Linux-audio-dev

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7516 discussions

[linux-audio-dev] USB-MIDI Transfer rate

by Jens M Andreasen

Hi! I am currently working on digesting the USB-MIDI Class Definition ... http://www.usb.org/developers/devclass_docs/midi10.pdf As I understand, you can have up to 16 USB MidiStreams (MS), each equal to a physical midi-cable (and each "cable" having 16 virtual midi-channels.) There is a bandwith limit of one 3-byte midievent/ms which makes sense given the bandwith of of a physical midi-cable The MIDI-USB device also have a control channel without any endpoints (without any physical midi-jacks.) Again only as far as I have understood; the control channel is not a MidiStream and should therefore be able to accept a significantly higher transfer rate than the physical MidiStreams. Question: How do I determine the max transfer rate for the control channel (bInterFaceSubClass == 1) as apposed to the physical midi-outs (bInterFaceSubClass == 3) ?+ This is for setting LEDs partially lid for more than one parameter, by pulsewidth-modulation over USB-MIDI. mvh // Jens M Andreasen

20 years, 1 month

[linux-audio-dev] compiling Rosegarden-4.1.0 problems

by Michael Stein

When trying to compile rosegarden, I am getting the following error during the .configure: checking if UIC has KDE plugins available... no configure: error: you need to install kdelibs first. I am running FC4 x86-64 with gcc4. Any idea? Has anyone encountered the sam problem? I think that it may be that redhat compiled QT without the -threads option.

20 years, 1 month

[linux-audio-dev] emu10k1 questions

by Garett Shulman

Hello, I am trying to figure out what my Live 512 & alsa are capable of. I have been trying to compare what I am seeing in the system with what I have been able to find in literature and posts. I was wondering if anyone might be able to offer any clarification. First of all, I am trying to figure out what the different I/O I see are. In /etc/asound/devices I see: 4: [0- 0]: hardware dependent 8: [0- 0]: raw midi 19: [0- 3]: digital audio playback This is digital mixer output? 18: [0- 2]: digital audio playback This is Synth or FX output? 26: [0- 2]: digital audio capture FX capture? 25: [0- 1]: digital audio capture Device 1? 16: [0- 0]: digital audio playback Is this the codec playback? 24: [0- 0]: digital audio capture Is this the codec capture? 0: [0- 0]: ctl 1: : sequencer 6: [0- 2]: hardware dependent 9: [0- 1]: raw midi 10: [0- 2]: raw midi Which midi devices are which? What do the different numbers represent? subdevice: [card- device]: ? This seems a bit reasonable but kind of contradicts the output I get from aplay -l: **** List of PLAYBACK Hardware Devices **** card 0: Live [SB Live [Unknown]], device 0: emu10k1 [ADC Capture/Standard PCM Playback] Subdevices: 32/32 Subdevice #0: subdevice #0 ... Subdevice #31: subdevice #31 card 0: Live [SB Live [Unknown]], device 2: emu10k1 efx [Multichannel Capture/PT Playback] Subdevices: 8/8 Subdevice #0: subdevice #0 ... Subdevice #7: subdevice #7 card 0: Live [SB Live [Unknown]], device 3: emu10k1 [Multichannel Playback] Subdevices: 1/1 Subdevice #0: subdevice #0 Here it would seem that device 2 does not have a subdevice 26 as suggested by my (perhaps wrong) interpretation of /proc/asound/devices Thanks. -Garett

20 years, 1 month

[linux-audio-dev] AES31 lin?

by Aaron

Hi I am writing a grant for a project I am doing. The potential funder requires the archived audio be in AES 31 format. Is there any application/lib etc on lin that outputs AES31? Thanks Aaron

20 years, 1 month

[linux-audio-dev] cs46xx mic recording

by Hans Fugal

My Turtle Beach Santa Cruz (cs46xx) card has a strange problem: mic recording in JACK is really distorted. Records fine (as fine as my cheap mic can) in Audacity. arecord sounds fine as well. When I record with TimeMachine in JACK, it sounds terribly distorted (maybe saturated is the word). Same mixer settings for all of the above. Has anyone else had luck recording with this card in JACK? -- Hans Fugal | If more of us valued food and cheer and http://hans.fugal.net/ | song above hoarded gold, it would be a http://gdmxml.fugal.net/ | merrier world. | -- J.R.R. Tolkien --------------------------------------------------------------------- GnuPG Fingerprint: 6940 87C5 6610 567F 1E95 CB5E FC98 E8CD E0AA D460

