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

[alsa-devel] ALSA driver for Native Instruments sound hardware
by Daniel Mack 23 Mar '08

23 Mar '08

Hi, some months after the last posting, we finally managed to finish a new version of the ALSA driver for Native Instruments' external USB sound cards such as "Rig Kontrol 2", "Kore Kontroller", "Audio Kontrol 1" and "Audio 8 DJ". Compared to the last release, the following things were changed: - the streaming logic was completely reworked so that multiple substreams are handled correctly - a Linux input device layer was added to support the knobs, buttons and pedals on "Rig Kontrol 2" and "AudioKontrol 1" - workqueues were entirely removed from the code - most data structures are now allocated within one big struct to avoid multiple, smaller allocations - many more cleanups according to Clemens' reply on my last posting[1] The driver was properly tested with all the mentioned hardware on both record and playback channels. The patchset should apply to the current mecurial alsa-kernel tree. Please let me know what you think. Best regards, Daniel [1] http://article.gmane.org/gmane.linux.alsa.devel/34324/ Signed-off-by: Daniel Mack <daniel(a)caiaq.org> Signed-off-by: Karsten Wiese <fzu(a)wemgehoertderstaat.de>

4 10

[alsa-devel] [PATCH] snd-aoa-i2sbus: use MODULE_DEVICE_TABLE instead of plain MODULE_ALIAS
by Johannes Berg 10 May '07

10 May '07

This patch changes snd-aoa-i2sbus to use MODULE_DEVICE_TABLE instead of a hardcoded MODULE_ALIAS. Thanks to Sylvain Munaut for pointing this out. Signed-off-by: Johannes Berg <johannes(a)sipsolutions.net> --- Please apply to the alsa tree. sound/aoa/soundbus/i2sbus/i2sbus-core.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) --- wireless-dev.orig/sound/aoa/soundbus/i2sbus/i2sbus-core.c 2007-03-28 13:08:25.903457795 +0200 +++ wireless-dev/sound/aoa/soundbus/i2sbus/i2sbus-core.c 2007-03-28 13:09:21.253457795 +0200 @@ -23,9 +23,6 @@ MODULE_LICENSE("GPL"); MODULE_AUTHOR("Johannes Berg <johannes(a)sipsolutions.net>"); MODULE_DESCRIPTION("Apple Soundbus: I2S support"); -/* for auto-loading, declare that we handle this weird - * string that macio puts into the relevant device */ -MODULE_ALIAS("of:Ni2sTi2sC"); static int force; module_param(force, int, 0444); @@ -37,6 +34,8 @@ static struct of_device_id i2sbus_match[ { } }; +MODULE_DEVICE_TABLE(of, i2sbus_match); + static int alloc_dbdma_descriptor_ring(struct i2sbus_dev *i2sdev, struct dbdma_command_mem *r, int numcmds)

2 6

[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

3 15

[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.

3 21

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

3 34

[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

3 12

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

4 20

[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

3 4

[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.

2 1

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

2 10

[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.

3 9

[alsa-devel] usleep() and nanosleep() timings seem inaccurate using ALSA
by Carlo Florendo 31 Mar '07

31 Mar '07

Good Day! After studying the intricacies of MIDI, I ended up writing an implementation of the MIDI protocol and file format. I then studied the ALSA sequencer API to be able to control a synthesizer keyboard and play MIDI files. I've used aplaymidi, aconnect, arecordmidi, and all those great ALSA utilities. They're very good! However, I wanted to have a simple ncurses based, command line MIDI sequencer meant for small Linux distributions such as DSL or Trustix so I began to write a command line sequencer using ALSA. I've encountered one problem about using usleep() and nanosleep() especially in 2.4 kernels. The core of my problem in my application is that the MIDI files slow down their tempo when pitch changes are made on a particular channel/track, while a NOTE-ON event is being played. For simplicity's sake, I'll give timing examples in terms of seconds and not bars nor measures. For example, if I've got a 2 second NOTE-ON playing a guitar note, all timings are ok. If the NOTE-ON is interspersed with CONTROLLER messages such as a pitch bend, the part which is supposed to sound in approximately 2 seconds, sounds much longer than that. The effect of this is that a ritardando is heard everytime short-duration CONTROLLER messages are sent. The problem disappears when I use the sequencer queues. However, the problem manifests itself when using direct output (i.e. not using queues, using snd_seq_event_output and snd_seq_drain_output one after the other). The reason why I would not want to use the queue is that I'd want the UI of my program (such as metronome ticks and counts) to somehow synch with the sound that is being heard. I've tried removing all UI elements, thinking that the UI controls might have caused the delay, but still the ritardando effect persists. The structure of my code is like this: parse_midi_file(); merge_all_tracks_to_one_single_track(); normalize_the_delta_times() for all events; do usleep(delta_time_for_each_event); play_the_event_using_alsa_direct_output(); done The ritardando effect is a bit lesser using 2.6 kernels. If I use queues, the problem disappears completely in 2.4 and 2.6 kernels. I don't hear the effect with simple MIDI files, such as with piano pieces. By the way, I intend to use my app with a 2.4 kernel. Does anybody have a solution to the delay problem? Please point out to me if I'm doing anything wrong since I've just been working with alsa-seq for a few months. Thank you very much. Best Regards, Carlo -- Carlo Florendo Softare Engineer/Network Co-Administrator Astra Philippines Inc. UP-Ayala Technopark, Diliman 1101, Quezon City Philippines http://www.astra.ph -- The Astra Group of Companies 5-3-11 Sekido, Tama City Tokyo 206-0011, Japan http://www.astra.co.jp

2 4

[alsa-devel] is statically linking libasound supported? because if i do it, i cant open the device!
by John Utz 31 Mar '07

31 Mar '07

i have a test app that works successfully with hw:0,1 on gnto if i link it dynamically: dmx-gnto jutz # ./atestminhw1 Playback device is hw:0,1 Stream parameters are 48000Hz, S16_LE, 6 channels Rate set to 48000Hz (requested 48000Hz) Buffer size range from 6 to 5461 Period size range from 3 to 2730 Using max buffer size 5460 Periods = 4 was set period_size = 1365 was set buffer_size = 5460 Channel: 0 Freq: 220.000000 but it fails to find hw:0,1 if i link it statically, is that expected behavior? dmx-gnto jutz # ./atestminhw1 Playback device is hw:0,1 Stream parameters are 48000Hz, S16_LE, 6 channels ALSA lib pcm.c:2109:(snd_pcm_open_conf) symbol _snd_pcm_hw_open is not defined inside (null) Failed: No such device or address this line succeeds with a dynamic linking and fails with if(0 > (iErr = snd_pcm_open(&pPcm, gs_pcNameDev, SND_PCM_STREAM_PLAYBACK, 0))) goto fin; here is my static link arguments: gcc -g -static -o atestminhw1 atestmin.c -lasound -lm -lpthread -ldl here is my dynamic link arguments: gcc -g -o atestminhw1 atestmin.c -lasound and, for completeness, here is my lame test app, derived from other test apps that come with the source: #include <stdio.h> #include <stdlib.h> #include <string.h> #include <sched.h> #include <errno.h> #include <getopt.h> #include <inttypes.h> #include <ctype.h> #include <byteswap.h> #define ALSA_PCM_NEW_HW_PARAMS_API #define ALSA_PCM_NEW_SW_PARAMS_API #include <alsa/asoundlib.h> #include <sys/time.h> #include <math.h> #define MAX_CHANNELS 16 static char *gs_pcNameDev = "hw:0,1"; /* playback device */ //static char *gs_pcNameDev = "plughw"; /* playback device */ //static char *gs_pcNameDev = "default"; /* playback device */ static snd_pcm_format_t gs_SndPcmFmt = SND_PCM_FORMAT_S16; /* sample format */ static unsigned int gs_uRate = 48000; /* stream rate */ static unsigned int gs_uCntChan = 6; /* count of channels */ static unsigned int gs_uTimBuf = 0; /* ring buffer length in us */ static unsigned int gs_uTimPer = 0; /* period time in us */ static unsigned int gs_uCntPer = 4; /* number of periods */ static double gs_dFreq = 440; /* sinusoidal wave frequency in Hz */ static snd_pcm_uframes_t gs_uSizBuf; static snd_pcm_uframes_t gs_uSizPer; static int set_hwparams(snd_pcm_t *handle, snd_pcm_hw_params_t *params, snd_pcm_access_t access) { unsigned int uRate; int iErr; snd_pcm_uframes_t period_size_min; snd_pcm_uframes_t period_size_max; snd_pcm_uframes_t buffer_size_min; snd_pcm_uframes_t buffer_size_max; /* choose all parameters */ if (0 > (iErr = snd_pcm_hw_params_any(handle, params))) { printf("Broken configuration for playback: no configurations available: %s\n", snd_strerror(iErr)); return iErr; } /* set the interleaved read/write format */ if (0 > (iErr = snd_pcm_hw_params_set_access(handle, params, access))) { printf("Access type not available for playback: %s\n", snd_strerror(iErr)); return iErr; } /* set the sample format */ if (0 > (iErr = snd_pcm_hw_params_set_format(handle, params, gs_SndPcmFmt))) { printf("Sample format not available for playback: %s\n", snd_strerror(iErr)); return iErr; } /* set the count of channels */ if (0 > (iErr = snd_pcm_hw_params_set_channels(handle, params, gs_uCntChan))) { printf("Channels count (%i) not available for playbacks: %s\n", gs_uCntChan, snd_strerror(iErr)); return iErr; } /* set the stream rate */ uRate = gs_uRate; if (0 > (iErr = snd_pcm_hw_params_set_rate(handle, params, gs_uRate, 0))) { printf("Rate %iHz not available for playback: %s\n", gs_uRate, snd_strerror(iErr)); return iErr; } if (uRate != gs_uRate) { printf("Rate doesn't match (requested %iHz, get %iHz, err %d)\n", gs_uRate, uRate, iErr); return -EINVAL; } printf("Rate set to %iHz (requested %iHz)\n", uRate, gs_uRate); /* set the buffer time */ iErr = snd_pcm_hw_params_get_buffer_size_min(params, &buffer_size_min); iErr = snd_pcm_hw_params_get_buffer_size_max(params, &buffer_size_max); iErr = snd_pcm_hw_params_get_period_size_min(params, &period_size_min, NULL); iErr = snd_pcm_hw_params_get_period_size_max(params, &period_size_max, NULL); printf("Buffer size range from %lu to %lu\n",buffer_size_min, buffer_size_max); printf("Period size range from %lu to %lu\n",period_size_min, period_size_max); if (gs_uTimPer > 0) { printf("Requested period time %u us\n", gs_uTimPer); if (0 > (iErr = snd_pcm_hw_params_set_period_time_near(handle, params, &gs_uTimPer, NULL))) { printf("Unable to set period time %u us for playback: %s\n", gs_uTimPer, snd_strerror(iErr)); return iErr; } } if (gs_uTimBuf > 0) { printf("Requested buffer time %u us\n", gs_uTimBuf); if (0 > (iErr = snd_pcm_hw_params_set_buffer_time_near(handle, params, &gs_uTimBuf, NULL))) { printf("Unable to set buffer time %u us for playback: %s\n", gs_uTimBuf, snd_strerror(iErr)); return iErr; } } if (! gs_uTimBuf && ! gs_uTimPer) { gs_uSizBuf = buffer_size_max; if (! gs_uTimPer) gs_uSizBuf = (gs_uSizBuf / gs_uCntPer) * gs_uCntPer; printf("Using max buffer size %lu\n", gs_uSizBuf); if (0 > (iErr = snd_pcm_hw_params_set_buffer_size_near(handle, params, &gs_uSizBuf))) { printf("Unable to set buffer size %lu for playback: %s\n", gs_uSizBuf, snd_strerror(iErr)); return iErr; } } if (! gs_uTimBuf || ! gs_uTimPer) { printf("Periods = %u\n", gs_uCntPer); if (0 > (iErr = snd_pcm_hw_params_set_periods_near(handle, params, &gs_uCntPer, NULL))) { printf("Unable to set nperiods %u for playback: %s\n", gs_uCntPer, snd_strerror(iErr)); return iErr; } } snd_pcm_hw_params_get_buffer_size(params, &gs_uSizBuf); snd_pcm_hw_params_get_period_size(params, &gs_uSizPer, NULL); printf("was set period_size = %lu\n",gs_uSizPer); printf("was set buffer_size = %lu\n",gs_uSizBuf); if (2*gs_uSizPer > gs_uSizBuf) { printf("buffer to small, could not use\n"); return iErr; } /* write the parameters to device */ iErr = snd_pcm_hw_params(handle, params); if (iErr < 0) { printf("Unable to set hw params for playback: %s\n", snd_strerror(iErr)); return iErr; } return 0; } static int set_swparams(snd_pcm_t *handle, snd_pcm_sw_params_t *swparams) { int iErr; /* get the current swparams */ if (0 > (iErr = snd_pcm_sw_params_current(handle, swparams))) { printf("Unable to determine current swparams for playback: %s\n", snd_strerror(iErr)); return iErr; } /* start the transfer when a buffer is full */ if (0 > (iErr = snd_pcm_sw_params_set_start_threshold(handle, swparams, gs_uSizBuf))) { printf("Unable to set start threshold mode for playback: %s\n", snd_strerror(iErr)); return iErr; } /* allow the transfer when at least period_size frames can be processed */ if (0 > (iErr = snd_pcm_sw_params_set_avail_min(handle, swparams, gs_uSizPer))) { printf("Unable to set avail min for playback: %s\n", snd_strerror(iErr)); return iErr; } /* align all transfers to 1 sample */ if (0 > (iErr = snd_pcm_sw_params_set_xfer_align(handle, swparams, 1))) { printf("Unable to set transfer align for playback: %s\n", snd_strerror(iErr)); return iErr; } /* write the parameters to the playback device */ if (0 > (iErr = snd_pcm_sw_params(handle, swparams))) { printf("Unable to set sw params for playback: %s\n", snd_strerror(iErr)); return iErr; } return 0; } static void generate_sine(uint8_t *pu8Frames, int iChannel, int iCntFrames, double *pdPhase, float fFreq) { double dPhase = *pdPhase; double dPhaseMax = 1.0 / fFreq; double dSizStep = 1.0 / (double)gs_uRate; double dRes; int iIdxChan; int32_t i32Res; int16_t *pi16Samp = (int16_t*) pu8Frames; while (iCntFrames-- > 0) { for(iIdxChan=0;iIdxChan<gs_uCntChan;iIdxChan++) if (iIdxChan==iChannel) { dRes = (sin((dPhase * 2 * M_PI) / dPhaseMax - M_PI)) * 0x03fffffff; /* Don't use MAX volume */ i32Res = dRes; *pi16Samp++ = i32Res >>16; } else *pi16Samp++ = 0; dPhase += dSizStep; if (dPhase >= dPhaseMax) dPhase -= dPhaseMax; } *pdPhase = dPhase; } /* * Underrun and suspend recovery */ static int xrun_recovery(snd_pcm_t *pPcm, int iErr) { if (iErr == -EPIPE) { /* under-run */ iErr = snd_pcm_prepare(pPcm); if (iErr < 0) printf("Can't recovery from underrun, prepare failed: %s\n", snd_strerror(iErr)); return 0; } else if (iErr == -ESTRPIPE) { while ((iErr = snd_pcm_resume(pPcm)) == -EAGAIN) sleep(1); /* wait until the suspend flag is released */ if (iErr < 0) { iErr = snd_pcm_prepare(pPcm); if (iErr < 0) printf("Can't recovery from suspend, prepare failed: %s\n", snd_strerror(iErr)); } return 0; } return iErr; } static int write_loop(snd_pcm_t *pPcm, int iChan, int iNumPer, uint8_t *pu8Frames) { double phase = 0; int iErr, iIdxPer, iCntFrames; uint8_t *pu8FramesLocal; float fFreq = gs_dFreq*(iChan+1)/2; // provide an audible chan change cue printf("Channel: %d Freq: %f\n", iChan, fFreq); for(iIdxPer = 0; iIdxPer < iNumPer; iIdxPer++) { iCntFrames = gs_uSizPer; pu8FramesLocal = pu8Frames; generate_sine(pu8FramesLocal, iChan, gs_uSizPer, &phase, fFreq); while (0 < iCntFrames) { if (-EAGAIN == (iErr = snd_pcm_writei(pPcm, pu8FramesLocal, iCntFrames))) continue; if (0 > iErr) { printf("Write error: %d,%s\n", iErr, snd_strerror(iErr)); if (0 > xrun_recovery(pPcm, iErr)) { printf("xrun_recovery failed: %d,%s\n", iErr, snd_strerror(iErr)); return -1; } break; // skip one period } pu8FramesLocal += snd_pcm_frames_to_bytes(pPcm, iErr); iCntFrames -= iErr; } } if (gs_uSizBuf > iIdxPer * gs_uSizPer) { snd_pcm_drain(pPcm); snd_pcm_prepare(pPcm ); } return 0; } int main(int argc, char *argv[]) { int iErr; int iIdxChan; uint8_t *pu8Frames = NULL; snd_pcm_t *pPcm = NULL; snd_pcm_hw_params_t *hwparams; snd_pcm_sw_params_t *swparams; snd_pcm_hw_params_alloca(&hwparams); snd_pcm_sw_params_alloca(&swparams); printf("Playback device is %s\n", gs_pcNameDev); printf("Stream parameters are %iHz, %s, %i channels\n", gs_uRate, snd_pcm_format_name(gs_SndPcmFmt), gs_uCntChan); if(0 > (iErr = snd_pcm_open(&pPcm, gs_pcNameDev, SND_PCM_STREAM_PLAYBACK, 0))) goto fin; if(0 > (iErr = set_hwparams(pPcm, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED))) goto fin; if(0 > (iErr = set_swparams(pPcm, swparams))) goto fin; if(!(pu8Frames = malloc(snd_pcm_frames_to_bytes(pPcm, gs_uSizPer)))) goto fin; while(1) for(iIdxChan = 0; iIdxChan < gs_uCntChan; iIdxChan++) if (0> (iErr = write_loop(pPcm, iIdxChan, (gs_uRate*3)/gs_uSizPer, pu8Frames))) goto fin; iErr = EXIT_SUCCESS; fin: if(iErr) printf("Failed: %s\n",snd_strerror(iErr)); if(pu8Frames) free(pu8Frames); if(pPcm) snd_pcm_close(pPcm); exit(iErr); }

