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[alsa-devel] Sigmatel patch for missing
by Klaus Schulz 17 Dec '08

17 Dec '08

Hi folks. Below patch seems to be missing in 1.0.18a! Note that also MCP51 seemed to work afterwards. It needs to be ported to 1.0.18.a though. https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3036 Can anybody let me know how the patch would look like in latest driver rev. THX Klaus --- patch_sigmatel.c.orig 2007-09-15 17:57:07.000000000 -0700 +++ patch_sigmatel.c 2007-09-15 17:50:26.000000000 -0700 @@ -2304,6 +2304,15 @@ stac92xx_auto_set_pinctl(codec, cfg->dig_in_pin, AC_PINCTL_IN_EN); + switch (codec->subsystem_id) { + case 0x107b0317: // Gateway MX3412, MT3423 + case 0x107b0318: // Gateway MT3707, ML3109 (ATI SB450) + case 0x10de026c: // Gateway MX3410 + snd_hda_codec_write(codec, 0x08, 0, AC_VERB_SET_POWER_STATE, 0x00); + snd_hda_codec_write(codec, 0x08, 0, AC_VERB_SET_EAPD_BTLENABLE, 0x02); + break; + } + if (spec->gpio_mute) { stac922x_gpio_mute(codec, 0, 0); stac922x_gpio_mute(codec, 1, 0);

2 1

[alsa-devel] 1.0.18: /proc/asound/card0/stream0 no longer existing?
by Klaus Schulz 17 Dec '08

17 Dec '08

Hi there. Can anybody tell me if /proc/asound/card0/stream0 is not existing any longer and what has replaced it? THX a lot Klaus

2 1

Re: [alsa-devel] Trying to figure out what went wrong with my HDMI.
by Clemens Ladisch 17 Dec '08

17 Dec '08

Mark Jenks wrote: > On Tue, Dec 16, 2008 at 7:24 AM, Clemens Ladisch <clemens(a)ladisch.de> wrote: > > Use "plughw:0,3" instead of "hw:0,3". > > What is the difference? Plughw automatically converts to a sample format that the device supports. > And why did hw:0,3 work before? Probably your old hardware did support mono playback. > If I want to use it in Myth, would it be ALSA:plughw:0,3 then? Yes. (It is possible that Myth can do resampling and channel rerouting by itself; in this case, hw would also work.) HTH Clemens

1 0

[alsa-devel] alsa-utils: alsactl restore not working after system restart
by Klaus Schulz 16 Dec '08

16 Dec '08

Hi there. I am running Ubuntu 8.10 - and latest ALSA. After restart amixer defaults to default settings. I am not really sure if this is Ubuntu/Debian specific or an Alsa issue. It seems that either a link /var/lib/alsa/asound.state to /etc/asound.state is needed or that /etc/init.d/alsa-utils needs to be updated within the restore section accordingly. With the link workaround I managed to get it working. Cheers Klaus

3 2

[alsa-devel] [PATCH] 1/2 ALSA: ca0106 support 44100Hz playback to spdif
by Ben Stanley 16 Dec '08

