similar to: speex voice seems to be bit breaking over long distance.

> Ok, what you suggest sound logical to me. Currently, I > have done a small trick to prevent this problem. What > I did is that whenever windows request a voice packet > from me and if I do not have the voice packet, I > repeat the previous packet. Hence, all the breaking > portion is filled with previous packet. This trick > seems to work so far. I am not sure what is the

Hi, Have anyone using speex experience the following effect: When I say "hello", it will be become "hellololololololo....". This does not always happened. The chance of happen is quite random. This effect does not happen in my LAN. It only happen when ping time is about 500 milisecond or more. Is it a speex problem? Any suggestion how to resolve this issue? Thanks in

Hi Jean, > > > Well, just trace it, how many times are you > calling > > > speex_decode_int()? > > > > Maximum is 51 times per second. Will this cause > any > > CPU high utilization? > > That's normal... What CPU are you using? If it's a > fixed-point CPU, then > the reason is probably just the fact that the packet > loss

Dear all, I have implemented speex. Under LAN environment, everything is working fine. However, when the source and destination is about 20 hrs away, with ping response time of about 800ms, the voice is breaking. Breaking means you can not hear a smooth voice. Like the voice is being "chopped" into many pieces. The amount of packet lost is less than 10%. I have tried 8KHz, 16KHz, 32KHz.

Hi Jean, > Actually, Speex has Packet Loss Concealment (PLC) > builtin. If a packet > is missing, instead of repeating the previous one, > just try decoding by > passing NULL instead of the SpeexBits struct. > Thanks, I have made the above changes and the effect seems to be better now. > > I think jitter buffering is more correct way to > solve > > this problem

Dear all, I have one simple question. I understand that speex_encode and speex_decode takes float * as an arguement to encode and decode the sound. However, when I get the PCM data from the sound card under win32, it is a just 16 bit array. May I know how do I convert this 16 bit value to speex float format and to convert back? Is there got any routine to do this? YueWeng

Hi all, I have deploy speex 1.1.6 in my application. With no option set, I can hear that the voice sounds a little bit artificial like robot. Any idea what causes this? I use openh323 with speex, but it seems ok. Is it neccessary for me to use more other filter prior to encode the sound or after decode my sound? yueweng __________________________________ Do you Yahoo!? Yahoo! Mail -

> 16 bit ints have a range of -32768 to 32767. If you divide > -32768 by 32767.0 you end up with -1.00003051850948 which > is a bad thing. > > Try normalizing with a value of 32768.0. No. Speex expects values in the +-32767 range, not +-1.0. Just converting from int16 to float *is* the right thing to do. Jean-Marc -- Jean-Marc Valin <Jean-Marc.Valin@USherbrooke.ca>

Dear all, First, my aim is to achieve VoIP using VBR and DTX under Win32. I face a problem using speex 1.0.4 and need some help. My voice is ok and no background noise when I do NOT normalize 16 bit value to floating value. Normalized means dividing the 16 bit value by 32767. Turning on VBR is also ok but DTX has no effect. However, the speak is has a continous background beep sound AFTER I

Dear all, I am using speex 1.17 at this moment, everything works great. However, I face a problem when no packet arrived from network for a few second, my CPU usage is 100%. I step though my code and seems that (not confirmed) the speaker callback WaveOutCallback() function which call speex_decode_int(decoder_state, NULL, shortData) (when no data arrived for PLC purpose) seems to consume a

Hi, > 1) Normal given the bandwidth/bit-rate used Do you mean the bit-rate that I should set in the speex codec? > 2) A conditioning problem with your audio (i.e. DC > not removed) What is DC? YueWeng __________________________________ Do you Yahoo!? Yahoo! Mail - now with 250MB free storage. Learn more. http://info.mail.yahoo.com/mail_250

Hi, I saw there is a new release of speex. What is this SPEEX_PLC_TUNING option for? And is there a more complete list of changes? Because I like to determine if it is interesting to update to 1.1.8 Greetings Jeroen de Kleijn (developer of VoIPerized)

Hi, When an audiopacket is received too late I could interpolate this frame. The problem is that I don't know if it is a true bufferloss or just the last audiopacket of a talkspurt. Now my question is if it's harmfull for the audioquality that at the end of the talkspurt one frame is interpolated? Or would this be almost inpossible to hear since the last audiopacket in the talkspurt

On Tue, 11 Oct 2005 10:36:51 +0200, Jean-Marc Valin <Jean-Marc.Valin@USherbrooke.ca> wrote: > Source code at: > http://people.xiph.org/~jm/speexclient/ I rewritten my program so it is more similar to yours, I grabbed your code for storing and retrieving echo, and I have better results, I think the echo is cancelled in about 50% but it still can be heared. I think there is a

Hi, I am using the speex 1.0.4 library from Windows. I have posted my problem before but didn't get a solution. I am doing an VOIP project in which i am recording sound and streaming it to the peer. I wanted to encode and decode wav files that brought me to this site. I am recording sound in the following format:- m_WaveFormatEx.wFormatTag = WAVE_FORMAT_PCM;

I use Speex with dialup modem users. Even if the dialup modem is "56K", you should assume that upstream bandwidth available is only 20-30kbps at the most. I use 16kHz wideband mode and VBR quality 2. Also, I send 80ms (4 frames) per packet, because there is an overhead of approximately 33 bytes per packet due to UDP (8 bytes), IP (20 bytes), and PPP (~5 bytes) headers. If you

Hi Jean-Marc, For the residue in the echo_cancel function I pass in a pointer to an array of floats. First I used an array the size of the framesize, but I discovered this causes crashes because the echo_cancel funtion writes framesize+1 floats to the pointer. I find this plus one strange, since almost all things are exactly the framesize. Is this an error in the echo_cancel function or simply

Hmmm... sorry i mislead you Tay... i didn't realise my encoder (libfishsound) was again shifting the data i gave it back to being 32767 based internally. Zen. ----- Original Message ----- From: "Jean-Marc Valin" <Jean-Marc.Valin@USherbrooke.ca> To: "Tay YueWeng" <yueweng@yahoo.com> Cc: "speex" <speex-dev@xiph.org> Sent: Wednesday, December

Good day. Some suggestions. 1. I think it would be useful to add support of "seekable streams" to Speex API. It seems many developers need functions like: GetDuration(), SeekToTime(...),... Something like VorbisFile for Ogg Vorbis. 2. It would be useful to add some samples on working with Speex files. E.g. something like vcut for Ogg Vorbis. Vitaly.

Hi, This applies to everyone having (or suspecting) problems with Speex. The first thing to do is to encode the file in wav format and use speexenc/speexdec on it. If you're getting something different with your application, it's likely buggy. If the result isn't OK, then it can be: 1) Normal given the bandwidth/bit-rate used 2) A conditioning problem with your audio (i.e. DC not