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

[alsa-devel] [PATCH] Another EMU0404 Board ID
by Veli-Matti Valtonen 07 Jan '08

07 Jan '08

Hey, This is based on pseudo-random playing around with the capabilities. With ca0102 this card gives no output atall, ca0108 appears to work fine, so it rather looks similar to the EMU1010b/EMU1010 changes. Some other people seem to have succeeded in using this aswell: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3496 diff -u -r alsa-driver-hg20071222/alsa-kernel/pci/emu10k1/emu10k1_main.c alsa-driver-hg20071222-0404b/alsa-kernel/pci/emu10k1/emu10k1_main.c --- alsa-driver-hg20071222/alsa-kernel/pci/emu10k1/emu10k1_main.c 2007-12-22 03:00:09.000000000 +0200 +++ alsa-driver-hg20071222-0404b/alsa-kernel/pci/emu10k1/emu10k1_main.c 2007-12-23 23:01:12.807047391 +0200 @@ -1419,6 +1419,14 @@ .ca0102_chip = 1, .spk71 = 1, .emu_model = EMU_MODEL_EMU1010}, /* Emu 1010 */ + /* EMU0404b */ + {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x40021102, + .driver = "Audigy2", .name = "E-mu 0404b [4002]", + .id = "EMU0404", + .emu10k2_chip = 1, + .ca0108_chip = 1, + .spk71 = 1, + .emu_model = EMU_MODEL_EMU0404}, /* EMU 0404 */ /* Tested by James(a)superbug.co.uk 20-3-2007. */ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x40021102, .driver = "Audigy2", .name = "E-mu 0404 [4002]",

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[alsa-devel] es18xx: Enable wavetable input from ESS chips
by Krzysztof Helt 07 Jan '08

07 Jan '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This patch enables wavetable chips ES689/ES69X connected to ESS ES18xx chips. The wavetable chip uses FM DAC if the clock signal from the wavetable is active. It has no effect if there is no ESS wavetable chip present. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- This patch has a gotcha. In theory, the OPL synthesis or wavetable input should use FM DAC depending on the active clock signal from the wavetable chip. Unfortunately, the default state of the wavetable chip from ESS (ES689/ES690/ES692) is to send clock signal all the time. This blocks OPL3 signal if the wavetable chip is present. This problem can be solved with sysex command 4 send to the wavetable chip. It sets the wavetable chip to auto-deactivate mode which switches the wavetable off if there are 5 seconds without midi input (according to the es1879 datasheet). I tested this patch on the es1868 chip without the wavetable. The OPL3 synthesis worked (no clock signal from the wavetable). I tested it on the card with es1868 chip and es690 wavetable. The OPL3 synthesis did not work, the wavetable worked. Both the OPL3 and wavetable worked if the sysex command 4 was sent to the wavetable chip (e.g. by amidi tool). Is it ok to add this sysex command sending to the driver? How? The wavetable clock signal can be detected (there is status bit which says if wavetable is active). Another question is about ES938 chip (bass, treble and 3D control) on the card. This chip is controlled by sysex midi commands. It can be detected (by reading settings with sysex commands). Is it possible to add this to the alsamixer or external tool for setting these controls is recommended? Regards, Krzysztof diff -urp linux-2.6.24/sound/isa/es18xx.c linux-new/sound/isa/es18xx.c --- linux-2.6.24/sound/isa/es18xx.c 2007-12-25 23:23:03.000000000 +0100 +++ linux-new/sound/isa/es18xx.c 2007-12-26 16:30:08.000000000 +0100 @@ -1442,6 +1442,8 @@ static int __devinit snd_es18xx_initiali snd_es18xx_write(chip, 0xB2, 0x50); /* Enable MPU and hardware volume interrupt */ snd_es18xx_mixer_write(chip, 0x64, 0x42); + /* Enable ESS wavetable input */ + snd_es18xx_mixer_bits(chip, 0x48, 0x10, 0x10); } else { int irqmask, dma1mask, dma2mask; ---------------------------------------------------------------------- Rozdajemy nagrody! Wygraj samochod! Sprawdz >> http://link.interia.pl/f1cb8

