> > As I said before, if I set them all to AUX, no input channels on my test
> > file are mixed to output.
>
> Yes, because the channel numbers are different and gavl doesn't know (yet)
> how to handle them.
>
> What I thought would make sense is to check if the input format
> has *only* GAVL_CHID_AUX and then initialize the matrix with:
>
> for(i = 0; i < out_format->num_channels; i++)
> {
> for(j = 0; j < in_format->num_channels; j++)
> {
> matrix[i][j] = (i == j) ? 1.0 : 0.0;
> }
> }
>
> If I didn't make a mistake, this should cover the cases you describe.
This looks correct to me and looks like the best way of handling a
default configuration. It is especially useful if you can assign the
mix matrix like you suggested before.
Where could I start to work on this? I'm a little blinded by all the
channel_id's .
> > Would it make sense to have a GAVL_CHID_PASS, so that channel input X is
> > mapped to channel output X?
>
> No, because the channel_ids can be different for input and output.
How does that happen if you don't assign an input or output channel_id?
Why does gavl try to guess this? It seems weird to me that a normal 8
or 16 channel wav file would have channel_id's.
I guess what is confusing me is all the LEFT, RIGHT, FRONT_CENTER, AUX,
etc. business. For someone who works with multichannel audio, these
things don't mean anything. It only means something if you have a home
entertainment system where the direction of the viewer/listener is
assumed to be singular and the speakers are adjusted accordingly. I
find AUX especially confusing.
Am I correct in assuming that MOST audio editors/writers will assign
these channel_ids wrong or at least arbitrarily? I mean, how do you
know if you want a specific channel to be left, right or center unless
it is for a specific speaker configuration?
Is there anywhere I can read up on this?
And, how does interleaving fit in? Right now, I am getting segfaults
and can't see exactly where (it is different with different audio
files)
For my puredata external, where the input can vary, but output format is
fixed I now have three audio frames: input, tmp, and output.
The input frame is created for each open file and can vary in all
aspects: channel, samplerate, bitdepth, interleaving, etc.
The output is set once to the desired out format with a fixed number of
chans, samplerate, etc.
The tmp frame is used as an in-between. How should I set the
interleaving and number of channels on this frame so that I can use it
to translate input to output ? Should it match the input on
interleaving, but the output in channels?
thanks -august.
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