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

[alsa-devel] [ASoc] TDM over one SSC channel on AT91 possible?
by Friedrich Lobenstock 24 Jun '11

24 Jun '11

Hello! The background: I'm just designing an ATMEL AT91RM9200 based board which got two TLV320AIC33 on board. It is either possible to hook up the two TLV320AIC33 to a separate SSC (synchronous serial control) channel each or use one SSC channel and configure it for TDM (time division multiplex) mode. There is already a driver for TLV320AIC14k/TLV320AIC2k in the development tree, so adding support for the TLV320AIC33 should be feasible. What I'm not sure about is the fact if ASOC is prepared for using channels, especially SSC, in TDM mode. Is the core ASOC structure prepared to handle that? Would we be better off instantiate one drive twice each time with different parameters for the SSC channel? Thanks for any feedback. -- MfG / Regards Friedrich Lobenstock _________________________________________________________ SCOTTY Group Austria GmbH a SCOTTY Group plc company, http://www.scottygroup.com/ _________________________________________________________

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Re: [alsa-devel] Patchmix-style application for EMU cards
by Camilo Polymeris 30 Aug '10

30 Aug '10

Hello > That is more or less what I had figured out, but it how are playback_1 to > > _16 channels routed to DSP0-15? It seems that it depends on the setting > of > > Front, Surround, LFE, etc. mixer controls. If they are turned up, I > don't > > understand what happens, sound to playback channels appears on various > DSP > > buses, at different levels. > > Really? Well, I didn't much experiment with surround mode. In source code, > I > can see that there are up to playback 8 channels at hw:0,0 with can be > controlled by faders in alsamixer. I tried surround40 and surroun51 on my > emu1616m and it worked as expected. The channles were mapped to first > couple > of DSP ports in a correct order. In fact this should work the same as > audigy > does, except the result is passed to DSP ports instead of physical > outputs. > On my card, playback channels 1 & 2 (hw:0,3) appear on all DSP busses associated with surround. Say you turn up the "Side" fader in alsamixer, channels 1 & 2 will appear at the relevant DSPs (6-7, I think) Whatever, for now I am just keeping all surround controls down (that is, Front, Surround, Center, Side & LFE). > Anyway, I think if there is another layer of routing done in audigy DSP, > it > shouldn't be used in EMU specific application. Or at least not for now. > The > most important thing is to provide all the switches and FPGA routing > scheme > for each EMU model. Or, to be precise, three routing schemes for each EMU > model, because for "double speed" (e.g. 96kHz) and "quadro speed" (e.g. > 192kHz) modes of operation the numebr of both physical and virtual > channels > changes. > I think the switches and enums for the card models are rather trivial, one can handle it with a customized version of qamix (done with a simple XML file), for example. A separate application would not be necessary for only that. > > If I turn the Front/Surround/LFE/etc. faders down, I can mix playback > > channels to DSP using the "Multichannel PCM Send/Routing/Volume" faders, > in > > a more or less intuitive way, besides that routing values do not map > > directly to DSP buses (i.e. 14-15 map to DSP0-1, but 2-3 map to DSP2-3). > Handling of this controls motivated me to write the application in the first place. No sound at all will appear if I don't set the appropiate controls for each playback channel: Multichannel PCM Volume (full), Multichannel PCM Send (an index of 8 controls, the first 4 select the target DSP) & Multichannel PCM Send Volume (again faders 0-3 of 8) By the way, only DSP0-7 work in this way. I have no idea how to make DSP8-31 work. On which card you did tested it? The driver was primary written for the > 1820M/1212M V1, so some things may not work well on other models and need > to > be fixed. I have a 1212m, don't know which version, but I think it's V1-- it does have firewire. I am using driver version 1.0.16. Thanks, Camilo.