16 Dec '08

>From 241939bdf80de14e204d6fcb43d850d33bdeda20 Mon Sep 17 00:00:00 2001 From: Ben Stanley <Ben.Stanley(a)exemail.com.au> Date: Tue, 16 Dec 2008 23:47:36 +1100 Subject: [PATCH] ALSA: ca0106 support 44100Hz playback to spdif This patch provides support for playback of 44100Hz sampled material in 16 and 32 bit sample depths through the digital spdif/iec958 connection/s available on the ca0106 card. I have re-worked the patch to address comments from a previous review by Takashi [1]. In particular, I have incorporated spin-locking to address a race condition that Takashi was concerned about. This code has been running on my MythTV box for the last fortnight without apparent ills. I have re-based the code a bit for submission. This patch is against alsa-kmirror. [1] http://article.gmane.org/gmane.linux.alsa.devel/55825/match=takashi+ca0106+… Signed-off-by: Ben Stanley <Ben.Stanley(a)exemail.com.au> --- pci/ca0106/ca0106.h | 13 ++- pci/ca0106/ca0106_main.c | 402 ++++++++++++++++++++++++++++++++++++++------- 2 files changed, 351 insertions(+), 64 deletions(-) diff --git a/pci/ca0106/ca0106.h b/pci/ca0106/ca0106.h index 3faccb6..2ffec74 100644 --- a/pci/ca0106/ca0106.h +++ b/pci/ca0106/ca0106.h @@ -1,7 +1,7 @@ /* * Copyright (c) 2004 James Courtier-Dutton <James(a)superbug.demon.co.uk> * Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit - * Version: 0.0.22 + * Version: 0.0.23 * * FEATURES currently supported: * See ca0106_main.c for features. @@ -49,6 +49,8 @@ * Implement support for Line-in capture on SB Live 24bit. * 0.0.22 * Add support for mute control on SB Live 24bit (cards w/ SPI DAC) + * 0.0.23 + * Add support for playback sampling rate and format constraints. * * * This code was initally based on code from ALSA's emu10k1x.c which is: @@ -644,6 +646,8 @@ #include "ca_midi.h" +#define DRVNAME "snd-ca0106" + struct snd_ca0106; struct snd_ca0106_channel { @@ -659,6 +663,7 @@ struct snd_ca0106_pcm { struct snd_pcm_substream *substream; int channel_id; unsigned short running; + unsigned short hw_reserved; }; struct snd_ca0106_details { @@ -688,6 +693,7 @@ struct snd_ca0106 { unsigned short model; /* subsystem id */ spinlock_t emu_lock; + spinlock_t pcm_lock; struct snd_ac97 *ac97; struct snd_pcm *pcm; @@ -707,6 +713,11 @@ struct snd_ca0106 { struct snd_ca_midi midi2; u16 spi_dac_reg[16]; + + unsigned char count_pb_44100_chan; + unsigned char count_pb_non_44100_chan; + unsigned char count_pb_S16_chan; + unsigned char count_pb_S32_chan; }; int snd_ca0106_mixer(struct snd_ca0106 *emu); diff --git a/pci/ca0106/ca0106_main.c b/pci/ca0106/ca0106_main.c index 6ac1936..bc78312 100644 --- a/pci/ca0106/ca0106_main.c +++ b/pci/ca0106/ca0106_main.c @@ -1,7 +1,7 @@ /* * Copyright (c) 2004 James Courtier-Dutton <James(a)superbug.demon.co.uk> * Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit - * Version: 0.0.25 + * Version: 0.0.26 * * FEATURES currently supported: * Front, Rear and Center/LFE. @@ -83,9 +83,14 @@ * Add support for mute control on SB Live 24bit (cards w/ SPI DAC) * 0.0.25 * Powerdown SPI DAC channels when not in use + * 0.0.26 Ben Stanley + * Added support for output at 44100Hz rate (SPDIF only). + * Implemented constraints system for output rate and format. * * BUGS: * Some stability problems when unloading the snd-ca0106 kernel module. + * Some programs fail to produce sound output (tested on SPDIF). See + * http://thread.gmane.org/gmane.linux.alsa.devel/55384/focus=55410 * -- * * TODO: @@ -145,6 +150,7 @@ #include <sound/core.h> #include <sound/initval.h> #include <sound/pcm.h> +#include <sound/pcm_params.h> #include <sound/ac97_codec.h> #include <sound/info.h> @@ -285,9 +291,9 @@ static struct snd_pcm_hardware snd_ca0106_playback_hw = { SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_SYNC_START, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, - .rates = (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | - SNDRV_PCM_RATE_192000), - .rate_min = 48000, + .rates = (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000), + .rate_min = 44100, .rate_max = 192000, .channels_min = 2, //1, .channels_max = 2, //6, @@ -319,6 +325,110 @@ static struct snd_pcm_hardware snd_ca0106_capture_hw = { .fifo_size = 0, }; +static unsigned int all_spdif_playback_rates[] = + {44100, 48000, 96000, 192000}; + +static int hw_rule_playback_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_ca0106 *chip = rule->private; + int mask; + unsigned long flags; + if (snd_BUG_ON(!chip)) + return -EINVAL; + + spin_lock_irqsave(&chip->pcm_lock, flags); + if (chip->spdif_enable) { + /* Compute