Log - f2e6fb3f66b75ee543e4ce13b74d0f7393b0af2d - fp2-dev/platform/external/chromium_org/third_party/webrtc

f2e6fb3 Include files from webrtc/.. paths in video_engine/ by pbos@webrtc.org · 11 years ago
db9d0be Enable WebRTC demo application on x86 Android by fischman@webrtc.org · 11 years ago
fc8382b Cleanup traces in WebRTC by hclam@chromium.org · 11 years ago
dc8c883 New VideoEngine API implementation on top of old one, first steps. by pbos@webrtc.org · 11 years ago
864f9d7 Remove SetOverUseDetectorOptions and cleaned ViESharedData. by mflodman@webrtc.org · 11 years ago
2b2e78c Adding a factory to remote bitrate estimator and allow it to be set via config. by andresp@webrtc.org · 11 years ago
4981e61 Fix typo in log statement. witdh should be width. by fbarchard@google.com · 11 years ago
302f731 Add more tracing for key frames. by justinlin@chromium.org · 11 years ago
f308b75 Increased the limit for KViEMaxCaptureDevices from 10 to 256. See issue 1343. by vikasmarwaha@webrtc.org · 11 years ago
34d0fec Updated WebRTC version to 3.31 TBR=wu@webrtc.org by elham@webrtc.org · 11 years ago
196ed2e Disabled flaky codec test (RunsCodecTestWithoutErrors) by phoglund@webrtc.org · 11 years ago
ad2b368 Wiring down config from video engine until video coding and remote bitrate estimator modules instantiation. by andresp@webrtc.org · 11 years ago
af6696e Remove TEXT(x) for BUILDINFO macros. by pbos@webrtc.org · 11 years ago
98a1ee2 This is the first step to convert building the Android WebRTC demo to a proper GYP target, android ndk toolchains is being used to build the jni cpp files instead of using ndk-build. by fischman@webrtc.org · 11 years ago
b181cac Since the layout of the Android WebRTC demo application is fixed, if we start the demo application in portrait postion, the activity will be destroyed and then created again, force the demo application to start in landscape position to avoid activity re-creation. by fischman@webrtc.org · 11 years ago
4ddb5bd WebRTCDemo Android doesn't hangle activity recreation correctly. by braveyao@webrtc.org · 11 years ago
f795df0 Add fischman into OWNERS of WebRTCDemo Android. by braveyao@webrtc.org · 11 years ago
d3d364e Fix compile errors in ViE with latest clang. by andrew@webrtc.org · 11 years ago
e155626 Clean creation of VideoEngine: by andresp@webrtc.org · 11 years ago
89f9266 Trigger a PLI if the duration of non-decodable frames exceeds a threshold. by stefan@webrtc.org · 11 years ago
34e0403 Fix clang errors in non-GYP_DEFINES=clang=1 build by pbos@webrtc.org · 11 years ago
e0aad3c Updated WebRTC version number to 3.30 by elham@webrtc.org · 11 years ago
a257915 Get rid of some unnecessary copying when sending REMBs. by solenberg@webrtc.org · 11 years ago
76318c5 Removing bad code resulting in flaky test. by pwestin@webrtc.org · 11 years ago
c06da8c Adding trace and changing pacing constants by pwestin@webrtc.org · 11 years ago
8f5edba Bugfix custom call stop. by pwestin@webrtc.org · 11 years ago
74161fc WebRTCDemo Android app to route audio to headphone when it's plugged in. by braveyao@webrtc.org · 11 years ago
a23b051 Consolidate common_audio into a single target. by andrew@webrtc.org · 11 years ago
c5fbd58 Fixing AV sync. Increased 2 const to allow for a bigger difference in AV sync. by pwestin@webrtc.org · 11 years ago
faec77d Adding buffered mode to loopback test by mikhal@webrtc.org · 11 years ago