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[alsa-devel] X-Fi support in ALSA.
by James Courtier-Dutton 18 Sep '08

18 Sep '08

Hi, I have now had a chance to read some of the specs of the X-Fi cards. aka. emu20k1. It turns out that some of the X-Fi cards have a compatibility mode. By default, they boot up in compatibility mode. The interesting point is that this "compatibility mode" is in fact the intel-HD standard so could use the intel-hda driver. For one X-Fi card with IDs: Device 1102:0009 Subsystem: 1102:0010 works a bit with the ALSA hda-intel driver. So far, the off the shelf hda-intel driver, with simple PCIs adjustment, works for the above card, but only sound capture works so far, so probably some quirk would be needed. This compatibility mode works with the Windows Vista UAA drivers. Not all X-Fi cards support with compatibility mode. For example, any card with (after a cold boot into Linux) Device 1102:0005 Subsystem: xxxx:xxxx will be X-Fi only. With regard to actual full X-Fi support, I am still in the progress of creating a GPL emu20k1.h file, that is the first step towards an snd-emu20k1 native ALSA driver. James

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[alsa-devel] What does 0dB refer to? (Logitech USB Speakers)
by Lennart Poettering 10 Jun '08

10 Jun '08

Heya! I think I discussed this with both James and Takashi a while back: it would be great if we could initialize the sound card mixers in a better way by default and not everything set to 0. i.e. for output only PCM should be active, at a sensible volume, no bass/treble/3d enabled or whatever. All other sources should be muted. And that for input the mic jack should be active and nothing else, again. Right now people too often have to invest a lot of time to find the right controls to toggle/change in the mixer to get something sensible out of the speakers. Some distributions have started to ship special scripts that try to initialize the volumes properly. I know for the Fedora case that this tool is very very simple, and just initializes PCM and Master to 70% and that's basically it. It would be highly preferable if we had a sane tool like this from alsa upstream, which would set the default values to 0dB, possibly as an extension to "alsactl". I started to hack something similar to this as part of PulseAudio: if PA finds that the PCM slider is not really useful for volume control (no attenuation to at least -60dB, too few steps, no seperate controls for all channels, ...) it falls back to software volume control, however tries to initialize the the slider to 0dB first. Now the thing is, this mostly works fine, however on one of my devices, a pair of external Logitech USB speakers (ID 046d:0a04 Logitech, Inc. V20 portable speakers), the dB scale seems to be totally bogus. According to ALSA volume ranges from -41 dB to +3dB. However, Every setting > -41 dB makes audio awfully loud. Really, really fucking loud that is. Disco loud. As loud that the speakers start jump around centimeters due to the massive vibrations. Setting the volume to 0dB with those speakers really makes you fear you are trashing them. I somehow expected that the magic "0dB" setting in ALSA would refer to some sane default, where the final output would have a sensible volume. In this case this is not so. I am thus wondering how ALSA actually defines 0dB? Is this a bug in the hardware that should be worked around in ALSA? Is this an ALSA bug? Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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[alsa-devel] [PATCH] improved snd-aloop quality when using certain samplerates and kernel HZ
by Ahmet İnan 02 Jun '08

02 Jun '08

after using a HZ:=300 kernel for some time now and needing snd-aloop again i found out, that i either had to correct the hardcoded number, or fix the problem permanently, which couses stuttering and other problems. i also removed the SYNC on start code, not just because its more or less useless and brings problems, but ive got a nice idea how to fix this overrun/underrun problem in a much nicer way. so stay tuned for the next patch, until its implemented. this patch here is more or less experimental, id like to hear some feedback. http://www.mathematik.uni-freiburg.de/IAM/homepages/ainan/alsa-driver-1.0.1… Signed-off-by: Ahmet İnan <ainan <at> mathematik.uni-freiburg.de> please also include my email address when responding - i have no intention to enable recieving emails from the list. ahmet -- admin der abteilung für angewandte mathematik, tel. 0761-203-5626

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[alsa-devel] MIDI on ice1724 - preliminary findings and questions
by Pavel Hofman 20 May '08