20 years, 1 month

[linux-audio-dev] Re: [Linuxsampler-devel] buffers, periods, & fragments

by Garett Shulman

Flo, Thanks for the jack suggestion. I definetly do need to spend some time working with jack on my system. Also, thanks for the great low latency documentation on tapas - ugh!. After getting 2.6 setup with | Preemptible Kernel (Low-Latency Desktop), fiddeling with my hardware to get the soundcard irq's right, and some tweeking with chrt all I can say is "wow. WOW!". I don't think there is anything you can do on a 2.4 kernel to get close to this level of performance and control. Sweet! I almost giggle as the entire system temporarily hangs while linuxsampler loads gig files at outrageous speed. Also, I wanted to let you know that after a bit more hacking I was able to get dshare working with the ice1712. It's awesome. I can send linuxsampler out of channels 1-4 and have fluidsynth playing out of channels 5 & 6, with 2 channels to spare for ecasound or something... all simultaneously, at very low latency, & hardly any processor load. I paid for the hardware... might as well take advantage of it. Besides, this way I can give my processor a break, under-clock it, & keep fan volume very low. Here is a snippet from asound.conf. -Garett |pcm_slave.66_slave { pcm "hw:1,0" channels 8 rate 44100 buffer_size 256 period_size 128 } pcm.66ch1234_dshare { type dshare ipc_key 18273645 slave 66_slave bindings.0 0 bindings.1 1 bindings.2 2 gindings.3 3 } pcm.66ch1234 { type plug slave.pcm "66ch1234_dshare" } pcm.66ch56_dshare { type dshare ipc_key 18273645 slave 66_slave bindings.0 4 bindings.1 5 } pcm.66ch56 type plug slave.pcm "66ch56_dshare" } Florian Schmidt wrote: > On Thu, 15 Sep 2005 22:40:32 -0600 > Garett Shulman <shulmang(a)colorado.edu> wrote: > >> Hello, I have been fooling around with my alsa asound.conf in an attempt >> to take advantage of the hardware mixer in my ice1712 and have multiple >> apps output to it. > > > The hw mixer on the ice1712 is not doing what one normally refers to as > hardware mixing with consumer grade cards (i.e. allowing several apps > concurrent access at the same time). It mixes and routes channels from > its single 10 in/12 out channel device. > > The way to get concurrent access with an ice1712 based card is _software > mixing_. That is, ALSA needs to do it (or whatever sounddriver you use). > > I would urge you to take a very long look at JACK though. LinuxSampler > has excellent jack support and most apps made for peolpe creating music > usually have jack support, too. You'll save yourself lots of hassles > (setting up jackd isn't so tough when you read some docs). It is very > simple to route specific LS patches to specific output channels on your > ice based card when using jack. > >> In order to acomplish this I need linuxsampler to be >> able to access devices I setup in asound.conf instead of just hw:x,x >> devices. This was easy enough to acomplish by nuking all of the '"hw:" >> +' code from AudioOutputDeviceAlsa.cpp. The next issue relates to the >> fact that the alsa dshare plugin requires that all of the "virtual >> devices" that I create from the ice1712 card share the same buffer_size, >> period_size, periods, & period_time settings. So, linuxsampler crashes >> when it tries to set buffersize and periods. I figured I would just find >> out what linuxsampler was trying to use for those values, setup the same >> values in the asound.conf and then comment the code that sets them in >> linuxsampler. This seems to work except that when linuxsampler connects >> to the device I notice click and pops from the device & the output from >> linuxsampler is rather distorted. I suspect that I'm just not setting >> the values quite right in asound.conf. However, I guess I also may be >> just mangleing linuxsampler beyond proper function. :) I was wondering >> if anywone has any suggestions. When linuxsampler tries to set the >> buffersize it is using FragmentSize=128 and Fragments=2. So I am >> interpreting this as period_size=128, periods=2, & buffer_size=256. The >> slave device in asound.conf looks like this: >> pcm_slave.66_slave { >> pcm "hw:0,0' >> channels 8 >> rate 48000 >> buffer_size 256 >> period_size 128 >> periods 2 #I also tried 1 here... counting from 0... >> period_time 0 >> } >> Then I create some dshare devices from this and some plug devices for >> each dshare device and connect linuxsampler to one of the plug devices. >> Any suggestions would be greatly appreciated. -Garett > > > dshare still won't allow you to have multiple apps use your ice1712 > card. You'll need dmix for that. Have a look at alsa.opensrc.org if you > want to go this route. dmix will probably kill latency though. > > But better take a look at jackit.sf.net. > > Regards, > Flo >