8a159ad Removing vie file related code from vie_custom_call by mikhal@webrtc.org · 11 years ago
fd7a1b7 Fixed remaining nits from Stefan by pwestin@webrtc.org · 11 years ago
71645c8 Fix the encoder pause logic. BUG=1691 by pwestin@webrtc.org · 11 years ago
2423690 Disabling avi file interface by mikhal@webrtc.org · 11 years ago
b35efcc Revert "Add a default RTT to CallStats and use different values for buffered/real-time mode." by stefan@webrtc.org · 11 years ago
65e6f91 Add a default RTT to CallStats and use different values for buffered/real-time mode. by stefan@webrtc.org · 11 years ago
7a14b35 Updated the sync module with a slow moving filter by pwestin@webrtc.org · 11 years ago
fe2bce3 Removed unused variable. by mflodman@webrtc.org · 11 years ago
fb5b5cb Fixing Coverity issues. by mflodman@webrtc.org · 11 years ago
1ccedf6 Ensure build_demo.py run subprocesses with bash shell. by kjellander@webrtc.org · 11 years ago
69b0d2c New ViE interface. by mflodman@webrtc.org · 11 years ago
e45d9af Remove vim/emacs modelines from .gypi files by pbos@webrtc.org · 11 years ago
06077c9 Adding a payload type for RTX. by mflodman@webrtc.org · 11 years ago
c4c16bf Change capture interface to use NTP capture time. by stefan@webrtc.org · 11 years ago
e90a0af Adding playout buffer status to the voe video sync by pwestin@webrtc.org · 11 years ago
8129077 Updated WebRTC version number to 3.29 by elham@webrtc.org · 11 years ago
b28e522 WebRTCDemo: Enable making multiple calls. by fischman@webrtc.org · 11 years ago
bffd956 More trace events by hclam@chromium.org · 11 years ago
41e3677 Revert "With these changes we will assume that the capture time of a frame is based on NTP time. This makes the interface of video engine more well defined and makes it easier and cleaner to handle user provided capture timestamps." by stefan@webrtc.org · 11 years ago
2a5d229 WebRtc_Word32 -> int32_t in video_engine/ by pbos@webrtc.org · 11 years ago
82e0d35 With these changes we will assume that the capture time of a frame is based on NTP time. This makes the interface of video engine more well defined and makes it easier and cleaner to handle user provided capture timestamps. by stefan@webrtc.org · 11 years ago
51868ad Always set render delay in ViEChannel::RegisterExternalDecoder. by pbos@webrtc.org · 11 years ago
713488f Remove the old unused udp_transport by pwestin@webrtc.org · 11 years ago
98e70d4 Clean packets on the network when closing + made loopback test actually run again. by mflodman@webrtc.org · 11 years ago
ad45772 Add GYP target for WebRTC Video demo for Android. by kjellander@webrtc.org · 11 years ago
3c48614 Permit arbitrary payload names for kVideoCodecGeneric. by pbos@webrtc.org · 11 years ago
47e4f00 Remove WEBRTC_*_ENGINE_NETWORK_API use by pwestin@webrtc.org · 11 years ago
0b8adb4 Adds event traces and counters for WebRTC receive side. by edjee@google.com · 11 years ago
e561f8c Remove UDP transport API from ViE by pwestin@webrtc.org · 11 years ago
2379013 Updated Webrtc version to 3.28 by elham@webrtc.org · 11 years ago
1ca9d42 Removed all code enclosed in WEBRTC_SRTP #ifdefs, and the unsupported VoE SRTP APIs. Test stubs are left in place as we still have the (De)RegisterExternalEncryption() APIs, although they are currently untested. by solenberg@webrtc.org · 11 years ago
fece2f5 Fix broken audio. by leozwang@webrtc.org · 11 years ago
c3ab830 Fix flakiness in network up/down event tests when running under memcheck. by stefan@webrtc.org · 11 years ago
