Log - b89eed3d8656c4a3e72a46746f81d7a4bd8496c1 - fp2-dev/platform/external/chromium_org/third_party/webrtc

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
0291c80 Removed ViE file API. by mflodman@webrtc.org · 11 years ago
0be9202 Do basic parsing of RTCP headers in PcapFileReader to enable log filtering. by solenberg@webrtc.org · 11 years ago
20cfda6 Remove unused multi stream bandwidth estimator. by solenberg@webrtc.org · 11 years ago
8ee45da Make sure padding packets are sent. by stefan@webrtc.org · 11 years ago
c8c333d mac: Mark kCGLPFAFullScreen as allowed in a 10.6-only function. by sergeyu@chromium.org · 11 years ago
0d35c78 Fix memory bot failure by hclam@chromium.org · 11 years ago
0f6f7cb Enqueue packet in pacer if sending fails by hclam@chromium.org · 11 years ago
44050b2 VCM: removing max jitter estimate by mikhal@webrtc.org · 11 years ago
f47d0f8 Remove unneeded *_NOT_SUPPORTED from VoEAudioProcessing. by andrew@webrtc.org · 11 years ago
7533659 Fixes some pacer/padding issues found while testing. by stefan@webrtc.org · 11 years ago
c9cb798 Use 3 threads for higher than 720p resolutions by fbarchard@google.com · 11 years ago
5b9adb0 Add a log message to see video delay break down by hclam@chromium.org · 11 years ago
b82ee51 Make ScreenCapturerMac work in versions of OSX before Lion. by sergeyu@chromium.org · 11 years ago
9ca71b1 Enable ScreenCapturer unittests by sergeyu@chromium.org · 11 years ago
ba458e2 Use intptr_t to represent window IDs on all platforms. by sergeyu@chromium.org · 11 years ago
69f7605 Wire up pacer-based padding. by stefan@webrtc.org · 11 years ago
67ca2b4 Revert r4145 "Revert 4127 "Switch frame list implementation to std::map."" by stefan@webrtc.org · 11 years ago
376ae3e Revert r4146 "Revert 4104 "Refactor jitter buffer to use separate lists for de..."" by stefan@webrtc.org · 11 years ago
b06dd93 Fix AV sync issue by hclam@chromium.org · 11 years ago
9540e2a Log current and target AV delay in ViESyncModule by hclam@chromium.org · 11 years ago
8dbc8ab Merge more tests into modules_{unit,integration}tests. by kjellander@webrtc.org · 11 years ago
6828566 WebRTCDemo: ensures that using front and back camera work as expected. by henrike@webrtc.org · 11 years ago
19b339a Fixes linker issue with no op trace. by henrike@webrtc.org · 11 years ago
7b2c430 Risk of division by zero. by turaj@webrtc.org · 11 years ago
cefb004 Revert 4211 "Build all java files into jar for each module on An..." by fischman@webrtc.org · 11 years ago
8ed5369 Disable ViEExtendedIntegrationTest.RunsCodecTestWithoutErrors test. by kjellander@webrtc.org · 11 years ago
83163e0 Disable ViEExtendedIntegrationTest.RunsCodecTestWithoutErrors test. by kjellander@webrtc.org · 11 years ago
789e98b Fix breakage due to test_fec conversion to gtest. by kjellander@webrtc.org · 11 years ago
f04f54a Convert test_fec to gtest by kjellander@webrtc.org · 11 years ago
f09f7b2 Disable ViEStandardIntegrationTest.RunsRtpRtcpTestWithoutErrors test. by kjellander@webrtc.org · 11 years ago
6ab2b1f G722_1/G722_1C codecs won't instantiate by tina.legrand@webrtc.org · 11 years ago
46cec2a Reorganize test targets in WebRTC by kjellander@webrtc.org · 11 years ago
0604490 Build all java files into jar for each module on Android by fischman@webrtc.org · 11 years ago
e8c9ecd Allow the screen capturer to capture oversized cursors and cursors without alpha channel (Windows). by alexeypa@chromium.org · 11 years ago
a463a4b Landing binary cursor image files to be used in a follow up CL. by alexeypa@chromium.org · 11 years ago
f1bcae0 Updated WebRTC version to 3.33 by elham@webrtc.org · 11 years ago
b4c89a4 Making no NACK mode work again in VideoEngine. by mflodman@webrtc.org · 11 years ago
63988b2 RW lock access to ssrc maps in VideoCall. by pbos@webrtc.org · 11 years ago
203d656 Add back the WEBRTC_DIRECT_TRACE flag. by solenberg@webrtc.org · 11 years ago