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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / 8ad10ec40b643e279d614d4f14d2b4d9accf863b / test / webrtc_test_common.gyp
  1. aacdb9f Integrate fake_network_pipe into direct_transport. by stefan@webrtc.org · 11 years ago
  2. c33d37c Add SwapFrame() to VideoSendStreamInput. by pbos@webrtc.org · 11 years ago
  3. b589c65 Improve VideoSendStreamTest::MaxPacketSize by sprang@webrtc.org · 11 years ago
  4. e388f9e Revert 5229 "Make VideoSendStreamTest::MaxPacketSize test a whol..." by andrew@webrtc.org · 11 years ago
  5. 532b8f7 Make VideoSendStreamTest::MaxPacketSize test a whole range of frame sizes, to make sure all corner cases are covered. by sprang@webrtc.org · 11 years ago
  6. 47f0c41 Adds support for sending redundant payloads over RTX. by stefan@webrtc.org · 11 years ago
  7. 4b50db1 Set local SSRC for VideoReceiveStream. by pbos@webrtc.org · 11 years ago
  8. e028410 Hook up audio/video sync to Call. by stefan@webrtc.org · 11 years ago
  9. 24e2089 Separate Call API/build files from video_engine/. by pbos@webrtc.org · 11 years ago