August 2008 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] X-Fi support in ALSA.
by James Courtier-Dutton 18 Sep '08

18 Sep '08

Hi, I have now had a chance to read some of the specs of the X-Fi cards. aka. emu20k1. It turns out that some of the X-Fi cards have a compatibility mode. By default, they boot up in compatibility mode. The interesting point is that this "compatibility mode" is in fact the intel-HD standard so could use the intel-hda driver. For one X-Fi card with IDs: Device 1102:0009 Subsystem: 1102:0010 works a bit with the ALSA hda-intel driver. So far, the off the shelf hda-intel driver, with simple PCIs adjustment, works for the above card, but only sound capture works so far, so probably some quirk would be needed. This compatibility mode works with the Windows Vista UAA drivers. Not all X-Fi cards support with compatibility mode. For example, any card with (after a cold boot into Linux) Device 1102:0005 Subsystem: xxxx:xxxx will be X-Fi only. With regard to actual full X-Fi support, I am still in the progress of creating a GPL emu20k1.h file, that is the first step towards an snd-emu20k1 native ALSA driver. James

3 3

[alsa-devel] [PATCH 1/2] ca0106 playback 44100Hz support to SPDIF and playback format & rate constraints
by Ben Stanley 10 Sep '08

10 Sep '08

Dear list, This patch provides the ability to play back digital audio via SPDIF at a sampling rate of 44100Hz. It also implements rate constraints to ensure that the hardware restrictions of the card are not violated. Note that if 44100Hz is used, then all used channels must use that frequency. For example, opening hw:0,0 at 48000 and then opening hw:0,1 at 44100 will fail due to constraints. However, you may successfully open both hw:0,0 and hw:0,1 at 44100 (presume no other channels open). Further details of the restriction may be found in the thread http://thread.gmane.org/gmane.linux.alsa.devel/52149/focus=52345 This patch also implements playback format restrictions, such that only one playback format may be in use at any time. I tested driver playback using the attached script (Test_ca0106_driver.sh). I get good audio output for all combinations of sampling rates and formats except for 192kHz S16 (as noted at the end of the script). Sometimes it works and sometimes it doesn't. The reason for the failure is not apparent at this time. It could have something to do with my receiver. I don't know. The attached script also checks and validates that the rate and format constraints operate as required. I periodically have trouble with this driver not producing sound output. This has been investigated and the cause is explained here http://thread.gmane.org/gmane.linux.alsa.devel/55384/focus=55410 I intend to submit a new patch to address this in the future. This patch has been in use for three months on my MythTV system (Ubuntu 8.04). I find it works reliably for me. I also use it with xine, aplay, mplayer, mame, and childsplay. I have checked with checkpatch.pl . Patch is against v1.0.18rc1. Signed-off-by: Ben Stanley Ben.Stanley(a)exemail.com.au This is the patch against alsa-kmirror. --- pci/ca0106/ca0106.h | 7 +- pci/ca0106/ca0106_main.c | 266 ++++++++++++++++++++++++++++++++++++++-------- 2 files changed, 229 insertions(+), 44 deletions(-) diff --git a/pci/ca0106/ca0106.h b/pci/ca0106/ca0106.h index 74175fc..376fa9c 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 { diff --git a/pci/ca0106/ca0106_main.c b/pci/ca0106/ca0106_main.c index 2f8b28a..33f992b 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> @@ -284,9 +290,9 @@ 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, @@ -318,6 +324,111 @@ 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 chi, any_44100 = 0, any_non_44100 = 0, mask = 0; + struct snd_ca0106_channel *chp = 0; + struct snd_pcm_runtime *runtime; + if (snd_BUG_ON(!chip)) + return -EINVAL; + + if (chip->spdif_enable) { + for (chi = 0; chi < 4; ++chi) { + chp = &(chip->playback_channels[chi]); + if (!chp->use) + continue; + if (snd_BUG_ON(!chp->epcm)) + return -EINVAL; + if (!chp->epcm->hw_reserved) + continue; + if (snd_BUG_ON(!chp->epcm->substream)) + return -EINVAL; + if (snd_BUG_ON(!chp->epcm->substream->runtime)) + return -EINVAL; + runtime = chp->epcm->substream->runtime; + snd_printd("snd_hw_rule_playback_rate: ch=%d, " + "rate=%d.