[linux-audio-dev] Beginner question about ring buffer and block processing

[David Cournapeau]

Jack O'Quin wrote:

>On 5/23/05, David Cournapeau <cournape at gmail.com> wrote:
>  
>
>>    I tried to read the source of jamin, which implements this kind of
>>scheme in io.c, but I couldn't manage to get the whole thing.
>>    
>>
>
>It's all in that one file.  What part did you not understand?
>  
>
There are 2 things which make it a bit difficult to read for me:

    - the first is all the thread synchronisation stuff (I intend to 
look at that problem later),
    - the 2 threads thing : dsp thread, jack thread (the "more" RT one).

My main problem is the 2 threads thing. If I understand correctly, the 
way it works is

     1) io_init starts it all (from the point of view of module io): it 
parses some cmd line options, register the callbacks to jack in 'normal 
mode' (ie no dummy mode), and calls process_init. My first problem: is 
dsp_block_size the *fixed* size of your actual dsp algorithm ? It looks 
like it, but I am not sure as it is defined in an other module.

    2) io_activate is the 2d function of io module called by main, which 
register jack ports, and creates a DSP thread is necessary

    3) If no dsp thread is created, once io_activate returns, jack may 
begin to call io_process.

io_process is the function where the buffering which I am interested in 
happens, right ? My problem is that I don't understand the DSP thread 
thing, and I am a bit confused. Can I just consider the case where no 
dsp thread is created to understand the buffering issue ? What is the 
dsp thread for exactly ?

Sorry if I sound stupid, but my experience in C is mainly in number 
crunching modules for matlab/octave extensions, and all this real time 
thing is really new to me :) Big thank you for writing a well documented 
code, by the way !

Cheers,

David