[FFmpeg-devel] ALAC encoder is not bitperfect

Baptiste Coudurier baptiste.coudurier
Tue Apr 14 01:02:19 CEST 2009

On 4/13/2009 1:59 PM, Baptiste Coudurier wrote:
> On 4/13/2009 1:50 PM, Justin Ruggles wrote:
>> Jai Menon wrote:
>>> On 4/13/09, Brent Huisman <brenthuisman at gmail.com> wrote:
>>>> Hey Jai,
>>>>  I've used several different builds, including the new ffmpeg 0.5. Any
>>>>  and all versions I tried exhibit this behaviour. Also any and all
>>>>  source wave files I use have this. Have you tried bitcomparing
>>>>  yourself?
>>> Yes, and the output is bitexact as far as I have seen. Please specify
>>> what exactly you are using to bitcompare. If its foobar2k bitcompare,
>>> then sadly thats a bug in fb2k's mp4 demuxer and should be reported to
>>> them. I think Justin posted something in this regard to HydrogenAudio.
>>> FFmpeg'a alac encoder writes out the no. of samples in every frame
>>> correctly which fb2k discards and instead pads the frame with zeroes.
>>> Try checking the bitexactness using itunes or ffalac.
>> The specific issue seems to be a combination of our mp4 muxer and how
>> fb2k handles it, but ALAC decoders other than fb2k read the actual ALAC
>> frames to determine the number of samples, while fb2k uses info from the
>> mp4 container.
>> Here is a quote on Hydrogenaudio from a user named Gregory S. Chudov:
>> ** start quote **
>> There are two places in mp4 container, where the length is stored.
>> First place is in moov.mvhd chunk (movie header).
>> iTunes encoder writes the approximate number of samples there.
>> ffmpeg encoder writes the approximate length in milliseconds.
>> This is not very reliable field and is ignored by fb2k.
> Well mvhd is scaled according to global timescale which is 1000, and set
> accordingly.
> I guess sample rate could be used if the file has only one audio track.
> But in any case, track time scale is sample rate and duration is in
> samples number.
>> Second place is moov.trak.mdia.minf.stbl.stts (sample table).
>> This is where iTunes encoder stores the correct length. This is what
>> fb2k uses.
>> This table contains array of struct { int sample_count; int
>> sample_duration }
>> Total length is a sum of sample_count*sample_duration.
>> Normally for iTunes-encoded file this table contains two entries.
>> First entry with sample_duration=4096 and sample_count=total_samples/4096
>> Second entry with sample_duration=total_samples%4096 and sample_count=1
>> For ffmpeg, this table sadly contains only one entry, so the total
>> sample length is rounded up to a multiple of 4096.
>> ** end quote **
> Is alac frame size different for the last sample ? If not, then it is
> _wrong_ to set it differently.

Well, it seems indeed alac last frame size is smaller, so in this case
pkt->duration must be set differently and this will be taken into
account by muxer.

Patch attached.

Btw can somebody explain why avctx->frame_size is changed back to its
old value ? Because that's what caused the problem, if frame_size was
kept to the new size, libavformat would have computed the duration

But of course setting pkt->duration is the correct solution.

Baptiste COUDURIER                              GnuPG Key Id: 0x5C1ABAAA
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FFmpeg maintainer                                  http://www.ffmpeg.org
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