\n", chi, runtime->rate); + any_44100 += runtime->rate == 44100; + any_non_44100 += runtime->rate != 44100; + } + if (snd_BUG_ON(any_44100 && any_non_44100)) + return -EINVAL; + if (any_44100) + mask = 0x1; + else if (any_non_44100) + mask = 0xE; + else + mask = 0xF; + } else { + /* 44100Hz is not supported for DAC (FIXME Why?) */ + mask = 0xE; + } + snd_printd("snd_hw_rule_playback_rate: any_44100=%d, " + "any_non_44100=%d, mask=0x%X, spdif=%d\n", + any_44100, any_non_44100, mask, chip->spdif_enable); + 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; + int chi, any_S16 = 0, any_S32 = 0; + struct snd_ca0106_channel *chp = 0; + struct snd_pcm_runtime *runtime; + struct snd_mask fmt, *f = hw_param_mask( + params, SNDRV_PCM_HW_PARAM_FORMAT); + int result; + if (snd_BUG_ON(!chip)) + return -EINVAL; + snd_mask_none(&fmt); + + for (chi = 0; chi < 4; ++chi) { + chp = &(chip->playback_channels[chi]); + if (!chp->use) + continue; + if (snd_BUG_ON(!chp->epcm)) + return -EINVAL; + if (!chp->epcm->hw_reserved) + continue; + if (snd_BUG_ON(!chp->epcm->substream)) + return -EINVAL; + if (snd_BUG_ON(!chp->epcm->substream->runtime)) + return -EINVAL; + runtime = chp->epcm->substream->runtime; + snd_printd("snd_hw_rule_playback_format: ch=%d, format=%d.\n", + chi, runtime->format); + any_S16 += runtime->format == SNDRV_PCM_FORMAT_S16_LE; + any_S32 += runtime->format == SNDRV_PCM_FORMAT_S32_LE; + } + if (snd_BUG_ON(any_S16 && any_S32)) + return -EINVAL; + if (any_S16) + snd_mask_set(&fmt, SNDRV_PCM_FORMAT_S16_LE); + else if (any_S32) + 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_printd("snd_hw_rule_playback_format: any_S16=%d, any_S32=%d, " + "refined_fmt=0x%X, avail_fmt=0x%X, changed=%d\n", + any_S16, any_S32, f->bits[0], fmt.bits[0], result); + return result; +} + unsigned int snd_ca0106_ptr_read(struct snd_ca0106 * emu, unsigned int reg, unsigned int chn) @@ -508,10 +619,24 @@ 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) + err = snd_pcm_hw_constraint_integer( + runtime, SNDRV_PCM_HW_PARAM_PERIODS); + if (err < 0) return err; - if ((err = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64)) < 0) + err = snd_pcm_hw_constraint_step( + runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64); + if (err < 0) return err; + 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) + return err; + 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) + return err; snd_pcm_set_sync(substream); if (chip->details->spi_dac && channel_id != PCM_FRONT_CHANNEL) { @@ -646,6 +771,9 @@ 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_pcm_runtime *runtime = substream->runtime; + struct snd_ca0106_pcm *epcm = runtime->private_data; + epcm->hw_reserved = 0; return snd_pcm_lib_free_pages(substream); } @@ -667,53 +795,103 @@ 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_pcm_runtime *runtime = substream->runtime, *runtimei = 0; struct snd_ca0106_pcm *epcm = runtime->private_data; - int channel = epcm->channel_id; + struct snd_ca0106_channel *chp = 0; + int channel = epcm->channel_id, chi, any_44100 = 0, any_non_44100 = 0; u32 *table_base = (u32 *)(emu->buffer.area+(8*16*channel)); 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; - //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); - /* 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; + epcm->hw_reserved = 1; + /* 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. */ + for (chi = 0; chi < 4; ++chi) { + chp = &(emu->playback_channels[chi]); + if (!chp->use) + continue; + if (snd_BUG_ON(!chp->epcm)) + return -EINVAL; + if (chi != channel && !chp->epcm->hw_reserved) + continue; + if (snd_BUG_ON(!chp->epcm->substream)) + return -EINVAL; + if (snd_BUG_ON(!chp->epcm->substream->runtime)) + return -EINVAL; + runtimei = chp->epcm->substream->runtime; + any_44100 += runtimei->rate == 44100; + any_non_44100 += runtimei->rate != 44100; + /* Rate can be set per channel. */ + /* reg40 control host to fifo */ + /* reg71 controls DAC rate. */ + switch (runtimei->rate) { + case 44100: + /* We only support 44100 to spdif, not to DAC. + (FIXME WHY?)*/ + 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: + printk(KERN_ERR DRVNAME + ": prepare_playback: " + "Bad sampling frequency %d.\n", + runtimei->rate); + } } - /* Format is a global setting */ - /* FIXME: Only let the first channel accessed set this. */ + printk(KERN_DEBUG DRVNAME ": prepare_playback: any_44100=%d, " + "any_non_44100=%d, spdif=%d.\n", + any_44100, any_non_44100, emu->spdif_enable); + /* 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; @@ -1363,8 +1541,8 @@ 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 +1556,9 @@ 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

