April 2007 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Functioning support for Prodigy 192
by dustin＠seznam.cz 12 Apr '07

12 Apr '07

Hi Takashi, I am finishing changes to prodigy192.c to make the card functional. I modified eeprom values (e.g. 49MHz crystal), fixed digital output of the card (requires GPIO), added support for MI/ODI/O and added input switch mic/line. So far I have not done any MIDI tests. Before posting corresponding patches I would like to resolve a few remaining issues. The ak4114 based miodio card is a separately sold add-on and provides SPDIF out (directly connected to SPDIFOUT pin of ice1724), SPDIF IN (TOSLINK to input 0, coax to input 1 of ak4114) + MIDI IN/OUT. SPDIF OUT required no code, SPDIF IN is fully working with my last patches, switching between toslink/coax works in my prodigy192.c. Windows drivers detect miodio automatically, offering corresponding controls when connected. What would be a proper way in the driver to activate the miodio support? There is one Xilinx FPGA chip onboard, perhaps it is used for MIDI (although ice1724 has a built-in support, right?), perhaps it informs original windows drivers about version and capabilities of the addon card. Another option would be to test communication with ak4114, whether we can read contents of a particular register. Perhaps a proper place to do so would be in snd_ak4114_create of ak4114.c which could return error when no communication could be established with ak4114 (I do not think this is the case now). What solution would you suggest? Another question concerns capture input switches, both analog and digital. For now I have used a single switch SNDRV_CTL_ELEM_TYPE_BOOLEAN for each. On the capture tab of alsamixer that is not intuitive (capture off = mice, capture on = line etc.). I would prefer standard "capture input radio buttons" like e.g. in ac97 codec (ich5 + AD1985). I would need two independent groups of two "radio buttons". Please could you point me to some self-explaining code implementing similar feature in another driver which a newbie could understand? Thanks a lot. Pavel.

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[alsa-devel] ALSA Wiki
by Jaroslav Kysela 12 Apr '07

12 Apr '07

Hello all, new Wiki running well known MediaWiki (wikipedia) which will replace ALSA WWW contents is prepared at URL http://bugtrack.alsa-project.org/main/ . I would like ask possible volunteers to start moving the current contents of ALSA WWW server to this Wiki. Also, if you like to be a WikiSysop to change at least some Wiki layout - you have a chance now. Also, we need sysops to rollback wrong changes. The edit policy is simple. You have to be logged in and you MUST acknowledge your e-mail address. I'm sorry, but we cannot take contents from anonymous users. Thanks, Jaroslav ----- Jaroslav Kysela <perex(a)suse.cz> Linux Kernel Sound Maintainer ALSA Project, SUSE Labs

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[alsa-devel] Unblock MIDI IN in blocking read mode
by Florian 12 Apr '07

12 Apr '07

Hi, for best performance, I use a blocking snd_rawmidi_read() on an ALSA MIDI input device, byte by byte. It works fine, except: 1) How do I close the device from a different thread? I.e. how can I unblock the blocking read() call when it's time to close the device and no MIDI data is currently coming in? Just calling snd_rawmidi_close() will cause a crash. Using snd_rawmidi_drop() did not release the blocking read() in the other thread. 2) How is a blocking read() typically implemented in the ALSA MIDI driver? Maybe I can resort to non-blocking mode and avoid the situation above. Thanks, Florian -- Florian Bomers bome.com ------------------------------------------------------- Music Software, Development Tools: http://www.bome.com Java Sound extensions, plugins: http://www.tritonus.org The Java Sound Resources: http://www.jsresources.org ------------------------------------------------------- Please quote this email in your reply. Thanks!

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[alsa-devel] [PATCH] ALSA: Support Media Vision Jazz16 chips
by Rask Ingemann Lambertsen 11 Apr '07

