similar to: Speex

> 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. Quite the opposite. Speex uses very few bits, so every bit you corrupt is likely to be heard. You would need to use "fractions of bits" at a time. Even then, it may be hard to hide data efficiently (so it's

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,

Hi, I have some questions regarding Speex. please reply as soon as possible. where are the two filter loops applied to the raw signal? where is the residue (innovation) signal calculated? where is the innovation mapped on a codebook entry (key)? which bits of a transmitted frame are the two filters, which are codebook keys? Is the codebook from the beginning on static (so already known by

Hello, Jean-Marc Valin schrieb: >>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. > > Quite the opposite. Speex uses very few bits, so every bit you corrupt > is likely to be heard. You would need to use "fractions of bits" at a > time.

Hi, ? Is there a fix for this issue??? ---> http://permalink.gmane.org/gmane.comp.audio.compression.speex.devel/2959 ? I am seeing the same thing when I compile speex in visualdsp ? These are the errors I get from using the assembly version of vq_nbest: ? ..\..\..\..\algorithms\voice\speex\src\vq.c [Error ea5004] "C:\Users\coder\AppData\Local\Temp\acc22e8547f000\acc22e8547f001.s":482

Hi, can someone help me understand and experiment with Speex in a better way than my knowledge. I want to change the LPC filter order of the Speex algorithm. I think I need to change the LPC order( or lpcSize) in modes.c file. but to make it work I need to modify the codebook also, can someone tell me how to go about changing the codebook entries to support the changed order of prediction.

I've been working on improving packet loss concealment in my VoIP client -- I hadn't actually realized speex performed some PLC until I was looking at nb_decode(). There are two things I'd like to do, and I think both will require some modification to nb_decode() and nb_decode_lost(). - I'd like to have my frames overlap by some number of samples. This is intended to reduce the

I've been working on improving packet loss concealment in my VoIP client -- I hadn't actually realized speex performed some PLC until I was looking at nb_decode(). There are two things I'd like to do, and I think both will require some modification to nb_decode() and nb_decode_lost(). - I'd like to have my frames overlap by some number of samples. This is intended to reduce the

dear sir/madam, i am runnin the speex codes for the TI-DSP in CCS. i have also given the same input file that has been mentioned in the readme-TIDSP. the code runs perfectly ( i think) but it takes a lot of time for the *.out file to "run". the output file, which is supposed to be "maleout5x" has a size much bigger than the input file "male.snd" (the input file is

Hi everyone, I have got some questions about Speex, I am sorry if my questions are too newbie: 1. For the LP analysis, did Speex use the AR (Autoregressive) model or the ARMA model? 2. Am I right to say that Speex use a multistage VQ (since I believe Speex employs two or more VQ consecutively - based on the manual it says that Speex uses dynamically selectable codebooks (linear

Hi, I am very glad to enter this mail list. i am a new guy,i am interested in speech coding.so is there anybody working on this ? please contact me,thank you.i am studing on fixed codebook search. and i have read almost all the articals(including patent on this) about codebook search. but i found that all the suboptimize methods have their own flaw. could anybody have some comments on

Dear Sir\Madam; Thanks for your source codes. You mentioned that your speex source code can be compiled for ti-c54 family; but you use linux command-line as following to compile speex for C54 family: % ./configure -prefix=<path> -enable-ti-c55x % make % make install Is it true?! Then what is the output file for loading on DSP chip (C54)?? As I remember the compiled file for C5x family

> I have one doubt again, that is Vorbis use DCT/MDCT based algorithm > and also use psychoacoustic model so this is lossy codec. Speex is also a lossy codec. > And I dont think it ca regenerate a better matching waveform than > speex. At bit-rates above 32 kbps, Vorbis tends to produce better results than Speex, even for speech. The only advantages of Speex over Vorbis at these

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 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

Thank you so much for your quick reply. It looks quite easy to change it. But, how about those codebook and excite table? I notice them have been fixed as the length of 160. Thanks again, Dennis -----Original Message----- From: Jean-Marc Valin [mailto:jean-marc.valin@usherbrooke.ca] Sent: Wednesday, February 13, 2008 4:03 PM To: Dennis Liu Cc: speex-dev@xiph.org Subject: Re: [Speex-dev] Can

u can play the packet in wireshark, wireshark has builtin player for playing audio packets in whtever format it is. in wireshark u have the option VoIP calls, select that u will c the player in it. On Thu, Sep 10, 2009 at 12:00 AM, <speex-dev-request at xiph.org> wrote: > Send Speex-dev mailing list submissions to > speex-dev at xiph.org > > To subscribe or unsubscribe

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 it 44.1kHz or tell it 32kHz and feed it 44.1kHz. What part of the

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

Sir, Sorry, I can not return your email immediately. I am out of office from May 18 to June 6, 2006. thanks for your patience, Jay Huang