Hi,
I'm using ffmpeg to convert aac files to m4a files without any other
conversion, so just demuxing and muxing. All of the aac files can be
played flawlessly with both players that I use (AIMP on Windows and
jetAudio on Adroid), and can be opened and edited with Audacity. So the
files seem ok to me.
But some of these files can't be converted by ffmpeg, stopping with
either the error " decoding for stream 0 failed" (usually with the error
"More than one AAC RDB per ADTS frame is not implemented. ..." but not
always) or "Error decoding AAC frame header". This is on Windows 10 with
ffmpeg version N-92752-g16ec62bbf4.
The command that I use is "ffmpeg -i example.aac -c copy -bitexact
example.m4a" (from a C++ application that goes over the directories). I
also tried without -bitexact, but the problem remains.
Further info: these aac files were generated by streamwriter
(https://streamwriter.org/). I have about 850 of these, 48 of them
couldn't be converted, stopping with one these two errors.
Should I deliver some of these files as an attachment or in another way?
Below the output of converting some of these files:
ffmpeg version N-92752-g16ec62bbf4 Copyright (c) 2000-2018 the FFmpeg
developers
built with gcc 8.2.1 (GCC) 20181201
configuration: --enable-gpl --enable-version3 --enable-sdl2
--enable-fontconfig --enable-gnutls --enable-iconv --enable-libass
--enable-libbluray --enable-libfreetype --enable-libmp3lame
--enable-libopencore-amrnb --enable-libopencore-amrwb
--enable-libopenjpeg --enable-libopus --enable-libshine
--enable-libsnappy --enable-libsoxr --enable-libtheora
--enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp
--enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg
--enable-lzma --enable-zlib --enable-gmp --enable-libvidstab
--enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa
--enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx
--enable-amf --enable-ffnvcodec --enable-cuvid --enable-d3d11va
--enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth
--enable-libopenmpt
libavutil 56. 24.101 / 56. 24.101
libavcodec 58. 42.104 / 58. 42.104
libavformat 58. 24.101 / 58. 24.101
libavdevice 58. 6.101 / 58. 6.101
libavfilter 7. 46.101 / 7. 46.101
libswscale 5. 4.100 / 5. 4.100
libswresample 3. 4.100 / 3. 4.100
libpostproc 55. 4.100 / 55. 4.100
[aac @ 000001c3f8623240] Error decoding AAC frame header.
[aac @ 000001c3f85fb080] Estimating duration from bitrate, this may be
inaccurate
Input #0, aac, from 'example1.aac':
Duration: 00:03:12.84, bitrate: 81 kb/s
Stream #0:0: Audio: aac (HE-AAC), 44100 Hz, stereo, fltp, 81 kb/s
Output #0, ipod, to 'example1.m4a':
Metadata:
title : example
encoder : Lavf58.24.101
Stream #0:0: Audio: aac (HE-AAC) (mp4a / 0x6134706D), 44100 Hz,
stereo, fltp, 81 kb/s
Stream mapping:
Stream #0:0 -> #0:0 (copy)
Press [q] to stop, [?] for help
[AVBSFContext @ 000001c3f8612380] Error parsing ADTS frame header!
[AVBSFContext @ 000001c3f8612380] Failed to send packet to filter
aac_adtstoasc for stream 0
example1.aac: Invalid data found when processing input
size= 8kB time=00:00:00.65 bitrate= 106.0kbits/s speed=N/A
video:0kB audio:8kB subtitle:0kB other streams:0kB global headers:0kB
muxing overhead: 5.422276%
ffmpeg version N-92752-g16ec62bbf4 Copyright (c) 2000-2018 the FFmpeg
developers
built with gcc 8.2.1 (GCC) 20181201
configuration: --enable-gpl --enable-version3 --enable-sdl2
--enable-fontconfig --enable-gnutls --enable-iconv --enable-libass
--enable-libbluray --enable-libfreetype --enable-libmp3lame
--enable-libopencore-amrnb --enable-libopencore-amrwb
--enable-libopenjpeg --enable-libopus --enable-libshine
--enable-libsnappy --enable-libsoxr --enable-libtheora
--enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp
--enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg
--enable-lzma --enable-zlib --enable-gmp --enable-libvidstab
--enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa
--enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx
--enable-amf --enable-ffnvcodec --enable-cuvid --enable-d3d11va
--enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth
--enable-libopenmpt
libavutil 56. 24.101 / 56. 24.101
libavcodec 58. 42.104 / 58. 42.104
libavformat 58. 24.101 / 58. 24.101
libavdevice 58. 6.101 / 58. 6.101
libavfilter 7. 46.101 / 7. 46.101
libswscale 5. 4.100 / 5. 4.100
libswresample 3. 4.100 / 3. 4.100
libpostproc 55. 4.100 / 55. 4.100
[aac @ 0000020af77e31c0] 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 occurs, it means that your file has a feature which
has not been implemented.
