Linux-audio-dev

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[linux-audio-dev] [ANN] Aqualung, a new music player for GNU/Linux

by Tom Szilagyi

Hi all, I am pleased to announce the first beta release (and the first public release, as well) of Aqualung, a music player for GNU/Linux -------------------------------------- Homepage: http://aqualung.sf.net Aqualung is a new music player for the GNU/Linux operating system. It plays audio files from your filesystem and has the feature of inserting _no_gaps_ between adjacent tracks. Aqualung is released under the GNU General Public License. Features at a glance ==================== Supported file formats: * Almost all sample-based, uncompressed formats (eg. WAV, AIFF, AU etc.), files encoded with FLAC (the Free Lossless Audio Codec), Ogg Vorbis and MPEG Audio files (including, but not limited to, MP3) are supported. Naturally, any of these files can be mono or stereo. Supported output devices: * OSS and ALSA driver interface, as well as support for connecting to the JACK Audio Connection Kit. Key features: * Continuous, gap-free playback of consecutive tracks! Your ears get exactly what is in the files -- no silence inserted in between. * Ability to convert sample rates between the input file and the output device, in high quality. (Thanks to libsamplerate!) * LADSPA plugin support -- you can use any suitable LADSPA plugin to enhance the music you are listening to. Some other niceties: * Internally working volume and balance controls (not touching the soundcard mixer). * Support for multiple skins; changing them is possible at any time. * Support for random seeking during playback. * Track repeat, List repeat and Shuffle mode (besides normal playback) * All windows are sizable. You can stretch the main window horizontally for more accurate seeking. * State persistence via XML config files. Aqualung will come up in the same state as it was when you closed it, including playback modes, volume & balance settings, currently processing LADSPA plugins, window sizes, positions & visibility, and other miscellaneous options. In addition to all this, Aqualung comes with a Music Store that is an XML-based music database, capable of storing various metadata about music on your computer (including, but not limited to, the names of artists, and the titles of records and tracks). This is much more efficient than the all-in-one Winamp/XMMS playlist. Hope you will like this program. Please report any problems. Tom

20 years

[linux-audio-dev] Re: [announce] [patch] Voluntary Kernel Preemption Patch

by Andrew Morton

Ingo Molnar <mingo(a)elte.hu> wrote: > > I took a > look at latencies and indeed 2.6.7 is pretty bad - latencies up to 50 > msec (!) can be easily triggered using common workloads, on fast 2GHz+ > x86 system - even when using the fully preemptible kernel! What were those workloads? Certainly 2.6+preempt is not as good as 2.4+LL at this time, but 2.6 isn't too bad either. Even under heavy filesystem load it's hard to exceed a 0.5 millisecond holdoff. There are still a few problem in the ext3 checkpoint buffer handling, but those seem pretty hard to hit. I doubt if the `Jack' testers were running `dbench 1000' during their testing. All of which makes me suspect that the problems which the `Jack' testers saw were not directly related to long periods of non-preemption in-kernel. At least, not in core kernel/fs/mm code. There have been problem in the past in places like i2c drivers, fbdev scrolling, etc. What we need to do is to encourage audio testers to use ALSA drivers, to enable CONFIG_SND_DEBUG in the kernel build and to set /proc/asound/*/*/xrun_debug and to send us the traces which result from underruns. As for the patch, well, sprinkling rescheduling points everywhere is still not the preferred approach. But adding more might_sleep() checks is a sneaky way of making it more attractive ;) Minor point: this: cond_resched(); function_which_might_sleep(); is less efficient than function_which_might_sleep(); cond_resched(); because if function_which_might_sleep() _does_ sleep, need_resched() will likely be false when we hit cond_resched(), thus saving a context switch. Unfortunately, might_sleep() calls tend to go at the entry to functions, whereas cond_resched() calls should be neat the exit point, or inside loop bodies.

