I am struggling to make AAC encoding work with libav. I have an example working for MP3. But I can't seem to figure out what is needed to make AAC encoding work. I get a whole bunch of errors when playing my encoder output.
I suspect I need some extra configuration for encoder context, but I have not been able to find hints that can help me. I have made a repo here with my code: https://github.com/taxfromdk/audio_encode_test.git The juicy code is here: https://github.com/taxfromdk/audio_encode_test/blob/master/encode_audio.cpp It can be run by checking the project out out and run "make test" I suspect it could be related to my workflow, so I will just note that I test the output data by creating a file with either the extension .aac or .mp3 where I dump the raw encoded packages and play through ffplay. I am on ffmpeg version n4.3.1 under Arch linux. The output from ffplay in case of the AAC file is listed below. I would appreciate pointers to what is needed for the codec context to spit out working AAC, as I need it to accompany some video. :) Kind regards Jesper ------ ffplay raw.aac -autoexit ffplay version n4.3.1 Copyright (c) 2003-2020 the FFmpeg developers [aac @ 0x7fe6f8000bc0] Format aac detected only with low score of 1, misdetection possible! [aac @ 0x7fe6f8002600] Error decoding AAC frame header. [aac @ 0x7fe6f8002600] More than one AAC RDB per ADTS frame is not implemented. Update your FFmpeg version to the newest one from Git. If the problem still o ccurs, it means that your file has a feature which has not been implemented. [aac @ 0x7fe6f8002600] Sample rate index in program config element does not match the sample rate index configured by the container. [aac @ 0x7fe6f8002600] Inconsistent channel configuration. [aac @ 0x7fe6f8002600] get_buffer() failed [aac @ 0x7fe6f8002600] invalid band type [aac @ 0x7fe6f8002600] channel element 2.14 is not allocated [aac @ 0x7fe6f8002600] Sample rate index in program config element does not match the sample rate index configured by the container. [aac @ 0x7fe6f8002600] Too large remapped id is not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented. [aac @ 0x7fe6f8002600] If you want to help, upload a sample of this file to https://streams.videolan.org/upload/ and contact the ffmpeg-devel mailing list. ( ffmpeg-de...@ffmpeg.org) [aac @ 0x7fe6f8002600] Multiple frames in a packet. [aac @ 0x7fe6f8002600] Reserved bit set. [aac @ 0x7fe6f8002600] Number of scalefactor bands in group (58) exceeds limit (47). [aac @ 0x7fe6f8002600] Reserved bit set. [aac @ 0x7fe6f8002600] Number of bands (24) exceeds limit (22). [aac @ 0x7fe6f8002600] channel element 1.0 is not allocated [aac @ 0x7fe6f8002600] Reserved bit set. [aac @ 0x7fe6f8002600] Number of scalefactor bands in group (15) exceeds limit (12). [aac @ 0x7fe6f8002600] channel element 1.10 is not allocated [aac @ 0x7fe6f8002600] SBR was found before the first channel element. [aac @ 0x7fe6f8002600] channel element 3.6 is not allocated [aac @ 0x7fe6f8000bc0] Packet corrupt (stream = 0, dts = NOPTS). [aac @ 0x7fe6f8002600] Sample rate index in program config element does not match the sample rate index configured by the container. [aac @ 0x7fe6f8002600] Inconsistent channel configuration. [aac @ 0x7fe6f8002600] get_buffer() failed [aac @ 0x7fe6f8002600] channel element 3.14 is not allocated [aac @ 0x7fe6f8002600] Sample rate index in program config element does not match the sample rate index configured by the container. [aac @ 0x7fe6f8002600] Inconsistent channel configuration. [aac @ 0x7fe6f8002600] get_buffer() failed [aac @ 0x7fe6f8000bc0] decoding for stream 0 failed [aac @ 0x7fe6f8000bc0] Estimating duration from bitrate, this may be inaccurate [aac @ 0x7fe6f8000bc0] Could not find codec parameters for stream 0 (Audio: aac (Main), mono, fltp, 58 kb/s): unspecified sample rate Consider increasing the value for the 'analyzeduration' and 'probesize' options Input #0, aac, from 'raw.aac': Duration: 00:00:04.26, bitrate: 58 kb/s Stream #0:0: Audio: aac (Main), mono, fltp, 58 kb/s Failed to open file 'raw.aac' or configure filtergraph
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