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[alsa-devel] [PATCH] [ALSA-KERNEL] Add IEC958 digital out support to AD1984 for Lenovo Thinkpads T61/X61
by Jerone Young 07 Jan '08

07 Jan '08

This patch adds IEC958 digital out support for the AD1984 sound card. This card can be found in Lenovo Thinkapds T61/X61. The digital out is not located on the Thinkpad, but optional docking station (it's coxial digital out). I've add this support as it is done the exact same way for the AD1983 & AD1884. I have tested this patch with my Lenovo Thinkpad T61 hooked up to a docking station (that has the digital coxial) and then run to my home theater reciever. Works like a charm :-) Patch is attached and also in this email. Signed-off-by: Jerone Young <jerone(a)gmail.com> diff -r 1227a1c12325 pci/hda/patch_analog.c --- a/pci/hda/patch_analog.c Mon Dec 24 14:40:56 2007 +0100 +++ b/pci/hda/patch_analog.c Thu Dec 27 17:41:13 2007 -0600 @@ -3095,6 +3095,16 @@ static struct snd_kcontrol_new ad1984_th .get = ad198x_mux_enum_get, .put = ad198x_mux_enum_put, }, + /* SPDIF controls */ + HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source", + /* identical with ad1983 */ + .info = ad1983_spdif_route_info, + .get = ad1983_spdif_route_get, + .put = ad1983_spdif_route_put, + }, { } /* end */ }; @@ -3197,7 +3207,7 @@ static int patch_ad1984(struct hda_codec codec->patch_ops.build_pcms = ad1984_build_pcms; break; case AD1984_THINKPAD: - spec->multiout.dig_out_nid = 0; + spec->multiout.dig_out_nid = AD1884_SPDIF_OUT; spec->input_mux = &ad1984_thinkpad_capture_source; spec->mixers[0] = ad1984_thinkpad_mixers; spec->init_verbs[spec->num_init_verbs++] = ad1984_thinkpad_init_verbs;

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[alsa-devel] [PATCH] hda: STAC927x VREF fix
by Matthew Ranostay 07 Jan '08

07 Jan '08

Some laptops incorrectly assume the front input jack as a line in instead of a microphone in. Which in turn disables the voltage reference, in which non-amplified input is not possible. This patch enables VREF80 for the input jack. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r eeed28ed6b7d pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Fri Jan 04 09:22:20 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Fri Jan 04 19:41:06 2008 -0700 @@ -3344,6 +3344,7 @@ static int patch_stac927x(struct hda_cod spec->mixer = stac927x_mixer; break; case STAC_DELL_BIOS: + stac92xx_set_config_reg(codec, 0x0e, 0x02a79130); case STAC_DELL_3ST: /* GPIO2 High = Enable EAPD */ spec->gpio_mask = spec->gpio_data = 0x00000004;

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[alsa-devel] ICE1724 chipset and envy24control
by Norval Watson 07 Jan '08

07 Jan '08

Hi, I just got an M-Audio Audiophile 192 soundcard with ICE1724 chipset. Works out of the box on debian sid. Is there any way to control this card with envy24control utility from alsa-tools? envy24control is written for ICE1712 chipset. Is there any project to develop control utility for ICE1724 card? Seasons greetings, Norval Watson Make the switch to the world's best email. Get the new Yahoo!7 Mail now. www.yahoo7.com.au/worldsbestemail

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[alsa-devel] python midi wrapper
by Aldrin Martoq 07 Jan '08

