similar to: beta 2

Hi, I have a track which when encoded with Mode_C or even Mode_E does not encode well at all ... It's a jazz flute track with a lot of plate echo. The flute (centred in the mix) almost disappears ... I am away for a few days, but if someone would like to look at this I would be grateful ... A small clip can be grabbed from my ftp server ... ftp.york.ac.uk in:

Hello Thomas and Amit, Thanks for your notice and the detailed decode performance report. I describe the details of my encode/decode test on STM32F407ZG. A. opus version: latest 1.2.1 (TI: opus 1.1.2) B. KEIL 5.23 (TI: ARM compiler tool chain 5.2.7) C. setup the encoder as the below (fs is the sampling frequency) enc = opus_encoder_create(fs, chans, OPUS_APPLICATION_AUDIO, &opus_err);

My last sentence still aplies to the bitrate. We usually use 80kbps, but sometimes we have used 64kbps (e.g. when sourcing from a cellular network). With music you can get the difference, but with voice there's no need to use more bitrate unless you want more quality in the audio itself. -- Xabier Oneca_,,_ El 27/02/2013 15:19, "Jos? Luis Artuch" <artuch at speedy.com.ar>

El 27/02/2013 16:12, "Jos? Luis Artuch" <artuch at speedy.com.ar> escribi?: > > We usually use 80kbps, but sometimes we have used 64kbps (e.g. when > > sourcing from a cellular network). > > > > With music you can get the difference, but with voice there's no need > > to use more bitrate unless you want more quality in the audio itself. > >

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Hi, Am I correct in understanding that the codebooks are *not* adaptive during compression? I see that Packet #3 is written to the stream in the beginning of the encode process with no modification. If the codebooks are not adaptive, then why are codebooks included in the stream at all? Why not pass the mode type (A or B or C...) instead of all the mode info and let the decoder load it's

Hi: I've been experimenting with streaming vorbis across my network from the windows box to an icecast2 server on my linux machine (the server I usually use is having some network issues at present). I've tested both ostream 0.7.1 and oddcast beta26. It seems that neither can encode on the fly on my P2-266 at 44100 fast enough to be able to stream it. Ostream looked the more promising

Hello, is somebody working on a good low-sampling rate / low-bitrate mode? I encoded today a mono/16KHz/16bit WAV (a TV-talkshow), using OggDrop. The quality of the '64kbps' mode was unacceptable, so I had to use '80kbps' mode. The bitrate averages around 42 kbps, which I found a bit high for this quality. In your opinion, what bitrate should I expect as Vorbis matures? 24 kbps?

> vorbis_encode_init( &vorbisInfo, 2, 44100, 96, 96, 96); > > which by all reasons should generate a 96 kb/s stream, as all > max_bitrate, nominal_bitrate and min_bitrate are set to 96. Strangely > enough, the generated stream's bitrate (according to XMMS and WinAmp) > varies between 49 and 59. Currently min and max are ignored by the library, since bounded bitrates

Hi, I'm experimenting with IceCast2, using DarkIce to generate the stream. I have found some peculiarities with the vorbis bitrates. In DarkIce, I call vorbis_encode_init() with about the following values: vorbis_encode_init( &vorbisInfo, 2, 44100, 96, 96, 96); which by all reasons should generate a 96 kb/s stream, as all max_bitrate, nominal_bitrate and min_bitrate are set to 96.

Firstly, please excuse the crosspost to both the icecast and vorbis lists, and the slightly commercial nature of this post. Virgin Radio Classic Rock, which is a sister station to Virgin Radio, is now also available in streaming Ogg Vorbis. We also have 'now playing' information working on both services. Links and more information at

Firstly, please excuse the crosspost to both the icecast and vorbis lists, and the slightly commercial nature of this post. Virgin Radio Classic Rock, which is a sister station to Virgin Radio, is now also available in streaming Ogg Vorbis. We also have 'now playing' information working on both services. Links and more information at

asked once before and didn't get a reply, but it was probably lost in my back-and-forth with Jack.. where is the encrypted password support for icecast 1.3.10? The documentation references a "mkpasswd.c" source file that doesn't exist anywhere in my icecast directory tree.. -------signature file------- PGP Key Fingerprint: 446B 7718 B219 9F1E 43DD 8E4A 6BE9 D739 CCC5 7FD7

Icecast allows using a fallback mount file which the listener will hear if the main mount stops. It is specified in icecast.xml as this example: <fallback-mount>/example2.ogg</fallback-mount> It is necessary that the file have the same bitrate and same format (single or two-channels) as the main mount. We were at a music festival last weekend and used butt (Broadcast Using This

The following block of code produces an exception. The testnew.ogg file is 10 seconds, containing pure silence encoded using mode E: http://www.cs.mun.ca/~chris3/testnew.ogg Other files and modes can cause the exception, but many files work fine. #include <stdio.h> #include "vorbis/vorbisfile.h" int main() { OggVorbis_File ov, ov2; FILE *f, *f2; f

Greetings, With all the press concerning the 'almost 1.0' release of vorbis; I see a lot of messages floating around as 'now 1.0, and -soon- it will include joint-stereo, Wavelets, low-bitrate, peeling, etc. etc.' Before I start 'advocating' vorbis and tell everybody what it all can do; is there actually a date or some other planning when all those

Hi! I would really appreciate if someone could try to help me out with this... i've been struggling with it for quite a while now. :( Anyway, I was trying to use ices to broadcast mp3s with icecast2 (a recent cvs snapshot of all 3 components from brendan's site). I have it working fine for some of my mp3s (encoded with lame -b 32) but with the other mp3s which are actually the ones that

Hey. I just wondered,... which is the recommended bitrate and further settings of opusenc (like complexity and framesize) for general music (e.g. including classical music) to achieve more or less transparency? Talking about CD level audio (16bit; 44,1 kHz; Stereo) The only source I could find regarding that was: http://wiki.hydrogenaud.io/index.php?title=Opus#Music_encoding_quality But that

Hi there, I encoded the hard to encode piece ":wumpscut: - mother" with the new prebeta4 encoder (windows). As I crashed my harddrive I've to reinstall linsuxx again and double check the results, maybe there's a difference between the platforms (as far as I remember there was such a thing before). This song is pretty good for fast listening-tests as there are plenty of

Hi there, I encoded the hard to encode piece ":wumpscut: - mother" with the new prebeta4 encoder (windows). As I crashed my harddrive I've to reinstall linsuxx again and double check the results, maybe there's a difference between the platforms (as far as I remember there was such a thing before). This song is pretty good for fast listening-tests as there are plenty of