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[alsa-devel] hda-intel channels hibernate bug
by Johannes Berg 15 Apr '09

15 Apr '09

Hi, On my macbook 5,1 I need to use model=mbp3, as you may still remember. I also need to set channels to 6 to hear anything at all, probably due to the codec mismatch -- it's not really an mbp3 so that only works 80%, but I haven't gotten around to writing a better codec description. However, when I hibernate with channels set to 6, I always need to reset the channels to 6 again after resume to hear anything -- it seems the channels control isn't reset to the hardware at resume time. I suppose this bugs affects any users of this codec, regardless of whether it is actually right or wrong. johannes PS: I wanted to file a bug on the alsa bugtracker but mantisbt decided I had not entered all fields correctly and didn't give me an option to edit, so I had to reenter, which frustrated me enough to give up on mantis.

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[alsa-devel] Gigabyte ALC889A Function ID: 0x01
by Jerry Geis 15 Apr '09

15 Apr '09

I am using the gigabyte MA78GM-US2H motherboard. it has the ALC889A audio. I am trying 1.0.19 and audio works out the analog back port but not over HDMI. Codec: Realtek ALC889A Address: 0 Function Id: 0x1 <--------------------------------- this is different Vendor Id: 0x10ec0885 Subsystem Id: 0x1458a002 Revision Id: 0x100101 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=2, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x11: Stereo I have been looking but havent found a solution yet. I tried grabbing the snapshot tar ball for today but I get errors on the compile. I had a similiar motherboard but it does not report Function ID: 01 : - HDMI audio on this board works correctly. Codec: Realtek ALC889A Address: 0 Vendor Id: 0x10ec0885 Subsystem Id: 0x1458a002 Revision Id: 0x100101 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 Any thoughts???? Jerry

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Re: [alsa-devel] 15 April unstable does not handle wave-in
by Susan Cragin 15 Apr '09

15 Apr '09

>At Wed, 15 Apr 2009 07:22:08 -0400 (GMT-04:00), >Susan Cragin wrote: >> >> I just downloaded and installed the unstable snapshot 15-April 9:55. >> Both my sound cards are shown by asoundconf, but neither works. The wave-in wave-out is still not recognized. > >I suppose you are referring to CA0110? Could you elaborate? > >The ca0110 codec support itself has no big change. The last one was >in Feb. 18. The rest is either HD-audio core or PCM core. But these >have little to do with the probing... > > >Takashi Please excuse my technical ignorance. Here is everything I know. My hda-intel card is also not being recognized after installing alsa-driver-unstable. The only thing that works is my usb microphone. I think that the program is not giving the instruction to have modules inserted on bootup. I have attached sample modprobe, aplay, asoundconf, modules, alsa-base.conf Susan ---------------------------------------------------------------------- $ modprobe snd-card-hda-intel WARNING: All config files need .conf: /etc/modprobe.d/oss-compat, it will be ignored in a future release. FATAL: Module snd_card_hda_intel not found. oss-compat is already the newest version. ----------------------------------------------------------------------- $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 6: Si3054 Modem [Si3054 Modem] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: system [iMic USB audio system], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 ------------------------------------------------------------------------ $ asoundconf list Names of available sound cards: Intel Generic (this is my Creative X-fi) system (this is my USB microphone) ------------------------------------------------------------------------ # /etc/modules: kernel modules to load at boot time. # # This file contains the names of kernel modules that should be loaded # at boot time, one per line. Lines beginning with "#" are ignored. loop lp ------------------------------------------------------------------------ alsabase.conf # autoloader aliases install sound-slot-0 /sbin/modprobe snd-card-0 install sound-slot-1 /sbin/modprobe snd-card-1 install sound-slot-2 /sbin/modprobe snd-card-2 install sound-slot-3 /sbin/modprobe snd-card-3 install sound-slot-4 /sbin/modprobe snd-card-4 install sound-slot-5 /sbin/modprobe snd-card-5 install sound-slot-6 /sbin/modprobe snd-card-6 install sound-slot-7 /sbin/modprobe snd-card-7 # Cause optional modules to be loaded above generic modules install snd /sbin/modprobe --ignore-install snd $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-ioctl32 ; /sbin/modprobe --quiet --use-blacklist snd-seq ; } # # Workaround at bug #499695 (reverted in Ubuntu see LP #319505) install snd-pcm /sbin/modprobe --ignore-install snd-pcm $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-pcm-oss ; : ; } install snd-mixer /sbin/modprobe --ignore-install snd-mixer $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-mixer-oss ; : ; } install snd-seq /sbin/modprobe --ignore-install snd-seq $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-seq-midi ; /sbin/modprobe --quiet --use-blacklist snd-se$ # install snd-rawmidi /sbin/modprobe --ignore-install snd-rawmidi $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-seq-midi ; : ; } # Cause optional modules to be loaded above sound card driver modules install snd-emu10k1 /sbin/modprobe --ignore-install snd-emu10k1 $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-emu10k1-synth ; } install snd-via82xx /sbin/modprobe --ignore-install snd-via82xx $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-seq ; } # Load saa7134-alsa instead of saa7134 (which gets dragged in by it anyway) install saa7134 /sbin/modprobe --ignore-install saa7134 $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist saa7134-alsa ; : ; } # Prevent abnormal drivers from grabbing index 0 options bt87x index=-2 options cx88_alsa index=-2 options saa7134-alsa index=-2 options snd-atiixp-modem index=-2 options snd-intel8x0m index=-2 options snd-via82xx-modem index=-2 options snd-usb-audio index=-2 options snd-usb-us122l index=-2 options snd-usb-usx2y index=-2 options snd-usb-caiaq index=-2 # Ubuntu #62691, enable MPU for snd-cmipci options snd-cmipci mpu_port=0x330 fm_port=0x388 # Keep snd-pcsp from beeing loaded as first soundcard options snd-pcsp index=-2

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[alsa-devel] [PATCH] Automatic driver detection for Amilo 3540
by Michalis Pappas 15 Apr '09

