Log - e75f2cea5ff42ebd2e49342f47dccfa861eb5ba1 - platform/external/webrtc

e75f2ce Add FORCE_HTTPS_COMMIT_URL to codereview.settings. by kjellander@webrtc.org · 10 years ago
cc7755b Whitespace change by kjellander@webrtc.org · 10 years ago
74499ef Add whitespace.txt file. by kjellander@webrtc.org · 10 years ago
2c13f65 Add a platform specific typedef for SOCKET in the peerconnection_server example since it's not universally 'int'. by tommi@webrtc.org · 10 years ago
83b5200 Add framerate for complete received frames to histogram stats: by asapersson@webrtc.org · 10 years ago
cc144de Make bands vector in SplittingFilter Analysis const by aluebs@webrtc.org · 10 years ago
8789376 Move ChannelBuffer class to channel_buffer file by aluebs@webrtc.org · 10 years ago
d87213a Remove unused RtpStatistics struct. by pbos@webrtc.org · 10 years ago
7d4e6d0 Roll chromium_revision d8c9041..309cf65 by kjellander@webrtc.org · 10 years ago
d952c40 Add receive bitrates to histogram stats: by asapersson@webrtc.org · 10 years ago
3e9ad26 Refactor iOS AppRTC parsing code. by tkchin@webrtc.org · 10 years ago
79b9eba Implement 3 band splitting filter bank by upsampling and splitting twice into 2 bands by aluebs@webrtc.org · 10 years ago
7806d8f Fix an ASSERT that fires in a browser test for renegotiation. by jiayl@webrtc.org · 10 years ago
a71bb60 Revert 7750 "Don't reset sequence number for a stream on deactiv..." by sprang@webrtc.org · 10 years ago
a56a2c5 Enabling building with NEON on ARM64 by andrew@webrtc.org · 10 years ago
31f7a0e Don't reset sequence number for a stream on deactivate/reactivate. by sprang@webrtc.org · 10 years ago
91d928e Rename RtpFileReader::Packet to RtpPacket and move out of RtpFileReader by henrik.lundin@webrtc.org · 10 years ago
2faf7ee Revert "Revert "This adds an Android apk for running tests on the Java layer of PeerConnection."" by perkj@webrtc.org · 10 years ago
58edb83 Add video encoder fps and bitrate statistics to Android AppRTCDemo UI. by glaznev@webrtc.org · 10 years ago
0087318 Implement settable min/start/max bitrates in Call. by pbos@webrtc.org · 10 years ago
b951eb1 Add back EXPECT_TRUEs. by pbos@webrtc.org · 10 years ago
ba25347 Reenable GetStats test. by pbos@webrtc.org · 10 years ago
dab5d92 Use mirror image for Android AppRTCDemo local preview. by glaznev@webrtc.org · 10 years ago
03499a0 Add wav output capability to neteq_rtpplay by henrik.lundin@webrtc.org · 10 years ago
aff1751 Add new test for VP8 packetizer to test tight partitions by henrik.lundin@webrtc.org · 10 years ago
dde19a6 sync_chromium.py: Check for chromium/src by kjellander@webrtc.org · 10 years ago
3398a4a PRESUBMIT: Only notify GN changes for GYP files in webrtc/* by kjellander@webrtc.org · 10 years ago
8562f23 OWNERS: Remove tomasl@ and mallinath@ by kjellander@webrtc.org · 10 years ago
4f16c87 Simplifying VideoReceiver and JitterBuffer. by pbos@webrtc.org · 10 years ago
9334ac2 Use vector of CSRCs for DeliverFrame & SetCSRCs. by pbos@webrtc.org · 10 years ago
308e7ff Revert "This adds an Android apk for running tests on the Java layer of PeerConnection." by kjellander@webrtc.org · 10 years ago
2751f2a This adds an Android apk for running tests on the Java layer of PeerConnection. by perkj@webrtc.org · 10 years ago
88d14f4 Remove expensive and unnecessary memory alloc for sending black frames on video by thorcarpenter@google.com · 10 years ago
