similar to: Sub-band filtering

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

>There's plenty of areas for improvements that don't require incompatible changes like this one. can u please tell me what do I do to make it more exact waveform coder for music rather than speech. I understand that its meant for speech, but I was just using it for music... I am interested in getting the residual as small as possible using speex. Can you please tell me the areas to

Hi, I chose speex initially because i had some work in VQ on speex i.e. modifying split VQ to GMM based parametric VQ and I thought If I train the GMM based VQ codebooks with audio signal and then do audio coding with speex, I probably get a better(smaller) residual signal even with speex. But I couldnt get that. I was trying to get a lossless bitstream by MUXing the speex-bitstream and the

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 Jean Marc and List, So I have started finally fiddling around with Ultra-wideband mode. It appears to be very similar in operation to Wide mode, except that when peering into the packet structure it looks like (and these are kind of questions as much as statements here): 1. update rate 0 is not used in UWB- only 1-4? 2. The total bits used for each UWB update rate seem to be as follows:

hi, basically I am anyway using GMM based parametric VQ for LSF quantization so that is not a problem. Can you plz. tell me how to retrain the pitch codebook and innovation codebook. what are the tools to train the pitch and innovation codebook? Regarding the pitch prediction for high-band, I understand your reason of not using it for high-band. There is one paper, "A 14 kb/s wideband

Le mercredi 10 mai 2006 ? 10:17 +0100, devilal sharma a ?crit : > 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. Sorry, you got your calculations wrong, it's 20 ms regardless of the modes. > What

Hi, JMV> ... none of these codecs can ever have open-source JMV> implementations due to patent issues. We use now these codecs for x86 in "C" source code running under Linux. They works very well. My problem is that we are not so familiar with Blackfin DSP. Concerning of patent issues, you are not right. The patent is for their using, not for the source code. Every one

Hi, Thank you. I will try it. Do you know some G.72x, GSM, and iLBC optimized for Blackfin ? I mean open source. -- Best regards, Miroslav mailto:miro@space-comm.com Wednesday, March 23, 2005, 9:05:11 PM, you wrote: JMV> Hi, JMV> As far as I understand, the last patch (for TI C5x) I merged in SVN also JMV> allows Blackfin to work, but I haven't

Dear Jean, The source code for G.729 can be download from ITU for free. Also, some developer can do yourself as open source G.729 codec without any help. In this case each who use this codec which source code is free and open source must pay, but not to the developer. Best Regards, Miroslav Nachev JMV> Le jeudi 24 mars 2005 ? 10:08 +0000, John Villar a ?crit : >>

Sir Can u kindly tell me what kind of coding does speex use in Ultra Wide Band ? And from where can I read the details of it ? Abhishek IIT - Madras, India. <p> ___________________________________________________________ Yahoo! Messenger - Communicate instantly..."Ping" your friends today! Download Messenger Now

Hi, I have one doubt again, that is Vorbis use DCT/MDCT based algorithm and also use psychoacoustic model so this is lossy codec. And I dont think it ca regenerate a better matching waveform than speex. Then there comes FLAC which is the perfect answer to my question, I suppose. But my concern is this that FLAC use simple prediction algorithm and doesnt use any CELP based algo which could have

Hello, Recently I realized that entering first 8 characters from my root password is enough to log in as root although I've enabled MD5 passwords in installation program. Also I've added two other users during installation. After looking through /etc/shadow it seems that root's password isn't md5 although passwords of other two users are md5. Is it correct behaviour? Eugene --

Hi all, One question that I hope someone on the list just knows the answer to without having to delve too deeply into the code: How does UWB mode divvy up the bandwidth and pack it in the bitstream? I know from the documentation that WB mode codes the first 0-4K kHz band as a Narrowband packet, and then adds on the 4-8 kHz band coded separately (so that a NB decoder can decode a WB bitstream

Hi list! There seem to be some oddities in using ABR. The first thing I notice is that because "abr_count" increases without bound, after a while the weight of the bitrate history will completely swamp any change in the current bitrate and the ABR adjustment will essentially stop happening. This seems to be true in any mode (not just UWB) and perhaps a solution is to cap

I am trying to join my Linux workstation to my ADS domain. Unfortunately, I'm not having much success. net ads join hangs forever (or at least for more than 12 hours) when run. The computer account is created in the domain, but the process never completes. tdbdump secrets.tdb shows no results, and wbinfo shows users and groups from the trusted domains but not from the domain I am trying to

Hi, I am currently encoding 32 Khz/Qual-10/UWB encode mode, with MANUAL_ALLOC enabled ( similar to the c5x/c6x configuration). In file sb_celp.c, I noticed the scratch memory grow during recursive (UWB -> WB) calls to 'sb_encode'. 1. 'stack' was not tracked (with 'tmp_stack' as done at other palaces) after/before - auto-correlation/Levinson-Durbin scope (~line

> Ok, here is cleaned up and fixed version. > > * Function is named speex_get_num_frames() now and return > number of frames, as you suggested. > * WB layers sizes are taken from wb_skip_table[], while NB frame > sizes are calculated with speex_mode_query(). Looking better. Just make sure to remove the stuff that isn't C99-compatible (e.g. // comments). > I've

Hi Jean-Marc, > Can you confirm I'm understanding everything correctly? You encode > with > the same encoder and then decode with either A) blackfin assembly and > fixed-point or B) fixed-point only on Blackfin. Then A) sounds bad and > B) sounds good. If you do the same in narrowband, it sounds OK. Is > that > correct? If that's the case, it's *probably* some

Speex tells me that the decoder is always 5 ms, but it says that the encoder is 5 ms for NB, 8.9375 ms for WB, and 10.90625 ms for UWB. Is there an extra frame of delay in the encoder that isn't otherwise accounted for? John Ridges Jean-Marc Valin wrote: > Quoting John Ridges <jridges at masque.com>: > >> I also need to know the precise delays from Speex but I used