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

Re: [alsa-devel] Problem setting mixer switch with amixer/alsactl.
by Mark Brown 24 Oct '08

24 Oct '08

On 23 Oct 2008, at 23:12, Oliver Ford <oliford(a)oliford.co.uk> wrote: > Oliver Ford wrote: > >>> name 'Speaker Playback Volume' >>> value 1 >>> } >>> > This is unrelated, I have just left it there from playing around. > > >> >>> One other thing that would be useful would be to put trace which >>> logs >>> the register writes in ac97_write() in the driver and then try >>> testing >>> both using alsamixer and then amixer to do the final enable of the >>> Speaker Mixer PCM Playback Switch. >>> > Well... > > In terms of the control's direct effect, setting it in alsamixer > gives: > pxa2xx_ac97_write: 0c = af08 > > setting it in amixer: > pxa2xx_ac97_write: 0c = af09 > > > > Am I right in understanding that setting all the controls in amixer > doesn't actually enable the control in the codec chip at that time > but that the powering on/off is done by soc-dapm.c when the the DAC > is actually enabled when playing some audio? This appears to be what > is happening when it does work (looking at the log). I hadn't > realised this. > > However, after running the offending amixer command this no longer > happens. It seems that the dapm_power_widgets() function determines > that the DAC is off and so doesn't power any widgets on at all. > > I've put some debugging in pxa2xx-ac97-write and in > dapm_power_widgets() and the log is attached (stripped of non-audio > stuff). > > > > Oliver > > Clean boot... > > > ** run speaker.sh setup without PCM->speaker setting... ** > pxa2xx_ac97_write: 000c = ef08 > pxa2xx_ac97_write: 001c = 0002 > pxa2xx_ac97_write: 001c = 1802 > pxa2xx_ac97_write: 001e = 0800 > pxa2xx_ac97_write: 0006 = 0081 > pxa2xx_ac97_write: 0002 = 0080 > > ** run alsamixer, set PCM->speaker on: ** > pxa2xx_ac97_write: 000c = af08 > > > # aplay /stuff/400Hz-wav.wav > +hpipaq214_startup() > hpipaq214-audio: Enabling iPAQ Speaker... > hpipaq214_startup, gpio96 -> on > -hpipaq214_startup() > > Playing WAVE '/stuff/400Hz-wav.wav' : Signed 16 bit Little Endian, > Rate 44100 Hz, Mono > pxa2xx_ac97_write: 002a = 0411 > pxa2xx_ac97_write: 002c = ac44 > DAPM: Path Widget 'DAC Inv Mux 2': off --> on > DAPM: Path Widget 'Out 4 Mux': off --> on > DAPM: Path Widget 'Left Speaker Out Mux': off --> on > DAPM: DAC Widget 'Right DAC': off --> on (w->active==1) > pxa2xx_ac97_write: 003c = 3abf > DAPM: DAC Widget 'Left DAC': off --> on (w->active==1) > pxa2xx_ac97_write: 003c = 3a3f > DAPM: Path Widget 'AC97 Mixer': off --> on > DAPM: Path Widget 'Speaker Mixer': off --> on > pxa2xx_ac97_write: 003c = 3a3d > DAPM: Path Widget 'Out 4': off --> on > pxa2xx_ac97_write: 003e = efff > DAPM: Path Widget 'Left Speaker': off --> on > pxa2xx_ac97_write: 003e = eeff > freq changed, set PCD to 5 (LCCR3 = 04700005, LCCR4 = 80000000) > DAPM: Path Widget 'iPAQ Speaker': off --> on > pxa2xx_ac97_write: 003c = 1a3d > hpipaq214_shutdown, gpio96 -> off > > ** aplay exits. ** > > ** ... few seconds delay ... ** > > DAPM: Path Widget 'iPAQ Speaker': on --> off > DAPM: Path Widget 'Out 4': on --> off > pxa2xx_ac97_write: 003e = feff > DAPM: Path Widget 'Left Speaker': on --> off > pxa2xx_ac97_write: 003e = ffff > DAPM: Path Widget 'AC97 Mixer': on --> off > DAPM: Path Widget 'Speaker Mixer': on --> off > pxa2xx_ac97_write: 003c = 1a3f > DAPM: Path Widget 'Right DAC': on --> off > pxa2xx_ac97_write: 003c = 1a7f > DAPM: Path Widget 'Left DAC': on --> off > pxa2xx_ac97_write: 003c = 1aff > DAPM: Path Widget 'DAC Inv Mux 2': on --> off > DAPM: Path Widget 'Out 4 Mux': on --> off > DAPM: Path Widget 'Left Speaker Out Mux': on --> off > pxa2xx_ac97_write: 003c = 1aff > pxa2xx_ac97_write: 0026 = 0000 > > > ** all quiet ** > > # > # > # amixer cset numid=62 1 > pxa2xx_ac97_write: 000c = af09 > numid=62,iface=MIXER,name='Speaker Mixer PCM Playback Swit' > ; type=BOOLEAN,access=rw---,values=1 > : values=on > > > > # aplay /stuff/400Hz-wav.wav > +hpipaq214_startup() > hpipaq214-audio: Enabling iPAQ Speaker... > hpipaq214_startup, gpio96 -> on > -hpipaq214_startup() > > Playing WAVE '/stuff/400Hz-wav.wav' : Signed 16 bit Little Endian, > Rate 44100 Hz, Mono > freq changed, set PCD to 5 (LCCR3 = 04700005, LCCR4 = 80000000) > pxa2xx_ac97_write: 002a = 0411 > pxa2xx_ac97_write: 002c = ac44 > DAPM: DAC Widget 'Right DAC': off --> off (w->active==1) > DAPM: DAC Widget 'Left DAC': off --> off (w->active==1) > pxa2xx_ac97_write: 003c = 1aff > hpipaq214_shutdown, gpio96 -> off > # > # > pxa2xx_ac97_write: 003c = 1aff > pxa2xx_ac97_write: 0026 = 0000

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[alsa-devel] [PATCH] aoa i2sbus: don't overwrite module parameter
by Johannes Berg 23 Oct '08

23 Oct '08

We shouldn't modify a global variable here. Signed-off-by: Johannes Berg <johannes(a)sipsolutions.net> --- sound/aoa/soundbus/i2sbus/i2sbus-core.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) --- everything.orig/sound/aoa/soundbus/i2sbus/i2sbus-core.c 2008-10-23 13:55:14.000000000 +0200 +++ everything/sound/aoa/soundbus/i2sbus/i2sbus-core.c 2008-10-23 13:55:33.000000000 +0200 @@ -159,7 +159,7 @@ static int i2sbus_add_dev(struct macio_d struct i2sbus_dev *dev; struct device_node *child = NULL, *sound = NULL; struct resource *r; - int i, layout = 0, rlen; + int i, layout = 0, rlen, ok = force; static const char *rnames[] = { "i2sbus: %s (control)", "i2sbus: %s (tx)", "i2sbus: %s (rx)" }; @@ -192,7 +192,7 @@ static int i2sbus_add_dev(struct macio_d layout = *layout_id; snprintf(dev->sound.modalias, 32, "sound-layout-%d", layout); - force = 1; + ok = 1; } } /* for the time being, until we can handle non-layout-id @@ -201,7 +201,7 @@ static int i2sbus_add_dev(struct macio_d * When there are two i2s busses and only one has a layout-id, * then this depends on the order, but that isn't important * either as the second one in that case is just a modem. */ - if (!force) { + if (!ok) { kfree(dev); return -ENODEV; }

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[alsa-devel] [PATCH] ASoC: tlv320aic3x: Fix DSP DAI format and signal polarities matching
by Jarkko Nikula 23 Oct '08

23 Oct '08

- Codec doesn't support to configure bit clock and frame sync polarities - Codec doesn't support DSP_A format but DSP_B with inverted bit clock polarity - Match also other formats with their signal polarities Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> --- sound/soc/codecs/tlv320aic3x.c | 16 ++++++++++------ 1 files changed, 10 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 05336ed..cff276e 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -863,17 +863,21 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - /* interface format */ - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_I2S: + /* + * match both interface format and signal polarities since they + * are fixed + */ + switch (fmt & (SND_SOC_DAIFMT_FORMAT_MASK | + SND_SOC_DAIFMT_INV_MASK)) { + case (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF): break; - case SND_SOC_DAIFMT_DSP_A: + case (SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF): iface_breg |= (0x01 << 6); break; - case SND_SOC_DAIFMT_RIGHT_J: + case (SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_NB_NF): iface_breg |= (0x02 << 6); break; - case SND_SOC_DAIFMT_LEFT_J: + case (SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF): iface_breg |= (0x03 << 6); break; default: -- 1.5.6.5

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[alsa-devel] [PATCH 0/4] ASoC: Blackfin updates
by Bryan Wu 23 Oct '08

23 Oct '08

Hi Mark and Takashi: We updates the patchset: - split out AD1980 change to a seperated patch - coding style change Thanks -Bryan

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[alsa-devel] Select source support in alsa soc
by naveen krishna ch 23 Oct '08

23 Oct '08

I am in a process of making a asoc driver for costum board. with OMAP3430 and twl4030 combination.. I have a driver now at a palce where i can play back with stereo headset and record through line in cable. I need some understanding of how can i give support for selecting output and input sources. I have understood the way it is done in ALSA but i feel the sane should not be used in ALSA SOC frame work. I would be thankful if someone can help me solve this issue. Thanks, Neyo

2 1

[alsa-devel] ASOC on OMAP3+twl4030
by naveen krishna ch 23 Oct '08

23 Oct '08

Hello all, I am reporting an issue for last 4 days and i am not getting enough response fromt he mailing list. My issue is with the asoc drivers for omap3 and twl4030 combination which are available in the latest git kernel. Audio is working on my custom board when i boot out of MMC but the same setup (x-laod, u-boot and uImage and hadware) the audio is not working when i boot out of NAND. Can any one help me in debugging the issue. Background: I am using 2.6.26 linux kernel i have ported soc stuff from 2.6.27 latest git kernel for omap3 mcbsp and twl4030 codec support.. the audio is working when i use it on OMAP3EVM even though it is booted out of NAND. I am getting very less replies against my issue am i posting to wrong people or what can any one tell me how can i make use of this mailing list to its fullest -- Thank You all, Neyo.

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Re: [alsa-devel] [PATCH] ALSA: emu10k1: fix device names for Live!/Audigy1/2/4/E-mu and coding style
by Vedran Miletić 23 Oct '08

23 Oct '08

Do you have the non-working old revision or working new revision? Because, if it already works, it will continue to work. However, if it doesn, then it won't change :-( It's a known hardware issue with old E-mu 0404 (EM8850 if I'm not mistaken). My code changes are mostly cleanups to adhere to standards and changes to make device names more user friendly and consistent, no big changes since I'm not a real programmer (yet! :-) who knows...). 2008/10/22 adrian dezeo <adezeo(a)gmail.com>: > Hi Vedran: > > My name is Adrian Dezeo and I'm from Argentina. > I have a emu 0404 and gladly will test any code from the emu10k1 alsa > driver if you need to. > Thanks in advance. > best regards. > > Adrian. > > > Vedran Miletić wrote: >> >> Hmm... I actually followed James here, he set standard of naming with >> CardBus 1616's name. But, oh well. I can fix it in my next run, and I >> will likely do it when I play around with porting over complete >> kx.inf's device list. >> >> 2008/10/22 Tony Vroon <tony(a)vroon.org>: >> >>> >>> On a related note, it looks to me as if the leading two characters (the >>> manufacturing location) have been erroneously included for E-MU >>> products. >>> You'll see this MA code on some sound cards as well (MASBxxxx) >>> >>> Regards, >>> Tony V. >>> >>> >> >> >> >> > > -- Vedran Miletić

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[alsa-devel] [PATCH 3/3] saa7146: Emagic Audiowerk8 low-level ALSA driver
by Matthias Nyffenegger 23 Oct '08

23 Oct '08

From: Matthias Nyffenegger <matthias.nyffenegger(a)bluewin.ch> Low-level ALSA driver for Emagic Audiowerk8 sound card. Project page: http://sourceforge.net/projects/aw8-alsa Built and tested with Vanilla 2.6.25.16 Signed-off-by: Matthias Nyffenegger <matthias.nyffenegger(a)bluewin.ch> --- This is a request for submission to ALSA-tree. diff -uprN ../alsa-driver-1.0.17/alsa-kernel/pci/saa7146/license.txt ../alsa-driver-1.0.17.mod/alsa-kernel/pci/saa7146/license.txt --- ../alsa-driver-1.0.17/alsa-kernel/pci/saa7146/license.txt 1970-01-01 01:00:00.000000000 +0100 +++ ../alsa-driver-1.0.17.mod/alsa-kernel/pci/saa7146/license.txt 2008-10-22 23:34:05.000000000 +0200 @@ -0,0 +1,339 @@ + GNU GENERAL PUBLIC LICENSE + Version 2, June 1991 + + Copyright (C) 1989, 1991 Free Software Foundation, Inc. + 675 Mass Ave, Cambridge, MA 02139, USA + Everyone is permitted to copy and distribute verbatim copies + of this license document, but changing it is not allowed. + + Preamble + + The licenses for most software are designed to take away your +freedom to share and change it. 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[alsa-devel] [PATCH 2/3] saa7146: Emagic Audiowerk8 low-level ALSA driver
by Matthias Nyffenegger 23 Oct '08

