similar to: 100% CPU usage

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

> 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>

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 -

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 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

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

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

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;

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

Hi I am using first time speex library, and this is my first problem. I need to decode AMR-NB packet to PCM. I read all manual instruction and I wrote these simple lines of code. bool CMMediaObj::AmrInit() { speex_bits_init(&bits); destate = speex_decoder_init(&speex_nb_mode); int tmp=1; speex_decoder_ctl(destate, SPEEX_SET_ENH,

I'm trying to use the speex API, and I think I'm missing something. I'm running on Mac OSX Tiger, and the attached microphone is a iSight. If I call the "speex_encode_int" function, and then immediately call the "speex_decode_int" function, shouldn't I get back the same data? i.e. in my example code, shouldn't the two arrays contain at least similar

On Mon, Apr 19, 2004 at 02:46:44AM -0400, Jean-Marc Valin wrote: > OK, I'll try to do *something* about it. It'll be either "support the > 1.0 ABI" or "make sure the link will break". While the former would be > nice, I'm not sure I want to really want to force 1.2 to have the same > ABI as 1.0.x. More on this when I have some time to think about it.

Hi, I am new to speex library. I am doing project in VC++. My project details are, I have to capture voice from mic and encode it then decode and play. I am capturing the voice and storing in a buffer from there i am reading frame by frame ( 160 bytes) . These bytes i am converting to short int then calling speex_encode-int(). The output buffer i am passing to doder speex_decode_int() function.

Do i always need to tell the decoder that some packets has been lost? I noticed that if i discard packets without passing NULL and advising the decoder, sometimes my system hangs and i need to reset my PDA. Since the decoder is mantaining a state, i suppose i always need to tell it a packet has been discarded through a sequence number check. Or can i just pass to speex_decode_int whatever i