[FFmpeg-trac] #7271(avformat:open): Converting aac fails (was: Converting m4a file to mp3 not wokring with ffmpeg 4.0)
FFmpeg
trac at avcodec.org
Fri Jun 22 14:45:35 EEST 2018
#7271: Converting aac fails
-------------------------------------+-------------------------------------
Reporter: eangelov | Owner:
Type: defect | Status: open
Priority: important | Component: avformat
Version: git-master | Resolution:
Keywords: aac | Blocked By:
regression | Reproduced by developer: 1
Blocking: |
Analyzed by developer: 0 |
-------------------------------------+-------------------------------------
Changes (by cehoyos):
* status: new => open
* component: ffmpeg => avformat
* priority: normal => important
* version: unspecified => git-master
* keywords: => aac regression
* reproduced: 0 => 1
Old description:
> After updating to version 4.0 converting m4a to mp3 broke
> PS D:\> .\ffmpeg.exe -i .\test.m4a test.mp3
> ffmpeg version 4.0 Copyright (c) 2000-2018 the FFmpeg developers
> built with gcc 7.3.0 (GCC)
> configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-
> bzlib --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
> libavutil 56. 14.100 / 56. 14.100
> libavcodec 58. 18.100 / 58. 18.100
> libavformat 58. 12.100 / 58. 12.100
> libavdevice 58. 3.100 / 58. 3.100
> libavfilter 7. 16.100 / 7. 16.100
> libswscale 5. 1.100 / 5. 1.100
> libswresample 3. 1.100 / 3. 1.100
> libpostproc 55. 1.100 / 55. 1.100
> [aac @ 0000028945cea440] Could not find codec parameters for stream 0
> (Audio: aac, 0 channels, fltp): unspecified sample rate
> Consider increasing the value for the 'analyzeduration' and 'probesize'
> options
> Input #0, aac, from '.\test.m4a':
> Duration: N/A, bitrate: N/A
> Stream #0:0: Audio: aac, 0 channels, fltp
> Output #0, mp3, to 'test.mp3':
> Output file #0 does not contain any stream
> [[BR]]
> [[BR]]
> [[BR]]
>
> When using ffmpeg 3.4.2 with the same input, the file in converted
> PS D:\> .\ffmpeg_3.exe -i .\test.m4a test.mp3
> ffmpeg version 3.4.2 Copyright (c) 2000-2018 the FFmpeg developers
> built with gcc 7.3.0 (GCC)
> configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-
> bzlib --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-libmfx --enable-
> cuda --enable-cuvid --enable-d3d11va --enable-nvenc --enable-dxva2
> --enable-avisynth
> libavutil 55. 78.100 / 55. 78.100
> libavcodec 57.107.100 / 57.107.100
> libavformat 57. 83.100 / 57. 83.100
> libavdevice 57. 10.100 / 57. 10.100
> libavfilter 6.107.100 / 6.107.100
> libswscale 4. 8.100 / 4. 8.100
> libswresample 2. 9.100 / 2. 9.100
> libpostproc 54. 7.100 / 54. 7.100
> [aac @ 000001ac8a913800] Input buffer exhausted before END element found
> [aac @ 000001ac89146ea0] Estimating duration from bitrate, this may be
> inaccurate
> Input #0, aac, from '.\test.m4a':
> Duration: 00:05:08.76, bitrate: 146 kb/s
> Stream #0:0: Audio: aac (LC), 48000 Hz, stereo, fltp, 146 kb/s
> Stream mapping:
> Stream #0:0 -> #0:0 (aac (native) -> mp3 (libmp3lame))
> Press [q] to stop, [?] for help
> [aac @ 000001ac8a9b4700] Input buffer exhausted before END element found
> Error while decoding stream #0:0: Invalid data found when processing
> input
> Output #0, mp3, to 'test.mp3':
> Metadata:
