Log - da9b40462d756896bcd942406d1f800af694328e - fp2-dev/platform/external/chromium_org/third_party/webrtc

60f1422 Move timestamp_extrapolator and rtp_to_ntp to system wrapper so that it can be shared by both audio and video engine. by wu@webrtc.org · 10 years ago
5e44f56 * Add 100ms network delay to test CaptureNtpTimeWithNetworkJitter. by wu@webrtc.org · 10 years ago
c54ff69 Add thread annotations to Call API. by pbos@webrtc.org · 10 years ago
267637b Disable flaky CaptureNtpTimeWithNetworkJitter. by pbos@webrtc.org · 10 years ago
c298835 Disabling flaky CanReceiveFec. by pbos@webrtc.org · 10 years ago
b991cd0 Calculate local/remote clock delta and capture ntp timestamp in receiver's timebase. by wu@webrtc.org · 10 years ago
dbebc39 Remove TraceCallback use from Call. by pbos@webrtc.org · 10 years ago
9d0f79f Rename Start/Stop in Video{Send,Receive}Streams. by pbos@webrtc.org · 10 years ago
69b14d5 Let A/V sync test use default AudioCoding module by henrik.lundin@webrtc.org · 10 years ago
6b1114a Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition. by fischman@webrtc.org · 10 years ago
19ca463 Disable UsesTraceCallback by pbos@webrtc.org · 10 years ago
d8b4d0f Implement FEC support in VideoReceiveStream. by pbos@webrtc.org · 10 years ago
4b0cd7f Convert logs in rtp rtcp module from WEBRTC_TRACE into LOG. by andresp@webrtc.org · 10 years ago
1982636 Clean up traces and logs in RemoteBitrateEstimator. by stefan@webrtc.org · 10 years ago
6c57efd Re-submit: rev5775 by andresp@webrtc.org · 10 years ago
765ea72 Revert 5775 "Modify bitrate controller to update bitrate based o..." by andrew@webrtc.org · 10 years ago
0b11715 Change sprintf format string from %zu to %i by henrik.lundin@webrtc.org · 10 years ago
539bbde Modify bitrate controller to update bitrate based on process call and not by andresp@webrtc.org · 10 years ago
5f804f8 VoE changes to allow forwarding of packets from VoE to ViE BWE. by solenberg@webrtc.org · 10 years ago
9b2b8ec Add AIMD option to BWE API. by stefan@webrtc.org · 10 years ago
a183edc Extend perf tests to perform rampup on single stream. by andresp@webrtc.org · 10 years ago
292e7f6 Disabling SendsSetSimulcastSsrcs. by pbos@webrtc.org · 10 years ago
16c3dcc Disable flaky CanSwitchToUseAllSsrcs. by pbos@webrtc.org · 10 years ago
bef6e62 Simplify pacer interface. by pbos@webrtc.org · 10 years ago
f39df52 Remove internal codecs from VideoSendStream. by pbos@webrtc.org · 10 years ago
ebae8bb Re-comitting r5711: "Fixing a flaky test in video_engine_tests" by henrik.lundin@webrtc.org · 10 years ago
8d3c410 Revert 5711 "Fixing a flaky test in video_engine_tests" by turaj@webrtc.org · 10 years ago
f9a6ab0 Fixing a flaky test in video_engine_tests by henrik.lundin@webrtc.org · 10 years ago
ca626eb Refactor rampup tests: by andresp@webrtc.org · 10 years ago
3c00b1c Stopping network threads before tearing down test by henrik.lundin@webrtc.org · 10 years ago
15cf717 Re-landing "Routing SuspendChange to VideoSendStream::Stats" by henrik.lundin@webrtc.org · 10 years ago
9420a1f Implement minimum transmit bitrate. by pbos@webrtc.org · 10 years ago
41da329 Enable all RampUpTest.UpDownUp* tests by henrik.lundin@webrtc.org · 10 years ago
c53e587 Replace labs with std::abs. by pbos@webrtc.org · 10 years ago
af634a2 Remove platform-specific code from new-API tests. by pbos@webrtc.org · 10 years ago
f951dfc Revert "Routing SuspendChange to VideoSendStream::Stats" by henrik.lundin@webrtc.org · 10 years ago
697cd78 Routing SuspendChange to VideoSendStream::Stats by henrik.lundin@webrtc.org · 10 years ago
f422ce1 Adding a link to issue by henrik.lundin@webrtc.org · 10 years ago
b5c0d2e NetEq4: Changing the behavior of playout_timestamp_ update by henrik.lundin@webrtc.org · 10 years ago
4368a8f Potential deadlock in VideoSendStreamTest::ProducesStats by sprang@webrtc.org · 10 years ago
c63f18d Use DISABLE_ instead of commenting out tests by henrik.lundin@webrtc.org · 10 years ago
0bf5a2f Adding a new ramp-up-down-up test by henrik.lundin@webrtc.org · 10 years ago
deb5d53 Fix compilation errors under clang 3.5. by pbos@webrtc.org · 10 years ago
3f3e951 Incorrect overhead calculation when using FEC + RTP extension headers. by sprang@webrtc.org · 10 years ago
f2c28a0 Make VideoReceiveStream::GetStats() const. by sprang@webrtc.org · 10 years ago
fa7c4c4 Wire up statistics in video receive stream of new API by sprang@webrtc.org · 10 years ago
4b1817f Add configuration for cpu overuse detection to video send stream. by asapersson@webrtc.org · 10 years ago
fdb30d1 Fix race when deleting video receive streams in Call. by solenberg@webrtc.org · 11 years ago
48ac0da Drop early packets when not sending in TransportAdapter. by sprang@webrtc.org · 11 years ago
ffd4269 Always initialize Trace in Call TraceDispatcher. by pbos@webrtc.org · 11 years ago