2 6

[alsa-devel] [2/4] 2.6.21-rc5: known regressions (v2)
by Adrian Bunk 31 Mar '07

31 Mar '07

This email lists some known regressions in Linus' tree compared to 2.6.20. If you find your name in the Cc header, you are either submitter of one of the bugs, maintainer of an affectected subsystem or driver, a patch of you caused a breakage or I'm considering you in any other way possibly involved with one or more of these issues. Due to the huge amount of recipients, please trim the Cc when answering. Subject : snd_hda_intel doesn't work with ASUS M2V mainboard References : http://bugzilla.kernel.org/show_bug.cgi?id=8273 Submitter : Hans-Georg Rist <hg.rist(a)web.de> Status : unknown Subject : snd_intel8x0: divide error: 0000 References : http://lkml.org/lkml/2007/3/5/252 Submitter : Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> Status : unknown Subject : hal daemon crashes after pulling a USB serial device References : http://www.opensubscriber.com/message/linux-usb-devel@lists.sourceforge.net… Submitter : Andi Kleen <ak(a)suse.de> Handled-By : Oliver Neukum <oneukum(a)suse.de> Status : problem is being debugged Subject : USB: iPod doesn't work (CONFIG_USB_SUSPEND) References : http://lkml.org/lkml/2007/3/21/320 Submitter : Tino Keitel <tino.keitel(a)gmx.de> Caused-By : Marcelo Tosatti <marcelo(a)kvack.org> commit 1d619f128ba911cd3e6d6ad3475f146eb92f5c27 Handled-By : Oliver Neukum <oneukum(a)suse.de> Status : problem is being debuggged Subject : USB: Oops when changing DVB-T adapter References : http://lkml.org/lkml/2007/3/9/212 Submitter : CIJOML <cijoml(a)volny.cz> Status : unknown Subject : forcedeth: sporadic under-load crashes References : http://lkml.org/lkml/2007/3/26/63 Submitter : Ingo Molnar <mingo(a)elte.hu> Handled-By : Ingo Molnar <mingo(a)elte.hu> Ayaz Abdulla <aabdulla(a)nvidia.com> Status : problem is being debugged

1 0

[alsa-devel] 14rc3 fails to compile on vanilla 2.6.20.3
by Andrea Gozzi 30 Mar '07

30 Mar '07

Hi. I recently compiled a new kernel, 2.6.20.3, but when I tried to build alsa 14rc3 it failed. I also had troubles with the ati fglrx driver (missing declarations in headers - solved) so it could maybe be related to the new 20.x kernels. I attached logs for both the "configure" and "make". Andrea

2 3

[alsa-devel] Questions on Bluetooth Audio & Alsa Relationship
by Allyn, Mark A 30 Mar '07

30 Mar '07

Hello: Where is the correct place to discuss issues pertaining to snd-bt-sco and its relationship with Alsa? I am having trouble getting snd-bt-sco to work with Alsa 1.0.14rc3. I am getting a kernel oops for a NULL pointer somewhere in snd_bt_sco_playback_open. I looked on the blutooth-alsa sourceforge site, it seems that snd-bt-sco is no longer maintained and that they are going toward a solution using something called plugz. Plugz, however requires a kernel patch that works for kernels 2.6.19 and above, but not with 2.6.14, which is the kernel in the distro that I am using. I have attempted to ask some questions on the bluez developer email list on sourceforge, but I only got one response saying that the snd-bt-sco solution is more mature than the plugz development effort. I then asked about the kernel oops and have not got any response. What is the state of bluetooth audio development? Is it slacking off? Perhaps is the a plugz email list that I don't know about? Thank you for your help. Truly, Mark Allyn

2 1

[alsa-devel] Custom ioctl
by intellij 30 Mar '07

30 Mar '07

hi, how can I send custom request to my driver? Looking into the sources I suspect that sending ioctl() on opened hardware pcm will not reach the playback_ops.ioctl in my driver. thank you very much.

2 1

[alsa-devel] [PATCH] snd-usb-caiaq: Make playback work
by Karsten Wiese 29 Mar '07

29 Mar '07

Hi Takashi, some typo fixes. best regards, Karsten -----------------------------------------------------------------------> From: Karsten Wiese <fzu(a)wemgehoertderstaat.de> Date: Thu, 29 Mar 2007 16:39:47 +0200 Subject: [PATCH] snd-usb-caiaq: Make playback work Signed-off-by: Karsten Wiese <fzu(a)wemgehoertderstaat.de> --- sound/usb/caiaq/caiaq-audio.c | 15 ++++++++------- 1 files changed, 8 insertions(+), 7 deletions(-) diff --git a/sound/usb/caiaq/caiaq-audio.c b/sound/usb/caiaq/caiaq-audio.c index e80c8db..0414d76 100644 --- a/sound/usb/caiaq/caiaq-audio.c +++ b/sound/usb/caiaq/caiaq-audio.c @@ -388,8 +388,8 @@ static void read_in_urb_mode2(struct snd_usb_caiaqdev *dev, struct snd_pcm_runtime *rt = sub->runtime; char *audio_buf = rt->dma_area; int sz = frames_to_bytes(rt, rt->buffer_size); - audio_buf[dev->audio_in_buf_pos[stream]++] - = usb_buf[i]; + audio_buf[dev->audio_in_buf_pos[stream]++] = + usb_buf[i]; dev->period_in_count[stream]++; if (dev->audio_in_buf_pos[stream] == sz) dev->audio_in_buf_pos[stream] = 0; @@ -436,19 +436,21 @@ static void fill_out_urb(struct snd_usb_caiaqdev *dev, spin_lock(&dev->spinlock); for (i = 0; i < iso->length;) { - for (stream = 0; stream < dev->n_streams; stream++) { + for (stream = 0; stream < dev->n_streams; stream++, i++) { sub = dev->sub_playback[stream]; if (sub) { struct snd_pcm_runtime *rt = sub->runtime; char *audio_buf = rt->dma_area; int sz = frames_to_bytes(rt, rt->buffer_size); - usb_buf[i++] - = audio_buf[dev->audio_out_buf_pos[stream]++]; + usb_buf[i] = + audio_buf[dev->audio_out_buf_pos[stream]]; + dev->period_out_count[stream]++; dev->audio_out_buf_pos[stream]++; if (dev->audio_out_buf_pos[stream] == sz) dev->audio_out_buf_pos[stream] = 0; } else - usb_buf[i++] = 0; + usb_buf[i] = 0; + } /* fill in the check bytes */ if (dev->spec.data_alignment == 2 && @@ -456,7 +458,6 @@ static void fill_out_urb(struct snd_usb_caiaqdev *dev, (dev->n_streams * CHANNELS_PER_STREAM)) for (stream = 0; stream < dev->n_streams; stream++, i++) usb_buf[i] = MAKE_CHECKBYTE(dev, stream, i); - } } spin_unlock(&dev->spinlock);

2 1

[alsa-devel] Alsa OSS - who is maintaining it?
by Stefan de Konink 29 Mar '07

29 Mar '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA512 Hello, I'm looking for the maintainers of the Alsa OSS layer. It seems it is getting more seriously broken each month. The last serious updates were almost 7 months ago by tiwai and perex. Dev must have noticed the 'skips' on the end of an OSS stream. The complete broken big endian emulation and so on. What is currently happening in this area? Yours Sincerely, Stefan de Konink -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.3 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGByXUYH1+F2Rqwn0RCmU3AJ9O25z5wcCpXLDUnx61idhB7MGcYACcD0fh Fl95KN+x77vuC006U0mexJ8= =guiG -----END PGP SIGNATURE-----

4 8

[alsa-devel] Function ap192_4wire_start in revo.c
by dustin＠seznam.cz 29 Mar '07

29 Mar '07

Hello, while working on support for AK4114 in Audiotrak MI/ODI/O, I have come accross a bug in revo.c (used for major inspiration :) )The function static unsigned char ap192_4wire_start(struct snd_ice1712 *ice) returns value of current GPIO bits, which is unsigned int. However, return value of the function is unsigned char. This results in clearing all GPIO8-22 in next snd_ice1712_gpio_write call of the 4-wire protocol communication which is a little bit inconvenient :). Thanks for fixing in HG. Best regards, Pavel.

2 1

Re: [alsa-devel] Status of CMI9738 and SiS7012
by Takashi Iwai 29 Mar '07

29 Mar '07

Hi, Please don't drop Cc to alsa-devel... At Wed, 28 Mar 2007 10:23:22 -0600, Rob Franklin wrote: > > On 3/28/07, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Tue, 27 Mar 2007 10:44:23 -0600, > > Rob Franklin wrote: > > > > > > On 3/27/07, Takashi Iwai <tiwai(a)suse.de> wrote: > > > > At Tue, 27 Mar 2007 08:49:05 -0600, > > > > Rob Franklin wrote: > > > > > > > > > > Hi All, > > > > > > > > > > I posted a question a couple of weeks ago to the alsa-users list > > > > > (Can't get spdif working with ALSA and FC6 on ECS L4S5MG/651+ > > > > > motherboard), but never got a response. > > > > > > > > > > I have been looking at the ALSA source code for the AC'97 codec on my > > > > > board (CMI9738) and the source code for the intel8x0 driver (for my > > > > > SiS 7012). However, as far as I can tell from looking at the source, > > > > > S/PDIF is not enabled/supported for this combination of devices. > > > > > Additionally, Googling around I haven't found any mention of people > > > > > getting spdif working with ALSA and a CM9738 / SiS7012 combo. > > > > > > > > > > I am wondering whether anyone on the developer list can confirm that > > > > > this is the case. Also, if this is the case can someone give me an > > > > > idea of what would be required to get this working. The system is > > > > > setup with dual boot and so I could grab some of the values from > > > > > windows using a PCI sniffer, and I might be able to do some hacking on > > > > > the source code and submit a patch if someone could help guide me in > > > > > the right direction. > > > > > > > > Are you sure it's CM9738? AFAIK, CM9738 doesn't support SPDIF at > > > > all, but CM9739 does... > > > > > > > That was my first thought too, but I pulled open the case last night > > > and verified that it was indeed a 9738. The download page from the > > > motherboard manufacturer has a driver for CMI9738 and a CMI9738/9739 > > > driver (http://www.ecsusa.com/downloads/drivers_sound.html). I am > > > wondering if there are two different versions of this chip. > > > > > > Here is the output from lspci -vv for the sound device: > > > > > > 00:02.7 Multimedia audio controller: Silicon Integrated Systems [SiS] > > > AC'97 Sound Controller (rev a0) > > > Subsystem: Elitegroup Computer Systems Unknown device 0a44 > > > 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: 32 (13000ns min, 2750ns max) > > > Interrupt: pin C routed to IRQ 22 > > > Region 0: I/O ports at d000 [size=256] > > > Region 1: I/O ports at d400 [size=128] > > > Capabilities: <access denied> > > > > > > One thought that I had was to force the AC'97 driver to detect the > > > chip as a 9739 instead of a 9738 to see whether my 9738 can work as a > > > 9739. But I'm not sure how to do that short of modifying the source > > > code and recompiling the kernel module. > > > > First check whether its codec id is CMI9738, see > > /proc/asound/card0/codec97#0/* files. If it's really CM9738, you can > > hack pci/ac97/ac97_codec.c to replace patch_cm9738 with patch_cm9739 > > in snd_ac97_codec_ids[]. > > > Here are the first couple of lines from /proc/asound/card0/codec97#0/ac97#0-0 : > > 0-0/0: C-Media Electronics CMI9738 > > PCI Subsys Vendor: 0x1019 > PCI Subsys Device: 0x0a44 > > Last night I hacked the ac97_codec.c and recompiled snd-ac97-codec.ko, > and now the file reads > > 0-0/0: C-Media Electronics -- rob CMI9739 > > Everything seems to be working the same as it was before, and I can > get stereo sound from the line-out jack just fine. However, when I > try to play back sound using the spdif I get the following: > > $ aplay -D plug:iec958 ac3_48khz_diatonis_soal.ac3 > ALSA lib setup.c:555:(add_elem) Cannot obtain info for CTL elem > (MIXER,'IEC958 Playback Default',0,0,0): No such file or directory I guess the control element wasn't created indeed because of lack of SPDIF feature on the codec. patch_cm9739() checks the availability of SPDIF bit (SPCV) and turn off the whole SPDIF feature if the bit isn't set. You can avoid that check just for testing. If this doesn't work, the possible situation is that the SPDIF is transmitted directly from the controller chip without AC97 codec. Unfortunately, we have no hardware information about SIS7012, so don't know whether it's even possible, and how to control it. > Is there a script I need to run or file I need to update my list of > CTLs (I have seen references to alsaconf and snddev on the web but > neither seems to be included with my FC5 install of alsa-utils)? % alsactl -f somefile store > Also, is there any easy way to reload the snd-ac97-codec.ko module. I > tried removing the modules with "rmmod snd-ac97-codec" and "modprobe > snd-ac97-codec". The module came out ok, but I got a bunch of errors > like this when I tried to reinsert the modules: > > Mar 27 21:16:46 XXXX kernel: snd_ac97_codec: Unknown symbol snd_info_register > Mar 27 21:16:46 XXXX kernel: snd_ac97_codec: disagrees about version > of symbol snd_ctl_add > > So right now I am rebooting the machine every time I want to replace > snd-ac97-codec. Seems that you still have the old modules on memory while installing the new modules. First unload all snd related modules, then reload again. Otherwise, something wrong with the build with your kernel tree. I never used FC or RedHat, so can't answer about it. Takashi

