Linux-audio-dev

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7520 discussions

[linux-audio-dev] Re: Converting a 24bit sample to 16bit

by Juhana Sadeharju

>From: James Courtier-Dutton <James(a)superbug.co.uk> > >Is there a standard way of converting a 24bit sample to 16bit? How your 24bit audio was made? If your audio card gives out 24-bit, then you may loose bits because there are no true 24-bit converters. At top you could use a smooth hard-limiter (which processes each sample separately). It takes out random high peaks and makes the audio more analogue. At bottom the dither noise may not be necessary if the audio source's noise floor is high enough. All dither tests seems to test against the mathematical sine wave which you don't get if you do acoustic music. Noiseshaping dither is suspectible as well because one may then wonder where the high frequency noise introduced by the dither goes to. No amplifier near me have a dither noise remover. Juhana -- http://music.columbia.edu/mailman/listinfo/linux-graphics-dev for developers of open source graphics software

19 years, 5 months

[linux-audio-dev] MIDI... know how to do bank/patch changes on yamaha Motif Rack?

by Stephen Cameron

So, I've got a yamaha motif rack, and the manual (which is here, btw: http://www2.yamaha.co.jp/manual/pdf/emi/english/synth/MOTIFRACKE2.pdf ) On P. 46 of that manual describes how, and it seems like it should be brain dead simple. There's a basic recieve channel on the Motif Rack, (it claims that it is set to "1" -- I don't know if that is zero or one based, so I tried both. I think, if I send a byte sequence like: 0xb1 0xf3 0x03 0xc1 0x07 (or maybe substitute 0 for any 1 in the above, I tried them all) It ought to do something. I *can* change patches within a bank, that is the Sending 0xc0 0x25 will change the current preset. Just can't figure out the bank change thing. Well, I'm retarded, or the spec is wrong, or the motif is broken... I think I must be retarded. At the moment, I've reduced things to a few lines of C that just open up /dev/snd/midiC1D0 and write some bytes out -- I know that much works because note-on and patch change work. Just wondering if anybody else out there has a Motif and if they've ever gotten it to change banks. -- steve __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

19 years, 5 months

[linux-audio-dev] Hello and Python bindings for LADSPA

by Jono Bacon

Hi all, My name is Jono Bacon (www.jonobacon.org) and I am one of the developers working on Jokosher, a GStreamer based multi-track audio editor written in Python for the GNOME desktop. Jokosher is designed to be really simple to use, and we release our first 0.1 release on Saturday. You can find out more at www.jokosher.org. although the brand new, much improved site is released on Saturday too. I am also one of the members of LUGRadio. Anyway, for Jokosher 0.2 I want to include LADSPA support. From what I understand, LADSPA not only provides an audio layer, but also a GUI layer, so the widgets in the plug-in can be displayed in the host toolkit. I know there is an antiquated GStreamer bridge for LADSPA, but in terms of getting the GUI elements to display, I assume I need Python bindings. Do such bindings exist? Oh, and by the way, if you are in the UK, we are running an event called LUGRadio Live 2006 with over 40 speakers, a room full of exhibitors, lots of BOF sessions and a party. It is held in Wolverhampton on the 22nd and 23rd July (next weekend). It would be great to see a fraternity of Linux audio developers there. Anyone want to come along? See www.lugradio.org/live/2006 for more details. Thanks folks! Jono

19 years, 5 months

[linux-audio-dev] LV2 Turtle requirement

by carmen

i notice this file: http://lv2plug.in/spec/lv2.ttl is in the nicely readable turtle format. my main question is, whether this will be transformed to RDF-XML during 'make install' or perhaps by the developer themselves (Eg, similar to leaving a configure script around for those who dont have autoconf). mainly beacuse, it seems RDF-XML is the dominant serialization format, and having this additional dependency stand in the way means, at the very least one can't use Pyrple out of the box. i installed Redland and the Ruby and TCL bindings, and both conspicuously left out the "Redland::Parser.turtle" method, presumably beacuse i dont have it installed (and it's not in portage...) cheers

19 years, 5 months

[linux-audio-dev] [ANN] netjack-0.12 - Low Latency Network Audio Driver

by torbenh＠gmx.de

hi. netjack-0.12 is released. Netjack is a jack-driver which uses the network card. On the other end of the network there is a normal jack-client. So its possible, to share a single soundcard between several laptops. This release finally handles the packet disordering UDP does. Thus high channel counts can now be achieved. However a 24ch in/out link over 100Mbit gave me a major "net xrun" storm on vanilla 2.6.15 kernel. At a roundtrip latency of 2.9ms that is. It was reliable with 5.8ms. 16 channels gave me some "net xruns", which i could not hear though. i expect this performance to increase when using an rt-kernel with the network-irq set to rt-prio. So please report back. Additionally to the audio transport, netjack provides sample accurate transport syncronisation. The roundtrip latency is compensated for. get it while its hot at: http://sourceforge.net/project/showfiles.php?group_id=140191 there is no link on http://netjack.sf.net because the project shell servers are down. -- torben Hohn http://galan.sourceforge.net -- The graphical Audio language