23 Oct '08

From: Matthias Nyffenegger <matthias.nyffenegger(a)bluewin.ch> Low-level ALSA driver for Emagic Audiowerk8 sound card. Project page: http://sourceforge.net/projects/aw8-alsa Built and tested with Vanilla 2.6.25.16 Signed-off-by: Matthias Nyffenegger <matthias.nyffenegger(a)bluewin.ch> --- This is a request for submission to ALSA-tree. diff -uprN ../alsa-driver-1.0.17/alsa-kernel/pci/saa7146/README.txt ../alsa-driver-1.0.17.mod/alsa-kernel/pci/saa7146/README.txt --- ../alsa-driver-1.0.17/alsa-kernel/pci/saa7146/README.txt 1970-01-01 01:00:00.000000000 +0100 +++ ../alsa-driver-1.0.17.mod/alsa-kernel/pci/saa7146/README.txt 2008-10-22 23:34:05.000000000 +0200 @@ -0,0 +1 @@ +See sourceforge.net project "Audiowerk8 ALSA driver" (aw8-alsa) Wiki for more information. diff -uprN ../alsa-driver-1.0.17/alsa-kernel/pci/saa7146/Makefile ../alsa-driver-1.0.17.mod/alsa-kernel/pci/saa7146/Makefile --- ../alsa-driver-1.0.17/alsa-kernel/pci/saa7146/Makefile 1970-01-01 01:00:00.000000000 +0100 +++ ../alsa-driver-1.0.17.mod/alsa-kernel/pci/saa7146/Makefile 2008-10-22 23:34:05.000000000 +0200 @@ -0,0 +1,8 @@ +# Makefile for Audiowerk module. +# run command in source directory: +# make -C /usr/src/linux M=`pwd` modules modules_install + +EXTRA_CFLAGS := -DSAA7146_SUBSYS_LOG_TAG=\"AW8\" +obj-m := snd-aw8.o +snd-aw8-objs := saa7146i2c.o saa7146i2s.o saa7146audio.o aw8.o aw8alsa.o + diff -uprN ../alsa-driver-1.0.17/alsa-kernel/pci/saa7146/aw8alsa.c ../alsa-driver-1.0.17.mod/alsa-kernel/pci/saa7146/aw8alsa.c --- ../alsa-driver-1.0.17/alsa-kernel/pci/saa7146/aw8alsa.c 1970-01-01 01:00:00.000000000 +0100 +++ ../alsa-driver-1.0.17.mod/alsa-kernel/pci/saa7146/aw8alsa.c 2008-10-22 23:34:05.000000000 +0200 @@ -0,0 +1,826 @@ +/* + * Emagic Audiowerk ALSA driver + * Copyright (c) 2006- by M. Nyffenegger <matthias.nyffenegger[AT]bluewin.ch> + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/sched.h> +#include <linux/pci.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include "log.h" +#include "saa7146audio.h" +#include "aw8.h" + +/* From Takashi Iwai's "Writing an ALSA Driver" developer guide */ +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; + +/* + * Module info + */ +MODULE_AUTHOR("Matthias Nyffenegger matthias.nyffenegger[AT]bluewin.ch"); +MODULE_DESCRIPTION(AW_MODULE_DESCRIPTION); +MODULE_LICENSE("GPL"); +MODULE_VERSION("0:0.2"); + +/** + * Module parameter 'input_type': select capture input digital or analog + * (default) + */ +int input_type; +module_param(input_type, int, 0444); +MODULE_PARM_DESC(input_type, "0=analog (default), 1=digital"); + +#define PCM_0 0 + +/** + * The ALSA 'chip' data structure. + */ +struct awalsa_t { + /* the alsa related stuff */ + struct snd_card *card; + struct snd_pcm *pcm_0; + struct snd_pcm_substream *pcm_in_substreams[MAX_IN_AUDIO_STREAMS]; + struct snd_pcm_substream *pcm_out_substreams[MAX_OUT_AUDIO_STREAMS]; + /* the SAA7146A stuff */ + struct saa7146audio chipaudio; + struct audio_stream *in_streams[MAX_IN_AUDIO_STREAMS]; + struct audio_stream *out_streams[MAX_OUT_AUDIO_STREAMS]; + /*/ the PCI stuff */ + struct pci_dev *pci; + int irq; + /* other useful things */ + spinlock_t isr_lock; +}; + +static int snd_aw_substream_open(struct snd_pcm_substream *substream, int in); +static int snd_aw_substream_close(struct snd_pcm_substream *substream, int in); +static int snd_aw_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params); +static int snd_aw_pcm_hw_free(struct snd_pcm_substream *substream); +static int snd_aw_pcm_prepare_substream(struct snd_pcm_substream *substream, + int in); +static int snd_aw_pcm_trigger_substream(struct snd_pcm_substream *substream, + int cmd, int in); +static snd_pcm_uframes_t snd_aw_pcm_pointer_substream( + struct snd_pcm_substream *substream, + int in); + +/** + * + * ALSA capture stream definitions and callbacks + * + */ +static struct snd_pcm_hardware snd_aw_capture_hw = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID), + .formats = (SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_S32_LE | + SNDRV_PCM_FMTBIT_S32_BE), + .rates = AW_PCM_RATE, + .rate_min = AW_FS_MIN, + .rate_max = AW_FS_MAX, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 16384, + .period_bytes_min = 64, /* HW restriction, see SAA7146A spec p.24 */ + .period_bytes_max = 8192, + .periods_min = 2, + .periods_max = 2, +}; + +/** + * ALSA capture stream open callback. + */ +static int snd_aw_capture_open(struct snd_pcm_substream *substream) +{ + return snd_aw_substream_open(substream, 1); +} + +/** + * ALSA capture stream close callback. + */ +static int snd_aw_capture_close(struct snd_pcm_substream *substream) +{ + return snd_aw_substream_close(substream, 1); +} + +/** + * Module scoped helper: converts alsa to Audiowerk capture sample format. + */ +static enum aw_formats alsa_2_aw_capture_format(snd_pcm_format_t fmt) +{ + switch (fmt) { + case SNDRV_PCM_FORMAT_S16_LE: + return fmt_s16_2le; + case SNDRV_PCM_FORMAT_S16_BE: + return fmt_s16_2be; + case SNDRV_PCM_FORMAT_S32_LE: + return fmt_s18_4le; + case SNDRV_PCM_FORMAT_S32_BE: + return fmt_s18_4be; + default: + LOG_ERROR("Unsupported capture format %d", fmt); + return fmt_s16_2le; + } +} + +/** + * ALSA capture stream prepare callback. + * Be careful that this callback will be called many times at each set up. + */ +static int snd_aw_pcm_prepare_capture(struct snd_pcm_substream *substream) +{ + return snd_aw_pcm_prepare_substream(substream, 1); +} + +/** + * ALSA capture stream trigger callback. [atomic]. + */ +static int snd_aw_pcm_trigger_capture(struct snd_pcm_substream *substream, + int cmd) +{ + return snd_aw_pcm_trigger_substream(substream, cmd, 1); +} + +/** + * ALSA capture stream hw-buffer pointer callback. [atomic] + */ +static snd_pcm_uframes_t snd_aw_pcm_pointer_capture( + struct snd_pcm_substream *substream) +{ + return snd_aw_pcm_pointer_substream(substream, 1); +} + +/** + * ALSA capture stream callback operations. + */ +static struct snd_pcm_ops snd_aw_capture_ops = { + .open = snd_aw_capture_open, + .close = snd_aw_capture_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_aw_pcm_hw_params, + .hw_free = snd_aw_pcm_hw_free, + .prepare = snd_aw_pcm_prepare_capture, + .trigger = snd_aw_pcm_trigger_capture, + .pointer = snd_aw_pcm_pointer_capture, +}; + +/** + * + * ALSA playback stream definitions and callbacks + * + */ +static struct snd_pcm_hardware snd_aw_playback_hw = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID), + .formats = (SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_S32_LE | + SNDRV_PCM_FMTBIT_S32_BE), + .rates = AW_PCM_RATE, + .rate_min = AW_FS_MIN, + .rate_max = AW_FS_MAX, + .channels_min = 2, + .channels_max = AW_MAX_PLAYBACK_CHANNELS_PER_STREAM, + .buffer_bytes_max = 16384, + .period_bytes_min = 64, /* HW restriction, see SAA7146A spec p.24 */ + .period_bytes_max = 8192, + .periods_min = 2, + .periods_max = 2, +}; + +/** + * ALSA playback stream open callback. + */ +static int snd_aw_playback_open(struct snd_pcm_substream *substream) +{ + return snd_aw_substream_open(substream, 0); +} + +/** + * ALSA playback stream close callback. + */ +static int snd_aw_playback_close(struct snd_pcm_substream *substream) +{ + return snd_aw_substream_close(substream, 0); +} + +/** + * Module scoped helper: converts alsa to Audiowerk playback sample format. + */ +static enum aw_formats alsa_2_aw_playback_format(snd_pcm_format_t fmt) +{ + switch (fmt) { + case SNDRV_PCM_FORMAT_S16_LE: + return fmt_s16_2le; + case SNDRV_PCM_FORMAT_S16_BE: + return fmt_s16_2be; + case SNDRV_PCM_FORMAT_S32_LE: + return fmt_s20_4le; + case SNDRV_PCM_FORMAT_S32_BE: + return fmt_s20_4be; + default: + LOG_ERROR("Unsupported playback format %d", fmt); + return fmt_s16_2le; + } +} + +/** + * ALSA playback stream prepare callback. + * Be careful that this callback will be called many times at each set up. + */ +static int snd_aw_pcm_prepare_playback(struct snd_pcm_substream *substream) +{ + return snd_aw_pcm_prepare_substream(substream, 0); +} + +/** + * ALSA playback stream trigger callback. [atomic]. + */ +static int snd_aw_pcm_trigger_playback(struct snd_pcm_substream *substream, + int cmd) +{ + return snd_aw_pcm_trigger_substream(substream, cmd, 0); +} + +/** + * ALSA playback stream hw-buffer pointer callback. [atomic]. + */ +static snd_pcm_uframes_t snd_aw_pcm_pointer_playback( + struct snd_pcm_substream *substream) +{ + return snd_aw_pcm_pointer_substream(substream, 0); +} + +/** + * ALSA playback stream callback operations. + */ +static struct snd_pcm_ops snd_aw_playback_ops = { + .open = snd_aw_playback_open, + .close = snd_aw_playback_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_aw_pcm_hw_params, + .hw_free = snd_aw_pcm_hw_free, + .prepare = snd_aw_pcm_prepare_playback, + .trigger = snd_aw_pcm_trigger_playback, + .pointer = snd_aw_pcm_pointer_playback, +}; + +static int snd_aw_substream_open(struct snd_pcm_substream *substream, int in) +{ + struct awalsa_t *awalsa = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + int device = substream->pcm->device; + struct snd_pcm_substream **alsa_pcm_substreams = NULL; + + if (device != PCM_0) { + LOG_ERROR("Invalid device index=%d", device); + return -EINVAL; + } + alsa_pcm_substreams = + (in ? awalsa->pcm_in_substreams : awalsa->pcm_out_substreams); + if (alsa_pcm_substreams[substream->number] != NULL) { + LOG_WARN("%s substream %d already open", + (in ? "Capture" : "Playback"), substream->number); + return -EAGAIN; + } + /* Remember substream for interrupt handler where we are given the chip + instance only. */ + alsa_pcm_substreams[substream->number] = substream; + runtime->hw = (in ? snd_aw_capture_hw : snd_aw_playback_hw); + return 0; +} + +static int snd_aw_substream_close(struct snd_pcm_substream *substream, int in) +{ + struct awalsa_t *awalsa = snd_pcm_substream_chip(substream); + int device = substream->pcm->device; + struct snd_pcm_substream **alsa_pcm_substreams = NULL; + struct audio_stream **saa7146_streams = NULL; + + if (device != PCM_0) { + LOG_ERROR("Invalid device index=%d", device); + return -EINVAL; + } + alsa_pcm_substreams = + (in ? awalsa->pcm_in_substreams : awalsa->pcm_out_substreams); + saa7146_streams = (in ? awalsa->in_streams : awalsa->out_streams); + alsa_pcm_substreams[substream->number] = NULL; + if (saa7146_streams[substream->number] != NULL) { + saa7146_stream_unprepare(&awalsa->chipaudio, + saa7146_streams[substream->number]); + } + return 0; +} + +/** + * ALSA stream hw_params callback. + */ +static int snd_aw_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + int err = 0; + int device = substream->pcm->device; + + if (device != PCM_0) { + LOG_ERROR("Invalid device index=%d", device); + return -EINVAL; + } + err = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + if (err < 0) { + LOG_ERROR("snd_pcm_lib_malloc_pages ERROR=%d", err); + return err; + } + return 0; +} + +/** + * ALSA stream hw_free callback. + */ +static int snd_aw_pcm_hw_free(struct snd_pcm_substream *substream) +{ + int err = 0; + int device = substream->pcm->device; + + if (device != PCM_0) { + LOG_ERROR("Invalid device index=%d", device); + return -EINVAL; + } + err = snd_pcm_lib_free_pages(substream); + if (err < 0) { + LOG_ERROR("snd_pcm_lib_free_pages ERROR=%d", err); + return err; + } + return 0; +} + +static int snd_aw_pcm_prepare_substream(struct snd_pcm_substream *substream, + int in) +{ + struct awalsa_t *awalsa = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct audio_stream **saa7146_streams = NULL; + struct audio_stream *stream = NULL; + int device = substream->pcm->device; + enum aw_formats fmt = fmt_s16_2le; + + if (device != PCM_0) { + LOG_ERROR("Invalid device index=%d", device); + return -EINVAL; + } + fmt = (in ? alsa_2_aw_capture_format(runtime->format) : + alsa_2_aw_playback_format(runtime->format)); + saa7146_streams = (in ? awalsa->in_streams : awalsa->out_streams); + /* as this function may be called many times we must clean up first */ + if (saa7146_streams[substream->number] != NULL) { + saa7146_stream_unprepare(&awalsa->chipaudio, + saa7146_streams[substream->number]); + } + /* try to prepare a new stream with the given parameters */ + if (in) { + stream = saa7146_stream_prepare_capture(&awalsa->chipaudio, + substream->number, + runtime->dma_addr, /* stream buffer phys. base addr. */ + 2 * snd_pcm_lib_period_bytes(substream), /* buf size */ + runtime->channels, + ((fmt == fmt_s16_2le || fmt == fmt_s16_2be) ? 2 : 4), + ((fmt == fmt_s16_2le || fmt == fmt_s18_4le) ? le : be)); + } else { + stream = saa7146_stream_prepare_playback(&awalsa->chipaudio, + substream->number, + runtime->dma_addr, /* stream buffer phys. base addr */ + 2 * snd_pcm_lib_period_bytes(substream), /* buf size */ + runtime->channels, + ((fmt == fmt_s16_2le || fmt == fmt_s16_2be) ? 2 : 4), + ((fmt == fmt_s16_2le || fmt == fmt_s20_4le) ? le : be)); + } + if (stream == NULL) + return -EINVAL; + saa7146_streams[substream->number] = stream; + if (aw8_set_samplerate(&awalsa->chipaudio.chipi2s.chip, runtime->rate) + != 0) + return -EINVAL; + return 0; +} + +static int snd_aw_pcm_trigger_substream(struct snd_pcm_substream *substream, + int cmd, int in) +{ + struct awalsa_t *awalsa = snd_pcm_substream_chip(substream); + int device = substream->pcm->device; + struct audio_stream **saa7146_streams = NULL; + + if (device != PCM_0) { + LOG_ERROR("Invalid device index=%d", device); + return -EINVAL; + } + saa7146_streams = (in ? awalsa->in_streams : awalsa->out_streams); + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + saa7146_stream_start(&awalsa->chipaudio, + saa7146_streams[substream->number]); + break; + case SNDRV_PCM_TRIGGER_STOP: + saa7146_stream_stop(&awalsa->chipaudio, + saa7146_streams[substream->number]); + break; + default: + LOG_ERROR("Unknown trigger command=%d", cmd); + return -EINVAL; + } + return 0; +} + +static snd_pcm_uframes_t snd_aw_pcm_pointer_substream( + struct snd_pcm_substream *substream, + int in) +{ + struct awalsa_t *awalsa = snd_pcm_substream_chip(substream); + struct audio_stream **saa7146_streams = NULL; + unsigned long hw_ptr = 0; + snd_pcm_uframes_t frames; + int device = substream->pcm->device; + + if (device != PCM_0) { + LOG_ERROR("Invalid device index=%d", device); + return -EINVAL; + } + saa7146_streams = (in ? awalsa->in_streams : awalsa->out_streams); + hw_ptr = saa7146_stream_get_hw_pointer(&awalsa->chipaudio, + saa7146_streams[substream->number]); + frames = bytes_to_frames(substream->runtime, + hw_ptr - (size_t)(substream->runtime->dma_addr)); + return frames; +} + +/** + * + * ALSA device initialization + * + */ + +/** + * ALSA Setup 1 PCM device: + * AW8 substreams: 1 capture /2 playback (1 analog/digital, 1 analog) + * AW2 substreams: 1 capture /2 playback (1 analog, 1 digital) + */ +static int __devinit snd_aw_new_pcm(struct awalsa_t *awalsa) +{ + struct snd_pcm *pcm; + int err; + int pages = 0; + + /* create the pcm device */ + err = snd_pcm_new(awalsa->card, "SAA7146A", /* device ID string */ + PCM_0, /* device index */ + 2, /* playback substreams AW2/8 */ + 1, /* capture substreams AW2/8 */ + &pcm); + if (err != 0) { + LOG_ERROR("snd_pcm_new ERROR=%d", err); + return err; + } + pcm->private_data = awalsa; + strcpy(pcm->name, "Multichannel Capture/Playback"); + awalsa->pcm_0 = pcm; + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_aw_playback_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_aw_capture_ops); + LOG_INFO("PCM device [Multichannel Capture/Playback]" + " successfully created"); + /* Pre-allocation 128k of dma buffer */ + pages = 1 << ((17 - PAGE_SHIFT > 0) ? 