20 May '08

Hi, I am trying to fix the ancient MIDI problem with ice1724 cards. I applied Takashi's patch from http://mailman.alsa-project.org/pipermail/alsa-devel/2007-April/000641.html and made some minor changes (MPU401_INFO_INPUT | MPU401_INFO_OUTPUT to mpu401_uart info_flags, added a few debugs for testing). I can test only MIDI input using my MIDI keyboard, I have no MIDI output device available. The input works fine now, the output most probably too since the method snd_vt1724_mpu401_write gets called when transmitting some data with amidi. But here is the problem I do not understand: When no MIDI input/output is used, the interrupt handler snd_vt1724_interrupt gets called with interrupt status 0x10 which correctly refers to the audio data interrupt (VT1724_IRQ_MTPCM). However, when MIDI input or output is used (opened) even for a short time after the module is loaded, any subsequent audio playback generates interrupt status of 0x30 which refers to VT1724_IRQ_MTPCM AND VT1724_IRQ_MPU_TX. I have no idea why VT1724_IRQ_MPU_TX gets generated. The first run of the snd_vt1724_interrupt loop tries to handle the MPU TX status, calling uart_interrupt_tx in mpu401_uart.c: if (test_bit(MPU401_MODE_BIT_OUTPUT, &mpu->mode) && test_bit(MPU401_MODE_BIT_OUTPUT_TRIGGER, &mpu->mode)) { spin_lock_irqsave(&mpu->output_lock, flags); snd_mpu401_uart_output_write(mpu); spin_unlock_irqrestore(&mpu->output_lock, flags); } However, since the MPU output is not open at this time (no MIDI is being transmitted), the call to snd_mpu401_uart_output_write is skipped. Even though I think the interrupt status should get cleared by outb(status, ICEREG1724(ice, IRQSTAT)) the loop repeats with the MPU TX interrupt status enabled until the timeout check breaks the while loop, returning to the interrupt handler the next moment again. Apr 19 22:20:47 nahore kernel: [ 2402.938245] uart interrupt: status 0x30 Apr 19 22:20:47 nahore kernel: [ 2402.938260] uart interrupt: status 0x20 Apr 19 22:20:47 nahore kernel: [ 2402.938267] uart interrupt: status 0x20 Apr 19 22:20:47 nahore kernel: [ 2402.938274] uart interrupt: status 0x20 Apr 19 22:20:47 nahore kernel: [ 2402.938281] uart interrupt: status 0x20 Apr 19 22:20:47 nahore kernel: [ 2402.938288] uart interrupt: status 0x20 Apr 19 22:20:47 nahore kernel: [ 2402.938295] uart interrupt: status 0x20 Apr 19 22:20:47 nahore kernel: [ 2402.938302] uart interrupt: status 0x20 Apr 19 22:20:47 nahore kernel: [ 2402.938309] uart interrupt: status 0x20 Apr 19 22:20:47 nahore kernel: [ 2402.938316] uart interrupt: status 0x20 Apr 19 22:20:47 nahore kernel: [ 2402.938323] uart interrupt: status 0x20 Apr 19 22:20:47 nahore kernel: [ 2402.938326] ice1724: Too long irq loop, status = 0x20 Apr 19 22:20:47 nahore kernel: [ 2402.961247] uart interrupt: status 0x30 Apr 19 22:20:47 nahore kernel: [ 2402.961278] uart interrupt: status 0x20 Apr 19 22:20:47 nahore kernel: [ 2402.961285] uart interrupt: status 0x20 Apr 19 22:20:47 nahore kernel: [ 2402.961292] uart interrupt: status 0x20 Apr 19 22:20:47 nahore kernel: [ 2402.961299] uart interrupt: status 0x20 Apr 19 22:20:47 nahore kernel: [ 2402.961306] uart interrupt: status 0x20 Apr 19 22:20:47 nahore kernel: [ 2402.961313] uart interrupt: status 0x20 Apr 19 22:20:47 nahore kernel: [ 2402.961320] uart interrupt: status 0x20 Apr 19 22:20:47 nahore kernel: [ 2402.961327] uart interrupt: status 0x20 Apr 19 22:20:47 nahore kernel: [ 2402.961334] uart interrupt: status 0x20 Apr 19 22:20:47 nahore kernel: [ 2402.961341] uart interrupt: status 0x20 Apr 19 22:20:47 nahore kernel: [ 2402.961344] ice1724: Too long irq loop, status = 0x20 After the playback stops, the interrupts are gone too. It seems as if the playback interrupt initiates the MPU TX interrupt. If we could avoid generating the MPU TX interrupt during regular playback, I believe the major problem would be resolved. Even if I mask the interrupts (CCS01) and do not explicitly unmask them (according to proc ice1724 the CCS01 register stays at 0xa0), the interrupt gets generated. I am enclosing the testing patch producing the above given results. Thanks a lot for any suggestion or help. Regards, Pavel.