11 Apr '07

From: Rask Ingemann Lambertsen <rask(a)sygehus.dk> - Add support for Jazz16 features to the common SoundBlaster code. - Add a front end for Jazz16 based sound cards. Tested with a Jazz16 based sound card and Linux 2.6.20. Signed-off-by: Rask Ingemann Lambertsen <rask(a)sygehus.dk> --- This patch is a prerequisite for the Jazz PnP patch, which needs the patched include/sound/sb.h. Documentation/sound/alsa/ALSA-Configuration.txt | 10 include/sound/sb.h | 5 sound/isa/Kconfig | 15 + sound/isa/sb/Makefile | 2 sound/isa/sb/jazz16.c | 193 ++++++++++++++ sound/isa/sb/sb8_main.c | 105 +++++++ sound/isa/sb/sb_common.c | 12 sound/isa/sb/sb_mixer.c | 3 8 files changed, 335 insertions(+), 10 deletions(-) --- linux-2.6.20/sound/isa/sb/Makefile.orig 2007-02-19 21:00:09.000000000 +0100 +++ linux-2.6.20/sound/isa/sb/Makefile 2007-03-02 20:14:48.000000000 +0100 @@ -12,6 +12,7 @@ snd-sb16-objs := sb16.o snd-sbawe-objs := sbawe.o emu8000.o snd-emu8000-synth-objs := emu8000_synth.o emu8000_callback.o emu8000_patch.o emu8000_pcm.o snd-es968-objs := es968.o +snd-jazz16-objs := jazz16.o # # this function returns: @@ -30,6 +31,7 @@ obj-$(CONFIG_SND_SB16) += snd-sb16.o snd obj-$(CONFIG_SND_SBAWE) += snd-sbawe.o snd-sb16-dsp.o snd-sb-common.o obj-$(CONFIG_SND_ES968) += snd-es968.o snd-sb8-dsp.o snd-sb-common.o obj-$(CONFIG_SND_ALS4000) += snd-sb-common.o +obj-$(CONFIG_SND_JAZZ16) += snd-jazz16.o snd-sb8-dsp.o snd-sb-common.o ifeq ($(CONFIG_SND_SB16_CSP),y) obj-$(CONFIG_SND_SB16) += snd-sb16-csp.o obj-$(CONFIG_SND_SBAWE) += snd-sb16-csp.o --- linux-2.6.20/sound/isa/sb/jazz16.c-orig 2007-02-17 20:01:40.000000000 +0100 +++ linux-2.6.20/sound/isa/sb/jazz16.c 2007-03-18 13:43:35.000000000 +0100 @@ -0,0 +1,193 @@ +/* Driver for Media Vision Jazz16 based boards. + * + * Copyright (c) 2007 by Rask Ingemann Lambertsen + * + * 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 + * + * The Jazz16 is an SB Pro compatible chip with a 16-bit mode and higher + * playback and capture rates added to it. It is not SB 16 compatible. + * There is also an MPU-401 interface. + * + * The IBM PPS Model 6015 has a Jazz16 chip on board. Please tell me if it + * works with this driver or not. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/errno.h> +#include <linux/slab.h> +#include <linux/init.h> +#include <linux/pnp.h> +#include <linux/ioport.h> +#include <asm/io.h> +#include <linux/delay.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/sb.h> +#include <sound/mpu401.h> +#include <sound/opl3.h> +#include <sound/initval.h> + +MODULE_AUTHOR ("Rask Ingemann Lambertsen <rask(a)sygehus.dk>"); +MODULE_DESCRIPTION ("Jazz16"); +MODULE_LICENSE("GPL"); + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable switches */ + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for Jazz16 soundcard."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for Jazz16 soundcard."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable Jazz16 soundcard."); + +#define PFX "jazz16: " +#define PFX_WARN KERN_WARNING PFX +#define PFX_ERR KERN_ERR PFX + +static struct pnp_driver snd_jazz16_pnp_driver; + +static irqreturn_t jazz16_interrupt(int irq, void *dev_id) +{ + struct snd_sb *chip = (struct snd_sb *) dev_id; + return snd_sb8dsp_interrupt(chip); +} + +static struct pnp_device_id snd_jazz16_pnpids[] = { + { .id = "PNPb00f" }, + { .id = "" } +}; +MODULE_DEVICE_TABLE(pnp, snd_jazz16_pnpids); + +static int __devinit snd_jazz16_pnp_probe(struct pnp_dev *pnp_dev, + const struct pnp_device_id *pnp_id) +{ struct snd_sb *chip; + struct snd_card *card; + struct snd_opl3 *opl3; + int err; + uint sbport, sbirq, sbdma8, sbdma16, mpuport, mpuirq; + static uint dev_num = 0; + + if (enable[dev_num]) + card = snd_card_new(index[dev_num], id[dev_num], THIS_MODULE, 0); + else + card = NULL; + dev_num ++; + if (card == NULL) + return -ENOMEM; + pnp_set_drvdata(pnp_dev, card); + /* TODO use pnp_port_valid(), pnp_port_flags(), pnp_port_length()... */ + sbport = pnp_port_start(pnp_dev, 0); + sbirq = pnp_irq(pnp_dev, 0); + sbdma8 = pnp_dma(pnp_dev, 0); + sbdma16 = pnp_dma(pnp_dev, 1); + err = snd_sbdsp_create(card, sbport, sbirq, jazz16_interrupt, + sbdma8, sbdma16, SB_HW_AUTO, &chip); + if (err < 0) { + snd_card_free(card); + return err; + } + /* TODO length limits - strncpy()/snprintf() and friends. */ + strcpy(card->driver, "Jazz16"); + strcpy(card->shortname, "Media Vision Jazz16"); + sprintf(card->longname, "%s at 0x%x, irq %u, dma8 %u, dma16 %u", + chip->name, sbport, sbirq, sbdma8, sbdma16); + + if (chip->hardware != SB_HW_JAZZ16) { + snd_printk(PFX_ERR "Not a Jazz16 chip at 0x%x.\n", sbport); + snd_card_free(card); + return -ENODEV; + } + err = snd_sb8dsp_pcm(chip, 0, NULL); + if (err < 0) { + snd_card_free(card); + return err; + } + err = snd_sbmixer_new(chip); + if (err < 0) { + snd_card_free(card); + return err; + } + err = snd_opl3_create(card, sbport, sbport + 2, + OPL3_HW_AUTO, 1, &opl3); + if (err < 0) { + snd_printk(PFX_WARN "No OPL device found, skipping.\n"); + } else { + err = snd_opl3_timer_new(opl3, 1, 2); + if (err < 0) { + snd_card_free(card); + return err; + } + err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); + if (err < 0) { + snd_card_free(card); + return err; + } + } + if (pnp_port_valid(pnp_dev, 1) + && !(pnp_port_flags(pnp_dev, 1) & IORESOURCE_DISABLED)) { + int mpuirqflags; + if (!pnp_irq_valid(pnp_dev, 1) + || pnp_irq_flags(pnp_dev, 1) & IORESOURCE_DISABLED) { + mpuirq = -1; + mpuirqflags = 0; + } else { + mpuirq = pnp_irq(pnp_dev, 1); + mpuirqflags = IRQF_DISABLED; + } + mpuport = pnp_port_start(pnp_dev, 1); + err = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, + mpuport, 0, mpuirq, + mpuirqflags, NULL); + if (err < 0) { + snd_printk(PFX_WARN "MPU-401 device not configured, skipping.\n"); + } + } + err = snd_card_register(card); + if (err < 0) { + snd_card_free(card); + return err; + } + return 0; +} + +static void __devexit snd_jazz16_pnp_remove(struct pnp_dev *dev) +{ + snd_card_free(pnp_get_drvdata(dev)); + pnp_set_drvdata(dev, NULL); +} + +static struct pnp_driver snd_jazz16_pnp_driver = { + .name = "Jazz16", + .id_table = snd_jazz16_pnpids, + .probe = snd_jazz16_pnp_probe, + .remove = __devexit_p(snd_jazz16_pnp_remove), +}; + +static int __devinit alsa_card_jazz16_init (void) +{ + return pnp_register_driver(&snd_jazz16_pnp_driver); +} + +static void __devexit alsa_card_jazz16_exit(void) +{ + pnp_unregister_driver(&snd_jazz16_pnp_driver); +} + +module_init (alsa_card_jazz16_init); +module_exit (alsa_card_jazz16_exit); --- linux-2.6.20/sound/isa/sb/sb_mixer.c.orig 2007-02-04 19:44:54.000000000 +0100 +++ linux-2.6.20/sound/isa/sb/sb_mixer.c 2007-02-25 01:09:05.000000000 +0100 @@ -811,6 +811,7 @@ int snd_sbmixer_new(struct snd_sb *chip) return err; break; case SB_HW_PRO: + case SB_HW_JAZZ16: if ((err = snd_sbmixer_init(chip, snd_sbpro_controls, ARRAY_SIZE(snd_sbpro_controls), @@ -946,6 +947,7 @@ void snd_sbmixer_suspend(struct snd_sb * save_mixer(chip, sb20_saved_regs, ARRAY_SIZE(sb20_saved_regs)); break; case SB_HW_PRO: + case SB_HW_JAZZ16: save_mixer(chip, sbpro_saved_regs, ARRAY_SIZE(sbpro_saved_regs)); break; case SB_HW_16: @@ -971,6 +973,7 @@ void snd_sbmixer_resume(struct snd_sb *c restore_mixer(chip, sb20_saved_regs, ARRAY_SIZE(sb20_saved_regs)); break; case SB_HW_PRO: + case SB_HW_JAZZ16: restore_mixer(chip, sbpro_saved_regs, ARRAY_SIZE(sbpro_saved_regs)); break; case SB_HW_16: --- linux-2.6.20/sound/isa/sb/sb_common.c.orig 2007-02-04 19:44:54.000000000 +0100 +++ linux-2.6.20/sound/isa/sb/sb_common.c 2007-03-03 12:50:32.000000000 +0100 @@ -146,8 +146,16 @@ static int snd_sbdsp_probe(struct snd_sb } break; case 3: - chip->hardware = SB_HW_PRO; - str = "Pro"; + spin_lock_irqsave(&chip->reg_lock, flags); + if (snd_sbdsp_command(chip, SB_DSP_GET_JAZZ_VERSION) && + SB_VERSION_IS_JAZZ16(snd_sbdsp_get_byte (chip))) { + chip->hardware = SB_HW_JAZZ16; + str = "Pro (Jazz16)"; + } else { + chip->hardware = SB_HW_PRO; + str = "Pro"; + } + spin_unlock_irqrestore (&chip->reg_lock, flags); break; case 4: chip->hardware = SB_HW_16; --- linux-2.6.20/sound/isa/sb/sb8_main.c.orig 2007-02-04 19:44:54.000000000 +0100 +++ linux-2.6.20/sound/isa/sb/sb8_main.c 2007-03-24 23:30:22.000000000 +0100 @@ -28,6 +28,9 @@ * * Wed Jul 12 22:02:55 CEST 2000 Uros Bizjak <uros(a)kss-loka.si> * Cleaned up and rewrote lowlevel routines. + * + * Sat Mar 24 23:29:20 CEST 2007 Rask Ingemann Lambertsen <rask(a)sygehus.dk> + * Added Jazz16 enhancements (ported from sound/oss/sb_common.c). */ #include <sound/driver.h> @@ -73,6 +76,15 @@ static struct