Summary of the bug:
An error occurred during aac playback in the latest version of the chrome browser from Feb 2018.
In some cases, changing bitrate option will return to normal.
This uploaded sound source file has all failed the aac with option 320k, 330k, and 340k bitrate.
Normally played in the 256k bitrate option.
Is it encoder problem of ffmpeg? decoder problem in chrome browser? or problem with the sound file?
Enviroment:
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OS : MS Windows 7
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Chrome version : 67.0.3396.62
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Drag and drop the aac sound file on chrome browser.
Chrom browser error log (in chrome://media-internals):
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00:00:15 657 event PLAY
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00:00:19 738 error Failed to send audio packet for decoding: timestamp=49644263 duration=23220 size=775 side_data_size=0 is_key_frame=1 encrypted=0 discard_padding (ms)=(0, 0)
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00:00:19 738 error audio decode error
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00:00:19 759 error audio error during playing, status: PIPELINE_ERROR_DECODE
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00:00:19 759 pipeline_error PIPELINE_ERROR_DECODE
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00:00:19 759 pipeline_state kStopping
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00:00:19 760 pipeline_state kStopped
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00:00:19 760 event PAUSE
How to reproduce:
% ffmpeg -i origin.flac -y -c:a aac -b:a 320k -ar 44100 -movflags +faststart -map 0:a -map_metadata -1 -report test.m4a
ffmpeg started on 2018-06-04 at 17:16:22
Report written to "ffmpeg-20180604-171622.log"
built with gcc 4.4.7 (GCC) 20120313 (Red Hat 4.4.7-17)
configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --enable-shared --disable-static --enable-runtime-cpudetect --enable-gpl --enable-version3 --enable-postproc --enable-avfilter --enable-pthreads --disable-avisynth --enable-libopencv --enable-libdc1394 --enable-libgsm --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libspeex --enable-libtheora --enable-bzlib --enable-libass --enable-libdc1394 --enable-libfreetype --enable-openal --enable-libopus --enable-libpulse --enable-libv4l2 --enable-libvpx --disable-debug --enable-libvorbis --enable-libxvid --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -fPIC' --disable-stripping --extra-libs=-lstdc++ --enable-libfdk-aac --enable-nonfree
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
Input #0, flac, from 'origin.flac':
Metadata:
COMMENT : 2c99cabad92fd9e30ad51951c0bd69ec
Duration: 00:02:25.79, start: 0.000000, bitrate: 1048 kb/s
Stream #0:0: Audio: flac, 44100 Hz, stereo, s16
Stream mapping:
Stream #0:0 -> #0:0 (flac (native) -> aac (native))
Press [q] to stop, [?] for help
Output #0, ipod, to 'test.m4a':
Metadata:
encoder : Lavf58.12.100
Stream #0:0: Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp (16 bit), 320 kb/s
Metadata:
encoder : Lavc58.18.100 aac
[ipod @ 0x12dbe40] Starting second pass: moving the moov atom to the beginning of the file
size= 5777kB time=00:02:25.79 bitrate= 324.6kbits/s speed=28.7x
video:0kB audio:5752kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.441327%
[aac @ 0x12dd540] Qavg: 836.753
Patches should be submitted to the ffmpeg-devel mailing list and not this bug tracker.
Hey cehoyos, appreciate your help always.
I thought bitrate option was causing a problem.
I didn't know it is encoder or decoder problem, so got to ask it here that I often visit.
As you say, I will also contact Google Chrome.
What do you think about is it a problem with the decoder in Chrome?