20 years

[linux-audio-dev] Re: OSC vs MIDI

by Kjetil Svalastog Matheussen <k.s.matheussen＠notam02.no>

Dave Robillard: > On Wed, 2004-09-01 at 07:48, martin rumori wrote: >> On Wed, Sep 01, 2004 at 11:31:01AM +0100, Steve Harris wrote: >> > On Wed, Sep 01, 2004 at 10:03:18 +0100, Dave Griffiths wrote: >> > > so if I'm writing a osc sequencer, is the best plan to leave the >> > > mapping open for the user to modify? >> > >> > I would say so yes, its possible that an OSC schema spe will be >> > standardised at soem point that would make it easier. >> >> not to mention the microtone-capabilities of your osc sequencer and >> the sophisticated envelope control functions, which are hard to cover >> with pure midi... :-)) > > Imagine a sequencer where, instead of little straight bars representing > notes, the 'piano roll' just allowed you to draw a line to represent > frequency.. with any angle, straight or curved (bezier), etc. Wow.. > > Control could be like that too, with overlay and everything, but having > that for pitch would be amazing.. has something like this ever been done > before? > You can draw lines to represent pitch (and everything else) in Radium, but its not a piano-roll though. http://www.notam02.no/radium/ (latest linux-(not-finished)-version: http://www.notam02.no/arkiv/src/)

20 years

[linux-audio-dev] audio stream (stereo)

by Kachalov Anton

Hi guys, I'm coding a software implementation of a buffer for PIO-DA16 (icpdas). I'd like to know what a stream of stereo PCM data looks like and how a Digital-toAnalog-Converter understands, which part of data belongs to which channel in a stereo PCM stream. Rgds, Anton

20 years

[linux-audio-dev] Howto grab keys?

by Pamplemousse_mk2

Hello, I'm using PortAudio library to try to program a loop-sampler software under Linux (and perhaps Windows). Actually, I would like to make a console-based software. I would like to know how could I manage entries from keyboards. I think that ncurses lib is a good choice. But my problem is in algorithm term. I think that making a code in the main() like this: while (true) { if (key is pressed) { switch (value of the key) { case 'A': proceed to FFT ; break; case 'B': proceed to time stretch break; case 'C: reduce volume break; ... ... } } wait for a certain time; } is not a good idea. I know that with MFC there is event management with WM messages. I don't know what is there under Linux, especially in console-based apps. With GUI libs like GTK, no problems I think. But what about console-based apps? Does anybody knows some internet sites, or some other libraries, or some methods (perhaps multithreads? seems to be heavy for just event managements...). I have look at ncurses docs but they shows an example with a kind of loop like the one I just talk about above. Any advice is welcome Bye.

20 years

[linux-audio-dev] libsndfile: type combinations and string data

by martin rumori

hi erik, i've got a question regarding type combinations in libsndfile: SF_FORMAT_FLOAT | SF_ENDIAN_CPU works with SF_FORMAT_AU on any platform, which is great. but it doesn't work with neither aiff nor wav. i expected that it would work with one of the both formats, depending on the host endianess. is there a special reason why it does not? some other questions regarding string data: snd does not explicitly allow using string data in the file format. however, i remember that it was done in the old NeXT days by increasing the data offset and using the space in between for a comment. do you think this could be implemented in libsndfile for the SF_STR_COMMENT thing or is that just too non-standard? with SF_FORMAT_AIFF, i can't write string data to the file in SFM_RDWR mode. the call to sf_set_string() succeeds, but the actual data in the file is not changed. the call to sf_set_string() succeeds even when the file is in SFM_READ mode. with SF_FORMAT_WAV, a file with string data can't be opened with SFM_RDWR anymore. this means there is no way to change the audio data or the comment of the file without copying it? i guess there is a special reason for that... thanks very much for your patience... bests, martin

20 years

[linux-audio-dev] wikipedia entries for Linux audio ?

by Dave Phillips

Greetings: While doing some research on VST/VSTi technology I checked the Wikipedia page for "VST". It's a good informative page, and there's even an entry regarding the fst/libfst project. However, there was no mention of Kjetil's vstserver and its clients. There is now. It occurred to me that while we're laboring over where & whether to use wikis, there's already a place for us to get some more information out to the public: the Wikipedia. I don't know how many other Linux audio projects are detailed. Ardour, ALSA, Pd, and others are already there, but I don't know how up-to-date that material is. So, would-be documenter writers, avail thyselves of this opportunity and start hitting the wikipedia ! Add new entries, correct the old ones, bring 'em up to date, but just do it ! ;-) Thank you for your attention. We now return you to your regularly scheduled programming. Best, dp