2 2

[alsa-devel] Enabling SPDIF output on Turtle Beach Audio Advantage Micro (CMEDIA CM102S)
by Eric Sandeen 28 Mar '07

28 Mar '07

I have this Turtle Beach Audio Advantage Micro usb sound card, er, stick. Inside it has pretty much nothing but a C-Media CM102S chip. TB claims, as does C-Media, that it can do S/PDIF out, and I believe them based on the glowing red TOSLINK light. :) I found previous emails on this list (http://www.mail-archive.com/alsa-devel@lists.sourceforge.net/msg09979.html) suggesting that there is probably some magic to be applied to tell usb hardware to go into S/PDIF mode. I've captured some traces under windows, linked from http://sandeen.net/tbaam/index.html Specifically, http://sandeen.net/tbaam/2spkr-spdif.txt seems to be a fairly short burst of USB activity when I used the control panel to switch from 2 speaker mode to spdif mode. I see the exact same activity when switching from headphone mode to spdif mode. Strangely, switching from, say, spdif back to 2 speaker seems to generate a lot more traffic. The full annotated trace is here: http://sandeen.net/tbaam/usbsnoop1.txt In any case, are these traces helpful to anyone who understands how to talk to a C-Media CM102S usb sound card? Does it look possible to poke this hardware into S/PDIF mode with this information? Thanks, -Eric

1 0

[alsa-devel] how to subscribe to new maillist?
by Songmao Tian 28 Mar '07

28 Mar '07

Hello, I have tried several times on the web page http://mailman.alsa-project.org/mailman/listinfo/alsa-devel, but i can't receive the confirm mail, so how to subscribe? Tian

4 3

[alsa-devel] New code for conexant chipsets in -mm tree
by Rodrigo Luiz 28 Mar '07

28 Mar '07

Hi. I have a HDA NVidia, with a Conexant chipset CX20549 Venice (CXT5045) I tried to test the last alsa-kernel in -mm tree to support my mic. (2.6.21-rc4-mm1 and 2.6.21-rc5-mm2) When I try to load the hda-intel, I receive this message: Mar 27 11:29:08 dig kernel: hda_intel: azx_get_response timeout, switching to polling mode Mar 27 11:29:09 dig kernel: hda_intel: azx_get_response timeout, switching to single_cmd mode After loaded, the speakers doesn't work. Headphone works ok. Mic too. Look in code, I apply this comment and now the speakers works for me: --- patch_conexant.c.orig 2007-03-27 11:37:43.000000000 -0300 +++ patch_conexant.c 2007-03-27 11:38:02.000000000 -0300 @@ -822,7 +822,7 @@ case CXT5045_FUJITSU: spec->input_mux = &cxt5045_capture_source; spec->num_init_verbs = 2; - spec->init_verbs[1] = cxt5045_mic_sense_init_verbs; +// spec->init_verbs[1] = cxt5045_mic_sense_init_verbs; spec->mixers[0] = cxt5045_mixers; codec->patch_ops.init = cxt5045_init; break; But, I don't understand why, because this code is not executed for my chipset. Thanks, Rodrigo Luiz

3 8

[alsa-devel] snd-es1968 capture broken
by Rene Herman 28 Mar '07

28 Mar '07

Good day. Capture is broken on snd-es1968, at least if it is driving a ES1970MS-3D (ESS Canyon 3D). The captured stream is "stuttering" like mad. Kernel 2.6.20 (ALSA 1.0.14-rc1) with alsa-lib 1.0.11 (my distribution's version). Does not happen with a CS4236 in the same machine and same software version(s). Don't know if this driver has a specific maintainer. If needed I guess I can try to pin it down to some change. I hardly use capture on this card but it wasn't always broken... Rene.

2 15

Re: [alsa-devel] Basic DMA question
by Sven Johnsson 28 Mar '07

28 Mar '07

Rene, I have already read the ALSA driver document. What I really like to understand is when DMA transfers are invoked, and what the source and destination for the transfer is. Regards, Sven 2007/3/26, Rene Herman <rene.herman(a)gmail.com>: > On 03/26/2007 03:13 AM, Sven Johnsson wrote: > > > I'm new to writing ALSA drivers, and I could use some help on > > clarifying how DMA is used in ALSA. > > "Writing an ALSA Driver" is a useful document. Google will find it... > > By the way, the alsa-devel mailinglist recently moved from > sourceforge.net to alsa-project.org directly and the new list is > moderated for non-subscribers. I just approved your message from the > moderation queue manually, but please consider subscribing, at: > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > Thanks, > Rene > > > > > > > > Regards, > > Sven > > _______________________________________________ > > Alsa-devel mailing list > > Alsa-devel(a)alsa-project.org > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > > >

3 2

Re: [alsa-devel] Status of CMI9738 and SiS7012
by Takashi Iwai 28 Mar '07

28 Mar '07

At Tue, 27 Mar 2007 10:44:23 -0600, Rob Franklin wrote: > > On 3/27/07, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Tue, 27 Mar 2007 08:49:05 -0600, > > Rob Franklin wrote: > > > > > > Hi All, > > > > > > I posted a question a couple of weeks ago to the alsa-users list > > > (Can't get spdif working with ALSA and FC6 on ECS L4S5MG/651+ > > > motherboard), but never got a response. > > > > > > I have been looking at the ALSA source code for the AC'97 codec on my > > > board (CMI9738) and the source code for the intel8x0 driver (for my > > > SiS 7012). However, as far as I can tell from looking at the source, > > > S/PDIF is not enabled/supported for this combination of devices. > > > Additionally, Googling around I haven't found any mention of people > > > getting spdif working with ALSA and a CM9738 / SiS7012 combo. > > > > > > I am wondering whether anyone on the developer list can confirm that > > > this is the case. Also, if this is the case can someone give me an > > > idea of what would be required to get this working. The system is > > > setup with dual boot and so I could grab some of the values from > > > windows using a PCI sniffer, and I might be able to do some hacking on > > > the source code and submit a patch if someone could help guide me in > > > the right direction. > > > > Are you sure it's CM9738? AFAIK, CM9738 doesn't support SPDIF at > > all, but CM9739 does... > > > That was my first thought too, but I pulled open the case last night > and verified that it was indeed a 9738. The download page from the > motherboard manufacturer has a driver for CMI9738 and a CMI9738/9739 > driver (http://www.ecsusa.com/downloads/drivers_sound.html). I am > wondering if there are two different versions of this chip. > > Here is the output from lspci -vv for the sound device: > > 00:02.7 Multimedia audio controller: Silicon Integrated Systems [SiS] > AC'97 Sound Controller (rev a0) > Subsystem: Elitegroup Computer Systems Unknown device 0a44 > 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: 32 (13000ns min, 2750ns max) > Interrupt: pin C routed to IRQ 22 > Region 0: I/O ports at d000 [size=256] > Region 1: I/O ports at d400 [size=128] > Capabilities: <access denied> > > One thought that I had was to force the AC'97 driver to detect the > chip as a 9739 instead of a 9738 to see whether my 9738 can work as a > 9739. But I'm not sure how to do that short of modifying the source > code and recompiling the kernel module. First check whether its codec id is CMI9738, see /proc/asound/card0/codec97#0/* files. If it's really CM9738, you can hack pci/ac97/ac97_codec.c to replace patch_cm9738 with patch_cm9739 in snd_ac97_codec_ids[]. Takashi

1 0

[alsa-devel] Status of CMI9738 and SiS7012
by Rob Franklin 27 Mar '07

27 Mar '07

Hi All, I posted a question a couple of weeks ago to the alsa-users list (Can't get spdif working with ALSA and FC6 on ECS L4S5MG/651+ motherboard), but never got a response. I have been looking at the ALSA source code for the AC'97 codec on my board (CMI9738) and the source code for the intel8x0 driver (for my SiS 7012). However, as far as I can tell from looking at the source, S/PDIF is not enabled/supported for this combination of devices. Additionally, Googling around I haven't found any mention of people getting spdif working with ALSA and a CM9738 / SiS7012 combo. I am wondering whether anyone on the developer list can confirm that this is the case. Also, if this is the case can someone give me an idea of what would be required to get this working. The system is setup with dual boot and so I could grab some of the values from windows using a PCI sniffer, and I might be able to do some hacking on the source code and submit a patch if someone could help guide me in the right direction. Thanks, Rob

2 1

[alsa-devel] Correct value for AK4114_DIF2 constant?
by dustin＠seznam.cz 27 Mar '07

27 Mar '07

Hello, While going through AK4114 support I have hit the constant AK4114_DIF2 in ak4114.h . I would say its value should be (1<<6) according to AK4114 datasheet. This is current situation: #define AK4114_DIF2 (1<<5) /* Audio Data Control */ #define AK4114_DIF1 (1<<5) /* Audio Data Control */ #define AK4114_DIF0 (1<<4) /* Audio Data Control */ I think cards using this code should experience related problems. E.g. in juli.c and revo.c the constant #define AK4114_DIF_I24I2S (AK4114_DIF2|AK4114_DIF1|AK4114_DIF0) /* STDO: I2S; LRCLK, BICK = Input */ is used, intended to set output serial data mode to I2S/slave perhaps compatible with ICE1712/24. This may be causing problems mentioned in revo.c: #if 0 /* FIXME: ak4114 makes the sound much lower due to some confliction, * so let's disable it right now... */ #define BUILD_AK4114_AP192 #endif I am trying to add support for SPDIF-IN of AudioTrak MI/ODI/O addon card (ice1724 + ak4114), hopefully this will be a little step forward. Thanks for any reaction. Best regards, Pavel.

2 1

[alsa-devel] capture line-out
by Tomas Carnecky 27 Mar '07

27 Mar '07

Is it possible to capture/record line-out (or simply 'what the user hears')? I'm writing 'fraps for linux' and so far I have video capturing implemented and that works great, but users are also requesting sound-capturing and I'm looking into how that could be done. thanks tom

2 10

[alsa-devel] [3/5] 2.6.21-rc5: known regressions
by Adrian Bunk 27 Mar '07

27 Mar '07

This email lists some known regressions in Linus' tree compared to 2.6.20. If you find your name in the Cc header, you are either submitter of one of the bugs, maintainer of an affectected subsystem or driver, a patch of you caused a breakage or I'm considering you in any other way possibly involved with one or more of these issues. Due to the huge amount of recipients, please trim the Cc when answering. Subject : Oops when changing DVB-T adapter References : http://lkml.org/lkml/2007/3/9/212 Submitter : CIJOML <cijoml(a)volny.cz> Status : unknown Subject : snd_intel8x0: divide error: 0000 References : http://lkml.org/lkml/2007/3/5/252 Submitter : Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> Status : unknown Subject : USB: iPod doesn't work (CONFIG_USB_SUSPEND) References : http://lkml.org/lkml/2007/3/21/320 Submitter : Tino Keitel <tino.keitel(a)gmx.de> Caused-By : Marcelo Tosatti <marcelo(a)kvack.org> commit 1d619f128ba911cd3e6d6ad3475f146eb92f5c27 Handled-By : Oliver Neukum <oneukum(a)suse.de> Status : problem is being debuggged

1 0

Re: [alsa-devel] [Alsa-devel] [RFC][RFT] Adding support for Jazz16 based sound cards
by Rask Ingemann Lambertsen 27 Mar '07

27 Mar '07

On Mon, Mar 12, 2007 at 12:28:00PM +0100, Takashi Iwai wrote: > At Mon, 12 Mar 2007 00:24:36 +0100, > Rask Ingemann Lambertsen wrote: > > > I tried to include u-law support, but it doesn't sound right and playback > > speed is about one third of what it should be. I could really use some > > documentation here. I'll remove it before submitting a final patch if I > > don't get it to work. > > OK. I never got u-law or a-law to work. I tried various ADPCM formats which gave correct playback speed, but much noise. There seems to be a Media Vision ADPCM format, which is probably what the Jazz16 wants, but I have no information about that format. FWIW, the playback_format/capture_format values I've tried are: #define SB_DSP_ALAW_OUTPUT_AUTO 0x1F #define SB_DSP_ALAW_INPUT_AUTO 0x2F #define SB_DSP_ULAW_OUTPUT_AUTO 0x7f #define SB_DSP_ULAW_INPUT_AUTO 0x8f #define SB_DSP_ADPCM_OUTPUT_AUTO 0x7C+1 #define SB_DSP_ADPCM_INPUT_AUTO 0x8C+1 > > Setting the irq of the MPU-401 port requires the SB part to be active and > > even worse, DSP commands need to be sent. Would it be better to just use the > > MPU-401 port without an irq and avoid the complexity? > > Isn't SB part already active? It's fine with non-irq version (which > uses timer) if the code gets too complicated, of course. It turned out that I can't disable just the MPU port. -- Rask Ingemann Lambertsen

1 0

[alsa-devel] [PATCH] PnP: Protocol for Media Vision Jazz16 sound cards
by Rask Ingemann Lambertsen 27 Mar '07

