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

[alsa-devel] snd-hda-intel & ALC268 codec
by Meelis Roos 28 Jul '07

28 Jul '07

Hi, I have a laptop with Realtek ALC268 codec connected to Intel HDA controller. The laptop is Compal OEM, specific module currently unknown (it's assembled by a local reseller as Ordi Enduro 7300BF). 2.6.22 didn't much understand the codec. 2.6.23-rc1 seems to know much more about it and it almost works - but not quite. On bootup, the following line is in dmesg: hda_codec: Unknown model for ALC268, trying auto-probe from BIOS... It initializes fine but the sound is so quiet that I can only hear it with active speakers when I turn the volume to max. No mixer sliders can make it louder. Additionally, there are some strange mixer switches like Off-hook and Caller ID. The only strings identifying the base board are from dmidecode: Handle 0x0002, DMI type 2, 8 bytes Base Board Information Manufacturer: COMPAL Product Name: IFT00 Version: REFERENCE Serial Number: N/A Additionally, the system has identified most PCI devices by subsystem vendor & ID 14c0:0023, like this: 00:1b.0 0403: 8086:284b (rev 03) Subsystem: 14c0:0023 Flags: bus master, fast devsel, latency 0, IRQ 23 Memory at f8400000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Capabilities: [70] Express Unknown type IRQ 0 ls -lR /proc/asound/Intel/: /proc/asound/Intel/: total 0 -r--r--r-- 1 root root 0 2007-07-28 17:26 codec#0 -r--r--r-- 1 root root 0 2007-07-28 17:26 codec#1 -r--r--r-- 1 root root 0 2007-07-28 17:26 id -rw-r--r-- 1 root root 0 2007-07-28 17:26 oss_mixer dr-xr-xr-x 4 root root 0 2007-07-28 17:26 pcm0c dr-xr-xr-x 3 root root 0 2007-07-28 17:26 pcm0p dr-xr-xr-x 4 root root 0 2007-07-28 17:26 pcm2c dr-xr-xr-x 3 root root 0 2007-07-28 17:26 pcm6c dr-xr-xr-x 3 root root 0 2007-07-28 17:26 pcm6p /proc/asound/Intel/pcm0c: total 0 -r--r--r-- 1 root root 0 2007-07-28 17:26 info -rw-r--r-- 1 root root 0 2007-07-28 17:26 oss dr-xr-xr-x 2 root root 0 2007-07-28 17:26 sub0 dr-xr-xr-x 2 root root 0 2007-07-28 17:26 sub1 /proc/asound/Intel/pcm0c/sub0: total 0 -r--r--r-- 1 root root 0 2007-07-28 17:26 hw_params -r--r--r-- 1 root root 0 2007-07-28 17:26 info -rw-r--r-- 1 root root 0 2007-07-28 17:26 prealloc -r--r--r-- 1 root root 0 2007-07-28 17:26 prealloc_max -r--r--r-- 1 root root 0 2007-07-28 17:26 status -r--r--r-- 1 root root 0 2007-07-28 17:26 sw_params /proc/asound/Intel/pcm0c/sub1: total 0 -r--r--r-- 1 root root 0 2007-07-28 17:26 hw_params -r--r--r-- 1 root root 0 2007-07-28 17:26 info -rw-r--r-- 1 root root 0 2007-07-28 17:26 prealloc -r--r--r-- 1 root root 0 2007-07-28 17:26 prealloc_max -r--r--r-- 1 root root 0 2007-07-28 17:26 status -r--r--r-- 1 root root 0 2007-07-28 17:26 sw_params /proc/asound/Intel/pcm0p: total 0 -r--r--r-- 1 root root 0 2007-07-28 17:26 info -rw-r--r-- 1 root root 0 2007-07-28 17:26 oss dr-xr-xr-x 2 root root 0 2007-07-28 17:26 sub0 /proc/asound/Intel/pcm0p/sub0: total 0 -r--r--r-- 1 root root 0 2007-07-28 17:26 hw_params -r--r--r-- 1 root root 0 2007-07-28 17:26 info -rw-r--r-- 1 root root 0 2007-07-28 17:26 prealloc -r--r--r-- 1 root root 0 2007-07-28 17:26 prealloc_max -r--r--r-- 1 root root 0 2007-07-28 17:26 status -r--r--r-- 1 root root 0 2007-07-28 17:26 sw_params /proc/asound/Intel/pcm2c: total 0 -r--r--r-- 1 root root 0 2007-07-28 17:26 info dr-xr-xr-x 2 root root 0 2007-07-28 17:26 sub0 dr-xr-xr-x 2 root root 0 2007-07-28 17:26 sub1 /proc/asound/Intel/pcm2c/sub0: total 0 -r--r--r-- 1 root root 