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[alsa-devel] Easiest way to test playback on an ASOC driver?
by Timur Tabi 04 Jul '07

04 Jul '07

I'm in the early stages of testing an ASOC driver, i.e. I'm trying to get playback to work. What is the easiest way to test playback? Is there a particular tool I should be using that does the most basic and direct playback of audio data through ALSA? -- Timur Tabi Linux Kernel Developer @ Freescale

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[alsa-devel] left/right bleed or crossover
by Howdy Pierce 04 Jul '07

04 Jul '07

I'm encountering the following problem on Fedora Core 6 running on a Dell Optiplex 745. This is a pretty vanilla computer with an ICH8 chipset and integrated AD1983 sound capabilities. I have wired up a funky line-input cable so that I have completely separate audio sources playing on the left and right inputs. When I either play these through to the speaker output directly (using alsamixer), or record the audio using arecord, the result is that there is a slight bit of bleeding or crossover, so I can faintly hear the left input on the right channel, and vice versa. If I record a wav file and then play it on another computer, the left/right bleeding follows the wav file, so it seems that the problem is in the capture process somehow. When I install Windows XP onto the same machine, the problem disappears, so I can rule out hardware errors. The application I'm working on relies on being able to reliably capture the right and left channels of audio independently. Since the problem can be seen using the standard arecord/aplay tools, though, I don't think it's my code. I've tried all the possible combinations inside alsamixer, and I don't think there's anything I've messed up there. I've put the relevant system details below. Thanks for any help you can provide. --Howdy ============================= Howdy Pierce Managing Partner Cardinal Peak, LLC work: (303) 665-3962 x101 cell: (303) 589-1645 email: howdy(a)cardinalpeak.com web: www.cardinalpeak.com ============================= [root@dundee ~]# cat /proc/asound/cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xdfdfc000 irq 169 [root@dundee ~]# cat /proc/asound/devices 2: : timer 3: : sequencer 4: [ 0- 1]: digital audio playback 5: [ 0- 0]: digital audio playback 6: [ 0- 0]: digital audio capture 7: [ 0] : control [root@dundee ~]# lspci 00:00.0 Host bridge: Intel Corporation 82Q963/Q965 Memory Controller Hub (rev 02) 00:01.0 PCI bridge: Intel Corporation 82Q963/Q965 PCI Express Root Port (rev 02) 00:02.0 Display controller: Intel Corporation 82Q963/Q965 Integrated Graphics Controller (rev 02) 00:02.1 Display controller: Intel Corporation 82Q963/Q965 Integrated Graphics Controller (rev 02) 00:1a.0 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI #4 (rev 02) 00:1a.1 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI #5 (rev 02) 00:1a.7 USB Controller: Intel Corporation 82801H (ICH8 Family) USB2 EHCI #2 (rev 02) 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 02) 00:1c.0 PCI bridge: Intel Corporation 82801H (ICH8 Family) PCI Express Port 1 (rev 02) 00:1c.4 PCI bridge: Intel Corporation 82801H (ICH8 Family) PCI Express Port 5 (rev 02) 00:1d.0 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI #1 (rev 02) 00:1d.1 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI #2 (rev 02) 00:1d.2 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI #3 (rev 02) 00:1d.7 USB Controller: Intel Corporation 82801H (ICH8 Family) USB2 EHCI #1 (rev 02) 00:1e.0 PCI bridge: Intel Corporation 82801 PCI Bridge (rev f2) 00:1f.0 ISA bridge: Intel Corporation 82801HB/HR (ICH8/R) LPC Interface Controller (rev 02) 00:1f.2 IDE interface: Intel Corporation 82801H (ICH8 Family) 4 port SATA IDE Controller (rev 02) 00:1f.3 SMBus: Intel Corporation 82801H (ICH8 Family) SMBus Controller (rev 02) 00:1f.5 IDE interface: Intel Corporation 82801H (ICH8 Family) 2 port SATA IDE Controller (rev 02) 03:00.