similar to: low sampling rate for Wideband ?

On Tue, Feb 18, 2003 at 06:09:43PM -0500, Jean-Marc Valin wrote: > Le mar 18/02/2003 ? 17:38, John Hayes a ?crit : > > If I encode something in ultra-wideband, can I decode it in wideband by > > chopping off bytes in every frame? > > All you have to do is use the --force-wb switch with speexdec. It will > decode as if the file were wideband, ignoring the ultra-wideband

If I encode something in ultra-wideband, can I decode it in wideband by chopping off bytes in every frame? John --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to 'speex-dev-request@xiph.org' containing only the word 'unsubscribe' in the body. No subject is needed.

It looks like I'll need to go further into the guts of speex to do this. I do, however, see some lines in nb_celp.c/nb_decode() that look interesting. I guess I'll play with it. I doubt that it will be terribly clean, though. Jean-Marc: Take a look at line 1195 in nb_celp.c (CVS). It reads "speex_warning ("More than to wideband layers found: corrupted

Hi, I found this list of Speex bitrates in the mail archive. http://www.xiph.org/archives/speex-dev/200306/0004.html Can somebody confirm that this list is correct? I am wondering about the following: - On the Speex website it says: "Speex is based on CELP and is designed to compress voice at bitrates ranging from 2 to 44 kbps." while the bitrates listed here are e.g. 84400 for

Hi all again, Thanks for the response to my previous question, Jean-Marc. It was of great help. Now another problem I encountered: How do I detect that comfort noise is being sent out (I have VAD, DTX on)? I could not find an API function for it, so tried frame rate. For nb, it gives 2950-3950 for silence, however it seems to also give low number sometimes even for speech periods. I

Hello, With high microphone gain, I seem to have problem making the silence detection work. The speech detection works well for the rare dish sample, which has very low noise amplitudes in silence regions. However, if the microphone gain is set to really high, noise samples are taken as speech, as indicated by the non-zero return value from speex_encode() calls. I had VAD turned on.

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

On Tue, Feb 18, 2003 at 09:06:16PM -0500, Jean-Marc Valin wrote: > BTW, when you have something working and stable, I could include it in > the main Speex distribution. Hmmm, define working and stable :) <braindump topic="speexcat"> It began as a merge between speexdec and speexenc from 1.0beta3, with the encoding/decoding removed, and simply piped in and out from ogg

Thanks! Btw, have you tried using SBR-technology or similar with speech codecs? That might be a good idea I thought.. But I don't know if it produces as good quality with speech codecs as it does for music codecs. Do you know if there is any open source variant of SBR? /Pontus -----Ursprungligt meddelande----- Från: owner-speex-dev@xiph.org [mailto:owner-speex-dev@xiph.org]För Jean-Marc

Hi, I was wondering if anyone has experimented with Speex's wideband (16kHz) mode at lower quality settings. In particular I have been using quality 3, and with wideband input files the resultant frequency spectrum is limited to about an upper end around 3.5kHz (almost telephony quality bandwidth). Has anyone tried increasing the spectral bandwidth at the expense of lowering the

It seems to work ok with the same audible quality as a standard sampling rate. Is there any way to test this? Will superimposing an inverse wave over the origional produce a meaningfull result? Thanks for your time, Ryan de Leeuw <p><p>>Sorry for the delay. I've been doing a couple tests >and what I'd suggest >is encoding using the narrowband (8 kHz normally)

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:

Le mar 18/11/2003 à 17:39, Allen Drennan a écrit : > Speaking of video conferencing in conjunction with Speex, we are > currently beta testing a solution we developed that offers multi-point, > multi-party video and audio using the Speex engine for voice. > > http://www.wiredred.com/downloads/ecsetup.exe > > The fair and good audio settings are Speex narrowband, high quality

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

> What is the maximum frame-length that libspeex will produce for narrow, > wide and ultrawide encoding? In normal operation (no in-band side information, like requests, ack, stereo, ...), the max size for a frame is 62 bytes in narrowband, 106 bytes for wideband and 110 bytes for ultra-wideband. Jean-Marc -- Jean-Marc Valin, M.Sc.A. LABORIUS (http://www.gel.usherb.ca/laborius)

FYI, the Asterisk software PBX <http://www.asterisk.org/> has now incorporated my recent patches to support dynamic RTP payload types. As a consequence, its SIP implementation now supports Speex, so if you have a Speex-compatible SIP client, you can use it to make calls using Asterisk. Some caveats: - Only narrowband (8 kHz) Speex is currently supported; not wideband. (Unfortunately,

Hi Jean-Marc, I thought at quality 3 (wideband) - wb_submode1 that the 4-8k band was not using a codebook table. From the code I can see that some sort of "lsp" encoding is performed. What exactly does this encode? (I assume lsp means line-spectral pairs) The reason I am asking is I'm comparing the "effective" spectral bandwidth of Speex against the

Le jeu 15/01/2004 à 15:30, Daniel Vogel a écrit : > Unrelated, but please use SSE/MMX/... intrinsics on Windows instead of using > inline assembly so you also get the speed benefit on Win64. OK, so here's a first start. I've translated to intrinsics the asm I sent 1-2 days ago. The result is about 5% slower than the pure asm approach, so it's not too bad (SSE asm is 2x faster

Dear Speex developers, I am currently experimenting with Speex on Symbian smartphones. I have compiled the Speex library, and I am now using it in the following way: 1. Record 320-byte buffers of data in PCM16 format, 8000 Hz sampling rate. 2. Feed the resulting buffer to an instance of a narrowband Speex encoder. 3. Send the encoded data over RTP. 4. Upon receiving on the other side, feed the

Hi, The ITU has been churning out a series of codecs which stretch older codecs to support wideband or super wideband and stereo. Does anyone know of these things being used in the real world? They push the idea of these things offering high compatibility, as the narrowband bitstream is embedded in the wideband stream. However, I'm skeptical this offers any real world advantage.