17 - PAGE_SHIFT : 0); + err = snd_pcm_lib_preallocate_pages_for_all(pcm, + SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(awalsa->pci), + pages * PAGE_SIZE, + pages * PAGE_SIZE); + if (err != 0) { + LOG_ERROR("snd_pcm_lib_preallocate_pages_for_all" + " ERROR=%d", err); + return err; + } + return 0; +} + +/** + * ALSA Interrupt Service Routine. + */ +static irqreturn_t snd_aw_interrupt(int irq, void *dev_id) +{ + struct awalsa_t *awalsa = dev_id; + uint32_t isr = 0; + + /* lock isr - block other interrupts */ + spin_lock(awalsa->isr_lock); + isr = saa7146_read(&awalsa->chipaudio.chipi2s.chip, ISR); + /* was it one of SAA7146A (remember we might share irq line ..) */ + if (!isr) { + spin_unlock(awalsa->isr_lock); + return IRQ_NONE; + } + /* clear all interrupts */ + saa7146_write(&awalsa->chipaudio.chipi2s.chip, ISR, isr); + if (isr & A1_in && awalsa->pcm_in_substreams[0] != NULL) + snd_pcm_period_elapsed(awalsa->pcm_in_substreams[0]); + if (isr & A1_out && awalsa->pcm_out_substreams[0] != NULL) + snd_pcm_period_elapsed(awalsa->pcm_out_substreams[0]); + if (isr & A2_out && awalsa->pcm_out_substreams[1] != NULL) + snd_pcm_period_elapsed(awalsa->pcm_out_substreams[1]); + spin_unlock(awalsa->isr_lock); + return IRQ_HANDLED; +} + +/** + * ALSA chip-specific destructor. + * (see "PCI Resource Managements") + */ +static int snd_aw_free(struct awalsa_t *awalsa) +{ + struct saa7146i2s *chipi2s = &awalsa->chipaudio.chipi2s; + + /* disable hardware here if any */ + aw_exit(chipi2s); + if (awalsa->irq >= 0) + free_irq(awalsa->irq, (void *)awalsa); + if (chipi2s->chip.iobase_virt) + iounmap(chipi2s->chip.iobase_virt); + pci_release_regions(awalsa->pci); + pci_disable_device(awalsa->pci); + kfree(awalsa); + return 0; +} + +/** + * ALSA component-destructor. + * (see "Management of Cards and Components") + */ +static int snd_aw_dev_free(struct snd_device *device) +{ + snd_aw_free(device->device_data); + return 0; +} + +/** + * ALSA chip-specific constructor. + * (see "Management of Cards and Components") + */ +static int __devinit snd_aw_create(struct snd_card *card, + struct pci_dev *pci, + struct awalsa_t **rchip) +{ + static struct snd_device_ops ops = { + .dev_free = snd_aw_dev_free, + }; + int err; + struct awalsa_t *awalsa; + *rchip = NULL; + + /* Initialize the PCI entry: This function switches the device 'on'. + In some cases, IRQ and I/O range is assigned. */ + err = pci_enable_device(pci); + if (err != 0) { + LOG_ERROR("pci_enable_device ERROR=%d", err); + return err; + } + pci_set_master(pci); + err = pci_set_dma_mask(pci, DMA_32BIT_MASK); + if (err != 0) { + LOG_ERROR("pci_set_dma_mask ERROR=%d", err); + goto snd_aw_create_exit0; + } + err = pci_set_consistent_dma_mask(pci, DMA_32BIT_MASK); + if (err != 0) { + LOG_ERROR("pci_set_consistent_dma_mask ERROR=%d", err); + goto snd_aw_create_exit0; + } + awalsa = kzalloc(sizeof(*awalsa), GFP_KERNEL); + if (awalsa == NULL) { + LOG_ERROR("Could not allocate memory"); + err = -ENOMEM; + goto snd_aw_create_exit0; + } + awalsa->card = card; + awalsa->pci = pci; + awalsa->irq = -1; + spin_lock_init(&awalsa->isr_lock); + /* (1) PCI resource allocation */ + err = pci_request_regions(pci, "SAA7146A"); + if (err != 0) { + LOG_ERROR("pci_request_regions ERROR=%d", err); + goto snd_aw_create_exit1; + } + /* Get the device's registers physical base address. The device register + space is not available for access by the driver now; it has to be + mapped to a virtual memory space first. */ + awalsa->chipaudio.chipi2s.chip.iobase_phys = pci_resource_start(pci, 0); + awalsa->chipaudio.chipi2s.chip.iolen = pci_resource_len(pci, 0); + /* Map device registers to virtual memory. 'nocache' means that we don't + want the device's registers to be read-cached by the CPU (e.g. as a + result of compiler optimizations). This is important since the CPU + does not necessarily know when a register value has changed (unless + this was by a write action of the driver). */ + awalsa->chipaudio.chipi2s.chip.iobase_virt = + ioremap_nocache(awalsa->chipaudio.chipi2s.chip.iobase_phys, + awalsa->chipaudio.chipi2s.chip.iolen); + err = request_irq(pci->irq, snd_aw_interrupt, IRQF_DISABLED|IRQF_SHARED, + "SAA7146A", (void *)awalsa); + if (err != 0) { + LOG_ERROR("Cannot grab irq %d, ERROR=%d", pci->irq, err); + err = -EBUSY; + goto snd_aw_create_exit2; + } + awalsa->irq = pci->irq; + /* (2) initialization of the chip hardware */ + if (aw_init(&awalsa->chipaudio.chipi2s, input_type) != 0) + goto snd_aw_create_exit2; + /* (3) create alsa device */ + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, awalsa, &ops); + if (err < 0) { + LOG_ERROR("snd_device_new ERROR=%d", err); + goto snd_aw_create_exit2; + } + snd_card_set_dev(card, &pci->dev); + *rchip = awalsa; + return 0; +snd_aw_create_exit2: + snd_aw_free(awalsa); + return err; +snd_aw_create_exit1: + kfree(awalsa); +snd_aw_create_exit0: + pci_disable_device(pci); + return err; +} + +/** + * ALSA device constructor -- see "Constructor" sub-section in Takashi Iwai's + * "Writing an ALSA Driver" developer guide. + */ +static int __devinit snd_aw_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ + static int dev; + int err; + struct snd_card *card; + struct awalsa_t *awalsa; + + /* (1) */ + if (dev >= SNDRV_CARDS) { + LOG_ERROR("No device available ERROR=%d", -ENODEV); + return -ENODEV; + } + if (!enable[dev]) { + dev++; + LOG_ERROR("No device entry ERROR=%d", -ENOENT); + return -ENOENT; + } + /* (2) */ + card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); + if (card == NULL) { + LOG_ERROR("snd_card_new ERROR=%d", -ENOMEM); + return -ENOMEM; + } + /* (3) */ + err = snd_aw_create(card, pci, &awalsa); + if (err < 0) { + LOG_ERROR("snd_aw_create ERROR=%d", err); + goto snd_aw_probe_exit0; + } + /* (4) */ + strcpy(card->driver, "SAA7146A"); + strcpy(card->shortname, AW_CARD_SHORT_NAME); + sprintf(card->longname, + "%s at 0x%lx [%i] irq %i", + card->shortname, + awalsa->chipaudio.chipi2s.chip.iobase_phys, + awalsa->chipaudio.chipi2s.chip.iolen, + awalsa->irq); + LOG_INFO("card-info[%s]", card->longname); + /* (5) */ + err = snd_aw_new_pcm(awalsa); + if (err != 0) { + LOG_ERROR("snd_aw_new_pcm ERROR=%d", err); + goto snd_aw_probe_exit0; + } + /* (6) */ + err = snd_card_register(card); + if (err < 0) { + LOG_ERROR("snd_card_register ERROR=%d", err); + goto snd_aw_probe_exit0; + } + /* (7) */ + pci_set_drvdata(pci, card); + dev++; + return 0; +snd_aw_probe_exit0: + snd_card_free(card); + return err; +} + +/** + * ALSA device destructor -- see "Destructor" sub-section + */ +static void __devexit snd_aw_remove(struct pci_dev *pci) +{ + snd_card_free(pci_get_drvdata(pci)); + pci_set_drvdata(pci, NULL); +} + +/** + * + * PCI specific stuff + * + */ + +/** + * Organisation of EEPROM of the SAA7146A: + * The data of the subsystem ID and the subsystem vendor ID is organized in + * the EEPROM in the following order: + * + * EE-Addr Value Configuration Space Addr. + * --------------------------------------------------------------------- + * 00 Subsys ID (high byte) 2C, bits: 31 - 24 + * 01 Subsys ID (low byte) 2C, bits: 23 - 16 + * 02 Subsys Vendor ID (high byte) 2C, bits: 15 - 8 + * 03 Subsys Vendor ID (low byte) 2C, bits: 7 - 0 + * 04 Max_Lat 3C, bits: 31 - 24 + * 05 Min_Gnt 3C, bits: 23 - 16 + * + * For AW8 Subsys ID and Subsys Vendor ID are 0xFF on my card. + * For AW2 Subsys ID and Subsys Vendor ID are 0x00 on my card. + */ + +static struct pci_device_id snd_aw_ids[] = { + { PCI_VENDOR_ID_PHILIPS, PCI_DEVICE_ID_PHILIPS_SAA7146, + PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, + { 0, } +}; + +MODULE_DEVICE_TABLE(pci, snd_aw_ids); + +/** + * pci_driver definition. + */ +static struct pci_driver driver = { + .name = AW_DRIVER_NAME, + .id_table = snd_aw_ids, + .probe = snd_aw_probe, + .remove = __devexit_p(snd_aw_remove), +}; + +/** + * initialization of the module. + */ +static int __init alsa_card_aw_init(void) +{ + int err; + + err = pci_register_driver(&driver); + if (err != 0) { + LOG_ERROR("pci_register_driver ERROR=%d", err); + return err; + } + return 0; +} + +/** + * clean up the module. + */ +static void __exit alsa_card_aw_exit(void) +{ + pci_unregister_driver(&driver); +} + +module_init(alsa_card_aw_init) +module_exit(alsa_card_aw_exit) diff -uprN ../alsa-driver-1.0.17/alsa-kernel/pci/saa7146/aw8.c ../alsa-driver-1.0.17.mod/alsa-kernel/pci/saa7146/aw8.c --- ../alsa-driver-1.0.17/alsa-kernel/pci/saa7146/aw8.c 1970-01-01 01:00:00.000000000 +0100 +++ ../alsa-driver-1.0.17.mod/alsa-kernel/pci/saa7146/aw8.c 2008-10-22 23:34:05.000000000 +0200 @@ -0,0 +1,185 @@ +/* + * Emagic Audiowerk specific functions + * Copyright (c) 2006- by M. Nyffenegger <matthias.nyffenegger[AT]bluewin.ch> + * Based on GPLed Emagic Audiowerk8 Windows driver provided by Martijn Sipkema. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ +#include <linux/types.h> +#include <linux/sched.h> +#include "log.h" +#include "saa7146.h" +#include "saa7146i2s.h" +#include "saa7146i2c.h" +#include "aw8.h" + +#define MAX(a, b) ((a) > (b) ? (a) : (b)) +#define SLEEP_DIGIN 100 /* wait [ms] after switching input type adc/daio */ +#define GPIO_0 GPIO_CTRL +#define MASK_GPIO_HI 0x50 +#define MASK_GPIO_LO 0x40 +#define SLEEP_PLL 500 /* wait [ms] after setting new fs */ +#define PLL_DEVICE_ADDR 0x62 /* I2C device address of AW8 PLL chip */ +#define PLL_FREF 1000 /* reference frequency for PLL */ + +/** + * TODO: Description + */ +enum aw_input_types {analog, digital}; + +/* forward declarations */ +static int aw8_device_init(struct saa7146i2s *chipi2s); +static void switch_input(struct saa7146reg *chip, enum aw_input_types type); + +/** + * see aw8.h + */ +int aw_init(struct saa7146i2s *chipi2s, int input_type) +{ + int err = 0; + + saa7146_write(&chipi2s->chip, IER, 0); /* disable all interrupts */ + saa7146_write(&chipi2s->chip, MC1, MRST_N<<16); /* reset SAA7146 */ + /* setup PCI arbitration ctrl (see SAA7146 spec. page 31 for details): + o 8DW burst for all FIFOs + o 8DW (empty DWs in FIFO) threshold for OUT-FIFOs + o 8DW (valid DWs in FIFO) threshold for IN-FIFOs + 0x0e0e0e0e => xxx0 1110 xxx0 1110 xxx0 1110 xxx0 1110 */ + saa7146_write(&chipi2s->chip, PCI_BT_A, 0x0e0e0e0e); + /* enable SAA7146 audio pins */ + saa7146_write(&chipi2s->chip, MC1, (EAP<<16 | EAP)); + /* setup audio clock source (from EMagic GPL driver) */ + saa7146_write(&chipi2s->chip, ACON2, ((0UL<<27) | /* A1_CLKSRC: BCLK1 */ + (1UL<<22) | /* A2_CLKSRC: BCLK1 */ + (0UL<<21) | /* INVERT_BCLK1 */ + (0UL<<20) | /* INVERT_BCLK2 */ + (1UL<<19) | /* BCLK1_OEN: input */ + (0UL<<18))); /* BCLK2_OEN: output */ + /* init ACON1: WS1-4 to output active low */ + chipi2s->acon1 = 0x0000cccc; + saa7146_write(&chipi2s->chip, ACON1, chipi2s->acon1); + if (input_type) + aw_switch_input_daio(&chipi2s->chip); + else + aw_switch_input_adc(&chipi2s->chip); + if (saa7146_i2s_init_audio_interface(chipi2s, a1, 4, 8) != 0 + || saa7146_i2s_init_audio_interface(chipi2s, a2, 4, 8) != 0) + return -1; + err = aw8_device_init(chipi2s); + /* enable interrupts */ + saa7146_write(&chipi2s->chip, IER, (A1_out | A2_out | A1_in)); + return err; +} + +/** + * see aw8.h + */ +void aw_exit(struct saa7146i2s *chipi2s) +{ + saa7146_write(&chipi2s->chip, IER, 0); /* disable all interrupts */ + saa7146_write(&chipi2s->chip, MC1, (MRST_N<<16)); /* reset SAA7146 */ +} + +/** + * see aw8.h + */ +void aw_switch_input_adc(struct saa7146reg *chip) +{ + switch_input(chip, analog); +} + +/** + * see aw8.h + */ +void aw_switch_input_daio(struct saa7146reg *chip) +{ + switch_input(chip, digital); +} + +/** + * see aw8.h + * Details regarding implementation see project wiki. + */ +int aw8_set_samplerate(struct saa7146reg *chip, unsigned int fs) +{ + uint32_t counter = 0; + uint8_t fs_prg_seq[4]; + + fs = fs > AW_FS_MAX ? AW_FS_MAX : fs; + fs = fs < AW_FS_MIN ? AW_FS_MIN : fs; + /* init TSA6060 counter */ + counter = (256 * fs * 4) / PLL_FREF; + fs_prg_seq[0] = counter << 1; + fs_prg_seq[1] = counter >> 7; + fs_prg_seq[2] = (counter >> 15 & 0x03) | 0x18; + fs_prg_seq[3] = 0; + if (saa7146_i2c_write(chip, bps_10k, PLL_DEVICE_ADDR, NULL, 0, + fs_prg_seq, 4) < 4) + return -1; + return 0; +} + +/** + * A1: A2: + * analog in: WS0/SD4 + * analog/digital out: WS1/SD0 analog out: WS3/SD1 + * analog out: WS2/SD2 analog out: WS4/SD3 + */ +static int aw8_device_init(struct saa7146i2s *chipi2s) +{ + if (saa7146_i2s_init_device(chipi2s, a1, in, ws0, sd4_i_a1) != 0 + || saa7146_i2s_init_device(chipi2s, a1, out, ws1, sd0_o_a1) != 0 + || saa7146_i2s_init_device(chipi2s, a1, out, ws2, sd2_io_ax) + != 0 + || saa7146_i2s_init_device(chipi2s, a2, out, ws3, sd1_io_ax) + != 0 + || saa7146_i2s_init_device(chipi2s, a2, out, ws4, sd3_io_ax) + != 0) + return -1; + return 0; +} + +/** + * TODO: Description + */ +static void switch_input(struct saa7146reg *chip, enum aw_input_types type) +{ + /* From EMagic driver... + // Switch SAA7146 to safe mode + setreg(chip, ACON1, 0L); + setreg(chip, ACON2, (12L<<27) | // A1 bitclock == ACLK/384 + (12L<<22) | // A2 bitclock == ACLK/384 + ( 1L<<19)); // BCLK1 is input + setreg(chip, MC1, 0x7fff0000); // Disable audio i/f + setreg(chip, A_TIME_SLOT1, (1L<<TSL_EOS)); + setreg(chip, A_TIME_SLOT2, (1L<<TSL_EOS)); + */ + if (type == analog) { + saa7146_write(chip, GPIO_0, MASK_GPIO_HI); + /* wait until relais is switched and PLL is locked */ + set_current_state(TASK_UNINTERRUPTIBLE); + schedule_timeout(MAX(1, SLEEP_PLL*HZ/1000)); + } else { + saa7146_write(chip, GPIO_0, MASK_GPIO_LO); + /* wait until relais is switched and digital-in is synced. */ + set_current_state(TASK_UNINTERRUPTIBLE); + schedule_timeout(MAX(1, SLEEP_DIGIN*HZ/1000)); + } + /* From EMagic driver... + SetReg(MC1, 0x7fff0200, true); // Enable audio i/f + */ +} + diff -uprN ../alsa-driver-1.0.17/alsa-kernel/pci/saa7146/aw8.h ../alsa-driver-1.0.17.mod/alsa-kernel/pci/saa7146/aw8.h --- ../alsa-driver-1.0.17/alsa-kernel/pci/saa7146/aw8.h 1970-01-01 01:00:00.000000000 +0100 +++ ../alsa-driver-1.0.17.mod/alsa-kernel/pci/saa7146/aw8.h 2008-10-22 23:34:05.000000000 +0200 @@ -0,0 +1,80 @@ +/* + * Emagic Audiowerk specific functions + * Copyright (c) 2006- by M. Nyffenegger <matthias.nyffenegger[AT]bluewin.ch> + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ +#ifndef AUDIOWERK_H_ +#define AUDIOWERK_H_ + +#include <linux/types.h> +#include "saa7146.h" +#include "saa7146i2s.h" + +# define AW_MODULE_DESCRIPTION "Emagic Audiowerk 8 Driver" +# define AW_CARD_SHORT_NAME "Audiowerk 8" +# define AW_DRIVER_NAME "SAA7146A Audiowerk 8" +# define AW_PCM_RATE SNDRV_PCM_RATE_CONTINUOUS +# define AW_FS_MIN 25000 /* fs lower limit for DAC TDA1305 (ADC 18kHz) */ +# define AW_FS_MAX 48000 /* fs upper limit for DAC TDA1305 (ADC 50kHz) */ +# define AW_MAX_PLAYBACK_CHANNELS_PER_STREAM 4 + +/** + * Audiwerk sampling formats. + */ +enum aw_formats {fmt_s16_2le, fmt_s16_2be, fmt_s18_4le, fmt_s18_4be, + fmt_s20_4le, fmt_s20_4be}; + +/** + * TODO: description + */ +int aw_init(struct saa7146i2s *chipi2s, int input_type); + +/** + * TODO: description + */ +void aw_exit(struct saa7146i2s *chipi2s); + +/** + * Switches the Audiowerk input to analog. + * There are two sysclock generators, switched via a relais on GPIO-0: + * o TSA 6060 PLL for analog input + * o TDA 1315 digital input + * @param chip Used for hardware access over PCI bus. + */ +void aw_switch_input_adc(struct saa7146reg *chip); + +/** + * Switches the Audiowerk input to digital. + * There are two sysclock generators, switched via a relais on GPIO-0: + * o TSA 6060 PLL for analog input + * o TDA 1315 digital input + * @param chip Used for hardware access over PCI bus. + */ +void aw_switch_input_daio(struct saa7146reg *chip); + +/** + * Set sample rate on AW8. + * AW8 supports continuous fs shared by all substreams -> last call to + * aw8_set_samplerate() wins. + * @param chip Used for hardware access over PCI bus. + * @param fs sampling frequency (25-48kHz). + * @return 0 in case of success or -1 in case of failure. + */ +int aw8_set_samplerate(struct saa7146reg *chip, unsigned int fs); + +#endif /*AUDIOWERK_H_*/