07 Jan '08

Hello, I need a midi/seq wrapper for Python. I've already created some code for the sequencer API and works great! Here is my python test code: -------------------------------------------------------------------- #!/usr/bin/env python import sys sys.path.append('build/lib.linux-i686-2.5') import alsaaudio print "Creating Sequencer..." seq = alsaaudio.Seq("hw", "Python seqtest.py", alsaaudio.SEQ_OPEN_DUPLEX, alsaaudio.SEQ_NONBLOCK) myid = seq.get_client_id() print "... seq:", myid, seq print "Creating Port..." myport = seq.create_simple_port("Processor port", alsaaudio.SEQ_PORT_CAP_WRITE | alsaaudio.SEQ_PORT_CAP_SUBS_WRITE, alsaaudio.SEQ_PORT_TYPE_APPLICATION) print "... port:", myport #queue = seq.create_queue() #print "... queue:", queue # dump clients and ports def dump_connection(): print print "Client/ports available:" for client in seq.get_client_ports(): clientname, clientid, portlist = client print " client: %3d %s" % (clientid, clientname) for port in portlist: portname, portid, connections = port print " port: %3d:%-2d +-%s" % (clientid, portid, portname) readconn, writeconn = connections for conn in readconn: print " connection to: %s" % conn for conn in writeconn: print " connection from: %s" % conn dump_connection() Is the front seq.connect_ports("0:1", "%d:%d" % (myid, myport)); seq.connect_ports("24:0", "%d:%d" % (myid, myport)); dump_connection() while True: events = seq.receive_events(60000) for ev in events: print ev print "\t", ev.data() -------------------------------------------------------------------- I have tried to contact the mantainer of PyAlsaAudio (cwi(a)unispeed.dk from the README file) for merging my code in his project, but have not received a response so far and the date of files seems old. Is anybody here interested in the merge? How can I put this effor in python-alsaaudio or may I create a new project for this specific task? Is the pyalsa project related to pyalsaaudio? Thanks, -- Aldrin Martoq

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[alsa-devel] function snd_mixer_load failed: Inappropriate ioctl for device
by Salatiel Filho 07 Jan '08

