Log - 0d03514a1a95cd83126ba35102c4277555603577 - fp2-dev/platform/external/chromium_org/third_party/webrtc

9b125e1 Delay Estimator: Converts a constant into a configurable parameter. by bjornv@webrtc.org · 11 years ago
f3b2148 Init to 16 kHz in the fixed-point profile. by andrew@webrtc.org · 11 years ago
48b9892 Ensure capture_levels_ is sized correctly at init time. by andrew@webrtc.org · 11 years ago
2c358e2 Now printing less output from compare_videos.py. by phoglund@webrtc.org · 11 years ago
e95dc25 Remove the requirement to call set_sample_rate_hz and friends. by andrew@webrtc.org · 11 years ago
b3b6049 Remove outdated DestroyVideoSendStream comment. by pbos@webrtc.org · 11 years ago
49812e6 Wire up statistics in video send stream of new video engine api by sprang@webrtc.org · 11 years ago
c902d88 Delay Estimator: robust_validation should be stored over a reset by bjornv@webrtc.org · 11 years ago
e66b5bc Add include guards to forward_error_correction_internal.h by braveyao@webrtc.org · 11 years ago
1a6b274 Fix the include guard in transmit_mixer.h by braveyao@webrtc.org · 11 years ago
5d7992f Fix the include guard in transmit_mixer.h by braveyao@webrtc.org · 11 years ago
29975da Android build: make it quiet on success and not overly noisy on failure. by fischman@webrtc.org · 11 years ago
fba1476 Fix the android clang bot for compiling with thread annotations. by andresp@webrtc.org · 11 years ago
c9faf10 Add thread_annotations for clang targets. by andresp@webrtc.org · 11 years ago
a9a7327 If the configured start bitrate is higher than the configures max by mflodman@webrtc.org · 11 years ago
9662535 Race condition in ViECapturer::RegisterObserver by sprang@webrtc.org · 11 years ago
91cebfc Update WebRTC to version 3.48 by tnakamura@webrtc.org · 11 years ago
4f1f5fa Add callbacks for receive channel RTCP statistics. by sprang@webrtc.org · 11 years ago
9edcdb0 Refactoring MediaOptimization so it can easily be turned into a thread-safe class. by andresp@webrtc.org · 11 years ago
aacdb9f Integrate fake_network_pipe into direct_transport. by stefan@webrtc.org · 11 years ago
a07c56f Remove metrics_unittests by kjellander@webrtc.org · 11 years ago
5596ac6 Remove media_file from VideoEngine dependencies. by pbos@webrtc.org · 11 years ago
5e0cbcf cpplint cleaning new API and its implementation files. by mflodman@webrtc.org · 11 years ago
6c172c5 Reduced execution time for CallTest::ReceivesPliAndRecovers, by dropping only one packet and made it predictable by removing rand(). by mflodman@webrtc.org · 11 years ago
5e252ac Speeding up CallTest.ReceivesAndRetransmitsNack and removed the random packet loss. by mflodman@webrtc.org · 11 years ago
eb9ce11 Make MouseCursor mutable by sergeyu@chromium.org · 11 years ago
8d14e06 audio_processing_unittest: unbreak clang compilation. by fischman@webrtc.org · 11 years ago
d138166 JNI Audio: remove dead members. by fischman@webrtc.org · 11 years ago
432e574 Revert "Make MouseCursor mutable" by sergeyu@chromium.org · 11 years ago
865be14 Make MouseCursor mutable by sergeyu@chromium.org · 11 years ago
5d13922 Stop transport in test SuspendBelowMinBitrate. by pbos@webrtc.org · 11 years ago
202d38d Added method for getting default module state and protect agains a by mflodman@webrtc.org · 11 years ago
d60137f Modify video_render/ to allow a single old frame. by pbos@webrtc.org · 11 years ago
471354f Delete capturers after destroying streams in test. by pbos@webrtc.org · 11 years ago
5041831 Revert 5285 "Revert 5228 "Use the RTT from RtcpRttStats class if..." by asapersson@webrtc.org · 11 years ago
0443f6c Simplification of histogram normalization in delay estimator. by bjornv@webrtc.org · 11 years ago
46f7288 Revert r5294 to re-roll r5293. by pbos@webrtc.org · 11 years ago
1b3b8cb Adds robust validation functionality to the delay estimator by bjornv@webrtc.org · 11 years ago
1eb1008 Incorrect iterator++ in ModuleRtpRtcpImpl::RegisterVideoBitrateObserver by sprang@webrtc.org · 11 years ago
c5a5713 Revert 5293 "Auto instantiate RBE depending on whether AST or TO..." by turaj@webrtc.org · 11 years ago
3bbc91e Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream. by solenberg@webrtc.org · 11 years ago
7d7e63d Disabled tests on Android. The issue 2723 is filed to investigate the reason for tests failing. by turaj@webrtc.org · 11 years ago
afceaca Measure pacer queue size based on when packets are inserted rather than captured. by stefan@webrtc.org · 11 years ago
f1b92fd Fix jitter buffer delay estimate. by turaj@webrtc.org · 11 years ago
79d6daf Update talk to 58174641 together with http://review.webrtc.org/4319005/. by wu@webrtc.org · 11 years ago
7ff4089 Adding REMB to receive stream configuration, the send side will always by mflodman@webrtc.org · 11 years ago
1465cef Revert 5228 "Use the RTT from RtcpRttStats class if provided whe..." by asapersson@webrtc.org · 11 years ago
f64791e Merge metrics_unittests into video_engine_tests. by pbos@webrtc.org · 11 years ago
f94ccd4 Move realtime tests to webrtc_perf_tests. by pbos@webrtc.org · 11 years ago
