and touched VoEBaseImpl::NeedMorePlayData and AudioCodingModuleImpl::PlayoutData10Ms(), for performance reasons in Android platforms. The two functions used about 6% of VoE originally. After the change, the percentage reduced to about 0.2%. Review URL: https://webrtc-codereview.appspot.com/379001 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1589 4adac7df-926f-26a2-2b94-8c16560cd09d
Adding libsrtp in libjingle.gyp and changing DEPS to retrieve libsrtp code from chromium repository.
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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