2 1

[alsa-devel] [PATCH 1/3] saa7146: Emagic Audiowerk8 low-level ALSA driver
by Matthias Nyffenegger 23 Oct '08

23 Oct '08

From: Matthias Nyffenegger <matthias.nyffenegger(a)bluewin.ch> Low-level ALSA driver for Emagic Audiowerk8 sound card. Project page: http://sourceforge.net/projects/aw8-alsa Built and tested with Vanilla 2.6.25.16 Signed-off-by: Matthias Nyffenegger <matthias.nyffenegger(a)bluewin.ch> --- This is a request for submission to ALSA-tree. diff -uprN ../alsa-driver-1.0.17/alsa-kernel/pci/saa7146/log.h ../alsa-driver-1.0.17.mod/alsa-kernel/pci/saa7146/log.h --- ../alsa-driver-1.0.17/alsa-kernel/pci/saa7146/log.h 1970-01-01 01:00:00.000000000 +0100 +++ ../alsa-driver-1.0.17.mod/alsa-kernel/pci/saa7146/log.h 2008-10-22 23:34:05.000000000 +0200 @@ -0,0 +1,43 @@ +/* + * Kernel log convenience macros + * Copyright (c) 2006- by M. Nyffenegger <matthias.nyffenegger[AT]bluewin.ch> + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ +#ifndef LOG_H_ +#define LOG_H_ +#define LOG_ENABLE +#ifndef SAA7146_SUBSYS_LOG_TAG +#define SAA7146_SUBSYS_LOG_TAG "SAA7146" +#endif +#ifdef LOG_ENABLE +# define LOG_ERROR(fmt, args...) \ + printk(KERN_ERR SAA7146_SUBSYS_LOG_TAG ": %s:%d:" fmt "\n", \ + __func__, __LINE__, ## args); +# define LOG_WARN(fmt, args...) \ + printk(KERN_WARNING SAA7146_SUBSYS_LOG_TAG ": %s:%d:" fmt "\n", \ + __func__, __LINE__, ## args); +# define LOG_INFO(fmt, args...) \ + printk(KERN_INFO SAA7146_SUBSYS_LOG_TAG ": %s:%d:" fmt "\n", \ + __func__, __LINE__, ## args); +#else +# define LOG_ERROR(fmt, args...) +# define LOG_WARN(fmt, args...) +# define LOG_INFO(fmt, args...) +#endif /* LOG_ENABLE */ + +#endif /*LOG_H_*/ diff -uprN ../alsa-driver-1.0.17/alsa-kernel/pci/saa7146/saa7146audio.c ../alsa-driver-1.0.17.mod/alsa-kernel/pci/saa7146/saa7146audio.c --- ../alsa-driver-1.0.17/alsa-kernel/pci/saa7146/saa7146audio.c 1970-01-01 01:00:00.000000000 +0100 +++ ../alsa-driver-1.0.17.mod/alsa-kernel/pci/saa7146/saa7146audio.c 2008-10-22 23:34:05.000000000 +0200 @@ -0,0 +1,343 @@ +/* + * SAA7146 audio stream abstraction layer + * Copyright (c) 2006- by M. Nyffenegger <matthias.nyffenegger[AT]bluewin.ch> + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ +#include <linux/types.h> +#include "log.h" +#include "saa7146.h" +#include "saa7146i2s.h" +#include "saa7146audio.h" + +/** + * Address maps for Audio DMA Control PCI_ADP, BaseAx_x, ProtAx_x, PageAx_x and + * MC1 Audio Transfer Enable: 1st dim. index corresponds to + * enum audio_interfaces's, 2nd to enum directions's. + * TODO: decouple from saa7146i2s.h by providing a function that makes a + * reliable mapping (independent of int values of enum members). + */ +static const int Base[2][2] = {{BaseA1_in, BaseA1_out}, + {BaseA2_in, BaseA2_out} }; +static const int Prot[2][2] = {{ProtA1_in, ProtA1_out}, + {ProtA2_in, ProtA2_out} }; +static const int Page[2][2] = {{PageA1_in, PageA1_out}, + {PageA2_in, PageA2_out} }; +static const int TR_E[2][2] = {{TR_E_A1_IN, TR_E_A1_OUT}, + {TR_E_A2_IN, TR_E_A2_OUT} }; +static const int PCI_ADP[2][2] = {{PCI_ADP2, PCI_ADP1}, + {PCI_ADP4, PCI_ADP3} }; + +/* forward declarations */ +static struct audio_stream *stream_prepare(struct saa7146audio *chipaudio, + unsigned int stream_nr, + unsigned long buffer_base_addr, + int buffer_size, + int channel_count, + int sample_length, + enum endian endian, + enum directions direction); +static int stream_check_resources(struct saa7146audio *chipaudio, + unsigned int stream_nr, + int channel_count, + int sample_length, + enum endian endian, + enum directions direction); +static int stream_init(struct saa7146audio *chipaudio, + struct audio_stream *stream, + unsigned int stream_nr, + unsigned long buffer_base_addr, + int buffer_size, + int channel_count, + int sample_length, + enum endian endian, + enum directions direction); +static int stream_setup_dma(struct saa7146audio *chipaudio, + struct audio_stream *audio_stream); +static int is_valid_period(int period); + +/** + * see saa7146audio.h + */ +struct audio_stream *saa7146_stream_prepare_capture( + struct saa7146audio *chipaudio, + unsigned int stream_nr, + unsigned long buffer_base_addr, + int buffer_size, + int channel_count, + int sample_length, + enum endian endian) +{ + return stream_prepare(chipaudio, stream_nr, buffer_base_addr, + buffer_size, channel_count, sample_length, endian, in); +} + +/** + * see saa7146audio.h + */ +struct audio_stream *saa7146_stream_prepare_playback( + struct saa7146audio *chipaudio, + unsigned int stream_nr, + unsigned long buffer_base_addr, + int buffer_size, + int channel_count, + int sample_length, + enum endian endian) +{ + return stream_prepare(chipaudio, stream_nr, buffer_base_addr, + buffer_size, channel_count, sample_length, endian, out); +} + +/** + * see saa7146audio.h + */ +void saa7146_stream_unprepare(struct saa7146audio *chipaudio, + struct audio_stream *stream) +{ + int i = 0; + struct i2s_device *device = NULL; + + if (stream != NULL) { + stream->number = -1; + stream->state = disabled; + stream->direction = in; + stream->channel_count = 0; + stream->sample_length = 0; + stream->endian = le; + stream->buffer_base_addr = 0; + stream->buffer_size = 0; + stream->audio_interface = NULL; + for (i = 0; i < MAX_I2S_DEVICES; i++) { + device = stream->i2s_devices[i]; + if (device != NULL) { + saa7146_i2s_disable_device(&chipaudio->chipi2s, + device); + stream->i2s_devices[i] = NULL; + } + } + } +} + +/** + * see saa7146audio.h + */ +void saa7146_stream_start(struct saa7146audio *chipaudio, + struct audio_stream *stream) +{ + int tr_e; + + if (stream == NULL) { + LOG_WARN("stream pointer is NULL"); + return; + } + /* enable DMA Ax IN/OUT */ + tr_e = TR_E[stream->audio_interface->id][stream->direction]; + saa7146_write(&chipaudio->chipi2s.chip, MC1, (tr_e<<16 | tr_e)); + saa7146_i2s_enable_audio_interface(&chipaudio->chipi2s, + stream->audio_interface); +} + +/** + * see saa7146audio.h + */ +void saa7146_stream_stop(struct saa7146audio *chipaudio, + struct audio_stream *stream) +{ + int tr_e; + + if (stream == NULL) { + LOG_WARN("stream pointer is NULL"); + return; + } + saa7146_i2s_disable_audio_interface(&chipaudio->chipi2s, + stream->audio_interface); + /* disable DMA Ax IN/OUT */ + tr_e = TR_E[stream->audio_interface->id][stream->direction]; + saa7146_write(&chipaudio->chipi2s.chip, MC1, (tr_e<<16)); +} + +/** + * see saa7146audio.h + */ +uint32_t saa7146_stream_get_hw_pointer(struct saa7146audio *chipaudio, + struct audio_stream *stream) +{ + uint32_t hw_ptr = 0; + + if (stream == NULL) { + LOG_WARN("stream pointer is NULL"); + return hw_ptr; + } + hw_ptr = saa7146_read(&chipaudio->chipi2s.chip, + PCI_ADP[stream->audio_interface->id][stream->direction]); + return hw_ptr; +} + +static struct audio_stream *stream_prepare(struct saa7146audio *chipaudio, + unsigned int stream_nr, + unsigned long buffer_base_addr, + int buffer_size, + int channel_count, + int sample_length, + enum endian endian, + enum directions direction) +{ + struct audio_stream *stream = NULL; + int max_streams = (direction == in ? + MAX_IN_AUDIO_STREAMS : MAX_OUT_AUDIO_STREAMS); + struct audio_stream *streams = (direction == in ? + chipaudio->in_streams : chipaudio->out_streams); + + if (stream_nr > max_streams || streams[stream_nr].state == enabled) { + LOG_WARN("Stream nr=%d not available", stream_nr); + return NULL; + } + stream = &streams[stream_nr]; + if (stream_check_resources(chipaudio, stream_nr, channel_count, + sample_length, endian, direction) != 0) + return NULL; + if (stream_init(chipaudio, stream, stream_nr, buffer_base_addr, + buffer_size, channel_count, sample_length, endian, direction) + != 0) + return NULL; + if (stream_setup_dma(chipaudio, stream) != 0) + return NULL; + return stream; +} + +/** + * Check if the requested resources (channels, sample-length, endian) are + * available. + */ +static int stream_check_resources(struct saa7146audio *chipaudio, + unsigned int stream_nr, + int channel_count, + int sample_length, + enum endian endian, + enum directions direction) +{ + int i = 0; + int available_channels = 0; + int available_frame_length = 0; + struct i2s_device *device = NULL; + struct audio_interface *ai = NULL; + + ai = &chipaudio->chipi2s.audio_interfaces[stream_nr]; + available_frame_length = ai->i2s_superframe_length; + for (i = 0; i < MAX_I2S_DEVICES; i++) { + device = ai->i2s_devices[i]; + if (device != NULL && device->direction == direction) { + if (device->state == disabled) + available_channels += 2; + else if (device->state == enabled && + device->direction == in && + device->endian != endian) { + LOG_WARN("Endian %s not available for capture" + " on audio-interface %s.", + (endian == le ? "le" : "be"), + ai->id == a1 ? "A1" : "A2"); + return -1; + } else { + available_frame_length -= + 2 * device->sample_length; + } + } + } + if (available_channels < channel_count || + available_frame_length < (channel_count * sample_length)) { + LOG_WARN("Not sufficient resources available."); + return -1; + } + return 0; +} + +static int stream_init(struct saa7146audio *chipaudio, + struct audio_stream *stream, + unsigned int stream_nr, + unsigned long buffer_base_addr, + int buffer_size, + int channel_count, + int sample_length, + enum endian endian, + enum directions direction) +{ + int i; + struct i2s_device *device = NULL; + + stream->number = stream_nr; + stream->state = enabled; + stream->direction = direction; + stream->channel_count = channel_count; + stream->sample_length = sample_length; + stream->endian = endian; + stream->buffer_base_addr = buffer_base_addr; + stream->buffer_size = buffer_size; + stream->audio_interface = + &chipaudio->chipi2s.audio_interfaces[stream_nr]; + for (i = 0; (i < MAX_I2S_DEVICES) && (channel_count > 0); i++) { + device = stream->audio_interface->i2s_devices[i]; + if (device != NULL && device->state == disabled && + device->direction == direction) { + if (saa7146_i2s_enable_device(&chipaudio->chipi2s, + device, + sample_length, + endian) != 0) + return -1; + /* use ws as index where device is stored in array */ + stream->i2s_devices[device->ws] = device; + channel_count -= 2; + } + } + return 0; +} + +static int stream_setup_dma(struct saa7146audio *chipaudio, + struct audio_stream *stream) +{ + int exp = 0; + int limit = 0; + enum audio_interfaces ai_nr = stream->audio_interface->id; + enum directions dir = stream->direction; + struct saa7146reg *chip = &chipaudio->chipi2s.chip; + + if (!is_valid_period(stream->buffer_size >> 1)) { + LOG_ERROR("Invalid buffer size %d", stream->buffer_size); + return -1; + } + /* find exponent for period, required to figure out the DMA IRQ limit */ + limit = stream->buffer_size >> 1; /* period is buffer_size/2 */ + for (exp = 0; limit > 0; limit >>= 1, exp++); + /* LimitAx_in = exp - 6 (minimum limit is 64bytes = 2^6) */ + limit = exp - 6; + saa7146_write(chip, Base[ai_nr][dir], stream->buffer_base_addr); + saa7146_write(chip, Prot[ai_nr][dir], + stream->buffer_base_addr + stream->buffer_size); + /* PageA1_in=0, MEA1_in=0, LimitA1_in=limit, PVA1_in=0 */ + saa7146_write(chip, Page[a1][dir], limit << 4); + return 0; +} + +/** + * Period size must be >= 64bytes <= 1Mb and a power of 2. + * This is a restriction imposed by the SAA7146 DMA capabilities: the period at + * which DMA IRQs are generated is 64*2^n where n={1-14}. + * @return 1 if the given period is valid. + */ +static int is_valid_period(int period) +{ + return (period >= 64) && (period <= (1 << 20)) && is_pow_of_2(period); +} diff -uprN ../alsa-driver-1.0.17/alsa-kernel/pci/saa7146/saa7146audio.h ../alsa-driver-1.0.17.mod/alsa-kernel/pci/saa7146/saa7146audio.h --- ../alsa-driver-1.0.17/alsa-kernel/pci/saa7146/saa7146audio.h 1970-01-01 01:00:00.000000000 +0100 +++ ../alsa-driver-1.0.17.mod/alsa-kernel/pci/saa7146/saa7146audio.h 2008-10-22 23:34:05.000000000 +0200 @@ -0,0 +1,120 @@ +/* + * SAA7146 audio stream abstraction layer + * Copyright (c) 2006- by M. Nyffenegger <matthias.nyffenegger[AT]bluewin.ch> + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ +#ifndef SAA7146AUDIO_H_ +#define SAA7146AUDIO_H_ + +#include <linux/types.h> +#include "saa7146i2s.h" + +#define MAX_IN_AUDIO_STREAMS 2 /* SAA7146 has 2 IN-DMA channels */ +#define MAX_OUT_AUDIO_STREAMS 2 /* SAA7146 has 2 OUT-DMA channels */ + +/** + * TODO: description + * Some of this struct's members also occur on i2s_device: this is because + * the 'stream abstraction' is built on top of the 'i2s abstraction'. + */ +struct audio_stream { + int number; /* stream number - corresponds to ALSA subdevice number */ + enum states state; + enum directions direction; + int channel_count; + int sample_length; /* all stream channels have equal sample length */ + enum endian endian; /* all stream channels have equal endian */ + unsigned long buffer_base_addr; + int buffer_size; + /* a stream is controlled by one audio-interface */ + struct audio_interface *audio_interface; + /* a stream can have multiple i2c devices (one per stereo channels) */ + struct i2s_device *i2s_devices[MAX_I2S_DEVICES]; +}; + +/** + * TODO: description + */ +struct saa7146audio { + struct saa7146i2s chipi2s; + /* we are the 'owner' of all streams */ + struct audio_stream in_streams[MAX_IN_AUDIO_STREAMS]; + struct audio_stream out_streams[MAX_OUT_AUDIO_STREAMS]; +}; + +/** + * TODO: description + * @param chipaudio used for SAA7146 I2S ctrl and register access over PCI bus, + * must not be NULL. + * @return A reference to an audio_stream structure, or NULL in case the + * stream could not be opened. + */ +struct audio_stream *saa7146_stream_prepare_capture( + struct saa7146audio *chipaudio, + unsigned int stream_nr, + unsigned long buffer_base_addr, + int buffer_size, + int channel_count, + int sample_length, + enum endian endian); + +/** + * TODO: description + * @param chipaudio used for SAA7146 I2S ctrl and register access over PCI bus, + * must not be NULL. + * @return A reference to an audio_stream structure, or NULL in case the + * stream could not be opened. + */ +struct audio_stream *saa7146_stream_prepare_playback( + struct saa7146audio *chipaudio, + unsigned int stream_nr, + unsigned long buffer_base_addr, + int buffer_size, + int channel_count, + int sample_length, + enum endian endian); + +/** + * TODO: description + * @param chipaudio used for SAA7146 I2S ctrl and register access over PCI bus, + * must not be NULL. + * @param stream A reference to an audio_stream structure, opened with + * saa7146_open_xxxxx_stream(). + */ +void saa7146_stream_unprepare(struct saa7146audio *chipaudio, + struct audio_stream *stream); + +/** + * TODO: description + */ +void saa7146_stream_start(struct saa7146audio *chipaudio, + struct audio_stream *stream); + +/** + * TODO: description + */ +void saa7146_stream_stop(struct saa7146audio *chipaudio, + struct audio_stream *stream); + +/** + * TODO: description + */ +uint32_t saa7146_stream_get_hw_pointer(struct saa7146audio *chipaudio, + struct audio_stream *stream); + +#endif /*SAA7146AUDIO_H_*/ diff -uprN ../alsa-driver-1.0.17/alsa-kernel/pci/saa7146/saa7146.h ../alsa-driver-1.0.17.mod/alsa-kernel/pci/saa7146/saa7146.h --- ../alsa-driver-1.0.17/alsa-kernel/pci/saa7146/saa7146.h 1970-01-01 01:00:00.000000000 +0100 +++ ../alsa-driver-1.0.17.mod/alsa-kernel/pci/saa7146/saa7146.h 2008-10-22 23:34:05.000000000 +0200 @@ -0,0 +1,216 @@ +/* + * SAA7146 register access abstraction layer + * Copyright (c) 2006- by M. Nyffenegger <matthias.nyffenegger[AT]bluewin.ch> + * Based on GPLed Emagic Audiowerk8 Windows driver provided by Martijn Sipkema. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ +#ifndef SAA7146_H_ +#define SAA7146_H_ + +#include <linux/types.h> +#include <linux/pci.h> + +/** + * SAA7146 chip access data structure + */ +struct saa7146reg { + unsigned long iobase_phys; + void __iomem *iobase_virt; + int iolen; +}; + +/** + * Read an SAA7146 register, mind that PCI is little-endian. + * @param chip used for SAA7146 register access over PCI bus + * @param offset register address offset + * @return contents of register at the given address offset + */ +static inline uint32_t saa7146_read(struct saa7146reg *chip, int offset) +{ + return __le32_to_cpu(readl(chip->iobase_virt + offset)); +} + +/** + * Write an SAA7146 register, mind that PCI is little-endian. + * @param chip used for SAA7146 register access over PCI bus + * @param offset register address offset + * @param data the data to be written at the given address offset + */ +static inline void saa7146_write(struct saa7146reg *chip, + int offset, + uint32_t data) +{ + writel(__cpu_to_le32(data), chip->iobase_virt + offset); +} + +/* SAA7146 register offset */ +#define PCI_BT_A 0x4C +#define IICTRF 0x8C +#define IICSTA 0x90 +#define BaseA1_in 0x94 +#define ProtA1_in 0x98 +#define PageA1_in 0x9C +#define BaseA1_out 0xA0 +#define ProtA1_out 0xA4 +#define PageA1_out 0xA8 +#define BaseA2_in 0xAC +#define ProtA2_in 0xB0 +#define PageA2_in 0xB4 +#define BaseA2_out 0xB8 +#define ProtA2_out 0xBC +#define PageA2_out 0xC0 +#define IER 0xDC +#define GPIO_CTRL 0xE0 +#define ACON1 0xF4 +#define ACON2 0xF8 +#define MC1 0xFC +#define MC2 0x100 +#define ISR 0x10C +#define PSR 0x110 +#define SSR 0x114 +#define PCI_ADP1 0x12C +#define PCI_ADP2 0x130 +#define PCI_ADP3 0x134 +#define PCI_ADP4 0x138 +#define LEVEL_REP 0x140 +#define FB_BUFFER1 0x144 +#define FB_BUFFER2 0x148 +#define TSL1 0x180 +#define TSL2 0x1C0 + +/* PSR/ISR/IER */ +#define PPEF (1UL << 31) +#define PABO (1UL << 30) +#define IIC_S (1UL << 17) +#define IIC_E (1UL << 16) +#define A2_in (1UL << 15) +#define A2_out (1UL << 14) +#define A1_in (1UL << 13) +#define A1_out (1UL << 12) +#define AFOU (1UL << 11) +#define PIN3 (1UL << 6) +#define PIN2 (1UL << 5) +#define PIN1 (1UL << 4) +#define PIN0 (1UL << 3) +#define ECS (1UL << 2) +#define EC3S (1UL << 1) +#define EC0S (1UL << 0) + +/* SSR */ +#define PRQ (1UL << 31) +#define PMA (1UL << 30) +#define IIC_EA (1UL << 21) +#define IIC_EW (1UL << 20) +#define IIC_ER (1UL << 19) +#define IIC_EL (1UL << 18) +#define IIC_EF (1UL << 17) +#define AF2_in (1UL << 10) +#define AF2_out (1UL << 9) +#define AF1_in (1UL << 8) +#define AF1_out (1UL << 7) +#define EC5S (1UL << 3) +#define EC4S (1UL << 2) +#define EC2S (1UL << 1) +#define EC1S (1UL << 0) + +/* PCI_BT_A */ +#define BurstA1_in (1UL << 26) +#define ThreshA1_in (1UL << 24) +#define BurstA1_out (1UL << 18) +#define ThreshA1_out (1UL << 16) +#define BurstA2_in (1UL << 10) +#define ThreshA2_in (1UL << 8) +#define BurstA2_out (1UL << 2) +#define ThreshA2_out (1UL << 0) + +/* MC1 */ +#define MRST_N (1UL << 15) +#define EAP (1UL << 9) +#define EI2C (1UL << 8) +#define TR_E_A2_OUT (1UL << 3) +#define TR_E_A2_IN (1UL << 2) +#define TR_E_A1_OUT (1UL << 1) +#define TR_E_A1_IN (1UL << 0) + +/* MC2 */ +#define UPLD_IIC (1UL << 0) + +/* ACON1 bit offsets */ +#define ACON1_AUDIO_MODE 29 +#define ACON1_MAXLEVEL 22 +#define ACON1_A1_SWAP 21 +#define ACON1_A2_SWAP 20 +#define ACON1_WS0_CTRL 18 +#define ACON1_WS0_SYNC 16 +#define ACON1_WS1_CTRL 14 +#define ACON1_WS1_SYNC 12 +#define ACON1_WS2_CTRL 10 +#define ACON1_WS2_SYNC 8 +#define ACON1_WS3_CTRL 6 +#define ACON1_WS3_SYNC 4 +#define ACON1_WS4_CTRL 2 +#define ACON1_WS4_SYNC 0 + +/* ACON2 bit offsets */ +#define ACON2_A1_CLKSRC 27 +#define ACON2_A2_CLKSRC 22 +#define ACON2_INVERT_BCLK1 21 +#define ACON2_INVERT_BCLK2 20 +#define ACON2_BCLK1_OEN 19 +#define ACON2_BCLK2_OEN 18 + +/* TSL bit offsets */ +#define TSL_EOS 0 +#define TSL_LOW_A2 1 +#define TSL_DOD_A2 2 +#define TSL_BSEL_A2 4 +#define TSL_LF_A2 7 +#define TSL_SF_A2 8 +#define TSL_SIB_A2 9 +#define TSL_SDW_A2 10 +#define TSL_DIS_A2 11 +#define TSL_LOW_A1 14 +#define TSL_DOD_A1 15 +#define TSL_BSEL_A1 17 +#define TSL_LF_A1 20 +#define TSL_SF_A1 21 +#define TSL_SIB_A1 22 +#define TSL_SDW_A1 23 +#define TSL_DIS_A1 24 +#define TSL_WS4 27 +#define TSL_WS3 28 +#define TSL_WS2 29 +#define TSL_WS1 30 +#define TSL_WS0 31 + +/* SD pin select for DIS_Ax and DOD_Ax */ +#define SD0_I_A2 0 +#define SD0_O_A1 0 +#define SD1_IO_Ax 1 +#define SD2_IO_Ax 2 +#define SD3_IO_Ax 3 +#define SD4_I_A1 0 +#define SD4_O_A2 0 + +/* ACON1 WSx_CTRL select */ +#define ACON1_WSx_CTRL_IN_TSLx 0 +#define ACON1_WSx_CTRL_OUT_TSL1 1 +#define ACON1_WSx_CTRL_OUT_TSL2 2 +#define ACON1_WSx_CTRL_LOW 3 +#define ACON1_WSx_SYNC_I2S 0 + +#endif /*SAA7146_H_*/ diff -uprN ../alsa-driver-1.0.17/alsa-kernel/pci/saa7146/saa7146i2c.c ../alsa-driver-1.0.17.mod/alsa-kernel/pci/saa7146/saa7146i2c.c --- ../alsa-driver-1.0.17/alsa-kernel/pci/saa7146/saa7146i2c.c 1970-01-01 01:00:00.000000000 +0100 +++ ../alsa-driver-1.0.17.mod/alsa-kernel/pci/saa7146/saa7146i2c.c 2008-10-22 23:34:05.000000000 +0200 @@ -0,0 +1,270 @@ +/* + * SAA7146 I2C-bus abstraction layer + * Copyright (c) 2006- by M. Nyffenegger <matthias.nyffenegger[AT]bluewin.ch> + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ +#include <linux/types.h> +#include <linux/sched.h> +#include "log.h" +#include "saa7146.h" +#include "saa7146i2c.h" + +/* Convenience macros */ +#define MAX(a, b) ((a) > (b) ? (a) : (b)) + +/** + * See SAA7146 tech. specification p.122 - 126 for details. + */ + +#define IICSTA_ABORT 0x80 +#define IICSTA_ERR_BSY 0x7f +#define MC1_EI2C 0x01000000 +#define MC2_UPLD_IIC 0x00010000 +#define START 3 +#define CONT 2 +#define STOP 1 +#define NOP 0 +#define RW 1 + +/** + * SAA7146 available i2c clock rates to Hz mapping: + * The i2c status register clock-rate value is equal to the array index of its + * Hz value => order matters! + */ +static const unsigned int i2c_clk_2_hz[] = { + 266000, /* PCI clock/120 at index 0 */ + 10000, /* PCI clock/3200 at index 1 */ + 400000, /* PCI clock/80 at index 2 */ + 533000, /* PCI clock/60 at index 3 */ + 66000, /* PCI clock/480 at index 4 */ + 5000, /* PCI clock/6400 at index 5 */ + 100000, /* PCI clock/320 at index 6 */ + 133000, /* PCI clock/240 at index 7 */ +}; + +/* forward declarations */ +static int i2c_open(struct saa7146reg *chip, + enum saa7146_i2c_clock clk, + int read, + uint8_t dev_addr, + uint8_t *sub_addr, + unsigned int sub_addr_size); +static int i2c_transmit(struct saa7146reg *chip, + enum saa7146_i2c_clock clk, + int read, + uint8_t *buf, + unsigned int buf_size); +static int i2c_transmit_byte(struct saa7146reg *chip, + enum saa7146_i2c_clock clk, + int read, + uint8_t *byte, + int cmd); +static void i2c_prepare(struct saa7146reg *chip, enum saa7146_i2c_clock clk); +static void i2c_upload(struct saa7146reg *chip, int offset, uint32_t data); +static void i2c_wait(enum saa7146_i2c_clock clk, int bytes_to_send); +static void i2c_abort(struct saa7146reg *chip); + +/** + * see saa7146i2c.h + */ +int saa7146_i2c_read(struct saa7146reg *chip, + enum saa7146_i2c_clock clk, + uint8_t dev_addr, + uint8_t *sub_addr, + int sub_addr_size, + uint8_t *buffer, + int size) +{ + uint32_t ret_val = 0; + + ret_val = i2c_open(chip, clk, RW, dev_addr, sub_addr, sub_addr_size); + if (ret_val >= 0) + ret_val = i2c_transmit(chip, clk, RW, buffer, size); + return ret_val; +} + +/** + * see saa7146i2c.h + */ +int saa7146_i2c_write(struct saa7146reg *chip, + enum saa7146_i2c_clock clk, + uint8_t dev_addr, + uint8_t *sub_addr, + int sub_addr_size, + uint8_t *data, + int size) +{ + uint32_t ret_val = 0; + + ret_val = i2c_open(chip, clk, !RW, dev_addr, sub_addr, sub_addr_size); + if (ret_val >= 0) + ret_val = i2c_transmit(chip, clk, !RW, data, size); + return ret_val; +} + +/** + * Open an i2c device: + * 1) transmit device-address, r/w == low and startcondition + * 2) optionally transmit subaddress + * 3) in case of READ operation, transmit device-address again with r/w == high + * and startcondition. + * @return number of address bytes sent or -1 in case of failure. + */ +static int i2c_open(struct saa7146reg *chip, + enum saa7146_i2c_clock clk, + int read, + uint8_t dev_addr, + uint8_t *sub_addr, + unsigned int sub_addr_size) +{ + int sub_addr_bytes_transmitted = 0; + uint8_t addr = 0; + + i2c_prepare(chip, clk); + addr = (dev_addr<<1)&~RW; + if (i2c_transmit_byte(chip, clk, read, &addr, START) != 0) { + LOG_ERROR("i2c address phase"); + return -1; + } + while (sub_addr_bytes_transmitted < sub_addr_size) { + if (i2c_transmit_byte(chip, clk, read, sub_addr++, CONT) != 0) { + LOG_ERROR("i2c subaddress phase"); + return -1; + } + sub_addr_bytes_transmitted++; + } + if (read) { + addr = (dev_addr<<1)|RW; + if (i2c_transmit_byte(chip, clk, read, &addr, START) != 0) { + LOG_ERROR("i2c prepare READ phase"); + return -1; + } + } + return sub_addr_bytes_transmitted; +} + +/** + * Transmit data on i2c-bus: transfer is closed when all data is transmitted. + * Call this function after i2c_open(). + * @return number of bytes transmitted or -1 in case of failure. + */ +static int i2c_transmit(struct saa7146reg *chip, + enum saa7146_i2c_clock clk, + int read, + uint8_t *buffer, + unsigned int size) +{ + int bytes_transmitted = 0; + int attr = 0; + + while (bytes_transmitted < size) { + attr = (bytes_transmitted++ < (size - 1) ? CONT : STOP); + if (i2c_transmit_byte(chip, clk, read, buffer++, attr) != 0) { + LOG_ERROR("i2c data transmission phase"); + return -1; + } + } + saa7146_write(chip, MC1, MC1_EI2C | 0); /* disable i2c */ + return bytes_transmitted; +} + +/** + * In order to keep the implementation simple, we transmit only one byte per + * i2c-upload (IICTRF BYTE2). + */ +static int i2c_transmit_byte(struct saa7146reg *chip, + enum saa7146_i2c_clock clk, + int read, + uint8_t *byte, + int attr) +{ + uint32_t iictrf = 0; /* BYTE0..2 = 0, ATTR0..2 = NOP */ + uint32_t iicsta = 0; + + iictrf = (*byte) << (8*3) | attr << (2*3); + i2c_upload(chip, IICTRF, iictrf); + i2c_wait(clk, 1); + iicsta = saa7146_read(chip, IICSTA); + if ((iicsta & IICSTA_ERR_BSY) != 0) { + i2c_abort(chip); + LOG_ERROR("IICSTA=%#010lx", (long)iicsta); + return -1; + } + if (read) { + iictrf = saa7146_read(chip, IICTRF); + *byte = iictrf >> (8*3); + } + return 0; +} + +/** + * Enable i2c, abort all pending operations, clear all errors (may be needed 2x + * after abort) and set clock rate. + */ +static void i2c_prepare(struct saa7146reg *chip, enum saa7146_i2c_clock clk) +{ + saa7146_write(chip, MC1, MC1_EI2C | EI2C); + i2c_upload(chip, IICSTA, IICSTA_ABORT); + i2c_upload(chip, IICSTA, 0); + i2c_upload(chip, IICSTA, (clk << 8)); +} + +/** + * Upload I2C register IICSTA and IICTRF from shadow RAM. + */ +static void i2c_upload(struct saa7146reg *chip, int offset, uint32_t data) +{ + if ((offset == IICSTA) || (offset == IICTRF)) { + saa7146_write(chip, offset, data); + saa7146_write(chip, MC2, (MC2_UPLD_IIC | UPLD_IIC)); + } +} + +/** + * Wait for a i2c transmission to complete. + * The required delay is calculated on the basis of the given clock-rate and + * the number of bytes to transmit: + * - each byte is followed by ACK i.e. there are 9 clock-cycles/byte, we add + * one for safety + * - the kernel's scheduler tick (HZ) is used as time base and we add one tick + * for safety + * + * delay[ms] = (bytes_to_send*10[bits]/clock[1/s]*1000)*HZ/1000 + 1 + * + * The function returns after the required delay, regardless of the i2c state. + */ +static void i2c_wait(enum saa7146_i2c_clock clk, int bytes_to_send) +{ + int delay = 0; + + delay = ((bytes_to_send*10*HZ)/i2c_clk_2_hz[clk]) + 1; + set_current_state(TASK_UNINTERRUPTIBLE); + schedule_timeout(MAX(1, delay)); +} + +/** + * Abort all pending operations, clear all errors (may be needed 2x after abort) + * set clock rate to lowset possible and finally disable i2c. + */ +static void i2c_abort(struct saa7146reg *chip) +{ + i2c_upload(chip, IICSTA, IICSTA_ABORT); + i2c_upload(chip, IICSTA, 0); + i2c_upload(chip, IICSTA, (bps_5k << 8)); + saa7146_write(chip, MC1, MC1_EI2C | 0); +} diff -uprN ../alsa-driver-1.0.17/alsa-kernel/pci/saa7146/saa7146i2c.h ../alsa-driver-1.0.17.mod/alsa-kernel/pci/saa7146/saa7146i2c.h --- ../alsa-driver-1.0.17/alsa-kernel/pci/saa7146/saa7146i2c.h 1970-01-01 01:00:00.000000000 +0100 +++ ../alsa-driver-1.0.17.mod/alsa-kernel/pci/saa7146/saa7146i2c.h 2008-10-22 23:34:05.000000000 +0200 @@ -0,0 +1,81 @@ +/* + * SAA7146 I2C-bus abstraction layer + * Copyright (c) 2006- by M. Nyffenegger <matthias.nyffenegger[AT]bluewin.ch> + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ +#ifndef SAA7146I2C_H_ +#define SAA7146I2C_H_ + +#include <linux/types.h> +#include "saa7146.h" + +/** + * SAA7146 available i2c clock rates (reg IICSTA IICCC[2:0] bit 8..10) + */ +enum saa7146_i2c_clock { + bps_5k = 5, /* PCI clock/6400 */ + bps_10k = 1, /* PCI clock/3200 */ + bps_66k = 4, /* PCI clock/480 */ + bps_100k = 6, /* PCI clock/320 */ + bps_133k = 7, /* PCI clock/240 */ + bps_266k = 0, /* PCI clock/120 */ + bps_400k = 2, /* PCI clock/80 */ + bps_533k = 3 /* PCI clock/60 */ +}; + +/** + * Write n bytes to i2c bus. + * @param chip Used for SAA7146 register access over PCI bus + * @param clock One of the SAA7146 available i2c bit rates + * @param dev_addr the i2c (slave-) device address 7-bit + * @param sub_addr the i2c (slave-) device sub-address + * @param sub_addr_size The size of the sub-address, can be 0 if no sub-address + * is provided + * @param data Data to be written + * @param size Size of data to be written, must be >= 0 + * @return Number of bytes written or -1 in case of failure + */ +int saa7146_i2c_write(struct saa7146reg *chip, + enum saa7146_i2c_clock clock, + uint8_t dev_addr, + uint8_t *sub_addr, + int sub_addr_size, + uint8_t *data, + int size); + +/** + * Read n bytes from i2c bus. + * @param chip Used for SAA7146 register access over PCI bus + * @param clock One of the SAA7146 available i2c bit rates + * @param dev_addr the i2c (slave-) device address 7-bit + * @param sub_addr the i2c (slave-) device sub-address + * @param sub_addr_size The size of the sub-address, can be 0 if no sub-address + * is provided + * @param buffer Data buffer for bytes to be read. + * @param size Size of data to be read, must be >= 0. + * @return Number of bytes read or -1 in case of failure. + */ +int saa7146_i2c_read(struct saa7146reg *chip, + enum saa7146_i2c_clock clock, + uint8_t dev_addr, + uint8_t *sub_addr, + int sub_addr_size, + uint8_t *buffer, + int size); + +#endif /*SAA7146I2C_H_*/ diff -uprN ../alsa-driver-1.0.17/alsa-kernel/pci/saa7146/saa7146i2s.c ../alsa-driver-1.0.17.mod/alsa-kernel/pci/saa7146/saa7146i2s.c --- ../alsa-driver-1.0.17/alsa-kernel/pci/saa7146/saa7146i2s.c 1970-01-01 01:00:00.000000000 +0100 +++ ../alsa-driver-1.0.17.mod/alsa-kernel/pci/saa7146/saa7146i2s.c 2008-10-22 23:34:05.000000000 +0200 @@ -0,0 +1,437 @@ +/* + * SAA7146 I2S and audio-interface abstraction layer + * Copyright (c) 2006- by M. Nyffenegger <matthias.nyffenegger[AT]bluewin.ch> + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ +#include <linux/types.h> +#include "log.h" +#include "saa7146.h" +#include "saa7146i2s.h" + +/* Convenience macros */ +#define MAX(a, b) ((a) > (b) ? (a) : (b)) +#define MIN(a, b) ((a) < (b) ? (a) : (b)) + +#define DWBUF_SIZE 4 /* SAA7146 audiointerface's DWORD buffer size */ +#define MAX_SUPERFRAME_LENGTH 16 /* 32 byte superframe is not supported */ +#define MIN_SUPERFRAME_LENGTH DWBUF_SIZE /* must be >= DWORD buffer length */ + +/** + * TSL address map: array pos. corresponds to enum audio_interfaces + */ +static const int TSLx[2] = {TSL1, TSL2}; + +/** + * TSL bit offse maps: array pos. corresponds to enum audio_interfaces + */ +static const int TSL_DIS_Ax[2] = {TSL_DIS_A1, TSL_DIS_A2}; +static const int TSL_SIB_Ax[2] = {TSL_SIB_A1, TSL_SIB_A2}; +static const int TSL_SDW_Ax[2] = {TSL_SDW_A1, TSL_SDW_A2}; +static const int TSL_SF_Ax[2] = {TSL_SF_A1, TSL_SF_A2}; +static const int TSL_LF_Ax[2] = {TSL_LF_A1, TSL_LF_A2}; +static const int TSL_BSEL_Ax[2] = {TSL_BSEL_A1, TSL_BSEL_A2}; +static const int TSL_DOD_Ax[2] = {TSL_DOD_A1, TSL_DOD_A2}; + +/** + * ACON1 SWAP_Ax bit offset map: array pos. corresponds to enum audio_interfaces + */ +static const int ACON1_Ax_SWAP[2] = {ACON1_A1_SWAP, ACON1_A2_SWAP}; + +/** + * ACON1 WSx_CTRL value map: array pos. correspond to enum audio_interfaces + */ +static const int ACON1_WSx_CTRL_OUT_TSLx[2] = {ACON1_WSx_CTRL_OUT_TSL1, + ACON1_WSx_CTRL_OUT_TSL2}; + +/** + * ACON1 WSx_CTRL bit offset map: array pos. corresponds to enum ws_lines + */ +static const int ACON1_WSx_CTRL[5] = {ACON1_WS0_CTRL, + ACON1_WS1_CTRL, + ACON1_WS2_CTRL, + ACON1_WS3_CTRL, + ACON1_WS4_CTRL}; + +/** + * ACON1 WSx_SYNC bit offset map: array pos. corresponds to enum ws_lines + */ +static const int ACON1_WSx_SYNC[5] = {ACON1_WS0_SYNC, + ACON1_WS1_SYNC, + ACON1_WS2_SYNC, + ACON1_WS3_SYNC, + ACON1_WS4_SYNC}; + +/** + * TSL WSx bit offset map: array pos. correspond to enum ws_lines + */ +static const int TSL_WSx[5] = {TSL_WS0, + TSL_WS1, + TSL_WS2, + TSL_WS3, + TSL_WS4}; + +/** + * TSL SDx bit value map: array pos. corresponds to enum sd_lines + */ +static const int TSL_SDx[7] = {SD0_I_A2, + SD0_O_A1, + SD1_IO_Ax, + SD2_IO_Ax, + SD3_IO_Ax, + SD4_I_A1, + SD4_O_A2}; + +/* forward declarations */ +static int tsl_update(struct saa7146i2s *chipi2s, + struct audio_interface *audio_interface); +static int tsl_build(struct audio_interface *audio_interface, uint32_t tsl[]); +static int tsl_align_devices(struct audio_interface *audio_interface); +static void tsl_prepare_capture_slot(struct i2s_device *dev, + uint32_t tsl[], + int slot, + int *total_bytes_cap); +static void tsl_prepare_playback_slot(struct i2s_device *dev, + uint32_t tsl[], + int slot, + int *total_bytes_pbk, + int *bsel); +static void setbits(uint32_t *reg, + unsigned int offset, + unsigned int len, + uint32_t val); + +/** + * see saa7146i2s.h + */ +int saa7146_i2s_init_audio_interface(struct saa7146i2s *chipi2s, + enum audio_interfaces ai_id, + int i2s_word_length, + int i2s_superframe_length) +{ + struct audio_interface *ai = NULL; + + if (i2s_word_length > DWBUF_SIZE || i2s_word_length <= 0 + || !is_pow_of_2(i2s_word_length)) { + LOG_ERROR("Invalid i2s word-length %d", i2s_word_length); + return -1; + } + if (i2s_superframe_length > MAX_SUPERFRAME_LENGTH || + i2s_superframe_length < MIN_SUPERFRAME_LENGTH || + !is_pow_of_2(i2s_superframe_length / i2s_word_length)) { + LOG_ERROR("Invalid i2s superframe-length %d", + i2s_superframe_length); + return -1; + } + ai = &chipi2s->audio_interfaces[ai_id]; + ai->id = ai_id; + ai->i2s_word_length = i2s_word_length; + ai->i2s_superframe_length = i2s_superframe_length; + return 0; +} + +/** + * see saa7146i2s.h + */ +void saa7146_i2s_enable_audio_interface(struct saa7146i2s *chipi2s, + struct audio_interface *ai) +{ + /* set AUDIO_MODE A1 and A2 independent */ + if (ai->id == a1) + setbits(&chipi2s->acon1, ACON1_AUDIO_MODE, 1, 1); + else if (ai->id == a2) + setbits(&chipi2s->acon1, ACON1_AUDIO_MODE + 1, 1, 1); + saa7146_write(&chipi2s->chip, ACON1, chipi2s->acon1); +} + +/** + * see saa7146i2s.h + */ +void saa7146_i2s_disable_audio_interface(struct saa7146i2s *chipi2s, + struct audio_interface *ai) +{ + /* set AUDIO_MODE A1 and A2 independent */ + if (ai->id == a1) + setbits(&chipi2s->acon1, ACON1_AUDIO_MODE, 1, 0); + else if (ai->id == a2) + setbits(&chipi2s->acon1, ACON1_AUDIO_MODE + 1, 1, 0); + saa7146_write(&chipi2s->chip, ACON1, chipi2s->acon1); +} + +/** + * see saa7146i2s.h + */ +int saa7146_i2s_init_device(struct saa7146i2s *chipi2s, + enum audio_interfaces ai_id, + enum directions dir, + enum ws_lines ws, + enum sd_lines sd) +{ + struct i2s_device *dev = NULL; + + if (ws < MAX_I2S_DEVICES) { + dev = &chipi2s->i2s_devices[ws]; + dev->ws = ws; + dev->sd = sd; + dev->sample_length = 0; + dev->endian = le; + dev->state = disabled; + dev->direction = dir; + dev->hw_mon = notmonitored; + dev->audio_interface = &chipi2s->audio_interfaces[ai_id]; + dev->audio_interface->i2s_devices[ws] = dev; + /* set WS-Line active low (disable) by default */ + setbits(&chipi2s->acon1, ACON1_WSx_CTRL[ws], 2, + ACON1_WSx_CTRL_LOW); + } else { + LOG_ERROR("invalid ws=%d", ws); + return -1; + } + return 0; +} + +/** + * see saa7146i2s.h + */ +int saa7146_i2s_enable_device(struct saa7146i2s *chipi2s, + struct i2s_device *dev, + int sample_len, + enum endian endian) +{ + dev->sample_length = sample_len; + dev->endian = endian; + dev->state = enabled; + if (dev->direction == in) { + /* i2s WS config - for capture devices we assume slave mode */ + setbits(&chipi2s->acon1, ACON1_WSx_CTRL[dev->ws], 2, + ACON1_WSx_CTRL_IN_TSLx); + /* handle endianness (see documentation in wiki) */ + setbits(&chipi2s->acon1, + ACON1_Ax_SWAP[dev->audio_interface->id], 1, + (dev->endian == le) ? 0 : 1); + } + if (dev->direction == out) { + /* i2s WS config - for playback devices we assume master mode */ + setbits(&chipi2s->acon1, ACON1_WSx_CTRL[dev->ws], 2, + ACON1_WSx_CTRL_OUT_TSLx[dev->audio_interface->id]); + } + setbits(&chipi2s->acon1, ACON1_WSx_SYNC[dev->ws], 2, + ACON1_WSx_SYNC_I2S); + saa7146_write(&chipi2s->chip, ACON1, chipi2s->acon1); + if (tsl_update(chipi2s, dev->audio_interface) != 0) + return -1; + return 0; +} + +/** + * see saa7146i2s.h + */ +int saa7146_i2s_disable_device(struct saa7146i2s *chipi2s, + struct i2s_device *dev) +{ + dev->sample_length = 0; + dev->endian = le; + dev->state = disabled; + dev->hw_mon = notmonitored; + /* set WS-Line active low (disable) by default */ + setbits(&chipi2s->acon1, ACON1_WSx_CTRL[dev->ws], 2, + ACON1_WSx_CTRL_LOW); + if (tsl_update(chipi2s, dev->audio_interface) != 0) + return -1; + return 0; +} + +/** + * TODO: description + * TSL1 is always associated with A1, TSL2 with A2. + */ +static int tsl_update(struct saa7146i2s *chipi2s, + struct audio_interface *ai) +{ + int slot = 0; + uint32_t tsl[ai->i2s_superframe_length]; + + if (tsl_align_devices(ai) != 0) + return -1; + /* reset TSL array! */ + for (slot = 0; slot < ai->i2s_superframe_length; tsl[slot++] = 0); + if (tsl_build(ai, tsl) == 0) { + for (slot = 0; slot < ai->i2s_superframe_length; slot++) { + saa7146_write(&chipi2s->chip, + TSLx[ai->id] + (4 * slot), tsl[slot]); + } + return 0; + } + return -1; +} + +/** + * Build the TSL for the given audio-interface. + * The TSL is built from scratch, based on the properties of the devices that + * are controlled by the associated audio-interface. + * This function does not write the TSL down to the HW. This is done in + * tsl_update() - i.e. if this function fails the current TSL is not changed. + */ +static int tsl_build(struct audio_interface *ai, uint32_t tsl[]) +{ + int i = 0; + int slot = 0; + int bsel = 0; + int total_bytes_cap = 0; + int total_bytes_pbk = 0; + struct i2s_device *dev = NULL; + int max_slots = ai->i2s_superframe_length; + + for (slot = 0; slot < max_slots; slot++) { + for (i = 0; i < MAX_I2S_DEVICES; i++) { + dev = ai->i2s_devices[i]; + if (dev != NULL && dev->state == enabled) { + if (dev->direction == in) { + tsl_prepare_capture_slot(dev, tsl, + slot, &total_bytes_cap); + } else if (dev->direction == out) { + tsl_prepare_playback_slot(dev, tsl, + slot, &total_bytes_pbk, &bsel); + } + } + } + /* Reset TSL pointer in the last timeslot */ + setbits(&tsl[slot], TSL_EOS, 1, (slot == max_slots-1 ? 1 : 0)); + } + if (total_bytes_cap > 0 && total_bytes_cap < DWBUF_SIZE) { + LOG_ERROR("Total sample-length of capture devices" + " on audio-interface %d < DWORD-buffer size => buffer" + " is never transfered to DMA", ai->id); + return -1; + } + if (total_bytes_pbk > 0 && total_bytes_pbk < DWBUF_SIZE) { + LOG_ERROR("Total sample-length of playback devices" + " on audio-interface %d < DWORD-buffer size => buffer" + " is never reloaded from DMA", ai->id); + return -1; + } + return 0; +} + +/** + * Assign a TSL slot to each i2s-device on the given audio-interface: this is + * the TSL slot where a given i2s-device's first sample byte of its first + * channel is captured/played back. + */ +static int tsl_align_devices(struct audio_interface *ai) +{ + int i = 0; + int next_pbk_dev_start_slot = 0; + int next_cap_dev_start_slot = 0; + struct i2s_device *dev = NULL; + + for (i = 0; i < MAX_I2S_DEVICES; i++) { + dev = ai->i2s_devices[i]; + if (dev != NULL && dev->direction == in) { + if (next_cap_dev_start_slot > ai->i2s_word_length) { + LOG_ERROR("Total sample-length of capture " + "devices on audio-interface %d > " + "i2s word-length", ai->id); + return -1; + } + dev->tsl_slot = next_cap_dev_start_slot; + next_cap_dev_start_slot += dev->sample_length; + } + if (dev != NULL && dev->direction == out) { + if (next_pbk_dev_start_slot > ai->i2s_word_length) { + LOG_ERROR("Total sample-length of playback " + "devices on audio-interface %d > " + "i2s word-length", ai->id); + return -1; + } + dev->tsl_slot = next_pbk_dev_start_slot; + next_pbk_dev_start_slot += dev->sample_length; + } + } + return 0; +} + +static void tsl_prepare_capture_slot(struct i2s_device *dev, + uint32_t tsl[], + int slot, + int *total_bytes_cap) +{ + int word_length = dev->audio_interface->i2s_word_length; + + if (slot%word_length >= dev->tsl_slot && + slot%word_length < (dev->tsl_slot + dev->sample_length)) { + setbits(&tsl[slot], TSL_DIS_Ax[dev->audio_interface->id], 2, + TSL_SDx[dev->sd]); + setbits(&tsl[slot], TSL_SDW_Ax[dev->audio_interface->id], 1, 1); + setbits(&tsl[slot], TSL_SIB_Ax[dev->audio_interface->id], 1, + (dev->hw_mon == source ? 1 : 0)); + (*total_bytes_cap)++; + setbits(&tsl[slot], TSL_SF_Ax[dev->audio_interface->id], 1, + (*total_bytes_cap%DWBUF_SIZE == 0 ? 1 : 0)); + } +} + +static void tsl_prepare_playback_slot(struct i2s_device *dev, + uint32_t tsl[], + int slot, + int *total_bytes_pbk, + int *bsel) +{ + unsigned int tmp = 0; + int ws_line = 0; + int bsel_offset = 0; + int word_length = dev->audio_interface->i2s_word_length; + + if (slot%word_length >= dev->tsl_slot && + slot%word_length < (dev->tsl_slot + dev->sample_length)) { + bsel_offset = (dev->hw_mon == destination ? 4 : 0); + /* see documentation in wiki for details regarding endianness */ + setbits(&tsl[slot], TSL_BSEL_Ax[dev->audio_interface->id], 3, + (dev->sample_length == 2 ? + (dev->endian == le ? (1 - *bsel) & 3 : *bsel) : + (dev->endian == le ? 3 - *bsel : *bsel)) + + bsel_offset); + setbits(&tsl[slot], TSL_DOD_Ax[dev->audio_interface->id], 2, + TSL_SDx[dev->sd]); + setbits(&tsl[slot], TSL_LF_Ax[dev->audio_interface->id], 1, + (*total_bytes_pbk%DWBUF_SIZE == 0 ? 1 : 0)); + (*total_bytes_pbk)++; + *bsel = (*bsel + 1)%DWBUF_SIZE; + } + /* WS: drive high until dev->tsl_slot, then drive low for word- + length slots, then drive high again */ + tmp = ((unsigned int)slot) - dev->tsl_slot; + ws_line = (tmp%(2 * word_length) < word_length ? 0 : 1); + setbits(&tsl[slot], TSL_WSx[dev->ws], 1, ws_line); +} + + +/** + * Set bits at a given offset to the given value. + * @param reg The variable where the bits are set + * @param offset The offset where the bits are set + * @param len The number of contiguous bits + * @param val The value to be set + */ +static void setbits(uint32_t *reg, + unsigned int offset, + unsigned int len, + uint32_t val) +{ + *reg &= ~(((1 << len) - 1) << offset); /* clear bits (!) */ + *reg |= (val << offset); +} diff -uprN ../alsa-driver-1.0.17/alsa-kernel/pci/saa7146/saa7146i2s.h ../alsa-driver-1.0.17.mod/alsa-kernel/pci/saa7146/saa7146i2s.h --- ../alsa-driver-1.0.17/alsa-kernel/pci/saa7146/saa7146i2s.h 1970-01-01 01:00:00.000000000 +0100 +++ ../alsa-driver-1.0.17.mod/alsa-kernel/pci/saa7146/saa7146i2s.h 2008-10-22 23:34:05.000000000 +0200 @@ -0,0 +1,197 @@ +/* + * SAA7146 I2S and audio-interface abstraction layer + * Copyright (c) 2006- by M. Nyffenegger <matthias.nyffenegger[AT]bluewin.ch> + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ +#ifndef SAA7146I2S_H_ +#define SAA7146I2S_H_ + +#include <linux/types.h> +#include "saa7146.h" + +#define MAX_I2S_DEVICES 5 /* SAA4146 supports a max. of 5 i2s devices */ +#define MAX_AUDIO_INTERFACES 2 /* SAA7146 has 2 independent audio-interfaces */ + +/* forward declarations due to mutual reference audio_interface<->i2s_device */ +struct audio_interface; + +/** + * Enumeration of SAA7146 audio-interfaces A1 and A2. + * Order matters (see head of saa7146i2s.c: address and bit offset maps) + */ +enum audio_interfaces {a1, a2}; + +/** + * Enumeration of SAA7146 i2c WS lines. + * Order matters (see head of saa7146i2s.c: address and bit offset maps) + */ +enum ws_lines {ws0, ws1, ws2, ws3, ws4}; + +/** + * Enumeration of SAA7146 i2c SD lines. + * sd0_i_a2: SD0 is always input on SAA7146 audiointerface A2 + * sd0_o_a1: SD0 is always output on SAA7146 audiointerface A1 + * sd4_i_a1: SD4 is always input on SAA7146 audiointerface A1 + * sd4_o_a2: SD4 is always output on SAA7146 audiointerface A2 + * Order matters (see head of saa7146i2s.c: address and bit offset maps) + */ +enum sd_lines {sd0_i_a2, sd0_o_a1, sd1_io_ax, sd2_io_ax, sd3_io_ax, sd4_i_a1, + sd4_o_a2}; + +enum endian {le, be}; + +enum directions {in, out}; + +enum states {disabled, enabled}; + +enum hw_mon {source, destination, notmonitored}; + +/** + * tsl_slot: SAA7146 TSL slot-number of 1st channel MSB + * audio_interface: convenience reference + * sample_length: sample length in bytes + */ +struct i2s_device { + enum ws_lines ws; + enum sd_lines sd; + unsigned int tsl_slot; + struct audio_interface *audio_interface; + int sample_length; + enum endian endian; + enum states state; + enum directions direction; + enum hw_mon hw_mon; +}; + +/** + * i2s_word_length: word length in bytes + * i2s_superframe_length: superframe length in bytes + * fs: each audio-interface can have its own samplerate + */ +struct audio_interface { + enum audio_interfaces id; + int i2s_word_length; + int i2s_superframe_length; + int fs; + struct i2s_device *i2s_devices[MAX_I2S_DEVICES]; +}; + +/** + * struct audio_interface and struct i2c_device instances are statically + * allocated and are 'owned' by this struct saa7146i2s -> this is the place + * where the actual data structures are defined (no dynamic allocation + * elsewhere). + */ +struct saa7146i2s { + struct saa7146reg chip; + struct audio_interface audio_interfaces[MAX_AUDIO_INTERFACES]; + struct i2s_device i2s_devices[MAX_I2S_DEVICES]; + uint32_t acon1; +}; + +/** + * Initialize an SAA7146's audio-interface. The parameter values depend on the + * hardware setup. As a rule, these are rather static settings and are + * established once in the module's lifecycle. + * @param chipi2s used for I2S device management and SAA7146 register access, + * must not be NULL. + * @param audio_interface_id the SAA7146 audio-interface to be initialized. + * @param i2s_word_length word-length of i2s devices operated by the given + * audio-interface. One audio-interface can handle more than one i2s device, + * but the word-length (i.e. 1/2 WS-period) must be the same for all. + * word-length 1, 2 and 4 bytes are supported. + * @param i2s_superframe_length a superframe contains a number of WS-periods. + * The following condition must match: + * + * (MIN_SUPERFRAME_LENGTH <= superframe-length <= MAX_SUPERFRAME_LENGTH) + * AND superframe-length == [2,4,8,16] * word-length + * + * @return 0 in case of success or -1 in case of failure. + */ +int saa7146_i2s_init_audio_interface(struct saa7146i2s *chipi2s, + enum audio_interfaces audio_interface_id, + int i2s_word_length, + int i2s_superframe_length); + +/** + * TODO: description + */ +void saa7146_i2s_enable_audio_interface(struct saa7146i2s *chipi2s, + struct audio_interface *ai); + +/** + * TODO: description + */ +void saa7146_i2s_disable_audio_interface(struct saa7146i2s *chipi2s, + struct audio_interface *ai); + +/** + * Initializes an i2s device. As a rule, init values are rather static, e.g. + * given by the HW wiring, and are established once in the module's lifecycle. + * WS determines the order in which the i2s devices (= stereo audio channels) + * are expected to appear in the host memory. + * @param chipi2s used for I2S device management and SAA7146 register access, + * must not be NULL. + * @param audio_interface_id the SAA7146 audio-interface which operates this + * i2s device. + * @param direction either 'in' or 'out'. + * @param ws the WS line to be used for this i2s device. + * @param sd the SD line to be used for this i2s device. + * @return 0 in case of success or -1 in case of failure. + */ +int saa7146_i2s_init_device(struct saa7146i2s *chipi2s, + enum audio_interfaces audio_interface_id, + enum directions direction, + enum ws_lines ws, + enum sd_lines sd); + +/** + * Enables an i2s device for capture or playback. + * @param chipi2s used for I2S device management and SAA7146 register access, + * must not be NULL. + * @param device A reference to the i2s_device reference returned by + * saa7146_i2s_init_device(), must not be NULL. + * @param sample_length the length of samples transferred over this i2s device. + * @param endian endianess of the captured bytes in the host memory. + * @return 0 for success or -1 for failure. + */ +int saa7146_i2s_enable_device(struct saa7146i2s *chipi2s, + struct i2s_device *device, + int sample_length, + enum endian endian); + +/** + * Disables an i2s device. + * @param chipi2s used for I2S device management and SAA7146 register access, + * must not be NULL. + * @param device A reference to the i2s_device reference returned by + * saa7146_i2s_init_device(), must not be NULL. + * @return 0 for success or -1 for failure. + */ +int saa7146_i2s_disable_device(struct saa7146i2s *chipi2s, + struct i2s_device *device); + +/** + * @return 1 if the given value is a power of 2. + */ +static inline int is_pow_of_2(int val) +{ + return ((val != 0) && ((val & (val - 1)) == 0)); +} + +#endif /*SAA7146I2S_H_*/