27 Mar '07

From: Rask Ingemann Lambertsen <rask(a)sygehus.dk> - Add code to detect and PnP protocol to manage Media Vision Jazz16 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 depends on the patch "ALSA: Support Media Vision Jazz16 chips" because the patched include/sound/sb.h is needed. drivers/pnp/Kconfig | 12 drivers/pnp/Makefile | 1 drivers/pnp/jazz16pnp.c | 555 ++++++++++++++++++++++++++++++++++++++ 3 files changed, 568 insertions(+) --- linux-2.6.20/drivers/pnp/Makefile-orig 2007-02-19 23:24:22.000000000 +0100 +++ linux-2.6.20/drivers/pnp/Makefile 2007-02-19 23:24:22.000000000 +0100 @@ -8,3 +8,4 @@ obj-$(CONFIG_PNPACPI) += pnpacpi/ obj-$(CONFIG_PNPACPI) += pnpacpi/ obj-$(CONFIG_PNPBIOS) += pnpbios/ obj-$(CONFIG_ISAPNP) += isapnp/ +obj-$(CONFIG_JAZZ16PNP) += jazz16pnp.o --- linux-2.6.20/drivers/pnp/jazz16pnp.c-orig 2007-03-11 22:11:58.000000000 +0100 +++ linux-2.6.20/drivers/pnp/jazz16pnp.c 2007-03-25 21:27:18.000000000 +0200 @@ -0,0 +1,555 @@ +/* + * Media Vision Jazz16 Plug & Play support. + * + * Copyright (c) 2007 Rask Ingemann Lambertsen <rask(a)sygehus.dk> + * + * 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 + * + * This driver manages two devices: + * + * Device 0: The game port. + * Device 1: The SoundBlaster look-alike part. + * + * Sources: + * sound/oss/sb_common.c + * sys/dev/isa/sbdsp.c (OpenBSD) + */ + +#include <linux/types.h> +#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> + +MODULE_AUTHOR("Rask Ingemann Lambertsen <rask(a)sygehus.dk>"); +MODULE_DESCRIPTION("Jazz16 Plug & Play support"); +MODULE_LICENSE("GPL"); + +int jazz16pnp_disable = false; /* Disable Jazz16 PnP. */ +module_param_named(disable, jazz16pnp_disable, bool, false); +MODULE_PARM_DESC(disable, "Jazz16 Plug & Play disable switch"); + +int jazz16pnp_verbose = false; /* Verbosity level. */ +module_param_named(verbose, jazz16pnp_verbose, bool, false); +MODULE_PARM_DESC(verbose, "Jazz16 Plug & Play verbosity level"); + +#define PFX "jazz16pnp: " +#define PFX_DEBUG KERN_DEBUG PFX +#define PFX_INFO KERN_INFO PFX +#define PFX_WARN KERN_WARNING PFX + +/* Jazz16 configuration. */ +#define SB_JAZZ16_CONFIG_PORT 0x201 +#define SB_JAZZ16_WAKEUP 0xaf +#define SB_JAZZ16_SET_PORTS 0x50 +#define SB_JAZZ16_SET_DMAINTR 0xFB + +extern void *pnp_alloc(long size); +extern void pnp_free_option(struct pnp_option *option); + +static struct pnp_dev *jazz16_gameport_dev = NULL; +static struct pnp_dev *jazz16_sb_dev = NULL; +static struct resource *jazz16pnp_fm_res = NULL; + +static uint jazz16_sb_port = 0, jazz16_sb_irq = 0; +static uint jazz16_sb_dma1 = 0, jazz16_sb_dma2 = 0; +static uint jazz16_mpu401_port = 0, jazz16_mpu401_irq = 0; + +static uint jazz16pnp_count = 0; + +static struct pnp_protocol jazz16pnp_protocol; + +static bool __init jazz16pnp_easy_option(struct pnp_dev *dev, uint pri, + struct pnp_port *port, uint nports, + struct pnp_irq *irq, uint nirqs, + struct pnp_dma *dma, uint ndmas) +{ + struct pnp_option *opt; + struct pnp_port *new_port; + struct pnp_irq *new_irq; + struct pnp_dma *new_dma; + uint i; + + switch (pri) { + case PNP_RES_PRIORITY_PREFERRED: + case PNP_RES_PRIORITY_ACCEPTABLE: + case PNP_RES_PRIORITY_FUNCTIONAL: + opt = pnp_register_dependent_option(dev, pri); + break; + + default: + opt = pnp_register_independent_option(dev); + break; + } + if (!opt) + return false; + + for (i = 0; i < nports; i ++) { + new_port = pnp_alloc(sizeof(*new_port)); + if (!new_port) + goto out_no_mem; + memcpy(new_port, &port[i], sizeof(*new_port)); + pnp_register_port_resource(opt, new_port); + } + for (i = 0; i < nirqs; i ++) { + new_irq = pnp_alloc(sizeof(*new_irq)); + if (!new_irq) + goto out_no_mem; + memcpy(new_irq, &irq[i], sizeof(*new_irq)); + pnp_register_irq_resource(opt, new_irq); + } + for (i = 0; i < ndmas; i ++) { + new_dma = pnp_alloc(sizeof(*new_dma)); + if (!new_dma) + goto out_no_mem; + memcpy(new_dma, &dma[i], sizeof(*new_dma)); + pnp_register_dma_resource(opt, new_dma); + } + return true; + +out_no_mem: + return false; +} + +static struct pnp_port jazz16pnp_game_ports[1] __initdata = { + { .min = SB_JAZZ16_CONFIG_PORT, + .max = SB_JAZZ16_CONFIG_PORT, + .size = 1, .align = 0, + .flags = PNP_PORT_FLAG_FIXED, }, +}; + +static int __init jazz16pnp_add_gameport(void) +{ + struct pnp_dev *dev; + struct pnp_id *dev_id; + + if (!(dev = pnp_alloc(sizeof(*dev)))) + goto out_no_dev; + if (!(dev_id = pnp_alloc(sizeof(*dev_id)))) + goto out_no_id; + if (!jazz16pnp_easy_option(dev, PNP_RES_PRIORITY_INVALID, + jazz16pnp_game_ports, 1, + NULL, 0, NULL, 0)) + goto out_no_option; + + dev->protocol = &jazz16pnp_protocol; + dev->capabilities = PNP_READ; + dev->active = 1; + dev->number = 0; + + pnp_init_resource_table(&dev->res); + dev->res.port_resource[0].start = SB_JAZZ16_CONFIG_PORT; + dev->res.port_resource[0].end = SB_JAZZ16_CONFIG_PORT; + dev->res.port_resource[0].flags = IORESOURCE_IO; + + memcpy(dev_id->id, "PNPb02f", PNP_ID_LEN); + pnp_add_id(dev_id, dev); + + jazz16_gameport_dev = dev; + pnp_add_device(dev); + return 0; + +out_no_option: + if (dev->independent) + pnp_free_option(dev->independent); + kfree(dev_id); +out_no_id: + kfree(dev); +out_no_dev: + return -ENOMEM; +}; + +static struct pnp_port jazz16pnp_ports[2] __initdata = { + { .min = 0x220, .max = 0x260, .align = 0x20, .size = 0x10 }, + { .min = 0x300, .max = 0x330, .align = 0x10, .size = 0x02 }, +}; + +static struct pnp_irq jazz16pnp_irqs[2] __initdata = { + { .map[0] = (1 << 3) | (1 << 5) | (1 << 7) | + (1 << 9) | (1 << 10) | (1 << 15), + .flags = IORESOURCE_IRQ_HIGHEDGE }, + { .map[0] = (1 << 3) | (1 << 5) | (1 << 7) | + (1 << 9) | (1 << 10) | (1 << 15), + .flags = IORESOURCE_IRQ_HIGHEDGE }, +}; + +static struct pnp_dma jazz16pnp_dmas[2] __initdata = { + { .map = (1 << 1) | (1 << 3), .flags = IORESOURCE_DMA_8BIT, }, + { .map = (1 << 5) | (1 << 7), .flags = IORESOURCE_DMA_16BIT, }, +}; + +static int __init jazz16pnp_add_sb(void) +{ + struct pnp_dev *sbdev; + struct pnp_id *sbid; + + if (!(sbdev = pnp_alloc(sizeof(*sbdev)))) + goto out_no_sbdev; + if (!(sbid = pnp_alloc(sizeof(*sbid)))) + goto out_no_sbid; + + if (!jazz16pnp_easy_option(sbdev, PNP_RES_PRIORITY_PREFERRED, + jazz16pnp_ports, 2, + jazz16pnp_irqs, 2, + jazz16pnp_dmas, 2)) + goto out_no_sboption; + if (!jazz16pnp_easy_option(sbdev, PNP_RES_PRIORITY_ACCEPTABLE, + jazz16pnp_ports, 2, + jazz16pnp_irqs, 1, + jazz16pnp_dmas, 2)) + goto out_no_sboption; + memcpy(sbid->id, "PNPb00f", PNP_ID_LEN); + pnp_add_id(sbid, sbdev); + + pnp_init_resource_table(&sbdev->res); + sbdev->capabilities = PNP_READ | PNP_WRITE | PNP_CONFIGURABLE | PNP_DISABLE; + sbdev->protocol = &jazz16pnp_protocol; + sbdev->number = 1; + + jazz16_sb_dev = sbdev; + pnp_add_device(sbdev); + return 0; + +out_no_sboption: + if (sbdev->dependent) + pnp_free_option(sbdev->dependent); + kfree(sbid); +out_no_sbid: + kfree(sbdev); +out_no_sbdev: + return -ENOMEM; +} + +/* Configure the SB and MPU ports. Passing 0 for sbport disables the Jazz16. */ +static void jazz16_port_setup(const uint sbport, const uint mpuport) +{ + u8 config; + + switch (sbport) { + case 0x220: + case 0x240: + case 0x260: + config = sbport & 0x70; + jazz16_sb_port = sbport; + break; + + default: + config = 0; + jazz16_sb_port = 0; + break; + } + + switch (mpuport) { + case 0x300: + case 0x310: + case 0x320: + case 0x330: + config |= (mpuport >> 4) & 0x07; + jazz16_mpu401_port = mpuport; + break; + + default: + if (jazz16_sb_port) + BUG(); + } + if (jazz16pnp_verbose) + printk(PFX_DEBUG "Setting SB port = 0x%x, MPU port = 0x%x.\n", + jazz16_sb_port, jazz16_mpu401_port); + outb(SB_JAZZ16_WAKEUP, SB_JAZZ16_CONFIG_PORT); + outb(SB_JAZZ16_SET_PORTS, SB_JAZZ16_CONFIG_PORT); + outb(config, SB_JAZZ16_CONFIG_PORT); +} + +static u8 jazz16_irq_config(uint irq) +{ + switch (irq) { + case 3: return 3; + case 5: return 1; + case 7: return 4; + case 9: return 2; + case 10: return 5; + case 15: return 6; + default: return 0; + } +} + +static u8 jazz16_dma_config(uint dma) +{ + switch (dma) { + case 1: return 1; + case 3: return 2; + case 5: return 3; + case 7: return 4; + default: return 0; + } +} + +/* The my_sbdsp_xxx functions were ripped from sound/isa/sb/sb_common.c. */ +#define BUSY_LOOPS 100000 + +static int my_sbdsp_command(unsigned int base, unsigned char val) +{ + int i; + for (i = BUSY_LOOPS; i; i--) + if ((inb(SBP1(base, STATUS)) & 0x80) == 0) { + outb(val, SBP1(base, COMMAND)); + return 1; + } + return 0; +} + +static int my_sbdsp_get_byte(unsigned int base) +{ + int val; + int i; + for (i = BUSY_LOOPS; i; i--) { + if (inb(SBP1(base, DATA_AVAIL)) & 0x80) { + val = inb(SBP1(base, READ)); + return val; + } + } + return -ENODEV; +} + +static int my_sbdsp_reset(unsigned int base) +{ + int i; + + outb(1, SBP1(base, RESET)); + udelay(10); + outb(0, SBP1(base, RESET)); + udelay(30); + for (i = BUSY_LOOPS; i; i--) + if (inb(SBP1(base, DATA_AVAIL)) & 0x80) { + if (inb(SBP1(base, READ)) == 0xaa) + return 0; + else + break; + } + return -ENODEV; +} + +static int jazz16_irq_dma_setup(unsigned int base, uint sbirq, uint mpuirq, uint dma8, uint dma16) +{ + u8 dma_config, irq_config; + + if (jazz16pnp_verbose) + printk(PFX_DEBUG "Setting SB irq %u, dma %u&%u, MPU irq %u.\n", + sbirq, dma8, dma16, mpuirq); + irq_config = jazz16_irq_config(sbirq) | jazz16_irq_config(mpuirq) << 4; + dma_config = jazz16_dma_config(dma8) | jazz16_dma_config(dma16) << 4; + if (!my_sbdsp_command(base, SB_JAZZ16_SET_DMAINTR)) + return 0; + + if (!my_sbdsp_command(base, dma_config)) + return 0; + jazz16_sb_dma1 = dma8; + jazz16_sb_dma2 = dma16; + + if (!my_sbdsp_command(base, irq_config)) + return 0; + jazz16_sb_irq = sbirq; + jazz16_mpu401_irq = mpuirq; + return 1; +} + +static int jazz16pnp_get_resources(struct pnp_dev *dev, struct pnp_resource_table *res) +{ + pnp_init_resource_table(res); + if (dev == jazz16_gameport_dev) { + res->port_resource[0].start = res->port_resource[0].end = SB_JAZZ16_CONFIG_PORT; + res->port_resource[0].flags = IORESOURCE_IO; + } + else if (dev == jazz16_sb_dev) { + if (!jazz16_sb_port) + return 0; + res->port_resource[0].start = jazz16_sb_port; + res->port_resource[0].end = jazz16_sb_port + 0x10 - 1; + res->port_resource[0].flags = IORESOURCE_IO; + + res->irq_resource[0].start = res->irq_resource[0].end = jazz16_sb_irq; + res->irq_resource[0].flags = IORESOURCE_IRQ; + + res->dma_resource[0].start = res->dma_resource[0].end = jazz16_sb_dma1; + res->dma_resource[0].flags = IORESOURCE_DMA; + res->dma_resource[1].start = res->dma_resource[1].end = jazz16_sb_dma2; + res->dma_resource[1].flags = IORESOURCE_DMA; + + res->port_resource[1].start = jazz16_mpu401_port; + res->port_resource[1].end = jazz16_mpu401_port + 2 - 1; + res->port_resource[1].flags = IORESOURCE_IO; + + if (!jazz16_mpu401_irq) + return 0; + res->irq_resource[1].start = res->irq_resource[0].end = jazz16_mpu401_irq; + res->irq_resource[1].flags = IORESOURCE_IRQ; + } + else + return -EINVAL; + return 0; +} + +static int jazz16pnp_set_resources(struct pnp_dev *dev, struct pnp_resource_table *res) +{ + unsigned int sbirq, mpuirq, dma1, dma2; + + if (dev == jazz16_gameport_dev) + return -ENOSYS; + + else if (dev == jazz16_sb_dev) { + if ((res->port_resource[0].flags & (IORESOURCE_IO | IORESOURCE_UNSET)) == IORESOURCE_IO) + jazz16_port_setup(res->port_resource[0].start, jazz16_mpu401_port); + if ((res->port_resource[1].flags & (IORESOURCE_IO | IORESOURCE_UNSET)) == IORESOURCE_IO) + jazz16_port_setup(jazz16_sb_port, res->port_resource[1].start); + if ((res->dma_resource[0].flags & (IORESOURCE_DMA | IORESOURCE_UNSET)) == IORESOURCE_DMA) + dma1 = res->dma_resource[0].start; + else + dma1 = jazz16_sb_dma1; + if ((res->dma_resource[1].flags & (IORESOURCE_DMA | IORESOURCE_UNSET)) == IORESOURCE_DMA) + dma2 = res->dma_resource[1].start; + else + dma2 = jazz16_sb_dma2; + if ((res->irq_resource[0].flags & (IORESOURCE_IRQ | IORESOURCE_UNSET)) == IORESOURCE_IRQ) + sbirq = res->irq_resource[0].start; + else + sbirq = jazz16_sb_irq; + if ((res->irq_resource[1].flags & (IORESOURCE_IRQ | IORESOURCE_UNSET)) == IORESOURCE_IRQ) + mpuirq = res->irq_resource[1].start; + else + mpuirq = jazz16_mpu401_irq; + if (jazz16_sb_port) { + /* Block the OPL3 alias to avoid conflicts. */ + jazz16pnp_fm_res = request_region(0x388, 4, "SoundBlaster FM"); + jazz16_irq_dma_setup(jazz16_sb_port, sbirq, mpuirq, dma1, dma2); + } else + return -EBUSY; + dev->active = true; + return 0; + } + else + return -EINVAL; +} + +static int jazz16pnp_disable_resources(struct pnp_dev *dev) +{ + if (dev == jazz16_gameport_dev) { + return -ENOSYS; + } + else if (dev == jazz16_sb_dev) { + jazz16_irq_dma_setup(jazz16_sb_port, 0, 0, 0, 0); + jazz16_port_setup(0, 0); + if (jazz16pnp_fm_res) { + release_resource(jazz16pnp_fm_res); + jazz16pnp_fm_res = NULL; + } + dev->active = false; + return 0; + } + else { + return -EINVAL; + } +} + +static struct pnp_protocol jazz16pnp_protocol = { + .name = "Jazz16 PnP", + .get = jazz16pnp_get_resources, + .set = jazz16pnp_set_resources, + .disable = jazz16pnp_disable_resources, +}; + +static uint __init jazz16_check(unsigned int base) +{ + if (my_sbdsp_reset(base)) + return 0; + if (!(my_sbdsp_command(base, SB_DSP_GET_JAZZ_VERSION))) + return 0; + return SB_VERSION_IS_JAZZ16(my_sbdsp_get_byte(base)); +} + +static int __init jazz16pnp_init(void) +{ + int err = 0; + unsigned int base, mpubase; + + if (jazz16pnp_disable == 1) { + printk(PFX_INFO "Jazz16 Plug & Play support disabled.\n"); + return 0; + } + if (!request_region(SB_JAZZ16_CONFIG_PORT, 1, "Jazz16 PnP probe")) + return -EBUSY; + + for (mpubase = 0x330; mpubase >= 0x300; mpubase -= 0x10) + if (request_region(mpubase, 2, "Jazz16 PnP probe")) + break; + if (mpubase < 0x300) + goto out_no_mpubase; + + if (jazz16pnp_verbose) + printk(PFX_DEBUG "Scanning for Jazz16 cards...\n"); + + for (base = 0x260; base >= 0x220; base -= 0x20) { + if (!request_region(base, 16, "Jazz16 PnP probe")) + continue; + if (jazz16pnp_verbose) + printk(PFX_DEBUG "Probing at 0x%x.\n", base); + jazz16_port_setup(base, mpubase); + if (jazz16_check(base)) + jazz16pnp_count++; + jazz16_irq_dma_setup(base, 0, 0, 0, 0); + jazz16_port_setup(0, 0); + release_region(base, 16); + if (jazz16pnp_count) + break; + } + if (base < 0x220) + goto out_no_sbbase; + if (!jazz16pnp_count) { + err = -ENODEV; + goto out_no_devices; + } + if (pnp_register_protocol(&jazz16pnp_protocol) < 0) + err = -ENOMEM; + else { + if (jazz16pnp_add_gameport()) + printk(PFX_WARN "Failed to add game port.\n"); + if (jazz16pnp_add_sb()) + printk(PFX_WARN "Failed to add SB port.\n"); + err = 0; + } +out_no_devices: + printk(PFX_INFO "%u Jazz16 card%s detected.\n", jazz16pnp_count, + jazz16pnp_count == 1 ? "" : "s"); +out_no_sbbase: + release_region(mpubase, 2); +out_no_mpubase: + release_region(SB_JAZZ16_CONFIG_PORT, 1); + return err; +} +module_init(jazz16pnp_init); + +static void __exit jazz16pnp_exit(void) +{ + pnp_unregister_protocol(&jazz16pnp_protocol); +} +module_exit(jazz16pnp_exit); --- linux-2.6.20/drivers/pnp/Kconfig-orig 2007-02-19 23:23:26.000000000 +0100 +++ linux-2.6.20/drivers/pnp/Kconfig 2007-03-25 21:44:48.000000000 +0200 @@ -39,5 +39,17 @@ source "drivers/pnp/pnpacpi/Kconfig" source "drivers/pnp/pnpacpi/Kconfig" +# +# Jazz16 Plug and Play configuration +# +config JAZZ16PNP + bool "Jazz16 sound card Plug and Play support" + depends on PNP && ISA + help + Say Y here if you would like support for automatic configuration + of I/O ports, DMA channels and IRQ lines of Jazz16 sound cards. + + Say Y if you have such a card. Otherwise, say N. + endmenu -- Rask Ingemann Lambertsen