0 2007-07-28 17:26 hw_params -r--r--r-- 1 root root 0 2007-07-28 17:26 info -rw-r--r-- 1 root root 0 2007-07-28 17:26 prealloc -r--r--r-- 1 root root 0 2007-07-28 17:26 prealloc_max -r--r--r-- 1 root root 0 2007-07-28 17:26 status -r--r--r-- 1 root root 0 2007-07-28 17:26 sw_params /proc/asound/Intel/pcm2c/sub1: total 0 -r--r--r-- 1 root root 0 2007-07-28 17:26 hw_params -r--r--r-- 1 root root 0 2007-07-28 17:26 info -rw-r--r-- 1 root root 0 2007-07-28 17:26 prealloc -r--r--r-- 1 root root 0 2007-07-28 17:26 prealloc_max -r--r--r-- 1 root root 0 2007-07-28 17:26 status -r--r--r-- 1 root root 0 2007-07-28 17:26 sw_params /proc/asound/Intel/pcm6c: total 0 -r--r--r-- 1 root root 0 2007-07-28 17:26 info dr-xr-xr-x 2 root root 0 2007-07-28 17:26 sub0 /proc/asound/Intel/pcm6c/sub0: total 0 -r--r--r-- 1 root root 0 2007-07-28 17:26 hw_params -r--r--r-- 1 root root 0 2007-07-28 17:26 info -rw-r--r-- 1 root root 0 2007-07-28 17:26 prealloc -r--r--r-- 1 root root 0 2007-07-28 17:26 prealloc_max -r--r--r-- 1 root root 0 2007-07-28 17:26 status -r--r--r-- 1 root root 0 2007-07-28 17:26 sw_params /proc/asound/Intel/pcm6p: total 0 -r--r--r-- 1 root root 0 2007-07-28 17:26 info dr-xr-xr-x 2 root root 0 2007-07-28 17:26 sub0 /proc/asound/Intel/pcm6p/sub0: total 0 -r--r--r-- 1 root root 0 2007-07-28 17:26 hw_params -r--r--r-- 1 root root 0 2007-07-28 17:26 info -rw-r--r-- 1 root root 0 2007-07-28 17:26 prealloc -r--r--r-- 1 root root 0 2007-07-28 17:26 prealloc_max -r--r--r-- 1 root root 0 2007-07-28 17:26 status -r--r--r-- 1 root root 0 2007-07-28 17:26 sw_params amixer controls gives the following output: numid=4,iface=MIXER,name='Headphone Playback Switch' ; type=BOOLEAN,access=rw------,values=2 : values=on,on numid=3,iface=MIXER,name='Headphone Playback Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=64,step=0 : values=45,45 | dBscale-min=-64.00dB,step=1.00dB,mute=0 numid=13,iface=MIXER,name='PCM Playback Volume' ; type=INTEGER,access=rw---RW-,values=2,min=0,max=255,step=0 : values=255,255 | dBscale-min=-51.00dB,step=0.20dB,mute=0 numid=2,iface=MIXER,name='Front Playback Switch' ; type=BOOLEAN,access=rw------,values=2 : values=on,on numid=1,iface=MIXER,name='Front Playback Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=64,step=0 : values=52,52 | dBscale-min=-64.00dB,step=1.00dB,mute=0 numid=6,iface=MIXER,name='Capture Switch' ; type=BOOLEAN,access=rw------,values=2 : values=on,on numid=8,iface=MIXER,name='Capture Switch',index=1 ; type=BOOLEAN,access=rw------,values=2 : values=on,on numid=5,iface=MIXER,name='Capture Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=31,step=0 : values=1,1 | dBscale-min=-16.50dB,step=1.50dB,mute=0 numid=7,iface=MIXER,name='Capture Volume',index=1 ; type=INTEGER,access=rw---R--,values=2,min=0,max=31,step=0 : values=1,1 | dBscale-min=-16.50dB,step=1.50dB,mute=0 numid=12,iface=MIXER,name='Caller ID Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=14,iface=MIXER,name='Digital Capture Volume' ; type=INTEGER,access=rw---RW-,values=2,min=0,max=120,step=0 : values=60,60 | dBscale-min=-30.00dB,step=0.50dB,mute=0 numid=9,iface=MIXER,name='Input Source' ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'Mic' ; Item #1 'Front Mic' : values=0 numid=10,iface=MIXER,name='Input Source',index=1 ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'Mic' ; Item #1 'Front Mic' : values=0 numid=11,iface=MIXER,name='Off-hook Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off -- Meelis Roos (mroos(a)linux.ee)