b70db6d Callback for send bitrate estimates - new roll by sprang@webrtc.org · 11 years ago
dadfc9e Make sure channels in the same call are in the same channel group. by mflodman@webrtc.org · 11 years ago
c49a3fa Making RemoteRateControl::min_configured_bit_rate_ configurable by henrik.lundin@webrtc.org · 11 years ago
efeb8ce Update talk to 58127566 together with by wu@webrtc.org · 11 years ago
da4d59e ACM 2 compatibility with ACM 1. by turaj@webrtc.org · 11 years ago
12553ad Revert 5274 "Update talk to 58113193 together with https://webrt..." by wu@webrtc.org · 11 years ago
984bee2 Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/. by wu@webrtc.org · 11 years ago
a48c91d Complete rewrite of demo application. by henrike@webrtc.org · 11 years ago
4fcb2f5 Remove overloaded CpuOveruseMeasure function. by asapersson@webrtc.org · 11 years ago
27f0841 Enables mixing and matching Java and native audio. It is used for getting best of both worlds capabilities (AEC and low latency). by henrike@webrtc.org · 11 years ago
6bf67db Fix common_video_unittests in apk_tests.gyp. by pbos@webrtc.org · 11 years ago
c33d37c Add SwapFrame() to VideoSendStreamInput. by pbos@webrtc.org · 11 years ago
ffea4ce Revert 5259 "Callback for send bitrate estimates" by sprang@webrtc.org · 11 years ago
ca63ad9 Roll chromium_revision 232627:238260 by kjellander@webrtc.org · 11 years ago
1430bc3 Callback for send bitrate estimates by sprang@webrtc.org · 11 years ago
b46e68d Removed unnecessary Pulse init from VoE startup. by fischman@webrtc.org · 11 years ago
aa9e768 Correctly define OVERRIDE when building with g++ 4.7 and C++11 support by andrew@webrtc.org · 11 years ago
b589c65 Improve VideoSendStreamTest::MaxPacketSize by sprang@webrtc.org · 11 years ago
2cafda4 Change uses of the obsolete armv7 setting to arm_version==7. by kjellander@webrtc.org · 11 years ago
5424c16 Fix compilation errors on Fedora 20. by fischman@webrtc.org · 11 years ago
241103f Ensure WEBRTC_MODULE_UTILITY_VIDEO is undefined for enable_video==0. by andrew@webrtc.org · 11 years ago
935c8c7 Add shape in DesktopFrame. by sergeyu@chromium.org · 11 years ago
8beba83 Add new method to MockAudioProcessing. by andrew@webrtc.org · 11 years ago
e388f9e Revert 5229 "Make VideoSendStreamTest::MaxPacketSize test a whol..." by andrew@webrtc.org · 11 years ago
7b72264 Allow opening an AEC dump from an existing file handle. by henrikg@webrtc.org · 11 years ago
4383539 Stop video capturers in multi-stream test. by pbos@webrtc.org · 11 years ago
539670c Fix use of uninitialized memory in RtpSenderTest::StreamDataCountersCallbacks by sprang@webrtc.org · 11 years ago
cebd1d7 Fraction lost statistics not being reported by sprang@webrtc.org · 11 years ago
532b8f7 Make VideoSendStreamTest::MaxPacketSize test a whole range of frame sizes, to make sure all corner cases are covered. by sprang@webrtc.org · 11 years ago
0b16527 Use the RTT from RtcpRttStats class if provided when sending/receiving NACK. by asapersson@webrtc.org · 11 years ago
b113981 Add callbacks for send channel rtp statistics by sprang@webrtc.org · 11 years ago
3adf058 Remove CallTest dependency on voice_engine/test/. by pbos@webrtc.org · 11 years ago
ee234be Add API to query video engine for the send-side delay. by stefan@webrtc.org · 11 years ago
a4fae33 Fixing the android build by henrik.lundin@webrtc.org · 11 years ago
d05597a Move implementation files out of the webrtc/ root. by pbos@webrtc.org · 11 years ago
db04941 Remove default implementations for SuspendBelowMinBitrate by henrik.lundin@webrtc.org · 11 years ago
d964bf5 Fix bug where fraction_lost is always set to 0 when getting received RTCP statistics. by stefan@webrtc.org · 11 years ago
9b30fd3 Add callbacks for send channel rtcp statistics by sprang@webrtc.org · 11 years ago
f03505e Make RTPSender::SendPadData public. by stefan@webrtc.org · 11 years ago
6d1a71b Remove unused ThreadData struct. by andrew@webrtc.org · 11 years ago
c7d7363 Remove the long disabled WEBRTC_SVNREVISION define. by andrew@webrtc.org · 11 years ago
1fc02eb Removing DropDeltaAfterKey functionality which is unused. by andresp@webrtc.org · 11 years ago
5fdd10a Add send frame rate statistics callback by sprang@webrtc.org · 11 years ago
f1630b1 Added a delay measurement, measures the time between an incoming captured frame until the frame is being processed. Measures the delay per second. by asapersson@webrtc.org · 11 years ago
c0d6b2d Fixes a crash in fullstack tests introduced with r5209. by stefan@webrtc.org · 11 years ago
a5be230 Small fixes to plot_neteq_delay.m by henrik.lundin@webrtc.org · 11 years ago
47f0c41 Adds support for sending redundant payloads over RTX. by stefan@webrtc.org · 11 years ago
76c6ac4 Fix a typo in neteq.gypi by henrik.lundin@webrtc.org · 11 years ago
3f0b77f Compile-out functions only used by the bit-exact test. by andrew@webrtc.org · 11 years ago
38aa817 Don't HANDLE_EINTR(close). Use IGNORE_EINTR(close). by fischman@webrtc.org · 11 years ago
af00735 Add baseline generation/verification to BWE test framework. by solenberg@webrtc.org · 11 years ago