similar to: Icecast2 - Decode/Encode an Mount to another

On Aug 07 08:30:31, hans at stare.cz wrote: > On Aug 07 00:41:52, petrparizek2000 at yahoo.com wrote: > > ????#1. To test encoding at low bitrates, I encoded a sine sweep at 12 kbps > > with Opusenc and then decoded the resulting file with Opusdec. > 1) Opusenc --bitrate 12 --downmix-mono Sweep50.wav Sweep50.opus Why are you using a stereo file containing the same sweep in both

I had expected that the default bitrate for the encoder would be the same as setting it to OPUS_AUTO, but I'm getting difference results: >opusenc --comp 4 sample.wav sample.opus Encoding using libopus 1.3-rc2 (audio) ----------------------------------------------------- Input: 8 kHz, 1 channel Output: 1 channel (1 uncoupled) 20ms packets, 25 kbit/s VBR Preskip: 312

Dear Vorbis Devteam, My name is Michael Feilen and I've been working on Digital Radio Mondiale (DRM) transmitters and receivers for quite a while now. DRM uses HE-AACv2 by Dolby to encode the audio content (see http://www.drm.org/uploads/media/es_201980v030101p.pdf - pages 23 ff). As I think Vorbis is an excellent alternative, I'd like to implement and define an interface for Vorbis

Good morning. I've ran a test against MP3 format. Code: (first convert tested audio file to 16 bit 48khz with sox.exe if needed) lame.exe -b 320 48khzfilein.wav -o fileout.mp3 lame --decode fileout.mp3 -o fileout.mp3.wav opusenc.exe --bitrate 320 48khzfilein.wav fileout.opus opusdec.exe fileout.opus fileout.opus.wav wavdiff.exe 48khzfilein.wav fileout.mp3.wav -diff fileout.mp3.delta.wav

Jean-Mark sarkasm. Jean-Markasm. (Bonus points for providing an actual noisy WAV! ^_^) On 30/10/2017 20:28, Jean-Marc Valin wrote: Hi, Before I comment on the graphics you posted to visualize the difference between two audio signals, I'd like to ask for your help in evaluating my JPEG encoder. I've encoded an image with JPEG and then computed the difference with the original. I then

On Aug 07 00:41:52, petrparizek2000 at yahoo.com wrote: > ????#1. To test encoding at low bitrates, I encoded a sine sweep at 12 kbps > with Opusenc and then decoded the resulting file with Opusdec. What sine sweep exactly? How did you obtain it, and how exactly did you encode and decode it? Jan > The strange > thing was that even though the output wave file was at 48 kHz, it

If I understand correctly, Ogg Vorbis is a lossless format. But then why does it support bitrates? Ogg Vorbis does not delete any information. When I encode a 128 kbit MP3 file to 32 kbit OggVorbis, the resulting file has a bitrate of about 110 kbit. And when I encode it to 320 kbit, the file will be bigger, but the quality won't improve because of the quality of the original. So what purpose

Hi! Playing with Opus 1.3 I converted a tone sweep with a sample rate of 96kHz (just for fun). Before I had converted that from WAV to FLAC, and to Vorbis without problems. With Opus I noticed that the file size for 48kHz and 48 kbps compared to 96kHz Vorbis at 31kbps is about double the size and it sounds even worse (than Vorbis) (there is a lot of noise in the lower frequencies when a low

Hi, Just wanted to clarify, can icecast2 not read OR broadcast any mp3, or can it just not broadcast mp3 (but still play from a playlist)? I know the quality would be dropped if it had to reencode them though. And what's up with that POS Winamp3, it doesn't even play gg streams?!? I use xmms, but we have winamp3 on a demo computer, it just sits and keeps prebuffering. I need something

Hello, thank you for responding. I'm sure I'm confusing something, but I did not explicitly set either kHz or kbps, I just set the mode to narrowband, and traced through libspeex's code to see that sample_rate was 8000 something, I'm guessing Hz. My assumption was: if you set PA to nb-mode, then you should encode 160 bytes to 21. Is that mistaken? ys On 6/13/07, Jean-Marc Valin

On 4 May 2003 at 10:49, Geoff Shang wrote: > I have done this in the past, though it was a few months back. First > thing to check - you need to change all occurances of > "application/x-ogg" to "application/ogg". These are in liboddcast.cpp > and transcurl.cpp. Change these and recompile. This is almost > certainly your problem, as streamTranscoder is

Looks good to me. Jean-Marc Alfred E. Heggestad a ?crit : > Hi > > Please find an updated version of the Speex I-D attached. The only > change is addition of the copyright conditions in Appendix A, > as requested by Ivo. > > Many thanks for your input. > > I will give you a few more days before submitting to AVT working group > > > /alfred > > Ivo

Geoff Shang said: > Hi: > > I don't think you'll be able to do this with EZStream, I don't think it has > the capability to send more than one stream instance, but I could be > wrong. > > Also, the reencode isn't going to match your standard input, as it's > looking for files ending in .mp3 which of course standard input won't do. > > Geoff.

Hello all, This is slightly off-topic, but should be of interest to some people on this list. I just released version 0.3.2 of the CELT ultra low-delay audio codec (http://www.celt-codec.org/). CELT is designed to encode high quality speech and music with less than 10 ms delay and at rates starting from around 32 kbit/s. This version is "special" in that it is the basis for some

Hi! Excuse the delay, but I had to deal with a corrupted NTFS file system that ate many important files on an USB stick... The FLAC version of the original is almost 6MB and it can be downloaded slowly from this time-limited link: https://sbr5vjid0jgmce4q.myfritz.net:40262/nas/filelink.lua?id=0ba5a10529a6fe7b On the meaning of a logarithmic sweep: If you use foobar2000 and the

hi, i want to reencode our ezstream. but it dosent work. mdplay and lame are installed on the server. the config for ezstream is in the bottom of this mail. i dont understand the problem. have you any ieda ? have a lot of thanks my config: <ezstream> <url>http://localhost:8000/backup</url> <sourcepassword>*****</sourcepassword>

Hi there, Any other versions of Google Chrome and all version of other browsers are working. I have done my best to set up the intro to match the live stream. http://185.139.168.34:8000/yleisohjelma - intro is 256 kbits/s 48 kHz ISO-MPEG2 L3 - live is 256 kbits/s 48 kHz ISO-MPEG2 L3 http://185.139.168.34:8000/vara - live is 128 kBits/s 48 kHz, possible ISO-MPEG2 L3 - there is no intro Of

I have a setup where Icecast 2.2.0 is serving a single Vorbis stream. The source is either a local ices2 or a remote Winamp/Oddcast. I'm then running a streamTranscoder 1.2.8 instance on the server to transcode the Vorbis stream to MP3 on the fly and send it back to Icecast, to be served at a different mountpoint. My config files are very basic with most settings at their defaults. The

thank you, for your fast answer. my problem with ogg is, that the mediaplayer don?t understand it. a lot of users are using ms mediaplayer unfortunately. greets Geoff Shang schrieb: > Carsten Henkel wrote: > >> we would like to use IceCast 2.3.1 as a mp3 streamer source for >> reencoding several sources of ices 0.4. >> >> It is working perfectly already by using

Hi there - I am currently in the middle of a project where I need to configure an Icecast Server and I am using Ices as the encoder. I want to configure Ices to take in a stream with a sample rate of 128 kHz and output it to the Icecast server. My question is, what are the minimum and maximum sample rates that Ices will continue to work with? Thank you, Heidi Young Software