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[alsa-devel] ca0106 Digital Output at 44.1kHz
by Ben Stanley 20 May '08

20 May '08

Dear List, Hello. I'm new here. This is my first attempt to hack an alsa driver. I recently purchased a "Sound Blaster Audigy Value!" card for use in a MythTV box. As I have a digital receiver, I was mainly concerned with getting a working S/PDIF output (coax or optical) and not too much more. Summary: I can make 44.1kHz digital output work, but I think that the driver model of the card having 3 or 4 digital output channels is incorrect. I believe the card only has 1 digital output channel. Details: The card I have is characterised by the following: http://www.soundblaster.com/products/product.asp?category=1&subcategory=205… Model: SB0570 serial: 100a1102 Also known as: Sound Blaster Audigy SE During initial testing, I noticed that 44.1kHz playback was not implemented. Subsequent testing shows that speaker-test works fine with 48kHz, 96kHz and 192kHz. (I only tested 16bit output so far.) I noticed in the source code that 44.1k was explicitly disabled. I added code in snd_ca0106_pcm_prepare_playback to set up this rate for S/PDIF output as per the comments in ca0106.h . Initial tests using hw:0,0 at 44.1kHz produced recognisable signals with some noisy corruption. Later I accidentally discovered that serially opening hw:0,2 , hw:0,1 and hw:0,0 at 44.1kHz then produces perfectly good 44.1kHz sampled digital audio output. Removing hw:0,1 or hw:0,2 from this sequence causes noisy corruption. It seems that channels 0-2 in reg40 must all be set to the same sampling frequency for S/PDIF to work where 44.1kHz is concerned. Conversely, to sucessfully output 48kHz again, I have to open hw:0,2 , hw:0,1 and hw:0,0 at 48kHz to restore proper output. I do not have such troubles with 96kHz and 192kHz, for which it suffices to just open hw:0,0 at the relevant sampling rate. Using these tricks, I can successfully output 44.1kHz encoded Dolby Digital and DTS signals (found at http://www.sr.se/cgi-bin/mall/index.asp?programid=2445 , as mentioned at http://alsa.opensrc.org/DigitalOut . ) Later, I examined the product manual (from a Windows compressed helpfile on the CD), and noticed that Creative instruct you to connect to the blue socket in only two ways: 1) with a mono 3.5mm jack plug to RCA plug -> S/PDIF input to external device, and 2) a proprietary digital I/O module (see http://www.soundblaster.com/products/product.asp?category=1&subcategory=16&… ) The digital I/O module only has one input and one output (the optical/coaxial pairs are duplicates of each other). The manual does not say how to connect a S/PDIF input signal other than through the digital I/O module. This suggests to me that the following comment near the top of ca0106_main.c is incorrect for the SB0570: * ( In theory one could output 3 different AC3 streams at once, to 3 different SPDIF outputs. ) I think you can only output one AC3 stream, and that there is only one S/PDIF output. Given this knowledge of the digital I/O module, I suggest that the pinout *may* possibly be: 1 (tip): S/PDIF output 2 : S/PDIF input 3 : power to digital I/O module 4 : gnd I have not been able to get any S/PDIF signal out of pin 2 (I only have a stereo cable to test with at the moment). I will attempt to test pin 2 as an input at some point. I think that this card can only output one S/PDIF channel, given the discovery of the required clock settings above, and that Creative's own hardware only gives you access to one output. Anyway, I don't have access to documentation. From reading around, I believe James has the docs under NDA from Creative Labs. Perhaps James could confirm these suspicions? Anyway, this is speculation. I'd love to have the docs. I haven't tried to get them. Is it likely/unlikely that I would get them? I'm currently working on kernel 2.6.22 from Ubuntu. They don't seem to have changed the ca0106 driver, and not much has happened there (for output) in the newer kernels. Please find attached my current patch, which is only suitable for the purposes of discussion. I need to clarify some things about channel arrangements with those who have the documentation before I begin to try to fix things up. This patch was written with a faulty understanding of the interaction between the channel model and the spdif_enable. I don't know how any of this affects other models in the ca0106 range. So, I think that before I go any further here, I need some feedback, and access to some documentation would be useful. Ben Stanley.

