[Libav-user] OPUS transcoding to AAC but 960 sample increase to 1024 with a nomalized blank data.

[강구철]

Im transcode voice comming from WebRTC through by RTP with h264 video.

received sound unit is 20msec OPUS stereo 48000 2channel sample per second

its good decoded to PCM32 FLTP type and good play. 

after decode I encode to AAC 48000 stereo frame nb_samples is 960. but
encoding ffmpeg aac function 

attach 64 samples every each decoded raw PCM samples. what should I do for
it to improving final aac product quality ?

 

now I found AAC Context be able to control cypher block size 1024 to 960.
some documents say aac encoder default block is 1024.

 

AACContext *ac = (AACContext*)aac_context->priv_data;

MPEG4AudioConfig *m4ac = &(ac->oc[0].m4ac);

m4ac->frame_length_short = 1;//1:960, 0:1024

 

is this right approching ? appriciate any kinds of oppinion of you guys!!

 

 

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