Log - 222efdce1bb10a28966523a868cefb5eb0c08742 - fp2-dev/platform/external/chromium_org/third_party/webrtc

a80d94b Change SetRTPAudioLevelIndicationStatus to ignore the id in the case of disabling. by wu@webrtc.org · 11 years ago
92bfbbd Replace the old resampler with SincResampler in the voice engine signal path. by andrew@webrtc.org · 11 years ago
379dce7 Remove ancient and unused CNG test. by andrew@webrtc.org · 11 years ago
2753b76 Add dummy audio NACK APIs by niklas.enbom@webrtc.org · 11 years ago
40954f0 Prevent excessive logging in jitter buffer by hclam@chromium.org · 11 years ago
8bf7456 Revert 4104 "Refactor jitter buffer to use separate lists for de..." by tnakamura@webrtc.org · 11 years ago
884ff69 Revert 4127 "Switch frame list implementation to std::map." by tnakamura@webrtc.org · 11 years ago
1aa406c MIPS optimizations for the following functions: by andrew@webrtc.org · 11 years ago
e477574 VCM/Timing: Setting clear names to members & methods by mikhal@webrtc.org · 11 years ago
9f62516 Fixes the frameRate stats by grouping the frames by timestamp. by jiayl@webrtc.org · 11 years ago
e3e4615 Use int for FPS instead of size_t. by pbos@webrtc.org · 11 years ago
cbd78ae Include files from webrtc/.. paths in rtp_rtcp/ by pbos@webrtc.org · 11 years ago
54b6ebc Correctly set SSRCs for extra send RTP modules. by stefan@webrtc.org · 11 years ago
23e3f44 Remove assert for aborting FrameGeneratorCapturer. by pbos@webrtc.org · 11 years ago
c1506a2 Fake VideoCapturer based on FrameGenerator by pbos@webrtc.org · 11 years ago
4e5f983 Fix a return value mismatch introduced in r4129. by stefan@webrtc.org · 11 years ago
50a4d9f Remove #pragma once by pbos@webrtc.org · 11 years ago
6696fba Breaking out RTP header parsing from the RTP module. by stefan@webrtc.org · 11 years ago
4988d94 Break video_engine/new_include/common.h into smaller parts. by pbos@webrtc.org · 11 years ago
afe587e Switch frame list implementation to std::map. by stefan@webrtc.org · 11 years ago
5221d1c Rename voice_engine_core -> voice_engine and move targets to voice_engine.gyp. by andrew@webrtc.org · 11 years ago
2dcf742 Add comment about test_packet_masks_metrics. by marpan@webrtc.org · 11 years ago
3990df2 Updated WebRTC version to 3.32 TBR=wu@webrtc.org by elham@webrtc.org · 11 years ago
e3b52e6 Don't return an estimated receive BW for channels not receiving video. by mflodman@webrtc.org · 11 years ago
bb6bef5 Include gflags with "gflags/gflags.h" instead of <> by pbos@webrtc.org · 11 years ago
8838f68 Include "gtest/gtest.h", not by full path, on WEBRTC_ANDROID_PLATFORM_BUILD by pbos@webrtc.org · 11 years ago
39784c4 Improve vie_autotest_rtp_rtcp by reenabling important tests and reducing flakiness. by stefan@webrtc.org · 11 years ago
0c836bf Include files from webrtc/.. paths in audio_conference_mixer/ by pbos@webrtc.org · 11 years ago
9fb1613 Include files from webrtc/.. paths in audio_processing/ by pbos@webrtc.org · 11 years ago
b2d1a40 Default constructors for new VideoEngine structs. by pbos@webrtc.org · 11 years ago
5437a2c 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
f40e9b6 - Created RemoteBitrateEstimator wrapper for use internally in (ViE) ChannelGroup. by solenberg@webrtc.org · 11 years ago
a93cbbf Adding Mac test renderer, some test refactoring and made cpplint pass. by mflodman@webrtc.org · 11 years ago
c6d6fed Include files from webrtc/.. paths in system_wrappers/ by pbos@webrtc.org · 11 years ago
96001c8 Include files from webrtc/.. paths in test/channel_transport/ by pbos@webrtc.org · 11 years ago
c4e10b8 Include files from webrtc/.. paths in video_processing/ by pbos@webrtc.org · 11 years ago
d5d709e Include files from webrtc/.. paths in remote_bitrate_estimator/ by pbos@webrtc.org · 11 years ago
f24ac59 Include files from webrtc/.. paths in common_audio/ by pbos@webrtc.org · 11 years ago
460e172 Disabling a flaky expectation in vie_autotest_rtp_rtcp.cc. by stefan@webrtc.org · 11 years ago
20a5c46 Include files from webrtc/.. paths in test/ by pbos@webrtc.org · 11 years ago
bb771bb Refactor jitter buffer to use separate lists for decodable and incomplete frames. by stefan@webrtc.org · 11 years ago
60142de Fix bugs in DesktopRegion::IntersectWith() and DesktopRect::IntersectWith(). by sergeyu@chromium.org · 11 years ago
294b789 Remove dead testRateControl.cc by pbos@webrtc.org · 11 years ago
68c4886 Removed dead testH263Parser.cc by pbos@webrtc.org · 11 years ago
aef3e5a Remove dead bitstreamTest.cc. by pbos@webrtc.org · 11 years ago
ad6cade Make sure GlxRenderer frees its resources. by pbos@webrtc.org · 11 years ago
eef4fd5 Adds integration test for RTX and fixes bugs found. by stefan@webrtc.org · 11 years ago
8f1d1a9 Fix regression where retransmission bitrate is no longer estimated. by stefan@webrtc.org · 11 years ago
42e1fe1 CreateEmptyFrame casts from size_t to int. by pbos@webrtc.org · 11 years ago
08f3ca9 FrameGenerator class for future fake capture device. by pbos@webrtc.org · 11 years ago