1 0

[alsa-devel] Mixing.
by Vasily Khoruzhick 28 Jul '07

28 Jul '07

Hi 2 all. I've just read many discussions about kernelspace and userspace sound mixing. Both sides have their arguments, etc. But why not to implement both? If you don't want to rewrite something there's easy way: just create a driver for dummy sound card, that will perform kernelspace mixing and will use default sound card for sound output. Kernelspace mixing will be very usefull for example for oss kernelspace oss emulation (as you know, userspace one doesn't work for huge amount of apps), or for dumb apps, that open hw:0 device for sound and have no option to change output device. Please, think about your users ;) Let them decide which mixing they wants Cheers, Vasily.

1 0

[alsa-devel] ALSA support for Teak DSP used in NXP LPC3180
by Jon Smirl 27 Jul '07

27 Jul '07

Is there an Alsa driver for the Teak DSP used in some ARM chips like the NXP LPC3180? I've been googling around and can't locate one. I have a GPL'd OSS driver if anyone is interested. It's 3,581 lines. -- Jon Smirl jonsmirl(a)gmail.com

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[alsa-devel] Intel HDA Driver - Bug
by Steven King 27 Jul '07

27 Jul '07

Hi, I have been trying to get the sound on my Dell Inspiron 1720 laptop running under Debian Etch (STAC 9205). Originally, I could only get sound through the main speakers, there was no headphone sound or mic. I downloaded the latest alsa-drivers package (1.0.14), compiled it and installed it. I received the following messages (and the device would not register itself): hda_intel: azx_get_response timeout, switching to polling mode... hda_intel: azx_get_response timeout, switching to single_cmd mode... I started digging into the driver code and found an issue with one of the routines going outside an array. The offending code is (all in patch_sigmatel): static void stac92xx_set_config_regs(struct hda_codec *codec) { // ... for (i = 0; i < spec->num_pins; i++) { // ... } And in: static int patch_stac9205(struct hda_codec *codec) { // .. spec->num_pins = 14 ; //ERROR // .. } However, there are only 12 pins with 12 corresponding NIDs and default configuration so in stac92xx_set_config_regs it accesses spec->pin_configs[i] and spec->pin_nids[i] outside of bounds. Fix: Add a define (STAC_9205_MAX_PINS = 12) and use for array defines and num_pins. The driver loads now, but I still only have sound out the main speaker. If I get rid of the configuration sent to the chipset altogether. I get sound through Headphone and Speakers (with detect working on HP), but still no MIC. How can I add support for my Dell as I don't know the configuration to use or am I missing something here? Thanks, Steve

2 4

[alsa-devel] BT87x ALSA driver patches
by Trent Piepho 27 Jul '07

27 Jul '07

On Mon, 23 Jul 2007, Mauro Carvalho Chehab wrote: > Em Sáb, 2007-07-21 às 17:36 -0700, Trent Piepho escreveu: > > 01/01: bttv: Fix Viewcast Osprey 440 support > > http://linuxtv.org/hg/~tap/v4l-dvb?cmd=changeset;node=8e72fe5675b3 > > > > I have two other changes that only affect the ALSA driver part. Should I > > send those to ALSA or to you? > > Please send to ALSA guys, c/c me. Ok, I'll do that. What about the two lines to bt87x.c that this patch added? I seemed logical to me that they should be part of the same patch at git as the changes to bttv-cards.c, but I suppose it's not 100% necessary to do so, the kernel will still compile and not crash. Do you plan to send this to git as one patch, or to strip out the ALSA change? I'm just wondering if I should send these two lines to the ALSA people as a separate patch, of if they will get them into their hg via a kernel-sync patch. My bt87x patches depend on this one, so somehow the ALSA hg needs to get this change before I can send them the next two. The problem really, is that having two divers for the same chip split into v4l-dvb and ALSA like this a pain. IMHO, it would make a lot more sense to put the bttv alsa driver into v4l-dvb, like the saa7134 alsa and oss and cx88 alsa drivers are. > > drivers/media/video/bt8xx/bttv-cards.c | 240 ++++++++++++++++++--------------- > > sound/pci/bt87x.c | 2 > > 2 files changed, 137 insertions(+), 105 deletions(-)

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[alsa-devel] another device for opl3sa2
by Paul Vojta 27 Jul '07

27 Jul '07

Folks: Here's a patch to add support for an (admittedly old) device: --- linux-2.6.21.5/sound/isa/opl3sa2.c.orig 2007-06-11 11:37:06.000000000 -0700 +++ linux-2.6.21.5/sound/isa/opl3sa2.c 2007-06-30 23:29:43.000000000 -0700 @@ -165,6 +165,8 @@ { .id = "YMH0801", .devs = { { "YMH0021" } } }, /* NeoMagic MagicWave 3DX */ { .id = "NMX2200", .devs = { { "YMH2210" } } }, + /* NeoMagic MagicWave 3D */ + { .id = "NMX2200", .devs = { { "NMX2210" } } }, /* --- */ { .id = "" } /* end */ }; (There may be a bug in the initialization code, though -- it doesn't work the first time after booting, just the second and subsequent times. For example, the first time I try running "aplay file.wav" it'll sit there silently instead of playing the file, but if I type "aplay file.wav" again then it'll play.) I don't know whether snd_opl3sa2_pnpbiosids[] should also be changed. Also, why does nm256.c direct users to the cs423x drivers and not opl3sa2? If it had mentioned opl3sa2 it would have saved me many hours. Sincerely, Paul Vojta vojta(a)math.berkeley.edu

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[alsa-devel] [PATCH] snd_usb_caiaq_input_free() fix
by Karsten Wiese 27 Jul '07

27 Jul '07

input_free_device()'s comment says: input_free_device() should only be used if input_register_device() was not called yet or if it failed. Once device was registered use input_unregister_device() and memory will be freed once last refrence to the device is dropped. Signed-off-by: Karsten Wiese <fzu(a)wemgehoertderstaat.de> --- diff -pur rt9/sound/usb/caiaq/caiaq-input.c rt9-kw/sound/usb/caiaq/caiaq-input.c --- rt9/sound/usb/caiaq/caiaq-input.c 2007-07-10 20:56:30.000000000 +0200 +++ rt9-kw/sound/usb/caiaq/caiaq-input.c 2007-07-27 11:45:52.000000000 +0200 @@ -238,7 +238,6 @@ void snd_usb_caiaq_input_free(struct snd return; input_unregister_device(dev->input_dev); - input_free_device(dev->input_dev); dev->input_dev = NULL; }