the mask applied to all_spdif_playback_rates */ + if (chip->count_pb_44100_chan) + mask = 0x1; + else if (chip->count_pb_non_44100_chan) + mask = 0xE; + else + mask = 0xF; + } else { + /* 44100Hz is not supported for DAC. + The DAC cannot be clocked at 44100, only 48000 and 96000 + from the CA0106 chip. The SPDIF can be clocked at 44100. + It is a hardware limitation. */ + mask = 0xE; + } + snd_printdd("snd_hw_rule_playback_rate: any_44100=%d, " + "any_non_44100=%d, mask=0x%X, spdif=%d\n", + chip->count_pb_44100_chan, + chip->count_pb_non_44100_chan, + mask, chip->spdif_enable); + spin_unlock_irqrestore(&chip->pcm_lock, flags); + return snd_interval_list(hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE), + ARRAY_SIZE(all_spdif_playback_rates), + all_spdif_playback_rates, mask); +} + +static int hw_rule_playback_format(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_ca0106 *chip = rule->private; + struct snd_mask fmt; + struct snd_mask *f = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + int result; + unsigned long flags; + if (snd_BUG_ON(!chip)) + return -EINVAL; + snd_mask_none(&fmt); + + spin_lock_irqsave(&chip->pcm_lock, flags); + if (chip->count_pb_S16_chan) + snd_mask_set(&fmt, SNDRV_PCM_FORMAT_S16_LE); + else if (chip->count_pb_S32_chan) + snd_mask_set(&fmt, SNDRV_PCM_FORMAT_S32_LE); + else { + /* No format yet chosen, so both formats are available. */ + snd_mask_set(&fmt, SNDRV_PCM_FORMAT_S16_LE); + snd_mask_set(&fmt, SNDRV_PCM_FORMAT_S32_LE); + } + result = snd_mask_refine(f, &fmt); + snd_printdd("snd_hw_rule_playback_format: any_S16=%d, any_S32=%d, " + "refined_fmt=0x%X, avail_fmt=0x%X, changed=%d\n", + chip->count_pb_S16_chan, + chip->count_pb_S32_chan, f->bits[0], fmt.bits[0], + result); + spin_unlock_irqrestore(&chip->pcm_lock, flags); + return result; +} + +void snd_ca0106_rebuild_playback_channel_counters(struct snd_ca0106 *chip, + int skip_channel) +{ + /* Re-build the counters of rate=44100 and the formats + for use in constraints processing. Does not process the + current_channel (this must be dealt with in the calling context). + To process all channels, set skip_channel=-1. + This must be called with chip->pcm_lock held. */ + + int chi; + struct snd_ca0106_channel *pchannel; + struct snd_pcm_runtime *runtimei; + + chip->count_pb_44100_chan = chip->count_pb_non_44100_chan = 0; + chip->count_pb_S16_chan = chip->count_pb_S32_chan = 0; + for (chi = 0; chi < 4; ++chi) { + if (chi == skip_channel) + continue; + pchannel = &(chip->playback_channels[chi]); + if (!pchannel->use || !pchannel->epcm + || !pchannel->epcm->hw_reserved) + continue; + runtimei = pchannel->epcm->substream->runtime; + chip->count_pb_44100_chan += runtimei->rate == 44100; + chip->count_pb_non_44100_chan += runtimei->rate != 44100; + chip->count_pb_S16_chan += + runtimei->format == SNDRV_PCM_FORMAT_S16_LE; + chip->count_pb_S32_chan += + runtimei->format == SNDRV_PCM_FORMAT_S32_LE; + } +} + unsigned int snd_ca0106_ptr_read(struct snd_ca0106 * emu, unsigned int reg, unsigned int chn) @@ -393,7 +503,7 @@ int snd_ca0106_i2c_write(struct snd_ca0106 *emu, int status; int retry; if ((reg > 0x7f) || (value > 0x1ff)) { - snd_printk(KERN_ERR "i2c_write: invalid values.\n"); + snd_printk(KERN_ERR DRVNAME "i2c_write: invalid values.\n"); return -EINVAL; } @@ -463,7 +573,20 @@ static void snd_ca0106_intr_disable(struct snd_ca0106 *emu, unsigned int intrenb static void snd_ca0106_pcm_free_substream(struct snd_pcm_runtime *runtime) { + struct snd_ca0106_pcm *epcm = runtime->private_data; + struct snd_ca0106 *chip = epcm->emu; + /* FIXME how to tell which case to use? */ + /* struct snd_ca0106_channel *channel = + &(chip->playback_channels[epcm->channel_id]); */ + /* struct snd_ca0106_channel *channel = + &(chip->capture_channels[epcm->channel_id]); */ + unsigned long flags; + + spin_lock_irqsave(&chip->pcm_lock, flags); kfree(runtime->private_data); + runtime->private_data = 0; + /* *channel = 0; */ + spin_unlock_irqrestore(&chip->pcm_lock, flags); } static const int spi_dacd_reg[] = { @@ -487,19 +610,21 @@ static int snd_ca0106_pcm_open_playback_channel(struct snd_pcm_substream *substr struct snd_ca0106_channel *channel = &(chip->playback_channels[channel_id]); struct snd_ca0106_pcm *epcm; struct snd_pcm_runtime *runtime = substream->runtime; + unsigned long flags; int err; epcm = kzalloc(sizeof(*epcm), GFP_KERNEL); if (epcm == NULL) return -ENOMEM; + spin_lock_irqsave(&chip->pcm_lock, flags); epcm->emu = chip; epcm->substream = substream; epcm->channel_id=channel_id; - + runtime->private_data = epcm; runtime->private_free = snd_ca0106_pcm_free_substream; - + runtime->hw = snd_ca0106_playback_hw; channel->emu = chip; @@ -509,10 +634,24 @@ static int snd_ca0106_pcm_open_playback_channel(struct snd_pcm_substream *substr //printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel); //channel->interrupt = snd_ca0106_pcm_channel_interrupt; channel->epcm = epcm; - if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) - return err; - if ((err = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64)) < 0) - return err; + err = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (err < 0) + goto error; + err = snd_pcm_hw_constraint_step(runtime, + 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64); + if (err < 0) + goto error; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + hw_rule_playback_rate, (void *)chip, + SNDRV_PCM_HW_PARAM_RATE, -1); + if (err < 0) + goto error; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, + hw_rule_playback_format, (void *)chip, + SNDRV_PCM_HW_PARAM_FORMAT, -1); + if (err < 0) + goto error; snd_pcm_set_sync(substream); if (chip->details->spi_dac && channel_id != PCM_FRONT_CHANNEL) { @@ -522,9 +661,12 @@ static int snd_ca0106_pcm_open_playback_channel(struct snd_pcm_substream *substr chip->spi_dac_reg[reg] &= ~spi_dacd_bit[channel_id]; err = snd_ca0106_spi_write(chip, chip->spi_dac_reg[reg]); if (err < 0) - return err; + goto error; } - return 0; + err = 0; +error: + spin_unlock_irqrestore(&chip->pcm_lock, flags); + return err; } /* close callback */ @@ -532,8 +674,12 @@ static int snd_ca0106_pcm_close_playback(struct snd_pcm_substream *substream) { struct snd_ca0106 *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_ca0106_pcm *epcm = runtime->private_data; + struct snd_ca0106_pcm *epcm; + unsigned long flags; + spin_lock_irqsave(&chip->pcm_lock, flags); + epcm = runtime->private_data; chip->playback_channels[epcm->channel_id].use = 0; + spin_unlock_irqrestore(&chip->pcm_lock, flags); if (chip->details->spi_dac && epcm->channel_id != PCM_FRONT_CHANNEL) { const int reg = spi_dacd_reg[epcm->channel_id]; @@ -574,6 +720,7 @@ static int snd_ca0106_pcm_open_capture_channel(struct snd_pcm_substream *substre struct snd_ca0106_channel *channel = &(chip->capture_channels[channel_id]); struct snd_ca0106_pcm *epcm; struct snd_pcm_runtime *runtime = substream->runtime; + unsigned long flags; int err; epcm = kzalloc(sizeof(*epcm), GFP_KERNEL); @@ -584,10 +731,11 @@ static int snd_ca0106_pcm_open_capture_channel(struct snd_pcm_substream *substre epcm->emu = chip; epcm->substream = substream; epcm->channel_id=channel_id; - + + spin_lock_irqsave(&chip->pcm_lock, flags); runtime->private_data = epcm; runtime->private_free = snd_ca0106_pcm_free_substream; - + runtime->hw = snd_ca0106_capture_hw; channel->emu = chip; @@ -596,13 +744,16 @@ static int snd_ca0106_pcm_open_capture_channel(struct snd_pcm_substream *substre channel->use = 1; //printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel); //channel->interrupt = snd_ca0106_pcm_channel_interrupt; - channel->epcm = epcm; + channel->epcm = epcm; if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) - return err; + goto error; //snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, &hw_constraints_capture_period_sizes); if ((err = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64)) < 0) - return err; - return 0; + goto error; + err = 0; +error: + spin_unlock_irqrestore(&chip->pcm_lock, flags); + return err; } /* close callback */ @@ -610,9 +761,13 @@ static int snd_ca0106_pcm_close_capture(struct snd_pcm_substream *substream) { struct snd_ca0106 *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_ca0106_pcm *epcm = runtime->private_data; + struct snd_ca0106_pcm *epcm; + unsigned long flags; + spin_lock_irqsave(&chip->pcm_lock, flags); + epcm = runtime->private_data; chip->capture_channels[epcm->channel_id].use = 0; /* FIXME: maybe zero others */ + spin_unlock_irqrestore(&chip->pcm_lock, flags); return 0; } @@ -647,6 +802,24 @@ static int snd_ca0106_pcm_hw_params_playback(struct snd_pcm_substream *substream /* hw_free callback */ static int snd_ca0106_pcm_hw_free_playback(struct snd_pcm_substream *substream) { + struct snd_ca0106 *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime; + struct snd_ca0106_pcm *epcm; + unsigned long flags; + + spin_lock_irqsave(&chip->pcm_lock, flags); + runtime = substream->runtime; + epcm = runtime->private_data; + epcm->hw_reserved = 0; + + snd_ca0106_rebuild_playback_channel_counters(chip, epcm->channel_id); + + snd_printdd(KERN_DEBUG DRVNAME ": free_playback: any_44100=%d, " + "any_non_44100=%d, spdif=%d.