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5754 Gigabit Ethernet PCI Express (rev 02) 04:02.0 VGA compatible controller: nVidia Corporation NV18 [GeForce4 MX 4000 AGP 8x] (rev c1) [root@dundee ~]# rpm --query --all | grep alsa alsa-lib-1.0.12-2.fc6 alsa-lib-devel-1.0.12-2.fc6 alsa-utils-1.0.12-3.fc6 [root@dundee Intel]# cat /proc/asound/Intel/codec#0 Codec: Analog Devices AD1983 Address: 0 Vendor Id: 0x11d41983 Subsystem Id: 0x102801da Revision Id: 0x100400 Default PCM: rates 0x07f, bits 0x0e, types 0x1 Default Amp-In caps: N/A Default Amp-Out caps: ofs=0x3d, nsteps=0x3f, stepsize=0x05, mute=1 Node 0x02 [Audio Output] wcaps 0x30311: Stereo Digital PCM: rates 0x060, bits 0x02, types 0x5 Connection: 2 0x01* 0x04 Node 0x03 [Audio Output] wcaps 0x441: Stereo Power: 0x0 Node 0x04 [Audio Input] wcaps 0x100501: Stereo Power: 0x0 Connection: 1 0x14 Node 0x05 [Pin Complex] wcaps 0x400185: Stereo Amp-Out Amp-Out caps: ofs=0x3d, nsteps=0x3f, stepsize=0x05, mute=1 Amp-Out vals: [0x32 0x32] Pincap 0x0817: OUT Detect Pin Default 0x01014010: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green Pin-ctls: 0x40: OUT Connection: 2 0x03 0x0e* Node 0x06 [Pin Complex] wcaps 0x400185: Stereo Amp-Out Amp-Out caps: ofs=0x3d, nsteps=0x3f, stepsize=0x05, mute=1 Amp-Out vals: [0x25 0x25] Pincap 0x081f: OUT HP Detect Pin Default 0x0221101f: [Jack] HP Out at Ext Front Conn = 1/8, Color = Black Pin-ctls: 0xc0: OUT HP Connection: 2 0x03 0x0e* Node 0x07 [Pin Complex] wcaps 0x400104: Mono Amp-Out Amp-Out caps: ofs=0x3d, nsteps=0x3f, stepsize=0x05, mute=1 Amp-Out vals: [0x80] Pincap 0x0810: OUT Pin Default 0x991301f0: [Fixed] Speaker at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x40: OUT Connection: 1 0x0f Node 0x08 [Pin Complex] wcaps 0x400081: Stereo Pincap 0x081727: IN Detect Pin Default 0x02a11020: [Jack] Mic at Ext Front Conn = 1/8, Color = Black Pin-ctls: 0x24: IN Node 0x09 [Pin Complex] wcaps 0x400081: Stereo Pincap 0x081727: IN Detect Pin Default 0x0181302e: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue Pin-ctls: 0x20: IN Node 0x0a [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x0810: OUT Pin Default 0x414511f0: [N/A] SPDIF Out at Ext Rear Conn = Optical, Color = Black Pin-ctls: 0x40: OUT Connection: 1 0x02 Node 0x0b [Audio Selector] wcaps 0x300101: Stereo Connection: 4 0x03 0x0c 0x0d 0x0e* Node 0x0c [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-Out vals: [0x00 0x00] Connection: 2 0x08* 0x09 Node 0x0d [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x09* 0x08 Node 0x0e [Audio Mixer] wcaps 0x200101: Stereo Connection: 3 0x11 0x12 0x13 Node 0x0f [Audio Mixer] wcaps 0x200100: Mono Connection: 1 0x0b Node 0x10 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x0f, nsteps=0x0f, stepsize=0x0b, mute=1 Amp-Out vals: [0x80] Node 0x11 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x97 0x17] Connection: 1 0x03 Node 0x12 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x80 0x80] Connection: 1 0x0c Node 0x13 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x9f 0x9f] Connection: 1 0x0d Node 0x14 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 Amp-Out vals: [0x0f 0x0f] Connection: 4 0x0c 0x0d* 0x0e 0x0f Node 0x15 [Power Widget] wcaps 0x500500: Mono Power: 0x0 Connection: 15 0x05 0x06* 0x07 0x08 0x09 0x0b 0x0c 0x0d 0x0e 0x0f 0x10 0x11 0x12 0x13 0x14

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[alsa-devel] PCMCIA Audigy 2 broken in 2.6.21?
by Chuck Ebbert 04 Jul '07

04 Jul '07

Multiple users report the PCMCIA Audigy 2 adapter hangs the machine in 2.6.21 while it work OK in 2.6.20: https://bugzilla.redhat.com/bugzilla/show_bug.cgi?id=242208

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[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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