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[alsa-devel] [PATCH] alsa-info.sh - Add another distro to search for
by Travis Place 20 May '08

20 May '08

Just allows finding of "Pardus" distro. diff -r e167861901c7 alsa-info.sh --- a/alsa-info.sh Tue Feb 26 10:55:47 2008 +0100 +++ b/alsa-info.sh Tue Apr 29 00:09:20 2008 +1000 @@ -1,6 +1,6 @@ #!/bin/bash -SCRIPT_VERSION=0.4.41 +SCRIPT_VERSION=0.4.42 CHANGELOG="http://hg.alsa-project.org/alsa/log/tip/alsa-info.sh" ################################################################################# @@ -270,7 +270,7 @@ mkdir $TEMPDIR 2>/dev/null mkdir $TEMPDIR 2>/dev/null #Fetch the info and store in temp files/variables -DISTRO=`grep -ihs "buntu\|SUSE\|Fedora\|PCLinuxOS\|MEPIS\|Mandriva\|Debian\| Damn\|Sabayon\|Slackware\|KNOPPIX\|Gentoo\|Zenwalk\|Mint\|Kubuntu\|FreeBSD\| Puppy\|Freespire\|Vector\|Dreamlinux\|CentOS\|Arch\|Xandros\|Elive\|SLAX\| Red\|BSD\|KANOTIX\|Nexenta\|Foresight\|GeeXboX\|Frugalware\|64\| SystemRescue\|Novell\|Solaris\|BackTrack\| KateOS" /etc/{issue,*release,*version}` +DISTRO=`grep -ihs "buntu\|SUSE\|Fedora\|PCLinuxOS\|MEPIS\|Mandriva\|Debian\| Damn\|Sabayon\|Slackware\|KNOPPIX\|Gentoo\|Zenwalk\|Mint\|Kubuntu\|FreeBSD\| Puppy\|Freespire\|Vector\|Dreamlinux\|CentOS\|Arch\|Xandros\|Elive\|SLAX\| Red\|BSD\|KANOTIX\|Nexenta\|Foresight\|GeeXboX\|Frugalware\|64\| SystemRescue\|Novell\|Solaris\|BackTrack\|KateOS\| Pardus" /etc/{issue,*release,*version}` KERNEL_VERSION=`uname -r` KERNEL_PROCESSOR=`uname -p` KERNEL_MACHINE=`uname -m` -- Happiness is like peeing your pants. Everyone can see it, but only you can feel its warmth

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[alsa-devel] [PATCH] snd-usb-us122l v0.2 for US-122L
by Karsten Wiese 20 May '08

20 May '08

Hi, attached is a fresh US-122L linux(test-)driver. If everything is installed on top a kernel 2.6.25-rc5 and alsa-lib 1.0.16, everything works, erm at least here. Without the ehci patch, it still works when connected to the uhci-hcd. (Maybe the ehci-patch makes it into .26) alsa-lib < 1.0.16 lets the plugin misbehave. PCM access needs an ~/.asoundrc containing whats in .asoundrc. Its established by patching/compiling/installing alsa-plugins. You need to adjust the path to the kernel's sound/usb/usx2y/usb_stream.h Only S24_3LE, 2 channels interleaved, 2 periods is supported. The userspace pcm devicename then is forexample "usb_stream:1" (To let alsa convert to other formats, use forexample "plug:usb_stream:1") Tested with aplay, arecord, amidi & jackd using alsa driver. Propably doesn't do 32bit PCM access on x86_64, tested on x86_64 only. Karsten

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[alsa-devel] Application doing forward/rewind
by Aggarwal, Anuj 18 May '08

18 May '08

Hi, I am new to ALSA framework and currently working on an embedded platform. My ALSA driver works fine with APLAY and ARECORD utilities, which are coming along with the alsa-utils package. But they don't support forward and rewind. I wrote my own app for the same but the APIs are returning me 0, although some frames have been forwarded. I searched a lot for some simple command line based audio player which can provide me this sort of functionality but I could not find one. Can someone help? If I have missed something, please let me know. Thanks & Regards, Anuj Aggarwal

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