. There > > > > are no channels readily observable to differentiate where sound goes and > > > > to which port with alsa-mixer. The behavior I would like to achieve is > > > > one that will allow me to mute onboard speakers while continuing to have > > > > the headphones receive audio signals. Has anyone ever done this? Any > > > > pointers to documentation will be appreciated. > > > > > > > > My board has the Conexant CX20549 Codec. I think is uses the ALC268 > > > > module. I checked gmane and saw that a fellow user got snd-hda-intel > > > > working with the NVidia MCP67 chipset with a patch received from > > > > linuxant.com. I tried the method and when patching i had encountered > > > > errors. Thanks for the reply. Here is the link http://pastebin.ca/886945

1 0

[alsa-devel] HDA - enable snoop on sch
by Tobin Davis 02 Feb '08

02 Feb '08

This patch enables snoop on Intel SCH chipset, eliminating static during playback. Signed off by Tobin Davis <tdavis(a)dsl-only.net> P.S. I know this is real close to the final release, but it is critically needed for product enabling. The patch has no effect on other components, and has been tested on production desktop and laptop systems, along with Intel's Mobile Internet Device. This is the last patch needed for release.

1 0

[alsa-devel] Fwd: Re: alsa-info.sh debugging script
by Travis Place 02 Feb '08

02 Feb '08

---------- Forwarded Message ---------- Subject: Re: [alsa-devel] alsa-info.sh debugging script Date: Sun, 3 Feb 2008 From: Travis Place <wishie(a)wishie.net> To: Andy Shevchenko <andy(a)smile.org.ua> On Sun, 3 Feb 2008 12:30:35 am you wrote: > Hi Travis Place! > > On Sat, Feb 02, 2008 at 11:34:42PM +1000, Travis Place wrote next: > > Any other ideas/changes that need to be made ? > > I don't remember, but if possible to use fileout instead of pastebin site > that will be good. There is already a --no-upload option, which saves the output to /tmp/alsa-info.txt but doesnt upload it. Is that sufficient ? -- Happiness is like peeing your pants. Everyone can see it, but only you can feel its warmth ------------------------------------------------------- -- Happiness is like peeing your pants. Everyone can see it, but only you can feel its warmth

1 0

[alsa-devel] Fwd: Re: alsa-info.sh debugging script
by Travis Place 02 Feb '08

02 Feb '08

---------- Forwarded Message ---------- Subject: Re: [alsa-devel] alsa-info.sh debugging script Date: Sun, 3 Feb 2008 From: Andy Shevchenko <andy(a)smile.org.ua> To: Travis Place <wishie(a)wishie.net> Hi Travis Place! On Sat, Feb 02, 2008 at 11:34:42PM +1000, Travis Place wrote next: > Any other ideas/changes that need to be made ? I don't remember, but if possible to use fileout instead of pastebin site that will be good. -- With best regards, Andy Shevchenko. mailto: andy(a)smile.org.ua ------------------------------------------------------- -- Happiness is like peeing your pants. Everyone can see it, but only you can feel its warmth

1 0

[alsa-devel] [PATCH] hda_intel - Add model quirk for Albatron KI690-AM2 motherboard
by Andrew Paprocki 02 Feb '08

02 Feb '08

This adds a quirk to the Realtek ALC883 table for the Albatron KI690-AM2 motherboard to use the 6stack-dig model. Signed-off-by: Andrew Paprocki <andrew(a)ishiboo.com> diff -r ba9ca6a13ef2 -r 380e8be9c65b pci/hda/patch_realtek.c --- a/pci/hda/patch_realtek.c Fri Feb 01 17:23:54 2008 +0100 +++ b/pci/hda/patch_realtek.c Sat Feb 02 02:11:00 2008 -0500 @@ -7635,6 +7635,7 @@ static struct snd_pci_quirk alc883_cfg_t SND_PCI_QUIRK(0x17aa, 0x3bfc, "Lenovo NB0763", ALC883_LENOVO_NB0763), SND_PCI_QUIRK(0x17aa, 0x3bfd, "Lenovo NB0763", ALC883_LENOVO_NB0763), SND_PCI_QUIRK(0x17c0, 0x4071, "MEDION MD2", ALC883_MEDION_MD2), + SND_PCI_QUIRK(0x17f2, 0x5000, "Albatron KI690-AM2", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1991, 0x5625, "Haier W66", ALC883_HAIER_W66), SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC883_3ST_6ch), {}

1 0

[alsa-devel] 1.0.16rc2
by Jaroslav Kysela 02 Feb '08

02 Feb '08

Hi, 1.0.16rc2 alsa-driver, alsa-lib, pyalsa packages are available for download. Full changelog from 1.0.16rc1 is on: http://www.alsa-project.org/main/index.php/Changes_v1.0.16rc1_v1.0.16rc2 Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

4 10

[alsa-devel] underruns and strange code in pcm_rate.c (and patch)
by Stas Sergeev 01 Feb '08

01 Feb '08

Hi. I spent 4 week-ends debugging the constant underruns and lock-ups with the libao-based programs. Now I've finally tracked the problem to some strange code in pcm_rate.c, namely snd_pcm_rate_poll_descriptors(). I am not sure what was the reason behind altering the "avail_min" every time. Looks like some heuristic was intended, say "see what amount of fragment part was written now, and make sure that amount of free space will be available next time, after poll". Whatever, this code gives underruns. It can increase avail_min by almost period_size. libao sets avail_min==period_size, so we can end up with avail_min==period_size*2. libao sets buffer_size==stop_threshold==period_size*2. So we end up with avail_min~=stop_threshold, which gives underruns. Since I don't know how this code was indended to work, I just removed it, and everything works perfectly now (also had to patch libao to check the returned rate). The patch is attached, any comments? Signed-off-by: Stas Sergeev <stsp(a)aknet.ru>

7 47

Re: [alsa-devel] Hey, wishie here.
by Tobin Davis 01 Feb '08

01 Feb '08

On Sat, 2008-02-02 at 00:29 +1000, Travis Place wrote: > Hey, gnubien gave me a rough overview I think.. > > If i can get exact details from you, of what you guys want added to > the script, ill get onto it straight away. > I was BLOWN AWAY that Eric Raymond used the script.. that is so > mind-blowing, considering i started this script out of boredom, and > laziness of asking users 1000's of questions.. It blew me away too. I also have seen it on internal mailings at Intel between developers working on different projects. > Anyway, get a list to me of what you want added, ill get on it, and we > can change the script to get updates from alsa-project.org etc.. What they were looking for was output from alsactl. Take a look at what it generates. Use the -f <filename> path to save a temporary file that can be appended to alsa-info.txt. The other changes you would need to discuss with the lead developers on alsa-devel (which I am cc'ng). > I am more than happy to help out where i can. > > Thanks for all the great help you have given along the way too.. > > Without the help of people like you, gnubien, and others, the script > wouldnt be where it is today. > > Travis (aka, wishie) > > -- > Happiness is like peeing your pants. Everyone can see it, but only you > can feel its warmth Love the sig. -- Tobin Davis There is only one word for aid that is genuinely without strings, and that word is blackmail. -- Colm Brogan -- Tobin Davis The best way to make a fire with two sticks is to make sure one of them is a match. -- Will Rogers

2 1

[alsa-devel] Salsa-lib add snd_pcm_get_params
by Matthieu CASTET 01 Feb '08

01 Feb '08

Hi, in 0.0.14, snd_pcm_get_params is missing. Is is not even in the stub form in pcm_macros.h. Why this function is missing ? As it is quite trivial, could it be added in pcm_macros.h or pcm_params.c. Thanks Matthieu

2 3

[alsa-devel] [PATCH/KCONFIG] - Add SUPERH depends to sound/soc/sh/Kconfig
by Kristoffer Ericson 01 Feb '08

01 Feb '08

Currently you will see an empty "SoC Audio support for SuperH" menu when building for other archs (example pxa). This patch adds "depends on SUPERH" to remove that empty menu. signed-off-by: Kristoffer Ericson <kristoffer.ericson(a)gmail.com> diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index f03220d..4c1e013 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -1,4 +1,5 @@ menu "SoC Audio support for SuperH" + depends on SUPERH config SND_SOC_PCM_SH7760 tristate "SoC Audio support for Renesas SH7760"

2 1

[alsa-devel] [PATCH] Mono voice playback volume for WM8753
by Mark Brown 01 Feb '08

01 Feb '08

From: Mike Montour <mail(a)mmontour.net> Voice playback volume is in register bits 0:2, not 4:6. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Werner Almesberger <werner(a)openmoko.org> --- I don't have a Signed-off-by from Mike but the change is sufficiently obvious. sound/soc/codecs/wm8753.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index ba16375..0f23659 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -280,7 +280,7 @@ SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8753_LOUT2V, WM8753_ROUT2V, 7, 1, 0 SOC_SINGLE("Mono Bypass Playback Volume", WM8753_MOUTM1, 4, 7, 1), SOC_SINGLE("Mono Sidetone Playback Volume", WM8753_MOUTM2, 4, 7, 1), -SOC_SINGLE("Mono Voice Playback Volume", WM8753_MOUTM2, 4, 7, 1), +SOC_SINGLE("Mono Voice Playback Volume", WM8753_MOUTM2, 0, 7, 1), SOC_SINGLE("Mono Playback ZC Switch", WM8753_MOUTV, 7, 1, 0), SOC_ENUM("Bass Boost", wm8753_enum[0]), -- 1.5.3.8

2 1

[alsa-devel] es1938 - patch trying to improve capture hw pointer reads
by Hermann Lauer 01 Feb '08

01 Feb '08

Hello, with the Solo1 (es1938) I got a lot of xrun's during capture on my machine. Tracing that down it seems to be comming from reading ocassionaly bad hw pointers from the chip. Appended is a patch against linux-2.6.23.12 which uses more checking to avoid that false pointer reads. Failed reads are giving back the last good value read instead of spinning in a tight loop, which seems more appropriate to me in an interrupt. I think I saw this trick used in another driver. On my machine xruns seems to be gone with that patch. If anybody is interested, mmaped capture seems also to be doable with some tricks, still testing this... Greetings Hermann -- Netzwerkadministration/Zentrale Dienste, Interdiziplinaeres Zentrum fuer wissenschaftliches Rechnen der Universitaet Heidelberg IWR; INF 368; 69120 Heidelberg; Tel: (06221)54-8236 Fax: -5224 Email: Hermann.Lauer(a)iwr.uni-heidelberg.de

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[alsa-devel] Roland UA-30 works
by Victor Roetman 01 Feb '08

01 Feb '08

I just tested a Roland UA-30 USB sound card, and it seems to work fine. I wanted to add it to the sound card matrix on http://www.alsa-project.org/main/index.php/Matrix:Vendor-Roland_Edirol but when I click edit, it says "This page has been locked to prevent editing." Can someone add this for me? Or tell me how to add it? On Ubuntu 7.10 it seems to work fine with playback and with recording, and when the 44.1/48 kHz switch is changed, the device seems to be removed and added again with the correct frequency shown in /proc/asound/cards.

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Re: [alsa-devel] M-Delta Audiophile 192 (ice1724) Recording
by Takashi Iwai 31 Jan '08

31 Jan '08

At Thu, 31 Jan 2008 18:11:38 +0100, CAG wrote: > > HI, > > Takashi Iwai schrieb: > > At Thu, 31 Jan 2008 16:44:06 +0100, > > CAG wrote: > >> Hi, > >> > >> we ran the alsa-info.sh script to provide more information on our setup: > >> http://pastebin.ca/886260 > >> > >> The output of /proc/asound/card0/ice1724 is at > >> http://pastebin.ca/884589 > > > > Thanks. One more thing - could you tell how did you test the analog > > recording exactly? I just need to make sure about your test method. > Well, we tried with arecord, ecasound, jack & audacity, but in the > alsamixer / mixer gui in kde, at capture there's just one "Multi Track > Peak" without capture switch. > > What would be the best record method to test ? Usually arecord is the best as a primary test. Simply run % arecord -fdat -vv somefile.wav To avoid any possible bugs in alsa-lib, also try % arecord -Dplughw:0 -fdat -vv somefile.wav > > The patch below enables AK4114, and possibly fixes some ADC setup. > > Give it a try. > Cool! we will try it this evening and give a feedback. It is a patch for > 1.0.16rc2 or 1.0.15 ? For 1.0.16rc2. Don't forget to update alsa-lib and alsa-utils, too. Takashi

2 2

[alsa-devel] M-Delta Audiophile 192 (ice1724) Recording
by CAG 31 Jan '08

31 Jan '08

Hi, we use the M-Delta Audiophile 192 with alsa-driver-1.0.15 and the ice1724 module. Playback works but we cannot record anything. I noticed that this maybe a bug according to the Alsa Changelog: From Changes v1.0.13 v1.0.14: > > - ice1724 - Add support of M-Audio Audiophile 192 > > Added the (experimental) support of M-Audio Audiophile 192 board. > Currently, the analog and the digital playbacks seem working fine. > The inputs seem not working as far as I've tested yet. How is the status for that card ? Is someone working on this ? I heard Recording with OSS works but actually we don't want to switch to OSS... Is it worth waiting for a new release ? We desperately need recording! Cheers from a free radio station, Toast

2 5

[alsa-devel] alsa-info.sh
by Jaroslav Kysela 31 Jan '08

31 Jan '08

On Wed, 30 Jan 2008, Tobin Davis wrote: > Try downloading alsa-driver-1.0.16rc2. It has some new patches in it. > > If that doesn't work, please report back with the link generated from > running http://bulletproof.servebeer.com/alsa/scripts/alsa-info.sh Hey this script is exactly something I'd like to show users in http://www.alsa-project.org/main/index.php/Help_To_Debug . Can I put this script to our HG tree in alsa/ repo and point users to it in mentioned page? Thanks. Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

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[alsa-devel] [PATCH] hda: STAC927x power down inactive DACs
by Matthew Ranostay 31 Jan '08

31 Jan '08

On several laptops that have STAC9228 codecs have unused DACs, this powers them down to a D3 state. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff -r 86c8c0a82bb9 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Wed Jan 30 11:52:33 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Wed Jan 30 08:58:37 2008 -0500 @@ -288,6 +288,10 @@ static hda_nid_t stac927x_adc_nids[3] = static hda_nid_t stac927x_mux_nids[3] = { 0x15, 0x16, 0x17 +}; + +static hda_nid_t stac927x_dac_nids[5] = { + 0x02, 0x03, 0x04, 0x05, 0x06 }; static hda_nid_t stac927x_dmux_nids[1] = { @@ -3537,7 +3541,7 @@ static int patch_stac927x(struct hda_cod static int patch_stac927x(struct hda_codec *codec) { struct sigmatel_spec *spec; - int err; + int err, i; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -3628,6 +3632,14 @@ static int patch_stac927x(struct hda_cod stac92xx_free(codec); return err; } + + /* on some 3-stack systems not all the DACs are used + * this powers down ones that aren't used + */ + for (i = 0; i < ARRAY_SIZE(stac927x_dac_nids); i++) + if (!is_in_dac_nids(spec, stac927x_dac_nids[i])) + snd_hda_codec_write_cache(codec, stac927x_dac_nids[i], + 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D3); codec->patch_ops = stac92xx_patch_ops;

2 3

[alsa-devel] HDA on Zepto Znote6224W - debug info codecgraph
by Philipp E. Letschert 31 Jan '08

31 Jan '08

Hello, using hg20080130 snapshot, sound is working now, without giving the model=toshiba option when loading the module. But it is still like coming from a tin can. attached is debugging info, as suggested on the web-page. Thanks! Phil Codec: Realtek ALC268 Address: 0 Vendor Id: 0x10ec0268 Subsystem Id: 0x11700040 Revision Id: 0x100003 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=4, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0 IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0 Node 0x02 [Audio Output] wcaps 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x16 0x16] Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x1f 0x1f] Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x05 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x07 [Audio Input] wcaps 0x100111: Stereo Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x24 Node 0x08 [Audio Input] wcaps 0x100111: Stereo Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0a [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0b [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0c [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0d [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0e [Audio Mixer] wcaps 0x20010a: Mono Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00] Connection: 1 0x02 Node 0x0f [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Connection: 2 0x02 0x1d Node 0x10 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x80 0x80] Connection: 3 0x03 0x1d 0x02 Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x12 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Node 0x13 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x081003c: IN OUT HP EAPD Detect EAPD 0x2: EAPD Pin Default 0x99130110: [Fixed] Speaker at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x0f Node 0x15 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x081003c: IN OUT HP EAPD Detect EAPD 0x2: EAPD Pin Default 0x0121401f: [Jack] HP Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0xc0: OUT HP Unsolicited: tag=04, enabled=1 Connection: 1 0x10 Node 0x16 [Pin Complex] wcaps 0x40010c: Mono Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80] Pincap 0x0810: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Connection: 1 0x0e Node 0x17 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x083734: IN OUT Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x01a19820: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Connection: 1 0x02 Node 0x19 [Pin Complex] wcaps 0x40008b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: Pincap 0x083724: IN Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x99a3092e: [Fixed] Mic at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0x2, Sequence = 0xe Misc = NO_PRESENCE Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x083734: IN OUT Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 1 0x02 Node 0x1b [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x0820: IN Pin Default 0x4005822d: [N/A] Line Out at Ext N/A Conn = Optical, Color = Purple DefAssociation = 0x2, Sequence = 0xd Pin-ctls: 0x20: IN Node 0x1e [Pin Complex] wcaps 0x400380: Mono Digital Pincap 0x0810: OUT Pin Default 0x01451130: [Jack] SPDIF Out at Ext Rear Conn = Optical, Color = Black DefAssociation = 0x3, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x06 Node 0x1f [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=10 Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x23 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x00 0x00] Connection: 7 0x18 0x19 0x1a 0x1c 0x14 0x15* 0x12 Node 0x24 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x01 0x01] Connection: 7 0x18 0x19 0x1a 0x1c 0x14* 0x15 0x13 00:1b.0 0403: 8086:284b (rev 03) Subsystem: 1170:0040 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 22 Region 0: Memory at f0500000 (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] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Address: 0000000000000000 Data: 0000 Capabilities: [70] Express Unknown type IRQ 0 Device: Supported: MaxPayload 128 bytes, PhantFunc 0, ExtTag- Device: Latency L0s <64ns, L1 <1us Device: Errors: Correctable- Non-Fatal- Fatal- Unsupported- Device: RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ Device: MaxPayload 128 bytes, MaxReadReq 128 bytes Link: Supported Speed unknown, Width x0, ASPM unknown, Port 0 Link: Latency L0s <64ns, L1 <1us Link: ASPM Disabled CommClk- ExtSynch- Link: Speed unknown, Width x0 Capabilities: [100] Virtual Channel Capabilities: [130] Unknown (5) 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03) Subsystem: Inventec Corporation Unknown device 0040 Flags: bus master, fast devsel, latency 0, IRQ 22 Memory at f0500000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Capabilities: [70] Express Unknown type IRQ 0 Capabilities: [100] Virtual Channel Capabilities: [130] Unknown (5)

1 0

[alsa-devel] ACER Aspier 5630 (snd-hda-intel ALC883)
by Marcin Floryan 30 Jan '08

30 Jan '08

Hello! I have had some problems using alsa sound on openSuse 10.3 wit the above hardware: sound output OK but mic not working. Have seen many relevant bugs and with the new driver the microphone is now working OK so thanks for all the effort. I was wondering now if there is anything I could do to help with driver development to fine-tune for this particular hardware setup. I can provide some further details or do some tests if guided a bit (I am fairly comfortable modifying source code and compiling the driver) Kind regards, -- Marcin Floryan http://marcin.floryan.pl/ Please consider the environment before printing this email.

3 5

[alsa-devel] alsa-firmware licensing
by Tim Jackson 30 Jan '08

30 Jan '08

As part of the Fedora Project [1], I'm trying to package [2] the ALSA firmware up for easy distribution to Fedora users. The trouble is that the licensing seems unclear. Specifically: a) Several firmwares claim to be under the GPL but don't specify a version and don't include source - see https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3411 b) More worryingly, the Emagic EMI 2|6 firmware appears to be under a "license" which prohibits ANY redistribution. Quoting from emi_26_62/license.txt (my emphasis): "The firmware contained herein is Copyright (c) 1999-2002 Emagic as an unpublished work. This notice does not imply unrestricted or public access to this firmware which is a trade secret of Emagic, and which ***may not be reproduced, used, sold or transferred to any third party without Emagic's written consent.***" Does the ALSA project have special authorisation ("written consent") to distribute this? Who within the ALSA project is responsible for the -firmware package and might be able to help resolve some of these queries? Does anyone have a contact at Emagic (which seems to have been bought by Apple) to discuss relaxing the firmware conditions slightly? Any information most welcome. We'd really like to get the firmware in Fedora. Thanks, Tim [1] http://www.fedoraproject.org/ [2] https://bugzilla.redhat.com/show_bug.cgi?id=217259

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[alsa-devel] Couple of ASOC v2 snd-core issues
by Jon Smirl 30 Jan '08

30 Jan '08

I had to fix these to get my drivers to load. diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1ead550..4358f74 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1330,7 +1331,7 @@ int snd_soc_codec_add_dai(struct snd_soc_codec *codec, if (dai_runtime == NULL) goto err; - dai_runtime->dai = dai; + dai_runtime->dai = dai++; dai_runtime->codec = codec; list_add(&dai_runtime->list, &codec->dai_list); @@ -1382,7 +1383,7 @@ int snd_soc_platform_add_dai(struct snd_soc_platform *platform, if (dai_runtime == NULL) goto err; - dai_runtime->dai = dai; + dai_runtime->dai = dai++; dai_runtime->platform = platform; list_add(&dai_runtime->list, &platform->dai_list); @@ -1479,6 +1480,7 @@ int snd_soc_codec_create(struct snd_soc_machine *machine, codec->dev.parent = machine->dev; codec->dev.release = codec_dev_release; codec->machine = machine; + codec->name = (char *)codec_id; mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dai_list); -- Jon Smirl jonsmirl(a)gmail.com

2 1

[alsa-devel] [PATCH] hda: Mic in as output changes
by Matthew Ranostay 30 Jan '08

30 Jan '08

Some laptop has an internal analog microphone that is 'fixed'. This patch prevents creating a 'Mic In as Output' switch for ports that can't be outputs. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff -r ec0316a4d108 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Tue Jan 29 15:28:44 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Tue Jan 29 17:33:58 2008 -0500 @@ -288,6 +288,10 @@ static hda_nid_t stac927x_adc_nids[3] = static hda_nid_t stac927x_mux_nids[3] = { 0x15, 0x16, 0x17 +}; + +static hda_nid_t stac927x_dac_nids[5] = { + 0x02, 0x03, 0x04, 0x05, 0x06 }; static hda_nid_t stac927x_dmux_nids[1] = { @@ -2283,15 +2287,23 @@ static int stac92xx_auto_create_multi_ou } if (spec->mic_switch) { + unsigned int def_conf; nid = cfg->input_pins[AUTO_PIN_MIC]; - pincap = snd_hda_param_read(codec, nid, - AC_PAR_PIN_CAP); - if (pincap & AC_PINCAP_OUT) { - err = stac92xx_add_control(spec, - STAC_CTL_WIDGET_IO_SWITCH, - "Mic as Output Switch", (nid << 8) | 1); - if (err < 0) - return err; + def_conf = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_CONFIG_DEFAULT, 0); + + /* some laptops have an internal analog microphone + * which can't be used as a output */ + if (get_defcfg_connect(def_conf) != AC_JACK_PORT_FIXED) { + pincap = snd_hda_param_read(codec, nid, + AC_PAR_PIN_CAP); + if (pincap & AC_PINCAP_OUT) { + err = stac92xx_add_control(spec, + STAC_CTL_WIDGET_IO_SWITCH, + "Mic as Output Switch", (nid << 8) | 1); + if (err < 0) + return err; + } } }

2 3

[alsa-devel] Gateway Model: MT3707 driver??
by Ya Bra 30 Jan '08

30 Jan '08

Dear Alsa Driver's Developers, I would like to express my gratitude to you all for your hard work developing the sound driver for linux users' computers. I am a beginner Ubuntu linux user. I recently tried to install the sound driver for my Gateway Laptop. It shows in the driver list as HDA Intel with an ATI manufacture. Please let me know if your project has extended to support for sound in such laptop's vendor like Gateway. Or if you have any solution on how to configure such sound card like to work, please let me know as well. I would like to thank you in advance for your help. Yours Sincerely, Nymo Lim,

2 1

[alsa-devel] 1.0.16 final time plan & mainstream sync
by Jaroslav Kysela 30 Jan '08

30 Jan '08

Hi all, I would like to release 1.0.16 final on Tuesday next week. Please, push only short fixes or compilation fixes to HG now. No big changes, please. Also, GIT update is prepared for next 2.6.25 mainstream sync (#linus branch). I will send a sync request to Linus probably tomorrow. Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

2 1

[alsa-devel] [PATCH] asihpi version 3.09.09
by Eliot Blennerhassett 30 Jan '08

30 Jan '08

Howdy Takashi, Code: http://audioscience.com/internet/download/beta/alsa_driver3507_asihpi30909.… Firmware: http://audioscience.com/internet/download/beta/dspbins30909.tar.bz2 here is an update to the asihpi driver, please apply to alsa Hg if you see fit. (BTW the previous version was only meant to be commented on, not committed!) Main changes in this set: * Redo comment formatting so block comments are properly indented with code (bulk of changes) hpifunc.c: * Remove HPI_UNUSED macro from hpifunc * Remove error->string translation, hence strcpy hpios_linux_kernel.* * Remove use of slab cache for DMA memory allocation tracking hpicmn.*: * Rename some functions to give Hpi prefix, made others static hpimod.c * Replace semaphore with mutex * Make driver cleanup more 'normal' hpi6205.c hpi6000.c hpimsgx.c * Pull some error handling up into hpimsgx.c Relevant DSP code changes (for alsa-firmware) add record format check for supported format. ASI6500/6500 - fix issue arising from back-to-back- adapter sample rate changes. ASI6000 - fix rare MP3 decode bug. ASI6585 - add support for clock source Livewire. ASI6416 - re-order control creation. Signed-off-by: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> regards -- Eliot Blennerhassett www.audioscience.com

1 0

[alsa-devel] HDA-Intel - Add support for Intel SCH
by Tobin Davis 30 Jan '08

30 Jan '08

This patch adds support for Intel's SCH mobile chipset. Signed off by Tobin Davis <tdavis(a)dsl-only.net>

2 1

[alsa-devel] ASOC v2 driver structure
by Jon Smirl 29 Jan '08

29 Jan '08

I'm trying to write an asoc v2 driver for my powerpc based SOC (mpc5200) hardware. We have two audio devices on the board, a WM8580 and a TAS5504. How do you do this with asoc v2? The card is created in snd_soc_machine_create(); there seems to be an assumption of a single card per machine. Or are you supposed to mix the different devices into a single pcm driver? What would happen in pxa2xx-pcm.c if more than one of these were defined? static int pxa2xx_pcm_probe(struct device *dev) { struct snd_soc_platform *platform = to_snd_soc_platform(dev); int ret; platform->pcm_ops = &pxa2xx_pcm_ops; platform->pcm_new = pxa2xx_pcm_new, platform->pcm_free = pxa2xx_pcm_free_dma_buffers, #if defined(CONFIG_SND_PXA2XX_SOC_I2S) ret = snd_soc_platform_add_dai(platform, &pxa2xx_i2s, 1); if (ret < 0) return ret; #endif #if defined(CONFIG_SND_PXA2XX_SOC_AC97) ret = snd_soc_platform_add_dai(platform, pxa2xx_ac97, 3); if (ret < 0) return ret; #endif #if defined(CONFIG_SND_PXA2XX_SOC_SSP) ret = snd_soc_platform_add_dai(platform, pxa2xx_ssp, 3); if (ret < 0) return ret; #endif ret = snd_soc_register_platform(platform); return ret; } -- Jon Smirl jonsmirl(a)gmail.com

4 6

[alsa-devel] [PATCH] hda: Add GPIO mute support to STAC9205
by Matthew Ranostay 29 Jan '08

29 Jan '08

Support added for detecting HP jack presence via GPIO on several laptop docks. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff -r b0d97ac73e0f pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Fri Jan 25 15:24:50 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Mon Jan 28 09:34:16 2008 -0500 @@ -122,7 +122,9 @@ struct sigmatel_spec { unsigned int alt_switch: 1; unsigned int hp_detect: 1; - unsigned int gpio_mask, gpio_data; + unsigned int gpio_en_mask, gpio_dir_mask; + unsigned int gpio_data; + unsigned int gpio_mute; unsigned char aloopback_mask; unsigned char aloopback_shift; @@ -2801,22 +2803,22 @@ static int stac9200_parse_auto_config(st * funky external mute control using GPIO pins. */ -static void stac_gpio_set(struct hda_codec *codec, unsigned int mask, - unsigned int data) +static void stac_gpio_set(struct hda_codec *codec, unsigned int en_mask, + unsigned int dir_mask, unsigned int data) { unsigned int gpiostate, gpiomask, gpiodir; gpiostate = snd_hda_codec_read(codec, codec->afg, 0, AC_VERB_GET_GPIO_DATA, 0); - gpiostate = (gpiostate & ~mask) | (data & mask); + gpiostate = (gpiostate & ~dir_mask) | (data & dir_mask); gpiomask = snd_hda_codec_read(codec, codec->afg, 0, AC_VERB_GET_GPIO_MASK, 0); - gpiomask |= mask; + gpiomask |= en_mask; gpiodir = snd_hda_codec_read(codec, codec->afg, 0, AC_VERB_GET_GPIO_DIRECTION, 0); - gpiodir |= mask; + gpiodir |= dir_mask; /* Configure GPIOx as CMOS */ snd_hda_codec_write(codec, codec->afg, 0, 0x7e7, 0); @@ -2911,7 +2913,8 @@ static int stac92xx_init(struct hda_code stac92xx_auto_set_pinctl(codec, cfg->dig_in_pin, AC_PINCTL_IN_EN); - stac_gpio_set(codec, spec->gpio_mask, spec->gpio_data); + stac_gpio_set(codec, spec->gpio_en_mask, + spec->gpio_dir_mask, spec->gpio_data); return 0; } @@ -3001,10 +3004,14 @@ static void stac92xx_hp_detect(struct hd int i, presence; presence = 0; + if (spec->gpio_mute) + presence = !(snd_hda_codec_read(codec, codec->afg, 0, + AC_VERB_GET_GPIO_DATA, 0) & spec->gpio_mute); + for (i = 0; i < cfg->hp_outs; i++) { - presence = get_hp_pin_presence(codec, cfg->hp_pins[i]); if (presence) break; + presence = get_hp_pin_presence(codec, cfg->hp_pins[i]); } if (presence) { @@ -3067,7 +3074,8 @@ static int stac92xx_resume(struct hda_co stac92xx_set_config_regs(codec); snd_hda_sequence_write(codec, spec->init); - stac_gpio_set(codec, spec->gpio_mask, spec->gpio_data); + stac_gpio_set(codec, spec->gpio_en_mask, + spec->gpio_dir_mask, spec->gpio_data); snd_hda_codec_resume_amp(codec); snd_hda_codec_resume_cache(codec); /* invoke unsolicited event to reset the HP state */ @@ -3301,7 +3309,8 @@ again: spec->num_dmuxes = ARRAY_SIZE(stac92hd73xx_dmux_nids); spec->dinput_mux = &stac92hd73xx_dmux; /* GPIO0 High = Enable EAPD */ - spec->gpio_mask = spec->gpio_data = 0x000001; + spec->gpio_en_mask = spec->gpio_dir_mask = 0x1; + spec->gpio_data = 0x01; spec->num_pwrs = ARRAY_SIZE(stac92hd73xx_pwr_nids); spec->pwr_nids = stac92hd73xx_pwr_nids; @@ -3375,7 +3384,8 @@ again: spec->aloopback_mask = 0x20; spec->aloopback_shift = 0; - spec->gpio_mask = spec->gpio_data = 0x00000001; /* GPIO0 High = EAPD */ + /* GPIO0 High = EAPD */ + spec->gpio_en_mask = spec->gpio_dir_mask = spec->gpio_data = 0x1; spec->mux_nids = stac92hd71bxx_mux_nids; spec->adc_nids = stac92hd71bxx_adc_nids; @@ -3431,7 +3441,8 @@ static int patch_stac922x(struct hda_cod stac922x_models, stac922x_cfg_tbl); if (spec->board_config == STAC_INTEL_MAC_V3) { - spec->gpio_mask = spec->gpio_data = 0x03; + spec->gpio_en_mask = spec->gpio_dir_mask = 0x03; + spec->gpio_data = 0x03; /* Intel Macs have all same PCI SSID, so we need to check * codec SSID to distinguish the exact models */ @@ -3559,7 +3570,8 @@ static int patch_stac927x(struct hda_cod case STAC_D965_3ST: case STAC_D965_5ST: /* GPIO0 High = Enable EAPD */ - spec->gpio_mask = spec->gpio_data = 0x00000001; + spec->gpio_en_mask = spec->gpio_dir_mask = 0x01; + spec->gpio_data = 0x01; spec->num_dmics = 0; spec->init = d965_core_init; @@ -3573,7 +3585,8 @@ static int patch_stac927x(struct hda_cod /* fallthru */ case STAC_DELL_3ST: /* GPIO2 High = Enable EAPD */ - spec->gpio_mask = spec->gpio_data = 0x00000004; + spec->gpio_en_mask = spec->gpio_dir_mask = 0x04; + spec->gpio_data = 0x04; spec->dmic_nids = stac927x_dmic_nids; spec->num_dmics = STAC927X_NUM_DMICS; @@ -3584,7 +3597,8 @@ static int patch_stac927x(struct hda_cod break; default: /* GPIO0 High = Enable EAPD */ - spec->gpio_mask = spec->gpio_data = 0x00000001; + spec->gpio_en_mask = spec->gpio_dir_mask = 0x1; + spec->gpio_data = 0x01; spec->num_dmics = 0; spec->init = stac927x_core_init; @@ -3679,15 +3693,25 @@ static int patch_stac9205(struct hda_cod stac92xx_set_config_reg(codec, 0x1f, 0x01441030); stac92xx_set_config_reg(codec, 0x20, 0x1c410030); - spec->gpio_mask = 0x0000000b; + /* Enable unsol response for GPIO4/Dock HP connection */ + snd_hda_codec_write(codec, codec->afg, 0, + AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK, 0x10); + snd_hda_codec_write_cache(codec, codec->afg, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + (AC_USRSP_EN | STAC_HP_EVENT)); + + spec->gpio_dir_mask = 0x0b; + spec->gpio_en_mask = 0x1b; + spec->gpio_mute = 0x10; /* GPIO0 High = EAPD, GPIO1 Low = Headphone Mute, - * GPIO3 High = DRM + * GPIO3 Low = DRM */ - spec->gpio_data = 0x00000009; + spec->gpio_data = 0x01; break; default: /* GPIO0 High = EAPD */ - spec->gpio_mask = spec->gpio_data = 0x00000001; + spec->gpio_en_mask = spec->gpio_dir_mask = 0x1; + spec->gpio_data = 0x01; break; }

2 3

[alsa-devel] [PATCH 2.6.24] hda_intel: ALSA HD Audio patch for Intel ICH10 DeviceID's
by Jason Gaston 29 Jan '08

29 Jan '08

This patch adds the Intel ICH10 HD Audio Controller DeviceID's. Signed-off-by: Jason Gaston <jason.d.gaston(a)intel.com> --- linux-2.6.24/sound/pci/hda/hda_intel.c.orig 2008-01-24 14:58:37.000000000 -0800 +++ linux-2.6.24/sound/pci/hda/hda_intel.c 2008-01-28 14:58:22.000000000 -0800 @@ -98,6 +98,7 @@ "{Intel, ESB2}," "{Intel, ICH8}," "{Intel, ICH9}," + "{Intel, ICH10}," "{ATI, SB450}," "{ATI, SB600}," "{ATI, RS600}," @@ -1936,6 +1937,8 @@ { 0x8086, 0x284b, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ICH }, /* ICH8 */ { 0x8086, 0x293e, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ICH }, /* ICH9 */ { 0x8086, 0x293f, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ICH }, /* ICH9 */ + { 0x8086, 0x3a3e, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ICH }, /* ICH10 */ + { 0x8086, 0x3a6e, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ICH }, /* ICH10 */ { 0x1002, 0x437b, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATI }, /* ATI SB450 */ { 0x1002, 0x4383, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATI }, /* ATI SB600 */ { 0x1002, 0x793b, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RS600 HDMI */

2 1

Re: [alsa-devel] [REPOST] opti93x: conversion to use cs4231 library
by krzysztof.h1＠poczta.fm 29 Jan '08

29 Jan '08

Takashi Iwai napisał(a): > At Sat, 26 Jan 2008 10:44:11 +0100, > Krzysztof Helt wrote: > > > > From: Krzysztof Helt >krzysztof.h1(a)wp.pl> > > > > This patch converts the Opti93x driver to use > > the cs4231 library instead of just duplicating the code. > > > > Signed-off-by: Krzysztof Helt >krzysztof.h1(a)wp.pl> > > This causes a compile error for snd-opti92x-cs4231. Also has some > coding style issues. > I am sorry for compilation error. I was sure I have tested compilation of all opti versions but obviously not. > The fixed patch is below. Please let me know this and previous > patches work so that I can push them to the upstream. > I don't have previous patches but I have few patches which depends on these two I posted (e.g. 3d control for opti 933, more coding style fixes). I will post them after these two are merged to have stable reference for diffs. Regards, Krzysztof ---------------------------------------------------------------------- Sprawdz, ktore komorki sa najmodniejsze! Kliknij >>> http://link.interia.pl/f1cd4

1 0

[alsa-devel] Migrating old application using ALSA v0.5.0 to use current ALSA v1.0.15
by Shaun Shirey 28 Jan '08

28 Jan '08

Good Afternoon, I am currently working on migrating an old application using an older version of ALSA v0.5.0 to a newer version v1.0.15. I have noticed that many functions have been removed entirely and the library itself has been completely re-structured. The problem is that I can't seem to find any documentation on the older library (the program also has little to no comments). I was wondering if anyone has any links to documentation or advice on porting ? Any help would be greatly appreciated.

1 0

[alsa-devel] Archive alsa versions
by Alan Horstmann 28 Jan '08

28 Jan '08

Are historic versions of ALSA available anywhere? I have been unable to locate any. It might be useful for me tracking down a bug to see the source code for alsa-driver-1.0.1 !! Alan

4 4

[alsa-devel] [REPOST] cs4231-lib: add support for Opti93x codec
by Krzysztof Helt 28 Jan '08

28 Jan '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This patch adds support for WSS compatible Opti93x codec to the cs4231-lib. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- This is the patch against the current alsa hg tree (which is the same as 2.6.24-rc8-mm1 kernel regarding files affected by the patch). It should apply to older kernels (2.6.24 final or -rcX) but not cleanly (shift by one line). diff -urp linux-mm-ref/include/sound/cs4231-regs.h linux-mm/include/sound/cs4231-regs.h --- linux-mm-ref/include/sound/cs4231-regs.h 2008-01-26 08:18:04.000000000 +0100 +++ linux-mm/include/sound/cs4231-regs.h 2008-01-26 08:45:44.312453695 +0100 @@ -177,4 +177,12 @@ #define CS4236_RIGHT_WAVE 0x1c /* right wavetable serial port volume */ #define CS4236_VERSION 0x9c /* chip version and ID */ +/* definitions for extended registers - OPTI93X */ +#define OPTi931_AUX_LEFT_INPUT 0x10 +#define OPTi931_AUX_RIGHT_INPUT 0x11 +#define OPTi93X_MIC_LEFT_INPUT 0x14 +#define OPTi93X_MIC_RIGHT_INPUT 0x15 +#define OPTi93X_OUT_LEFT 0x16 +#define OPTi93X_OUT_RIGHT 0x17 + #endif /* __SOUND_CS4231_REGS_H */ diff -urp linux-mm-ref/include/sound/cs4231.h linux-mm/include/sound/cs4231.h --- linux-mm-ref/include/sound/cs4231.h 2008-01-26 08:18:04.000000000 +0100 +++ linux-mm/include/sound/cs4231.h 2008-01-26 08:49:44.157843596 +0100 @@ -58,6 +58,7 @@ /* compatible, but clones */ #define CS4231_HW_INTERWAVE 0x1000 /* InterWave chip */ #define CS4231_HW_OPL3SA2 0x1101 /* OPL3-SA2 chip, similar to cs4231 */ +#define CS4231_HW_OPTI93X 0x1102 /* Opti 930/931/933 */ /* defines for codec.hwshare */ #define CS4231_HWSHARE_IRQ (1<<0) @@ -120,6 +121,8 @@ unsigned char snd_cs4236_ext_in(struct s void snd_cs4231_mce_up(struct snd_cs4231 *chip); void snd_cs4231_mce_down(struct snd_cs4231 *chip); +void snd_cs4231_overrange(struct snd_cs4231 *chip); + irqreturn_t snd_cs4231_interrupt(int irq, void *dev_id); const char *snd_cs4231_chip_id(struct snd_cs4231 *chip); diff -urp linux-mm-ref/sound/isa/cs423x/cs4231_lib.c linux-mm/sound/isa/cs423x/cs4231_lib.c --- linux-mm-ref/sound/isa/cs423x/cs4231_lib.c 2008-01-26 08:18:04.000000000 +0100 +++ linux-mm/sound/isa/cs423x/cs4231_lib.c 2008-01-26 08:45:44.324456487 +0100 @@ -119,6 +119,42 @@ static unsigned char snd_cs4231_original 0x00, /* 1f/31 - cbrl */ }; +static unsigned char snd_opti93x_original_image[32] = +{ + 0x00, /* 00/00 - l_mixout_outctrl */ + 0x00, /* 01/01 - r_mixout_outctrl */ + 0x88, /* 02/02 - l_cd_inctrl */ + 0x88, /* 03/03 - r_cd_inctrl */ + 0x88, /* 04/04 - l_a1/fm_inctrl */ + 0x88, /* 05/05 - r_a1/fm_inctrl */ + 0x80, /* 06/06 - l_dac_inctrl */ + 0x80, /* 07/07 - r_dac_inctrl */ + 0x00, /* 08/08 - ply_dataform_reg */ + 0x00, /* 09/09 - if_conf */ + 0x00, /* 0a/10 - pin_ctrl */ + 0x00, /* 0b/11 - err_init_reg */ + 0x0a, /* 0c/12 - id_reg */ + 0x00, /* 0d/13 - reserved */ + 0x00, /* 0e/14 - ply_upcount_reg */ + 0x00, /* 0f/15 - ply_lowcount_reg */ + 0x88, /* 10/16 - reserved/l_a1_inctrl */ + 0x88, /* 11/17 - reserved/r_a1_inctrl */ + 0x88, /* 12/18 - l_line_inctrl */ + 0x88, /* 13/19 - r_line_inctrl */ + 0x88, /* 14/20 - l_mic_inctrl */ + 0x88, /* 15/21 - r_mic_inctrl */ + 0x80, /* 16/22 - l_out_outctrl */ + 0x80, /* 17/23 - r_out_outctrl */ + 0x00, /* 18/24 - reserved */ + 0x00, /* 19/25 - reserved */ + 0x00, /* 1a/26 - reserved */ + 0x00, /* 1b/27 - reserved */ + 0x00, /* 1c/28 - cap_dataform_reg */ + 0x00, /* 1d/29 - reserved */ + 0x00, /* 1e/30 - cap_upcount_reg */ + 0x00 /* 1f/31 - cap_lowcount_reg */ +}; + /* * Basic I/O functions */ @@ -895,7 +931,7 @@ static int snd_cs4231_capture_prepare(st return 0; } -static void snd_cs4231_overrange(struct snd_cs4231 *chip) +void snd_cs4231_overrange(struct snd_cs4231 *chip) { unsigned long flags; unsigned char res; @@ -1054,8 +1090,11 @@ static int snd_cs4231_probe(struct snd_c chip->image[CS4231_IFACE_CTRL] = (chip->image[CS4231_IFACE_CTRL] & ~CS4231_SINGLE_DMA) | (chip->single_dma ? CS4231_SINGLE_DMA : 0); - chip->image[CS4231_ALT_FEATURE_1] = 0x80; - chip->image[CS4231_ALT_FEATURE_2] = chip->hardware == CS4231_HW_INTERWAVE ? 0xc2 : 0x01; + if (chip->hardware != CS4231_HW_OPTI93X) { + chip->image[CS4231_ALT_FEATURE_1] = 0x80; + chip->image[CS4231_ALT_FEATURE_2] = + chip->hardware == CS4231_HW_INTERWAVE ? 0xc2 : 0x01; + } ptr = (unsigned char *) &chip->image; snd_cs4231_mce_down(chip); spin_lock_irqsave(&chip->reg_lock, flags); @@ -1376,6 +1415,7 @@ const char *snd_cs4231_chip_id(struct sn case CS4231_HW_INTERWAVE: return "AMD InterWave"; case CS4231_HW_OPL3SA2: return chip->card->shortname; case CS4231_HW_AD1845: return "AD1845"; + case CS4231_HW_OPTI93X: return "OPTi 93x"; default: return "???"; } } @@ -1401,7 +1441,12 @@ static int snd_cs4231_new(struct snd_car chip->rate_constraint = snd_cs4231_xrate; chip->set_playback_format = snd_cs4231_playback_format; chip->set_capture_format = snd_cs4231_capture_format; - memcpy(&chip->image, &snd_cs4231_original_image, sizeof(snd_cs4231_original_image)); + if (chip->hardware == CS4231_HW_OPTI93X) + memcpy(&chip->image, &snd_opti93x_original_image, + sizeof(snd_opti93x_original_image)); + else + memcpy(&chip->image, &snd_cs4231_original_image, + sizeof(snd_cs4231_original_image)); *rchip = chip; return 0; @@ -1790,6 +1835,32 @@ CS4231_SINGLE("Loopback Capture Switch", CS4231_SINGLE("Loopback Capture Volume", 0, CS4231_LOOPBACK, 2, 63, 1) }; +static struct snd_kcontrol_new snd_opti93x_controls[] __devinitdata = { +CS4231_DOUBLE("Master Playback Switch", 0, OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 7, 7, 1, 1), +CS4231_DOUBLE("Master Playback Volume", 0, OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 1, 1, 31, 1), +CS4231_DOUBLE("PCM Playback Switch", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), +CS4231_DOUBLE("PCM Playback Volume", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 31, 1), +CS4231_DOUBLE("FM Playback Switch", 0, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), +CS4231_DOUBLE("FM Playback Volume", 0, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 1, 1, 15, 1), +CS4231_DOUBLE("Line Playback Switch", 0, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), +CS4231_DOUBLE("Line Playback Volume", 0, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 15, 1), +CS4231_DOUBLE("Mic Playback Switch", 0, OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 7, 7, 1, 1), +CS4231_DOUBLE("Mic Playback Volume", 0, OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 1, 1, 15, 1), +CS4231_DOUBLE("Mic Boost", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 5, 5, 1, 0), +CS4231_DOUBLE("CD Playback Switch", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), +CS4231_DOUBLE("CD Playback Volume", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 1, 1, 15, 1), +CS4231_DOUBLE("Aux Playback Switch", 0, OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 7, 7, 1, 1), +CS4231_DOUBLE("Aux Playback Volume", 0, OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 1, 1, 15, 1), +CS4231_DOUBLE("Capture Volume", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 0, 0, 15, 0), +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = snd_cs4231_info_mux, + .get = snd_cs4231_get_mux, + .put = snd_cs4231_put_mux, +} +}; + int snd_cs4231_mixer(struct snd_cs4231 *chip) { struct snd_card *card; @@ -1802,10 +1873,22 @@ int snd_cs4231_mixer(struct snd_cs4231 * strcpy(card->mixername, chip->pcm->name); - for (idx = 0; idx < ARRAY_SIZE(snd_cs4231_controls); idx++) { - if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_cs4231_controls[idx], chip))) < 0) - return err; - } + if (chip->hardware == CS4231_HW_OPTI93X) + for (idx = 0; idx < ARRAY_SIZE(snd_opti93x_controls); idx++) { + err = snd_ctl_add(card, + snd_ctl_new1(&snd_opti93x_controls[idx], + chip)); + if (err < 0) + return err; + } + else + for (idx = 0; idx < ARRAY_SIZE(snd_cs4231_controls); idx++) { + err = snd_ctl_add(card, + snd_ctl_new1(&snd_cs4231_controls[idx], + chip)); + if (err < 0) + return err; + } return 0; } @@ -1815,6 +1898,7 @@ EXPORT_SYMBOL(snd_cs4236_ext_out); EXPORT_SYMBOL(snd_cs4236_ext_in); EXPORT_SYMBOL(snd_cs4231_mce_up); EXPORT_SYMBOL(snd_cs4231_mce_down); +EXPORT_SYMBOL(snd_cs4231_overrange); EXPORT_SYMBOL(snd_cs4231_interrupt); EXPORT_SYMBOL(snd_cs4231_chip_id); EXPORT_SYMBOL(snd_cs4231_create); ---------------------------------------------------------------------- Lezac w wannie skorzystala z ... 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[alsa-devel] [PATCH] replace logical- by bit-and in alc_subsystem_id(), sound/pci/hda/patch_realtek.c
by Roel Kluin 28 Jan '08

28 Jan '08

This was found in linus' git tree, am I right here? -- replace logical- by bit-and Signed-off-by: Roel Kluin <12o3l(a)tiscali.nl> --- diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1c50278..0a13c53 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -739,7 +739,7 @@ static void alc_subsystem_id(struct hda_codec *codec, /* check sum */ tmp = 0; for (i = 1; i < 16; i++) { - if ((ass >> i) && 1) + if ((ass >> i) & 1) tmp++; } if (((ass >> 16) & 0xf) != tmp)

2 1

[alsa-devel] A Weird issue on Gigabyte board
by Jiang zhe 28 Jan '08

28 Jan '08

Hi, I meet with a weird issue. Please refer to the bug 0003674 on bugtrack.alsa-project.org. It use alc889 codec but the codec vendor id is 0885. With the alsa-driver-hg20080110.tar.bz2, it works well with option "model=6stack-dig". But after I simply added the this line into the alc882_cfg_tbl[]: SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte P35-DS3R", ALC882_6ST_DIG), The dmesg still show "hda_codec: Unknown model for ALC882, trying auto-probe from BIOS...". and driver will crash. With this patch, even loading it with option "model=6stack-dig" doesn't help, driver will still crash. I found a line in the dmesg: "hda_intel: azx_get_response timeout, switching to polling mode: last cmd=0xbe0f0009." The Cad is 0xb. The codec_mask can change from 0x4 to 0x1: alsa-kernel/pci/hda/hda_intel.c:732: codec_mask = 0x4 alsa-kernel/pci/hda/hda_intel.c:732: codec_mask = 0x1 The os is kubuntu,kernel version is 2.6.22-14. Here is some dmesg info: [ 36.934143] ALSA /home/matt/Data/alsa-driver-hg20080110/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:732: codec_mask = 0x4 [ 36.935975] hda-codec: In the pacth_alc882,board_config is -1 [ 36.935976] hda_codec: Unknown model for ALC882, trying auto-probe from BIOS... [ 36.936469] ALSA /home/matt/Data/alsa-driver-hg20080110/pci/hda/hda_codec.c:2794: autoconfig: line_outs=4 (0x14/0x15/0x16/0x17/0x0) [ 36.936471] ALSA /home/matt/Data/alsa-driver-hg20080110/pci/hda/hda_codec.c:2798: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 36.936473] ALSA /home/matt/Data/alsa-driver-hg20080110/pci/hda/hda_codec.c:2802: hp_outs=1 (0x1b/0x0/0x0/0x0/0x0) [ 36.936476] ALSA /home/matt/Data/alsa-driver-hg20080110/pci/hda/hda_codec.c:2810: inputs: mic=0x18, fmic=0x19, line=0x1a, fline=0x0, cd=0x0, aux=0x0 [ 37.935491] ALSA /home/matt/Data/alsa-driver-hg20080110/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:581: hda_intel: azx_get_response timeout, switching to polling mode: last cmd=0xe1cf0009 [ 38.936857] ALSA /home/matt/Data/alsa-driver-hg20080110/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:588: hda_intel: azx_get_response timeout, switching to single_cmd mode: last cmd=0xe1cf0009 [ 38.936937] ALSA /home/matt/Data/alsa-driver-hg20080110/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:644: hda-intel: get_response timeout: IRS=0x1 [ 38.936940] ALSA /home/matt/Data/alsa-driver-hg20080110/pci/hda/hda_codec.c:2065: BUG? (info->nid) [ 38.936946] [<f8c19bc0>] snd_hda_build_pcms+0x1e0/0x220 [snd_hda_intel] [ 38.936962] [<f8c1598c>] azx_probe+0x71c/0xa50 [snd_hda_intel] [ 38.936974] [<f8c14850>] azx_send_cmd+0x0/0x130 [snd_hda_intel] [ 38.936981] [<f8c14980>] azx_get_response+0x0/0x200 [snd_hda_intel] [ 38.936991] [<f8c146c0>] azx_power_notify+0x0/0x70 [snd_hda_intel] [ 38.937002] [<c0209a66>] pci_device_probe+0x56/0x80 [ 38.937008] [<c026110e>] driver_probe_device+0x8e/0x190 [ 38.937013] [<c026137e>] __driver_attach+0x9e/0xa0 [ 38.937017] [<c02604fb>] bus_for_each_dev+0x3b/0x60 [ 38.937022] [<c0260f86>] driver_attach+0x16/0x20 [ 38.937024] [<c02612e0>] __driver_attach+0x0/0xa0 [ 38.937026] [<c02608ca>] bus_add_driver+0x8a/0x1b0 [ 38.937031] [<c0209c13>] __pci_register_driver+0x53/0xa0 [ 38.937035] [<c014a7d1>] sys_init_module+0x151/0x1a00 [ 38.937038] [<c01fb39f>] prio_tree_insert+0x1f/0x250 [ 38.937052] [<c01041d2>] sysenter_past_esp+0x6b/0xa9 [ 38.937058] ======================= [ 38.937097] ACPI: PCI interrupt for device 0000:00:1b.0 disabled [ 38.937102] HDA Intel: probe of 0000:00:1b.0 failed with error -22 [ 38.937124] ACPI: PCI Interrupt 0000:01:00.1[B] -> GSI 17 (level, low) -> IRQ 23 [ 38.937134] PCI: Setting latency timer of device 0000:01:00.1 to 64 [ 38.952813] ALSA /home/matt/Data/alsa-driver-hg20080110/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:732: codec_mask = 0x1

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[alsa-devel] a7839e96 (PNP: increase max resources) breaks my ALSA intel8x0 sound card
by Avuton Olrich 28 Jan '08

28 Jan '08

With v2.6.24 my second ALSA sound device stopped working. After bisection it says this was the offending commit. a7839e960675b549f06209d18283d5cee2ce9261 is first bad commit commit a7839e960675b549f06209d18283d5cee2ce9261 Author: Zhao Yakui <yakui.zhao(a)intel.com> Date: Wed Nov 28 16:21:21 2007 -0800 PNP: increase the maximum number of resources On some systems the number of resources(IO,MEM) returnedy by PNP device is greater than the PNP constant, for example motherboard devices. It brings that some resources can't be reserved and resource confilicts. This will cause PCI resources are assigned wrongly in some systems, and cause hang. This is a regression since we deleted ACPI motherboard driver and use PNP system driver. [akpm(a)linux-foundation.org: fix text and coding-style a bit] Signed-off-by: Li Shaohua <shaohua.li(a)intel.com> Signed-off-by: Zhao Yakui <yakui.zhao(a)intel.com> Cc: Bjorn Helgaas <bjorn.helgaas(a)hp.com> Cc: Thomas Renninger <trenn(a)suse.de> Cc: <stable(a)kernel.org> Signed-off-by: Andrew Morton <akpm(a)linux-foundation.org> Signed-off-by: Linus Torvalds <torvalds(a)linux-foundation.org> The audio device is 00:1b.0 (see my lspci -vvv output), the other audio device works fine. http://avuton.googlepages.com/v2.6-before (dmesg revision before) http://avuton.googlepages.com/v2.6-after (dmesg broken revision) http://avuton.googlepages.com/lspci-vvv http://avuton.googlepages.com/config (from the broken revision) http://avuton.googlepages.com/iomem http://avuton.googlepages.com/ioports -- avuton -- Anyone who quotes me in their sig is an idiot. -- Rusty Russell.

3 2

[alsa-devel] Broken seq
by Alan Horstmann 27 Jan '08

27 Jan '08

Can someone help with this... During repeated testing of MIDI, something has broken the seq system, so when loading previously working sound drivers, I get a message: /proc/asound/seq/clients File not found (or similar) and sure enough it is not there. Why has it gone and how can I get it back? Thanks Alan

3 3

[alsa-devel] installation steps for us-122l tascam driver
by Xavier garabedian 27 Jan '08

27 Jan '08

hello i'm a linux beginner and i'm trying to make the us 122l work on ubuntu. i have seen the patches that you published, but i don't know the steps to install them and make the tascam work. is there anyone who can discribe me the process, step by step ? thank you very much in advance best regards xavier garabedian

2 1

[alsa-devel] Intel HDA (ICH8) unsupported model?
by Philipp E. Letschert 27 Jan '08

27 Jan '08

Hello, This is for the Zepto Znote 6224W Notebook (and probably others): lspci -v 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03) Subsystem: Inventec Corporation Unknown device 0040 Flags: bus master, fast devsel, latency 0, IRQ 22 Memory at f0500000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Capabilities: [70] Express Unknown type IRQ 0 Capabilities: [100] Virtual Channel Capabilities: [130] Unknown (5) lspci -vvn 00:1b.0 0403: 8086:284b (rev 03) Subsystem: 1170:0040 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 22 Region 0: Memory at f0500000 (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] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Address: 0000000000000000 Data: 0000 Capabilities: [70] Express Unknown type IRQ 0 Device: Supported: MaxPayload 128 bytes, PhantFunc 0, ExtTag- Device: Latency L0s <64ns, L1 <1us Device: Errors: Correctable- Non-Fatal- Fatal- Unsupported- Device: RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ Device: MaxPayload 128 bytes, MaxReadReq 128 bytes Link: Supported Speed unknown, Width x0, ASPM unknown, Port 0 Link: Latency L0s <64ns, L1 <1us Link: ASPM Disabled CommClk- ExtSynch- Link: Speed unknown, Width x0 Capabilities: [100] Virtual Channel Capabilities: [130] Unknown (5) Enabling sound caused some headaches. I finally ended with kernel 2.6.24 and alsa 1.0.16rc1. The card was autodetected as Realtec ALC268, but no sound output. Using option model=toshiba instead, when loading the snd-hda-intel module did work, but sound has a really poor quality, with lots of crackling noises. Enabling the samplerate plugin didnt help... Please let me know, if this card is already supported and I just have errors in my configuration, or further informations are helpful for develepment. Thanks, Phil

1 0

[alsa-devel] Usb driver correction
by LCID Fire 27 Jan '08

27 Jan '08

Hi. I hope you guys can help me with 3 problems I have. I have a usb device which has 3 streams - which all work fine (that's not the problem yet ;)). Aside from this there is also a mixer created. Alsa cannot retrieve any information from the mixer - not the minimum, nor the maximum or current value. Basicly alsa leaves the mixer in a state where it cannot be used - which leads to a question why not omnit the mixer? What I am ending up with are these 2 volumes: [4] FU [PCM Capture Volume] ch = 1, val = 0/1/1 [9] FU [PCM Capture Volume] ch = 1, val = 0/1/1 What would I have to do to disable these? Is it enough to define a usbmix_name_map in usbmixer_maps.c with {4, NULL} and {9, NULL}? Also, the usb device has a switch - from line input to phono input. I know what control message I have to send to change the switch - but where has such an additional switch to be implemented? Is it similar to snd_audigy2nx_led_get in usbmixer.c? Would be great if someone could explain these mysteries to me ;)

1 0

[alsa-devel] Microphone dead: Gateway MX6543, hda-intel, STAC925x
by Zoran Rilak 27 Jan '08

27 Jan '08

Hello, I've searched the archives and haven't found a solution for this particular model (STAC9250). Front mic does not work. I have tried other models, aside from m2-2, and all four possible values for position_fix as well, but to no avail. If anyone is willing to tackle this, please point me to what other information I should provide in order to get to the bottom of this. I am using Ubuntu kernel 2.6.22-14 and ALSA 1.0.15 compiled from source. Here are my /proc/asound/card0/codec#* files: --- /proc/asound/card0/codec#0 --- Codec: SigmaTel STAC9250 Address: 0 Vendor Id: 0x83847634 Subsystem Id: 0x107b0367 Revision Id: 0x100101 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: N/A Default Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Node 0x02 [Audio Output] wcaps 0xd0401: Stereo Power: 0x0 Node 0x03 [Audio Input] wcaps 0x1d0541: Stereo Power: 0x0 Connection: 1 0x14 Node 0x04 [Audio Input] wcaps 0x140311: Stereo Digital PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Connection: 1 0x07 Node 0x05 [Audio Output] wcaps 0x40211: Stereo Digital PCM: rates [0x1e0]: 44100 48000 88200 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Node 0x06 [Audio Selector] wcaps 0x300901: Stereo Connection: 3 0x02* 0x07 0x14 Node 0x07 [Pin Complex] wcaps 0x430681: Stereo Digital Pincap 0x0810024: IN EAPD Detect Pin Default 0x40c003f3: [N/A] SPDIF In at Ext N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x00: Power: 0x0 Node 0x08 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x0810: OUT Pin Default 0x424503f2: [N/A] SPDIF Out at Ext Front Conn = Optical, Color = Unknown Pin-ctls: 0x00: Connection: 2 0x05* 0x14 Node 0x09 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=0 Amp-Out vals: [0x0f 0x0f] Connection: 1 0x0f Node 0x0a [Pin Complex] wcaps 0x400181: Stereo Pincap 0x083f: IN OUT HP Detect Pin Default 0x04180011: [Jack] Speaker at Ext Right Conn = DIN, Color = Unknown Pin-ctls: 0x40: OUT Connection: 1 0x0e Node 0x0b [Pin Complex] wcaps 0x400181: Stereo Pincap 0x081737: IN OUT Detect Pin Default 0x02a19020: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink Pin-ctls: 0x24: IN Connection: 1 0x0e Node 0x0c [Pin Complex] wcaps 0x400181: Stereo Pincap 0x081737: IN OUT Detect Pin Default 0x50a103f0: [N/A] Mic at Int N/A Conn = 1/8, Color = Unknown Pin-ctls: 0x20: IN Connection: 1 0x0e Node 0x0d [Pin Complex] wcaps 0x400181: Stereo Pincap 0x083f: IN OUT HP Detect Pin Default 0x90100212: [Fixed] Speaker at Int N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x40: OUT Connection: 1 0x0e Node 0x0e [Audio Selector] wcaps 0x300105: Stereo Amp-Out Amp-Out caps: N/A Amp-Out vals: [0x0d 0x0d] Connection: 1 0x06 Node 0x0f [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-Out vals: [0x04 0x04] Connection: 5 0x0b* 0x0c 0x0d 0x0a 0x11 Node 0x10 [Pin Complex] wcaps 0x400104: Mono Amp-Out Amp-Out caps: N/A Amp-Out vals: [0x00] Pincap 0x0810: OUT Pin Default 0x400003f1: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x00: Connection: 1 0x12 Node 0x11 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x9033032e: [Fixed] CD at Int N/A Conn = ATAPI, Color = Unknown Pin-ctls: 0x20: IN Node 0x12 [Audio Mixer] wcaps 0x200100: Mono Connection: 1 0x06 Node 0x13 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x03, nsteps=0x03, stepsize=0x17, mute=1 Amp-Out vals: [0x00] Node 0x14 [Audio Selector] wcaps 0x30090d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Connection: 1 0x09 --- end codec#0 --- --- /proc/asound/card0/codec#1 --- Codec: Motorola Si3054 Address: 1 Vendor Id: 0x10573057 Subsystem Id: 0x10001 Revision Id: 0x100100 Modem Function Group: 0x1 --- end codec#1 ---

1 0

[alsa-devel] [PATCH] Add missing additional verbs for SPDIF with AD1984 & thinkpad
by Jerone Young 26 Jan '08

26 Jan '08

Additional verbs are missing for IEC958 code for the AD1984 in thinkpads. This is identical to the verbs in AD19884. I missed these in my patch last month. Signed-off-by: Jerone Young <jerone(a)gmail.com> diff -r dcc1b9d533d3 alsa-kernel/pci/hda/patch_analog.c --- a/alsa-kernel/pci/hda/patch_analog.c Fri Jan 25 19:49:19 2008 -0600 +++ b/alsa-kernel/pci/hda/patch_analog.c Fri Jan 25 19:51:05 2008 -0600 @@ -3188,6 +3188,9 @@ static struct hda_verb ad1984_thinkpad_i {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* enable EAPD bit */ {0x12, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, + /* SPDIF output selector */ + {0x02, AC_VERB_SET_CONNECT_SEL, 0x0}, /* PCM */ + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */ }; /* Digial MIC ADC NID 0x05 + 0x06 */

2 2

[alsa-devel] [PATCH] Remove psuedo master from pci/hda/analog_patch.c . It does not function well!
by Jerone Young 26 Jan '08

26 Jan '08

Hi I noticed after installing 1.0.16rc-1 today that my Thinkpad T61 with AD1984 had a Master mixer now. Problem is it's not very good in that it 1) does not follow with PCM (the real master) volume well. It causes a volume disparity and does not follow or even boost PCM signal. So it just a bad mister. 2) it does not mute the volume (since PCM is the real master mixer you have to go to PCM to mute the volume!) I have fixed a problem in Gstreamer so that Gnome (gnome volume mixer) does finally work correctly for the AD1984 which does not have a Master. When it looks for a master mixer it now will fallback to PCM as a master mixer (the code is upsteam too). You can find this here: http://bugzilla.gnome.org/show_bug.cgi?id=506928 I have a feeling this is why this patch was put in. I have fixed the problem in Gnome, KDE has allways worked. just fine. But I would honestly love it if this code was removed!!! If you need a Master can you rename PCM to master for the AD1984! I have attached a patch to remove this code .. because it is so does not function well! Signed-off-by : Jerone Young <jerone(a)gmail.com> diff -r 5dfc8324ae17 alsa-kernel/pci/hda/patch_analog.c --- a/alsa-kernel/pci/hda/patch_analog.c Fri Jan 25 00:12:14 2008 -0600 +++ b/alsa-kernel/pci/hda/patch_analog.c Fri Jan 25 01:04:13 2008 -0600 @@ -170,26 +170,6 @@ static int ad198x_build_controls(struct } if (spec->dig_in_nid) { err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid); - if (err < 0) - return err; - } - - /* if we have no master control, let's create it */ - if (!snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) { - snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid, - HDA_OUTPUT, spec->vmaster_tlv); - err = snd_hda_add_vmaster(codec, "Master Playback Volume", - spec->vmaster_tlv, - (spec->slave_vols ? - spec->slave_vols : ad_slave_vols)); - if (err < 0) - return err; - } - if (!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) { - err = snd_hda_add_vmaster(codec, "Master Playback Switch", - NULL, - (spec->slave_sws ? - spec->slave_sws : ad_slave_sws)); if (err < 0) return err; }

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Re: [alsa-devel] Error when compiling from HG on Ubuntu 7.10
by Pavel Hofman 25 Jan '08

25 Jan '08

I am sorry for the false alarm. Clean hg clone and hgcompile works flawlessly. Pavel Hofman. > Hello, > > While trying to compile the latest alsa-kernel + alsa-drivers from HG > using ./hgcompile, I am hitting the following error: > > CC [M] /home/pavel/projects/alsa/alsa-driver/i2c/other/tea575x-tuner.o > In file included from > /home/pavel/projects/alsa/alsa-driver/i2c/other/tea575x-tuner.c:23: > /home/pavel/projects/alsa/alsa-driver/include/sound/driver.h:1:2: > warning: #warning "This file is deprecated" > /home/pavel/projects/alsa/alsa-driver/i2c/other/tea575x-tuner.c:60: > error: redefinition of ‘video_get_drvdata’ > include/media/v4l2-dev.h:383: error: previous definition of > ‘video_get_drvdata’ was here

1 0

[alsa-devel] [PATCH] hda: In-Amp support for 92HD7xxx codecs.
by Matthew Ranostay 25 Jan '08

25 Jan '08

Some 92HD7xxx codecs have amps on the ports to volume control and/or mute certain ports. Also this makes stac92hd71bxx unmute amps lines in the init not needed. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff -r 5bf4c5d02f4b pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Thu Jan 24 15:32:15 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Thu Jan 24 11:25:36 2008 -0500 @@ -577,10 +577,6 @@ static struct hda_verb stac92hd71bxx_cor /* connect headphone jack to dac1 */ { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01}, { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Speaker */ - /* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */ - { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, }; static struct hda_verb stac92hd71bxx_analog_core_init[] = { @@ -594,11 +590,6 @@ static struct hda_verb stac92hd71bxx_ana { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Speaker */ /* unmute dac0 input in audio mixer */ { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0x701f}, - /* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */ - { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {} }; static struct hda_verb stac925x_core_init[] = { @@ -2215,6 +2206,37 @@ static int create_controls(struct sigmat return 0; } +/* add playback controls for ports that have amps */ +static int stac92xx_create_amp_ctls(struct hda_codec *codec, + hda_nid_t nid, char *pfx, int idx) +{ + struct sigmatel_spec *spec = codec->spec; + int err; + char name[48]; + u32 caps = query_amp_caps(codec, nid, HDA_INPUT); + if (idx) + sprintf(name, "%s %d", pfx, idx); + else + strcpy(name, pfx); + + if ((caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT) { + sprintf(name, "%s Playback Volume", name); + err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, name, + HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT)); + if (err < 0) + return err; + } + + if ((caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT) { + sprintf(name, "%s Playback Switch", name); + err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, name, + HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT)); + if (err < 0) + return err; + } + return 0; +} + /* add playback controls from the parsed DAC table */ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) @@ -2262,13 +2284,39 @@ static int stac92xx_auto_create_multi_ou } } - if (spec->line_switch) - if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_IO_SWITCH, "Line In as Output Switch", cfg->input_pins[AUTO_PIN_LINE] << 8)) < 0) + if (spec->line_switch) { + int val = cfg->input_pins[AUTO_PIN_LINE] << 8; + wid_caps = get_wcaps(codec, val >> 8); + + err = stac92xx_add_control(spec, STAC_CTL_WIDGET_IO_SWITCH, + "Line In as Output Switch", val); + if (err < 0) return err; - if (spec->mic_switch) - if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_IO_SWITCH, "Mic as Output Switch", (cfg->input_pins[AUTO_PIN_MIC] << 8) | 1)) < 0) + if (wid_caps & AC_WCAP_IN_AMP) { + err = stac92xx_create_amp_ctls(codec, val >> 8, + "Line In as Output Gain", 0); + if (err < 0) + return err; + } + } + + if (spec->mic_switch) { + int val = cfg->input_pins[AUTO_PIN_MIC] << 8; + wid_caps = get_wcaps(codec, val >> 8); + + err = stac92xx_add_control(spec, STAC_CTL_WIDGET_IO_SWITCH, + "Mic as Output Switch", val | 1); + if (err < 0) return err; + + if (wid_caps & AC_WCAP_IN_AMP) { + err = stac92xx_create_amp_ctls(codec, val >> 8, + "Mic as Output Gain", 0); + if (err < 0) + return err; + } + } return 0; } @@ -2311,6 +2359,13 @@ static int stac92xx_auto_create_hp_ctls( spec->hp_detect = 1; nid = snd_hda_codec_read(codec, cfg->hp_pins[i], 0, AC_VERB_GET_CONNECT_LIST, 0) & 0xff; + if (wid_caps & AC_WCAP_IN_AMP) { + err = stac92xx_create_amp_ctls(codec, + cfg->hp_pins[i], + "Headphone Gain", i); + if (err < 0) + return err; + } if (check_in_dac_nids(spec, nid)) nid = 0; if (! nid) @@ -2320,6 +2375,14 @@ static int stac92xx_auto_create_hp_ctls( for (i = 0; i < cfg->speaker_outs; i++) { nid = snd_hda_codec_read(codec, cfg->speaker_pins[i], 0, AC_VERB_GET_CONNECT_LIST, 0) & 0xff; + if (get_wcaps(codec, cfg->speaker_pins[i]) & AC_WCAP_IN_AMP) { + err = stac92xx_create_amp_ctls(codec, + cfg->speaker_pins[i], + "Speaker Gain", i); + if (err < 0) + return err; + } + if (check_in_dac_nids(spec, nid)) nid = 0; if (! nid) @@ -2329,6 +2392,13 @@ static int stac92xx_auto_create_hp_ctls( for (i = 0; i < cfg->line_outs; i++) { nid = snd_hda_codec_read(codec, cfg->line_out_pins[i], 0, AC_VERB_GET_CONNECT_LIST, 0) & 0xff; + if (get_wcaps(codec, cfg->line_out_pins[i]) & AC_WCAP_IN_AMP) { + err = stac92xx_create_amp_ctls(codec, + cfg->line_out_pins[i], + "Line Out Gain", i); + if (err < 0) + return err; + } if (check_in_dac_nids(spec, nid)) nid = 0; if (! nid)

2 3

[alsa-devel] [PATCH] hda: STAC92xx Line In/Mic as output check
by Matthew Ranostay 25 Jan '08

25 Jan '08

This patch checks to see the Line In/Mic port have the ability to do output before creating the the control switches. The 92hd71bxx series of codecs has this issue with the port 0xe, which only allows input. Signed-off-by: Matthew Ranostay <mranostay(a)embededalley.com> --- diff -r 303f894ac2a8 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Thu Jan 24 18:46:42 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Thu Jan 24 22:41:27 2008 -0500 @@ -2226,7 +2226,7 @@ static int stac92xx_auto_create_multi_ou int i, err; struct sigmatel_spec *spec = codec->spec; - unsigned int wid_caps; + unsigned int wid_caps, pincap; for (i = 0; i < cfg->line_outs; i++) { @@ -2262,13 +2262,31 @@ static int stac92xx_auto_create_multi_ou } } - if (spec->line_switch) - if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_IO_SWITCH, "Line In as Output Switch", cfg->input_pins[AUTO_PIN_LINE] << 8)) < 0) - return err; - - if (spec->mic_switch) - if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_IO_SWITCH, "Mic as Output Switch", (cfg->input_pins[AUTO_PIN_MIC] << 8) | 1)) < 0) - return err; + if (spec->line_switch) { + nid = cfg->input_pins[AUTO_PIN_LINE]; + pincap = snd_hda_param_read(codec, nid, + AC_PAR_PIN_CAP); + if (pincap & AC_PINCAP_OUT) { + err = stac92xx_add_control(spec, + STAC_CTL_WIDGET_IO_SWITCH, + "Line In as Output Switch", nid << 8); + if (err < 0) + return err; + } + } + + if (spec->mic_switch) { + nid = cfg->input_pins[AUTO_PIN_MIC]; + pincap = snd_hda_param_read(codec, nid, + AC_PAR_PIN_CAP); + if (pincap & AC_PINCAP_OUT) { + ` err = stac92xx_add_control(spec, + STAC_CTL_WIDGET_IO_SWITCH, + "Mic as Output Switch", (nid << 8) | 1); + if (err < 0) + return err; + } + } return 0; }

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Re: [alsa-devel] 1.0.16rc1 release -- error: 'system_utsname' undeclared
by Takashi Iwai 25 Jan '08

25 Jan '08

At Fri, 25 Jan 2008 22:48:43 +1100, Rick Measham wrote: > > > Rick Measham wrote: > >> dmesg tells me there are zero speaker_outs. So I figure it didn't > >> auto-detect them, and there are no pins specified for this exact setup. > > Takashi Iwai wrote: > > OK, it means that either the BIOS set up is broken or the preset > > pincfg is broken. Could you check which board_config is chosen for > > your device? It's selected via PCI SSID (see lspci -nv output). > > > > If it's a bugin preset pincfg, it'd be realtivelyeasy to fix (once > > after we figure out which pin corresponds to the spekaer output). > > I'm certainly hoping for the latter .. and it looks hopeful > > 00:1b.0 0403: 8086:27d8 (rev 01) > Subsystem: 1028:01bf > Flags: bus master, fast devsel, latency 0, IRQ 21 > Memory at dfffc000 (64-bit, non-prefetchable) [size=16K] > Capabilities: [50] Power Management version 2 > Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ > Queue=0/0 Enable- > Capabilities: [70] Express Unknown type IRQ 0 > > Lines up with: > /* > STAC 922X pin configs for > 102801BF > */ > static unsigned int dell_922x_m81_pin_configs[10] = { > 0x0321101f, 0x01112024, 0x01111222, 0x91174220, > 0x03a11050, 0x01116221, 0x90a70330, 0x01452340, > 0x40C003f1, 0x405003f0, > }; > > > So it certainly looks like we might have some hope? > > It's listed as "unknown Dell" .. so let me fill that blank: It's the XPS > M2010. > > Now for the billion dollar question: How do I work out what the > pin_configs SHOULD be? I can play guess-the-number, but that's going to > take some time and I'd guess a lot of crashing! That pin config actually looks strange. It defines 4 "speaker" outputs. The autocfg routine in hda-intel driver switches these speaker outputs to line-outs (just for compatibility issue). That's why you don't see any speaker there. The patch below is a quick fix to change the jack types for 3 outputs. Give it a try. Now more questions: - Does your machine have 3 output jacks in addition to HP? - What is the content of /proc/asound/card0/codec#* when you load the driver with model=auto? After setting it, reboot so that BIOS resets the configuration. - Show both codec#* files for model=auto and without model option. thanks, Takashi --- diff -r 19029da61288 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Fri Jan 25 11:56:57 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Fri Jan 25 13:04:20 2008 +0100 @@ -1273,8 +1273,8 @@ static unsigned int dell_922x_d82_pin_co 102801BF */ static unsigned int dell_922x_m81_pin_configs[10] = { - 0x0321101f, 0x01112024, 0x01111222, 0x91174220, - 0x03a11050, 0x01116221, 0x90a70330, 0x01452340, + 0x0321101f, 0x01012024, 0x01011222, 0x91174220, + 0x03a11050, 0x01016221, 0x90a70330, 0x01452340, 0x40C003f1, 0x405003f0, };

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[alsa-devel] [PATCH] Add model for Acer Aspire 5310
by Andy Shevchenko 25 Jan '08

25 Jan '08

Hi! Simplify usage of the Acer Aspire 5310 laptop with the ALC268 based codec sound card via add correct PCI SSID. Signed-off-by: Andy Shevchenko <andy(a)smile.org.ua> --- alsa-driver-hg20080125/alsa-kernel/pci/hda/patch_realtek.c.acer5310 2008-01-25 03:00:07.000000000 +0200 +++ alsa-driver-hg20080125/alsa-kernel/pci/hda/patch_realtek.c 2008-01-25 13:05:39.000000000 +0200 @@ -9925,6 +9925,7 @@ static const char *alc268_models[ALC268_ static struct snd_pci_quirk alc268_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0126, "Acer", ALC268_ACER), + SND_PCI_QUIRK(0x1025, 0x012e, "Acer Aspire 5310", ALC268_ACER), SND_PCI_QUIRK(0x1025, 0x0130, "Acer Extensa 5210", ALC268_ACER), SND_PCI_QUIRK(0x1025, 0x0136, "Acer Aspire 5315", ALC268_ACER), SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL), P.S. Sorry for separate patch. I've been noticed today about this one. -- With best regards, Andy Shevchenko. mailto: andy(a)smile.org.ua

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[alsa-devel] [Fwd: Re: Code relsease to alsa.]
by Tobin Davis 25 Jan '08

25 Jan '08

Forwarded from the good people at Linuxant. Tobin -------- Forwarded Message -------- From: Marc Boucher <marc(a)linuxant.com> To: Tobin Davis <tdavis(a)dsl-only.net> Cc: Linuxant support (Jonathan) <support(a)linuxant.com> Subject: Re: Code relsease to alsa. Date: Mon, 21 Jan 2008 10:33:03 -0500 Tobin, As promised here is our current tentative patch (relative to the current ALSA development tree in hg) for merging with ALSA. It is also available from: http://www.linuxant.com/alsa-hg-20080120.patch Naturally the changes are released under the GPL, so feel free to integrate them. Comments and improvements are welcome. Kind regards, Marc -- Marc Boucher Linuxant inc. http://www.linuxant.com On 19-Jan-08, at 6:40 PM, Marc Boucher wrote: > > Hi Tobin, > > We have isolated the modifications in patch form and are currently > working to merge them with the latest ALSA tree / submit them back > upstream. > > By the way, all changes and code under the modules/GPL/ directory > (including the hda subdirectory) of our modem driver are released > under the GPL, so you would be free to take these fixes if you'd like. > > However you would be also welcome to reviewing the patches that we > are about to submit. I will arrange to send you a copy asap (should > be next week). > > Kind regards, (and many thanks for your great contributions to ALSA) > > Marc > > PS: We could also try to help you getting any missing info on the > CX20561 or other things from Conexant... > > -- > Marc Boucher > Linuxant inc. > http://www.linuxant.com > > On 19-Jan-08, at 1:07 PM, Tobin Davis wrote: > >> Hello, >> >> I'm the developer that put the current audio support for the >> Conexant HD Audio codecs in alsa. I worked many hours on this, >> with testers around the world sending me feedback. Most of the >> work was done prior to Conexant offering me any documentation. I >> have since received documentation for both CX20549 (Venice), and >> CX20551 (Waikiki), but am waiting for the latest info on the new >> CX20561, which you apparently already have. >> >> I have also seen in your mailing list that you are providing >> several fixes to the audio portion of my code, but only making them >> available to users of your code, which requires a license agreement >> that I believe is incompatible with the GPL. At the least, it >> prohibits someone like me taking any of these fixes and putting >> them in the mainstream alsa, where more users would benefit from >> them. >> >> What I'm asking, is if you could at least provide the audio >> modifications back upstream to alsa? It would help out far more >> users (modem usage is on the decline - sorry), and you would have >> some benefit in that your name would be in the source as well. >> >> >> -- >> Tobin Davis <tdavis(a)dsl-only.net> > -- Tobin Davis no maintenance: Impossible to fix.

