- 81e21c6 Added libjingle_peerconnection_java_unittest to buildbot_tests.py by phoglund@webrtc.org · 11 years ago
- fe8ba4d Move internal aec_core defines out of header. by andrew@webrtc.org · 11 years ago
- 3ea4830 Add svn:ignore properties for all spuriously-removed dirs on Linux64 Release (internal). by fischman@webrtc.org · 11 years ago
- b36d452 Correcting Turaj's email. by turaj@webrtc.org · 11 years ago
- 80df10d Fix some chromium-style warnings in webrtc/modules/video_coding/ by pbos@webrtc.org · 11 years ago
- ae6d494 Fix some chromium-style warnings in webrtc/test/ by pbos@webrtc.org · 11 years ago
- 9bf2b46 Fix some chromium-style warnings in webrtc/tools/ by pbos@webrtc.org · 11 years ago
- e142b98 Fix some chromium-style warnings in webrtc/modules/audio_device/ by pbos@webrtc.org · 11 years ago
- 46688dd Sets up framework for decoding with errors: collects frame sizes (in number of packets) in JB and passes this information to VCMSessionInfo with rtt_ms as FrameData. by agalusza@google.com · 11 years ago
- 03bfae8 PeerConnectionTest.java: make the test work for the bots' v4l2loopback. by fischman@webrtc.org · 11 years ago
- 2a61170 Land http://webrtc-codereview.appspot.com/1632005/ by niklas.enbom@webrtc.org · 11 years ago
- 0b8a595 Updated WebRTC version to 3.37 TBR=tnakamura@webrtc.org by elham@webrtc.org · 11 years ago
- 263411c Improved error messages when binaries are missing. Also stderr = stdout now. by phoglund@webrtc.org · 11 years ago
- 6429cdb To fix a bug in InverseFFTAndWindow() function in AECM. by kma@webrtc.org · 11 years ago
- 7b97b16 Updated WebRtcNsx_PrepareSpectrumNeon() in accordance with the new real FFT interface in APM. For reference, you can check https://webrtc-codereview.appspot.com/1830004/diff/92001/webrtc/modules/audio_processing/ns/nsx_core.c, line 594 "static void PrepareSpectrumC()". by kma@webrtc.org · 11 years ago
- 1095587 Access receiving_ under receive_cs critical section by braveyao@webrtc.org · 11 years ago
- dbdcf16 Don't set clang_use_chrome_plugins in common.gypi by sergeyu@chromium.org · 11 years ago
- 0bd4d89 Fixes resources and data path in modules_unittests.isolate. by henrike@webrtc.org · 11 years ago
- fd87865 Downstream latest Chromium SincResampler changes. by andrew@webrtc.org · 11 years ago
- b31f64f Update include paths in device_info_external.cc by sergeyu@chromium.org · 11 years ago
- d13f24b Add a Config class interface to AudioProcessing for passing options. by andrew@webrtc.org · 11 years ago
- 2cf4d85 Fix include path in video_capture_external.cc by niklas.enbom@webrtc.org · 11 years ago
- a3c7fa2 Formalized Real 16-bit FFT for APM. by kma@webrtc.org · 11 years ago
- 51f7c7e Fix ScreenCapturerLinux not to use XDamage when requested. by sergeyu@chromium.org · 11 years ago
- 530f40f webrtc/common_types.h: Document bitrate fields' units. by fischman@webrtc.org · 11 years ago
- 7b87e6b Makes webrtc and libjingle build from the same gyp-file. Also, the libjingle and webrtc DEPS revisions were mismatching. This cl takes the most recent revision of mismatches. Also disables 64 bit Mac builds for libjingle by henrike@webrtc.org · 11 years ago
- ecbeb2b Hooking up first simple CPU adaptation version. by mflodman@webrtc.org · 11 years ago
- cb2fb3f Revert 4382 "Makes webrtc and libjingle build from the same gyp-..." by henrike@webrtc.org · 11 years ago
- 4432261 Makes webrtc and libjingle build from the same gyp-file. Also, the libjingle and webrtc DEPS revisions were mismatching. This cl takes the most recent revision of mismatches. Also disables 64 bit Mac builds for libjingle by henrike@webrtc.org · 11 years ago