1 0

[alsa-devel] Can't subscribe to the list
by Ed Wright 26 Mar '07

26 Mar '07

Hi, I've tried 3 times over the last week or so to subscribe to this list with no success. I can't believe I'm doing anything wrong (how hard is it?), but I don't see any posts (in the archives) indicating any problems subscribing, so I'm at a loss. Any suggestions? Or can you subscribe me manually? Thanks, Ed

2 2

[alsa-devel] Support for Compal EL81 (Intel HDA/Realtek alc883)
by Carlo E. Prelz 26 Mar '07

26 Mar '07

Good evening. I am the lucky owner of a nice Compal EL80 laptop, sporting an Intel HDA audio controller with realtek alc883. Here is what I get with lspci -vvv: 00:1b.0 Audio device [0403]: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller [8086:27d8] (rev 02) Subsystem: COMPAL Electronics Inc Unknown device [14c0:0017] Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 22 Region 0: Memory at d2500000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Address: 0000000000000000 Data: 0000 Capabilities: [70] Express Unknown type IRQ 0 Device: Supported: MaxPayload 128 bytes, PhantFunc 0, ExtTag- Device: Latency L0s <64ns, L1 <1us Device: Errors: Correctable- Non-Fatal- Fatal- Unsupported- Device: RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ Device: MaxPayload 128 bytes, MaxReadReq 128 bytes Link: Supported Speed unknown, Width x0, ASPM unknown, Port 0 Link: Latency L0s <64ns, L1 <1us Link: ASPM Disabled CommClk- ExtSynch- Link: Speed unknown, Width x0 Capabilities: [100] Virtual Channel Capabilities: [130] Unknown (5) I had very elementary operativity, but missed much, including for example microphone boost. I applied the following patch: --8<-----8<----8<----8<----8<----8<----8<----8<----8<----8<---8<-- --- /usr/src/alsa/alsa-kernel/pci/hda/patch_realtek.c 2007-02-15 20:00:04.000000000 +0100 +++ ./patch_realtek.c 2007-03-15 08:36:30.000000000 +0100 @@ -141,6 +141,7 @@ ALC883_ACER, ALC883_MEDION, ALC883_LAPTOP_EAPD, + ALC883_LAPTOP_COMPAL, ALC883_AUTO, ALC883_MODEL_LAST, }; @@ -5582,6 +5583,7 @@ [ALC883_ACER] = "acer", [ALC883_MEDION] = "medion", [ALC883_LAPTOP_EAPD] = "laptop-eapd", + [ALC883_LAPTOP_COMPAL] = "laptop-compal", [ALC883_AUTO] = "auto", }; @@ -5607,6 +5609,7 @@ SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION), SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC883_LAPTOP_EAPD), SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC883_3ST_6ch), + SND_PCI_QUIRK(0x14c0, 0x0017, "Compal EL80", ALC883_LAPTOP_COMPAL), {} }; @@ -5747,6 +5750,76 @@ .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, }, +#if 1 + [ALC883_LAPTOP_COMPAL] = { + .mixers = { alc883_base_mixer, + alc883_chmode_mixer }, + .init_verbs = { alc883_init_verbs, alc882_eapd_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), + .adc_nids = alc883_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), + .channel_mode = alc883_3ST_2ch_modes, + .input_mux = &alc883_capture_source, + }, +#else + [ALC883_LAPTOP_COMPAL] = { + .mixers = { alc883_fivestack_mixer, --- /usr/src/alsa/alsa-kernel/pci/hda/patch_realtek.c 2007-02-15 20:00:04.000000000 +0100 +++ ./patch_realtek.c 2007-03-15 08:36:30.000000000 +0100 @@ -141,6 +141,7 @@ ALC883_ACER, ALC883_MEDION, ALC883_LAPTOP_EAPD, + ALC883_LAPTOP_COMPAL, ALC883_AUTO, ALC883_MODEL_LAST, }; @@ -5582,6 +5583,7 @@ [ALC883_ACER] = "acer", [ALC883_MEDION] = "medion", [ALC883_LAPTOP_EAPD] = "laptop-eapd", + [ALC883_LAPTOP_COMPAL] = "laptop-compal", [ALC883_AUTO] = "auto", }; @@ -5607,6 +5609,7 @@ SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION), SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC883_LAPTOP_EAPD), SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC883_3ST_6ch), + SND_PCI_QUIRK(0x14c0, 0x0017, "Compal EL80", ALC883_LAPTOP_COMPAL), {} }; @@ -5747,6 +5750,76 @@ .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, }, +#if 1 + [ALC883_LAPTOP_COMPAL] = { + .mixers = { alc883_base_mixer, + alc883_chmode_mixer }, + .init_verbs = { alc883_init_verbs, alc882_eapd_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), + .adc_nids = alc883_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), + .channel_mode = alc883_3ST_2ch_modes, + .input_mux = &alc883_capture_source, + }, +#else + [ALC883_LAPTOP_COMPAL] = { + .mixers = { alc883_fivestack_mixer, + alc883_chmode_mixer }, + .init_verbs = { alc883_init_verbs, + alc883_medion_eapd_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), + .adc_nids = alc883_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes), + .channel_mode = alc883_sixstack_modes, + .input_mux = &alc883_capture_source, + }, + [ALC883_LAPTOP_COMPAL] = { + .mixers = { alc883_3ST_2ch_mixer }, + .init_verbs = { alc883_init_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .dig_out_nid = ALC883_DIGOUT_NID, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), + .adc_nids = alc883_adc_nids, + .dig_in_nid = ALC883_DIGIN_NID, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), + .channel_mode = alc883_3ST_2ch_modes, + .input_mux = &alc883_capture_source, + }, + [ALC883_LAPTOP_COMPAL] = { + .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer }, + .init_verbs = { alc883_init_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .dig_out_nid = ALC883_DIGOUT_NID, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), + .adc_nids = alc883_adc_nids, + .dig_in_nid = ALC883_DIGIN_NID, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes), + .channel_mode = alc883_3ST_6ch_modes, + .need_dac_fix = 1, + .input_mux = &alc883_capture_source, + }, + [ALC883_LAPTOP_COMPAL] = { + .mixers = { alc883_tagra_mixer, alc883_chmode_mixer }, + .init_verbs = { alc883_init_verbs, alc883_tagra_verbs}, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .dig_out_nid = ALC883_DIGOUT_NID, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), + .adc_nids = alc883_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes), + .channel_mode = alc883_3ST_6ch_modes, + .need_dac_fix = 1, + .input_mux = &alc883_capture_source, + .unsol_event = alc883_tagra_unsol_event, + .init_hook = alc883_tagra_automute, + }, +#endif }; --8<-----8<----8<----8<----8<----8<----8<----8<----8<----8<---8<-- As you can see, I did various tests with possible 'alc_config_preset' structures. I could only copy from other existing examples, since I have no idea how you would obtain the correct values to insert there. With the first example, the only one that is commented in (copied from LAPTOP_EAPD), I could obtain a working microphone boost. The thing is far from perfect, though. Suffice to say that I find the main volume in the cursor labeled 'Surround' in alsamixer. Is there any instruction about how to obtain the right information to configure this laptop of mine? Can you help? In this case, what other information may I provide? Thanks in advance Carlo - Den Haag, NL -- * Se la Strada e la sua Virtu' non fossero state messe da parte, * K * Carlo E. Prelz - fluido(a)fluido.as che bisogno ci sarebbe * di parlare tanto di amore e di rettitudine? (Chuang-Tzu)

2 2

[alsa-devel] ftp.alsa-project.org
by Rene Herman 26 Mar '07

26 Mar '07

Hi Jaroslav. When you currently want to download a complete new release from ftp.alsa-project.org you have to scrounge around all the directories manually. Would it be easy to automate to have a /pub/alsa/<version>/ directory containing links to all the relevant packages? Ie: /pub/alsa/1.0.13/alsa-driver-1.0.13.tar.bz2 -> ../../driver/alsa-driver-1.13.tar.bz2 and same: alsa-lib-1.0.13.tar.bz2 -> ../../lib/alsa-lib-1.13.tar.bz2 alsa-plugins-1.0.13.tar.bz2 -> ../../plugins/alsa-plugins-1.13.tar.bz2 alsa-utils-1.0.13.tar.bz2 -> ../../utils/alsa-utils-1.13.tar.bz2 alsa-tools-1.0.13.tar.bz2 -> ../../tools/alsa-tools-1.13.tar.bz2 alsa-firmware-1.0.13.tar.bz2 -> .../firmware/alsa-firmware-1.13.tar.bz2 alsa-oss-lib-1.0.12.tar.bz2 -> ../../oss-lib/alsa-oss-lib-1.12.tar.bz2 with probably links /pub/alsa/stable -> 1.0.13 and /pub/alsa/devel -> 1.14rc3 and stuff like that. (by the way, as far as I'm concerned, the /pub/alsa locations wouldn't need to be links but could just as well be the main locations but if you want to keep the current layout in place...) Would make it a bit nicer to download a new release... Rene.

