Log - 266d5f6452ff0db16d2609a0c8461aa591c05a38 - fp2-dev/platform/external/chromium_org/third_party/webrtc

266d5f6 Break video_engine/new_include/common.h into smaller parts. by pbos@webrtc.org · 11 years ago
251c209 Rename voice_engine_core -> voice_engine and move targets to voice_engine.gyp. by andrew@webrtc.org · 11 years ago
65a5f3f Updated WebRTC version to 3.32 TBR=wu@webrtc.org by elham@webrtc.org · 11 years ago
3bf9290 Don't return an estimated receive BW for channels not receiving video. by mflodman@webrtc.org · 11 years ago
cab277d Include gflags with "gflags/gflags.h" instead of <> by pbos@webrtc.org · 11 years ago
d927a4d Improve vie_autotest_rtp_rtcp by reenabling important tests and reducing flakiness. by stefan@webrtc.org · 11 years ago
a6ca12e Default constructors for new VideoEngine structs. by pbos@webrtc.org · 11 years ago
66b8717 Remove libvpx_intrinsics_sse4_1.a in Android.mk since this target is no longer generated in libvpx by fischman@webrtc.org · 11 years ago
8a3b04d - Created RemoteBitrateEstimator wrapper for use internally in (ViE) ChannelGroup. by solenberg@webrtc.org · 11 years ago
9bacbf4 Adding Mac test renderer, some test refactoring and made cpplint pass. by mflodman@webrtc.org · 11 years ago
104218e Disabling a flaky expectation in vie_autotest_rtp_rtcp.cc. by stefan@webrtc.org · 11 years ago
77fa22e Make sure GlxRenderer frees its resources. by pbos@webrtc.org · 11 years ago
1a37edd Adds integration test for RTX and fixes bugs found. by stefan@webrtc.org · 11 years ago
06721fc Fix regression where retransmission bitrate is no longer estimated. by stefan@webrtc.org · 11 years ago
c0dba24 CreateEmptyFrame casts from size_t to int. by pbos@webrtc.org · 11 years ago
d9f9185 FrameGenerator class for future fake capture device. by pbos@webrtc.org · 11 years ago
3d6a8bf Control new VideoEngine tests with gflags. by pbos@webrtc.org · 11 years ago
2660072 Adds print out of incoming resolution. by henrike@webrtc.org · 11 years ago
6de406d API to control target delay in NetEq jitter buffer. NetEq maintains the given delay unless channel conditions require a higher delay. by turaj@webrtc.org · 11 years ago
d822fe4 - Changed RemoteBitrateEstimator::IncomingPacket() to include a const WebRtcRTPHeader& and remove ssrc, rtp_timestamp. by solenberg@webrtc.org · 11 years ago
08dbe39 Remove <iostream> usage from loopback.cc by pbos@webrtc.org · 11 years ago
9a15fc3 Suffix VcmCapturer's privates with underscore_ by pbos@webrtc.org · 11 years ago
e37160f Log error in ViESender::SendRTCPPacket by hclam@chromium.org · 11 years ago
967320b Add functions to ViE API to enable/disable the absolute send time header extension. by solenberg@webrtc.org · 11 years ago
0313a3a Avoid NPE crash on Android platforms that don't support getting preview framerate. by fischman@webrtc.org · 11 years ago
ecd1591 Include gflags properly and X11 include order in VideoEngine. by pbos@webrtc.org · 11 years ago
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