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07.19.2007 at 12:28AM PDT, ID: 22706282
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G.729 20ms Payloads in RTP

Tags: rtp
Hi Experts,
I'm working on decoding G.729 received by RTP packets from Cisco Call Manager. I'm able to playback the sound, although the voice sounds almost alien or like a robot, with some clicking noises. I'm using the Open G.729 lib from VoiceAge.

I supose my question is: can I simply take the payload from the RTP packet and split it in half, sending the first 10 Bytes to the decoder, then the second 10 Bytes to the decoder? Or is there some byte order that I must format the payload from the RTP packet into?

For example, let's say the payload looks like this:
64 61 75 af 35 6b d4 31 8e 96 62 05 a8 a6 bc bb dc 42 b7 51

Can I send the following "64 61 75 af 35 6b d4 31 8e 96" (in this order as an array of 1 Byte unsigned chars) to the decoder first then getting back PCM1 (an array of 2 Byte words), then send "62 05 a8 a6 bc bb dc 42 b7 51" (just like the first 10 Bytes) to the decoder then getting back PCM2? Once I have PCM1 and PCM2 just concatinating the arrays together?

I'm not sure if that example made sense, but I'm curious if because the RTP payload is 20ms of G.729 (which is normally decoded in 10ms) is there a special way in which this payload needs to be extracted and ordered before converting to PCM? I'm thinking this might be my problem, but I'm not sure.

Also my wave format for the data once decoded is:

nBlockAlign = 2;
wBitsPerSample = nBlockAlign * 8;
wFormatTag = 1;
nChannels = 1;
nSamplesPerSec = 8000;
nAvgBytesPerSec = (nSamplesPerSec * nChannels * wBitsPerSample)/8;


Ok, so I realize this is a pretty big question, but really any help I can get will be greatly appreciated!

Thanks in advance,
Brandon
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Question Stats
Zone: Programming
Question Asked By: bdunz19
Solution Provided By: MerijnB
Participating Experts: 2
Solution Grade: A
Views: 72
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07.19.2007 at 01:19AM PDT, ID: 19520143

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07.19.2007 at 02:12AM PDT, ID: 19520331

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07.19.2007 at 02:15AM PDT, ID: 19520341

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07.19.2007 at 02:19AM PDT, ID: 19520355

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07.19.2007 at 03:32AM PDT, ID: 19520670

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07.19.2007 at 03:38AM PDT, ID: 19520716

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07.19.2007 at 04:46AM PDT, ID: 19521010

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07.19.2007 at 04:47AM PDT, ID: 19521016

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07.19.2007 at 01:19AM PDT, ID: 19520143
I don't have experience with this particular codec, but I would start by just dumping what you receive to disk, and try to decode that with an external application to see if that works ok.

You probably can send the data in blocks to the codec, but it's probably best to send it in blocks of whole samples, I don't know what that is in this case, but probably blocks of 8 or 16 bytes, so sending 10 bytes might not be a good idea.

If it's decoded correctly, you can concatenate the PCM blocks without problems, assuming the codec gives back blocks of whole samples again.

Do you have some documentation on this codec you are using? You will probably find more info there.
Accepted Solution
 
07.19.2007 at 02:12AM PDT, ID: 19520331
Hey thanks MerijnB,

It looks like your suggestion of writing the whole file to disk and using the codec to decode it worked great. I don't know why I didn't think of testing that (that would tell me if the RTP data is formatted)!? The wav pcm file My program outputed played back in windows media player perfectly.

So now, it looks like I have to concentrate on my code that is doing the streaming playback. If you have any experiance with that please let me know.

Based on my PCM Wave format above, wouldn't I have full samples returned by the decoder? One PCM sample is 2 Bytes, which the decoder for every 10 Bytes of g.729 will return 160 Bytes of PCM, meaning there will be a total of 160 PCM samples for every 20 Bytes of g.729 right?? So is that 160 samples enough to playback in realtime? Or am I missing something?

Thanks man,
Brandon
 
07.19.2007 at 02:15AM PDT, ID: 19520341
I'm quite sure if you playback with blocks of 160 samples you will get choppy audio, try buffering pcm data until you have blocks of 8kb or something. Just experiment somewhat.
 
07.19.2007 at 02:19AM PDT, ID: 19520355
I just checked some PCM play code of myself, and I'm using blocks of 16kb at the moment.
 
07.19.2007 at 03:32AM PDT, ID: 19520670
Hey MerijnB,
Turns out my play back code was the issue... Although I'm not sure what was the problem, because I just simply rewrote it and it worked! Because I'm using the program for realtime audio I don't really have the luxury of buffering the frames, though I might get away with buffering 320 Bytes or even 480 without any problems (it apears multiples of 160 Bytes works).

Thanks for you help man! It sounds like you enjoy this area or work... I just got started in audio myself, but already it is challenging and that's what I like.

Anyway, thanks again and your first post was most helpful to get me back to the basics to verify the RTP streams were not specially formatted.

Cheers,
Brandon
 
07.19.2007 at 03:38AM PDT, ID: 19520716
I forgot that the 16kb is used for 44kHz 16 bit audio, which causes slightly less delay then in your situation.
Yes, I do enjoy this kind of work, check our website to see our link with audio: www.aviavox.com :)

cheers!
 
07.19.2007 at 04:46AM PDT, ID: 19521010
Hey MerijnB,
I checked out you web site. Those voice demos are amazing! Great work if you were part in creating that program!

I will definatly keep your company in mind if mine ever needs artificial voice. That is probably one of the best I've ever heard (if not the best).
 
07.19.2007 at 04:47AM PDT, ID: 19521016
thanks! :)
 
 
12.10.2007 at 08:35PM PST, ID: 20447323
hi guys!

I'm developing a record call system. I need to record to a file every RTP stream that I receive from a CTI Port in the CallManager.
Do you know how to record the rtp stream to a file? Should I use the byte buffer of the media locator?
Thank tou!
 
 
 
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