The current use of rtc::FifoBuffer can lead to reading across DTLS packet boundaries which could cause packets to not being processed correctly. This CL introduces the new class rtc::BufferQueue and changes the StreamInterfaceChannel to use it instead of the rtc::FifoBuffer. BUG=chromium:447431 R=juberti@google.com Review URL: https://webrtc-codereview.appspot.com/52509004 Cr-Commit-Position: refs/heads/master@{#9254}
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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