5 15

Re: [alsa-devel] snd-soc-cs4270: Convert to a new-style i2c driver (work in progress)
by Jean Delvare 04 Sep '08

04 Sep '08

Hi Timur, On Sun, 31 Aug 2008 11:28:08 -0500, Timur Tabi wrote: > On Sun, Aug 31, 2008 at 9:18 AM, Jean Delvare <khali(a)linux-fr.org> wrote: > > Hi Tibur, > > It's "Timur" Oops, apparently I mixed your first name and last name. Sorry about that. > > I am in the process of converting your cs4270 codec driver from the > > legacy i2c model to the new (standard) one. > > It has already been converted. > > http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=commit;h=0d… Ah, excellent. One less thing for me to do :) Thanks! Two comments about your patch: * Your driver now lacks a remove method. Unless I miss something, if the snd-soc-cs4270 driver is unloaded, you will leave dangling resources behind (codec->reg_cache in particular.) * I2C_DRIVERID_CS4270 must be removed from i2c-id.h. > I don't know why this commit hasn't been pushed upstream, though. As I understand it, without that patch the mpc8610_hpcd doesn't work, as the I2C address of the sound codec will be made busy by the platform code and thus the snd-soc-cs4270 driver won't be able to attach to it. If I am correct then I suggest that you ask Takashi to push the patch to Linus now to fix that. > > This is work in progress. > > The patch below converts the cs4270 driver itself. However we also need > > to convert its users. As far as I can see there's only one user at this > > point: mpc8610_hpcd. > > That's correct. I've never seen any interest in the CS4270 outside of > Freescale. > > > The problem is that this driver doesn't look like the other codec > > drivers I have already converted. So, we need to add code to > > instantiate the cs4270 i2c device, but I don't know where this should > > happen. Given that the mpc8610_hpcd is apparently based on Open > > Firmware, I guess that the i2c device should be instantiated directly > > by the platform code. I see that the device is declared in > > mpc8610_hpcd.dts, so maybe it's already done and my patch should work > > already? What do you think? > > I think you need to use the right git repository for your development. :-) Very good point indeed. I'll make sure to check Takashi's sound-2.6 tree before attempting to convert any other SoC codec driver. Do you know off the top of your head if other drivers have already been converted? Thanks, -- Jean Delvare

4 9

Re: [alsa-devel] [PATCH 1/4] ASOC: Blackfin driver for ALSA SoC framework
by Mark Brown 03 Sep '08

