Linux-audio-dev

linux-audio-dev@lists.linuxaudio.org

7513 discussions

[linux-audio-dev] 8bit sound wav playing to a 16bit sound card...

by Derrick

I'm new to OSS Programming, and I'm attempting to play some 8bit wav files. However OSS is telling me that my sound card will not play 8bit , only 16bit. If I force it. The sound changes pitch, and is very fast. ( obviously ). Is there anyway to convert 8bit to 16bit on the fly? I've noticed that XMMS also fails to play the 8bit wav file correctly. I've even tryed to convert the file from 8bit to 16bit using SOX. But with the same results. I would like to support 8bit file wavs in my program as MOST of the wavs available are in 8bit format... Any one Have some pointers? PS: The command I used with sox is " sox -V -r 11025 -w -c 1 backup.wav temp.wav " -- >From the moment I picked your book up until I put it down I was convulsed with laughter. Some day I intend reading it. -- Groucho Marx, from "The Book of Insults"

22 years, 3 months

[linux-audio-dev] Q-Audio 1.0, Q-Midi 1.10 released

by Dr.Graef＠t-online.de

Q-Audio is a digital audio library for the Q programming language, which interfaces to Phil Burk's PortAudio and Erik de Castro Lopo's libsndfile libraries. Q-Midi is a MIDI interface for Q, built on top of Grame's MidiShare. Q is an interpreted functional programming language based on symbolic expression rewriting, providing a high-level interactive programming environment for scientific, computer music and other advanced applications. All software is distributed under the GPL. Sources, binaries and further information can be found on the Q homepage: http://www.musikwissenschaft.uni-mainz.de/~ag/q/ Enjoy! Albert Graef -- Dr. Albert Gr"af Email: Dr.Graef(a)t-online.de, ag(a)muwiinfa.geschichte.uni-mainz.de WWW: http://www.musikwissenschaft.uni-mainz.de/~ag

22 years, 3 months

[linux-audio-dev] Re: [MusE] More ideas :) [Track freeze] - Alsa Sequencer

by Robert Jonsson

tisdagen den 10 juni 2003 13.21 skrev Frank van de Pol: > On Tue, Jun 10, 2003 at 08:30:39AM +0200, Robert Jonsson wrote: > > Hi, > > > > > In fact the bounce feature in MusE is "realtime". It means that you > > > have to wait the real duration of the track to be rendered. > > > In a non "realtime" mode the track is rendered as fast as computer can. > > > > AFAICT the realtimeness of the bounce feature is like that because of > > design constraints. Okay, bouncing wavetracks should be possible in > > non-realtime, but not when using softsynths. > > > > This is because all softsynths use alsa-sequencer as the input interface. > > And if I'm not missing anything, this interface is strictly realtime > > based. (perhaps it can be tweaked by timestamping every note and sending > > them in batches? it seems very hard though.) > > You are right, with the current alsa-sequencer the softsynths are driven by > realtime events. Though an application can enqueue the events to the > priority queues with delivery timestamp, the scheduling is handled > internally by the alsa sequencer. This causes some problems (especially for > sample accurate synchronisation with JACK or LADSPA synth plugins (XAP?)), > but also for network transparency and support for MIDI interfaces which > accepts timing hints (Steinberg LTB or Emagic AMT ... if specs of the > protocol were available :-( ). > > During the LAD meeting at Karlsruhe we discussed this and sketched a > alsa-sequencer roadmap that focusses on transition of the alsa-sequencer > from kernel to userspace and better integration with softsynths / JACK. > A few things from this are very much related to your track bouncing / > off-line rendering thing: > > - Provide facility to delegate scheduling to the client. The implementation > would be to deliver the events directly (without queuing) with the > timestamp attached to the registered client port. This would allow the > client to get the events before the deadline (time at which the event > should be played) and use that additional time to put the events at the > right sample position. > > Note that for the softsynth to get advantage of this the application > should enqueue the events (a bit) ahead of time and pass the timestamp. > Some of the current applications (including MusE) use the alsa-sequencer > only as event router and drive it real-time. > > Since the softsynth/plugin has no notion of the acutal time (only the > media time and sample position), rendering at arbitrary speed should be > possible: bounce faster than realtime or even slower than realtime for > those complex patches. > > - JACK is real-time, and bound to the sample rate of the soundcard. Since > the audio sample rate can also be used as a clock master for the alsa > sequencer this would be a good option to ensure synchronisation. The > transport of JACK and alsa sequencer can be tied together (either one of > the two acting as master, a run-time configurable option) to provide > uniform transport and media time amongst the applications that hook into > the JACK and/or alsa sequencer framework. > > For the offline rendering no nice scheme has been worked out yet; I guess > it would be something along the lines where the application that owns the > sequencer queue has full control on the transport, moving media time at the > speed the frames are actually rendered, and the app(s) generating the > events keeping at least one sample frame ahead of time. > > Frank. Okay, I didn't know that this had been up on the table, how far has this work progressed, was it just the Karlsruhe meeting or has more thinking occured? (fyi I'm CC:ing LAD, it might be a more appropriate place for this discussion..). Regards, Robert

22 years, 3 months

[linux-audio-dev] horgand-0.92

by holborn

Hi! new release ...horgand-092 Program is released GNU/GPL version 2. Horgand is a organ, Jack capable and recognizes MIDI Program Change (1-32). Horgand is tested on a PIII 966 (Debian Sid ) and PII 300 (Gentoo). Changes in 0.92 ---------------------- -Added Reverb at this moment only presets available. -All the sliders and dials widget reponse in realtime. -Solved bug in keyboard level scaling, that reduced noises, unfortunatelly sound changes a little bit. -Small look changes and other minor bugs solved. REQUERIMENTS: * FAST COMPUTER * LINUX * ALSA * JACK * FLTK 1.1 Web Page : http://personal.telefonica.terra.es/web/soudfontcombi/ Josep