\n", + chip->count_pb_44100_chan, chip->count_pb_non_44100_chan, + emu->spdif_enable); + spin_unlock_irqrestore(&chip->pcm_lock, flags); + return snd_pcm_lib_free_pages(substream); } @@ -668,53 +841,149 @@ static int snd_ca0106_pcm_hw_free_capture(struct snd_pcm_substream *substream) static int snd_ca0106_pcm_prepare_playback(struct snd_pcm_substream *substream) { struct snd_ca0106 *emu = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_ca0106_pcm *epcm = runtime->private_data; - int channel = epcm->channel_id; - u32 *table_base = (u32 *)(emu->buffer.area+(8*16*channel)); + struct snd_pcm_runtime *runtime = substream->runtime, *runtimei; + struct snd_ca0106_pcm *epcm; + struct snd_ca0106_channel *pchannel; + int channel, chi; + unsigned char count_pb_44100_chan, count_pb_non_44100_chan; + unsigned char count_pb_S16_chan, count_pb_S32_chan; + u32 *table_base; u32 period_size_bytes = frames_to_bytes(runtime, runtime->period_size); u32 hcfg_mask = HCFG_PLAYBACK_S32_LE; u32 hcfg_set = 0x00000000; u32 hcfg; - u32 reg40_mask = 0x30000 << (channel<<1); + u32 reg40_mask = 0xFF0000; u32 reg40_set = 0; u32 reg40; - /* FIXME: Depending on mixer selection of SPDIF out or not, select the spdif rate or the DAC rate. */ - u32 reg71_mask = 0x03030000 ; /* Global. Set SPDIF rate. We only support 44100 to spdif, not to DAC. */ + u32 reg71_mask; + u32 reg71_shift; u32 reg71_set = 0; u32 reg71; int i; + unsigned long flags; - //snd_printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, periods=%u, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, runtime->periods, frames_to_bytes(runtime, 1)); - //snd_printk("dma_addr=%x, dma_area=%p, table_base=%p\n",runtime->dma_addr, runtime->dma_area, table_base); - //snd_printk("dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",emu->buffer.addr, emu->buffer.area, emu->buffer.bytes); + /* FIXME CLEAN UP IF spdif_enable IS CHANGED WHILE CHANNELS ARE OPENED + * OR PREVENT THIS FROM HAPPENING. */ + if (emu->spdif_enable) + reg71_shift = 24; /* SPDIF Output Rate */ + else + reg71_shift = 16; /* I2S Output Rate */ + reg71_mask = 0x3 << reg71_shift; + + /*printk(KERN_DEBUG DRVNAME ": prepare_playback: " + "channel_number=%d, rate=%d, format=0x%x, channels=%d, " + "buffer_size=%ld,period_size=%ld, periods=%u, " + "frames_to_bytes=%d\n", + channel, runtime->rate, runtime->format, runtime->channels, + runtime->buffer_size, runtime->period_size, runtime->periods, + frames_to_bytes(runtime, 1));*/ + /*printk("dma_addr=%x, dma_area=%p, table_base=%p\n", + runtime->dma_addr,runtime->dma_area, table_base);*/ + /*printk("dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n", + emu->buffer.addr,emu->buffer.area, emu->buffer.bytes);*/ + /* We are forced to build the settings for all the channels. */ + spin_lock_irqsave(&emu->pcm_lock, flags); + epcm = runtime->private_data; + channel = epcm->channel_id; + + snd_ca0106_rebuild_playback_channel_counters(emu, channel); + + /* Determine the effects of the new settings for validation. + Note that nothing is recorded permanently + until the new settings are validated. */ + count_pb_44100_chan = + emu->count_pb_44100_chan + runtime->rate == 44100; + count_pb_non_44100_chan = + emu->count_pb_non_44100_chan + runtime->rate != 44100; + count_pb_S16_chan = emu->count_pb_S16_chan + + runtime->format == SNDRV_PCM_FORMAT_S16_LE; + count_pb_S32_chan = emu->count_pb_S32_chan + + runtime->format == SNDRV_PCM_FORMAT_S32_LE; + /* Rate can be set per channel. */ /* reg40 control host to fifo */ /* reg71 controls DAC rate. */ - switch (runtime->rate) { - case 44100: - reg40_set = 0x10000 << (channel<<1); - reg71_set = 0x01010000; - break; - case 48000: - reg40_set = 0; - reg71_set = 0; - break; - case 96000: - reg40_set = 0x20000 << (channel<<1); - reg71_set = 0x02020000; - break; - case 192000: - reg40_set = 0x30000 << (channel<<1); - reg71_set = 0x03030000; - break; - default: - reg40_set = 0; - reg71_set = 0; - break; + for (chi = 0; chi < 4; ++chi) { + pchannel = &(emu->playback_channels[chi]); + if (!pchannel->use || !pchannel->epcm) + continue; + if (!pchannel->epcm->hw_reserved && chi != channel) + continue; + runtimei = pchannel->epcm->substream->runtime; + /* Rate can be set per channel. */ + /* reg40 control host to fifo */ + /* reg71 controls DAC rate. */ + switch (runtimei->rate) { + case 44100: + /* 44100Hz is not supported for DAC. The DAC cannot be + clocked at 44100, only 48000 and 96000 from the CA0106 + chip. The SPDIF can be clocked at 44100. + It is a hardware limitation. */ + if (emu->spdif_enable) { + /* When using 44100, *all* channels + must be set to that rate. */ + reg40_set |= 0x550000; + reg71_set |= 0x1 << reg71_shift; + break; + } else { + printk(KERN_ERR DRVNAME + "prepare_playback: " + "44100Hz is invalid for DAC.