Using “audio_fifo” to cache samples, When the samples number in fifo is enough to 1024, and then encode it. The last, If the number of samples is less than 1024 in fifo, fill with mute
look at “doc/examples/transcode_aac.c” LeiHec [email protected]<mailto:[email protected]> 在 2018年10月26日，下午4:34，강구철 <[email protected]<mailto:[email protected]>> 写道： 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!! _______________________________________________ Libav-user mailing list [email protected]<mailto:[email protected]> http://ffmpeg.org/mailman/listinfo/libav-user
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