[Alsa-devel] how to cross compile the alsa util?
Hello alls: I cross compiled the alsa-lib-1.04 to arm platform. The 2.6.6 kernel including alsa driver is ok on our ARM board, so I want to test the driver. how to do it? I think maybe the first step is to crosscompile the alsa utils to play a wave audio file. CC=arm-linux-gcc ./configure --host=i686-linux --target=arm-linux make make install then compiled and installed the libasound.*/ asound.h to /usr/lib and /usr/include/sys. Then I try to cross compile the alsa-utils: CC=arm-linux-gcc ./configure --host=arm-linux --target=arm-linux checking for libasound headers version = 0.9.0... not present. configure: error: Sufficiently new version of libasound not found. So how to make it and is there any simple method to test the alsa? Thanks a lot Best Regards Bryan Wu _ Do You Yahoo!? http://cn.rd.yahoo.com/mail_cn/tag/10m/*http://cn.mail.yahoo.com/event/10m.html --- This SF.Net email is sponsored by: Oracle 10g Get certified on the hottest thing ever to hit the market... Oracle 10g. Take an Oracle 10g class now, and we'll give you the exam FREE. http://ads.osdn.com/?ad_id=3149alloc_id=8166op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] PCM Params
ok paul i had another look at the driver playback/capture tutorials. I have one question.Do applicationsinitialize and use a pcm_handle of the alsa driver when they begin interaction with it?Surely not I thought that the driver would work totally independently of my VoIP and this is why I asked the question is it possible to change driver parameters directly. Brian. Paul Davis [EMAIL PROTECTED] wrote: Im running OpenH323 a voice over ip application.What Iwere you asking about this on alsa-devel a couple of weeks ago?have found out from using it is that the alsa driveralways puts 32 msecs(256 bytes) of data into thesender buffer, or takes the same amount from thereceiver buffer.32 msecs is driver default if it's not set by theapplication? I thought if i change the relevantdefault params in alsa-driver/alsa-kernel/ andrecompile, it wud do the trick. ...Instead i should include this function call to setthe driver period..snd_pcm_hw_params_set_period_size()within my OpenH323 app with relevant header files???yes, although i am not sure you understand enough about how this allworks. and i am pretty sure i don't have time to explain. did you readeither o! f the tutorials on the ALSA site for programmers? Yahoo! Messenger - Communicate instantly..."Ping" your friends today! Download Messenger Now
[Alsa-devel] ALSA support for AudigyLS
Here is a status update. I have received an Audigy LS sound card which was kindly donated by Greg Turpin. So, far, I have managed to get some sound out of the Front speakers. My speaker-test program outputs a nice constant tone. When playing in xine, it is not perfect sound, it stops and starts, so I have some work to do there, but I just wanted to update people. Cheers James --- This SF.Net email is sponsored by: Oracle 10g Get certified on the hottest thing ever to hit the market... Oracle 10g. Take an Oracle 10g class now, and we'll give you the exam FREE. http://ads.osdn.com/?ad_id=3149alloc_id=8166op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] MAYA44 USB audio device doesn't work with 1.0.4.