snd_ratnum stereo_clocks[] } }; +/* For stereo playback and capture, the denominator is divided by two, so it + * must be even to get the intended sample rate. */ +static struct snd_ratnum jazz16_stereo_clock = { + .num = SB8_CLOCK, + .den_min = 2, + .den_max = 512, + .den_step = 2, +}; + static int snd_sb8_hw_constraint_rate_channels(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { @@ -101,14 +113,53 @@ static int snd_sb8_hw_constraint_channel return 0; } +static int snd_jazz16_hw_constraint_rate_channels(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_interval *c = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_interval *r = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + int err = 0; + + if (c->max > 1) { + unsigned int num = 0, den = 0; + err = snd_interval_ratnum(r, 1, &jazz16_stereo_clock, &num, &den); + if (err >= 0 && den) { + params->rate_num = num; + params->rate_den = den; + } + } + return err; +} + +static int snd_jazz16_hw_constraint_channels_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_interval *r = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + int err = 0; + + /* Force mono mode if the sample rate interval doesn't allow stereo. */ + if (SB8_DEN(r->min) == SB8_DEN(r->max) + && SB8_DEN(r->min) & 1) + { + struct snd_interval t = { .min = 1, .max = 1 }; + struct snd_interval *c; + + c = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + err = snd_interval_refine(c, &t); + } + return err; +} + static int snd_sb8_playback_prepare(struct snd_pcm_substream *substream) { unsigned long flags; struct snd_sb *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; - unsigned int mixreg, rate, size, count; + unsigned int mixreg, rate, size, count, dma; + unsigned char stereo, format; rate = runtime->rate; + stereo = runtime->channels > 1; switch (chip->hardware) { case SB_HW_PRO: if (runtime->channels > 1) { @@ -117,6 +168,7 @@ static int snd_sb8_playback_prepare(stru break; } /* fallthru */ + case SB_HW_JAZZ16: case SB_HW_201: if (rate > 23000) { chip->playback_format = SB_DSP_HI_OUTPUT_AUTO; @@ -134,8 +186,17 @@ static int snd_sb8_playback_prepare(stru } size = chip->p_dma_size = snd_pcm_lib_buffer_bytes(substream); count = chip->p_period_size = snd_pcm_lib_period_bytes(substream); + if (runtime->format == SNDRV_PCM_FORMAT_S16_LE) { + dma = chip->dma16; + format = stereo ? SB_DSP_STEREO_16BIT : SB_DSP_MONO_16BIT; + } else { + dma = chip->dma8; + format = stereo ? SB_DSP_STEREO_8BIT : SB_DSP_MONO_8BIT; + } spin_lock_irqsave(&chip->reg_lock, flags); snd_sbdsp_command(chip, SB_DSP_SPEAKER_ON); + if (chip->hardware >= SB_HW_PRO) + snd_sbdsp_command(chip, format); if (runtime->channels > 1) { /* set playback stereo mode */ spin_lock(&chip->mixer_lock); @@ -173,7 +234,7 @@ static int snd_sb8_playback_prepare(stru snd_sbdsp_command(chip, count >> 8); } spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_dma_program(chip->dma8, runtime->dma_addr, + snd_dma_program(dma, runtime->dma_addr, size, DMA_MODE_WRITE | DMA_AUTOINIT); return 0; } @@ -232,9 +293,11 @@ static int snd_sb8_capture_prepare(struc unsigned long flags; struct snd_sb *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; - unsigned int mixreg, rate, size, count; + unsigned int mixreg, rate, size, count, dma; + unsigned char stereo, format; rate = runtime->rate; + stereo = runtime->channels > 1; switch (chip->hardware) { case SB_HW_PRO: if (runtime->channels > 1) { @@ -242,6 +305,8 @@ static int snd_sb8_capture_prepare(struc chip->capture_format = SB_DSP_HI_INPUT_AUTO; break; } + /* fall through */ + case SB_HW_JAZZ16: chip->capture_format = (rate > 23000) ? SB_DSP_HI_INPUT_AUTO : SB_DSP_LO_INPUT_AUTO; break; case SB_HW_201: @@ -261,10 +326,17 @@ static int snd_sb8_capture_prepare(struc } size = chip->c_dma_size = snd_pcm_lib_buffer_bytes(substream); count = chip->c_period_size = snd_pcm_lib_period_bytes(substream); + if (runtime->format == SNDRV_PCM_FORMAT_S16_LE) { + dma = chip->dma16; + format = stereo ? SB_DSP_STEREO_16BIT : SB_DSP_MONO_16BIT; + } else { + dma = chip->dma8; + format = stereo ? SB_DSP_STEREO_8BIT : SB_DSP_MONO_8BIT; + } spin_lock_irqsave(&chip->reg_lock, flags); snd_sbdsp_command(chip, SB_DSP_SPEAKER_OFF); - if (runtime->channels > 1) - snd_sbdsp_command(chip, SB_DSP_STEREO_8BIT); + if (chip->hardware >= SB_HW_PRO) + snd_sbdsp_command(chip, format); snd_sbdsp_command(chip, SB_DSP_SAMPLE_RATE); if (runtime->channels > 1) { snd_sbdsp_command(chip, 256 - runtime->rate_den / 2); @@ -285,7 +357,7 @@ static int snd_sb8_capture_prepare(struc snd_sbdsp_command(chip, count >> 8); } spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_dma_program(chip->dma8, runtime->dma_addr, + snd_dma_program(dma, runtime->dma_addr, size, DMA_MODE_READ | DMA_AUTOINIT); return 0; } @@ -358,10 +430,12 @@ static snd_pcm_uframes_t snd_sb8_playbac { struct snd_sb *chip = snd_pcm_substream_chip(substream); size_t ptr; + int dma; if (chip->mode != SB_MODE_PLAYBACK_8) return 0; - ptr = snd_dma_pointer(chip->dma8, chip->p_dma_size); + dma = (substream->runtime->format == SNDRV_PCM_FORMAT_S16_LE) ? chip->dma16 : chip->dma8; + ptr = snd_dma_pointer(dma, chip->p_dma_size); return bytes_to_frames(substream->runtime, ptr); } @@ -369,10 +443,12 @@ static snd_pcm_uframes_t snd_sb8_capture { struct snd_sb *chip = snd_pcm_substream_chip(substream); size_t ptr; + int dma; if (chip->mode != SB_MODE_CAPTURE_8) return 0; - ptr = snd_dma_pointer(chip->dma8, chip->c_dma_size); + dma = (substream->runtime->format == SNDRV_PCM_FORMAT_S16_LE) ? chip->dma16 : chip->dma8; + ptr = snd_dma_pointer(dma, chip->c_dma_size); return bytes_to_frames(substream->runtime, ptr); } @@ -443,6 +519,19 @@ static int snd_sb8_open(struct snd_pcm_s runtime->hw = snd_sb8_capture; } switch (chip->hardware) { + case SB_HW_JAZZ16: + runtime->hw.formats |= SNDRV_PCM_FMTBIT_S16_LE; + runtime->hw.rates |= SNDRV_PCM_RATE_8000_44100; + runtime->hw.rate_max = 45455; + runtime->hw.channels_max = 2; + snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + snd_jazz16_hw_constraint_rate_channels, NULL, + SNDRV_PCM_HW_PARAM_CHANNELS, + SNDRV_PCM_HW_PARAM_RATE, -1); + snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + snd_jazz16_hw_constraint_channels_rate, NULL, + SNDRV_PCM_HW_PARAM_RATE, -1); + break; case SB_HW_PRO: runtime->hw.rate_max = 44100; runtime->hw.channels_max = 2; --- linux-2.6.20/sound/isa/Kconfig.orig 2007-02-04 19:44:54.000000000 +0100 +++ linux-2.6.20/sound/isa/Kconfig 2007-03-25 21:49:44.000000000 +0200 @@ -248,6 +248,21 @@ config SND_INTERWAVE_STB To compile this driver as a module, choose M here: the module will be called snd-interwave-stb. +config SND_JAZZ16 + tristate "Media Vision Jazz16" + depends on SND + select SND_OPL3_LIB + select SND_PCM + help + Say Y here to include support for Media Vision Jazz16 based + soundcards. + + You probably want to enable Jazz16 Plug and Play support + (CONFIG_JAZZ16PNP) also. + + To compile this driver as a module, choose M here: the module + will be called snd-jazz16. + config SND_OPL3SA2 tristate "Yamaha OPL3-SA2/SA3" depends on SND --- linux-2.6.20/Documentation/sound/alsa/ALSA-Configuration.txt-orig 2007-02-04 19:44:54.000000000 +0100 +++ linux-2.6.20/Documentation/sound/alsa/ALSA-Configuration.txt 2007-03-25 20:35:31.000000000 +0200 @@ -1120,6 +1120,16 @@ Prior to version 0.9.0rc4 options had a This module supports multiple cards, autoprobe and ISA PnP. + Module snd-jazz16 + ----------------- + + Module for sound cards based on Media Vision Jazz16 chip (PnP only) + + This module supports multiple cards and PnP. + + Note: You will most likely want to enable Jazz16 PnP support + (CONFIG_JAZZ16PNP) in the kernel. + Module snd-korg1212 ------------------- --- linux-2.6.20/include/sound/sb.h.orig 2007-02-19 20:57:51.000000000 +0100 +++ linux-2.6.20/include/sound/sb.h 2007-03-25 21:37:01.000000000 +0200 @@ -33,6 +33,7 @@ enum sb_hw_type { SB_HW_20, SB_HW_201, SB_HW_PRO, + SB_HW_JAZZ16, /* Media Vision Jazz16 chip */ SB_HW_16, SB_HW_16CSP, /* SB16 with CSP chip */ SB_HW_ALS100, /* Avance Logic ALS100 chip */ @@ -142,6 +143,7 @@ struct snd_sb { #define SB_DSP_HI_INPUT_AUTO 0x98 #define SB_DSP_IMMED_INT 0xf2 #define SB_DSP_GET_VERSION 0xe1 +#define SB_DSP_GET_JAZZ_VERSION 0xfa #define SB_DSP_SPEAKER_ON 0xd1 #define SB_DSP_SPEAKER_OFF 0xd3 #define SB_DSP_DMA8_OFF 0xd0 @@ -265,6 +267,9 @@ struct snd_sb { #define SB_DMASETUP_DMA6 0x40 #define SB_DMASETUP_DMA7 0x80 +/* Check the output of SB_DSP_GET_JAZZ_VERSION. */ +#define SB_VERSION_IS_JAZZ16(x) ((x) == 0x12) + /* * */ -- Rask Ingemann Lambertsen

