Skip to main content


Please note that most of the software linked on this forum is likely to be safe to use. If you are unsure, feel free to ask in the relevant topics, or send a private message to an administrator or moderator. To help curb the problems of false positives, or in the event that you do find actual malware, you can contribute through the article linked here.
Topic: AMR-WB+ audio encoding (Read 8415 times) previous topic - next topic
0 Members and 1 Guest are viewing this topic.

AMR-WB+ audio encoding

I have downloaded 3gpp's AMR-WB+ open source code and would like to encoder a file at 16KHz stereo. My input is a 16KHz stereo file. I have given these encoder options:
mi=24 and isf=0.625 corresponding to 16KHz stereo modes. But I see that the encoder is running @ 25.6KHz. Shouldn't it be @ 16K rate? Also there is an oversampling being done @48k before encoder loop.
However the encoded bitstream has the isf factor as 3 which corresponds to 16KHz. When I decoded this file, the output is a 48KHz rather that a 16KHz. Does AMR-WB+ encoder and decoder not support native sampling rate encoding / decoding? It would be great if experts can throw some light on this.

SimplePortal 1.0.0 RC1 © 2008-2021