- 8b06200 Include files from webrtc/.. paths in utility/. by pbos@webrtc.org · 11 years ago
- 0ed57c5 Remove dead code testAPI.cc. by pbos@webrtc.org · 11 years ago
- 5aa3f1b Include files from webrtc/.. paths in video_render/. by pbos@webrtc.org · 11 years ago
- 5b10d8f Fix some voe_auto_test uninitialised-value errors. by pbos@webrtc.org · 11 years ago
- ffe16bd trunk/talk: removes empty folders. by henrike@webrtc.org · 11 years ago
- 811269d Include files from webrtc/.. paths in audio_device/. by pbos@webrtc.org · 11 years ago
- db6e3f8 Fix root-relative includes for pacing/. by pbos@webrtc.org · 11 years ago
- e4736ee Fixes a crash when sending SR reports from a sender only module. by stefan@webrtc.org · 11 years ago
- aeba6e8 ModuleRTPRTCP call rtcp_sender_.TMMBR() directly instead of calling its own API. by braveyao@webrtc.org · 11 years ago
- 96edd56 Sorted headers under rtp_rtcp/. by pbos@webrtc.org · 11 years ago
- 69215d8 Include files from webrtc/.. paths in video_engine/. by pbos@webrtc.org · 11 years ago
- adf23a5 Direct3D renderer for new VideoEngine API tests. by pbos@webrtc.org · 11 years ago
- 717d147 Support sending multiple report blocks and keeping track of statistics on several SSRCs. by stefan@webrtc.org · 11 years ago
- 9de89a6 Fix uninitialized value warning in rtp_payload_registry and make sure we return an error if the payload type isn't registered. by stefan@webrtc.org · 11 years ago
- 452d853 Fix three uninitialized members in rtp_receiver_impl.cc. by stefan@webrtc.org · 11 years ago
- 08933a5 Initialize payload-type frequency in channel.cc. by pbos@webrtc.org · 11 years ago
- 28e2075 Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk by henrike@webrtc.org · 11 years ago
- 6aa6229 Update version number to 3.35 by tnakamura@webrtc.org · 11 years ago
- c79b929 Update version number to 3.34 by tnakamura@webrtc.org · 11 years ago
- fc496d9 Add root_path_android.cc to webrtc/test/Android.mk. by pbos@webrtc.org · 11 years ago
- f3f1358 Fixed implicit-int-conversion bugs. by pbos@webrtc.org · 11 years ago
- cab716c Fix a circular dependency by removing an unnecessary dependency, add a missing include_tests check and missing lib references for android. by stefan@webrtc.org · 11 years ago
- f56d612 Create gyp target for bwe components. by stefan@webrtc.org · 11 years ago
- af8d5af Initial port of FullStackTest to new VideoEngine API. by pbos@webrtc.org · 11 years ago
- 5fc4d34 Arguments need to be separated when implementing gyp-actions. by henrike@webrtc.org · 11 years ago
- 1a7b9b9 Cleanup WebRTC tracing by hclam@chromium.org · 11 years ago
- e80a934 Added modules_unittests.isolate for ndk-apk builds. by henrike@webrtc.org · 11 years ago
- a950300b Disables unit tests that don't work on Android for Android. by henrike@webrtc.org · 11 years ago
- a2073af Fixes build breakage when building WebRTC in Chromium and having include_tests=1. by henrike@webrtc.org · 11 years ago
- bd3eee3 Fixes broken gyp-condition. by henrike@webrtc.org · 11 years ago
- 34773d9 Unreverts revert: Makes it possible to find files used by some unit tests when running them as Chrome native tests. by henrike@webrtc.org · 11 years ago
- 1932fe1 Use scoped_ptr<> for loopback.cc by pbos@webrtc.org · 11 years ago
- 66b2e5c Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the by stefan@webrtc.org · 11 years ago
- d4d9480 Added gum4.html, a multiple camera opening demo, each opening with a different resolution and/or frame rate. by mcasas@webrtc.org · 11 years ago
- db7d82f Revert 4298 "Makes it possible to find files used by some unit t..." by pbos@webrtc.org · 11 years ago