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[alsa-devel] snd_mixer_selem_channel_id -- should i be using this? if not, what should i use?
by John L. Utz III 11 Apr '07

11 Apr '07

Hello Everybody; mixer.h indicates that snd_mixer_selem_channel_id is a legacy api. so, what is the non legacy api for setting volumes on individual channels, ie Master L&R, PCM L&R and Surround L&R ? tnx! johnu

1 0

[alsa-devel] Interrupts and kmalloc in alsa modules
by Ciaccia 11 Apr '07

11 Apr '07

Hi all, I am working with an ARM embedded system and I porting an alsa application to that system. Cirrus (the manufacturer of the board) wrote a generic alsa driver for "their" kernel (a kernel based on the 2.6.8 with some patches) and I ported the module to the vanilla kernel 2.6.19. The module works fine (play and capture both work), but the function that allocates the memory prints a strange message in the kernel log when the buffers are allocated (the module calls kmalloc, which prints "sleeping function called from invalid context" to the log). The buffers are allocated when the streams are started, not when the module is loaded, I think this is standard... The buffers are allocated with the following line: s->buffers = (audio_buf_t *) kmalloc(sizeof(audio_buf_t) * s->nbfrags, GFP_KERNEL); When the line here above is executed, slab.c prints the error the kernel log. The error comes from the "might_sleep_if" line: static inline void cache_alloc_debugcheck_before(struct kmem_cache *cachep, gfp_t flags) { might_sleep_if(flags & __GFP_WAIT); #if DEBUG kmem_flagcheck(cachep, flags); #endif } Playing with the module, I discovered that if I enable the interrupts before calling kmalloc, the error disappears. I wonder if this is correct, or if something is wrong... Could someone of you explain to me why I have to enable the interrupts before calling kmalloc? The strange thing is that the module never explicitly turns them off (I attached the file to the mail) Any help is greatly appreciated ;-) Bye Andrea ____________________________________________________________________________________ Food fight? Enjoy some healthy debate in the Yahoo! Answers Food & Drink Q&A. http://answers.yahoo.com/dir/?link=list&sid=396545367

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[alsa-devel] sound on asus p5ld2-se
by Andrius 10 Apr '07

10 Apr '07

Hi, It says "If the default configuration doesn't work and one of the above matches with your device, report it together with the PCI subsystem ID (output of "lspci -nv") to ALSA BTS or alsa-devel ML" in ALSA-configuration.txt so, here it is: i'm using asus p5ld2-se mobo with intel ich7 chipset, and just have installed new debian stable (4.0). And encountered a problem: onboard audio sound from one of the speakers was just static noise, and if i adjusted volume or anything in alsamixer, the sound kept dissappearing. I tried module=3stack option in modprobe, added line options snd-hda-intel index=0 model=3stack in /etc/modprobe.d/sound and everything is ok now. i attached lspci -nv output to this letter p.s. on ubuntu live cd and other distros i've tried there were same problem

2 1

[alsa-devel] Driver for user-space sound source/sink(s)
by Alex Makarov 10 Apr '07

10 Apr '07

Hi We're considering implementing the driver for user-space sound input/output. For example: "Line in" is provided by some application by streaming data to a file in /dev/..., "Line out" can be read by any application from /dev/... It would be also nice to have arbitrary number of streams, for example to have different applications to "capture" audio from different sources. I wonder if we reinvent the wheel, or maybe there's an ongoing project for the similar driver. If not, could anyone advise how this can be achieved. We had no experience with ALSA before, but pretty strong in C/C++ development. Thanks! Alex

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[alsa-devel] Volume control channels mixed
by Simon van der Linden 10 Apr '07

10 Apr '07

Hello, I use Alsa in the latest Ubuntu bêta, with my MacBook. lspci gives the audio card as: 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 02) Sound is working pretty good. However, volume control channels (in alsamixer) are mixed: with laptop speakers: Master controls treble volume Front controls bass volume with headphones: Master has no effect Front controls the volume I'm unfamiliar with Alsa, but I guess it shouldn't be too hard to correct that. Browsing the web didn't give me more informations. Perhaps some of you could help me? -- Simon van der Linden jid: simon(a)vanderlinden.eu.org

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Re: [alsa-devel] [Audacity-devel] distortion with alsa I/O - no distortion with oss emulation
by Simon Lewis 10 Apr '07