1 0

[alsa-devel] Patch to p5vdc-x motherboard's sound
by sergiodaniel＠inf.ufg.br 26 Mar '07

26 Mar '07

This is a description to correct the problem of p5vdc-x motherboard's. In file hda_codec.c the line 1727 is: if (codec->bus->modelname && models) { the variable codec->bus->modelname not is reconized for asus pc5vdc-x motherboards. Your ship is "Analog Devices AD1986A" i correct this add the lines in the funcion static int patch_ad1986a(struct hda_codec *codec) . . . default: spec->num_mixers = 2; spec->mixers[1] = ad1986a_3st_mixers; spec->num_init_verbs = 3; spec->init_verbs[1] = ad1986a_3st_init_verbs; spec->init_verbs[2] = ad1986a_ch2_init; spec->channel_mode = ad1986a_modes; spec->num_channel_mode = ARRAY_SIZE(ad1986a_modes); spec->need_dac_fix = 1; spec->multiout.max_channels = 2; spec->multiout.num_dacs = 1; break; } The complete code is in www.inf.ufg.br/~sergiodaniel/alsa-driver-1.0.14_PATCH_rc3-p5vdc-x.tar.gz thank's.. Sérgio Daniel ---------------------------------------------------------------- This message was sent using IMP, the Internet Messaging Program.

2 1

Re: [alsa-devel] [PATCH] azt3328.c: small cleanup patch
by Takashi Iwai 26 Mar '07

26 Mar '07

At Sun, 25 Mar 2007 21:06:56 +0200, Andreas Mohr wrote: > > Hi, > > - change "PCM" mixer control (pre/post 3D) to "PCM Output Route" > - improve snd_azf3328_debug_show_ports > - less aggressive module init message > - document Bass/Treble non-bug (prompted by user report - Thank You!!) > - add some items to card description > - add some I/O register documentation > - enhance copyright > > Patch done and hardware-tested on 2.6.21-rc4-mm1, hopefully applies cleanly > to current ALSA. > > Signed-off-by: Andreas Mohr <andi(a)lisas.de> Thanks, applied to HG tree now. (BTW, the address of alsa-devel ML was changed to alsa-devel(a)alsa-project.org) Takashi

1 0

[alsa-devel] [PATCH -mm] sound/pci/au88x0/au88x0_a3d.c remove unused code
by Michal Piotrowski 26 Mar '07

26 Mar '07

This patch is against the latest -mm snapshot. Regards, Michal -- Michal K. K. Piotrowski LTG - Linux Testers Group (PL) (http://www.stardust.webpages.pl/ltg/) LTG - Linux Testers Group (EN) (http://www.stardust.webpages.pl/linux_testers_group_en/) Signed-off-by: Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> --- linux-work5/sound/pci/au88x0/au88x0_a3d.c 2007-02-04 19:44:54.000000000 +0100 +++ linux-work3/sound/pci/au88x0/au88x0_a3d.c 2007-03-25 16:23:46.000000000 +0200 @@ -44,15 +44,6 @@ a3dsrc_SetTimeConsts(a3dsrc_t * a, short a3d_addrA(a->slice, a->source, A3D_A_CoeffTrackTC), CTrack); } -#if 0 -static void -a3dsrc_GetTimeConsts(a3dsrc_t * a, short *HrtfTrack, short *ItdTrack, - short *GTrack, short *CTrack) -{ - // stub! -} - -#endif /* Atmospheric absorbtion. */ static void @@ -95,41 +86,6 @@ a3dsrc_SetAtmosState(a3dsrc_t * a, short hwwrite(vortex->mmio, a3d_addrA(a->slice, a->source, A3D_A_y2), y2); } -#if 0 -static void -a3dsrc_GetAtmosTarget(a3dsrc_t * a, short *aa, short *b, short *c, - short *d, short *e) -{ -} -static void -a3dsrc_GetAtmosCurrent(a3dsrc_t * a, short *bb01, short *ab01, short *b2, - short *aa12, short *ba12) -{ - vortex_t *vortex = (vortex_t *) (a->vortex); - *aa12 = - hwread(vortex->mmio, - a3d_addrA(a->slice, a->source, A3D_A_A12Current)); - *ba12 = - hwread(vortex->mmio, - a3d_addrB(a->slice, a->source, A3D_B_A12Current)); - *ab01 = - hwread(vortex->mmio, - a3d_addrA(a->slice, a->source, A3D_A_B01Current)); - *bb01 = - hwread(vortex->mmio, - a3d_addrB(a->slice, a->source, A3D_B_B01Current)); - *b2 = - hwread(vortex->mmio, - a3d_addrA(a->slice, a->source, A3D_A_B2Current)); -} - -static void -a3dsrc_GetAtmosState(a3dsrc_t * a, short *x1, short *x2, short *y1, short *y2) -{ - -} - -#endif /* HRTF */ static void @@ -180,71 +136,6 @@ static void a3dsrc_SetHrtfOutput(a3dsrc_ a3d_addrA(a->slice, a->source, A3D_A_HrtfOutR), right); } -#if 0 -static void a3dsrc_GetHrtfTarget(a3dsrc_t * a, a3d_Hrtf_t aa, a3d_Hrtf_t b) -{ - vortex_t *vortex = (vortex_t *) (a->vortex); - int i; - - for (i = 0; i < HRTF_SZ; i++) - aa[i] = - hwread(vortex->mmio, - a3d_addrA(a->slice, a->source, - A3D_A_HrtfTarget + (i << 2))); - for (i = 0; i < HRTF_SZ; i++) - b[i] = - hwread(vortex->mmio, - a3d_addrB(a->slice, a->source, - A3D_B_HrtfTarget + (i << 2))); -} - -static void a3dsrc_GetHrtfCurrent(a3dsrc_t * a, a3d_Hrtf_t aa, a3d_Hrtf_t b) -{ - vortex_t *vortex = (vortex_t *) (a->vortex); - int i; - - for (i = 0; i < HRTF_SZ; i++) - aa[i] = - hwread(vortex->mmio, - a3d_addrA(a->slice, a->source, - A3D_A_HrtfCurrent + (i << 2))); - for (i = 0; i < HRTF_SZ; i++) - b[i] = - hwread(vortex->mmio, - a3d_addrB(a->slice, a->source, - A3D_B_HrtfCurrent + (i << 2))); -} - -static void a3dsrc_GetHrtfState(a3dsrc_t * a, a3d_Hrtf_t aa, a3d_Hrtf_t b) -{ - vortex_t *vortex = (vortex_t *) (a->vortex); - int i; - // FIXME: verify this! - for (i = 0; i < HRTF_SZ; i++) - aa[i] = - hwread(vortex->mmio, - a3d_addrA(a->slice, a->source, - A3D_A_HrtfDelayLine + (i << 2))); - for (i = 0; i < HRTF_SZ; i++) - b[i] = - hwread(vortex->mmio, - a3d_addrB(a->slice, a->source, - A3D_B_HrtfDelayLine + (i << 2))); -} - -static void a3dsrc_GetHrtfOutput(a3dsrc_t * a, short *left, short *right) -{ - vortex_t *vortex = (vortex_t *) (a->vortex); - *left = - hwread(vortex->mmio, - a3d_addrA(a->slice, a->source, A3D_A_HrtfOutL)); - *right = - hwread(vortex->mmio, - a3d_addrA(a->slice, a->source, A3D_A_HrtfOutR)); -} - -#endif - /* Interaural Time Difference. * "The other main clue that humans use to locate sounds, is called * Interaural Time Difference (ITD). The differences in distance from @@ -297,43 +188,6 @@ static void a3dsrc_SetItdDline(a3dsrc_t A3D_A_ITDDelayLine) + (i << 2), dline[i]); } -#if 0 -static void a3dsrc_GetItdTarget(a3dsrc_t * a, short *litd, short *ritd) -{ - vortex_t *vortex = (vortex_t *) (a->vortex); - *ritd = - hwread(vortex->mmio, - a3d_addrA(a->slice, a->source, A3D_A_ITDTarget)); - *litd = - hwread(vortex->mmio, - a3d_addrB(a->slice, a->source, A3D_B_ITDTarget)); -} - -static void a3dsrc_GetItdCurrent(a3dsrc_t * a, short *litd, short *ritd) -{ - vortex_t *vortex = (vortex_t *) (a->vortex); - - *ritd = - hwread(vortex->mmio, - a3d_addrA(a->slice, a->source, A3D_A_ITDCurrent)); - *litd = - hwread(vortex->mmio, - a3d_addrB(a->slice, a->source, A3D_B_ITDCurrent)); -} - -static void a3dsrc_GetItdDline(a3dsrc_t * a, a3d_ItdDline_t dline) -{ - vortex_t *vortex = (vortex_t *) (a->vortex); - int i; - - for (i = 0; i < DLINE_SZ; i++) - dline[i] = - hwread(vortex->mmio, - a3d_addrA(a->slice, a->source, - A3D_A_ITDDelayLine + (i << 2))); -} - -#endif /* This is may be used for ILD Interaural Level Difference. */ static void a3dsrc_SetGainTarget(a3dsrc_t * a, short left, short right) @@ -352,37 +206,6 @@ static void a3dsrc_SetGainCurrent(a3dsrc (right << 0x10) | left); } -#if 0 -static void a3dsrc_GetGainTarget(a3dsrc_t * a, short *left, short *right) -{ - vortex_t *vortex = (vortex_t *) (a->vortex); - *right = - hwread(vortex->mmio, - a3d_addrA(a->slice, a->source, A3D_A_GainTarget)); - *left = - hwread(vortex->mmio, - a3d_addrB(a->slice, a->source, A3D_B_GainTarget)); -} - -static void a3dsrc_GetGainCurrent(a3dsrc_t * a, short *left, short *right) -{ - vortex_t *vortex = (vortex_t *) (a->vortex); - *right = - hwread(vortex->mmio, - a3d_addrA(a->slice, a->source, A3D_A_GainCurrent)); - *left = - hwread(vortex->mmio, - a3d_addrB(a->slice, a->source, A3D_B_GainCurrent)); -} - -/* CA3dIO this func seems to be inlined all over this place. */ -static void CA3dIO_WriteReg(a3dsrc_t * a, unsigned long addr, short aa, short b) -{ - vortex_t *vortex = (vortex_t *) (a->vortex); - hwwrite(vortex->mmio, addr, (aa << 0x10) | b); -} - -#endif /* Generic A3D stuff */ static void a3dsrc_SetA3DSampleRate(a3dsrc_t * a, int sr) @@ -422,28 +245,6 @@ static void a3dsrc_SetA3DPointerReg(a3ds hwwrite(vortex->mmio, A3D_SLICE_Pointers + ((a->slice) << 0xd), ptr); } -#if 0 -static void a3dsrc_GetA3DSampleRate(a3dsrc_t * a, int *sr) -{ - vortex_t *vortex = (vortex_t *) (a->vortex); - *sr = ((hwread(vortex->mmio, A3D_SLICE_Control + (a->slice << 0xd)) - >> 3) & 0x1f); - //*sr = ((hwread(vortex->mmio, 0x19C38 + (this08<<0xd))>>3)&0x1f); -} - -static void a3dsrc_GetA3DControlReg(a3dsrc_t * a, unsigned long *ctrl) -{ - vortex_t *vortex = (vortex_t *) (a->vortex); - *ctrl = hwread(vortex->mmio, A3D_SLICE_Control + ((a->slice) << 0xd)); -} - -static void a3dsrc_GetA3DPointerReg(a3dsrc_t * a, unsigned long *ptr) -{ - vortex_t *vortex = (vortex_t *) (a->vortex); - *ptr = hwread(vortex->mmio, A3D_SLICE_Pointers + ((a->slice) << 0xd)); -} - -#endif static void a3dsrc_ZeroSliceIO(a3dsrc_t * a) { vortex_t *vortex = (vortex_t *) (a->vortex); @@ -528,25 +329,6 @@ static void a3dsrc_ProgramPipe(a3dsrc_t //a3dsrc_SetHrtfTarget(a, A3dHrirSatTest, A3dHrirSatTest); } -/* VDB = Vortex audio Dataflow Bus */ -#if 0 -static void a3dsrc_ClearVDBData(a3dsrc_t * a, unsigned long aa) -{ - vortex_t *vortex = (vortex_t *) (a->vortex); - - // ((aa >> 2) << 8) - (aa >> 2) - hwwrite(vortex->mmio, - a3d_addrS(a->slice, A3D_SLICE_VDBDest) + (a->source << 2), 0); - hwwrite(vortex->mmio, - a3d_addrS(a->slice, - A3D_SLICE_VDBDest + 4) + (a->source << 2), 0); - /* - hwwrite(vortex->mmio, 0x19c00 + (((aa>>2)*255*4)+aa)*8, 0); - hwwrite(vortex->mmio, 0x19c04 + (((aa>>2)*255*4)+aa)*8, 0); - */ -} -#endif - /* A3D HwSource stuff. */ static void vortex_A3dSourceHw_Initialize(vortex_t * v, int source, int slice) @@ -622,7 +404,6 @@ static void vortex_Vort3D_connect(vortex return; #endif -#if 1 /* Alloc Xtalk mixin resources */ v->mixxtlk[0] = vortex_adb_checkinout(v, v->fixed_res, en, VORTEX_RESOURCE_MIXIN); @@ -638,7 +419,6 @@ static void vortex_Vort3D_connect(vortex ("vortex: vortex_Vort3D: ERROR: not enough free mixer resources.\n"); return; } -#endif /* Connect A3D -> XTALK */ for (i = 0; i < 4; i++) { @@ -646,10 +426,7 @@ static void vortex_Vort3D_connect(vortex vortex_route(v, en, 0x11, ADB_A3DOUT(i * 2), ADB_XTALKIN(i)); vortex_route(v, en, 0x11, ADB_A3DOUT(i * 2) + 1, ADB_XTALKIN(5 + i)); } -#if 0 - vortex_route(v, en, 0x11, ADB_XTALKOUT(0), ADB_EQIN(2)); - vortex_route(v, en, 0x11, ADB_XTALKOUT(1), ADB_EQIN(3)); -#else + /* Connect XTalk -> mixer */ vortex_route(v, en, 0x11, ADB_XTALKOUT(0), ADB_MIXIN(v->mixxtlk[0])); vortex_route(v, en, 0x11, ADB_XTALKOUT(1), ADB_MIXIN(v->mixxtlk[1])); @@ -671,7 +448,6 @@ static void vortex_Vort3D_connect(vortex v->mixxtlk[1], en ? MIX_DEFIGAIN : VOL_MIN); } -#endif } /* Initialize one single A3D source. */

4 5

[alsa-devel] Basic DMA question
by Sven Johnsson 26 Mar '07

26 Mar '07

Hello all, I'm new to writing ALSA drivers, and I could use some help on clarifying how DMA is used in ALSA. Regards, Sven