07 Jan '08

*** please let me in the to or cc fields. I am not on the list *** I am having a big trouble to make sound work on my ARM machine. I am running debian 4. # uname -a Linux LS-GL7D6 2.6.16.57-arm2 #7 Wed Dec 12 18:14:51 BRT 2007 armv5tejl GNU/Linux # cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.15. Compiled on Dec 12 2007 for kernel 2.6.16.57-arm2. Modules load successfully # lsmod Module Size Used by snd_pcm_oss 47680 0 snd_mixer_oss 15040 1 snd_pcm_oss snd_usb_audio 79904 0 snd_pcm 75877 2 snd_pcm_oss,snd_usb_audio snd_timer 20164 1 snd_pcm snd_page_alloc 8264 1 snd_pcm snd_usb_lib 15808 1 snd_usb_audio snd_rawmidi 21376 1 snd_usb_lib snd_seq_device 7116 1 snd_rawmidi snd_hwdep 7460 1 snd_usb_audio snd 54368 9 snd_pcm_oss,snd_mixer_oss,snd_usb_audio,snd_pcm,snd_timer,snd_usb_lib,snd_rawmidi,snd_seq_device,snd_hwdep soundcore 7684 1 snd nfs 102044 0 lockd 61780 1 nfs sunrpc 142808 3 nfs,lockd dm_mod 48176 0 usbhid 38628 0 # cat /proc/asound/cards 0 [default ]: USB-Audio - C-Media USB Headphone Set C-Media USB Headphone Set at usb-ehci_platform.4523-1, full speed # cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.15. Compiled on Dec 12 2007 for kernel 2.6.16.57-arm2. but everytime i try to use alsamixer i get: # alsamixer alsamixer: function snd_mixer_load failed: Inappropriate ioctl for device # aplay test.wav Playing WAVE 'test.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono aplay: xrun:1090: read/write error, state = PREPARED # strace alsamixer ... stat64("/usr//share/alsa/alsa.conf", {st_mode=S_IFREG|0644, st_size=8611, ...}) = 0 open("/usr//share/alsa/alsa.conf", O_RDONLY) = 3 fstat64(3, {st_mode=S_IFREG|0644, st_size=8611, ...}) = 0 mmap2(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) = 0x40017000 read(3, "#\n# ALSA library configuration "..., 4096) = 4096 read(3, "r\n\t\t\t\tname defaults.pcm.device\n\t"..., 4096) = 4096 read(3, "{\n\t\t\t@func refer\n\t\t\tname default"..., 4096) = 419 read(3, "", 4096) = 0 read(3, "", 4096) = 0 close(3) = 0 munmap(0x40017000, 4096) = 0 access("/etc/asound.conf", R_OK) = -1 ENOENT (No such file or directory) access("/root/.asoundrc", R_OK) = -1 ENOENT (No such file or directory) open("/dev/snd/controlC0", O_RDONLY) = 3 close(3) = 0 open("/dev/snd/controlC0", O_RDWR) = 3 ioctl(3, USBDEVFS_CONTROL, 0xbec2da64) = 0 ioctl(3, UI_DEV_CREATE, 0xbec2db2c) = 0 close(3) = 0 stat64("/usr//share/alsa/alsa.conf", {st_mode=S_IFREG|0644, st_size=8611, ...}) = 0 open("/dev/snd/controlC0", O_RDONLY) = 3 close(3) = 0 open("/dev/snd/controlC0", O_RDWR) = 3 ioctl(3, USBDEVFS_CONTROL, 0xbec2da44) = 0 fcntl64(3, F_GETFL) = 0x2 (flags O_RDWR) fcntl64(3, F_SETFL, O_RDWR|O_NONBLOCK) = 0 ioctl(3, USBDEVFS_RELEASEINTERFACE, 0xbec2dac0) = 0 ioctl(3, USBDEVFS_RELEASEINTERFACE, 0xbec2dac0) = 0 ioctl(3, USBDEVFS_CONNECTINFO, 0xbec2d86c) = 0 ioctl(3, USBDEVFS_IOCTL, 0xbec2d42c) = -1 ENOTTY (Inappropriate ioctl for device) fstat64(1, {st_mode=S_IFCHR|0620, st_rdev=makedev(136, 2), ...}) = 0 mmap2(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) = 0x40017000 write(1, "\n", 1 ) = 1 write(2, "alsamixer: function snd_mixer_lo"..., 74alsamixer: function snd_mixer_load failed: Inappropriate ioctl for device ) = 74 munmap(0x40017000, 4096) = 0 io_submit(0x2, 0, 0x8 <unfinished ... exit status 2> any help? The same card works ok on my x86 machine using . So the problem is not the card but something else. -- []'s Salatiel "O maior prazer do inteligente é bancar o idiota diante de um idiota que banca o inteligente".

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[alsa-devel] Test music sampled at 192K
by Jon Smirl 06 Jan '08

06 Jan '08

Can anyone point me to some uncompressed test music/tones sampled at 192K/sec? -- Jon Smirl jonsmirl(a)gmail.com

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[alsa-devel] 5.1 surround--no front center speaker sound--[NVidia CK804]
by Steve Thornley 05 Jan '08

