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(and then removed the talk/ impact)
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5903 4adac7df-926f-26a2-2b94-8c16560cd09d
Instead have ProcessStream transparently handle changes to the stream
audio parameters (sample rate and channels). This removes two locks
per 10 ms ProcessStream call taken by VoiceEngine (four total with the
audio level indicator.)
Also, prepare future improvements by having the splitting filter take
a length parameter. This will allow it to work at different sample
rates. Remove the useless splitting_filter wrapper.
TESTED=voe_cmd_test with audio processing enabled and switching between
codecs; unit tests.
R=aluebs@webrtc.org, bjornv@webrtc.org, turaj@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5346 4adac7df-926f-26a2-2b94-8c16560cd09d
Test:
-manual test with voe_cmd_test.
-manual test with RTPEncode & NetEqRTPPlay.
-manual test with simpleKenny.
-Bit-exact test of iSAC-swb and iSAC-wb with head revision of trunk. The bit-exactness is confirmed on all files generated by running webrtc/modules/audio_coding/codecs/isac/main/test/QA/runiSACLongtest.txt
Review URL: https://webrtc-codereview.appspot.com/937025
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3226 4adac7df-926f-26a2-2b94-8c16560cd09d
