This also is long, but simple.
First, we switch to floats everywhere. This allows to work with any rate, is
more flexible with channel layout, and is a stable API (see audio_processing.h
in webrtc.org).
Then, 10ms worth of audio (already at the graph rate) are poped from the
lock-free queue (fed on the other end by the MSG mixer), and does the following:
- Down mixing to stereo (if needed)
- De-interleaving into planar buffer
- Prepare input and output config
- Actually make the API call
- Free the data
Now, first, we should use a ring buffer, and not have to free any data. Then we
also should not use a lock-free queue, and synchronously process the
reverse-stream, but this is enough code already.
Then, the actual mic data processing:
- Pop a packet from the packetizer (that gives us 10ms worth of audio, note that
we switch from int16_t to float, i.e. we don't do this conversion anymore).
- We convert to planar buffers, deinterleaving
- Prepare input and output config
- Allocate a SharedBuffer of the right size
- Process the data with the processing algorithm selected in UpdateSingleSource
- Append to the a MediaSegment, and append to the right MediaStreamTrack for the
correct SourceMediaStream (the data is already planar and all well).
MozReview-Commit-ID: 2IjgHP0GAmw
--HG--
extra : rebase_source : 1e08c4a781db8778e0532f9ef1a8e369513a2c66
extra : source : 0107b3feb84bbe0e643f505ec58e303dfd94e1a7
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