Hi, I am working on an MP4 file encapsulation module which use libavformat and libavcodec.
I have some issues with AAC audio, which leads to some segfault into ffmpeg code, in some cases. I am not comfortable with AAC and MPEG specifications so I am not sure which aspects I am misunderstanding. Basicly, segfault happens because ffmpeg is trying to withdraw ADTS header of my AAC packet in ff_rtp_send_aac, before making the memcpy. I do not understand why withdraw the ADTS header here, as it seems to be done in aac_adtstoasc_bsf function that I have to call earlier in my code. Then in the case of an AAC packet size of 6 bytes (without ADTS header), it crashes because of a memcpy of size -1. This happens only in rare cases, in other cases the file is well generated and the audio track is ok. Thanks for your help. Chris.
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