Hi again all I don't know what became of the info I sent to the list previously, but below is the information I get when attempting to use this soundcard under a 2.6.6 kernel with alsa drivers 1.0.5rc1 and hotplug. dmes: usb 2-2: new full speed USB device using address 3 usbaudio: device 3 audiocontrol interface 0 has 1 input and 1 output AudioStreaming interfaces usbaudio: device 3 interface 2 altsetting 1 unsupported channels 4 framesize 2 usbaudio: device 3 interface 1 altsetting 1 unsupported channels 4 framesize 2 usbaudio: constructing mixer for Terminal 8 type 0x0301 usbaudio: registered mixer 14,16 usbaudio: constructing mixer for Terminal 12 type 0x0101 usbaudio: unit 11 invalid MIXER_UNIT descriptor (bitmap too small) usbaudio: registered mixer 14,32 usb_audio_parsecontrol: usb_audio_state at f7b00f5c hiddev96: USB HID v1.00 Device [AUDIOTRAK MAYA44 USB] on usb-:00:1d.0-2 lsusb -v Bus 004 Device 001: ID : Virtual Hub Device Descriptor: bLength18 bDescriptorType 1 bcdUSB 1.10 bDeviceClass9 Hub bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 8 idVendor 0x Virtual idProduct 0x Hub bcdDevice2.06 iManufacturer 3 Linux 2.6.6 uhci_hcd iProduct2 Intel Corp. 82801DB USB (Hub #3) iSerial 1 :00:1d.2 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 25 bNumInterfaces 1 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x40 Self Powered MaxPower0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber0 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 9 Hub bInterfaceSubClass 0 bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes3 Transfer TypeInterrupt Synch Type none wMaxPacketSize 2 bInterval 255 Language IDs: (length=4) 0409 English(US) Bus 003 Device 001: ID : Virtual Hub Device Descriptor: bLength18 bDescriptorType 1 bcdUSB 1.10 bDeviceClass9 Hub bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 8 idVendor 0x Virtual idProduct 0x Hub bcdDevice2.06 iManufacturer 3 Linux 2.6.6 uhci_hcd iProduct2 Intel Corp. 82801DB USB (Hub #2) iSerial 1 :00:1d.1 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 25 bNumInterfaces 1 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x40 Self Powered MaxPower0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber0 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 9 Hub bInterfaceSubClass 0 bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes3 Transfer TypeInterrupt Synch Type none wMaxPacketSize 2 bInterval 255 Language IDs: (length=4) 0409 English(US) Bus 002 Device 003: ID 0a92:0091 Device Descriptor: bLength18 bDescriptorType 1 bcdUSB 1.00 bDeviceClass0 Interface bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 8 idVendor 0x0a92 idProduct 0x0091 bcdDevice1.00 iManufacturer 1 AUDIOTRAK iProduct2 MAYA44 USB iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 295 bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x00 MaxPower 250mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol
Re: [Alsa-devel] ATI IXP SPDIF
At Wed, 26 May 2004 15:19:05 -0700, Alex Song wrote: hi, i happened to have that ASUS mobo (P4R800-VM) which has an ATIIXP/AD1888 and both PCM and ac3 pass through works with SPDIF. but the board i am trying to get working has an ATIIXP/ALC655 and i had a look at the realtek site and they had some drivers (http://www.realtek.com.tw/downloads/dlac97-2.aspx?lineid=5famid=12series= 8Software=True) which looks like they were based off alsa-1.0.4. just doing a quick diff i see that they added some SPDIF stuff amongst other things. after testing and some code diffing this is what i figured: alsa-1.0.4broken atiixp-spdif broken alc655-spdif realtek-alsa-1.0.4broken atiixp-spdif working alc655-spdif alsa-cvs working atiixp-spdifbroken alc655-spdif i am going to try and hack together some combination of realtek-alsa-1.0.4 and alsa-cvs and see if what i figured is right or not. it would be great if one of the alsa developers can look into realtek-alsa-1.0.4 and merge those changes into alsa. ok then the problem seems alc655 specific. after a quick look, i haven't found relavant changes for the spdif. they implemented additional switches for category but these should be already handled if you set IEC958 status bits. how did you test your board? Takashi --- This SF.Net email is sponsored by: Oracle 10g Get certified on the hottest thing ever to hit the market... Oracle 10g. Take an Oracle 10g class now, and we'll give you the exam FREE. http://ads.osdn.com/?ad_id=3149alloc_id=8166op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] PCM Params
ok paul i had another look at the driver playback/capture tutorials. I have on e question.Do applications initialize and use a pcm_handle of the alsa driver when they begin interaction with it?Surely not surely yes. ALSA isn't structured the way windows seems to concieve of audio. We don't have separate tools to configure the driver, and then have applications just use it. Each application configures the driver to the setup that it requires. It might leave many things to their default settings, but some basic configuration is nearly always done. I thought that the driver would work totally independently of my VoIP and this is why I asked the question is it possible to change driver parameters direct ly. The application changes them. There are some h/w parameters that get changed by utilities, but not the basic stuff like h/w buffer size, interrupt interval (period size), sample rates and so on. If you want a different model, in which no h/w configuration is done and you are presented with a highly abstract audio API, JACK is worth looking at. --p --- This SF.Net email is sponsored by: Oracle 10g Get certified on the hottest thing ever to hit the market... Oracle 10g. Take an Oracle 10g class now, and we'll give you the exam FREE. http://ads.osdn.com/?ad_id=3149alloc_id=8166op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] MAYA44 USB audio device doesn't work with 1.0.4.