[aac @ 0000020af77e31c0] Multiple frames in a packet.
[aac @ 0000020af77e31c0] SBR was found before the first channel element.
Last message repeated 1 times
[aac @ 0000020af77e31c0] channel element 1.10 is not allocated
[aac @ 0000020af77bb040] decoding for stream 0 failed
[aac @ 0000020af77bb040] Estimating duration from bitrate, this may be
inaccurate
[aac @ 0000020af77bb040] Could not find codec parameters for stream 0
(Audio: aac (LTP), 3.0, fltp, 259 kb/s): unspecified sample rate
Consider increasing the value for the 'analyzeduration' and 'probesize'
options
Input #0, aac, from 'example2.aac':
Duration: 00:14:43.76, bitrate: 259 kb/s
Stream #0:0: Audio: aac (LTP), 3.0, fltp, 259 kb/s
[ipod @ 0000020af77e5b80] sample rate not set
Could not write header for output file #0 (incorrect codec parameters
?): Invalid argument
Stream mapping:
Stream #0:0 -> #0:0 (copy)
Last message repeated 1 times
ffmpeg version N-92752-g16ec62bbf4 Copyright (c) 2000-2018 the FFmpeg
developers
built with gcc 8.2.1 (GCC) 20181201
configuration: --enable-gpl --enable-version3 --enable-sdl2
--enable-fontconfig --enable-gnutls --enable-iconv --enable-libass
--enable-libbluray --enable-libfreetype --enable-libmp3lame
--enable-libopencore-amrnb --enable-libopencore-amrwb
--enable-libopenjpeg --enable-libopus --enable-libshine
--enable-libsnappy --enable-libsoxr --enable-libtheora
--enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp
--enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg
--enable-lzma --enable-zlib --enable-gmp --enable-libvidstab
--enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa
--enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx
--enable-amf --enable-ffnvcodec --enable-cuvid --enable-d3d11va
--enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth
--enable-libopenmpt
libavutil 56. 24.101 / 56. 24.101
libavcodec 58. 42.104 / 58. 42.104
libavformat 58. 24.101 / 58. 24.101
libavdevice 58. 6.101 / 58. 6.101
libavfilter 7. 46.101 / 7. 46.101
libswscale 5. 4.100 / 5. 4.100
libswresample 3. 4.100 / 3. 4.100
libpostproc 55. 4.100 / 55. 4.100
[aac @ 00000237bd033640] Reserved bit set.
[aac @ 00000237bd033640] Number of scalefactor bands in group (63)
exceeds limit (49).
[aac @ 00000237bd00ad00] decoding for stream 0 failed
[aac @ 00000237bd00ad00] Estimating duration from bitrate, this may be
inaccurate
[aac @ 00000237bd00ad00] Could not find codec parameters for stream 0
(Audio: aac (LTP), 5.0, fltp, 1920 kb/s): unspecified sample rate
Consider increasing the value for the 'analyzeduration' and 'probesize'
options
Input #0, aac, from 'example3.aac':
Duration: 00:00:07.10, bitrate: 1920 kb/s
Stream #0:0: Audio: aac (LTP), 5.0, fltp, 1920 kb/s
[ipod @ 00000237bd420940] sample rate not set
Could not write header for output file #0 (incorrect codec parameters
?): Invalid argument
Stream mapping:
Stream #0:0 -> #0:0 (copy)
Last message repeated 1 times
Kind regards,
--
Mark
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