c766775 Wire up RTX in VideoReceiveStream. by pbos@webrtc.org · 11 years ago
bb55b6d Set NACKed packet to -1 in TestNackRetransmission. by pbos@webrtc.org · 11 years ago
64339f0 Add configuration and test for extended RTCP reference time reports to new video api. by asapersson@webrtc.org · 11 years ago
df9f099 Permitting double start/stopping of streams. by pbos@webrtc.org · 11 years ago
c92ae91 Add another test case for AST/TOF switching. by solenberg@webrtc.org · 11 years ago
ca72300 Wire up statistics in video send stream of new video engine api by sprang@webrtc.org · 11 years ago
acc2e43 Add callbacks for receive channel RTCP statistics. by sprang@webrtc.org · 11 years ago
cd117d2 Integrate fake_network_pipe into direct_transport. by stefan@webrtc.org · 11 years ago
2a4595a cpplint cleaning new API and its implementation files. by mflodman@webrtc.org · 11 years ago
b409d78 Reduced execution time for CallTest::ReceivesPliAndRecovers, by dropping only one packet and made it predictable by removing rand(). by mflodman@webrtc.org · 11 years ago
f22f12a Speeding up CallTest.ReceivesAndRetransmitsNack and removed the random packet loss. by mflodman@webrtc.org · 11 years ago
4db3691 Stop transport in test SuspendBelowMinBitrate. by pbos@webrtc.org · 11 years ago
620d9e5 Modify video_render/ to allow a single old frame. by pbos@webrtc.org · 11 years ago
4494516 Delete capturers after destroying streams in test. by pbos@webrtc.org · 11 years ago
39139dc Revert r5294 to re-roll r5293. by pbos@webrtc.org · 11 years ago
0af1d21 Revert 5293 "Auto instantiate RBE depending on whether AST or TO..." by turaj@webrtc.org · 11 years ago
ee867fa Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream. by solenberg@webrtc.org · 11 years ago
ab6ccbc Adding REMB to receive stream configuration, the send side will always by mflodman@webrtc.org · 11 years ago
d1dd1d2 Move realtime tests to webrtc_perf_tests. by pbos@webrtc.org · 11 years ago
e4d538a Make sure channels in the same call are in the same channel group. by mflodman@webrtc.org · 11 years ago
e6dc4ff Making RemoteRateControl::min_configured_bit_rate_ configurable by henrik.lundin@webrtc.org · 11 years ago
3a4fc4b Update talk to 58127566 together with by wu@webrtc.org · 11 years ago
9b3d2bf Revert 5274 "Update talk to 58113193 together with https://webrt..." by wu@webrtc.org · 11 years ago
cde78d6 Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/. by wu@webrtc.org · 11 years ago
7123a80 Add SwapFrame() to VideoSendStreamInput. by pbos@webrtc.org · 11 years ago
090f37f Improve VideoSendStreamTest::MaxPacketSize by sprang@webrtc.org · 11 years ago
e4d591a Revert 5229 "Make VideoSendStreamTest::MaxPacketSize test a whol..." by andrew@webrtc.org · 11 years ago
9e40eba Stop video capturers in multi-stream test. by pbos@webrtc.org · 11 years ago
6dccf13 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
f0d9b20 Remove CallTest dependency on voice_engine/test/. by pbos@webrtc.org · 11 years ago
3d70641 Move implementation files out of the webrtc/ root. by pbos@webrtc.org · 11 years ago
da3ae7c Adds support for sending redundant payloads over RTX. by stefan@webrtc.org · 11 years ago
309b2c8 Set local SSRC for VideoReceiveStream. by pbos@webrtc.org · 11 years ago
169a27a Disable check for all sent SSRCs being valid. by pbos@webrtc.org · 11 years ago
9105cbd Set up SSRCs correctly after switching codec. by pbos@webrtc.org · 11 years ago
4a9843f Implement and test EncodedImageCallback in new ViE API. by sprang@webrtc.org · 11 years ago
586becf Add test for automatically disabling padding when no video is being captured. by stefan@webrtc.org · 11 years ago
d7d60c8 Connect pacer/padding to SuspendBelowMinBitrate by henrik.lundin@webrtc.org · 11 years ago
f8486d0 Rename DestroyStream methods to include Video. by pbos@webrtc.org · 11 years ago
04bcc9d Deliver I420VideoFrames from VideoRender module. by pbos@webrtc.org · 11 years ago
f3b4602 Rename newapi::Transport::SendRTP()->SendRtp(). by pbos@webrtc.org · 11 years ago
60108c2 Rename RTP-extension constants. by pbos@webrtc.org · 11 years ago
48cc9dc Rename video streams' start/stop methods. by pbos@webrtc.org · 11 years ago
162021c Rename Call::Create{Receive,Send}Stream(). by pbos@webrtc.org · 11 years ago
8fdf191 Rename AutoMute to SuspendBelowMinBitrate by henrik.lundin@webrtc.org · 11 years ago
26a736f Hook up audio/video sync to Call. by stefan@webrtc.org · 11 years ago
5eca0c7 Fix breakage after introducing new test. by stefan@webrtc.org · 11 years ago
f8c47a1 Improve Call tests for RTX. by stefan@webrtc.org · 11 years ago
8f2997c Implement VideoSendStream::SetCodec(). by pbos@webrtc.org · 11 years ago
9a1635a Make video/ only depend on video_engine_core. by pbos@webrtc.org · 11 years ago