Luke Yelavich wrote: below is the information I get when attempting to use this soundcard under a 2.6.6 kernel with alsa drivers 1.0.5rc1 and hotplug. dmes: usbaudio: device 3 audiocontrol interface 0 has 1 input and 1 output AudioStreaming interfaces This is the OSS driver (audio). Put the line audio into the file /etc/hotplug/blacklist to prevent it from loading (or deactivate it in the kernel config). HTH Clemens --- This SF.Net email is sponsored by: Oracle 10g Get certified on the hottest thing ever to hit the market... Oracle 10g. Take an Oracle 10g class now, and we'll give you the exam FREE. http://ads.osdn.com/?ad_id=3149alloc_id=8166op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] RME 9632 Mixer...
Thomas Charbonnel wrote: Ed Wildgoose wrote: Can anyone suggest how to script the controls to default to some known values? This is not a standard mixer control, alsactl is of no use here. Ed, here's a script for 1:1 routing (hdspmixer preset 1) Aha, at last the penny drops... I had mucked around with this based on the alsa wiki instructions and hadn't got it working. I (wrongly) seem to remember a post where it said this was not working anymore, and gave up. The trick is clearly that the normal alsa sound channels start from 16, and outputs start from 0. A couple of questions while I have your attention: 1) Is 32678 really 0dB? Should it be 32767? (I want to avoid any software rescaling of values where possible) 2) How can I set the output mixer values? 3) Is there any way to start hdspmixer and not load the defaults, ie so that I can experiment and reload hdspmixer in order to determine the current state of the mixer? 4) Why does the OSS emulation write to all (16) channels when an app opens it in 2 channel mode? It's very easy to do something really silly as a result (I got a rather nasty burning smell from my expensive speakers after inadvertently putting a lot of clipped signal through them at high power...) Is this an OSS issue, or just the way this driver works? (I don't have any other multichannel cards to compare against!) 5) Can you please point me to the relevant chunk of code which does the software mixing in the matrix mixer, also on the output channels. I want to understand which if any levels can be set on the card rather than by dropping bits in the driver (or if you have the time, please feel free to enlighten me). I'm trying to get highest quality signal, and wondering whether there is any kind of output based volume control on this card on the outside of the DAC, as opposed to just feeding it fewer bits. Thanks Thomas, I really appreciate the work you have done making this all work. I will try and add some notes on the alsa site for the RME 9632 on the mixer settings Ed W --- This SF.Net email is sponsored by: Oracle 10g Get certified on the hottest thing ever to hit the market... Oracle 10g. Take an Oracle 10g class now, and we'll give you the exam FREE. http://ads.osdn.com/?ad_id=3149alloc_id=8166op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] RME 9632 Mixer...
