- 53304e8 Merge webrtc_utility_unittests into modules_unittests. by kjellander@webrtc.org · 11 years ago
- 266fc69 Restore relative include paths to libyuv. by andrew@webrtc.org · 11 years ago
- 9b82368 Issue 1847, memcopy is wrong and unnecessary, it is sufficient to store the pointer before clearing the instance, and write back the pointer. by turaj@webrtc.org · 11 years ago
- ae05178 resolve b9050210. Avoid pushing sync packet before any packet received. Do not turn on AV-sync if initial delay is zero. by turaj@webrtc.org · 11 years ago
- 6c82a7e Move screen capturers from chromium to webrtc. by sergeyu@chromium.org · 11 years ago
- 12bce3b Refactor padding and rtp header functionality. by stefan@webrtc.org · 11 years ago
- d8ecee5 Update the remote bitrate estimator before passing the packet to the RTP module. by stefan@webrtc.org · 11 years ago
- e54928f Remove XvRenderer. by pbos@webrtc.org · 11 years ago
- 751253d Fix build error introduced with r4168. by stefan@webrtc.org · 11 years ago
- 695ff2a Add support for padding in pacer. by stefan@webrtc.org · 11 years ago
- 026d1ce Include files from webrtc/.. paths in common_video/ by pbos@webrtc.org · 11 years ago
- cff5c03 Include files from webrtc/.. paths in tools/ by pbos@webrtc.org · 11 years ago
- 1cc4ed7 Disable neteq_unittests on Win x64 in code. by kjellander@webrtc.org · 11 years ago
- ec5caf3 Disable audio_decoder_unittests on Win x64 in code. by kjellander@webrtc.org · 11 years ago
- f4fc8ba Disable audio_coding_unittests on Win x64 in code. by kjellander@webrtc.org · 11 years ago
- 3b35ec6 Do not hold a lock when calling VCMReceiveCallback::FrameToRender. by fischman@webrtc.org · 11 years ago
- aec1bc8 Optimized DesktopRegion implementation. by sergeyu@chromium.org · 11 years ago
- ad9ee0d Removed unused class members to enable clang=1 android build. by fischman@webrtc.org · 11 years ago
- 0016110 Setting SSRC in vie_loopback_test by mikhal@webrtc.org · 11 years ago
- 915ca75 Fix error in mixing test for supported sample rates. by andrew@webrtc.org · 11 years ago
- a80d94b Change SetRTPAudioLevelIndicationStatus to ignore the id in the case of disabling. by wu@webrtc.org · 11 years ago
- 92bfbbd Replace the old resampler with SincResampler in the voice engine signal path. by andrew@webrtc.org · 11 years ago
- 379dce7 Remove ancient and unused CNG test. by andrew@webrtc.org · 11 years ago
- 2753b76 Add dummy audio NACK APIs by niklas.enbom@webrtc.org · 11 years ago
- 40954f0 Prevent excessive logging in jitter buffer by hclam@chromium.org · 11 years ago
- 8bf7456 Revert 4104 "Refactor jitter buffer to use separate lists for de..." by tnakamura@webrtc.org · 11 years ago
- 884ff69 Revert 4127 "Switch frame list implementation to std::map." by tnakamura@webrtc.org · 11 years ago
- 1aa406c MIPS optimizations for the following functions: by andrew@webrtc.org · 11 years ago
- e477574 VCM/Timing: Setting clear names to members & methods by mikhal@webrtc.org · 11 years ago
- 9f62516 Fixes the frameRate stats by grouping the frames by timestamp. by jiayl@webrtc.org · 11 years ago
- e3e4615 Use int for FPS instead of size_t. by pbos@webrtc.org · 11 years ago
- cbd78ae Include files from webrtc/.. paths in rtp_rtcp/ by pbos@webrtc.org · 11 years ago
- 54b6ebc Correctly set SSRCs for extra send RTP modules. by stefan@webrtc.org · 11 years ago
- 23e3f44 Remove assert for aborting FrameGeneratorCapturer. by pbos@webrtc.org · 11 years ago