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[alsa-devel] HDMI Status
by Zsolt Barat 25 Jan '08

25 Jan '08

hi list, since i don't got any response on the users list, i try it here. i own a MSI board, K9AGM2, with north bridge S690G/690V and a as southbridge SB600. the hda-intel driver works perfectly for the most part. i just want to ask, how complete is the support for hdmi-audio. it seems that the hdmi-hardware is at least detected since alsamixer shows me as chip: Generic 1002 ATI RS690/780 HDMI. is it also supposed to put out some audio? i couldn't test it cause of lack of hdmi-sound capable hardware. best regards zsolt

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[alsa-devel] 1.0.16rc1 release
by Jaroslav Kysela 25 Jan '08

25 Jan '08

Hello all, 1.0.16rc1 release is available for download on http://www.alsa-project.org . I expect to release 1.0.16rc2 on next Tuesday. Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

5 13

[alsa-devel] Wall clock interrrupts
by James Courtier-Dutton 25 Jan '08

25 Jan '08

Hi, Do all sound cards hardware have wall clocks that one can generate interrupts from? I know all Creative and E-Mu cards do. I am looking to use the wall clock interrupt for triggering events in application space, but would like to know what other sound cards support it. James

2 1

[alsa-devel] Error when compiling from HG on Ubuntu 7.10
by Pavel Hofman 25 Jan '08

25 Jan '08

Hello, While trying to compile the latest alsa-kernel + alsa-drivers from HG using ./hgcompile, I am hitting the following error: CC [M] /home/pavel/projects/alsa/alsa-driver/i2c/other/tea575x-tuner.o In file included from /home/pavel/projects/alsa/alsa-driver/i2c/other/tea575x-tuner.c:23: /home/pavel/projects/alsa/alsa-driver/include/sound/driver.h:1:2: warning: #warning "This file is deprecated" /home/pavel/projects/alsa/alsa-driver/i2c/other/tea575x-tuner.c:60: error: redefinition of ‘video_get_drvdata’ include/media/v4l2-dev.h:383: error: previous definition of ‘video_get_drvdata’ was here /home/pavel/projects/alsa/alsa-driver/i2c/other/tea575x-tuner.c:65: error: redefinition of ‘video_set_drvdata’ include/media/v4l2-dev.h:388: error: previous definition of ‘video_set_drvdata’ was here /home/pavel/projects/alsa/alsa-driver/i2c/other/tea575x-tuner.c:101:5: warning: "LINUX_VERSION_CODE" is not defined /home/pavel/projects/alsa/alsa-driver/i2c/other/tea575x-tuner.c:101:26: warning: "KERNEL_VERSION" is not defined /home/pavel/projects/alsa/alsa-driver/i2c/other/tea575x-tuner.c:101:40: error: missing binary operator before token "(" /home/pavel/projects/alsa/alsa-driver/i2c/other/tea575x-tuner.c:112:5: warning: "LINUX_VERSION_CODE" is not defined /home/pavel/projects/alsa/alsa-driver/i2c/other/tea575x-tuner.c:112:27: warning: "KERNEL_VERSION" is not defined /home/pavel/projects/alsa/alsa-driver/i2c/other/tea575x-tuner.c:112:41: error: missing binary operator before token "(" /home/pavel/projects/alsa/alsa-driver/i2c/other/tea575x-tuner.c:120:5: warning: "LINUX_VERSION_CODE" is not defined /home/pavel/projects/alsa/alsa-driver/i2c/other/tea575x-tuner.c:120:27: warning: "KERNEL_VERSION" is not defined /home/pavel/projects/alsa/alsa-driver/i2c/other/tea575x-tuner.c:120:41: error: missing binary operator before token "(" /home/pavel/projects/alsa/alsa-driver/i2c/other/tea575x-tuner.c:124:5: warning: "LINUX_VERSION_CODE" is not defined /home/pavel/projects/alsa/alsa-driver/i2c/other/tea575x-tuner.c:124:27: warning: "KERNEL_VERSION" is not defined /home/pavel/projects/alsa/alsa-driver/i2c/other/tea575x-tuner.c:124:41: error: missing binary operator before token "(" /home/pavel/projects/alsa/alsa-driver/i2c/other/tea575x-tuner.c:204:5: warning: "LINUX_VERSION_CODE" is not defined /home/pavel/projects/alsa/alsa-driver/i2c/other/tea575x-tuner.c:204:27: warning: "KERNEL_VERSION" is not defined /home/pavel/projects/alsa/alsa-driver/i2c/other/tea575x-tuner.c:204:41: error: missing binary operator before token "(" /home/pavel/projects/alsa/alsa-driver/i2c/other/tea575x-tuner.c:227:5: warning: "LINUX_VERSION_CODE" is not defined /home/pavel/projects/alsa/alsa-driver/i2c/other/tea575x-tuner.c:227:26: warning: "KERNEL_VERSION" is not defined /home/pavel/projects/alsa/alsa-driver/i2c/other/tea575x-tuner.c:227:40: error: missing binary operator before token "(" /home/pavel/projects/alsa/alsa-driver/i2c/other/tea575x-tuner.c:230:5: warning: "LINUX_VERSION_CODE" is not defined /home/pavel/projects/alsa/alsa-driver/i2c/other/tea575x-tuner.c:230:27: warning: "KERNEL_VERSION" is not defined /home/pavel/projects/alsa/alsa-driver/i2c/other/tea575x-tuner.c:230:41: error: missing binary operator before token "(" /home/pavel/projects/alsa/alsa-driver/i2c/other/tea575x-tuner.c:234:5: warning: "LINUX_VERSION_CODE" is not defined /home/pavel/projects/alsa/alsa-driver/i2c/other/tea575x-tuner.c:234:26: warning: "KERNEL_VERSION" is not defined /home/pavel/projects/alsa/alsa-driver/i2c/other/tea575x-tuner.c:234:40: error: missing binary operator before token "(" /home/pavel/projects/alsa/alsa-driver/i2c/other/tea575x-tuner.c: In function ‘snd_tea575x_init’: /home/pavel/projects/alsa/alsa-driver/i2c/other/tea575x-tuner.c:243: warning: assignment from incompatible pointer type make[4]: *** [/home/pavel/projects/alsa/alsa-driver/i2c/other/tea575x-tuner.o] Error 1 The system runs on 2.6.22-14-rt kernel, corresponding package with headers installed correctly. Perhaps I am doing something wrong. Thanks a lot for any help. Regards, Pavel Hofman.

1 0

[alsa-devel] [PATCH] usb/caiaq: decrease period_bytes_min
by Daniel Mack 24 Jan '08

24 Jan '08

This patch increases the snd_pcm_hardware->period_bytes_min field in the caiaq/usb audio driver. The hardware can actually handle as few as 128 bytes, depending on the system. So it makes no sense to keep applications from actually using such values. Signed-off-by: Daniel Mack <daniel(a)caiaq.de>

2 2

[alsa-devel] Splitting up the HDA-INTEL modules
by Marco Schuster 24 Jan '08

24 Jan '08

Hi all, when I looked today through the files of the snd-hda-intel module, I was a bit shocked by this complex amount of code (patch_realtek.c for example). I would think about splitting up the code in a way each chipset gets its own .c file - which will make debugging, patching or editing a lot easier IMO, and for second people can then comment on just their specific chipset in config. What do you think about this? Marco PS: If there are no objections, I'll start working on realtek.c on weekend.

5 19

[alsa-devel] [PATCH] Add mute support for FM-only card with FM801 PCI to tuner bridge
by Andy Shevchenko 24 Jan '08

24 Jan '08

This is improvement of the early support of the FM-only cards where the fm801 chip represents the PCI to tuner bridge. The tuner initialization isn't included the mute on as well as mute support via V4L request. Proposed patch should fix this at least for 64-PCR model. Signed-off-by: Andy Shevchenko <andy(a)smile.org.ua> --- alsa-driver-hg20080124/alsa-kernel/i2c/other/tea575x-tuner.c.mute-fix 2008-01-09 03:00:14.000000000 +0200 +++ alsa-driver-hg20080124/alsa-kernel/i2c/other/tea575x-tuner.c 2008-01-24 19:15:48.000000000 +0200 @@ -158,6 +158,10 @@ static int snd_tea575x_ioctl(struct inod struct video_audio v; if(copy_from_user(&v, arg, sizeof(v))) return -EFAULT; + if (tea->ops->mute) + tea->ops->mute(tea, + (v.flags & + VIDEO_AUDIO_MUTE) ? 1 : 0); if(v.audio) return -EINVAL; return 0; @@ -205,6 +209,10 @@ void snd_tea575x_init(struct snd_tea575x tea->freq = 90500 * 16; /* 90.5Mhz default */ snd_tea575x_set_freq(tea); + + /* mute on init */ + if (tea->ops->mute) + tea->ops->mute(tea, 1); } void snd_tea575x_exit(struct snd_tea575x *tea) --- alsa-driver-hg20080124/alsa-kernel/pci/fm801.c.mute-fix 2008-01-09 03:00:15.000000000 +0200 +++ alsa-driver-hg20080124/alsa-kernel/pci/fm801.c 2008-01-24 19:15:48.000000000 +0200 @@ -978,21 +978,45 @@ static unsigned int snd_fm801_tea575x_64 return val; } +static void snd_fm801_tea575x_64pcr_mute(struct snd_tea575x *tea, + unsigned int mute) +{ + struct fm801 *chip = tea->private_data; + unsigned short reg; + + spin_lock_irq(&chip->reg_lock); + + reg = inw(FM801_REG(chip, GPIO_CTRL)); + if (mute) + /* 0xf800 (mute) */ + reg &= ~FM801_GPIO_GP(TEA_64PCR_WRITE_ENABLE); + else + /* 0xf802 (unmute) */ + reg |= FM801_GPIO_GP(TEA_64PCR_WRITE_ENABLE); + outw(reg, FM801_REG(chip, GPIO_CTRL)); + udelay(1); + + spin_unlock_irq(&chip->reg_lock); +} + static struct snd_tea575x_ops snd_fm801_tea_ops[3] = { { /* 1 = MediaForte 256-PCS */ .write = snd_fm801_tea575x_256pcs_write, .read = snd_fm801_tea575x_256pcs_read, + .mute = NULL, }, { /* 2 = MediaForte 256-PCPR */ .write = snd_fm801_tea575x_256pcpr_write, .read = snd_fm801_tea575x_256pcpr_read, + .mute = NULL, }, { /* 3 = MediaForte 64-PCR */ .write = snd_fm801_tea575x_64pcr_write, .read = snd_fm801_tea575x_64pcr_read, + .mute = snd_fm801_tea575x_64pcr_mute, } }; #endif --- alsa-driver-hg20080124/alsa-kernel/include/tea575x-tuner.h.mute-fix 2007-10-16 03:00:12.000000000 +0300 +++ alsa-driver-hg20080124/alsa-kernel/include/tea575x-tuner.h 2008-01-24 19:15:48.000000000 +0200 @@ -30,6 +30,7 @@ struct snd_tea575x; struct snd_tea575x_ops { void (*write)(struct snd_tea575x *tea, unsigned int val); unsigned int (*read)(struct snd_tea575x *tea); + void (*mute)(struct snd_tea575x *tea, unsigned int mute); }; struct snd_tea575x { -- With best regards, Andy Shevchenko. mailto: andy(a)smile.org.ua

2 1

[alsa-devel] [PATCH] pyalsa sequencer binding - 1/3
by Aldrin Martoq 24 Jan '08

24 Jan '08

Hi hackers, The patch below is the first of a series for implementing the pyalsa (python alsa) binding for the sequencer API of libasound2 that I hope be included in alsa. The following features are implemented: 1.- Opening a sequencer 1.1 support for SEQ_OPEN_{INPUT,OUTPUT,DUPLEX} 1.2 support for SEQ_NONBLOCK or blocking mode (SEQ_BLOCK) 1.3 support for alsa hw name and clientname 2.- Creating a simple port 2.1 support for port name 2.2 support for port capabilities bits SEQ_PORT_CAP_* 2.3 support for port types bits SEQ_PORT_TYPE_* 3.- Listing all connected client and their ports 3.1 client id's and names 3.2 port lists 3.3 read/write connections of ports 4.- Getting port info by clientid, portid 4.1 name of port 4.2 port capabilities 4.3 port type I *encourage to test and stress* the current implementation, which is a rewrite of what I wrote for the old pyalsaaudio project. The features presented are sufficient for creating a simple (or graphically!) aconnect python port, there is a sample 'seqtest1.py' using all the features. Planned features for next release: 1. alsaseq.SequencerError -- base exception for all exceptions (right now all method raises RuntimeError) 2. alsaseq.SequenceEvent -- for receiving and sending events 3. Fixing missing documentation (marked as TODO in alsaseq.c) Thank you for all your hard and good work. ----- Signed-off-by: Aldrin Martoq <amartoq(a)dcc.uchile.cl> diff -r 5080500e6cd3 pyalsa/alsaseq.c --- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/pyalsa/alsaseq.c Tue Jan 08 14:32:01 2008 -0300 @@ -0,0 +1,846 @@ +/* + * Python binding for the ALSA library - sequencer + * Copyright (c) 2007 by Aldrin Martoq <amartoq(a)dcc.uchile.cl> + * + * This library is free software; you can redistribute it and/or modify + * it under the terms of the GNU Lesser General Public License as + * published by the Free Software Foundation; either version 2.1 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 Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with this library; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include "Python.h" +#include <alsa/asoundlib.h> +#include <stdio.h> + +///////////////////////////////////////////////////////////////////// +// some helper #define go here... +///////////////////////////////////////////////////////////////////// + +/* temporal debug (will be deleted in last patch, promise!) */ +#define ddebug(x, args...) fprintf(stderr, x "\n",##args); + +/* check passed object is integer and can't be deleted */ +#define SETCHECKPYINT(attr, val) \ + if (val == NULL) { \ + PyErr_SetString(PyExc_AttributeError, "attribute " attr " can't be deleted!"); \ + return -1; \ + } \ + if (!PyInt_Check(val)) { \ + PyErr_SetString(PyExc_TypeError, "int value expected for " attr); \ + return -1; \ + } + +/* check if alsaseq.SEQ* constants is being used */ +/* + #define CHECKCONSTANT(attr, obj, type) \ + if (!PyObject_TypeCheck(obj, type)) { \ + PyErr_Warn(PyExc_RuntimeWarning, "please use only constants from alsaseq.SEQ*"); \ + } + */ +#define CHECKCONSTANT(attr, obj, type) + +/* check passed object is string and can't be deleted */ +#define SETCHECKPYSTR(attr, val) \ + if (val == NULL) { \ + PyErr_SetString(PyExc_AttributeError, "attribute " attr " can't be deleted!"); \ + return -1; \ + } \ + if (!PyString_Check(val)) { \ + PyErr_SetString(PyExc_TypeError, "string value expected for " attr); \ + return -1; \ + } + +/* free only if pointer is not NULL */ +#define FREECHECKED(name, pointer) \ + if (pointer != NULL) { \ + free(pointer); \ + pointer = NULL; \ + } + +/* add constant as integer to module */ +#define CONST(module, name, value) \ + if (PyModule_AddIntConstant(module, name, (long) value) < 0) { \ + return; \ + } + +/* define constant dict by type */ +#define TDICT(subtype, name) \ + PyObject *_dictPYALSASEQ_CONST_##subtype = tmp = PyDict_New(); \ + if (PyModule_AddObject(module, name, tmp) < 0) { \ + return; \ + } \ + +/* create typed constant and add to module */ +#define TCONST(module, subtype, name, value) \ + tmp = Constant_create(name, value, PYALSASEQ_CONST_##subtype); \ + if (PyModule_AddObject(module, name, tmp) < 0) { \ + return; \ + } \ + PyDict_SetItem(_dictPYALSASEQ_CONST_##subtype, PyInt_FromLong(value), tmp); + +/* num protocol support for binary Constant operations */ +#define NUMPROTOCOL2(name, oper) \ +static PyObject *Constant_##name (PyObject *v, PyObject *w) { \ + int type = 0; \ + long val = 0; \ + /* both have to be a int */ \ + if (!PyInt_Check(v) || !PyInt_Check(w)) { \ + Py_INCREF(Py_NotImplemented); \ + return Py_NotImplemented; \ + } \ + val = PyInt_AS_LONG(v) oper PyInt_AS_LONG(w); \ + /* always asume left will be the type */ \ + if (PyObject_TypeCheck(v, &ConstantType)) { \ + type = ((ConstantObject *) v)->type; \ + } else if (PyObject_TypeCheck(w, &ConstantType)) { \ + type = ((ConstantObject *) w)->type; \ + } \ + PyObject *self = Constant_create(#oper, val, type); \ + return self; \ +} +/* num protocol support for unary Constant operations */ +#define NUMPROTOCOL1(name, oper) \ +static PyObject *Constant_##name (PyObject *v) { \ + int type = 0; \ + long val = 0; \ + if (!PyInt_Check(v)) { \ + Py_INCREF(Py_NotImplemented); \ + return Py_NotImplemented; \ + } \ + val = oper PyInt_AS_LONG(v); \ + if (PyObject_TypeCheck(v, &ConstantType)) { \ + type = ((ConstantObject *) v)->type; \ + } \ + PyObject *self = Constant_create(#oper, val, type); \ + return self; \ +} + +//#define ddebug(x) {} + +// alsaseq.Constant type definitions +enum { + PYALSASEQ_CONST_STREAMS, + PYALSASEQ_CONST_MODE, + PYALSASEQ_CONST_PORT_CAP, + PYALSASEQ_CONST_PORT_TYPE +}; + +///////////////////////////////////////////////////////////////////// +// alsaseq.Constant implementation +///////////////////////////////////////////////////////////////////// + +/** alsaseq.Constant __doc__ */ +// TODO: write doc +PyDoc_STRVAR(Constant__doc__, + "Constant type: TODO: write doc" +); + +/** alsaseq.Constant object structure type */ +typedef struct { + PyObject_HEAD + ; + + /* value of constant */ + long value; + /* name of constant */ + const char *name; + /* type of constant */ + int type; +} ConstantObject; + +/** alsaseq.Constant type (initialized later...) */ +static PyTypeObject ConstantType; + +/** alsaseq.Constant internal create */ +static PyObject *Constant_create(const char *name, long value, int type) { + ConstantObject *self= PyObject_New(ConstantObject, &ConstantType); + if (self == NULL) { + return NULL; + } + + self->value = value; + self->name = name; + self->type = type; + + return (PyObject *)self; +} + +/** alsaseq.Constant tp_repr */ +static PyObject *Constant_repr(ConstantObject *self) { + return PyString_FromFormat("<alsaseq.Constant %s (%ld)", self->name, + self->value); +} + +/** alsaseq.Constant tp_str */ +static PyObject *Constant_str(ConstantObject *self) { + return PyString_FromFormat("%s", self->name); +} + +/** alsaseq.Constant Number protocol support (note: not all operations) */ +NUMPROTOCOL2(Add, +) +NUMPROTOCOL2(Subtract, -) +NUMPROTOCOL2(Xor, ^) +NUMPROTOCOL2(Or, |) +NUMPROTOCOL2(And, &) +NUMPROTOCOL1(Invert, ~) + +static PyNumberMethods Constant_as_number = { nb_add: (binaryfunc)Constant_Add, nb_subtract: (binaryfunc)Constant_Subtract, nb_xor: (binaryfunc)Constant_Xor, nb_or: (binaryfunc)Constant_Or, nb_and: (binaryfunc)Constant_And, nb_invert: (unaryfunc)Constant_Invert }; + +/** alsaseq.Constant type */ +static PyTypeObject ConstantType = { PyObject_HEAD_INIT(NULL) +tp_name: "alsaseq.Constant", +tp_base: &PyInt_Type, +tp_basicsize: sizeof(ConstantObject), +tp_flags: Py_TPFLAGS_HAVE_GETCHARBUFFER | Py_TPFLAGS_HAVE_CLASS | Py_TPFLAGS_CHECKTYPES, +tp_doc: Constant__doc__, +//tp_dealloc: (destructor)PyObject_Del, +tp_as_number: &Constant_as_number, +tp_free: PyObject_Del, tp_str: (reprfunc)Constant_str, +tp_repr: (reprfunc)Constant_repr, }; + +///////////////////////////////////////////////////////////////////// +// alsaseq.Sequencer implementation +///////////////////////////////////////////////////////////////////// + +/** alsaseq.Sequencer __doc__ */ +// TODO: write doc +PyDoc_STRVAR(Sequencer__doc__, + "Sequencer class: TODO: write doc" +); + +/** alsaseq.Sequencer object structure type */ +typedef struct { + PyObject_HEAD + ; + + /* name of device */ + char *name; + /* name of client */ + char *clientname; + /* client id */ + int client_id; + /* streams */ + int streams; + /* mode */ + int mode; + + /* alsa handler */ + snd_seq_t *handle; + /* max fd's */ + int receive_max; + /* receive poll fd's */ + struct pollfd *receive_fds; + /* max events for returning in receive_events() */ + int receive_max_events; +} SequencerObject; + +/** alsaseq.Sequencer type (initialized later...) */ +static PyTypeObject SequencerType; + +/** alsaseq.Sequencer tp_init */ +static int Sequencer_init(SequencerObject *self, PyObject *args, PyObject *kwds) { + int ret; + /* defaults */ + char *name = "default"; + char *clientname = "pyalsa"; + self->streams = SND_SEQ_OPEN_DUPLEX; + self->mode = SND_SEQ_NONBLOCK; + int maxreceiveevents = 10; + + char *kwlist[] = { "name", "clientname", "streams", "mode", + "maxreceiveevents", NULL }; + + if (!PyArg_ParseTupleAndKeywords(args, kwds, "|ssiii", kwlist, &name, + &clientname, &self->streams, &self->mode, + &maxreceiveevents)) { + return -1; + } + + self->name = strdup(name); + self->clientname = strdup(clientname); + self->receive_fds = NULL; + self->receive_max = 0; + self->receive_max_events + = (self->mode == SND_SEQ_NONBLOCK ? maxreceiveevents + : 0); + + ret = snd_seq_open(&(self->handle), self->name, self->streams, + self->mode); + if (ret < 0) { + // TODO: raise exception + return -1; + } + ret = snd_seq_set_client_name(self->handle, self->clientname); + if (ret < 0) { + // TODO: raise exception + return -1; + } + self->client_id = snd_seq_client_id(self->handle); + + return 0; +} + +/** alsaseq.Sequencer tp_dealloc */ +static void Sequencer_dealloc(SequencerObject *self) { + FREECHECKED("name", self->name) + FREECHECKED("clientname", self->clientname) + FREECHECKED("receive_fds", self->receive_fds) + + if (self->handle) { + snd_seq_close(self->handle); + self->handle = NULL; + } + + self->ob_type->tp_free(self); +} + +/** alsaseq.Sequencer name attribute: __doc__ */ +PyDoc_STRVAR(Sequencer_name__doc__ , + "name -- (string) the hardware name of this alsaseq.Sequencer (default: \"default\"). " + "Note: read-only attribute."); + +/** + * alsaseq.Sequencer name attribute: tp_get getter() + * + * @param self -- the SeqEventObject + * @returns the alsa hardware name of this Sequencer. + */ +static PyObject *Sequencer_get_name(SequencerObject *self) { + return PyString_FromString(self->name); +} + +/** alsaseq.Sequencer clientname attribute: __doc__ */ +PyDoc_STRVAR(Sequencer_clientname__doc__ , + "clientname -- (string) the client name of this alsaseq.Sequencer (default: \"pyalsa\")."); + +/** + * alsaseq.Sequencer clientname attribute: tp_get getter() + * + * @param self -- the SeqEventObject + * @returns the client name of this Sequencer. + */ +static PyObject *Sequencer_get_clientname(SequencerObject *self) { + return PyString_FromString(self->clientname); +} + +/** + * alsaseq.Sequencer clientname attribute: tp_get setter() + * + * @param self -- the SeqEventObject + * @param val -- the new value to set (must be a string) + * @returns 0 on success, -1 on error + * @raises TypeError -- if the val parameter is not a String + */ +static int Sequencer_set_clientname(SequencerObject *self, PyObject *val) { + int ret; + char *buff; + + SETCHECKPYSTR("clientname", val) + + buff = PyString_AsString(val); + + ret = snd_seq_set_client_name(self->handle, buff); + if (ret == 0) { + FREECHECKED("clientname", self->clientname) + self->clientname = strdup(buff); + } else { + // TODO: raise exception + return -1; + } + return 0; +} + +/** alsaseq.Sequencer streams attribute: __doc__ */ +PyDoc_STRVAR(Sequencer_streams__doc__ , + "streams -- (int) the streams of this alsaseq.Sequencer (default: alsaseq.SEQ_OPEN_DUPLEX). " + "Posible values: alsaseq.SEQ_OPEN_OUTPUT, alsaseq.SEQ_OPEN_INPUT, alsaseq.SEQ_OPEN_DUPLEX. " + "Note: read-only attribute."); + +/** + * alsaseq.Sequencer streams attribute: tp_get getter() + * + * @param self -- the SeqEventObject + * @returns the stremas of this Sequencer. + */ +static PyObject *Sequencer_get_streams(SequencerObject *self) { + return PyInt_FromLong(self->streams); +} + +/** alsaseq.Sequencer mode attribute: __doc__ */ +PyDoc_STRVAR(Sequencer_mode__doc__ , + "mode -- (int) the blocking mode of this alsaseq.Sequencer (default: alsaseq.SEQ_NONBLOCK). " + "Posible values: alsaseq.SEQ_BLOCK, alsaseq.SEQ_NONBLOCK. "); + +/** + * alsaseq.Sequencer mode attribute: tp_get getter() + * + * @param self -- the SeqEventObject + * @returns the blocking mode of this Sequencer. + */ +static PyObject *Sequencer_get_mode(SequencerObject *self) { + return PyInt_FromLong(self->mode); +} + +/** + * alsaseq.Sequencer mode attribute: tp_get setter() + * + * @param self -- the SeqEventObject + * @param val -- the new value to set (must be a integer) + * @returns 0 on success, -1 on error + * @raises TypeError -- if the val parameter is not a integer + * @raises ValueError -- if the val parameter is not 0 (SEQ_BLOCK) or 1 (SEQ_NONBLOCK). + */ +static int Sequencer_set_mode(SequencerObject *self, PyObject *val) { + int ret, mode; + + SETCHECKPYINT("mode", val) + +CHECKCONSTANT("mode", val, &ConstantType) + + mode = (int) PyInt_AsLong(val); + + if (mode != 0 && mode != SND_SEQ_NONBLOCK) { + PyErr_SetString(PyExc_ValueError, "Invalid value for mode."); + return -1; + } + + ret = snd_seq_nonblock(self->handle, mode); + if (ret == 0) { + self->mode = mode; + } else { + // TODO: raise exception + return -1; + } + return 0; +} + +/** alsaseq.Sequencer client_id attribute: __doc__ */ +PyDoc_STRVAR(Sequencer_client_id__doc__ , + "client_id -- (int) the client id of this alsaseq.Sequencer. " + "Note: read-only attribute."); + +/** + * alsaseq.Sequencer client_id attribute: tp_get getter() + * + * @param self -- the SeqEventObject + * @returns the client id of this Sequencer. + */ +static PyObject *Sequencer_get_client_id(SequencerObject *self) { + return PyInt_FromLong(self->client_id); +} + +/** alsaseq.Sequencer tp_getset list*/ +static PyGetSetDef Sequencer_getset[] = { { "name", (getter)Sequencer_get_name, + NULL, Sequencer_name__doc__, NULL }, { "clientname", + (getter)Sequencer_get_clientname, + (setter)Sequencer_set_clientname, Sequencer_clientname__doc__, + NULL }, { "streams", (getter) Sequencer_get_streams, NULL, + Sequencer_streams__doc__, NULL }, { "mode", + (getter) Sequencer_get_mode, (setter) Sequencer_set_mode, + Sequencer_mode__doc__, NULL }, { "client_id", + (getter) Sequencer_get_client_id, NULL, + Sequencer_client_id__doc__, NULL }, { NULL } }; + +/** alsaseq.Sequencer tp_repr */ +static PyObject *Sequencer_repr(SequencerObject *self) { + return PyString_FromFormat( + "<alsaseq.Sequencer client_id=%d name=%s clientname=%s streams=%d mode=%d at 0x%p>", + self->client_id, self->name, self->clientname, + self->streams, self->mode, self); +} + +/** alsaseq.Sequencer create_simple_port(): __doc__ */ +PyDoc_STRVAR(Sequencer_create_simple_port__doc__, + "create_simple_port(name, caps, type) -- Creates a port for receiving or sending events.\n\n" + "Parameters:\n" + " name -- (string) name of the port\n" + " caps -- (int) capabilites of the port (use bitwise alsaseq.SEQ_PORT_CAP_* constants)\n" + " type -- (int) type of port (use one of the alsaseq.SEQ_PORT_TYPE_* constants)" + "Returns:\n" + " (int) the port id.\n" + "Raises:\n" + " TypeError: an invalid type was used in a parameter\n" + " ValueError: an invalid value was used in a parameter\n" +); + +/** + * alsaseq.Sequencer create_simple_port() + */ +static PyObject *Sequencer_create_simple_port(SequencerObject *self, + PyObject *args) { + char *name; + unsigned int caps; + unsigned int type; + int port; + + if (!PyArg_ParseTuple(args, "sII", &name, &caps, &type)) { + return NULL; + } + port = snd_seq_create_simple_port(self->handle, name, caps, type); + if (port < 0) { + // TODO: Throw exception + return NULL; + } + + return PyInt_FromLong(port); +} + +/** alsaseq.Sequencer get_client_ports(): __doc__ */ +PyDoc_STRVAR(Sequencer_get_client_ports__doc__, + "get_client_ports() -- List current clients and their ports connected to alsa.\n\n" + "Returns:\n" + " (list) a list of tuples: client_name, client_id, port_list.\n" + " client_name -- the client's name\n" + " client_id -- the client's id\n" + " port_list -- a list of tuples: port_name, port_id, connection_list\n" + " port_name -- the name of the port\n" + " port_id -- the port id\n" + " connection_list -- a list of tuples: read_conn, write_conn\n" + " read_conn -- a list of tuples with the client_id, port_id this port is connected to (sends events)\n" + " write_conn -- a list of tuples with the client_id, port_id this port is connected from (receives events)\n" +); + +static PyObject *_query_connections_list(snd_seq_t *handle, + snd_seq_query_subscribe_t *query, + int type) { + + PyObject *list = PyList_New(0); + int index = 0; + snd_seq_query_subscribe_set_type(query, type); + snd_seq_query_subscribe_set_index(query, index); + while (snd_seq_query_port_subscribers(handle, query) >= 0) { + const snd_seq_addr_t *addr = + snd_seq_query_subscribe_get_addr(query); + + /* create tuple for clientid, portid */ + PyObject *tuple= PyTuple_New(2); + PyTuple_SetItem(tuple, 0, PyInt_FromLong(addr->client)); + PyTuple_SetItem(tuple, 1, PyInt_FromLong(addr->port)); + + PyList_Append(list, tuple); + snd_seq_query_subscribe_set_index(query, ++index); + } + return list; +} + +static PyObject *_query_connections(snd_seq_t *handle, + const snd_seq_addr_t *addr) { + snd_seq_query_subscribe_t *query; + snd_seq_query_subscribe_alloca(&query); + snd_seq_query_subscribe_set_root(query, addr); + + /* create tuple for read,write lists */ + PyObject *tuple = PyTuple_New(2); + PyObject *readlist = _query_connections_list(handle, query, + SND_SEQ_QUERY_SUBS_READ); + PyObject *writelist = _query_connections_list(handle, query, + SND_SEQ_QUERY_SUBS_WRITE); + PyTuple_SetItem(tuple, 0, readlist); + PyTuple_SetItem(tuple, 1, writelist); + + return tuple; +} + +/** + * alsaseq.Sequencer get_client_ports() + */ +static PyObject *Sequencer_get_client_ports(SequencerObject *self, + PyObject *args) { + snd_seq_client_info_t *cinfo; + snd_seq_port_info_t *pinfo; + PyObject *list = PyList_New(0); + + if (list == NULL) { + return NULL; + } + + snd_seq_client_info_alloca(&cinfo); + snd_seq_port_info_alloca(&pinfo); + snd_seq_client_info_set_client(cinfo, -1); + while (snd_seq_query_next_client(self->handle, cinfo) >= 0) { + /* reset query info */ + snd_seq_port_info_set_client(pinfo, + snd_seq_client_info_get_client(cinfo)); + snd_seq_port_info_set_port(pinfo, -1); + + /* create tuple for client info */ + PyObject *tuple= PyTuple_New(3); + PyObject *portlist = PyList_New(0); + + PyTuple_SetItem(tuple, 0, PyString_FromFormat("%s", + snd_seq_client_info_get_name(cinfo))); + PyTuple_SetItem( + tuple, + 1, + PyInt_FromLong(snd_seq_client_info_get_client(cinfo))); + + while (snd_seq_query_next_port(self->handle, pinfo) >= 0) { + /* create tuple for port info */ + PyObject *porttuple = PyTuple_New(3); + PyTuple_SetItem(porttuple, 0, PyString_FromFormat("%s", + snd_seq_port_info_get_name(pinfo))); + PyTuple_SetItem( + porttuple, + 1, + PyInt_FromLong(snd_seq_port_info_get_port(pinfo))); + /* create tuple for read,write connections */ + PyObject *conntuple= _query_connections(self->handle, + snd_seq_port_info_get_addr(pinfo)); + PyTuple_SetItem(porttuple, 2, conntuple); + PyList_Append(portlist, porttuple); + } + PyTuple_SetItem(tuple, 2, portlist); + + /* append list of port tuples */ + PyList_Append(list, tuple); + } + + return list; +} + +/** alsaseq.Sequencer get_port_info(): __doc__ */ +PyDoc_STRVAR(Sequencer_get_port_info__doc__, + "get_port_info(port_id, client_id = self.client_id) -- Retrieve info about an existing port of a client.\n\n" + "Parameters:\n" + " port_id -- (int) the port id\n" + " client_id -- (int) the client id (defaults to: self.client_id)\n" + "Returns:\n" + " (dict) a dictionary with the following values:\n" + " name -- (string) the port name\n" + " type -- (int) the port type bit flags\n" + " capability -- (int) the port capability bit flags\n" +); + +/** + * alsaseq.Sequencer get_port_info() + */ +static PyObject *Sequencer_get_port_info(SequencerObject *self, PyObject *args, + PyObject *kwds) { + snd_seq_port_info_t *pinfo; + int port_id; + int client_id = self->client_id; + const char *tmpchar; + long tmplong; + int ret; + + char *kwlist[] = { "port_id", "client_id", NULL }; + + if (!PyArg_ParseTupleAndKeywords(args, kwds, "i|i", kwlist, &port_id, + &client_id)) { + return NULL; + } + + PyObject *dict = PyDict_New(); + if (dict == NULL) { + return NULL; + } + + snd_seq_port_info_alloca(&pinfo); + /* query info */ + ret + = snd_seq_get_any_port_info(self->handle, client_id, + port_id, pinfo); + if (ret < 0) { + ddebug("FIXME: Throw exception: %s", snd_strerror(ret)); + // TODO: Throw exception + return NULL; + } + tmpchar = snd_seq_port_info_get_name(pinfo); + PyDict_SetItem(dict, PyString_FromString("name"), + PyString_FromString(tmpchar)); + + tmplong = snd_seq_port_info_get_capability(pinfo); + PyDict_SetItem(dict, PyString_FromString("capability"), + PyInt_FromLong(tmplong)); + tmplong = snd_seq_port_info_get_type(pinfo); + PyDict_SetItem(dict, PyString_FromString("type"), + PyInt_FromLong(tmplong)); + + return dict; +} + +/** alsaseq.Sequencer connect_ports(): __doc__ */ +PyDoc_STRVAR(Sequencer_connect_ports__doc__, + "connect_ports(srcaddr, dstaddr) -- Connect the ports specified by srcaddr and dstaddr.\n\n" + "Parameters:\n" + " srcaddr -- (tuple) the client_id, port_id tuple for the source port (the port sending events)\n" + " dstaddr -- (tuple) the client_id, port_id tuple for the destination port (the port receiveng events)\n" +); + +/** + * alsaseq.Sequencer connect_ports() + */ +static PyObject *Sequencer_connect_ports(SequencerObject *self, PyObject *args) { + snd_seq_addr_t sender, dest; + snd_seq_port_subscribe_t *sinfo; + int ret; + + if (!PyArg_ParseTuple(args, "(BB)(BB)", + &(sender.client), &(sender.port), &(dest.client), &(dest.port))) { + return NULL; + } + + snd_seq_port_subscribe_alloca(&sinfo); + snd_seq_port_subscribe_set_sender(sinfo, &sender); + snd_seq_port_subscribe_set_dest(sinfo, &dest); + + ret = snd_seq_subscribe_port(self->handle, sinfo); + if (ret < 0) { + ddebug("FIXME: Raise Exception: %s", snd_strerror(ret)); + return NULL; + } + + Py_INCREF(Py_None); + + return Py_None; +} + +/** alsaseq.Sequencer disconnect_ports(): __doc__ */ +PyDoc_STRVAR(Sequencer_disconnect_ports__doc__, + "disconnect_ports(srcaddr, dstaddr) -- Disconnect the ports specified by srcaddr and dstaddr.\n\n" + "Parameters:\n" + " srcaddr -- (tuple) the client_id, port_id tuple for the source port (the port sending events)\n" + " dstaddr -- (tuple) the client_id, port_id tuple for the destination port (the port receiveng events)\n" +); + +/** + * alsaseq.Sequencer disconnect_ports() + */ +static PyObject *Sequencer_disconnect_ports(SequencerObject *self, PyObject *args) { + snd_seq_addr_t sender, dest; + snd_seq_port_subscribe_t *sinfo; + int ret; + + if (!PyArg_ParseTuple(args, "(BB)(BB)", + &(sender.client), &(sender.port), &(dest.client), &(dest.port))) { + return NULL; + } + + snd_seq_port_subscribe_alloca(&sinfo); + snd_seq_port_subscribe_set_sender(sinfo, &sender); + snd_seq_port_subscribe_set_dest(sinfo, &dest); + + ret = snd_seq_unsubscribe_port(self->handle, sinfo); + if (ret < 0) { + ddebug("FIXME: Raise Exception: %s", snd_strerror(ret)); + return NULL; + } + + Py_INCREF(Py_None); + + return Py_None; +} + +/** alsaseq.Sequencer tp_methods */ +static PyMethodDef Sequencer_methods[] = { { "create_simple_port", + (PyCFunction) Sequencer_create_simple_port, METH_VARARGS, + Sequencer_create_simple_port__doc__ }, { "get_client_ports", + (PyCFunction) Sequencer_get_client_ports, METH_VARARGS, + Sequencer_get_client_ports__doc__ }, { "get_port_info", + (PyCFunction) Sequencer_get_port_info, + METH_VARARGS, Sequencer_get_port_info__doc__ }, + {"connect_ports", (PyCFunction) Sequencer_connect_ports, METH_VARARGS, Sequencer_connect_ports__doc__ }, + {"disconnect_ports", (PyCFunction) Sequencer_disconnect_ports, METH_VARARGS, Sequencer_disconnect_ports__doc__ }, + { NULL } }; + +/** alsaseq.Sequencer type */ +static PyTypeObject SequencerType = { PyObject_HEAD_INIT(NULL) +tp_name: "alsaseq.Sequencer", +tp_basicsize: sizeof(SequencerObject), +tp_dealloc: (destructor)Sequencer_dealloc, +tp_flags: Py_TPFLAGS_DEFAULT | Py_TPFLAGS_BASETYPE, +tp_doc: Sequencer__doc__, +tp_init: (initproc)Sequencer_init, +tp_new: PyType_GenericNew, +tp_alloc: PyType_GenericAlloc, +tp_free: PyObject_Del, tp_repr: (reprfunc)Sequencer_repr, +tp_methods: Sequencer_methods, tp_getset: Sequencer_getset +}; + +///////////////////////////////////////////////////////////////////// +// alsaseq module implementation +///////////////////////////////////////////////////////////////////// + +/** alsaseq module: __doc__ */ +// TODO: write documentation +PyDoc_STRVAR(alsaseq__doc__, "alsaseq: TODO: write documentation"); + +/** alsaseq module methods */ +static PyMethodDef alsaseq_methods[] = { { NULL }, +}; + +PyMODINIT_FUNC +initalsaseq(void) { + PyObject *module; + PyObject *tmp; + + if (PyType_Ready(&ConstantType) < 0) { + return; + } + + if (PyType_Ready(&SequencerType) < 0) { + return; + } + + module = Py_InitModule3("alsaseq", alsaseq_methods, alsaseq__doc__); + + Py_INCREF(&SequencerType); + PyModule_AddObject(module, "Sequencer", (PyObject *) &SequencerType); + Py_INCREF(&ConstantType); + PyModule_AddObject(module, "Constant", (PyObject *) &ConstantType); + + // constants definition + + TDICT(STREAMS, "_const_streams"); + TDICT(MODE, "_const_mode"); + TDICT(PORT_CAP, "_const_port_cap"); + TDICT(PORT_TYPE, "_const_port_type"); + + /* streams */ + + TCONST(module, STREAMS, "SEQ_OPEN_OUTPUT", SND_SEQ_OPEN_OUTPUT); + TCONST(module, STREAMS, "SEQ_OPEN_INPUT", SND_SEQ_OPEN_INPUT); + TCONST(module, STREAMS, "SEQ_OPEN_DUPLEX", SND_SEQ_OPEN_DUPLEX); + + /* blocking mode */ + TCONST(module, MODE, "SEQ_BLOCK", 0); + TCONST(module, MODE, "SEQ_NONBLOCK", SND_SEQ_NONBLOCK); + + /* port cap */ + TCONST(module, PORT_CAP, "SEQ_PORT_CAP_WRITE", SND_SEQ_PORT_CAP_WRITE); + TCONST(module, PORT_CAP, "SEQ_PORT_CAP_SYNC_WRITE", SND_SEQ_PORT_CAP_SYNC_WRITE); + TCONST(module, PORT_CAP, "SEQ_PORT_CAP_SYNC_READ", SND_SEQ_PORT_CAP_SYNC_READ); + TCONST(module, PORT_CAP, "SEQ_PORT_CAP_SUBS_WRITE", SND_SEQ_PORT_CAP_SUBS_WRITE); + TCONST(module, PORT_CAP, "SEQ_PORT_CAP_SUBS_READ", SND_SEQ_PORT_CAP_SUBS_READ); + TCONST(module, PORT_CAP, "SEQ_PORT_CAP_READ", SND_SEQ_PORT_CAP_READ); + TCONST(module, PORT_CAP, "SEQ_PORT_CAP_NO_EXPORT", SND_SEQ_PORT_CAP_NO_EXPORT); + TCONST(module, PORT_CAP, "SEQ_PORT_CAP_DUPLEX", SND_SEQ_PORT_CAP_DUPLEX); + + /* port type */ + TCONST(module, PORT_TYPE, "SEQ_PORT_TYPE_SYNTHESIZER", SND_SEQ_PORT_TYPE_SYNTHESIZER); + TCONST(module, PORT_TYPE, "SEQ_PORT_TYPE_SYNTH", SND_SEQ_PORT_TYPE_SYNTH); + TCONST(module, PORT_TYPE, "SEQ_PORT_TYPE_SPECIFIC", SND_SEQ_PORT_TYPE_SPECIFIC); + TCONST(module, PORT_TYPE, "SEQ_PORT_TYPE_SOFTWARE", SND_SEQ_PORT_TYPE_SOFTWARE); + TCONST(module, PORT_TYPE, "SEQ_PORT_TYPE_SAMPLE", SND_SEQ_PORT_TYPE_SAMPLE); + TCONST(module, PORT_TYPE, "SEQ_PORT_TYPE_PORT", SND_SEQ_PORT_TYPE_PORT); + TCONST(module, PORT_TYPE, "SEQ_PORT_TYPE_MIDI_XG", SND_SEQ_PORT_TYPE_MIDI_XG); + TCONST(module, PORT_TYPE, "SEQ_PORT_TYPE_MIDI_MT32", SND_SEQ_PORT_TYPE_MIDI_MT32); + TCONST(module, PORT_TYPE, "SEQ_PORT_TYPE_MIDI_GS", SND_SEQ_PORT_TYPE_MIDI_GS); + TCONST(module, PORT_TYPE, "SEQ_PORT_TYPE_MIDI_GM2", SND_SEQ_PORT_TYPE_MIDI_GM2); + TCONST(module, PORT_TYPE, "SEQ_PORT_TYPE_MIDI_GM", SND_SEQ_PORT_TYPE_MIDI_GM); + TCONST(module, PORT_TYPE, "SEQ_PORT_TYPE_MIDI_GENERIC", SND_SEQ_PORT_TYPE_MIDI_GENERIC); + TCONST(module, PORT_TYPE, "SEQ_PORT_TYPE_HARDWARE", SND_SEQ_PORT_TYPE_HARDWARE); + TCONST(module, PORT_TYPE, "SEQ_PORT_TYPE_DIRECT_SAMPLE", SND_SEQ_PORT_TYPE_DIRECT_SAMPLE); + TCONST(module, PORT_TYPE, "SEQ_PORT_TYPE_APPLICATION", SND_SEQ_PORT_TYPE_APPLICATION); + +} diff -r 5080500e6cd3 setup.py --- a/setup.py Sat Nov 03 11:25:08 2007 +0100 +++ b/setup.py Tue Jan 08 14:32:01 2008 -0300 @@ -45,6 +45,11 @@ setup( include_dirs=[], library_dirs=[], libraries=['asound']), + Extension('pyalsa.alsaseq', + ['pyalsa/alsaseq.c'], + include_dirs=[], + library_dirs=[], + libraries=['asound']), ], packages=['pyalsa'], scripts=[] @@ -52,7 +57,7 @@ setup( uname = os.uname() a = 'build/lib.%s-%s-%s' % (uname[0].lower(), uname[4], sys.version[:3]) -for f in ['alsacard.so', 'alsacontrol.so', 'alsahcontrol.so', 'alsamixer.so']: +for f in ['alsacard.so', 'alsacontrol.so', 'alsahcontrol.so', 'alsamixer.so', 'alsaseq.so']: if not os.path.exists('pyalsa/%s' % f): a = '../build/lib.%s-%s-%s/pyalsa/%s' % (uname[0].lower(), uname[4], sys.version[:3], f) diff -r 5080500e6cd3 test/alsamemdebug.py --- a/test/alsamemdebug.py Sat Nov 03 11:25:08 2007 +0100 +++ b/test/alsamemdebug.py Tue Jan 08 14:32:01 2008 -0300 @@ -6,6 +6,7 @@ import gc def debuginit(): gc.set_debug(gc.DEBUG_LEAK) + print "LEAK: init" def debug(what): from sys import getrefcount @@ -13,7 +14,8 @@ def debug(what): for o in what: if getrefcount(o) - 3 != 1 or \ len(get_referrers(o)) - 2 != 1: - print 'LEAK:', hex(id(o)), type(o), getrefcount(o)-3, len(get_referrers(o))-2 + print 'LEAK:', o, hex(id(o)), type(o), getrefcount(o)-3, len(get_referrers(o))-2 def debugdone(): + print "LEAK: done" None diff -r 5080500e6cd3 test/seqtest1.py --- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/test/seqtest1.py Tue Jan 08 14:32:01 2008 -0300 @@ -0,0 +1,135 @@ +#! /usr/bin/python +# Sample code for pyalsa Sequencer binding +# by Aldrin Martoq <amartoq(a)dcc.uchile.cl> +# version 2008010801 (UTC: 1199812236 secs) +# +# This code is in the public domain, +# use it as base for creating your pyalsa' +# alsaseq application. + +import sys +sys.path.insert(0, '../pyalsa') +del sys +from alsamemdebug import debuginit, debug, debugdone +import alsaseq + + +def dump_portinfo(dict): + def dl(dict, key): + if dict.has_key(key): + return "'" + str(dict[key]) + "'" + return "N/A" + def da(dict, key, search): + tmp = None + if dict.has_key(key): + for k in search: + if k & dict[key]: + if tmp == None: + tmp = str(search[k]) + else: + tmp += " " + str(search[k]) + if tmp == None: + tmp = "N/A" + return tmp + + return "name=%s capability=%s type=%s" % (dl(dict, 'name'), da(dict, 'capability', alsaseq._const_port_cap), da(dict, 'type', alsaseq._const_port_type)) + +debuginit() + +print "01:Creating Sequencer ==============================" +sequencer = alsaseq.Sequencer() +# other examples: +# sequencer = alsaseq.Sequencer("hw", "myfancyapplication", alsaseq.SEQ_OPEN_INPUT, alsaseq.SEQ_BLOCK) +# sequencer = alsaseq.Sequencer(clientname='python-test', streams=alsaseq.SEQ_OPEN_OUTPUT) +print " sequencer: %s" % sequencer +print " name: %s" % sequencer.name +print " clientname: %s" % sequencer.clientname +print " streams: %d (%s)" % (sequencer.streams, alsaseq._const_streams[sequencer.streams]) +print " mode: %d (%s)" % (sequencer.mode, alsaseq._const_mode[sequencer.mode]) +print " client_id: %s" % sequencer.client_id +print + +print "02:Changing some parameters ========================" +sequencer.clientname = 'pepito' +sequencer.mode = alsaseq.SEQ_BLOCK +print " clientname: %s" % sequencer.clientname +print " mode: %d (%s)" % (sequencer.mode, alsaseq._const_mode[sequencer.mode]) +print + +print "03:Creating simple port ============================" +port_id = sequencer.create_simple_port("myport", alsaseq.SEQ_PORT_CAP_WRITE | alsaseq.SEQ_PORT_CAP_SUBS_WRITE, alsaseq.SEQ_PORT_TYPE_APPLICATION) +print " port_id: %s" % port_id +print + +print "04:Getting port info ===============================" +port_info = sequencer.get_port_info(port_id) +print " --> %s" % dump_portinfo(port_info) +print " --> %s" % port_info +print + +print "05:Listing all client ports ========================" +connections = sequencer.get_client_ports() +print " %s" % (connections) +print + +print "06:Listing all client ports (lenghtly version) ========" +connections = sequencer.get_client_ports() +print " %s" % (connections) +for clientports in connections: + clientname, clientid, portlist = clientports + print " client: %3d %s" % (clientid, clientname) + for port in portlist: + portname, portid, connections = port + portinfo = sequencer.get_port_info(portid, clientid) + print " port: %3d:%-2d +-%s\t[%s]" % (clientid, portid, portname, dump_portinfo(portinfo)) + readconn, writeconn = connections + for c, p in readconn: + print " connection to: %d:%d" % (c,p) + for c,p in writeconn: + print " connection from: %d:%d" % (c,p) +print + +print "07:Connecting 'arbitrary' ports... =================" +sequencer.connect_ports((0, 1), (sequencer.client_id, port_id)) + +print "08:Listing all client ports (fancy version) ========" +connections = sequencer.get_client_ports() +print " %s" % (connections) +for clientports in connections: + clientname, clientid, portlist = clientports + print " client: %3d %s" % (clientid, clientname) + for port in portlist: + portname, portid, connections = port + readconn, writeconn = connections + for c, p in readconn: + print " connection to: %d:%d" % (c,p) + for c,p in writeconn: + print " connection from: %d:%d" % (c,p) +print + +print "09:Disconnecting previous 'arbitrary' port =========" +sequencer.disconnect_ports((0, 1), (sequencer.client_id, port_id)) + +print "10:Listing all client ports (fancy version) ========" +connections = sequencer.get_client_ports() +print " %s" % (connections) +for clientports in connections: + clientname, clientid, portlist = clientports + print " client: %3d %s" % (clientid, clientname) + for port in portlist: + portname, portid, connections = port + readconn, writeconn = connections + for c, p in readconn: + print " connection to: %d:%d" % (c,p) + for c,p in writeconn: + print " connection from: %d:%d" %init (c,p) +print + + +print "99:Removing sequencer ==============================" +debug([sequencer]) +del sequencer +print + +debugdone() +print "All Done." -- Aldrin Martoq

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[alsa-devel] [PATCH] Simplify to use Acer Aspire 5315
by Andy Shevchenko 24 Jan '08

24 Jan '08

Hi! Simplify usage of the Acer Aspire 5315 laptop with the ALC268 based codec sound card via add correct PCI SSID. Signed-off-by: Andy Shevchenko <andy(a)smile.org.ua> --- alsa-driver-hg20080124/alsa-kernel/pci/hda/patch_realtek.c.acer 2008-01-16 03:00:07.000000000 +0200 +++ alsa-driver-hg20080124/alsa-kernel/pci/hda/patch_realtek.c 2008-01-24 18:34:11.000000000 +0200 @@ -9896,6 +9896,7 @@ static const char *alc268_models[ALC268_ static struct snd_pci_quirk alc268_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0126, "Acer", ALC268_ACER), SND_PCI_QUIRK(0x1025, 0x0130, "Acer Extensa 5210", ALC268_ACER), + SND_PCI_QUIRK(0x1025, 0x0136, "Acer Aspire 5315", ALC268_ACER), SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL), SND_PCI_QUIRK(0x103c, 0x30cc, "TOSHIBA", ALC268_TOSHIBA), SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST), -- With best regards, Andy Shevchenko. mailto: andy(a)smile.org.ua

2 1

[alsa-devel] [PATCH] Add mute support to FM-only card with FM801 PCI to tuner bridge
by Andy Shevchenko 24 Jan '08

24 Jan '08

Hi! This is improvement of the early support of the FM-only cards where the fm801 chip represents the PCI to tuner bridge. The tuner initialization isn't included the mute on as well as mute support via V4L request. Proposed patch should fix this at least for 64-PCR model. Signed-off-by: Andy Shevchenko <andy(a)smile.org.ua> --- alsa-driver-hg20080124/alsa-kernel/i2c/other/tea575x-tuner.c.mute-fix 2008-01-09 03:00:14.000000000 +0200 +++ alsa-driver-hg20080124/alsa-kernel/i2c/other/tea575x-tuner.c 2008-01-24 16:58:25.000000000 +0200 @@ -53,6 +53,9 @@ MODULE_LICENSE("GPL"); #define TEA575X_BIT_DUMMY (1<<15) /* buffer */ #define TEA575X_BIT_FREQ_MASK 0x7fff +#define TEA575X_UNMUTE 0 /* just FALSE */ +#define TEA575X_MUTE 1 /* just TRUE */ + /* * lowlevel part */ @@ -158,6 +161,10 @@ static int snd_tea575x_ioctl(struct inod struct video_audio v; if(copy_from_user(&v, arg, sizeof(v))) return -EFAULT; + if (v.flags & VIDEO_AUDIO_MUTE) + tea->ops->mute(tea, TEA575X_MUTE); + else + tea->ops->mute(tea, TEA575X_UNMUTE); if(v.audio) return -EINVAL; return 0; @@ -205,6 +212,9 @@ void snd_tea575x_init(struct snd_tea575x tea->freq = 90500 * 16; /* 90.5Mhz default */ snd_tea575x_set_freq(tea); + + /* mute on init */ + tea->ops->mute(tea, TEA575X_MUTE); } void snd_tea575x_exit(struct snd_tea575x *tea) --- alsa-driver-hg20080124/alsa-kernel/pci/fm801.c.mute-fix 2008-01-09 03:00:15.000000000 +0200 +++ alsa-driver-hg20080124/alsa-kernel/pci/fm801.c 2008-01-24 17:02:17.000000000 +0200 @@ -802,6 +802,11 @@ static unsigned int snd_fm801_tea575x_25 return val; } +static void snd_fm801_tea575x_256pcs_mute(struct snd_tea575x *tea, unsigned int mute) +{ + /* TODO: Should be implemented */ +} + /* 256PCPR GPIO numbers */ #define TEA_256PCPR_BUS_CLOCK 0 #define TEA_256PCPR_DATA 1 @@ -890,6 +895,11 @@ static unsigned int snd_fm801_tea575x_25 return val; } +static void snd_fm801_tea575x_256pcpr_mute(struct snd_tea575x *tea, unsigned int mute) +{ + /* TODO: Should be implemented */ +} + /* 64PCR GPIO numbers */ #define TEA_64PCR_BUS_CLOCK 0 #define TEA_64PCR_WRITE_ENABLE 1 /* inverted */ @@ -978,21 +988,42 @@ static unsigned int snd_fm801_tea575x_64 return val; } +static void snd_fm801_tea575x_64pcr_mute(struct snd_tea575x *tea, unsigned int mute) +{ + struct fm801 *chip = tea->private_data; + unsigned short reg; + + spin_lock_irq(&chip->reg_lock); + + reg = inw(FM801_REG(chip, GPIO_CTRL)); + if (mute) + reg &= ~FM801_GPIO_GP(TEA_64PCR_WRITE_ENABLE); // 0xf800 (mute) + else + reg |= FM801_GPIO_GP(TEA_64PCR_WRITE_ENABLE); // 0xf802 (unmute) + outw(reg, FM801_REG(chip, GPIO_CTRL)); + udelay(1); + + spin_unlock_irq(&chip->reg_lock); +} + static struct snd_tea575x_ops snd_fm801_tea_ops[3] = { { /* 1 = MediaForte 256-PCS */ .write = snd_fm801_tea575x_256pcs_write, .read = snd_fm801_tea575x_256pcs_read, + .mute = snd_fm801_tea575x_256pcs_mute, }, { /* 2 = MediaForte 256-PCPR */ .write = snd_fm801_tea575x_256pcpr_write, .read = snd_fm801_tea575x_256pcpr_read, + .mute = snd_fm801_tea575x_256pcpr_mute, }, { /* 3 = MediaForte 64-PCR */ .write = snd_fm801_tea575x_64pcr_write, .read = snd_fm801_tea575x_64pcr_read, + .mute = snd_fm801_tea575x_64pcr_mute, } }; #endif --- alsa-driver-hg20080124/alsa-kernel/include/tea575x-tuner.h.mute-fix 2007-10-16 03:00:12.000000000 +0300 +++ alsa-driver-hg20080124/alsa-kernel/include/tea575x-tuner.h 2008-01-24 16:58:25.000000000 +0200 @@ -30,6 +30,7 @@ struct snd_tea575x; struct snd_tea575x_ops { void (*write)(struct snd_tea575x *tea, unsigned int val); unsigned int (*read)(struct snd_tea575x *tea); + void (*mute)(struct snd_tea575x *tea, unsigned int mute); }; struct snd_tea575x { -- With best regards, Andy Shevchenko. mailto: andy(a)smile.org.ua

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[alsa-devel] Problems with ALC882
by Christophe Osuna 24 Jan '08

24 Jan '08

Hi, I am having a bad time trying to get an ALC882 [1] to work. I hope that you can help me. Currently I am using alsa-driver-1.0.14 but it would probably be the same with 1.0.15 since there has been no change in /pci/hda/. The ALC882 has got three ADC, and for every one of them it is possible to select one input source. amixer shows my current settings: ~# amixer get 'Input Source',0 Simple mixer control 'Input Source',0 Capabilities: enum Items: 'Mic' 'Front Mic' 'Line' 'CD' Item0: 'Mic' ~# amixer get 'Input Source',1 Simple mixer control 'Input Source',1 Capabilities: enum Items: 'Mic' 'Front Mic' 'Line' 'CD' Item0: 'Front Mic' ~# amixer get 'Input Source',2 Simple mixer control 'Input Source',2 Capabilities: enum Items: 'Mic' 'Front Mic' 'Line' 'CD' Item0: 'Line' Put more simply: Mic -> Input 0 (1st ADC) Front Mic -> Input 1 (2nd ADC) Line In -> Input 2 (3rd ADC) A grep in the proc filesystem shows that there are three ADC and one digital input. ~# cat /proc/asound/card0/codec\#2 | grep Input Node 0x07 [Audio Input] wcaps 0x10011b: Stereo Amp-In Node 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In Node 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In Node 0x0a [Audio Input] wcaps 0x100391: Stereo Digital So far, so good. But then I try to use the PCM system and things start to go worse. ~# cat /proc/asound/pcm 00-02: ALC882 Analog : ALC882 Analog : capture 2 00-01: ALC882 Digital : ALC882 Digital : playback 1 : capture 1 00-00: ALC882 Analog : ALC882 Analog : playback 1 : capture 2 If my ALSA understanding is right, that means I have two analog input devices (hw:0,0 and hw:0,2). One is missing! If I ask arecord to list the capture devices, it shows the following: ~# ./arecord -l **** List of CAPTURE Hardware Devices **** card 0: Intel [HDA Intel], device 0: ALC882 Analog [ALC882 Analog] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 card 0: Intel [HDA Intel], device 1: ALC882 Digital [ALC882 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 2: ALC882 Analog [ALC882 Analog] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 I have tried to pipe arecord into aplay several times: 1) arecord -f dat -D hw:0,0,0 | aplay -f dat -D hw:0,0,0 2) arecord -f dat -D hw:0,0,1 | aplay -f dat -D hw:0,0,0 3) arecord -f dat -D hw:0,1,0 | aplay -f dat -D hw:0,0,0 (fails) 4) arecord -f dat -D hw:0,2,0 | aplay -f dat -D hw:0,0,0 5) arecord -f dat -D hw:0,2,1 | aplay -f dat -D hw:0,0,0 6) arecord -f dat -D hw:0,0,0 > /dev/null (fails) and arecord -f dat -D hw:0,2,0 > /dev/null (fails) and it shows that: 1) hw:0,0,0 maps to 1st ADC: OK 2) hw:0,0,1 maps to 2nd ADC: OK 3) fails probably because digital input is not activated 4) hw:0,2,0 maps to 1st ADC (same as hw:0,0,0) 5) hw:0,2,1 maps to 2nd ADC (same as hw:0,0,1) 6) when I start the second arecord the first stops (I/O error) and then the second one stops (error), probably because they are working on the same ADC. So it appears that while the "control" ALSA layer detects the 3 ADC the "PCM" layer does not. Do you have any idea of what could be wrong? Besides that I have another smaller problem. According to the ALC882 documentation, the codec can handle 8 kHz signals (by sending setting only one packet on every six packets of the HDA bus which operates at 48 kHz). However the driver will not let me use that sample rate. Example from /proc: Node 0x07 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x08, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x14 0x14] PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x24 Is this a limit of the HDA implementation? Thanks for your help. [1] http://www.realtek.com.tw/products/productsView.aspx?Langid=1&PFid=28&Level…

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[alsa-devel] [PATCH] hda: STAC9205 GPIO line fix
by Matthew Ranostay 24 Jan '08

24 Jan '08

Fixed issue that the incorrect GPIO line was being pulled high for some STAC9205 based laptops. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff -r 32b13f326915 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Thu Jan 24 11:49:21 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Thu Jan 24 09:08:23 2008 -0500 @@ -3661,11 +3661,11 @@ static int patch_stac9205(struct hda_cod stac92xx_set_config_reg(codec, 0x1f, 0x01441030); stac92xx_set_config_reg(codec, 0x20, 0x1c410030); - spec->gpio_mask = 0x00000007; /* GPIO0-2 */ - /* GPIO0 High = EAPD, GPIO1 Low = DRM, - * GPIO2 High = Headphone Mute + spec->gpio_mask = 0x0000000b; + /* GPIO0 High = EAPD, GPIO1 Low = Headphone Mute, + * GPIO3 High = DRM */ - spec->gpio_data = 0x00000005; + spec->gpio_data = 0x00000009; break; default: /* GPIO0 High = EAPD */

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Re: [alsa-devel] 1.0.16rc1 release -- error: ‘system_utsname’ undeclared
by Takashi Iwai 24 Jan '08