- 675eead Modified the presubmit checks such that difference license templates are checked for in webrtc and talk folder. by henrike@webrtc.org · 11 years ago
- 95a4477 Correctly rebuild WebRTCDemo after jni/ source file changes by yujie.mao@webrtc.org · 11 years ago
- 331a9b2 Revert 4372 "Makes webrtc and libjingle build from the same gyp-..." by henrike@webrtc.org · 11 years ago
- 30a83c1 Makes webrtc and libjingle build from the same gyp-file. Also, the libjingle and webrtc DEPS revisions were mismatching. This cl takes the most recent revision of mismatches. by henrike@webrtc.org · 11 years ago
- 7c4152b AppRTCDemo: build fixes for iOS build in webrtc by fischman@webrtc.org · 11 years ago
- 382ef1e Undo libvpx include changes in r4348 to fix build. by tnakamura@webrtc.org · 11 years ago
- a491674 Default constructor for RtcpAppHandler. by pbos@webrtc.org · 11 years ago
- 62365d0 clean up incomplete revert in r4357 Also revert r4319, will follow up with pbos by tnakamura@webrtc.org · 11 years ago
- 1628267 Revert r4301 by tnakamura@webrtc.org · 11 years ago
- 8eea45a Fixes: Resolves conflict that will happen when merging libjingle's and WebRTC's supplemental.gyp. By separating build_with_chromium and build_with_libjingle one can now just define build_with_libjingle in libjingle's supplemental.gyp. Once that is done it will be possible to merge the two supplemental.gyp-files. I.e. in WebRTC the supplemental.gyp would only set build_with_chromium to 0 since there is no longer any reason to disable logging and tests as they will be accessible in the same repository as libjingle. by henrike@webrtc.org · 11 years ago
- c64c74c Include files from webrtc/.. paths in signal_processing/. by pbos@webrtc.org · 11 years ago
- 93c9ccb Include files from webrtc/.. paths in media_file/. by pbos@webrtc.org · 11 years ago
- d92e5df Make sure first RTP packet counts as in-order. by pbos@webrtc.org · 11 years ago
- 3fa41a6 Include files from webrtc/.. paths in bitrate_controller/. by pbos@webrtc.org · 11 years ago
- 2a2e7ff Include files from webrtc/.. paths in video_coding/. by pbos@webrtc.org · 11 years ago
- 1a9da45 Revert r4320 "Fix three uninitialized members in rtp_receiver_impl.cc" by elham@webrtc.org · 11 years ago
- 11c4464 Revert r4321 "Fix uninitialized value warning in rtp_payload_registry and make sure we return an error if the payload type isn't registered" by elham@webrtc.org · 11 years ago
- 35c7707 Revert r4322 "Support sending multiple report blocks and keeping track of statistics on by elham@webrtc.org · 11 years ago
- 9706ba7 Revert r4328 by elham@webrtc.org · 11 years ago
- 71e2b87 Updated WebRTC version to 3.36 TBR=tnakamura@webrtc.org by elham@webrtc.org · 11 years ago
- fa0f507 Remove dead video_capture for QuickTime. by pbos@webrtc.org · 11 years ago
- adb1418 Include files from webrtc/.. paths in video_capture/. by pbos@webrtc.org · 11 years ago
- fab7ea2 Include files from webrtc/.. paths in utility/. by pbos@webrtc.org · 11 years ago
- 3aefd72 Remove dead code testAPI.cc. by pbos@webrtc.org · 11 years ago
- 9c45631 Include files from webrtc/.. paths in video_render/. by pbos@webrtc.org · 11 years ago
- f58779b Fix some voe_auto_test uninitialised-value errors. by pbos@webrtc.org · 11 years ago
- 7c413cd Include files from webrtc/.. paths in audio_device/. by pbos@webrtc.org · 11 years ago
- 784d202 Fix root-relative includes for pacing/. by pbos@webrtc.org · 11 years ago
- 5f999a3 Fixes a crash when sending SR reports from a sender only module. by stefan@webrtc.org · 11 years ago
- c80c63d ModuleRTPRTCP call rtcp_sender_.TMMBR() directly instead of calling its own API. by braveyao@webrtc.org · 11 years ago
- 633c018 Sorted headers under rtp_rtcp/. by pbos@webrtc.org · 11 years ago