20 years, 1 month

[linux-audio-dev] [OT] OGG on Tiger

by Hans Fugal

On Sat, 17 Sep 2005 at 11:20 +0200, Adrian Prantl wrote: > i'm afraid this is slightly offtopic, but does anyone know of an ogg/ > vorbis plugin for the new iTunes running on 10.4? XMMS through darwin ports works, and that is what I currently use. I've heard good things about VLC, too. While we're on the subject, I've done a little bit of research on the problem with quicktime. The qtcomponents project that worked before qt7 did things as a component, when they arguably should have made a codec from the start. Apple hadn't (and hasn't) solidified the API, and so things stopped working in Tiger and also with QT7 on Panther. I'm not sure, but if it had been done as a codec to begin with it might still be working. In any case it looks like doing it as a codec is the way to go at this stage. I think this will require basically taking Apple's AudioCodec example and wiring it up to libvorbisfile. I and at least one other person intend to do this "when I get time." If anyone out there is good with Apple codecs or good with libvorbisfile, help would be appreciated, and speed up the process. See this discussion for more information: http://tinyurl.com/8z6rb Also this bug reporter got some good info from Apple: http://tinyurl.com/cy35h -- Hans Fugal | If more of us valued food and cheer and http://hans.fugal.net/ | song above hoarded gold, it would be a http://gdmxml.fugal.net/ | merrier world. | -- J.R.R. Tolkien --------------------------------------------------------------------- GnuPG Fingerprint: 6940 87C5 6610 567F 1E95 CB5E FC98 E8CD E0AA D460

20 years, 1 month

Re: [linux-audio-dev] convenient ogg vorbis wrapper

by Conrad Parker

On Sat, Sep 17, 2005 at 08:49:27AM +1000, Erik de Castro Lopo wrote: > Richard Spindler wrote: > > > Hi, > > > > I'm looking for a convenient wrapper library for ogg vorbis, because > > what I've found on the xiph.org pages looks a little overengineered to > > me. > > > > I've used libsndfile most of the time, so this is the API Style that > > I'd prefer, basically I need 4 functions I believe: > > Conrad Parker has been working on adding Ogg Vorbis and Speex support > to libsndfile. Its been very close to working for some time now :-). aye, if you'd like to play with it, it's in an arch repository at: http://www.metadecks.org/arch/ The library dependencies etc. were described in this email (which references Erik's libsndfile--hack branch): http://music.columbia.edu/pipermail/linux-audio-dev/2005-July/013288.html cheers, Conrad.

20 years, 1 month

[linux-audio-dev] convenient ogg vorbis wrapper

by Richard Spindler

Hi, I'm looking for a convenient wrapper library for ogg vorbis, because what I've found on the xiph.org pages looks a little overengineered to me. I've used libsndfile most of the time, so this is the API Style that I'd prefer, basically I need 4 functions I believe: what_samplerate_are_you( vorbis_file ); how_long_are_you( vorbis_file ); seek_to_position( vorbis_file, 23456 ); read_data( vorbis_file, buffer ); libvorbis and libogg however have to interact in some obscure way and I honestly don't want to waste my time reinventing the wheel, while I could be hacking on a new asskicking application ;) -Richard

20 years, 1 month

[linux-audio-dev] research on lau / lad -statistics

by Michael Iber

Hello administrators, hello list members, for my research for a lecture I will be holding on opensource and audio in October in Stuttgart the following informations would be very interesting: How many people are subscribed to the lau / lad -lists. How many participate actively ? Is it possible to roughly estimate, where they are from (Amerika, US, Asia, part of Europe, Germany). Is anybody capable of providing those informations? Thank you very much, Michael

20 years, 1 month

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Linux-audio-dev