- 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
- 6c35e0b Reorganize test targets in WebRTC by kjellander@webrtc.org · 11 years ago
- 6d6d95e Add support for test disable files in webrtc_tests.py by kjellander@webrtc.org · 11 years ago
- 1374965 Build all java files into jar for each module on Android by fischman@webrtc.org · 11 years ago
- 4af0878 Allow the screen capturer to capture oversized cursors and cursors without alpha channel (Windows). by alexeypa@chromium.org · 11 years ago
- 5e03f8a Landing binary cursor image files to be used in a follow up CL. by alexeypa@chromium.org · 11 years ago
- dfa1c4a libyuv r722 for OWNERS file for chromium, white space fix for lint, unittests on scale use randomize to reduce overhead, and neon change from vld1.u8 to vld1.8 for better compiler portability. by fbarchard@google.com · 11 years ago
- fe6b571 AppRTCDemo: delete hosted android_channel.html now that it's no longer necessary. by fischman@webrtc.org · 11 years ago
- 5137b97 Updated WebRTC version to 3.33 by elham@webrtc.org · 11 years ago
- 509754c Making no NACK mode work again in VideoEngine. by mflodman@webrtc.org · 11 years ago
- 1819fd7 RW lock access to ssrc maps in VideoCall. by pbos@webrtc.org · 11 years ago
- adb51f5 Add back the WEBRTC_DIRECT_TRACE flag. by solenberg@webrtc.org · 11 years ago
- 83a062c AudioDeviceAndroidOpenSLES: NULL variables might be referenced in StopPlayout() by braveyao@webrtc.org · 11 years ago
- 569fdef Revert some variables to uint32_t to fix compile errors on Mac gcc. by andrew@webrtc.org · 11 years ago
- 6f69eb7 Allow audio devices with up to 64 channels on Mac. by andrew@webrtc.org · 11 years ago
- 1064cf0 Fixed Rtp/Rtcp tests by pwestin@webrtc.org · 11 years ago
- 6367fe8 Fix relative path to .gitignore and other minor changes. by andrew@webrtc.org · 11 years ago
- 3ba883f Removing functionality for inserting pre-encoded frames instead of raw by mflodman@webrtc.org · 11 years ago
- b69cc15 Add script for appending entries to .gitignore. by andrew@webrtc.org · 11 years ago
- da71044 Fix size_t to int conversion error on Win64. by andrew@webrtc.org · 11 years ago
- 7e4ff35 Remove fake screen capturer because it's not used anywhere. by sergeyu@chromium.org · 11 years ago
- 8d80fa8 Fix for STL vector function data not available. by pwestin@webrtc.org · 11 years ago
- d30859e Connect ACM with RTP module for audio NACK. by pwestin@webrtc.org · 11 years ago
- a305e96 Nack for audio. by turaj@webrtc.org · 11 years ago
- d9c4658 Fix leaks in DesktopRegion by sergeyu@chromium.org · 11 years ago
- 2b3a29a Implement DetectNumberOfCores on Android and make it consistent on Linux and Android by fischman@webrtc.org · 11 years ago
- db24995 Wire up Nack for Voe by pwestin@webrtc.org · 11 years ago
- 7f1b0ae Fix init list for VideoSendStream::Config::Rtp. by pbos@webrtc.org · 11 years ago
- 025f4f1 Stats+Config moved into VideoSend/ReceiveStreams. by pbos@webrtc.org · 11 years ago
- fec34d7 Merge webrtc_utility_unittests into modules_unittests. by kjellander@webrtc.org · 11 years ago
- b2d29bd Restore relative include paths to libyuv. by andrew@webrtc.org · 11 years ago
- 3942f3a 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
- 16d78bd Fix scale.cc build error with mingw64 -m32 gcc by fbarchard@google.com · 11 years ago
- 9238de9 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
- 3d34f66 Move screen capturers from chromium to webrtc. by sergeyu@chromium.org · 11 years ago
- b7a8f43 Roll chromium_revision in webrtc 199267:203806 by fischman@webrtc.org · 11 years ago
- 430464c Add WebKit/Tools/Scripts to support Android test execution. by kjellander@webrtc.org · 11 years ago
- a817962 Refactor padding and rtp header functionality. by stefan@webrtc.org · 11 years ago
- de98478 Update the remote bitrate estimator before passing the packet to the RTP module. by stefan@webrtc.org · 11 years ago
- 6998c8e Remove XvRenderer. by pbos@webrtc.org · 11 years ago
- 8ad3ec9 Fix build error introduced with r4168. by stefan@webrtc.org · 11 years ago
- c3cc375 Add support for padding in pacer. by stefan@webrtc.org · 11 years ago
- c69ae69 Include files from webrtc/.. paths in common_video/ by pbos@webrtc.org · 11 years ago
- ba7f6a8 Include files from webrtc/.. paths in tools/ by pbos@webrtc.org · 11 years ago