- 3265bbd Include files from webrtc/.. paths in video_engine/. by pbos@webrtc.org · 11 years ago
- 8d671f5 Direct3D renderer for new VideoEngine API tests. by pbos@webrtc.org · 11 years ago
- 45ab259 Support sending multiple report blocks and keeping track of statistics on several SSRCs. by stefan@webrtc.org · 11 years ago
- 2211018 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
- aa64b24 Fix three uninitialized members in rtp_receiver_impl.cc. by stefan@webrtc.org · 11 years ago
- efa245a Initialize payload-type frequency in channel.cc. by pbos@webrtc.org · 11 years ago
- c2ae12e Update version number to 3.35 by tnakamura@webrtc.org · 11 years ago
- faf2dba Update version number to 3.34 by tnakamura@webrtc.org · 11 years ago
- 0114e3d Add root_path_android.cc to webrtc/test/Android.mk. by pbos@webrtc.org · 11 years ago
- 31c750c Fixed implicit-int-conversion bugs. by pbos@webrtc.org · 11 years ago
- 7ca2327 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
- 6dd15dc Create gyp target for bwe components. by stefan@webrtc.org · 11 years ago
- 12d5ede Initial port of FullStackTest to new VideoEngine API. by pbos@webrtc.org · 11 years ago
- 5ca7ffd Arguments need to be separated when implementing gyp-actions. by henrike@webrtc.org · 11 years ago
- 7183bbb Cleanup WebRTC tracing by hclam@chromium.org · 11 years ago
- 80fa00b Added modules_unittests.isolate for ndk-apk builds. by henrike@webrtc.org · 11 years ago
- 6f44ab3 Disables unit tests that don't work on Android for Android. by henrike@webrtc.org · 11 years ago
- 5b871f8 Fixes build breakage when building WebRTC in Chromium and having include_tests=1. by henrike@webrtc.org · 11 years ago
- 13efe02 Fixes broken gyp-condition. by henrike@webrtc.org · 11 years ago
- 0642536 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
- b7f287d Use scoped_ptr<> for loopback.cc by pbos@webrtc.org · 11 years ago
- a430fef Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the by stefan@webrtc.org · 11 years ago
- ad63306 Revert 4298 "Makes it possible to find files used by some unit t..." by pbos@webrtc.org · 11 years ago
- c82b35c 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
- 1d06d1a Adding Stefan as VideoEngine owner, removing Per. by mflodman@webrtc.org · 11 years ago
- e590835 In AudioDeviceWindowsCore::_EnumerateEndpointDevicesAll(), continue enumerating if one individual device failed. by braveyao@webrtc.org · 11 years ago
- 6504a1d 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
- 5ab7b93 Proper spacing for end-of-namespace comments. by pbos@webrtc.org · 11 years ago
- a539d8e In call to Opus decoder: frame length too large by tina.legrand@webrtc.org · 11 years ago
- 817d63c Possible divide by 0 in ACM. by tina.legrand@webrtc.org · 11 years ago
- 3b7be22 Error in update of read index in ACM by tina.legrand@webrtc.org · 11 years ago
- eaf7428 Rename unit_test.{cc,h} under module_unittest. by pbos@webrtc.org · 11 years ago
- 363852e Remove log of undefined input values in GetCodec. by pbos@webrtc.org · 11 years ago
- 787640d Diff NTP and internal once in VideoCaptureImpl. by pbos@webrtc.org · 11 years ago
- f808b77 Build all java files into jar for each module on Android by fischman@webrtc.org · 11 years ago
- 5934a6a WebRTCViEDemo: Use global reference when passing variables across different threads by yujie.mao@webrtc.org · 11 years ago
- cdfff5b Android opengles renderer: add thread sync to swap frame and draw native. by braveyao@webrtc.org · 11 years ago
- c9ba795 Suppress excessive logging in video_coding by hclam@chromium.org · 11 years ago
- 0c9b40d 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
- 399baf7 Removes unused main function that is poluting the build. by henrike@webrtc.org · 11 years ago