24 Jan '08

At Fri, 25 Jan 2008 00:30:31 +1300, Brad Milne wrote: > > On Jan 25, 2008 12:23 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Fri, 25 Jan 2008 00:18:33 +1300, > > > > Brad Milne wrote: > > > > > > On Jan 24, 2008 11:58 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > > > > At Thu, 24 Jan 2008 23:40:55 +1300, > > > > > > > > Brad Milne wrote: > > > > > > > > > > > From tiwai at suse.de Wed Jan 23 08:31:32 2008 > > > > > > From: tiwai at suse.de (Takashi Iwai) > > > > > > Date: Wed, 23 Jan 2008 08:31:32 +0100 > > > > > > Subject: [alsa-devel] > > > > > > =?iso-8859-7?q?1=2E0=2E16rc1_release_--_error=3A_=A1?= > > > > > > =?iso-8859-7?q?system=5Futsname=A2_undeclared?= > > > > > > In-Reply-To: <s5htzl5t45y.wl%tiwai(a)suse.de> > > > > > > References: < Pine.LNX.4.61.0801221247570.9698(a)tm8103.perex-int.cz> > > > > > > <47964342.1060608(a)measham.id.au> <s5htzl5t45y.wl%tiwai(a)suse.de > > > > > > > Message-ID: <s5hsl0pt2uj.wl%tiwai(a)suse.de> > > > > > > > > > > > At Wed, 23 Jan 2008 08:03:05 +0100, > > > > > > I wrote: > > > > > > > > > > > > > > At Wed, 23 Jan 2008 06:25:54 +1100, > > > > > > > Rick Measham wrote: > > > > > > > > > > > > > > > > [1 <multipart/signed (7bit)>] > > > > > > > > [1.1 <text/plain; UTF-8 (quoted-printable)>] > > > > > > > > Jaroslav Kysela wrote: > > > > > > > > > 1.0.16rc1 release is available for download on > > > > > > > > > > > > > > > > Running Kubuntu: > > > > > > > > > > > > > > > > # uname -a > > > > > > > > Linux lugtop 2.6.20-16-386 #2 Sun Sep 23 19:47:10 UTC 2007 i686 GNU/Linux > > > > > > > > > > > > > > > > > > > > > > > > During the make: > > > > > > > > > > > > > > > > In file included from > > > > > > > > /usr/src/alsa/alsa-driver-1.0.16rc1/acore/info_oss.c:7: > > > > > > > > /usr/src/alsa/alsa-driver-1.0.16rc1/acore/../alsa-kernel/core/info_oss.c: > > > > > > > > In function ?snd_sndstat_proc_read?: > > > > > > > > /usr/src/alsa/alsa-driver-1.0.16rc1/acore/../alsa-kernel/core/info_oss.c:96: > > > > > > > > error: ?system_utsname? undeclared (first use in this function) > > > > > > > > /usr/src/alsa/alsa-driver-1.0.16rc1/acore/../alsa-kernel/core/info_oss.c:96: > > > > > > > > error: (Each undeclared identifier is reported only once > > > > > > > > /usr/src/alsa/alsa-driver-1.0.16rc1/acore/../alsa-kernel/core/info_oss.c:96: > > > > > > > > error: for each function it appears in.) > > > > > > > > make[3]: *** [/usr/src/alsa/alsa-driver-1.0.16rc1/acore/info_oss.o] Error 1 > > > > > > > > make[2]: *** [/usr/src/alsa/alsa-driver-1.0.16rc1/acore] Error 2 > > > > > > > > make[1]: *** [_module_/usr/src/alsa/alsa-driver-1.0.16rc1] Error 2 > > > > > > > > make[1]: Leaving directory `/usr/src/linux-headers-2.6.20-16-386' > > > > > > > > make: *** [compile] Error 2 > > > > > > > > > > > > > > Isn't it fixed on HG version? > > > > > > > > > > > Nevermind, I myself can confirm that it's broken. > > > > > > The patch below should fix the problem. > > > > > > > > > > > > > > > > Takashi > > > > > > > > > > > diff -r a5f603cb254a configure.in > > > > > > --- a/configure.in Tue Jan 22 17:52:34 2008 +0100 > > > > > > +++ b/configure.in Wed Jan 23 08:28:56 2008 +0100 > > > > > > @@ -2571,7 +2571,7 @@ init_utsname="0" > > > > > > init_utsname="0" > > > > > > ac_save_CFLAGS="$CFLAGS" > > > > > > ac_save_CC="$CC" > > > > > > -CFLAGS="$KERNEL_CHECK_CFLAGS -Werror" > > > > > > +CFLAGS="$KERNEL_CHECK_CFLAGS -Werror=implicit-function-declaration" > > > > > > CC=$KCC > > > > > > AC_TRY_COMPILE([ > > > > > > #define __KERNEL__ > > > > > > > > > > It doesn't seem that it is fixed in HG or by this patch (though I do get: > > > > > patching file configure.in > > > > > Hunk #1 succeeded at 1579 with fuzz 2 (offset > > > > > -992 lines). > > > > > patch unexpectedly ends in middle of line > > > > > when patching), I get the exact same error in both. > > > > > > > > And did you run hgcompile after that? > > > > > > If I run: > > > sudo ./configure > > > then: > > > sudo ./hgcompile > > > > You don't have to be root for compiling the driver. The root priv is > > needed only for installation, i.e. make install. > > > > > on the latest hg code, I still get the same error: > > > > > from /usr/src/alsa/alsa-driver/acore/info_oss.c:7: > > > include/linux/utsname.h:77: error: expected '=', ',', ';', 'asm' or > > > '__attribute__' before '{' token > > > ... > > > /usr/src/alsa/alsa-driver/acore/../alsa-kernel/core/info_oss.c: In > > > function 'snd_sndstat_proc_read': > > > /usr/src/alsa/alsa-driver/acore/../alsa-kernel/core/info_oss.c:96: > > > error: 'system_utsname' undeclared (first use in this function) > > > ... > > > make[3]: *** [/usr/src/alsa/alsa-driver/acore/info_oss.o] Error 1 > > > ... > > > > Interesting. You're using 2.6.20 kernel, right? > > > > First check include/config.h. whether it defines > > CONFIG_HAVE_INIT_UTSNAME. If it's defined, then the build must go > > well. If it's undefined (commented out), then check configure.in > > whether the patch was properly applied and -Werror was changed to > > -Werror=implicit-function-declaration. And see config.log at the > > place where init_utsname is checked. You must have a compile error > > there. Show that part. > > brad@cyborg:/usr/src/alsa/alsa-driver$ uname -a > Linux cyborg 2.6.22-14-generic #1 SMP Tue Dec 18 08:02:57 UTC 2007 > i686 GNU/Linux > > include/config.h: > /* 2.6 kernel */ > #define CONFIG_SND_NESTED_CLASS_DEVICE 1 > /* #undef CONFIG_SND_OLD_DRIVER_SUSPEND */ > #define CONFIG_SND_REMOVE_PAGE_RESERVE 1 > #define CONFIG_SND_NEW_IRQ_HANDLER 1 > #define CONFIG_HAVE_PNP_SUSPEND 1 > #define CONFIG_HAVE_GFP_T 1 > /* #undef CONFIG_HAVE_INIT_UTSNAME */ <----!! > > configure.in (hg version): > dnl Check for init_utsname... > AC_MSG_CHECKING(for init_utsname) > init_utsname="0" > ac_save_CFLAGS="$CFLAGS" > ac_save_CC="$CC" > CFLAGS="$KERNEL_CHECK_CFLAGS -Werror=implicit-function-declaration" > CC=$KCC > AC_TRY_COMPILE([ > #define __KERNEL__ > > config.log: > configure:11638: checking for init_utsname > configure:11673: gcc -c -Wall -fno-strict-aliasing -g -O2 > -I/lib/modules/2.6.22-14-generic/build/include > -I/lib/modules/2.6.22-14-generic/build/include/asm-i386/mach-default > -Iinclude -nostdinc -iwithprefix include -DKBUILD_STR(s)=#s > -DKBUILD_BASENAME=KBUILD_STR(dummy) -DKBUILD_MODNAME=KBUILD_BASENAME > -Werror=implicit-function-declaration conftest.c >&5 > cc1: error: unrecognized command line option > "-Werror=implicit-function-declaration" Grrr, it looks like this option isn't available for all gcc versions. Then try the patch below and run hgcompile. It should work, at least, for your system. thanks, Takashi --- diff -r e80f59e90231 configure.in --- a/configure.in Wed Jan 23 08:33:10 2008 +0100 +++ b/configure.in Thu Jan 24 12:36:35 2008 +0100 @@ -2567,13 +2567,16 @@ fi fi dnl Check for init_utsname... +if test "$kversion.$kpatchlevel" = "2.6" -a $ksublevel -ge 19; then + CONFIG_HAVE_INIT_UTSNAME=1 +else AC_MSG_CHECKING(for init_utsname) -init_utsname="0" -ac_save_CFLAGS="$CFLAGS" -ac_save_CC="$CC" -CFLAGS="$KERNEL_CHECK_CFLAGS -Werror=implicit-function-declaration" -CC=$KCC -AC_TRY_COMPILE([ + init_utsname="0" + ac_save_CFLAGS="$CFLAGS" + ac_save_CC="$CC" + CFLAGS="$KERNEL_CHECK_CFLAGS -Werror" + CC=$KCC + AC_TRY_COMPILE([ #define __KERNEL__ #include <linux/config.h> #include <linux/utsname.h> @@ -2585,9 +2588,10 @@ AC_TRY_COMPILE([ AC_MSG_RESULT(no);init_utsname="0", AC_MSG_RESULT(unknown);init_utsname="0" ) -CFLAGS=$ac_save_CFLAGS -CC=$ac_save_CC -CONFIG_HAVE_INIT_UTSNAME=$init_utsname + CFLAGS=$ac_save_CFLAGS + CC=$ac_save_CC + CONFIG_HAVE_INIT_UTSNAME=$init_utsname +fi dnl AC_SUBST(CONFIG_HAVE_INIT_UTSNAME) if test "$CONFIG_HAVE_INIT_UTSNAME" = "1"; then AC_DEFINE(CONFIG_HAVE_INIT_UTSNAME)

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Re: [alsa-devel] 1.0.16rc1 release -- error: ‘system_utsname’ undeclared
by Brad Milne 24 Jan '08

24 Jan '08

> From tiwai at suse.de Wed Jan 23 08:31:32 2008 > From: tiwai at suse.de (Takashi Iwai) > Date: Wed, 23 Jan 2008 08:31:32 +0100 > Subject: [alsa-devel] > =?iso-8859-7?q?1=2E0=2E16rc1_release_--_error=3A_=A1?= > =?iso-8859-7?q?system=5Futsname=A2_undeclared?= > In-Reply-To: <s5htzl5t45y.wl%tiwai(a)suse.de> > References: < Pine.LNX.4.61.0801221247570.9698(a)tm8103.perex-int.cz> > <47964342.1060608(a)measham.id.au> <s5htzl5t45y.wl%tiwai(a)suse.de > > Message-ID: <s5hsl0pt2uj.wl%tiwai(a)suse.de> > At Wed, 23 Jan 2008 08:03:05 +0100, > I wrote: > > > > At Wed, 23 Jan 2008 06:25:54 +1100, > > Rick Measham wrote: > > > > > > [1 <multipart/signed (7bit)>] > > > [1.1 <text/plain; UTF-8 (quoted-printable)>] > > > Jaroslav Kysela wrote: > > > > 1.0.16rc1 release is available for download on > > > > > > Running Kubuntu: > > > > > > # uname -a > > > Linux lugtop 2.6.20-16-386 #2 Sun Sep 23 19:47:10 UTC 2007 i686 GNU/Linux > > > > > > > > > During the make: > > > > > > In file included from > > > /usr/src/alsa/alsa-driver-1.0.16rc1/acore/info_oss.c:7: > > > /usr/src/alsa/alsa-driver-1.0.16rc1/acore/../alsa-kernel/core/info_oss.c: > > > In function ?snd_sndstat_proc_read?: > > > /usr/src/alsa/alsa-driver-1.0.16rc1/acore/../alsa-kernel/core/info_oss.c:96: > > > error: ?system_utsname? undeclared (first use in this function) > > > /usr/src/alsa/alsa-driver-1.0.16rc1/acore/../alsa-kernel/core/info_oss.c:96: > > > error: (Each undeclared identifier is reported only once > > > /usr/src/alsa/alsa-driver-1.0.16rc1/acore/../alsa-kernel/core/info_oss.c:96: > > > error: for each function it appears in.) > > > make[3]: *** [/usr/src/alsa/alsa-driver-1.0.16rc1/acore/info_oss.o] Error 1 > > > make[2]: *** [/usr/src/alsa/alsa-driver-1.0.16rc1/acore] Error 2 > > > make[1]: *** [_module_/usr/src/alsa/alsa-driver-1.0.16rc1] Error 2 > > > make[1]: Leaving directory `/usr/src/linux-headers-2.6.20-16-386' > > > make: *** [compile] Error 2 > > > > Isn't it fixed on HG version? > Nevermind, I myself can confirm that it's broken. > The patch below should fix the problem. > Takashi > diff -r a5f603cb254a configure.in > --- a/configure.in Tue Jan 22 17:52:34 2008 +0100 > +++ b/configure.in Wed Jan 23 08:28:56 2008 +0100 > @@ -2571,7 +2571,7 @@ init_utsname="0" > init_utsname="0" > ac_save_CFLAGS="$CFLAGS" > ac_save_CC="$CC" > -CFLAGS="$KERNEL_CHECK_CFLAGS -Werror" > +CFLAGS="$KERNEL_CHECK_CFLAGS -Werror=implicit-function-declaration" > CC=$KCC > AC_TRY_COMPILE([ > #define __KERNEL__ It doesn't seem that it is fixed in HG or by this patch (though I do get: patching file configure.in Hunk #1 succeeded at 1579 with fuzz 2 (offset -992 lines). patch unexpectedly ends in middle of line when patching), I get the exact same error in both. I don't pretend to know a lot about alsa workings, but is this patch in the right area as the error complains about 'system_utsname', not 'init_utsname'? Brad

2 3

[alsa-devel] [PATCH1/2] hda: Added mono_out_pin to autoconfig
by Matthew Ranostay 24 Jan '08

24 Jan '08

Added a mono_out_pin field to autocfg struct, and code to parse for the mono_out_line. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff -r 5492010913f0 pci/hda/hda_codec.c --- a/pci/hda/hda_codec.c Tue Jan 22 15:37:13 2008 +0100 +++ b/pci/hda/hda_codec.c Wed Jan 23 10:55:11 2008 -0500 @@ -769,7 +769,7 @@ get_alloc_amp_hash(struct hda_codec *cod /* * query AMP capabilities for the given widget and direction */ -static u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction) +u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction) { struct hda_amp_info *info; @@ -2708,6 +2708,10 @@ int snd_hda_parse_pin_def_config(struct case AC_JACK_LINE_OUT: seq = get_defcfg_sequence(def_conf); assoc = get_defcfg_association(def_conf); + + if (!(wid_caps & AC_WCAP_STEREO)) + if (!cfg->mono_out_pin) + cfg->mono_out_pin = nid; if (!assoc) continue; if (!assoc_line_out) @@ -2856,6 +2860,7 @@ int snd_hda_parse_pin_def_config(struct cfg->hp_outs, cfg->hp_pins[0], cfg->hp_pins[1], cfg->hp_pins[2], cfg->hp_pins[3], cfg->hp_pins[4]); + snd_printd(" mono: mono_out=0x%x\n", cfg->mono_out_pin); snd_printd(" inputs: mic=0x%x, fmic=0x%x, line=0x%x, fline=0x%x," " cd=0x%x, aux=0x%x\n", cfg->input_pins[AUTO_PIN_MIC], diff -r 5492010913f0 pci/hda/hda_local.h --- a/pci/hda/hda_local.h Tue Jan 22 15:37:13 2008 +0100 +++ b/pci/hda/hda_local.h Wed Jan 23 10:56:40 2008 -0500 @@ -332,6 +332,7 @@ struct auto_pin_cfg { hda_nid_t input_pins[AUTO_PIN_LAST]; hda_nid_t dig_out_pin; hda_nid_t dig_in_pin; + hda_nid_t mono_out_pin; }; #define get_defcfg_connect(cfg) \ @@ -374,6 +375,7 @@ static inline u32 get_wcaps(struct hda_c return codec->wcaps[nid - codec->start_nid]; } +u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction); int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir, unsigned int caps);

2 1

[alsa-devel] Fixing my sound driver
by Adrian McMenamin 24 Jan '08

24 Jan '08

I am still trying to find out what is wrong with my sound driver - for the Sega Dreamcast (sound/sh/aica.c). This works perfectly with the OSS emulation layer but will not work with the ALSA layer per se. As this shows - there is something wrong with how it interacts with the PCM layer of alsa-lib: $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: AICA [AICA], device 0: AICA PCM [AICA PCM] Subdevices: 1/1 Subdevice #0: subdevice #0 $ $ $ aplay -L null Discard all samples (playback) or generate zero samples (capture) I have no reason to think that my build of alsa-lib is wrong - both the automagic buildroot build and my own by-hand build behave the same way. So therefore I assume it must be something in the driver?! Some questions: The driver implements nothing for capture (the AICA device is pure playback) - could that have an effect? What information must the driver pass to the PCM layer? What could make the alsa-lib layer detect the card but then report "null" is the only PCM device? Adrian

2 3

[alsa-devel] Specific question about ICE1724 + AK4114
by Pavel Hofman 24 Jan '08

24 Jan '08

Hello, I would like to add support for automatic detection of incoming SPDIF sample rate for ICE cards with AK4114 DIR, specifically ESI Juli and Prodigy192 with MI/IODI/O card. Juli windows drivers support the detection, Prodigy drivers don't (broken in some other way too), but the hardware is basically identical. AK4114 support detection by comparing the recovered clock from SPDIF with another clock of 11, 12 or 24MHz, supplied by a crystal, or externally. In both cards this functionality seems to be provided by Xilinx XC9536 CPLD, together with PLL frequency multiplier MK73-1. It all fits together, the CPLD provides the freqency division for PLL feedback of the multiplier. External reference clock input of AK4114 is connected to CPLD, as well as setup pins XTL0/1. Output frequency of MK73-1 specified in datasheet is in the range of 10-50MHz Here is my question: For the PLL multiplier to work, its datasheet says it must be fed with input pulses of 20-50kHz. This frequency could be easily provided by ICE1724, BUT in master mode only. When in slave mode (which is the case for active SPDIF input - clock provided by AK4114), all the clocks are derived from the external clock. This signal obviously cannot be used as reference for the sample rate detection in AK4114 - i.e. basically of itself. According to the ICE1724 datasheet, only I2C and MIDI clocks are derived from the 48kHz mode crystal in slave mode. The datasheed says I2C clock is output only during I2C read/write activity, MIDI clock is not directly output at all. The cards have no other visible crystals/clock modules apart of those two connected directly to ICE1724. I could not trace any connections from the actual crystals to CPLD. Now here I am, absolutely wondering where the CPLD gets its low frequency stable signal for the PLL multiplier, to provide reference clock for SPDIF rate detection. I am afraid without this knowledge I cannot make the detection work in the alsa driver. Hopefully someone experienced in this area will throw a ray of light upon this mystery. Thanks in advance for any ideas, information, suggestions. Best regards, Pavel Hofman.

1 0

[alsa-devel] [PATCH2/2] hda: Add dynamic mono mixer support for STAC92xx codecs
by Matthew Ranostay 23 Jan '08

23 Jan '08

Allows for dynamically creating mono out mixer controls and well as mono mux controls. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff -r 5492010913f0 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Tue Jan 22 15:37:13 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Tue Jan 22 21:22:52 2008 -0500 @@ -756,7 +756,6 @@ static struct snd_kcontrol_new stac92hd7 static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { STAC_INPUT_SOURCE(2), - STAC_MONO_MUX, HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), @@ -768,15 +767,12 @@ static struct snd_kcontrol_new stac92hd7 HDA_CODEC_MUTE("Analog Loopback 1", 0x17, 0x3, HDA_INPUT), HDA_CODEC_MUTE("Analog Loopback 2", 0x17, 0x4, HDA_INPUT), - - HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x14, 0x1, 0, HDA_INPUT), { } /* end */ }; static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { STAC_INPUT_SOURCE(2), STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2), - STAC_MONO_MUX, HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), @@ -785,8 +781,6 @@ static struct snd_kcontrol_new stac92hd7 HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1d, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x0, HDA_OUTPUT), - - HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x14, 0x1, 0, HDA_INPUT), { } /* end */ }; @@ -1157,7 +1151,7 @@ static struct snd_pci_quirk stac9200_cfg static unsigned int ref925x_pin_configs[8] = { 0x40c003f0, 0x424503f2, 0x01813022, 0x02a19021, - 0x90a70320, 0x02214210, 0x400003f1, 0x9033032e, + 0x90a70320, 0x02214210, 0x01019020, 0x9033032e, }; static unsigned int stac925x_MA6_pin_configs[8] = { @@ -1561,7 +1555,7 @@ static struct snd_pci_quirk stac927x_cfg static unsigned int ref9205_pin_configs[12] = { 0x40000100, 0x40000100, 0x01016011, 0x01014010, - 0x01813122, 0x01a19021, 0x40000100, 0x40000100, + 0x01813122, 0x01a19021, 0x01019020, 0x40000100, 0x90a000f0, 0x90a000f0, 0x01441030, 0x01c41030 }; @@ -2018,6 +2012,7 @@ enum { enum { STAC_CTL_WIDGET_VOL, STAC_CTL_WIDGET_MUTE, + STAC_CTL_WIDGET_MONO_MUX, STAC_CTL_WIDGET_IO_SWITCH, STAC_CTL_WIDGET_CLFE_SWITCH }; @@ -2025,6 +2020,7 @@ static struct snd_kcontrol_new stac92xx_ static struct snd_kcontrol_new stac92xx_control_templates[] = { HDA_CODEC_VOLUME(NULL, 0, 0, 0), HDA_CODEC_MUTE(NULL, 0, 0, 0), + STAC_MONO_MUX, STAC_CODEC_IO_SWITCH(NULL, 0), STAC_CODEC_CLFE_SWITCH(NULL, 0), }; @@ -2388,7 +2384,9 @@ static int stac92xx_auto_create_mono_out mono_mux->items[mono_mux->num_items].index = i; mono_mux->num_items++; } - return 0; + + return stac92xx_add_control(spec, STAC_CTL_WIDGET_MONO_MUX, + "Mono Mux", spec->mono_nid); } /* labels for dmic mux inputs */ @@ -2569,6 +2567,50 @@ static int stac92xx_parse_auto_config(st sizeof(spec->autocfg.hp_pins)); spec->autocfg.line_outs = spec->autocfg.hp_outs; hp_speaker_swap = 1; + } + if (spec->autocfg.mono_out_pin) { + int dir = (get_wcaps(codec, spec->autocfg.mono_out_pin) + & AC_WCAP_OUT_AMP) ? HDA_OUTPUT : HDA_INPUT; + u32 caps = query_amp_caps(codec, + spec->autocfg.mono_out_pin, dir); + hda_nid_t conn_list[1]; + + /* get the mixer node and then the mono mux if it exists */ + if (snd_hda_get_connections(codec, + spec->autocfg.mono_out_pin, conn_list, 1) && + snd_hda_get_connections(codec, conn_list[0], + conn_list, 1)) { + + int wcaps = get_wcaps(codec, conn_list[0]); + int wid_type = (wcaps & AC_WCAP_TYPE) + >> AC_WCAP_TYPE_SHIFT; + /* LR swap check, some stac925x have a mux that + * changes the DACs output path instead of the + * mono-mux path. + */ + if (wid_type == AC_WID_AUD_SEL && + !(wcaps & AC_WCAP_LR_SWAP)) + spec->mono_nid = conn_list[0]; + } + /* all mono outs have a least a mute/unmute switch */ + err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, + "Mono Playback Switch", + HDA_COMPOSE_AMP_VAL(spec->autocfg.mono_out_pin, + 1, 0, dir)); + if (err < 0) + return err; + /* check to see if there is volume support for the amp */ + if ((caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT) { + err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, + "Mono Playback Volume", + HDA_COMPOSE_AMP_VAL(spec->autocfg.mono_out_pin, + 1, 0, dir)); + if (err < 0) + return err; + } + + stac92xx_auto_set_pinctl(codec, spec->autocfg.mono_out_pin, + AC_PINCTL_OUT_EN); } if ((err = stac92xx_add_dyn_out_pins(codec, &spec->autocfg)) < 0) @@ -3317,7 +3359,6 @@ again: spec->gpio_mask = spec->gpio_data = 0x00000001; /* GPIO0 High = EAPD */ - spec->mono_nid = 0x15; spec->mux_nids = stac92hd71bxx_mux_nids; spec->adc_nids = stac92hd71bxx_adc_nids; spec->dmic_nids = stac92hd71bxx_dmic_nids;

1 0

[alsa-devel] problems with Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03)
by forwind 23 Jan '08

23 Jan '08

Hi folks, I am having difficulty getting a friends on-board speakers working on her laptop. After some reading of forums I'm not sure whether there is a solution for this or not ? The setup and details can be read below. The headphones work fine but the speakers are mute even though nothing is muted. I installed from source alsa version 1.0.15rc3 but to no avail. ALSA Audio Debug v0.1.0 - Thu Jan 17 23:21:23 GMT 2008 http://alsa.opensrc.org/aadebug http://www.gnu.org/licenses/gpl.txt Kernel ---------------------------------------------------- Linux laura-laptop 2.6.22-14-rt #1 SMP PREEMPT RT Tue Dec 18 10:01:34 UTC 2007 i686 GNU/Linux Loaded Modules -------------------------------------------- snd_rtctimer 4384 1 snd_hda_intel 291616 1 snd_pcm_oss 43264 0 snd_mixer_oss 17664 1 snd_pcm_oss snd_pcm 80132 2 snd_hda_intel,snd_pcm_oss snd_page_alloc 11400 2 snd_hda_intel,snd_pcm snd_hwdep 10500 1 snd_hda_intel snd_seq_dummy 4740 0 snd_seq_oss 33792 0 snd_seq_midi 9600 0 snd_rawmidi 25728 1 snd_seq_midi snd_seq_midi_event 8576 2 snd_seq_oss,snd_seq_midi snd_seq 54128 7 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_seq_midi_event snd_timer 24324 3 snd_rtctimer,snd_pcm,snd_seq snd_seq_device 9228 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_rawmidi,snd_seq snd 55428 13 snd_hda_intel,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_hwdep,snd_seq_oss,snd_rawmidi,snd_seq,snd_timer,snd_seq_device Modprobe Conf --------------------------------------------- Warning: module config file does not exist This means any kernel modules will not be auto loaded See your linux distro docs on how to create this file Proc Asound ----------------------------------------------- Advanced Linux Sound Architecture Driver Version 1.0.15rc3. 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xfc200000 irq 22 2: : timer 3: : sequencer 4: [ 0- 2]: digital audio capture 5: [ 0- 1]: digital audio playback 6: [ 0- 0]: digital audio playback 7: [ 0- 0]: digital audio capture 8: [ 0- 1]: hardware dependent 9: [ 0- 0]: hardware dependent 10: [ 0] : control 00-01: HDA Codec 1 00-00: HDA Codec 0 00-02: ALC883 Analog : ALC883 Analog : capture 2 00-01: ALC883 Digital : ALC883 Digital : playback 1 00-00: ALC883 Analog : ALC883 Analog : playback 1 : capture 2 Client info cur clients : 4 peak clients : 4 max clients : 192 Client 0 : "System" [Kernel] Port 0 : "Timer" (Rwe-) Port 1 : "Announce" (R-e-) Connecting To: 15:0 Client 14 : "Midi Through" [Kernel] Port 0 : "Midi Through Port-0" (RWe-) Client 15 : "OSS sequencer" [Kernel] Port 0 : "Receiver" (-we-) Connected From: 0:1 Client 128 : "TiMidity" [User] Port 0 : "TiMidity port 0" (-We-) Port 1 : "TiMidity port 1" (-We-) Port 2 : "TiMidity port 2" (-We-) Port 3 : "TiMidity port 3" (-We-) Output pool : Pool size : 500 Cells in use : 0 Peak cells in use : 0 Alloc success : 0 Alloc failures : 0 Input pool : Pool size : 1000 Cells in use : 0 Peak cells in use : 0 Alloc success : 0 Alloc failures : 0 Dev Snd --------------------------------------------------- controlC0 hwC0D0 hwC0D1 pcmC0D0c pcmC0D0p pcmC0D1p pcmC0D2c seq timer CPU ------------------------------------------------------- model name : Intel(R) Core(TM)2 Duo CPU T7300 @ 2.00GHz cpu MHz : 2001.000 model name : Intel(R) Core(TM)2 Duo CPU T7300 @ 2.00GHz cpu MHz : 2001.000 RAM ------------------------------------------------------- MemTotal: 1028132 kB SwapTotal: 3028212 kB Hardware -------------------------------------------------- 00:00.0 Host bridge: Intel Corporation Mobile PM965/GM965/GL960 Memory Controller Hub (rev 03) does this sound familiar or is everybody blue in the face from hearing about this 'bug' :)? Best regards, Conor

3 5

[alsa-devel] The state of Digigram cards
by Mark Hills 23 Jan '08

23 Jan '08

Hi, I've been trying to use Digigram cards with ALSA using the drivers specific to these cards: * VX222 (detected as vx222/Old) * miXArt 8 Since about 1 year ago, I've tried both cards in various machines with various software configurations, and my experiences have ranged from complete system lockups to problems opening the device. Using more than one CPU seems to be a certain way to get system lockups. Is anybody actually using the Digigram drivers? Searching the various archives shows interest in the cards from a number of years ago. Is it possible that more recent developments to ALSA and the kernel cause the drivers to be unstable? Thanks, Mark

1 0

[alsa-devel] S16_LE + softvol = bad
by Alexander E. Patrakov 23 Jan '08

23 Jan '08

Hello developers, some cards (most notably, Intel HD audio) have the PCM control as softvol, and default to S16_LE as a format for dmix. Since softvol is configured to attenuate signal, and not many people set this control to 100% (especially those who have no other volume controls), this effectively leaves less than 16 bits per sample. The card can surely produce sound of better quality in most cases, if one changes defaults.pcm.dmix.format in ~/.asoundrc. However, one should now search a lot of forums in order to know about this setting. Not everyone does this, and not everyone is even able to diagnose the "noisy sound in Linux" problem properly as "too few bits per sample in the default configuration". A much better situation would be if ALSA defaulted to good quality (i.e.: S24_LE or S32_LE on all cards that require softvol and support this sample format). Could you please change the files in /usr/share/alsa/cards accordingly? -- Alexander E. Patrakov

3 7

Re: [alsa-devel] how do you people debug ALSA?
by mazarick＠bellsouth.net 23 Jan '08

23 Jan '08

"Mihaela Vitalariu" <admiralwolverine(a)gmail.com> wrote: > I meant what tools do you use, what Debuggers? > In my post i wasn't complaining about ALSA, i was just curios about the > debugging tools that you use, from a developer's standpoint... The group that I'm working with (to develop an alsa driver for the Gadget Labs soundcard) would be very interested in any input on this subject. The team is made up of skilled C/C++ coders, but there is very little to no experience with writing a linux hardware driver. It would be fine to say "vi is the development environment, and we sprinkle the code with printk statements and take them out when the soup's done". It would save us a lot of overhead in figuring out the best way to get on the same page with each other since it's a pretty new team. Any other words of wisdom from those who've been there would be very welcomed and appreciated. If you use KDE, Eclipse, or something else, that's fine too. Some particular questions (in no particular order): 1. Do you regularly use the "alsasound" kernel function during development so you can take the driver in and out of the kernel? If it's used, is is used from the beginning or in the last stages of development? (I'm not sure if it's alsasound, sndconfig, etc). 2. Are the process statements given in the 'How-To' (snd_printk, snd_assert, snd_BUG) the main ones or are there others that are 'pre-built' and widely useful? http://www.alsa-project.org/~tiwai/writing-an-alsa-driver/c1696.htm 3. Are there tools that are almost always used when writing a sound card driver like gdb, ddd, etc? 4. Since it's easy to do, is an interface into the proc file system useful from the start? http://www.alsa-project.org/~tiwai/writing-an-alsa-driver/c1493.htm 5. Is the alsa wiki on debugging the sum total of what's needed or is there more we don't know about? (and we don't know that much!!) http://www.alsa-project.org/main/index.php/Driver_Compilation Of course, I could bring up the current Hot and Burning question we have (they are more important now): http://wavepro-driver.svn.sourceforge.net/viewvc/wavepro-driver/ 1. We've having difficulty getting the Altera PLD to load. Any suggestions on how to do this, how people without cards can help, an easy way to know when you are successful, or an incremental approach to the problem? 2. The GL card is powered from an Altera PLD (FPGA) (it has similar function as the Envy ICE chip once it's loaded). For the card to function at all, the Altera PLD has to be loaded when power is applied to the computer. Should the 'load the Altera' function be part of the alsa driver, part of alsa firmware, or should we just load it up when the computer boots (not in alsa)? 3. We're planning on using FC8/PlanetCCRMA as a 'beta test' environment. I've just noticed the new FC8 specific instructions on how to overcome my many bad habits when fooling around with the kernel. I'm putting it here for reference so I don't forget it (like I just did when trying to figure out if it was alsasound, procsound, or whatever to unload the driver). http://fedoraproject.org/wiki/Docs/CustomKernel If you would like to just put your two cents worth in to the very small mailing list (for the next 5 days), send a note to: wavepro-driver-developers(a)lists.sourceforge.net (in 5 days, you'll have to be subscribed to send a note) -Mike Mazarick

4 3

[alsa-devel] alsa-lib(1.0.12) crash
by Cliff Cai 23 Jan '08

23 Jan '08

Hi all, I'm experiencing weird crash of alsa-lib. the sound card has 16/24 bit ADC. The ADC works on 24 bit mode, hw->formats= SNDRV_PCM_FMTBIT_S32_LE,and use 'copy' playback to copy DMA buffer to user space's buffer. I ran: arecord -t wav -c 2 -f U8 -r 8000 -d 5 a.wav then arecord crash with following information: Data access misaligned address violation - Attempted misaligned data memory or data cache access. RETE: /* Maybe null pointer? */ RETN: /* unknown address */ RETX: [ /usr/lib/libasound.so.2 + 0x46da8 ] RETS: [ /usr/lib/libasound.so.2 + 0x46aec ] PC : [ /usr/lib/libasound.so.2 + 0x46da8 ] DCPLB_FAULT_ADDR: [ arecord + 0x2ac ] ICPLB_FAULT_ADDR: [ /usr/lib/libasound.so.2 + 0x46da8 ] Hardware Trace: 0 Target : { _trap_c + 0x0 } Source : { _exception_to_level5 + 0xb4 } 1 Target : { _exception_to_level5 + 0x0 } Source : { _ex_trap_c + 0x5c } 2 Target : { _ex_trap_c + 0x0 } Source : { _trap + 0x28 } 3 Target : { _trap + 0x0 } Source : [ /usr/lib/libasound.so.2 + 0x46da6 ] 4 Target : [ /usr/lib/libasound.so.2 + 0x46d9a ] Source : [ /usr/lib/libasound.so.2 + 0x46b54 ] 5 Target : [ /usr/lib/libasound.so.2 + 0x46b46 ] Source : [ /usr/lib/libasound.so.2 + 0x46b12 ] 6 Target : [ /usr/lib/libasound.so.2 + 0x46aec ] Source : [ /usr/lib/libasound.so.2 + 0x4d16a ] 7 Target : [ /usr/lib/libasound.so.2 + 0x4d166 ] Source : [ /usr/lib/libasound.so.2 + 0x4d17e ] 8 Target : [ /usr/lib/libasound.so.2 + 0x4d17c ] Source : [ /usr/lib/libasound.so.2 + 0x4d0b2 ] 9 Target : [ /usr/lib/libasound.so.2 + 0x4d09c ] Source : [ /usr/lib/libasound.so.2 + 0x278ec ] 10 Target : [ /usr/lib/libasound.so.2 + 0x278e4 ] Source : [ /usr/lib/libasound.so.2 + 0x46ae8 ] 11 Target : [ /usr/lib/libasound.so.2 + 0x46a74 ] Source : [ /usr/lib/libasound.so.2 + 0x28602 ] 12 Target : [ /usr/lib/libasound.so.2 + 0x285fa ] Source : [ /usr/lib/libasound.so.2 + 0x46eac ] 13 Target : [ /usr/lib/libasound.so.2 + 0x46e94 ] Source : [ /usr/lib/libasound.so.2 + 0x46e8a ] 14 Target : [ /usr/lib/libasound.so.2 + 0x46e88 ] Source : [ /usr/lib/libasound.so.2 + 0x46e38 ] 15 Target : [ /usr/lib/libasound.so.2 + 0x46e0e ] Source : [ /usr/lib/libasound.so.2 + 0x4d16a ] Then I modified the driver ,set the ADC to 16bit mode and set hw->formats= SNDRV_PCM_FMTBIT_S16_LE no crash any more. Any idea is appreciated. Thanks a lot in advance! Cliff _________________________________________________________________ 天凉了，添衣了，心动了，“七件”了 http://get.live.cn

3 2

[alsa-devel] [PATCH] ASoC documentation updates
by Mark Brown 23 Jan '08

23 Jan '08

Update the ASoC documentation. Along with minor formatting and grammar cleanups it moves the ASoC overview into the present tense to reflect the fact that it has now been merged. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- Documentation/sound/alsa/soc/DAI.txt | 6 +- Documentation/sound/alsa/soc/clocking.txt | 10 ++-- Documentation/sound/alsa/soc/codec.txt | 53 +++++++++++++------------- Documentation/sound/alsa/soc/dapm.txt | 51 +++++++++++++------------ Documentation/sound/alsa/soc/machine.txt | 12 +++--- Documentation/sound/alsa/soc/overview.txt | 42 +++++++++++---------- Documentation/sound/alsa/soc/platform.txt | 6 +- Documentation/sound/alsa/soc/pops_clicks.txt | 10 ++-- 8 files changed, 97 insertions(+), 93 deletions(-) diff --git a/Documentation/sound/alsa/soc/DAI.txt b/Documentation/sound/alsa/soc/DAI.txt index 3feeb9e..0ebd7ea 100644 --- a/Documentation/sound/alsa/soc/DAI.txt +++ b/Documentation/sound/alsa/soc/DAI.txt @@ -1,5 +1,5 @@ ASoC currently supports the three main Digital Audio Interfaces (DAI) found on -SoC controllers and portable audio CODECS today, namely AC97, I2S and PCM. +SoC controllers and portable audio CODECs today, namely AC97, I2S and PCM. AC97 @@ -25,7 +25,7 @@ left/right clock (LRC) synchronise the link. I2S is flexible in that either the controller or CODEC can drive (master) the BCLK and LRC clock lines. Bit clock usually varies depending on the sample rate and the master system clock (SYSCLK). LRCLK is the same as the sample rate. A few devices support separate -ADC and DAC LRCLK's, this allows for simultaneous capture and playback at +ADC and DAC LRCLKs, this allows for simultaneous capture and playback at different sample rates. I2S has several different operating modes:- @@ -35,7 +35,7 @@ I2S has several different operating modes:- o Left Justified - MSB is transmitted on transition of LRC. - o Right Justified - MSB is transmitted sample size BCLK's before LRC + o Right Justified - MSB is transmitted sample size BCLKs before LRC transition. PCM diff --git a/Documentation/sound/alsa/soc/clocking.txt b/Documentation/sound/alsa/soc/clocking.txt index 1493088..b130016 100644 --- a/Documentation/sound/alsa/soc/clocking.txt +++ b/Documentation/sound/alsa/soc/clocking.txt @@ -13,7 +13,7 @@ or SYSCLK). This audio master clock can be derived from a number of sources (e.g. crystal, PLL, CPU clock) and is responsible for producing the correct audio playback and capture sample rates. -Some master clocks (e.g. PLL's and CPU based clocks) are configurable in that +Some master clocks (e.g. PLLs and CPU based clocks) are configurable in that their speed can be altered by software (depending on the system use and to save power). Other master clocks are fixed at a set frequency (i.e. crystals). @@ -41,11 +41,11 @@ BCLK = LRC * x BCLK = LRC * Channels * Word Size This relationship depends on the codec or SoC CPU in particular. In general -it's best to configure BCLK to the lowest possible speed (depending on your -rate, number of channels and wordsize) to save on power. +it is best to configure BCLK to the lowest possible speed (depending on your +rate, number of channels and word size) to save on power. -It's also desirable to use the codec (if possible) to drive (or master) the -audio clocks as it's usually gives more accurate sample rates than the CPU. +It is also desirable to use the codec (if possible) to drive (or master) the +audio clocks as it usually gives more accurate sample rates than the CPU. diff --git a/Documentation/sound/alsa/soc/codec.txt b/Documentation/sound/alsa/soc/codec.txt index 1e766ad..1e95342 100644 --- a/Documentation/sound/alsa/soc/codec.txt +++ b/Documentation/sound/alsa/soc/codec.txt @@ -9,7 +9,7 @@ code should be added to the platform and machine drivers respectively. Each codec driver *must* provide the following features:- 1) Codec DAI and PCM configuration - 2) Codec control IO - using I2C, 3 Wire(SPI) or both API's + 2) Codec control IO - using I2C, 3 Wire(SPI) or both APIs 3) Mixers and audio controls 4) Codec audio operations @@ -19,7 +19,7 @@ Optionally, codec drivers can also provide:- 6) DAPM event handler. 7) DAC Digital mute control. -It's probably best to use this guide in conjunction with the existing codec +Its probably best to use this guide in conjunction with the existing codec driver code in sound/soc/codecs/ ASoC Codec driver breakdown @@ -27,8 +27,8 @@ ASoC Codec driver breakdown 1 - Codec DAI and PCM configuration ----------------------------------- -Each codec driver must have a struct snd_soc_codec_dai to define it's DAI and -PCM's capabilities and operations. This struct is exported so that it can be +Each codec driver must have a struct snd_soc_codec_dai to define its DAI and +PCM capabilities and operations. This struct is exported so that it can be registered with the core by your machine driver. e.g. @@ -67,18 +67,18 @@ EXPORT_SYMBOL_GPL(wm8731_dai); 2 - Codec control IO -------------------- -The codec can usually be controlled via an I2C or SPI style interface (AC97 -combines control with data in the DAI). The codec drivers will have to provide -functions to read and write the codec registers along with supplying a register -cache:- +The codec can usually be controlled via an I2C or SPI style interface +(AC97 combines control with data in the DAI). The codec drivers provide +functions to read and write the codec registers along with supplying a +register cache:- /* IO control data and register cache */ - void *control_data; /* codec control (i2c/3wire) data */ - void *reg_cache; + void *control_data; /* codec control (i2c/3wire) data */ + void *reg_cache; -Codec read/write should do any data formatting and call the hardware read write -below to perform the IO. These functions are called by the core and alsa when -performing DAPM or changing the mixer:- +Codec read/write should do any data formatting and call the hardware +read write below to perform the IO. These functions are called by the +core and ALSA when performing DAPM or changing the mixer:- unsigned int (*read)(struct snd_soc_codec *, unsigned int); int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); @@ -131,7 +131,7 @@ Defines a stereo enumerated control 4 - Codec Audio Operations -------------------------- -The codec driver also supports the following alsa operations:- +The codec driver also supports the following ALSA operations:- /* SoC audio ops */ struct snd_soc_ops { @@ -142,15 +142,15 @@ struct snd_soc_ops { int (*prepare)(struct snd_pcm_substream *); }; -Please refer to the alsa driver PCM documentation for details. +Please refer to the ALSA driver PCM documentation for details. http://www.alsa-project.org/~iwai/writing-an-alsa-driver/c436.htm 5 - DAPM description. --------------------- -The Dynamic Audio Power Management description describes the codec's power -components, their relationships and registers to the ASoC core. Please read -dapm.txt for details of building the description. +The Dynamic Audio Power Management description describes the codec power +components and their relationships and registers to the ASoC core. +Please read dapm.txt for details of building the description. Please also see the examples in other codec drivers. @@ -158,8 +158,8 @@ Please also see the examples in other codec drivers. 6 - DAPM event handler ---------------------- This function is a callback that handles codec domain PM calls and system -domain PM calls (e.g. suspend and resume). It's used to put the codec to sleep -when not in use. +domain PM calls (e.g. suspend and resume). It is used to put the codec +to sleep when not in use. Power states:- @@ -175,13 +175,14 @@ Power states:- SNDRV_CTL_POWER_D3cold: /* Everything Off, without power */ -7 - Codec DAC digital mute control. ------------------------------------- -Most codecs have a digital mute before the DAC's that can be used to minimise -any system noise. The mute stops any digital data from entering the DAC. +7 - Codec DAC digital mute control +---------------------------------- +Most codecs have a digital mute before the DACs that can be used to +minimise any system noise. The mute stops any digital data from +entering the DAC. -A callback can be created that is called by the core for each codec DAI when the -mute is applied or freed. +A callback can be created that is called by the core for each codec DAI +when the mute is applied or freed. i.e. diff --git a/Documentation/sound/alsa/soc/dapm.txt b/Documentation/sound/alsa/soc/dapm.txt index ab0766f..c784a18 100644 --- a/Documentation/sound/alsa/soc/dapm.txt +++ b/Documentation/sound/alsa/soc/dapm.txt @@ -4,20 +4,20 @@ Dynamic Audio Power Management for Portable Devices 1. Description ============== -Dynamic Audio Power Management (DAPM) is designed to allow portable Linux devices -to use the minimum amount of power within the audio subsystem at all times. It -is independent of other kernel PM and as such, can easily co-exist with the -other PM systems. +Dynamic Audio Power Management (DAPM) is designed to allow portable +Linux devices to use the minimum amount of power within the audio +subsystem at all times. It is independent of other kernel PM and as +such, can easily co-exist with the other PM systems. -DAPM is also completely transparent to all user space applications as all power -switching is done within the ASoC core. No code changes or recompiling are -required for user space applications. DAPM makes power switching decisions based -upon any audio stream (capture/playback) activity and audio mixer settings -within the device. +DAPM is also completely transparent to all user space applications as +all power switching is done within the ASoC core. No code changes or +recompiling are required for user space applications. DAPM makes power +switching decisions based upon any audio stream (capture/playback) +activity and audio mixer settings within the device. -DAPM spans the whole machine. It covers power control within the entire audio -subsystem, this includes internal codec power blocks and machine level power -systems. +DAPM spans the whole machine. It covers power control within the entire +audio subsystem, this includes internal codec power blocks and machine +level power systems. There are 4 power domains within DAPM @@ -34,7 +34,7 @@ There are 4 power domains within DAPM Automatically set when mixer and mux settings are changed by the user. e.g. alsamixer, amixer. - 4. Stream domain - DAC's and ADC's. + 4. Stream domain - DACs and ADCs. Enabled and disabled when stream playback/capture is started and stopped respectively. e.g. aplay, arecord. @@ -51,7 +51,7 @@ widgets hereafter. Audio DAPM widgets fall into a number of types:- o Mixer - Mixes several analog signals into a single analog signal. - o Mux - An analog switch that outputs only 1 of it's inputs. + o Mux - An analog switch that outputs only one of many inputs. o PGA - A programmable gain amplifier or attenuation widget. o ADC - Analog to Digital Converter o DAC - Digital to Analog Converter @@ -78,14 +78,14 @@ parameters for stream name and kcontrols. 2.1 Stream Domain Widgets ------------------------- -Stream Widgets relate to the stream power domain and only consist of ADC's -(analog to digital converters) and DAC's (digital to analog converters). +Stream Widgets relate to the stream power domain and only consist of ADCs +(analog to digital converters) and DACs (digital to analog converters). Stream widgets have the following format:- SND_SOC_DAPM_DAC(name, stream name, reg, shift, invert), -NOTE: the stream name must match the corresponding stream name in your codecs +NOTE: the stream name must match the corresponding stream name in your codec snd_soc_codec_dai. e.g. stream widgets for HiFi playback and capture @@ -97,7 +97,7 @@ SND_SOC_DAPM_ADC("HiFi ADC", "HiFi Capture", REG, 2, 1), 2.2 Path Domain Widgets ----------------------- -Path domain widgets have a ability to control or effect the audio signal or +Path domain widgets have a ability to control or affect the audio signal or audio paths within the audio subsystem. They have the following form:- SND_SOC_DAPM_PGA(name, reg, shift, invert, controls, num_controls) @@ -149,7 +149,7 @@ SND_SOC_DAPM_MIC("Mic Jack", spitz_mic_bias), 2.4 Codec Domain ---------------- -The Codec power domain has no widgets and is handled by the codecs DAPM event +The codec power domain has no widgets and is handled by the codecs DAPM event handler. This handler is called when the codec powerstate is changed wrt to any stream event or by kernel PM events. @@ -158,8 +158,8 @@ stream event or by kernel PM events. ------------------- Sometimes widgets exist in the codec or machine audio map that don't have any -corresponding register bit for power control. In this case it's necessary to -create a virtual widget - a widget with no control bits e.g. +corresponding soft power control. In this case it is necessary to create +a virtual widget - a widget with no control bits e.g. SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_DAPM_NOPM, 0, 0, NULL, 0), @@ -172,13 +172,14 @@ subsystem individually with a call to snd_soc_dapm_new_control(). 3. Codec Widget Interconnections ================================ -Widgets are connected to each other within the codec and machine by audio -paths (called interconnections). Each interconnection must be defined in order -to create a map of all audio paths between widgets. +Widgets are connected to each other within the codec and machine by audio paths +(called interconnections). Each interconnection must be defined in order to +create a map of all audio paths between widgets. + This is easiest with a diagram of the codec (and schematic of the machine audio system), as it requires joining widgets together via their audio signal paths. -i.e. from the WM8731 codec's output mixer (wm8731.c) +e.g., from the WM8731 output mixer (wm8731.c) The WM8731 output mixer has 3 inputs (sources) diff --git a/Documentation/sound/alsa/soc/machine.txt b/Documentation/sound/alsa/soc/machine.txt index 72bd222..f370e7d 100644 --- a/Documentation/sound/alsa/soc/machine.txt +++ b/Documentation/sound/alsa/soc/machine.txt @@ -16,7 +16,7 @@ struct snd_soc_machine { int (*remove)(struct platform_device *pdev); /* the pre and post PM functions are used to do any PM work before and - * after the codec and DAI's do any PM work. */ + * after the codec and DAIs do any PM work. */ int (*suspend_pre)(struct platform_device *pdev, pm_message_t state); int (*suspend_post)(struct platform_device *pdev, pm_message_t state); int (*resume_pre)(struct platform_device *pdev); @@ -38,7 +38,7 @@ probe/remove are optional. Do any machine specific probe here. suspend()/resume() ------------------ The machine driver has pre and post versions of suspend and resume to take care -of any machine audio tasks that have to be done before or after the codec, DAI's +of any machine audio tasks that have to be done before or after the codec, DAIs and DMA is suspended and resumed. Optional. @@ -49,10 +49,10 @@ The machine specific audio operations can be set here. Again this is optional. Machine DAI Configuration ------------------------- -The machine DAI configuration glues all the codec and CPU DAI's together. It can +The machine DAI configuration glues all the codec and CPU DAIs together. It can also be used to set up the DAI system clock and for any machine related DAI initialisation e.g. the machine audio map can be connected to the codec audio -map, unconnnected codec pins can be set as such. Please see corgi.c, spitz.c +map, unconnected codec pins can be set as such. Please see corgi.c, spitz.c for examples. struct snd_soc_dai_link is used to set up each DAI in your machine. e.g. @@ -67,7 +67,7 @@ static struct snd_soc_dai_link corgi_dai = { .ops = &corgi_ops, }; -struct snd_soc_machine then sets up the machine with it's DAI's. e.g. +struct snd_soc_machine then sets up the machine with it's DAIs. e.g. /* corgi audio machine driver */ static struct snd_soc_machine snd_soc_machine_corgi = { @@ -110,4 +110,4 @@ details. Machine Controls ---------------- -Machine specific audio mixer controls can be added in the dai init function. \ No newline at end of file +Machine specific audio mixer controls can be added in the DAI init function. diff --git a/Documentation/sound/alsa/soc/overview.txt b/Documentation/sound/alsa/soc/overview.txt index c47ce95..1e4c6d3 100644 --- a/Documentation/sound/alsa/soc/overview.txt +++ b/Documentation/sound/alsa/soc/overview.txt @@ -1,25 +1,26 @@ ALSA SoC Layer ============== -The overall project goal of the ALSA System on Chip (ASoC) layer is to provide -better ALSA support for embedded system-on-chip processors (e.g. pxa2xx, au1x00, -iMX, etc) and portable audio codecs. Currently there is some support in the -kernel for SoC audio, however it has some limitations:- +The overall project goal of the ALSA System on Chip (ASoC) layer is to +provide better ALSA support for embedded system-on-chip processors (e.g. +pxa2xx, au1x00, iMX, etc) and portable audio codecs. Prior to the ASoC +subsystem there was some support in the kernel for SoC audio, however it +had some limitations:- - * Currently, codec drivers are often tightly coupled to the underlying SoC - CPU. This is not ideal and leads to code duplication i.e. Linux now has 4 - different wm8731 drivers for 4 different SoC platforms. + * Codec drivers were often tightly coupled to the underlying SoC + CPU. This is not ideal and leads to code duplication - for example, + Linux had different wm8731 drivers for 4 different SoC platforms. - * There is no standard method to signal user initiated audio events (e.g. + * There was no standard method to signal user initiated audio events (e.g. Headphone/Mic insertion, Headphone/Mic detection after an insertion event). These are quite common events on portable devices and often require machine specific code to re-route audio, enable amps, etc., after such an event. - * Current drivers tend to power up the entire codec when playing - (or recording) audio. This is fine for a PC, but tends to waste a lot of - power on portable devices. There is also no support for saving power via - changing codec oversampling rates, bias currents, etc. + * Drivers tended to power up the entire codec when playing (or + recording) audio. This is fine for a PC, but tends to waste a lot of + power on portable devices. There was also no support for saving + power via changing codec oversampling rates, bias currents, etc. ASoC Design @@ -31,12 +32,13 @@ features :- * Codec independence. Allows reuse of codec drivers on other platforms and machines. - * Easy I2S/PCM audio interface setup between codec and SoC. Each SoC interface - and codec registers it's audio interface capabilities with the core and are - subsequently matched and configured when the application hw params are known. + * Easy I2S/PCM audio interface setup between codec and SoC. Each SoC + interface and codec registers it's audio interface capabilities with the + core and are subsequently matched and configured when the application + hardware parameters are known. * Dynamic Audio Power Management (DAPM). DAPM automatically sets the codec to - it's minimum power state at all times. This includes powering up/down + its minimum power state at all times. This includes powering up/down internal power blocks depending on the internal codec audio routing and any active streams. @@ -45,16 +47,16 @@ features :- signals the codec when to change power states. * Machine specific controls: Allow machines to add controls to the sound card - (e.g. volume control for speaker amp). + (e.g. volume control for speaker amplifier). To achieve all this, ASoC basically splits an embedded audio system into 3 components :- * Codec driver: The codec driver is platform independent and contains audio - controls, audio interface capabilities, codec dapm definition and codec IO + controls, audio interface capabilities, codec DAPM definition and codec IO functions. - * Platform driver: The platform driver contains the audio dma engine and audio + * Platform driver: The platform driver contains the audio DMA engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. * Machine driver: The machine driver handles any machine specific controls and @@ -81,4 +83,4 @@ machine.txt: Machine driver internals. pop_clicks.txt: How to minimise audio artifacts. -clocking.txt: ASoC clocking for best power performance. \ No newline at end of file +clocking.txt: ASoC clocking for best power performance. diff --git a/Documentation/sound/alsa/soc/platform.txt b/Documentation/sound/alsa/soc/platform.txt index d4678b4..b681d17 100644 --- a/Documentation/sound/alsa/soc/platform.txt +++ b/Documentation/sound/alsa/soc/platform.txt @@ -8,7 +8,7 @@ specific code. Audio DMA ========= -The platform DMA driver optionally supports the following alsa operations:- +The platform DMA driver optionally supports the following ALSA operations:- /* SoC audio ops */ struct snd_soc_ops { @@ -38,7 +38,7 @@ struct snd_soc_platform { struct snd_pcm_ops *pcm_ops; }; -Please refer to the alsa driver documentation for details of audio DMA. +Please refer to the ALSA driver documentation for details of audio DMA. http://www.alsa-project.org/~iwai/writing-an-alsa-driver/c436.htm An example DMA driver is soc/pxa/pxa2xx-pcm.c @@ -52,7 +52,7 @@ Each SoC DAI driver must provide the following features:- 1) Digital audio interface (DAI) description 2) Digital audio interface configuration 3) PCM's description - 4) Sysclk configuration + 4) SYSCLK configuration 5) Suspend and resume (optional) Please see codec.txt for a description of items 1 - 4. diff --git a/Documentation/sound/alsa/soc/pops_clicks.txt b/Documentation/sound/alsa/soc/pops_clicks.txt index 3371bd9..e1e74da 100644 --- a/Documentation/sound/alsa/soc/pops_clicks.txt +++ b/Documentation/sound/alsa/soc/pops_clicks.txt @@ -15,11 +15,11 @@ click every time a component power state is changed. Minimising Playback Pops and Clicks =================================== -Playback pops in portable audio subsystems cannot be completely eliminated atm, -however future audio codec hardware will have better pop and click suppression. -Pops can be reduced within playback by powering the audio components in a -specific order. This order is different for startup and shutdown and follows -some basic rules:- +Playback pops in portable audio subsystems cannot be completely eliminated +currently, however future audio codec hardware will have better pop and click +suppression. Pops can be reduced within playback by powering the audio +components in a specific order. This order is different for startup and +shutdown and follows some basic rules:- Startup Order :- DAC --> Mixers --> Output PGA --> Digital Unmute -- 1.5.3.8

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[alsa-devel] how do you people debug ALSA?
by Mihaela Vitalariu 22 Jan '08

22 Jan '08

Hello everyone! How do you ALSA developers debug the ALSA code? I am working on a large Linux project and i am using gdb which is not very helpful sometimes. I also use KDevelop to do graphical debugging(its' debugger is a frontend for gdb too), but it's not so good for multithreaded programs. If you have any hints please reply. Have a nice day!

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Re: [alsa-devel] [alsa-cvslog] alsa-lib: Remove indirect control access
by Thierry Vignaud 22 Jan '08

22 Jan '08

Takashi Iwai <tiwai(a)suse.de> writes: > changeset: 2374:d0e6c6e079eb > user: tiwai > date: Tue Jan 08 18:34:36 2008 +0100 > files: include/sound/asound.h src/control/control.c > description: > Remove indirect control access > > The indirect control access is removed from the kernel. > This patch cleans the corresponding alsa-lib part. Shouldn't this be kept for compatibility with older kernels? Eg: if someone update distro foobar from version X to version Y, he usually can boot either the old kernel from X or the new one from Y. With such a change, booting the older kernel would result in missing features, wouldn't it?

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[alsa-devel] [PATCH 2/2] hda: Mono mux mixer support
by Matthew Ranostay 22 Jan '08

22 Jan '08

Add support for the mono mux on several 92HD7xxx codecs. Creates a dynamic mixer for the mux selection. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff -r a035d5e4801c pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Fri Jan 18 15:32:32 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Mon Jan 21 11:58:25 2008 -0500 @@ -131,6 +131,8 @@ struct sigmatel_spec { hda_nid_t *pwr_nids; /* playback */ + struct hda_input_mux *mono_mux; + unsigned int cur_mmux; struct hda_multi_out multiout; hda_nid_t dac_nids[5]; @@ -144,6 +146,7 @@ struct sigmatel_spec { hda_nid_t *dmux_nids; unsigned int num_dmuxes; hda_nid_t dig_in_nid; + hda_nid_t mono_nid; /* pin widgets */ hda_nid_t *pin_nids; @@ -174,6 +177,7 @@ struct sigmatel_spec { struct snd_kcontrol_new *kctl_alloc; struct hda_input_mux private_dimux; struct hda_input_mux private_imux; + struct hda_input_mux private_mono_mux; /* virtual master */ unsigned int vmaster_tlv[4]; @@ -399,6 +403,34 @@ static int stac92xx_mux_enum_put(struct return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, spec->mux_nids[adc_idx], &spec->cur_mux[adc_idx]); +} + +static int stac92xx_mono_mux_enum_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + return snd_hda_input_mux_info(spec->mono_mux, uinfo); +} + +static int stac92xx_mono_mux_enum_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + + ucontrol->value.enumerated.item[0] = spec->cur_mmux; + return 0; +} + +static int stac92xx_mono_mux_enum_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + + return snd_hda_input_mux_put(codec, spec->mono_mux, ucontrol, + spec->mono_nid, &spec->cur_mmux); } #define stac92xx_aloopback_info snd_ctl_boolean_mono_info @@ -603,6 +635,16 @@ static struct hda_verb stac9205_core_ini {} }; +#define STAC_MONO_MUX \ + { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = "Mono Mux", \ + .count = 1, \ + .info = stac92xx_mono_mux_enum_info, \ + .get = stac92xx_mono_mux_enum_get, \ + .put = stac92xx_mono_mux_enum_put, \ + } + #define STAC_INPUT_SOURCE(cnt) \ { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ @@ -714,6 +756,7 @@ static struct snd_kcontrol_new stac92hd7 static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { STAC_INPUT_SOURCE(2), + STAC_MONO_MUX, HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), @@ -733,6 +776,7 @@ static struct snd_kcontrol_new stac92hd7 static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { STAC_INPUT_SOURCE(2), STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2), + STAC_MONO_MUX, HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), @@ -1180,7 +1224,7 @@ static struct snd_pci_quirk stac92hd73xx static unsigned int ref92hd71bxx_pin_configs[10] = { 0x02214030, 0x02a19040, 0x01a19020, 0x01014010, - 0x0181302e, 0x01114010, 0x01a19020, 0x90a000f0, + 0x0181302e, 0x01114010, 0x01019020, 0x90a000f0, 0x90a000f0, 0x01452050, }; @@ -2318,6 +2362,35 @@ static int stac92xx_auto_create_hp_ctls( return 0; } +/* labels for mono mux outputs */ +static const char *stac92xx_mono_labels[3] = { + "DAC0", "DAC1", "Mixer" +}; + +/* create mono mux for mono out on capable codecs */ +static int stac92xx_auto_create_mono_output_ctls(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + struct hda_input_mux *mono_mux = &spec->private_mono_mux; + int i, num_cons; + hda_nid_t con_lst[HDA_MAX_NUM_INPUTS]; + + num_cons = snd_hda_get_connections(codec, + spec->mono_nid, + con_lst, + HDA_MAX_NUM_INPUTS); + if (!num_cons) + return -EINVAL; + + for (i = 0; i < num_cons; i++) { + mono_mux->items[mono_mux->num_items].label = + stac92xx_mono_labels[i]; + mono_mux->items[mono_mux->num_items].index = i; + mono_mux->num_items++; + } + return 0; +} + /* labels for dmic mux inputs */ static const char *stac92xx_dmic_labels[5] = { "Analog Inputs", "Digital Mic 1", "Digital Mic 2", @@ -2532,6 +2605,12 @@ static int stac92xx_parse_auto_config(st if (err < 0) return err; + if (spec->mono_nid > 0) { + err = stac92xx_auto_create_mono_output_ctls(codec); + if (err < 0) + return err; + } + if (spec->num_dmics > 0) if ((err = stac92xx_auto_create_dmic_input_ctls(codec, &spec->autocfg)) < 0) @@ -2552,6 +2631,7 @@ static int stac92xx_parse_auto_config(st spec->input_mux = &spec->private_imux; if (!spec->dinput_mux) spec->dinput_mux = &spec->private_dimux; + spec->mono_mux = &spec->private_mono_mux; return 1; } @@ -3237,6 +3317,7 @@ again: spec->gpio_mask = spec->gpio_data = 0x00000001; /* GPIO0 High = EAPD */ + spec->mono_nid = 0x15; spec->mux_nids = stac92hd71bxx_mux_nids; spec->adc_nids = stac92hd71bxx_adc_nids; spec->dmic_nids = stac92hd71bxx_dmic_nids;

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Re: [alsa-devel] conversion of opti93x driver to use cs4231-lib
by krzysztof.h1＠poczta.fm 22 Jan '08

22 Jan '08

> On 08-01-08 15:36, Rene Herman wrote: > > >> Missed this one originally. I'll look into this in some detail but > >> will need a few days. Obviously, if it works, it's desirable. Maybe > >> Jaroslav knows why it's not that way now? > > > > Oh, and by the way, I suppose it'll be trivial but a rediff would be > > good for testing. > I'll do a new diff on weekend. Thanks, Krzysztof PS. I tested it on 931 and 933. ---------------------------------------------------------------------- Sprawdz, ktore komorki sa najmodniejsze! Kliknij >>> http://link.interia.pl/f1cd4

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[alsa-devel] Support for Compaq Presario F700 - model 755US - NVidia MCP67 Audio - Codec Conexant 5051
by Philippe Paravicini 21 Jan '08

21 Jan '08

I recently purchased the laptop listed above and installed kubuntu-7.10-desktop-amd64 Feisty on it. The sound card works, but, much to my chagrin, plugging in the headphone jack does not mute the speaker. <wining> This is less than ideal for me because I use Skype quite a bit, and work in a no-walls small office with other people. Having the ability to use my headset for talking on the phone and drowning out ambient noise is important. </wining><deepbreath/> I have googled high and low and have tried various suggested solutions, including changing: options snd-hda-intel model=laptop in /etc/modprobe.d/alsa-base to various settings (laptop-hp, laptop-eahd, test). I have also downloaded, compiled and installed the 1.0.15 driver and utils according to the instructions at: https://help.ubuntu.com/community/HdaIntelSoundHowto without success. I consulted the page at: http://www.mjmwired.net/kernel/Documentation/sound/alsa/ALSA-Configuration.… dated 2007-10-11 and could note see support for the Conexant 5051 Codec or NVidia MCP67 Audio. Has anyone had experience with this soundcard/codec and gotten the headphones jack to mute the speakers correctly ? Any help would be greatly appreciated. I am a java programmer with very high reverence for linux C-programmers and would be happy to help test a patch for this device (I may need a little help to get going), or trade for java programming/consulting. Please let me know if you additional information other than the one listed below. Thank you kindly for any help. Sincerely, Philippe Paravicini lspci -v outputs: ... 00:07.0 Audio device: nVidia Corporation MCP67 High Definition Audio (rev a1) Subsystem: Hewlett-Packard Company Unknown device 30ea ... lspci -vn ... 00:07.0 0403: 10de:055c (rev a1) Subsystem: 103c:30ea Flags: bus master, 66MHz, fast devsel, latency 0, IRQ 21 Memory at f6480000 (32-bit, non-prefetchable) [size=16K] Capabilities: [44] Power Management version 2 Capabilities: [50] Message Signalled Interrupts: Mask+ 64bit+ Queue=0/0 Enable- Capabilities: [6c] HyperTransport: MSI Mapping ... cat /proc/asound/card0/codec#0 Codec: Conexant ID 5051 Address: 0 Vendor Id: 0x14f15051 Subsystem Id: 0x103c30ea Revision Id: 0x100000 Modem Function Group: 0x2 Default PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A Node 0x10 [Audio Output] wcaps 0xc1d: Stereo Amp-Out Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=0 Amp-Out vals: [0x3a 0x3a] PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: 0x0 Node 0x11 [Audio Output] wcaps 0xc1d: Stereo Amp-Out Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=0 Amp-Out vals: [0x4a 0x4a] PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: 0x0 Node 0x12 [Audio Output] wcaps 0x211: Stereo Digital PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Node 0x13 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x03, nsteps=0x03, stepsize=0x17, mute=0 Amp-Out vals: [0x00] Node 0x14 [Audio Input] wcaps 0x100d1b: Stereo Amp-In Amp-In caps: ofs=0x4a, nsteps=0x50, stepsize=0x03, mute=0 Amp-In vals: [0x06 0x06] [0x35 0x35] PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: 0x0 Connection: 2 0x1d 0x17* Node 0x15 [Audio Input] wcaps 0x100d1b: Stereo Amp-In Amp-In caps: ofs=0x4a, nsteps=0x50, stepsize=0x03, mute=0 Amp-In vals: [0x4a 0x4a] PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: 0x0 Connection: 1 0x18 Node 0x16 [Pin Complex] wcaps 0x400581: Stereo Pincap 0x081c: OUT HP Detect Pin Default 0x02211030: [Jack] HP Out at Ext Front Conn = 1/8, Color = Black Pin-ctls: 0xc0: OUT HP Power: 0x0 Connection: 2 0x10* 0x11 Node 0x17 [Pin Complex] wcaps 0x40048b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-In vals: Pincap 0x081224: IN Detect Pin Default 0x02a19020: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink Pin-ctls: 0x24: IN Power: 0x0 Node 0x18 [Pin Complex] wcaps 0x40048b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-In vals: Pincap 0x081224: IN Detect Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x00: Power: 0x0 Node 0x19 [Pin Complex] wcaps 0x400581: Stereo Pincap 0x0814: OUT Detect Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x00: Power: 0x0 Connection: 2 0x10 0x11* Node 0x1a [Pin Complex] wcaps 0x400501: Stereo Pincap 0x0810010: OUT EAPD Pin Default 0x95170110: [Fixed] Speaker at Int Top Conn = Analog, Color = Unknown Pin-ctls: 0x40: OUT Power: 0x0 Connection: 2 0x10* 0x11 Node 0x1b [Pin Complex] wcaps 0x400500: Mono Pincap 0x0810010: OUT EAPD Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x40: OUT Power: 0x0 Connection: 2 0x10* 0x11 Node 0x1c [Pin Complex] wcaps 0x400701: Stereo Digital Pincap 0x0810: OUT Pin Default 0x224511f0: [Jack] SPDIF Out at Sep Front Conn = Optical, Color = Black Pin-ctls: 0x00: Power: 0x0 Connection: 1 0x12 Node 0x1d [Pin Complex] wcaps 0x40040b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-In vals: Pincap 0x0820: IN Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x00: Power: 0x0 Node 0x1e [Vendor Defined Widget] wcaps 0xf00000: Mono

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[alsa-devel] Patch for ALC882
by Kip Warner 21 Jan '08

21 Jan '08

So after many months of wrestling with Realtek's technical support, I finally managed to get all four of my speakers to work in my ASUS A7J laptop. The patch was intended to replace alsa-kernel/pci/hda/patch_realtek.c in 1.0.14rc4 and that was the last version I successfully tested it on. Since then, I haven't managed to get all four speaker to work. I am not sure why Realtek never posted this patch, but I figured that perhaps some of the ALSA brains might be able to make sense of it and integrate properly into HEAD. -- Kip Warner -- Software Engineer OpenPGP encrypted/signed mail preferred http://www.thevertigo.com

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[alsa-devel] [PATCH 1/2] hda: STAC92xx mono mixer controls
by Matthew Ranostay 21 Jan '08

21 Jan '08

Added mono mixer volume/switch controls for STAC9205 and STAC925x codecs. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff -r 4177ec41bbef pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Mon Jan 21 14:33:37 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Mon Jan 21 11:37:49 2008 -0500 @@ -572,6 +572,7 @@ static struct hda_verb stac925x_core_ini static struct hda_verb stac925x_core_init[] = { /* set dac0mux for dac converter */ { 0x06, AC_VERB_SET_CONNECT_SEL, 0x00}, + { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {} }; @@ -600,6 +601,7 @@ static struct hda_verb stac9205_core_ini static struct hda_verb stac9205_core_init[] = { /* set master volume and direct control */ { 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, + { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {} }; @@ -748,6 +750,9 @@ static struct snd_kcontrol_new stac92hd7 static struct snd_kcontrol_new stac925x_mixer[] = { STAC_INPUT_SOURCE(1), + HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x10, 0x1, 0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x10, 0x1, 0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x09, 0, HDA_OUTPUT), HDA_CODEC_VOLUME("Capture Mux Volume", 0x0f, 0, HDA_OUTPUT), @@ -757,6 +762,9 @@ static struct snd_kcontrol_new stac9205_ static struct snd_kcontrol_new stac9205_mixer[] = { STAC_INPUT_SOURCE(2), STAC_ANALOG_LOOPBACK(0xFE0, 0x7E0, 1), + + HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x14, 0x1, 0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x14, 0x1, 0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1b, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1d, 0x0, HDA_OUTPUT),

2 2

[alsa-devel] [PATCH 01/17] Update MAINTAINERS for ALSA SoC
by Mark Brown 21 Jan '08

21 Jan '08

Add myself as a point of contact for the ALSA SoC subsystem and add a reference to the development GIT tree. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> --- MAINTAINERS | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index b4f611c..10c3af3 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -3514,6 +3514,9 @@ S: Maintained SOUND - SOC LAYER / DYNAMIC AUDIO POWER MANAGEMENT P: Liam Girdwood M: liam.girdwood(a)wolfsonmicro.com +P: Mark Brown +M: broonie(a)opensource.wolfsonmicro.com +T: git opensource.wolfsonmicro.com/linux-2.6-asoc L: alsa-devel(a)alsa-project.org (subscribers-only) S: Supported -- 1.5.3.8

4 24

[alsa-devel] [PATCH] hda: Added mono mux mixer for some STAC92HD7XXX family codecs
by Matthew Ranostay 21 Jan '08