05 Jan '08

Nforce 4 MSI motherboard integrated sound----since my Creative X-Fi isn't supported, YET can not get the front center speaker to work. jlt:/home/steven# aplay -l **** List of PLAYBACK Hardware Devices **** card 0: CK804 [NVidia CK804], device 0: Intel ICH [NVidia CK804] Subdevices: 0/1 Subdevice #0: subdevice #0 card 0: CK804 [NVidia CK804], device 2: Intel ICH - IEC958 [NVidia CK804 - IEC958] Subdevices: 1/1 Subdevice #0: subdevice #0 Here is my present asound.conf file that I created in /etc for my Debian etch 4.0----amd64 system X2-4200-cpu # Set default sound card # Useful so that all settings can be changed to a different card here. pcm.snd_card { type hw card 0 } pcm.dmix6 { type dmix ipc_key 1024 ipc_key_add_uid false # let multiple users share ipc_perm 0660 # IPC permissions (octal, default 0600) slave { pcm snd_card # see below rate 48000 channels 6 period_time 0 period_size 1024 buffer_time 0 buffer_size 4096 } } #route for stereo only pcm.dmix20 { type route slave.pcm "dmix6" slave.channels 6 ttable.0.0 1 ttable.1.1 1 } #route for stereo sound+center+subwoofer pcm.ch31dup { type route slave.pcm "dmix6" slave.channels 6 ttable.0.0 1 ttable.1.1 1 ttable.0.4 0.5 ttable.1.4 0.5 ttable.0.5 0.5 ttable.1.5 0.5 } pcm.ch51dup { type route slave.pcm dmix6 slave.channels 6 ttable.0.0 1 ttable.1.1 1 ttable.0.2 1 ttable.1.3 1 ttable.0.4 0.5 ttable.1.4 0.5 ttable.0.5 0.5 ttable.1.5 0.5 } pcm.duplex { type asym playback.pcm "ch51dup" # playback.pcm "dmix2" # capture.pcm "mydsnoop" capture.pcm "snd_card" } ################### # CONVERSION PLUG # ################### # Setting the default pcm device allows the conversion # rate to be selected on the fly. # duplex mode allows any alsa enabled app to read/write # to the dmix plug (Fixes a problem with wine). pcm.!default { type plug slave.pcm "duplex" } pcm.dsp "duplex" pcm.dsp1 "duplex" pcm.fr { type route slave.pcm "dmix6" slave.channels 6 ttable.0.0 1 ttable.1.1 1 } pcm.rr { type route slave.pcm "dmix6" slave.channels 6 ttable.0.2 1 ttable.1.3 1 } pcm.lfe { type route slave.pcm "dmix6" slave.channels 6 ttable.0.4 1 ttable.1.5 1 } Here is my back up with other configurations for the file that created the same 4 speakers and subwoofer sounds but no front center speaker sounds pcm.ch51dup { type route slave.pcm dmix6 slave.channels 6 ttable.0.0 1 ttable.1.1 1 ttable.0.2 1 ttable.1.3 1 ttable.0.4 1 ttable.1.4 1 ttable.0.5 1 ttable.1.5 1 } ########Old Setup # pcm.51to40 # { # type route # slave.pcm surround51 # slave.channels 6 # # # Front and rear, at 75% of original signal strength # ttable.0.0 0.75 # ttable.1.1 0.75 # ttable.2.2 0.75 # ttable.3.3 0.75 # # Center channel routing (routed to front-left and front-right), # 6dB gaindrop (gain half of main channels) per channel # ttable.4.0 0.375 # ttable.4.1 0.375 # LFE channel routing (routed to front-left and front-right), # 6dB gaindrop (gain half of main channels) per channel # ttable.5.0 0.375 # ttable.5.1 0.375 # } When I changed a setting from Autodetect to ALSA I could not get any sound--so I compiled everything from source with the latests tars and still nothing, and once I turned the Autodetect in Multimedia Systems Selector<Preferences<Desktop for the Default output plugin, I still had no sound in my center front speaker. Whew. I'd love it for my Creative X-Fi card to be supported, but right now, I really like my front center speaker back. I'm at the end of my list of possible fixes, maybe someone who understands can take a look at this an have a go ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

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[alsa-devel] fujitsu siemens amilo la 1703 vt1807 hda intel card problem
by Norayr Chilingaryan 05 Jan '08

05 Jan '08

Hello I decided to write here, firtly, because my messages in aksa-user have been ingored, and secondly, because I understood that my card most probably just unsupported and I must contact developers. I have tried to pass options model=fujitsu, because I have a fujitsu siements notebook etc I have also tried to pass amny other options in /etc/modules.d/alsa but withnot effect I use gentoo, and the only thing I noticed with latest kernels and alsa drivers 1.0.15 is that I can hear sound through the headphones, but with low volume. It is not possible to get volume higher or speaker to make sounds now. So, please, this is information about the card 04:01.0 Audio device: VIA Technologies, Inc. VIA High Definition Audio Controller (rev 10) kakraphoon ~ # cat /proc/asound/card0/codec#* | grep Codec Codec: VIA VIA VT1708 Codec: Motorola Si3054 Thank you very much Norayr

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