Aha, at last the penny drops... I had mucked around with this based on the alsa wiki instructions and hadn't got it working. I (wrongly) seem to remember a post where it said this was not working anymore, and gave up. There were simple mixer controls (using the MIXER api, compatible with alsamixer) that have been dropped, because they were conflicting with hdspmixer's representation. The matrix 'mixer' ctl is implemented using the HWDEP (hardware dependant) interface because it is non-standard (and thus doesn't show up in alsamixer). The trick is clearly that the normal alsa sound channels start from 16, and outputs start from 0. The syntax is amixer -c X numid=5 in,out,gain where * X is your hdsp's alsa card number * in is the input (physical: 0-15, playback 16-31 for the hdsp 9632) * out is the physical output targetted (0-15 for the 9632) * gain is the gain between 0 and 65535 (-infinity to +6dB, 32768 being 0dB) A couple of questions while I have your attention: 1) Is 32678 really 0dB? Should it be 32767? (I want to avoid any software rescaling of values where possible) 0dB is 32768. It's true that 32767 would sound more logical, but this appears to work (Jesse Chappell successfully fed an AC-3 external receiver through raw spdif, which only works if the stream is totally unaltered). 2) How can I set the output mixer values? You can't, that's the whole problem, and that's why it's not trivial to implement simple mixer controls the proper way (this is also an answer to your last post about this on alsa-user). The mixer control is just a wrapper around the way the hardware works. There is no output attenuation stage in hardware. It is simulated in hdspmixer, but in the end when you move an output fader it translates to simpler calls to the mixer ctl. This leads to your next point. 3) Is there any way to start hdspmixer and not load the defaults, ie so that I can experiment and reload hdspmixer in order to determine the current state of the mixer? No, precisely because the representation hdspmixer proposes doesn't match the hardware. There is no way for hdspmixer to reconstruct a coherent view from the matrix mixer cache in the driver. So for now there is no way to have hdspmixer and other programs access the mixer in a cooperative way. The solution (already discussed on alsa-devel) is to write a daemon to control the mixer and have all the other apps use it. I even can think of a crooked use of the alsa ctl callback mechanism to make it compatible with standard alsa mixer apps (define dummy simple mixer ctls, and have the daemon register a callback for them. when an application requires a change, the driver does nothing but the daemon is notified and does the job). 4) Why does the OSS emulation write to all (16) channels when an app opens it in 2 channel mode? It's very easy to do something really silly as a result (I got a rather nasty burning smell from my expensive speakers after inadvertently putting a lot of clipped signal through them at high power...) Is this an OSS issue, or just the way this driver works? (I don't have any other multichannel cards to compare against!) I'm aware of this problem, but still have to track it down. I guess it's related to the fact that the card can only be opened using all available channels. I don't know if oss emulation uses the plughw facility, but there must be something similar because non-interleaved streams are handled properly, there just lacks something to deal with the number of channel. 5) Can you please point me to the relevant chunk of code which does the software mixing in the matrix mixer, also on the output channels. I want to understand which if any levels can be set on the card rather than by dropping bits in the driver (or if you have the time, please feel free to enlighten me). I'm trying to get highest quality signal, and wondering whether there is any kind of output based volume control on this card on the outside of the DAC, as opposed to just feeding it fewer bits. There is no software mixing, only hardware mixing. Even hdspmixer's output stage, which is a software construction, translates to hardware calls. Thanks Thomas, I really appreciate the work you have done making this all work. I will try and add some notes on the alsa site for the RME 9632 on the mixer settings Ed W You're welcome. It would be indeed nice to add some 9632 specific notes on the alsa site, thanks. Thomas --- This SF.Net email is sponsored by: Oracle 10g Get certified on the hottest thing ever to hit the market... Oracle 10g. Take an Oracle 10g class now, and we'll give you the exam FREE. http://ads.osdn.com/?ad_id=3149alloc_id=8166op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
