Log - 9c8f391af3144ba2f749b44f6b4b84d8612bc2ac - fp2-dev/platform/external/chromium_org/third_party/webrtc

9c8f391 Removes usage of ListWrapper from several files. by henrike@webrtc.org · 11 years ago
4861e69 Revert "Activate ACM test for Android in modules_tests." (rev5364). by andresp@webrtc.org · 11 years ago
087bcfa Temporarily disabling audio processing tests. by aluebs@webrtc.org · 11 years ago
0782a57 Increasing simulation time for NetEqPerformanceTest by henrik.lundin@webrtc.org · 11 years ago
62ca1c6 Enables robust delay validation in AEC delay logging. by bjornv@webrtc.org · 11 years ago
457e101 Minor voice engine improvements around AGC. by andrew@webrtc.org · 11 years ago
842d07a Android: Fixes crash when exiting WebRTCDemo. by henrike@webrtc.org · 11 years ago
303c52b Activate ACM test for Android in modules_tests. by turaj@webrtc.org · 11 years ago
df9f099 Permitting double start/stopping of streams. by pbos@webrtc.org · 11 years ago
1cc1166 Adding NetEq performance test to webrtc_perf_tests by henrik.lundin@webrtc.org · 11 years ago
15ba589 Delay Estimator: Adds unittests for robust validation. by bjornv@webrtc.org · 11 years ago
9fd4d83 Fixing lint errors in NetEq4 by henrik.lundin@webrtc.org · 11 years ago
401e046 Make code simpler on VCMEncodedCallback. by andresp@webrtc.org · 11 years ago
080eeee Isolate register post encode callback in video coding module to simplify code and critical sections. by andresp@webrtc.org · 11 years ago
da08e77 Isolate debug recording from video sender into a thread safe small class. by andresp@webrtc.org · 11 years ago
c92ae91 Add another test case for AST/TOF switching. by solenberg@webrtc.org · 11 years ago
11dddc0 Delay Estimator: Converts a constant into a configurable parameter. by bjornv@webrtc.org · 11 years ago
d3c0b85 Init to 16 kHz in the fixed-point profile. by andrew@webrtc.org · 11 years ago
c5eb922 Ensure capture_levels_ is sized correctly at init time. by andrew@webrtc.org · 11 years ago
859b462 Now printing less output from compare_videos.py. by phoglund@webrtc.org · 11 years ago
926e88a Remove the requirement to call set_sample_rate_hz and friends. by andrew@webrtc.org · 11 years ago
f45e5b2 Remove outdated DestroyVideoSendStream comment. by pbos@webrtc.org · 11 years ago
ca72300 Wire up statistics in video send stream of new video engine api by sprang@webrtc.org · 11 years ago
59a26d2 Delay Estimator: robust_validation should be stored over a reset by bjornv@webrtc.org · 11 years ago
efdfe16 Add include guards to forward_error_correction_internal.h by braveyao@webrtc.org · 11 years ago
9a7cb02 Fix the include guard in transmit_mixer.h by braveyao@webrtc.org · 11 years ago
a3ae4d1 Fix the include guard in transmit_mixer.h by braveyao@webrtc.org · 11 years ago
9456776 Android build: make it quiet on success and not overly noisy on failure. by fischman@webrtc.org · 11 years ago
748625e Fix the android clang bot for compiling with thread annotations. by andresp@webrtc.org · 11 years ago
7e4053c Add thread_annotations for clang targets. by andresp@webrtc.org · 11 years ago
d3f0617 If the configured start bitrate is higher than the configures max by mflodman@webrtc.org · 11 years ago
4a185e9 Race condition in ViECapturer::RegisterObserver by sprang@webrtc.org · 11 years ago
e83367b Update WebRTC to version 3.48 by tnakamura@webrtc.org · 11 years ago
acc2e43 Add callbacks for receive channel RTCP statistics. by sprang@webrtc.org · 11 years ago
e9c9d54 Refactoring MediaOptimization so it can easily be turned into a thread-safe class. by andresp@webrtc.org · 11 years ago