03 Sep '08

On Wed, Aug 27, 2008 at 05:39:25PM +0800, Bryan Wu wrote: > From: Cliff Cai <cliff.cai(a)analog.com> > > Signed-off-by: Cliff Cai <cliff.cai(a)analog.com> > Signed-off-by: Bryan Wu <cooloney(a)kernel.org> As with the other patches in the series this looks good but it has been affected by changes in the ASoC APIs and needs to be updated to reflect current ALSA. The topic/asoc branch of Takashi's tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git is what's currently queued for 2.6.28. Other than that, checkpatch-style things and a few more minor issues the main things are the register write interface in /proc and the hardware parameters for the I2S driver. I've not flagged all the checkpatch stuff here. Can I also suggest splitting this up into a series of patches for easier review - for example, adding the SPORT support first, followed by the AC97 and I2S drivers and then the machine drivers? I'd certainly have appreciated having seen the SPORT support (which appears at the end of the patch) before looking at the AC97 and I2S stuff. > + depends on BLACKFIN && SND_SOC > + help > + Say Y or M if you want to add support for codecs attached to > + the Blackfin SPORT (synchronous serial ports) interface in slot 16 > + mode (pseudo AC97 interface). > + You will also need to select the audio interfaces to support below. > + > + Note: > + AC'97 codecs which do not implment the slot-16 mode will not function > + properly with this driver. This driver is known to work with the > + Analog Devices line of AC97 codecs. Your spelling of AC97 isn't consistent here. The kernel seems to prefer AC97, as do you. > + * File: sound/soc/blackfin/bf5xx-ac97-pcm.c > + * Author: Cliff Cai <Cliff.Cai(a)analog.com> > + * > + * Created: Tue June 06 2008 > + * Description: Driver for SSM2602 sound chip built in ADSP-BF52xC Hopefully the driver is more generic than that? > +static int bf5xx_pcm_hw_params(struct snd_pcm_substream *substream, > + struct snd_pcm_hw_params *params) > +{ > + size_t size = bf5xx_pcm_hardware.buffer_bytes_max * sizeof(struct ac97_frame)/4; Checkpatch picks up on this and quite a few other places being over 80 columns. > +static int bf5xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) > +{ > + struct snd_pcm_runtime *runtime = substream->runtime; > + struct sport_device *sport = runtime->private_data; > + int ret = 0; > + > + pr_debug("%s %s\n", substream->stream?"Capture":"Playback", \ > + cmd?" start":" stop"); This looks wrong - there's rather more options for cmd, for example, and you're making assumptions about the value of SNDRV_PMC_STREAM_PLAYBACK. > +/*Need to allocate local buffer when enable MMAP for SPORT working in TMD mode(include AC97).*/ Indentation? > diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.h b/sound/soc/blackfin/bf5xx-ac97-pcm.h > new file mode 100644 > index 0000000..5831300 > --- /dev/null > +++ b/sound/soc/blackfin/bf5xx-ac97-pcm.h > +struct bf5xx_gpio { > + u32 sys; > + u32 rx; > + u32 tx; > + u32 clk; > + u32 frm; > +}; That should be a space before rx, not a tab. > diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c > new file mode 100644 > index 0000000..685a494 > --- /dev/null > +++ b/sound/soc/blackfin/bf5xx-ac97.c > +#include <sound/driver.h> This header is obsolte and should be removed. > +static unsigned short bf5xx_ac97_read(struct snd_ac97 *ac97, > + unsigned short reg) > +{ > + struct ac97_frame out_frame[2], in_frame[2]; > + > + pr_debug("%s enter 0x%x\n", __func__, reg); > + > + /* When dma descriptor is enabled, the register should not be read */ > + if (sport_handle->tx_run || sport_handle->rx_run) { > + printk(KERN_ERR "Could you send a mail to author " > + "to report this?\n"); > + return -EFAULT; > + } Might be nice to say what to report and who the author is - by itself the printout is going to be rather obscure :) > +static int bf5xx_ac97_resume(struct platform_device *pdev, > + struct snd_soc_cpu_dai *dai) struct snd_soc_cpu_dai and struct snd_soc_codec_dai have been merged into a single struct snd_soc_dai in current versions. > + ret = sport_config_rx(sport_handle, IRFS, 0xF, 0, (16*16-1)); > + > + if (ret) { > + printk(KERN_ERR "SPORT is busy!\n"); > + return -EBUSY; > + } > + ret = sport_config_tx(sport_handle, ITFS, 0xF, 0, (16*16-1)); > + > + if (ret) { > + printk(KERN_ERR "SPORT is busy!