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hbos authored
Changes: https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+log/05fb319..4b544de $ git log 05fb319..4b544de --date=short --no-merges --format=%ad %ae %s 2017-04-07 alessiob@webrtc.org MultiEndCall::CheckTiming() verifies that a set of audio tracks and timing information is valid to simulate conversational speech. Unordered turns are rejected. Self cross-talk and cross-talk with 3 or more speakers are not permitted since it would require mixing at the simulation step. 2017-04-07 zhihuang@webrtc.org Added the GetSources() to the RtpReceiverInterface and implemented it for the AudioRtpReceiver. TBR= CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_archive_rel_ng;master.tryserver.chromium.mac:mac_chromium_archive_rel_ng BUG= Review-Url: https://codereview.chromium.org/2804353002 Cr-Commit-Position: refs/heads/master@{#463005}
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