10 Apr '07

Dear Alsa developers Audacity seems to work perfectly with AOSS and Jack but not directly with ALSA on some linux PCs. Trying to use ALSA directly leads to severe distortion due to lack of sync with Alsa. (I hope that is the correct terminology.) I received this reply from the Audacity developers, what tools are available from ALSA to investigate this problem? How can I determine what the real "Alsa configuration" is that portaudio has used setting up communication with Alsa. Best Regards, Simon > Markus Meyer schrieb: > > > As for Simon's problems with "synchronisation", I'm not so sure that it > is a Portaudio issue, or a software issue at all. Software issues tend > to be of the "on/off" type, and Portaudio does work well with ALSA in > the general case. Apart from being a hardware issue (defective sound > card, main board or something) this could also well be a race condition > in the ALSA driver (kernel module) which is only triggered by certain > buffer sizes, order of driver requests etc. Because Audacity is > multi-threaded it certainly stresses the driver/kernel subsystem more > than just, say, "arecord", which just gets blocks from the driver and > writes it out to disk. > > > Markus > > >> Simon Lewis schrieb: >> >> The main issue is that on some linux machines, mine included, there is >> severe distortion when Audacity is used with ALSA drivers. >> >> You will be pleased to hear that I have found the root cause of the >> problem. It is nothing to do with the options with which Audacity is >> compiled. The root cause is that portaudio loses synchronisation with >> ALSA's data stream. >> >> I was able to verify this yesterday, as I could start Audacity with >> Jack, and through Jack set up a number of simulation conditions and >> monitor the results. >> >> With my standard Jack settings of 96kHz / 256 Frames per period / 2 >> periods per buffer (latency 5.33 msecs.), ALSA initialises with sample >> format 32bit little-endian, 2 periods for capture and 2 periods for >> playback. >> >> Under these conditions Audacity cannot synchronise with the data stream >> and severe distortion is the result. >> >> However, increasing the frames per period to 1024 (latency 21.3 msec) >> and Audacity (1.3.2 with resampling disabled) runs perfectly and I can >> record for a long time. (CPU load is 9.6%) >> >> Reducing the frames per period to 512 and the same version of Audacity >> starts OK but after a few seconds, sometimes longer losses sync with the >> data stream. >> >> The standard 1.3.2 version of Audacity with resampling enabled does not >> sync on start with 512 frames per period but runs OK with Jack set to >> 1024 frames per period. >> >> The 1.3.3 version of Audacity complied with libsamplerate without >> portmixer runs OK with 512 frames per period under jack. >> >> NONE of the Audacity versions will run directly with the ALSA drivers. >> The Version with resampling disabled often starts OK, but after 30 sec >> to a few minutes of recording sync is lost and naturally distortion is >> the result. >> >> The conclusion is that portaudio is very poor at syncing with the >> alsadata stream, something which Jack can do with very little buffering. >> >> >> Ciao, Simon

3 2

[alsa-devel] Correct use of ak4114.c?
by dustin＠seznam.cz 10 Apr '07

10 Apr '07

Hello, while working on support for AK4114 in Audiotrak MI/ODI/O card, I used corresponding code from revo.c (and similarly from juli.c), calling function snd_ak4114_create. However, this function does not initialize array ak4114->kctls[idx], leading to consequent kernel oops after the periodically called snd_ak4114_check_rate_and_errors, specifically in function snd_ctl_notify called by snd_ctl_notify(ak4114->card, SNDRV_CTL_EVENT_MASK_VALUE, &ak4114->kctls[0]->id) Temporarily I resolved the problem by commenting out the snd_ctl_notify calls. However, it seems author of ak4114.c intended to create the device with snd_ak4114_build instead which defines all the controls. I did not hit the problem until fixing the 4-wire communication in revo.c - that explaing why nobody had this problem with revo cards. I do not understand how the code can work in juli.c - perhaps I have overlooked something there. My question: Is the correct way to add null pointer checks before snd_ctl_notify calls only, or to use snd_ak4114_build insted of snd_ak4114_create? If using snd_ak4114_build, where do I get the parameters ply_substream and cap_substream? int snd_ak4114_build(struct ak4114 *ak4114, struct snd_pcm_substream *playback_substream, struct snd_pcm_substream *capture_substream); Thanks for any suggestions. Pavel.

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[alsa-devel] seq: timing unstable on slow computer when busy
by Gerald Grabner 09 Apr '07

09 Apr '07

Hi there, I'm trying to use a simple (self written) sequencer application on an old laptop with a 350 MHz CPU. It works, but the timing of output events becomes unstable when my program is busy with things like scheduling events for the subsequent loop or an unmuted track. I'm wondering about the best approach to tackle this problem. Depending on how the priority of the kernel's event scheduling is related to the priority of an application, I could imagine some of the following measures: * Running my process with lower or higher priority * Using (priorized) threads for the event scheduling * Setting Linux kernel config options * Setting ALSA related options (driver or lib) * ... While playing midi, I'm scheduling an echo event a few ticks before the end of every sequence, upon which a function to schedule the events of the next sequence is called. What is the typical length of this scheduling span: Just one bar? Or longer? Many thanks in advance, Gerald BTW: My operation system is Gentoo's Linux 2.6.19 with alsa-driver 1.0.14_rc1.

2 1

[alsa-devel] Popping noise on HP dv6000z (CX20549) running x86 kernel
by enigma_0Z 08 Apr '07

08 Apr '07

Here's my situation: If I play a game (any game, in particular, however, neverball/putt or sauerbraten), the sound has these little soft "pop" noises happening so quick per second I cannot count them. This is when I'm running on an x86 kernel, however, when I run an x86_64 kernel, these are not present. I've got a 64-bit processor, specifically an AMD Sempron 3500+ Does Anyone know why? I've got a conexant CX20549 in my laptop. Thanks, enigma

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[alsa-devel] dont knw what this error means
by Ashlesha Shintre 05 Apr '07

05 Apr '07

Hi, I am getting the following error when I try to playback a raw sample file: "ALSA lib pcm.c:7216snd_pcm_set_params) Unable to get period size for PLAYBACK: Invalid argument" Does anyone know what this means? Thanks Ashlesha.

2 1

[alsa-devel] alsa+saa7134 sound problem
by aa 05 Apr '07

05 Apr '07

please read marco007 posts my english is so bad http://forums.gentoo.org/viewtopic-t-527329-postdays-0-postorder-asc-start-… tnx

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[alsa-devel] Sampling rate limited to 48 kHz
by Marek Ławicki 05 Apr '07

