similar to: narrowband embedded in wideband

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 Alexander, > Here is first attempt. I'm sending it here for review - am I understand > you correctly? If general approach is correct, I'll change code to use > modes from modes.c. I haven't tested, but it looks sane. Just a few comments: - printf("\tERROR not enough bits left after wb\n"); this is actually now an error. It just means you got to the end of the

I know speex is not supposed to do a great job compressing music, but I've noticed that the new VBR code chokes completely when you try to compress horns. I've placed a particularly offensive example up at http://www.utdallas.edu/~matthias/ . Take a look at a-16m*{ogg,spx}. a.ogg is the first minute of an ogg created from the source media (in 44khz stereo). The rest have been mixed down

Take a look at this: peexenc --vbr -V --quality 4 audiodump.wav a.spx Encoding 32000 Hz audio using ultra-wideband (sub-band CELP) mode (stereo) Bitrate is use: 22400 bps (average -1268444817 bps) I'll take a look at what the problem is later today or tomorrow, but if anyone beats me to the bug, let me know before I spend too much cscope time playing with it. Regards, Matthias --

I downloaded speex-1.2beta3-win32 binaries, and used the speexenc.exe and speexdec to process the file male.wav download from http://speex.org/samples I used both the narrowband and ultra-wideband mode: >speexenc -n --quality 3 male.wav n3.spx Encoding 8000 Hz audio using narrowband mode (mono) >speexdec n3.spx n3.wav Decoding 8000 Hz audio using narrowband mode (mono) Encoded with Speex

Hello all, I'm testing the Speex codec for my diploma thesis on a BF-533 Blackfin under uCLinux (2005R3 RC3 release). I successfully compiled the Speex (1.1.11-svn) and I can encode/decode wav-files on my STAMP-board using the speexenc/speexdec sample apps. But I encountered that the decoded file sounds strange/noisy, when compiling with "--enable-blackfin-asm" +

Hello, playing a little around with speexenc/speexdec, I discovered that if I pass the "-n" or "-w" command line option to speexenc (version 1.1.10), I get a segmentation fault. Looking closely at the code, I found that the line: fprintf (stderr, "Warning: Trying to encode in %s at %d Hz. I'll do it but I suggest you try ultra-wideband instead\n",

Hello, To start with, thanks a lot for making such a great voice codec available! Having recently upgrading to speex rc1, It occurred to us that there seems to have been a regression in the quality of encoding since version beta1. We are compressing some 22khz wave files in wb mode with maximum quality / complexity in VBR, and the result was really great with speex beta1. With rc1 (or beta3),

But using the ACM codec I just need to put the speex content into the wave file data chunk and then windows will play it? Miguel Gomes -----Mensagem original----- De: owner-speex-dev@xiph.org em nome de Matthias Granberry Enviada: qui 04-03-2004 11:53 Para: speex-dev@xiph.org Cc: Assunto: Re: [speex-dev] Speex, what container?

Hi everyone, I just released Speex version 1.0.1 with a major bugfix to the wideband mode. It's highly recommended for people using wideband Speex to upgrade. Also, since the old encoder and decoder had the bug, you may notice that files encoded with 1.0 have a lower quality when decoded with 1.0.1. For that reason, re-encoding might be a good idea. Note that narrowband is not affected by

I'm trying to get asterisk working w/ packet8 (incoming and outgoing) and a dta310 so I can have more control over voicemail. I've looked at the data stream coming from the dta310 and from packet8, but I haven't managed to get the phone to actually place invites through asterisk. On the asterisk end with chan_oss.so, I can make it dial and I hear ringing and the first second of

Hi, Just to let you know that Speex 1.0 RC3 is out. It should be the last RC until 1.0 is released (tentative release date is March 16th). Please give RC3 a try and report bugs so they can be fixed before 1.0. As usual, download from http://www.speex.org/download.html Jean-Marc -- Jean-Marc Valin, M.Sc.A. LABORIUS (http://www.gel.usherb.ca/laborius) Université de Sherbrooke, Québec,

Use the -V option when encoding and you should get bitrates. "Daniel Vogel" <vogel@epicgames.com> writes: > Could someone post a mapping of quality to bitrate or provide me with a link > where to find this information? > > Thanks, > > -- Daniel, Epic Games Inc. > > --- >8 ---- > List archives: http://www.xiph.org/archives/ > Ogg project homepage:

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

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

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 Jean-Marc, yes, problem exists in narrowband-mode, too. I already twiddled with max_gain, but did not have real success. I changed line 337 of ltp.c (function pitch_gain_search_3tap_vq) if (sum>best_sum && gain_sum<=max_gain) { to if (sum>best_sum && gain_sum<max_gain) { -- that stabilizes speex for 2000 Hz and 2200 Hz input on quality setting 7 (23800

Hello, In the internet draft describing the SDP parameters for speex I find this: mode: Speex encoding mode. Can be {1,2,3,4,5,6,any} defaults to 3 in narrowband, 6 in wide and ultra-wide. In the documentation I can only find the following request: SPEEX_SET_MODE There is no description though what this request does. I think the mode somehow maps to the bitrate, but I

Hi, Can someone please tell me how should I go about changing the frame size which is hardcoded to 160 for NB and WB and 320 for UWB. For NB speech(8KHz) the framesize of 160 is 20ms frame but for WB and UWB its 10ms. What are the parameters being affected by simply changing the framesize and sub-frame size in "modes.c" How to change the buffer size and how its affected. can we have a

Hi, I got my hands on the LCA 2003 CD today and I tried to listen to it on my powerbook running Linux. Instead of getting speech, I got an earfull of noise! Attempting the same experiment on an i386 with the same CD produced understandable speech. It seems that the output routine in speexdec converts the output data to little endian short format, which is incorrect on the ppc architecture. The