Log - a048d7cb0a5bad5ca49bbcc5273cb4cca28c1710 - platform/external/webrtc

a048d7c Include files from webrtc/.. paths in rtp_rtcp/ by pbos@webrtc.org · 11 years ago
eea2622 Correctly set SSRCs for extra send RTP modules. by stefan@webrtc.org · 11 years ago
7bdfff3 Remove assert for aborting FrameGeneratorCapturer. by pbos@webrtc.org · 11 years ago
26d1210 Fake VideoCapturer based on FrameGenerator by pbos@webrtc.org · 11 years ago
08994cc Fix a return value mismatch introduced in r4129. by stefan@webrtc.org · 11 years ago
9aca5b3 Remove #pragma once by pbos@webrtc.org · 11 years ago
a5cb98c Breaking out RTP header parsing from the RTP module. by stefan@webrtc.org · 11 years ago
1ecee9a Break video_engine/new_include/common.h into smaller parts. by pbos@webrtc.org · 11 years ago
ace7ad2 Switch frame list implementation to std::map. by stefan@webrtc.org · 11 years ago
f791b1c Rename voice_engine_core -> voice_engine and move targets to voice_engine.gyp. by andrew@webrtc.org · 11 years ago
a6ae644 Add comment about test_packet_masks_metrics. by marpan@webrtc.org · 11 years ago
fe6a75e Updated WebRTC version to 3.32 TBR=wu@webrtc.org by elham@webrtc.org · 11 years ago
a066cbf Don't return an estimated receive BW for channels not receiving video. by mflodman@webrtc.org · 11 years ago
4079c31 Include gflags with "gflags/gflags.h" instead of <> by pbos@webrtc.org · 11 years ago
8c34cee Include "gtest/gtest.h", not by full path, on WEBRTC_ANDROID_PLATFORM_BUILD by pbos@webrtc.org · 11 years ago
3496ef1 Improve vie_autotest_rtp_rtcp by reenabling important tests and reducing flakiness. by stefan@webrtc.org · 11 years ago
15c1c61 Include files from webrtc/.. paths in audio_conference_mixer/ by pbos@webrtc.org · 11 years ago
7fad4b8 Include files from webrtc/.. paths in audio_processing/ by pbos@webrtc.org · 11 years ago
eceb532 Default constructors for new VideoEngine structs. by pbos@webrtc.org · 11 years ago
68c05f4 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
a6db54d - Created RemoteBitrateEstimator wrapper for use internally in (ViE) ChannelGroup. by solenberg@webrtc.org · 11 years ago
7f944f3 Adding Mac test renderer, some test refactoring and made cpplint pass. by mflodman@webrtc.org · 11 years ago
acaf3a1 Include files from webrtc/.. paths in system_wrappers/ by pbos@webrtc.org · 11 years ago
1e50231 Include files from webrtc/.. paths in test/channel_transport/ by pbos@webrtc.org · 11 years ago
6f3d8fc Include files from webrtc/.. paths in video_processing/ by pbos@webrtc.org · 11 years ago
47ce120 Include files from webrtc/.. paths in remote_bitrate_estimator/ by pbos@webrtc.org · 11 years ago
aa30bb7 Include files from webrtc/.. paths in common_audio/ by pbos@webrtc.org · 11 years ago
0afd840 Disabling a flaky expectation in vie_autotest_rtp_rtcp.cc. by stefan@webrtc.org · 11 years ago
34741c8 Include files from webrtc/.. paths in test/ by pbos@webrtc.org · 11 years ago
7f3f8bc Refactor jitter buffer to use separate lists for decodable and incomplete frames. by stefan@webrtc.org · 11 years ago
ead3c6d Fix bugs in DesktopRegion::IntersectWith() and DesktopRect::IntersectWith(). by sergeyu@chromium.org · 11 years ago
8665da8 Remove dead testRateControl.cc by pbos@webrtc.org · 11 years ago
a01f7f6 Removed dead testH263Parser.cc by pbos@webrtc.org · 11 years ago
c1f0eb2 Remove dead bitstreamTest.cc. by pbos@webrtc.org · 11 years ago