\n"); + } + case 48000: + /* reg40_set &= !(0x1 << (chi<<1)); */ + /* reg71_set &= !(0x1 << reg71_shift); */ + break; + case 96000: + reg40_set |= 0x20000 << (chi<<1); + reg71_set |= 0x2 << reg71_shift; + break; + case 192000: + reg40_set |= 0x30000 << (chi<<1); + reg71_set |= 0x3 << reg71_shift; + break; + default: + spin_unlock_irqrestore(&emu->pcm_lock, flags); + printk(KERN_ERR DRVNAME + ": prepare_playback: " + "Bad sampling frequency %dHz.\n", + runtime->rate); + return -EINVAL; + } } - /* Format is a global setting */ - /* FIXME: Only let the first channel accessed set this. */ + if (count_pb_44100_chan && count_pb_non_44100_chan) { + spin_unlock_irqrestore(&emu->pcm_lock, flags); + printk(KERN_ERR DRVNAME + "prepare_playback: requested sampling rate %dHz" + " conflicts with other selected sampling rates.\n", + runtime->rate); + return -EINVAL; + } + if (count_pb_S16_chan && count_pb_S32_chan) { + spin_unlock_irqrestore(&emu->pcm_lock, flags); + printk(KERN_ERR DRVNAME + "prepare_playback: requested sample format %d" + " conflicts with other selected sample formats.\n", + runtime->format); + return -EINVAL; + } + /* Requested sample rate and format are now validated. + Commit the change. */ + emu->count_pb_44100_chan = count_pb_44100_chan; + emu->count_pb_non_44100_chan = count_pb_non_44100_chan; + emu->count_pb_S16_chan = count_pb_S16_chan; + emu->count_pb_S32_chan = count_pb_S32_chan; + epcm->hw_reserved = 1; + + snd_printdd(KERN_DEBUG DRVNAME ": prepare_playback: any_44100=%d, " + "any_non_44100=%d, spdif=%d.\n", + emu->count_pb_44100_chan, emu->count_pb_non_44100_chan, + emu->spdif_enable); + spin_unlock_irqrestore(&emu->pcm_lock, flags); + + /* Format is a global setting. */ + /* Only the first channel accessed can set this + (enforced by constraints). */ switch (runtime->format) { case SNDRV_PCM_FORMAT_S16_LE: hcfg_set = 0; @@ -736,10 +1005,13 @@ static int snd_ca0106_pcm_prepare_playback(struct snd_pcm_substream *substream) reg71 = (reg71 & ~reg71_mask) | reg71_set; snd_ca0106_ptr_write(emu, 0x71, 0, reg71); - /* FIXME: Check emu->buffer.size before actually writing to it. */ - for(i=0; i < runtime->periods; i++) { - table_base[i*2] = runtime->dma_addr + (i * period_size_bytes); - table_base[i*2+1] = period_size_bytes << 16; + table_base = (u32 *)(emu->buffer.area+(8*16*channel)); + if (!snd_BUG_ON(8*16*channel+runtime->periods*2 > emu->buffer.bytes)) { + for (i = 0; i < runtime->periods; i++) { + table_base[i*2] = runtime->dma_addr + + (i * period_size_bytes); + table_base[i*2+1] = period_size_bytes << 16; + } } snd_ca0106_ptr_write(emu, PLAYBACK_LIST_ADDR, channel, emu->buffer.addr+(8*16*channel)); @@ -1335,6 +1607,7 @@ static int __devinit snd_ca0106_create(int dev, struct snd_card *card, chip->irq = -1; spin_lock_init(&chip->emu_lock); + spin_lock_init(&chip->pcm_lock); chip->port = pci_resource_start(pci, 0); if ((chip->res_port = request_region(chip->port, 0x20, @@ -1351,8 +1624,9 @@ static int __devinit snd_ca0106_create(int dev, struct snd_card *card, return -EBUSY; } chip->irq = pci->irq; - - /* This stores the periods table. */ + + /* This stores the periods table. FIXME Only the first 512 bytes + appear to be used (for playback buffers) */ if(snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), 1024, &chip->buffer) < 0) { snd_ca0106_free(chip); return -ENOMEM; @@ -1363,8 +1637,8 @@ static int __devinit snd_ca0106_create(int dev, struct snd_card *card, pci_read_config_dword(pci, PCI_SUBSYSTEM_VENDOR_ID, &chip->serial); pci_read_config_word(pci, PCI_SUBSYSTEM_ID, &chip->model); #if 1 - printk(KERN_INFO "snd-ca0106: Model %04x Rev %08x Serial %08x\n", chip->model, - pci->revision, chip->serial); + printk(KERN_INFO DRVNAME ": Model %04x Rev %08x Serial %08x\n", + chip->model, pci->revision, chip->serial); #endif strcpy(card->driver, "CA0106"); strcpy(card->shortname, "CA0106"); @@ -1378,7 +1652,9 @@ static int __devinit snd_ca0106_create(int dev, struct snd_card *card, } chip->details = c; if (subsystem[dev]) { - printk(KERN_INFO "snd-ca0106: Sound card name=%s, subsystem=0x%x. Forced to subsystem=0x%x\n", + printk(KERN_INFO DRVNAME + ": Sound card name=%s, subsystem=0x%x. " + "Forced to subsystem=0x%x\n", c->name, chip->serial, subsystem[dev]); } -- 1.5.4.3

