similar to: Fwd: Re: Flac multi channel

I see :( That what I would call a good struct size optimisation. Please tell me there was another reason behind this being only 3 instead of 8 or 16 bits, right ? 2017-01-25 18:30 GMT+01:00 Tor-Einar Jarnbjo <tor-einar at jarnbjo.name>: > Hello Olivier, > > the limitation is in the file format itself, as the number of channels is > encoded in a 3 bit field in the streaminfo

Hi All, I am getting this error when i try to compile the "Linphone" package by typing----- make. please help me i am feeling very frustrated with this error pasdt from 7 days i am getting this error. please help me. speexec.c: In function `speex_ec_process': speexec.c:112: `spx_int32_t' undeclared (first use in this function) speexec.c:112: (Each undeclared identifier is

Flash does support interactive audio/video, not only streaming. And there is a new protocol called RTMFP in flash 10, which is on top of UDP. For voice application, see Connect Now (www.acrobat.com) Jozsef Message: 2 Date: Fri, 03 Oct 2008 16:54:53 +0200 From: Tor-Einar Jarnbjo <tor-einar at jarnbjo.name> Subject: Re: [Speex-dev] Fwd: [Vorbis-dev] Flash Vorbis player To: Ivo Emanuel

--- Tor-Einar Jarnbjo <Tor-Einar_Jarnbjo@grosch-link.de> wrote: > > -----Ursprungliche Nachricht----- > > Von: Josh Coalson > > > > --- Tor-Einar Jarnbjo <Tor-Einar_Jarnbjo@grosch-link.de> wrote: > > > Hi, > > > > > > have I just overseen a link, or is the format description on > > > http://flac.sourceforge.net/format.html

On Fri, Jan 7, 2011 at 5:58 PM, Tor-Einar Jarnbjo <tor-einar at jarnbjo.name> wrote: > Am 07.01.2011 23:38, schrieb David Richards: >> >> I'm also interested in another concept of lossless streaming with >> flac. Lets call it broadcast flac. A problem with streaming for long >> periods of time is that the sending and receiving computers clocks go >> out

On Thu, 25 Jun 2020 at 23:11, Tor-Einar Jarnbjo <tor-einar at jarnbjo.name> wrote: > > Isn't the FLAC encoder 'tuned' for the compression of audio data at common sample rates anyway? Does it make sense to use FLAC to compress arbitrary analog data at very high sample rates as opposed to other general purpose compression algorithms? The idea is to actually use it for

Torsdag, 23 januar 2003, skrev Tor-Einar Jarnbjo <Tor-Einar_Jarnbjo@grosch- link.de>: >According to the format description, the coding method has to be 0. > >I've been using libFLAC 1.0.4 to encode the stream. I've checked my interpretation against "flac -a" and it seems to read 17 bits for each warmup sample. Here is its output: frame=168 blocksize=4608

I think the software is ripe enough for a broader audience now, so I would just like to inform you about the Java library J-Ogg, available from http://www.j-ogg.de Features: - Open source & free. - Decoding of Ogg-encapsulated Vorbis and FLAC streams. - Support for local files, single files over HTTP, icecast streams over HTTP. - Plugins for JMF (Java Media Framework). - J2ME (Java 2 Micro

> -----Ursprungliche Nachricht----- > Von: Josh Coalson > > --- Tor-Einar Jarnbjo <Tor-Einar_Jarnbjo@grosch-link.de> wrote: > > Hi, > > > > have I just overseen a link, or is the format description on > > http://flac.sourceforge.net/format.html lacking information on how to > > actually decode the residual partitions to PCM sample data? > > you

Hi! I don't find any detailed information about this in the specification, so I try a question here. If two plain, unmultiplexed Ogg/Vorbis-files are chained into one file, you end up with an Ogg-file with two logical Vorbis streams. When playing this file, would it be correct for a decoder to synchronize the two streams according to the granule position of the Ogg pages (playing them

The issue is that silent frames compress to a very small size, and the Ogg packeting layer can put more than one FLAC frame into a page. So if you have an extended period of silence with a live or rate-limited input stream, the client buffers may exhaust themselves before a new page can be put together and transmitted. > This thread has raised several good topics. It's surprising that

Torsdag, 23 januar 2003, skrev "Tor-Einar Jarnbjo" <Tor-Einar_Jarnbjo@grosch- link.de>: >- My implementation seems to decode fixed subframes now, but the lpc >subframes contains nothing but noise, and the bitstream is out of sync after >reading one. The source code I'm using for decoding the lpc subframe is >available here:

--- Tor-Einar Jarnbjo <Tor-Einar_Jarnbjo@grosch-link.de> wrote: > Torsdag, 23 januar 2003, skrev Miroslav Lichvar <lichvarm@phoenix. > inf.upol.cz>: > > >If input is 16 bit, side channel will be 17 bit (16bit - 16bit is > >17bit number). And warmup samples will be (17 - wasted_bits) bit. > > Voila, this was the source of all my frustration, sync problems

On Tue, 2008-12-23 at 00:55 +0100, Tor-Einar Jarnbjo wrote: > the Android media API (android.media) does not offer the required > capabilities to play software generated PCM samples. The first Android > SDK releases contained non-functional classes from the standard > JavaSound API (javax.sound), but these were removed in later releases. I > opened a bug regarding the JavaSound

On Fri, Jan 7, 2011 at 7:36 PM, Brian Willoughby <brianw at sounds.wa.com> wrote: > This thread has raised several good topics. ?It's surprising that the > FLAC-Dev list has been silent for years, and now suddenly there are several > good ideas to discuss. I'll take credit for this, toot toot toot :D > > On Jan 7, 2011, at 15:04, David Richards wrote: >>

Hi, Thank you both for your replies. I will try to put together a repro case when I get a chance. In our case, we are recording audio directly into FLAC files, so they can end up with very short durations. Since I sent my first email, I have also seen this reproduce with longer files. Thanks, Luke On Mon, Nov 21, 2016 at 3:07 PM, Federico Miyara <fmiyara at fceia.unr.edu.ar> wrote:

Hi all, Thanks in advance for your help, and sorry for the slow reply. I've created a small OS X example project <https://drive.google.com/file/d/0B8vFUUth7NpvRFI0MmI2bVJNNEU/view?usp=sharing> that reproduces the issue. (I've run into the size limit for emails on this list so I've put it on Google Drive.) The program just initializes a libFLAC++ stream decoder for a given file

Here's the patch submit message for 4GB+ windows barrier. http://lists.xiph.org/pipermail//flac-dev/2013-March/003804.html On Wed, Jun 12, 2013 at 3:00 PM, <flac-dev-request at xiph.org> wrote: > Send flac-dev mailing list submissions to > flac-dev at xiph.org > > To subscribe or unsubscribe via the World Wide Web, visit >

I'm trying to use a paper by Hardin & Rocke: http://handel.cipic.ucdavis.edu/~dmrocke/Robdist5.pdf as a guide for a function to identify outliers in multivariate data. Attached below is a function that is my attempt to reproduce their method and also a test to see what fraction of the data are identified as outliers. Using this function I am able to reproduce their results regarding the

Op do 3 nov. 2022 om 19:39 schreef Federico Miyara <fmiyara at fceia.unr.edu.ar>: > > > Martijn, > > Currently FLAC already stores and restores most kinds of metadata corruption without problems, so in most cases the conversion is already bit-accurate. However, there are some kinds of corruption it cannot handle. These are the kinds of corruption that invalidate your