1 0

[alsa-devel] hopeless about snd-hda-intel driver.
by Ustun ERGENOGLU 25 Mar '07

25 Mar '07

Hello everybody, for more than one month I have been crawling through the net to find a cure for my endless torment about ATI SB450 Realtek ALC861 soundcard. I have no problem with playback, but when it comes recording there is nothing wrong in the beginning but after sometime recording stops and I see a message about time out in dmesg output. I have made a bug report in the ALSA bug tracking system but got no response :(. It's here: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=2970 Can anybody, please, tell me where to begin to solve this problem. I can even start hacking ALSA code with my next to nothing knowledge about kernel driver programming as I have more than average c programming experience. This is more than enough motivation to start some coding, I suppose. I would greatly appreciate any recommendation or help. Thanks Ustun

1 1

[alsa-devel] [PATCH -mm] sound/pci/trident/trident_synth.c remove unused code
by Michal Piotrowski 25 Mar '07

25 Mar '07

This patch is against the latest -mm snapshot. Regards, Michal -- Michal K. K. Piotrowski LTG - Linux Testers Group (PL) (http://www.stardust.webpages.pl/ltg/) LTG - Linux Testers Group (EN) (http://www.stardust.webpages.pl/linux_testers_group_en/) Signed-off-by: Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> --- linux-work5/sound/pci/trident/trident_synth.c 2007-02-04 19:44:54.000000000 +0100 +++ linux-work3/sound/pci/trident/trident_synth.c 2007-03-25 00:41:40.000000000 +0100 @@ -772,19 +772,7 @@ static int snd_trident_synth_use(void *p voice->Vol = 0x3ff; voice->EC = 0x0fff; } -#if 0 - for (idx = 0; idx < info->midi_voices; idx++) { - port->midi_has_voices = 1; - voice = snd_trident_alloc_voice(trident, SNDRV_TRIDENT_VOICE_TYPE_MIDI, info->sender.client, info->sender.port); - if (voice == NULL) { - snd_trident_synth_free_voices(trident, info->sender.client, info->sender.port); - spin_unlock_irqrestore(&trident->reg_lock, flags); - return -EBUSY; - } - voice->Vol = 0x3ff; - voice->EC = 0x0fff; - } -#endif + spin_unlock_irqrestore(&trident->reg_lock, flags); return 0; }

1 0

[alsa-devel] [PATCH -mm] sound/pci/trident/trident_main.c remove unused code
by Michal Piotrowski 25 Mar '07

25 Mar '07

This patch is against the latest -mm snapshot. Regards, Michal -- Michal K. K. Piotrowski LTG - Linux Testers Group (PL) (http://www.stardust.webpages.pl/ltg/) LTG - Linux Testers Group (EN) (http://www.stardust.webpages.pl/linux_testers_group_en/) Signed-off-by: Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> --- linux-work5/sound/pci/trident/trident_main.c 2007-03-25 00:31:00.000000000 +0100 +++ linux-work3/sound/pci/trident/trident_main.c 2007-03-25 00:51:49.000000000 +0100 @@ -63,49 +63,6 @@ static int snd_trident_free(struct snd_t * common I/O routines */ - -#if 0 -static void snd_trident_print_voice_regs(struct snd_trident *trident, int voice) -{ - unsigned int val, tmp; - - printk("Trident voice %i:\n", voice); - outb(voice, TRID_REG(trident, T4D_LFO_GC_CIR)); - val = inl(TRID_REG(trident, CH_LBA)); - printk("LBA: 0x%x\n", val); - val = inl(TRID_REG(trident, CH_GVSEL_PAN_VOL_CTRL_EC)); - printk("GVSel: %i\n", val >> 31); - printk("Pan: 0x%x\n", (val >> 24) & 0x7f); - printk("Vol: 0x%x\n", (val >> 16) & 0xff); - printk("CTRL: 0x%x\n", (val >> 12) & 0x0f); - printk("EC: 0x%x\n", val & 0x0fff); - if (trident->device != TRIDENT_DEVICE_ID_NX) { - val = inl(TRID_REG(trident, CH_DX_CSO_ALPHA_FMS)); - printk("CSO: 0x%x\n", val >> 16); - printk("Alpha: 0x%x\n", (val >> 4) & 0x0fff); - printk("FMS: 0x%x\n", val & 0x0f); - val = inl(TRID_REG(trident, CH_DX_ESO_DELTA)); - printk("ESO: 0x%x\n", val >> 16); - printk("Delta: 0x%x\n", val & 0xffff); - val = inl(TRID_REG(trident, CH_DX_FMC_RVOL_CVOL)); - } else { // TRIDENT_DEVICE_ID_NX - val = inl(TRID_REG(trident, CH_NX_DELTA_CSO)); - tmp = (val >> 24) & 0xff; - printk("CSO: 0x%x\n", val & 0x00ffffff); - val = inl(TRID_REG(trident, CH_NX_DELTA_ESO)); - tmp |= (val >> 16) & 0xff00; - printk("Delta: 0x%x\n", tmp); - printk("ESO: 0x%x\n", val & 0x00ffffff); - val = inl(TRID_REG(trident, CH_NX_ALPHA_FMS_FMC_RVOL_CVOL)); - printk("Alpha: 0x%x\n", val >> 20); - printk("FMS: 0x%x\n", (val >> 16) & 0x0f); - } - printk("FMC: 0x%x\n", (val >> 14) & 3); - printk("RVol: 0x%x\n", (val >> 7) & 0x7f); - printk("CVol: 0x%x\n", val & 0x7f); -} -#endif - /*--------------------------------------------------------------------------- unsigned short snd_trident_codec_read(struct snd_ac97 *ac97, unsigned short reg) @@ -495,15 +452,6 @@ void snd_trident_write_voice_regs(struct outl(regs[2], TRID_REG(trident, CH_START + 8)); outl(regs[3], TRID_REG(trident, CH_START + 12)); outl(regs[4], TRID_REG(trident, CH_START + 16)); - -#if 0 - printk("written %i channel:\n", voice->number); - printk(" regs[0] = 0x%x/0x%x\n", regs[0], inl(TRID_REG(trident, CH_START + 0))); - printk(" regs[1] = 0x%x/0x%x\n", regs[1], inl(TRID_REG(trident, CH_START + 4))); - printk(" regs[2] = 0x%x/0x%x\n", regs[2], inl(TRID_REG(trident, CH_START + 8))); - printk(" regs[3] = 0x%x/0x%x\n", regs[3], inl(TRID_REG(trident, CH_START + 12))); - printk(" regs[4] = 0x%x/0x%x\n", regs[4], inl(TRID_REG(trident, CH_START + 16))); -#endif } EXPORT_SYMBOL(snd_trident_write_voice_regs); @@ -972,12 +920,7 @@ static int snd_trident_playback_prepare( voice->CVol = mix->cvol; voice->Pan = mix->pan; voice->Attribute = 0; -#if 0 - voice->Attribute = (1<<(30-16))|(2<<(26-16))| - (0<<(24-16))|(0x1f<<(19-16)); -#else voice->Attribute = 0; -#endif snd_trident_write_voice_regs(trident, voice); @@ -996,12 +939,7 @@ static int snd_trident_playback_prepare( evoice->Vol = 0x3ff; /* mute */ evoice->RVol = evoice->CVol = 0x7f; /* mute */ evoice->Pan = 0x7f; /* mute */ -#if 0 - evoice->Attribute = (1<<(30-16))|(2<<(26-16))| - (0<<(24-16))|(0x1f<<(19-16)); -#else evoice->Attribute = 0; -#endif snd_trident_write_voice_regs(trident, evoice); evoice->isync2 = 1; evoice->isync_mark = runtime->period_size; @@ -3804,14 +3742,7 @@ static irqreturn_t snd_trident_interrupt snd_trident_write_eso_reg(trident, voice, voice->ESO); snd_trident_start_voice(trident, voice->number); } -#if 0 - if (voice->extra) { - /* update CSO for extra voice to preserve sync */ - snd_trident_stop_voice(trident, voice->extra->number); - snd_trident_write_cso_reg(trident, voice->extra, 0); - snd_trident_start_voice(trident, voice->extra->number); - } -#endif + spin_unlock(&trident->reg_lock); snd_pcm_period_elapsed(voice->substream); spin_lock(&trident->reg_lock);

1 0

[alsa-devel] [PATCH -mm] sound/pci/pcxhr/pcxhr.c remove unused code
by Michal Piotrowski 25 Mar '07

25 Mar '07

This patch is against the latest -mm snapshot. Regards, Michal -- Michal K. K. Piotrowski LTG - Linux Testers Group (PL) (http://www.stardust.webpages.pl/ltg/) LTG - Linux Testers Group (EN) (http://www.stardust.webpages.pl/linux_testers_group_en/) Signed-off-by: Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> --- linux-work5/sound/pci/pcxhr/pcxhr.c 2007-03-25 00:31:00.000000000 +0100 +++ linux-work3/sound/pci/pcxhr/pcxhr.c 2007-03-25 14:45:33.000000000 +0200 @@ -477,26 +477,6 @@ static int pcxhr_update_r_buffer(struct return err; } - -#if 0 -static int pcxhr_pipe_sample_count(struct pcxhr_stream *stream, snd_pcm_uframes_t *sample_count) -{ - struct pcxhr_rmh rmh; - int err; - pcxhr_t *chip = snd_pcm_substream_chip(stream->substream); - pcxhr_init_rmh(&rmh, CMD_PIPE_SAMPLE_COUNT); - pcxhr_set_pipe_cmd_params(&rmh, stream->pipe->is_capture, 0, 0, - 1<<stream->pipe->first_audio); - err = pcxhr_send_msg(chip->mgr, &rmh); - if (err == 0) { - *sample_count = ((snd_pcm_uframes_t)rmh.stat[0]) << 24; - *sample_count += (snd_pcm_uframes_t)rmh.stat[1]; - } - snd_printdd("PIPE_SAMPLE_COUNT = %lx\n", *sample_count); - return err; -} -#endif - static inline int pcxhr_stream_scheduled_get_pipe(struct pcxhr_stream *stream, struct pcxhr_pipe **pipe) {

1 0

[alsa-devel] [PATCH -mm] sound/pci/pcxhr/pcxhr_hwdep.c remove unused code
by Michal Piotrowski 25 Mar '07

25 Mar '07

This patch is against the latest -mm snapshot. Regards, Michal -- Michal K. K. Piotrowski LTG - Linux Testers Group (PL) (http://www.stardust.webpages.pl/ltg/) LTG - Linux Testers Group (EN) (http://www.stardust.webpages.pl/linux_testers_group_en/) Signed-off-by: Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> --- linux-work5/sound/pci/pcxhr/pcxhr_hwdep.c 2007-02-04 19:44:54.000000000 +0100 +++ linux-work3/sound/pci/pcxhr/pcxhr_hwdep.c 2007-03-25 14:39:47.000000000 +0200 @@ -175,38 +175,6 @@ static int pcxhr_dsp_allocate_pipe( stru return 0; } -/* - * free playback/capture pipe (pcmp0/pcmc0) - */ -#if 0 -static int pcxhr_dsp_free_pipe( struct pcxhr_mgr *mgr, struct pcxhr_pipe *pipe) -{ - struct pcxhr_rmh rmh; - int capture_mask = 0; - int playback_mask = 0; - int err = 0; - - if (pipe->is_capture) - capture_mask = (1 << pipe->first_audio); - else - playback_mask = (1 << pipe->first_audio); - - /* stop one pipe */ - err = pcxhr_set_pipe_state(mgr, playback_mask, capture_mask, 0); - if (err < 0) - snd_printk(KERN_ERR "error stopping pipe!\n"); - /* release the pipe */ - pcxhr_init_rmh(&rmh, CMD_FREE_PIPE); - pcxhr_set_pipe_cmd_params(&rmh, pipe->is_capture, pipe->first_audio, 0, 0); - err = pcxhr_send_msg(mgr, &rmh); - if (err < 0) - snd_printk(KERN_ERR "error pipe release (CMD_FREE_PIPE) err(%x)\n", err); - pipe->status = PCXHR_PIPE_UNDEFINED; - return err; -} -#endif - - static int pcxhr_config_pipes(struct pcxhr_mgr *mgr) { int err, i, j;

1 0

[alsa-devel] [PATCH -mm] sound/pci/ice1712/pontis.c remove unused code
by Michal Piotrowski 25 Mar '07

25 Mar '07

This patch is against the latest -mm snapshot. Regards, Michal -- Michal K. K. Piotrowski LTG - Linux Testers Group (PL) (http://www.stardust.webpages.pl/ltg/) LTG - Linux Testers Group (EN) (http://www.stardust.webpages.pl/linux_testers_group_en/) Signed-off-by: Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> --- linux-work5/sound/pci/ice1712/pontis.c 2007-03-25 00:31:00.000000000 +0100 +++ linux-work3/sound/pci/ice1712/pontis.c 2007-03-25 00:56:33.000000000 +0100 @@ -767,12 +767,6 @@ static int __devinit pontis_init(struct WM_DAC_CTRL2, 0x0000, /* no deemphasis, no ZFLG */ WM_ADC_ATTEN_L, 0x0000, /* ADC muted */ WM_ADC_ATTEN_R, 0x0000, /* ADC muted */ -#if 0 - WM_ALC_CTRL1, 0x007b, /* */ - WM_ALC_CTRL2, 0x0000, /* */ - WM_ALC_CTRL3, 0x0000, /* */ - WM_NOISE_GATE, 0x0000, /* */ -#endif WM_DAC_MUTE, 0x0000, /* DAC unmute */ WM_ADC_MUX, 0x0003, /* ADC unmute, both CD/Line On */ };

1 0

[alsa-devel] [PATCH -mm] sound/pci/ice1712/prodigy192.c remove unused code
by Michal Piotrowski 25 Mar '07