15 Apr '09

A patch to fix the issue described in #4344 and #4352. Signed-off-by: Michalis Pappas <mepapp(a)noc.uoa.gr> --- a/alsa-kernel/pci/hda/patch_realtek.c 2009-01-19 13:08:58.000000000 +0200 +++ b/alsa-kernel/pci/hda/patch_realtek.c 2009-04-11 14:15:44.000000000 +0300 @@ -8516,6 +8516,8 @@ SND_PCI_QUIRK(0x1734, 0x1108, "Fujitsu AMILO Pi2515", ALC883_FUJITSU_PI2515), SND_PCI_QUIRK(0x1734, 0x113d, "Fujitsu AMILO Xa3530", ALC888_FUJITSU_XA3530), + SND_PCI_QUIRK(0x1734, 0x1137, "Fujitsu AMILO Xa3540", + ALC888_FUJITSU_XA3530), SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo 101e", ALC883_LENOVO_101E_2ch), SND_PCI_QUIRK(0x17aa, 0x2085, "Lenovo NB0763", ALC883_LENOVO_NB0763), SND_PCI_QUIRK(0x17aa, 0x3bfc, "Lenovo NB0763", ALC883_LENOVO_NB0763),

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Re: [alsa-devel] Ubuntu sound problem - http://ubuntuforums.org/showthread.php?t=1073090
by crimsunkg＠yahoo.com 15 Apr '09

15 Apr '09

--- On Mon, 4/13/09, jesse fillion <jessefillion24(a)hotmail.com> wrote: > I have attached my alsa-info output. You should try the latest alsa-driver snapshot (http://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver-sn…), which contains quite a few fixes.

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[alsa-devel] [PATCH] AOA: Convert onyx and tas codecs to new-style i2c drivers (v3)
by Jean Delvare 15 Apr '09