2 1

[alsa-devel] [PATCH] snd_usb_caiaq_input_free() fix
by Karsten Wiese 27 Jul '07

27 Jul '07

input_free_device()'s comment says: input_free_device() should only be used if input_register_device() was not called yet or if it failed. Once device was registered use input_unregister_device() and memory will be freed once last refrence to the device is dropped. Signed-off-by: Karsten Wiese <fzu(a)wemgehoertderstaat.de> --- diff -pur rt9/sound/usb/caiaq/caiaq-input.c rt9-kw/sound/usb/caiaq/caiaq-input.c --- rt9/sound/usb/caiaq/caiaq-input.c 2007-07-10 20:56:30.000000000 +0200 +++ rt9-kw/sound/usb/caiaq/caiaq-input.c 2007-07-27 11:45:52.000000000 +0200 @@ -238,7 +238,6 @@ void snd_usb_caiaq_input_free(struct snd return; input_unregister_device(dev->input_dev); - input_free_device(dev->input_dev); dev->input_dev = NULL; }

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[alsa-devel] Arecord not working
by Pharaoh . 27 Jul '07

27 Jul '07

Hi The arecord says following: arecord -f cd -t wav -c 2 -D dsnoop 16bit.raw ALSA lib confmisc.c:670:(snd_func_card_driver) cannot find card '0' ALSA lib conf.c:3500:(_snd_config_evaluate) function snd_func_card_driver returned error: No such device ALSA lib confmisc.c:391:(snd_func_concat) error evaluating strings ALSA lib conf.c:3500:(_snd_config_evaluate) function snd_func_concat returned error: No such device ALSA lib confmisc.c:1070:(snd_func_refer) error evaluating name ALSA lib conf.c:3500:(_snd_config_evaluate) function snd_func_refer returned error: No such device ALSA lib conf.c:3968:(snd_config_expand) Evaluate error: No such device ALSA lib pcm.c:2143:(snd_pcm_open_noupdate) Unknown PCM dsnoop arecord: main:550: audio open error: No such device #arecord -L arecord -L PCM list: hw { @args.0 CARD @args.1 DEV @args.2 SUBDEV @args.CARD { type string default { @func getenv vars { 0 ALSA_PCM_CARD 1 ALSA_CARD } default { @func refer name 'defaults.pcm.card' } } } @args.DEV { type integer default { @func igetenv vars { 0 ALSA_PCM_DEVICE } default { @func refer name 'defaults.pcm.device' } } } @args.SUBDEV { type integer default { @func refer name 'defaults.pcm.subdevice' } } type hw card $CARD device $DEV subdevice $SUBDEV } plughw { @args.0 CARD @args.1 DEV @args.2 SUBDEV @args.CARD { type string default { @func getenv vars { 0 ALSA_PCM_CARD 1 ALSA_CARD } default { @func refer name 'defaults.pcm.card' } } } @args.DEV { type integer default { @func igetenv vars { 0 ALSA_PCM_DEVICE } default { @func refer name 'defaults.pcm.device' } } } @args.SUBDEV { type integer default { @func refer name 'defaults.pcm.subdevice' } } type plug slave.pcm { type hw card $CARD device $DEV subdevice $SUBDEV } } plug { @args.0 SLAVE @args.SLAVE { type string } type plug slave.pcm $SLAVE } shm { @args.0 SOCKET @args.1 PCM @args.SOCKET { type string } @args.PCM { type string } type shm server $SOCKET pcm $PCM } tee { @args.0 SLAVE @args.1 FILE @args.2 FORMAT @args.SLAVE { type string } @args.FILE { type string } @args.FORMAT { type string default raw } type file slave.pcm $SLAVE file $FILE format $FORMAT } file { @args.0 FILE @args.1 FORMAT @args.FILE { type string } @args.FORMAT { type string default raw } type file slave.pcm null file $FILE format $FORMAT } null { type null } cards 'cards.pcm' front 'cards.pcm.front' rear 'cards.pcm.rear' center_lfe 'cards.pcm.center_lfe' side 'cards.pcm.side' surround40 'cards.pcm.surround40' surround41 'cards.pcm.surround41' surround50 'cards.pcm.surround50' surround51 'cards.pcm.surround51' surround71 'cards.pcm.surround71' iec958 'cards.pcm.iec958' spdif 'cards.pcm.iec958' modem 'cards.pcm.modem' phoneline 'cards.pcm.phoneline' dmix 'cards.pcm.dmix' dsnoop 'cards.pcm.dsnoop' default { type plug slave.pcm dmix } headset { @args.0 BDADDR @args.1 TIMEOUT @args.BDADDR { type string default '00:07:A4:5C:64:F0' } @args.TIMEOUT { type integer default 6000 } type sco bdaddr $BDADDR timeout $TIMEOUT } a2dpd { type a2dpd } I tried to incorporate dsnoop in this, but it doesnt seem to work. How should I merge it in the alsa.conf?

