[FFmpeg-soc] libavfilter audio work - qualification task

[S.N. Hemanth Meenakshisundaram]

On 04/05/2010 02:43 AM, Stefano Sabatini wrote:
> On date Thursday 2010-04-01 11:53:42 -0700, Baptiste Coudurier encoded:
>    
>> On 03/31/2010 10:44 PM, S.N. Hemanth Meenakshisundaram wrote:
>>      
>>> Hi All,
>>>
>>> Based on the posts above, here's what I was thinking of doing as a
>>> qualification task :
>>>
>>> 1. Fix the configure_filters function in ffmpeg.c to work for both video
>>> and audio filters.
>>>
>>> 2. Bring the af_null and af_vol filters from soc/afilters to
>>> soc/avfilter and generalize avfilter framework to work with the af_*
>>> filters. This should involve a minimal design for the avfilter audio
>>> changes.
>>>
>>> 3. af_vol currently just scales input by two. Modify af_vol to have some
>>> basic volume filter functionality.
>>>
>>> Please let me know if I have understood the tasks right and if this will
>>> work as a qualification task.
>>>        
> I believe that is even too burden as a qualification task, much of the
> libavfilter audio design has yet to be done.
>
>    
>> A qualification task could be to add AVCodecContext get_buffer
>> functionality to audio decoders.
>>      
> As alternative to this, any video libavfilter task should be OK
> (e.g. port a filter from MPlayer/libmpcodecs or write a new one), note
> that vf_imlib.c is *not* a valid qualification task (see
> https://roundup.ffmpeg.org/issue302), this should help you to get
> confident with the FFmpeg/libavfilter API.
>
> I especially recommend: a deinterlace filter (check the code in
> libavcodec/imgconvert.c), a libfreetype wrapper (check as reference
> vhook/drawtext.c in ffmpeg-0.5).
>
> Ask here or on IRC if you have questions.
>
>    
Hi,

For the deinterlace filter, do we want just a vf_* wrapper for 
imgconvert functions or should the code be ported to the vf_deinterlace 
filter?

I have been working on getting the audio filters from soc/afilters to 
work more like the current video filters and wanted to check that am on 
the right track :

1. Client will use a 'asrc_buffer_add_samples' function to give buffers 
to the source input filter or define an input filter that can fetch 
audio data.
2. Client then calls request_buffer on output filter, request_buffers 
propagate all the way to source (input filter).
3. The buffer is then passed along the filter chain and processed.

Also have a couple of questions :

1. Should the formats for audio be from the SampleFormat enum 
(libavcodec/avcodec.h) or the PCM formats in the CodecID enum. The old 
audio filter seems to be wrongly using both in two different places.
2. While video filters work a single frame at a time, audio would need 
to work on a buffer. Is it ok to use a single call similar to the 
current 'filter_buffer' to get this done instead of the start_frame, 
draw_slice, end_frame sequence in video?

I am planning to do the deinterlace, libfreetype first and then continue 
with this.

Thanks,
Hemanth