05 Apr '07

Hi, I have Turtle Beach Catalina Card (Envy24HT-S, VT1616 + WM8728) which I belive is exactly the same as Mad Dog Entertainer 7.1 ( http://techreport.com/reviews/2004q1/maddog-entertainer/index.x?pg=2 ) Card is recognized as Chaintech AV-710 but works very well ( so far I tested analog out through WM8728 and digital out ). My problem is : I can't set sampling rate higher than 48000 on SPDIF out. I know this card can go higher because : 1. It can go higher under Windows (High Sample Rate 2-channel mode) 2. If I "flash" eeprom with M-Audio Revolution 7.1 settings card can sync with my external DAC up to 96 kHz (DAC limit) and I can play 96kHz .wav file. This is not the way to go for me (analog part and SPDIF-in is lost) I found this : ice1724.c:390:static int get_max_rate(struct snd_ice1712 *ice) ice1724.c-391-{ ice1724.c-392- if (ice->eeprom.data[ICE_EEP2_ACLINK] & VT1724_CFG_PRO_I2S) { ice1724.c-393- if ((ice->eeprom.data[ICE_EEP2_I2S] & 0x08) && !ice->vt1720) ice1724.c-394- return 192000; ice1724.c-395- else ice1724.c-396- return 96000; ice1724.c-397- } else ice1724.c-398- return 48000; ice1724.c-399-} ice1724.c-400- Tried to play with it, but no luck - I'm not developer :-) How can I help with making ice1724 driver even better ? Thanks, Marek # cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.14rc1. Compiled on Apr 2 2007 for kernel 2.6.18.6. # cat /proc/asound/card0/ice1724 Chaintech AV-710 at 0xd000, irq 11 EEPROM: Subvendor : 0x12142417 Size : 28 bytes Version : 1 System Config : 0x42 ACLink : 0x2 I2S : 0x10 S/PDIF : 0xc3 GPIO direction : 0xff GPIO mask : 0xff GPIO state : 0xff Extra #18 : 0x1 Extra #19 : 0x8 Extra #20 : 0x0 Extra #21 : 0x0 Extra #22 : 0x0 Extra #23 : 0x0 Extra #24 : 0x0 Extra #25 : 0x0 Extra #26 : 0x0 Extra #27 : 0x0 Registers: PSDOUT03 : 0x00000000 CCS00 : 0x00 CCS01 : 0xa0 CCS02 : 0x00 CCS03 : 0x00 CCS04 : 0x42 CCS05 : 0x02 CCS06 : 0x10 CCS07 : 0xc3 CCS08 : 0x00 CCS09 : 0x00 CCS0a : 0x00 CCS0b : 0x00 CCS0c : 0x00 CCS0d : 0x0a CCS0e : 0x00 CCS0f : 0x00 CCS10 : 0x37 CCS11 : 0x06 CCS12 : 0x01 CCS13 : 0x80 CCS14 : 0x00 CCS15 : 0xea CCS16 : 0xff CCS17 : 0x00 CCS18 : 0xff CCS19 : 0x00 CCS1a : 0x00 CCS1b : 0x00 CCS1c : 0x00 CCS1d : 0x00 CCS1e : 0x2a CCS1f : 0x00 MT00 : 0x00 MT01 : 0x08 MT02 : 0x00 MT03 : 0x08 MT04 : 0x5a MT05 : 0x08 MT06 : 0x21 MT07 : 0x82 MT08 : 0x00 MT09 : 0x00 MT0a : 0x00 MT0b : 0x00 MT0c : 0x00 MT0d : 0x00 MT0e : 0x00 MT0f : 0x00 MT10 : 0x40 MT11 : 0x1f MT12 : 0xa8 MT13 : 0x0d MT14 : 0x2f MT15 : 0x78 MT16 : 0x00 MT17 : 0x00 MT18 : 0x80 MT19 : 0x03 MT1a : 0x00 MT1b : 0x00 MT1c : 0xff MT1d : 0x07 MT1e : 0x00 MT1f : 0x00 MT20 : 0x20 MT21 : 0x01 MT22 : 0xac MT23 : 0x0d MT24 : 0xb7 MT25 : 0x7f MT26 : 0xff MT27 : 0x07 MT28 : 0x00 MT29 : 0x00 MT2a : 0x00 MT2b : 0x00 MT2c : 0x00 MT2d : 0x00 MT2e : 0x00 MT2f : 0x00 # lspci -vvv 00:0b.0 Multimedia audio controller: VIA Technologies Inc. VT1720/24 [Envy24PT/HT] PCI Multi-Channel Audio Controller (rev 01) Subsystem: VIA Technologies Inc. AMP Ltd AUDIO2000 Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 64 Interrupt: pin A routed to IRQ 11 Region 0: I/O ports at d000 [size=32] Region 1: I/O ports at cc00 [size=128] Capabilities: [80] Power Management version 1 Flags: PMEClk- DSI- D1- D2+ AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME-

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Re: [alsa-devel] [Alsa-devel] Quality resampling code for libasound
by Takashi Iwai 05 Apr '07

05 Apr '07

At Wed, 21 Mar 2007 08:35:22 +1100, Jean-Marc Valin wrote: > > >> Well, I use that type because that's what is used in the API. And the > >> reason I use it in the API is because this code is meant to work even on > >> platforms with 16-bit integers (e.g. TI C5X DSPs) and no C99 compiler. > >> If that's causing problem, I guess you could still use unsigned int with > >> no problem... > > > > Don't worry, alsa-lib would be very unlikely ported to such an > > architecture. > > Well, my code has to run on those architectures, and it's much simpler > if I don't need to change it for ALSA. Now if you don't want the > spx_uint32_t in the plugin code, then feel free to change that. OK, as long as it's in alsa-plugins repository, I basically don't care. But, for merging it to alsa-lib, I'd like to change the code very much (including coding-style unification). Anyway, I applied your latest codes to HG tree now (with the removal of samplerate.h). Thanks! Takashi

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Re: [alsa-devel] Alsa OSS - who is maintaining it?
by Stefan de Konink 04 Apr '07

04 Apr '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA512 Johannes Berg wrote: > On Wed, 2007-04-04 at 20:42 +0200, Stefan de Konink wrote: > >>> /* >>> * For input, we need to do a graceful stop; if we abort >>> * the DMA, we end up with leftover bytes that corrupt >>> * the next recording. To do this we set the S0 status >>> * bit and wait for the DMA controller to stop. Each >>> * command has a branch condition to >>> * make it branch to a stop command if S0 is set. >>> * On input we also need to wait for the S7 bit to be >>> * set before turning off the DMA controller. >>> * In fact we do the graceful stop for output as well. >>> */ >> Check, I have that. > > Hm. Then I don't know what could be causing this. Does it happen all the > time or just intermittently? With every OSS app out there... but native alsa works fine. Stefan -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.3 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGFBwQYH1+F2Rqwn0RCi2dAJ9Vlel3cz0PAUMvgYi7L2Nd3XCc9wCeJG9H NT+AsNRL91Dm1y9Dlch75PU= =xxif -----END PGP SIGNATURE-----

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Re: [alsa-devel] Alsa OSS - who is maintaining it?
by Stefan de Konink 04 Apr '07

04 Apr '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA512 Johannes Berg wrote: > On Wed, 2007-04-04 at 20:31 +0200, Stefan de Konink wrote: > >> In lsmod powermac is not loaded. The 'strange' thing I found, is that >> even without snd-pcm-oss or aoss I get garbage. At that moment I wonder >> what emulation layer is used. > > Oh, I see. There was a bug in the i2sbus setup, does the i2sbus-pcm.c > file (sound/aoa/soundbus/i2sbus/i2sbus-pcm.c) contain the following > comment at around line 380: Something I must note: recording works. Stefan -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.3 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGE/mcYH1+F2Rqwn0RCiYeAJ9h+F1tCdS/cyNbaXoVueFBXX09PgCdGfAp rRH4y2U/E/SBIHtwobJjVI4= =FdSb -----END PGP SIGNATURE-----

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Re: [alsa-devel] Alsa OSS - who is maintaining it?
by Stefan de Konink 04 Apr '07