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[alsa-devel] Effect of CONFIG_SND_PCM_OSS_PLUGINS
by Adrian McMenamin 27 Jul '07

27 Jul '07

Unless CONFIG_SND_PCM_OSS_PLUGINS is set then with my driver the userspace (and OSS) app mpg123 doesn't work properly - not outputting any sound. Is this likely to be a problem with my driver or with the app? Adrian

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[alsa-devel] VT1618 driver
by Niels Gram Jeppesen 27 Jul '07

27 Jul '07

Hi, I have a VIA CN1000G mini-ITX board with a VT1618 that I would like to have up and running with sound next week. I see (https://bugtrack.alsa-project.org/wiki/wikka.php?wakka=VT1618&show_comments…) that there exist a still unreleased driver for the chip. If possible, I would like to play arround with that driver next week. Best regards and thanks in advance, Niels Gram Jeppesen

1 0

[alsa-devel] Intel HDA Driver - Bug
by Steven King 26 Jul '07

26 Jul '07

Hi, I have been trying to get the sound on my Dell Inspiron 1720 laptop running under Debian Etch (STAC 9205). Originally, I could only get sound through the main speakers, there was no headphone sound or mic. I downloaded the latest alsa-drivers package (1.0.14), compiled it and installed it. I received the following messages (and the device would not register itself): hda_intel: azx_get_response timeout, switching to polling mode... hda_intel: azx_get_response timeout, switching to single_cmd mode... I started digging into the driver code and found an issue with one of the routines going outside an array. The offending code is (all in patch_sigmatel): static void stac92xx_set_config_regs(struct hda_codec *codec) { // ... for (i = 0; i < spec->num_pins; i++) { // ... } And in: static int patch_stac9205(struct hda_codec *codec) { // .. spec->num_pins = 14 ; //ERROR // .. } However, there are only 12 pins with 12 corresponding NIDs and default configuration so in stac92xx_set_config_regs it accesses spec->pin_configs[i] and spec->pin_nids[i] outside of bounds. Fix: Add a define (STAC_9205_MAX_PINS = 12) and use for array defines and num_pins. The driver loads now, but I still only have sound out the main speaker. If I get rid of the configuration sent to the chipset altogether. I get sound through Headphone and Speakers (with detect working on HP), but still no MIC. How can I add support for my Dell as I don't know the configuration to use or am I missing something here? Thanks, Steve

1 0

[alsa-devel] [patch] asihpi: mmap and link streams
by Eliot Blennerhassett 26 Jul '07

26 Jul '07

Greetings, asihpi version 3.07.13 : Enable MMAP dma capable cards. Enable linked streams for cards that have hardware support for this. Add TLV for level and volume controls. Register asihpi class for HPI part of driver Add documentation, remove unused code. Signed-off-by: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> ---- Patch (4798 lines) is here: http://audioscience.com/internet/download/beta/alsa_driver3349_asihpi30713.… Matching firmware is here: http://audioscience.com/internet/download/firmware/dspbins30713.zip FYI: Original source package here: http://audioscience.com/internet/download/hpklinux-3.07.13.tar.bz2 asihpi.c | 1517 ++++++++++++++++++++++++++------------------------- Implement MMAP and linked streams. Merge some capture and playback functions Add dB TLV for level and volume controls hpi6000.c | 7 Clean hpi6205.c | 498 +++++++++------- Changes to allow attaching external DMA buffer (for ALSA) hpidebug.c | 67 +- hpidebug.h | 349 +++++------ Changes to support unicode hpimod.c | 283 ++------- Cleanup, add class registration hpimsgx.c | 50 + Add 2 new subsys functions hpios_linux_kernel.c | 36 - hpios_linux_kernel.h | 15 Minor debug changes. Add pci dev ptr to allow alsa to allocate dma buffer hpi.h | 54 - hpi56301.c | 1 hpidspcd.c | 2 hpifunc.c | 121 ++-- Documentation updates 13 files changed, 1506 insertions(+), 1494 deletions(-)

2 1

[alsa-devel] ALC262 - HDA-Intel
by Andy 26 Jul '07

26 Jul '07

I have a ALC262 hda-intel card which doesn't function quite correctly. Details can be found at https://bugs.launchpad.net/ubuntu/+source/linux-source-2.6.20/+bug/110145 I got bored of waiting for it to be fixed, and had a go myself. My knowledge of ALSA is very limited, but nevertheless it was really simple to add my device ids to the quirks list in the patch_realtek.c file - eg: ... SND_PCI_QUIRK(0x104d, 0x900e, "Sony ASSAMD", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x1f00, "Sony ASSAMD", ALC262_SONY_ASSAMD), /* My card... */ SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD), ... I found by trial and error that ALC262_SONY_ASSAMD almost works. Connecting headphones disables built-in speakers and unplugging them re-enables them. However, something does not get initialised until this is done. That is, every bootup using ALC262_SONY_ASSAMD, I must plug in and remove my headphones before the device is initialised properly. Until this is done, I get no sound and the little GUI mixer in XFCE doesn't show the list of supported outputs on the device. I'm posting this to the devel list as I'm more interested the code side of things rather than actually having working sound :) Thanks Andy. ___________________________________________________________ Yahoo! Mail is the world's favourite email. Don't settle for less, sign up for your free account today http://uk.rd.yahoo.com/evt=44106/*http://uk.docs.yahoo.com/mail/winter07.ht…