- c1506a2 Fake VideoCapturer based on FrameGenerator by pbos@webrtc.org · 11 years ago
- 4e5f983 Fix a return value mismatch introduced in r4129. by stefan@webrtc.org · 11 years ago
- 50a4d9f Remove #pragma once by pbos@webrtc.org · 11 years ago
- 6696fba Breaking out RTP header parsing from the RTP module. by stefan@webrtc.org · 11 years ago
- 4988d94 Break video_engine/new_include/common.h into smaller parts. by pbos@webrtc.org · 11 years ago
- afe587e Switch frame list implementation to std::map. by stefan@webrtc.org · 11 years ago
- 5221d1c Rename voice_engine_core -> voice_engine and move targets to voice_engine.gyp. by andrew@webrtc.org · 11 years ago
- 2dcf742 Add comment about test_packet_masks_metrics. by marpan@webrtc.org · 11 years ago
- 3990df2 Updated WebRTC version to 3.32 TBR=wu@webrtc.org by elham@webrtc.org · 11 years ago
- e3b52e6 Don't return an estimated receive BW for channels not receiving video. by mflodman@webrtc.org · 11 years ago
- bb6bef5 Include gflags with "gflags/gflags.h" instead of <> by pbos@webrtc.org · 11 years ago
- 8838f68 Include "gtest/gtest.h", not by full path, on WEBRTC_ANDROID_PLATFORM_BUILD by pbos@webrtc.org · 11 years ago
- 39784c4 Improve vie_autotest_rtp_rtcp by reenabling important tests and reducing flakiness. by stefan@webrtc.org · 11 years ago
- 0c836bf Include files from webrtc/.. paths in audio_conference_mixer/ by pbos@webrtc.org · 11 years ago
- 9fb1613 Include files from webrtc/.. paths in audio_processing/ by pbos@webrtc.org · 11 years ago
- b2d1a40 Default constructors for new VideoEngine structs. by pbos@webrtc.org · 11 years ago
- 5437a2c Remove libvpx_intrinsics_sse4_1.a in Android.mk since this target is no longer generated in libvpx by fischman@webrtc.org · 11 years ago
- f40e9b6 - Created RemoteBitrateEstimator wrapper for use internally in (ViE) ChannelGroup. by solenberg@webrtc.org · 11 years ago
- a93cbbf Adding Mac test renderer, some test refactoring and made cpplint pass. by mflodman@webrtc.org · 11 years ago
- c6d6fed Include files from webrtc/.. paths in system_wrappers/ by pbos@webrtc.org · 11 years ago
- 96001c8 Include files from webrtc/.. paths in test/channel_transport/ by pbos@webrtc.org · 11 years ago
- c4e10b8 Include files from webrtc/.. paths in video_processing/ by pbos@webrtc.org · 11 years ago
- d5d709e Include files from webrtc/.. paths in remote_bitrate_estimator/ by pbos@webrtc.org · 11 years ago
- f24ac59 Include files from webrtc/.. paths in common_audio/ by pbos@webrtc.org · 11 years ago
- 460e172 Disabling a flaky expectation in vie_autotest_rtp_rtcp.cc. by stefan@webrtc.org · 11 years ago
- 20a5c46 Include files from webrtc/.. paths in test/ by pbos@webrtc.org · 11 years ago
- bb771bb Refactor jitter buffer to use separate lists for decodable and incomplete frames. by stefan@webrtc.org · 11 years ago
- 60142de Fix bugs in DesktopRegion::IntersectWith() and DesktopRect::IntersectWith(). by sergeyu@chromium.org · 11 years ago
- 294b789 Remove dead testRateControl.cc by pbos@webrtc.org · 11 years ago
- 68c4886 Removed dead testH263Parser.cc by pbos@webrtc.org · 11 years ago
- aef3e5a Remove dead bitstreamTest.cc. by pbos@webrtc.org · 11 years ago
- ad6cade Make sure GlxRenderer frees its resources. by pbos@webrtc.org · 11 years ago
- eef4fd5 Adds integration test for RTX and fixes bugs found. by stefan@webrtc.org · 11 years ago
- 8f1d1a9 Fix regression where retransmission bitrate is no longer estimated. by stefan@webrtc.org · 11 years ago