28556f5 Make sure GlxRenderer frees its resources. by pbos@webrtc.org · 11 years ago
c74c3c2 Adds integration test for RTX and fixes bugs found. by stefan@webrtc.org · 11 years ago
5c58f63 Fix regression where retransmission bitrate is no longer estimated. by stefan@webrtc.org · 11 years ago
d445d22 CreateEmptyFrame casts from size_t to int. by pbos@webrtc.org · 11 years ago
9b30348 FrameGenerator class for future fake capture device. by pbos@webrtc.org · 11 years ago
771cdcb Control new VideoEngine tests with gflags. by pbos@webrtc.org · 11 years ago
191c596 Adds print out of incoming resolution. by henrike@webrtc.org · 11 years ago
a7dc37d Log the type of recycled frames. by stefan@webrtc.org · 11 years ago
8c49c1e Log a message when a key frame packet is received by hclam@chromium.org · 11 years ago
46db413 Fix failing tests on 32 bit Linux. by solenberg@webrtc.org · 11 years ago
e46c8d3 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
561990f - Changed RemoteBitrateEstimator::IncomingPacket() to include a const WebRtcRTPHeader& and remove ssrc, rtp_timestamp. by solenberg@webrtc.org · 11 years ago
6ec2507 Disable WindowCapturer tests on OSX and Linux by sergeyu@chromium.org · 11 years ago
6ebfd34 Add direct_dependent_settings in common.gypi. by sergeyu@chromium.org · 11 years ago
5f8f112 Not to request to TURN server for local tests. Follow-up work to issue1197. by braveyao@webrtc.org · 11 years ago
106afff Roll libvpx to 196669. -pick up libvpx roll to 9981006d by marpan@webrtc.org · 11 years ago
2eaf98b Refactor VCM/Timing. No changes in functionality. by mikhal@webrtc.org · 11 years ago
3417eb4 Consolidate GetFrame and InsertPacket and move NACK list processing to after a packet has been successfully inserted. by stefan@webrtc.org · 11 years ago
956aa7e Include files from webrtc/.. paths in voice_engine/ by pbos@webrtc.org · 11 years ago
8a025e2 Make sure VoiceEngine tests only include one test framework. by pbos@webrtc.org · 11 years ago
d2541e8 Remove <iostream> usage from loopback.cc by pbos@webrtc.org · 11 years ago
375deb4 Suffix VcmCapturer's privates with underscore_ by pbos@webrtc.org · 11 years ago
0d540c3 Log timestamp of the frame when it's dropped from the render module by hclam@chromium.org · 11 years ago
69bb348 Log error in ViESender::SendRTCPPacket by hclam@chromium.org · 11 years ago
ac0ef48 Revert 4067 "libyuv roll to r698 for Core Media fourccs for OSX ..." by andrew@webrtc.org · 11 years ago
f9825e5 Revert 4000 "Reverting r3978" by andrew@webrtc.org · 11 years ago
225f2b8 Revert 4001 "Revert 3977" by andrew@webrtc.org · 11 years ago
c0352d5 Fix assertions in rtp_header_extension.h caused by not handling the AudioLevel extension. Added unit tests to do basic checks of the AudioLevel extension. by solenberg@webrtc.org · 11 years ago
e5794cb Recalibrate point sample expectation by fbarchard@google.com · 11 years ago
a58d729 libyuv roll to r698 for Core Media fourccs for OSX camtwist support and performance improvements in ARGB scaler. by fbarchard@google.com · 11 years ago
cb9cff0 Add functions to ViE API to enable/disable the absolute send time header extension. by solenberg@webrtc.org · 11 years ago
b10ccbe Window capturer implementation for Windows. by sergeyu@chromium.org · 11 years ago
5e2a1bb AppRTC: make requestTurn() failure non-fatal to call establishment. by fischman@webrtc.org · 11 years ago