21 Jan '08

Some 92HD7XXX family codecs have a mono mux, added support for selecting the input path to the mono out. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff -r a035d5e4801c pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Fri Jan 18 15:32:32 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Fri Jan 18 14:53:36 2008 -0500 @@ -131,6 +131,8 @@ struct sigmatel_spec { hda_nid_t *pwr_nids; /* playback */ + struct hda_input_mux *mono_mux; + unsigned int cur_mmux; struct hda_multi_out multiout; hda_nid_t dac_nids[5]; @@ -144,6 +146,7 @@ struct sigmatel_spec { hda_nid_t *dmux_nids; unsigned int num_dmuxes; hda_nid_t dig_in_nid; + hda_nid_t mono_nid; /* pin widgets */ hda_nid_t *pin_nids; @@ -399,6 +402,31 @@ static int stac92xx_mux_enum_put(struct return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, spec->mux_nids[adc_idx], &spec->cur_mux[adc_idx]); +} + +static int stac92xx_mono_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + return snd_hda_input_mux_info(spec->mono_mux, uinfo); +} + +static int stac92xx_mono_mux_enum_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + + ucontrol->value.enumerated.item[0] = spec->cur_mmux; + return 0; +} + +static int stac92xx_mono_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + + return snd_hda_input_mux_put(codec, spec->mono_mux, ucontrol, + spec->mono_nid, &spec->cur_mmux); } #define stac92xx_aloopback_info snd_ctl_boolean_mono_info @@ -603,6 +631,16 @@ static struct hda_verb stac9205_core_ini {} }; +#define STAC_MONO_MUX \ + { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = "Mono Mux", \ + .count = 1, \ + .info = stac92xx_mono_mux_enum_info, \ + .get = stac92xx_mono_mux_enum_get, \ + .put = stac92xx_mono_mux_enum_put, \ + } + #define STAC_INPUT_SOURCE(cnt) \ { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ @@ -714,6 +752,7 @@ static struct snd_kcontrol_new stac92hd7 static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { STAC_INPUT_SOURCE(2), + STAC_MONO_MUX, HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), @@ -733,6 +772,7 @@ static struct snd_kcontrol_new stac92hd7 static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { STAC_INPUT_SOURCE(2), STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2), + STAC_MONO_MUX, HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), @@ -1180,7 +1220,7 @@ static struct snd_pci_quirk stac92hd73xx static unsigned int ref92hd71bxx_pin_configs[10] = { 0x02214030, 0x02a19040, 0x01a19020, 0x01014010, - 0x0181302e, 0x01114010, 0x01a19020, 0x90a000f0, + 0x0181302e, 0x01114010, 0x01019020, 0x90a000f0, 0x90a000f0, 0x01452050, }; @@ -3188,6 +3228,16 @@ again: return 0; } +static struct hda_input_mux stac92hd71bxx_mono_mux = { + /* count set later */ + .items = { + { "DAC0", 0x0 }, + { "DAC1", 0x1 }, + /* not an option on all cards */ + { "Mixer", 0x2 }, + }, +}; + static int patch_stac92hd71bxx(struct hda_codec *codec) { struct sigmatel_spec *spec; @@ -3224,10 +3274,12 @@ again: case 0x111d76b7: case 0x111d76b4: /* 6 Port without Analog Mixer */ case 0x111d76b5: + stac92hd71bxx_mono_mux.num_items = 2, spec->mixer = stac92hd71bxx_mixer; spec->init = stac92hd71bxx_core_init; break; default: + stac92hd71bxx_mono_mux.num_items = 3, spec->mixer = stac92hd71bxx_analog_mixer; spec->init = stac92hd71bxx_analog_core_init; } @@ -3237,6 +3289,8 @@ again: spec->gpio_mask = spec->gpio_data = 0x00000001; /* GPIO0 High = EAPD */ + spec->mono_mux = &stac92hd71bxx_mono_mux; + spec->mono_nid = 0x15; spec->mux_nids = stac92hd71bxx_mux_nids; spec->adc_nids = stac92hd71bxx_adc_nids; spec->dmic_nids = stac92hd71bxx_dmic_nids;

2 7

[alsa-devel] Building for 2.4 kernel..
by marcus hall 21 Jan '08

21 Jan '08

I support a product that is running the 2.4.18 linux kernel, and was trying to install alsa-driver-1.0.15, and ran into a problem. The alsa-kernel/pci/cs5535audio/cs5535audio_pm.c contains: if (pci_save_state(pci)) { ... and a similar line for pci_restore_state(). But, in include/adriver.h, pci_save_state is #defined to snd_pci_compat_save_state, which is declared to return void, so the compiler complains that the void return isn't being ignored. Now, pci_save_state() appears to be supposed to return status, although in the 2.4 kernel at least, it returns a constant 0. Nevertheless, to get things to work, I changed snd_pci_compat_save_state(), snd_pci_orig_save_state(), and the corresponding restore functions to return an int with the following patch. (The other obvious solution would be to remove the test in cs5535audio_pm.c, but pci_save_state() is *supposed* to return an int...) -- Marcus Hall marcus(a)tuells.org --------- Index: alsa-driver/acore/misc_driver.c =================================================================== RCS file: /usr/local/cvs_repos/almond/alsa/alsa-driver/acore/misc_driver.c,v retrieving revision 1.1 retrieving revision 1.2 diff -u -r1.1 -r1.2 --- alsa-driver/acore/misc_driver.c 20 Jan 2008 16:52:35 -0000 1.1 +++ alsa-driver/acore/misc_driver.c 20 Jan 2008 17:14:23 -0000 1.2 @@ -105,14 +105,14 @@ #ifdef CONFIG_PCI #ifndef CONFIG_HAVE_NEW_PCI_SAVE_STATE #ifdef CONFIG_HAVE_PCI_SAVED_CONFIG -void snd_pci_compat_save_state(struct pci_dev *pci) +int snd_pci_compat_save_state(struct pci_dev *pci) { - snd_pci_orig_save_state(pci, pci->saved_config_space); + return snd_pci_orig_save_state(pci, pci->saved_config_space); } EXPORT_SYMBOL(snd_pci_compat_save_state); -void snd_pci_compat_restore_state(struct pci_dev *pci) +int snd_pci_compat_restore_state(struct pci_dev *pci) { - snd_pci_orig_restore_state(pci, pci->saved_config_space); + return snd_pci_orig_restore_state(pci, pci->saved_config_space); } EXPORT_SYMBOL(snd_pci_compat_restore_state); #else /* !CONFIG_HAVE_PCI_SAVED_CONFIG */ @@ -122,7 +122,7 @@ }; static struct saved_config_tbl saved_tbl[16]; -void snd_pci_compat_save_state(struct pci_dev *pci) +int snd_pci_compat_save_state(struct pci_dev *pci) { int i; /* FIXME: mutex needed for race? */ @@ -130,14 +130,15 @@ if (! saved_tbl[i].pci) { saved_tbl[i].pci = pci; snd_pci_orig_save_state(pci, saved_tbl[i].config); - return; + return 0; } } printk(KERN_DEBUG "snd: no pci config space found!\n"); + return 1; } EXPORT_SYMBOL(snd_pci_compat_save_state); -void snd_pci_compat_restore_state(struct pci_dev *pci) +int snd_pci_compat_restore_state(struct pci_dev *pci) { int i; /* FIXME: mutex needed for race? */ @@ -145,10 +146,11 @@ if (saved_tbl[i].pci == pci) { saved_tbl[i].pci = NULL; snd_pci_orig_restore_state(pci, saved_tbl[i].config); - return; + return 0; } } printk(KERN_DEBUG "snd: no saved pci config!\n"); + return 1; } EXPORT_SYMBOL(snd_pci_compat_restore_state); #endif /* CONFIG_HAVE_PCI_SAVED_CONFIG */ Index: alsa-driver/include/adriver.h =================================================================== RCS file: /usr/local/cvs_repos/almond/alsa/alsa-driver/include/adriver.h,v retrieving revision 1.1 retrieving revision 1.2 diff -u -r1.1 -r1.2 --- alsa-driver/include/adriver.h 20 Jan 2008 16:52:37 -0000 1.1 +++ alsa-driver/include/adriver.h 20 Jan 2008 17:14:23 -0000 1.2 @@ -1165,15 +1165,15 @@ #if LINUX_VERSION_CODE >= KERNEL_VERSION(2, 4, 0) #ifdef CONFIG_PCI #ifndef CONFIG_HAVE_NEW_PCI_SAVE_STATE -void snd_pci_compat_save_state(struct pci_dev *pci); -void snd_pci_compat_restore_state(struct pci_dev *pci); -static inline void snd_pci_orig_save_state(struct pci_dev *pci, u32 *buffer) +int snd_pci_compat_save_state(struct pci_dev *pci); +int snd_pci_compat_restore_state(struct pci_dev *pci); +static inline int snd_pci_orig_save_state(struct pci_dev *pci, u32 *buffer) { - pci_save_state(pci, buffer); + return pci_save_state(pci, buffer); } -static inline void snd_pci_orig_restore_state(struct pci_dev *pci, u32 *buffer) +static inline int snd_pci_orig_restore_state(struct pci_dev *pci, u32 *buffer) { - pci_restore_state(pci, buffer); + return pci_restore_state(pci, buffer); } #undef pci_save_state

2 1

[alsa-devel] [PATCH] Fix lockdep warning in ASoC machine probe
by Mark Brown 21 Jan '08

21 Jan '08

Don't take the codec mutex during machine probe until we have registered with ALSA, fixing a lockdep warning reported by Dmitry Baryshkov. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Dmitry Baryshkov <dbaryshkov(a)gmail.com> --- sound/soc/soc-core.c | 7 +++++-- 1 files changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2eb9b20..78d1662 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1098,7 +1098,6 @@ int snd_soc_register_card(struct snd_soc_device *socdev) struct snd_soc_machine *machine = socdev->machine; int ret = 0, i, ac97 = 0, err = 0; - mutex_lock(&codec->mutex); for(i = 0; i < machine->num_links; i++) { if (socdev->machine->dai_link[i].init) { err = socdev->machine->dai_link[i].init(codec); @@ -1124,12 +1123,14 @@ int snd_soc_register_card(struct snd_soc_device *socdev) goto out; } + mutex_lock(&codec->mutex); #ifdef CONFIG_SND_SOC_AC97_BUS if (ac97) { ret = soc_ac97_dev_register(codec); if (ret < 0) { printk(KERN_ERR "asoc: AC97 device register failed\n"); snd_card_free(codec->card); + mutex_unlock(&codec->mutex); goto out; } } @@ -1142,8 +1143,10 @@ int snd_soc_register_card(struct snd_soc_device *socdev) err = device_create_file(socdev->dev, &dev_attr_codec_reg); if (err < 0) printk(KERN_WARNING "asoc: failed to add codec sysfs entries\n"); -out: + mutex_unlock(&codec->mutex); + +out: return ret; } EXPORT_SYMBOL_GPL(snd_soc_register_card); -- 1.5.3.8

2 1

Re: [alsa-devel] [alsa-cvslog] alsa-driver: Added CONFIG_HAVE_INIT_UTSNAME test
by Takashi Iwai 21 Jan '08

21 Jan '08

At Mon, 21 Jan 2008 11:48:06 +0100 (CET), Jaroslav Kysela wrote: > > changeset: 3491:94a0c6eeb244 > tag: tip > user: perex > date: Mon Jan 21 11:48:04 2008 +0100 > files: acore/info_oss.c configure.in include/config.h.in > description: > Added CONFIG_HAVE_INIT_UTSNAME test > > > diff -r f8ead0afa48d -r 94a0c6eeb244 acore/info_oss.c > --- a/acore/info_oss.c Mon Jan 21 10:42:45 2008 +0100 > +++ b/acore/info_oss.c Mon Jan 21 11:48:04 2008 +0100 > @@ -1,6 +1,6 @@ > #define __NO_VERSION__ > #include <linux/version.h> > -#if LINUX_VERSION_CODE < KERNEL_VERSION(2, 6, 19) > +#ifdef CONFIG_HAVE_INIT_UTSNAME > #define init_utsname() (&system_utsname) > #endif > #include "adriver.h" > diff -r f8ead0afa48d -r 94a0c6eeb244 configure.in > --- a/configure.in Mon Jan 21 10:42:45 2008 +0100 > +++ b/configure.in Mon Jan 21 11:48:04 2008 +0100 > @@ -2566,6 +2566,32 @@ if test "$CONFIG_HAVE_GFP_T" = "1"; then > AC_DEFINE(CONFIG_HAVE_GFP_T) > fi > > +dnl Check for init_utsname... > +AC_MSG_CHECKING(for init_utsname) > +gfp_t="0" > +ac_save_CFLAGS="$CFLAGS" > +ac_save_CC="$CC" > +CFLAGS="$KERNEL_CHECK_CFLAGS" > +CC=$KCC > +AC_TRY_COMPILE([ > +#define __KERNEL__ > +#include <linux/config.h> > +#include <linux/utsname.h> > +],[ > + void *x = init_utsname(); > +], I'm afraid it won't work. In C, it's fine to call undeclared functions. It's supposed to be an int function. Takashi

2 1

[alsa-devel] [PATCH] dapm_new_mixer and 32 bytes limitation
by Robert Jarzmik 21 Jan '08

21 Jan '08

Dapm mixer names are truncated to 32 bytes, while alsa control names are 44 bytes wide. Subsequent usage of functions like snd_ctl_find_id fail because of the mismatch. Unless there's a limitation I don't understand, the following patch fixes the limit back to 44 bytes. -- Robert

3 3

[alsa-devel] alsaconf usefulness
by Alexander E. Patrakov 21 Jan '08

21 Jan '08

Hello developers, Debian applies the following patch to alsaconf: Index: alsa-utils-1.0.15/alsaconf/alsaconf.in =================================================================== --- alsa-utils-1.0.15.orig/alsaconf/alsaconf.in 2007-10-21 14:57:02.000000000 +0200 +++ alsa-utils-1.0.15/alsaconf/alsaconf.in 2007-10-21 14:57:02.000000000 +0200 @@ -458,8 +458,16 @@ Advanced Linux Sound Architecture (ALSA) driver. - You should stop all sound applications now." $version) - $DIALOG --msgbox "$msg" 20 63 || acex 0 + You should stop all sound applications now. + + Please notice, that sound configuration with this script + is not needed if you are running testing/sid with an up + to date kernel and your sound device is a supported PCI + one. In that case *udev* is predestinated to load your + driver ;) + But any way, you can finish this procedure with a working + PCI-Sound at the end." $version) + $DIALOG --msgbox "$msg" 24 63 || acex 0 } # FAREWELL The added note is completely valid: in most cases, there is nothing to do, even for ISA-PnP cards (because all self-respecting distributions already contain a rule for loading ISA-PnP drivers). However, the very presence of alsaconf generates a lot of support traffic on forums in the form "I have tried alsaconf and it didn't help" (or worse, ruined a working setup). Also, it cannot deal with more than one sound card. Could you please either apply the above patch (of course, replacing the wording with a distro-neutral one), or just drop this obsolete program and replace it with the "alsa-unmute" script which essentially contains the same as the set_mixers() function (but allows to specify the card to unmute)? -- Alexander E. Patrakov

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[alsa-devel] [PATCH] wm9713 and virtual Line mixer
by Robert Jarzmik 21 Jan '08

21 Jan '08

Trivial patch for the forgotten virtual line mixer in wm9713. Missing this mixer causes Mono Mixer and Speaker Mixer lack of support for input from Line Left and Line Right signals. -- Robert

2 1

[alsa-devel] "Help to debug" in ALSA wiki
by Jaroslav Kysela 21 Jan '08

21 Jan '08

Hi all, I would like to introduce "Help to debug" link in the ALSA wiki. The primary motivation is to provide information to users what we need to know to help us to debug things. Also next information is for potential developers to debug our code providing links to hardware specification and debug tools. Actually, I feel that especially for HDA support information is split in a group of main HDA developers, but we are lacking other links like how to compile ALSA driver from sources using a distribution or so. I would like to ask all developers to add and improve texts in this section of wiki, so we can just redirect users there (if required, of course). Thanks, Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

1 0

[alsa-devel] [Fwd: problem with audigy2 ZS remote]
by Stefano Radaelli 21 Jan '08

21 Jan '08

The problem still exists. Also the mic on the livedrive panel don't works without initialization. If I start windows and reboot in linux the panel is initialized and works(that confirm it is not an hardware problem) Thanks -------- Messaggio Originale -------- Oggetto: problem with audigy2 ZS remote Data: Mon, 25 Jun 2007 18:47:59 +0200 Da: Stefano Radaelli <puttext(a)tiscali.it> A: alsa-devel(a)alsa-project.org Hi all! Sorry for my bad english. I have a problem with the livedrive of an audigy2 ZS. After some upgrades (on a gentoo linux system) the eternal panel seems to stop working. Even after the "initialization" (with echo on the midi device), the IR led not blink on remote action. This worked correctly in the past but after some upgrade (i don't know exactly what) it stoipped working. I don't known where i have to look (logs?) or what type of debugging it's necessary. Help me please

1 0

Re: [alsa-devel] Si3054 and Lenovo3000 n100
by Brian Hinz 20 Jan '08

20 Jan '08

11c1:1040 is not an SiLabs chip, it's an agere - see alsa bug #0003316: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3316 and agere still has not released a driver or documentation yet: http://linmodems.technion.ac.il/archive-seventh/msg03029.html it probably should be removed from patch_si3054.c though... -brian ------------------------------ Message: 3 Date: Mon, 14 Jan 2008 12:07:15 +0100 From: Takashi Iwai Subject: Re: [alsa-devel] Si3054 and Lenovo3000 n100 To: Vasily Khoruzhick Cc: Andy Shevchenko , alsa-devel(a)alsa-project.org Message-ID: Content-Type: text/plain; charset=US-ASCII At Sun, 13 Jan 2008 19:10:44 +0200, Vasily Khoruzhick wrote: > > On 13 January 2008 18:15:06 Andy Shevchenko wrote: >> Hi Vasily Khoruzhick! >> >> On Sun, Jan 13, 2008 at 04:49:45PM +0200, Vasily Khoruzhick wrote next: >>> Hi, I have problems with setting up internal modem (regular si3054 >>> sitting on hda bus) with subj notebook. >> >> Do you belive that the Si3054 clone? What is ids of your modem codec? >> If you get 11c1:1040 you lose this game. >> >> P.S. IIRC the n200 has unsupported modem codec. I guess your model too. > > Btw, if it isn't compatible with Si3054, why its id added to patch_si3054.c > file? Possibly mistakenly :) Takashi

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[alsa-devel] conversion of opti93x driver to use cs4231-lib
by Krzysztof Helt 20 Jan '08

20 Jan '08

Hi, I posted some time ago two patches which converts opti93x driver to use cs4231-lib library. The opti93x chips are almost compatible with the cs4231 differing mostly by mixer. This change greatly reduces the source of the opti drivers. Here are patches: [addition of opti93x details to the cs4231-lib] http://mailman.alsa-project.org/pipermail/alsa-devel/2007-September/003395.… and [conversion of opti93x driver to use cs4231-lib] http://mailman.alsa-project.org/pipermail/alsa-devel/2007-September/003396.… I will rediff the second patch as there were some changes to opti93x driver after sending the patch. If it is not a good idea to do such conversion, please reply and drop patches. Regards, Krzysztof ------------------------------------------------------------ Walcz w kosmicznej wojnie w nowej grze! Kliknij >>> http://link.interia.pl/f1cc1

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[alsa-devel] Last call - get on the linux dev Gadget Labs card mailing list if you want to be notified
by Mike Mazarick 19 Jan '08

19 Jan '08

There is a project to develop a Gadget Labs Alsa driver for Linux audio. A mailing list has been established for this project so people could track the progress and release to the public of the linux driver for the GL card. There are currently ~10 people subscribed (it's a very small list). If you have any interest in knowing when a linux driver is available for this card, I'd highly recommend following the progress by subscribing to the mailing list. This will be significant to you primarily if you have a Gadget Labs card and want to use it with Mac OS X, because it will start to be feasible (and relatively easy) to have a Mac OS X driver when a linux driver is complete. It is also significant to you if you want an inexpensive way to get 8 channels of analog in/out from your computer with a high quality audio using linux audio tools. The Gadget Labs card was 'state of the art' 8 years ago, and is currently the least expensive way to get more than 4 channels of studio quality input/output to/from your computer (however, the other cards available are also excellent and close in price). Of course, we are always interested in anyone that can help with this effort by providing beta testing of the software, assistance with coding the driver, assistance with documentation, general 'cheering us on' and encouragement, etc. The list is scheduled to go 'private' and available for subscribers only in 5 days or less. If you are interested in this, please subscribe before the list goes 'private'. You will be able to subscribe after the list goes private, but won't be able to see postings with the latest information until after you have subscribed and been approved by the mailing list moderator. The way to subscribe to the mailing list is here: https://lists.sourceforge.net/lists/listinfo/wavepro-driver-developers The project summary page is here: http://sourceforge.net/projects/wavepro-driver/ The project web page is here: http://wavepro-driver.sourceforge.net/ -Mike Mazarick

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Re: [alsa-devel] [PATCH] Change of cold reset and ac97 register cache
by Robert Jarzmik 19 Jan '08

19 Jan '08

Mark Brown <broonie(a)opensource.wolfsonmicro.com> writes: > Just to clarify, it is the initial reset on chip probe that is causing > problems for you rather than a call later on? If the driver state and > chip state are out of sync after that then that suggests that there is > an error in the register defaults or some other similar issue. It's the initial on chip probe in my case. > > After that, I need to setup a specific GPIO to trigger loud speaker > > amplificator. The trouble is that I do not want to trigger another GPIO which > > powers another amplificator. > > My code basically reads de GPIO register, put the right bit, and stores back. > > But because of cache/chip difference, I trigger the other GPIO. > > How exactly are you doing these accesses and which registers is it that > trigger the problem - is it setting the GPIO state or configuring the > GPIO pin modes? I'm setting up a wm9713 GPIO (GPIO8) as input, and then set the pull-up register to power my speaker amplification. As we said, I'll implement a machine dependent solution, assuming there'll be in the future a mecanism in ASoC v2 allowing hook inscription for the reset phase. I'll synchronize the cache in this specific function, and will wait ASoC to catch up or propose a patch. -- Robert PS: Sorry for the list, in my last reply I forgot alsa-devel.

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[alsa-devel] [PATCH] hda_codec: Added power management for ADC/DAC's
by Matthew Ranostay 19 Jan '08

19 Jan '08

Added power management support to muted DACs and ADCs. They are set into D3 state when both channels are muted, and D0 when unmuted. We specify the ADC in the HDA_CODEC_MUTE_IDX instead of the ADC mux, so that we can easily know which node to power manage, and we use the connection list to get the needed mux node. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff -r 1cb5fe1b2443 pci/hda/hda_codec.c --- a/pci/hda/hda_codec.c Thu Jan 17 17:44:49 2008 +0100 +++ b/pci/hda/hda_codec.c Fri Jan 18 11:16:36 2008 -0500 @@ -1091,11 +1091,19 @@ int snd_hda_mixer_amp_switch_get(struct { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); hda_nid_t nid = get_amp_nid(kcontrol); + hda_nid_t conn_list[1]; int chs = get_amp_channels(kcontrol); int dir = get_amp_direction(kcontrol); int idx = get_amp_index(kcontrol); + int wcaps = get_wcaps(codec, nid); + int wtype = (wcaps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; long *valp = ucontrol->value.integer.value; + /* if pin is a adc get the mux connection */ + if (wtype == AC_WID_AUD_IN) { + snd_hda_get_connections(codec, nid, conn_list, 1); + nid = conn_list[0]; + } if (chs & 1) *valp++ = (snd_hda_codec_amp_read(codec, nid, 0, dir, idx) & HDA_AMP_MUTE) ? 0 : 1; @@ -1109,24 +1117,52 @@ int snd_hda_mixer_amp_switch_put(struct struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + hda_nid_t ctl_nid = get_amp_nid(kcontrol); hda_nid_t nid = get_amp_nid(kcontrol); + hda_nid_t conn_list[1]; int chs = get_amp_channels(kcontrol); int dir = get_amp_direction(kcontrol); int idx = get_amp_index(kcontrol); + int wcaps = get_wcaps(codec, nid); + int wtype = (wcaps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; long *valp = ucontrol->value.integer.value; - int change = 0; + int change = 0, val = 0; snd_hda_power_up(codec); + /* if pin is a adc get the mux connection */ + if (wtype == AC_WID_AUD_IN) { + snd_hda_get_connections(codec, nid, conn_list, 1); + nid = conn_list[0]; + } if (chs & 1) { change = snd_hda_codec_amp_update(codec, nid, 0, dir, idx, HDA_AMP_MUTE, *valp ? 0 : HDA_AMP_MUTE); + val = !*valp; valp++; } - if (chs & 2) + if (chs & 2) { change |= snd_hda_codec_amp_update(codec, nid, 1, dir, idx, HDA_AMP_MUTE, *valp ? 0 : HDA_AMP_MUTE); + val &= !*valp; + } + + /* check for power management support */ + if ((wcaps & AC_WCAP_POWER) && (wcaps != AC_WID_AUD_MIX)) { + if ((chs != 3) && (wcaps & AC_WCAP_STEREO)) { + struct hda_amp_info *info; + info = get_alloc_amp_hash(codec, + HDA_HASH_KEY(nid, dir, idx)); + /* get the other channel value */ + val &= !(get_vol_mute(codec, info, nid, + ~chs & 3, dir, idx) & HDA_AMP_MUTE); + } + /* power down adc/dac if both channels are muted */ + snd_hda_codec_write(codec, ctl_nid, 0, AC_VERB_SET_POWER_STATE, + val ? AC_PWRST_D3 : 0); + } + #ifdef CONFIG_SND_HDA_POWER_SAVE if (codec->patch_ops.check_power_status) codec->patch_ops.check_power_status(codec, nid); diff -r 1cb5fe1b2443 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Thu Jan 17 17:44:49 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Thu Jan 17 20:17:50 2008 -0500 @@ -638,10 +638,10 @@ static struct snd_kcontrol_new stac92hd7 STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 3), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1a, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1b, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Front Mic Mixer Capture Volume", 0x1d, 0, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Mixer Capture Switch", 0x1d, 0, HDA_INPUT), @@ -664,10 +664,10 @@ static struct snd_kcontrol_new stac92hd7 STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 4), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1a, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1b, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Front Mic Mixer Capture Volume", 0x1d, 0, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Mixer Capture Switch", 0x1d, 0, HDA_INPUT), @@ -690,10 +690,10 @@ static struct snd_kcontrol_new stac92hd7 STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 5), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1a, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1b, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Front Mic Mixer Capture Volume", 0x1d, 0, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Mixer Capture Switch", 0x1d, 0, HDA_INPUT), @@ -716,11 +716,11 @@ static struct snd_kcontrol_new stac92hd7 STAC_INPUT_SOURCE(2), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x12, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x0, 0x1a, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x13, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Analog Loopback 1", 0x17, 0x3, HDA_INPUT), @@ -735,11 +735,11 @@ static struct snd_kcontrol_new stac92hd7 STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x12, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x0, 0x1a, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x13, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x14, 0x1, 0, HDA_INPUT),

2 1

[alsa-devel] HDA codec graphing
by Claudio Matsuoka 19 Jan '08

19 Jan '08

After a lot of time published only as a git repository, I created a page and tarball for Eduardo Habkost's codecgraph tool to graph the layout of HDA codecs. It may be useful for those tweaking or troubleshooting existing drivers. The page has also examples of different codec layouts rendered in SVG. See details at http://helllabs.org/codecgraph/.

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[alsa-devel] [PATCH v4] [POWERPC] Update MPC8610 HPCD to support audio drivers
by Timur Tabi 18 Jan '08

18 Jan '08

Update the MPC8610 HPCD files to support the audio driver. Update booting-without-of.txt with information on the SSI device. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- Rebased for Kumar's repo. Documentation/powerpc/booting-without-of.txt | 41 ++++++ arch/powerpc/boot/dts/mpc8610_hpcd.dts | 113 +++++++++++++++++- arch/powerpc/configs/mpc8610_hpcd_defconfig | 171 +++++++++++++++++++++++++- arch/powerpc/platforms/86xx/mpc8610_hpcd.c | 15 +++ 4 files changed, 336 insertions(+), 4 deletions(-) diff --git a/Documentation/powerpc/booting-without-of.txt b/Documentation/powerpc/booting-without-of.txt index a13d457..6ac726d 100644 --- a/Documentation/powerpc/booting-without-of.txt +++ b/Documentation/powerpc/booting-without-of.txt @@ -56,6 +56,7 @@ Table of Contents m) Chipselect/Local Bus n) 4xx/Axon EMAC ethernet nodes o) Xilinx IP cores + p) Freescale Synchronous Serial Interface VII - Specifying interrupt information for devices 1) interrupts property @@ -2575,6 +2576,46 @@ platforms are moved over to use the flattened-device-tree model. Requred properties: - current-speed : Baud rate of uartlite + p) Freescale Synchronous Serial Interface + + The SSI is a serial device that communicates with audio codecs. It can + be programmed in AC97, I2S, left-justified, or right-justified modes. + + Required properties: + - compatible : compatible list, containing "fsl,ssi" + - cell-index : the SSI, <0> = SSI1, <1> = SSI2, and so on + - reg : offset and length of the register set for the device + - interrupts : <a b> where a is the interrupt number and b is a + field that represents an encoding of the sense and + level information for the interrupt. This should be + encoded based on the information in section 2) + depending on the type of interrupt controller you + have. + - interrupt-parent : the phandle for the interrupt controller that + services interrupts for this device. + - fsl,mode : the operating mode for the SSI interface + "i2s-slave" - I2S mode, SSI is clock slave + "i2s-master" - I2S mode, SSI is clock master + "lj-slave" - left-justified mode, SSI is clock slave + "lj-master" - l.j. mode, SSI is clock master + "rj-slave" - right-justified mode, SSI is clock slave + "rj-master" - r.j., SSI is clock master + "ac97-slave" - AC97 mode, SSI is clock slave + "ac97-master" - AC97 mode, SSI is clock master + + Optional properties: + - codec-handle : phandle to a 'codec' node that defines an audio + codec connected to this SSI. This node is typically + a child of an I2C or other control node. + + Child 'codec' node required properties: + - compatible : compatible list, contains the name of the codec + + Child 'codec' node optional properties: + - clock-frequency : The frequency of the input clock, which typically + comes from an on-board dedicated oscillator. + + More devices will be defined as this spec matures. VII - Specifying interrupt information for devices diff --git a/arch/powerpc/boot/dts/mpc8610_hpcd.dts b/arch/powerpc/boot/dts/mpc8610_hpcd.dts index 01040a7..d98715c 100644 --- a/arch/powerpc/boot/dts/mpc8610_hpcd.dts +++ b/arch/powerpc/boot/dts/mpc8610_hpcd.dts @@ -1,7 +1,7 @@ /* * MPC8610 HPCD Device Tree Source * - * Copyright 2007 Freescale Semiconductor Inc. + * Copyright 2007-2008 Freescale Semiconductor 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 @@ -49,6 +49,7 @@ #size-cells = <1>; #interrupt-cells = <2>; device_type = "soc"; + compatible = "fsl,mpc8610-immr", "simple-bus"; ranges = <0 e0000000 00100000>; reg = <e0000000 1000>; bus-frequency = <0>; @@ -62,6 +63,13 @@ interrupts = <2b 2>; interrupt-parent = <&mpic>; dfsrr; + + cs4270:codec@4f { + compatible = "cirrus,cs4270"; + reg = <4f>; + /* MCLK source is a stand-alone oscillator */ + clock-frequency = <bb8000>; + }; }; i2c@3100 { @@ -111,6 +119,109 @@ reg = <e0000 1000>; fsl,has-rstcr; }; + + i2s@16000 { + compatible = "fsl,mpc8610-ssi"; + cell-index = <0>; + reg = <16000 100>; + interrupt-parent = <&mpic>; + interrupts = <3e 2>; + fsl,mode = "i2s-slave"; + codec-handle = <&cs4270>; + }; + + ssi@16100 { + compatible = "fsl,mpc8610-ssi"; + cell-index = <1>; + reg = <16100 100>; + interrupt-parent = <&mpic>; + interrupts = <3f 2>; + }; + + dma@21300 { + #address-cells = <1>; + #size-cells = <1>; + compatible = "fsl,mpc8610-dma", "fsl,eloplus-dma"; + cell-index = <0>; + reg = <21300 4>; /* DMA general status register */ + ranges = <0 21100 200>; + + dma-channel@0 { + compatible = "fsl,mpc8610-dma-channel", + "fsl,eloplus-dma-channel"; + cell-index = <0>; + reg = <0 80>; + interrupt-parent = <&mpic>; + interrupts = <14 2>; + }; + dma-channel@1 { + compatible = "fsl,mpc8610-dma-channel", + "fsl,eloplus-dma-channel"; + cell-index = <1>; + reg = <80 80>; + interrupt-parent = <&mpic>; + interrupts = <15 2>; + }; + dma-channel@2 { + compatible = "fsl,mpc8610-dma-channel", + "fsl,eloplus-dma-channel"; + cell-index = <2>; + reg = <100 80>; + interrupt-parent = <&mpic>; + interrupts = <16 2>; + }; + dma-channel@3 { + compatible = "fsl,mpc8610-dma-channel", + "fsl,eloplus-dma-channel"; + cell-index = <3>; + reg = <180 80>; + interrupt-parent = <&mpic>; + interrupts = <17 2>; + }; + }; + + dma@c300 { + #address-cells = <1>; + #size-cells = <1>; + compatible = "fsl,mpc8610-dma", "fsl,mpc8540-dma"; + cell-index = <1>; + reg = <c300 4>; /* DMA general status register */ + ranges = <0 c100 200>; + + dma-channel@0 { + compatible = "fsl,mpc8610-dma-channel", + "fsl,mpc8540-dma-channel"; + cell-index = <0>; + reg = <0 80>; + interrupt-parent = <&mpic>; + interrupts = <3c 2>; + }; + dma-channel@1 { + compatible = "fsl,mpc8610-dma-channel", + "fsl,mpc8540-dma-channel"; + cell-index = <1>; + reg = <80 80>; + interrupt-parent = <&mpic>; + interrupts = <3d 2>; + }; + dma-channel@2 { + compatible = "fsl,mpc8610-dma-channel", + "fsl,mpc8540-dma-channel"; + cell-index = <2>; + reg = <100 80>; + interrupt-parent = <&mpic>; + interrupts = <3e 2>; + }; + dma-channel@3 { + compatible = "fsl,mpc8610-dma-channel", + "fsl,mpc8540-dma-channel"; + cell-index = <3>; + reg = <180 80>; + interrupt-parent = <&mpic>; + interrupts = <3f 2>; + }; + }; + }; pci0: pci@e0008000 { diff --git a/arch/powerpc/configs/mpc8610_hpcd_defconfig b/arch/powerpc/configs/mpc8610_hpcd_defconfig index 9614d24..2500ef4 100644 --- a/arch/powerpc/configs/mpc8610_hpcd_defconfig +++ b/arch/powerpc/configs/mpc8610_hpcd_defconfig @@ -696,7 +696,7 @@ CONFIG_SERIAL_8250_RSA=y CONFIG_SERIAL_CORE=y CONFIG_SERIAL_CORE_CONSOLE=y # CONFIG_SERIAL_JSM is not set -CONFIG_SERIAL_OF_PLATFORM=y +# CONFIG_SERIAL_OF_PLATFORM is not set CONFIG_UNIX98_PTYS=y # CONFIG_LEGACY_PTYS is not set # CONFIG_IPMI_HANDLER is not set @@ -708,7 +708,60 @@ CONFIG_UNIX98_PTYS=y # CONFIG_RAW_DRIVER is not set # CONFIG_TCG_TPM is not set CONFIG_DEVPORT=y -# CONFIG_I2C is not set +CONFIG_I2C=y +CONFIG_I2C_BOARDINFO=y +# CONFIG_I2C_CHARDEV is not set + +# +# I2C Algorithms +# +# CONFIG_I2C_ALGOBIT is not set +# CONFIG_I2C_ALGOPCF is not set +# CONFIG_I2C_ALGOPCA is not set + +# +# I2C Hardware Bus support +# +# 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 is not set +# CONFIG_I2C_I810 is not set +# CONFIG_I2C_PIIX4 is not set +CONFIG_I2C_MPC=y +# CONFIG_I2C_NFORCE2 is not set +# CONFIG_I2C_OCORES is not set +# CONFIG_I2C_PARPORT_LIGHT is not set +# CONFIG_I2C_PROSAVAGE is not set +# CONFIG_I2C_SAVAGE4 is not set +# CONFIG_I2C_SIMTEC is not set +# CONFIG_I2C_SIS5595 is not set +# CONFIG_I2C_SIS630 is not set +# CONFIG_I2C_SIS96X is not set +# CONFIG_I2C_TAOS_EVM is not set +# CONFIG_I2C_VIA is not set +# CONFIG_I2C_VIAPRO is not set +# CONFIG_I2C_VOODOO3 is not set + +# +# Miscellaneous I2C Chip support +# +# CONFIG_SENSORS_DS1337 is not set +# CONFIG_SENSORS_DS1374 is not set +# CONFIG_DS1682 is not set +# CONFIG_SENSORS_EEPROM is not set +# CONFIG_SENSORS_PCF8574 is not set +# CONFIG_SENSORS_PCA9539 is not set +# CONFIG_SENSORS_PCF8591 is not set +# CONFIG_SENSORS_M41T00 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 # # SPI support @@ -763,7 +816,119 @@ CONFIG_DUMMY_CONSOLE=y # # Sound # -# CONFIG_SOUND is not set +CONFIG_SOUND=y + +# +# Advanced Linux Sound Architecture +# +CONFIG_SND=y +CONFIG_SND_TIMER=y +CONFIG_SND_PCM=y +# CONFIG_SND_SEQUENCER is not set +CONFIG_SND_OSSEMUL=y +CONFIG_SND_MIXER_OSS=y +CONFIG_SND_PCM_OSS=y +# CONFIG_SND_PCM_OSS_PLUGINS is not set +# CONFIG_SND_DYNAMIC_MINORS is not set +# CONFIG_SND_SUPPORT_OLD_API is not set +CONFIG_SND_VERBOSE_PROCFS=y +# CONFIG_SND_VERBOSE_PRINTK is not set +# CONFIG_SND_DEBUG is not set + +# +# Generic devices +# +# CONFIG_SND_DUMMY is not set +# CONFIG_SND_MTPAV is not set +# CONFIG_SND_SERIAL_U16550 is not set +# CONFIG_SND_MPU401 is not set + +# +# PCI devices +# +# CONFIG_SND_AD1889 is not set +# CONFIG_SND_ALS300 is not set +# CONFIG_SND_ALS4000 is not set +# CONFIG_SND_ALI5451 is not set +# CONFIG_SND_ATIIXP is not set +# CONFIG_SND_ATIIXP_MODEM is not set +# CONFIG_SND_AU8810 is not set +# CONFIG_SND_AU8820 is not set +# CONFIG_SND_AU8830 is not set +# CONFIG_SND_AZT3328 is not set +# CONFIG_SND_BT87X is not set +# CONFIG_SND_CA0106 is not set +# CONFIG_SND_CMIPCI is not set +# CONFIG_SND_CS4281 is not set +# CONFIG_SND_CS46XX is not set +# CONFIG_SND_CS5530 is not set +# CONFIG_SND_DARLA20 is not set +# CONFIG_SND_GINA20 is not set +# CONFIG_SND_LAYLA20 is not set +# CONFIG_SND_DARLA24 is not set +# CONFIG_SND_GINA24 is not set +# CONFIG_SND_LAYLA24 is not set +# CONFIG_SND_MONA is not set +# CONFIG_SND_MIA is not set +# CONFIG_SND_ECHO3G is not set +# CONFIG_SND_INDIGO is not set +# CONFIG_SND_INDIGOIO is not set +# CONFIG_SND_INDIGODJ is not set +# CONFIG_SND_EMU10K1 is not set +# CONFIG_SND_EMU10K1X is not set +# CONFIG_SND_ENS1370 is not set +# CONFIG_SND_ENS1371 is not set +# CONFIG_SND_ES1938 is not set +# CONFIG_SND_ES1968 is not set +# CONFIG_SND_FM801 is not set +# CONFIG_SND_HDA_INTEL is not set +# CONFIG_SND_HDSP is not set +# CONFIG_SND_HDSPM is not set +# CONFIG_SND_ICE1712 is not set +# CONFIG_SND_ICE1724 is not set +# CONFIG_SND_INTEL8X0 is not set +# CONFIG_SND_INTEL8X0M is not set +# CONFIG_SND_KORG1212 is not set +# CONFIG_SND_MAESTRO3 is not set +# CONFIG_SND_MIXART is not set +# CONFIG_SND_NM256 is not set +# CONFIG_SND_PCXHR is not set +# CONFIG_SND_RIPTIDE is not set +# CONFIG_SND_RME32 is not set +# CONFIG_SND_RME96 is not set +# CONFIG_SND_RME9652 is not set +# CONFIG_SND_SONICVIBES is not set +# CONFIG_SND_TRIDENT is not set +# CONFIG_SND_VIA82XX is not set +# CONFIG_SND_VIA82XX_MODEM is not set +# CONFIG_SND_VX222 is not set +# CONFIG_SND_YMFPCI is not set + +# +# ALSA PowerMac devices +# + +# +# ALSA PowerPC devices +# + +# +# System on Chip audio support +# +CONFIG_SND_SOC=y + +# +# SoC Audio support for SuperH +# + +# +# ALSA SoC audio for Freescale SOCs +# +CONFIG_SND_SOC_MPC8610=y +CONFIG_SND_SOC_MPC8610_HPCD=y +CONFIG_SND_SOC_CS4270=y +CONFIG_SND_SOC_CS4270_VD33_ERRATA=y + CONFIG_HID_SUPPORT=y CONFIG_HID=y # CONFIG_HID_DEBUG is not set diff --git a/arch/powerpc/platforms/86xx/mpc8610_hpcd.c b/arch/powerpc/platforms/86xx/mpc8610_hpcd.c index c6d2f48..0b07485 100644 --- a/arch/powerpc/platforms/86xx/mpc8610_hpcd.c +++ b/arch/powerpc/platforms/86xx/mpc8610_hpcd.c @@ -34,9 +34,24 @@ #include <asm/mpic.h> +#include <linux/of_platform.h> #include <sysdev/fsl_pci.h> #include <sysdev/fsl_soc.h> +static struct of_device_id __initdata mpc8610_ids[] = { + { .compatible = "fsl,mpc8610-immr", }, + {} +}; + +static int __init mpc8610_declare_of_platform_devices(void) +{ + /* Without this call, the SSI device driver won't get probed. */ + of_platform_bus_probe(NULL, mpc8610_ids, NULL); + + return 0; +} +machine_device_initcall(mpc86xx_hpcd, mpc8610_declare_of_platform_devices); + void __init mpc86xx_hpcd_init_irq(void) { -- 1.5.2.4

2 1

[alsa-devel] [PATCH 0/2] ASoC drivers for the Freescale MPC8610 SoC
by Timur Tabi 18 Jan '08

18 Jan '08

These patches add ALSA SoC device drivers for the Freescale MPC8610 SoC and the MPC8610-HPCD reference board. The first patch updates some files in arch/powerpc and booting-without-of.txt. The second patch adds the new audio driver files to sound/soc. Both must be applied in order to avoid a build break.

8 22

[alsa-devel] alsa-1.0.16
by Thierry Vignaud 18 Jan '08

18 Jan '08

Hi, Do you plan to release alsa-1.0.16? Especially: - so that userland match kernel-2.6.24's subsystem - since alsa-lib has accumulated a number of bug fixes Releasing would be helpful for distro than plan to release in the next few monthes... Thanks.

6 7

[alsa-devel] [PATCH] hda_proc - Add a number of new settings to proc codec output
by Andrew Paprocki 18 Jan '08

18 Jan '08

This patch adds additional output to the /proc codec#X info. The following pieces of information are added to the output: - Balanced, L/R swap, trigger, impedance sense pin capabilities - Vref pin capabilities - Current Vref pin widget control setting - Default configuration association, sequence, and misc bit test - EAPD/BTL bits conveying balanced mode, EAPD, and L/R swap - Power state modified to show state name as well as setting vs actual value - GPIO parameter output on Audio Function Group, including enumeration of IO pins which are indicated present (Any I and O pins are not output at this time) - Stripe and L/R swap widget capabilities - All digital converter bits: enable, validity, validity config, preemphasis, copyright, non-audio, professional, generation level, and content category - Converter stream and channel values for in/out widgets - SDI select value for in widgets - Unsolicited response widget capability tag and enabled bit - Delay widget capability value - Processing widget capability benign bit and number of coefficients - Realtek Define Registers: processing coefficient, coefficient index Signed-off-by: Andrew Paprocki <andrew(a)ishiboo.com> diff -r 1e5c4ccf093c -r b86f0a07dd7d pci/hda/hda_codec.h --- a/pci/hda/hda_codec.h Sun Jan 13 12:03:53 2008 +0100 +++ b/pci/hda/hda_codec.h Fri Jan 18 02:05:04 2008 -0500 @@ -84,7 +84,9 @@ enum { #define AC_VERB_GET_GPIO_DATA 0x0f15 #define AC_VERB_GET_GPIO_MASK 0x0f16 #define AC_VERB_GET_GPIO_DIRECTION 0x0f17 +#define AC_VERB_GET_GPIO_WAKE_MASK 0x0f18 #define AC_VERB_GET_GPIO_UNSOLICITED_RSP_MASK 0x0f19 +#define AC_VERB_GET_GPIO_STICKY_MASK 0x0f1a #define AC_VERB_GET_CONFIG_DEFAULT 0x0f1c /* f20: AFG/MFG */ #define AC_VERB_GET_SUBSYSTEM_ID 0x0f20 @@ -112,7 +114,9 @@ enum { #define AC_VERB_SET_GPIO_DATA 0x715 #define AC_VERB_SET_GPIO_MASK 0x716 #define AC_VERB_SET_GPIO_DIRECTION 0x717 +#define AC_VERB_SET_GPIO_WAKE_MASK 0x718 #define AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK 0x719 +#define AC_VERB_SET_GPIO_STICKY_MASK 0x71a #define AC_VERB_SET_CONFIG_DEFAULT_BYTES_0 0x71c #define AC_VERB_SET_CONFIG_DEFAULT_BYTES_1 0x71d #define AC_VERB_SET_CONFIG_DEFAULT_BYTES_2 0x71e @@ -185,6 +189,27 @@ enum { #define AC_SUPFMT_FLOAT32 (1<<1) #define AC_SUPFMT_AC3 (1<<2) +/* GP I/O count */ +#define AC_GPIO_IO_COUNT (0xff<<0) +#define AC_GPIO_O_COUNT (0xff<<8) +#define AC_GPIO_O_COUNT_SHIFT 8 +#define AC_GPIO_I_COUNT (0xff<<16) +#define AC_GPIO_I_COUNT_SHIFT 16 +#define AC_GPIO_UNSOLICITED (1<<30) +#define AC_GPIO_WAKE (1<<31) + +/* Converter stream, channel */ +#define AC_CONV_CHANNEL (0xf<<0) +#define AC_CONV_STREAM (0xf<<4) +#define AC_CONV_STREAM_SHIFT 4 + +/* Input converter SDI select */ +#define AC_SDI_SELECT (0xf<<0) + +/* Unsolicited response */ +#define AC_UNSOL_TAG (0x3f<<0) +#define AC_UNSOL_ENABLED (1<<7) + /* Pin widget capabilies */ #define AC_PINCAP_IMP_SENSE (1<<0) /* impedance sense capable */ #define AC_PINCAP_TRIG_REQ (1<<1) /* trigger required */ @@ -230,6 +255,9 @@ enum { #define AC_PWRST_D3SUP (1<<3) /* Power state values */ +#define AC_PWRST_SETTING (0xf<<0) +#define AC_PWRST_ACTUAL (0xf<<4) +#define AC_PWRST_ACTUAL_SHIFT 4 #define AC_PWRST_D0 0x00 #define AC_PWRST_D1 0x01 #define AC_PWRST_D2 0x02 @@ -238,6 +266,7 @@ enum { /* Processing capabilies */ #define AC_PCAP_BENIGN (1<<0) #define AC_PCAP_NUM_COEF (0xff<<8) +#define AC_PCAP_NUM_COEF_SHIFT 8 /* Volume knobs capabilities */ #define AC_KNBCAP_NUM_STEPS (0x7f<<0) @@ -273,6 +302,9 @@ enum { #define AC_DIG1_NONAUDIO (1<<5) #define AC_DIG1_PROFESSIONAL (1<<6) #define AC_DIG1_LEVEL (1<<7) + +/* DIGITAL2 bits */ +#define AC_DIG2_CC (0x7f<<0) /* Pin widget control - 8bit */ #define AC_PINCTL_VREFEN (0x7<<0) @@ -288,12 +320,22 @@ enum { /* Unsolicited response - 8bit */ #define AC_USRSP_EN (1<<7) +/* Pin sense - 32bit */ +#define AC_PINSENSE_IMPEDANCE_MASK (0x7fffffff) +#define AC_PINSENSE_PRESENCE (1<<31) + +/* EAPD/BTL enable - 32bit */ +#define AC_EAPDBTL_BALANCED (1<<0) +#define AC_EAPDBTL_EAPD (1<<1) +#define AC_EAPDBTL_LR_SWAP (1<<2) + /* configuration default - 32bit */ #define AC_DEFCFG_SEQUENCE (0xf<<0) #define AC_DEFCFG_DEF_ASSOC (0xf<<4) #define AC_DEFCFG_ASSOC_SHIFT 4 #define AC_DEFCFG_MISC (0xf<<8) #define AC_DEFCFG_MISC_SHIFT 8 +#define AC_DEFCFG_MISC_NO_PRESENCE (1<<0) #define AC_DEFCFG_COLOR (0xf<<12) #define AC_DEFCFG_COLOR_SHIFT 12 #define AC_DEFCFG_CONN_TYPE (0xf<<16) diff -r 1e5c4ccf093c -r b86f0a07dd7d pci/hda/hda_proc.c --- a/pci/hda/hda_proc.c Sun Jan 13 12:03:53 2008 +0100 +++ b/pci/hda/hda_proc.c Fri Jan 18 02:05:04 2008 -0500 @@ -202,7 +202,8 @@ static const char *get_jack_color(u32 cf } static void print_pin_caps(struct snd_info_buffer *buffer, - struct hda_codec *codec, hda_nid_t nid) + struct hda_codec *codec, hda_nid_t nid, + int *supports_vref) { static char *jack_conns[4] = { "Jack", "N/A", "Fixed", "Both" }; static char *jack_types[16] = { @@ -226,7 +227,47 @@ static void print_pin_caps(struct snd_in snd_iprintf(buffer, " EAPD"); if (caps & AC_PINCAP_PRES_DETECT) snd_iprintf(buffer, " Detect"); + if (caps & AC_PINCAP_BALANCE) + snd_iprintf(buffer, " Balanced"); + if (caps & AC_PINCAP_LR_SWAP) + snd_iprintf(buffer, " R/L"); + if (caps & AC_PINCAP_TRIG_REQ) + snd_iprintf(buffer, " Trigger"); + if (caps & AC_PINCAP_IMP_SENSE) + snd_iprintf(buffer, " ImpSense"); snd_iprintf(buffer, "\n"); + if (caps & AC_PINCAP_VREF) { + unsigned int vref = + (caps & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT; + snd_iprintf(buffer, " Vref caps:"); + if (vref & AC_PINCAP_VREF_HIZ) + snd_iprintf(buffer, " HIZ"); + if (vref & AC_PINCAP_VREF_50) + snd_iprintf(buffer, " 50"); + if (vref & AC_PINCAP_VREF_GRD) + snd_iprintf(buffer, " GRD"); + if (vref & AC_PINCAP_VREF_80) + snd_iprintf(buffer, " 80"); + if (vref & AC_PINCAP_VREF_100) + snd_iprintf(buffer, " 100"); + snd_iprintf(buffer, "\n"); + *supports_vref = 1; + } + else { + *supports_vref = 0; + } + if (caps & AC_PINCAP_EAPD) { + val = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_EAPD_BTLENABLE, 0); + snd_iprintf(buffer, " EAPD 0x%x:", val); + if (val & AC_EAPDBTL_BALANCED) + snd_iprintf(buffer, " BALANCED"); + if (val & AC_EAPDBTL_EAPD) + snd_iprintf(buffer, " EAPD"); + if (val & AC_EAPDBTL_LR_SWAP) + snd_iprintf(buffer, " R/L"); + snd_iprintf(buffer, "\n"); + } caps = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); snd_iprintf(buffer, " Pin Default 0x%08x: [%s] %s at %s %s\n", caps, jack_conns[(caps & AC_DEFCFG_PORT_CONN) >> AC_DEFCFG_PORT_CONN_SHIFT], @@ -236,13 +277,71 @@ static void print_pin_caps(struct snd_in snd_iprintf(buffer, " Conn = %s, Color = %s\n", get_jack_connection(caps), get_jack_color(caps)); - if (caps & AC_PINCAP_EAPD) { - val = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_EAPD_BTLENABLE, 0); - snd_iprintf(buffer, " EAPD: 0x%x\n", val); - } -} - + /* Default association and sequence values refer to default grouping + * of pin complexes and their sequence within the group. This is used + * for priority and resource allocation. + */ + snd_iprintf(buffer, " DefAssociation = 0x%x, Sequence = 0x%x\n", + (caps & AC_DEFCFG_DEF_ASSOC) >> AC_DEFCFG_ASSOC_SHIFT, + caps & AC_DEFCFG_SEQUENCE); + if (((caps & AC_DEFCFG_MISC) >> AC_DEFCFG_MISC_SHIFT) & + AC_DEFCFG_MISC_NO_PRESENCE) { + /* Miscellaneous bit indicates external hardware does not + * support presence detection even if the pin complex + * indicates it is supported. + */ + snd_iprintf(buffer, " Misc = NO_PRESENCE\n"); + } +} + +static const char *get_pwr_state(u32 state) +{ + static const char *buf[4] = { + "D0", "D1", "D2", "D3" + }; + if (state < 4) + return buf[state]; + return "UNKNOWN"; +} + +static void print_gpio(struct snd_info_buffer *buffer, + struct hda_codec *codec, hda_nid_t nid) +{ + unsigned int gpio = + snd_hda_param_read(codec, codec->afg, AC_PAR_GPIO_CAP); + unsigned int enable, direction, wake, unsol, sticky, data; + int i, max; + snd_iprintf(buffer, "GPIO: io=%d, o=%d, i=%d, " + "unsolicited=%d, wake=%d\n", + gpio & AC_GPIO_IO_COUNT, + (gpio & AC_GPIO_O_COUNT) >> AC_GPIO_O_COUNT_SHIFT, + (gpio & AC_GPIO_I_COUNT) >> AC_GPIO_I_COUNT_SHIFT, + (gpio & AC_GPIO_UNSOLICITED) ? 1 : 0, + (gpio & AC_GPIO_WAKE) ? 1 : 0); + max = gpio & AC_GPIO_IO_COUNT; + enable = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_GPIO_MASK, 0); + direction = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_GPIO_DIRECTION, 0); + wake = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_GPIO_WAKE_MASK, 0); + unsol = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_GPIO_UNSOLICITED_RSP_MASK, 0); + sticky = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_GPIO_STICKY_MASK, 0); + data = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_GPIO_DATA, 0); + for (i = 0; i < max; ++i) + snd_iprintf(buffer, + " IO[%d]: enable=%d, dir=%d, wake=%d, " + "sticky=%d, data=%d\n", i, + (enable & (1<<i)) ? 1 : 0, + (direction & (1<<i)) ? 1 : 0, + (wake & (1<<i)) ? 1 : 0, + (sticky & (1<<i)) ? 1 : 0, + (data & (1<<i)) ? 1 : 0); + /* FIXME: add GPO and GPI pin information */ +} static void print_codec_info(struct snd_info_entry *entry, struct snd_info_buffer *buffer) @@ -280,13 +379,16 @@ static void print_codec_info(struct snd_ snd_hda_power_down(codec); return; } + + print_gpio(buffer, codec, codec->afg); + for (i = 0; i < nodes; i++, nid++) { unsigned int wid_caps = snd_hda_param_read(codec, nid, AC_PAR_AUDIO_WIDGET_CAP); unsigned int wid_type = (wid_caps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; - int conn_len = 0; + int conn_len = 0; hda_nid_t conn[HDA_MAX_CONNECTIONS]; unsigned int pinctls; @@ -302,6 +404,10 @@ static void print_codec_info(struct snd_ snd_iprintf(buffer, " Amp-In"); if (wid_caps & AC_WCAP_OUT_AMP) snd_iprintf(buffer, " Amp-Out"); + if (wid_caps & AC_WCAP_STRIPE) + snd_iprintf(buffer, " Stripe"); + if (wid_caps & AC_WCAP_LR_SWAP) + snd_iprintf(buffer, " R/L"); snd_iprintf(buffer, "\n"); /* volume knob is a special widget that always have connection @@ -330,8 +436,9 @@ static void print_codec_info(struct snd_ } switch (wid_type) { - case AC_WID_PIN: - print_pin_caps(buffer, codec, nid); + case AC_WID_PIN: { + int supports_vref; + print_pin_caps(buffer, codec, nid, &supports_vref); pinctls = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); @@ -342,8 +449,29 @@ static void print_codec_info(struct snd_ snd_iprintf(buffer, " OUT"); if (pinctls & AC_PINCTL_HP_EN) snd_iprintf(buffer, " HP"); + if (supports_vref) { + int vref = pinctls & AC_PINCTL_VREFEN; + switch (vref) { + case AC_PINCTL_VREF_HIZ: + snd_iprintf(buffer, " VREF_HIZ"); + break; + case AC_PINCTL_VREF_50: + snd_iprintf(buffer, " VREF_50"); + break; + case AC_PINCTL_VREF_GRD: + snd_iprintf(buffer, " VREF_GRD"); + break; + case AC_PINCTL_VREF_80: + snd_iprintf(buffer, " VREF_80"); + break; + case AC_PINCTL_VREF_100: + snd_iprintf(buffer, " VREF_100"); + break; + } + } snd_iprintf(buffer, "\n"); break; + } case AC_WID_VOL_KNB: pinctls = snd_hda_param_read(codec, nid, AC_PAR_VOL_KNB_CAP); @@ -356,19 +484,85 @@ static void print_codec_info(struct snd_ (pinctls >> 7) & 1, pinctls & 0x7f); break; case AC_WID_AUD_OUT: - case AC_WID_AUD_IN: + case AC_WID_AUD_IN: { + int conv = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_CONV, 0); + snd_iprintf(buffer, + " Converter: stream=%d, channel=%d\n", + (conv & AC_CONV_STREAM) >> + AC_CONV_STREAM_SHIFT, + conv & AC_CONV_CHANNEL); + + if (wid_type == AC_WID_AUD_IN && + (conv & AC_CONV_CHANNEL) == 0) { + int sdi = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_SDI_SELECT, + 0); + snd_iprintf(buffer, " SDI-Select: %d\n", + sdi & AC_SDI_SELECT); + } + + if (wid_caps & AC_WCAP_DIGITAL) { + unsigned int digi1 = + snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_DIGI_CONVERT_1, 0); + unsigned int digi2 = + snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_DIGI_CONVERT_2, 0); + snd_iprintf(buffer, " Digital:"); + if (digi1 & AC_DIG1_ENABLE) + snd_iprintf(buffer, " Enabled"); + if (digi1 & AC_DIG1_V) + snd_iprintf(buffer, " Validity"); + if (digi1 & AC_DIG1_VCFG) + snd_iprintf(buffer, " ValidityCfg"); + if (digi1 & AC_DIG1_EMPHASIS) + snd_iprintf(buffer, " Preemphasis"); + if (digi1 & AC_DIG1_COPYRIGHT) + snd_iprintf(buffer, " Copyright"); + if (digi1 & AC_DIG1_NONAUDIO) + snd_iprintf(buffer, " Non-Audio"); + if (digi1 & AC_DIG1_PROFESSIONAL) + snd_iprintf(buffer, " Pro"); + if (digi1 & AC_DIG1_LEVEL) + snd_iprintf(buffer, " GenLevel"); + snd_iprintf(buffer, "\n"); + snd_iprintf(buffer, + " Digital category: 0x%x\n", + digi2 & AC_DIG2_CC); + } + if (wid_caps & AC_WCAP_FORMAT_OVRD) { snd_iprintf(buffer, " PCM:\n"); print_pcm_caps(buffer, codec, nid); } break; + } } - if (wid_caps & AC_WCAP_POWER) - snd_iprintf(buffer, " Power: 0x%x\n", - snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_POWER_STATE, - 0)); + if (wid_caps & AC_WCAP_UNSOL_CAP) { + int unsol = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_UNSOLICITED_RESPONSE, 0); + snd_iprintf(buffer, + " Unsolicited: tag=%02x, enabled=%d\n", + unsol & AC_UNSOL_TAG, + (unsol & AC_UNSOL_ENABLED) ? 1 : 0); + } + + if (wid_caps & AC_WCAP_POWER) { + int pwr = + snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_POWER_STATE, 0); + snd_iprintf(buffer, " Power: setting=%s, actual=%s\n", + get_pwr_state(pwr & AC_PWRST_SETTING), + get_pwr_state((pwr & AC_PWRST_ACTUAL) >> + AC_PWRST_ACTUAL_SHIFT)); + } + + if (wid_caps & AC_WCAP_DELAY) + snd_iprintf(buffer, " Delay: %d samples\n", + (wid_caps & AC_WCAP_DELAY) >> + AC_WCAP_DELAY_SHIFT); if (wid_caps & AC_WCAP_CONN_LIST) { int c, curr = -1; @@ -386,6 +580,31 @@ static void print_codec_info(struct snd_ snd_iprintf(buffer, "\n"); } } + + if (wid_caps & AC_WCAP_PROC_WID) { + unsigned int proc_caps = + snd_hda_param_read(codec, nid, + AC_PAR_PROC_CAP); + snd_iprintf(buffer, + " Processing caps: benign=%d, ncoeff=%d\n", + proc_caps & AC_PCAP_BENIGN, + (proc_caps & AC_PCAP_NUM_COEF) >> + AC_PCAP_NUM_COEF_SHIFT); + } + + /* NID 0x20 == Realtek Define Registers */ + if (codec->vendor_id == 0x10ec && nid == 0x20) { + int coeff = + snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PROC_COEF, 0); + snd_iprintf(buffer, + " Processing Coefficient: 0x%02x\n", + coeff); + coeff = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_COEF_INDEX, 0); + snd_iprintf(buffer, + " Coefficient Index: 0x%02x\n", coeff); + } } snd_hda_power_down(codec); }

2 2

[alsa-devel] [PATCH] hda_proc - Add a number of new settings to proc codec output
by Andrew Paprocki 18 Jan '08

18 Jan '08

This patch adds additional output to the /proc codec#X info. The following pieces of information are added to the output: - Balanced, L/R swap, trigger, impedance sense pin capabilities - Vref pin capabilities - Current Vref pin widget control setting - Default configuration association, sequence, and misc bit test - EAPD/BTL bits conveying balanced mode, EAPD, and L/R swap - Power state modified to show state name as well as setting vs actual value - GPIO parameter output on Audio Function Group, including enumeration of IO pins which are indicated present (Any I and O pins are not output at this time) - Stripe and L/R swap widget capabilities - All digital converter bits: enable, validity, validity config, preemphasis, copyright, non-audio, professional, generation level, and content category - Converter stream and channel values for in/out widgets - SDI select value for in widgets - Unsolicited response widget capability tag and enabled bit - Delay widget capability value - Processing widget capability benign bit and number of coefficients - Realtek Define Registers: processing coefficient, coefficient index Signed-off-by: Andrew Paprocki <andrew(a)ishiboo.com> diff -r 1e5c4ccf093c -r f343e1eb02f6 pci/hda/hda_codec.h --- a/pci/hda/hda_codec.h Sun Jan 13 12:03:53 2008 +0100 +++ b/pci/hda/hda_codec.h Fri Jan 18 01:54:07 2008 -0500 @@ -84,7 +84,9 @@ enum { #define AC_VERB_GET_GPIO_DATA 0x0f15 #define AC_VERB_GET_GPIO_MASK 0x0f16 #define AC_VERB_GET_GPIO_DIRECTION 0x0f17 +#define AC_VERB_GET_GPIO_WAKE_MASK 0x0f18 #define AC_VERB_GET_GPIO_UNSOLICITED_RSP_MASK 0x0f19 +#define AC_VERB_GET_GPIO_STICKY_MASK 0x0f1a #define AC_VERB_GET_CONFIG_DEFAULT 0x0f1c /* f20: AFG/MFG */ #define AC_VERB_GET_SUBSYSTEM_ID 0x0f20 @@ -112,7 +114,9 @@ enum { #define AC_VERB_SET_GPIO_DATA 0x715 #define AC_VERB_SET_GPIO_MASK 0x716 #define AC_VERB_SET_GPIO_DIRECTION 0x717 +#define AC_VERB_SET_GPIO_WAKE_MASK 0x718 #define AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK 0x719 +#define AC_VERB_SET_GPIO_STICKY_MASK 0x71a #define AC_VERB_SET_CONFIG_DEFAULT_BYTES_0 0x71c #define AC_VERB_SET_CONFIG_DEFAULT_BYTES_1 0x71d #define AC_VERB_SET_CONFIG_DEFAULT_BYTES_2 0x71e @@ -185,6 +189,27 @@ enum { #define AC_SUPFMT_FLOAT32 (1<<1) #define AC_SUPFMT_AC3 (1<<2) +/* GP I/O count */ +#define AC_GPIO_IO_COUNT (0xff<<0) +#define AC_GPIO_O_COUNT (0xff<<8) +#define AC_GPIO_O_COUNT_SHIFT 8 +#define AC_GPIO_I_COUNT (0xff<<16) +#define AC_GPIO_I_COUNT_SHIFT 16 +#define AC_GPIO_UNSOLICITED (1<<30) +#define AC_GPIO_WAKE (1<<31) + +/* Converter stream, channel */ +#define AC_CONV_CHANNEL (0xf<<0) +#define AC_CONV_STREAM (0xf<<4) +#define AC_CONV_STREAM_SHIFT 4 + +/* Input converter SDI select */ +#define AC_SDI_SELECT (0xf<<0) + +/* Unsolicited response */ +#define AC_UNSOL_TAG (0x3f<<0) +#define AC_UNSOL_ENABLED (1<<7) + /* Pin widget capabilies */ #define AC_PINCAP_IMP_SENSE (1<<0) /* impedance sense capable */ #define AC_PINCAP_TRIG_REQ (1<<1) /* trigger required */ @@ -230,6 +255,9 @@ enum { #define AC_PWRST_D3SUP (1<<3) /* Power state values */ +#define AC_PWRST_SETTING (0xf<<0) +#define AC_PWRST_ACTUAL (0xf<<4) +#define AC_PWRST_ACTUAL_SHIFT 4 #define AC_PWRST_D0 0x00 #define AC_PWRST_D1 0x01 #define AC_PWRST_D2 0x02 @@ -238,6 +266,7 @@ enum { /* Processing capabilies */ #define AC_PCAP_BENIGN (1<<0) #define AC_PCAP_NUM_COEF (0xff<<8) +#define AC_PCAP_NUM_COEF_SHIFT 8 /* Volume knobs capabilities */ #define AC_KNBCAP_NUM_STEPS (0x7f<<0) @@ -273,6 +302,9 @@ enum { #define AC_DIG1_NONAUDIO (1<<5) #define AC_DIG1_PROFESSIONAL (1<<6) #define AC_DIG1_LEVEL (1<<7) + +/* DIGITAL2 bits */ +#define AC_DIG2_CC (0x7f<<0) /* Pin widget control - 8bit */ #define AC_PINCTL_VREFEN (0x7<<0) @@ -288,12 +320,22 @@ enum { /* Unsolicited response - 8bit */ #define AC_USRSP_EN (1<<7) +/* Pin sense - 32bit */ +#define AC_PINSENSE_IMPEDANCE_MASK (0x7fffffff) +#define AC_PINSENSE_PRESENCE (1<<31) + +/* EAPD/BTL enable - 32bit */ +#define AC_EAPDBTL_BALANCED (1<<0) +#define AC_EAPDBTL_EAPD (1<<1) +#define AC_EAPDBTL_LR_SWAP (1<<2) + /* configuration default - 32bit */ #define AC_DEFCFG_SEQUENCE (0xf<<0) #define AC_DEFCFG_DEF_ASSOC (0xf<<4) #define AC_DEFCFG_ASSOC_SHIFT 4 #define AC_DEFCFG_MISC (0xf<<8) #define AC_DEFCFG_MISC_SHIFT 8 +#define AC_DEFCFG_MISC_NO_PRESENCE (1<<0) #define AC_DEFCFG_COLOR (0xf<<12) #define AC_DEFCFG_COLOR_SHIFT 12 #define AC_DEFCFG_CONN_TYPE (0xf<<16) diff -r 1e5c4ccf093c -r f343e1eb02f6 pci/hda/hda_proc.c --- a/pci/hda/hda_proc.c Sun Jan 13 12:03:53 2008 +0100 +++ b/pci/hda/hda_proc.c Fri Jan 18 01:54:07 2008 -0500 @@ -202,7 +202,8 @@ static const char *get_jack_color(u32 cf } static void print_pin_caps(struct snd_info_buffer *buffer, - struct hda_codec *codec, hda_nid_t nid) + struct hda_codec *codec, hda_nid_t nid, + int *supports_vref) { static char *jack_conns[4] = { "Jack", "N/A", "Fixed", "Both" }; static char *jack_types[16] = { @@ -226,7 +227,35 @@ static void print_pin_caps(struct snd_in snd_iprintf(buffer, " EAPD"); if (caps & AC_PINCAP_PRES_DETECT) snd_iprintf(buffer, " Detect"); + if (caps & AC_PINCAP_BALANCE) + snd_iprintf(buffer, " Balanced"); + if (caps & AC_PINCAP_LR_SWAP) + snd_iprintf(buffer, " R/L"); + if (caps & AC_PINCAP_TRIG_REQ) + snd_iprintf(buffer, " Trigger"); + if (caps & AC_PINCAP_IMP_SENSE) + snd_iprintf(buffer, " ImpSense"); snd_iprintf(buffer, "\n"); + if (caps & AC_PINCAP_VREF) { + unsigned int vref = + (caps & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT; + snd_iprintf(buffer, " Vref caps:"); + if (vref & AC_PINCAP_VREF_HIZ) + snd_iprintf(buffer, " HIZ"); + if (vref & AC_PINCAP_VREF_50) + snd_iprintf(buffer, " 50"); + if (vref & AC_PINCAP_VREF_GRD) + snd_iprintf(buffer, " GRD"); + if (vref & AC_PINCAP_VREF_80) + snd_iprintf(buffer, " 80"); + if (vref & AC_PINCAP_VREF_100) + snd_iprintf(buffer, " 100"); + snd_iprintf(buffer, "\n"); + *supports_vref = 1; + } + else { + *supports_vref = 0; + } caps = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); snd_iprintf(buffer, " Pin Default 0x%08x: [%s] %s at %s %s\n", caps, jack_conns[(caps & AC_DEFCFG_PORT_CONN) >> AC_DEFCFG_PORT_CONN_SHIFT], @@ -236,13 +265,83 @@ static void print_pin_caps(struct snd_in snd_iprintf(buffer, " Conn = %s, Color = %s\n", get_jack_connection(caps), get_jack_color(caps)); + /* Default association and sequence values refer to default grouping + * of pin complexes and their sequence within the group. This is used + * for priority and resource allocation. + */ + snd_iprintf(buffer, " DefAssociation = 0x%x, Sequence = 0x%x\n", + (caps & AC_DEFCFG_DEF_ASSOC) >> AC_DEFCFG_ASSOC_SHIFT, + caps & AC_DEFCFG_SEQUENCE); + if (((caps & AC_DEFCFG_MISC) >> AC_DEFCFG_MISC_SHIFT) & + AC_DEFCFG_MISC_NO_PRESENCE) { + /* Miscellaneous bit indicates external hardware does not + * support presence detection even if the pin complex + * indicates it is supported. + */ + snd_iprintf(buffer, " Misc = NO_PRESENCE\n"); + } if (caps & AC_PINCAP_EAPD) { val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_EAPD_BTLENABLE, 0); - snd_iprintf(buffer, " EAPD: 0x%x\n", val); - } -} - + snd_iprintf(buffer, " EAPD 0x%x:", val); + if (val & AC_EAPDBTL_BALANCED) + snd_iprintf(buffer, " BALANCED"); + if (val & AC_EAPDBTL_EAPD) + snd_iprintf(buffer, " EAPD"); + if (val & AC_EAPDBTL_LR_SWAP) + snd_iprintf(buffer, " R/L"); + snd_iprintf(buffer, "\n"); + } +} + +static const char *get_pwr_state(u32 state) +{ + static const char *buf[4] = { + "D0", "D1", "D2", "D3" + }; + if (state < 4) + return buf[state]; + return "UNKNOWN"; +} + +static void print_gpio(struct snd_info_buffer *buffer, + struct hda_codec *codec, hda_nid_t nid) +{ + unsigned int gpio = + snd_hda_param_read(codec, codec->afg, AC_PAR_GPIO_CAP); + unsigned int enable, direction, wake, unsol, sticky, data; + int i, max; + snd_iprintf(buffer, "GPIO: io=%d, o=%d, i=%d, " + "unsolicited=%d, wake=%d\n", + gpio & AC_GPIO_IO_COUNT, + (gpio & AC_GPIO_O_COUNT) >> AC_GPIO_O_COUNT_SHIFT, + (gpio & AC_GPIO_I_COUNT) >> AC_GPIO_I_COUNT_SHIFT, + (gpio & AC_GPIO_UNSOLICITED) ? 1 : 0, + (gpio & AC_GPIO_WAKE) ? 1 : 0); + max = gpio & AC_GPIO_IO_COUNT; + enable = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_GPIO_MASK, 0); + direction = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_GPIO_DIRECTION, 0); + wake = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_GPIO_WAKE_MASK, 0); + unsol = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_GPIO_UNSOLICITED_RSP_MASK, 0); + sticky = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_GPIO_STICKY_MASK, 0); + data = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_GPIO_DATA, 0); + for (i = 0; i < max; ++i) + snd_iprintf(buffer, + " IO[%d]: enable=%d, dir=%d, wake=%d, " + "sticky=%d, data=%d\n", i, + (enable & (1<<i)) ? 1 : 0, + (direction & (1<<i)) ? 1 : 0, + (wake & (1<<i)) ? 1 : 0, + (sticky & (1<<i)) ? 1 : 0, + (data & (1<<i)) ? 1 : 0); + /* FIXME: add GPO and GPI pin information */ +} static void print_codec_info(struct snd_info_entry *entry, struct snd_info_buffer *buffer) @@ -280,13 +379,16 @@ static void print_codec_info(struct snd_ snd_hda_power_down(codec); return; } + + print_gpio(buffer, codec, codec->afg); + for (i = 0; i < nodes; i++, nid++) { unsigned int wid_caps = snd_hda_param_read(codec, nid, AC_PAR_AUDIO_WIDGET_CAP); unsigned int wid_type = (wid_caps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; - int conn_len = 0; + int conn_len = 0; hda_nid_t conn[HDA_MAX_CONNECTIONS]; unsigned int pinctls; @@ -302,6 +404,10 @@ static void print_codec_info(struct snd_ snd_iprintf(buffer, " Amp-In"); if (wid_caps & AC_WCAP_OUT_AMP) snd_iprintf(buffer, " Amp-Out"); + if (wid_caps & AC_WCAP_STRIPE) + snd_iprintf(buffer, " Stripe"); + if (wid_caps & AC_WCAP_LR_SWAP) + snd_iprintf(buffer, " R/L"); snd_iprintf(buffer, "\n"); /* volume knob is a special widget that always have connection @@ -330,8 +436,9 @@ static void print_codec_info(struct snd_ } switch (wid_type) { - case AC_WID_PIN: - print_pin_caps(buffer, codec, nid); + case AC_WID_PIN: { + int supports_vref; + print_pin_caps(buffer, codec, nid, &supports_vref); pinctls = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); @@ -342,8 +449,29 @@ static void print_codec_info(struct snd_ snd_iprintf(buffer, " OUT"); if (pinctls & AC_PINCTL_HP_EN) snd_iprintf(buffer, " HP"); + if (supports_vref) { + int vref = pinctls & AC_PINCTL_VREFEN; + switch (vref) { + case AC_PINCTL_VREF_HIZ: + snd_iprintf(buffer, " VREF_HIZ"); + break; + case AC_PINCTL_VREF_50: + snd_iprintf(buffer, " VREF_50"); + break; + case AC_PINCTL_VREF_GRD: + snd_iprintf(buffer, " VREF_GRD"); + break; + case AC_PINCTL_VREF_80: + snd_iprintf(buffer, " VREF_80"); + break; + case AC_PINCTL_VREF_100: + snd_iprintf(buffer, " VREF_100"); + break; + } + } snd_iprintf(buffer, "\n"); break; + } case AC_WID_VOL_KNB: pinctls = snd_hda_param_read(codec, nid, AC_PAR_VOL_KNB_CAP); @@ -356,19 +484,85 @@ static void print_codec_info(struct snd_ (pinctls >> 7) & 1, pinctls & 0x7f); break; case AC_WID_AUD_OUT: - case AC_WID_AUD_IN: + case AC_WID_AUD_IN: { + int conv = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_CONV, 0); + snd_iprintf(buffer, + " Converter: stream=%d, channel=%d\n", + (conv & AC_CONV_STREAM) >> + AC_CONV_STREAM_SHIFT, + conv & AC_CONV_CHANNEL); + + if (wid_type == AC_WID_AUD_IN && + (conv & AC_CONV_CHANNEL) == 0) { + int sdi = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_SDI_SELECT, + 0); + snd_iprintf(buffer, " SDI-Select: %d\n", + sdi & AC_SDI_SELECT); + } + + if (wid_caps & AC_WCAP_DIGITAL) { + unsigned int digi1 = + snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_DIGI_CONVERT_1, 0); + unsigned int digi2 = + snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_DIGI_CONVERT_2, 0); + snd_iprintf(buffer, " Digital:"); + if (digi1 & AC_DIG1_ENABLE) + snd_iprintf(buffer, " Enabled"); + if (digi1 & AC_DIG1_V) + snd_iprintf(buffer, " Validity"); + if (digi1 & AC_DIG1_VCFG) + snd_iprintf(buffer, " ValidityCfg"); + if (digi1 & AC_DIG1_EMPHASIS) + snd_iprintf(buffer, " Preemphasis"); + if (digi1 & AC_DIG1_COPYRIGHT) + snd_iprintf(buffer, " Copyright"); + if (digi1 & AC_DIG1_NONAUDIO) + snd_iprintf(buffer, " Non-Audio"); + if (digi1 & AC_DIG1_PROFESSIONAL) + snd_iprintf(buffer, " Pro"); + if (digi1 & AC_DIG1_LEVEL) + snd_iprintf(buffer, " GenLevel"); + snd_iprintf(buffer, "\n"); + snd_iprintf(buffer, + " Digital category: 0x%x\n", + digi2 & AC_DIG2_CC); + } + if (wid_caps & AC_WCAP_FORMAT_OVRD) { snd_iprintf(buffer, " PCM:\n"); print_pcm_caps(buffer, codec, nid); } break; + } } - if (wid_caps & AC_WCAP_POWER) - snd_iprintf(buffer, " Power: 0x%x\n", - snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_POWER_STATE, - 0)); + if (wid_caps & AC_WCAP_UNSOL_CAP) { + int unsol = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_UNSOLICITED_RESPONSE, 0); + snd_iprintf(buffer, + " Unsolicited: tag=%02x, enabled=%d\n", + unsol & AC_UNSOL_TAG, + (unsol & AC_UNSOL_ENABLED) ? 1 : 0); + } + + if (wid_caps & AC_WCAP_POWER) { + int pwr = + snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_POWER_STATE, 0); + snd_iprintf(buffer, " Power: setting=%s, actual=%s\n", + get_pwr_state(pwr & AC_PWRST_SETTING), + get_pwr_state((pwr & AC_PWRST_ACTUAL) >> + AC_PWRST_ACTUAL_SHIFT)); + } + + if (wid_caps & AC_WCAP_DELAY) + snd_iprintf(buffer, " Delay: %d samples\n", + (wid_caps & AC_WCAP_DELAY) >> + AC_WCAP_DELAY_SHIFT); if (wid_caps & AC_WCAP_CONN_LIST) { int c, curr = -1; @@ -386,6 +580,31 @@ static void print_codec_info(struct snd_ snd_iprintf(buffer, "\n"); } } + + if (wid_caps & AC_WCAP_PROC_WID) { + unsigned int proc_caps = + snd_hda_param_read(codec, nid, + AC_PAR_PROC_CAP); + snd_iprintf(buffer, + " Processing caps: benign=%d, ncoeff=%d\n", + proc_caps & AC_PCAP_BENIGN, + (proc_caps & AC_PCAP_NUM_COEF) >> + AC_PCAP_NUM_COEF_SHIFT); + } + + /* NID 0x20 == Realtek Define Registers */ + if (codec->vendor_id == 0x10ec && nid == 0x20) { + int coeff = + snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PROC_COEF, 0); + snd_iprintf(buffer, + " Processing Coefficient: 0x%02x\n", + coeff); + coeff = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_COEF_INDEX, 0); + snd_iprintf(buffer, + " Coefficient Index: 0x%02x\n", coeff); + } } snd_hda_power_down(codec); }