RE: [Alsa-devel] ATI IXP SPDIF
-Original Message- From: Takashi Iwai [mailto:[EMAIL PROTECTED] Sent: Thursday, May 27, 2004 5:45 AM To: Alex Song Cc: '[EMAIL PROTECTED]' Subject: Re: [Alsa-devel] ATI IXP SPDIF At Wed, 26 May 2004 15:19:05 -0700, Alex Song wrote: hi, i happened to have that ASUS mobo (P4R800-VM) which has an ATIIXP/AD1888 and both PCM and ac3 pass through works with SPDIF. but the board i am trying to get working has an ATIIXP/ALC655 and i had a look at the realtek site and they had some drivers (http://www.realtek.com.tw/downloads/dlac97-2.aspx?lineid=5fa mid=12series= 8Software=True) which looks like they were based off alsa-1.0.4. just doing a quick diff i see that they added some SPDIF stuff amongst other things. after testing and some code diffing this is what i figured: alsa-1.0.4 broken atiixp-spdif broken alc655-spdif realtek-alsa-1.0.4 broken atiixp-spdif working alc655-spdif alsa-cvsworking atiixp-spdifbroken alc655-spdif i am going to try and hack together some combination of realtek-alsa-1.0.4 and alsa-cvs and see if what i figured is right or not. it would be great if one of the alsa developers can look into realtek-alsa-1.0.4 and merge those changes into alsa. ok then the problem seems alc655 specific. after a quick look, i haven't found relavant changes for the spdif. they implemented additional switches for category but these should be already handled if you set IEC958 status bits. how did you test your board? Takashi what do you mean set the IEC958 status bits? who sets it? atiixp.c? my board is a gigabyte GA-8TRS300M with ATIIXP/ALC655 and i test by running aplay with a wav file and ac3dec with an ac3 file. with alsa-cvs using bith aplay and ac3dec, software says it is playing but i get no sound, the optical spdif light is not lit. with realtek-alsa-1.0.4, the software says it is playing, still no sound but the optical spdif light is on. i am loading the atiixp module without options, just defaults. alex --- This SF.Net email is sponsored by: Oracle 10g Get certified on the hottest thing ever to hit the market... Oracle 10g. Take an Oracle 10g class now, and we'll give you the exam FREE. http://ads.osdn.com/?ad_id=3149alloc_id=8166op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] [PATCH] Audigy LS support.
Here is my first go at Audigy LS support.
It can play sound to the front speakers.
Only one mixer volume control does anything, that is the Front Volume
control in alsamixer.
I attach 2 files. One is a new audigyls.c, and the other is a patch to
get alsa-driver to build with it.
audigyls.c should be added to directory ./alsa-driver/pci/emu10k1/
Support for more output channels will be added later, but this is just a
proof of concept, so that other people wishing to write audigyls drivers
for other operating systems can make a start.
Cheers
James
Index: alsa-driver/pci/Kconfig
===
RCS file: /cvsroot/alsa/alsa-driver/pci/Kconfig,v
retrieving revision 1.19
diff -u -r1.19 Kconfig
--- alsa-driver/pci/Kconfig 24 May 2004 13:27:31 - 1.19
+++ alsa-driver/pci/Kconfig 27 May 2004 18:35:31 -
@@ -26,6 +26,14 @@
Say 'Y' or 'M' to include support for Sound Blaster Live Dell
OEM version.
+config SND_AUDIGYLS
+ tristate SB Audigy LS
+ depends on SND
+ select SND_AC97_CODEC
+ help
+ Say 'Y' or 'M' to include support for Sound Blaster Live Dell
+ OEM version.
+
config SND_ATIIXP_MODEM
tristate ATI IXP 150/200/250 Modem
depends on SND
Index: alsa-driver/pci/emu10k1/Makefile
===
RCS file: /cvsroot/alsa/alsa-driver/pci/emu10k1/Makefile,v
retrieving revision 1.6
diff -u -r1.6 Makefile
--- alsa-driver/pci/emu10k1/Makefile14 May 2004 13:42:49 - 1.6
+++ alsa-driver/pci/emu10k1/Makefile27 May 2004 18:35:31 -
@@ -6,8 +6,10 @@
include $(SND_TOPDIR)/Makefile.conf
snd-emu10k1x-objs := emu10k1x.o
+snd-audigyls-objs := audigyls.o
obj-$(CONFIG_SND_EMU10K1X) += snd-emu10k1x.o
+obj-$(CONFIG_SND_AUDIGYLS) += snd-audigyls.o
export-objs := emu10k1_main.o
/*
* Copyright (c) by James Courtier-Dutton [EMAIL PROTECTED]
* Driver AUDIGYLS chips
*
* BUGS:
*--
*
* TODO:
*Surround and Center/LFE playback.