8d6eb56 Avoid NPE crash on Android platforms that don't support getting preview framerate. by fischman@webrtc.org · 11 years ago
5a602d7 Enable WebRTC demo application on x86 Android by fischman@webrtc.org · 11 years ago
2163212 Include gflags properly and X11 include order in VideoEngine. by pbos@webrtc.org · 11 years ago
f5d4cb1 Include files from webrtc/.. paths in video_engine/ by pbos@webrtc.org · 11 years ago
9f557c1 Improve wraparound handling in the render time extrapolator. by stefan@webrtc.org · 11 years ago
14d7700 Moved command line parsing to internal tools and moved back the mic volume thingie. by phoglund@webrtc.org · 11 years ago
e874a8f Enable WebRTC demo application on x86 Android by fischman@webrtc.org · 11 years ago
8630cfe Guarding certain operations, e.g. bandwidth estimation, RTCP statistics update etc., not to be run on sync RTPS. by turaj@webrtc.org · 11 years ago
fe307e1 Add one unit test for NACKing a key frame by hclam@chromium.org · 11 years ago
b3e5acf Cleanup traces in WebRTC by hclam@chromium.org · 11 years ago
b9bb3d1 Avoid resetting encoder on identical settings. by pbos@webrtc.org · 11 years ago
890f609 Bugfix: VCM would report wrong sentBitrate by marpan@webrtc.org · 11 years ago
9919ad5 Formatted FEC stuff. by phoglund@webrtc.org · 11 years ago
5c1948d Moved force_volume_max to its own gyp file to avoid a circular dependency. by phoglund@webrtc.org · 11 years ago
61d3c55 Wrote a small portable tool for forcing the mic volume to 100%. by phoglund@webrtc.org · 11 years ago
29d5839 New VideoEngine API implementation on top of old one, first steps. by pbos@webrtc.org · 11 years ago
2038214 Log too long non-decodable duration events. by stefan@webrtc.org · 11 years ago
4dee309 Remove SetOverUseDetectorOptions and cleaned ViESharedData. by mflodman@webrtc.org · 11 years ago
7ebbea1 Add handling of the absolute send time header extension to the rtp_rtcp module. by solenberg@webrtc.org · 11 years ago
59a0667 Updated apprtc demo to interop with firefox. by vikasmarwaha@webrtc.org · 11 years ago
40298d4 Added webaudio-and-webtrc.html to the demos index.html. by vikasmarwaha@webrtc.org · 11 years ago
8c2e78b Roll chromium_revision 193311:199267 by fischman@webrtc.org · 11 years ago
6cfa390 Updating NACK RTX test by mikhal@webrtc.org · 11 years ago
cb20a5b VCM/JB: Bug fix in ExtractAndSetDecode BUG=1771 R=stefan@webrtc.org by mikhal@webrtc.org · 11 years ago
5add4ad RemoteBitrateEstimatorTest::TestRateIncreaseReordering sent in arrival timestamps in non monotonically increasing order. Fixed. by solenberg@webrtc.org · 11 years ago
c93b1d0 CoreAudio Win: release resources safely under certain rare circumstance in GTalkplugin by braveyao@webrtc.org · 11 years ago
e2a8006 Linux support for typing detection by niklas.enbom@webrtc.org · 11 years ago
4ce8389 Address sanitizer out of bounds read in iSAC by turaj@webrtc.org · 11 years ago
6bee05a Remove const for plain data types in common_video/ by pbos@webrtc.org · 11 years ago
29b2219 Adding a factory to remote bitrate estimator and allow it to be set via config. by andresp@webrtc.org · 11 years ago
1673481 Fixes a bug where the render buffer size (and indirectly the non-continuous duration) was computed incorrectly. by stefan@webrtc.org · 11 years ago
736c6f7 Fixed more perf expectations. by phoglund@webrtc.org · 11 years ago
80c7e3b Adjusted perf expectations for mac large tests. by phoglund@webrtc.org · 11 years ago