1 0

[alsa-devel] [PATCH] hda_proc - Add output of Vref, additional pcaps, and proccaps to proc output
by Andrew Paprocki 18 Jan '08

18 Jan '08

This adds additional pincaps which were not previously output. Also, Vref capabilities are output per-pin along with the current Vref pinctl setting if the NID supports Vref. Also added: processing widgets and proccaps, as well as processing coefficient and coefficient index from the Realtek Define Registers per the ALC883 datasheet. Signed-off-by: Andrew Paprocki <andrew(a)ishiboo.com> diff -r 1e5c4ccf093c -r 8a9218bae32f pci/hda/hda_proc.c --- a/pci/hda/hda_proc.c Sun Jan 13 12:03:53 2008 +0100 +++ b/pci/hda/hda_proc.c Thu Jan 17 09:44:07 2008 -0500 @@ -202,7 +202,8 @@ static const char *get_jack_color(u32 cf } static void print_pin_caps(struct snd_info_buffer *buffer, - struct hda_codec *codec, hda_nid_t nid) + struct hda_codec *codec, hda_nid_t nid, + int *supports_vref) { static char *jack_conns[4] = { "Jack", "N/A", "Fixed", "Both" }; static char *jack_types[16] = { @@ -226,7 +227,35 @@ static void print_pin_caps(struct snd_in snd_iprintf(buffer, " EAPD"); if (caps & AC_PINCAP_PRES_DETECT) snd_iprintf(buffer, " Detect"); - snd_iprintf(buffer, "\n"); + if (caps & AC_PINCAP_BALANCE) + snd_iprintf(buffer, " Balanced"); + if (caps & AC_PINCAP_LR_SWAP) + snd_iprintf(buffer, " R/L"); + if (caps & AC_PINCAP_TRIG_REQ) + snd_iprintf(buffer, " Trigger"); + if (caps & AC_PINCAP_IMP_SENSE) + snd_iprintf(buffer, " ImpSense"); + snd_iprintf(buffer, "\n"); + if (caps & AC_PINCAP_VREF) { + unsigned int vref = + (caps & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT; + snd_iprintf(buffer, " Vref:"); + if (vref & AC_PINCAP_VREF_HIZ) + snd_iprintf(buffer, " HIZ"); + if (vref & AC_PINCAP_VREF_50) + snd_iprintf(buffer, " 50"); + if (vref & AC_PINCAP_VREF_GRD) + snd_iprintf(buffer, " GRD"); + if (vref & AC_PINCAP_VREF_80) + snd_iprintf(buffer, " 80"); + if (vref & AC_PINCAP_VREF_100) + snd_iprintf(buffer, " 100"); + snd_iprintf(buffer, "\n"); + *supports_vref = 1; + } + else { + *supports_vref = 0; + } caps = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); snd_iprintf(buffer, " Pin Default 0x%08x: [%s] %s at %s %s\n", caps, jack_conns[(caps & AC_DEFCFG_PORT_CONN) >> AC_DEFCFG_PORT_CONN_SHIFT], @@ -330,8 +359,9 @@ static void print_codec_info(struct snd_ } switch (wid_type) { - case AC_WID_PIN: - print_pin_caps(buffer, codec, nid); + case AC_WID_PIN: { + int supports_vref; + print_pin_caps(buffer, codec, nid, &supports_vref); pinctls = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); @@ -342,8 +372,29 @@ static void print_codec_info(struct snd_ snd_iprintf(buffer, " OUT"); if (pinctls & AC_PINCTL_HP_EN) snd_iprintf(buffer, " HP"); + if (supports_vref) { + int vref = pinctls & AC_PINCTL_VREFEN; + switch (vref) { + case AC_PINCTL_VREF_HIZ: + snd_iprintf(buffer, " VREF_HIZ"); + break; + case AC_PINCTL_VREF_50: + snd_iprintf(buffer, " VREF_50"); + break; + case AC_PINCTL_VREF_GRD: + snd_iprintf(buffer, " VREF_GRD"); + break; + case AC_PINCTL_VREF_80: + snd_iprintf(buffer, " VREF_80"); + break; + case AC_PINCTL_VREF_100: + snd_iprintf(buffer, " VREF_100"); + break; + } + } snd_iprintf(buffer, "\n"); break; + } case AC_WID_VOL_KNB: pinctls = snd_hda_param_read(codec, nid, AC_PAR_VOL_KNB_CAP); @@ -386,6 +437,31 @@ static void print_codec_info(struct snd_ snd_iprintf(buffer, "\n"); } } + + if (wid_caps & AC_WCAP_PROC_WID) { + unsigned int proc_caps = + snd_hda_param_read(codec, nid, + AC_PAR_PROC_CAP); + snd_iprintf(buffer, + " Processing Caps: benign=%d ncoeff=%d\n", + proc_caps & AC_PCAP_BENIGN, + (proc_caps & AC_PCAP_NUM_COEF) >> + AC_PCAP_NUM_COEF_SHIFT); + } + + /* NID 0x20 == Realtek Define Registers */ + if (codec->vendor_id == 0x10ec && nid == 0x20) { + int coeff = + snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PROC_COEF, 0); + snd_iprintf(buffer, + " Processing Coefficient: 0x%02x\n", + coeff); + coeff = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_COEF_INDEX, 0); + snd_iprintf(buffer, + " Coefficient Index: 0x%02x\n", coeff); + } } snd_hda_power_down(codec); }

3 5

Re: [alsa-devel] intel-hd on an intel G33 motherboard
by De Fuuss 18 Jan '08

18 Jan '08

Hi, Indeed it works. As you mght have noticed I'm new to ALSA ;-) and I hadn't noticed that some of the channels could be muted. I thought they were muted only when at 0%. I also get "hda_codec_setup_stream:NID=" lines in dmesg now. I'm really sorry about that, it's really a beginner's mistake. At least one good thing came out of it: I can confirm that hda_intel works with model=6stack-hp on the XPC Shuttle SG33G5M . I will continue to carry out tests to see if there are any issues with Mic, line out, spdifs etc.. I'd like to thank all those that helped me for their continued patience and understanding. ALSA rocks because of you! Regards, Patrick > From: zhe.jiang(a)intel.com > To: fuuss(a)hotmail.com > Date: Thu, 17 Jan 2008 11:36:32 +0800 > CC: cmatsuoka(a)gmail.com; alsa-devel(a)alsa-project.org > Subject: Re: [alsa-devel] intel-hd on an intel G33 motherboard > > On Wed, 2008-01-16 at 23:06 +0000, De Fuuss wrote: >> Ups my bad, somehow the model number got commented out by an update script I ran yesterday. The drivers were recompiled with the "with-debug=detect" option set. I've attached the dmesg, codec#0 output and amixer contents. >> Dmesg is not very talkative, it did not show "hda_codec_setup_stream:NID=", however I get something of the type "hda_generic: no proper output path found". >> About the testing, I performed speaker-test every time I changed something (still only output on "Front L/R channels"). >> > > I guess that the "hda_generic: no proper output" is caused during > detect another codec. > > It's strange that there is no messages like > "hda_codec_setup_stream:NID=" when you run speaker-test. > >>From you amixer1.txt, I found that you muted the Surround,C/LFE and > Side.Please unmute them. > > Did you just run the "speaker-test -Dplug:surround71 -c8 -sX" with the > X=0 and X=1? > If it is true, you should change the X from 0 to 8. > > Best regards! > > >> Patrick >> >>> From: zhe.jiang(a)intel.com >>> To: fuuss(a)hotmail.com >>> Date: Wed, 16 Jan 2008 13:49:43 +0800 >>> CC: cmatsuoka(a)gmail.com; alsa-devel(a)alsa-project.org >>> Subject: Re: [alsa-devel] intel-hd on an intel G33 motherboard >>> >>> On Wed, 2008-01-16 at 02:22 +0800, De Fuuss wrote: >>>> >>>> Hi, >>>> >>>> I have recompiled the driver with --with-debug=detect. Dmesg output is >>>> attached to this mail. ALSA specificities seem to be present this >>>> time. >>> Hi, >>> >>>>From the dmesg, there is "hda_codec: Unknown model for ALC883, trying >>> auto-probe from BIOS" >>> It looks like that no "model=" is assigned. >>> Did you assign model for the driver? >>> >>> Can you play some music? >>> Then there will be some info in the dmesg like >>> "hda_codec_setup_stream:NID= ". >>> Please paste such info and give the result of >>> "cat /proc/asound/card0/codec#0". >>> >>> BTW, how did you do the testing? >>> You run speaker-test or by other way? >>> >>>>From the codec info, it seems that there are paths from DAC to Jack. >>> So it's a little weird that all other jacks are silent. >>> >>> Thanks! >>> >>>> Hope this helps. >>>> >>>> Best regards, >>>> >>>> Patrick >>>> >>>>> From: zhe.jiang(a)intel.com >>>>> To: fuuss(a)hotmail.com >>>>> Date: Tue, 15 Jan 2008 09:32:35 +0800 >>>>> CC: cmatsuoka(a)gmail.com; alsa-devel(a)alsa-project.org >>>>> Subject: Re: [alsa-devel] intel-hd on an intel G33 motherboard >>>>> >>>>> On Mon, 2008-01-14 at 17:11 +0000, De Fuuss wrote: >>>>>> The jumper is now set to enable onboard (Intel HDA) audio. >>>>>> At the back of the barebone there are 5 jacks (4 channels for 7.1 >>>> and 1 line in channel) + 1 spdif out. >>>>>> From these, using 6stack-hp, only the "Front L/R" jack works, none >>>> of the others >>>>>> On the front, there are 2 jacks (1 headphones and 1 microphone). >>>> The headphone jack works. >>>>>> >>>>> Can you tell us how you did the testing? >>>>>>From the driver code, it's not likely that all other jacks don't >>>> work. >>>>> >>>>> Because the driver support up-mixing, you can play a 2-channel music >>>> and >>>>> get sound from 4 jacks on the back panel. >>>>> >>>>> I read the dmesg output that you sent to me. >>>>> But there is no alsa debug info in it. >>>>> Did you really configure the driver with the option >>>>> "--with-debug=detect"? >>>>> If not, please try it and paste the dmesg info, it is helpful to >>>> trace >>>>> the issue. >>>>> >>>>> Best regards! >>>>> Jiang,zhe >>>>>> All the channels have been set to 100% in alsamixer. >>>>>> >>>>>> Considering that a week ago I had 0.0 and today I'm at 2.0, it's >>>> already a improvement. ;-) >>>>>> >>>>>> Patrick >>>>>> >>>>>>> Subject: Re: [alsa-devel] intel-hd on an intel G33 motherboard >>>>>>> From: zhe.jiang(a)intel.com >>>>>>> To: fuuss(a)hotmail.com >>>>>>> CC: cmatsuoka(a)gmail.com; alsa-devel(a)alsa-project.org >>>>>>> Date: Mon, 14 Jan 2008 16:43:00 +0800 >>>>>>> >>>>>>> On Mon, 2008-01-14 at 02:06 +0800, De Fuuss wrote: >>>>>>>> >>>>>>>> Following your advice, I tested per channel output. Using either >>>>>>>> 6stack-dig or 6stack-hp, I get noise from Front Left and Right >>>> jacks, >>>>>>>> using speaker-test (X=1 and 2) but not on any other channels. >>>>>>> >>>>>>> Hi, >>>>>>> Did you mean that after you changed the 'Jumper', >>>>>>> you could get the sound from the Front jack on the back channel? >>>>>>> >>>>>>>> Patrick >>>>>>>> >>>>>>>>> Date: Sun, 13 Jan 2008 01:01:50 -0200 >>>>>>>>> From: cmatsuoka(a)gmail.com >>>>>>>>> To: fuuss(a)hotmail.com >>>>>>>>> CC: zhe.jiang(a)intel.com; alsa-devel(a)alsa-project.org >>>>>>>>> Subject: Re: [alsa-devel] intel-hd on an intel G33 motherboard >>>>>>>>> >>>>>>>>> 2008/1/12 De Fuuss : >>>>>>>>> >>>>>>>>>> I get the uneasy feeling that something is messed up with this >>>>>>>> board. Anyway, for completeness' sake, I attached the results of >>>>>>>> amixer contents, cat /proc/asound/card0/codec#0 and dmesg for the >>>>>>>> module loaded as 6stack-dig. I have to try 6stack-hp again to >>>> confirm >>>>>>>> if I get something there. >>>>>>>>> >>>>>>>>> I fixed the Inspiron output, and I'm adding the SG33G5M codec to >>>> the >>>>>>>>> codecgraph codecs collection (git repository at >>>>>>>>> http://helllabs.org/git/codecgraph.git). If nodes 0x14, 0x15, >>>> 0x16 >>>>>>>> and >>>>>>>>> 0x17 are the front, rear, clfe and side connectors in the back >>>>>>>> panel, >>>>>>>>> 6stack-hp should work (at least for the outputs). You can test >>>> the >>>>>>>>> outputs using "speaker-test -Dplug:surround71 -c8 -sX" (where X >>>>>>>> varies >>>>>>>>> from 1 to 8) and a pair of headphones. >>>>>>>>> _______________________________________________ >>>>>>>>> Alsa-devel mailing list >>>>>>>>> Alsa-devel(a)alsa-project.org >>>>>>>>> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >>>>>>>> >>>>>>>> _________________________________________________________________ >>>>>>>> Express yourself instantly with MSN Messenger! Download today >>>> it's >>>>>>>> FREE! >>>>>>>> http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ >>>>>>>> _______________________________________________ >>>>>>>> Alsa-devel mailing list >>>>>>>> Alsa-devel(a)alsa-project.org >>>>>>>> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >>>>>>>> >>>>>>>> >>>>>>> >>>>>> >>>>>> _________________________________________________________________ >>>>>> Express yourself instantly with MSN Messenger! Download today it's >>>> FREE! >>>>>> http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ >>>>> >>>>> _______________________________________________ >>>>> Alsa-devel mailing list >>>>> Alsa-devel(a)alsa-project.org >>>>> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >>>> >>>> _________________________________________________________________ >>>> Express yourself instantly with MSN Messenger! Download today it's >>>> FREE! >>>> http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ >>>> >>>> >>> >>> _______________________________________________ >>> Alsa-devel mailing list >>> Alsa-devel(a)alsa-project.org >>> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >> >> _________________________________________________________________ >> Express yourself instantly with MSN Messenger! Download today it's FREE! >> http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ > > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

3 2

[alsa-devel] [PATCH] [ALSA] mpc8610: Add mmap support
by Timur Tabi 17 Jan '08

17 Jan '08

Enable mmap support in the MPC8610 ASoC driver. The driver can use ALSA's default mmap functionality, it was just not enabled previously. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/fsl/fsl_dma.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 2173203..652514f 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -136,7 +136,9 @@ struct fsl_dma_private { */ static const struct snd_pcm_hardware fsl_dma_hardware = { - .info = SNDRV_PCM_INFO_INTERLEAVED, + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID, .formats = FSLDMA_PCM_FORMATS, .rates = FSLDMA_PCM_RATES, .rate_min = 5512, -- 1.5.2.4

4 6

[alsa-devel] HDA NIDs 0x25 & 0x26 on ALC883
by Andrew Paprocki 17 Jan '08

17 Jan '08

Takashi, Do you know anything about NIDs 0x25 and 0x26 on the ALC883 chip? 0x25 is an Audio Output and 0x26 is an Audio Mixer. They appear completely separate from the 0x0c-0x0f mixers, but when I map one of my pins to use the 0x26 mixer, I hear audio over the jack whenever I play to the standard 'Green' 0x0c mixer. The 0x26 mixer seems unused in the current driver, so I'm not sure if there is something special about it. I'm basically trying to use it as a 5th output. Also, when I play audio, the 0x0c mixer has a nice full sound (the "expected" sound). The 0x0d & 0x0e mixers have much less bass and sound like someone turned up the treble. The 0x0f and 0x26 mixers have an even more "tinny" sound. I've verified that all the mixers/pins are configured the same exact way and all have full volume (0x1f). Do you know of any reason why the audio output over the pins would differ? I'm not sure what I could be missing here to get them to all sound the same. I appreciate any info you could provide. Thanks, -Andrew This is my 0x26 / 0x25 config, mapped to pin 0x1a which I've reconfigured as an output jack: Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x08173e: IN OUT HP Detect Pin Default 0x0181344f: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue EAPD: 0x0 Pin-ctls: 0x40: OUT Connection: 5 0x0c 0x0d 0x0e 0x0f 0x26* Node 0x26 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Connection: 2 0x25 0x0b Node 0x25 [Audio Output] wcaps 0x11: Stereo PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM

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Re: [alsa-devel] S16_LE + softvol = bad
by Alexander E. Patrakov 17 Jan '08

17 Jan '08

[sorry for a possible duplicate] 2008/1/17, Clemens Ladisch <cladisch(a)fastmail.net>: > Alexander E. Patrakov wrote: > > some cards (most notably, Intel HD audio) have the PCM control as softvol, and > > default to S16_LE as a format for dmix. > > HDA-Intel.conf doesn't set a sample format, and when none is set, dmix > tries S32 before S16. You get S16 only if the device doesn't support > S32. Sorry, this doesn't appear to be the case here. # with explicit defaults.pcm.dmix.format S32_LE patrakov@home:~$ aplay -vv /usr/share/sounds/alsa/Front_Left.wav Playing WAVE '/usr/share/sounds/alsa/Front_Left.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono Plug PCM: Route conversion PCM (sformat=S32_LE) Transformation table: 0 <- 0 1 <- 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 8192 period_size : 1024 period_time : 21333 tick_time : 0 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 1024 xfer_align : 1024 start_threshold : 8192 stop_threshold : 8192 silence_threshold: 0 silence_size : 0 boundary : 1073741824 Slave: Soft volume PCM Control: PCM Playback Volume min_dB: -51 max_dB: 0 resolution: 256 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 32 buffer_size : 8192 period_size : 1024 period_time : 21333 tick_time : 0 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 1024 xfer_align : 1024 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 : S32_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 32 buffer_size : 8192 period_size : 1024 period_time : 21333 tick_time : 0 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 1024 xfer_align : 1024 start_threshold : 8192 stop_threshold : 8192 silence_threshold: 0 silence_size : 0 boundary : 1073741824 Hardware PCM card 0 'HDA Intel' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 32 buffer_size : 8192 period_size : 1024 period_time : 21333 tick_time : 4000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 1024 xfer_align : 1024 start_threshold : 1 stop_threshold : 1073741824 silence_threshold: 0 silence_size : 1073741824 boundary : 1073741824 #+ | 00% # The same without defaults.pcm.dmix.format patrakov@home:~$ aplay -vv /usr/share/sounds/alsa/Front_Left.wav Playing WAVE '/usr/share/sounds/alsa/Front_Left.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono Plug PCM: Route conversion PCM (sformat=S16_LE) Transformation table: 0 <- 0 1 <- 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 16384 period_size : 1024 period_time : 21333 tick_time : 0 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 1024 xfer_align : 1024 start_threshold : 16384 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 1073741824 Slave: Soft volume PCM Control: PCM Playback Volume min_dB: -51 max_dB: 0 resolution: 256 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 16384 period_size : 1024 period_time : 21333 tick_time : 0 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 1024 xfer_align : 1024 start_threshold : 16384 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 1073741824 Slave: Direct Stream Mixing PCM Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 16384 period_size : 1024 period_time : 21333 tick_time : 0 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 1024 xfer_align : 1024 start_threshold : 16384 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 1073741824 Hardware PCM card 0 'HDA Intel' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 16384 period_size : 1024 period_time : 21333 tick_time : 4000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 1024 xfer_align : 1024 start_threshold : 1 stop_threshold : 1073741824 silence_threshold: 0 silence_size : 1073741824 boundary : 1073741824 # + | 49% This is with libasound2 1.0.15-3 from Debian. -- Alexander E. Patrakov

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[alsa-devel] [PATCH] Change of cold reset and ac97 register cache
by Robert Jarzmik 17 Jan '08

17 Jan '08

This patch synchronizes wm9713 AC97 register cache and hardware registers upon reset. The actual implementation of the wm9713 codec provides a cache of AC97 registers. After cold reset, the cache and hardware are not in sync. When using wm9713 logic, oring a GPIO triggers many other GPIOS. The solution of this patch reads from hardware to cache upon cold reset. Another one would be to write the cache to hardware upon init. If decided so, I would gladly change my patch. -- Robert Jarzmik

2 1

[alsa-devel] [PATCH]
by Jiang zhe 17 Jan '08

17 Jan '08

Hi, For codec conexant 5045, I found that the name of "Capture Source Items" is different from the name of mixer. The mixer is: HDA_CODEC_VOLUME("Ext Mic Playback Volume", 0x17, 0x2, HDA_INPUT), HDA_CODEC_MUTE("Ext Mic Playback Switch", 0x17, 0x2, HDA_INPUT), But the capture source item is : static struct hda_input_mux cxt5045_capture_source = { .num_items = 2, .items = { { "IntMic", 0x1 }, { "LineIn", 0x2 }, } }; I think that it's better to change the name of capture_source to avoid misunderstanding. Best regards! Signed-off-by: Jiang zhe <zhe.jiang(a)intel.com> --- alsa-driver-hg20080115/alsa-kernel/pci/hda/patch_conexant.c 2008-01-09 09:00:15.000000000 +0800 +++ b/alsa-kernel/pci/hda/patch_conexant.c 2008-01-17 10:50:06.000000000 +0800 @@ -453,7 +453,7 @@ static struct hda_input_mux cxt5045_capt .num_items = 2, .items = { { "IntMic", 0x1 }, - { "LineIn", 0x2 }, + { "ExtMic", 0x2 }, } };

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[alsa-devel] [PATCH]New model for conexant 5045 codec to support benq r55e
by Jiang zhe 17 Jan '08

17 Jan '08

Hi, Please refer to bug 0003636. The benq r55e laptop have 3 jacks on the front panel. One for HP, one for Line In and one for Mic In. This patch implemented a new model to support it. Best regards! Jiang,zhe Signed-off-by: Jiang Zhe <zhe.jiang(a)intel.com> --- alsa-driver-hg20080115/alsa-kernel/pci/hda/patch_conexant.c 2008-01-09 09:00:15.000000000 +0800 +++ b/alsa-kernel/pci/hda/patch_conexant.c 2008-01-17 10:33:44.000000000 +0800 @@ -457,6 +457,15 @@ static struct hda_input_mux cxt5045_capt } }; +static struct hda_input_mux cxt5045_capture_source_benq = { + .num_items = 3, + .items = { + { "IntMic", 0x1 }, + { "ExtMic", 0x2 }, + { "LineIn", 0x3 }, + } +}; + /* turn on/off EAPD (+ mute HP) as a master switch */ static int cxt5045_hp_master_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -576,6 +585,15 @@ static struct snd_kcontrol_new cxt5045_m {} }; +static struct snd_kcontrol_new cxt5045_benq_mixers[] = { + HDA_CODEC_VOLUME("Line In Capture Volume", 0x1a, 0x03, HDA_INPUT), + HDA_CODEC_MUTE("Line In Capture Switch", 0x1a, 0x03, HDA_INPUT), + HDA_CODEC_VOLUME("Line In Playback Volume", 0x17, 0x3, HDA_INPUT), + HDA_CODEC_MUTE("Line In Playback Switch", 0x17, 0x3, HDA_INPUT), + + {} +}; + static struct hda_verb cxt5045_init_verbs[] = { /* Line in, Mic */ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, @@ -601,6 +619,30 @@ static struct hda_verb cxt5045_init_verb { } /* end */ }; +static struct hda_verb cxt5045_benq_init_verbs[] = { + /* Int Mic, Mic */ + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_80 }, + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_80 }, + /* Line In,HP, Amp */ + {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x10, AC_VERB_SET_CONNECT_SEL, 0x1}, + {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x11, AC_VERB_SET_CONNECT_SEL, 0x1}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + /* Record selector: Int mic */ + {0x1a, AC_VERB_SET_CONNECT_SEL,0x1}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, + AC_AMP_SET_INPUT|AC_AMP_SET_RIGHT|AC_AMP_SET_LEFT|0x17}, + /* SPDIF route: PCM */ + { 0x13, AC_VERB_SET_CONNECT_SEL, 0x0 }, + /* EAPD */ + {0x10, AC_VERB_SET_EAPD_BTLENABLE, 0x2 }, /* default on */ + { } /* end */ +}; static struct hda_verb cxt5045_hp_sense_init_verbs[] = { /* pin sensing on HP jack */ @@ -741,6 +783,7 @@ static int cxt5045_init(struct hda_codec enum { CXT5045_LAPTOP, /* Laptops w/ EAPD support */ CXT5045_FUJITSU, /* Laptops w/ EAPD support */ + CXT5045_BENQ, #ifdef CONFIG_SND_DEBUG CXT5045_TEST, #endif @@ -750,6 +793,7 @@ enum { static const char *cxt5045_models[CXT5045_MODELS] = { [CXT5045_LAPTOP] = "laptop", [CXT5045_FUJITSU] = "fujitsu", + [CXT5045_BENQ] = "benq", #ifdef CONFIG_SND_DEBUG [CXT5045_TEST] = "test", #endif @@ -763,6 +807,7 @@ static struct snd_pci_quirk cxt5045_cfg_ SND_PCI_QUIRK(0x103c, 0x30cd, "HP DV Series", CXT5045_LAPTOP), SND_PCI_QUIRK(0x103c, 0x30d5, "HP 530", CXT5045_LAPTOP), SND_PCI_QUIRK(0x103c, 0x30d9, "HP Spartan", CXT5045_LAPTOP), + SND_PCI_QUIRK(0x152d, 0x0753, "Benq R55E", CXT5045_BENQ), SND_PCI_QUIRK(0x1734, 0x10ad, "Fujitsu Si1520", CXT5045_FUJITSU), SND_PCI_QUIRK(0x1734, 0x10cb, "Fujitsu Si3515", CXT5045_LAPTOP), SND_PCI_QUIRK(0x1734, 0x110e, "Fujitsu V5505", CXT5045_LAPTOP), @@ -819,6 +864,16 @@ static int patch_cxt5045(struct hda_code spec->mixers[0] = cxt5045_mixers; codec->patch_ops.init = cxt5045_init; break; + case CXT5045_BENQ: + codec->patch_ops.unsol_event = cxt5045_hp_unsol_event; + spec->input_mux = &cxt5045_capture_source_benq; + spec->num_init_verbs = 1; + spec->init_verbs[0] =cxt5045_benq_init_verbs; + spec->mixers[0] = cxt5045_mixers; + spec->mixers[1] = cxt5045_benq_mixers; + spec->num_mixers = 2; + codec->patch_ops.init = cxt5045_init; + break; #ifdef CONFIG_SND_DEBUG case CXT5045_TEST: spec->input_mux = &cxt5045_test_capture_source;

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Re: [alsa-devel] intel-hd on an intel G33 motherboard
by De Fuuss 17 Jan '08

17 Jan '08

Ups my bad, somehow the model number got commented out by an update script I ran yesterday. The drivers were recompiled with the "with-debug=detect" option set. I've attached the dmesg, codec#0 output and amixer contents. Dmesg is not very talkative, it did not show "hda_codec_setup_stream:NID=", however I get something of the type "hda_generic: no proper output path found". About the testing, I performed speaker-test every time I changed something (still only output on "Front L/R channels"). Patrick > From: zhe.jiang(a)intel.com > To: fuuss(a)hotmail.com > Date: Wed, 16 Jan 2008 13:49:43 +0800 > CC: cmatsuoka(a)gmail.com; alsa-devel(a)alsa-project.org > Subject: Re: [alsa-devel] intel-hd on an intel G33 motherboard > > On Wed, 2008-01-16 at 02:22 +0800, De Fuuss wrote: >> >> Hi, >> >> I have recompiled the driver with --with-debug=detect. Dmesg output is >> attached to this mail. ALSA specificities seem to be present this >> time. > Hi, > >>From the dmesg, there is "hda_codec: Unknown model for ALC883, trying > auto-probe from BIOS" > It looks like that no "model=" is assigned. > Did you assign model for the driver? > > Can you play some music? > Then there will be some info in the dmesg like > "hda_codec_setup_stream:NID= ". > Please paste such info and give the result of > "cat /proc/asound/card0/codec#0". > > BTW, how did you do the testing? > You run speaker-test or by other way? > >>From the codec info, it seems that there are paths from DAC to Jack. > So it's a little weird that all other jacks are silent. > > Thanks! > >> Hope this helps. >> >> Best regards, >> >> Patrick >> >>> From: zhe.jiang(a)intel.com >>> To: fuuss(a)hotmail.com >>> Date: Tue, 15 Jan 2008 09:32:35 +0800 >>> CC: cmatsuoka(a)gmail.com; alsa-devel(a)alsa-project.org >>> Subject: Re: [alsa-devel] intel-hd on an intel G33 motherboard >>> >>> On Mon, 2008-01-14 at 17:11 +0000, De Fuuss wrote: >>>> The jumper is now set to enable onboard (Intel HDA) audio. >>>> At the back of the barebone there are 5 jacks (4 channels for 7.1 >> and 1 line in channel) + 1 spdif out. >>>> From these, using 6stack-hp, only the "Front L/R" jack works, none >> of the others >>>> On the front, there are 2 jacks (1 headphones and 1 microphone). >> The headphone jack works. >>>> >>> Can you tell us how you did the testing? >>>>From the driver code, it's not likely that all other jacks don't >> work. >>> >>> Because the driver support up-mixing, you can play a 2-channel music >> and >>> get sound from 4 jacks on the back panel. >>> >>> I read the dmesg output that you sent to me. >>> But there is no alsa debug info in it. >>> Did you really configure the driver with the option >>> "--with-debug=detect"? >>> If not, please try it and paste the dmesg info, it is helpful to >> trace >>> the issue. >>> >>> Best regards! >>> Jiang,zhe >>>> All the channels have been set to 100% in alsamixer. >>>> >>>> Considering that a week ago I had 0.0 and today I'm at 2.0, it's >> already a improvement. ;-) >>>> >>>> Patrick >>>> >>>>> Subject: Re: [alsa-devel] intel-hd on an intel G33 motherboard >>>>> From: zhe.jiang(a)intel.com >>>>> To: fuuss(a)hotmail.com >>>>> CC: cmatsuoka(a)gmail.com; alsa-devel(a)alsa-project.org >>>>> Date: Mon, 14 Jan 2008 16:43:00 +0800 >>>>> >>>>> On Mon, 2008-01-14 at 02:06 +0800, De Fuuss wrote: >>>>>> >>>>>> Following your advice, I tested per channel output. Using either >>>>>> 6stack-dig or 6stack-hp, I get noise from Front Left and Right >> jacks, >>>>>> using speaker-test (X=1 and 2) but not on any other channels. >>>>> >>>>> Hi, >>>>> Did you mean that after you changed the 'Jumper', >>>>> you could get the sound from the Front jack on the back channel? >>>>> >>>>>> Patrick >>>>>> >>>>>>> Date: Sun, 13 Jan 2008 01:01:50 -0200 >>>>>>> From: cmatsuoka(a)gmail.com >>>>>>> To: fuuss(a)hotmail.com >>>>>>> CC: zhe.jiang(a)intel.com; alsa-devel(a)alsa-project.org >>>>>>> Subject: Re: [alsa-devel] intel-hd on an intel G33 motherboard >>>>>>> >>>>>>> 2008/1/12 De Fuuss : >>>>>>> >>>>>>>> I get the uneasy feeling that something is messed up with this >>>>>> board. Anyway, for completeness' sake, I attached the results of >>>>>> amixer contents, cat /proc/asound/card0/codec#0 and dmesg for the >>>>>> module loaded as 6stack-dig. I have to try 6stack-hp again to >> confirm >>>>>> if I get something there. >>>>>>> >>>>>>> I fixed the Inspiron output, and I'm adding the SG33G5M codec to >> the >>>>>>> codecgraph codecs collection (git repository at >>>>>>> http://helllabs.org/git/codecgraph.git). If nodes 0x14, 0x15, >> 0x16 >>>>>> and >>>>>>> 0x17 are the front, rear, clfe and side connectors in the back >>>>>> panel, >>>>>>> 6stack-hp should work (at least for the outputs). You can test >> the >>>>>>> outputs using "speaker-test -Dplug:surround71 -c8 -sX" (where X >>>>>> varies >>>>>>> from 1 to 8) and a pair of headphones. >>>>>>> _______________________________________________ >>>>>>> Alsa-devel mailing list >>>>>>> Alsa-devel(a)alsa-project.org >>>>>>> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >>>>>> >>>>>> _________________________________________________________________ >>>>>> Express yourself instantly with MSN Messenger! Download today >> it's >>>>>> FREE! >>>>>> http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ >>>>>> _______________________________________________ >>>>>> Alsa-devel mailing list >>>>>> Alsa-devel(a)alsa-project.org >>>>>> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >>>>>> >>>>>> >>>>> >>>> >>>> _________________________________________________________________ >>>> Express yourself instantly with MSN Messenger! Download today it's >> FREE! >>>> http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ >>> >>> _______________________________________________ >>> Alsa-devel mailing list >>> Alsa-devel(a)alsa-project.org >>> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >> >> _________________________________________________________________ >> Express yourself instantly with MSN Messenger! Download today it's >> FREE! >> http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ >> >> > > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

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Re: [alsa-devel] hdsp driver broken for Multiface II Rev 35
by Jean Batrel 17 Jan '08

17 Jan '08

> Date: Mon, 14 Jan 2008 02:21:42 +0100 > From: Andreas Degert <ad(a)papyrus-gmbh.de> > Subject: [alsa-devel] hdsp driver broken for Multiface II Rev 35 > To: alsa-devel(a)alsa-project.org > Cc: tiwai(a)suse.de > Message-ID: <20080114022142.51d6ae44(a)pluto.noname> > Content-Type: text/plain; charset=US-ASCII > > Hi folks, > > changeset 5326 from 2007/08/31 broke the driver for this card revision. > There was an older thread about this problem which didn't seem to come > to a conclusion. > (...) > Similar changes must be made to hdsp_input_to_output_key. Does anyone > know which firmware revision exist for which card (io_type) ? > > thank you > Andreas Degert > ------------------------------ Bonjour ! You are asking for firmware revision and (RME) cards : I use a HDSP9632 RME card with issues : my card is Rev. 1.53 and only the 1.51 version is working with Linux, AFAIK... The card is detected, there is some activity in the middle of HDSPmixer, but I can hear no sound... A lspci says : 02:01.0 Multimedia audio controller: Xilinx Corporation RME Hammerfall DSP (rev 99) Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap- 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 255 Interrupt: pin A routed to IRQ 20 Region 0: Memory at feaf0000 (32-bit, non-prefetchable) [size=64K] I have no very skills with Linux yet, but I very hope getting this card working with Linux. Can you help me please ? I very apologize if this mail is not relevant in this list. But I very thank you in advance (I use Fedora 8 and the latest CCRMA kernel) ! Jean Batrel - jean.batrel(a)voila.fr

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Re: [alsa-devel] ALC268 auto-parsing
by Jonathan Woithe 17 Jan '08

17 Jan '08

Marco Schuster wrote: > I am a bit interested in how the Realtek Windows driver gets its pin > config. There IMO are two options: > a) it uses the bios autoconfig - as parsing this doesn't work in alsa > here, there must be something special in it > b) it uses a config file - it should be easy to get out the pin config. I have no concrete evidence for this but my feeling based on a poke around the windoze drivers a few years ago is that they were configured for the particular laptop by the manufacturer. This configuration information is not in an obvious location though - if they go this way it's either embedded within the driver itself or located in a binary format file somewhere. In both cases this makes it almost impossible to utilise. Regards jonathan

1 0

[alsa-devel] SNDRV_PCM_INFO_MMAP - what do I need for it?
by Timur Tabi 16 Jan '08

16 Jan '08

My 8610 ASoC driver has this: static const struct snd_pcm_hardware fsl_dma_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED, This means that MMAP is not enabled. I understand what MMAP is, but not from an ALSA driver perspective. If I change the above line to this: .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID, what does ALSA do differently? Do I need to define any new functions? From an application perspective, what is different? Also, does DMIX need MMAP support? -- Timur Tabi Linux kernel developer at Freescale

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[alsa-devel] ALC268 auto-parsing
by Marco Schuster 16 Jan '08

16 Jan '08

Hi all, I am a bit interested in how the Realtek Windows driver gets its pin config. There IMO are two options: a) it uses the bios autoconfig - as parsing this doesn't work in alsa here, there must be something special in it b) it uses a config file - it should be easy to get out the pin config. Any opinions what is correct way? Marco

2 1

[alsa-devel] PNP_DRIVER_RES_DISABLE breaks swsusp at least with snd_cs4236
by Ondrej Zary 16 Jan '08

16 Jan '08

Hello, as hibernation (swsusp) started to work with my CPU, I found that my Turtle Beach Malibu stops working after resume from hibernation. It's caused by fact that the card is not enabled on the pnp layer during resume - and thus card registers are inaccessible (reads return FFs, writes go nowhere). During resume, pnp_bus_resume() in drivers/pnp/driver.c is called for each pnp device. This function calls pnp_start_dev() only when the PNP_DRIVER_RES_DO_NOT_CHANGE bit is NOT seting pnp_drv->flags. But the cs4236 driver in sound/isa/cs423x/cs4236.c explicitly sets the .flags to PNP_DRIVER_RES_DISABLE - it's value is 3 and that includes PNP_DRIVER_RES_DO_NOT_CHANGE bit. The same .flags value is present in many of the ALSA ISA sound drivers. Removing that .flags line caused this to appear inlog when loading snd_cs4236 module: CS4236+ WSS PnP manual resources are invalid, using auto config CS4236+ CTRL PnP manual resources are invalid, using auto config CS4236+ MPU401 PnP manual resources are invalid, using auto config and the sound now works after resume! So the question is: why is this line present? Is this a bug? What's the correct fix? -- Ondrej Zary

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[alsa-devel] TLV support for PCM (in driver)
by Jaroslav Kysela 16 Jan '08

16 Jan '08

Hi, I am actually thinking to add TLV support to each PCM stream. The reason is to obtain various "static" information from the driver without even opening the PCM stream. The current API lack: a) get channel info / name / position (e.g. Front Left Speaker) - not sure how to handle multichannel cards - I'm sure that this is something to be user configurable b) get possible channels c) get possible formats d) various inter-PCM-device constraint description (e.g. if S/PDIF PCM stream can be used with analog etc.) e) eventually more information for device name hinting for userspace (current problems for HDA, AC97, USB devices are that we have only one .conf file per driver covering all hardware - we need to detect real device capabilities and show only devices which can be really used) Opinions? Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

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[alsa-devel] hdsp driver broken for Multiface II Rev 35
by Andreas Degert 16 Jan '08

16 Jan '08

Hi folks, changeset 5326 from 2007/08/31 broke the driver for this card revision. There was an older thread about this problem which didn't seem to come to a conclusion. ioctl SNDRV_HDSP_IOCTL_GET_VERSION returns io_type = 1 (Multiface) firmware_rev = 0x35 which is correct. The old code for hdsp_playback_to_output_key looks like this: static int hdsp_playback_to_output_key (struct hdsp *hdsp, int in, int out) { switch (hdsp->firmware_rev) { case 0xa: return (64 * out) + (32 + (in)); case 0x96: case 0x97: return (32 * out) + (16 + (in)); default: return (52 * out) + (26 + (in)); } } For my card the result is "(52 * out) + (26 + (in))". The new code looks like this: static int hdsp_playback_to_output_key (struct hdsp *hdsp, int in, int out) { switch (hdsp->io_type) { case Multiface: case Digiface: default: return (64 * out) + (32 + (in)); case H9632: return (32 * out) + (16 + (in)); case H9652: return (52 * out) + (26 + (in)); } } which gives the wrong result "(64 * out) + (32 + (in))". The old code returns this when firmware_rev == 0xa, which might be the revision of the older Multiface (not Multiface II). I changed the code to static int hdsp_playback_to_output_key (struct hdsp *hdsp, int in, int out) { switch (hdsp->io_type) { case Multiface: case Digiface: default: if (hdsp->firmware_rev == 0xa) return (64 * out) + (32 + (in)); else return (52 * out) + (26 + (in)); case H9632: return (32 * out) + (16 + (in)); case H9652: return (52 * out) + (26 + (in)); } } Another possibility would be to go back to the old version of the code and only make changes for io_type H9632/H9652 with new firmware revisions: static int hdsp_playback_to_output_key (struct hdsp *hdsp, int in, int out) { switch (hdsp->firmware_rev) { case 0xa: return (64 * out) + (32 + (in)); case 0x96: case 0x97: return (32 * out) + (16 + (in)); } switch (hdsp->io_type) { case H9632: return (32 * out) + (16 + (in)); case Multiface: case H9652: return (52 * out) + (26 + (in)); case Digiface: ??? is there a Digiface with revision != 0xa ??? default: ??? HDSP_IO_Type Undefined ??? } } Similar changes must be made to hdsp_input_to_output_key. Does anyone know which firmware revision exist for which card (io_type) ? thank you Andreas Degert

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Re: [alsa-devel] Bug#436723: Playing sound hands on powerbook3, 5 with 2.6.21-2 kernel
by maximilian attems 16 Jan '08

16 Jan '08

[ adding alsa upstream on cc ] On Mon, Jan 14, 2008 at 12:21:11AM +0100, th.huth(a)googlemail.com wrote: > > Please have a look at > https://bugs.launchpad.net/ubuntu/+source/linux-source-2.6.20/+bug/87652/co… > and the corresponding patch to see wether this patch solves this sound > issue. woow, i thought to fire up "try newer kernel", but your patch still applies to current linus git: ~/src/linux-2.6$ patch -p0 --dry-run < ~/pmac.c.diff patching file sound/ppc/pmac.c Hunk #5 succeeded at 868 (offset -24 lines). Hunk #6 succeeded at 1207 (offset -47 lines). so bringing that to upstream alsa radar. -- maks

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[alsa-devel] I/O possible
by Mihaela Vitalariu 16 Jan '08

16 Jan '08

Hello everyone! I am coding a Linux sound application using the PCM interface. It plays fine but at unexpected intervals i sometimes get a SIGIO directly in my program, which exits printing the message "I/O possible". What can be happening? Any help would be greatly appreciated. Simbasaurus.

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[alsa-devel] hda_intel messages showing up in dmesg
by Andrew Paprocki 16 Jan '08

16 Jan '08

I'm seeing these messages show up in dmesg every once in a while. If anyone has ideas as to why I'd be seeing these, I can help track down any potential problems. I am having problems where ALSA playback stutters while using the OSS-compatibility layer for ALSA works fine with no stuttering. After inserting the module w/ model=6stack-dig: ACPI: PCI Interrupt 0000:00:14.2[A] -> GSI 16 (level, low) -> IRQ 16 hda_intel: azx_get_response timeout, switching to polling mode: last cmd=0x225f0d00 hda_intel: azx_get_response timeout, switching to single_cmd mode: last cmd=0x225f0d00 ACPI: PCI Interrupt 0000:01:05.2[B] -> GSI 19 (level, low) -> IRQ 18 hda-intel: Invalid position buffer, using LPIB read method instead. After inserting the module w/ no model defined: ACPI: PCI Interrupt 0000:00:14.2[A] -> GSI 16 (level, low) -> IRQ 16 hda_codec: Unknown model for ALC883, trying auto-probe from BIOS... ACPI: PCI Interrupt 0000:01:05.2[B] -> GSI 19 (level, low) -> IRQ 18 hda-intel: Invalid position buffer, using LPIB read method instead. I've only seen the azx_get_response messages print out once. I see the LPIB message print out every once in a while. Thanks, -Andrew

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[alsa-devel] [PATCH] remove 11c1:1040 from patch_si3054.c id list
by Vasily Khoruzhick 16 Jan '08

16 Jan '08

Codec with id 11c1:1040 sitting on hda bus isn't si3054-compatible. It should be removed from patch_si3054.c id list

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Re: [alsa-devel] intel-hd on an intel G33 motherboard
by De Fuuss 16 Jan '08

16 Jan '08

Hi, I have recompiled the driver with --with-debug=detect. Dmesg output is attached to this mail. ALSA specificities seem to be present this time. Hope this helps. Best regards, Patrick > From: zhe.jiang(a)intel.com > To: fuuss(a)hotmail.com > Date: Tue, 15 Jan 2008 09:32:35 +0800 > CC: cmatsuoka(a)gmail.com; alsa-devel(a)alsa-project.org > Subject: Re: [alsa-devel] intel-hd on an intel G33 motherboard > > On Mon, 2008-01-14 at 17:11 +0000, De Fuuss wrote: >> The jumper is now set to enable onboard (Intel HDA) audio. >> At the back of the barebone there are 5 jacks (4 channels for 7.1 and 1 line in channel) + 1 spdif out. >> From these, using 6stack-hp, only the "Front L/R" jack works, none of the others >> On the front, there are 2 jacks (1 headphones and 1 microphone). The headphone jack works. >> > Can you tell us how you did the testing? >>From the driver code, it's not likely that all other jacks don't work. > > Because the driver support up-mixing, you can play a 2-channel music and > get sound from 4 jacks on the back panel. > > I read the dmesg output that you sent to me. > But there is no alsa debug info in it. > Did you really configure the driver with the option > "--with-debug=detect"? > If not, please try it and paste the dmesg info, it is helpful to trace > the issue. > > Best regards! > Jiang,zhe >> All the channels have been set to 100% in alsamixer. >> >> Considering that a week ago I had 0.0 and today I'm at 2.0, it's already a improvement. ;-) >> >> Patrick >> >>> Subject: Re: [alsa-devel] intel-hd on an intel G33 motherboard >>> From: zhe.jiang(a)intel.com >>> To: fuuss(a)hotmail.com >>> CC: cmatsuoka(a)gmail.com; alsa-devel(a)alsa-project.org >>> Date: Mon, 14 Jan 2008 16:43:00 +0800 >>> >>> On Mon, 2008-01-14 at 02:06 +0800, De Fuuss wrote: >>>> >>>> Following your advice, I tested per channel output. Using either >>>> 6stack-dig or 6stack-hp, I get noise from Front Left and Right jacks, >>>> using speaker-test (X=1 and 2) but not on any other channels. >>> >>> Hi, >>> Did you mean that after you changed the 'Jumper', >>> you could get the sound from the Front jack on the back channel? >>> >>>> Patrick >>>> >>>>> Date: Sun, 13 Jan 2008 01:01:50 -0200 >>>>> From: cmatsuoka(a)gmail.com >>>>> To: fuuss(a)hotmail.com >>>>> CC: zhe.jiang(a)intel.com; alsa-devel(a)alsa-project.org >>>>> Subject: Re: [alsa-devel] intel-hd on an intel G33 motherboard >>>>> >>>>> 2008/1/12 De Fuuss : >>>>> >>>>>> I get the uneasy feeling that something is messed up with this >>>> board. Anyway, for completeness' sake, I attached the results of >>>> amixer contents, cat /proc/asound/card0/codec#0 and dmesg for the >>>> module loaded as 6stack-dig. I have to try 6stack-hp again to confirm >>>> if I get something there. >>>>> >>>>> I fixed the Inspiron output, and I'm adding the SG33G5M codec to the >>>>> codecgraph codecs collection (git repository at >>>>> http://helllabs.org/git/codecgraph.git). If nodes 0x14, 0x15, 0x16 >>>> and >>>>> 0x17 are the front, rear, clfe and side connectors in the back >>>> panel, >>>>> 6stack-hp should work (at least for the outputs). You can test the >>>>> outputs using "speaker-test -Dplug:surround71 -c8 -sX" (where X >>>> varies >>>>> from 1 to 8) and a pair of headphones. >>>>> _______________________________________________ >>>>> Alsa-devel mailing list >>>>> Alsa-devel(a)alsa-project.org >>>>> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >>>> >>>> _________________________________________________________________ >>>> Express yourself instantly with MSN Messenger! Download today it's >>>> FREE! >>>> http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ >>>> _______________________________________________ >>>> Alsa-devel mailing list >>>> Alsa-devel(a)alsa-project.org >>>> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >>>> >>>> >>> >> >> _________________________________________________________________ >> Express yourself instantly with MSN Messenger! Download today it's FREE! >> http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ > > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

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Re: [alsa-devel] Splitting up the HDA-INTEL modules
by Jonathan Woithe 16 Jan '08

16 Jan '08

Marco Schuster wrote: > when I looked today through the files of the snd-hda-intel module, I > was a bit shocked by this complex amount of code (patch_realtek.c for > example). I would think about splitting up the code in a way each > chipset gets its own .c file - which will make debugging, patching or > editing a lot easier IMO, and for second people can then comment on > just their specific chipset in config. The problem with such a split is, as Tobin went on to say, that there are a lot of interdependencies and shared code within each chipset's family: > I don't think that it would be very easy, as there are several instances > where code relies on other codec definitions. One could argue that therefore we have toplevel realtek/sigmatek/etc "librarys" which are used by each chipset-specific driver but IMHO that is hardly going to count as a simplification. Tobin then said: > In all actuality, what is needed is a more generic driver that is driven > by userspace tools. ... Maybe we can get a core group together and > actually define and implement an API for this. I think this idea has merit and may be worth pursuing. I would be interested in assisting and providing input to the process although I haven't time at present to take a lead role. > My idea is a driver that takes settings from a config file that are > either 32bit register values (aka Sigmatel), or individual Node > parameters (Realtek, Conexant, etc). This way we can define model > config files, and associate their PCI Quirk in a central config file > that can be edited in userspace. This would be much easier to support, > for users, distro vendors, hardware vendors, and us. Agreed, at least in theory. There may be complications due to the varying topography of the chips; the userspace API may have to be relatively complex to support things like record source switch options (which are vastly different for diffenent chips). Even so, if we can get this to work the ongoing support effort for HDA will be much reduced so it's probably worth the effort. > I'll try to write up a proposal in the next few weeks, if we can get > others on board to work on this. I think this would be worth looking at. Regards jonathan

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[alsa-devel] HDA: Enable chipset gcap usage
by Tobin Davis 15 Jan '08

15 Jan '08

I tweaked this as promised, and ran it through the checkpatch.pl utility from kernel git tree. I have tested it on my Asus Z62F laptop as well as a few systems in my work environment without issue. Summary: hda_intel.c use gcap register to determine number of streams supported by southbridge. This patch quieries the southbridge for the number of streams supported by the southbridge in most chipsets, and reads these values from the chipset directly. Most systems are hardwired for 4 streams in each direction, but newer chipsets change that capability. As a fallback, the old hardcoded method is still available. Signed off by Tobin Davis <tdavis(a)dsl-only.net>

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[alsa-devel] Fwd: surround51 lfe/center/front channel cross-talk with intel8x0 board
by Matt Hirsch 15 Jan '08

15 Jan '08

I sent this to alsa-users and didn't get any response. I'd be grateful if someone had some suggestions for getting around the problem described below. Thanks very much, Matt ---------- Forwarded message ---------- From: Matt Hirsch <matthew.hirsch(a)gmail.com> Date: Jan 5, 2008 1:20 AM Subject: surround51 lfe/center/front channel cross-talk with intel8x0 board To: alsa-user(a)lists.sourceforge.net I have an AC'97 based onboard sound card in my system (snd_intel8x0 driver). I'm using it in 6ch mode to drive two front speakers, two rear speakers and a subwoofer and center channel. Using speaker-test with the surround51 plug (or any other alsa program with that plug) to send audio to channel 5 (center in alsamixer) produces an attenuated signal on channel 2 (right front in alsamixer) and likewise, sending audio to channel 6 (lfe in alsamixer) produces an attenuated signal on channel 1 (left front in alsamixer). In summary: speaker-test -tsine -f200 -Dsurround51 -c6 -s1 tone on left front speaker only speaker-test -tsine -f200 -Dsurround51 -c6 -s2 tone on right front speaker only speaker-test -tsine -f200 -Dsurround51 -c6 -s3 tone on left rear speaker only speaker-test -tsine -f200 -Dsurround51 -c6 -s4 tone on right rear speaker only speaker-test -tsine -f200 -Dsurround51 -c6 -s5 tone on center speaker, and quieter tone on right front speaker speaker-test -tsine -f200 -Dsurround51 -c6 -s6 tone on lfe, and quieter tone on left front speaker Any help or suggestions as to getting rid of this crosstalk would be greatly appreciated! Here's what I've tried so far: If I don't use the surround51 plug, and use default (hw:0) instead, the channels are all in a different order, but the same effect occurs (crosstalk between two channels). I tried sequentially setting and un-setting all of the modes and buttons in the alsa mixer, such as high-pass filter (on/off), surround jack mode (shared/independent), line jack sense (on/off), exchanging front/surround, etc. Nothing seemed to have any effect. When I exchange the front and surround speakers, the cross talk is also exchanged (i.e. it shows up on the rear speakers rather than the front speakers). I'm running fedora 8, so I started with the latest fedora 8 alsa rpms (1.15). When this didn't work I compiled a recent CVS snapshot (12/25/07) against my kernel and installed it. Same results as above. My hardware shows up in lspci as: 00:02.7 Multimedia audio controller: Silicon Integrated Systems [SiS] AC'97 Sound Controller (rev a0) My kernel is 2.6.23.9-85.fc8 Thanks, Matt

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[alsa-devel] ALC268 test model not working
by Marco Schuster 15 Jan '08

15 Jan '08

Hi all, finally I got the patch applied to kernel src :) But it doesn't work correctly, because when running alsamixer, dmesg tells "hda_codec: num_steps = 0 for NID=0x1d", and alsamixer fails with alsamixer: function snd_mixer_load failed: Invalid argument. Is this some obscure bug in the notebook hardware or some driver bug? Marco (note, I apply against .24-rc6 still)

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[alsa-devel] Red Hat
by Jaroslav Kysela 15 Jan '08

15 Jan '08

Hello all, I would like to annouce that I work for Red Hat, Inc. since beginning of year. You might noticed this in my signature for awhile already. I would like to thank to my previous employer - Novell, Inc. (SUSE) - for the support for the ALSA project and looking forward to continue with the ALSA project development under new employer. Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

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Re: [alsa-devel] intel-hd on an intel G33 motherboard
by De Fuuss 15 Jan '08

15 Jan '08

Hi, I reopened my shuttle case, closely inspected the motherboard and I think I found the issue. The onboard sound was factory set off (jumper), something which I must admit I did not check. However, I find it hard to imagine that XPC would sell their barebones with onboard sound off as a default. Even more weird, the headphones work when the onboard sound is "off". I get the uneasy feeling that something is messed up with this board. Anyway, for completeness' sake, I attached the results of amixer contents, cat /proc/asound/card0/codec#0 and dmesg for the module loaded as 6stack-dig. I have to try 6stack-hp again to confirm if I get something there. Patrick > From: zhe.jiang(a)intel.com > To: fuuss(a)hotmail.com > Date: Fri, 11 Jan 2008 12:58:07 +0800 > CC: alsa-devel(a)alsa-project.org > Subject: Re: [alsa-devel] intel-hd on an intel G33 motherboard > > On Thu, 2008-01-10 at 16:09 +0000, De Fuuss wrote: >> Hi, >> >> 6stack-dig, 6stack-hp: I tested all channels, and indeed I get sound but only from the headphone jack on the front. All other channels remain silent (front, center,rear,...) >> > Hi, > > It's a little weird that the sound only come from headphone. > Seems that the 0x1b is the hp pin and 0x14 is the Front line out pin. > Both of them are connected to 0x0c widget. > If there is sound on hp, there should be sound on Front line out. > > May you rebuild the alsa-driver with debug option to get some info? > ./configure --with-debug=detect; > make;make install; > > After reload the driver, > please show the result of "amixer contents", > "cat /proc/asound/card0/codec#0" and "dmesg". > > Thanks! >> Regards, >> >> Patrick >> >>> From: zhe.jiang(a)intel.com >>> To: fuuss(a)hotmail.com >>> Date: Thu, 10 Jan 2008 13:46:42 +0800 >>> CC: alsa-devel(a)alsa-project.org >>> Subject: Re: [alsa-devel] intel-hd on an intel G33 motherboard >>> >>> On Fri, 2007-12-28 at 19:09 +0800, De Fuuss wrote: >>>> >>>> Hi, >>>> >>>> Here's the pastebin information http://pastebin.ca/834640 >>>> >>>> That was using 3stack-6ch-dig >>> The 3stack-6ch-dig won't work because it use 0x18 and 0x1a as >>> the effective pins, which is different from your pins layout. >>>> >>>> For experimentation's sake: >>>> >>>> If I use 6stack-dig I get the following http://pastebin.ca/834644 >>>> If I use 6stack-hp I get the following http://pastebin.ca/834646 >>>> >>> Did you do any testing with those two configures? >>> After analyzing your information, >>> I guess that you can get voice from the output jacks with those >>> configurations. >>> >>> >>>> Thanks for helping me out. It would be nice if it worked :-) >>>> >>>> Patrick >>>> >>>>> From: tdavis(a)dsl-only.net >>>>> To: fuuss(a)hotmail.com >>>>> Date: Thu, 27 Dec 2007 20:22:57 -0800 >>>>> CC: alsa-devel(a)alsa-project.org >>>>> Subject: Re: [alsa-devel] intel-hd on an intel G33 motherboard >>>>> >>>>> Ok, need to step back in the thread a bit. Now that we've >>>> established >>>>> that this is not an Intel manufactured motherboard (not that they >>>> make >>>>> their own anymore), I need more detailed information on what the OS >>>>> see's. Please run >>>>> http://bulletproof.servebeer.com/alsa/scripts/alsa-info.sh and post >>>> the >>>>> link that it generates. This will give me a detailed snapshot of >>>> your >>>>> hardware that I can use for debugging. >>>>> >>>>> I will begin working on this as soon as I get this info. I hope to >>>> get >>>>> you enabled this weekend if possible, this year at the latest. :) >>>>> >>>>> Thanks, >>>>> >>>>> Tobin >>>> ---------------------------------------- >>>>> From: tdavis(a)dsl-only.net >>>>> To: fuuss(a)hotmail.com >>>>> Date: Thu, 27 Dec 2007 20:22:57 -0800 >>>>> CC: alsa-devel(a)alsa-project.org >>>>> Subject: Re: [alsa-devel] intel-hd on an intel G33 motherboard >>>>> >>>>> Ok, need to step back in the thread a bit. Now that we've >>>> established >>>>> that this is not an Intel manufactured motherboard (not that they >>>> make >>>>> their own anymore), I need more detailed information on what the OS >>>>> see's. Please run >>>>> http://bulletproof.servebeer.com/alsa/scripts/alsa-info.sh and post >>>> the >>>>> link that it generates. This will give me a detailed snapshot of >>>> your >>>>> hardware that I can use for debugging. >>>>> >>>>> I will begin working on this as soon as I get this info. I hope to >>>> get >>>>> you enabled this weekend if possible, this year at the latest. :) >>>>> >>>>> Thanks, >>>>> >>>>> Tobin >>>>> >>>>> >>>>> On Thu, 2007-12-27 at 00:33 +0000, De Fuuss wrote: >>>>> >>>>>> Hi Tobin, >>>>>> >>>>>> Thanks for you quick reply. >>>>>> >>>>>> I tried both 3stack-6ch and 3stack-6ch-dig (the card has a spdif >>>> out). I found out that when putting on headphones on a separate >>>> headphone jack I get some sound with both settings. Hoever, the front >>>> out l/r, center/bass sourround back l/r and side sourround l/r port >>>> remain silent. >>>>>> >>>>>> I set all channels to 80-100% in alsamixer. >>>>>> >>>>>> A few more details from my setup: >>>>>> Shuttle XPC SG33G5M >>>>>> Intel G33 Express + ICH9DH >>>>>> >>>>>> Interestingly, when I run alsamixer, it identifies the Chip as >>>> "Generic 1095 ID 1392", in other words in points to codec#1. I thought >>>> it should point to codec#0, e.g. RealTek ALC888 >>>>>> >>>>>> I have the strong impression that the channels are not truly set by >>>> alsamixer as they point to the wrong codec. I'm only a layman in ALSA, >>>> and I have no idea how to tell alsamixer to consider codec#0 instead >>>> codec#1. >>>>>> >>>>>> Do you have any idea on how to crack this nut? >>>>>> >>>>>> Below is the output of both codecs: >>>>>> >>>>>> Thank you. >>>>>> >>>>>> Best regards, >>>>>> >>>>>> Patrick >>>>>> >>>>>> ____________________ >>>>>> #cat /proc/asound/cards >>>>>> >>>>>> 0 [Intel ]: HDA-Intel - HDA Intel >>>>>> HDA Intel at 0xfdff8000 irq 22 >>>>>> >>>>>> # cat /proc/asound/Intel/codec#0 >>>>>> Codec: Realtek ALC888 >>>>>> Address: 0 >>>>>> Vendor Id: 0x10ec0888 >>>>>> Subsystem Id: 0x12970888 >>>>>> Revision Id: 0x100001 >>>>>> No Modem Function Group found >>>>>> Default PCM: >>>>>> rates [0x560]: 44100 48000 96000 192000 >>>>>> bits [0xe]: 16 20 24 >>>>>> formats [0x1]: PCM >>>>>> Default Amp-In caps: N/A >>>>>> Default Amp-Out caps: N/A >>>>>> Node 0x02 [Audio Output] wcaps 0x11: Stereo >>>>>> PCM: >>>>>> rates [0x560]: 44100 48000 96000 192000 >>>>>> bits [0xe]: 16 20 24 >>>>>> formats [0x1]: PCM >>>>>> Node 0x03 [Audio Output] wcaps 0x11: Stereo >>>>>> PCM: >>>>>> rates [0x560]: 44100 48000 96000 192000 >>>>>> bits [0xe]: 16 20 24 >>>>>> formats [0x1]: PCM >>>>>> Node 0x04 [Audio Output] wcaps 0x11: Stereo >>>>>> PCM: >>>>>> rates [0x560]: 44100 48000 96000 192000 >>>>>> bits [0xe]: 16 20 24 >>>>>> formats [0x1]: PCM >>>>>> Node 0x05 [Audio Output] wcaps 0x11: Stereo >>>>>> PCM: >>>>>> rates [0x560]: 44100 48000 96000 192000 >>>>>> bits [0xe]: 16 20 24 >>>>>> formats [0x1]: PCM >>>>>> Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital >>>>>> PCM: >>>>>> rates [0x5e0]: 44100 48000 88200 96000 192000 >>>>>> bits [0x1e]: 16 20 24 32 >>>>>> formats [0x1]: PCM >>>>>> Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono >>>>>> Node 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In >>>>>> Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 >>>>>> Amp-In vals: [0x9f 0x9f] >>>>>> PCM: >>>>>> rates [0x160]: 44100 48000 96000 >>>>>> bits [0x6]: 16 20 >>>>>> formats [0x1]: PCM >>>>>> Connection: 1 >>>>>> 0x23 >>>>>> Node 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In >>>>>> Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 >>>>>> Amp-In vals: [0x9f 0x9f] >>>>>> PCM: >>>>>> rates [0x160]: 44100 48000 96000 >>>>>> bits [0x6]: 16 20 >>>>>> formats [0x1]: PCM >>>>>> Connection: 1 >>>>>> 0x22 >>>>>> Node 0x0a [Audio Input] wcaps 0x100391: Stereo Digital >>>>>> PCM: >>>>>> rates [0x560]: 44100 48000 96000 192000 >>>>>> bits [0x1e]: 16 20 24 32 >>>>>> formats [0x1]: PCM >>>>>> Connection: 1 >>>>>> 0x1f >>>>>> Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In >>>>>> Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 >>>>>> Amp-In vals: [0x80 0x80] [0x80 0x80] [0x9d 0x9d] [0x80 0x80] >>>> [0x9f 0x9f] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 >>>> 0x80] >>>>>> Connection: 10 >>>>>> 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 >>>>>> Node 0x0c [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out >>>>>> Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >>>>>> Amp-In vals: [0x00 0x00] [0x00 0x00] >>>>>> Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 >>>>>> Amp-Out vals: [0x1d 0x1d] >>>>>> Connection: 2 >>>>>> 0x02 0x0b >>>>>> Node 0x0d [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out >>>>>> Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >>>>>> Amp-In vals: [0x00 0x00] [0x00 0x00] >>>>>> Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 >>>>>> Amp-Out vals: [0x1f 0x1f] >>>>>> Connection: 2 >>>>>> 0x03 0x0b >>>>>> Node 0x0e [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out >>>>>> Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >>>>>> Amp-In vals: [0x00 0x00] [0x00 0x00] >>>>>> Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 >>>>>> Amp-Out vals: [0x1f 0x1f] >>>>>> Connection: 2 >>>>>> 0x04 0x0b >>>>>> Node 0x0f [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out >>>>>> Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >>>>>> Amp-In vals: [0x80 0x80] [0x80 0x80] >>>>>> Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 >>>>>> Amp-Out vals: [0x00 0x00] >>>>>> Connection: 2 >>>>>> 0x05 0x0b >>>>>> Node 0x10 [Vendor Defined Widget] wcaps 0xf00000: Mono >>>>>> Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono >>>>>> Node 0x12 [Vendor Defined Widget] wcaps 0xf00000: Mono >>>>>> Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono >>>>>> Node 0x14 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out >>>>>> Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 >>>>>> Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] >>>> [0x00 0x00] >>>>>> Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >>>>>> Amp-Out vals: [0x00 0x00] >>>>>> Pincap 0x083e: IN OUT HP Detect >>>>>> Pin Default 0x01014010: [Jack] Line Out at Ext Rear >>>>>> Conn = 1/8, Color = Green >>>>>> Pin-ctls: 0x40: OUT >>>>>> Connection: 5 >>>>>> 0x0c* 0x0d 0x0e 0x0f 0x26 >>>>>> Node 0x15 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out >>>>>> Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 >>>>>> Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] >>>> [0x00 0x00] >>>>>> Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >>>>>> Amp-Out vals: [0x00 0x00] >>>>>> Pincap 0x083e: IN OUT HP Detect >>>>>> Pin Default 0x01011012: [Jack] Line Out at Ext Rear >>>>>> Conn = 1/8, Color = Black >>>>>> Pin-ctls: 0x40: OUT >>>>>> Connection: 5 >>>>>> 0x0c 0x0d* 0x0e 0x0f 0x26 >>>>>> Node 0x16 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out >>>>>> Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 >>>>>> Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] >>>> [0x00 0x00] >>>>>> Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >>>>>> Amp-Out vals: [0x00 0x00] >>>>>> Pincap 0x0836: IN OUT Detect >>>>>> Pin Default 0x01016011: [Jack] Line Out at Ext Rear >>>>>> Conn = 1/8, Color = Orange >>>>>> Pin-ctls: 0x40: OUT >>>>>> Connection: 5 >>>>>> 0x0c 0x0d 0x0e* 0x0f 0x26 >>>>>> Node 0x17 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out >>>>>> Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 >>>>>> Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] >>>> [0x00 0x00] >>>>>> Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >>>>>> Amp-Out vals: [0x00 0x00] >>>>>> Pincap 0x0836: IN OUT Detect >>>>>> Pin Default 0x01012014: [Jack] Line Out at Ext Rear >>>>>> Conn = 1/8, Color = Grey >>>>>> Pin-ctls: 0x40: OUT >>>>>> Connection: 5 >>>>>> 0x0c 0x0d 0x0e 0x0f* 0x26 >>>>>> Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out >>>>>> Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 >>>>>> Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] >>>> [0x00 0x00] >>>>>> Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >>>>>> Amp-Out vals: [0x00 0x00] >>>>>> Pincap 0x08373e: IN OUT HP Detect >>>>>> Pin Default 0x01813040: [Jack] Line In at Ext Rear >>>>>> Conn = 1/8, Color = Blue >>>>>> Pin-ctls: 0x40: OUT >>>>>> Connection: 5 >>>>>> 0x0c 0x0d 0x0e* 0x0f 0x26 >>>>>> Node 0x19 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out >>>>>> Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 >>>>>> Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] >>>> [0x00 0x00] >>>>>> Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >>>>>> Amp-Out vals: [0x80 0x80] >>>>>> Pincap 0x08373e: IN OUT HP Detect >>>>>> Pin Default 0x02a19850: [Jack] Mic at Ext Front >>>>>> Conn = 1/8, Color = Pink >>>>>> Pin-ctls: 0x24: IN >>>>>> Connection: 5 >>>>>> 0x0c* 0x0d 0x0e 0x0f 0x26 >>>>>> Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out >>>>>> Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 >>>>>> Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] >>>> [0x00 0x00] >>>>>> Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >>>>>> Amp-Out vals: [0x00 0x00] >>>>>> Pincap 0x08373e: IN OUT HP Detect >>>>>> Pin Default 0x9993014f: [Fixed] Aux at Int ATAPI >>>>>> Conn = ATAPI, Color = Unknown >>>>>> Pin-ctls: 0x40: OUT >>>>>> Connection: 5 >>>>>> 0x0c 0x0d* 0x0e 0x0f 0x26 >>>>>> Node 0x1b [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out >>>>>> Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 >>>>>> Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] >>>> [0x00 0x00] >>>>>> Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >>>>>> Amp-Out vals: [0x00 0x00] >>>>>> Pincap 0x08373e: IN OUT HP Detect >>>>>> Pin Default 0x02214020: [Jack] HP Out at Ext Front >>>>>> Conn = 1/8, Color = Green >>>>>> Pin-ctls: 0xc0: OUT HP >>>>>> Connection: 5 >>>>>> 0x0c* 0x0d 0x0e 0x0f 0x26 >>>>>> Node 0x1c [Pin Complex] wcaps 0x400001: Stereo >>>>>> Pincap 0x0820: IN >>>>>> Pin Default 0x411111f0: [N/A] Speaker at Ext Rear >>>>>> Conn = 1/8, Color = Black >>>>>> Pin-ctls: 0x20: IN >>>>>> Node 0x1d [Pin Complex] wcaps 0x400000: Mono >>>>>> Pincap 0x0820: IN >>>>>> Pin Default 0x4005a603: [N/A] Line Out at Ext N/A >>>>>> Conn = Optical, Color = UNKNOWN >>>>>> Pin-ctls: 0x20: IN >>>>>> Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital >>>>>> Pincap 0x0810: OUT >>>>>> Pin Default 0x01451130: [Jack] SPDIF Out at Ext Rear >>>>>> Conn = Optical, Color = Black >>>>>> Pin-ctls: 0x40: OUT >>>>>> Connection: 1 >>>>>> 0x06 >>>>>> Node 0x1f [Pin Complex] wcaps 0x400200: Mono Digital >>>>>> Pincap 0x0820: IN >>>>>> Pin Default 0x01c59160: [Jack] SPDIF In at Ext Rear >>>>>> Conn = Optical, Color = Pink >>>>>> Pin-ctls: 0x20: IN >>>>>> Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono >>>>>> Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono >>>>>> Node 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In >>>>>> Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >>>>>> Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x80 0x80] >>>> [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 >>>> 0x80] [0x80 0x80] >>>>>> Connection: 11 >>>>>> 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b >>>>>> Node 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In >>>>>> Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >>>>>> Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x80 0x80] >>>> [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 >>>> 0x80] [0x80 0x80] >>>>>> Connection: 11 >>>>>> 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b >>>>>> Node 0x24 [Vendor Defined Widget] wcaps 0xf00000: Mono >>>>>> Node 0x25 [Audio Output] wcaps 0x11: Stereo >>>>>> PCM: >>>>>> rates [0x560]: 44100 48000 96000 192000 >>>>>> bits [0xe]: 16 20 24 >>>>>> formats [0x1]: PCM >>>>>> Node 0x26 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out >>>>>> Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >>>>>> Amp-In vals: [0x00 0x00] [0x80 0x80] >>>>>> Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 >>>>>> Amp-Out vals: [0x00 0x00] >>>>>> Connection: 2 >>>>>> 0x25 0x0b >>>>>> >>>>>> A second codec sits at /proc/asound/Intel/codec#0. A cat results >>>> in: >>>>>> >>>>>> Codec: Generic 1095 ID 1392 >>>>>> Address: 1 >>>>>> Vendor Id: 0x10951392 >>>>>> Subsystem Id: 0xffffffff >>>>>> Revision Id: 0x100000 >>>>>> No Modem Function Group found >>>>>> Default PCM: >>>>>> rates [0x0]: >>>>>> bits [0x0]: >>>>>> formats [0x0]: >>>>>> Default Amp-In caps: N/A >>>>>> Default Amp-Out caps: N/A >>>>>> Node 0x02 [Audio Output] wcaps 0x6211: Stereo Digital >>>>>> PCM: >>>>>> rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 >>>>>> bits [0x1e]: 16 20 24 32 >>>>>> formats [0x5]: PCM AC3 >>>>>> Node 0x03 [Pin Complex] wcaps 0x40738d: Stereo Digital Amp-Out >>>>>> Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >>>>>> Amp-Out vals: [0x80 0x80] >>>>>> Pincap 0x0894: OUT Detect >>>>>> Pin Default 0x18560010: [Jack] Digital Out at Int HDMI >>>>>> Conn = Digital, Color = Unknown >>>>>> Pin-ctls: 0x00: >>>>>> Connection: 1 >>>>>> 0x02 >>>>>> >>>>>> >>>>>>> From: tdavis(a)dsl-only.net >>>>>>> To: fuuss(a)hotmail.com >>>>>>> Date: Wed, 26 Dec 2007 11:30:28 -0800 >>>>>>> CC: alsa-devel(a)alsa-project.org >>>>>>> Subject: Re: [alsa-devel] intel-hd on an intel G33 motherboard >>>>>>> >>>>>>> Try "model=3stack-6ch". Also, if you look in the >>>>>>> >>>> alsa-driver-1.0.15/alsa-kernel/Documentation/Alsa-Configuration.txt >>>>>>> file, there are other models you can select for the alc883/888 >>>> codec. >>>>>>> >>>>>>> Let me know if this works and I'll add your system to the driver. >>>>>>> >>>>>>> Tobin >>>>>>> >>>>>>> On Wed, 2007-12-26 at 18:42 +0000, De Fuuss wrote: >>>>>>> >>>>>>>> Hi, >>>>>>>> >>>>>>>> Maybe I'm missing something, but I've failed to get even the >>>> slightest >>>>>>>> noise out of the onboard sound card from a Intel G33 motherboard. >>>> I >>>>>>>> tried the alsa drivers from the alsa-driver-1.0.15 package as >>>> well as >>>>>>>> the the alsa drivers provided by kernel 2.6.23 (built as >>>> modules), to >>>>>>>> no avail. >>>>>>>> >>>>>>>> The sound chip is identified as a Intel HDA / Realtek ALC888 on >>>> the Intel G33 ICH9 motherboard. >>>>>>>> >>>>>>>> After reading some of the documentation from the ALSA site and >>>> googling through half the internet I haven't found out yet if this >>>> specific sound card is supported by ALSA and if some module options >>>> must be passed (such as options snd-intel-hda model=.....). >>>>>>>> >>>>>>>> Could somebody be so kind as to point me into the right >>>> direction? >>>>>>>> >>>>>>>> Thank you. >>>>>>>> >>>>>>>> Best regards, >>>>>>>> >>>>>>>> Patrick >>>>>>>> >>>>>>>> _________________________________________________________________ >>>>>>>> Express yourself instantly with MSN Messenger! Download today >>>> it's FREE! >>>>>>>> http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ >>>>>>>> _______________________________________________ >>>>>>>> Alsa-devel mailing list >>>>>>>> Alsa-devel(a)alsa-project.org >>>>>>>> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >>>>>>> >>>>>>> -- >>>>>>> Tobin Davis >>>>>>> >>>>>>> >>>>>>> TONY RANDALL! Is YOUR life a PATIO of FUN?? >>>>>>> _______________________________________________ >>>>>>> Alsa-devel mailing list >>>>>>> Alsa-devel(a)alsa-project.org >>>>>>> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >>>>>> >>>>>> _________________________________________________________________ >>>>>> Express yourself instantly with MSN Messenger! Download today it's >>>> FREE! >>>>>> http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ >>>>>> _______________________________________________ >>>>>> Alsa-devel mailing list >>>>>> Alsa-devel(a)alsa-project.org >>>>>> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >>>>> >>>>> -- >>>>> Tobin Davis >>>>> >>>>> >>>>> Authors are easy to get on with -- if you're fond of children. >>>>> -- Michael Joseph, "Observer" >>>> >>>> _________________________________________________________________ >>>> Express yourself instantly with MSN Messenger! Download today it's >>>> FREE! >>>> http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ >>>> _______________________________________________ >>>> Alsa-devel mailing list >>>> Alsa-devel(a)alsa-project.org >>>> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >>>> >>>> >>> >>> _______________________________________________ >>> Alsa-devel mailing list >>> Alsa-devel(a)alsa-project.org >>> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >> >> _________________________________________________________________ >> Express yourself instantly with MSN Messenger! Download today it's FREE! >> http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ > > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