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[alsa-devel] [PATCH] ALSA: ca0106 MSI K8N Diamond MB spi_dac 2->1
by Ben Stanley 16 Dec '08

16 Dec '08

>From fe2e6692ea6fcdf93d724b8aa48e7433994e011c Mon Sep 17 00:00:00 2001 From: Ben Stanley <Ben.Stanley(a)exemail.com.au> Date: Mon, 15 Dec 2008 23:38:12 +1100 Subject: [PATCH] ALSA: ca0106 MSI K8N Diamond MB spi_dac 2->1 This patch removes an inconsistency that became apparent when I documented the fields of snd_ca0106_details. spi_dac is always used in a 'boolean' sense, so this cleanup should make no difference. Signed-off-by: Ben Stanley <Ben.Stanley(a)exemail.com.au> --- pci/ca0106/ca0106_main.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/pci/ca0106/ca0106_main.c b/pci/ca0106/ca0106_main.c index 88fbf28..6ac1936 100644 --- a/pci/ca0106/ca0106_main.c +++ b/pci/ca0106/ca0106_main.c @@ -254,7 +254,7 @@ static struct snd_ca0106_details ca0106_chip_details[] = { .name = "MSI K8N Diamond MB", .gpio_type = 2, .i2c_adc = 1, - .spi_dac = 2 } , + .spi_dac = 1 } , /* Shuttle XPC SD31P which has an onboard Creative Labs * Sound Blaster Live! 24-bit EAX * high-definition 7.1 audio processor". -- 1.5.4.3

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[alsa-devel] [PATCH] ALSA: ca0106 Add comments to snd_ca0106_details struct
by Ben Stanley 16 Dec '08

16 Dec '08

>From dada1c7fe866b860f96d56315648cd5704675aa1 Mon Sep 17 00:00:00 2001 From: Ben Stanley <Ben.Stanley(a)exemail.com.au> Date: Fri, 12 Dec 2008 09:47:13 +1100 Subject: [PATCH] ALSA: ca0106 Add comments to snd_ca0106_details struct Takashi wrote an email [1] explaining the fields of snd_ca0106_details, so I captured the information into the ca0106.h header file. [1] http://article.gmane.org/gmane.linux.alsa.devel/56783/match=takashi+gpio_ty… Signed-off-by: Ben Stanley <Ben.Stanley(a)exemail.com.au> --- pci/ca0106/ca0106.h | 12 ++++++++---- 1 files changed, 8 insertions(+), 4 deletions(-) diff --git a/pci/ca0106/ca0106.h b/pci/ca0106/ca0106.h index 74175fc..3faccb6 100644 --- a/pci/ca0106/ca0106.h +++ b/pci/ca0106/ca0106.h @@ -664,10 +664,14 @@ struct snd_ca0106_pcm { struct snd_ca0106_details { u32 serial; char * name; - int ac97; - int gpio_type; - int i2c_adc; - int spi_dac; + int ac97; /* ac97 = 0 -> Select MIC, Line in, TAD in, AUX in. + ac97 = 1 -> Default to AC97 in. */ + int gpio_type; /* gpio_type = 1 -> shared mic-in/line-in + gpio_type = 2 -> shared side-out/line-in. */ + int i2c_adc; /* with i2c_adc=1, the driver adds some capture volume + controls, phone, mic, line-in and aux. */ + int spi_dac; /* spi_dac=1 adds the mute switch for each analog + output, front, rear, etc. */ }; // definition of the chip-specific record -- 1.5.4.3

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[alsa-devel] Trying to figure out what went wrong with my HDMI.
by Mark Jenks 16 Dec '08