\n"); > + return -EBUSY; > + } Shouldn't this undo the previous sport_confix_rx() on error? Also, the use of blank lines is a bit funny here - I'd expect the blank before rather than after the function calls. > +static struct proc_dir_entry *ac_entry; > + > +/* For test purpose, read a register from codec */ > +static int proc_write(struct file *file, const char __user *buffer, > + unsigned long count, void *data) > +{ I think it's probably better to drop this from mainline - it doesn't seem like something that we should be supporting long term and obviously it could do damage if used. If you do want to put it in mainline then it ought to go in debugfs rather than proc. > + /*SPORT works in TDM mode to simulate AC97 transfers*/ > + ret = sport_set_multichannel(sport_handle, 16, 0x1F, 1); > + > + if (ret) { > + printk(KERN_ERR "SPORT is busy!\n"); > + return -EBUSY; > + } > + ret = sport_config_rx(sport_handle, IRFS, 0xF, 0, (16*16-1)); > + > + if (ret) { > + printk(KERN_ERR "SPORT is busy!\n"); > + return -EBUSY; > + } Again, odd blank lines and presumably there needs to be something to undo things on error? > + * File: sound/soc/blackfin/bf5xx-ad1980.c > + * Author: Cliff Cai <Cliff.Cai(a)analog.com> > + * > + * Created: Tue June 06 2008 o> + * Description: Driver for SSM2602 sound chip built in ADSP-BF52xC This comment is wrong - it's for the AD1980. > +static int bf5xx_board_startup(struct snd_pcm_substream *substream) > +{ > + struct snd_soc_pcm_runtime *rtd = substream->private_data; > + struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; > + > + pr_debug("%s enter\n", __func__); > + cpu_dai->private_data = sport_handle; > + return 0; > +} > --- /dev/null > +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c > @@ -0,0 +1,293 @@ > +/* > + * File: sound/soc/blackfin/bf5xx-i2s-pcm.c > + * Author: Cliff Cai <Cliff.Cai(a)analog.com> > + * > + * Created: Tue June 06 2008 > + * Description: Driver for SSM2602 sound chip built in ADSP-BF52xC This comment is wrong too :) > +#include <sound/driver.h> Obsolete. > + pr_debug("%s %s\n", substream->stream?"Capture":"Playback", \ > + cmd?" start":" stop"); Same comment as for AC97. Shouldn't more of this code be shared with the AC97 DMA driver? > +static int bf5xx_pcm_close(struct snd_pcm_substream *substream) > +{ > + pr_debug("%s enter\n", __func__); > + /*Nothing need to be cleared here*/ > + > + return 0; > +} You should just be able to remove this if it's empty. > +#include <sound/driver.h> Obsolete. > + /* TX and RX are not independent,they are enabled at the same time, > + * even if only one side is running.So,we need to configure both of them in advance. > + * CPU DAI format:I2S,word length:32 bit,slave mode. > + */ Are the RX and TX clocks independant? If not you should probably tell user space about the constraint in startup(), forcing the second stream that's opened to use the same configuration as the first - take a look at the fsl_ssi driver or the wm8903 driver for examples of how to do this. I'm also not seeing the code that configures the sample rate anywhere - but then it looks like the driver only support slave mode ATM? That's what the machine driver is using. There should still be a set_fmt() to document what's supported if nothing else. > +void decfrag(struct sport_device *sport, int *frag, int tx) > +{ > + --(*frag); > + if (tx == 1 && *frag == 0) > + *frag = sport->tx_frags; > + > + if (tx == 0 && *frag == 0) > + *frag = sport->rx_frags; > +} > +EXPORT_SYMBOL(decfrag); This symbol should be namespaced. > diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c > new file mode 100644 > index 0000000..41e161f > --- /dev/null > +++ b/sound/soc/blackfin/bf5xx-ssm2602.c This driver needs to follow the ssm2602 codec driver so either the series needs reordering or at least this bit needs to be split out into another patch. > +#include <sound/driver.h> Obsolete. > + * WARNING - TODO > + * > + * This code assumes there is a variable clocksource for the SSM2602. > + * i.e. it supplies MCLK depending on rate. > + * > + * If you are using a crystal source then modify the below case > + * statement with a static frequency. > + * > + * If you are using the SPORT to generate clocking then this is > + * where to do it. > + */ > + > + switch (params_rate(params)) { > + case 8000: > + case 16000: > + case 48000: > + case 96000: > + case 11025: > + case 22050: > + case 44100: > + clk = 12000000; > + break; > + } I'm not sure that comment quite agrees with the code here...

