Log - 1df6cc71692eb1bc2512ce53a291bfd78a0df9e0 - fp2-dev/platform/external/chromium_org/third_party/webrtc

1df6cc7 Reorganize test targets in WebRTC by kjellander@webrtc.org · 11 years ago
f92d9ad Build all java files into jar for each module on Android by fischman@webrtc.org · 11 years ago
3da595e Allow the screen capturer to capture oversized cursors and cursors without alpha channel (Windows). by alexeypa@chromium.org · 11 years ago
a83e538 Landing binary cursor image files to be used in a follow up CL. by alexeypa@chromium.org · 11 years ago
ccc21d2 Updated WebRTC version to 3.33 by elham@webrtc.org · 11 years ago
091c4f8 Making no NACK mode work again in VideoEngine. by mflodman@webrtc.org · 11 years ago
7556bbe RW lock access to ssrc maps in VideoCall. by pbos@webrtc.org · 11 years ago
a6e8ec3 Add back the WEBRTC_DIRECT_TRACE flag. by solenberg@webrtc.org · 11 years ago
ffce2b1 AudioDeviceAndroidOpenSLES: NULL variables might be referenced in StopPlayout() by braveyao@webrtc.org · 11 years ago
6e5b871 Revert some variables to uint32_t to fix compile errors on Mac gcc. by andrew@webrtc.org · 11 years ago
4477bd5 Allow audio devices with up to 64 channels on Mac. by andrew@webrtc.org · 11 years ago
b42bf4b Fixed Rtp/Rtcp tests by pwestin@webrtc.org · 11 years ago
9c49814 Fix relative path to .gitignore and other minor changes. by andrew@webrtc.org · 11 years ago
a052973 Removing functionality for inserting pre-encoded frames instead of raw by mflodman@webrtc.org · 11 years ago
306e331 Add script for appending entries to .gitignore. by andrew@webrtc.org · 11 years ago
4dca856 Fix size_t to int conversion error on Win64. by andrew@webrtc.org · 11 years ago
1632b97 Remove fake screen capturer because it's not used anywhere. by sergeyu@chromium.org · 11 years ago
f22cc80 Fix for STL vector function data not available. by pwestin@webrtc.org · 11 years ago
f94ddea Connect ACM with RTP module for audio NACK. by pwestin@webrtc.org · 11 years ago
a629133 Nack for audio. by turaj@webrtc.org · 11 years ago
a1e84f1 Fix leaks in DesktopRegion by sergeyu@chromium.org · 11 years ago
018870d Implement DetectNumberOfCores on Android and make it consistent on Linux and Android by fischman@webrtc.org · 11 years ago
4673a99 Wire up Nack for Voe by pwestin@webrtc.org · 11 years ago
767ca95 Fix init list for VideoSendStream::Config::Rtp. by pbos@webrtc.org · 11 years ago
2c343fc Stats+Config moved into VideoSend/ReceiveStreams. by pbos@webrtc.org · 11 years ago
561fe8b Merge webrtc_utility_unittests into modules_unittests. by kjellander@webrtc.org · 11 years ago
14035f1 Restore relative include paths to libyuv. by andrew@webrtc.org · 11 years ago
fd5d808 Issue 1847, memcopy is wrong and unnecessary, it is sufficient to store the pointer before clearing the instance, and write back the pointer. by turaj@webrtc.org · 11 years ago
07e10ab resolve b9050210. Avoid pushing sync packet before any packet received. Do not turn on AV-sync if initial delay is zero. by turaj@webrtc.org · 11 years ago
910a3c6 Move screen capturers from chromium to webrtc. by sergeyu@chromium.org · 11 years ago
71fe9ac Refactor padding and rtp header functionality. by stefan@webrtc.org · 11 years ago
4dc727b Update the remote bitrate estimator before passing the packet to the RTP module. by stefan@webrtc.org · 11 years ago
84119ff Remove XvRenderer. by pbos@webrtc.org · 11 years ago
5eb0408 Fix build error introduced with r4168. by stefan@webrtc.org · 11 years ago
39278fb Add support for padding in pacer. by stefan@webrtc.org · 11 years ago
1ad8620 Include files from webrtc/.. paths in common_video/ by pbos@webrtc.org · 11 years ago
7ddad3e Include files from webrtc/.. paths in tools/ by pbos@webrtc.org · 11 years ago
76070bf Disable neteq_unittests on Win x64 in code. by kjellander@webrtc.org · 11 years ago
aa28a43 Disable audio_decoder_unittests on Win x64 in code. by kjellander@webrtc.org · 11 years ago
9d86b92 Disable audio_coding_unittests on Win x64 in code. by kjellander@webrtc.org · 11 years ago
9d59eaf Do not hold a lock when calling VCMReceiveCallback::FrameToRender. by fischman@webrtc.org · 11 years ago