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[alsa-devel] [PATCH 1/2] Add SPI devices to ALSA Kconfig and Makefile
by Hans-Christian Egtvedt 26 Jul '07

26 Jul '07

This patch adds SPI devices in the ALSA diretory, including the Kconfig and Makefile. Signed-off-by: Hans-Christian Egtvedt <hcegtvedt(a)atmel.com> --- sound/Kconfig | 6 +++++- sound/Makefile | 3 ++- sound/spi/Kconfig | 6 ++++++ sound/spi/Makefile | 2 ++ 4 files changed, 15 insertions(+), 2 deletions(-) create mode 100644 sound/spi/Kconfig create mode 100644 sound/spi/Makefile diff --git a/sound/Kconfig b/sound/Kconfig index 9ea4738..9217739 100644 --- a/sound/Kconfig +++ b/sound/Kconfig @@ -63,9 +63,13 @@ source "sound/aoa/Kconfig" source "sound/arm/Kconfig" +if SPI +source "sound/spi/Kconfig" +endif + source "sound/mips/Kconfig" -# the following will depend on the order of config. +# tee following will depend on the order of config. # here assuming USB is defined before ALSA source "sound/usb/Kconfig" diff --git a/sound/Makefile b/sound/Makefile index b7c7fb7..581ea5e 100644 --- a/sound/Makefile +++ b/sound/Makefile @@ -5,7 +5,8 @@ obj-$(CONFIG_SOUND) += soundcore.o obj-$(CONFIG_SOUND_PRIME) += sound_firmware.o obj-$(CONFIG_SOUND_PRIME) += oss/ obj-$(CONFIG_DMASOUND) += oss/ -obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ synth/ usb/ sparc/ parisc/ pcmcia/ mips/ soc/ +obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ synth/ usb/ \ + sparc/ spi/ parisc/ pcmcia/ mips/ soc/ obj-$(CONFIG_SND_AOA) += aoa/ # This one must be compilable even if sound is configured out diff --git a/sound/spi/Kconfig b/sound/spi/Kconfig new file mode 100644 index 0000000..60fbb2a --- /dev/null +++ b/sound/spi/Kconfig @@ -0,0 +1,6 @@ +#SPI drivers + +menu "SPI devices" + depends on SND != n + +endmenu diff --git a/sound/spi/Makefile b/sound/spi/Makefile new file mode 100644 index 0000000..30d8ec9 --- /dev/null +++ b/sound/spi/Makefile @@ -0,0 +1,2 @@ +# Makefile for SPI drivers + -- 1.5.2.3

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[alsa-devel] How to get correct midi timings from ALSA using the library only
by Carlo Florendo 26 Jul '07

26 Jul '07

Hi, (1) I've written a command line MIDI sequencer for lightweight systems and am successful in making it work using the ALSA queue API. However, one drawback of the API is its lack of callback functions. I wish to be able to track events as they are drained by the queue. (2) I know and have successfully worked on a work around whereby the application itself subscribes to the output port so as to see events as they are played. However, I wish to be able to make the sequencer or player work without the use of the ALSA queue nor the workaround in (2). Here's the pseudo-code of the relevant MIDI player routine: for (i = 0; i < number_of_events; i++) { usleep(event[i].delta_time_in_microseconds); output_and_drain_event(event[i]); } This routine gives a non-bearable latency on 2.4 kernels but not so much on 2.6 kernels. How could I get the app to <u|nano>sleep() in the most accurate way in userspace without using the ALSA queue nor the extra subscription to an output port? Or, is there a drain or output routine that supports callbacks? If so, I will be grateful if you could point them out. I seem not to find any output callback routine under the docs. 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

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[alsa-devel] dmix and sounds less than 64kb
by Aaron "Caustik" Robinson 25 Jul '07

25 Jul '07

We have been having one heck of a time tracking down the cause of some ALSA audio playback issues. It appears that any time we play a very short sound (<64kb), either the sound doesn't play at all, or only a fragment of it plays. We haven't had much luck finding a solution to this issue, then I found this thread on the history of this email group: http://mailman.alsa-project.org/pipermail/alsa-devel/2007-April/000526.html Our problem is pretty much exactly like his issues relating to <64k sounds. My question is - has anybody else run into this problem? To the OP of the above thread, if you're still listening - did you ever find a solution to your problem? I'd be very interested to hear any help you can provide. Thanks much, caustik

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[alsa-devel] Testing play and record simultaneously.
by Pharaoh . 25 Jul '07

25 Jul '07

Very naive questions. 1. I want to test the recording using arecord, but how will I make sure that the data is actually being recorded? I mean is there any way of storing the data recorded, say in a file? Off course there must be some way, its just that I dont know it. Can some one throw some light on this? 2. Also, I want to test my card in full duplex mode as well, i.e. playing and recording simultaneously. How can test this using aplay and arecord? -Pharaoh.