04 Apr '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA512 Johannes Berg wrote: > On Wed, 2007-04-04 at 20:31 +0200, Stefan de Konink wrote: > >> In lsmod powermac is not loaded. The 'strange' thing I found, is that >> even without snd-pcm-oss or aoss I get garbage. At that moment I wonder >> what emulation layer is used. > > Oh, I see. There was a bug in the i2sbus setup, does the i2sbus-pcm.c > file (sound/aoa/soundbus/i2sbus/i2sbus-pcm.c) contain the following > comment at around line 380: > /* > * For input, we need to do a graceful stop; if we abort > * the DMA, we end up with leftover bytes that corrupt > * the next recording. To do this we set the S0 status > * bit and wait for the DMA controller to stop. Each > * command has a branch condition to > * make it branch to a stop command if S0 is set. > * On input we also need to wait for the S7 bit to be > * set before turning off the DMA controller. > * In fact we do the graceful stop for output as well. > */ Check, I have that. Stefan -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.3 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGE/F+YH1+F2Rqwn0RCtRLAJ0VTYJlaivU3gC0toB7cKI7AYBz6QCfV5AW BO+B/PhvRdxr+asrfI8f9uA= =0LCT -----END PGP SIGNATURE-----

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Re: [alsa-devel] Alsa OSS - who is maintaining it?
by Takashi Iwai 04 Apr '07

04 Apr '07

At Wed, 04 Apr 2007 19:18:48 +0200, Johannes Berg wrote: > > On Wed, 2007-04-04 at 19:10 +0200, Takashi Iwai wrote: > > > > > To make sure, check /proc/asound/card0/pcm0p/sub0/hw_params has no > > > > "OSS" entry when you're playing via aoss. If a line like "OSS Format" > > > > appears there, it is still kernel oss-emulation. > > > > > > OSS Format is not in there. > > > > Fine. Then it smells more. > > > > Johannes, any known / possible problem with OSS emulation using aoa > > driver (especially regarding endianness)? > > snd-pcm-oss you mean? Both snd-pcm-oss and aoss wrapper seem not working for Stefan's machine. Maybe a problem specific to a certain model? Takashi > No, it's working fine, the contents of that file is (mplayer -ao oss): > > access: RW_INTERLEAVED > format: S16_BE > subformat: STD > channels: 2 > rate: 44100 (44100/1) > period_size: 1024 > buffer_size: 16384 > tick_time: 10000 > OSS format: S16_BE > OSS channels: 2 > OSS rate: 44100 > OSS period bytes: 4096 > OSS periods: 16 > OSS period frames: 1024 > > while with alsa it is (rhythmbox/gstreamer): > > access: MMAP_INTERLEAVED > format: S16_BE > subformat: STD > channels: 2 > rate: 44100 (44100/1) > period_size: 1024 > buffer_size: 8192 > tick_time: 10000 > > and with aoss (aoa mplayer -ao oss, snd-pcm-oss not loaded): > access: MMAP_INTERLEAVED > format: S16_BE > subformat: STD > channels: 2 > rate: 44100 (44100/1) > period_size: 1024 > buffer_size: 8192 > tick_time: 10000 > > > Don't see a problem at all, working well for me.

2 1

Re: [alsa-devel] Alsa OSS - who is maintaining it?
by Takashi Iwai 04 Apr '07

04 Apr '07

At Wed, 04 Apr 2007 19:12:49 +0200, Stefan de Konink wrote: > > Takashi Iwai wrote: > > I meant the comparison between the working state with ALSA apps and > > the non-working state with OSS apps... > > Lets put it this way: > mplayer -ao oss: noise > mplayer -ao alsa: nice sound And the comparison of parameters? You can get it in /proc/asound/card0/pcm0p/sub0/* files as I already mentioned. Diff them between two different stats. Takashi

1 0

Re: [alsa-devel] Alsa OSS - who is maintaining it?
by Takashi Iwai 04 Apr '07

04 Apr '07

At Wed, 04 Apr 2007 18:57:11 +0200, Stefan de Konink wrote: > > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA512 > > Takashi Iwai wrote: > > Hm, then I suspect rather it's a problem in the aoa driver side. > > > > To make sure, check /proc/asound/card0/pcm0p/sub0/hw_params has no > > "OSS" entry when you're playing via aoss. If a line like "OSS Format" > > appears there, it is still kernel oss-emulation. > > OSS Format is not in there. Fine. Then it smells more. Johannes, any known / possible problem with OSS emulation using aoa driver (especially regarding endianness)? > > After you confirm it's a native ALSA in the driver execution, compare > > the parameters there between the working status (using ALSA API) and > > non-working status (OSS API). > > I never have had the aoa driver working with OSS. I'm using 1.0.14_rc3. I meant the comparison between the working state with ALSA apps and the non-working state with OSS apps... Takashi

1 0

[alsa-devel] Individual PCM stream control
by Nagalla, Hari 04 Apr '07

04 Apr '07

Hello, I am new to ALSA, so some help from experienced folks would help.. We are trying to develop ALSA support for our DSP based mixer. In our architecture DSP performs the mixing, sample rate conversion etc. My question is how, can does the user space applications exercise controlling individual PCM stream to the mixer? Ex: individual stream gain control, individual stream mute, etc.. Any user space example code would be useful too.. Thanks Hari

2 3

[alsa-devel] ALSA, GStreamer and surround
by Christian F.K. Schaller 04 Apr '07

04 Apr '07

Hi, We are still having issues resolving how to deal with Surround sound and ALSA in GStreamer. If anyone knowledgeable about ALSA would be willing to comment on this bug: http://bugzilla.gnome.org/show_bug.cgi?id=410403 We would much appreciate it. Christian

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[alsa-devel] usb midi disconnect -> kernel oops
by Dmitry Baikov 04 Apr '07

04 Apr '07

Hi! If I have a running application which connects (via alsa seq) to usb midi device, and device is disconnected, I get a kernel oops. kernels tried: 2.6.20-rt8 (in-kernel alsa), 2.6.21-rc5-rt3 (in-kernel alsa, and 1.0.14_rc3) Kernel dump attached. Dmitry.

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Re: [alsa-devel] SPDIF/IEC958 sample rate on HDA/ALC882
by Dag Lem 03 Apr '07