2 2

[alsa-devel] [PATCH 0/4] Blackfin supports for ALSA/ASOC
by Bryan Wu 02 Sep '08

02 Sep '08

Hi folks, It's very happy to release this ASOC Blackfin ports. Please review our drivers and feel free to ask us update it. In this patchset, we post ASOC Blackfin supports, SSM2602 audio codec driver and other 2 ALSA/ASOC patches. Thanks a lot -Bryan

8 21

[alsa-devel] get rid of controls with snd_ctl_remove
by Harald Radke 02 Sep '08

02 Sep '08

Hi there! Please be patient with me, I am quite inexperienced with alsa programming, so this might be a dumb question: We have a WM8750 working in our PDA, the codec is working so far but I want to remove some of the control since they aren't needed. Instead of copying the wm8750.c file and adjusting it, I want to write another module which basically get rid of those controls, something like: [...] static const char* unused_controls[] = {"AAAA", "BBBB,"CCCC"}; struct snd_kcontrol *ctl; list_for_each_entry(ctl, &card->controls, list) for(i=0; i < ARRAY_SIZE(unused_controls); i++) if (!strcmp(unused_controls[i],ctl->id.name)) { down_write(&card->controls_rwsem); snd_ctl_remove(card,ctl); up_write(&card->controls_rwsem); break; } unfortunatly this doesn't work (module segfaults...."of course" you will say *g*) My question: is it possible at all and I am just not deep nuff in understanding the sound system or do I have to adjust the original code? Thx Harry

4 14

[alsa-devel] [PATCH] ASoC: Convert wm8731 to a new-style i2c driver (testers wanted)
by Jean Delvare 01 Sep '08