20 years

[linux-audio-dev] libasound.so in /lib?

by Mario Lang

Hi. I recently wrote ALSA PCM and seq support for BRLTTY[1]. However, the fact that libasound.so.2 lives in /usr/lib does present a little problem for us. BRLTTY needs to be run as early as possible during startup, most likely before /usr is even mounted. Linking against -lasound the traditional way therefore is quite a bug, since the executable now doesn't start anymore if /usr is missing. We've solved this now by dlopen() and dlsym()'ing all the symbols we actually need, and call our private function pointers instead of the real library symbols. However, this solution is kind of icky. At the same time, we wrote support for QNX, which seems to have a early fork of ALSA as the QNX Sound Architecture (also called libasound). QNX apparently already puts its libasound.so in /lib, most probably because of this problem. I'd like to ask the community what you think about moving libasound from /usr/lib/ to /lib/? With OSS, that wasn't a problem at all, since the ioctl calls were simple enough that there is/was no need for a wrapper lib. But since ALSA kind of relies on that, wouldn't it make more sense to have the lib in /lib so that early boot programs can output sound too? -- CYa, Mario

20 years

[linux-audio-dev] Cakewalk file format

by Pedro Lopez-Cabanillas

Hi, I've posted the following message in Rosegarden-devel mailing list, and Michael pointed me to an old thread from you about the same subject: http://www.mail-archive.com/linux-audio-dev@music.columbia.edu/msg03322 Attached is a little program summarizing what I have found about the file format at this moment. I would like to know if you have more information, ideas or want to collaborate in this task. Regards, Pedro On Sunday 29 August 2004 01:50, Pedro Lopez-Cabanillas wrote: > Hi, > > I have many old songs, stored as Cakewalk "WRK" files, and I don't use > Cakewalk or Sonar anymore. I'm not planning to do so in the future, 'cause > I'm sure I will use Rosegarden. Of course. > > So, I need to convert my old WRK files to Rosegarden. But the WRK file is a > closed format, and not publicly documented. It will be necessary to reverse > engineer it, to rescue my old loved melodies along with the pile of shit. > > My plan is to first write a 'cake2rg' standalone utility, to start learning > the file format and meanwhile to produce some practical results. After > that, I hope to be able to import the files directly into Rosegarden. This > won't happen before 1.0-final. > > If somebody had the same idea and has the work almost finished, please let > me know ;-). Let's avoid duplicated work. > > Comments, please. > > Regards, > Pedro

20 years

[linux-audio-dev] S/PDIF, PCM, and AC-3 Capturing

by Braman, Paul

It may be that this topic is already covered somewhere, but after much searching over the past few days I have been unable to come up with a comprehensive guide to what I am attempting to do. (Note that I've never written something like this before but know enough about Linux and how things work that I would probably understand a higher-level discussion on the topic...maybe. [grins]) I am attempting to write a program that will capture and record digital audio (in the form of PCM/AC-3 over S/PDIF). What I *do* with that data is a secondary issue and not something I want to get into now. The input to the "system" is a USB digital audio capture device. (I currently have an Edirol UA-1D (http://www.edirol.com/products/info/ua1d.html) and Creative Labs MP3+ (http://www.creative.com/products/product.asp?prodid=154) at my disposal.) What I'm not sure of is what I am looking for (or what I should expect to find). The S/PDIF stream that will be coming into the device may have PCM or AC-3. My basic assumptions on this topic are that I will have to find some /dev entry to open and read from to do the capturing itself, but from everything I've read so far it doesn't look like the standard /dev/audio or /dev/dsp devices would suffice. So, there you have it. I'm looking for some pointers to get the ball rolling and I'm positive I'll be able to pick it up from there...it's always the first few steps that are daunting. The kind of questions I am asking are: 1) With the USB audio device plugged in, what device would I need to open to read the raw digital data? 2) Is there an API already available for reading this information? (I've looked at OSS, but haven't gotten much into ALSA yet.) 3) Assuming I can get at *a* bitstream, what will it contain? Will it be the S/PDIF stream from which I will need to extract PCM/AC-3, or will it be the raw PCM/AC-3 itself? Thanks. Paul Braman Paul dot Braman at NielsenMedia dot com 813-366-5053

20 years

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Linux-audio-dev