1153322 Build fix for MIPS Android Webview build. by andrew@webrtc.org · 10 years ago
bdcf38c cricket::VideoFrame: Refactor ConvertToRgbBuffer into base class by magjed@webrtc.org · 10 years ago
ad0e71c Update mock_frame_dropper.h to use size_t by kjellander@webrtc.org · 10 years ago
4591fbd Use size_t more consistently for packet/payload lengths. by pkasting@chromium.org · 10 years ago
edc6e57 Support loopback mode and command line execution by glaznev@webrtc.org · 10 years ago
6ff3ac1 Fix problems if first packet into NetEq is rejected by henrik.lundin@webrtc.org · 10 years ago
ed91068 Create a NetEq test for when the first incoming payload type is unknown by henrik.lundin@webrtc.org · 10 years ago
049e4ec Change default values for CpuOveruseOptions. by asapersson@webrtc.org · 10 years ago
f58b455 cricket::VideoAdapter: Drop frames before spending time converting/scaling, not after. by magjed@webrtc.org · 10 years ago
40af3a5 Revert "Add DCHECK to ensure that NetEq's packet buffer is not empty" by henrik.lundin@webrtc.org · 10 years ago
6f6ef72 Add DCHECK to ensure that NetEq's packet buffer is not empty by henrik.lundin@webrtc.org · 10 years ago
2176db3 AppRTCDemoActivity: Add a config CheckBox for enabling/disabling CPU overuse adaptation. (re-land) by henrika@webrtc.org · 10 years ago
c56814f Roll chromium_revision 91f1781..d8c9041 by kjellander@webrtc.org · 10 years ago
087da13 Add empty 3 band splitting filter API by aluebs@webrtc.org · 10 years ago
2656bf8 Fix ExpectedQueueTimeMs() to avoid truncation or overflow. by pkasting@chromium.org · 10 years ago
930e004 Add jmi field for packets discarded due to network error by guoweis@webrtc.org · 10 years ago
c72a22c Add preliminary empty file videoframefactory.cc by magjed@webrtc.org · 10 years ago
f5b56fb Annotate COMPILE_ASSERT with __attribute__((unused)). by pbos@webrtc.org · 10 years ago
4ef22d1 Setting Opus FEC as default by minyue@webrtc.org · 10 years ago
966a708 Use RtpFileSource in NetEqDecodingTest by henrik.lundin@webrtc.org · 10 years ago
4ec19e3 Revert 7707 "cricket::VideoAdapter: Drop frames before spending ..." by tommi@webrtc.org · 10 years ago
858dbbc cricket::VideoAdapter: Drop frames before spending time converting/scaling, not after. by magjed@webrtc.org · 10 years ago
6a782c2 Revert 7693 "Add jmi field for packets discarded due to network error" breaks chromium's webrtc_cases. by henrike@webrtc.org · 10 years ago
be05c74 Wrap the splitting filter in its own class by aluebs@webrtc.org · 10 years ago
67c2247 Disable EndToEnd.GetStats test. by pbos@webrtc.org · 10 years ago
a73d746 Revert 7702 "cricket::VideoAdapter: Drop frames before spending ..." by magjed@webrtc.org · 10 years ago
bbd8cad cricket::VideoAdapter: Drop frames before spending time converting/scaling, not after. by magjed@webrtc.org · 10 years ago
ece3890 Report total bitrate for all streams in GetStats. by pbos@webrtc.org · 10 years ago
35c1ace Revert 7698 "WebRtcVideoMediaChannel::SetSendParams: Don't cap r..." by magjed@webrtc.org · 10 years ago
a1f5b96 Remove unnecessary copying of libjingle resource files. by kjellander@webrtc.org · 10 years ago
52da44b WebRtcVideoMediaChannel::SetSendParams: Don't cap resolution by magjed@webrtc.org · 10 years ago
49ff40e Make SetREMBData accept vector of SSRCs. by pbos@webrtc.org · 10 years ago
a9c2d45 Fix and enable CanReceiveFec test. by pbos@webrtc.org · 10 years ago