01 Sep '08

Convert the wm8731 codec driver to the new (standard) device driver binding model. Signed-off-by: Jean Delvare <khali(a)linux-fr.org> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- Warning: this assumes that the corgi, poodle and eti_b1 have their codec chips on I2C bus 0, which may or may not be the case. Someone with these machines please test and report. Note that you can safely ignore patch failures in i2c-id.h. include/linux/i2c-id.h | 1 sound/soc/at91/eti_b1_wm8731.c | 1 sound/soc/codecs/wm8731.c | 111 +++++++++++++++++++--------------------- sound/soc/codecs/wm8731.h | 1 sound/soc/pxa/corgi.c | 1 sound/soc/pxa/poodle.c | 1 6 files changed, 59 insertions(+), 57 deletions(-) --- linux-2.6.27-rc4.orig/include/linux/i2c-id.h 2008-08-26 16:41:12.000000000 +0200 +++ linux-2.6.27-rc4/include/linux/i2c-id.h 2008-08-26 17:02:29.000000000 +0200 @@ -64,7 +64,6 @@ #define I2C_DRIVERID_SAA717X 80 /* saa717x video encoder */ #define I2C_DRIVERID_DS1672 81 /* Dallas/Maxim DS1672 RTC */ #define I2C_DRIVERID_TLV320AIC23B 87 /* TI TLV320AIC23B audio codec */ -#define I2C_DRIVERID_WM8731 89 /* Wolfson WM8731 audio codec */ #define I2C_DRIVERID_WM8753 91 /* Wolfson WM8753 audio codec */ #define I2C_DRIVERID_LM4857 92 /* LM4857 Audio Amplifier */ #define I2C_DRIVERID_VP27SMPX 93 /* Panasonic VP27s tuner internal MPX */ --- linux-2.6.27-rc4.orig/sound/soc/at91/eti_b1_wm8731.c 2008-08-26 16:40:44.000000000 +0200 +++ linux-2.6.27-rc4/sound/soc/at91/eti_b1_wm8731.c 2008-08-26 17:02:20.000000000 +0200 @@ -243,6 +243,7 @@ static struct snd_soc_machine snd_soc_ma }; static struct wm8731_setup_data eti_b1_wm8731_setup = { + .i2c_bus = 0, .i2c_address = 0x1a, }; --- linux-2.6.27-rc4.orig/sound/soc/codecs/wm8731.c 2008-08-26 16:40:44.000000000 +0200 +++ linux-2.6.27-rc4/sound/soc/codecs/wm8731.c 2008-08-26 17:11:50.000000000 +0200 @@ -570,86 +570,87 @@ static struct snd_soc_device *wm8731_soc * low = 0x1a * high = 0x1b */ -static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; -/* Magic definition of all other variables and things */ -I2C_CLIENT_INSMOD; - -static struct i2c_driver wm8731_i2c_driver; -static struct i2c_client client_template; - -/* If the i2c layer weren't so broken, we could pass this kind of data - around */ - -static int wm8731_codec_probe(struct i2c_adapter *adap, int addr, int kind) +static int wm8731_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) { struct snd_soc_device *socdev = wm8731_socdev; - struct wm8731_setup_data *setup = socdev->codec_data; struct snd_soc_codec *codec = socdev->codec; - struct i2c_client *i2c; int ret; - if (addr != setup->i2c_address) - return -ENODEV; - - client_template.adapter = adap; - client_template.addr = addr; - - i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); - if (i2c == NULL) - return -ENOMEM; - i2c_set_clientdata(i2c, codec); - codec->control_data = i2c; - - ret = i2c_attach_client(i2c); - if (ret < 0) { - pr_err("failed to attach codec at addr %x\n", addr); - goto err; - } ret = wm8731_init(socdev); - if (ret < 0) { + if (ret < 0) pr_err("failed to initialise WM8731\n"); - goto err; - } - return ret; -err: - kfree(i2c); return ret; } -static int wm8731_i2c_detach(struct i2c_client *client) +static int wm8731_i2c_remove(struct i2c_client *client) { struct snd_soc_codec *codec = i2c_get_clientdata(client); - i2c_detach_client(client); kfree(codec->reg_cache); - kfree(client); return 0; } -static int wm8731_i2c_attach(struct i2c_adapter *adap) -{ - return i2c_probe(adap, &addr_data, wm8731_codec_probe); -} +static const struct i2c_device_id wm8731_i2c_id[] = { + { "wm8731", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8731_i2c_id); -/* corgi i2c codec control layer */ static struct i2c_driver wm8731_i2c_driver = { .driver = { .name = "WM8731 I2C Codec", .owner = THIS_MODULE, }, - .id = I2C_DRIVERID_WM8731, - .attach_adapter = wm8731_i2c_attach, - .detach_client = wm8731_i2c_detach, - .command = NULL, + .probe = wm8731_i2c_probe, + .remove = wm8731_i2c_remove, + .id_table = wm8731_i2c_id, }; -static struct i2c_client client_template = { - .name = "WM8731", - .driver = &wm8731_i2c_driver, -}; +static int wm8731_add_i2c_device(struct platform_device *pdev, + const struct wm8731_setup_data *setup) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct i2c_board_info info; + struct i2c_adapter *adapter; + struct i2c_client *client; + int ret; + + ret = i2c_add_driver(&wm8731_i2c_driver); + if (ret != 0) { + dev_err(&pdev->dev, "can't add i2c driver\n"); + return ret; + } + + memset(&info, 0, sizeof(struct i2c_board_info)); + info.addr = setup->i2c_address; + strlcpy(info.type, "wm8731", I2C_NAME_SIZE); + + adapter = i2c_get_adapter(setup->i2c_bus); + if (!adapter) { + dev_err(&pdev->dev, "can't get i2c adapter %d\n", + setup->i2c_bus); + goto err_driver; + } + + client = i2c_new_device(adapter, &info); + i2c_put_adapter(adapter); + if (!client) { + dev_err(&pdev->dev, "can't add i2c device at 0x%x\n", + (unsigned int)info.addr); + goto err_driver; + } + socdev->codec->control_data = client; + + return 0; + +err_driver: + i2c_del_driver(&wm8731_i2c_driver); + return -ENODEV; +} #endif static int wm8731_probe(struct platform_device *pdev) @@ -682,11 +683,8 @@ static int wm8731_probe(struct platform_ wm8731_socdev = socdev; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) if (setup->i2c_address) { - normal_i2c[0] = setup->i2c_address; codec->hw_write = (hw_write_t)i2c_master_send; - ret = i2c_add_driver(&wm8731_i2c_driver); - if (ret != 0) - printk(KERN_ERR "can't add i2c driver"); + ret = wm8731_add_i2c_device(pdev, setup); } #else /* Add other interfaces here */ @@ -711,6 +709,7 @@ static int wm8731_remove(struct platform snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_unregister_device(codec->control_data); i2c_del_driver(&wm8731_i2c_driver); #endif kfree(codec->private_data); --- linux-2.6.27-rc4.orig/sound/soc/codecs/wm8731.h 2008-08-26 16:40:44.000000000 +0200 +++ linux-2.6.27-rc4/sound/soc/codecs/wm8731.h 2008-08-26 17:02:20.000000000 +0200 @@ -35,6 +35,7 @@ #define WM8731_DAI 0 struct wm8731_setup_data { + int i2c_bus; unsigned short i2c_address; }; --- linux-2.6.27-rc4.orig/sound/soc/pxa/corgi.c 2008-08-26 16:40:44.000000000 +0200 +++ linux-2.6.27-rc4/sound/soc/pxa/corgi.c 2008-08-26 17:02:20.000000000 +0200 @@ -330,6 +330,7 @@ static struct snd_soc_machine snd_soc_ma /* corgi audio private data */ static struct wm8731_setup_data corgi_wm8731_setup = { + .i2c_bus = 0, .i2c_address = 0x1b, }; --- linux-2.6.27-rc4.orig/sound/soc/pxa/poodle.c 2008-08-26 16:40:44.000000000 +0200 +++ linux-2.6.27-rc4/sound/soc/pxa/poodle.c 2008-08-26 17:02:20.000000000 +0200 @@ -284,6 +284,7 @@ static struct snd_soc_machine snd_soc_ma /* poodle audio private data */ static struct wm8731_setup_data poodle_wm8731_setup = { + .i2c_bus = 0, .i2c_address = 0x1b, }; -- Jean Delvare

