[Libav-user] AAC decode problems

[Power Pan]

is that VBR ? sampling rate changed during play ?
BTW, how we could get current play time ?

On Thu, Sep 14, 2017 at 7:22 PM, Anton Shekhovtsov <shekh.anton at gmail.com>
wrote:

>
>
> 2017-09-14 14:06 GMT+03:00 Info || Non-Lethal Applications <
> info at non-lethal-applications.com>:
>
>> Hi guys,
>>
>> I just received a movie file with AAC audio which doesn't play correctly
>> in my player using the ffmpeg libraries.
>> It sounded “robotic” and the audio ran out of sync rather quickly.
>>
>> Ffprobe output:
>>
>> ffprobe version 3.3.1 Copyright (c) 2007-2017 the FFmpeg developers
>>   built with llvm-gcc 4.2.1 (LLVM build 2336.11.00)
>>   configuration: --prefix=/Volumes/Ramdisk/sw --enable-gpl
>> --enable-pthreads --enable-version3 --enable-libspeex --enable-libvpx
>> --disable-decoder=libvpx --enable-libmp3lame --enable-libtheora
>> --enable-libvorbis --enable-libx264 --enable-avfilter
>> --enable-libopencore_amrwb --enable-libopencore_amrnb --enable-filters
>> --enable-libgsm --enable-libvidstab --enable-libx265 --disable-doc
>> --arch=x86_64 --enable-runtime-cpudetect
>>   libavutil      55. 58.100 / 55. 58.100
>>   libavcodec     57. 89.100 / 57. 89.100
>>   libavformat    57. 71.100 / 57. 71.100
>>   libavdevice    57.  6.100 / 57.  6.100
>>   libavfilter     6. 82.100 /  6. 82.100
>>   libswscale      4.  6.100 /  4.  6.100
>>   libswresample   2.  7.100 /  2.  7.100
>>   libpostproc    54.  5.100 / 54.  5.100
>> Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/Users/flo/Downloads/wetransfer-6ec47f/IQT
>> Capture 170911 Unity Round1.mov':
>>   Metadata:
>>     major_brand     : qt
>>     minor_version   : 512
>>     compatible_brands: qt
>>     encoder         : Lavf57.66.102
>>   Duration: 00:07:58.81, start: 0.000000, bitrate: 2672 kb/s
>>     Stream #0:0(eng): Video: h264 (High) (avc1 / 0x31637661), yuv420p,
>> 1280x720, 2499 kb/s, 29.97 fps, 29.97 tbr, 30k tbn, 59.94 tbc (default)
>>     Metadata:
>>       handler_name    : DataHandler
>>     Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz,
>> stereo, fltp, 162 kb/s (default)
>>     Metadata:
>>       handler_name    : DataHandler
>>
>> By examining the file I found out that the packet durations of the AAC
>> stream were not 1024 as they are with all the other files I have but rather
>> either 1000 or 1100.
>> While this by itself would not be a problem, it’s very confusing that the
>> AAC decoder outputs AVFrames with a duration of 1024.
>>
>> So now I’m a bit stuck as I really don’t know what to do.
>> If the packets were all 1000 I would try discarding the last 24 samples
>> of each AVFrame but as the frame apparently only holds 1024 samples
>> although the packet says it contains 1100 samples, I’m stuck.
>> I tried sending a NULL packet in the case of longer packets hoping to get
>> the remains 76 samples out of the decoder but I got this error message:
>>
>> *[aac @ 0x10539da00] Got unexpected packet size after a partial decode*
>>
>> The file plays fine in both QuickTime and VLC so there must be a correct
>> way of handling these kind of files..
>>
>> I’d be very grateful for any insight anyone of you might have!
>>
>> Thanks and best,
>>
>> Flo
>>
>> _______________________________________________
>> Libav-user mailing list
>> Libav-user at ffmpeg.org
>> http://ffmpeg.org/mailman/listinfo/libav-user
>>
>>
> What do you mean "packet contains 1100 samples", how is that specified?
> AVPacket->duration is in time_base units, not in samples.
>
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