15 Apr '09

The legacy i2c binding model is going away soon, so convert the AOA codec drivers to the new model or they'll break. Signed-off-by: Jean Delvare <khali(a)linux-fr.org> Cc: Johannes Berg <johannes(a)sipsolutions.net> Cc: Benjamin Herrenschmidt <benh(a)kernel.crashing.org> Tested-by: Andreas Schwab <schwab(a)linux-m68k.org> --- Johannes, this is a reworked patch which assumes that the onyx codec probing can fail. It's not exactly clean but that's the easiest approach for now. Can you please give it a try? Thanks. sound/aoa/codecs/onyx.c | 76 ++++++++++++++++++++++++++++++++--------------- sound/aoa/codecs/tas.c | 66 +++++++++++++++++++++++++--------------- 2 files changed, 95 insertions(+), 47 deletions(-) --- linux-2.6.30-rc2.orig/sound/aoa/codecs/onyx.c 2009-04-15 09:55:20.000000000 +0200 +++ linux-2.6.30-rc2/sound/aoa/codecs/onyx.c 2009-04-15 14:07:50.000000000 +0200 @@ -47,7 +47,7 @@ MODULE_DESCRIPTION("pcm3052 (onyx) codec struct onyx { /* cache registers 65 to 80, they are write-only! */ u8 cache[16]; - struct i2c_client i2c; + struct i2c_client *i2c; struct aoa_codec codec; u32 initialised:1, spdif_locked:1, @@ -72,7 +72,7 @@ static int onyx_read_register(struct ony *value = onyx->cache[reg-FIRSTREGISTER]; return 0; } - v = i2c_smbus_read_byte_data(&onyx->i2c, reg); + v = i2c_smbus_read_byte_data(onyx->i2c, reg); if (v < 0) return -1; *value = (u8)v; @@ -84,7 +84,7 @@ static int onyx_write_register(struct on { int result; - result = i2c_smbus_write_byte_data(&onyx->i2c, reg, value); + result = i2c_smbus_write_byte_data(onyx->i2c, reg, value); if (!result) onyx->cache[reg-FIRSTREGISTER] = value; return result; @@ -996,12 +996,45 @@ static void onyx_exit_codec(struct aoa_c onyx->codec.soundbus_dev->detach_codec(onyx->codec.soundbus_dev, onyx); } -static struct i2c_driver onyx_driver; - static int onyx_create(struct i2c_adapter *adapter, struct device_node *node, int addr) { + struct i2c_board_info info; + struct i2c_client *client; + + memset(&info, 0, sizeof(struct i2c_board_info)); + strlcpy(info.type, "aoa_codec_onyx", I2C_NAME_SIZE); + info.addr = addr; + info.platform_data = node; + client = i2c_new_device(adapter, &info); + if (!client) + return -ENODEV; + + /* + * We know the driver is already loaded, so the device should be + * already bound. If not it means binding failed, which suggests + * the device doesn't really exist and should be deleted. + * Ideally this would be replaced by better checks _before_ + * instantiating the device. + */ + if (!client->driver) { + i2c_unregister_device(client); + return -ENODEV; + } + + /* + * Let i2c-core delete that device on driver removal. + * This is safe because i2c-core holds the core_lock mutex for us. + */ + list_add_tail(&client->detected, &client->driver->clients); + return 0; +} + +static int onyx_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct device_node *node = client->dev.platform_data; struct onyx *onyx; u8 dummy; @@ -1011,20 +1044,12 @@ static int onyx_create(struct i2c_adapte return -ENOMEM; mutex_init(&onyx->mutex); - onyx->i2c.driver = &onyx_driver; - onyx->i2c.adapter = adapter; - onyx->i2c.addr = addr & 0x7f; - strlcpy(onyx->i2c.name, "onyx audio codec", I2C_NAME_SIZE); - - if (i2c_attach_client(&onyx->i2c)) { - printk(KERN_ERR PFX "failed to attach to i2c\n"); - goto fail; - } + onyx->i2c = client; + i2c_set_clientdata(client, onyx); /* we try to read from register ONYX_REG_CONTROL * to check if the codec is present */ if (onyx_read_register(onyx, ONYX_REG_CONTROL, &dummy) != 0) { - i2c_detach_client(&onyx->i2c); printk(KERN_ERR PFX "failed to read control register\n"); goto fail; } @@ -1036,14 +1061,14 @@ static int onyx_create(struct i2c_adapte onyx->codec.node = of_node_get(node); if (aoa_codec_register(&onyx->codec)) { - i2c_detach_client(&onyx->i2c); goto fail; } printk(KERN_DEBUG PFX "created and attached onyx instance\n"); return 0; fail: + i2c_set_clientdata(client, NULL); kfree(onyx); - return -EINVAL; + return -ENODEV; } static int onyx_i2c_attach(struct i2c_adapter *adapter) @@ -1080,28 +1105,33 @@ static int onyx_i2c_attach(struct i2c_ad return onyx_create(adapter, NULL, 0x47); } -static int onyx_i2c_detach(struct i2c_client *client) +static int onyx_i2c_remove(struct i2c_client *client) { - struct onyx *onyx = container_of(client, struct