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

[alsa-devel] [PATCH] Improve behaviour of Dreamcast aica ALSA driver in poor resource conditions
by Adrian McMenamin 04 Jul '07

04 Jul '07

This patch stops the driver from crashing in certain situations (eg if the network fails when NFS mounted), please apply. Signed-off by: Adrian McMenamin <adrian(a)mcmen.demon.co.uk> --- alsa-kernel/sh/aica.c 2007-06-23 15:25:55.000000000 +0100 +++ linux-2.6.21/sound/sh/aica.c 2007-07-03 22:28:16.000000000 +0100 @@ -366,7 +366,9 @@ *substream) { struct snd_card_aica *dreamcastcard = substream->pcm->private_data; + flush_workqueue(aica_queue); del_timer(&dreamcastcard->timer); + aica_chn_halt(); kfree(dreamcastcard->channel); spu_disable(); return 0; @@ -402,16 +404,11 @@ static int snd_aicapcm_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { - struct snd_card_aica *dreamcastcard; switch (cmd) { case SNDRV_PCM_TRIGGER_START: spu_begin_dma(substream); break; case SNDRV_PCM_TRIGGER_STOP: - dreamcastcard = substream->pcm->private_data; - if (dreamcastcard->timer.data) - del_timer(&dreamcastcard->timer); - aica_chn_halt(); break; default: return -EINVAL;

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[alsa-devel] Errors in application building
by Paul Kavan 03 Jul '07

03 Jul '07

Hello all: I am working on my first bit of application code for ALSA and am not sure how to properly compile it against the libs. I have to cross compile the application on my host machine i686 PC for use on a ARM development board. I copied the alsa include directory into my staging include directory (this is where the cross-compiler works from), which allowed the program to find alsa/asoundlib.h However, I am still getting some errors that indicate to me it cannot find everything it needs. I am getting the following errors: grhuser@pcL7:$ arm-linux-gcc -Wall -o grh_audio grh_audio.c grh_audio.c: In function 'grh_capture': grh_audio.c:113: warning: pointer targets in passing argument 3 of 'snd_pcm_hw_params_set_rate_near' differ in signedness grh_audio.c: In function 'grh_playback': grh_audio.c:208: warning: pointer targets in passing argument 3 of 'snd_pcm_hw_params_set_rate_near' differ in signedness /tmp/ccMn48zN.o: In function `grh_capture': grh_audio.c:(.text+0x16c): undefined reference to `snd_pcm_open' grh_audio.c:(.text+0x198): undefined reference to `snd_strerror' grh_audio.c:(.text+0x1c0): undefined reference to `snd_pcm_hw_params_malloc' grh_audio.c:(.text+0x1e4): undefined reference to `snd_strerror' grh_audio.c:(.text+0x214): undefined reference to `snd_pcm_hw_params_any' grh_audio.c:(.text+0x238): undefined reference to `snd_strerror' grh_audio.c:(.text+0x26c): undefined reference to `snd_pcm_hw_params_set_access' grh_audio.c:(.text+0x290): undefined reference to `snd_strerror' grh_audio.c:(.text+0x2c4): undefined reference to `snd_pcm_hw_params_set_format' grh_audio.c:(.text+0x2e8): undefined reference to `snd_strerror' grh_audio.c:(.text+0x324): undefined reference to `snd_pcm_hw_params_set_rate_near' grh_audio.c:(.text+0x348): undefined reference to `snd_strerror' grh_audio.c:(.text+0x380): undefined reference to `snd_pcm_hw_params_set_channels' grh_audio.c:(.text+0x3a4): undefined reference to `snd_strerror' grh_audio.c:(.text+0x3d4): undefined reference to `snd_pcm_hw_params' grh_audio.c:(.text+0x3f8): undefined reference to `snd_strerror' grh_audio.c:(.text+0x420): undefined reference to `snd_pcm_hw_params_free' grh_audio.c:(.text+0x42c): undefined reference to `snd_pcm_prepare' grh_audio.c:(.text+0x450): undefined reference to `snd_strerror' grh_audio.c:(.text+0x494): undefined reference to `snd_pcm_readi' grh_audio.c:(.text+0x4c0): undefined reference to `snd_strerror' grh_audio.c:(.text+0x500): undefined reference to `snd_pcm_close' /tmp/ccMn48zN.o: In function `grh_playback': grh_audio.c:(.text+0x5b8): undefined reference to `snd_pcm_open' grh_audio.c:(.text+0x5e4): undefined reference to `snd_strerror' grh_audio.c:(.text+0x60c): undefined reference to `snd_pcm_hw_params_malloc' grh_audio.c:(.text+0x630): undefined reference to `snd_strerror' grh_audio.c:(.text+0x660): undefined reference to `snd_pcm_hw_params_any' grh_audio.c:(.text+0x684): undefined reference to `snd_strerror' grh_audio.c:(.text+0x6b8): undefined reference to `snd_pcm_hw_params_set_access' grh_audio.c:(.text+0x6dc): undefined reference to `snd_strerror' grh_audio.c:(.text+0x710): undefined reference to `snd_pcm_hw_params_set_format' grh_audio.c:(.text+0x734): undefined reference to `snd_strerror' grh_audio.c:(.text+0x770): undefined reference to `snd_pcm_hw_params_set_rate_near' grh_audio.c:(.text+0x794): undefined reference to `snd_strerror' grh_audio.c:(.text+0x7cc): undefined reference to `snd_pcm_hw_params_set_channels' grh_audio.c:(.text+0x7f0): undefined reference to `snd_strerror' grh_audio.c:(.text+0x820): undefined reference to `snd_pcm_hw_params' grh_audio.c:(.text+0x844): undefined reference to `snd_strerror' grh_audio.c:(.text+0x86c): undefined reference to `snd_pcm_hw_params_free' grh_audio.c:(.text+0x878): undefined reference to `snd_pcm_prepare' grh_audio.c:(.text+0x89c): undefined reference to `snd_strerror' grh_audio.c:(.text+0x8e0): undefined reference to `snd_pcm_writei' grh_audio.c:(.text+0x90c): undefined reference to `snd_strerror' grh_audio.c:(.text+0x94c): undefined reference to `snd_pcm_close' collect2: ld returned 1 exit status My application code is attached. If anyone could give me some pointers as to why these errors are occurring, I would appreciate it greatly. Thanks. Paul -- *************************************** Paul David Kavan Project Engineer GRH Electronics, Inc. 402-734-4900 pkavan(a)gmail.com ***************************************

