similar to: Speex joins Xiph.org, releases beta 1

What sould be the capture and playback buffer size in 8,16,32 khz for the Alsa system? Can this also causing latency? In the server side I have: -A thread that reads the input from mic (capture) ands copies to main buffer. -The main loop encodes and sends it to the client ( it's read the data from the main buffer) Client: -A thread for

Ohh... Nice! This is new in 1.0.1, isn't it? It doesn't seem to be included in the reference manual yet, though. Thanks! Tom <p>Jean-Marc Valin (jean-marc.valin@hermes.usherb.ca) wrote: > > OK, this is how it works: > > The encoder calls speex_encode any number of times and then calls > speex_bits_insert_terminator before sending the bits. > > The

Hi, I'd like to announce Speex beta 3: This is the third beta for Speex, implementing what should be the last new features before 1.0. These new features are a new "ultra-wideband" mode for encoding at 32 kHz (up to 48 kHz) and an intensity stereo mode. Both of these are implemented to preserve both backward and forward compatibility with other releases. This means that it is now

Le mer 26/02/2003 à 15:43, Rick Kane a écrit : > I was also wondering if there is a standard set of input sequences people > are using to test Speex. I haven't stumbled upon it/them yet. I've got a few samples at: http://www.speex.org/audio/samples/ Jean-Marc > > -----Original Message----- > > From: owner-speex-dev@xiph.org [mailto:owner-speex-dev@xiph.org]On

Hi, I uploaded a pre-release of beta3 for which I'd like to get feedback. There are some new features like a new "ultra-wideband" mode for 32 kHz operation (up to 48 kHz) and a (intensity) stereo mode. You can get the source at: http://www.speex.org/download/Speex-1.0beta3cvs.tar.gz So please test that code and report any bug or inconsistency you may find. Jean-Marc --

Hi everyone, I've just released version 1.1.4. This includes some code cleanup and improvements to the fixed-point port and SSE optimizations. All the SSE code has been converted to intrinsics and some new functions have been implemented with SSE. Overall, the speed has been increased by up to ~30% with SSE. Jean-Marc -- Jean-Marc Valin, M.Sc.A., ing. jr. LABORIUS

Hello Jean-Marc >If you want to do it, I can show you >what functions (there are 2-3) to port. Otherwise I might do it >eventually, but it's not a top priority (there's already an SSE version >though). I would indeed like to know which functions can be used to improve K6-2 performance through 3DNow. Cheers Bjoern D. Rasmussen <p><p><p>>From: Jean-Marc

This works really well for white noise reduction. However what I've noticed was the amplitudes of normal speech samples also get reduced. Is this something by design, or is there a way to automatically recover the original speech sample volumes ? <p>Thanks. <p>Tongbiao <p>-----Original Message----- From: Jean-Marc Valin [mailto:jean-marc.valin@hermes.usherb.ca] Sent:

> Am I right with the assumption, that currently you have to enable > processor specific optimizations with compile/configure options? > > How difficult would it be to add support for runtime CPU detection? > Is this a feature you might consider adding? Pretty complicated because of some annoying decisions taken by the gcc team. The problem is that gcc won't let you use

At 07:55 PM 3/11/03, Jean-Marc Valin wrote: > > - Only narrowband (8 kHz) Speex is currently supported; not > > wideband. (Unfortunately, the assumption that audio sample rate == 8 kHz > > is riddled throughout the Asterisk code.) > >Perhaps it's still possible to send wideband, while telling Asterisk >it's narrowband (the bit-stream is such that you can decode

Hi, just to let you know that unstable version 1.1.1 is out. It includes the latest fixed-point changes which can be enabled with --enable-fixed-point (as configure option) or -DENABLE_FIXED_POINT (for win32). The port is not complete, but most of the floating-point operations have been converted. Please give it a try and report any difference with previous versions (both for float and

(replying to the new list: speex-dev@xiph.org) > I'm having trouble building the 'speexdec' executable in the win32 port of > Speex. I'm using MSVC++ 6.0 on both a Win2000 and a WinXP box (the error > occurs on both machines). The sources are fresh from CVS as of today. Can you check whether or not you have the same problem in beta 2? > I had no trouble building the

Hi, Speex is progressing and I've started thinking about the next 1.2 release (don't hold your breath). Though the bit-stream won't change, the API likely will. The API for 1.1.x already differs from 1.0.x because the speex_encode and speex_decode now use shorts instead of floats. Now, since I'm changing the API anyway, I thought I might as well fix things that might be annoying

I tried feeding in the 3 encoded frame in ONE BLOCK, and calling speex_decode() 3 times in a roll. Only the 1st frames came out perfectly. For the other 2, I got "corrupt" frame warning. I was supposed to get 38 bytes consumed each frame (narrow-band, VBR off). I tried speex_bits_remaining() to peek on the # of bits consumed, and got variable (clearly wrong)#s returned. But if I

> I'm about finished developing a QuickTime component that supports Speex > (on > MacOS X and Windows).. As it is now the user can set complexity > (SPEEX_SET_COMPLEXITY) and quality (SPEEX_SET_QUALITY / > SPEEX_SET_VBR_QUALITY) and to wether to use VBR or not. Will these > options > make it possible to produce all combinations of bitrates/qualities? Or > should I

Jean-Marc, I was wondering if you could add a check to ensure that memory is actually allocated during the nb_encoder_init and sb_encoder_init functions. We have been looking at using Speex on a DSP and noticed that if we didn't allocate a large enough heap space memory segment that the DSP would crash. I would recommend something like: if (!st->stack) fprintf(stderr,"ERROR

Le ven 10/01/2003 à 14:39, John Hayes a écrit : > I've been playing with speex for use in a VoIP application between PC's. One > thing I've found (correlating to the documentation) it that speex runs much > faster and produced much better output when it's fed a 32kHz signal instead > of a 44.1kHz sample rate. This is whether I tell it a 44.1kHz sample rate > and feed

> - Only narrowband (8 kHz) Speex is currently supported; not > wideband. (Unfortunately, the assumption that audio sample rate == 8 kHz > is riddled throughout the Asterisk code.) Perhaps it's still possible to send wideband, while telling Asterisk it's narrowband (the bit-stream is such that you can decode a wideband frame even if you think it's narrowband). > - Some

Hi, Well, there are already Speex plugins for widely used GUI players. I don't see what an "official" standalone player would add. Of course you can still write it, but I don't see a point in making one player "the official player". As for cross-platform, it would basically mean it would have to be written in Java, but even that is inconvenient for platforms that

Just to let you know I released 1.1: This is an unstable release. It brings many new features, some of which are still experimental. The new features are: * a denoiser that removes most of the background noise and can be used before encoding (available as --denoise in speexenc) * adaptive gain control (AGC), which adjusts the volume to a constant level (available as