3 7

[alsa-devel] [PATCH] cs4231: remove one busy wait
by Krzysztof Helt 14 Jan '08

14 Jan '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This busy_wait is not needed after latest changes to the cs4231-lib Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- Just after this busy wait is 1ms sleep then waiting loop. --- linux-2.6.22/sound/isa/cs423x/cs4231_lib.c 2007-09-11 00:02:15.000000000 +0200 +++ linux-2.6.23/sound/isa/cs423x/cs4231_lib.c 2007-09-11 00:03:36.619095346 +0200 @@ -332,7 +332,6 @@ void snd_cs4231_mce_down(struct snd_cs42 !(chip->hardware & (CS4231_HW_CS4231_MASK | CS4231_HW_CS4232_MASK))) { return; } - snd_cs4231_busy_wait(chip); /* * Wait for (possible -- during init auto-calibration may not be set) ---------------------------------------------------------------------- Chcesz kupic aparat? Sprawdz, ktory jest najlepszy! Kliknij >>> http://link.interia.pl/f1cd3

2 1

[alsa-devel] Si3054 and Lenovo3000 n100
by Vasily Khoruzhick 14 Jan '08

14 Jan '08

Hi, I have problems with setting up internal modem (regular si3054 sitting on hda bus) with subj notebook. During boot I get following message in dmesg: ALSA /var/tmp/portage/media-sound/alsa-driver-9999/work/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/patch_si3054.c:237: si3054: cannot initialize. EXT MID = 0000 And then when I try to use slmodem I get following repeating messages from slmodemd: <598.472031> main: alsa xrun: try to recover... <598.475735> main: alsa xrun: recovered. <598.475792> main: dev read = 0 I've tried to use alsa-driver-1.0.15 and latest alsa-driver from hg tree I can provide any additional info, just ask ;) Regards Vasily

3 4

Re: [alsa-devel] [Alsa-user] Intel HDMI Audio Output
by Lee Revell 14 Jan '08

14 Jan '08

On Jan 13, 2008 5:46 AM, Paul McEnery <pmcenery(a)gmail.com> wrote: > Following an email to the manufacturers European support representatives, I > was asked to try the following custom version of ALSA: > > > http://www.realtek.com.tw/downloads/downloadsView.aspx?Langid=1&PFid=24&Lev… > > > I am working on building and trying these drivers, but have a couple of > questions in the meantime: > > 1. Does anyone know what the status of HDMI audio support is (in particular > Intel)? > > 2. Any idea why Realtek are maintaining their own version of drivers? Is > this a legal/licensing issue? Jaroslav and Takashi-san, Any idea why Realtek is still maintaining their forked ALSA? I thought you guys raised the issue with them a while back and they agreed to submit their changes through the standard process. Lee

2 1

[alsa-devel] [PATCH] hda_intel: Fix multiple device support by incrementing device count
by Andrew Paprocki 13 Jan '08

13 Jan '08

Fixes multiple device support by incrementing the static device counter at the end of the azx_probe() call. Without this, subsequent probes would always use the index specified for the first card. Signed-off-by: Andrew Paprocki <andrew(a)ishiboo.com> diff -r acdc98363786 -r bde9dbdbd2ba pci/hda/hda_intel.c --- a/pci/hda/hda_intel.c Fri Jan 11 18:15:26 2008 +0100 +++ b/pci/hda/hda_intel.c Sat Jan 12 18:59:58 2008 -0500 @@ -1933,6 +1933,7 @@ static int __devinit azx_probe(struct pc chip->running = 1; power_down_all_codecs(chip); + dev++; return err; }

2 1

[alsa-devel] [PATCH] hda: Fix 5.1 sound in Dell 6stack ALC888 HDA
by Claudio Matsuoka 13 Jan '08

13 Jan '08

This patch fixes 5.1 surround output and headphone detection in the Dell Inspiron 530 and possibly other Dell systems using the ALC888 codec (mode 6stack-dell). This configuration is very similar to 6stack-hp, which also has rear/clfe channels swapped if compared to the standard ALC883 6stack configuration. (This is my first mercurial-generated patch, I added the signed-off line by hand and the user name in the header is an internal address, please correct that if needed.)

2 1

[alsa-devel] Realtek ALC883 support on Albatron KI690-AM2 motherboard
by Andrew Paprocki 13 Jan '08

13 Jan '08

Hello everyone, I'm currently trying to get the ALC883 on the Albatron KI690-AM2 motherboard working under ALSA and I'm running into problems with some of the ports. From looking at the patch_realtek.c file inside the intel_hda driver, it appears there are many variants of hookups to this chipset which all require some manual configuration in the driver. I've tried what appears to be the closest match (ALC883_6ST_DIG or 6stack-dig), but it still does not enable all of the ports. The motherboard webpage is here: http://www.albatron.com.tw/English/product/mb/pro_detail.asp?rlink=Specific… The board has two audio stacks on the back panel. Facing the jack inputs, the left-hand stack contains: <S/PDIF output> <TOSLINK w/ green cover> Next to that stack is a larger 6-stack with the following configuration: <blue> <black> <green> <pink> <grey> <TOSLINK w/ pink cover> The box is running a stock debian 2.6.18-4-486 kernel. The kernel outputs this at startup if the "6stack-dig" model is not explicitly supplied: hda_codec: Unknown model for ALC883, trying auto-probe from BIOS... hda-intel: Invalid position buffer, using LPIB read method instead. When using the "6stack-dig" mode, ALSA sees two devices: card 0: SB [HDA ATI SB], device 0: ALC883 Analog [ALC883 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: SB [HDA ATI SB], device 2: ALC883 Digital [ALC883 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 When attempting to use ALSA to open the first device in raw mode to output to each stereo pair separately, 5 pairs are detected, and the outputs appear as follows... Pair 1: <green> (working) Pair 2: <pink> (working) Pair 3: <black> (working) Pair 4: (no audio) Pair 5: (no audio) No sound can be heard from the blue or grey jacks in any configuration. I'm wondering if a special entry is needed in patch_realtek.c to handle this board's usage of the ALC883. I am willing to try changing the code and testing if someone can point me to what might need to be done. 01:05.2 Audio device: ATI Technologies Inc Unknown device 7919 Subsystem: Albatron Corp. Unknown device 5000 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 64, Cache Line Size: 4 bytes Interrupt: pin B routed to IRQ 3 Region 0: Memory at fdffc000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Address: 0000000000000000 Data: 0000 Thanks, -Andrew

2 6

[alsa-devel] Hotplug plugin
by Erik Slagter 13 Jan '08

13 Jan '08

Sorry for crossposting, but this is a problem I am not quite sure might be solved by users or developers. Also I am not too lazy to do a little development myself, if that appears to be necessary. This is my setup: - internal snd-hda soundcard connected to speakers in the bathroom playing 24x7 music; - jvc amplifier in living room with integrated USB sound device. The USB sound device in the amplifier vanishes when the amplifier is powered down. So... this is my problem. I want to have 24x7 music on the internal soundcard and on the amplifier. When the amplifier is switched off, no audio should be send there (of course) but playback should resume as soon as the amplifier is switched on and the USB device re-appears. I created a "tee" construction using asound.conf which adresses both soundcards, this works. At least... until the amplifer is switched off and the USB device disappears. Then the sound playing application gets an error and quits. I tried these approaches for a solution: - pulseaudio, using hal-detection and "combined" device: doesn't work because a removed device is never reinserted into the combined device; - alsa extplug: easy to create, no worries about the "slave" device, you get it for free, but I abandoned the idea as there is probably no way to recreate the slave pcm while the "owner" remains active. - alsa ioplug, attempt 1: there is probably no way to to address and recreate the "slave" (which I open myself in the plugin using snd_pcm_open) in such a way that it is completely transparent to the application that it is talking to the "slave" (not a "real" alsa slave) instead of the ioplug plugin - alsa ioplug, attempt 2: I tried to make the callback functions of the ioplug call the callbacks of the "slave" directly, but that probably isn't going to work either (too much callbacks in the pcm struct that have no equivalent in the ioplug struct) - alsa ioplug, attempt 3: open and write to the hardware devices directly. Abonded it as well because this cannot practically be done, as you'd need parts copies of header files from the kernel and alsalib for hidden data structures. Anyone ideas to solve this in a neat way? (preferrably using an alsa plugin; for the moment I am using a self-made external program that alsa dumps it sound to (file plugin) and uses two alsa-player processes to get the audio to the devices)

2 4

[alsa-devel] Disabling buffer fill level preprocessing by ALSA
by Lennart Poettering 12 Jan '08

12 Jan '08

Hi! In PulseAudio I want to schedule on my own when I need to write audio data into the device and when not. To achieve that I want to be notified via poll() whenever a period boundary is passed (i.e. when an IRQ happens), but only then. That's different from the usual mode where you are notified via poll() whether there is space in the playback buffer that needs to be filled up. On OSS the mmap() mode enables a mode like I described above. After enabling mmap() the application can decide by itself what it considers full and what empty in the dma buffer, and use GETOPTR to query the playback position. poll() on the OSS fd will directly reflect the sound card IRQs and is not influenced if you ever wrote data to device or not. I assume that I can enable a mode like that with one of the SW params. But quite frankly the docs for it are not enlighening at all. 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] ALSA on top of OpenMAX
by Brett Keating 11 Jan '08

11 Jan '08

Hi! We are interested in providing ALSA on our embedded linux platform. Our interface to the sound devices is OpenMAX, through a multimedia coprocessor. The host processor runs Linux and ALSA would hopefully be able to "talk" to the coprocessor through OpenMAX. I see many examples of using ALSA as a sink for OpenMAX, but not any examples of OpenMAX acting as a device that ALSA can drive. Is there any information available along these lines in terms of mapping ALSA to OpenMAX? Thanks for any help, Brett

1 0

[alsa-devel] dmix without mmap support in soundcard driver
by cailinhua 11 Jan '08

11 Jan '08

Hi,all: I want to enable dmix for our sound card.Unfortunately for some reason the sound card driver don't support 'mmap' way,it only support 'copy' callback to copy data from upper level to DMA.So ,I think maybe I can modify the alsa-lib dmix source code to let it use 'copy' way to send data to the driver. Is it possible？ Could you please give me some ideas? Thanks a lot in advance. Cliff _________________________________________________________________ Windows Live Photo gallery 数码相机的超级伴侣，轻松管理和编辑照片，还能制作全景美图！ http://get.live.cn/product/photo.html

4 5

[alsa-devel] ATI Technologies Inc SBx00 Azalia
by Markus Rechberger 11 Jan '08

11 Jan '08

Hi, there seems to be an issue with this ATI SBx00 Azalia driver. nmixer (which uses the oss emulated interface shows up 4 controls) Vol (volume which seems to control the whole output volume) PCM2 (maybe not routed on that notebook, it practically has no effect) IGain (guess for the microphone which I haven't tested yet) Digital1 (don't know) if I try to start alsamixer, it doesn't seem to work and I get an entry in dmesg whenever I try to run it: $ alsamixer alsamixer: function snd_mixer_load failed: Invalid argument $ dmesg ... [ 1269.997724] hda_codec: num_steps = 0 for NID=0x12 $ lspci 00:14.2 Audio device: ATI Technologies Inc SBx00 Azalia $ lspci -vvvn 00:14.2 0403: 1002:4383 Subsystem: 1002:4383 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=slow >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 64, Cache Line Size: 32 bytes Interrupt: pin ? routed to IRQ 16 Region 0: Memory at f0300000 (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- ---- $ arecord -l **** List of CAPTURE Hardware Devices **** card 0: SB [HDA ATI SB], device 0: ALC262 Analog [ALC262 Analog] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 card 0: SB [HDA ATI SB], device 2: ALC262 Analog [ALC262 Analog] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 ---- $ arecord -L default:CARD=SB HDA ATI SB, ALC262 Analog Default Audio Device front:CARD=SB,DEV=0 HDA ATI SB, ALC262 Analog Front speakers surround40:CARD=SB,DEV=0 HDA ATI SB, ALC262 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=SB,DEV=0 HDA ATI SB, ALC262 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=SB,DEV=0 HDA ATI SB, ALC262 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=SB,DEV=0 HDA ATI SB, ALC262 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=SB,DEV=0 HDA ATI SB, ALC262 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=SB,DEV=0 HDA ATI SB, ALC262 Digital IEC958 (S/PDIF) Digital Audio Output null Discard all samples (playback) or generate zero samples (capture) ---- $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: SB [HDA ATI SB], device 0: ALC262 Analog [ALC262 Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 card 0: SB [HDA ATI SB], device 1: ALC262 Digital [ALC262 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 ---- $ aplay -L **** List of PLAYBACK Hardware Devices **** card 0: SB [HDA ATI SB], device 0: ALC262 Analog [ALC262 Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 card 0: SB [HDA ATI SB], device 1: ALC262 Digital [ALC262 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 root@osrc-laptop:/devel/alsa/alsa-driver# aplay -L default:CARD=SB HDA ATI SB, ALC262 Analog Default Audio Device front:CARD=SB,DEV=0 HDA ATI SB, ALC262 Analog Front speakers surround40:CARD=SB,DEV=0 HDA ATI SB, ALC262 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=SB,DEV=0 HDA ATI SB, ALC262 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=SB,DEV=0 HDA ATI SB, ALC262 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=SB,DEV=0 HDA ATI SB, ALC262 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=SB,DEV=0 HDA ATI SB, ALC262 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers null Discard all samples (playback) or generate zero samples (capture) ---- root@laptop:/proc/asound# cat cards 0 [SB ]: HDA-Intel - HDA ATI SB HDA ATI SB at 0xf0300000 irq 16 root@laptop:/proc/asound# cat devices 0: [ 0] : control 1: : sequencer 7: [ 0- 3]: hardware dependent 16: [ 0- 0]: digital audio playback 17: [ 0- 1]: digital audio playback 24: [ 0- 0]: digital audio capture 26: [ 0- 2]: digital audio capture 33: : timer root@laptop:/proc/asound# cat hwdep 00-03: HDA Codec 3 root@laptop:/proc/asound# cat meminfo kmalloc: 53165 bytes root@osrc-laptop:/proc/asound# cat modules 0 snd_hda_intel root@laptop:/proc/asound# cat pcm 00-02: ALC262 Analog : ALC262 Analog : capture 2 00-01: ALC262 Digital : ALC262 Digital : playback 1 00-00: ALC262 Analog : ALC262 Analog : playback 1 : capture 2 root@laptop:/proc/asound# cat version Advanced Linux Sound Architecture Driver Version 1.0.15. Compiled on Jan 11 2008 for kernel 2.6.23-rc6 (SMP). maybe someone has more insight in that intel-hda driver. thanks & best regards, Markus

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Re: [alsa-devel] [alsa-cvslog] alsa-lib: SND_PCM_TSTAMP_MMAP -> SND_PCM_TSTAMP_ENABLE change
by Clemens Ladisch 11 Jan '08

11 Jan '08

Jaroslav Kysela wrote: > description: > SND_PCM_TSTAMP_MMAP -> SND_PCM_TSTAMP_ENABLE change > > diff -r 3c87e22e70d8 -r 2519fcd54901 include/pcm.h > --- a/include/pcm.h Thu Jan 10 10:01:14 2008 +0100 > +++ b/include/pcm.h Fri Jan 11 08:54:07 2008 +0100 > @@ -281,9 +281,9 @@ > typedef enum _snd_pcm_tstamp { > /** No timestamp */ > SND_PCM_TSTAMP_NONE = 0, > - /** Update mmap'ed timestamp */ > - SND_PCM_TSTAMP_MMAP, > - SND_PCM_TSTAMP_LAST = SND_PCM_TSTAMP_MMAP > + /** Update timestamp at every hardware position update */ > + SND_PCM_TSTAMP_ENABLE, > + SND_PCM_TSTAMP_LAST = SND_PCM_TSTAMP_ENABLE > } snd_pcm_tstamp_t; This change breaks backward compatibility. The v19-devel branch of PortAudio uses SND_PCM_TSTAMP_MMAP. Regards, Clemens

3 2

Re: [alsa-devel] ALSA : Can't open capture device
by Clemens Ladisch 11 Jan '08

11 Jan '08

Alexandre BOUIN wrote: > I'm working on an intern project, using an AT91SAM9260EK-based board, > embedded system. > Our kernel contains a full ALSA 1.14 with ASoC layer. > Our ASoC driver works well in capture and playback, if we are using OSS > layer or directly /dev/dsp. > So I need some tips on ALSA layer side : > > here you can see that my device is created : > ll /dev/snd/ > crwxrwxrwx 1 root root 116, 0 Dec 26 2007 controlC0 > crwxrwxrwx 1 root root 116, 4 Dec 26 2007 hwC0D0 > crwxrwxrwx 1 root root 116, 14 Dec 26 2007 pcmC0D0c > crwxrwxrwx 1 root root 116, 16 Dec 26 2007 pcmC0D0p > ll /dev/dsp > crwxrwxrwx 1 root root 14, 3 Dec 26 2007 /dev/dsp > > A cat on playback device works (I added some traces in kernel) : > cat < /dev/snd/pcmC0D0p > snd_pcm_open : enter > snd_pcm_open_file : before snd_pcm_open_substream > snd_pcm_open_file : after snd_pcm_open_substream > cat: read error: Invalid argument > > But a cat on capture device does not work : > cat < /dev/snd/pcmC0D0c > -sh: cannot open /dev/snd/pcmC0D0c: No such device Does the capture device appear in /proc/asound/devices or in the output of "arecord -l"? Regards, Clemens

2 1

[alsa-devel] intel-hd on an intel G33 motherboard
by De Fuuss 11 Jan '08

11 Jan '08

Hi, Maybe I'm missing something, but I've failed to get even the slightest noise out of the onboard sound card from a Intel G33 motherboard. I tried the alsa drivers from the alsa-driver-1.0.15 package as well as the the alsa drivers provided by kernel 2.6.23 (built as modules), to no avail. The sound chip is identified as a Intel HDA / Realtek ALC888 on the Intel G33 ICH9 motherboard. After reading some of the documentation from the ALSA site and googling through half the internet I haven't found out yet if this specific sound card is supported by ALSA and if some module options must be passed (such as options snd-intel-hda model=.....). Could somebody be so kind as to point me into the right direction? Thank you. Best regards, Patrick _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

5 20

[alsa-devel] ALSA : Can't open capture device
by Alexandre BOUIN 11 Jan '08

11 Jan '08

Hi list, I'm working on an intern project, using an AT91SAM9260EK-based board, embedded system. Our kernel contains a full ALSA 1.14 with ASoC layer. Our ASoC driver works well in capture and playback, if we are using OSS layer or directly /dev/dsp. So I need some tips on ALSA layer side : here you can see that my device is created : ll /dev/snd/ crwxrwxrwx 1 root root 116, 0 Dec 26 2007 controlC0 crwxrwxrwx 1 root root 116, 4 Dec 26 2007 hwC0D0 crwxrwxrwx 1 root root 116, 14 Dec 26 2007 pcmC0D0c crwxrwxrwx 1 root root 116, 16 Dec 26 2007 pcmC0D0p ll /dev/dsp crwxrwxrwx 1 root root 14, 3 Dec 26 2007 /dev/dsp A cat on playback device works (I added some traces in kernel) : cat < /dev/snd/pcmC0D0p snd_pcm_open : enter snd_pcm_open_file : before snd_pcm_open_substream snd_pcm_open_file : after snd_pcm_open_substream cat: read error: Invalid argument But a cat on capture device does not work : cat < /dev/snd/pcmC0D0c -sh: cannot open /dev/snd/pcmC0D0c: No such device Any suggestion will be greatfull !! Regards, Alex.

1 0

[alsa-devel] [PATCH] hda: Add new STAC9205 PCI_QUIRK
by Matthew Ranostay 11 Jan '08

11 Jan '08

Added a new STAC 9205 quirk for Vostro 1500. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r 1e968b2c8617 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Thu Jan 10 14:50:34 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Thu Jan 10 15:55:23 2008 -0500 @@ -1468,6 +1468,7 @@ static unsigned int ref9205_pin_configs[ 102801FD 10280204 1028021F + 10280228 (Dell Vostro 1500) */ static unsigned int dell_9205_m42_pin_configs[12] = { 0x0321101F, 0x03A11020, 0x400003FA, 0x90170310, @@ -1551,6 +1552,8 @@ static struct snd_pci_quirk stac9205_cfg "unknown Dell", STAC_9205_DELL_M42), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x021f, "Dell Inspiron", STAC_9205_DELL_M44), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0228, + "Dell Vostro 1500", STAC_9205_DELL_M42), {} /* terminator */ };

2 1

[alsa-devel] [PATCH] ASoC drivers for the Freescale MPC8610 SoC
by Timur Tabi 10 Jan '08