*Capture.
*SPDIF playback.
*Other rates apart from 48khz.
*MIDI
*--
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*
*/
#include sound/driver.h
#include linux/init.h
#include linux/interrupt.h
#include linux/pci.h
#include linux/slab.h
#include linux/moduleparam.h
#include sound/core.h
#include sound/initval.h
#include sound/pcm.h
#include sound/ac97_codec.h
#include sound/info.h
MODULE_AUTHOR(James Courtier-Dutton [EMAIL PROTECTED]);
MODULE_DESCRIPTION(AUDIGYLS);
MODULE_LICENSE(GPL);
MODULE_CLASSES({sound});
MODULE_DEVICES({{Creative SB Audigy LS});
// module parameters (see Module Parameters)
static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;
static int boot_devs;
module_param_array(index, int, boot_devs, 0444);
MODULE_PARM_DESC(index, Index value for the AUDIGYLS soundcard.);
MODULE_PARM_SYNTAX(index, SNDRV_INDEX_DESC);
module_param_array(id, charp, boot_devs, 0444);
MODULE_PARM_DESC(id, ID string for the AUDIGYLS soundcard.);
MODULE_PARM_SYNTAX(id, SNDRV_ID_DESC);
module_param_array(enable, bool, boot_devs, 0444);
MODULE_PARM_DESC(enable, Enable the AUDIGYLS soundcard.);
MODULE_PARM_SYNTAX(enable, SNDRV_ENABLE_DESC);
//
/* PCI function 0 registers, address = val + PCIBASE0 */
//
#define PTR 0x00 /* Indexed register set pointer register */
/* NOTE: The CHANNELNUM and ADDRESS words can */
/* be modified independently of each other. */
#define DATA 0x04 /* Indexed register set data register */
#define IPR 0x08 /* Global interrupt pending register */
/* Clear pending interrupts by writing a 1 to */
/* the relevant bits and zero to the other bits */
#define IPR_CH_0_LOOP 0x0800 /* Channel 0 loop */
#define IPR_CH_0_HALF_LOOP 0x0100 /* Channel 0 half loop */
#define INTE 0x0c /* Interrupt enable register */
#define INTE_CH_0_LOOP
[Alsa-devel] [ANN] Open Source US-X2Y firmware
Hi, if someone is interested in a free (I mean GPL) firmware for the Tascam US-X2Y devices I can offer an open source replacement for the second stage loader alsa-firmware/usx2yloader/tascam_loader.ihx. All three US-X2Y devices need this file. But I have only tried it out with an US-122 where it works. Be careful. This software comes with ABSOLUTELY NO WARRANTY! http://www.langerland.de/us122/usx2y-fw-0.1.tar.bz2 martin -- Living on earth may be expensive, but it includes an annual free trip around the sun. --- This SF.Net email is sponsored by: Oracle 10g Get certified on the hottest thing ever to hit the market... Oracle 10g. Take an Oracle 10g class now, and we'll give you the exam FREE. http://ads.osdn.com/?ad_id=3149alloc_id=8166op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] MAYA44 USB audio device doesn't work with 1.0.4.