19 years, 5 months

[linux-audio-dev] diskstream and jack

by conrad berhörster

Hello all, i try to implement the reading of audiofiles with ringbuffers. For this, i need two threads: a main process thread , which is triggered from jack every 1024 samples for example. This reads from a ringbuffer. And a worker thread which fills this ringbuffer. Now the question: - The worker thread is faster then the jackthread. (Sure, it should be). What is the usual way to pause the workerthread and wake up again, when the ringbuffer needs more data. -Which size should the ringbuffer have. - How should the ringbuffer be initilized when starting to play? - Any small program to have look into or a good inroduction into pthreads?. thanks c~

19 years, 5 months

[linux-audio-dev] Creative EMU1212m

by James Courtier-Dutton

Hi, Are there any people on this list who would be able to help test a Linux ALSA driver for the EMU1212m. I am now in a position to write a driver that works for this sound card. I currently have SPDIF output working, but other support will follow soon after that. I do not have an AudioDock or any ADAT equipment, so I would need particular help with testing that. Kind Regards James

19 years, 5 months

[linux-audio-dev] ANNOUNCE: Rosegarden 1.2.4 released

by Chris Cannam

ROSEGARDEN 1.2.4 RELEASED Miscellaneous locations -- Bastille day, 2006 The Rosegarden team are pleased to announce the release of version 1.2.4 of Rosegarden, an audio and MIDI sequencer and musical notation editor for Linux. http://www.rosegardenmusic.com/ The 1.2.4 release addresses several issues with the prior 1.2.3 feature release. 1.2.4 introduces no new application features. Fixes in this release include, briefly: * Avoid crash on startup if /dev/snd/seq does not exist * Fix incorrect sequencer status report ("no driver") * Fix MIDI Text Marker export * Fix text encoding for Lilypond 2.6 (UTF8 instead of ISO-8859-1) * Fix stuck notes in matrix after pressing a stop button * Fix crash when erasing a duplicated key signature * Fix crash when switching documents with a tempo editor window open * Fix incorrect sorting and insertion logic in marker editor * Fix hang in main canvas when a segment has zero duration * Fix audio preview display for repeating audio segments * Update percussion matrix when a different drum mapping is selected * Avoid crash when deleting a device with percussion matrix open * Fix matrix display for notes outside range of current key mapping * Ensure correct segment is acted on when clicking overlapping segments * Fix sequencer crash when playing back tiny audio files * Avoid display hang when too many segments overlap * Fix several build system bugs, and compilation with gcc-4.1.2 This release also includes several new MIDI device definition (.rgd) files, as well as updates for Catalan, Russian, Swedish, Czech and Italian translations, and a completely new Finnish translation from Heikki Johannes Junes. Special thanks go to Pedro Lopez-Cabanillas for preparing the release. For more information about Rosegarden and what it can do for you, please see http://www.rosegardenmusic.com/ Rosegarden is Free Software under the GNU General Public License.

19 years, 5 months

[linux-audio-dev] memory-mapped wav files

by Stephen Sinclair

A thought occured to me recently... If I am writing an application which needs to stream a large wav file, I am having to write something which reserves some memory, and loads pieces of the wav file from disk on request. Say I need to be able to jump around the file a bit, I would have to detect when the piece is not available and load it in as appropriate. ... which I realized is exactly what the OS VM paging system does. So, has anyone tried using memory-mapped files for streaming audio data? (obviously, I mean, in a non-realtime thread, using a buffer). Or would this be totally inefficient? I was thinking it could really simplify programming, just directly accessing parts of the wav file as needed, and letting the OS load it up into physical memory when it wants to. Just curious. Steve

19 years, 5 months

[linux-audio-dev] realtimeness: pthread_cond_signal vs. pipe write

by Stefan Westerfeld

Hi! I am trying to notify a high priority (nice -20 or nice -19) thread from a realtime thread (from a jack callback to be precise). Of course I want the realtime thread to not block, but I want the high priority thread to react as soon as possible. For the first implementation, I used a condition, and pthread_cond_signal. But as I can't aquire a mutex within the realtime thread, at least not always (at most with trylock()), this results in racy code. Also, it doesn't integrate well with poll(), which the high priority process should be able to use. So basically I want to switch to a pipe write() instead of pthread_cond_signal. Now the question is: am I producing priority inversion, then? I've seen that the kernel code of pipe write has code like mutex_lock(PIPE_MUTEX(*inode)); (in fs/pipe.c, linux-2.6.16.16), so can this mutex cause priority inversion? However, on the other hand, I examined the code of glibc-2.3.6/ntpl/sysdeps/pthread, and found in pthread_cond_signal.c: __pthread_cond_signal (cond) pthread_cond_t *cond; { /* Make sure we are alone. */ lll_mutex_lock (cond->__data.__lock); /* Are there any waiters to be woken? */ if (cond->__data.__total_seq > cond->__data.__wakeup_seq) { /* Yes. Mark one of them as woken. */ ++cond->__data.__wakeup_seq; ++cond->__data.__futex; /* Wake one. */ lll_futex_wake (&cond->__data.__futex, 1); } /* We are done. */ lll_mutex_unlock (cond->__data.__lock); return 0; } So it looks like there is some kind of low level mutex locking involved even when signalling a condition. Can't this also lead to priority inversion? So what is the best way to achieve that some lower priority thread will definitely wakeup from a realtime priority thread, without causing priority inversion, under recent linux kernels? Cu... Stefan -- Stefan Westerfeld, Hamburg/Germany, http://space.twc.de/~stefan

19 years, 5 months

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Linux-audio-dev