- d8daa60 Fix some chromium-style warnings in webrtc/modules/audio_device/ by pbos@webrtc.org · 11 years ago
- 99199e5 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
- 2dd408e PeerConnectionTest.java: make the test work for the bots' v4l2loopback. by fischman@webrtc.org · 11 years ago
- 09121dc Land http://webrtc-codereview.appspot.com/1632005/ by niklas.enbom@webrtc.org · 11 years ago
- 45e69ce Updated WebRTC version to 3.37 TBR=tnakamura@webrtc.org by elham@webrtc.org · 11 years ago
- 3081f6d Improved error messages when binaries are missing. Also stderr = stdout now. by phoglund@webrtc.org · 11 years ago
- d91e5ee To fix a bug in InverseFFTAndWindow() function in AECM. by kma@webrtc.org · 11 years ago
- 33f81b1 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
- d69e2f4 Access receiving_ under receive_cs critical section by braveyao@webrtc.org · 11 years ago
- 0496413 Don't set clang_use_chrome_plugins in common.gypi by sergeyu@chromium.org · 11 years ago
- f09b0e3 Fixes resources and data path in modules_unittests.isolate. by henrike@webrtc.org · 11 years ago
- 05f2131 Downstream latest Chromium SincResampler changes. by andrew@webrtc.org · 11 years ago
- 5199024 Update include paths in device_info_external.cc by sergeyu@chromium.org · 11 years ago
- 25613ea Add a Config class interface to AudioProcessing for passing options. by andrew@webrtc.org · 11 years ago
- f9373bb Fix include path in video_capture_external.cc by niklas.enbom@webrtc.org · 11 years ago
- d3b8af7 Formalized Real 16-bit FFT for APM. by kma@webrtc.org · 11 years ago
- fb0aa4e Fix ScreenCapturerLinux not to use XDamage when requested. by sergeyu@chromium.org · 11 years ago
- 5dcb4e3 webrtc/common_types.h: Document bitrate fields' units. by fischman@webrtc.org · 11 years ago
- 00c3b1e 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
- bf76ae2 Hooking up first simple CPU adaptation version. by mflodman@webrtc.org · 11 years ago
- 3df426b Revert 4382 "Makes webrtc and libjingle build from the same gyp-..." by henrike@webrtc.org · 11 years ago
- fa6c54c 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
- 175b0c0 Modified the presubmit checks such that difference license templates are checked for in webrtc and talk folder. by henrike@webrtc.org · 11 years ago
- 237fd31 Correctly rebuild WebRTCDemo after jni/ source file changes by yujie.mao@webrtc.org · 11 years ago
- 9183fad Revert 4372 "Makes webrtc and libjingle build from the same gyp-..." by henrike@webrtc.org · 11 years ago
- 282ca8e 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
- a86790e AppRTCDemo: build fixes for iOS build in webrtc by fischman@webrtc.org · 11 years ago
- f03fa34 Undo libvpx include changes in r4348 to fix build. by tnakamura@webrtc.org · 11 years ago
- 6349e17 Default constructor for RtcpAppHandler. by pbos@webrtc.org · 11 years ago
- 1c8d5a0 clean up incomplete revert in r4357 Also revert r4319, will follow up with pbos by tnakamura@webrtc.org · 11 years ago
- 0ba496b Revert r4301 by tnakamura@webrtc.org · 11 years ago
- 2eb721e 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
- 3936bb1 Include files from webrtc/.. paths in signal_processing/. by pbos@webrtc.org · 11 years ago
- 3eba48c Include files from webrtc/.. paths in media_file/. by pbos@webrtc.org · 11 years ago
- d0cfc70 Make sure first RTP packet counts as in-order. by pbos@webrtc.org · 11 years ago
- 4736a26 Include files from webrtc/.. paths in bitrate_controller/. by pbos@webrtc.org · 11 years ago
- a557f43 Include files from webrtc/.. paths in video_coding/. by pbos@webrtc.org · 11 years ago
- cb0c159 Revert r4320 "Fix three uninitialized members in rtp_receiver_impl.cc" by elham@webrtc.org · 11 years ago
- 9d788a1 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
- b89eed3 Revert r4322 "Support sending multiple report blocks and keeping track of statistics on by elham@webrtc.org · 11 years ago
- 4981d3c Revert r4328 by elham@webrtc.org · 11 years ago
- 5526bfb Updated WebRTC version to 3.36 TBR=tnakamura@webrtc.org by elham@webrtc.org · 11 years ago
- a2ef48c Remove dead video_capture for QuickTime. by pbos@webrtc.org · 11 years ago
- 87c29b5 Include files from webrtc/.. paths in video_capture/. by pbos@webrtc.org · 11 years ago
- f72d6b0 Include files from webrtc/.. paths in utility/. by pbos@webrtc.org · 11 years ago
- 6293e68 Remove dead code testAPI.cc. by pbos@webrtc.org · 11 years ago
- 7e5dc87 Include files from webrtc/.. paths in video_render/. by pbos@webrtc.org · 11 years ago
- 6a4acb9 Fix some voe_auto_test uninitialised-value errors. by pbos@webrtc.org · 11 years ago
- bc669ac Include files from webrtc/.. paths in audio_device/. by pbos@webrtc.org · 11 years ago
- 86c5732 Fix root-relative includes for pacing/. by pbos@webrtc.org · 11 years ago
