[FFmpeg-devel] [PATCH] HE-AAC v1 decoder try 4

Vladimir Pantelic pan
Wed Feb 17 19:06:44 CET 2010

Vladimir Pantelic wrote:
> Alexander Strange wrote:
>>  On Feb 16, 2010, at 6:19 PM, Alex Converse wrote:
>>>   On Tue, Feb 16, 2010 at 6:01 PM, Alex Converse<alex.converse at gmail.com>wrote:
>>>>   Notes:
>>>>   *All the computation time is spent in ff_sbr_apply() and it's
>>>>   children. If it isn't called from ff_sbr_apply() making it 100% faster
>>>>   isn't going to buy us anything.
>>>>   *Right now the synthesis filterbank is written on top on an MDCT. With
>>>>   appropriate SIMD functions it may make sense to move it to an FFT.
>>>>   Right now the MDCT version is much faster.
>>>>   *SIMD placeholder patch not included
>>>   Wrong patch.
>>>   try 4.1 :)
>>>   <sbr.diff>
>>  I tried it with ffplay and it plays 2x too slow. If ffplay needs to be fixed, I wonder if other players will too? Well, it's unavoidable anyway.
>>  ffmpeg to wav sounds fine.
> guess its the issue that the container signals the smaller (non-sbr) sample rate, e.g. 24khz and the decoder applies SBR
> and returns 48kHz. I had to fix my player to switch to the decoder returned sample rate too...

ok, it is more the fact that avctx->sample_rate is 0
after we perform avcodec_open(), so SDL defaults to

it is correct before the avcodec_open() call, but that
might not always be the case I guess.

totally lame patch attached to detects any change in
sample_rate by the decoder and re-open SDL audio...

(my first patch (yay), more for RFC than an attempt to
get instant fame...)
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