09e8463 WebRTCDemo: remove unnecessary stop & start during orientation change which isn't necessary since API v14. by fischman@webrtc.org · 11 years ago
e3eea1b Add interface to signal a network down event. by stefan@webrtc.org · 11 years ago
e760243 Updated WebRTC version to 3.27 by elham@webrtc.org · 11 years ago
d3eb512 Bugfix for extended RTP/RTCP test by pwestin@webrtc.org · 11 years ago
9c3b7bd Move the VIE tests to use external transport instead of the built in udp transport by pwestin@webrtc.org · 11 years ago
90fa4a1 Add min and target bitrate to VideoCodec. by marpan@webrtc.org · 11 years ago
06d1e8f Follow-up fix for r3681. by stefan@webrtc.org · 11 years ago
035c96a Updated WebRTC version number to 3.26 by elham@webrtc.org · 11 years ago
3be5a98 Change VCM interface to take target bitrate in bits per second. by stefan@webrtc.org · 11 years ago
a2e9124 Generic video-codec support. by pbos@webrtc.org · 11 years ago
072c9b6 Adding RTX on source by mikhal@webrtc.org · 11 years ago
9a7b9f7 Revert r3667 and r3665 Review URL: https://webrtc-codereview.appspot.com/1199004 by pwestin@webrtc.org · 11 years ago
a891566 Added destructors for tests to control destruct order by pwestin@webrtc.org · 11 years ago
25023aa Increasing size of nack list in buffered mode. by mikhal@webrtc.org · 11 years ago
66ccc6e Removed the engine API:s related to transport such as SetSendDestination, the functionality is now provided via the test frame work. by pwestin@webrtc.org · 11 years ago
2a3949f Lazy capture_device_info acquisition. by pbos@webrtc.org · 11 years ago
ace0823 Enabling bufffering mode with no sync module or VoE by mikhal@webrtc.org · 11 years ago
87d8f2d Updated version number to 3.25 by elham@webrtc.org · 11 years ago
3da576e Update integration tests for idempotent RTP header settings. by bemasc@google.com · 11 years ago
1dcba31 Destroy VCM and VPM instead of delete. by mflodman@webrtc.org · 11 years ago
ca65c51 Handle multiple calls to set initial delay by mikhal@webrtc.org · 11 years ago
213217c Stop and restart fix. by mflodman@webrtc.org · 11 years ago
2325284 Fixed typo in vie_autotest_loopback.cc. by pbos@webrtc.org · 11 years ago
cb139b1 Rename webrtc::StatsObserver to webrtc::CallStatsObserver by fischman@webrtc.org · 11 years ago
432bc1a fixing nack list size calculation by mikhal@webrtc.org · 11 years ago
39eb955 Updated version number to 3.24 by elham@webrtc.org · 11 years ago
85e2e0e Refactoring temporal layers implementation and adding VideoCodecMode for easier control of codec settings. by stefan@webrtc.org · 11 years ago
ce3f2ca Add VoE interface to VieRTP test by mikhal@webrtc.org · 11 years ago
4db69af Adding a receive side API for buffering mode. by mikhal@webrtc.org · 11 years ago
64506e2 Roll Chromium revision 176094:182149 by kjellander@webrtc.org · 11 years ago
e740a7b Remove MultiStreamMode from test. by stefan@webrtc.org · 11 years ago
4c6689a Reset ssrc when calling SetSendCodec. by mflodman@webrtc.org · 11 years ago
33c6e92 Sync libvpx and its gyp wrapper from Chromium. by andrew@webrtc.org · 11 years ago
1fb8372 Increase maximum resolution to 4k x 3k. by fbarchard@google.com · 11 years ago
9c4707e Android NDK build tools by kjellander@webrtc.org · 11 years ago
4da62e0 Set SingleStream BWE in unittests. by stefan@webrtc.org · 11 years ago
6cd34e5 Updates to send side streaming mode: by mikhal@webrtc.org · 11 years ago