similar to: UDP in flash

-------- Forwarded Message -------- Subject: Re: [flac-dev] Flac multi channel Date: Wed, 25 Jan 2017 15:55:14 -0300 From: Federico Miyara <fmiyara at fceia.unr.edu.ar> To: Tor-Einar Jarnbjo <tor-einar at jarnbjo.name> Besides the fact that many multichannel signals largely exceed 8 channels, there are examples of signal packaging with many channels which are actually

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

--- 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

The link is http://www.adobe.com/devnet/rtmp/. TC Message stands for TinCan message. It is 11 bytes long, first byte is message type, three bytes of payload length four bytes of timestamp and three bytes of stream ID. The first byte of the payload for audio message is the format byte and the rest of the byte is the payload. Jozsef ----- Original Message ---- From: Jeff Ramin <jeff.ramin

Hello everybody, Fist a little about our setup. We have an Debian (7) Wheezy, now upgraded to Debian (testing) Jessie with Samba4 as PDC, Kerberos and LDAP - all provided through Samba4, and bind9 and isc-dhcp server for DDNS and DHCP, our environment is a mix of Linux (Debian Jessie), Mac (Maverick and Yosemite) and Windows 7 and 8.1 clients. The Windows clients use Samba and are all part of

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: >>

On 2/2/08, ogg-dev-request@xiph.org <ogg-dev-request@xiph.org> wrote: > > Send ogg-dev mailing list submissions to > ogg-dev@xiph.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.xiph.org/mailman/listinfo/ogg-dev > or, via email, send a message with subject or body 'help' to > ogg-dev-request@xiph.org >

Jean-Marc, you might like to simply copy the skeleton.c from liboggz, as described below. cheers, Conrad. ---------- Forwarded message ---------- From: Conrad Parker <conrad@metadecks.org> Date: 2 Feb 2008 16:41 Subject: Re: [ogg-dev] [PATCH] skeleton.c To: Ivo Emanuel Gon?alves <justivo@gmail.com> Cc: "ogg.k.ogg.k@googlemail.com" <ogg.k.ogg.k@googlemail.com>,

On 12/04/2008, Ivo Emanuel Gon?alves <justivo at gmail.com> wrote: > On 4/11/08, Ivo Emanuel Gon?alves <justivo at gmail.com> wrote: > > > uuencode ^_^ > > > > Hmm... > > Yes! A Vorbis Comment tag called ART or ALBUMART with a Base64 string > would do the trick and it would not choke existing players. No, vorbiscomments are meant to be