2 1

[alsa-devel] ALSA Programming Query
by santosh pattar 23 Oct '08

23 Oct '08

Dear All, I have some doubts on the ALSA programming, 1)My ALSA driver supports minimum Period Bytes size as 32 bytes, but when i set the period size as 512 bytes it is always setting to 1024 bytes.I am using the function "snd_pcm_hw_params_set_period_size_near()" . What might be wrong here?Is the function what i am using correct? 2)How the Library functions and the Driver functions interact, what is the interface between the library functions and driver functions?Is there any document which explains this. I got a document which explains how to write ALSA driver.But it does not mention the interface with the library. 3)How Can we achive MIC and LINEIN switching using the mixer?What library function do i need to use to achive this? 4)If i want to get the Timestamp for each buffer captured, what library function should be used? 5)I am trying to capture and play the same audio (ALSA duplex). When i capture and store it in a file and play from the file i am able to hear the sound properly. If i capture in a buffer and play the same buffer directly it says Broken PIPE ERROR in the playout. I tried the capture and playout in two different threads, i am able to hear the sound and i am not getting any Broken PIPE error. But along with the audio i am hearing some noise. Do we need to use some kind of synchronisation between the threads for capture and playout? Any help on these things is appreciated. Thank you in advance. Regards, santosh

1 0

[alsa-devel] View alsa-lib plugins setup for any app?
by Eliot Blennerhassett 23 Oct '08

23 Oct '08

Greetings, running aplay -v will show the plugin chain setup. /proc/asound/card/pcm/* shows hw and sw params for the card driver when in use by any app Is there any way to show the alsa-lib plugin settings a-la aplay for an arbitrary application? thanks -- Eliot

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[alsa-devel] Fwd: multiseat and alsa audio systems (usb headsets and usb audio adapters)
by Sean McNamara 22 Oct '08

22 Oct '08

Oops! Forgot to include the ML on this reply. Maybe someone else will find this info useful, or be able to add constructive comments to what has been said :) Sean ---------- Forwarded message ---------- From: Sean McNamara <smcnam(a)gmail.com> Date: Wed, Oct 22, 2008 at 4:17 PM Subject: Re: [alsa-devel] multiseat and alsa audio systems (usb headsets and usb audio adapters) To: Jelle de Jong <jelledejong(a)powercraft.nl> On Wed, Oct 22, 2008 at 1:48 PM, Jelle de Jong <jelledejong(a)powercraft.nl> wrote: > Hello everybody, > > I have a four to eighth muliseat debian linux system here and it is > missing sound for the users. I was hoping the alsa team can help me with > this issue. All sound issues on Linux are extremely application-specific. Getting sound to work without any specific application in mind is asking for trouble, because there is no single configuration of the sound stack (which, if you expand into historical sound APIs and servers, contains about 20 different ways to play sound) that will satisfy any application without further configuration. Since this is not a really well-formed question, I will proceed with the assumption that you want to be able to run, for example, aplay /usr/share/sounds/pop.wav in the console, and have it produce sound in the "appropriate" headset. Here's how you proceed: 1. Each person on the multiseat box must have their own UNIX user account. 2. Each user account must have an ~/.asoundrc file. 3. Each ~/.asoundrc file must redefine the "default" device. How you redefine it depends on a lot of things: (a) Do you want _direct_ hardware access to the sound card, i.e. one application at a time? (b) Do you want to use ALSA's built-in software mixing, i.e. dmix? (c) How many channels? (d) Do you want to use PulseAudio? (e) Do you want to use JACK? (f) For each USB headset, what is its corresponding "raw device string"? This will be something like hw:n, where n is a number starting from 0, depending on the order in which the USB drivers initialize each USB controller. All of the above questions, and others, will determine exactly how each user configures their ~/.asoundrc file. The *extremely naive* (not recommended) example of an ~/.asoundrc is like: pcm.!default { card 3 } to have ALSA clients play sound, by default, to the fourth sound device that happened to be initialized. > > I have usb headsets and usb audio devices and all users have there own > usb hub. There are no internal audio cards in the multiseat server. > > How can we configure an system that the user can listen to audio only to > the devices connected to his usb hub. Well, this is guaranteed by design, isn't it? Without physical access to someone else's headset, how would I get access to the audio streams being played to that headset? I don't think this really expresses what your problem is. If we flip your question around, it might be a little more interesting: "How can we configure a system so that each user's applications will only play audio to that user's corresponding headset?" You can interpret this question in two ways: 1. "How can we _prevent_ users from playing sound to each other's sound cards?" [mutually untrust[ed/ing] users] or 2. "How can we configure the default settings so that, if untampered with, a user's applications will play sound to that user's own headset and not anyone else's?" [mutually trust[ed/ing] users] The lack of hardware mixing actually comes back to being a security benefit to user space control in this kind of situation. Recall that if I run aplay --device=hw:0 /usr/share/sounds/pop.wav then as long as the aplay program is playing sound to hw:0, it is *impossible* (short of some really devious hacking in alsa-lib or the kernel) for another application -- from this user, or another -- to also play sound to hw:0 at the same time. I'll call this claim "No Native SW Mixing". Now, let's look at what is required to satisfy the question in each of the two above cases. Trusted: 1. We know that users aren't going to maliciously try and hurt one another's ears by setting someone else's mixer really loud and playing static to them. 2. We know that users won't try to tamper with eachother's processes or configuration files; once things are configured correctly, it'll stay that way. 3. By 'No Native SW Mixing', we know that users can *theoretically* hog another user's sound device whenever that user has zero applications currently using it. 4. Since users trust each other, they won't do this. 5. We can use something like dmix, or just use hw:0 if users don't need simultaneous app output, and things will be fine. 6. dmix is a viable option to implement software mixing. 7. PulseAudio is also a viable option to implement software mixing. Untrusted: 1. Users may want to maliciously try and hurt one another's ears by setting someone else's mixer really loud and playing static to them. 2. Users may try to tamper with eachother's processes or configuration files; once things are configured correctly, it could get messed up again. [There's not much you can do to prevent this ... at least not based on existing open source tools.] 3. By 'No Native SW Mixing', we know that users can *theoretically* hog another user's sound device whenever that user has zero applications currently using it. 4. Since users do not trust each other, there is potential for 3 to happen. 5. We must use 'No Native SW Mixing' to our advantage, to create a walled garden around each user's sound experience. We need a process that, once started by a user, will permanently "hog" the sound card that user wants to use. This process should, ideally, allow other processes run by that user (and by that user only) to access their USB headset for output or input. We need a "gatekeeper" that is sensitive to the context in which an app is being run. 6. dmix is a NOT viable option to implement software mixing, because: (a) Once all dmix applications close their streams, the audio device hw:n is available, and some malicious user can decide to acquire it, abuse it, and prevent its rightful user from accessing it with his own applications. (b) dmix does not restrict users from outputting sound based on their UID or session. It is NOT a gatekeeper, just an open gate. 7. PulseAudio is [perhaps the best] viable solution to implement software mixing for each respective user. It is also a viable solution to preventing users from taking control of one another's sound devices. Configuring pulseaudio for a multi-soundcard multi-user system is out of the scope of this mailing list, but suffice it to say that it can be done. You will want to run one PulseAudio daemon per user, and each daemon will hold exclusive control over exactly one USB headset. Each daemon will only accept clients' requests for audio I/O from the user who started the daemon. Do make sure that you disable module-suspend-on-idle, or PulseAudio will actually give up the sound device after a period of inactivity, making it insecure in an untrusted environment. BTW, if this project is for a commercial interest, you usually pay someone for this kind of in-depth analysis. If you're that someone, you might want to read up on some of the underlying technologies so that you can develop this sort of reasoning on your own. I can envision a very real situation for both the trusted and the untrusted environment, so there is definitely a demand for this kind of specialist. Trusted environments are popular at workplaces, and untrusted multi-seat boxes are popular at computer labs, public libraries, etc. > > Any help is really appreciated. Please let me know if this information was helpful -- it will help me gauge, in turn, whether I should spend my time writing up these sorts of answers on the ML. FWIW, I searched around for existing articles touching this subject, but I couldn't find anyone who addresses this specific issue accurately or in enough detail. Sean P.S. - I once spoke to a user on IRC who was having intermittent problems with multiple USB headsets. It turns out that the USB controllers in the computer didn't have enough bandwidth, or power (or both) to allow all of the headsets to play at once. If, after configuring, you run into issues where only so many users can play sound at once, then you should invest in additional PCI or (preferably) PCI-Express USB controllers. Note that adding hubs upstream of a USB controller does absolutely nothing to lessen the load on the USB controllers that are integrated into your motherboard. > > Thank in advance, > > Jelle > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >

1 0

[alsa-devel] multiseat and alsa audio systems (usb headsets and usb audio adapters)
by Jelle de Jong 22 Oct '08

22 Oct '08

Hello everybody, I have a four to eighth muliseat debian linux system here and it is missing sound for the users. I was hoping the alsa team can help me with this issue. I have usb headsets and usb audio devices and all users have there own usb hub. There are no internal audio cards in the multiseat server. How can we configure an system that the user can listen to audio only to the devices connected to his usb hub. Any help is really appreciated. Thank in advance, Jelle

1 0

Re: [alsa-devel] Nvidia hdmi problems
by Takashi Iwai 22 Oct '08

22 Oct '08

At Tue, 21 Oct 2008 14:46:19 +0200, Nir Tzachar wrote: > > >> The bad news are that mplayer misbehaves and will get stuck every time > >> I try to change the volume. However, this does not happen in vlc. Any > >> ideas? > > > > Well, we need to sort out where the problem comes from. > > Try the vanilla Linus tree without sound-2.6 whether the problem > > persists. > > I'll try to remove the sound-2.6 tree and report. Nir, do you have any results? I tried to reproduce the problem by myself, but I couldn't. So I wonder whether it's a codec specific or I missed something... thanks, Takashi

2 1

[alsa-devel] [PATCH] ASoC: OMAP: Continue fixing DSP DAI format in McBSP DAI driver
by Jarkko Nikula 22 Oct '08

22 Oct '08

Fix "ASoC: OMAP: Fix DSP DAI format in McBSP DAI driver" was not correct due misunderstanding of DSP_A format and similar error in TLV320AIC33 codec which was used to test the original fix. This patch corrects now DSP_A format in OMAP McBSP DAI driver and is verified with TLV320AIC23 codec that's implementing DSP_A correctly. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> Cc: Arun KS <arunks(a)mistralsolutions.com> Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> --- sound/soc/omap/omap-mcbsp.c | 7 ++----- 1 files changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 853b33a..8485a8a 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -265,7 +265,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, break; case SND_SOC_DAIFMT_DSP_A: regs->srgr2 |= FPER(wlen * 2 - 1); - regs->srgr1 |= FWID(0); + regs->srgr1 |= FWID(wlen * 2 - 2); break; } @@ -284,7 +284,6 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, { struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; - unsigned int temp_fmt = fmt; if (mcbsp_data->configured) return 0; @@ -307,8 +306,6 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, /* 0-bit data delay */ regs->rcr2 |= RDATDLY(0); regs->xcr2 |= XDATDLY(0); - /* Invert bit clock and FS polarity configuration for DSP_A */ - temp_fmt ^= SND_SOC_DAIFMT_IB_IF; break; default: /* Unsupported data format */ @@ -332,7 +329,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, } /* Set bit clock (CLKX/CLKR) and FS polarities */ - switch (temp_fmt & SND_SOC_DAIFMT_INV_MASK) { + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: /* * Normal BCLK + FS. -- 1.5.6.5

2 1

[alsa-devel] [PATCH 2.6.28] ALSA: Ensure PXA runtime data is initialised
by Mark Brown 22 Oct '08

22 Oct '08

The rest of the code relies on the runtime data being zero initialised so we need to use kzalloc() to allocate it. Reported-by: Oliver Ford <ipaqlinux(a)oliford.co.uk> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/arm/pxa2xx-pcm-lib.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index 1c93eb7..75a0d74 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -194,7 +194,7 @@ int __pxa2xx_pcm_open(struct snd_pcm_substream *substream) goto out; ret = -ENOMEM; - rtd = kmalloc(sizeof(*rtd), GFP_KERNEL); + rtd = kzalloc(sizeof(*rtd), GFP_KERNEL); if (!rtd) goto out; rtd->dma_desc_array = -- 1.5.6.5

2 1

Re: [alsa-devel] [PATCH] ALSA: emu10k1: fix device names for Live!/Audigy1/2/4/E-mu and coding style
by Tony Vroon 22 Oct '08

22 Oct '08

On a related note, it looks to me as if the leading two characters (the manufacturing location) have been erroneously included for E-MU products. You'll see this MA code on some sound cards as well (MASBxxxx) Regards, Tony V.