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[alsa-devel] [PATCH] CA0106 SPI DAC powerdown
by Trent Piepho 25 Jul '07

25 Jul '07

Here are three patches for the CA0106 SPI DAC. The first patch powers down the DACs (other than the front dac) when they are not in use. The front dac serves as the output for the line-in pass-through / capture feedback, so it's not powered down. The second patch adds a bunch of symbol names for SPI registers, and converts the existing init sequence from magic numbers to these names. The sequence doesn't change at all. The third patch cleans up some unnecessary repeated sets to the same registers in the SPI init sequence. Of course, I have tested all these patches and they work.

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[alsa-devel] How do I tell ALSA that my driver supports every possible sample rate?
by Timur Tabi 25 Jul '07

25 Jul '07

I'm building an ASOC driver, and I want to tell ALSA that my PCM and my I2S drivers support every possible sample rate, because it's the codec that determines the real limitations. What SNDRV_PCM_RATE_xxx bit-pattern do I specify that will always match? -- Timur Tabi Linux Kernel Developer @ Freescale

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Re: [alsa-devel] support for Medion laptops
by albox＠free.fr 25 Jul '07

25 Jul '07

Realtek published a driver using ALSA 1.0.14-4. It seems that they added a new model for medion ("medion-md2"). I tried this model but it doesn't work either on my laptop. I sent them an email to know more about that. Does anybody has more information ? I updated the bug report : https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3160 Thanks

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[alsa-devel] YMFPCI: IEC958 volume does not work anymore (2.6.18->2.6.21)
by Jan Zuchhold 24 Jul '07

24 Jul '07

Hello, i'm using a Hoontech Soundtrack Digital-XG (YMF744) and after i've upgraded to kernel 2.6.21, i can't control the IEC958 volume (using "Wave volume") any longer. It seems to be always at maximum volume. The volume control of the analog output port works fine. This is working in 2.6.18, that i've used before. Something must have changed in between. Is there anything I can do to help to get this working again? Thanks alot, Jan

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Re: [alsa-devel] snd-hda-intel, AD 1988b: SPDIF output not working correctly, volume low for analog output
by Takashi Iwai 24 Jul '07

