similar to: Speex: Open-source, patent-free speech coding

Hello everyone! Please bear with me, I am a complete audio-newbie, and I apologise if I am asking a question that I should know. As most of you all know, DVD's have 5.1 audio, and I want to preserve that in my voribs files. So, to ask the question : Does it support Surround sound? Thank you very much!

You're asking the wrong question. The question is not "why does it would bad with Speex?", but "why does it sound good with LPC10 and MELP?". And the answer is that both are vocoders. Try dropping frames/subframes with anything else (Vorbis, MP3, G.729, u-law, ...) and it'll sound terrible as well. The only reason it sounds good with vocoders is because the

-----Original Message----- From: Jean-Marc Valin [mailto:jean-marc.valin@usherbrooke.ca] Sent: Tuesday, June 19, 2007 6:38 PM To: Michael Shatz Cc: speex-dev@xiph.org Subject: Re: [Speex-dev] Blackfin inline assembler and VisualDSP++ toolchain >> Yes, data footprint in the new version is quite manageable. Still I would >> wish better documentation for speex_alloc_scratch(). >

Hello Jean-Marc: I've been watching the speex development from its inception because I and several Tech Startup Connection members have a very important application for this voice encoding/decoding. Further, we are quite familiar with ACELP as implemented in G.729. As far as I know Speex is also ACELP ... yes/no? Question ... what do you see as the advantages of ACELP compared to MP3? I

I was wondering how vorbis fares as a speech codec? I mean can it give similar or better quality as GSM/Toast at the same kind of bitrates/filesize? I'm not worried about streaming but I'd love to be able to create really small voice files. Can anyone give me an idea of filesize for recording time for pure speech based use? love Freya __________________________________________________

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I have an AVI file (~900MB) that I want to convert to OGM format, with a ~ DivX 5 video and uncompressed AC3 (5.1ch) audio stream. Nandub doesn't want to compress the audio, so I was wondering if anyone could recommend a (Windows) tool to help. Thanks for your time! -Chris - -- "Everyone has to find their own reason." ~ -Cloud,

Jean-Marc, I am building a tool for producing the highest possible quality Internet interviews for "podcasting" applications. The goal is to produce a perfect recording of an interview or conference -- and giving the participants a glitch-free experience is secondary. My approach, therefore, is to build a Windows "wave" file asynchronously by using a streaming

-----Original Message----- From: Jean-Marc Valin [mailto:jean-marc.valin@usherbrooke.ca] Sent: Thursday, June 14, 2007 11:17 PM To: Michael Shatz Cc: speex-dev@xiph.org Subject: Re: [Speex-dev] Blackfin inline assembler and VisualDSP++ toolchain Michael Shatz a ?crit : >>> Actually, you're the first I know using the VisualDSP++ toolchain >>> :-) >> >> I guess

In increasing the computation time and bit rate with VBR: Has anyone considered implementing the standard audiological recognition technique of using the duration of zero energy (voice onset timing) to identify the presence of voiceless sounds? I would like a means of determining whether or not a given window will be full of voiced speech or not. Matt -- The swallow may fly south with the sun

Hello kind developers, I am planning on creating a steganographic extension for speex/CELP. Since speex/CELP is lossy, there should be quite many bits that one can use for hiding data. I'm familiar with the principles of CELP (two filter loops, one for pitch, the other for formants, let the raw speech frame run through those filter loops, take residue signal, map it on a codebook entry,

Hello Jean-Marc and all, I recently had a talk with somebody about CELP. He said, there is this federal standard 1016 (4.8kbps) with a reference implementation of the Department of Defense (only on Sun, unfortunately, if I got this right). This one is noticed in the manual already. He also said, since there is this implementation of the DoD, nobody would voluntarily re-implement CELP. If I read

Hi. I'm Bill Cox, and I volunteer a bit for the Vinux project, which is Linux for people with vision impairments. Most blind users use a closed-source speech synthesis tool called voxin, as it's very easy to understand at high speed. I would like to make TTS synthesizers based on large recorded vocabularies of actual speech, but to make it useful for the blind, I need to be able to

hi all, thanks Jean for the codec. I am new to speech coding. Am interested to implement speex on a TI dsp, say C54. just downloaded speex v 1.1.6. when I removed optimisation flags for win32 in libspeex.dsp, I get a slightly different output for decoding using speexdec.exe (debug version). no prob for speexenc.exe except for some differences in the ogg header. for anyone interested, i

Hi all, Yes, it's true. A new version of rehuff, the tool that losslessly compresses Vorbis files: one that is easy to compile, and that works with newer-than-two-years-ago streams, too! On 1.0 streams, you get about 3% size reduction, and the headers get _much_ smaller (which helps for fast-start network streams). Building it should be easy (you might have to add some -I and -L for

Hi all, Yes, it's true. A new version of rehuff, the tool that losslessly compresses Vorbis files: one that is easy to compile, and that works with newer-than-two-years-ago streams, too! On 1.0 streams, you get about 3% size reduction, and the headers get _much_ smaller (which helps for fast-start network streams). Building it should be easy (you might have to add some -I and -L for

Hi, I was just trying to use speex for sampling frequency >48KHz. In the original Speex-1.0.5 its restricted only upto 48KHz. I tired to modify it by changing the boundary conditions( the error conditions, i.e. if sampling freq >48KHz, it gives error) in /src/speexenc.c and then it atleast doesnt give the error, there is flow in decoding or encoding(i think). I suspect there are other

Hi everyone, I have been using vorbis for ages as audio codec for archiving movies and I have always been happy with it, but now I have been thinking about two things: I) Multi-channel Up until now I mostly encoded to stereo; now I am thinking of encoding more movies to 4.0 or 5.1. I heard that vorbis does not optimize for channels > 2, i.e. that 6 channels is just stereo*3. AAC is

Hi everyone, I have been using vorbis for ages as audio codec for archiving movies and I have always been happy with it, but now I have been thinking about two things: I) Multi-channel Up until now I mostly encoded to stereo; now I am thinking of encoding more movies to 4.0 or 5.1. I heard that vorbis does not optimize for channels > 2, i.e. that 6 channels is just stereo*3. AAC is

Hi All, First of all, I'm sorry if my question is offtopic on this list. In such case please ignore this post and/or contact me directly. I'm asking my questions there because I feel you had similar problem before starting developing Speex. My story: my friend developed 3gpp content creator and he would distribute it in binary form. But there is problem with AMR licensing (the terms

This idea is so simple that I'm sure it must have been thought of before, and discarded, since AFAIK it's not used anywhere. I did a quick web search but that didn't turn up much, so I figured I'd put it up for discussion here anyway. How about using large-scale repetition in audio compression? I'm thinking of redundancy in repeated pieces of a song, ie a chorus.