10 Jan '08

This patch adds ALSA SoC device drivers for the Freescale MPC8610 SoC and the MPC8610-HPCD reference board. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- This patch applies on top of git://git.kernel.org/pub/scm/linux/kernel/git/perex/alsa.git. Documentation/powerpc/booting-without-of.txt | 40 ++ arch/powerpc/boot/dts/mpc8610_hpcd.dts | 107 ++++ arch/powerpc/configs/mpc8610_hpcd_defconfig | 171 ++++++- arch/powerpc/platforms/86xx/mpc8610_hpcd.c | 18 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 2 +- sound/soc/fsl/Kconfig | 21 + sound/soc/fsl/Makefile | 6 + sound/soc/fsl/fsl_dma.c | 819 ++++++++++++++++++++++++++ sound/soc/fsl/fsl_dma.h | 149 +++++ sound/soc/fsl/fsl_ssi.c | 614 +++++++++++++++++++ sound/soc/fsl/fsl_ssi.h | 224 +++++++ sound/soc/fsl/mpc8610_hpcd.c | 621 +++++++++++++++++++ 13 files changed, 2789 insertions(+), 4 deletions(-) create mode 100644 sound/soc/fsl/Kconfig create mode 100644 sound/soc/fsl/Makefile create mode 100644 sound/soc/fsl/fsl_dma.c create mode 100644 sound/soc/fsl/fsl_dma.h create mode 100644 sound/soc/fsl/fsl_ssi.c create mode 100644 sound/soc/fsl/fsl_ssi.h create mode 100644 sound/soc/fsl/mpc8610_hpcd.c diff --git a/Documentation/powerpc/booting-without-of.txt b/Documentation/powerpc/booting-without-of.txt index e9a3cb1..826af91 100644 --- a/Documentation/powerpc/booting-without-of.txt +++ b/Documentation/powerpc/booting-without-of.txt @@ -53,6 +53,7 @@ Table of Contents j) CFI or JEDEC memory-mapped NOR flash k) Global Utilities Block l) Xilinx IP cores + p) Freescale Synchronous Serial Interface VII - Specifying interrupt information for devices 1) interrupts property @@ -2514,6 +2515,45 @@ platforms are moved over to use the flattened-device-tree model. Requred properties: - current-speed : Baud rate of uartlite + p) Freescale Synchronous Serial Interface + + The SSI is a serial device that communicates with audio codecs. It can + be programmed in AC97, I2S, left-justified, or right-justified modes. + + Required properties: + - compatible : compatible list, containing "fsl,ssi" + - cell-index : the SSI, <0> = SSI1, <1> = SSI2, and so on + - reg : offset and length of the register set for the device + - interrupts : <a b> where a is the interrupt number and b is a + field that represents an encoding of the sense and + level information for the interrupt. This should be + encoded based on the information in section 2) + depending on the type of interrupt controller you + have. + - interrupt-parent : the phandle for the interrupt controller that + services interrupts for this device. + - fsl,mode : the operating mode for the SSI interface + "i2s-slave" - I2S mode, SSI is clock slave + "i2s-master" - I2S mode, SSI is clock master + "lj-slave" - left-justified mode, SSI is clock slave + "lj-master" - l.j. mode, SSI is clock master + "rj-slave" - right-justified mode, SSI is clock slave + "rj-master" - r.j., SSI is clock master + "ac97-slave" - AC97 mode, SSI is clock slave + "ac97-master" - AC97 mode, SSI is clock master + + Optional properties: + - codec : child node that defines an audio codec connected + to this SSI + + Child 'codec' node required properties: + - compatible : compatible list, contains the name of the codec + + Child 'codec' node optional properties: + - bus-frequency : The frequency of the input clock, which typically + comes from an on-board dedicated oscillator. + + More devices will be defined as this spec matures. VII - Specifying interrupt information for devices diff --git a/arch/powerpc/boot/dts/mpc8610_hpcd.dts b/arch/powerpc/boot/dts/mpc8610_hpcd.dts index 966edf1..4f12ede 100644 --- a/arch/powerpc/boot/dts/mpc8610_hpcd.dts +++ b/arch/powerpc/boot/dts/mpc8610_hpcd.dts @@ -103,6 +103,113 @@ reg = <e0000 1000>; fsl,has-rstcr; }; + + ssi@16000 { + compatible = "fsl,ssi"; + cell-index = <0>; + reg = <16000 100>; + interrupt-parent = <&mpic>; + interrupts = <3e 2>; + fsl,mode = "i2s-slave"; + codec { + compatible = "cirrus,cs4270"; + /* MCLK source is a stand-alone oscillator */ + bus-frequency = <bb8000>; + }; + }; + + ssi@16100 { + compatible = "fsl,ssi"; + cell-index = <1>; + reg = <16100 100>; + interrupt-parent = <&mpic>; + interrupts = <3f 2>; + }; + + dma@21300 { + #address-cells = <1>; + #size-cells = <1>; + compatible = "fsl,mpc8610-dma", "fsl,mpc8540-dma"; + cell-index = <0>; + reg = <21300 4>; /* DMA general status register */ + ranges = <0 21100 200>; + + dma-channel@0 { + compatible = "fsl,mpc8610-dma-channel", + "fsl,mpc8540-dma-channel"; + cell-index = <0>; + reg = <0 80>; + interrupt-parent = <&mpic>; + interrupts = <14 2>; + }; + dma-channel@1 { + compatible = "fsl,mpc8610-dma-channel", + "fsl,mpc8540-dma-channel"; + cell-index = <1>; + reg = <80 80>; + interrupt-parent = <&mpic>; + interrupts = <15 2>; + }; + dma-channel@2 { + compatible = "fsl,mpc8610-dma-channel", + "fsl,mpc8540-dma-channel"; + cell-index = <2>; + reg = <100 80>; + interrupt-parent = <&mpic>; + interrupts = <16 2>; + }; + dma-channel@3 { + compatible = "fsl,mpc8610-dma-channel", + "fsl,mpc8540-dma-channel"; + cell-index = <3>; + reg = <180 80>; + interrupt-parent = <&mpic>; + interrupts = <17 2>; + }; + }; + + dma@c300 { + #address-cells = <1>; + #size-cells = <1>; + compatible = "fsl,mpc8610-dma", "fsl,mpc8540-dma"; + cell-index = <1>; + reg = <c300 4>; /* DMA general status register */ + ranges = <0 c100 200>; + + dma-channel@0 { + compatible = "fsl,mpc8610-dma-channel", + "fsl,mpc8540-dma-channel"; + cell-index = <0>; + reg = <0 80>; + interrupt-parent = <&mpic>; + interrupts = <3c 2>; + }; + dma-channel@1 { + compatible = "fsl,mpc8610-dma-channel", + "fsl,mpc8540-dma-channel"; + cell-index = <1>; + reg = <80 80>; + interrupt-parent = <&mpic>; + interrupts = <3d 2>; + }; + dma-channel@2 { + compatible = "fsl,mpc8610-dma-channel", + "fsl,mpc8540-dma-channel"; + cell-index = <2>; + reg = <100 80>; + interrupt-parent = <&mpic>; + interrupts = <3e 2>; + }; + dma-channel@3 { + compatible = "fsl,mpc8610-dma-channel", + "fsl,mpc8540-dma-channel"; + cell-index = <3>; + reg = <180 80>; + interrupt-parent = <&mpic>; + interrupts = <3f 2>; + }; + }; + }; pci@e0008000 { diff --git a/arch/powerpc/configs/mpc8610_hpcd_defconfig b/arch/powerpc/configs/mpc8610_hpcd_defconfig index 0483211..fb68886 100644 --- a/arch/powerpc/configs/mpc8610_hpcd_defconfig +++ b/arch/powerpc/configs/mpc8610_hpcd_defconfig @@ -684,7 +684,7 @@ CONFIG_SERIAL_8250_RSA=y CONFIG_SERIAL_CORE=y CONFIG_SERIAL_CORE_CONSOLE=y # CONFIG_SERIAL_JSM is not set -CONFIG_SERIAL_OF_PLATFORM=y +# CONFIG_SERIAL_OF_PLATFORM is not set CONFIG_UNIX98_PTYS=y # CONFIG_LEGACY_PTYS is not set # CONFIG_IPMI_HANDLER is not set @@ -699,7 +699,60 @@ CONFIG_UNIX98_PTYS=y # CONFIG_RAW_DRIVER is not set # CONFIG_TCG_TPM is not set CONFIG_DEVPORT=y -# CONFIG_I2C is not set +CONFIG_I2C=y +CONFIG_I2C_BOARDINFO=y +# CONFIG_I2C_CHARDEV is not set + +# +# I2C Algorithms +# +# CONFIG_I2C_ALGOBIT is not set +# CONFIG_I2C_ALGOPCF is not set +# CONFIG_I2C_ALGOPCA is not set + +# +# I2C Hardware Bus support +# +# 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 is not set +# CONFIG_I2C_I810 is not set +# CONFIG_I2C_PIIX4 is not set +CONFIG_I2C_MPC=y +# CONFIG_I2C_NFORCE2 is not set +# CONFIG_I2C_OCORES is not set +# CONFIG_I2C_PARPORT_LIGHT is not set +# CONFIG_I2C_PROSAVAGE is not set +# CONFIG_I2C_SAVAGE4 is not set +# CONFIG_I2C_SIMTEC is not set +# CONFIG_I2C_SIS5595 is not set +# CONFIG_I2C_SIS630 is not set +# CONFIG_I2C_SIS96X is not set +# CONFIG_I2C_TAOS_EVM is not set +# CONFIG_I2C_VIA is not set +# CONFIG_I2C_VIAPRO is not set +# CONFIG_I2C_VOODOO3 is not set + +# +# Miscellaneous I2C Chip support +# +# CONFIG_SENSORS_DS1337 is not set +# CONFIG_SENSORS_DS1374 is not set +# CONFIG_DS1682 is not set +# CONFIG_SENSORS_EEPROM is not set +# CONFIG_SENSORS_PCF8574 is not set +# CONFIG_SENSORS_PCA9539 is not set +# CONFIG_SENSORS_PCF8591 is not set +# CONFIG_SENSORS_M41T00 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 # # SPI support @@ -746,7 +799,119 @@ CONFIG_DUMMY_CONSOLE=y # # Sound # -# CONFIG_SOUND is not set +CONFIG_SOUND=y + +# +# Advanced Linux Sound Architecture +# +CONFIG_SND=y +CONFIG_SND_TIMER=y +CONFIG_SND_PCM=y +# CONFIG_SND_SEQUENCER is not set +CONFIG_SND_OSSEMUL=y +CONFIG_SND_MIXER_OSS=y +CONFIG_SND_PCM_OSS=y +# CONFIG_SND_PCM_OSS_PLUGINS is not set +# CONFIG_SND_DYNAMIC_MINORS is not set +# CONFIG_SND_SUPPORT_OLD_API is not set +CONFIG_SND_VERBOSE_PROCFS=y +# CONFIG_SND_VERBOSE_PRINTK is not set +# CONFIG_SND_DEBUG is not set + +# +# Generic devices +# +# CONFIG_SND_DUMMY is not set +# CONFIG_SND_MTPAV is not set +# CONFIG_SND_SERIAL_U16550 is not set +# CONFIG_SND_MPU401 is not set + +# +# PCI devices +# +# CONFIG_SND_AD1889 is not set +# CONFIG_SND_ALS300 is not set +# CONFIG_SND_ALS4000 is not set +# CONFIG_SND_ALI5451 is not set +# CONFIG_SND_ATIIXP is not set +# CONFIG_SND_ATIIXP_MODEM is not set +# CONFIG_SND_AU8810 is not set +# CONFIG_SND_AU8820 is not set +# CONFIG_SND_AU8830 is not set +# CONFIG_SND_AZT3328 is not set +# CONFIG_SND_BT87X is not set +# CONFIG_SND_CA0106 is not set +# CONFIG_SND_CMIPCI is not set +# CONFIG_SND_CS4281 is not set +# CONFIG_SND_CS46XX is not set +# CONFIG_SND_CS5530 is not set +# CONFIG_SND_DARLA20 is not set +# CONFIG_SND_GINA20 is not set +# CONFIG_SND_LAYLA20 is not set +# CONFIG_SND_DARLA24 is not set +# CONFIG_SND_GINA24 is not set +# CONFIG_SND_LAYLA24 is not set +# CONFIG_SND_MONA is not set +# CONFIG_SND_MIA is not set +# CONFIG_SND_ECHO3G is not set +# CONFIG_SND_INDIGO is not set +# CONFIG_SND_INDIGOIO is not set +# CONFIG_SND_INDIGODJ is not set +# CONFIG_SND_EMU10K1 is not set +# CONFIG_SND_EMU10K1X is not set +# CONFIG_SND_ENS1370 is not set +# CONFIG_SND_ENS1371 is not set +# CONFIG_SND_ES1938 is not set +# CONFIG_SND_ES1968 is not set +# CONFIG_SND_FM801 is not set +# CONFIG_SND_HDA_INTEL is not set +# CONFIG_SND_HDSP is not set +# CONFIG_SND_HDSPM is not set +# CONFIG_SND_ICE1712 is not set +# CONFIG_SND_ICE1724 is not set +# CONFIG_SND_INTEL8X0 is not set +# CONFIG_SND_INTEL8X0M is not set +# CONFIG_SND_KORG1212 is not set +# CONFIG_SND_MAESTRO3 is not set +# CONFIG_SND_MIXART is not set +# CONFIG_SND_NM256 is not set +# CONFIG_SND_PCXHR is not set +# CONFIG_SND_RIPTIDE is not set +# CONFIG_SND_RME32 is not set +# CONFIG_SND_RME96 is not set +# CONFIG_SND_RME9652 is not set +# CONFIG_SND_SONICVIBES is not set +# CONFIG_SND_TRIDENT is not set +# CONFIG_SND_VIA82XX is not set +# CONFIG_SND_VIA82XX_MODEM is not set +# CONFIG_SND_VX222 is not set +# CONFIG_SND_YMFPCI is not set + +# +# ALSA PowerMac devices +# + +# +# ALSA PowerPC devices +# + +# +# System on Chip audio support +# +CONFIG_SND_SOC=y + +# +# SoC Audio support for SuperH +# + +# +# ALSA SoC audio for Freescale SOCs +# +CONFIG_SND_SOC_MPC8610=y +CONFIG_SND_SOC_MPC8610_HPCD=y +CONFIG_SND_SOC_CS4270=y +CONFIG_SND_SOC_CS4270_VD33_ERRATA=y + CONFIG_HID_SUPPORT=y CONFIG_HID=y # CONFIG_HID_DEBUG is not set diff --git a/arch/powerpc/platforms/86xx/mpc8610_hpcd.c b/arch/powerpc/platforms/86xx/mpc8610_hpcd.c index 6390895..6e1bde3 100644 --- a/arch/powerpc/platforms/86xx/mpc8610_hpcd.c +++ b/arch/powerpc/platforms/86xx/mpc8610_hpcd.c @@ -34,9 +34,27 @@ #include <asm/mpic.h> +#include <asm/of_platform.h> #include <sysdev/fsl_pci.h> #include <sysdev/fsl_soc.h> +static struct of_device_id mpc8610_ids[] = { + { .type = "soc", }, + {} +}; + +static int __init mpc8610_declare_of_platform_devices(void) +{ + if (!machine_is(mpc86xx_hpcd)) + return 0; + + /* Without this call, the SSI device driver won't get probed. */ + of_platform_bus_probe(NULL, mpc8610_ids, NULL); + + return 0; +} +device_initcall(mpc8610_declare_of_platform_devices); + void __init mpc86xx_hpcd_init_irq(void) { diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 97b2552..43bbc60 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -28,6 +28,7 @@ source "sound/soc/at91/Kconfig" source "sound/soc/pxa/Kconfig" source "sound/soc/s3c24xx/Kconfig" source "sound/soc/sh/Kconfig" +source "sound/soc/fsl/Kconfig" # Supported codecs source "sound/soc/codecs/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 3041403..4869c9a 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,4 +1,4 @@ snd-soc-core-objs := soc-core.o soc-dapm.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o -obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ s3c24xx/ sh/ +obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ s3c24xx/ sh/ fsl/ diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig new file mode 100644 index 0000000..4a5bbd2 --- /dev/null +++ b/sound/soc/fsl/Kconfig @@ -0,0 +1,21 @@ +menu "ALSA SoC audio for Freescale SOCs" + +config SND_SOC_MPC8610 + bool "ALSA SoC support for the MPC8610 SOC" + depends on SND_SOC #&& MPC8610_HPCD + default y #if MPC8610 + help + Say Y if you want to add support for codecs attached to the SSI + device on an MPC8610. + +config SND_SOC_MPC8610_HPCD + # ALSA SoC support for Freescale MPC8610HPCD + bool "ALSA SoC support for the Freescale MPC8610 HPCD board" + depends on SND_SOC_MPC8610 + select SND_SOC_CS4270 + select SND_SOC_CS4270_VD33_ERRATA + default y #if MPC8610_HPCD + help + Say Y if you want to enable audio on the Freescale MPC8610 HPCD. + +endmenu diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile new file mode 100644 index 0000000..62f680a --- /dev/null +++ b/sound/soc/fsl/Makefile @@ -0,0 +1,6 @@ +# MPC8610 HPCD Machine Support +obj-$(CONFIG_SND_SOC_MPC8610_HPCD) += mpc8610_hpcd.o + +# MPC8610 Platform Support +obj-$(CONFIG_SND_SOC_MPC8610) += fsl_ssi.o fsl_dma.o + diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c new file mode 100644 index 0000000..6b86be0 --- /dev/null +++ b/sound/soc/fsl/fsl_dma.c @@ -0,0 +1,819 @@ +/* + * Freescale DMA ALSA SoC PCM driver + * + * Author: Timur Tabi <timur(a)freescale.com> + * + * Copyright 2007 Freescale Semiconductor, Inc. This file is licensed under + * the terms of the GNU General Public License version 2. This program + * is licensed "as is" without any warranty of any kind, whether express + * or implied. + * + * This driver implements ASoC support for the Elo DMA controller, which is + * the DMA controller on Freescale 83xx, 85xx, and 86xx SOCs. In ALSA terms, + * the PCM driver is what handles the DMA buffer. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/dma-mapping.h> +#include <linux/interrupt.h> +#include <linux/delay.h> + +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <asm/io.h> + +#include "fsl_dma.h" + +/* + * The formats that the DMA controller supports, which is anything + * that is 8, 16, or 32 bits. + */ +#define FSLDMA_PCM_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_U8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_U16_BE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S24_BE | \ + SNDRV_PCM_FMTBIT_U24_LE | \ + SNDRV_PCM_FMTBIT_U24_BE | \ + SNDRV_PCM_FMTBIT_S32_LE | \ + SNDRV_PCM_FMTBIT_S32_BE | \ + SNDRV_PCM_FMTBIT_U32_LE | \ + SNDRV_PCM_FMTBIT_U32_BE) + +#define FSLDMA_PCM_RATES (SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_192000 | \ + SNDRV_PCM_RATE_CONTINUOUS) + +/* DMA global data. This structure is used by fsl_dma_open() to determine + * which DMA channels to assign to a substream. Unfortunately, ASoC V1 does + * not allow the machine driver to provide this information to the PCM + * driver in advance, and there's no way to differentiate between the two + * DMA controllers. So for now, this driver only supports one SSI device + * using two DMA channels. We cannot support multiple DMA devices. + * + * ssi_stx_phys: bus address of SSI STX register + * ssi_srx_phys: bus address of SSI SRX register + * dma_channel: pointer to the DMA channel's registers + * irq: IRQ for this DMA channel + * assigned: set to 1 if that DMA channel is assigned to a substream + */ +static struct { + dma_addr_t ssi_stx_phys; + dma_addr_t ssi_srx_phys; + struct ccsr_dma_channel __iomem *dma_channel[2]; + unsigned int irq[2]; + unsigned int assigned[2]; +} dma_global_data; + +/* + * The number of DMA links to use. Two is the bare minimum, but if you + * have really small links you might need more. + */ +#define NUM_DMA_LINKS 2 + +/* Per-substream DMA data + * + * Each substream has a 1-to-1 association with a DMA channel. + * + * The link[] array is first because it needs to be aligned on a 32-byte + * boundary, so putting it first will ensure alignment without padding the + * structure. + * + * link[]: array of link descriptors + * controller_id: which DMA controller (0, 1, ...) + * channel_id: which DMA channel on the controller (0, 1, 2, ...) + * dma_channel: pointer to the DMA channel's registers + * irq: IRQ for this DMA channel + * substream: pointer to the substream object, needed by the ISR + * ssi_sxx_phys: bus address of the STX or SRX register to use + * ld_buf_phys: physical address of the LD buffer + * current_link: index into link[] of the link currently being processed + * dma_buf_phys: physical address of the DMA buffer + * dma_buf_next: physical address of the next period to process + * dma_buf_end: physical address of the byte after the end of the DMA buffer + * period_size: the size of a single period + */ +struct fsl_dma_private { + struct fsl_dma_link_descriptor link[NUM_DMA_LINKS]; + unsigned int controller_id; + unsigned int channel_id; + struct ccsr_dma_channel __iomem *dma_channel; + unsigned int irq; + struct snd_pcm_substream *substream; + dma_addr_t ssi_sxx_phys; + dma_addr_t ld_buf_phys; + unsigned int current_link; + dma_addr_t dma_buf_phys; + dma_addr_t dma_buf_next; + dma_addr_t dma_buf_end; + size_t period_size; + unsigned int num_periods; +}; + +/* + * Define the hardware characteristics for the PCM hardware. + * + * The PCM hardware is the Freescale DMA controller. This structure defines + * the capabilities of that hardware. + * + * Since the sampling rate and data format are not controlled by the DMA + * controller, we specify no limits for those values. The only exception is + * period_bytes_min, which is set to a reasonably low value to prevent the + * DMA controller from generating too many interrupts per second. + * + * Since each link descriptor has a 32-bit byte count field, we set + * period_bytes_max to the largest 32-bit number. We also have no maximum + * number of periods. + */ +static const struct snd_pcm_hardware fsl_dma_hardware = { + + .info = SNDRV_PCM_INFO_INTERLEAVED, + .formats = FSLDMA_PCM_FORMATS, + .rates = FSLDMA_PCM_RATES, + .rate_min = 5512, + .rate_max = 192000, + .period_bytes_min = 512, /* A reasonable limit */ + .period_bytes_max = (u32) -1, + .periods_min = NUM_DMA_LINKS, + .periods_max = (unsigned int) -1, + .buffer_bytes_max = 128 * 1024, /* A reasonable limit */ +}; + +/* + * Tell ALSA that the DMA transfer has aborted + * + * This function should be called by the ISR whenever the DMA controller + * halts data transfer. + */ +static void fsl_dma_abort_stream(struct snd_pcm_substream *substream) +{ + unsigned long flags; + + snd_pcm_stream_lock_irqsave(substream, flags); + + if (snd_pcm_running(substream)) + snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + + snd_pcm_stream_unlock_irqrestore(substream, flags); +} + +/* + * Update the link descriptor pointers to point to the next period buffer. +*/ +static void fsl_dma_update_pointers(struct fsl_dma_private *dma_private) +{ + struct fsl_dma_link_descriptor *link = + &dma_private->link[dma_private->current_link]; + + /* Update our link descriptors to point to the next period */ + if (dma_private->substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + link->source_addr = + cpu_to_be32(dma_private->dma_buf_next); + else + link->dest_addr = + cpu_to_be32(dma_private->dma_buf_next); + + /* Update our variables for next time */ + dma_private->dma_buf_next += dma_private->period_size; + + if (dma_private->dma_buf_next >= dma_private->dma_buf_end) + dma_private->dma_buf_next = dma_private->dma_buf_phys; + + if (++dma_private->current_link >= NUM_DMA_LINKS) + dma_private->current_link = 0; +} + +/* + * Interrupt handler for the DMA controller + */ +static irqreturn_t fsl_dma_isr(int irq, void *dev_id) +{ + struct fsl_dma_private *dma_private = dev_id; + struct ccsr_dma_channel __iomem *dma_channel = dma_private->dma_channel; + irqreturn_t ret = IRQ_NONE; + u32 sr, sr2 = 0; + + /* We got an interrupt, so read the status register to see what we + were interrupted for. + */ + sr = in_be32(&dma_channel->sr); + + if (sr & CCSR_DMA_SR_TE) { + dev_err(dma_private->substream->pcm->card->dev, + "DMA transmit error (controller=%u channel=%u irq=%u\n", + dma_private->controller_id, + dma_private->channel_id, irq); + fsl_dma_abort_stream(dma_private->substream); + sr2 |= CCSR_DMA_SR_TE; + ret = IRQ_HANDLED; + } + + if (sr & CCSR_DMA_SR_CH) + ret = IRQ_HANDLED; + + if (sr & CCSR_DMA_SR_PE) { + dev_err(dma_private->substream->pcm->card->dev, + "DMA%u programming error (channel=%u irq=%u)\n", + dma_private->controller_id, + dma_private->channel_id, irq); + fsl_dma_abort_stream(dma_private->substream); + sr2 |= CCSR_DMA_SR_PE; + ret = IRQ_HANDLED; + } + + if (sr & CCSR_DMA_SR_EOLNI) { + sr2 |= CCSR_DMA_SR_EOLNI; + ret = IRQ_HANDLED; + } + + if (sr & CCSR_DMA_SR_CB) + ret = IRQ_HANDLED; + + if (sr & CCSR_DMA_SR_EOSI) { + struct snd_pcm_substream *substream = dma_private->substream; + + /* Tell ALSA we completed a period. */ + snd_pcm_period_elapsed(substream); + + /* + * Update our link descriptors to point to the next period. We + * only need to do this if the number of periods is not equal to + * the number of links. + */ + if (dma_private->num_periods != NUM_DMA_LINKS) + fsl_dma_update_pointers(dma_private); + + sr2 |= CCSR_DMA_SR_EOSI; + ret = IRQ_HANDLED; + } + + if (sr & CCSR_DMA_SR_EOLSI) { + sr2 |= CCSR_DMA_SR_EOLSI; + ret = IRQ_HANDLED; + } + + /* Clear the bits that we set */ + if (sr2) + out_be32(&dma_channel->sr, sr2); + + return ret; +} + +/* + * Initialize this PCM driver. + * + * This function is called when the codec driver calls snd_soc_new_pcms(), + * once for each .dai_link in the machine driver's snd_soc_machine + * structure. + */ +static int fsl_dma_new(struct snd_card *card, struct snd_soc_codec_dai *dai, + struct snd_pcm *pcm) +{ + static u64 fsl_dma_dmamask = 0xffffffff; + int ret; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &fsl_dma_dmamask; + + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = fsl_dma_dmamask; + + ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, + fsl_dma_hardware.buffer_bytes_max, + &pcm->streams[0].substream->dma_buffer); + if (ret) { + dev_err(card->dev, + "Can't allocate playback DMA buffer (size=%u)\n", + fsl_dma_hardware.buffer_bytes_max); + return -ENOMEM; + } + + ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, + fsl_dma_hardware.buffer_bytes_max, + &pcm->streams[1].substream->dma_buffer); + if (ret) { + snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer); + dev_err(card->dev, + "Can't allocate capture DMA buffer (size=%u)\n", + fsl_dma_hardware.buffer_bytes_max); + return -ENOMEM; + } + + return 0; +} + +/* + * Open a new stream. + * + * Each stream has its own DMA buffer. + */ +static int fsl_dma_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsl_dma_private *dma_private; + dma_addr_t ld_buf_phys; + unsigned int channel; + int ret = 0; + + /* + * Reject any DMA buffer whose size is not a multiple of the period + * size. We need to make sure that the DMA buffer can be evenly divided + * into periods. + */ + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) { + dev_err(substream->pcm->card->dev, "invalid buffer size\n"); + return ret; + } + + channel = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; + + if (dma_global_data.assigned[channel]) { + dev_err(substream->pcm->card->dev, + "DMA channel already assigned\n"); + return -EBUSY; + } + + dma_private = dma_alloc_coherent(substream->pcm->dev, + sizeof(struct fsl_dma_private), &ld_buf_phys, GFP_KERNEL); + if (!dma_private) { + dev_err(substream->pcm->card->dev, + "can't allocate DMA private data\n"); + return -ENOMEM; + } + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dma_private->ssi_sxx_phys = dma_global_data.ssi_stx_phys; + else + dma_private->ssi_sxx_phys = dma_global_data.ssi_srx_phys; + + dma_private->dma_channel = dma_global_data.dma_channel[channel]; + dma_private->irq = dma_global_data.irq[channel]; + dma_private->substream = substream; + dma_private->ld_buf_phys = ld_buf_phys; + dma_private->dma_buf_phys = substream->dma_buffer.addr; + + /* We only support one DMA controller for now */ + dma_private->controller_id = 0; + dma_private->channel_id = channel; + + ret = request_irq(dma_private->irq, fsl_dma_isr, 0, "DMA", dma_private); + if (ret) { + dev_err(substream->pcm->card->dev, + "can't register ISR for IRQ %u (ret=%i)\n", + dma_private->irq, ret); + dma_free_coherent(substream->pcm->dev, + sizeof(struct fsl_dma_private), + dma_private, dma_private->ld_buf_phys); + return ret; + } + + dma_global_data.assigned[channel] = 1; + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + snd_soc_set_runtime_hwparams(substream, &fsl_dma_hardware); + runtime->private_data = dma_private; + + return 0; +} + +/* + * Allocate the DMA buffer and the DMA link descriptors. + * + * ALSA divides the DMA buffer into N periods. We create NUM_DMA_LINKS link + * descriptors that ping-pong from one period to the next. For example, if + * there are six periods and two link descriptors, this is how they look + * before playback starts: + * + * The last link descriptor + * ____________ points back to the first + * | | + * V | + * ___ ___ | + * | |->| |->| + * |___| |___| + * | | + * | | + * V V + * _________________________________________ + * | | | | | | | The DMA buffer is + * | | | | | | | divided into 6 parts + * |______|______|______|______|______|______| + * + * and here's how they look after the first period is finished playing: + * + * ____________ + * | | + * V | + * ___ ___ | + * | |->| |->| + * |___| |___| + * | | + * |______________ + * | | + * V V + * _________________________________________ + * | | | | | | | + * | | | | | | | + * |______|______|______|______|______|______| + * + * The first link descriptor now points to the third period. The DMA + * controller is currently playing the second period. When it finishes, it + * will jump back to the first descriptor and play the third period. + * + * There are four reasons we do this: + * + * 1. The only way to get the DMA controller to automatically restart the + * transfer when it gets to the end of the buffer is to use chaining + * mode. Basic direct mode doesn't offer that feature. + * 2. We need to receive an interrupt at the end of every period. The DMA + * controller can generate an interrupt at the end of every link transfer + * (aka segment). Making each period into a DMA segment will give us the + * interrupts we need. + * 3. By creating only two link descriptors, regardless of the number of + * periods, we do not need to reallocate the link descriptors if the + * number of periods changes. + * 4. All of the audio data is still stored in a single, contiguous DMA + * buffer, which is what ALSA expects. We're just dividing it into + * contiguous parts, and creating a link descriptor for each one. + * + * Note that due to a quirk of the SSI's STX register, the target address + * for the DMA operations depends on the sample size. So we don't program + * the dest_addr (for playback -- source_addr for capture) fields in the + * link descriptors here. We do that in fsl_dma_prepare() + */ +static int fsl_dma_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsl_dma_private *dma_private = runtime->private_data; + struct ccsr_dma_channel __iomem *dma_channel = dma_private->dma_channel; + + dma_addr_t temp_addr; /* Pointer to next period */ + u64 temp_link; /* Pointer to next link descriptor */ + u32 mr; /* Temporary variable for MR register */ + + unsigned int i; + + /* Get all the parameters we need */ + size_t buffer_size = params_buffer_bytes(hw_params); + size_t period_size = params_period_bytes(hw_params); + + /* Initialize our DMA tracking variables */ + dma_private->period_size = period_size; + dma_private->num_periods = params_periods(hw_params); + dma_private->dma_buf_end = dma_private->dma_buf_phys + buffer_size; + dma_private->dma_buf_next = dma_private->dma_buf_phys + + (NUM_DMA_LINKS * period_size); + if (dma_private->dma_buf_next >= dma_private->dma_buf_end) + dma_private->dma_buf_next = dma_private->dma_buf_phys; + + /* + * Initialize each link descriptor. + * + * The actual address in STX0 (destination for playback, source for + * capture) is based on the sample size, but we don't know the sample + * size in this function, so we'll have to adjust that later. See + * comments in fsl_dma_prepare(). + * + * The DMA controller does not have a cache, so the CPU does not + * need to tell it to flush its cache. However, the DMA + * controller does need to tell the CPU to flush its cache. + * That's what the SNOOP bit does. + * + * Also, even though the DMA controller supports 36-bit addressing, for + * simplicity we currently support only 32-bit addresses for the audio + * buffer itself. + */ + temp_addr = substream->dma_buffer.addr; + temp_link = dma_private->ld_buf_phys + + sizeof(struct fsl_dma_link_descriptor); + + for (i = 0; i < NUM_DMA_LINKS; i++) { + struct fsl_dma_link_descriptor *link = &dma_private->link[i]; + + link->count = cpu_to_be32(period_size); + link->source_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); + link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); + link->next = cpu_to_be64(temp_link); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + link->source_addr = cpu_to_be32(temp_addr); + else + link->dest_addr = cpu_to_be32(temp_addr); + + temp_addr += period_size; + temp_link += sizeof(struct fsl_dma_link_descriptor); + } + /* The last link descriptor points to the first */ + dma_private->link[i - 1].next = cpu_to_be64(dma_private->ld_buf_phys); + + /* Tell the DMA controller where the first link descriptor is */ + out_be32(&dma_channel->clndar, + CCSR_DMA_CLNDAR_ADDR(dma_private->ld_buf_phys)); + out_be32(&dma_channel->eclndar, + CCSR_DMA_ECLNDAR_ADDR(dma_private->ld_buf_phys)); + + /* The manual says the BCR must be clear before enabling EMP */ + out_be32(&dma_channel->bcr, 0); + + /* Program the mode register for interrupts, external master control, + and source/destination hold. Also clear the Channel Abort bit. + */ + mr = in_be32(&dma_channel->mr) & + ~(CCSR_DMA_MR_CA | CCSR_DMA_MR_DAHE | CCSR_DMA_MR_SAHE); + + /* + * We want External Master Start and External Master Pause enabled, + * because the SSI is controlling the DMA controller. We want the DMA + * controller to be set up in advance, and then we signal only the SSI + * to start transfering. + * + * We want End-Of-Segment Interrupts enabled, because this will generate + * an interrupt at the end of each segment (each link descriptor + * represents one segment). Each DMA segment is the same thing as an + * ALSA period, so this is how we get an interrupt at the end of every + * period. + * + * We want Error Interrupt enabled, so that we can get an error if + * the DMA controller is mis-programmed somehow. + */ + mr |= CCSR_DMA_MR_EOSIE | CCSR_DMA_MR_EIE | CCSR_DMA_MR_EMP_EN | + CCSR_DMA_MR_EMS_EN; + + /* For playback, we want the destination address to be held. For + capture, set the source address to be held. */ + mr |= (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + CCSR_DMA_MR_DAHE : CCSR_DMA_MR_SAHE; + + out_be32(&dma_channel->mr, mr); + + return 0; +} + +/* + * Prepare the DMA registers for playback. + * + * This function is called after the specifics of the audio data are known, + * i.e. snd_pcm_runtime is initialized. + * + * In this function, we finish programming the registers of the DMA + * controller that are dependent on the sample size. + * + * One of the drawbacks with big-endian is that when copying integers of + * different sizes to a fixed-sized register, the address to which the + * integer must be copied is dependent on the size of the integer. + * + * For example, if P is the address of a 32-bit register, and X is a 32-bit + * integer, then X should be copied to address P. However, if X is a 16-bit + * integer, then it should be copied to P+2. If X is an 8-bit register, + * then it should be copied to P+3. + * + * So for playback of 8-bit samples, the DMA controller must transfer single + * bytes from the DMA buffer to the last byte of the STX0 register, i.e. + * offset by 3 bytes. For 16-bit samples, the offset is two bytes. + * + * For 24-bit samples, the offset is 1 byte. However, the DMA controller + * does not support 3-byte copies (the DAHTS register supports only 1, 2, 4, + * and 8 bytes at a time). So we do not support packed 24-bit samples. + * 24-bit data must be padded to 32 bits. + */ +static int fsl_dma_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsl_dma_private *dma_private = runtime->private_data; + struct ccsr_dma_channel __iomem *dma_channel = dma_private->dma_channel; + u32 mr; + unsigned int i; + dma_addr_t ssi_sxx_phys; /* Bus address of SSI STX register */ + unsigned int frame_size; /* Number of bytes per frame */ + + ssi_sxx_phys = dma_private->ssi_sxx_phys; + + mr = in_be32(&dma_channel->mr) & ~(CCSR_DMA_MR_BWC_MASK | + CCSR_DMA_MR_SAHTS_MASK | CCSR_DMA_MR_DAHTS_MASK); + + switch (runtime->sample_bits) { + case 8: + mr |= CCSR_DMA_MR_DAHTS_1 | CCSR_DMA_MR_SAHTS_1; + ssi_sxx_phys += 3; + break; + case 16: + mr |= CCSR_DMA_MR_DAHTS_2 | CCSR_DMA_MR_SAHTS_2; + ssi_sxx_phys += 2; + break; + case 32: + mr |= CCSR_DMA_MR_DAHTS_4 | CCSR_DMA_MR_SAHTS_4; + break; + default: + dev_err(substream->pcm->card->dev, + "unsupported sample size %u\n", runtime->sample_bits); + return -EINVAL; + } + + frame_size = runtime->frame_bits / 8; + /* + * BWC should always be a multiple of the frame size. BWC determines + * how many bytes are sent/received before the DMA controller checks the + * SSI to see if it needs to stop. For playback, the transmit FIFO can + * hold three frames, so we want to send two frames at a time. For + * capture, the receive FIFO is triggered when it contains one frame, so + * we want to receive one frame at a time. + */ + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + mr |= CCSR_DMA_MR_BWC(2 * frame_size); + else + mr |= CCSR_DMA_MR_BWC(frame_size); + + out_be32(&dma_channel->mr, mr); + + /* + * Program the address of the DMA transfer to/from the SSI. + */ + for (i = 0; i < NUM_DMA_LINKS; i++) { + struct fsl_dma_link_descriptor *link = &dma_private->link[i]; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + link->dest_addr = cpu_to_be32(ssi_sxx_phys); + else + link->source_addr = cpu_to_be32(ssi_sxx_phys); + } + + return 0; +} + +/* + * fsl_dma_pointer - 'pointer' PCM operation callback function + * + * This function is called by ALSA when ALSA wants to know where in the + * stream buffer the hardware currently is. + * + * For playback, the SAR register contains the physical address of the most + * recent DMA transfer. For capture, the value is in the DAR register. + * + * The base address of the buffer is stored in the source_addr field of the + * first link descriptor. + */ +static snd_pcm_uframes_t fsl_dma_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsl_dma_private *dma_private = runtime->private_data; + struct ccsr_dma_channel __iomem *dma_channel = dma_private->dma_channel; + dma_addr_t position; + snd_pcm_uframes_t frames; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + position = in_be32(&dma_channel->sar); + else + position = in_be32(&dma_channel->dar); + + frames = bytes_to_frames(runtime, position - dma_private->dma_buf_phys); + + /* + * If the current address is just past the end of the buffer, wrap it + * around. + */ + if (frames == runtime->buffer_size) + frames = 0; + + return frames; +} + +/* + * Release the resources allocated in fsl_dma_hw_params() and + * de-program the registers. This function can be called multiple times. + */ +static int fsl_dma_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsl_dma_private *dma_private = runtime->private_data; + + if (dma_private) { + struct ccsr_dma_channel __iomem *dma_channel; + + dma_channel = dma_private->dma_channel; + + dump_dma_regs(__FUNCTION__, __LINE__, dma_channel); + dump_lds(dma_private); + + /* Stop the DMA */ + out_be32(&dma_channel->mr, CCSR_DMA_MR_CA); + out_be32(&dma_channel->mr, 0); + + /* Reset all the other registers */ + out_be32(&dma_channel->sr, -1); + out_be32(&dma_channel->clndar, 0); + out_be32(&dma_channel->eclndar, 0); + out_be32(&dma_channel->satr, 0); + out_be32(&dma_channel->sar, 0); + out_be32(&dma_channel->datr, 0); + out_be32(&dma_channel->dar, 0); + out_be32(&dma_channel->bcr, 0); + out_be32(&dma_channel->nlndar, 0); + out_be32(&dma_channel->enlndar, 0); + } + + return 0; +} + +/* + * Close the stream. + */ +static int fsl_dma_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsl_dma_private *dma_private = runtime->private_data; + int dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; + + if (dma_private) { + if (dma_private->irq) + free_irq(dma_private->irq, dma_private); + + if (dma_private->ld_buf_phys) { + dma_unmap_single(substream->pcm->dev, + dma_private->ld_buf_phys, + sizeof(dma_private->link), DMA_TO_DEVICE); + } + + /* Deallocate the fsl_dma_private structure */ + dma_free_coherent(substream->pcm->dev, + sizeof(struct fsl_dma_private), + dma_private, dma_private->ld_buf_phys); + substream->runtime->private_data = NULL; + } + + dma_global_data.assigned[dir] = 0; + + return 0; +} + +/* + * Remove this PCM driver. + */ +static void fsl_dma_free_dma_buffers(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + unsigned int i; + + for (i = 0; i < ARRAY_SIZE(pcm->streams); i++) { + substream = pcm->streams[i].substream; + if (substream) { + snd_dma_free_pages(&substream->dma_buffer); + substream->dma_buffer.area = NULL; + substream->dma_buffer.addr = 0; + } + } +} + +static struct snd_pcm_ops fsl_dma_ops = { + .open = fsl_dma_open, + .close = fsl_dma_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = fsl_dma_hw_params, + .hw_free = fsl_dma_hw_free, + .prepare = fsl_dma_prepare, + .pointer = fsl_dma_pointer, +}; + +struct snd_soc_platform fsl_soc_platform = { + .name = "fsl-dma", + .pcm_ops = &fsl_dma_ops, + .pcm_new = fsl_dma_new, + .pcm_free = fsl_dma_free_dma_buffers, +}; +EXPORT_SYMBOL_GPL(fsl_soc_platform); + +int fsl_dma_configure(struct fsl_dma_info *dma_info) +{ + static int initialized; + + /* We only support one DMA controller for now */ + if (initialized) + return 0; + + dma_global_data.ssi_stx_phys = dma_info->ssi_stx_phys; + dma_global_data.ssi_srx_phys = dma_info->ssi_srx_phys; + dma_global_data.dma_channel[0] = dma_info->dma_channel[0]; + dma_global_data.dma_channel[1] = dma_info->dma_channel[1]; + dma_global_data.irq[0] = dma_info->dma_irq[0]; + dma_global_data.irq[1] = dma_info->dma_irq[1]; + dma_global_data.assigned[0] = 0; + dma_global_data.assigned[1] = 0; + + initialized = 1; + return 1; +} +EXPORT_SYMBOL_GPL(fsl_dma_configure); + +MODULE_AUTHOR("Timur Tabi <timur(a)freescale.com>"); +MODULE_DESCRIPTION("Freescale FSL Elo DMA ASoC PCM module"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/fsl/fsl_dma.h b/sound/soc/fsl/fsl_dma.h new file mode 100644 index 0000000..430a6ce --- /dev/null +++ b/sound/soc/fsl/fsl_dma.h @@ -0,0 +1,149 @@ +/* + * mpc8610-pcm.h - ALSA PCM interface for the Freescale MPC8610 SoC + * + * 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 _MPC8610_PCM_H +#define _MPC8610_PCM_H + +struct ccsr_dma { + u8 res0[0x100]; + struct ccsr_dma_channel { + __be32 mr; /* Mode register */ + __be32 sr; /* Status register */ + __be32 eclndar; /* Current link descriptor extended addr reg */ + __be32 clndar; /* Current link descriptor address register */ + __be32 satr; /* Source attributes register */ + __be32 sar; /* Source address register */ + __be32 datr; /* Destination attributes register */ + __be32 dar; /* Destination address register */ + __be32 bcr; /* Byte count register */ + __be32 enlndar; /* Next link descriptor extended address reg */ + __be32 nlndar; /* Next link descriptor address register */ + u8 res1[4]; + __be32 eclsdar; /* Current list descriptor extended addr reg */ + __be32 clsdar; /* Current list descriptor address register */ + __be32 enlsdar; /* Next list descriptor extended address reg */ + __be32 nlsdar; /* Next list descriptor address register */ + __be32 ssr; /* Source stride register */ + __be32 dsr; /* Destination stride register */ + u8 res2[0x38]; + } channel[4]; + __be32 dgsr; +}; + +#define CCSR_DMA_MR_BWC_DISABLED 0x0F000000 +#define CCSR_DMA_MR_BWC_SHIFT 24 +#define CCSR_DMA_MR_BWC_MASK 0x0F000000 +#define CCSR_DMA_MR_BWC(x) \ + ((ilog2(x) << CCSR_DMA_MR_BWC_SHIFT) & CCSR_DMA_MR_BWC_MASK) +#define CCSR_DMA_MR_EMP_EN 0x00200000 +#define CCSR_DMA_MR_EMS_EN 0x00040000 +#define CCSR_DMA_MR_DAHTS_MASK 0x00030000 +#define CCSR_DMA_MR_DAHTS_1 0x00000000 +#define CCSR_DMA_MR_DAHTS_2 0x00010000 +#define CCSR_DMA_MR_DAHTS_4 0x00020000 +#define CCSR_DMA_MR_DAHTS_8 0x00030000 +#define CCSR_DMA_MR_SAHTS_MASK 0x0000C000 +#define CCSR_DMA_MR_SAHTS_1 0x00000000 +#define CCSR_DMA_MR_SAHTS_2 0x00004000 +#define CCSR_DMA_MR_SAHTS_4 0x00008000 +#define CCSR_DMA_MR_SAHTS_8 0x0000C000 +#define CCSR_DMA_MR_DAHE 0x00002000 +#define CCSR_DMA_MR_SAHE 0x00001000 +#define CCSR_DMA_MR_SRW 0x00000400 +#define CCSR_DMA_MR_EOSIE 0x00000200 +#define CCSR_DMA_MR_EOLNIE 0x00000100 +#define CCSR_DMA_MR_EOLSIE 0x00000080 +#define CCSR_DMA_MR_EIE 0x00000040 +#define CCSR_DMA_MR_XFE 0x00000020 +#define CCSR_DMA_MR_CDSM_SWSM 0x00000010 +#define CCSR_DMA_MR_CA 0x00000008 +#define CCSR_DMA_MR_CTM 0x00000004 +#define CCSR_DMA_MR_CC 0x00000002 +#define CCSR_DMA_MR_CS 0x00000001 + +#define CCSR_DMA_SR_TE 0x00000080 +#define CCSR_DMA_SR_CH 0x00000020 +#define CCSR_DMA_SR_PE 0x00000010 +#define CCSR_DMA_SR_EOLNI 0x00000008 +#define CCSR_DMA_SR_CB 0x00000004 +#define CCSR_DMA_SR_EOSI 0x00000002 +#define CCSR_DMA_SR_EOLSI 0x00000001 + +/* ECLNDAR takes bits 32-36 of the CLNDAR register */ +static inline u32 CCSR_DMA_ECLNDAR_ADDR(u64 x) +{ + return (x >> 32) & 0xf; +} + +#define CCSR_DMA_CLNDAR_ADDR(x) ((x) & 0xFFFFFFFE) +#define CCSR_DMA_CLNDAR_EOSIE 0x00000008 + +/* SATR and DATR, combined */ +#define CCSR_DMA_ATR_PBATMU 0x20000000 +#define CCSR_DMA_ATR_TFLOWLVL_0 0x00000000 +#define CCSR_DMA_ATR_TFLOWLVL_1 0x06000000 +#define CCSR_DMA_ATR_TFLOWLVL_2 0x08000000 +#define CCSR_DMA_ATR_TFLOWLVL_3 0x0C000000 +#define CCSR_DMA_ATR_PCIORDER 0x02000000 +#define CCSR_DMA_ATR_SME 0x01000000 +#define CCSR_DMA_ATR_NOSNOOP 0x00040000 +#define CCSR_DMA_ATR_SNOOP 0x00050000 +#define CCSR_DMA_ATR_ESAD_MASK 0x0000000F + +/** + * List Descriptor for extended chaining mode DMA operations. + * + * The CLSDAR register points to the first (in a linked-list) List + * Descriptor. Each object must be aligned on a 32-byte boundary. Each + * list descriptor points to a linked-list of link Descriptors. + */ +struct fsl_dma_list_descriptor { + __be64 next; /* Address of next list descriptor */ + __be64 first_link; /* Address of first link descriptor */ + __be32 source; /* Source stride */ + __be32 dest; /* Destination stride */ + u8 res[8]; /* Reserved */ +} __attribute__ ((aligned(32), packed)); + +/** + * Link Descriptor for basic and extended chaining mode DMA operations. + * + * A Link Descriptor points to a single DMA buffer. Each link descriptor + * must be aligned on a 32-byte boundary. + */ +struct fsl_dma_link_descriptor { + __be32 source_attr; /* Programmed into SATR register */ + __be32 source_addr; /* Programmed into SAR register */ + __be32 dest_attr; /* Programmed into DATR register */ + __be32 dest_addr; /* Programmed into DAR register */ + __be64 next; /* Address of next link descriptor */ + __be32 count; /* Byte count */ + u8 res[4]; /* Reserved */ +} __attribute__ ((aligned(32), packed)); + +/* DMA information needed to create a snd_soc_cpu_dai object + * + * ssi_stx_phys: bus address of SSI STX register to use + * ssi_srx_phys: bus address of SSI SRX register to use + * dma[0]: points to the DMA channel to use for playback + * dma[1]: points to the DMA channel to use for capture + * dma_irq[0]: IRQ of the DMA channel to use for playback + * dma_irq[1]: IRQ of the DMA channel to use for capture + */ +struct fsl_dma_info { + dma_addr_t ssi_stx_phys; + dma_addr_t ssi_srx_phys; + struct ccsr_dma_channel __iomem *dma_channel[2]; + unsigned int dma_irq[2]; +}; + +extern struct snd_soc_platform fsl_soc_platform; + +int fsl_dma_configure(struct fsl_dma_info *dma_info); + +#endif diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c new file mode 100644 index 0000000..a219e58 --- /dev/null +++ b/sound/soc/fsl/fsl_ssi.c @@ -0,0 +1,614 @@ +/* + * Freescale SSI ALSA SoC Digital Audio Interface (DAI) driver + * + * Author: Timur Tabi <timur(a)freescale.com> + * + * Copyright 2007 Freescale Semiconductor, Inc. This file is licensed under + * the terms of the GNU General Public License version 2. This program + * is licensed "as is" without any warranty of any kind, whether express + * or implied. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/interrupt.h> +#include <linux/device.h> +#include <linux/delay.h> + +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include <asm/immap_86xx.h> + +#include "fsl_ssi.h" + +/* + * The sample rates supported by the I2S + * + * This driver currently only supports the SSI running in I2S slave mode, + * which means the codec determines the sample rate. Therefore, we tell + * ALSA that we support all rates and let the codec driver decide what rates + * are really supported. + */ +#define FSLSSI_I2S_RATES (SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_192000 | \ + SNDRV_PCM_RATE_CONTINUOUS) + +/* + * The audio formats supported by the SSI + * + * This driver currently only supports the SSI running in I2S slave mode. + * + * The SSI has a limitation in that the samples must be in the same byte + * order as the host CPU. This is because when multiple bytes are written + * to the STX register, the bytes and bits must be written in the same + * order. The STX is a shift register, so all the bits need to be aligned + * (bit-endianness must match byte-endianness). Processors typically write + * the bits within a byte in the same order that the bytes of a word are + * written in. So if the host CPU is big-endian, then only big-endian + * samples will be written to STX properly. + */ +#ifdef __BIG_ENDIAN +#define FSLSSI_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_S18_3BE | SNDRV_PCM_FMTBIT_S20_3BE | \ + SNDRV_PCM_FMTBIT_S24_3BE | SNDRV_PCM_FMTBIT_S24_BE) +#else +#define FSLSSI_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE) +#endif + +/* SSI information. + * + * name: short name for this device ("SSI0", "SSI1", etc) + * ssi: pointer to the SSI's registers + * ssi_phys: physical address of the SSI registers + * irq: IRQ of this SSI + * dev: struct device pointer + * playback: the number of playback streams opened + * capture: the number of capture streams opened + */ +struct fsl_ssi_private { + char name[8]; + struct ccsr_ssi __iomem *ssi; + dma_addr_t ssi_phys; + unsigned int irq; + struct device *dev; + unsigned int playback; + unsigned int capture; + struct snd_soc_cpu_dai cpu_dai; + struct device_attribute dev_attr; + + struct { + unsigned int rfrc; + unsigned int tfrc; + unsigned int cmdau; + unsigned int cmddu; + unsigned int rxt; + unsigned int rdr1; + unsigned int rdr0; + unsigned int tde1; + unsigned int tde0; + unsigned int roe1; + unsigned int roe0; + unsigned int tue1; + unsigned int tue0; + unsigned int tfs; + unsigned int rfs; + unsigned int tls; + unsigned int rls; + unsigned int rff1; + unsigned int rff0; + unsigned int tfe1; + unsigned int tfe0; + } stats; +}; + +/* + * SSI interrupt handler. + * + * Do we even want an SSI interrupt handler? Maybe for error handling, but + * buffer processing is handled via DMA interrupts. + */ +static irqreturn_t fsl_ssi_isr(int irq, void *dev_id) +{ + struct fsl_ssi_private *ssi_private = dev_id; + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + irqreturn_t ret = IRQ_NONE; + __be32 sisr; + __be32 sisr2 = 0; + + /* We got an interrupt, so read the status register to see what we + were interrupted for. We mask it with the Interrupt Enable register + so that we only check for events that we're interested in. + */ + sisr = in_be32(&ssi->sisr) & in_be32(&ssi->sier); + + if (sisr & CCSR_SSI_SISR_RFRC) { + ssi_private->stats.rfrc++; + sisr2 |= CCSR_SSI_SISR_RFRC; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_TFRC) { + ssi_private->stats.tfrc++; + sisr2 |= CCSR_SSI_SISR_TFRC; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_CMDAU) { + ssi_private->stats.cmdau++; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_CMDDU) { + ssi_private->stats.cmddu++; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_RXT) { + ssi_private->stats.rxt++; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_RDR1) { + ssi_private->stats.rdr1++; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_RDR0) { + ssi_private->stats.rdr0++; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_TDE1) { + ssi_private->stats.tde1++; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_TDE0) { + ssi_private->stats.tde0++; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_ROE1) { + ssi_private->stats.roe1++; + sisr2 |= CCSR_SSI_SISR_ROE1; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_ROE0) { + ssi_private->stats.roe0++; + sisr2 |= CCSR_SSI_SISR_ROE0; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_TUE1) { + ssi_private->stats.tue1++; + sisr2 |= CCSR_SSI_SISR_TUE1; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_TUE0) { + ssi_private->stats.tue0++; + sisr2 |= CCSR_SSI_SISR_TUE0; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_TFS) { + ssi_private->stats.tfs++; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_RFS) { + ssi_private->stats.rfs++; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_TLS) { + ssi_private->stats.tls++; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_RLS) { + ssi_private->stats.rls++; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_RFF1) { + ssi_private->stats.rff1++; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_RFF0) { + ssi_private->stats.rff0++; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_TFE1) { + ssi_private->stats.tfe1++; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_TFE0) { + ssi_private->stats.tfe0++; + ret = IRQ_HANDLED; + } + + /* Clear the bits that we set */ + if (sisr2) + out_be32(&ssi->sisr, sisr2); + + return ret; +} + +/* + * Startup + * + * This is the first function called when a stream is opened. + * + * If this is the first stream open, then grab the IRQ and program most of + * the SSI registers. + */ +static int fsl_ssi_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data; + + /* + * If this is the first stream opened, then request the IRQ + * and initialize the SSI registers. + */ + if (!ssi_private->playback && !ssi_private->capture) { + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + int ret; + + ret = request_irq(ssi_private->irq, fsl_ssi_isr, 0, + ssi_private->name, ssi_private); + if (ret < 0) { + dev_err(substream->pcm->card->dev, + "could not claim irq %u\n", ssi_private->irq); + return ret; + } + + /* + * Section 16.5 of the MPC8610 reference manual says that the + * SSI needs to be disabled before updating the registers we set + * here. + */ + clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + + /* + * Program the SSI into I2S Slave Non-Network Synchronous mode. + * Also enable the transmit and receive FIFO. + * + * FIXME: Little-endian samples require a different shift dir + */ + clrsetbits_be32(&ssi->scr, CCSR_SSI_SCR_I2S_MODE_MASK, + CCSR_SSI_SCR_TFR_CLK_DIS | + CCSR_SSI_SCR_I2S_MODE_SLAVE | CCSR_SSI_SCR_SYN); + + out_be32(&ssi->stcr, + CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFEN0 | + CCSR_SSI_STCR_TFSI | CCSR_SSI_STCR_TEFS | + CCSR_SSI_STCR_TSCKP); + + out_be32(&ssi->srcr, + CCSR_SSI_SRCR_RXBIT0 | CCSR_SSI_SRCR_RFEN0 | + CCSR_SSI_SRCR_RFSI | CCSR_SSI_SRCR_REFS | + CCSR_SSI_SRCR_RSCKP); + + /* + * The DC and PM bits are only used if the SSI is the clock + * master. + */ + + /* 4. Enable the interrupts and DMA requests */ + out_be32(&ssi->sier, + CCSR_SSI_SIER_TFRC_EN | CCSR_SSI_SIER_TDMAE | + CCSR_SSI_SIER_TIE | CCSR_SSI_SIER_TUE0_EN | + CCSR_SSI_SIER_TUE1_EN | CCSR_SSI_SIER_RFRC_EN | + CCSR_SSI_SIER_RDMAE | CCSR_SSI_SIER_RIE | + CCSR_SSI_SIER_ROE0_EN | CCSR_SSI_SIER_ROE1_EN); + + /* + * Set the watermark for transmit FIFI 0 and receive FIFO 0. We + * don't use FIFO 1. Since the SSI only supports stereo, the + * watermark should never be an odd number. + */ + out_be32(&ssi->sfcsr, + CCSR_SSI_SFCSR_TFWM0(6) | CCSR_SSI_SFCSR_RFWM0(2)); + + /* + * We keep the SSI disabled because if we enable it, then the + * DMA controller will start. It's not supposed to start until + * the SCR.TE (or SCR.RE) bit is set, but it does anyway. The + * DMA controller will transfer one "BWC" of data (i.e. the + * amount of data that the MR.BWC bits are set to). The reason + * this is bad is because at this point, the PCM driver has not + * finished initializing the DMA controller. + */ + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + ssi_private->playback++; + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + ssi_private->capture++; + + return 0; +} + +/* + * Prepare the SSI. + * + * Most of the SSI registers have been programmed in the startup function, + * but the word length must be programmed here. Unfortunately, programming + * the SxCCR.WL bits requires the SSI to be temporarily disabled. This can + * cause a problem with supporting simultaneous playback and capture. If + * the SSI is already playing a stream, then that stream may be temporarily + * stopped when you start capture. + * + * Note: The SxCCR.DC and SxCCR.PM bits are only used if the SSI is the + * clock master. + */ +static int fsl_ssi_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data; + + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + u32 wl; + + wl = CCSR_SSI_SxCCR_WL(snd_pcm_format_width(runtime->format)); + + clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + clrsetbits_be32(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl); + else + clrsetbits_be32(&ssi->srccr, CCSR_SSI_SxCCR_WL_MASK, wl); + + setbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + + return 0; +} + +/* + * Start and stop the DMA transfer. + * + * This function is called by ALSA to start, stop, pause, and resume the DMA + * transfer of data. + * + * The SSI has put the DMA in external master start and pause mode, which + * means the SSI completely controls the flow of data. + */ +static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data; + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + + 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) { + setbits32(&ssi->scr, CCSR_SSI_SCR_TE); + } else { + setbits32(&ssi->scr, CCSR_SSI_SCR_RE); + + /* + * I have no idea why we need a delay here. Without it, + * the DMA won't start (or the SSI won't tell the DMA to + * start). + */ + msleep(1); + } + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + clrbits32(&ssi->scr, CCSR_SSI_SCR_TE); + else + clrbits32(&ssi->scr, CCSR_SSI_SCR_RE); + break; + + default: + return -EINVAL; + } + + return 0; +} + +/* + * Shutdown. Clear DMA parameters and shutdown the SSI if there + * are no other substreams open. + */ +static void fsl_ssi_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + ssi_private->playback--; + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + ssi_private->capture--; + + /* + * If this is the last active substream, disable the SSI and release + * the IRQ. + */ + if (!ssi_private->playback && !ssi_private->capture) { + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + + clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + + free_irq(ssi_private->irq, ssi_private); + } +} + +/* Set the clock frequency and direction + * + * This function is called by the machine driver to tell us what the clock + * frequency and direction are. + * + * Currently, we only support operating as a clock slave (SND_SOC_CLOCK_IN), + * and we don't care about the frequency. Return an error if the direction + * is not SND_SOC_CLOCK_IN. + */ +static int fsl_ssi_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + + return (dir == SND_SOC_CLOCK_IN) ? 0 : -EINVAL; +} + +/* Set the serial format. + * + * This function is called by the machine driver to tell us what serial + * format to use. + * + * Currently, we only support I2S mode. Return an error if the format is + * not SND_SOC_DAIFMT_I2S. + */ +static int fsl_ssi_set_fmt(struct snd_soc_cpu_dai *cpu_dai, unsigned int format) +{ + return (format == SND_SOC_DAIFMT_I2S) ? 0 : -EINVAL; +} + +/* + * Template CPU DAI for the SSI + */ +static struct snd_soc_cpu_dai fsl_ssi_dai_template = { + .playback = { + /* The SSI does not support monaural audio. */ + .channels_min = 2, + .channels_max = 2, + .rates = FSLSSI_I2S_RATES, + .formats = FSLSSI_I2S_FORMATS, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .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, + }, + .dai_ops = { + .set_sysclk = fsl_ssi_set_sysclk, + .set_fmt = fsl_ssi_set_fmt, + }, +}; + +static ssize_t fsl_sysfs_ssi_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct fsl_ssi_private *ssi_private = + container_of(attr, struct fsl_ssi_private, dev_attr); + ssize_t length; + + length = sprintf(buf, "rfrc=%u", ssi_private->stats.rfrc); + length += sprintf(buf + length, "\ttfrc=%u", ssi_private->stats.tfrc); + length += sprintf(buf + length, "\tcmdau=%u", ssi_private->stats.cmdau); + length += sprintf(buf + length, "\tcmddu=%u", ssi_private->stats.cmddu); + length += sprintf(buf + length, "\trxt=%u", ssi_private->stats.rxt); + length += sprintf(buf + length, "\trdr1=%u", ssi_private->stats.rdr1); + length += sprintf(buf + length, "\trdr0=%u", ssi_private->stats.rdr0); + length += sprintf(buf + length, "\ttde1=%u", ssi_private->stats.tde1); + length += sprintf(buf + length, "\ttde0=%u", ssi_private->stats.tde0); + length += sprintf(buf + length, "\troe1=%u", ssi_private->stats.roe1); + length += sprintf(buf + length, "\troe0=%u", ssi_private->stats.roe0); + length += sprintf(buf + length, "\ttue1=%u", ssi_private->stats.tue1); + length += sprintf(buf + length, "\ttue0=%u", ssi_private->stats.tue0); + length += sprintf(buf + length, "\ttfs=%u", ssi_private->stats.tfs); + length += sprintf(buf + length, "\trfs=%u", ssi_private->stats.rfs); + length += sprintf(buf + length, "\ttls=%u", ssi_private->stats.tls); + length += sprintf(buf + length, "\trls=%u", ssi_private->stats.rls); + length += sprintf(buf + length, "\trff1=%u", ssi_private->stats.rff1); + length += sprintf(buf + length, "\trff0=%u", ssi_private->stats.rff0); + length += sprintf(buf + length, "\ttfe1=%u", ssi_private->stats.tfe1); + length += sprintf(buf + length, "\ttfe0=%u\n", ssi_private->stats.tfe0); + + return length; +} + +/* Create a snd_soc_cpu_dai structure + * + * This function is called by the machine driver to create a snd_soc_cpu_dai + * structure. A private data structure, with SSI hardware information, is + * appended to it. + */ +struct snd_soc_cpu_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info) +{ + struct snd_soc_cpu_dai *fsl_ssi_dai; + struct fsl_ssi_private *ssi_private; + int ret = 0; + struct device_attribute *dev_attr; + + ssi_private = kzalloc(sizeof(struct fsl_ssi_private), GFP_KERNEL); + if (!ssi_private) { + dev_err(ssi_info->dev, "could not allocate DAI object\n"); + return NULL; + } + memcpy(&ssi_private->cpu_dai, &fsl_ssi_dai_template, + sizeof(struct snd_soc_cpu_dai)); + + fsl_ssi_dai = &ssi_private->cpu_dai; + dev_attr = &ssi_private->dev_attr; + + sprintf(ssi_private->name, "ssi%u", (u8) ssi_info->id); + ssi_private->ssi = ssi_info->ssi; + ssi_private->ssi_phys = ssi_info->ssi_phys; + ssi_private->irq = ssi_info->irq; + ssi_private->dev = ssi_info->dev; + + ssi_private->dev->driver_data = fsl_ssi_dai; + + /* Initialize the the device_attribute structure */ + dev_attr->attr.name = "ssi-stats"; + dev_attr->attr.mode = S_IRUGO; + dev_attr->show = fsl_sysfs_ssi_show; + + ret = device_create_file(ssi_private->dev, dev_attr); + if (ret) { + dev_err(ssi_info->dev, "could not create sysfs %s file\n", + ssi_private->dev_attr.attr.name); + kfree(fsl_ssi_dai); + return NULL; + } + + fsl_ssi_dai->private_data = ssi_private; + fsl_ssi_dai->name = ssi_private->name; + fsl_ssi_dai->id = ssi_info->id; + + return fsl_ssi_dai; +} +EXPORT_SYMBOL_GPL(fsl_ssi_create_dai); + +void fsl_ssi_destroy_dai(struct snd_soc_cpu_dai *fsl_ssi_dai) +{ + struct fsl_ssi_private *ssi_private = + container_of(fsl_ssi_dai, struct fsl_ssi_private, cpu_dai); + + device_remove_file(ssi_private->dev, &ssi_private->dev_attr); + + kfree(ssi_private); +} +EXPORT_SYMBOL_GPL(fsl_ssi_destroy_dai); + +MODULE_AUTHOR("Timur Tabi <timur(a)freescale.com>"); +MODULE_DESCRIPTION("Freescale Synchronous Serial Interface (SSI) ASoC Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h new file mode 100644 index 0000000..f6e33af --- /dev/null +++ b/sound/soc/fsl/fsl_ssi.h @@ -0,0 +1,224 @@ +/* + * fsl_ssi.h - ALSA SSI interface for the Freescale MPC8610 SoC + * + * Author: Timur Tabi <timur(a)freescale.com> + * + * Copyright 2007 Freescale Semiconductor, Inc. This file is licensed under + * the terms of the GNU General Public License version 2. This program + * is licensed "as is" without any warranty of any kind, whether express + * or implied. + */ + +#ifndef _MPC8610_I2S_H +#define _MPC8610_I2S_H + +/* SSI Register Map */ +struct ccsr_ssi { + __be32 stx0; /* 0x.0000 - SSI Transmit Data Register 0 */ + __be32 stx1; /* 0x.0004 - SSI Transmit Data Register 1 */ + __be32 srx0; /* 0x.0008 - SSI Receive Data Register 0 */ + __be32 srx1; /* 0x.000C - SSI Receive Data Register 1 */ + __be32 scr; /* 0x.0010 - SSI Control Register */ + __be32 sisr; /* 0x.0014 - SSI Interrupt Status Register Mixed */ + __be32 sier; /* 0x.0018 - SSI Interrupt Enable Register */ + __be32 stcr; /* 0x.001C - SSI Transmit Configuration Register */ + __be32 srcr; /* 0x.0020 - SSI Receive Configuration Register */ + __be32 stccr; /* 0x.0024 - SSI Transmit Clock Control Register */ + __be32 srccr; /* 0x.0028 - SSI Receive Clock Control Register */ + __be32 sfcsr; /* 0x.002C - SSI FIFO Control/Status Register */ + __be32 str; /* 0x.0030 - SSI Test Register */ + __be32 sor; /* 0x.0034 - SSI Option Register */ + __be32 sacnt; /* 0x.0038 - SSI AC97 Control Register */ + __be32 sacadd; /* 0x.003C - SSI AC97 Command Address Register */ + __be32 sacdat; /* 0x.0040 - SSI AC97 Command Data Register */ + __be32 satag; /* 0x.0044 - SSI AC97 Tag Register */ + __be32 stmsk; /* 0x.0048 - SSI Transmit Time Slot Mask Register */ + __be32 srmsk; /* 0x.004C - SSI Receive Time Slot Mask Register */ + __be32 saccst; /* 0x.0050 - SSI AC97 Channel Status Register */ + __be32 saccen; /* 0x.0054 - SSI AC97 Channel Enable Register */ + __be32 saccdis; /* 0x.0058 - SSI AC97 Channel Disable Register */ +}; + +#define CCSR_SSI_SCR_RFR_CLK_DIS 0x00000800 +#define CCSR_SSI_SCR_TFR_CLK_DIS 0x00000400 +#define CCSR_SSI_SCR_TCH_EN 0x00000100 +#define CCSR_SSI_SCR_SYS_CLK_EN 0x00000080 +#define CCSR_SSI_SCR_I2S_MODE_MASK 0x00000060 +#define CCSR_SSI_SCR_I2S_MODE_NORMAL 0x00000000 +#define CCSR_SSI_SCR_I2S_MODE_MASTER 0x00000020 +#define CCSR_SSI_SCR_I2S_MODE_SLAVE 0x00000040 +#define CCSR_SSI_SCR_SYN 0x00000010 +#define CCSR_SSI_SCR_NET 0x00000008 +#define CCSR_SSI_SCR_RE 0x00000004 +#define CCSR_SSI_SCR_TE 0x00000002 +#define CCSR_SSI_SCR_SSIEN 0x00000001 + +#define CCSR_SSI_SISR_RFRC 0x01000000 +#define CCSR_SSI_SISR_TFRC 0x00800000 +#define CCSR_SSI_SISR_CMDAU 0x00040000 +#define CCSR_SSI_SISR_CMDDU 0x00020000 +#define CCSR_SSI_SISR_RXT 0x00010000 +#define CCSR_SSI_SISR_RDR1 0x00008000 +#define CCSR_SSI_SISR_RDR0 0x00004000 +#define CCSR_SSI_SISR_TDE1 0x00002000 +#define CCSR_SSI_SISR_TDE0 0x00001000 +#define CCSR_SSI_SISR_ROE1 0x00000800 +#define CCSR_SSI_SISR_ROE0 0x00000400 +#define CCSR_SSI_SISR_TUE1 0x00000200 +#define CCSR_SSI_SISR_TUE0 0x00000100 +#define CCSR_SSI_SISR_TFS 0x00000080 +#define CCSR_SSI_SISR_RFS 0x00000040 +#define CCSR_SSI_SISR_TLS 0x00000020 +#define CCSR_SSI_SISR_RLS 0x00000010 +#define CCSR_SSI_SISR_RFF1 0x00000008 +#define CCSR_SSI_SISR_RFF0 0x00000004 +#define CCSR_SSI_SISR_TFE1 0x00000002 +#define CCSR_SSI_SISR_TFE0 0x00000001 + +#define CCSR_SSI_SIER_RFRC_EN 0x01000000 +#define CCSR_SSI_SIER_TFRC_EN 0x00800000 +#define CCSR_SSI_SIER_RDMAE 0x00400000 +#define CCSR_SSI_SIER_RIE 0x00200000 +#define CCSR_SSI_SIER_TDMAE 0x00100000 +#define CCSR_SSI_SIER_TIE 0x00080000 +#define CCSR_SSI_SIER_CMDAU_EN 0x00040000 +#define CCSR_SSI_SIER_CMDDU_EN 0x00020000 +#define CCSR_SSI_SIER_RXT_EN 0x00010000 +#define CCSR_SSI_SIER_RDR1_EN 0x00008000 +#define CCSR_SSI_SIER_RDR0_EN 0x00004000 +#define CCSR_SSI_SIER_TDE1_EN 0x00002000 +#define CCSR_SSI_SIER_TDE0_EN 0x00001000 +#define CCSR_SSI_SIER_ROE1_EN 0x00000800 +#define CCSR_SSI_SIER_ROE0_EN 0x00000400 +#define CCSR_SSI_SIER_TUE1_EN 0x00000200 +#define CCSR_SSI_SIER_TUE0_EN 0x00000100 +#define CCSR_SSI_SIER_TFS_EN 0x00000080 +#define CCSR_SSI_SIER_RFS_EN 0x00000040 +#define CCSR_SSI_SIER_TLS_EN 0x00000020 +#define CCSR_SSI_SIER_RLS_EN 0x00000010 +#define CCSR_SSI_SIER_RFF1_EN 0x00000008 +#define CCSR_SSI_SIER_RFF0_EN 0x00000004 +#define CCSR_SSI_SIER_TFE1_EN 0x00000002 +#define CCSR_SSI_SIER_TFE0_EN 0x00000001 + +#define CCSR_SSI_STCR_TXBIT0 0x00000200 +#define CCSR_SSI_STCR_TFEN1 0x00000100 +#define CCSR_SSI_STCR_TFEN0 0x00000080 +#define CCSR_SSI_STCR_TFDIR 0x00000040 +#define CCSR_SSI_STCR_TXDIR 0x00000020 +#define CCSR_SSI_STCR_TSHFD 0x00000010 +#define CCSR_SSI_STCR_TSCKP 0x00000008 +#define CCSR_SSI_STCR_TFSI 0x00000004 +#define CCSR_SSI_STCR_TFSL 0x00000002 +#define CCSR_SSI_STCR_TEFS 0x00000001 + +#define CCSR_SSI_SRCR_RXEXT 0x00000400 +#define CCSR_SSI_SRCR_RXBIT0 0x00000200 +#define CCSR_SSI_SRCR_RFEN1 0x00000100 +#define CCSR_SSI_SRCR_RFEN0 0x00000080 +#define CCSR_SSI_SRCR_RFDIR 0x00000040 +#define CCSR_SSI_SRCR_RXDIR 0x00000020 +#define CCSR_SSI_SRCR_RSHFD 0x00000010 +#define CCSR_SSI_SRCR_RSCKP 0x00000008 +#define CCSR_SSI_SRCR_RFSI 0x00000004 +#define CCSR_SSI_SRCR_RFSL 0x00000002 +#define CCSR_SSI_SRCR_REFS 0x00000001 + +/* STCCR and SRCCR */ +#define CCSR_SSI_SxCCR_DIV2 0x00040000 +#define CCSR_SSI_SxCCR_PSR 0x00020000 +#define CCSR_SSI_SxCCR_WL_SHIFT 13 +#define CCSR_SSI_SxCCR_WL_MASK 0x0001E000 +#define CCSR_SSI_SxCCR_WL(x) \ + (((((x) / 2) - 1) << CCSR_SSI_SxCCR_WL_SHIFT) & CCSR_SSI_SxCCR_WL_MASK) +#define CCSR_SSI_SxCCR_DC_SHIFT 8 +#define CCSR_SSI_SxCCR_DC_MASK 0x00001F00 +#define CCSR_SSI_SxCCR_DC(x) \ + ((((x) - 1) << CCSR_SSI_SxCCR_DC_SHIFT) & CCSR_SSI_SxCCR_DC_MASK) +#define CCSR_SSI_SxCCR_PM_SHIFT 0 +#define CCSR_SSI_SxCCR_PM_MASK 0x000000FF +#define CCSR_SSI_SxCCR_PM(x) \ + ((((x) - 1) << CCSR_SSI_SxCCR_PM_SHIFT) & CCSR_SSI_SxCCR_PM_MASK) + +/* + * The xFCNT bits are read-only, and the xFWM bits are read/write. Use the + * CCSR_SSI_SFCSR_xFCNTy() macros to read the FIFO counters, and use the + * CCSR_SSI_SFCSR_xFWMy() macros to set the watermarks. + */ +#define CCSR_SSI_SFCSR_RFCNT1_SHIFT 28 +#define CCSR_SSI_SFCSR_RFCNT1_MASK 0xF0000000 +#define CCSR_SSI_SFCSR_RFCNT1(x) \ + (((x) & CCSR_SSI_SFCSR_RFCNT1_MASK) >> CCSR_SSI_SFCSR_RFCNT1_SHIFT) +#define CCSR_SSI_SFCSR_TFCNT1_SHIFT 24 +#define CCSR_SSI_SFCSR_TFCNT1_MASK 0x0F000000 +#define CCSR_SSI_SFCSR_TFCNT1(x) \ + (((x) & CCSR_SSI_SFCSR_TFCNT1_MASK) >> CCSR_SSI_SFCSR_TFCNT1_SHIFT) +#define CCSR_SSI_SFCSR_RFWM1_SHIFT 20 +#define CCSR_SSI_SFCSR_RFWM1_MASK 0x00F00000 +#define CCSR_SSI_SFCSR_RFWM1(x) \ + (((x) << CCSR_SSI_SFCSR_RFWM1_SHIFT) & CCSR_SSI_SFCSR_RFWM1_MASK) +#define CCSR_SSI_SFCSR_TFWM1_SHIFT 16 +#define CCSR_SSI_SFCSR_TFWM1_MASK 0x000F0000 +#define CCSR_SSI_SFCSR_TFWM1(x) \ + (((x) << CCSR_SSI_SFCSR_TFWM1_SHIFT) & CCSR_SSI_SFCSR_TFWM1_MASK) +#define CCSR_SSI_SFCSR_RFCNT0_SHIFT 12 +#define CCSR_SSI_SFCSR_RFCNT0_MASK 0x0000F000 +#define CCSR_SSI_SFCSR_RFCNT0(x) \ + (((x) & CCSR_SSI_SFCSR_RFCNT0_MASK) >> CCSR_SSI_SFCSR_RFCNT0_SHIFT) +#define CCSR_SSI_SFCSR_TFCNT0_SHIFT 8 +#define CCSR_SSI_SFCSR_TFCNT0_MASK 0x00000F00 +#define CCSR_SSI_SFCSR_TFCNT0(x) \ + (((x) & CCSR_SSI_SFCSR_TFCNT0_MASK) >> CCSR_SSI_SFCSR_TFCNT0_SHIFT) +#define CCSR_SSI_SFCSR_RFWM0_SHIFT 4 +#define CCSR_SSI_SFCSR_RFWM0_MASK 0x000000F0 +#define CCSR_SSI_SFCSR_RFWM0(x) \ + (((x) << CCSR_SSI_SFCSR_RFWM0_SHIFT) & CCSR_SSI_SFCSR_RFWM0_MASK) +#define CCSR_SSI_SFCSR_TFWM0_SHIFT 0 +#define CCSR_SSI_SFCSR_TFWM0_MASK 0x0000000F +#define CCSR_SSI_SFCSR_TFWM0(x) \ + (((x) << CCSR_SSI_SFCSR_TFWM0_SHIFT) & CCSR_SSI_SFCSR_TFWM0_MASK) + +#define CCSR_SSI_STR_TEST 0x00008000 +#define CCSR_SSI_STR_RCK2TCK 0x00004000 +#define CCSR_SSI_STR_RFS2TFS 0x00002000 +#define CCSR_SSI_STR_RXSTATE(x) (((x) >> 8) & 0x1F) +#define CCSR_SSI_STR_TXD2RXD 0x00000080 +#define CCSR_SSI_STR_TCK2RCK 0x00000040 +#define CCSR_SSI_STR_TFS2RFS 0x00000020 +#define CCSR_SSI_STR_TXSTATE(x) ((x) & 0x1F) + +#define CCSR_SSI_SOR_CLKOFF 0x00000040 +#define CCSR_SSI_SOR_RX_CLR 0x00000020 +#define CCSR_SSI_SOR_TX_CLR 0x00000010 +#define CCSR_SSI_SOR_INIT 0x00000008 +#define CCSR_SSI_SOR_WAIT_SHIFT 1 +#define CCSR_SSI_SOR_WAIT_MASK 0x00000006 +#define CCSR_SSI_SOR_WAIT(x) (((x) & 3) << CCSR_SSI_SOR_WAIT_SHIFT) +#define CCSR_SSI_SOR_SYNRST 0x00000001 + +/* Instantiation data for an SSI interface + * + * This structure contains all the information that the the SSI driver needs + * to instantiate an SSI interface with ALSA. The machine driver should + * create this structure, fill it in, call fsl_ssi_create_dai(), and then + * delete the structure. + * + * id: which SSI this is (0, 1, etc. ) + * ssi: pointer to the SSI's registers + * ssi_phys: physical address of the SSI registers + * irq: IRQ of this SSI + * dev: struct device, used to create the sysfs statistics file +*/ +struct fsl_ssi_info { + unsigned int id; + struct ccsr_ssi __iomem *ssi; + dma_addr_t ssi_phys; + unsigned int irq; + struct device *dev; +}; + +struct snd_soc_cpu_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info); +void fsl_ssi_destroy_dai(struct snd_soc_cpu_dai *fsl_ssi_dai); + +#endif + diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c new file mode 100644 index 0000000..5421a5c --- /dev/null +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -0,0 +1,621 @@ +/** + * Freescale MPC8610HPCD ALSA SoC Fabric driver + * + * Author: Timur Tabi <timur(a)freescale.com> + * + * Copyright 2007 Freescale Semiconductor, Inc. This file is licensed under + * the terms of the GNU General Public License version 2. This program + * is licensed "as is" without any warranty of any kind, whether express + * or implied. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <asm/of_device.h> +#include <asm/of_platform.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <asm/immap_86xx.h> + +#include "../codecs/cs4270.h" +#include "fsl_dma.h" +#include "fsl_ssi.h" + +/* + * Fabric-specific ASoC device data + * + * This structure contains data for a single sound platform device on an + * MPC8610 HPCD. Some of the data is taken from the device tree. + */ +struct mpc8610hpcd_data { + struct snd_soc_device sound_devdata; + struct snd_soc_dai_link dai; + struct snd_soc_machine machine; + unsigned int dai_format; + unsigned int codec_clk_direction; + unsigned int cpu_clk_direction; + unsigned int clk_frequency; + struct ccsr_guts __iomem *guts; + struct ccsr_ssi __iomem *ssi; + unsigned int ssi_id; /* 0 = SSI1, 1 = SSI2, etc */ + unsigned int ssi_irq; + unsigned int dma_id; /* 0 = DMA1, 1 = DMA2, etc */ + unsigned int dma_irq[2]; + struct ccsr_dma_channel __iomem *dma[2]; + unsigned int dma_channel_id[2]; /* 0 = ch 0, 1 = ch 1, etc*/ +}; + +/* + * Initalize the board + * + * This function is called when platform_device_add() is called. It is used + * to initialize the board-specific hardware. + * + * Here we program the DMACR and PMUXCR registers. + */ +int mpc8610_hpcd_machine_probe(struct platform_device *sound_device) +{ + struct mpc8610hpcd_data *machine_data = sound_device->dev.platform_data; + + /* Program the signal routing between the SSI and the DMA */ + guts_set_dmacr(machine_data->guts, machine_data->dma_id + 1, + machine_data->dma_channel_id[0], CCSR_GUTS_DMACR_DEV_SSI); + guts_set_dmacr(machine_data->guts, machine_data->dma_id + 1, + machine_data->dma_channel_id[1], CCSR_GUTS_DMACR_DEV_SSI); + + guts_set_pmuxcr_dma(machine_data->guts, machine_data->dma_id, + machine_data->dma_channel_id[0], 0); + guts_set_pmuxcr_dma(machine_data->guts, machine_data->dma_id, + machine_data->dma_channel_id[1], 0); + + guts_set_pmuxcr_dma(machine_data->guts, 1, 0, 0); + guts_set_pmuxcr_dma(machine_data->guts, 1, 3, 0); + guts_set_pmuxcr_dma(machine_data->guts, 0, 3, 0); + + switch (machine_data->ssi_id) { + case 0: + clrsetbits_be32(&machine_data->guts->pmuxcr, + CCSR_GUTS_PMUXCR_SSI1_MASK, CCSR_GUTS_PMUXCR_SSI1_SSI); + break; + case 1: + clrsetbits_be32(&machine_data->guts->pmuxcr, + CCSR_GUTS_PMUXCR_SSI2_MASK, CCSR_GUTS_PMUXCR_SSI2_SSI); + break; + } + + return 0; +} + +/* + * Program the board based on various hardware parameters + */ +static int mpc8610hpcd_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct mpc8610hpcd_data *machine_data = rtd->socdev->dev->platform_data; + int ret = 0; + + /* Tell the CPU driver what the serial protocol is. */ + if (cpu_dai->dai_ops.set_fmt) { + ret = cpu_dai->dai_ops.set_fmt(cpu_dai, + machine_data->dai_format); + if (ret < 0) { + dev_err(substream->pcm->card->dev, + "could not set CPU driver audio format\n"); + return ret; + } + } + + /* Tell the codec driver what the serial protocol is. */ + if (codec_dai->dai_ops.set_fmt) { + ret = codec_dai->dai_ops.set_fmt(codec_dai, + machine_data->dai_format); + if (ret < 0) { + dev_err(substream->pcm->card->dev, + "could not set codec driver audio format\n"); + return ret; + } + } + + /* + * Tell the CPU driver what the clock frequency is, and whether it's a + * slave or master. + */ + if (cpu_dai->dai_ops.set_sysclk) { + ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, 0, + machine_data->clk_frequency, + machine_data->cpu_clk_direction); + if (ret < 0) { + dev_err(substream->pcm->card->dev, + "could not set CPU driver clock parameters\n"); + return ret; + } + } + + /* + * Tell the codec driver what the MCLK frequency is, and whether it's + * a slave or master. + */ + if (codec_dai->dai_ops.set_sysclk) { + ret = codec_dai->dai_ops.set_sysclk(codec_dai, 0, + machine_data->clk_frequency, + machine_data->codec_clk_direction); + if (ret < 0) { + dev_err(substream->pcm->card->dev, + "could not set codec driver clock params\n"); + return ret; + } + } + + return 0; +} + +/* + * Remove the sound device + * + * This function is called to remove the sound device for one SSI. We + * de-program the DMACR and PMUXCR register. + */ +int mpc8610_hpcd_machine_remove(struct platform_device *sound_device) +{ + struct mpc8610hpcd_data *machine_data = sound_device->dev.platform_data; + + /* Restore the signal routing */ + + guts_set_dmacr(machine_data->guts, machine_data->dma_id + 1, + machine_data->dma_channel_id[0], 0); + guts_set_dmacr(machine_data->guts, machine_data->dma_id + 1, + machine_data->dma_channel_id[1], 0); + + switch (machine_data->ssi_id) { + case 0: + clrsetbits_be32(&machine_data->guts->pmuxcr, + CCSR_GUTS_PMUXCR_SSI1_MASK, CCSR_GUTS_PMUXCR_SSI1_LA); + break; + case 1: + clrsetbits_be32(&machine_data->guts->pmuxcr, + CCSR_GUTS_PMUXCR_SSI2_MASK, CCSR_GUTS_PMUXCR_SSI1_LA); + break; + } + + return 0; +} + +/* + * ASoC fabric driver operations + */ +static struct snd_soc_ops mpc8610hpcd_ops = { + .startup = mpc8610hpcd_startup, +}; + +/* + * ASoC machine + */ +static struct snd_soc_machine mpc8610_hpcd_machine = { + .probe = mpc8610_hpcd_machine_probe, + .remove = mpc8610_hpcd_machine_remove, + .name = "MPC8610 HPCD", + .num_links = 1, +}; + +/* + * Probe function for the fabric driver driver. + * + * This function gets called when an SSI node is found in the device tree. + * + * Although this is a fabric driver, the SSI node is the "master" node with + * respect to audio hardware connections. Therefore, we create a new ASoC + * device for each new SSI node that has a codec attached. + * + * FIXME: Currently, we only support one DMA controller, so if there are + * multiple SSI nodes with codecs, only the first will be supported. + * + * FIXME: Even if we did support multiple DMA controllers, we have no + * mechanism for assigning DMA controllers and channels to the individual + * SSI devices. We also probably aren't compatible with the generic Elo DMA + * device driver. + */ +static int mpc8610_hpcd_probe(struct of_device *ofdev, + const struct of_device_id *match) +{ + struct device_node *np = ofdev->node; + struct device_node *codec_np = NULL; + struct device_node *guts_np = NULL; + struct device_node *dma_np = NULL; + struct device_node *dma_channel_np = NULL; + const char *sprop; + const u32 *iprop; + struct resource res; + struct platform_device *sound_device = NULL; + struct mpc8610hpcd_data *machine_data; + struct fsl_ssi_info ssi_info; + struct fsl_dma_info dma_info; + int ret = -ENODEV; + + machine_data = kzalloc(sizeof(struct mpc8610hpcd_data), GFP_KERNEL); + if (!machine_data) + return -ENOMEM; + + memset(&ssi_info, 0, sizeof(ssi_info)); + memset(&dma_info, 0, sizeof(dma_info)); + + ssi_info.dev = &ofdev->dev; + + /* + * We are only interested in SSIs with a codec child node in them, so + * let's make sure this SSI has one. + */ + while ((codec_np = of_get_next_child(np, codec_np)) != NULL) { + if (strcmp(codec_np->name, "codec") == 0) { + /* Most drivers forget the final of_node_put() call */ + of_node_put(codec_np); + break; + } + } + + if (!codec_np) + goto error; + + /* The MPC8610 HPCD only knows about the CS4270 codec, so reject + anything else. */ + if (!of_device_is_compatible(codec_np, "cirrus,cs4270")) + goto error; + + /* Get the device ID */ + iprop = of_get_property(np, "cell-index", NULL); + if (!iprop) { + dev_err(&ofdev->dev, "cell-index property not found\n"); + ret = -EINVAL; + goto error; + } + machine_data->ssi_id = *iprop; + ssi_info.id = *iprop; + + /* Get the serial format and clock direction. */ + sprop = of_get_property(np, "fsl,mode", NULL); + if (!sprop) { + dev_err(&ofdev->dev, "fsl,mode property not found\n"); + ret = -EINVAL; + goto error; + } + + if (strcasecmp(sprop, "i2s-slave") == 0) { + machine_data->dai_format = SND_SOC_DAIFMT_I2S; + machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; + machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; + + /* In i2s-slave mode, the codec has its own clock source, so + we need to get the frequency from the device tree and pass + it to the codec driver. */ + iprop = of_get_property(codec_np, "bus-frequency", NULL); + if (!iprop || !*iprop) { + dev_err(&ofdev->dev, "codec bus-frequency property " + "is missing or invalid\n"); + ret = -EINVAL; + goto error; + } + machine_data->clk_frequency = *iprop; + } else if (strcasecmp(sprop, "i2s-master") == 0) { + machine_data->dai_format = SND_SOC_DAIFMT_I2S; + machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; + machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; + } else if (strcasecmp(sprop, "lj-slave") == 0) { + machine_data->dai_format = SND_SOC_DAIFMT_LEFT_J; + machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; + machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; + } else if (strcasecmp(sprop, "lj-master") == 0) { + machine_data->dai_format = SND_SOC_DAIFMT_LEFT_J; + machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; + machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; + } else if (strcasecmp(sprop, "rj-master") == 0) { + machine_data->dai_format = SND_SOC_DAIFMT_RIGHT_J; + machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; + machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; + } else if (strcasecmp(sprop, "rj-master") == 0) { + machine_data->dai_format = SND_SOC_DAIFMT_RIGHT_J; + machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; + machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; + } else if (strcasecmp(sprop, "ac97-slave") == 0) { + machine_data->dai_format = SND_SOC_DAIFMT_AC97; + machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; + machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; + } else if (strcasecmp(sprop, "ac97-master") == 0) { + machine_data->dai_format = SND_SOC_DAIFMT_AC97; + machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; + machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; + } else { + dev_err(&ofdev->dev, + "unrecognized fsl,mode property \"%s\"\n", sprop); + ret = -EINVAL; + goto error; + } + + if (!machine_data->clk_frequency) { + dev_err(&ofdev->dev, "unknown clock frequency\n"); + ret = -EINVAL; + goto error; + } + + /* Read the SSI information from the device tree */ + ret = of_address_to_resource(np, 0, &res); + if (ret) { + dev_err(&ofdev->dev, "could not obtain SSI address\n"); + goto error; + } + if (!res.start) { + dev_err(&ofdev->dev, "invalid SSI address\n"); + goto error; + } + ssi_info.ssi_phys = res.start; + + machine_data->ssi = ioremap(ssi_info.ssi_phys, sizeof(struct ccsr_ssi)); + if (!machine_data->ssi) { + dev_err(&ofdev->dev, "could not map SSI address %x\n", + ssi_info.ssi_phys); + ret = -EINVAL; + goto error; + } + ssi_info.ssi = machine_data->ssi; + + + /* Get the IRQ of the SSI */ + machine_data->ssi_irq = irq_of_parse_and_map(np, 0); + if (!machine_data->ssi_irq) { + dev_err(&ofdev->dev, "could not get SSI IRQ\n"); + ret = -EINVAL; + goto error; + } + ssi_info.irq = machine_data->ssi_irq; + + + /* Map the global utilities registers. */ + guts_np = of_find_compatible_node(NULL, NULL, "fsl,mpc8610-guts"); + if (!guts_np) { + dev_err(&ofdev->dev, "could not obtain address of GUTS\n"); + ret = -EINVAL; + goto error; + } + machine_data->guts = of_iomap(guts_np, 0); + of_node_put(guts_np); + if (!machine_data->guts) { + dev_err(&ofdev->dev, "could not map GUTS\n"); + ret = -EINVAL; + goto error; + } + + /* Find the DMA channels to use. For now, we always use the first DMA + controller. */ + for_each_compatible_node(dma_np, NULL, "fsl,mpc8610-dma") { + iprop = of_get_property(dma_np, "cell-index", NULL); + if (iprop && (*iprop == 0)) { + of_node_put(dma_np); + break; + } + } + if (!dma_np) { + dev_err(&ofdev->dev, "could not find DMA node\n"); + ret = -EINVAL; + goto error; + } + machine_data->dma_id = *iprop; + + /* + * Find the DMA channels to use. For now, we always use DMA channel 0 + * for playback, and DMA channel 1 for capture. + */ + while ((dma_channel_np = of_get_next_child(dma_np, dma_channel_np))) { + iprop = of_get_property(dma_channel_np, "cell-index", NULL); + /* Is it DMA channel 0? */ + if (iprop && (*iprop == 0)) { + /* dma_channel[0] and dma_irq[0] are for playback */ + dma_info.dma_channel[0] = of_iomap(dma_channel_np, 0); + dma_info.dma_irq[0] = + irq_of_parse_and_map(dma_channel_np, 0); + machine_data->dma_channel_id[0] = *iprop; + continue; + } + if (iprop && (*iprop == 1)) { + /* dma_channel[1] and dma_irq[1] are for capture */ + dma_info.dma_channel[1] = of_iomap(dma_channel_np, 0); + dma_info.dma_irq[1] = + irq_of_parse_and_map(dma_channel_np, 0); + machine_data->dma_channel_id[1] = *iprop; + continue; + } + } + if (!dma_info.dma_channel[0] || !dma_info.dma_channel[1] || + !dma_info.dma_irq[0] || !dma_info.dma_irq[1]) { + dev_err(&ofdev->dev, "could not find DMA channels\n"); + ret = -EINVAL; + goto error; + } + + dma_info.ssi_stx_phys = ssi_info.ssi_phys + + offsetof(struct ccsr_ssi, stx0); + dma_info.ssi_srx_phys = ssi_info.ssi_phys + + offsetof(struct ccsr_ssi, srx0); + + /* We have the DMA information, so tell the DMA driver what it is */ + if (!fsl_dma_configure(&dma_info)) { + dev_err(&ofdev->dev, "could not instantiate DMA device\n"); + ret = -EBUSY; + goto error; + } + + /* + * Initialize our DAI data structure. We should probably get this + * information from the device tree. + */ + machine_data->dai.name = "CS4270"; + machine_data->dai.stream_name = "CS4270"; + + machine_data->dai.cpu_dai = fsl_ssi_create_dai(&ssi_info); + machine_data->dai.codec_dai = &cs4270_dai; /* The codec_dai we want */ + machine_data->dai.ops = &mpc8610hpcd_ops; + + mpc8610_hpcd_machine.dai_link = &machine_data->dai; + + /* Allocate a new audio platform device structure */ + sound_device = platform_device_alloc("soc-audio", -1); + if (!sound_device) { + dev_err(&ofdev->dev, "platform device allocation failed\n"); + ret = -ENOMEM; + goto error; + } + + machine_data->sound_devdata.machine = &mpc8610_hpcd_machine; + machine_data->sound_devdata.codec_dev = &soc_codec_device_cs4270; + machine_data->sound_devdata.platform = &fsl_soc_platform; + + sound_device->dev.platform_data = machine_data; + + + /* Set the platform device and ASoC device to point to each other */ + platform_set_drvdata(sound_device, &machine_data->sound_devdata); + + machine_data->sound_devdata.dev = &sound_device->dev; + + + /* Tell ASoC to probe us. This will call mpc8610_hpcd_machine.probe(), + if it exists. */ + ret = platform_device_add(sound_device); + + if (ret) { + dev_err(&ofdev->dev, "platform device add failed\n"); + goto error; + } + + dev_set_drvdata(&ofdev->dev, sound_device); + + return 0; + +error: + if (sound_device) + platform_device_unregister(sound_device); + + if (machine_data->dai.cpu_dai) + fsl_ssi_destroy_dai(machine_data->dai.cpu_dai); + + if (ssi_info.ssi) + iounmap(ssi_info.ssi); + + if (ssi_info.irq) + irq_dispose_mapping(ssi_info.irq); + + if (dma_info.dma_channel[0]) + iounmap(dma_info.dma_channel[0]); + + if (dma_info.dma_channel[1]) + iounmap(dma_info.dma_channel[1]); + + if (dma_info.dma_irq[0]) + irq_dispose_mapping(dma_info.dma_irq[0]); + + if (dma_info.dma_irq[1]) + irq_dispose_mapping(dma_info.dma_irq[1]); + + if (machine_data->guts) + iounmap(machine_data->guts); + + kfree(machine_data); + + return ret; +} + +static int mpc8610_hpcd_remove(struct of_device *ofdev) +{ + struct platform_device *sound_device = dev_get_drvdata(&ofdev->dev); + struct mpc8610hpcd_data *machine_data = sound_device->dev.platform_data; + + platform_device_unregister(sound_device); + + if (machine_data->dai.cpu_dai) + fsl_ssi_destroy_dai(machine_data->dai.cpu_dai); + + if (machine_data->ssi) + iounmap(machine_data->ssi); + + if (machine_data->dma[0]) + iounmap(machine_data->dma[0]); + + if (machine_data->dma[1]) + iounmap(machine_data->dma[1]); + + if (machine_data->dma_irq[0]) + irq_dispose_mapping(machine_data->dma_irq[0]); + + if (machine_data->dma_irq[1]) + irq_dispose_mapping(machine_data->dma_irq[1]); + + if (machine_data->guts) + iounmap(machine_data->guts); + + kfree(machine_data); + sound_device->dev.platform_data = NULL; + + dev_set_drvdata(&ofdev->dev, NULL); + + return 0; +} + +static struct of_device_id mpc8610_hpcd_match[] = { + { + .compatible = "fsl,ssi", + }, + {} +}; +MODULE_DEVICE_TABLE(of, mpc8610_hpcd_match); + +static struct of_platform_driver mpc8610_hpcd_of_driver = { + .owner = THIS_MODULE, + .name = "mpc8610_hpcd", + .match_table = mpc8610_hpcd_match, + .probe = mpc8610_hpcd_probe, + .remove = mpc8610_hpcd_remove, +}; + +/** + * Fabric driver initialization. + * + * This function is called when this module is loaded. + */ +static int __init mpc8610hpcd_init(void) +{ + int ret; + + printk(KERN_INFO "Freescale MPC8610 HPCD ALSA SoC fabric driver\n"); + + ret = of_register_platform_driver(&mpc8610_hpcd_of_driver); + + if (ret) + printk(KERN_ERR + "mpc8610-hpcd: failed to register platform driver\n"); + + return ret; +} + +/** + * Fabric driver exit + * + * This function is called when this driver is unloaded. + */ +static void __exit mpc8610hpcd_exit(void) +{ + of_unregister_platform_driver(&mpc8610_hpcd_of_driver); +} + +module_init(mpc8610hpcd_init); +module_exit(mpc8610hpcd_exit); + +MODULE_AUTHOR("Timur Tabi <timur(a)freescale.com>"); +MODULE_DESCRIPTION("Freescale MPC8610 HPCD ALSA SoC fabric driver"); +MODULE_LICENSE("GPL"); -- 1.5.2.4

12 88

[alsa-devel] [PATCH 0/2] ASoC drivers for the Freescale MPC8610 SoC
by Timur Tabi 10 Jan '08

10 Jan '08

These patches add ALSA SoC device drivers for the Freescale MPC8610 SoC and the MPC8610-HPCD reference board. The first patch updates some files in arch/powerpc and booting-without-of.txt. The second patch adds the new audio driver files to sound/soc. Both must be applied in order to avoid a build break.

4 7

[alsa-devel] [PATCH] hda: 92HD7XXX power management support
by Matthew Ranostay 10 Jan '08

10 Jan '08

Added support for advanced power management support for output ports on 92HD7xxx family of codecs. Inactive output ports are powered down when the pin sense doesn't detect a connection, and powered back up when a connection is sensed. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r 60aa7d37dcd5 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Tue Jan 08 17:19:22 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Wed Jan 09 11:13:21 2008 -0500 @@ -35,7 +35,8 @@ #include "hda_local.h" #define NUM_CONTROL_ALLOC 32 -#define STAC_HP_EVENT 0x37 +#define STAC_PWR_EVENT 0x20 +#define STAC_HP_EVENT 0x30 enum { STAC_REF, @@ -127,6 +128,10 @@ struct sigmatel_spec { unsigned char aloopback_mask; unsigned char aloopback_shift; + /* power management */ + unsigned int num_pwrs; + hda_nid_t *pwr_nids; + /* playback */ struct hda_multi_out multiout; hda_nid_t dac_nids[5]; @@ -185,6 +190,11 @@ static hda_nid_t stac9200_dac_nids[1] = 0x02, }; +static hda_nid_t stac92hd73xx_pwr_nids[8] = { + 0x0a, 0x0b, 0x0c, 0xd, 0x0e, + 0x0f, 0x10, 0x11 +}; + static hda_nid_t stac92hd73xx_adc_nids[2] = { 0x1a, 0x1b }; @@ -205,6 +215,10 @@ static hda_nid_t stac92hd73xx_mux_nids[4 static hda_nid_t stac92hd73xx_dmux_nids[2] = { 0x20, 0x21, +}; + +static hda_nid_t stac92hd71bxx_pwr_nids[3] = { + 0x0a, 0x0d, 0x0f }; static hda_nid_t stac92hd71bxx_adc_nids[2] = { @@ -546,7 +560,7 @@ static struct hda_verb stac92hd71bxx_ana /* connect ports 0d and 0f to audio mixer */ { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x2}, { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2}, - { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Speaker */ /* unmute dac0 input in audio mixer */ { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0x701f}, /* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */ @@ -714,6 +728,8 @@ static struct snd_kcontrol_new stac92hd7 HDA_CODEC_MUTE("Analog Loopback 1", 0x17, 0x3, HDA_INPUT), HDA_CODEC_MUTE("Analog Loopback 2", 0x17, 0x4, HDA_INPUT), + + HDA_CODEC_MUTE_MONO("Mono Switch", 0x14, 0x1, 0, HDA_INPUT), { } /* end */ }; @@ -728,6 +744,8 @@ static struct snd_kcontrol_new stac92hd7 HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1d, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x0, HDA_OUTPUT), + + HDA_CODEC_MUTE_MONO("Mono Switch", 0x14, 0x1, 0, HDA_INPUT), { } /* end */ }; @@ -2651,11 +2669,26 @@ static void enable_pin_detect(struct hda static void enable_pin_detect(struct hda_codec *codec, hda_nid_t nid, unsigned int event) { + int pinctl = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + if (pinctl & AC_PINCTL_IN_EN) + return; + if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_UNSOLICITED_ENABLE, (AC_USRSP_EN | event)); } + +static int is_nid_hp_pin(struct auto_pin_cfg *cfg, hda_nid_t nid) +{ + int i; + for (i = 0; i < cfg->hp_outs; i++) + if (cfg->hp_pins[i] == nid) + return 1; /* nid is a HP-Out */ + + return 0; /* nid is not a HP-Out */ +}; static int stac92xx_init(struct hda_codec *codec) { @@ -2696,6 +2729,14 @@ static int stac92xx_init(struct hda_code for (i = 0; i < spec->num_dmics; i++) stac92xx_auto_set_pinctl(codec, spec->dmic_nids[i], AC_PINCTL_IN_EN); + if (spec->num_pwrs > 0) { + for (i = 0; i < spec->num_pwrs; i++) { + int event = is_nid_hp_pin(cfg, spec->pwr_nids[i]) + ? STAC_HP_EVENT : STAC_PWR_EVENT; + enable_pin_detect(codec, spec->pwr_nids[i], event | i); + codec->patch_ops.unsol_event(codec, (event | i) << 26); + } + } if (cfg->dig_out_pin) stac92xx_auto_set_pinctl(codec, cfg->dig_out_pin, @@ -2822,12 +2863,37 @@ static void stac92xx_hp_detect(struct hd } } +static void stac92xx_pin_sense(struct hda_codec *codec, int idx) +{ + struct sigmatel_spec *spec = codec->spec; + int presence, val; + + val = snd_hda_codec_read(codec, codec->afg, 0, 0x0fec, 0x0) + & 0x000000ff; + presence = get_hp_pin_presence(codec, spec->pwr_nids[idx]); + idx = 1 << idx; + + if (presence) + val &= ~idx; + else + val |= idx; + + /* power down unused output ports */ + snd_hda_codec_write(codec, codec->afg, 0, 0x7ec, val); +}; + static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res) { - switch (res >> 26) { + struct sigmatel_spec *spec = codec->spec; + int idx = res >> 26 & 0x0f; + + switch ((res >> 26) & 0x30) { case STAC_HP_EVENT: stac92xx_hp_detect(codec, res); - break; + /* fallthru */ + case STAC_PWR_EVENT: + if (spec->num_pwrs > 0) + stac92xx_pin_sense(codec, idx); } } @@ -2898,6 +2964,7 @@ static int patch_stac9200(struct hda_cod spec->num_muxes = 1; spec->num_dmics = 0; spec->num_adcs = 1; + spec->num_pwrs = 0; if (spec->board_config == STAC_9200_GATEWAY) spec->init = stac9200_eapd_init; @@ -2953,6 +3020,7 @@ static int patch_stac925x(struct hda_cod spec->mux_nids = stac925x_mux_nids; spec->num_muxes = 1; spec->num_adcs = 1; + spec->num_pwrs = 0; switch (codec->vendor_id) { case 0x83847632: /* STAC9202 */ case 0x83847633: /* STAC9202D */ @@ -3076,6 +3144,9 @@ again: spec->gpio_mask = spec->gpio_data = 0x000001; stac92xx_enable_gpio_mask(codec); + spec->num_pwrs = 8; + spec->pwr_nids = stac92hd73xx_pwr_nids; + err = stac92xx_parse_auto_config(codec, 0x22, 0x24); if (!err) { @@ -3157,6 +3228,9 @@ again: spec->num_adcs = ARRAY_SIZE(stac92hd71bxx_adc_nids); spec->num_dmics = STAC92HD71BXX_NUM_DMICS; spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids); + + spec->num_pwrs = 3; + spec->pwr_nids = stac92hd71bxx_pwr_nids; spec->multiout.num_dacs = 2; spec->multiout.hp_nid = 0x11; @@ -3252,6 +3326,7 @@ static int patch_stac922x(struct hda_cod spec->num_muxes = ARRAY_SIZE(stac922x_mux_nids); spec->num_adcs = ARRAY_SIZE(stac922x_adc_nids); spec->num_dmics = 0; + spec->num_pwrs = 0; spec->init = stac922x_core_init; spec->mixer = stac922x_mixer; @@ -3356,6 +3431,7 @@ static int patch_stac927x(struct hda_cod spec->mixer = stac927x_mixer; } + spec->num_pwrs = 0; spec->aloopback_mask = 0x40; spec->aloopback_shift = 0; @@ -3417,6 +3493,7 @@ static int patch_stac9205(struct hda_cod spec->num_dmics = STAC9205_NUM_DMICS; spec->dmux_nids = stac9205_dmux_nids; spec->num_dmuxes = ARRAY_SIZE(stac9205_dmux_nids); + spec->num_pwrs = 0; spec->init = stac9205_core_init; spec->mixer = stac9205_mixer; @@ -3679,6 +3756,7 @@ static int patch_stac9872(struct hda_cod spec->multiout.hp_nid = VAIO_HP_DAC; spec->num_adcs = ARRAY_SIZE(vaio_adcs); spec->adc_nids = vaio_adcs; + spec->num_pwrs = 0; spec->input_mux = &vaio_mux; spec->mux_nids = vaio_mux_nids; codec->patch_ops = stac9872_vaio_patch_ops; @@ -3692,6 +3770,7 @@ static int patch_stac9872(struct hda_cod spec->multiout.dac_nids = vaio_dacs; spec->multiout.hp_nid = VAIO_HP_DAC; spec->num_adcs = ARRAY_SIZE(vaio_adcs); + spec->num_pwrs = 0; spec->adc_nids = vaio_adcs; spec->input_mux = &vaio_mux; spec->mux_nids = vaio_mux_nids;

2 5

[alsa-devel] [PATCH] Device ID for Macbook sound card
by Jiang zhe 10 Jan '08

10 Jan '08

Hi, Please refer to the [0003680] on bugtracking system. The user found that "model=mbp3" works and provided the ID. --- alsa-driver-hg20080110/alsa-kernel/pci/hda/patch_realtek.c 2008-01-09 09:00:15.000000000 +0800 +++ b/alsa-kernel/pci/hda/patch_realtek.c 2008-01-10 16:07:20.000000000 +0800 @@ -6122,6 +6122,7 @@ static int patch_alc882(struct hda_codec case 0x106b1000: /* iMac 24 */ board_config = ALC885_IMAC24; break; + case 0x106b00a1: /* Macbook */ case 0x106b2c00: /* Macbook Pro rev3 */ board_config = ALC885_MBP3; break;

2 1

[alsa-devel] [PATCH] hda: STAC9228 VT fixes
by Matthew Ranostay 10 Jan '08

10 Jan '08

Moved 2 systems PCI_QUIRK values to STAC_DELL_BIOS. Also the second front HP jack is incorrect defined in the BIOS VT's for some laptops, this patch corrects this. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r d09f88bf00e8 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Wed Jan 09 12:30:20 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Wed Jan 09 12:34:59 2008 -0500 @@ -1428,13 +1428,13 @@ static struct snd_pci_quirk stac927x_cfg SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2002, "Intel D965", STAC_D965_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2001, "Intel D965", STAC_D965_3ST), /* Dell 3 stack systems */ - SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f3, "Dell Inspiron 1420", STAC_DELL_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f7, "Dell XPS M1730", STAC_DELL_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01dd, "Dell Dimension E520", STAC_DELL_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01ed, "Dell ", STAC_DELL_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f4, "Dell ", STAC_DELL_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0227, "Dell Vostro 1400 ", STAC_DELL_3ST), /* Dell 3 stack systems with verb table in BIOS */ + 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, 0x022f, "Dell ", STAC_DELL_BIOS), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x022e, "Dell ", STAC_DELL_BIOS), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0242, "Dell ", STAC_DELL_BIOS), @@ -3332,6 +3332,8 @@ static int patch_stac927x(struct hda_cod spec->mixer = stac927x_mixer; break; case STAC_DELL_BIOS: + /* correct the front output jack as a hp out */ + stac92xx_set_config_reg(codec, 0x0f, 0x02270110); /* correct the front input jack as a mic */ stac92xx_set_config_reg(codec, 0x0e, 0x02a79130); /* fallthru */

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[alsa-devel] can not access the alsa driver
by Ning Zhang 10 Jan '08

10 Jan '08

Dear all, I wrote an alsa device driver for my arm_based platform. Then I cross compiled the alsa-lib-1.0.15, and build an application (static) with the lib. I build the driver in the kernel, so after the kernel booted, I can get these information: # cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.14rc1 (Tue Jan 09 09:56:17 2007 UTC). cat /proc/asound/devices 0: [ 0] : control 16: [ 0- 0]: digital audio playback 24: [ 0- 0]: digital audio capture 33: : timer # cat /proc/asound/cards 0 [XYZ ]: XYZ - ARM XYZ at 0xc485e000 0xc4870000 irq 57 ( I did not found /dev/snd or /dev/dsp ) But when I run the application, I got these errors: # /tmp/pcm_play_arm_s default ALSA lib pcm.c:2145:(snd_pcm_open_noupdate) Unknown PCM default Playback open error: No such file or directory I googled the Internet, and add two files manually : # cat /etc/alsa/alsa.conf pcm.default{ type hw card 0 } # cat ~/.asoundrc pcm.default { type hw card 0 } But I still can not let the application to access the pcm device, this time I got the following information: # /tmp/pcm_play_arm_s default cannot open audio device default (No such file or directory) Here are my questions: 1, When we install the alsa-lib, we can get some libs, besides those, do we also get some config files according to the local linux environment and alsa-lib will use these config files to access the real driver? If that, how can I install the alsa-lib to a embedded platform directly or what config files should I port to embedded platform? 2, Must I edit ~/.asoundrc and /etc/alsa/alsa.conf ? In my situation, what should I do ? Is somebody so kind can give me a simple template? Because I am a newbie in alsa world, I must make some mistakes, so I am lookinf forward to your advice. Thanks in advance for your help. David.

