So in the HandleStreamFormatChange() callback, we need to re-initiate the playout as same as what we do in InitPlayout(). Here we merely copy those codes out from InitPlayout() into a new SetDesiredPlayoutFormat() function for the invoking from the two places. Previously, HandleStreamFormatChange only re-creates the AudioConverter, which is not enough. We also need to reset the buffer size and refresh the latency. BUG=4240 TEST=Manual Test R=andrew@webrtc.org, henrika@webrtc.org Review URL: https://webrtc-codereview.appspot.com/36029004 Cr-Commit-Position: refs/heads/master@{#8815} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8815 4adac7df-926f-26a2-2b94-8c16560cd09d
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
Languages
C++
90.3%
Java
2.9%
C
2.2%
Objective-C++
2%
Python
1.3%
Other
1%