074eb20 Control new VideoEngine tests with gflags. by pbos@webrtc.org · 11 years ago
b59962f Adds print out of incoming resolution. by henrike@webrtc.org · 11 years ago
2e37985 Log the type of recycled frames. by stefan@webrtc.org · 11 years ago
9de67da Log a message when a key frame packet is received by hclam@chromium.org · 11 years ago
de0b5fa Fix failing tests on 32 bit Linux. by solenberg@webrtc.org · 11 years ago
d557734 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
b7716d8 - Changed RemoteBitrateEstimator::IncomingPacket() to include a const WebRtcRTPHeader& and remove ssrc, rtp_timestamp. by solenberg@webrtc.org · 11 years ago
0204219 Disable WindowCapturer tests on OSX and Linux by sergeyu@chromium.org · 11 years ago
ced13a5 Add direct_dependent_settings in common.gypi. by sergeyu@chromium.org · 11 years ago
389bb40 Refactor VCM/Timing. No changes in functionality. by mikhal@webrtc.org · 11 years ago
3740808 Consolidate GetFrame and InsertPacket and move NACK list processing to after a packet has been successfully inserted. by stefan@webrtc.org · 11 years ago
471ae72 Include files from webrtc/.. paths in voice_engine/ by pbos@webrtc.org · 11 years ago
8510750 Make sure VoiceEngine tests only include one test framework. by pbos@webrtc.org · 11 years ago
141a00c Remove <iostream> usage from loopback.cc by pbos@webrtc.org · 11 years ago
6169712 Suffix VcmCapturer's privates with underscore_ by pbos@webrtc.org · 11 years ago
15bdfdf Log timestamp of the frame when it's dropped from the render module by hclam@chromium.org · 11 years ago
cca5086 Log error in ViESender::SendRTCPPacket by hclam@chromium.org · 11 years ago
366d158 Revert 4000 "Reverting r3978" by andrew@webrtc.org · 11 years ago
9038990 Revert 4001 "Revert 3977" by andrew@webrtc.org · 11 years ago
0e15695 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
6595271 Recalibrate point sample expectation by fbarchard@google.com · 11 years ago
453f9c0 Add functions to ViE API to enable/disable the absolute send time header extension. by solenberg@webrtc.org · 11 years ago
e032f9f Window capturer implementation for Windows. by sergeyu@chromium.org · 11 years ago
b9e5732 Avoid NPE crash on Android platforms that don't support getting preview framerate. by fischman@webrtc.org · 11 years ago
9ea8c99 Include gflags properly and X11 include order in VideoEngine. by pbos@webrtc.org · 11 years ago
281cff8 Include files from webrtc/.. paths in video_engine/ by pbos@webrtc.org · 11 years ago
c7979e0 Improve wraparound handling in the render time extrapolator. by stefan@webrtc.org · 11 years ago
de93f2c Moved command line parsing to internal tools and moved back the mic volume thingie. by phoglund@webrtc.org · 11 years ago
e8dc588 Enable WebRTC demo application on x86 Android by fischman@webrtc.org · 11 years ago
8787048 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
673f50d Add one unit test for NACKing a key frame by hclam@chromium.org · 11 years ago
5a22c40 Cleanup traces in WebRTC by hclam@chromium.org · 11 years ago
f6cb4b6 Avoid resetting encoder on identical settings. by pbos@webrtc.org · 11 years ago
d434891 Bugfix: VCM would report wrong sentBitrate by marpan@webrtc.org · 11 years ago
4efbd60 Formatted FEC stuff. by phoglund@webrtc.org · 11 years ago
fefc490 Moved force_volume_max to its own gyp file to avoid a circular dependency. by phoglund@webrtc.org · 11 years ago
a4f0d20 Wrote a small portable tool for forcing the mic volume to 100%. by phoglund@webrtc.org · 11 years ago
2a9108f New VideoEngine API implementation on top of old one, first steps. by pbos@webrtc.org · 11 years ago
069f626 Log too long non-decodable duration events. by stefan@webrtc.org · 11 years ago
7645e4d Remove SetOverUseDetectorOptions and cleaned ViESharedData. by mflodman@webrtc.org · 11 years ago
a0b0025 Add handling of the absolute send time header extension to the rtp_rtcp module. by solenberg@webrtc.org · 11 years ago
b960975 Updating NACK RTX test by mikhal@webrtc.org · 11 years ago
9c94651 VCM/JB: Bug fix in ExtractAndSetDecode BUG=1771 R=stefan@webrtc.org by mikhal@webrtc.org · 11 years ago
8129578 RemoteBitrateEstimatorTest::TestRateIncreaseReordering sent in arrival timestamps in non monotonically increasing order. Fixed. by solenberg@webrtc.org · 11 years ago
88ba0a5 CoreAudio Win: release resources safely under certain rare circumstances. by braveyao@webrtc.org · 11 years ago
32b4e2f Linux support for typing detection by niklas.enbom@webrtc.org · 11 years ago
f6e0404 Address sanitizer out of bounds read in iSAC by turaj@webrtc.org · 11 years ago
2cc529f Remove const for plain data types in common_video/ by pbos@webrtc.org · 11 years ago
0425392 Adding a factory to remote bitrate estimator and allow it to be set via config. by andresp@webrtc.org · 11 years ago
5ba433c Fixes a bug where the render buffer size (and indirectly the non-continuous duration) was computed incorrectly. by stefan@webrtc.org · 11 years ago