25 Mar '07

This patch is against the latest -mm snapshot. Regards, Michal -- Michal K. K. Piotrowski LTG - Linux Testers Group (PL) (http://www.stardust.webpages.pl/ltg/) LTG - Linux Testers Group (EN) (http://www.stardust.webpages.pl/linux_testers_group_en/) Signed-off-by: Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> --- linux-work5/sound/pci/ice1712/prodigy192.c 2007-03-25 00:31:00.000000000 +0100 +++ linux-work3/sound/pci/ice1712/prodigy192.c 2007-03-25 14:32:42.000000000 +0200 @@ -234,129 +234,6 @@ static int stac9460_adc_vol_put(struct s return change; } -#if 0 -/* - * Headphone Amplifier - */ -static int aureon_set_headphone_amp(struct snd_ice1712 *ice, int enable) -{ - unsigned int tmp, tmp2; - - tmp2 = tmp = snd_ice1712_gpio_read(ice); - if (enable) - tmp |= AUREON_HP_SEL; - else - tmp &= ~ AUREON_HP_SEL; - if (tmp != tmp2) { - snd_ice1712_gpio_write(ice, tmp); - return 1; - } - return 0; -} - -static int aureon_get_headphone_amp(struct snd_ice1712 *ice) -{ - unsigned int tmp = snd_ice1712_gpio_read(ice); - - return ( tmp & AUREON_HP_SEL )!= 0; -} - -static int aureon_bool_info(struct snd_kcontrol *k, struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} - -static int aureon_hpamp_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - - ucontrol->value.integer.value[0] = aureon_get_headphone_amp(ice); - return 0; -} - - -static int aureon_hpamp_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - - return aureon_set_headphone_amp(ice,ucontrol->value.integer.value[0]); -} - -/* - * Deemphasis - */ -static int aureon_deemp_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - ucontrol->value.integer.value[0] = (wm_get(ice, WM_DAC_CTRL2) & 0xf) == 0xf; - return 0; -} - -static int aureon_deemp_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - int temp, temp2; - temp2 = temp = wm_get(ice, WM_DAC_CTRL2); - if (ucontrol->value.integer.value[0]) - temp |= 0xf; - else - temp &= ~0xf; - if (temp != temp2) { - wm_put(ice, WM_DAC_CTRL2, temp); - return 1; - } - return 0; -} - -/* - * ADC Oversampling - */ -static int aureon_oversampling_info(struct snd_kcontrol *k, struct snd_ctl_elem_info *uinfo) -{ - static char *texts[2] = { "128x", "64x" }; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 2; - - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - - return 0; -} - -static int aureon_oversampling_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - ucontrol->value.enumerated.item[0] = (wm_get(ice, WM_MASTER) & 0x8) == 0x8; - return 0; -} - -static int aureon_oversampling_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - int temp, temp2; - struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - - temp2 = temp = wm_get(ice, WM_MASTER); - - if (ucontrol->value.enumerated.item[0]) - temp |= 0x8; - else - temp &= ~0x8; - - if (temp != temp2) { - wm_put(ice, WM_MASTER, temp); - return 1; - } - return 0; -} -#endif - static const DECLARE_TLV_DB_SCALE(db_scale_dac, -19125, 75, 0); static const DECLARE_TLV_DB_SCALE(db_scale_adc, 0, 150, 0); @@ -424,36 +301,6 @@ static struct snd_kcontrol_new stac_cont .put = stac9460_adc_vol_put, .tlv = { .p = db_scale_adc } }, -#if 0 - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Route", - .info = wm_adc_mux_info, - .get = wm_adc_mux_get, - .put = wm_adc_mux_put, - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Headphone Amplifier Switch", - .info = aureon_bool_info, - .get = aureon_hpamp_get, - .put = aureon_hpamp_put - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "DAC Deemphasis Switch", - .info = aureon_bool_info, - .get = aureon_deemp_get, - .put = aureon_deemp_put - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "ADC Oversampling", - .info = aureon_oversampling_info, - .get = aureon_oversampling_get, - .put = aureon_oversampling_put - }, -#endif }; static int __devinit prodigy192_add_controls(struct snd_ice1712 *ice)

1 0

[alsa-devel] [PATCH -mm] sound/pci/ice1712/juli.c remove unused code
by Michal Piotrowski 25 Mar '07

25 Mar '07

This patch is against the latest -mm snapshot. Regards, Michal -- Michal K. K. Piotrowski LTG - Linux Testers Group (PL) (http://www.stardust.webpages.pl/ltg/) LTG - Linux Testers Group (EN) (http://www.stardust.webpages.pl/linux_testers_group_en/) Signed-off-by: Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> --- linux-work5/sound/pci/ice1712/juli.c 2007-03-25 00:31:00.000000000 +0100 +++ linux-work3/sound/pci/ice1712/juli.c 2007-03-25 14:27:33.000000000 +0200 @@ -160,13 +160,6 @@ static int __devinit juli_init(struct sn int err; struct snd_akm4xxx *ak; -#if 0 - for (err = 0; err < 0x20; err++) - juli_ak4114_read(ice, err); - juli_ak4114_write(ice, 0, 0x0f); - juli_ak4114_read(ice, 0); - juli_ak4114_read(ice, 1); -#endif err = snd_ak4114_create(ice->card, juli_ak4114_read, juli_ak4114_write, @@ -175,14 +168,7 @@ static int __devinit juli_init(struct sn if (err < 0) return err; -#if 0 - /* it seems that the analog doughter board detection does not work - reliably, so force the analog flag; it should be very rare - to use Juli@ without the analog doughter board */ - ice->spec.juli.analog = (ice->gpio.get_data(ice) & GPIO_ANALOG_PRESENT) ? 0 : 1; -#else ice->spec.juli.analog = 1; -#endif if (ice->spec.juli.analog) { printk(KERN_INFO "juli@: analog I/O detected\n");

1 0

[alsa-devel] [PATCH -mm] sound/pci/au88x0/au88x0_xtalk.c remove unused code
by Michal Piotrowski 25 Mar '07

25 Mar '07

This patch is against the latest -mm snapshot. Regards, Michal -- Michal K. K. Piotrowski LTG - Linux Testers Group (PL) (http://www.stardust.webpages.pl/ltg/) LTG - Linux Testers Group (EN) (http://www.stardust.webpages.pl/linux_testers_group_en/) Signed-off-by: Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> --- linux-work5/sound/pci/au88x0/au88x0_xtalk.c 2007-02-04 19:44:54.000000000 +0100 +++ linux-work3/sound/pci/au88x0/au88x0_xtalk.c 2007-03-25 16:52:23.000000000 +0200 @@ -369,148 +369,6 @@ vortex_XtalkHw_SetRightXTStates(vortex_t hwwrite(vortex->mmio, 0x24534 + i * 0x24, arg_0[3]); } -#if 0 -static void -vortex_XtalkHw_GetLeftEQ(vortex_t * vortex, short *arg_0, short *arg_4, - xtalk_coefs_t coefs) -{ - int i; - - for (i = 0; i < 5; i++) { - coefs[i][0] = hwread(vortex->mmio, 0x24200 + i * 0x24); - coefs[i][1] = hwread(vortex->mmio, 0x24204 + i * 0x24); - coefs[i][2] = hwread(vortex->mmio, 0x24208 + i * 0x24); - coefs[i][3] = hwread(vortex->mmio, 0x2420c + i * 0x24); - coefs[i][4] = hwread(vortex->mmio, 0x24210 + i * 0x24); - } - *arg_0 = hwread(vortex->mmio, 0x24538) & 0xffff; - *arg_4 = hwread(vortex->mmio, 0x2453c) & 0xffff; -} - -static void -vortex_XtalkHw_GetRightEQ(vortex_t * vortex, short *arg_0, short *arg_4, - xtalk_coefs_t coefs) -{ - int i; - - for (i = 0; i < 5; i++) { - coefs[i][0] = hwread(vortex->mmio, 0x242b4 + i * 0x24); - coefs[i][1] = hwread(vortex->mmio, 0x242b8 + i * 0x24); - coefs[i][2] = hwread(vortex->mmio, 0x242bc + i * 0x24); - coefs[i][3] = hwread(vortex->mmio, 0x242c0 + i * 0x24); - coefs[i][4] = hwread(vortex->mmio, 0x242c4 + i * 0x24); - } - *arg_0 = hwread(vortex->mmio, 0x24540) & 0xffff; - *arg_4 = hwread(vortex->mmio, 0x24544) & 0xffff; -} - -static void -vortex_XtalkHw_GetLeftXT(vortex_t * vortex, short *arg_0, short *arg_4, - xtalk_coefs_t coefs) -{ - int i; - - for (i = 0; i < 5; i++) { - coefs[i][0] = hwread(vortex->mmio, 0x24368 + i * 0x24); - coefs[i][1] = hwread(vortex->mmio, 0x2436C + i * 0x24); - coefs[i][2] = hwread(vortex->mmio, 0x24370 + i * 0x24); - coefs[i][3] = hwread(vortex->mmio, 0x24374 + i * 0x24); - coefs[i][4] = hwread(vortex->mmio, 0x24378 + i * 0x24); - } - *arg_0 = hwread(vortex->mmio, 0x24548) & 0xffff; - *arg_4 = hwread(vortex->mmio, 0x2454C) & 0xffff; -} - -static void -vortex_XtalkHw_GetRightXT(vortex_t * vortex, short *arg_0, short *arg_4, - xtalk_coefs_t coefs) -{ - int i; - - for (i = 0; i < 5; i++) { - coefs[i][0] = hwread(vortex->mmio, 0x2441C + i * 0x24); - coefs[i][1] = hwread(vortex->mmio, 0x24420 + i * 0x24); - coefs[i][2] = hwread(vortex->mmio, 0x24424 + i * 0x24); - coefs[i][3] = hwread(vortex->mmio, 0x24428 + i * 0x24); - coefs[i][4] = hwread(vortex->mmio, 0x2442C + i * 0x24); - } - *arg_0 = hwread(vortex->mmio, 0x24550) & 0xffff; - *arg_4 = hwread(vortex->mmio, 0x24554) & 0xffff; -} - -static void -vortex_XtalkHw_GetLeftEQStates(vortex_t * vortex, xtalk_instate_t arg_0, - xtalk_state_t coefs) -{ - int i; - - for (i = 0; i < 5; i++) { - coefs[i][0] = hwread(vortex->mmio, 0x24214 + i * 0x24); - coefs[i][1] = hwread(vortex->mmio, 0x24218 + i * 0x24); - coefs[i][2] = hwread(vortex->mmio, 0x2421C + i * 0x24); - coefs[i][3] = hwread(vortex->mmio, 0x24220 + i * 0x24); - } - arg_0[0] = hwread(vortex->mmio, 0x244F8 + i * 0x24); - arg_0[1] = hwread(vortex->mmio, 0x244FC + i * 0x24); - arg_0[2] = hwread(vortex->mmio, 0x24500 + i * 0x24); - arg_0[3] = hwread(vortex->mmio, 0x24504 + i * 0x24); -} - -static void -vortex_XtalkHw_GetRightEQStates(vortex_t * vortex, xtalk_instate_t arg_0, - xtalk_state_t coefs) -{ - int i; - - for (i = 0; i < 5; i++) { - coefs[i][0] = hwread(vortex->mmio, 0x242C8 + i * 0x24); - coefs[i][1] = hwread(vortex->mmio, 0x242CC + i * 0x24); - coefs[i][2] = hwread(vortex->mmio, 0x242D0 + i * 0x24); - coefs[i][3] = hwread(vortex->mmio, 0x242D4 + i * 0x24); - } - arg_0[0] = hwread(vortex->mmio, 0x24508 + i * 0x24); - arg_0[1] = hwread(vortex->mmio, 0x2450C + i * 0x24); - arg_0[2] = hwread(vortex->mmio, 0x24510 + i * 0x24); - arg_0[3] = hwread(vortex->mmio, 0x24514 + i * 0x24); -} - -static void -vortex_XtalkHw_GetLeftXTStates(vortex_t * vortex, xtalk_instate_t arg_0, - xtalk_state_t coefs) -{ - int i; - - for (i = 0; i < 5; i++) { - coefs[i][0] = hwread(vortex->mmio, 0x2437C + i * 0x24); - coefs[i][1] = hwread(vortex->mmio, 0x24380 + i * 0x24); - coefs[i][2] = hwread(vortex->mmio, 0x24384 + i * 0x24); - coefs[i][3] = hwread(vortex->mmio, 0x24388 + i * 0x24); - } - arg_0[0] = hwread(vortex->mmio, 0x24518 + i * 0x24); - arg_0[1] = hwread(vortex->mmio, 0x2451C + i * 0x24); - arg_0[2] = hwread(vortex->mmio, 0x24520 + i * 0x24); - arg_0[3] = hwread(vortex->mmio, 0x24524 + i * 0x24); -} - -static void -vortex_XtalkHw_GetRightXTStates(vortex_t * vortex, xtalk_instate_t arg_0, - xtalk_state_t coefs) -{ - int i; - - for (i = 0; i < 5; i++) { - coefs[i][0] = hwread(vortex->mmio, 0x24430 + i * 0x24); - coefs[i][1] = hwread(vortex->mmio, 0x24434 + i * 0x24); - coefs[i][2] = hwread(vortex->mmio, 0x24438 + i * 0x24); - coefs[i][3] = hwread(vortex->mmio, 0x2443C + i * 0x24); - } - arg_0[0] = hwread(vortex->mmio, 0x24528 + i * 0x24); - arg_0[1] = hwread(vortex->mmio, 0x2452C + i * 0x24); - arg_0[2] = hwread(vortex->mmio, 0x24530 + i * 0x24); - arg_0[3] = hwread(vortex->mmio, 0x24534 + i * 0x24); -} - -#endif /* Gains */ static void @@ -529,16 +387,6 @@ vortex_XtalkHw_SetGainsAllChan(vortex_t vortex_XtalkHw_SetGains(vortex, asXtalkGainsAllChan); } -#if 0 -static void vortex_XtalkHw_GetGains(vortex_t * vortex, xtalk_gains_t gains) -{ - int i; - - for (i = 0; i < XTGAINS_SZ; i++) - gains[i] = hwread(vortex->mmio, 0x244D0 + i * 4); -} - -#endif /* Delay parameters */ static void @@ -577,53 +425,6 @@ vortex_XtalkHw_SetRightDline(vortex_t * } } -#if 0 -static void -vortex_XtalkHw_GetDelay(vortex_t * vortex, unsigned short *right, - unsigned short *left) -{ - int esp0; - - esp0 = hwread(vortex->mmio, 0x24660); - *right = (esp0 >> 8) & 0x1f; - *left = (esp0 >> 0xd) & 0x1f; -} - -static void vortex_XtalkHw_GetLeftDline(vortex_t * vortex, xtalk_dline_t dline) -{ - int i; - - for (i = 0; i < 0x20; i++) { - dline[i] = - (hwread(vortex->mmio, 0x24000 + (i << 2)) & 0xffff) | - (hwread(vortex->mmio, 0x24080 + (i << 2)) << 0x10); - } -} - -static void vortex_XtalkHw_GetRightDline(vortex_t * vortex, xtalk_dline_t dline) -{ - int i; - - for (i = 0; i < 0x20; i++) { - dline[i] = - (hwread(vortex->mmio, 0x24100 + (i << 2)) & 0xffff) | - (hwread(vortex->mmio, 0x24180 + (i << 2)) << 0x10); - } -} - -#endif -/* Control/Global stuff */ - -#if 0 -static void vortex_XtalkHw_SetControlReg(vortex_t * vortex, u32 ctrl) -{ - hwwrite(vortex->mmio, 0x24660, ctrl); -} -static void vortex_XtalkHw_GetControlReg(vortex_t * vortex, u32 *ctrl) -{ - *ctrl = hwread(vortex->mmio, 0x24660); -} -#endif static void vortex_XtalkHw_SetSampleRate(vortex_t * vortex, u32 sr) { u32 temp; @@ -633,13 +434,6 @@ static void vortex_XtalkHw_SetSampleRate hwwrite(vortex->mmio, 0x24660, temp); } -#if 0 -static void vortex_XtalkHw_GetSampleRate(vortex_t * vortex, u32 *sr) -{ - *sr = (hwread(vortex->mmio, 0x24660) >> 3) & 0x1f; -} - -#endif static void vortex_XtalkHw_Enable(vortex_t * vortex) { u32 temp;

1 0