Hello, I’m encoding my wav files to HE AAC v2 in 44kHz/32 Kbps and 44/64.
Ffprobe is correctly showing that the audio is HE AAC v2 stereo with the correct sampling rate and nitrate. When I open the audio in Exoplayer or Apple’s AVURLAsset code, they report that the sampling rate is 22050kHz instead. When we looked into it, it looks like there may be an atom in the MP4 container that is misrepresenting the sampling rate. I tried looking for it but couldn’t find it. Is this a known problem? In addition, I’ve learned that the time stamps of the audio samples are different for a 44/32 HEAAC file vs a 44/128 LCAAC one. This prevents seamless HLS Adaptive Streaming between the two. Is this by design? Is it because of the way HE AAC works? I’ve filed an issue about this to Exoplayer if it would help to better understand the problem. https://github.com/google/ExoPlayer/issues/3971 Thanks Ronak Sent from my iPhone _______________________________________________ ffmpeg-user mailing list [email protected] http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email [email protected] with subject "unsubscribe".