1bcdae5 Optimized DesktopRegion implementation. by sergeyu@chromium.org · 11 years ago
26c4698 Removed unused class members to enable clang=1 android build. by fischman@webrtc.org · 11 years ago
52f0bce Setting SSRC in vie_loopback_test by mikhal@webrtc.org · 11 years ago
dff82e7 Fix error in mixing test for supported sample rates. by andrew@webrtc.org · 11 years ago
5276226 Change SetRTPAudioLevelIndicationStatus to ignore the id in the case of disabling. by wu@webrtc.org · 11 years ago
8f515b1 Replace the old resampler with SincResampler in the voice engine signal path. by andrew@webrtc.org · 11 years ago
a654283 Remove ancient and unused CNG test. by andrew@webrtc.org · 11 years ago
ad09c1a Add dummy audio NACK APIs by niklas.enbom@webrtc.org · 11 years ago
ebe0069 Prevent excessive logging in jitter buffer by hclam@chromium.org · 11 years ago
0bc7fa0 Revert 4104 "Refactor jitter buffer to use separate lists for de..." by tnakamura@webrtc.org · 11 years ago
8d4927d Revert 4127 "Switch frame list implementation to std::map." by tnakamura@webrtc.org · 11 years ago
e113280 MIPS optimizations for the following functions: by andrew@webrtc.org · 11 years ago
a6ba491 VCM/Timing: Setting clear names to members & methods by mikhal@webrtc.org · 11 years ago
7d7a7bb Fixes the frameRate stats by grouping the frames by timestamp. by jiayl@webrtc.org · 11 years ago
c98be72 Use int for FPS instead of size_t. by pbos@webrtc.org · 11 years ago
6c0fab5 Include files from webrtc/.. paths in rtp_rtcp/ by pbos@webrtc.org · 11 years ago
064e3f2 Correctly set SSRCs for extra send RTP modules. by stefan@webrtc.org · 11 years ago
043fef6 Remove assert for aborting FrameGeneratorCapturer. by pbos@webrtc.org · 11 years ago
c1cf6bd Fake VideoCapturer based on FrameGenerator by pbos@webrtc.org · 11 years ago
8687453 Fix a return value mismatch introduced in r4129. by stefan@webrtc.org · 11 years ago
b3afc18 Remove #pragma once by pbos@webrtc.org · 11 years ago
028ec72 Breaking out RTP header parsing from the RTP module. by stefan@webrtc.org · 11 years ago
266d5f6 Break video_engine/new_include/common.h into smaller parts. by pbos@webrtc.org · 11 years ago
f378d7c Switch frame list implementation to std::map. by stefan@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
9162bbc Add comment about test_packet_masks_metrics. by marpan@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
ef32e73 Include "gtest/gtest.h", not by full path, on WEBRTC_ANDROID_PLATFORM_BUILD 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
464f9be Include files from webrtc/.. paths in audio_conference_mixer/ by pbos@webrtc.org · 11 years ago
382c8b3 Include files from webrtc/.. paths in audio_processing/ by pbos@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
a960785 Include files from webrtc/.. paths in system_wrappers/ by pbos@webrtc.org · 11 years ago
6c9726a Include files from webrtc/.. paths in test/channel_transport/ by pbos@webrtc.org · 11 years ago
519d7cf Include files from webrtc/.. paths in video_processing/ by pbos@webrtc.org · 11 years ago
d8f7f53 Include files from webrtc/.. paths in remote_bitrate_estimator/ by pbos@webrtc.org · 11 years ago
abf0cd8 Include files from webrtc/.. paths in common_audio/ by pbos@webrtc.org · 11 years ago
104218e Disabling a flaky expectation in vie_autotest_rtp_rtcp.cc. by stefan@webrtc.org · 11 years ago
56041ab Include files from webrtc/.. paths in test/ by pbos@webrtc.org · 11 years ago
9e0d3ec Refactor jitter buffer to use separate lists for decodable and incomplete frames. by stefan@webrtc.org · 11 years ago
3f2091a Fix bugs in DesktopRegion::IntersectWith() and DesktopRect::IntersectWith(). by sergeyu@chromium.org · 11 years ago
087f8c6 Remove dead testRateControl.cc by pbos@webrtc.org · 11 years ago
2f30cee Removed dead testH263Parser.cc by pbos@webrtc.org · 11 years ago
2ea6127 Remove dead bitstreamTest.cc. by pbos@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
53e452d Log the type of recycled frames. by stefan@webrtc.org · 11 years ago
6bd2847 Log a message when a key frame packet is received by hclam@chromium.org · 11 years ago
1286255 Fix failing tests on 32 bit Linux. by solenberg@webrtc.org · 11 years ago