onyx, i2c); - int err; + struct onyx *onyx = i2c_get_clientdata(client); - if ((err = i2c_detach_client(client))) - return err; aoa_codec_unregister(&onyx->codec); of_node_put(onyx->codec.node); if (onyx->codec_info) kfree(onyx->codec_info); + i2c_set_clientdata(client, onyx); kfree(onyx); return 0; } +static const struct i2c_device_id onyx_i2c_id[] = { + { "aoa_codec_onyx", 0 }, + { } +}; + static struct i2c_driver onyx_driver = { .driver = { .name = "aoa_codec_onyx", .owner = THIS_MODULE, }, .attach_adapter = onyx_i2c_attach, - .detach_client = onyx_i2c_detach, + .probe = onyx_i2c_probe, + .remove = onyx_i2c_remove, + .id_table = onyx_i2c_id, }; static int __init onyx_init(void) --- linux-2.6.30-rc2.orig/sound/aoa/codecs/tas.c 2009-04-15 09:55:20.000000000 +0200 +++ linux-2.6.30-rc2/sound/aoa/codecs/tas.c 2009-04-15 09:56:39.000000000 +0200 @@ -82,7 +82,7 @@ MODULE_DESCRIPTION("tas codec driver for struct tas { struct aoa_codec codec; - struct i2c_client i2c; + struct i2c_client *i2c; u32 mute_l:1, mute_r:1 , controls_created:1 , drc_enabled:1, @@ -108,9 +108,9 @@ static struct tas *codec_to_tas(struct a static inline int tas_write_reg(struct tas *tas, u8 reg, u8 len, u8 *data) { if (len == 1) - return i2c_smbus_write_byte_data(&tas->i2c, reg, *data); + return i2c_smbus_write_byte_data(tas->i2c, reg, *data); else - return i2c_smbus_write_i2c_block_data(&tas->i2c, reg, len, data); + return i2c_smbus_write_i2c_block_data(tas->i2c, reg, len, data); } static void tas3004_set_drc(struct tas *tas) @@ -882,12 +882,34 @@ static void tas_exit_codec(struct aoa_co } -static struct i2c_driver tas_driver; - static int tas_create(struct i2c_adapter *adapter, struct device_node *node, int addr) { + struct i2c_board_info info; + struct i2c_client *client; + + memset(&info, 0, sizeof(struct i2c_board_info)); + strlcpy(info.type, "aoa_codec_tas", I2C_NAME_SIZE); + info.addr = addr; + info.platform_data = node; + + client = i2c_new_device(adapter, &info); + if (!client) + return -ENODEV; + + /* + * Let i2c-core delete that device on driver removal. + * This is safe because i2c-core holds the core_lock mutex for us. + */ + list_add_tail(&client->detected, &client->driver->clients); + return 0; +} + +static int tas_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct device_node *node = client->dev.platform_data; struct tas *tas; tas = kzalloc(sizeof(struct tas), GFP_KERNEL); @@ -896,17 +918,11 @@ static int tas_create(struct i2c_adapter return -ENOMEM; mutex_init(&tas->mtx); - tas->i2c.driver = &tas_driver; - tas->i2c.adapter = adapter; - tas->i2c.addr = addr; + tas->i2c = client; + i2c_set_clientdata(client, tas); + /* seems that half is a saner default */ tas->drc_range = TAS3004_DRC_MAX / 2; - strlcpy(tas->i2c.name, "tas audio codec", I2C_NAME_SIZE); - - if (i2c_attach_client(&tas->i2c)) { - printk(KERN_ERR PFX "failed to attach to i2c\n"); - goto fail; - } strlcpy(tas->codec.name, "tas", MAX_CODEC_NAME_LEN); tas->codec.owner = THIS_MODULE; @@ -915,14 +931,12 @@ static int tas_create(struct i2c_adapter tas->codec.node = of_node_get(node); if (aoa_codec_register(&tas->codec)) { - goto detach; + goto fail; } printk(KERN_DEBUG "snd-aoa-codec-tas: tas found, addr 0x%02x on %s\n", - addr, node->full_name); + (unsigned int)client->addr, node->full_name); return 0; - detach: - i2c_detach_client(&tas->i2c); fail: mutex_destroy(&tas->mtx); kfree(tas); @@ -970,14 +984,11 @@ static int tas_i2c_attach(struct i2c_ada return -ENODEV; } -static int tas_i2c_detach(struct i2c_client *client) +static int tas_i2c_remove(struct i2c_client *client) { - struct tas *tas = container_of(client, struct tas, i2c); - int err; + struct tas *tas = i2c_get_clientdata(client); u8 tmp = TAS_ACR_ANALOG_PDOWN; - if ((err = i2c_detach_client(client))) - return err; aoa_codec_unregister(&tas->codec); of_node_put(tas->codec.node); @@ -989,13 +1000,20 @@ static int tas_i2c_detach(struct i2c_cli return 0; } +static const struct i2c_device_id tas_i2c_id[] = { + { "aoa_codec_tas", 0 }, + { } +}; + static struct i2c_driver tas_driver = { .driver = { .name = "aoa_codec_tas", .owner = THIS_MODULE, }, .attach_adapter = tas_i2c_attach, - .detach_client = tas_i2c_detach, + .probe = tas_i2c_probe, + .remove = tas_i2c_remove, + .id_table = tas_i2c_id, }; static int __init tas_init(void) -- Jean Delvare