4 4

[alsa-devel] [w/ formating this time] Any alsa cards with PLD/FPGA? (similar to Gadget Labs)
by mazarick＠bellsouth.net 09 Jan '08

09 Jan '08

[that's what I get for sending mail from a lame webmail browser] First, I wanted to report that there is good progress being made on developing an alsa linux driver for the Gadget Labs card. It's not 'done yet' but there is steady progress being made. As was mentioned earlier, any interested developers, testers, or other parties are welcomed to join in the discussions or coding, however the mailing list will go 'private' in the next week (to eliminate the fear of spam and email address havesting). The archive is busted right now; not sure if/when it will be fixed. We're using the mailing list for any interested parties, and are making project members of people who would like to contribute to the effort. http://sourceforge.net/projects/wavepro-driver/ Since the small team has a variety of experience and familiarity with writing drivers, an experimental 'branch' of the SVN repo has been set up to place reference material (like existing, known good alsa drivers, known good windows code, etc) and eventually, the skeleton of what is to be done next. The 'good stuff' lives in the trunk and there will be a merge at an appropriate time in the future (when 'soups done'). The heart of the Gadget Labs card is an Altera PLD, and it has a bus interface chip PLX9052 (and some on-board memory, DAC/ADC, etc). We are planning on putting the alsa code for the Delta 1010 in the branch because it is the most similar from a user perspective. However, it uses a vlsi ICE chip to do the job, and it doesn't have to have the PLD/FPGA code downloaded each time it is initialized. If there is a known good driver that can/should have it's code used as a reference, please let us know via this list. A good technical reference to us would be cards that have to download PLD/FPGA as part of initialization and/or cards that use the PLX9052. There are a couple of wireless modem cards, etc with linux drivers that fit the description, but it is unknown if there is already an alsa driver for an existing card that would be good reference code. As you are aware, this effort is not what repos are normally used for, but it was felt that having this would allow people at a variety of different skill levels to quickly pick up on what is trying to be accomplished without having to hunt all over the net, read a bunch of different emails, etc in order to make a contribution. -Mike Mazarick

1 0

[alsa-devel] Any alsa cards with PLD/FPGA? (similar to Gadget Labs)
by mazarick＠bellsouth.net 09 Jan '08

09 Jan '08

First, I wanted to report that there is good progress being made on developing an alsa linux driver for the Gadget Labs card. It's not 'done yet' but there is steady progress being made. As was mentioned earlier, any interested developers, testers, or other parties are welcomed to join in the discussions or coding, however the mailing list will go 'private' in the next week (to eliminate the fear of spam and email address havesting). The archive is busted right now; not sure if/when it will be fixed. We're using the mailing list for any interested parties, and are making project members of people who would like to contribute to the effort. http://sourceforge.net/projects/wavepro-driver/ Since the small team has a variety of experience and familiarity with writing drivers, an experimental 'branch' of the SVN repo has been set up to place reference material (like existing, known good alsa drivers, known good windows code, etc) and eventually, the skeleton of what is to be done next. The 'good stuff' lives in the trunk and there will be a merge at an appropriate time in the future (when 'soups done'). The heart of the Gadget Labs card is an Altera PLD, and it has a bus interface chip PLX9052 (and some on-board memory, DAC/ADC, etc). We are planning on putting the alsa code for the Delta 1010 in the branch because it is the most similar from a user perspective. However, it uses a vlsi ICE chip to do the job, and it doesn't have to have the PLD/FPGA code downloaded each time it is initialized. If there is a known good driver that can/should have it's code used as a reference, please let us know via this list. A good technical reference to us would be cards that have to download PLD/FPGA as part of initialization and/or cards that use the PLX9052. There are a couple of wireless modem cards, etc with linux drivers that fit the description, but it is unknown if there is already an alsa driver for an existing card that would be good reference code. As you are aware, this effort is not what repos are normally used for, but it was felt that having this would allow people at a variety of different skill levels to quickly pick up on what is trying to be accomplished without having to hunt all over the net, read a bunch of different emails, etc in order to make a contribution. -Mike Mazarick

1 0

Re: [alsa-devel] snd-hda-intel support for SPDIF-in wiht vt1708 card?
by Takashi Iwai 09 Jan '08

09 Jan '08

At Wed, 09 Jan 2008 02:06:53 -0800, John Stile wrote: > > On Wed, 2008-01-09 at 08:39 +0100, Takashi Iwai wrote: > > Thanks. > You are welcome. Thank you for your patience. > > > You can put alsa-driver and alsa-kernel in the same directory level. > > Make in alsa-driver will create a symlink automatically from > > ../alsa-kernel. > My procedure is changed: > cd /usr/src/ > rm -rf alsa-lib > rm -rf alsa-utils > rm -rf alsa-driver > rm -rf alsa-driver > hg clone http://hg-mirror.alsa-project.org/alsa-lib alsa-lib > hg clone http://hg-mirror.alsa-project.org/alsa-utils alsa-utils > hg clone http://hg-mirror.alsa-project.org/alsa-kernel alsa-kernel > hg clone http://hg-mirror.alsa-project.org/alsa-driver alsa-driver > pushd alsa-lib > ./hgcompile && make install To be sure: you are using 32bit version, right? For 64bit, the library path might be /usr/lib64, instead of /usr/lib (depending on distro). > popd > pushd alsa-utils > ./hgcompile && make install > popd > cd alsa-driver > ./hgcompile && make install > popd > # Build completed successfully. Yay. > /usr/sbin/alsaconf > # Chose 'hda-intel VIA Technologies, Inc. VIA High Definition Audio Controller" > /usr/bin/alsamixer -Vall So far so good. > > IEC958 shows > ┌──┐ > │MM│ > L└──┘R > CAPTUR > > There are 2 'CAPTURE' columns where I maxed levels. > 'Input So' is set to 'Stereo M' Attach the generated file via "alsactl -f somefile store" instead of ascii art. This contains the all mixer information. > Play tests all work > songs="/usr/share/sounds/alsa/Front_Right.wav > /usr/share/sounds/alsa/Rear_Right.wav > /usr/share/sounds/alsa/Rear_Left.wav > /usr/share/sounds/alsa/Rear_Center.wav > /usr/share/sounds/alsa/Front_Left.wav > /usr/share/sounds/alsa/Noise.wav > /usr/share/sounds/alsa/Side_Right.wav > /usr/share/sounds/alsa/Side_Left.wav > /usr/share/sounds/alsa/Front_Center.wav" > echo "not setting hardware" > for song in $songs; > do > aplay $song > done > > arecord -l > **** List of CAPTURE Hardware Devices **** > card 0: VT82xx [HDA VIA VT82xx], device 0: VT1708 Analog [VT1708 Analog] > Subdevices: 2/2 > Subdevice #0: subdevice #0 > Subdevice #1: subdevice #1 > card 0: VT82xx [HDA VIA VT82xx], device 1: VT1708 Digital [VT1708 > Digital] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > > One digital device is listed with: arecord -L > iec958:CARD=VT82xx,DEV=0 > HDA VIA VT82xx, VT1708 Digital > IEC958 (S/PDIF) Digital Audio Output > null > Discard all samples (playback) or generate zero samples (capture) > > Arecord without specifying -D makes a pulsing screech. > arecord -d 10 -f dat -t wav |aplay > > This records 10 seconds and ends in an I/O error: > arecord -D hw:0,1 -d 10 -f dat -t wav |aplay > > Recording WAVE 'stdin' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo > Playing WAVE 'stdin' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo > arecord: pcm_read:1324: read error: Input/output error > > All these record tests fail. > arecord -D cards.pcm.Digital -d 10 -f dat -t wav |aplay > ALSA lib pcm.c:2121:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.Digital > arecord: main:540: audio open error: No such file or directory > aplay: playback:1995: read error > > arecord -D cards.pcm.iec958 -d 10 -f dat -t wav |aplay > > ALSA lib dlmisc.c:118:(snd_dlsym_verify) unable to verify version for symbol _snd_pcm_hook_ctl_elems_install > ALSA lib pcm_hooks.c:392:(snd_pcm_hook_add_conf) symbol _snd_pcm_hook_ctl_elems_install is not defined inside (null) > arecord: main:540: audio open error: No such device or address > aplay: playback:1995: read error > > arecord -D hwplug:0,1 -d 10 -f dat -t wav |aplay > > ALSA lib pcm.c:2121:(snd_pcm_open_noupdate) Unknown PCM hwplug:0,1 > arecord: main:540: audio open error: No such file or directory > aplay: playback:1995: read error It's a user error :) The usage below, -D plug:spdif, is correct, though. > arecord -D plug:spdif -d 10 -f dat -t wav |aplay > > ALSA lib dlmisc.c:118:(snd_dlsym_verify) unable to verify version for symbol _snd_pcm_hook_ctl_elems_install OK, this is a bug in the recent alsa-lib. I fixed it now on HG tree. Please update your alsa-lib repo (maybe better from hg.alsa-project.org). Takashi

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[alsa-devel] snd_pcm_htimestamp & more PCM mmap timestamping in driver
by Jaroslav Kysela 09 Jan '08

09 Jan '08

Hi all, I just added snd_pcm_htimestamp() function to alsa-lib: int snd_pcm_htimestamp(snd_pcm_t *pcm, snd_pcm_uframes_t *avail, snd_htimestamp_t *tstamp); If you have any complaints, please, do now. Also, I would like to change SNDRV_PCM_TSTAMP_MMAP semantics to SNDRV_PCM_TSTAMP_ENABLE - when hw_ptr is updated in the driver, a timestamp would be always grabbed to ensure that we have always a valid timestamp & position pair. In current implementation, if an application updates hw_ptr in driver (using for example PCM delay ioctl), timestamp grabbed in the interrupt handler won't be associated with any real ring buffer position which makes things complicated and not too much abstract. Opinions? Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

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[alsa-devel] [PATCH] hda: 92HD71BXX Mono Mute Support
by Matthew Ranostay 09 Jan '08

09 Jan '08

Added a mono output mute mixer for the 92hd71bxx family of codecs, this also removes the need for the mono out node to explicitly unmuted in the core init. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r b3915b3a8a22 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Tue Jan 08 12:33:19 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Tue Jan 08 17:58:12 2008 -0500 @@ -534,8 +534,6 @@ static struct hda_verb stac92hd71bxx_cor { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - /* unmute mono out node */ - { 0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, }; static struct hda_verb stac92hd71bxx_analog_core_init[] = { @@ -553,8 +551,6 @@ static struct hda_verb stac92hd71bxx_ana { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - /* unmute mono out node */ - { 0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {} }; @@ -714,6 +710,8 @@ static struct snd_kcontrol_new stac92hd7 HDA_CODEC_MUTE("Analog Loopback 1", 0x17, 0x3, HDA_INPUT), HDA_CODEC_MUTE("Analog Loopback 2", 0x17, 0x4, HDA_INPUT), + + HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x14, 0x1, 0, HDA_INPUT), { } /* end */ }; @@ -728,6 +726,8 @@ static struct snd_kcontrol_new stac92hd7 HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1d, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x0, HDA_OUTPUT), + + HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x14, 0x1, 0, HDA_INPUT), { } /* end */ };

2 1

[alsa-devel] No sound over HDMI on HDA-Intel - HDA ATI HDMI
by Georgi Chorbadzhiyski 09 Jan '08

09 Jan '08

Hi, I have an Asus MB with integrated ATI X1200 audio controller connected over HDMI to an LCD TV. intel-hda driver correctly identifies and drives the card, but I get no sound over HDMI only over the standard stereo output jack. I've unmuted iec958 control on the HDMI card. I'm testing the card like this: root@browser:~# aplay -D plughw:1,0 test.wav Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 16000 Hz, Stereo No sound. root@browser:~# aplay -D plughw:0,0 test.wav Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 16000 Hz, Stereo Sound is ok. Is there some patches that I could try to get the HDMI output working? Similar reports: http://www.mail-archive.com/ubuntu-bugs@lists.ubuntu.com/msg590477.html http://www.redhat.com/archives/rhl-list/2007-November/msg00372.html aplay -l **** List of PLAYBACK Hardware Devices **** card 0: SB [HDA ATI SB], device 0: HDA Generic [HDA Generic] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: HDMI [HDA ATI HDMI], device 0: ATI HDMI [ATI HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 /proc/asound/cards 0 [SB ]: HDA-Intel - HDA ATI SB HDA ATI SB at 0xfe8f4000 irq 16 1 [HDMI ]: HDA-Intel - HDA ATI HDMI HDA ATI HDMI at 0xfeae8000 irq 19 lspci 00:00.0 Host bridge: ATI Technologies Inc RS690 Host Bridge 00:01.0 PCI bridge: ATI Technologies Inc RS690 PCI to PCI Bridge (Internal gfx) 00:06.0 PCI bridge: ATI Technologies Inc RS690 PCI to PCI Bridge (PCI Express Port 2) 00:07.0 PCI bridge: ATI Technologies Inc RS690 PCI to PCI Bridge (PCI Express Port 3) 00:12.0 SATA controller: ATI Technologies Inc SB600 Non-Raid-5 SATA 00:13.0 USB Controller: ATI Technologies Inc SB600 USB (OHCI0) 00:13.1 USB Controller: ATI Technologies Inc SB600 USB (OHCI1) 00:13.2 USB Controller: ATI Technologies Inc SB600 USB (OHCI2) 00:13.3 USB Controller: ATI Technologies Inc SB600 USB (OHCI3) 00:13.4 USB Controller: ATI Technologies Inc SB600 USB (OHCI4) 00:13.5 USB Controller: ATI Technologies Inc SB600 USB Controller (EHCI) 00:14.0 SMBus: ATI Technologies Inc SBx00 SMBus Controller (rev 14) 00:14.1 IDE interface: ATI Technologies Inc SB600 IDE 00:14.2 Audio device: ATI Technologies Inc SBx00 Azalia 00:14.3 ISA bridge: ATI Technologies Inc SB600 PCI to LPC Bridge 00:14.4 PCI bridge: ATI Technologies Inc SBx00 PCI to PCI Bridge 00:18.0 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] HyperTransport Technology Configuration 00:18.1 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Address Map 00:18.2 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] DRAM Controller 00:18.3 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Miscellaneous Control 01:05.0 VGA compatible controller: ATI Technologies Inc RS690 [Radeon X1200 Series] 01:05.2 Audio device: ATI Technologies Inc Radeon X1200 Series Audio Controller 03:00.0 Ethernet controller: Attansic Technology Corp. L1 Gigabit Ethernet Adapter (rev b0) lspci -vv 01:05.2 Audio device: ATI Technologies Inc Radeon X1200 Series Audio Controller Subsystem: ASUSTeK Computer Inc. Unknown device 8287 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 64, Cache Line Size: 64 bytes Interrupt: pin B routed to IRQ 19 Region 0: Memory at feae8000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Address: 0000000000000000 Data: 0000 dmesg: Linux version 2.6.24-rc6 (gf@gf) (gcc version 4.1.2) #32 SMP PREEMPT Tue Jan 8 19:08:44 EET 2008 BIOS-provided physical RAM map: BIOS-e820: 0000000000000000 - 000000000009fc00 (usable) BIOS-e820: 000000000009fc00 - 00000000000a0000 (reserved) BIOS-e820: 00000000000e4000 - 0000000000100000 (reserved) BIOS-e820: 0000000000100000 - 0000000037fb0000 (usable) BIOS-e820: 0000000037fb0000 - 0000000037fbe000 (ACPI data) BIOS-e820: 0000000037fbe000 - 0000000037fe0000 (ACPI NVS) BIOS-e820: 0000000037fe0000 - 0000000038000000 (reserved) BIOS-e820: 00000000fff00000 - 0000000100000000 (reserved) 895MB LOWMEM available. found SMP MP-table at 000ff780 Entering add_active_range(0, 0, 229296) 0 entries of 256 used Zone PFN ranges: DMA 0 -> 4096 Normal 4096 -> 229296 Movable zone start PFN for each node early_node_map[1] active PFN ranges 0: 0 -> 229296 On node 0 totalpages: 229296 DMA zone: 32 pages used for memmap DMA zone: 0 pages reserved DMA zone: 4064 pages, LIFO batch:0 Normal zone: 1759 pages used for memmap Normal zone: 223441 pages, LIFO batch:31 Movable zone: 0 pages used for memmap DMI present. ATI board detected. Disabling timer routing over 8254. Intel MultiProcessor Specification v1.4 Virtual Wire compatibility mode. OEM ID: ASUS Product ID: APIC at: 0xFEE00000 Processor #0 15:15 APIC version 16 I/O APIC #1 Version 33 at 0xFEC00000. Enabling APIC mode: Flat. Using 1 I/O APICs Processors: 1 Allocating PCI resources starting at 40000000 (gap: 38000000:c7f00000) Built 1 zonelists in Zone order, mobility grouping on. Total pages: 227505 Kernel command line: auto BOOT_IMAGE=Linux ro root=801 quiet mapped APIC to ffffb000 (fee00000) mapped IOAPIC to ffffa000 (fec00000) Enabling fast FPU save and restore... done. Enabling unmasked SIMD FPU exception support... done. Initializing CPU#0 PID hash table entries: 4096 (order: 12, 16384 bytes) Detected 2009.066 MHz processor. Console: colour dummy device 80x25 console [tty0] enabled Dentry cache hash table entries: 131072 (order: 7, 524288 bytes) Inode-cache hash table entries: 65536 (order: 6, 262144 bytes) Memory: 906216k/917184k available (1460k kernel code, 10468k reserved, 560k data, 172k init, 0k highmem) virtual kernel memory layout: fixmap : 0xfffba000 - 0xfffff000 ( 276 kB) vmalloc : 0xf8800000 - 0xfffb8000 ( 119 MB) lowmem : 0xc0000000 - 0xf7fb0000 ( 895 MB) .init : 0xc02ff000 - 0xc032a000 ( 172 kB) .data : 0xc026d14d - 0xc02f9154 ( 560 kB) .text : 0xc0100000 - 0xc026d14d (1460 kB) Checking if this processor honours the WP bit even in supervisor mode... Ok. SLUB: Genslabs=11, HWalign=64, Order=0-1, MinObjects=4, CPUs=1, Nodes=1 Calibrating delay using timer specific routine.. 4109.60 BogoMIPS (lpj=2054803) Mount-cache hash table entries: 512 CPU: After generic identify, caps: 078bfbff ebd3fbff 00000000 00000000 00002001 00000000 0000001d 00000000 CPU: L1 I Cache: 64K (64 bytes/line), D cache 64K (64 bytes/line) CPU: L2 Cache: 512K (64 bytes/line) CPU: After all inits, caps: 078bfbff ebd3fbff 00000000 00000410 00002001 00000000 0000001d 00000000 Compat vDSO mapped to ffffe000. Checking 'hlt' instruction... OK. SMP alternatives: switching to UP code Freeing SMP alternatives: 8k freed CPU0: AMD Athlon(tm) 64 Processor 3200+ stepping 02 Total of 1 processors activated (4109.60 BogoMIPS). ExtINT not setup in hardware but reported by MP table ENABLING IO-APIC IRQs ..TIMER: vector=0x31 apic1=0 pin1=2 apic2=0 pin2=0 Brought up 1 CPUs net_namespace: 64 bytes NET: Registered protocol family 16 PCI: PCI BIOS revision 3.00 entry at 0xf0031, last bus=4 Setting up standard PCI resources SCSI subsystem initialized libata version 3.00 loaded. usbcore: registered new interface driver usbfs usbcore: registered new interface driver hub usbcore: registered new device driver usb PCI: Probing PCI hardware PCI: Probing PCI hardware (bus 00) PCI: Transparent bridge - 0000:00:14.4 PCI: Using IRQ router default [1002/438d] at 0000:00:14.3 PCI->APIC IRQ transform: 0000:00:12.0[A] -> IRQ 22 PCI->APIC IRQ transform: 0000:00:13.0[A] -> IRQ 16 PCI->APIC IRQ transform: 0000:00:13.1[B] -> IRQ 17 PCI->APIC IRQ transform: 0000:00:13.2[C] -> IRQ 18 PCI->APIC IRQ transform: 0000:00:13.3[B] -> IRQ 17 PCI->APIC IRQ transform: 0000:00:13.4[C] -> IRQ 18 PCI->APIC IRQ transform: 0000:00:13.5[D] -> IRQ 19 PCI->APIC IRQ transform: 0000:00:14.1[A] -> IRQ 16 PCI->APIC IRQ transform: 0000:00:14.2[A] -> IRQ 16 PCI->APIC IRQ transform: 0000:01:05.0[A] -> IRQ 18 PCI->APIC IRQ transform: 0000:01:05.2[B] -> IRQ 19 PCI->APIC IRQ transform: 0000:03:00.0[A] -> IRQ 19 PCI: Bridge: 0000:00:01.0 IO window: e000-efff MEM window: fe900000-feafffff PREFETCH window: f0000000-f7ffffff PCI: Bridge: 0000:00:06.0 IO window: disabled. MEM window: disabled. PREFETCH window: disabled. PCI: Bridge: 0000:00:07.0 IO window: disabled. MEM window: feb00000-febfffff PREFETCH window: disabled. PCI: Bridge: 0000:00:14.4 IO window: disabled. MEM window: disabled. PREFETCH window: disabled. PCI: Setting latency timer of device 0000:00:06.0 to 64 PCI: Setting latency timer of device 0000:00:07.0 to 64 NET: Registered protocol family 2 Time: tsc clocksource has been installed. IP route cache hash table entries: 32768 (order: 5, 131072 bytes) TCP established hash table entries: 131072 (order: 8, 1048576 bytes) TCP bind hash table entries: 65536 (order: 7, 786432 bytes) TCP: Hash tables configured (established 131072 bind 65536) TCP reno registered io scheduler noop registered (default) Boot video device is 0000:01:05.0 vesafb: framebuffer at 0xf0000000, mapped to 0xf8800000, using 2000k, total 16384k vesafb: mode is 640x400x32, linelength=2560, pages=15 vesafb: protected mode interface info at c000:a034 vesafb: pmi: set display start = c00ca0be, set palette = c00ca17c vesafb: scrolling: redraw vesafb: Truecolor: size=0:8:8:8, shift=0:16:8:0 Console: switching to colour frame buffer device 80x25 fb0: VESA VGA frame buffer device Real Time Clock Driver v1.12ac Linux agpgart interface v0.102 Serial: 8250/16550 driver $Revision: 1.90 $ 4 ports, IRQ sharing disabled PCI: Setting latency timer of device 0000:03:00.0 to 64 atl1 0000:03:00.0: version 2.0.7 ahci 0000:00:12.0: version 3.0 ahci 0000:00:12.0: controller can't do 64bit DMA, forcing 32bit ahci 0000:00:12.0: controller can't do PMP, turning off CAP_PMP ahci 0000:00:12.0: AHCI 0001.0100 32 slots 4 ports 3 Gbps 0xf impl SATA mode ahci 0000:00:12.0: flags: ncq sntf ilck pm led clo pio slum part scsi0 : ahci scsi1 : ahci scsi2 : ahci scsi3 : ahci ata1: SATA max UDMA/133 abar m1024@0xfe8ff800 port 0xfe8ff900 irq 22 ata2: SATA max UDMA/133 abar m1024@0xfe8ff800 port 0xfe8ff980 irq 22 ata3: SATA max UDMA/133 abar m1024@0xfe8ff800 port 0xfe8ffa00 irq 22 ata4: SATA max UDMA/133 abar m1024@0xfe8ff800 port 0xfe8ffa80 irq 22 ata1: SATA link down (SStatus 0 SControl 300) ata2: SATA link up 1.5 Gbps (SStatus 113 SControl 300) ata2.00: ATA-7: ST380815AS, 3.AAD, max UDMA/133 ata2.00: 156301488 sectors, multi 16: LBA48 NCQ (depth 31/32) ata2.00: configured for UDMA/133 ata3: SATA link down (SStatus 0 SControl 300) ata4: SATA link down (SStatus 0 SControl 300) scsi 1:0:0:0: Direct-Access ATA ST380815AS 3.AA PQ: 0 ANSI: 5 sd 1:0:0:0: [sda] 156301488 512-byte hardware sectors (80026 MB) 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 sd 1:0:0:0: [sda] 156301488 512-byte hardware sectors (80026 MB) 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: sda1 sd 1:0:0:0: [sda] Attached SCSI disk ehci_hcd 0000:00:13.5: EHCI Host Controller ehci_hcd 0000:00:13.5: new USB bus registered, assigned bus number 1 ehci_hcd 0000:00:13.5: debug port 1 ehci_hcd 0000:00:13.5: irq 19, io mem 0xfe8ff000 ehci_hcd 0000:00:13.5: USB 2.0 started, EHCI 1.00, driver 10 Dec 2004 usb usb1: configuration #1 chosen from 1 choice hub 1-0:1.0: USB hub found hub 1-0:1.0: 10 ports detected ohci_hcd: 2006 August 04 USB 1.1 'Open' Host Controller (OHCI) Driver ohci_hcd 0000:00:13.0: OHCI Host Controller ohci_hcd 0000:00:13.0: new USB bus registered, assigned bus number 2 ohci_hcd 0000:00:13.0: irq 16, io mem 0xfe8fe000 usb usb2: configuration #1 chosen from 1 choice hub 2-0:1.0: USB hub found hub 2-0:1.0: 2 ports detected ohci_hcd 0000:00:13.1: OHCI Host Controller ohci_hcd 0000:00:13.1: new USB bus registered, assigned bus number 3 ohci_hcd 0000:00:13.1: irq 17, io mem 0xfe8fd000 usb usb3: configuration #1 chosen from 1 choice hub 3-0:1.0: USB hub found hub 3-0:1.0: 2 ports detected ohci_hcd 0000:00:13.2: OHCI Host Controller ohci_hcd 0000:00:13.2: new USB bus registered, assigned bus number 4 ohci_hcd 0000:00:13.2: irq 18, io mem 0xfe8fc000 usb usb4: configuration #1 chosen from 1 choice hub 4-0:1.0: USB hub found hub 4-0:1.0: 2 ports detected ohci_hcd 0000:00:13.3: OHCI Host Controller ohci_hcd 0000:00:13.3: new USB bus registered, assigned bus number 5 ohci_hcd 0000:00:13.3: irq 17, io mem 0xfe8fb000 usb usb5: configuration #1 chosen from 1 choice hub 5-0:1.0: USB hub found hub 5-0:1.0: 2 ports detected usb 3-2: new low speed USB device using ohci_hcd and address 2 ohci_hcd 0000:00:13.4: OHCI Host Controller ohci_hcd 0000:00:13.4: new USB bus registered, assigned bus number 6 ohci_hcd 0000:00:13.4: irq 18, io mem 0xfe8fa000 usb 3-2: configuration #1 chosen from 1 choice usb usb6: configuration #1 chosen from 1 choice hub 6-0:1.0: USB hub found hub 6-0:1.0: 2 ports detected USB Universal Host Controller Interface driver v3.0 serio: i8042 KBD port at 0x60,0x64 irq 1 serio: i8042 AUX port at 0x60,0x64 irq 12 mice: PS/2 mouse device common for all mice input: Logitech USB-PS/2 Optical Mouse as /devices/pci0000:00/0000:00:13.1/usb3/3-2/3-2:1.0/input/input0 input: USB HID v1.10 Mouse [Logitech USB-PS/2 Optical Mouse] on usb-0000:00:13.1-2 usbcore: registered new interface driver usbhid drivers/hid/usbhid/hid-core.c: v2.6:USB HID core driver Advanced Linux Sound Architecture Driver Version 1.0.15 (Tue Nov 20 19:16:42 2007 UTC). input: AT Translated Set 2 keyboard as /devices/platform/i8042/serio0/input/input1 ALSA device list: #0: HDA ATI SB at 0xfe8f4000 irq 16 #1: HDA ATI HDMI at 0xfeae8000 irq 19 TCP cubic registered NET: Registered protocol family 1 NET: Registered protocol family 17 Using IPI Shortcut mode VFS: Mounted root (ext2 filesystem) readonly. Freeing unused kernel memory: 172k freed atl1 0000:03:00.0: eth0 link is up 100 Mbps full duplex -- Georgi Chorbadzhiyski http://georgi.unixsol.org/ - To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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Re: [alsa-devel] snd-hda-intel support for SPDIF-in wiht vt1708 card?
by forumjunk＠comcast.net 09 Jan '08

09 Jan '08

Ok, here's the current state of affairs... both the compilation and driver seem to work well now, though I haven't tested SPDIF yet. I don't know why both John Stile and I were getting ioctl errors, perhaps if it is still happening to him he can elucidate, but it isn't happening to me any more with the latest build. I obtained and built a clean mercurial clone this morning (07 Jan 08): %> cd /usr/src %> rm -rf alsa-driver %> hg clone http://hg-mirror.alsa-project.org/alsa-driver alsa-driver %> cd alsa-driver %> hg clone http://hg-mirror.alsa-project.org/alsa-kernel alsa-kernel %> hgcompile [...] compilation proceeds cleanly... %> make install %> alsaconf [...] select card and configure it via alsamixer Lo and belold, it all worked! I have a clean compilation and good response from aplay and other user space tools. I haven't applied any patches, and I haven't been able to test SPDIF input yet but I'm betting John Stile should be able to report that faster than I can get to it... Thank you very much for your help with this and for all of the work you and the ALSA team do! -- Darryl -------------- Original message ---------------------- From: Takashi Iwai <tiwai(a)suse.de> > At Wed, 02 Jan 2008 22:31:19 +0000, > Darryl House wrote: > > > > > > -------------- Original message ---------------------- > > [...] > > > > I'd like to get this card's spdif working. What else can I do to help? > > > > Although the driver loads, the user space tools can't use the card any > > > > more. Thank you. > > > > > > > Should I assume at this point no more work will be done on the > > > snd-hda-intel driver to fixing SPDIF-in for the vt1708 sound card? > > > Thank you for all your work so far. > > > > > > > I think it is a bit early to make that assumption. There isn't a lot > > of demand for this particular card, but it doesn't seem that it > > will take too much work to get it going again... after all, it did > > basically work before the last patch, and I am hoping that Takashi > > will find some time soon to address the issues. > > Hm, I see no obvious error in the patch. Are you sure that the latest > HG driver works properly if you don't apply the patch? > > Let's begin with that. > > > Takashi > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

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[alsa-devel] Problem using arecord and aplay
by conn intel 08 Jan '08

08 Jan '08

Dear Friends, I am using kernel 2.6.22.2, and following alsa library and utilities : alsa-lib-1.0.14a & alsa-utils-1.0.14. I am having problem executing arecord and aplay alsa utilities. After executing below command it returns immeaditely to command prompt with following messages. The command and output:: # arecord -c 2 -d 10 -f S16_LE -r 44100 -t wav /home/test.wav Recording WAVE '/home/test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, St ereo ALSA sound/arm/as352x-audio.c:1153: len:15888 arecord: set_params:1012: unable to install sw params: start_mode: DATA xrun_mode: STOP tstamp_mode: NONE period_step: 1 sleep_min: 0 avail_min: 0 xfer_align: 10 silence_threshold: 0 silence_size: 0 boundary: 1445068800 # waiting for any comments or suggestions. Thank you, Ankur.

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Re: [alsa-devel] no input/output analog channels on rme multiface I
by rmgls＠free.fr 08 Jan '08

08 Jan '08

On Fri, 21 Dec 2007 13:41:42 +0100 Takashi Iwai <tiwai(a)suse.de> wrote >> on Sun, 09 Dec 2007 23:39:46 +0100, rmgls(a)free.fr wrote: >> Hi all, >> >> the multiface 1, does not work for me. >> it is recognized as hammerfal DSP, but the amixer shows no analog in/out >> (there are 8 analog I/O in the box) and of course no volume control for >> any channel. > Did you try hdspmixer? > Takashi yes i tried it, without success, hdspmixer does not start with an (inaccesible) message. as i mentionned earlier, the firmware is correctly loaded, and the aslamixer doesn't show any analog I/O. of course one can't play any analog sound, and the midi part is not detected: (is it the same problem?° Note: the kernel is 22.3, running on debian, with the hg version from 20080107: drivers, libs, and all. thanks for your help. Raoul rmgls(a)free.Fr

1 0

[alsa-devel] [RFC] alsa-kernel clean-ups
by Takashi Iwai 08 Jan '08

08 Jan '08

Hi, ALSA kernel tree has lots of unused and rotten codes. I'd like to clean up this. Here we go... 1. Indirect control element access This is what I already suggested. This hasn't been used (we have no proper API in alsa-lib), and 32/64bit wrapper doesn't work at all. Also, the idea to copy such a big data area via ioctl is bad. If we need a big matrix mixer, let make each matrix element accessible individually. Copying the whole matrix at each time isn't efficient. 2. PCM xfer_align parameter This sw_params parameter has never been used in a sane manner, and no one understands what this does exactly. The current implementation looks also buggy because it allows write of shorter size than xfer_align. So, if you do partial writes, the write isn't actually aligned at all. Removing this parameter will make some pcm_lib_* code more readable (and less buggy). 3. PCM sleep_min and tick The "tick" in PCM is set (again) via sw_params. And, nobody uses this feature at all except for a command line option of aplay. (This is literally "nobody", as I checked alsa-lib API calls in all programs in some major distros.) Above all, if we need finer wake-ups for the position update, it's basically an issue that the driver should solve, not tuned by each application. 4. sound/driver.h This header file exists only for some hacks to adapt alsa-driver tree. It's useless for building in the kernel. Let's move a few lines in it to sound/core.h and remove it. For building the modules on alsa-driver external tree, we can simply add #include "adriver.h" to each build stub (alsa-driver/*/*.c) before inclusion of alsa-kernel codes. I'll post some patches to clean the above thing up in the following posts. Please speak up if you have objections. Takashi

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[alsa-devel] Driver - PCM MMAP mode timestamps
by Jaroslav Kysela 08 Jan '08

08 Jan '08

Hi, it seems that we have another inconsistency with PCM MMAP mode timestamps. If I read code correctly, snd_pcm_status() returns timestamp from last interrupt BUT updates hw_ptr to current (calling snd_pcm_update_hw_ptr() again). It means, that application will get unrelated timestamp to hw position returned in snd_pcm_status() in case of mmap mode. I don't see any way to use this timestamp now. To fix this issue, I would like to create a next status ioctl which will return only mmap timestamp plus last related interrupt ring buffer position and remove 'if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_MMAP)' condition from snd_pcm_status() in driver completely. Any opinion? Other idea? Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

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Re: [alsa-devel] ALC268 "test" model
by Jonathan Woithe 08 Jan '08

08 Jan '08

Hi Tobin > > I try to use a patch from J. Woithe to put a test model into the > > ALC268 driver [1] (enabling all pins etc.); unfortunately it gives a > > lot of compiling errors ([2]) which I am unable to understand as I > > don't have sufficient knowledge of the internal API of HDA codec. > > > > Can someone please look over it and correct the mistakes? > > > > Marco Schuster > > > > [1]: http://harddisk.is-a-geek.org/patch.txt > > [2]: http://harddisk.is-a-geek.org/compiling-errors.txt > > Hi, and thanks for the effort. I'll fix this up a bit and send out an > updated patch this weekend. I have an ALC268 test board to work with > now, so I should be able to get it figured out. I'm not entirely sure what's going on here with this patch. As is indicated by the patch itself it's against ALSA in the 2.6.24-rc6 kernel and it applies and compiles fine for me. The last few error messages at the second URL seems to suggest that the patch was in some way corrupted because I'm seeing non-ASCII characters in the error messages. I should note that the patch as referenced by Marco was not believed by me to be ready for release yet - that is why I haven't forwarded it on to the list for consideration yet. I sent it to Marco privately and I had hoped to utilise Marco's system to verify basic functionality of the patch before posting it for consideration. It was prepared in a hurry at the end of last year just before I went on leave and have not had a chance to revisit the patch or the issues that Marco saw until now. Since you have an ALC268 board it's probably sensible for you to take over the patch at this point - I don't have an ALC268 system. However, understand that by making this initial patch available to Marco privately it was not my intention to create a lot of extra work for yourself (or any other developer for that matter). I had intended to see it through to operational status before bothering the list with it. Are you happy to take over management of this patch or would you like me to resume dialog with Marco? If the patch at the above URL is in some way corrupted drop me a line and I'll forward you the original. Regards jonathan

3 5

[alsa-devel] [PATCH] hda: Dynamically create digital gain mixers
by Matthew Ranostay 08 Jan '08

08 Jan '08

Dynamically create digital gain mixers for dmics that have out-amp support. Also some 92HD73xx's codecs don't have DMIC gains, so this also prevents creating dead mixers. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r 37956a45462d pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Mon Jan 07 15:16:37 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Mon Jan 07 15:04:42 2008 -0500 @@ -463,9 +463,6 @@ static struct hda_verb stac92hd73xx_6ch_ /* setup adcs to point to mixer */ { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b}, { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b}, - { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Front Mic */ - { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Mic */ - { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Line In */ { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, @@ -490,9 +487,6 @@ static struct hda_verb stac92hd73xx_8ch_ /* setup adcs to point to mixer */ { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b}, { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b}, - { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Front Mic */ - { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Mic */ - { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Line In */ { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, @@ -519,9 +513,6 @@ static struct hda_verb stac92hd73xx_10ch /* setup adcs to point to mixer */ { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b}, { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b}, - { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Front Mic */ - { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Mic */ - { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Line In */ { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, @@ -635,10 +626,6 @@ static struct snd_kcontrol_new stac92hd7 static struct snd_kcontrol_new stac92hd73xx_6ch_mixer[] = { STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 3), - /* hardware gain controls */ - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x13, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), @@ -665,10 +652,6 @@ static struct snd_kcontrol_new stac92hd7 static struct snd_kcontrol_new stac92hd73xx_8ch_mixer[] = { STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 4), - /* hardware gain controls */ - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x13, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), @@ -695,10 +678,6 @@ static struct snd_kcontrol_new stac92hd7 static struct snd_kcontrol_new stac92hd73xx_10ch_mixer[] = { STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 5), - /* hardware gain controls */ - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x13, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), @@ -725,10 +704,6 @@ static struct snd_kcontrol_new stac92hd7 static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { STAC_INPUT_SOURCE(2), - /* hardware gain controls */ - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x18, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x19, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x0, 0x1a, 0x0, HDA_OUTPUT), @@ -745,10 +720,6 @@ static struct snd_kcontrol_new stac92hd7 static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { STAC_INPUT_SOURCE(2), STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2), - - /* hardware gain controls */ - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x18, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x19, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), @@ -2313,15 +2284,18 @@ static int stac92xx_auto_create_dmic_inp struct sigmatel_spec *spec = codec->spec; struct hda_input_mux *dimux = &spec->private_dimux; hda_nid_t con_lst[HDA_MAX_NUM_INPUTS]; - int i, j; + int err, i, j; + char name[32]; dimux->items[dimux->num_items].label = stac92xx_dmic_labels[0]; dimux->items[dimux->num_items].index = 0; dimux->num_items++; for (i = 0; i < spec->num_dmics; i++) { + hda_nid_t nid; int index; int num_cons; + unsigned int wcaps; unsigned int def_conf; def_conf = snd_hda_codec_read(codec, @@ -2332,17 +2306,32 @@ static int stac92xx_auto_create_dmic_inp if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE) continue; + nid = spec->dmic_nids[i]; num_cons = snd_hda_get_connections(codec, spec->dmux_nids[0], con_lst, HDA_MAX_NUM_INPUTS); for (j = 0; j < num_cons; j++) - if (con_lst[j] == spec->dmic_nids[i]) { + if (con_lst[j] == nid) { index = j; goto found; } continue; found: + wcaps = get_wcaps(codec, nid); + + if (wcaps & AC_WCAP_OUT_AMP) { + sprintf(name, "%s Capture Volume", + stac92xx_dmic_labels[dimux->num_items]); + + err = stac92xx_add_control(spec, + STAC_CTL_WIDGET_VOL, + name, + HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + } + dimux->items[dimux->num_items].label = stac92xx_dmic_labels[dimux->num_items]; dimux->items[dimux->num_items].index = index;

2 1

[alsa-devel] HDA: Enable chipset gcap usage
by Tobin Davis 08 Jan '08

08 Jan '08

Summary: hda_intel.c use gcap register to determine number of streams supported by southbridge. This patch removes hardcoded values for the number of streams supported by the southbridge in most chipsets, and reads these values from the chipset directly. Most systems are hardwired for 4 streams in each direction, but newer chipsets change that capability. Signed off by Tobin Davis <tdavis(a)dsl-only.net>

2 5

Re: [alsa-devel] fujitsu siemens amilo la 1703 vt1807 hda intel card problem
by Tobin Davis 07 Jan '08

07 Jan '08

The information you provided is good, we're just a bit behind. I'm trying to get some programming specs for this codec now. In the mean time, could you try using a tip build of the alsa driver? You can download it here: ftp://ftp.suse.com/pub/projects/alsa/snapshot/driver/ This may have some newer code that will enable your audio. Also, try the drivers on http://viaarena.com. THey may have an updated version that hasn't hit the main tree yet. Tobin On Mon, 2008-01-07 at 21:59 +0100, Norayr Chilingaryan wrote: > Excuse me for bothering you. > May be I need to provide other info? > Did you receive the link? > Di you have any ideas? > > Thank you > > Norayr > > > On Jan 5, 2008 5:53 PM, Norayr Chilingaryan < gnorayr(a)gmail.com> > wrote: > > http://pastebin.ca/842745 > > Thank you very much! > > > > > > On Jan 5, 2008 3:56 AM, Tobin Davis <tdavis(a)dsl-only.net> > wrote: > > Could you download and run > http://bulletproof.servebeer.com/alsa/scripts/alsa-info.sh > > This script will provide a lot of useful information. > After running it, please reply to this email with the > link that the script generates, so we can retrieve the > data, thanks. > > Tobin > > > > > On Sat, 2008-01-05 at 02:48 +0100, Norayr Chilingaryan > wrote: > > > > Hello I decided to write here, firtly, because my > > messages in aksa-user have been ingored, and > > secondly, because I understood that my card most > > probably just unsupported and I must contact > > developers. I have tried to pass options > > model=fujitsu, because I have a fujitsu siements > > notebook etc I have also tried to pass amny other > > options in /etc/modules.d/alsa but withnot effect I > > use gentoo, and the only thing I noticed with latest > > kernels and alsa drivers 1.0.15 is that I can hear > > sound through the headphones, but with low volume. > > It is not possible to get volume higher or speaker > > to make sounds now. So, please, this is information > > about the card 04:01.0 Audio device: VIA > > Technologies, Inc. VIA High Definition Audio > > Controller (rev 10) kakraphoon ~ # > > cat /proc/asound/card0/codec#* | grep Codec Codec: > > VIA VIA VT1708 Codec: Motorola Si3054 Thank you very > > much Norayr > > > > _______________________________________________ > > Alsa-devel mailing list > > Alsa-devel(a)alsa-project.org > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > -- Tobin Davis > > > Hain't we got all the fools in town on our side? And hain't that a big > enough majority in any town? > -- Mark Twain, "Huckleberry Finn" > > > > > -- Tobin Davis Anything anybody can say about America is true. -- Emmett Grogan

1 0

[alsa-devel] Cross compiling alsa-lib (again)
by Adrian McMenamin 07 Jan '08

07 Jan '08

Well, I searched through 96 weeks of archived alsa-devel postings to get to this one: http://article.gmane.org/gmane.linux.alsa.devel/37120 (Which shows I am doing the right thing in any case.) This post recommends this: CFLAGS="-Os -m4 -ml" ./configure --prefix=/usr --host=sh4-unknown-linux-gnu --target=sh4-unknown-linux-gnu --build=i686-pc-linux-gnu I admit to not using the additional cflags stuff before, but even if I use this: CFLAGS="-Os -m4 -ml" ./configure --host=sh4-linux --prefix=/usr CC=/home/adrian/dreamy-linux/buildroot/build_sh4/staging_dir/bin/sh4-linux-gcc --disable-python --disable-rawmidi --enable-static --with-confdir=/home/adrian/dreamy-linux/buildroot/project_build_sh4/dreamcast/root/usr/share/alsa --disable-mixer --enable-static=yes --enable-shared=no --target=sh4-linux --build=i686-linux Followed by: make and make DESTDIR=/home/adrian/dreamy-linux/buildroot/project_build_sh4/dreamcast/root install I still get this: $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: AICA [AICA], device 0: AICA PCM [AICA PCM] Subdevices: 1/1 Subdevice #0: subdevice #0 Followed by: $ aplay -L null Discard all samples (playback) or generate zero samples (capture) $ aplay /aine-email.wav Playing WAVE '/aine-email.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono ALSA lib pcm_plug.c:773:(snd_pcm_plug_hw_refine_schange) Unable to find an usable access for '(null)' aplay: set_params:895: Access type not available $ aplay -D hw:0,0 /aine-email.wav Playing WAVE '/aine-email.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono aplay: set_params:895: Access type not available Just as before. Any clues?

4 8

Re: [alsa-devel] [PATCH] hda: Advanced power management for codecs.
by Takashi Iwai 07 Jan '08

07 Jan '08

At Mon, 07 Jan 2008 12:31:49 -0500, Matthew Ranostay wrote: > > >> @@ -2830,7 +2883,12 @@ static void stac92xx_hp_detect(struct hd > >> for (i = 0; i < cfg->speaker_outs; i++) > >> stac92xx_set_pinctl(codec, cfg->speaker_pins[i], > >> AC_PINCTL_OUT_EN); > >> - } > >> + for (i = 0; i < cfg->hp_outs; i++) > >> + pwr_state |= 1 << (cfg->hp_pins[i] - 0xa); > >> + } > >> + > >> + /* power down unused ports */ > >> + snd_hda_codec_write(codec, codec->afg, 0, 0x7ec, pwr_state); > >> > > > > Is this verb available on every STAC codec? I took a look at some > > specs but they don't show this verb at all. > > > Ah no this a new one for recent codecs, probably need to some sort of > check for this. Or is it equivalent to power down each unused pin widget? If so, we can do it somehow better in the common code. Takashi

1 0

[alsa-devel] [PATCH] hda: Advanced power management for codecs.
by Matthew Ranostay 07 Jan '08

07 Jan '08

Added support for advanced power management features in new IDT codecs. Inactive ADCs and DACs are changed to D3 mode when not in playback or capture, also inactive line-out/headphone ports are disabled when jack detect does not sense a connection. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r 5b03c0176aa5 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Mon Jan 07 13:33:45 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Mon Jan 07 11:22:56 2008 -0500 @@ -160,6 +160,7 @@ struct sigmatel_spec { /* i/o switches */ unsigned int io_switch[2]; + unsigned int io_state[2]; unsigned int clfe_swap; unsigned int aloopback; @@ -438,6 +439,52 @@ static int stac92xx_aloopback_put(struct kcontrol->private_value >> 16, dac_mode); return 1; +} + +static int is_in_dac_nids(struct sigmatel_spec *spec, hda_nid_t nid) +{ + int i; + + for (i = 0; i < spec->multiout.num_dacs; i++) { + if (spec->multiout.dac_nids[i] == nid) + return 1; + } + + return 0; +} + +static int stac92xx_mixer_amp_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + hda_nid_t nid = kcontrol->private_value & 0xffff; + int idx = (kcontrol->private_value >> 19) & 0xf; + int chs = (kcontrol->private_value >> 16) & 0x3; + long *valp = ucontrol->value.integer.value; + unsigned int wid_caps, val, change; + + change = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); +#ifdef CONFIG_SND_HDA_POWER_SAVE + if (!is_in_dac_nids(spec, nid)) + nid = spec->adc_nids[idx]; + + val = *valp++; + wid_caps = get_wcaps(codec, nid); + if (wid_caps & AC_WCAP_POWER) { + if (chs == 3) + val |= *valp; + else { /* Center/LFE mixers workaround */ + spec->io_state[--chs] = val; + val |= spec->io_state[chs ? 0 : 1]; + } + + /* set to power state D3 mode if DAC/ADC isn't being used */ + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, + val ? 0: AC_PWRST_D3); + } +#endif + return change; } static struct hda_verb stac9200_core_init[] = { @@ -622,6 +669,20 @@ static struct hda_verb stac9205_core_ini .private_value = verb_read | (verb_write << 16), \ } +#define STAC_CODEC_MUTE_IDX(xname, xcidx, nid, xindex, direction) \ + { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .index = xcidx, \ + .info = snd_hda_mixer_amp_switch_info, \ + .get = snd_hda_mixer_amp_switch_get, \ + .put = stac92xx_mixer_amp_switch_put, \ + .private_value = HDA_COMPOSE_AMP_VAL(nid, 3, xindex, direction), \ + } + +#define STAC_CODEC_MUTE(xname, nid, xindex, direction) \ + STAC_CODEC_MUTE_IDX(xname, 0, nid, xindex, direction) + static struct snd_kcontrol_new stac9200_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0xb, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Master Playback Switch", 0xb, 0, HDA_OUTPUT), @@ -640,10 +701,10 @@ static struct snd_kcontrol_new stac92hd7 HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), - - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x0, HDA_OUTPUT), + STAC_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x1, HDA_OUTPUT), + STAC_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x1, HDA_OUTPUT), HDA_CODEC_VOLUME("Front Mic Mixer Capture Volume", 0x1d, 0, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Mixer Capture Switch", 0x1d, 0, HDA_INPUT), @@ -670,10 +731,10 @@ static struct snd_kcontrol_new stac92hd7 HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), - - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x0, HDA_OUTPUT), + STAC_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x1, HDA_OUTPUT), + STAC_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x1, HDA_OUTPUT), HDA_CODEC_VOLUME("Front Mic Mixer Capture Volume", 0x1d, 0, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Mixer Capture Switch", 0x1d, 0, HDA_INPUT), @@ -700,10 +761,10 @@ static struct snd_kcontrol_new stac92hd7 HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), - - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x0, HDA_OUTPUT), + STAC_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x1, HDA_OUTPUT), + STAC_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x1, HDA_OUTPUT), HDA_CODEC_VOLUME("Front Mic Mixer Capture Volume", 0x1d, 0, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Mixer Capture Switch", 0x1d, 0, HDA_INPUT), @@ -730,12 +791,12 @@ static struct snd_kcontrol_new stac92hd7 HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x19, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), + STAC_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x0, 0x1a, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1d, 0x1, HDA_OUTPUT), + STAC_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x1, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x1, HDA_OUTPUT), HDA_CODEC_MUTE("Analog Loopback 1", 0x17, 0x3, HDA_INPUT), HDA_CODEC_MUTE("Analog Loopback 2", 0x17, 0x4, HDA_INPUT), @@ -751,12 +812,12 @@ static struct snd_kcontrol_new stac92hd7 HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x19, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), + STAC_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x0, 0x1a, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1d, 0x1, HDA_OUTPUT), + STAC_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x1, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x1, HDA_OUTPUT), { } /* end */ }; @@ -857,7 +918,7 @@ static int stac92xx_build_controls(struc if (err < 0) return err; } - return 0; + return 0; } static unsigned int ref9200_pin_configs[8] = { @@ -1962,7 +2023,7 @@ enum { static struct snd_kcontrol_new stac92xx_control_templates[] = { HDA_CODEC_VOLUME(NULL, 0, 0, 0), - HDA_CODEC_MUTE(NULL, 0, 0, 0), + STAC_CODEC_MUTE(NULL, 0, 0, 0), STAC_CODEC_IO_SWITCH(NULL, 0), STAC_CODEC_CLFE_SWITCH(NULL, 0), }; @@ -2059,19 +2120,6 @@ static int stac92xx_add_dyn_out_pins(str return 0; } - -static int is_in_dac_nids(struct sigmatel_spec *spec, hda_nid_t nid) -{ - int i; - - for (i = 0; i < spec->multiout.num_dacs; i++) { - if (spec->multiout.dac_nids[i] == nid) - return 1; - } - - return 0; -} - /* * Fill in the dac_nids table from the parsed pin configuration * This function only works when every pin in line_out_pins[] @@ -2806,6 +2854,7 @@ static void stac92xx_hp_detect(struct hd struct sigmatel_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; int i, presence; + unsigned char pwr_state = 0; presence = 0; for (i = 0; i < cfg->hp_outs; i++) { @@ -2816,12 +2865,16 @@ static void stac92xx_hp_detect(struct hd if (presence) { /* disable lineouts, enable hp */ - for (i = 0; i < cfg->line_outs; i++) + for (i = 0; i < cfg->line_outs; i++) { stac92xx_reset_pinctl(codec, cfg->line_out_pins[i], AC_PINCTL_OUT_EN); - for (i = 0; i < cfg->speaker_outs; i++) + pwr_state |= 1 << (cfg->line_out_pins[i] - 0xa); + } + for (i = 0; i < cfg->speaker_outs; i++) { stac92xx_reset_pinctl(codec, cfg->speaker_pins[i], AC_PINCTL_OUT_EN); + pwr_state |= 1 << (cfg->speaker_pins[i] - 0xa); + } } else { /* enable lineouts, disable hp */ for (i = 0; i < cfg->line_outs; i++) @@ -2830,7 +2883,12 @@ static void stac92xx_hp_detect(struct hd for (i = 0; i < cfg->speaker_outs; i++) stac92xx_set_pinctl(codec, cfg->speaker_pins[i], AC_PINCTL_OUT_EN); - } + for (i = 0; i < cfg->hp_outs; i++) + pwr_state |= 1 << (cfg->hp_pins[i] - 0xa); + } + + /* power down unused ports */ + snd_hda_codec_write(codec, codec->afg, 0, 0x7ec, pwr_state); } static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res)

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[alsa-devel] snd_pcm_status_get_tstamp()
by Lennart Poettering 07 Jan '08

07 Jan '08

Hi! The documentation for snd_pcm_status_get_tstamp() is not really enlightening. After a call to snd_pcm_status() what data does the tstamp field actually carry? Some kind of system clock timestamp when the data in the status struct was valid, or some kind of sound card clock timestamp that is relative to the point in time where the device was opened/started? 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] Status of snd_pcm_rewind()
by Lennart Poettering 07 Jan '08

07 Jan '08

Hi! What's the status of snd_pcm_rewind()? Is it available on all devices? I assume not (at least dmix doesn't...). But I guess I can safely assume it is available when mmap is used? Is there any API function to test if it is available? I am working on adding "glitch-free" support to PulseAudio. I.e. have a huge playback buffer (> 2s), and schedule playback based on system timers instead of sound card interrupts, thus allowing us to dynamically adjust the inetrrupt frequency to the clients that are connected, and thus save power. I can enable this technique only if I can rewrite the playback buffer whenever I need to, thus I need to in advance if snd_pcm_rewind() will work or not. 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] Hello and providing ALSA support to sdl
by M-.-n 07 Jan '08

07 Jan '08

Hi there, My name is Marc and I'm writing a tracker software originally geared towards gameconsole (first GP32 but now GP2X, PC, Mac) and since I got this lovely eee, I'm planning on porting it to debian-derived platforms. Actually it works already but the SDL version I have uses oss and it seems to hickup sometimes although the cpu power consumption seems fine. Other programs (vlc/audacity) seems no to reflect this problem so I guess it's due to the fact they use ALSA while the SDL I got doesn't. When configuring SDL, it seems not to include ALSA (by not finding alsa/soundlib.h) and I'd like to add it but I'm a little confused on the step to follow. ALSA seems to be installed but I guess I miss the includes and compiling libs. Does it mean I need to configure/compile alsa-lib ? Do I risk corrupting my system if the installed libraries are not compatible ? Sorry for the rambling, but I'm only a beginner in terms of developping audio apps under linux. Regards, Marc.

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Re: [alsa-devel] snd-hda-intel support for SPDIF-in wiht vt1708 card?
by forumjunk＠comcast.net 07 Jan '08

07 Jan '08

-------------- Original message ---------------------- [...] > > I'd like to get this card's spdif working. What else can I do to help? > > Although the driver loads, the user space tools can't use the card any > > more. Thank you. > > > Should I assume at this point no more work will be done on the > snd-hda-intel driver to fixing SPDIF-in for the vt1708 sound card? > Thank you for all your work so far. > I think it is a bit early to make that assumption. There isn't a lot of demand for this particular card, but it doesn't seem that it will take too much work to get it going again... after all, it did basically work before the last patch, and I am hoping that Takashi will find some time soon to address the issues. As of this moment, I assume you're in the same boat I'm in, where the user space tools report an inappropriate ioctl for the device. -- Darryl

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[alsa-devel] Conexant CX2051 on HP Spartan
by Claudio Matsuoka 07 Jan '08

07 Jan '08

Hi, I'm working with Herton Krzesinski on a HP Spartan laptop which has a Conexant CX2051 configured with internal and external microphones connected to two different ADC nodes (0x14 and 0x15 in http://helllabs.org/tmp/hp-spartan-ng.png). We were able to get input from one or other ADC, but we weren't able to switch from one capture source to another (switching between 0x17 and 0x1d would be simple, but HP chose to route the external and internal mics to two different ADCs). What is the recommended way to handle this kind of setup?

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[alsa-devel] [PATCH] Another EMU0404 Board ID
by Veli-Matti Valtonen 07 Jan '08

07 Jan '08

Hey, This is based on pseudo-random playing around with the capabilities. With ca0102 this card gives no output atall, ca0108 appears to work fine, so it rather looks similar to the EMU1010b/EMU1010 changes. Some other people seem to have succeeded in using this aswell: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3496 diff -u -r alsa-driver-hg20071222/alsa-kernel/pci/emu10k1/emu10k1_main.c alsa-driver-hg20071222-0404b/alsa-kernel/pci/emu10k1/emu10k1_main.c --- alsa-driver-hg20071222/alsa-kernel/pci/emu10k1/emu10k1_main.c 2007-12-22 03:00:09.000000000 +0200 +++ alsa-driver-hg20071222-0404b/alsa-kernel/pci/emu10k1/emu10k1_main.c 2007-12-23 23:01:12.807047391 +0200 @@ -1419,6 +1419,14 @@ .ca0102_chip = 1, .spk71 = 1, .emu_model = EMU_MODEL_EMU1010}, /* Emu 1010 */ + /* EMU0404b */ + {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x40021102, + .driver = "Audigy2", .name = "E-mu 0404b [4002]", + .id = "EMU0404", + .emu10k2_chip = 1, + .ca0108_chip = 1, + .spk71 = 1, + .emu_model = EMU_MODEL_EMU0404}, /* EMU 0404 */ /* Tested by James(a)superbug.co.uk 20-3-2007. */ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x40021102, .driver = "Audigy2", .name = "E-mu 0404 [4002]",

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[alsa-devel] es18xx: Enable wavetable input from ESS chips
by Krzysztof Helt 07 Jan '08

07 Jan '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This patch enables wavetable chips ES689/ES69X connected to ESS ES18xx chips. The wavetable chip uses FM DAC if the clock signal from the wavetable is active. It has no effect if there is no ESS wavetable chip present. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- This patch has a gotcha. In theory, the OPL synthesis or wavetable input should use FM DAC depending on the active clock signal from the wavetable chip. Unfortunately, the default state of the wavetable chip from ESS (ES689/ES690/ES692) is to send clock signal all the time. This blocks OPL3 signal if the wavetable chip is present. This problem can be solved with sysex command 4 send to the wavetable chip. It sets the wavetable chip to auto-deactivate mode which switches the wavetable off if there are 5 seconds without midi input (according to the es1879 datasheet). I tested this patch on the es1868 chip without the wavetable. The OPL3 synthesis worked (no clock signal from the wavetable). I tested it on the card with es1868 chip and es690 wavetable. The OPL3 synthesis did not work, the wavetable worked. Both the OPL3 and wavetable worked if the sysex command 4 was sent to the wavetable chip (e.g. by amidi tool). Is it ok to add this sysex command sending to the driver? How? The wavetable clock signal can be detected (there is status bit which says if wavetable is active). Another question is about ES938 chip (bass, treble and 3D control) on the card. This chip is controlled by sysex midi commands. It can be detected (by reading settings with sysex commands). Is it possible to add this to the alsamixer or external tool for setting these controls is recommended? Regards, Krzysztof diff -urp linux-2.6.24/sound/isa/es18xx.c linux-new/sound/isa/es18xx.c --- linux-2.6.24/sound/isa/es18xx.c 2007-12-25 23:23:03.000000000 +0100 +++ linux-new/sound/isa/es18xx.c 2007-12-26 16:30:08.000000000 +0100 @@ -1442,6 +1442,8 @@ static int __devinit snd_es18xx_initiali snd_es18xx_write(chip, 0xB2, 0x50); /* Enable MPU and hardware volume interrupt */ snd_es18xx_mixer_write(chip, 0x64, 0x42); + /* Enable ESS wavetable input */ + snd_es18xx_mixer_bits(chip, 0x48, 0x10, 0x10); } else { int irqmask, dma1mask, dma2mask; ---------------------------------------------------------------------- Rozdajemy nagrody! Wygraj samochod! Sprawdz >> http://link.interia.pl/f1cb8

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[alsa-devel] [PATCH] [ALSA-KERNEL] Add IEC958 digital out support to AD1984 for Lenovo Thinkpads T61/X61
by Jerone Young 07 Jan '08

07 Jan '08

This patch adds IEC958 digital out support for the AD1984 sound card. This card can be found in Lenovo Thinkapds T61/X61. The digital out is not located on the Thinkpad, but optional docking station (it's coxial digital out). I've add this support as it is done the exact same way for the AD1983 & AD1884. I have tested this patch with my Lenovo Thinkpad T61 hooked up to a docking station (that has the digital coxial) and then run to my home theater reciever. Works like a charm :-) Patch is attached and also in this email. Signed-off-by: Jerone Young <jerone(a)gmail.com> diff -r 1227a1c12325 pci/hda/patch_analog.c --- a/pci/hda/patch_analog.c Mon Dec 24 14:40:56 2007 +0100 +++ b/pci/hda/patch_analog.c Thu Dec 27 17:41:13 2007 -0600 @@ -3095,6 +3095,16 @@ static struct snd_kcontrol_new ad1984_th .get = ad198x_mux_enum_get, .put = ad198x_mux_enum_put, }, + /* SPDIF controls */ + HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source", + /* identical with ad1983 */ + .info = ad1983_spdif_route_info, + .get = ad1983_spdif_route_get, + .put = ad1983_spdif_route_put, + }, { } /* end */ }; @@ -3197,7 +3207,7 @@ static int patch_ad1984(struct hda_codec codec->patch_ops.build_pcms = ad1984_build_pcms; break; case AD1984_THINKPAD: - spec->multiout.dig_out_nid = 0; + spec->multiout.dig_out_nid = AD1884_SPDIF_OUT; spec->input_mux = &ad1984_thinkpad_capture_source; spec->mixers[0] = ad1984_thinkpad_mixers; spec->init_verbs[spec->num_init_verbs++] = ad1984_thinkpad_init_verbs;

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[alsa-devel] [PATCH] hda: STAC927x VREF fix
by Matthew Ranostay 07 Jan '08

07 Jan '08

Some laptops incorrectly assume the front input jack as a line in instead of a microphone in. Which in turn disables the voltage reference, in which non-amplified input is not possible. This patch enables VREF80 for the input jack. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r eeed28ed6b7d pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Fri Jan 04 09:22:20 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Fri Jan 04 19:41:06 2008 -0700 @@ -3344,6 +3344,7 @@ static int patch_stac927x(struct hda_cod spec->mixer = stac927x_mixer; break; case STAC_DELL_BIOS: + stac92xx_set_config_reg(codec, 0x0e, 0x02a79130); case STAC_DELL_3ST: /* GPIO2 High = Enable EAPD */ spec->gpio_mask = spec->gpio_data = 0x00000004;

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[alsa-devel] ICE1724 chipset and envy24control
by Norval Watson 07 Jan '08

07 Jan '08

Hi, I just got an M-Audio Audiophile 192 soundcard with ICE1724 chipset. Works out of the box on debian sid. Is there any way to control this card with envy24control utility from alsa-tools? envy24control is written for ICE1712 chipset. Is there any project to develop control utility for ICE1724 card? Seasons greetings, Norval Watson Make the switch to the world's best email. Get the new Yahoo!7 Mail now. www.yahoo7.com.au/worldsbestemail

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[alsa-devel] python midi wrapper
by Aldrin Martoq 07 Jan '08

07 Jan '08

Hello, I need a midi/seq wrapper for Python. I've already created some code for the sequencer API and works great! Here is my python test code: -------------------------------------------------------------------- #!/usr/bin/env python import sys sys.path.append('build/lib.linux-i686-2.5') import alsaaudio print "Creating Sequencer..." seq = alsaaudio.Seq("hw", "Python seqtest.py", alsaaudio.SEQ_OPEN_DUPLEX, alsaaudio.SEQ_NONBLOCK) myid = seq.get_client_id() print "... seq:", myid, seq print "Creating Port..." myport = seq.create_simple_port("Processor port", alsaaudio.SEQ_PORT_CAP_WRITE | alsaaudio.SEQ_PORT_CAP_SUBS_WRITE, alsaaudio.SEQ_PORT_TYPE_APPLICATION) print "... port:", myport #queue = seq.create_queue() #print "... queue:", queue # dump clients and ports def dump_connection(): print print "Client/ports available:" for client in seq.get_client_ports(): clientname, clientid, portlist = client print " client: %3d %s" % (clientid, clientname) for port in portlist: portname, portid, connections = port print " port: %3d:%-2d +-%s" % (clientid, portid, portname) readconn, writeconn = connections for conn in readconn: print " connection to: %s" % conn for conn in writeconn: print " connection from: %s" % conn dump_connection() Is the front seq.connect_ports("0:1", "%d:%d" % (myid, myport)); seq.connect_ports("24:0", "%d:%d" % (myid, myport)); dump_connection() while True: events = seq.receive_events(60000) for ev in events: print ev print "\t", ev.data() -------------------------------------------------------------------- I have tried to contact the mantainer of PyAlsaAudio (cwi(a)unispeed.dk from the README file) for merging my code in his project, but have not received a response so far and the date of files seems old. Is anybody here interested in the merge? How can I put this effor in python-alsaaudio or may I create a new project for this specific task? Is the pyalsa project related to pyalsaaudio? Thanks, -- Aldrin Martoq

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[alsa-devel] function snd_mixer_load failed: Inappropriate ioctl for device
by Salatiel Filho 07 Jan '08

07 Jan '08

*** please let me in the to or cc fields. I am not on the list *** I am having a big trouble to make sound work on my ARM machine. I am running debian 4. # uname -a Linux LS-GL7D6 2.6.16.57-arm2 #7 Wed Dec 12 18:14:51 BRT 2007 armv5tejl GNU/Linux # cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.15. Compiled on Dec 12 2007 for kernel 2.6.16.57-arm2. Modules load successfully # lsmod Module Size Used by snd_pcm_oss 47680 0 snd_mixer_oss 15040 1 snd_pcm_oss snd_usb_audio 79904 0 snd_pcm 75877 2 snd_pcm_oss,snd_usb_audio snd_timer 20164 1 snd_pcm snd_page_alloc 8264 1 snd_pcm snd_usb_lib 15808 1 snd_usb_audio snd_rawmidi 21376 1 snd_usb_lib snd_seq_device 7116 1 snd_rawmidi snd_hwdep 7460 1 snd_usb_audio snd 54368 9 snd_pcm_oss,snd_mixer_oss,snd_usb_audio,snd_pcm,snd_timer,snd_usb_lib,snd_rawmidi,snd_seq_device,snd_hwdep soundcore 7684 1 snd nfs 102044 0 lockd 61780 1 nfs sunrpc 142808 3 nfs,lockd dm_mod 48176 0 usbhid 38628 0 # cat /proc/asound/cards 0 [default ]: USB-Audio - C-Media USB Headphone Set C-Media USB Headphone Set at usb-ehci_platform.4523-1, full speed # cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.15. Compiled on Dec 12 2007 for kernel 2.6.16.57-arm2. but everytime i try to use alsamixer i get: # alsamixer alsamixer: function snd_mixer_load failed: Inappropriate ioctl for device # aplay test.wav Playing WAVE 'test.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono aplay: xrun:1090: read/write error, state = PREPARED # strace alsamixer ... stat64("/usr//share/alsa/alsa.conf", {st_mode=S_IFREG|0644, st_size=8611, ...}) = 0 open("/usr//share/alsa/alsa.conf", O_RDONLY) = 3 fstat64(3, {st_mode=S_IFREG|0644, st_size=8611, ...}) = 0 mmap2(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) = 0x40017000 read(3, "#\n# ALSA library configuration "..., 4096) = 4096 read(3, "r\n\t\t\t\tname defaults.pcm.device\n\t"..., 4096) = 4096 read(3, "{\n\t\t\t@func refer\n\t\t\tname default"..., 4096) = 419 read(3, "", 4096) = 0 read(3, "", 4096) = 0 close(3) = 0 munmap(0x40017000, 4096) = 0 access("/etc/asound.conf", R_OK) = -1 ENOENT (No such file or directory) access("/root/.asoundrc", R_OK) = -1 ENOENT (No such file or directory) open("/dev/snd/controlC0", O_RDONLY) = 3 close(3) = 0 open("/dev/snd/controlC0", O_RDWR) = 3 ioctl(3, USBDEVFS_CONTROL, 0xbec2da64) = 0 ioctl(3, UI_DEV_CREATE, 0xbec2db2c) = 0 close(3) = 0 stat64("/usr//share/alsa/alsa.conf", {st_mode=S_IFREG|0644, st_size=8611, ...}) = 0 open("/dev/snd/controlC0", O_RDONLY) = 3 close(3) = 0 open("/dev/snd/controlC0", O_RDWR) = 3 ioctl(3, USBDEVFS_CONTROL, 0xbec2da44) = 0 fcntl64(3, F_GETFL) = 0x2 (flags O_RDWR) fcntl64(3, F_SETFL, O_RDWR|O_NONBLOCK) = 0 ioctl(3, USBDEVFS_RELEASEINTERFACE, 0xbec2dac0) = 0 ioctl(3, USBDEVFS_RELEASEINTERFACE, 0xbec2dac0) = 0 ioctl(3, USBDEVFS_CONNECTINFO, 0xbec2d86c) = 0 ioctl(3, USBDEVFS_IOCTL, 0xbec2d42c) = -1 ENOTTY (Inappropriate ioctl for device) fstat64(1, {st_mode=S_IFCHR|0620, st_rdev=makedev(136, 2), ...}) = 0 mmap2(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) = 0x40017000 write(1, "\n", 1 ) = 1 write(2, "alsamixer: function snd_mixer_lo"..., 74alsamixer: function snd_mixer_load failed: Inappropriate ioctl for device ) = 74 munmap(0x40017000, 4096) = 0 io_submit(0x2, 0, 0x8 <unfinished ... exit status 2> any help? The same card works ok on my x86 machine using . So the problem is not the card but something else. -- []'s Salatiel "O maior prazer do inteligente é bancar o idiota diante de um idiota que banca o inteligente".

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[alsa-devel] Test music sampled at 192K
by Jon Smirl 06 Jan '08

06 Jan '08

Can anyone point me to some uncompressed test music/tones sampled at 192K/sec? -- Jon Smirl jonsmirl(a)gmail.com

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[alsa-devel] 5.1 surround--no front center speaker sound--[NVidia CK804]
by Steve Thornley 05 Jan '08

05 Jan '08

Nforce 4 MSI motherboard integrated sound----since my Creative X-Fi isn't supported, YET can not get the front center speaker to work. jlt:/home/steven# aplay -l **** List of PLAYBACK Hardware Devices **** card 0: CK804 [NVidia CK804], device 0: Intel ICH [NVidia CK804] Subdevices: 0/1 Subdevice #0: subdevice #0 card 0: CK804 [NVidia CK804], device 2: Intel ICH - IEC958 [NVidia CK804 - IEC958] Subdevices: 1/1 Subdevice #0: subdevice #0 Here is my present asound.conf file that I created in /etc for my Debian etch 4.0----amd64 system X2-4200-cpu # Set default sound card # Useful so that all settings can be changed to a different card here. pcm.snd_card { type hw card 0 } pcm.dmix6 { type dmix ipc_key 1024 ipc_key_add_uid false # let multiple users share ipc_perm 0660 # IPC permissions (octal, default 0600) slave { pcm snd_card # see below rate 48000 channels 6 period_time 0 period_size 1024 buffer_time 0 buffer_size 4096 } } #route for stereo only pcm.dmix20 { type route slave.pcm "dmix6" slave.channels 6 ttable.0.0 1 ttable.1.1 1 } #route for stereo sound+center+subwoofer pcm.ch31dup { type route slave.pcm "dmix6" slave.channels 6 ttable.0.0 1 ttable.1.1 1 ttable.0.4 0.5 ttable.1.4 0.5 ttable.0.5 0.5 ttable.1.5 0.5 } pcm.ch51dup { type route slave.pcm dmix6 slave.channels 6 ttable.0.0 1 ttable.1.1 1 ttable.0.2 1 ttable.1.3 1 ttable.0.4 0.5 ttable.1.4 0.5 ttable.0.5 0.5 ttable.1.5 0.5 } pcm.duplex { type asym playback.pcm "ch51dup" # playback.pcm "dmix2" # capture.pcm "mydsnoop" capture.pcm "snd_card" } ################### # CONVERSION PLUG # ################### # Setting the default pcm device allows the conversion # rate to be selected on the fly. # duplex mode allows any alsa enabled app to read/write # to the dmix plug (Fixes a problem with wine). pcm.!default { type plug slave.pcm "duplex" } pcm.dsp "duplex" pcm.dsp1 "duplex" pcm.fr { type route slave.pcm "dmix6" slave.channels 6 ttable.0.0 1 ttable.1.1 1 } pcm.rr { type route slave.pcm "dmix6" slave.channels 6 ttable.0.2 1 ttable.1.3 1 } pcm.lfe { type route slave.pcm "dmix6" slave.channels 6 ttable.0.4 1 ttable.1.5 1 } Here is my back up with other configurations for the file that created the same 4 speakers and subwoofer sounds but no front center speaker sounds pcm.ch51dup { type route slave.pcm dmix6 slave.channels 6 ttable.0.0 1 ttable.1.1 1 ttable.0.2 1 ttable.1.3 1 ttable.0.4 1 ttable.1.4 1 ttable.0.5 1 ttable.1.5 1 } ########Old Setup # pcm.51to40 # { # type route # slave.pcm surround51 # slave.channels 6 # # # Front and rear, at 75% of original signal strength # ttable.0.0 0.75 # ttable.1.1 0.75 # ttable.2.2 0.75 # ttable.3.3 0.75 # # Center channel routing (routed to front-left and front-right), # 6dB gaindrop (gain half of main channels) per channel # ttable.4.0 0.375 # ttable.4.1 0.375 # LFE channel routing (routed to front-left and front-right), # 6dB gaindrop (gain half of main channels) per channel # ttable.5.0 0.375 # ttable.5.1 0.375 # } When I changed a setting from Autodetect to ALSA I could not get any sound--so I compiled everything from source with the latests tars and still nothing, and once I turned the Autodetect in Multimedia Systems Selector<Preferences<Desktop for the Default output plugin, I still had no sound in my center front speaker. Whew. I'd love it for my Creative X-Fi card to be supported, but right now, I really like my front center speaker back. I'm at the end of my list of possible fixes, maybe someone who understands can take a look at this an have a go ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

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