- caf2fcc Makes it possible to find files used by some unit tests when running them as Chrome native tests. by henrike@webrtc.org · 11 years ago
- 21beaf9 Adding Stefan as VideoEngine owner, removing Per. by mflodman@webrtc.org · 11 years ago
- 0b8636a In AudioDeviceWindowsCore::_EnumerateEndpointDevicesAll(), continue enumerating if one individual device failed. by braveyao@webrtc.org · 11 years ago
- 1303af3 Makes it possible to build ndk-apks of native unit tests if the workspace is inside a chromium checkout. by henrike@webrtc.org · 11 years ago
- d900e8b Proper spacing for end-of-namespace comments. by pbos@webrtc.org · 11 years ago
- 45426ea In call to Opus decoder: frame length too large by tina.legrand@webrtc.org · 11 years ago
- f6f033f Possible divide by 0 in ACM. by tina.legrand@webrtc.org · 11 years ago
- b1698ab Error in update of read index in ACM by tina.legrand@webrtc.org · 11 years ago
- ecd3c80 Add Magnus to root owners. by tommi@webrtc.org · 11 years ago
- c66aaaf Rename unit_test.{cc,h} under module_unittest. by pbos@webrtc.org · 11 years ago
- 510dfad Update myself in webrtc watchlist by yujie.mao@webrtc.org · 11 years ago
- 65a1f2c Remove log of undefined input values in GetCodec. by pbos@webrtc.org · 11 years ago
- 504af45 Diff NTP and internal once in VideoCaptureImpl. by pbos@webrtc.org · 11 years ago
- 546c91d Build all java files into jar for each module on Android by fischman@webrtc.org · 11 years ago
- d4803ce WebRTCViEDemo: Use global reference when passing variables across different threads by yujie.mao@webrtc.org · 11 years ago
- 90cc3b9 Android opengles renderer: add thread sync to swap frame and draw native. by braveyao@webrtc.org · 11 years ago
- 5616aba Suppress excessive logging in video_coding by hclam@chromium.org · 11 years ago
- 2a7fd53 Moves tools/update.py to trunk/webrtc/tools and updates it so that it no longer pulls any information from the DEPS file. by henrike@webrtc.org · 11 years ago
- 83cebb2 Removes unused main function that is poluting the build. by henrike@webrtc.org · 11 years ago
- 0021632 Re-add WebRTCDemo dependencies as dependencies (not just inputs) because they also need to be built for this target! by fischman@webrtc.org · 11 years ago
- 1d4a2d5 Move TickTime::QueryOsForTicks out-of-line by fischman@webrtc.org · 11 years ago
- 4cf1a8a Removes kStateFree and kStateDecoding, added a free_frames_ list which simplifies finding a free frame. by stefan@webrtc.org · 11 years ago
- 7bcc7e3 Fixed bad parameter passing in compare_videos.py by phoglund@webrtc.org · 11 years ago
- 2de80dd Fix unnamed-type-template-args warnings on clang. by pbos@webrtc.org · 11 years ago
- 3145a64 Correctly rebuild WebRTCDemo-debug.apk after modules/ source file changes. by fischman@webrtc.org · 11 years ago
- e6168f5 Adding a first simple version of overuse detection, but not hooked up. by mflodman@webrtc.org · 11 years ago
- 1c986e7 Removed ViE file API. by mflodman@webrtc.org · 11 years ago
- a5fd2f1 Do basic parsing of RTCP headers in PcapFileReader to enable log filtering. by solenberg@webrtc.org · 11 years ago
- 892d750 Add *.DS_Store to .gitignore so that ._.DS_Store is ignored too. by solenberg@webrtc.org · 11 years ago
- 91811e2 Remove unused multi stream bandwidth estimator. by solenberg@webrtc.org · 11 years ago
- a4c5abb Make sure padding packets are sent. by stefan@webrtc.org · 11 years ago
- bb25256 Added changes in apprtc demo to ignore turn address through query string for FF. Also made sure that the iceServers array doesnot include transport parameter in turn url for FF. Finally removed turn: from the turn_url when creating iceservers for pre-M28 chrome. by vikasmarwaha@webrtc.org · 11 years ago