> TSSE : Lavf57.83.100
> Stream #0:0: Audio: mp3 (libmp3lame), 48000 Hz, stereo, fltp
> Metadata:
> encoder : Lavc57.107.100 libmp3lame
> size= 4697kB time=00:05:00.57 bitrate= 128.0kbits/s speed=17.3x
> video:0kB audio:4696kB subtitle:0kB other streams:0kB global headers:0kB
> muxing overhead: 0.004928%
>
> Both ffmpeg versions are static builds for windows 10 64 bit
New description:
After updating to version 4.0 converting m4a to mp3 broke
{{{
PS D:\> .\ffmpeg.exe -i .\test.m4a test.mp3
ffmpeg version 4.0 Copyright (c) 2000-2018 the FFmpeg developers
built with gcc 7.3.0 (GCC)
configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-
bzlib --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
libavutil 56. 14.100 / 56. 14.100
libavcodec 58. 18.100 / 58. 18.100
libavformat 58. 12.100 / 58. 12.100
libavdevice 58. 3.100 / 58. 3.100
libavfilter 7. 16.100 / 7. 16.100
libswscale 5. 1.100 / 5. 1.100
libswresample 3. 1.100 / 3. 1.100
libpostproc 55. 1.100 / 55. 1.100
[aac @ 0000028945cea440] Could not find codec parameters for stream 0
(Audio: aac, 0 channels, fltp): unspecified sample rate
Consider increasing the value for the 'analyzeduration' and 'probesize'
options
Input #0, aac, from '.\test.m4a':
Duration: N/A, bitrate: N/A
Stream #0:0: Audio: aac, 0 channels, fltp
Output #0, mp3, to 'test.mp3':
Output file #0 does not contain any stream
}}}
[[BR]]
[[BR]]
[[BR]]
When using ffmpeg 3.4.2 with the same input, the file in converted
{{{
PS D:\> .\ffmpeg_3.exe -i .\test.m4a test.mp3
ffmpeg version 3.4.2 Copyright (c) 2000-2018 the FFmpeg developers
built with gcc 7.3.0 (GCC)
configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-
bzlib --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-libmfx --enable-cuda --enable-
cuvid --enable-d3d11va --enable-nvenc --enable-dxva2 --enable-avisynth
libavutil 55. 78.100 / 55. 78.100
libavcodec 57.107.100 / 57.107.100
libavformat 57. 83.100 / 57. 83.100
libavdevice 57. 10.100 / 57. 10.100
libavfilter 6.107.100 / 6.107.100
libswscale 4. 8.100 / 4. 8.100
libswresample 2. 9.100 / 2. 9.100
libpostproc 54. 7.100 / 54. 7.100
[aac @ 000001ac8a913800] Input buffer exhausted before END element found
[aac @ 000001ac89146ea0] Estimating duration from bitrate, this may be
inaccurate
Input #0, aac, from '.\test.m4a':
Duration: 00:05:08.76, bitrate: 146 kb/s
Stream #0:0: Audio: aac (LC), 48000 Hz, stereo, fltp, 146 kb/s
Stream mapping:
Stream #0:0 -> #0:0 (aac (native) -> mp3 (libmp3lame))
Press [q] to stop, [?] for help
[aac @ 000001ac8a9b4700] Input buffer exhausted before END element found
Error while decoding stream #0:0: Invalid data found when processing input
Output #0, mp3, to 'test.mp3':
Metadata:
TSSE : Lavf57.83.100
Stream #0:0: Audio: mp3 (libmp3lame), 48000 Hz, stereo, fltp
Metadata:
encoder : Lavc57.107.100 libmp3lame
size= 4697kB time=00:05:00.57 bitrate= 128.0kbits/s speed=17.3x
video:0kB audio:4696kB subtitle:0kB other streams:0kB global headers:0kB
muxing overhead: 0.004928%
}}}
Both ffmpeg versions are static builds for windows 10 64 bit
--
Comment:
Regression since e023334661e6eafcf638ffc2a780fd495fc25ec9
--
Ticket URL: <https://trac.ffmpeg.org/ticket/7271#comment:1>
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