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

[Vladimir Pantelic]

Michael Niedermayer wrote:
> On Sun, Feb 21, 2010 at 09:36:10PM +0100, Vladimir Pantelic wrote:
>> Michael Niedermayer wrote:
>>> On Wed, Feb 17, 2010 at 07:06:44PM +0100, Vladimir Pantelic wrote:
>>>> 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
>>>> 22050.
>>> i guess unconditionally setting it to 0 in avcodec_open() is not ideal
>>>
>>>
>>>> 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...)
>>>>    ffplay.c |   33 ++++++++++++++++++++++++++++++++-
>>>>    1 file changed, 32 insertions(+), 1 deletion(-)
>>>> 8951327565cd5e449a94c431caa46767cfc84ae2  ffplay_he_aac_sample_rate.diff
>>>> Index: ffplay.c
>>>> ===================================================================
>>>> --- ffplay.c	(revision 21861)
>>>> +++ ffplay.c	(working copy)
>>>> @@ -146,6 +146,8 @@
>>>>        AVPacket audio_pkt_temp;
>>>>        AVPacket audio_pkt;
>>>>        enum SampleFormat audio_src_fmt;
>>>> +    int audio_sample_rate;
>>>> +    int audio_sample_rate_changed;
>>>>        AVAudioConvert *reformat_ctx;
>>>>
>>>>        int show_audio; /* if true, display audio samples */
>>>> @@ -1670,7 +1672,10 @@
>>>>                pkt_temp->size -= len1;
>>>>                if (data_size<= 0)
>>>>                    continue;
>>>> -
>>>> +            if (dec->sample_rate != is->audio_sample_rate) {
>>>> +                is->audio_sample_rate = dec->sample_rate;
>>>> +                is->audio_sample_rate_changed = 1;
>>>> +            }
>>>>                if (dec->sample_fmt != is->audio_src_fmt) {
>>>>                    if (is->reformat_ctx)
>>>>                        av_audio_convert_free(is->reformat_ctx);
>>>> @@ -1848,6 +1853,8 @@
>>>>            }
>>>>            is->audio_hw_buf_size = spec.size;
>>>>            is->audio_src_fmt= SAMPLE_FMT_S16;
>>>> +        is->audio_sample_rate = spec.freq;
>>>> +        is->audio_sample_rate_changed = 0;
>>>>        }
>>>>
>>>>        ic->streams[stream_index]->discard = AVDISCARD_DEFAULT;
>>>> @@ -2136,6 +2143,30 @@
>>>>                eof= 0;
>>>>            }
>>>>
>>>> +        if( is->audio_sample_rate_changed ) {
>>>> +            SDL_AudioSpec wanted_spec, spec;
>>>> +fprintf(stderr, "sample rate changed to: %d\n", is->audio_sample_rate);
>>>> +            is->audio_sample_rate_changed = 0;
>>>> +
>>>> +            SDL_CloseAudio();
>>>> +	
>>> indention is off, trailing whitespace and you maybe could use
>>>    stream_component_open()
>> that I tried 1st, but stream_component_open() drops the whole audio packet
>> queue, so the sound is totally off afterwards. But of course I could factor
>> out the SDL audio related pieces and use them here and in
>> stream_component_open()..
>
> anything is better than duplicating code (with its bugs)

right. btw, is the place I perform the change of sample rate the right one?

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