- 3348ae2 mac: Mark kCGLPFAFullScreen as allowed in a 10.6-only function. by sergeyu@chromium.org · 11 years ago
- bb4f225 Roll libvpx to 207593. -pick up libvpx roll to c259af4f. by marpan@webrtc.org · 11 years ago
- 6eb53f7 Fix memory bot failure by hclam@chromium.org · 11 years ago
- 2e402ce Enqueue packet in pacer if sending fails by hclam@chromium.org · 11 years ago
- 9ca7360 VCM: removing max jitter estimate by mikhal@webrtc.org · 11 years ago
- 0851df8 Remove unneeded *_NOT_SUPPORTED from VoEAudioProcessing. by andrew@webrtc.org · 11 years ago
- 8ccb9f9 Fixes some pacer/padding issues found while testing. by stefan@webrtc.org · 11 years ago
- 2d7617a Add dummy Android test APK to be used for buildbot automation testing. by kjellander@webrtc.org · 11 years ago
- d7148c8 Use 3 threads for higher than 720p resolutions by fbarchard@google.com · 11 years ago
- 30fb7b8 Add a log message to see video delay break down by hclam@chromium.org · 11 years ago
- 6cfe178 Chromium Android tools for test execution. by kjellander@webrtc.org · 11 years ago
- a20eb91 Make ScreenCapturerMac work in versions of OSX before Lion. by sergeyu@chromium.org · 11 years ago
- 9e18279 Enable ScreenCapturer unittests by sergeyu@chromium.org · 11 years ago
- a590b41 Use intptr_t to represent window IDs on all platforms. by sergeyu@chromium.org · 11 years ago
- 508a84b Wire up pacer-based padding. by stefan@webrtc.org · 11 years ago
- 50fb4af Revert r4145 "Revert 4127 "Switch frame list implementation to std::map."" by stefan@webrtc.org · 11 years ago
- c8b29a2 Revert r4146 "Revert 4104 "Refactor jitter buffer to use separate lists for de..."" by stefan@webrtc.org · 11 years ago
- 7262ad1 Fix AV sync issue by hclam@chromium.org · 11 years ago
- 9b23ecb Log current and target AV delay in ViESyncModule by hclam@chromium.org · 11 years ago
- 63e9888 Merge more tests into modules_{unit,integration}tests. by kjellander@webrtc.org · 11 years ago
- f27389c WebRTCDemo: ensures that using front and back camera work as expected. by henrike@webrtc.org · 11 years ago
- d4ed1a3 Fixes linker issue with no op trace. by henrike@webrtc.org · 11 years ago
- a193339 Apprtc CSS: Add flip to local view of FireFox and remove warning of Canary by braveyao@webrtc.org · 11 years ago
- fee739c Risk of division by zero. by turaj@webrtc.org · 11 years ago
- dd97ef4 Revert 4211 "Build all java files into jar for each module on An..." by fischman@webrtc.org · 11 years ago
- 20a993f Disable ViEExtendedIntegrationTest.RunsCodecTestWithoutErrors test. by kjellander@webrtc.org · 11 years ago
- 935d705 Disable ViEExtendedIntegrationTest.RunsCodecTestWithoutErrors test. by kjellander@webrtc.org · 11 years ago
- 04996cd Fix breakage due to test_fec conversion to gtest. by kjellander@webrtc.org · 11 years ago
- 22bbbdf Convert test_fec to gtest by kjellander@webrtc.org · 11 years ago
- 7124dd8 Disable ViEStandardIntegrationTest.RunsRtpRtcpTestWithoutErrors test. by kjellander@webrtc.org · 11 years ago
- 18275a8 Update bots to make LKGR progress. by kjellander@webrtc.org · 11 years ago
- b097670 G722_1/G722_1C codecs won't instantiate by tina.legrand@webrtc.org · 11 years ago
- 2ef9513 libyuv r723 with convert util -attenuate feature used to fix transparent pixels used by Effects. By attenuating and then unattenuating, any transparent pixels will have RGB value of black, which will filter correctly when bilinear resized. by fbarchard@google.com · 11 years ago