Re: [Osalp-dev] Application Design Clarification

[Darrick S. <da...@dc...>]

My thoughts:

1.  You'll get the audio data in buffers.  So this means the file modules 
will need to handle a stdin stream.  I could very easily do this.  Another 
option is to setup the file modules to accept a data buffer and read from 
that rather then opening a file.

2.  Output is much more simple.  You open a aflibAudioFile with the input as 
parent after setting the config to what you want.  You should be able to 
choose what format you want too.  Write it to /dev/null.  Currently we don't 
have a raw 8kHz ULAW format.  The sox module can handle it but it's not 
currently listed in the exported formats.  I'll fix that.  

so

aflibAudioFile* output = new aflibAudioFile( input, "8kHz U-LAW", 
"/dev/null", config, status);

then:

aflibData* data;

while( no_samples){
	data = output->process(status,-1,no_samples,FALSE);
	do_something_with_data( data );
}


On Monday 20 August 2001 10:52, Anthony Zawacki wrote:
> Hello,
>
> I'm seeking a bit of clarification on how to use OSALP in my application.
>
> A brief background:
> My module will receive audio data in buffers, read from an HTTP source
> (though I am not responsible for the reading.)  This data must then be
> converted to 8kHz AU-ULAW with the header stripped.  What I was planning to
> do is build an audio device that requires 8khz AU-ULAW.  It would then
> start by stripping off the header, and write the data out to a set of
> buffers.  Other portions of the application will read those buffers and do
> the actual writing to the audio device through it's API.  I thought that an
> aflibAudioMemoryInput class would do the trick, but as far as I can
> understand it, the data must already be in the "correct" format as noted in
> section 4.2.10 of the Audio Library Programmers Manual.  It's not clear to
> me what is the "correct" format.
>
> My input data could be one of:
> Raw (headerless) 8kHz 8-bit mu-law [PCM] single channel*
> Raw (headerless) 8kHz 8-bit a-law [PCM] single channel*
> WAV (RIFF header) 8kHz 8-bit mu-law [PCM] single channel*
> WAV (RIFF header) 8kHz 8-bit a-law [PCM] single channel*
> Others we specify as supported(MP3, ...)
>
> The item's marked by an (*) are required.  I need to convert these formats
> to 8kHz Raw (headerless) 8-bit mu-law [PCM] single channel, which is what
> our audio device understands.  The input data format I was hoping for is
> all of the required items, plus additional formats as supported by OSALP
> (plus any additional we may want to create supporting our audio device.)
>
> In general, I know which of the input types I will be getting from the
> content-type in the header.  The question is, after I recognize that the
> file is MP3, how do I put an MP3 decoder into the audio chain so that the
> device writer can get the correct input.
>
> If anything is not clear, let me know, and I'll try to explain better.
>
> Thanks,
> Anthony
>
>
> _______________________________________________
> Osalp-dev mailing list
> Osa...@li...
> http://lists.sourceforge.net/lists/listinfo/osalp-dev