- 7b66e14 Fixes a crash when sending SR reports from a sender only module. by stefan@webrtc.org · 11 years ago
- b48a4a9 ModuleRTPRTCP call rtcp_sender_.TMMBR() directly instead of calling its own API. by braveyao@webrtc.org · 11 years ago
- d3756f7 Sorted headers under rtp_rtcp/. by pbos@webrtc.org · 11 years ago
- e835019 Include files from webrtc/.. paths in video_engine/. by pbos@webrtc.org · 11 years ago
- 778a172 Direct3D renderer for new VideoEngine API tests. by pbos@webrtc.org · 11 years ago
- 46088d2 Support sending multiple report blocks and keeping track of statistics on several SSRCs. by stefan@webrtc.org · 11 years ago
- 446ea2e 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
- e37c2cd Fix three uninitialized members in rtp_receiver_impl.cc. by stefan@webrtc.org · 11 years ago
- d5e5863 Initialize payload-type frequency in channel.cc. by pbos@webrtc.org · 11 years ago
- 73a2ba6 Update version number to 3.35 by tnakamura@webrtc.org · 11 years ago
- 44cff54 Update version number to 3.34 by tnakamura@webrtc.org · 11 years ago
- f9c7018 Add root_path_android.cc to webrtc/test/Android.mk. by pbos@webrtc.org · 11 years ago
- e9bd299 Fixed implicit-int-conversion bugs. by pbos@webrtc.org · 11 years ago
- 57dbdbd 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
- f6d9630 Create gyp target for bwe components. by stefan@webrtc.org · 11 years ago
- 2f02da8 Initial port of FullStackTest to new VideoEngine API. by pbos@webrtc.org · 11 years ago
- 609e332 Arguments need to be separated when implementing gyp-actions. by henrike@webrtc.org · 11 years ago
- 9c0f14d Cleanup WebRTC tracing by hclam@chromium.org · 11 years ago
- d5f14d5 Added modules_unittests.isolate for ndk-apk builds. by henrike@webrtc.org · 11 years ago
- 7537dde Disables unit tests that don't work on Android for Android. by henrike@webrtc.org · 11 years ago
- 7e27a0f Fixes build breakage when building WebRTC in Chromium and having include_tests=1. by henrike@webrtc.org · 11 years ago
- 1f4fa04 Fixes broken gyp-condition. by henrike@webrtc.org · 11 years ago
- e25e28f 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
- b16cb05 Use scoped_ptr<> for loopback.cc by pbos@webrtc.org · 11 years ago
- a32d18f Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the by stefan@webrtc.org · 11 years ago
- 00d566e Revert 4298 "Makes it possible to find files used by some unit t..." by pbos@webrtc.org · 11 years ago
- 222efdc 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
- 39445b0 Adding Stefan as VideoEngine owner, removing Per. by mflodman@webrtc.org · 11 years ago
- 531a99b In AudioDeviceWindowsCore::_EnumerateEndpointDevicesAll(), continue enumerating if one individual device failed. by braveyao@webrtc.org · 11 years ago
- a0ebe97 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
- 3b89e10 Proper spacing for end-of-namespace comments. by pbos@webrtc.org · 11 years ago
- cbb3966 In call to Opus decoder: frame length too large by tina.legrand@webrtc.org · 11 years ago
- 1bd3140 Possible divide by 0 in ACM. by tina.legrand@webrtc.org · 11 years ago
- cbb535a Error in update of read index in ACM by tina.legrand@webrtc.org · 11 years ago
- 98ac1e8 Rename unit_test.{cc,h} under module_unittest. by pbos@webrtc.org · 11 years ago
- 438be80 Remove log of undefined input values in GetCodec. by pbos@webrtc.org · 11 years ago
- 5cf0fd1 Diff NTP and internal once in VideoCaptureImpl. by pbos@webrtc.org · 11 years ago
- 50c1aef Build all java files into jar for each module on Android by fischman@webrtc.org · 11 years ago
- 5e742a8 WebRTCViEDemo: Use global reference when passing variables across different threads by yujie.mao@webrtc.org · 11 years ago
- e63c003 Android opengles renderer: add thread sync to swap frame and draw native. by braveyao@webrtc.org · 11 years ago
- af60a80 Suppress excessive logging in video_coding by hclam@chromium.org · 11 years ago
- 8a5cb95 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
- c1624d5 Removes unused main function that is poluting the build. by henrike@webrtc.org · 11 years ago
- c7eab28 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
- d305e11 Move TickTime::QueryOsForTicks out-of-line by fischman@webrtc.org · 11 years ago
- 8148118 Removes kStateFree and kStateDecoding, added a free_frames_ list which simplifies finding a free frame. by stefan@webrtc.org · 11 years ago
- c10fc53 Fixed bad parameter passing in compare_videos.py by phoglund@webrtc.org · 11 years ago
- 3656192 Fix unnamed-type-template-args warnings on clang. by pbos@webrtc.org · 11 years ago
- 555f1cd Correctly rebuild WebRTCDemo-debug.apk after modules/ source file changes. by fischman@webrtc.org · 11 years ago
- f8f91d6 Adding a first simple version of overuse detection, but not hooked up. by mflodman@webrtc.org · 11 years ago