At 11:14 PM 27/05/2004, Clemens Ladisch wrote: Luke Yelavich wrote: below is the information I get when attempting to use this soundcard under a 2.6.6 kernel with alsa drivers 1.0.5rc1 and hotplug. dmes: usbaudio: device 3 audiocontrol interface 0 has 1 input and 1 output AudioStreaming interfaces This is the OSS driver (audio). Put the line audio into the file /etc/hotplug/blacklist to prevent it from loading (or deactivate it in the kernel config). Clemens This made no difference. I completely disabled hotplug and loaded the modules manually. First the usb controller module, and then the snd-usb-audio module. The output of dmesg is below usbcore: registered new driver usbfs usbcore: registered new driver hub USB Universal Host Controller Interface driver v2.2 uhci_hcd :00:1d.0: Intel Corp. 82801DB USB (Hub #1) PCI: Setting latency timer of device :00:1d.0 to 64 uhci_hcd :00:1d.0: irq 16, io base d800 uhci_hcd :00:1d.0: new USB bus registered, assigned bus number 1 hub 1-0:1.0: USB hub found hub 1-0:1.0: 2 ports detected uhci_hcd :00:1d.1: Intel Corp. 82801DB USB (Hub #2) PCI: Setting latency timer of device :00:1d.1 to 64 uhci_hcd :00:1d.1: irq 19, io base d400 uhci_hcd :00:1d.1: new USB bus registered, assigned bus number 2 hub 2-0:1.0: USB hub found hub 2-0:1.0: 2 ports detected uhci_hcd :00:1d.2: Intel Corp. 82801DB USB (Hub #3) PCI: Setting latency timer of device :00:1d.2 to 64 uhci_hcd :00:1d.2: irq 18, io base d000 uhci_hcd :00:1d.2: new USB bus registered, assigned bus number 3 hub 3-0:1.0: USB hub found hub 3-0:1.0: 2 ports detected usb 1-2: new full speed USB device using address 2 usb 1-2: device not accepting address 2, error -71 usb 1-2: new full speed USB device using address 3 usb 1-2: device not accepting address 3, error -71 usbcore: registered new driver snd-usb-audio -- Luke Yelavich http://www.audioslack.com [EMAIL PROTECTED] --- This SF.Net email is sponsored by: Oracle 10g Get certified on the hottest thing ever to hit the market... Oracle 10g. Take an Oracle 10g class now, and we'll give you the exam FREE. http://ads.osdn.com/?ad_id=3149alloc_id=8166op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] [alsa-dev]Question regarding period/buffer and error handling
Hi there, For my research, I need to use audio with matlab under linux, and sound support of matlab is kind of... well, crappy (basically, it is opening the /dev/audio file and write to it; on my computer, it doesn't seem to work). That's why I am thinking about using a wrapper to alsa. Thanks to some tutorials (by the way, thanks for your tutorial, Paul), I managed to play sound with matlab and alsa, but I would like my wrapper to be full proof against errors. Basically, here is my method : - parse the input arguments, convert from matlab representation to 'normal C' representation (that is, vectors of double for samples). - To see if any soundcard is present, I am using snd_card_next(card), with card value to -1. - I allocate memory for a snd_pcm_hw_params_t structure, open a pcm device in NON_BLOCK_MODE, set the mode to blocking mode, set the format (16 bits LE), sampling rate, interleaved format and number of channels - I write to the pcm device using writei. My first problem is with writei: I cut my audio input in blocks of a small size before sending them to writei( blocks are 1024 frames long, for example), and depending of the sampling rate, writei doesn't write the whole block. For example, at 44100 Hz, if the block is 1024 frames long, only the 940 first frames are written. I think this problem is related to period size and so on, but I don't really understand the differences between period and buffer. Where can I find some informations about that ? My second problem is related to snd_card_next: If the card value returned by snd_card_next is different from -1, does that always mean than a alsa audio device is present on the system ? Is it the best method to probe a soundcard with alsa support? My third problem is related to the error handling: right now, with some hack, my wrapper works. But if I stop the wrapper with ctl+C, the pcm device is not closed properly, which means I cannot use the pcm device again (except by closing matlab and launching it again). Is there a way to close the pcm device before opening it ? ( I first thought about catching the ctrl+C 'signal' from matlab in my wrapper, to exit cleanly, but for now, I didn't find any way to do it). Thank you, David --- This SF.Net email is sponsored by: Oracle 10g Get certified on the hottest thing ever to hit the market... Oracle 10g. Take an Oracle 10g class now, and we'll give you the exam FREE. http://ads.osdn.com/?ad_id=3149alloc_id=8166op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