24 Jul '07

At Mon, 23 Jul 2007 23:02:57 +0200, Matthias Dahl wrote: > > Hello Takashi. > > On Friday 20 July 2007 15:47:55 Takashi Iwai wrote: > > > This is expected behavior. The PCM volume is implemented via softvol > > plugin, and it creates dynamically at the first run. But, usually, > > it's restored via alsactl at the next boot. > > It is. Thanks for the info. > > > Yes, it'll involve with patch_analog.c deeply. Basically you need to > > modify init verb table and the mixer element table. Check the AD1988B > > datasheet, and try to assign the different PIN (and corresponding > > mixer / selector widgets) in init verb and mixer tables. Then, > > rebuild driver, and try the driver until you get the complete mapping > > of pin widgets. > > I am currently a bit limited in time (end of semester exams) but after a few > hours of playing around (and reading the specs) I figured the following: > > 1) From time to time no analog output is routed to the spdif anymore. The only > clue I could get is the "IEC958 Playback Default" which is set to > > 06820002 [... many more zeros ...] > > and thus restored to that value after reboots. When things work like they > should (analog output is routed to spdif), its value is > > 04820002 [... many more zeros ...] > > Manually setting it to this value naturally solves the problem. I have yet > to figure out how to trigger this problem?! It seems to happen rather > randomly. In both cases, ac3 passthrough works just fine btw. Maybe some apps change the IEC958 status bits and doesn't restore the original value at close. (Or interrupting via ctrl-C may cause it.) You can see the meaning of these bits via iecset utility included in alsa-utils package. > 2) I have come a step closer in solving the analog-to-spdif volume/distortions > problem... at least I hope. > > The following patch fixes the volume problem for me. Without it, I had to > set the IEC958 volume somewhere around 58... increasing it would result in > noticable clipping and even short audio gaps (noise). > > Honestly, I don't quite understand the purpose of the code I removed in the > first place. From what I figured reading the specs and the code, the spdif > in is mixed with the spdif out stream... but that's all I could gather. But > still it doesn't seem quite right to me. > > --- patch_analog.c.orig 2007-07-23 22:44:25.000000000 +0200 > +++ patch_analog.c 2007-07-23 22:45:15.000000000 +0200 > @@ -2037,12 +2037,8 @@ > static struct hda_verb ad1988_spdif_init_verbs[] = { > /* SPDIF out sel */ > {0x02, AC_VERB_SET_CONNECT_SEL, 0x0}, /* PCM */ > - {0x0b, AC_VERB_SET_CONNECT_SEL, 0x0}, /* ADC1 */ > - {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, > - {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, > /* SPDIF out pin */ > {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */ > - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x17}, /* 0dB */ > > { } > }; > > The audio distortions in the background have become a little less noisy > but are still there. (a bass-like pumping, some white noise and sometimes > some high frequency tone) But the volume works just fine now... (iec958 > vol at 100%) and the audio sounds somewhat "clearer"! So, the culprit looks like the analog loopback. Could you try the patch below, and make sure that you set "PCM" as the "IEC958 Playback Source"? thanks, Takashi diff -r c2078875c0cd pci/hda/patch_analog.c --- a/pci/hda/patch_analog.c Tue Jul 24 12:10:34 2007 +0200 +++ b/pci/hda/patch_analog.c Tue Jul 24 12:47:42 2007 +0200 @@ -1889,16 +1889,19 @@ static int ad1988_spdif_playback_source_ struct hda_codec *codec = snd_kcontrol_chip(kcontrol); unsigned int sel; - sel = snd_hda_codec_read(codec, 0x02, 0, AC_VERB_GET_CONNECT_SEL, 0); - if (sel > 0) { + sel = snd_hda_codec_read(codec, 0x1d, 0, AC_VERB_GET_AMP_GAIN_MUTE, + AC_AMP_GET_INPUT); + if (!(sel & 0x80)) + ucontrol->value.enumerated.item[0] = 0; + else { sel = snd_hda_codec_read(codec, 0x0b, 0, AC_VERB_GET_CONNECT_SEL, 0); if (sel < 3) sel++; else sel = 0; + ucontrol->value.enumerated.item[0] = sel; } - ucontrol->value.enumerated.item[0] = sel; return 0; } @@ -1910,17 +1913,32 @@ static int ad1988_spdif_playback_source_ int change; val = ucontrol->value.enumerated.item[0]; - sel = snd_hda_codec_read(codec, 0x02, 0, AC_VERB_GET_CONNECT_SEL, 0); if (!val) { - change = sel != 0; - if (change || codec->in_resume) - snd_hda_codec_write(codec, 0x02, 0, - AC_VERB_SET_CONNECT_SEL, 0); + sel = snd_hda_codec_read(codec, 0x1d, 0, + AC_VERB_GET_AMP_GAIN_MUTE, + AC_AMP_GET_INPUT); + change = sel & 0x80; + if (change || codec->in_resume) { + snd_hda_codec_write(codec, 0x1d, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_IN_UNMUTE(0)); + snd_hda_codec_write(codec, 0x1d, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_IN_MUTE(1)); + } } else { - change = sel == 0; - if (change || codec->in_resume) - snd_hda_codec_write(codec, 0x02, 0, - AC_VERB_SET_CONNECT_SEL, 1); + sel = snd_hda_codec_read(codec, 0x1d, 0, + AC_VERB_GET_AMP_GAIN_MUTE, + AC_AMP_GET_INPUT | 0x01); + change = sel & 0x80; + if (change || codec->in_resume) { + snd_hda_codec_write(codec, 0x1d, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_IN_MUTE(0)); + snd_hda_codec_write(codec, 0x1d, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_IN_UNMUTE(1)); + } sel = snd_hda_codec_read(codec, 0x0b, 0, AC_VERB_GET_CONNECT_SEL, 0) + 1; change |= sel != val; @@ -2039,10 +2057,9 @@ static struct hda_verb ad1988_spdif_init {0x02, AC_VERB_SET_CONNECT_SEL, 0x0}, /* PCM */ {0x0b, AC_VERB_SET_CONNECT_SEL, 0x0}, /* ADC1 */ {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* SPDIF out pin */ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */ - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x17}, /* 0dB */ { } };

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[alsa-devel] Query regarding ALSA control interface
by Pharaoh . 24 Jul '07

24 Jul '07

Hi I am writing a pcm control interface for my alsa driver. I am confused about how the switches are used. I have implemented simple cases like following and they are working well. static int audio_path_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { /*Count is 256 since the values can vary from 0-255. But values fall in this range only*/ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 256; uinfo->value.integer.min = 0; uinfo->value.integer.max = 255; FN_EXIT; return 0; } static int audio_path_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { my_mixer_t *mixer_chip = snd_kcontrol_chip(kcontrol); ucontrol->value.integer.value[0] = mixer_chip->echo; ucontrol->value.integer.value[1] = mixer_chip->echo; return 0; } static int audio_path_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { my_mixer_t *mixer_chip = snd_kcontrol_chip(kcontrol); ----- ---- } Now, I have to implement a control which takes four different values..namely 0,1.100 and 101. In this case what shoud be the type of control? It should be ENUMERATED or INTEGER? I am confused here since the values are not exactly in a range. -pharaoh.

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[alsa-devel] [PATCH] Patch for AICA
by Adrian McMenamin 24 Jul '07

24 Jul '07

I submitted this yesterday via lkml and it was accepted by the SH maintainer. Probably should have copied in the ALSA list too - though the change is so small and consequential of another issue I was dealing with I forgot. The change is necessary because of a fix in the SH DMA API (see http://lkml.org/lkml/2007/7/23/445 for the larger patch) Signed-off by: Adrian McMenamin ,adrian(a)mcmen.demon.co.uk> diff --git a/sound/sh/aica.h b/sound/sh/aica.h index 8c11e3d..d098baa 100644 --- a/sound/sh/aica.h +++ b/sound/sh/aica.h @@ -52,7 +52,7 @@ #define AICA_CHANNEL1_OFFSET 0x21000 #define CHANNEL_OFFSET 0x10000 -#define AICA_DMA_CHANNEL 0 +#define AICA_DMA_CHANNEL 5 #define AICA_DMA_MODE 5

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