16 Dec '08

I have a M3N78 Pro motherboard, and am running alsa 1.0.18a on it. I had HDMI audio working on it, but for some reason it has stopped. I really didn't change too much about the computer, other then take out a hvr-1800, and put in a hvr-1600. I did try to go up to .27 kernel, but had other problems and went back to .26. Recompiled alsa 1.0.18a and even tried multiple versions of the nvidia drivers. (173, 177).. And now I have just tried the snapshot with the same results. I believe it was recognised as a ALC883 before, and not is is a ALC1200, but I'm not completely sure about it. Help! :) -Mark aplay -l shows: card 0: NVidia [HDA NVidia], device 0: ALC1200 Analog [ALC1200 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: NVidia [HDA NVidia], device 1: ALC1200 Digital [ALC1200 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: NVidia [HDA NVidia], device 3: NVIDIA HDMI [NVIDIA HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 So, everything looks good there. If I try and do an aplay, I get. #aplay -D hw:0,3 /usr/share/sounds/alsa/Noise.wav Playing WAVE '/usr/share/sounds/alsa/Noise.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono aplay: set_params:966: Channels count non available I noticed that I am missing hdmi in my aplay -L. When it worked, it was there. # aplay -L front:CARD=NVidia,DEV=0 HDA NVidia, ALC1200 Analog Front speakers surround40:CARD=NVidia,DEV=0 HDA NVidia, ALC1200 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=NVidia,DEV=0 HDA NVidia, ALC1200 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=NVidia,DEV=0 HDA NVidia, ALC1200 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=NVidia,DEV=0 HDA NVidia, ALC1200 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=NVidia,DEV=0 HDA NVidia, ALC1200 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=NVidia,DEV=0 HDA NVidia, ALC1200 Digital IEC958 (S/PDIF) Digital Audio Output null Discard all samples (playback) or generate zero samples (capture) pulse PulseAudio Sound Server default Default

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[alsa-devel] ASoC: help needed for capture DAPM (twl4030)
by Peter Ujfalusi 16 Dec '08

16 Dec '08

Hello, I need some help/advice on how to implement the DAPM routings for twl4030 capture path. There are at least two issues which I'm wondering at the moment. [1] Mux for the analog paths As with other parts, this is implemented as a bitfield, where only one bit can be enabled at the time (0x0 being no routed inputs). ADCL <-|- Main Mix (0x1) |- Headset mic (0x2) |- AUXL, LineL (0x4) |- Carkit mic (0x8) Invalid values: 0x3, 0x5, 0x6, 0x7 ADCR <-|- Sub mic (0x1) |- AUXR, LineR (0x4) Invalid values: 0x2, 0x3 Since the playback path also has similar 'mux as not enum but bitfield' types, I'm thinking of creating a new type for soc based on the soc_enum. This new type would store the valid enum texts and the corresponding value in the register (with mask). This could be used in very flexible way. It can be also be used instead of the normal enum types (altrough it does not make sense). But this can wait for now. [2] ADC versus digital microphone TWL has two ADCs (ADCL, ADCR), the routing looks something like this, when the ADCs are in use: InputL mux -> AmpliL -> ADCL |-> ATXL1_PGA -> Mic channel1 (I2S, TDM) |-> ATXL2_PGA -> Mic channel2 (TDM) InputR mux -> AmpliR -> ADCR |-> ATXR1_PGA -> Mic channel3 (I2S, TDM) |-> ATXR2_PGA -> Mic channel4 (TDM) When the digital microphones are in use: Digimic0 Left -> ATXL1_PGA -> Mic channel1 (I2S, TDM) Digimic0 Right -> ATXR1_PGA -> Mic channel3 (I2S, TDM) Digimic1 Left -> ATXL2_PGA -> Mic channel2 (TDM) Digimic1 Right -> ATXR2_PGA -> Mic channel4 (TDM) There is a single point of selection for Mic channel1,3 and for channel2,4. So ATXL1_PGA and ATXR1_PGA receives the data either from the ADCs or from the digimic0, ATXL2_PGA and ATXR2_PGA receives the data either from the ADCs or from the digimic1. As you can see, if the digital mic is selected the ADC can be powered off, since they are not needed. How to map this with DAPM? Is it mandatory to have the SND_SOC_DAPM_INPUT(digimic) to be connected to SND_SOC_DAPM_ADC? I have been thinking to create four 'ghost ADC' for the four channel. Than when building the route the real ADCs can receive switches to handle the power management. Thank you, -- Péter

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[alsa-devel] [PATCH] ALSA: hda - Add automatic model setting for Samsung Q45
by Luke Yelavich 16 Dec '08

16 Dec '08

Have the Samsung Q45 (144d:c510) select ALC262_HIPPO by default Signed-off-by: Luke Yelavich <themuso(a)ubuntu.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4afa0c9..3b2a793 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10541,6 +10541,7 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FUJITSU), SND_PCI_QUIRK(0x144d, 0xc032, "Samsung Q1 Ultra", ALC262_ULTRA), SND_PCI_QUIRK(0x144d, 0xc039, "Samsung Q1U EL", ALC262_ULTRA), + SND_PCI_QUIRK(0x144d, 0xc510, "Samsung Q45", ALC262_HIPPO), SND_PCI_QUIRK(0x17aa, 0x384e, "Lenovo 3000 y410", ALC262_LENOVO_3000), SND_PCI_QUIRK(0x17ff, 0x0560, "Benq ED8", ALC262_BENQ_ED8), SND_PCI_QUIRK(0x17ff, 0x058d, "Benq T31-16", ALC262_BENQ_T31), -- 1.5.6.3

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