Hi Hi, Im using the ffmpeg API to re-encode video over a network stream (HTTP). Is it possible to pad out AAC audio samples to a specific size? Is it a case of appending hex 0 to samples or is there a construct for AAC like AVC Filler NAL to tell the player to ignore this data. I predict frame sizes up front for MP4 files to stream this content via HTTP
also posted this question ont he #ffmpeg on IRC channel earlier. Thanks
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