22 years, 3 months

[linux-audio-dev] Newbie wants to record 4 channels underwater

by Jo Evans

I am Computer Support technician at the Marine Research Lab, near Leigh, North Island, New Zealand. We are a remote department of the University of Auckland. We are building a computer based on a JENLOGIX miniature industrial motherboard into an underwater housing in order to record on the HDD the output from 4 hydrophones. We have a MIDIMAN DELTA 44 four-channel sound card and hope to be able to write a simple script to record and save the sound streams to disk automatically whenever the computer wakes up. An external timer would wake up the computer for 10mins every hour. I am a newbie to Linux but have just ordered Redhat9 and intend to get the system working on a desktop (DELL OptiplexGXa, 233MHz PII) before installing it on the U/W computer. I have found the snd-ice1712 driver module on the ALSA site. Does anyone know if there are scripts already written that would do what we want or could be readily modified? Any other advice welcomed. - - - - - - - - - - - - - - - - - - - - - - - - - - - - - Jo Evans, Network Manager, Computer Support ph ext.3601 University of Auckland ph 64 - 9 - 422 6111 Leigh Marine Laboratory fax 64 - 9 - 422 6113 P.O. Box 349, Warkworth j.evans(a)auckland.ac.nz NEW ZEALAND - - - - - - - - - - - - - - - - - - - - - -

22 years, 3 months

[linux-audio-dev] DSP hacking under Linux: Synthesizer of Death

by Jay Vaughan

Last week there was a thread on LAD about DSP kits and integration with Linux, and I mentioned that I knew of a few small DSP projects with promise, but couldn't find the URL. Found the URL today, and it is: http://www.gweep.net/~shifty/death/ And: http://www.gweep.net/~shifty/ezkit/ Seems like this might give some of you some inspiration in the "put a cheap DSP into Linux, integrate it ..." department. -- ; Jay Vaughan r&d>>music:technology:synthesizers - www.access-music.de/

22 years, 3 months

[linux-audio-dev] libakai

by Christian Schoenebeck

Hi! I have ported Sebastien Metrot's libakai to Linux some couple of weeks ago. Until it will be in CVS one day you can get it from: http://stud.fh-heilbronn.de/~cschoene/projects/libakai/ I added some code to the demo application to extract samples from a Akai disc yesterday, so you can now actually not only see but also hear what's on a disc. You will notice that there's still a small bug, I will fix it ASAP, but I won't complain if someone else will look for it ;) Best regards. Christian

22 years, 3 months

[linux-audio-dev] [OT] Question about some definition..

by Juan Linietsky

Hi! i'm doing some studying on dsp, and one thing I could never properly understand is the term of "excitation signal" I seem to find it associated to environment or natural sources, but I cant really find a definition. Could someone with enough knowledge on this subject please give me a brief on this? Thanks in advance! Juan Linietsky

22 years, 3 months

[linux-audio-dev] JACK 0.72.4 released

by Taybin

JACK 0.72.4 JACK is a low-latency audio server, written primarily for the GNU/Linux operating system. It can connect a number of different applications to an audio device, as well as allowing them to share audio between themselves. Its clients can run in their own processes (ie. as normal applications), or can they can run within the JACK server (ie. as a "plugin"). JACK is different from other audio server efforts in that it has been designed from the ground up to be suitable for professional audio work. This means that it focuses on two key areas: synchronous execution of all clients, and low latency operation. **CHANGES** * Updated documentation * Bug fixes * MacOSX port. Includes a ProjectBuilder file to help compilation. Requires PortAudio to be installed. * Ringbuffer example files added * New example client: simple transport master to demonstrate Jack's transport API. Requires GNU readline to compile. * Removed software monitoring and improved hardware monitoring semantics. Taybin Rutkin

22 years, 3 months

Re: [linux-audio-dev] 8bit sound wav playing to a 16bit sound card...

by Derrick

> I think someone else (Paul?) hit it on the head. I fyou load 2 8bit > samples and pass them to a soundcard that is expecting 16-bit samples, > you'll get a LOT of garbage - like white noise - and it will be 1/2 as long > as your input sample. > > See, each pair of 8 bits will become a 16 bit sample. > > input 8 8bit samples: > dec: 0 64 127 64 0 -64 -127 -64 > hex: 0x00 0x40 0x7f 0x40 0x00 0xc0 0x81 0xc0 > > read as 4 16bit samples (ignore endianness): > hex: 0x0040 0x7f40 0x00c0 0x81c0 > dec: 64 32576 192 -32320 > > > Notice how the waveforms don't resemble each other at all. > > > Clearer, now? I See.. So if I wanted to convert to a 16bit, how would you recommend I do this? It would seam I would need some type of Filler.... err. white noise or just blank noise to fill in the extra 8bits. I guess I could convert every bit read from the file from an 8bit (unsigned char) to a 16bit (signed short) then write it to the /dev/dsp? Bare with me here, I'm not a verteran.. So I would have to convert the unsigned char to a signed char, then to a signed short... So Not knowing how the conversion is done in the OS, I'm assuming that the resulting signed short would be padded with 'off' bits. which would come out as silence correct? ( but it's in the same sample, so you really wouldn't hear the silence ) Werd? -- It looked like something resembling white marble, which was probably what it was: something resembling white marble. -- Douglas Adams, "The Hitchhikers Guide to the Galaxy"

22 years, 3 months

Jump to page:

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

Linux-audio-dev