similar to: 100% CPU usage

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

Hi, 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 side effect. I think jitter

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

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, > 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 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;

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

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

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

> 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, 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

Hello, Could you please explain or send an example for decoding a packet that contains multiple Speex frames. According to the API document the speex_decode_int() decodes only one frame of speech. Best regards, Danny Peled R&D Software Engineer Intelligence Solutions NICE Systems Ltd. ISRAEL *+ (972) 9 769 7161 ? + (972) 54 4 602840 *Danny.Peled at nice.com <mailto:*Danny.Peled at

Hello I have lots of folders of individual Scottish river catchments on my uni One Drive. Each folder is labelled with the river name eg "Tay" and they are all in a folder named "Scotland" I want to list the folders on One Drive so that I can cross check that I have them all against a list of folders on my laptop. Can I somehow use list.files() - I've tried various things

Jean-Marc Valin wrote: > > > The decoder will return -1 if it can't decode anything else, so you just > do something like: > > while(1) { > err = speex_decode_int(state, bits, pcm); > if (err==-1) > break; > output_data(pcm); > } I think I still didn't fully understand. As a test I encoded some frames (wideband, with quality set to 8),

>> Consider a device that only has enough ROM to store one set of >> quantization tables (the limitation could also be about speed, network, >> ...). If you specify MUST be able to decode, then it means that this >> device simply *cannot* implement the spec *at all*. This is bad for >> interoperability. > > For me: device which don't have all mode

Please compare samples: http://www.ivo.pl/ivo2/sound/krwawa_org.wav http://www.ivo.pl/ivo2/sound/krwawa_fixed.wav http://www.ivo.pl/ivo2/sound/krwawa_float.wav krwawa_org.wav is original file. Both output files was created using same program by compressing from wav to speex, then decompressing from speex to wav (using speex_encode_int and speex_decode_int). Only difference was used speex

Hi, I've been looking at using speex over RTP. Section 3.3 of draft-ietf-avt-rtp-speex-05 states that: "For the purposes of packetizing the bit stream in RTP, it is only necessary to consider the sequence of bits as output by the Speex encoder [speex_manual], and present the same sequence to the decoder." So, I assume that I just pass the whole contents of the RTP packet (less the