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[alsa-devel] namehint problem...
by Fausto Carvalho Marques SIlva 03 Jul '07

03 Jul '07

Compiling alsa-lib-1.0.14rc4 with the above options: ./configure --disable-aload --disable-alisp --disable-instr --disable-seq --disable-hwdep --disable-rawmidi --with-debug=no --with-pcm-plugins=hw --with-ctl-plugins=hw After the build of asound and a linking against a simple program: #include <stdio.h> #include <stdlib.h> #include <alsa/asoundlib.h> main (int argc, char *argv[]) { int i; int err; short buf[128]; snd_pcm_t *playback_handle; snd_pcm_hw_params_t *hw_params; if ((err = snd_pcm_open (&playback_handle, argv[1], SND_PCM_STREAM_PLAYBACK, 0)) < 0) { fprintf (stderr, "cannot open audio device %s (%s)\n", argv[1], snd_strerror (err)); exit (1); } snd_pcm_close (playback_handle); exit (0); } gcc -o teste teste.c -I./include src/.libs/libasound.so.2.0.0 -ldl src/.libs/libasound.so.2.0.0: undefined reference to `snd_hwdep_info_get_name' src/.libs/libasound.so.2.0.0: undefined reference to `snd_rawmidi_info_get_name' src/.libs/libasound.so.2.0.0: undefined reference to `snd_rawmidi_info_sizeof' src/.libs/libasound.so.2.0.0: undefined reference to `snd_rawmidi_info_set_device' src/.libs/libasound.so.2.0.0: undefined reference to `snd_hwdep_info_set_device' src/.libs/libasound.so.2.0.0: undefined reference to `snd_hwdep_info_sizeof' src/.libs/libasound.so.2.0.0: undefined reference to `snd_rawmidi_info_set_stream' collect2: ld returned 1 exit status After some verification discoverd that the problem is on src/control/namehint.c that references this functions but the configure removed rawmidi and hwdep. How can we resolve this problem? Fausto Carvalho Marques Silva

