similar to: Conversion problem in encoding and decoding

Hi, I am new to this newsgroup. I have a problem using the Speex library. I have recorded sound data using a character buffer like char recMessg[BUFFERSIZE]. I have problem converting this to the float array for encoding. Also help me in converting the decoded information in float array back to character array. Can anybody give me function codes for this conversion. Thanks in

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;

openh323 speexcodec.cxx has: BOOL SpeexCodec::EncodeFrame(BYTE * buffer, unsigned & length) { // convert PCM to float float floatData[SAMPLES_PER_FRAME]; PINDEX i; for (i = 0; i < SAMPLES_PER_FRAME; i++) floatData[i] = sampleBuffer[i]; // encode PCM data in sampleBuffer to buffer speex_bits_reset(bits); speex_encode(coder_state, floatData, bits);

Vinod Vijayan wrote: >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:- > >

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

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

> 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

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

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

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 -

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, 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

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

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

On 2016-07-12 9:34 AM, Aleksandr Mikoff wrote: > If I encode each frame, decode it and send it back to the speakers I > hear a very noisy signal - it is so large, so it is almost impossible to > distinguish words in my speech... This effect can be caused by some data type issues: - If you are using opus_encode() and opus_decode(), make sure to use 16-bit signed integer samples in the

Hello Aleksandar You mentioned "Now I have the clear enough sound with 30% MCU load for full duplex sound." How are you calculating the load??? Regards Amit -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.xiph.org/pipermail/opus/attachments/20160717/a85f6ab0/attachment.html>