2 2

[alsa-devel] 15 April unstable does not handle wave-in
by Susan Cragin 15 Apr '09

15 Apr '09

I just downloaded and installed the unstable snapshot 15-April 9:55. Both my sound cards are shown by asoundconf, but neither works. The wave-in wave-out is still not recognized. (USB sound does work.) System: Asus G1S running Ubuntu Jaunty with all updates (most recent kernel is only 2.6.28-11-Generic). Sound cards present: hda-intel / Realtek ALC660-VD hda-audio-generic / Creative CA0110-IBG (Creative X-fi)

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[alsa-devel] ASoC: DMA Platform driver for i.mx27
by javier Martin 15 Apr '09

15 Apr '09

This still does not support bidirectional audio but works when used playback or capture separatelly. SDMA API used in i.mx31 dma driver has been moved to DMA API. Please, have patience with me, this is a very experimental version and I am new to this world (audio codecs and linux in general). Signed-off-by: Javier Martin <javier.martin(a)vista-silicon.com> /* * linux/sound/arm/mx27-pcm.c -- ALSA SoC interface for the Freescale i.MX27 CPU * * Copyright 2009 Vista Silicon S.L. * Author: Javier Martin * javier.martin(a)vista-silicon.com * * Based on mxc-pcm.c by Liam Girdwood, (C) 2006 Wolfson Microelectronics PLC. * and on mxc-alsa-mc13783 (C) 2006 Freescale. * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as * published by the Free Software Foundation. * * Revision history * 14th April 2009 Initial version. * */ #include <linux/module.h> #include <linux/init.h> #include <linux/platform_device.h> #include <linux/slab.h> #include <linux/dma-mapping.h> #include <sound/driver.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> #include <asm/dma.h> #include <asm/hardware.h> #include "imx27-pcm.h" #include "imx27-ssi.h" //For debugging /* debug */ #define IMX_DEBUG 0 #if IMX_DEBUG #define dbg(format, arg...) printk(format, ## arg) #else #define dbg(format, arg...) #endif /* Taken from drivers/mxc/ssi/registers.h just provisionally */ #define MXC_SSI1STX0 0x00 #define MXC_SSI1SRX0 0x08 #define MXC_SSI2STX0 0x00 #define MXC_SSI2SRX0 0x08 static const struct snd_pcm_hardware mx27_pcm_hardware = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID), .formats = SNDRV_PCM_FMTBIT_S16_LE, .buffer_bytes_max = 32 * 1024, .period_bytes_min = 64, .period_bytes_max = 8 * 1024, .periods_min = 2, .periods_max = 255, .fifo_size = 0, }; struct mx27_runtime_data { int dma_ch_tx; //DMA channel number for tx int dma_ch_rx; //DMA channel number for rx int active; //Is the stream active? unsigned int period; //Period number unsigned int periods; //¿? int tx_spin; //¿? spinlock_t dma_lock; //DMA spinlock struct mx27_pcm_dma_param *dma_params; //FIXME this is SDMA api }; /*! * This function stops the current dma transfer for playback * and clears the dma pointers. * * @param substream pointer to the structure of the current stream. * */ static int audio_stop_dma(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct mx27_runtime_data *prtd = runtime->private_data; unsigned int dma_size = frames_to_bytes(runtime, runtime->period_size); unsigned int offset = dma_size * prtd->periods; unsigned long flags; spin_lock_irqsave(&prtd->dma_lock, flags); // dbg("MX27 : audio_stop_dma active = 0\n"); prtd->active = 0; prtd->period = 0; prtd->periods = 0; /* this stops the dma channel and clears the buffer ptrs */ if(substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { if(mxc_dma_disable(prtd->dma_ch_tx)<0) printk("audio_stop_dma: error when stopping dma transfer in channel %d \n", prtd->dma_ch_tx); } else { if(mxc_dma_disable(prtd->dma_ch_rx)<0) printk("audio_stop_dma: error when stopping dma transfer in channel %d \n", prtd->dma_ch_rx); } spin_unlock_irqrestore(&prtd->dma_lock, flags); return 0; } /*! * This function is called whenever a new audio block needs to be * transferred to wm8974. The function receives the address and the size * of the new block and start a new DMA transfer. * * @param substream pointer to the structure of the current stream. * */ static int dma_new_period(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct mx27_runtime_data *prtd = runtime->private_data; unsigned int dma_size; unsigned int offset; int ret=0; mxc_dma_requestbuf_t dma_request; if (prtd->active){ memset(&dma_request, 0, sizeof(mxc_dma_requestbuf_t));; dma_size = frames_to_bytes(runtime, runtime->period_size); printk("dma_new_period: prtd->period (%x) runtime->periods (%d)\n", prtd->period,runtime->periods); printk("dma_new_period: runtime->period_size (%d) dma_size (%d)\n", (unsigned int)runtime->period_size, runtime->dma_bytes); offset = dma_size * prtd->period; snd_assert(dma_size <= mx27_pcm_hardware.period_bytes_max); if(substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { printk("\ndma_new_period (playback): runtime->dma_area = %x\noffset = %x\ndma_size = %x\n", runtime->dma_area, offset, dma_size); printk("\ndma_new_period : dma_addr is %x\n", (runtime->dma_addr + offset)); dma_request.src_addr = (dma_addr_t)(runtime->dma_addr + offset); #ifdef CONFIG_SND_SOC_MX27_SSI1 dma_request.dst_addr = (dma_addr_t) (SSI1_BASE_ADDR + MXC_SSI1STX0); printk("dst_addr is %x\n", dma_request.dst_addr); printk("src_addr is %x\n", dma_request.src_addr); #else dma_request.dst_addr = (dma_addr_t) (SSI2_BASE_ADDR + MXC_SSI2STX0); #endif } else { printk("\ndma_new_period (capture): runtime->dma_area = %x\noffset = %x\ndma_size = %x\n", runtime->dma_area, offset, dma_size); printk("\ndma_new_period: dma_addr is %x\n", (runtime->dma_addr + offset)); dma_request.dst_addr = (dma_addr_t) (runtime->dma_addr + offset); #ifdef CONFIG_SND_SOC_MX27_SSI1 dma_request.src_addr = (dma_addr_t) (SSI1_BASE_ADDR + MXC_SSI1SRX0); #else dma_request.src_addr = (dma_addr_t) (SSI2_BASE_ADDR + MXC_SSI2SRX0); #endif } dma_request.num_of_bytes = dma_size; //8 printk("dma_new_period: Start DMA offset (%d) size (%d)\n", offset, runtime->dma_bytes); if(substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { printk("dma_neq_period: configuring tx dma\n"); ret = mxc_dma_config(prtd->dma_ch_tx, &dma_request, 1, MXC_DMA_MODE_WRITE); if(ret<0) { printk("dma_new_period: buffer could not be added for playback transfer\n"); return ret; } printk("dma_new_period: enable tx dma transfer\n"); ret = mxc_dma_enable(prtd->dma_ch_tx); if(ret<0) { printk("dma_new_period: cannot queue DMA buffer\ (%i)\n", ret); return ret; } } else { ret = mxc_dma_config(prtd->dma_ch_rx, &dma_request, 