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[alsa-devel] Application Coding
by Paul Kavan 03 Jul '07

03 Jul '07

Hello all: Beginning to write my application code now that I have alsa SoC working fairly well on my ARM development board. My audio application is fairly simple. I will be doing raw, S8, samples at an 8k sample rate. Essentially I need to do playback of a stream when signaled by an application to do so and capture when signaled by a button press. Capture takes precedent over playback. I am really new to the alsa api and I am planning on using aplay as a template. However, if anyone knows of an easier piece of code that may work as a template, please feel free to point it out to me. I am using the API documentation from the alsa-project website. If anyone knows of another good piece of documentation, please feel free to let me know. Thanks. Paul -- *************************************** Paul David Kavan Project Engineer GRH Electronics, Inc. 402-734-4900 pkavan(a)gmail.com ***************************************

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[alsa-devel] SALSA-Lib: snd_pcm_hw_params fails with MMAP
by J. Scott Merritt 03 Jul '07

03 Jul '07

Dear List, Using SALSA-Lib 0.0.5 ... After calling snd_pcm_hw_params_set_access with an access mode of SND_PCM_ACCESS_MMAP_INTERLEAVED, the subsequent call to snd_pcm_hw_params returns a snd_strerror error code of "Invalid argument". This same program operates properly in the target environment (ARM PXA270) when compiled and linked with the standard ALSA LIB 1.0.13. Thanks, Scott.

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Re: [alsa-devel] ALSA - snd-hda-intel - Realtek ALC268
by Peter Poulsen 03 Jul '07

03 Jul '07

Citat af Chris Cheney <chris.cheney(a)canonical.com>: <snip> > I already have alc268 working fine on my box for sound, just tested it. > :) I am now working on making sure all the other functionality is > properly implemented. Sounds awesome :) looking forward to trying out your code /peter

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[alsa-devel] roland synth sh-201
by carlo beccaria 03 Jul '07

03 Jul '07

hi all, with this change in usbquirks.h , i can use my synth roland sh-201 { /* Roland SH-201 */ USB_DEVICE(0x0582, 0x00ad), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .vendor_name = "Roland", .product_name = "SH-201", .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 0, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { .format = SNDRV_PCM_FORMAT_S24_3LE, .channels = 2, .iface = 0, .altsetting = 1, .altset_idx = 1, .attributes = 0, .endpoint = 0x01, .ep_attr = 0x01, .rates = SNDRV_PCM_RATE_CONTINUOUS, .rate_min = 44100, .rate_max = 44100, } }, { .ifnum = 1, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { .format = SNDRV_PCM_FORMAT_S24_3LE, .channels = 2, .iface = 1, .altsetting = 1, .altset_idx = 1, .attributes = 0, .endpoint = 0x81, .ep_attr = 0x01, .rates = SNDRV_PCM_RATE_CONTINUOUS, .rate_min = 44100, .rate_max = 44100, } }, { .ifnum = 2, .type = QUIRK_MIDI_FIXED_ENDPOINT, .data = & (const struct snd_usb_midi_endpoint_info) { .out_cables = 0x0007, .in_cables = 0x0007 } }, { .ifnum = -1 } } } }, my kernel is 2.6.21.5-rt17, (plus preempt patch) and i use debian testing the hardware seem to be very similar to Roland SC-D70 (windows .inf file are very similar) this is the device id Bus 001 Device 003: ID 0582:00ad Roland Corp. in attach you can see the lsusb (verbose) output becks@vaix:~$ amidi -l Dir Device Name IO hw:1,0,0 SH-201 MIDI 1 IO hw:1,0,1 SH-201 MIDI 2 IO hw:1,0,2 SH-201 MIDI 3 muse works fine with jack and midi output becks@vaix:~$ aseqdump -p 20:0 Waiting for data. Press Ctrl+C to end. Source_ Event_________________ Ch _Data__ 20:0 Note on 1 52 17 20:0 Note off 1 52 55 also midi-in works becks@vaix:~$ cat /proc/asound/cards 0 [A5451 ]: ALI5451 - ALI 5451 ALI 5451 at 0x8400, irq 9 1 [SH201 ]: USB-Audio - SH-201 Roland SH-201 at usb-0000:00:0c.0-1, full speed when i use my synth like a sound card, it don't work, i don't know exactly why (if is a patch problem, or rate/format) becks@vaix:~$ aplay -D hw:1,0 /usr/share/sounds/KDE_Window_UnSticky.wav Playing WAVE '/usr/share/sounds/KDE_Window_UnSticky.wav' : Signed 16 bit Little Endian, Rate 22050 Hz, Mono aplay: set_params:904: Sample format non available let me know if you need more info thanks -- carlo beccaria