- 42e1fe1 CreateEmptyFrame casts from size_t to int. by pbos@webrtc.org · 11 years ago
- 08f3ca9 FrameGenerator class for future fake capture device. by pbos@webrtc.org · 11 years ago
- 074eb20 Control new VideoEngine tests with gflags. by pbos@webrtc.org · 11 years ago
- b59962f Adds print out of incoming resolution. by henrike@webrtc.org · 11 years ago
- 2e37985 Log the type of recycled frames. by stefan@webrtc.org · 11 years ago
- 9de67da Log a message when a key frame packet is received by hclam@chromium.org · 11 years ago
- de0b5fa Fix failing tests on 32 bit Linux. by solenberg@webrtc.org · 11 years ago
- d557734 API to control target delay in NetEq jitter buffer. NetEq maintains the given delay unless channel conditions require a higher delay. by turaj@webrtc.org · 11 years ago
- b7716d8 - Changed RemoteBitrateEstimator::IncomingPacket() to include a const WebRtcRTPHeader& and remove ssrc, rtp_timestamp. by solenberg@webrtc.org · 11 years ago
- 0204219 Disable WindowCapturer tests on OSX and Linux by sergeyu@chromium.org · 11 years ago
- ced13a5 Add direct_dependent_settings in common.gypi. by sergeyu@chromium.org · 11 years ago
- 389bb40 Refactor VCM/Timing. No changes in functionality. by mikhal@webrtc.org · 11 years ago
- 3740808 Consolidate GetFrame and InsertPacket and move NACK list processing to after a packet has been successfully inserted. by stefan@webrtc.org · 11 years ago
- 471ae72 Include files from webrtc/.. paths in voice_engine/ by pbos@webrtc.org · 11 years ago
- 8510750 Make sure VoiceEngine tests only include one test framework. by pbos@webrtc.org · 11 years ago
- 141a00c Remove <iostream> usage from loopback.cc by pbos@webrtc.org · 11 years ago
- 6169712 Suffix VcmCapturer's privates with underscore_ by pbos@webrtc.org · 11 years ago
- 15bdfdf Log timestamp of the frame when it's dropped from the render module by hclam@chromium.org · 11 years ago
- cca5086 Log error in ViESender::SendRTCPPacket by hclam@chromium.org · 11 years ago
- 366d158 Revert 4000 "Reverting r3978" by andrew@webrtc.org · 11 years ago
- 9038990 Revert 4001 "Revert 3977" by andrew@webrtc.org · 11 years ago
- 0e15695 Fix assertions in rtp_header_extension.h caused by not handling the AudioLevel extension. Added unit tests to do basic checks of the AudioLevel extension. by solenberg@webrtc.org · 11 years ago
- 6595271 Recalibrate point sample expectation by fbarchard@google.com · 11 years ago
- 453f9c0 Add functions to ViE API to enable/disable the absolute send time header extension. by solenberg@webrtc.org · 11 years ago
- e032f9f Window capturer implementation for Windows. by sergeyu@chromium.org · 11 years ago
- b9e5732 Avoid NPE crash on Android platforms that don't support getting preview framerate. by fischman@webrtc.org · 11 years ago
- 9ea8c99 Include gflags properly and X11 include order in VideoEngine. by pbos@webrtc.org · 11 years ago
- 281cff8 Include files from webrtc/.. paths in video_engine/ by pbos@webrtc.org · 11 years ago
- c7979e0 Improve wraparound handling in the render time extrapolator. by stefan@webrtc.org · 11 years ago
- de93f2c Moved command line parsing to internal tools and moved back the mic volume thingie. by phoglund@webrtc.org · 11 years ago
- e8dc588 Enable WebRTC demo application on x86 Android by fischman@webrtc.org · 11 years ago
- 8787048 Guarding certain operations, e.g. bandwidth estimation, RTCP statistics update etc., not to be run on sync RTPS. by turaj@webrtc.org · 11 years ago