[gst-devel] More fun with homemade src's

[Iain *]

Hi

Still working on slowly porting Marlin to 0.10, I'll get there
eventually. I've been writing my channel src over the last while and
think I've got it working more or less, but it'll link to fakesink ok,
but I cannot for the life of me get it to play after its been linked
to alsasink.

Basically pipeline is

channel-src -----> audioconvert -------> alsasink

The channel-src outputs audio/x-raw-float, channels = 1, rate = 16000
The link between audioconvert and alsasink is filtered to audio/x-raw-int
It all links ok, but when set playing audioconvert complains that
negotiation never happened.

Error: Internal data flow error.
Debug: gstbasesrc.c(1569): gst_base_src_loop (): /test-src/src:
streaming task paused, reason not-negotiated (-4)

There's a debug log available at
http://www.openedhand.com/~iain/marlin-channel-src.log.gz

And attached are the source files that I'm using to test
marlin-channel-src.c and test-sample.c with test_src being the
function used to create the pipeline.

Any hints would be welcome.
thanks
iain
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