03 Apr '07

Takashi Iwai <tiwai <at> suse.de> writes: > > Hi, > > Sorry, looks like I overlooked this post... > > At Wed, 01 Nov 2006 17:58:55 +0100, > Erik Slagter wrote: > > > > Sorry if this is considered to be "user" question, but I got no response > > at all from the "user" list. Guess the real cracks hang around here. > > > > I have lots of questions I cannot resolve from all of the docs, but for > > the moment I'll stick to the most important one. > > > > The combination of HDA/ALC882 should be able to deliver sound at 96 kHz > > (hda..., right...) Also the ALC882 datasheet says it can deliver sound > > at 96 kHz. > > > > In practise, though, I cannot get it to output any other sample rate > > than 48 kHz. Any other sample rate handed to hw:0,0 is simply converted > > (somewhere??? and badly...) to 48 kHz and the SPDIF is still driven at > > 48 kHz. > > > > iecset allows me to set the rate to 32/44.1/48 kHz, the spdif output is > > still at 48 kHz. Some other flags from iecset are actually honoured > > (like "data" and "emphasis"). > > It's simply because iecset program doesn't support the rates over > 48kHz. But the hda-intel driver supports the rate, AFAIK. > (At least, there is no particular code that restricts over-48kHz.) > > One thing to be noted is that if you use "hw" PCM device, you have to > set up the SPDIF status bits _manually_ via control API. If you use > "iec958" or "spdif" PCM device, you can pass these bits as optional > arguments at opening the PCM. > > > I can ask aplay to play at various rates (from a suitable PCM file), but > > it complains at any other rate than 44.1 or 48 (notably 32) kHz that the > > rate is not supported. It plays at 44.1 kHz though. And again converts > > it to 48 kHz! Sigh... > > Who complains? At least, the driver won't. > > > Maybe I am using the wrong "hw", there is also a "hw:0,2" device, which > > I cannot make work properly at all (only one channel is output, large > > chunks are discarded, much much clipping). > > The first PCM device is for the multi-output PCM. It's for both > analog and digital. The dedicated SPDIF is the secondary one. > > Takashi > > ------------------------------------------------------------------------- > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share your > opinions on IT & business topics through brief surveys - and earn cash > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV > Hi, I was googling around and found this (old) post; I can confirm that there are severe problems with the ALC882 and/or the driver code. I'm experiencing the same problem as Erik - I cannot get the ALC882 to transmit at any other sample rate than 48kHz over S/PDIF. When I attempt to play 44.1kHz sound, the S/PDIF connection is still locked at 48kHz (shown by my amplifier), and the resulting sound is less than pleasing. It would seem like a 44.1kHz signal packed in a 48kHz frame rate played back directly - high-pitched sounds suffer badly from what my ears recognize as straightforward sample picking as opposed to resampling. Looking at hda_codec.c:convert_from_spdif_status() it seems that the IEC958 parameter bit for 44.1kHz/48kHz is not used for anything (correct me if I'm wrong). This would mean that the rate is only set in snd_hda_codec_setup_stream(), and as far as I can tell from looking at the array rate_bits and the ALC882 Series Datasheet the setting is correct for 44.1kHz. Since I can't find any obvious mistake in the HDA code, I guess there is some problem in patch_realtek.c, or an initialization incompatibility with the HDA code and the ALC882 chip. There are certainly initialization problems somewhere, as the S/PDIF output is sometimes not able to send AC3 data correctly (amplifier cannot detect AC3 and stays silent), and attempting to play audio (even 48kHz) results in intermittent screeches. This is solved by setting and resetting the IEC958 control widget, trying again and again until it finally works. When it does work, it works without a hitch until the next reboot. Any ideas? Is there anything I can help to test in order to nail down the problem? My motherboard is an Asus P5W DH Deluxe with an 82801G HDA controller, in case this has any relevance. -- Best regards, Dag Lem

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[alsa-devel] Vanilla Blocked I/O probs
by Gordon Miller 03 Apr '07

03 Apr '07

Hi, I am calling rc = snd_pcm_readi (handle, buffer, MAX_LEN) but rc is always 4 short of MAX_LEN (1020 instead of 1024). I thought that it would just block until MAX_LEN frames were available. This is really doing my head in, and any help would be appreciated. Cheers

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Re: [alsa-devel] Alsa OSS - who is maintaining it?
by Takashi Iwai 03 Apr '07

03 Apr '07

At Tue, 03 Apr 2007 00:35:21 +0200, Stefan de Konink wrote: > > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA512 > > Takashi Iwai wrote: > > At Wed, 28 Mar 2007 13:08:09 +0200 (CEST), > > Stefan de Konink wrote: > >> On Wed, 28 Mar 2007, Takashi Iwai wrote: > >> > >>> Did you try other drivers, such as usb-audio? > >> I did not. > > > > Then try it if you have one. > > Don't have one. It's a pity. > >>> And, you're using user-space alsa-oss wrapper (e.g. via aoss wrapper), > >>> right? The user-space emulation and kernel-space emulation are > >>> totally different, so I have to know what you actually tried. > >> I'm only using the in kernel translation. > > > > Then try aoss, too. > > aoss program gives the the same garbage. Hm, then I suspect rather it's a problem in the aoa driver side. To make sure, check /proc/asound/card0/pcm0p/sub0/hw_params has no "OSS" entry when you're playing via aoss. If a line like "OSS Format" appears there, it is still kernel oss-emulation. After you confirm it's a native ALSA in the driver execution, compare the parameters there between the working status (using ALSA API) and non-working status (OSS API). Takashi

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[alsa-devel] USB audio class v2.0
by Daniel Mack 03 Apr '07

03 Apr '07

Hi, is there any efford taken to support devices conforming to the USB audio class in version 2.0? I'm just curious if anybody ever started work on that since I couldn't find any hint in the sources. Greets, Daniel

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[alsa-devel] dereferencing pointer error in app but not in lib
by intellij 03 Apr '07

03 Apr '07

In the library, like pcm.c I can code like const char *snd_pcm_name(snd_pcm_t pcm) { return pcm->name; } At the same time, the same code in the application causes "dereferencing pointer to incomplete type" error. Note, the compiler does not tell that there is no such field in the pcm structure. It just thinks that the name has another type. Which one? thank you.

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[alsa-devel] PCM low level - need help
by ODINOT Jean-Marc ETRALI 02 Apr '07

02 Apr '07

Hi Alsa gurus, I've to write an alsa driver for a custom card. I readed "writing an alsa driver" and other paper about alsa and drivers. Well I got a result, but not as good as I think it's possible with alsa. My card have 30 output slots (and 7 input slots), can be 8Khz or 16Khz (same rate for every slots), 16bits little endian by sample, can have one or more sample for each slot (same for every slots) in a frame, an hardware IT every frame (a frame is 30 slots :-). The purpose is to get, for each IT, a frame with all the datas. IT -> frame [channel1][channel2][channel3]...[channel30] channelX [sample1][sample2]...[sampleN] sampleX [16bitsle] At this time, I've something like a frame if I use channel30 with speaker-test -c30, but if I use channel1, alsa give me a buffer with many samples for channel1. And, just after oppening, I get an extra word. BTW I use .copy to get the buffer. My questions: -why the extra word (can I avoid it?)? -how set the buffer len to always have all my channels (used or not)? -is it possible to have more than a sample by channel contiguous? Think's it's all for today, but maybe your answers will give me more ideas :-) Thanx for replies (and please, forgive my awful english :-(, JM ********************************* This message and any attachments (the "message") are confidential and intended solely for the addressees. Any unauthorised use or dissemination is prohibited. Messages are susceptible to alteration. France Telecom Group shall not be liable for the message if altered, changed or falsified. If you are not the intended addressee of this message, please cancel it immediately and inform the sender. ********************************

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Re: [alsa-devel] [Alsa-devel] [i8x0] docking station audio-out ceased to work
by Takashi Iwai 02 Apr '07

02 Apr '07

At Wed, 21 Mar 2007 23:57:32 +0100, Andreas Happe wrote: > > On Mon, 2007-03-19 at 18:53 +0100, Takashi Iwai wrote: > > Interesting. Usually AC97-SPSA specifies the AC97 slot for SPDIF > > output. The full (= 3) means the slot 10/11, 0 means slot 3/4. > > IIUC, AC97-SPSA must be 3 for docking station output working? > > If i remember correctly - but I was on holiday before the hardware > stopped to work (it was either set to 0 or 3, but I think that it was > 3). > > > If yes, try to set spdif_aclink=1 with the latest driver (HG > > version)... > > do you mean alsa-driver or alsa-kernel? I'm somehow reluctant to replace > my distribution's userspace utilities with self-compiled stuff (while i > would happily build a custom kernel). You need both trees to compile the driver. > BTW: a short link to a tutorial howto build hg kernels (or against what > kernel release). I checked it out, copied over the sound/ directory and > rebuild my 2.6.21-rc4 kernel. Copying to Linus kernel may work in most cases but sometimes not since alsa-kernel repo is updated for mm tree. The building HG driver is easy. Clone (both) repos, and run hgcompile script instead of configure. See alsa-driver/INSTALL. Takashi

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[alsa-devel] how does hw:0, 1 eventually get mapped to /dev/snd/controlC0 and company? is this on the wiki somewhere?
by John Utz 02 Apr '07

02 Apr '07

how does this work? an explicit explanation would be great, but i'd even settle for a vague handwavy one at this point.

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