ee30082 Set correct sample rate in far_frame in audioproc tool. by bjornv@webrtc.org · 10 years ago
52bb521 Update isolate files for Android APK tests. by kjellander@webrtc.org · 10 years ago
312614a Add jmi field for packets discarded due to network error by guoweis@webrtc.org · 10 years ago
90b9b08 Fix a platform check to use WEBRTC_WIN instead of OS_WIN. by jiayl@webrtc.org · 10 years ago
6ca6190 Fix a SCTP message reordering issue in datachannel.cc. by jiayl@webrtc.org · 10 years ago
ea73ff7 webrtc::Scaler: Preserve aspect ratio by magjed@webrtc.org · 10 years ago
0b3d89b VideoSendStreamTest.SwapsI420VideoFrames: Initialize frame memory to avoid drmemory errors by magjed@webrtc.org · 10 years ago
14ea50a Change the static_library("webrtc") to a source set in the GN build. by kjellander@webrtc.org · 10 years ago
0e37b89 replace inline assembly WebRtcAecm_CalcLinearEnergiesNeon by intrinsics. by andrew@webrtc.org · 10 years ago
e497be3 replace inline assembly WebRtcAecm_StoreAdaptiveChannelNeon by intrinsics. by andrew@webrtc.org · 10 years ago
0e71070 Use ScreenCapturer to capture the whole and clip to the window rect when the shared window is on the top. by jiayl@webrtc.org · 10 years ago
a367aea Bump to version 40 by tnakamura@webrtc.org · 10 years ago
f7c5d4f Revert 7679 "webrtc::Scaler: Preserve aspect ratio" by magjed@webrtc.org · 10 years ago
525baea Add PROJECT to codereview.settings by kjellander@webrtc.org · 10 years ago
944fb57 Roll chromium_revision 375f736..91f1781 by kjellander@webrtc.org · 10 years ago
809986b webrtc::Scaler: Preserve aspect ratio by magjed@webrtc.org · 10 years ago
cd621a8 Add thread annotations to overuse_frame_detector class. by asapersson@webrtc.org · 10 years ago
8038d42 Follow-up fixes for G722 by henrik.lundin@webrtc.org · 10 years ago
1431e4d Revert 7675 "Make an AudioEncoder subclass for iSAC" by turaj@webrtc.org · 10 years ago
05feff0 Make an AudioEncoder subclass for iSAC by kwiberg@webrtc.org · 10 years ago
33045ab Change from talk/p2p (r7664) "(Auto)update libjingle 79414100-> 79428003". by henrike@webrtc.org · 10 years ago
43e033e Change from talk/p2p (r7572): "Improve the logging when a TCP connection is deleted." by henrike@webrtc.org · 10 years ago
4ffc734 replace inline assembly WebRtcAecm_ResetAdaptiveChannelNeon by intrinsics. by andrew@webrtc.org · 10 years ago
d024f75 clear asm code and unused functions in audio processing module by andrew@webrtc.org · 10 years ago
c492231 Removes talk/xmllite, talk/xmpp and talk/p2p as they are no longer used by gyp/gn builds. by henrike@webrtc.org · 10 years ago
d819803 Wire up DSCP support in WebRtcVideoEngine2. by pbos@webrtc.org · 10 years ago
83d4804 Put send-side bwe probing under finch experiment. by stefan@webrtc.org · 10 years ago
957e802 Refactor SetDefaultEncoderConfig to work on existing codecs. by pbos@webrtc.org · 10 years ago
a5d29fc Add unit to dropped frames. by pbos@webrtc.org · 10 years ago
bd495fa .gitignore updates by kjellander@webrtc.org · 10 years ago
3c1970f (Auto)update libjingle 79414100-> 79428003 by buildbot@webrtc.org · 10 years ago
188d3b2 Enable VP9 video codec support on webrtcvideoengine behind a field trial. by andresp@webrtc.org · 10 years ago
f85dbce Reapply "Advertise G722 as 8 kHz rather than 16 kHz"" by henrik.lundin@webrtc.org · 10 years ago
d105cc8 Change dummy address to use 0.0.0.0 instead of :: by perkj@webrtc.org · 10 years ago