1, MXC_DMA_MODE_READ); if(ret<0) { printk("dma_new_period: buffer could not be added for capture transfer\n"); return ret; } printk("dma_new_period: enable dma transfer\n"); ret = mxc_dma_enable(prtd->dma_ch_rx); if(ret<0) { printk("dma_new_period: cannot queue DMA buffer\ (%i)\n", ret); return ret; } } printk("dma_new_period: transfer enabled\n src_addr = %x\n dst_addr = %x\n num_of_byes = %d\n", (void *) dma_request.src_addr, (void *) dma_request.dst_addr, dma_request.num_of_bytes); prtd->tx_spin = 1; /* FGA little trick to retrieve DMA pos */ prtd->period++; prtd->period %= runtime->periods; } return ret; } /*! * This is a callback which will be called * when a TX transfer finishes. The call occurs * in interrupt context. * * @param dat pointer to the structure of the current stream. * */ static void audio_dma_irq(void *data, int error, unsigned int count) { struct snd_pcm_substream *substream; struct snd_pcm_runtime *runtime; struct mx27_runtime_data *prtd; unsigned int dma_size; unsigned int previous_period; unsigned int offset; substream = data; runtime = substream->runtime; prtd = runtime->private_data; previous_period = prtd->periods; dma_size = frames_to_bytes(runtime, runtime->period_size); offset = dma_size * previous_period; prtd->tx_spin = 0; prtd->periods++; prtd->periods %= runtime->periods; // printk("audio_dma_irq: irq per %d offset %x\n", prtd->periods, offset); /* * Give back to the CPU the access to the non cached memory */ // if(substream->stream == SNDRV_PCM_STREAM_PLAYBACK) // dma_unmap_single(NULL, runtime->dma_addr + offset, dma_size, // DMA_TO_DEVICE); // else // dma_unmap_single(NULL, runtime->dma_addr + offset, dma_size, // DMA_FROM_DEVICE); /* * If we are getting a callback for an active stream then we inform * the PCM middle layer we've finished a period */ if (prtd->active) snd_pcm_period_elapsed(substream); /* * Trig next DMA transfer */ dma_new_period(substream); } /*! * This function configures the hardware to allow audio * playback operations. It is called by ALSA framework. * * @param substream pointer to the structure of the current stream. * * @return 0 on success, -1 otherwise. */ static int snd_mxc_prepare(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct mx27_runtime_data *prtd = runtime->private_data; prtd->period = 0; prtd->periods = 0; return 0; } static int mxc_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { struct snd_pcm_runtime *runtime = substream->runtime; struct mx27_runtime_data *prtd = runtime->private_data; int ret; if((ret=snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params))) < 0) { printk("imx27-pcm: failed to malloc pcm pages\n"); return ret; } printk("mxc_pcm_hw_params: snd_pcm_lib_malloc pages has return OK\n"); printk("IMX27: snd_imx27_audio_hw_params runtime->dma_addr 0x(%x)\n", (unsigned int)runtime->dma_addr); printk("IMX27: snd_imx27_audio_hw_params runtime->dma_area 0x(%x)\n", (unsigned int)runtime->dma_area); printk("IMX27: snd_imx27_audio_hw_params runtime->dma_bytes 0x(%x)\n", (unsigned int)runtime->dma_bytes); return ret; } static int mxc_pcm_hw_free(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct mx27_runtime_data *prtd = runtime->private_data; mxc_dma_free(prtd->dma_ch_tx); mxc_dma_free(prtd->dma_ch_rx); snd_pcm_lib_free_pages(substream); return 0; } static int mxc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { struct mx27_runtime_data *prtd = substream->runtime->private_data; int ret = 0; switch (cmd) { case SNDRV_PCM_TRIGGER_START: prtd->tx_spin = 0; /* requested stream startup */ prtd->active = 1; printk("mxc_pcm_trigger: starting dma_new_period\n"); ret = dma_new_period(substream); break; case SNDRV_PCM_TRIGGER_STOP: /* requested stream shutdown */ printk("mxc_pcm_trigger: stopping dma transfer\n"); ret = audio_stop_dma(substream); break; default: ret = -EINVAL; break; } return ret; } static snd_pcm_uframes_t mxc_pcm_pointer(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct mx27_runtime_data *prtd = runtime->private_data; unsigned int offset = 0; printk("mxc_pcm_pointer:\n runtime->period_size=%d\n prtd->periods=%d\n runtime->buffer_size=%d\n prtd->tx_spin=%d\n", runtime->period_size, prtd->periods, runtime->buffer_size, prtd->tx_spin); /* tx_spin value is used here to check if a transfer is active */ if (prtd->tx_spin){ offset = (runtime->period_size * (prtd->periods)) + (runtime->period_size >> 1); if (offset >= runtime->buffer_size) offset = runtime->period_size >> 1; } else { offset = (runtime->period_size * (prtd->periods)); if (offset >= runtime->buffer_size) offset = 0; } printk("mxc_pcm_pointer: pointer offset %x\n", offset); return offset; } //Alocation of substream->runtime->private_data static int mxc_pcm_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct mx27_runtime_data *prtd; int ret; snd_soc_set_runtime_hwparams(substream, &mx27_pcm_hardware); //?? if ((ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) return ret; if((prtd = kzalloc(sizeof(struct mx27_runtime_data), GFP_KERNEL)) == NULL) { ret = -ENOMEM; goto out; } runtime->private_data = prtd; if(substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { #ifdef CONFIG_SND_SOC_MX27_SSI1 printk("mxc_pcm_open: Requesting MXC_DMA_SSI1_16BIT_TX0 dma channel\n"); prtd->dma_ch_tx = mxc_dma_request(MXC_DMA_SSI1_16BIT_TX0, "ALSA TX DMA"); //MXC_DMA_TEST_RAM2RAM #else prtd->dma_ch_tx = mxc_dma_request(MXC_DMA_SSI2_16BIT_TX0, "ALSA TX DMA"); #endif if(prtd->dma_ch_tx < 0) { printk("error requesting a write(dma to device) dma channel\n"); return ret; } //Set dma callback printk("mxc_pcm_open: Setting tx dma callback function\n"); ret = mxc_dma_callback_set(prtd->dma_ch_tx, (mxc_dma_callback_t) audio_dma_irq, (void *)substream); if(ret<0) { printk("error setting dma callback function\n"); return ret; } return 0; } else { #ifdef CONFIG_SND_SOC_MX27_SSI1 printk("mxc_pcm_open: Requesting MXC_DMA_SSI1_16BIT_RX0 dma channel\n"); prtd->dma_ch_rx = mxc_dma_request(MXC_DMA_SSI1_16BIT_RX0, "ALSA RX DMA"); #else prtd->dma_ch_rx = mxc_dma_request(MXC_DMA_SSI2_16BIT_RX0, "ALSA RX DMA"); #endif if(prtd->dma_ch_rx < 0) { printk("error requesting a read(dma from device) dma channel\n"); return ret; } //Set dma callback printk("mxc_pcm_open: Setting rx dma callback function\n"); if(mxc_dma_callback_set(prtd->dma_ch_rx, (mxc_dma_callback_t) audio_dma_irq, (void *)substream)<0) printk("error setting dma callback function\n"); return 0; } out: return ret; } static int mxc_pcm_close(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct mxc_runtime_data *prtd = runtime->private_data; kfree(prtd); return 0; } static int mxc_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma) { struct snd_pcm_runtime *runtime = substream->runtime; return dma_mmap_writecombine(substream->pcm->card->dev, vma, runtime->dma_area, runtime->dma_addr, runtime->dma_bytes); } struct snd_pcm_ops mxc_pcm_ops = { .open = mxc_pcm_open, .close = mxc_pcm_close, .ioctl = snd_pcm_lib_ioctl, .hw_params = mxc_pcm_hw_params, .hw_free = mxc_pcm_hw_free, .prepare = snd_mxc_prepare, .trigger = mxc_pcm_trigger, .pointer = mxc_pcm_pointer, .mmap = mxc_pcm_mmap, }; static u64 mxc_pcm_dmamask = 0xffffffff; static int imx31_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) { struct snd_pcm_substream *substream = pcm->streams[stream].substream; struct snd_dma_buffer *buf = &substream->dma_buffer; size_t size = mx27_pcm_hardware.buffer_bytes_max; buf->dev.type = SNDRV_DMA_TYPE_DEV; buf->dev.dev = pcm->card->dev; buf->private_data = NULL; //Reserve uncached-buffered memory area for DMA buf->area = dma_alloc_writecombine(pcm->card->dev, size, &buf->addr, GFP_KERNEL); printk("imx27_pcm: preallocate_dma_buffer: area=%p, addr=%p, size=%d\n", (void *) buf->area, (void *) buf->addr, size); if (!buf->area) return -ENOMEM; buf->bytes = size; return 0; } static void imx31_pcm_free_dma_buffers(struct snd_pcm *pcm) { struct snd_pcm_substream *substream; struct snd_dma_buffer *buf; int stream; for (stream = 0; stream < 2; stream++) { substream = pcm->streams[stream].substream; if (!substream) continue; buf = &substream->dma_buffer; if (!buf->area) continue; dma_free_writecombine(pcm->card->dev, buf->bytes, buf->area, buf->addr); buf->area = NULL; } } int mxc_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) { int ret = 0; if (!card->dev->dma_mask) card->dev->dma_mask = &mxc_pcm_dmamask; if (!card->dev->coherent_dma_mask) card->dev->coherent_dma_mask = 0xffffffff; if (dai->playback.channels_min) { ret = imx31_pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK); printk("mxc_pcm_new: preallocate playback buffer\n"); if (ret) goto out; } if (dai->capture.channels_min) { ret = imx31_pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_CAPTURE); printk("mxc_pcm_new: preallocate capture buffer\n"); if (ret) goto out; } out: return ret; } struct snd_soc_platform imx27_soc_platform = { .name = "mxc-audio", .pcm_ops = &mxc_pcm_ops, .pcm_new = mxc_pcm_new, .pcm_free = imx31_pcm_free_dma_buffers, }; EXPORT_SYMBOL_GPL(imx27_soc_platform); MODULE_AUTHOR("Javier Martin"); MODULE_DESCRIPTION("Freescale i.MX27 PCM DMA module"); MODULE_LICENSE("GPL"); /* * mx27-pcm.h :- ASoC platform header for Freescale i.MX27 * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as * published by the Free Software Foundation. */ #ifndef _MXC_PCM_H #define _MXC_PCM_H #include <asm/arch/dma.h> /* AUDMUX regs definition - MOVE to asm/arch when stable #define AUDMUX_IO_BASE_ADDR IO_ADDRESS(AUDMUX_BASE_ADDR) #define DAM_PTCR1 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x00))) #define DAM_PDCR1 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x04))) #define DAM_PTCR2 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x08))) #define DAM_PDCR2 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x0C))) #define DAM_PTCR3 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x10))) #define DAM_PDCR3 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x14))) #define DAM_PTCR4 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x18))) #define DAM_PDCR4 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x1C))) #define DAM_PTCR5 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x20))) #define DAM_PDCR5 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x24))) #define DAM_PTCR6 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x28))) #define DAM_PDCR6 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x2C))) #define DAM_PTCR7 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x30))) #define DAM_PDCR7 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x34))) #define DAM_CNMCR (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x38))) #define DAM_PTCR(a) (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + a*8))) #define DAM_PDCR(a) (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 4 + a*8))) #define AUDMUX_PTCR_TFSDIR (1 << 31) #define AUDMUX_PTCR_TFSSEL(x, y) ((x << 30) | (((y - 1) & 0x7) << 27)) #define AUDMUX_PTCR_TCLKDIR (1 << 26) #define AUDMUX_PTCR_TCSEL(x, y) ((x << 25) | (((y - 1) & 0x7) << 22)) #define AUDMUX_PTCR_RFSDIR (1 << 21) #define AUDMUX_PTCR_RFSSEL(x, y) ((x << 20) | (((y - 1) & 0x7) << 17)) #define AUDMUX_PTCR_RCLKDIR (1 << 16) #define AUDMUX_PTCR_RCSEL(x, y) ((x << 15) | (((y - 1) & 0x7) << 12)) #define AUDMUX_PTCR_SYN (1 << 11) #define AUDMUX_FROM_TXFS 0 #define AUDMUX_FROM_RXFS 1 #define AUDMUX_PDCR_RXDSEL(x) (((x - 1) & 0x7) << 13) #define AUDMUX_PDCR_TXDXEN (1 << 12) #define AUDMUX_PDCR_MODE(x) (((x) & 0x3) << 8) #define AUDMUX_PDCR_INNMASK(x) (((x) & 0xff) << 0) #define AUDMUX_CNMCR_CEN (1 << 18) #define AUDMUX_CNMCR_FSPOL (1 << 17) #define AUDMUX_CNMCR_CLKPOL (1 << 16) #define AUDMUX_CNMCR_CNTHI(x) (((x) & 0xff) << 8) #define AUDMUX_CNMCR_CNTLOW(x) (((x) & 0xff) << 0) */ /* Previous defines are only valid for imx31 */ #define AUDMUX_IO_BASE_ADDR IO_ADDRESS(AUDMUX_BASE_ADDR) #define DAM_HPCR1 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x00))) #define DAM_HPCR2 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x04))) #define DAM_HPCR3 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x08))) #define DAM_PPCR1 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x10))) #define DAM_PPCR2 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x14))) #define DAM_PPCR3 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x1C))) #define AUDMUX_HPCR_TFSDIR (1 << 31) #define AUDMUX_HPCR_TCLKDIR (1 << 30) #define AUDMUX_HPCR_TFCSEL(x) (((x) & 0xff) << 26) #define AUDMUX_HPCR_RXDSEL(x) (((x) & 0x7) << 13) #define AUDMUX_HPCR_SYN (1 << 12) #define AUDMUX_PPCR_TFSDIR (1 << 31) #define AUDMUX_PPCR_TCLKDIR (1 << 30) #define AUDMUX_PPCR_TFCSEL(x) (((x) & 0xff) << 26) #define AUDMUX_PPCR_RXDSEL(x) (((x) & 0x7) << 13) #define AUDMUX_PPCR_SYN (1 << 12) struct mx27_pcm_dma_params { char *name; /* stream identifier */ mxc_dma_requestbuf_t buf_params; /*DMA buffer params */ }; extern struct snd_soc_dai mxc_ssi_dai[3]; /* platform data */ extern struct snd_soc_platform imx27_soc_platform; #endif