2 1

[alsa-devel] [PATCH] ASoC: OMAP: Fix DSP DAI format in McBSP DAI driver
by Jarkko Nikula 22 Oct '08

22 Oct '08

Fix word clock length which must equal to one bit clock cycle in DSP mode. Surprisingly McBSP is able synchronize into wrong length when it's slave but e.g. TLV320AIC33 codec in slave configuration is outputting some amount of noise if word clock length is longer than one bit clock cycle. Fix also bit clock and frame sync polarities in DSP mode since they are opposite from I2S. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> Cc: Arun KS <arunks(a)mistralsolutions.com> Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> --- sound/soc/omap/omap-mcbsp.c | 24 ++++++++++++++++++++---- 1 files changed, 20 insertions(+), 4 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 0a063a9..853b33a 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -43,6 +43,7 @@ struct omap_mcbsp_data { unsigned int bus_id; struct omap_mcbsp_reg_cfg regs; + unsigned int fmt; /* * Flags indicating is the bus already activated and configured by * another substream @@ -200,6 +201,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; int dma, bus_id = mcbsp_data->bus_id, id = cpu_dai->id; + int wlen; unsigned long port; if (cpu_class_is_omap1()) { @@ -244,19 +246,29 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: /* Set word lengths */ + wlen = 16; regs->rcr2 |= RWDLEN2(OMAP_MCBSP_WORD_16); regs->rcr1 |= RWDLEN1(OMAP_MCBSP_WORD_16); regs->xcr2 |= XWDLEN2(OMAP_MCBSP_WORD_16); regs->xcr1 |= XWDLEN1(OMAP_MCBSP_WORD_16); - /* Set FS period and length in terms of bit clock periods */ - regs->srgr2 |= FPER(16 * 2 - 1); - regs->srgr1 |= FWID(16 - 1); break; default: /* Unsupported PCM format */ return -EINVAL; } + /* Set FS period and length in terms of bit clock periods */ + switch (mcbsp_data->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + regs->srgr2 |= FPER(wlen * 2 - 1); + regs->srgr1 |= FWID(wlen - 1); + break; + case SND_SOC_DAIFMT_DSP_A: + regs->srgr2 |= FPER(wlen * 2 - 1); + regs->srgr1 |= FWID(0); + break; + } + omap_mcbsp_config(bus_id, &mcbsp_data->regs); mcbsp_data->configured = 1; @@ -272,10 +284,12 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, { struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; + unsigned int temp_fmt = fmt; if (mcbsp_data->configured) return 0; + mcbsp_data->fmt = fmt; memset(regs, 0, sizeof(*regs)); /* Generic McBSP register settings */ regs->spcr2 |= XINTM(3) | FREE; @@ -293,6 +307,8 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, /* 0-bit data delay */ regs->rcr2 |= RDATDLY(0); regs->xcr2 |= XDATDLY(0); + /* Invert bit clock and FS polarity configuration for DSP_A */ + temp_fmt ^= SND_SOC_DAIFMT_IB_IF; break; default: /* Unsupported data format */ @@ -316,7 +332,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, } /* Set bit clock (CLKX/CLKR) and FS polarities */ - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + switch (temp_fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: /* * Normal BCLK + FS. -- 1.5.6.5

4 8

[alsa-devel] [PATCH] ALSA: emu10k1: fix device names for Live!/Audigy1/2/4/E-mu and coding style
by Vedran Miletić 22 Oct '08

22 Oct '08

ALSA: emu10k1: fix device names for Live!/Audigy1/2/4/E-mu and coding style Changelog: * added missing SBxxxx, CTxxxx, PCxxx and MAEMxxxx where they were missing, and fixed some of them which were wrong (according to kx.inf, which is pretty accurate compared to anything out there) * fixed device names to make them more consistent across various cards * fixed lots of coding style errors and some warnings, now it is down to: checkpatch.pl-0.24 --no-tree --file --strict --terse emu10k1_main.c total: 0 errors, 62 warnings, 7 checks, 2074 lines checked Signed-off-by: Vedran Miletic <rivanvx(a)gmail.com> -- Vedran Miletić

3 7

[alsa-devel] help test sbxfi
by William Zumwalt 22 Oct '08

22 Oct '08

I've been very interested in a working driver for the X-Fi for a long time, saw work was finally started, and would like to help test things out. Can someone list where the source is and the steps needed to get this working on linux? Kernel version, which X-Fi card, etc... I'd like to help and follow along if possible.

3 2

[alsa-devel] Intel dg45id motherboard - ICH10 HD audio issues
by Anton Blanchard 22 Oct '08

22 Oct '08

Hi, I was unable to get sound to work (analog or SPDIF) on my dg45id using 2.6.27. After looking at the pinout information I made a quick hack to get SPDIF out going (it's a mythtv box so thats all I need): Index: linux-source-2.6.27/sound/pci/hda/patch_sigmatel.c =================================================================== --- linux-source-2.6.27.orig/sound/pci/hda/patch_sigmatel.c 2008-08-31 11:06:37.000000000 +1000 +++ linux-source-2.6.27/sound/pci/hda/patch_sigmatel.c 2008-08-31 11:07:14.000000000 +1000 @@ -3591,7 +3591,7 @@ spec->num_pwrs = ARRAY_SIZE(stac92hd73xx_pwr_nids); spec->pwr_nids = stac92hd73xx_pwr_nids; - err = stac92xx_parse_auto_config(codec, 0x22, 0x24); + err = stac92xx_parse_auto_config(codec, 0x26, 0x24); if (!err) { if (spec->board_config < 0) { And pointed mythtv at ALSA:plughw:0,1. Quick and dirty I know :) Details of the chipset are below. Anton -- # lspci -vvnn 00:1b.0 Audio device [0403]: Intel Corporation 82801JI (ICH10 Family) HD Audio Controller [8086:3a3e] Subsystem: Intel Corporation Device [8086:5002] Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 22 Region 0: Memory at fe820000 (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 (v1) Root Complex Integrated Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s <64ns, L1 <1us ExtTag- RBE- FLReset+ DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr+ TransPend- LnkCap: Port #0, Speed unknown, Width x0, ASPM unknown, Latency L0 <64ns, L1 <1us ClockPM- Suprise- LLActRep- BwNot- LnkCtl: ASPM Disabled; Disabled- Retrain- CommClk- ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed unknown, Width x0, TrErr- Train- SlotClk- DLActive- BWMgmt- ABWMgmt- Capabilities: [100] Virtual Channel <?> Capabilities: [130] Root Complex Link <?> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel # aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: STAC92xx Digital [STAC92xx Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 # aplay -L default:CARD=Intel HDA Intel, STAC92xx Analog Default Audio Device front:CARD=Intel,DEV=0 HDA Intel, STAC92xx Analog Front speakers surround40:CARD=Intel,DEV=0 HDA Intel, STAC92xx Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=Intel,DEV=0 HDA Intel, STAC92xx Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=Intel,DEV=0 HDA Intel, STAC92xx Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=Intel,DEV=0 HDA Intel, STAC92xx Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=Intel,DEV=0 HDA Intel, STAC92xx Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers hdmi:CARD=Intel,DEV=0 HDA Intel HDMI Audio Output null Discard all samples (playback) or generate zero samples (capture) # cat /proc/asound/card0/codec#2 Codec: IDT 92HD73E1X5 Address: 2 Vendor Id: 0x111d7676 Subsystem Id: 0x80865002 Revision Id: 0x100202 No Modem Function Group found Default PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Default Amp-Out caps: ofs=0x7f, nsteps=0x7f, stepsize=0x02, mute=1 GPIO: io=8, o=0, i=0, unsolicited=1, wake=1 IO[0]: enable=1, dir=1, wake=0, sticky=0, data=1 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 IO[4]: enable=0, dir=0, wake=0, sticky=0, data=0 IO[5]: enable=0, dir=0, wake=0, sticky=0, data=0 IO[6]: enable=0, dir=0, wake=0, sticky=0, data=0 IO[7]: enable=0, dir=0, wake=0, sticky=0, data=0 Node 0x0a [Pin Complex] wcaps 0x400183: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Pincap 0x08173f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x0321101f: [Jack] HP Out at Ext Left Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0xc0: OUT HP VREF_HIZ Unsolicited: tag=30, enabled=1 Connection: 6 0x15 0x16* 0x17 0x18 0x19 0x1e Node 0x0b [Pin Complex] wcaps 0x400183: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Pincap 0x08173f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x4f00000f: [N/A] Line Out at Ext UNKNOWN Conn = Unknown, Color = Unknown DefAssociation = 0x0, Sequence = 0xf Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 6 0x15* 0x16 0x17 0x18 0x19 0x1e Node 0x0c [Pin Complex] wcaps 0x400183: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Pincap 0x081737: IN OUT Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x4f0000f0: [N/A] Line Out at Ext UNKNOWN Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 6 0x15* 0x16 0x17 0x18 0x19 0x1e Node 0x0d [Pin Complex] wcaps 0x400183: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Pincap 0x083f: IN OUT HP Detect Trigger ImpSense Pin Default 0x90170110: [Fixed] Speaker at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 6 0x15* 0x16 0x17 0x18 0x19 0x1e Node 0x0e [Pin Complex] wcaps 0x400183: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Pincap 0x081737: IN OUT Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x03a11020: [Jack] Mic at Ext Left Conn = 1/8, Color = Black DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=30, enabled=1 Connection: 6 0x15* 0x16 0x17 0x18 0x19 0x1e Node 0x0f [Pin Complex] wcaps 0x400183: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Pincap 0x0837: IN OUT Detect Trigger ImpSense Pin Default 0x0321101f: [Jack] HP Out at Ext Left Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0x40: OUT Unsolicited: tag=35, enabled=1 Connection: 6 0x15 0x16 0x17* 0x18 0x19 0x1e Node 0x10 [Pin Complex] wcaps 0x400183: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Pincap 0x0837: IN OUT Detect Trigger ImpSense Pin Default 0x4f0000f0: [N/A] Line Out at Ext UNKNOWN Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 6 0x15* 0x16 0x17 0x18 0x19 0x1e Node 0x11 [Pin Complex] wcaps 0x400183: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Pincap 0x0837: IN OUT Detect Trigger ImpSense Pin Default 0x4f0000f0: [N/A] Line Out at Ext UNKNOWN Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 6 0x15* 0x16 0x17 0x18 0x19 0x1e Node 0x12 [Pin Complex] wcaps 0x400081: Stereo Pincap 0x0824: IN Detect Pin Default 0x4f0000f0: [N/A] Line Out at Ext UNKNOWN Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Node 0x13 [Pin Complex] wcaps 0x400003: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Pincap 0x0820: IN Pin Default 0x90a60160: [Fixed] Mic at Int N/A Conn = Digital, Color = Unknown DefAssociation = 0x6, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Node 0x14 [Pin Complex] wcaps 0x400003: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Pincap 0x0820: IN Pin Default 0x4f0000f0: [N/A] Line Out at Ext UNKNOWN Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x00: Node 0x15 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Amp-Out caps: N/A Amp-Out vals: [0x40 0x40] Converter: stream=0, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x16 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Amp-Out caps: N/A Amp-Out vals: [0xd9 0xd9] Converter: stream=0, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x17 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Amp-Out caps: N/A Amp-Out vals: [0xd9 0xd9] Converter: stream=0, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x18 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Amp-Out caps: N/A Amp-Out vals: [0xff 0xff] Converter: stream=0, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x19 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Amp-Out caps: N/A Amp-Out vals: [0x19 0x19] Converter: stream=0, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x1a [Audio Input] wcaps 0x1d0541: Stereo Converter: stream=0, channel=0 SDI-Select: 0 Power: setting=D0, actual=D0 Delay: 13 samples Connection: 1 0x20 Processing caps: benign=0, ncoeff=0 Node 0x1b [Audio Input] wcaps 0x1d0541: Stereo Converter: stream=0, channel=0 SDI-Select: 0 Power: setting=D0, actual=D0 Delay: 13 samples Connection: 1 0x21 Processing caps: benign=0, ncoeff=0 Node 0x1c [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x03, nsteps=0x03, stepsize=0x17, mute=1 Amp-Out vals: [0x00] Node 0x1d [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x97 0x97] [0x97 0x97] [0x97 0x97] [0x97 0x97] [0x97 0x97] Connection: 5 0x28 0x29 0x2a 0x2b 0x12 Node 0x1e [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x9f 0x9f] Connection: 1 0x1d Node 0x1f [Volume Knob Widget] wcaps 0x600000: Mono Volume-Knob: delta=1, steps=127, direct=1, val=108 Connection: 5 0x15* 0x16 0x17 0x18 0x19 Node 0x20 [Audio Selector] wcaps 0x30090d: Stereo Amp-Out R/L Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 Amp-Out vals: [0x80 0x80] Connection: 12 0x0a 0x0b 0x0c 0x0d 0x0e 0x0f 0x10 0x11 0x12 0x13 0x14 0x1d* Node 0x21 [Audio Selector] wcaps 0x30090d: Stereo Amp-Out R/L Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 Amp-Out vals: [0x80 0x80] Connection: 12 0x0a 0x0b 0x0c 0x0d 0x0e 0x0f 0x10 0x11 0x12 0x13 0x14 0x1d* Node 0x22 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x0810: OUT Pin Default 0x4f0000f0: [N/A] Line Out at Ext UNKNOWN Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x00: Connection: 3 0x25* 0x20 0x21 Node 0x23 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x0810: OUT Pin Default 0x01451140: [Jack] SPDIF Out at Ext Rear Conn = Optical, Color = Black DefAssociation = 0x4, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 3 0x26* 0x20 0x21 Node 0x24 [Pin Complex] wcaps 0x400681: Stereo Digital Pincap 0x0810024: IN EAPD Detect EAPD 0x0: Pin Default 0x40f000f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Node 0x25 [Audio Output] wcaps 0x4021d: Stereo Digital Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Delay: 4 samples Node 0x26 [Audio Output] wcaps 0x4021d: Stereo Digital Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Delay: 4 samples Node 0x27 [Audio Input] wcaps 0x14031b: Stereo Digital Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 Digital: Digital category: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Delay: 4 samples Connection: 1 0x24 Node 0x28 [Audio Selector] wcaps 0x300101: Stereo Connection: 4 0x0a 0x0b* 0x0d 0x0f Node 0x29 [Audio Selector] wcaps 0x300101: Stereo Connection: 4 0x0a 0x0e* 0x10 0x11 Node 0x2a [Audio Selector] wcaps 0x300101: Stereo Connection: 4 0x0b 0x0c* 0x10 0x11 Node 0x2b [Audio Selector] wcaps 0x300101: Stereo Connection: 4 0x15* 0x16 0x17 0x18

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[alsa-devel] quick question, BT codec???
by John Kacur 22 Oct '08

22 Oct '08

Hello, quick question, both s3c24xx-pcm.c and pxa2xx-pcm.c contain the following comment and code segment, what does the "BT codec" refer to? /* return if this is a bufferless transfer e.g. * codec <--> BT codec or GSM modem -- lg FIXME */ if (!dma) return 0;

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[alsa-devel] 20K1 PCI bars, and mode switching info.
by James Courtier-Dutton 22 Oct '08

22 Oct '08

This info should help the current snd-sbxfi driver to actually work with the "Vista compatible" cards that it currently does not work with. I believe the true UAA to 20K1 mode switch code is missing from the current alsa snd-sbxfi driver. PCI BARs UAA mode, or Vista Compatibility mode. BAR0: 16K for UAA 14h BAR1: 2M for 20k1 (low 32 bit) 18h 2M for 20k1 (high 32 bit) 1Ch BAR2: Between 2 and 256M for 20k1 (low 32 bit) 20h Between 2 and 256M for 20k1 (high 32 bit) 24h BAR3: 32 bit I/O for 20K1 20K1 Mode BAR0: 32 bit I/O for 20K1 14h BAR1: 2M for 20k1 (low 32 bit) 18h 2M for 20k1 (high 32 bit) 1Ch BAR2: Between 2 and 256M for 20k1 (low 32 bit) 20h Between 2 and 256M for 20k1 (high 32 bit) 24h "UAA/20K1 mode" and "config space selection" work independently of each other. 1) Configuration space selection: This allows the driver to talk to both the UAA and 20K1 configuration space whichever mode (UAA or 20K1) the card is in. Location: PCI Register 0xA0 for "20K1 Configuration Space Flag" register. Value: 0x12345678 to select 20K1 without BAR0 exposed; Value: 0x87654321 to select 20K1 with BAR0 exposed; (should disable immediately after using.) Value: 0x00000000 to select UAA. 2) UAA/20K1 MODE CHANGE (controls who the bus master is): The "Mode Change" register is located in the UAA configuration space at location offset 0x3FFC. This location stores 4 values in a ring. To switch from 20K1 mode to UAA mode write each DW value: CTLA, CTLZ, CTLL, CTLA. To switch from UAA mode to 20K1 mode write each DW value: CTLX, CTL-, CTLF, CTLi To read the current mode, the 4 value ring may be in any position, so one might read for example: CTLL, CTLA, CTLA, CTLZ At power on, the card is in UAA mode. In UAA mode, the UAA chip is the bus master. In 20K1 mode, the 20K1 chip is the bus master. Under either UAA mode or 20K1 mode, the driver can read and write the configuration space of both UAA chip and 20K1 chip.

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Re: [alsa-devel] Nvidia hdmi problems
by Takashi Iwai 22 Oct '08

22 Oct '08

At Tue, 21 Oct 2008 14:46:19 +0200, Nir Tzachar wrote: > > Hello > > On Tue, Oct 21, 2008 at 2:40 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Tue, 21 Oct 2008 13:28:25 +0200, > > Nir Tzachar wrote: > >> > >> Hello. > >> > >> Updated to the latest linus-git tree, and pulled in the sound-2.6 tree. > >> > >> The good news are that by following Wei's instructions (adding the > >> device id in patch_nvhdmi.c), I managed to get audio through the hdmi. > >> Goodie. > > > > Could you give details? > > > > I just did this: > > > You can try to add device id (0x10de8001) to snd_hda_preset_nvhdmi > > table in patch_nvhdmi.c. > > You also need to install latest display driver. > > and told alsa to play through the third pcm device. Thanks. So it's just a missing entry. Wei, do you have a patch for this addition? > >> The bad news are that mplayer misbehaves and will get stuck every time > >> I try to change the volume. However, this does not happen in vlc. Any > >> ideas? > > > > Well, we need to sort out where the problem comes from. > > Try the vanilla Linus tree without sound-2.6 whether the problem > > persists. > > I'll try to remove the sound-2.6 tree and report. That'll be helpful. Thanks. Takashi

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