Log - d3d364eb3382e3058f2d26f5e63431b23873101c - fp2-dev/platform/external/chromium_org/third_party/webrtc

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
6bcf2ab Update version number to 3.23 by tnakamura@webrtc.org · 11 years ago
75e6669 Made it possible to render custom call output to file. by phoglund@webrtc.org · 11 years ago
89c3de3 Don't report an error for GetEstimatedReceiveBandwidth if there is no valid by mflodman@webrtc.org · 11 years ago
34d1110 Enable indefinitely running vie_auto_test option by kjellander@webrtc.org · 11 years ago
db325e2 Updated version number to 3.22 by elham@webrtc.org · 11 years ago
cc895d1 Fixing/disabling Windows x64 warnings by kjellander@webrtc.org · 11 years ago
d6739c8 Adding a send side API for streaming by mikhal@webrtc.org · 11 years ago
a7761c7 Fix mismatch between different NACK list lengths and packet buffers. by stefan@webrtc.org · 11 years ago
3442158 Break out RemoteBitrateEstimator from RtpRtcp module and make RemoteBitrateEstimator::Process trigger new REMB messages. by stefan@webrtc.org · 11 years ago
fd2dd1a Set frame length for frame converting in external renderer by braveyao@webrtc.org · 11 years ago
8d759af VP8: Making key frame interval a tunnable parameter by mikhal@webrtc.org · 11 years ago
b4575c1 Fix webrtc compilation errors for Chrome Win64 by andrew@webrtc.org · 11 years ago
ceca869 Moving ViE test files and deleting files no longer used. by mflodman@webrtc.org · 11 years ago
3d7848b Updated version number to 3.21 by elham@webrtc.org · 11 years ago
81cfcb5 Remove '<(library)' in gyp files. by wjia@webrtc.org · 11 years ago
fc37398 Convert psnr and ssim to strings before printing them. by stefan@webrtc.org · 11 years ago
087c593 Removing outdated comment by mikhal@webrtc.org · 11 years ago