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[alsa-devel] General protection fault when unloading snd_hda_intel on 2.6.30-rc1+
by Carlos R. Mafra 15 Apr '09

15 Apr '09

If I try on my Vaio VGN-FZ240E, using the latest kernel 2.6.30-rc1-00191-gd848223 modprobe -r snd_hda_intel I get the following trace (copied from dmesg): general protection fault: 0000 [#1] SMP last sysfs file: /sys/devices/system/cpu/cpu0/cpufreq/scaling_cur_freq CPU 1 Modules linked in: nvram uvcvideo videodev v4l1_compat v4l2_compat_ioctl32 snd_hda_codec_idt usbhid snd_hda_intel(-) snd_hda_codec snd_hwdep snd_pcm sr_mod sky2 i2c_i801 iwlagn snd_timer sg snd_page_alloc evdev ata_piix ahci libata sd_mod scsi_mod uhci_hcd ohci_hcd ehci_hcd usbcore [last unloaded: scsi_wait_scan] Pid: 9063, comm: modprobe Not tainted 2.6.30-rc1-00191-gd848223 #48 VGN-FZ240E RIP: 0010:[<ffffffff8044bf90>] [<ffffffff8044bf90>] input_event+0x40/0xa0 RSP: 0018:ffff88006f283a78 EFLAGS: 00010293 RAX: 0000000000000008 RBX: 0000000000000003 RCX: 0000000000000000 RDX: 0000000000000007 RSI: 0000000000000005 RDI: 697665646632785c RBP: ffff88006f283aa8 R08: 0000000004d6f76c R09: 0000000000000001 R10: 0000000000000000 R11: 0000000000000001 R12: ffffffff805bb97c R13: 0000000000000005 R14: 697665646632785c R15: 0000000000000000 FS: 00007fc6c1e1c6f0(0000) GS:ffff880001028000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 000000008005003b CR2: 000000000071bf40 CR3: 000000007e2d4000 CR4: 00000000000006e0 DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 DR3: 0000000000000000 DR6: 00000000ffff0ff0 DR7: 0000000000000400 Process modprobe (pid: 9063, threadinfo ffff88006f282000, task ffff88006f206840) Stack: ffff88006f283aa8 0000000000000003 ffffffff805bb97c ffff88007e811cc0 0000000000000001 0000000000000000 ffff88006f283ae8 ffffffff80475612 ffff88007f241ae0 0000000000000002 ffff88007d5eb010 0000000000000000 Call Trace: [<ffffffff80475612>] snd_jack_report+0x92/0xa0 [<ffffffffa0160047>] stac92xx_unsol_event+0x2e7/0x5e0 [snd_hda_codec_idt] [<ffffffffa015f528>] stac_issue_unsol_event+0x58/0x70 [snd_hda_codec_idt] [<ffffffffa0160796>] stac92xx_init+0x186/0x660 [snd_hda_codec_idt] [<ffffffffa0160c90>] stac92xx_resume+0x20/0x60 [snd_hda_codec_idt] [<ffffffffa013c973>] hda_call_codec_resume+0x53/0x90 [snd_hda_codec] [<ffffffffa013ca50>] snd_hda_power_up+0x40/0x70 [snd_hda_codec] [<ffffffffa013c2b5>] snd_hda_codec_write+0x45/0xa0 [snd_hda_codec] [<ffffffffa013c37a>] restore_pincfgs+0x6a/0xa0 [snd_hda_codec] [<ffffffffa013ed75>] snd_hda_codec_free+0x35/0x120 [snd_hda_codec] [<ffffffffa01403f8>] snd_hda_bus_free+0x68/0xb0 [snd_hda_codec] [<ffffffffa01405e4>] snd_hda_bus_dev_free+0x14/0x20 [snd_hda_codec] [<ffffffff80473e61>] snd_device_free+0x71/0xd0 [<ffffffff80473f24>] snd_device_free_all+0x64/0x70 [<ffffffff8046e16a>] snd_card_do_free+0x3a/0xd0 [<ffffffff8046ee39>] snd_card_free+0xa9/0xd0 [<ffffffffa015031e>] azx_remove+0x18/0x2a [snd_hda_intel] [<ffffffff8036506f>] pci_device_remove+0x2f/0x60 [<ffffffff8040ec8c>] __device_release_driver+0x6c/0xb0 [<ffffffff8040ed78>] driver_detach+0xa8/0xb0 [<ffffffff8040de48>] bus_remove_driver+0x98/0xd0 [<ffffffff8040f357>] driver_unregister+0x47/0x60 [<ffffffff8036530c>] pci_unregister_driver+0x4c/0xc0 [<ffffffffa0150304>] alsa_card_azx_exit+0x10/0x12 [snd_hda_intel] [<ffffffff8026787b>] sys_delete_module+0x18b/0x240 [<ffffffff80257be9>] ? up_write+0x9/0x10 [<ffffffff8020be2b>] system_call_fastpath+0x16/0x1b Code: 89 5d d8 4c 89 65 e0 49 89 fe 41 89 f5 41 89 cf 83 fe 1f 76 16 48 8b 5d d8 4c 8b 65 e0 4c 8b 6d e8 4c 8b 75 f0 4c 8b 7d f8 c9 c3 <0f> a3 77 20 19 c0 85 c0 74 e0 48 8d 9f f8 06 00 00 89 55 d0 48 RIP [<ffffffff8044bf90>] input_event+0x40/0xa0 RSP <ffff88006f283a78> ---[ end trace 7fa0517f56c40085 ]--- After that, 'cat /proc/modules' contains this line: snd_hda_intel 28648 0 - Unloading 0xffffffffa014e000 and the HDA driver never finishes that advertised "Unloading". It also stops working, [mafra@Pilar:linux-2.6]$ alsamixer alsamixer: function snd_ctl_open failed for default: No such file or directory The complete dmesg is available at http://www.aei.mpg.de/~crmafra/dmesg_hda_intel.txt and from lspci -vvnn I see this 00:1b.0 Audio device [0403]: Intel Corporation 82801H (ICH8 Family) HD Audio Controller [8086:284b] (rev 03) Subsystem: Sony Corporation Vaio VGN-FZ260E [104d:9005] Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 22 Region 0: Memory at fc400000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Address: 0000000000000000 Data: 0000 Capabilities: [70] Express (v1) Root Complex Integrated Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s <64ns, L1 <1us ExtTag- RBE- FLReset- DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr+ TransPend- LnkCap: Port #0, Speed unknown, Width x0, ASPM unknown, Latency L0 <64ns, L1 <1us ClockPM- Suprise- LLActRep- BwNot- LnkCtl: ASPM Disabled; Disabled- Retrain- CommClk- ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed unknown, Width x0, TrErr- Train- SlotClk- DLActive- BWMgmt- ABWMgmt- Capabilities: [100] Virtual Channel <?> Capabilities: [130] Root Complex Link <?> Kernel modules: snd-hda-intel When the audio is working alsamixer says it is using Sigmatel STAC9872AK. Anything else I can do to help? -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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[alsa-devel] getting HDAudio IDs
by Scott Serr 15 Apr '09

15 Apr '09

Hi All, I've found a great way to get HDAudio vendor/device IDs when audio drivers are loaded: /proc/asound/card?/codec? But to remake my particular (barebones) distro with with audio support, it will be repetitive and time consuming. I'm wondering what would be involved in making a user space utility that would return the HDAudio ID? A guy on #alsa came up with some good ideas. Potentially following the lead of others directly accessing the hardware from user space http://samba.org/picturebook/ but using libpci instead. Maybe we can use "single cmd mode" so we don't need interrupts. My original idea: It might be that a smart developer could grab a bunch of alsa/hdaudio driver source and torture it into userland.... just to get these IDs. Should I quit now? Is it doable? Are there people here with the skills/experience to do this? BTW, in the kernel config: #CONFIG_SOUND is not set Thanks, Scott

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