[FFmpeg-devel] ALAC encoder is not bitperfect

[Baptiste Coudurier]

On 4/13/2009 4:09 PM, Michael Niedermayer wrote:
> On Mon, Apr 13, 2009 at 04:02:19PM -0700, Baptiste Coudurier wrote:
>> 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.
> 
> probably ok
> 

Applied. Btw, it seems alac does not have regressions tests ?
It would be useful to have a alac in .m4a test I think.

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