5 9

[alsa-devel] [PATCH] hda: Distortion fix for dell_m6_core_init
by Matthew Ranostay 01 Sep '08

01 Sep '08

Added the EQ distortion fix to the dell_m6_core_init. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index 6ad6ef3..9968ee4 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -606,8 +606,9 @@ static struct hda_verb dell_eq_core_init[] = { }; static struct hda_verb dell_m6_core_init[] = { - /* set master volume and direct control */ - { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, + /* set master volume to max value without distortion + * and direct control */ + { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xec}, /* setup audio connections */ { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00}, { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01},

2 2

[alsa-devel] [PATCH] ASoC: fix pxa2xx-i2s clk_get call
by Dmitry Baryshkov 01 Sep '08

01 Sep '08

pxa2xx-isa: probe actual device and use it for clk_get call thus fixing error during startup hook Signed-off-by: Dmitry Baryshkov <dbaryshkov(a)gmail.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/pxa/pxa2xx-i2s.c | 40 +++++++++++++++++++++++++++++++++++++++- 1 files changed, 39 insertions(+), 1 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 8548818..c796b18 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -16,6 +16,7 @@ #include <linux/device.h> #include <linux/delay.h> #include <linux/clk.h> +#include <linux/platform_device.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/initval.h> @@ -81,7 +82,6 @@ static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - clk_i2s = clk_get(NULL, "I2SCLK"); if (IS_ERR(clk_i2s)) return PTR_ERR(clk_i2s); @@ -152,6 +152,7 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream, pxa_gpio_mode(gpio_bus[pxa_i2s.master].tx); pxa_gpio_mode(gpio_bus[pxa_i2s.master].frm); pxa_gpio_mode(gpio_bus[pxa_i2s.master].clk); + BUG_ON(IS_ERR(clk_i2s)); clk_enable(clk_i2s); pxa_i2s_wait(); @@ -317,6 +318,43 @@ struct snd_soc_dai pxa_i2s_dai = { EXPORT_SYMBOL_GPL(pxa_i2s_dai); +static int pxa2xx_i2s_probe(struct platform_device *dev) +{ + clk_i2s = clk_get(&dev->dev, "I2SCLK"); + return IS_ERR(clk_i2s) ? PTR_ERR(clk_i2s) : 0; +} + +static int __devexit pxa2xx_i2s_remove(struct platform_device *dev) +{ + clk_put(clk_i2s); + clk_i2s = ERR_PTR(-ENOENT); + return 0; +} + +static struct platform_driver pxa2xx_i2s_driver = { + .probe = pxa2xx_i2s_probe, + .remove = __devexit_p(pxa2xx_i2s_remove), + + .driver = { + .name = "pxa2xx-i2s", + .owner = THIS_MODULE, + }, +}; + +static int __init pxa2xx_i2s_init(void) +{ + clk_i2s = ERR_PTR(-ENOENT); + return platform_driver_register(&pxa2xx_i2s_driver); +} + +static void __exit pxa2xx_i2s_exit(void) +{ + platform_driver_unregister(&pxa2xx_i2s_driver); +} + +module_init(pxa2xx_i2s_init); +module_exit(pxa2xx_i2s_exit); + /* Module information */ MODULE_AUTHOR("Liam Girdwood, liam.girdwood(a)wolfsonmicro.com, www.wolfsonmicro.com"); MODULE_DESCRIPTION("pxa2xx I2S SoC Interface"); -- 1.5.6.3

5 6

[alsa-devel] [PATCH] Automatically use ALC268_TOSHIBA for Toshiba L305 Laptop
by Travis Place 01 Sep '08

01 Sep '08

Changelog: Added Subsystem IDs (0x1179, 0xff64) for the Toshiba Satellite L305 laptop, so it automatically uses the ALC268_TOSHIBA quirk. Signed-off-by: Travis Place <wishie(a)wishie.net> Patch Attached. Thanks, Travis Place (wishie)

2 1