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Re: [alsa-devel] [HDA-Intel/ALC268] no sound
by Steven Zakulec 02 Jul '07

02 Jul '07

I've attached the file. I tried using aplay and ffplay- both had no trouble opening and playing a file, there was just no sound. --------------------------------------- Original E-mail From: Takashi Iwai [tiwai(a)suse.de] Date: 07/02/2007 01:25 PM To: Steven Zakulec [spzakulec(a)mail.widener.edu] Subject: Re: [alsa-devel] [HDA-Intel/ALC268] no sound At Wed, 27 Jun 2007 17:34:15 -0400 (EDT), Steven Zakulec wrote: > > (Continued from this thread: http://mailman.alsa-project.org/pipermail/alsa-devel/2007-June/001459.html) > > I've included the result of alsactl -f somefile store. I thought it got sent to the list earlier, but apparently not. Below this I also added the alsa things from dmesg. And the contents of /proc/asound/card0/codec#* files, please. If you change the mixer status, these files also changed, too. Also, please inform what you tested more exactly. In which hardware configuration and which application with what parameters... Takashi > > state.Intel { > control.1 { > comment.access 'read write' > comment.type INTEGER > comment.count 2 > comment.range '0 - 64' > iface MIXER > name 'Front Playback Volume' > value.0 57 > value.1 57 > } > control.2 { > comment.access 'read write' > comment.type BOOLEAN > comment.count 2 > iface MIXER > name 'Front Playback Switch' > value.0 true > value.1 true > } > control.3 { > comment.access 'read write' > comment.type INTEGER > comment.count 2 > comment.range '0 - 64' > iface MIXER > name 'Headphone Playback Volume' > value.0 63 > value.1 63 > } > control.4 { > comment.access 'read write' > comment.type BOOLEAN > comment.count 2 > iface MIXER > name 'Headphone Playback Switch' > value.0 true > value.1 true > } > control.5 { > comment.access 'read write' > comment.type INTEGER > comment.count 2 > comment.range '0 - 31' > iface MIXER > name 'Capture Volume' > value.0 1 > value.1 1 > } > control.6 { > comment.access 'read write' > comment.type BOOLEAN > comment.count 2 > iface MIXER > name 'Capture Switch' > value.0 false > value.1 false > } > control.7 { > comment.access 'read write' > comment.type INTEGER > comment.count 2 > comment.range '0 - 31' > iface MIXER > name 'Capture Volume' > index 1 > value.0 1 > value.1 1 > } > control.8 { > comment.access 'read write' > comment.type BOOLEAN > comment.count 2 > iface MIXER > name 'Capture Switch' > index 1 > value.0 false > value.1 false > } > control.9 { > comment.access 'read write' > comment.type ENUMERATED > comment.count 1 > comment.item.0 Mic > comment.item.1 'Front Mic' > iface MIXER > name 'Input Source' > value Mic > } > control.10 { > comment.access 'read write' > comment.type ENUMERATED > comment.count 1 > comment.item.0 Mic > comment.item.1 'Front Mic' > iface MIXER > name 'Input Source' > index 1 > value Mic > } > control.11 { > comment.access 'read write' > comment.type BOOLEAN > comment.count 1 > iface MIXER > name 'Off-hook Switch' > value false > } > control.12 { > comment.access 'read write' > comment.type BOOLEAN > comment.count 1 > iface MIXER > name 'Caller ID Switch' > value false > } > control.13 { > comment.access 'read write user' > comment.type INTEGER > comment.count 2 > comment.range '0 - 255' > comment.tlv '0000000100000008ffffec1400000014' > iface MIXER > name 'PCM Playback Volume' > value.0 255 > value.1 255 > } > } > hda_codec: Unknown model for ALC268, trying auto-probe from BIOS... > [ 18.512000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/hda_codec.c:2339: autoconfig: line_outs=1 (0x14/0x0/0x0/0x0/0x0) > [ 18.512000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/hda_codec.c:2343: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) > [ 18.512000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/hda_codec.c:2347: hp_outs=1 (0x15/0x0/0x0/0x0/0x0) > [ 18.512000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/hda_codec.c:2355: inputs: mic=0x19, fmic=0x18, line=0x0, fline=0x0, cd=0x0, aux=0x0 > [ 19.236000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/patch_si3054.c:244: si3054: cannot initialize. EXT MID = 0000 > [ 19.244000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/patch_si3054.c:256: Link Frame Detect(FDT) is not ready (line status: 0000) > > > > comment.count 2 > : > > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >

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[alsa-devel] HDA Intel/ALC268
by Peter Poulsen 02 Jul '07

02 Jul '07

Is there anybody that has had any luck with this, or I'm a fighting a loosing battle? /peter

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[alsa-devel] [HDA-Intel/ALC268] no sound
by Steven Zakulec 02 Jul '07

02 Jul '07

(Continued from this thread: http://mailman.alsa-project.org/pipermail/alsa-devel/2007-June/001459.html) I've included the result of alsactl -f somefile store. I thought it got sent to the list earlier, but apparently not. Below this I also added the alsa things from dmesg. state.Intel { control.1 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 64' iface MIXER name 'Front Playback Volume' value.0 57 value.1 57 } control.2 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Front Playback Switch' value.0 true value.1 true } control.3 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 64' iface MIXER name 'Headphone Playback Volume' value.0 63 value.1 63 } control.4 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Headphone Playback Switch' value.0 true value.1 true } control.5 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' iface MIXER name 'Capture Volume' value.0 1 value.1 1 } control.6 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Capture Switch' value.0 false value.1 false } control.7 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' iface MIXER name 'Capture Volume' index 1 value.0 1 value.1 1 } control.8 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Capture Switch' index 1 value.0 false value.1 false } control.9 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 Mic comment.item.1 'Front Mic' iface MIXER name 'Input Source' value Mic } control.10 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 Mic comment.item.1 'Front Mic' iface MIXER name 'Input Source' index 1 value Mic } control.11 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Off-hook Switch' value false } control.12 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Caller ID Switch' value false } control.13 { comment.access 'read write user' comment.type INTEGER comment.count 2 comment.range '0 - 255' comment.tlv '0000000100000008ffffec1400000014' iface MIXER name 'PCM Playback Volume' value.0 255 value.1 255 } } hda_codec: Unknown model for ALC268, trying auto-probe from BIOS... [ 18.512000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/hda_codec.c:2339: autoconfig: line_outs=1 (0x14/0x0/0x0/0x0/0x0) [ 18.512000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/hda_codec.c:2343: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 18.512000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/hda_codec.c:2347: hp_outs=1 (0x15/0x0/0x0/0x0/0x0) [ 18.512000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/hda_codec.c:2355: inputs: mic=0x19, fmic=0x18, line=0x0, fline=0x0, cd=0x0, aux=0x0 [ 19.236000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/patch_si3054.c:244: si3054: cannot initialize. EXT MID = 0000 [ 19.244000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/patch_si3054.c:256: Link Frame Detect(FDT) is not ready (line status: 0000) comment.count 2 :

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