[FFmpeg-devel] [PATCH] HE-AAC v1 decoder try 4
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:
>>>> *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 :)
>> 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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