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I am having a spot of difficulty with the g.722.1 decoder sample source code.
We have an application which receives 722.1 rtp packets and it attempts to decode these into raw audio information to write to file. It seems as though I have mostly done things correctly however the final raw audio seems to be bad data. Silence followed by bursts of random noise.
Is there anything in particular that I should be doing with the data from the rtp payload before passing it onto the apiG722Decode function. I am a bit confused by the phrase (from rfc3047) "The G.722.1 encoder bit stream is split into a sequence of octets (60 or 80 depending on the bit rate), and each octet is in turn mapped into an RTP octet." If the output from the encoder is put in left to right order MSB to LSB directly into the rtp payload then I have something else wrong here...
Any thoughts?
Cheers,
Jesse Baker
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We have tested this against several units sending 1, 2, and 3 frames per packet. I have also tried reordering incoming words and bytes without any success. I havn't, however, tried reversing words on the unit which is sending me 1 frame per packet. I'll get onto that and get back to you.
Thanks again.
Jesse
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An interesting by-product of this was a windows media writer was consistantly overwriting the last half of each audio sample whilst a real media writer would simply stack them one after another. The final result of this was a wmv file where the audio track was the same length as the video track and a rm file where the audio track was twice as long. And of course the audio track from the realmedia file could be extracted, speed up and listened to as normal sound, however the extracted and speed up audio track from the windows media file was horribly corrupt.
Anyway, thanks for your help. Now, onto the encoder...
Jesse
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Hello vbaranni,
at first, thanks for your fast answer.
I dont know the meaning of acronyms "vmw" and "rmw". Please explain it or write the full name of the acronyms.
I thought the input of G722.1 is 16Bit linear pcm. Ifsome clientuse before alaw- or ulaw-compression,he have to expand to 16Bit linear before using another compression sheme like G7221. Same does Intel IPPSamplesif output shopuld be a-law or u-law. After encodingsample codecsmay compresses resultwith a-law or u-law.
40 Octeds are right on 16Bit linear coded with SIREN (G722.1 16000 kbit/s)
(Octeds 40/60/80 on G722.1 with 16000/24000/32000 kbit/s)
Therefore I dont unterstand your answer. Ill be glad if you describe it more detailed.
And in which way this problemcan be solved.
Regards
telefonie
Message Edited by telefonie on 10-13-2005 06:09 AM
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Hello,
ill try the swap-function as mentioned, but I tryed to rearange the bytes in the same kind like mentioned before and it dont work, so I think this wont work also.
After the rearagement the stream should be:
byte2, byte1, byte4, byte3, byte6, byte5, ....
Regards
Joachim
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Hi! I am have a problem at use of functions of coding - decoding G726. (I use IPP4.0 on WindowsXP SP2). I code input PCM a signal function ippsEncode_G726_16s8u 16 kbps and decode the target buffer function ippsDecode_G726_8u16s. As a result after decoding to an initial sound the crash and handicapes is added. In what here a problem? And what on volume should be used the buffer for these functions?
Thanks for the help
Esli chto - mojno po russki na mail
Alexander
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Hello Rick,
I see you decode G722.1 from Microsoft. Am I right?Sirren should be the same at 16KBit/s, but it didnt work. So Microsoft do somthing strange with Sirren. Do you tryed also the ACM-Codec Vivo-Active Sirren?
You can also contactme directy to joachimneumann@t-online.de
Regards
Joachim
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Hello Alexander,
Wrong discussion Group (This is G722.1).
Try the demo decoder decoderG726 and encoder to see correct working. Downloadable here at intel "codec samples" for ipps.
The PCM Signal should be linear 16Bit.
Initalization habe to be done for -16 or-24 or-32 or-40 before encoding/decoding.
Hint: There is another Topic for rtp packetizationof G726 at this forum. Search for G726.
Regards
Joachim
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