cd117d2 Integrate fake_network_pipe into direct_transport. by stefan@webrtc.org · 11 years ago
0d8474d Remove metrics_unittests by kjellander@webrtc.org · 11 years ago
ef1f6c3 Remove media_file from VideoEngine dependencies. by pbos@webrtc.org · 11 years ago
2a4595a cpplint cleaning new API and its implementation files. by mflodman@webrtc.org · 11 years ago
b409d78 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
f22f12a Speeding up CallTest.ReceivesAndRetransmitsNack and removed the random packet loss. by mflodman@webrtc.org · 11 years ago
cc407fd Make MouseCursor mutable by sergeyu@chromium.org · 11 years ago
32a0f69 audio_processing_unittest: unbreak clang compilation. by fischman@webrtc.org · 11 years ago
6b89cba JNI Audio: remove dead members. by fischman@webrtc.org · 11 years ago
32705ce Revert "Make MouseCursor mutable" by sergeyu@chromium.org · 11 years ago
0c7efa2 Make MouseCursor mutable by sergeyu@chromium.org · 11 years ago
4db3691 Stop transport in test SuspendBelowMinBitrate. by pbos@webrtc.org · 11 years ago
6f43aa7 Added method for getting default module state and protect agains a by mflodman@webrtc.org · 11 years ago
620d9e5 Modify video_render/ to allow a single old frame. by pbos@webrtc.org · 11 years ago
4494516 Delete capturers after destroying streams in test. by pbos@webrtc.org · 11 years ago
f3aed2f Revert 5285 "Revert 5228 "Use the RTT from RtcpRttStats class if..." by asapersson@webrtc.org · 11 years ago
b06cca3 Simplification of histogram normalization in delay estimator. by bjornv@webrtc.org · 11 years ago
39139dc Revert r5294 to re-roll r5293. by pbos@webrtc.org · 11 years ago
2ec8a62 Adds robust validation functionality to the delay estimator by bjornv@webrtc.org · 11 years ago
beb643b Incorrect iterator++ in ModuleRtpRtcpImpl::RegisterVideoBitrateObserver by sprang@webrtc.org · 11 years ago
0af1d21 Revert 5293 "Auto instantiate RBE depending on whether AST or TO..." by turaj@webrtc.org · 11 years ago
ee867fa Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream. by solenberg@webrtc.org · 11 years ago
b8dc2e2 Disabled tests on Android. The issue 2723 is filed to investigate the reason for tests failing. by turaj@webrtc.org · 11 years ago
f34e39b Measure pacer queue size based on when packets are inserted rather than captured. by stefan@webrtc.org · 11 years ago
b50a841 Fix jitter buffer delay estimate. by turaj@webrtc.org · 11 years ago
7f0519e Update talk to 58174641 together with http://review.webrtc.org/4319005/. by wu@webrtc.org · 11 years ago
ab6ccbc Adding REMB to receive stream configuration, the send side will always by mflodman@webrtc.org · 11 years ago
9b3321f Revert 5228 "Use the RTT from RtcpRttStats class if provided whe..." by asapersson@webrtc.org · 11 years ago
9d9f138 Merge metrics_unittests into video_engine_tests. by pbos@webrtc.org · 11 years ago
d1dd1d2 Move realtime tests to webrtc_perf_tests. by pbos@webrtc.org · 11 years ago
0e4512b Callback for send bitrate estimates - new roll by sprang@webrtc.org · 11 years ago
e4d538a Make sure channels in the same call are in the same channel group. by mflodman@webrtc.org · 11 years ago
e6dc4ff Making RemoteRateControl::min_configured_bit_rate_ configurable by henrik.lundin@webrtc.org · 11 years ago
3a4fc4b Update talk to 58127566 together with by wu@webrtc.org · 11 years ago
b9cf1de ACM 2 compatibility with ACM 1. by turaj@webrtc.org · 11 years ago
9b3d2bf Revert 5274 "Update talk to 58113193 together with https://webrt..." by wu@webrtc.org · 11 years ago
cde78d6 Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/. by wu@webrtc.org · 11 years ago
a4670a1 Complete rewrite of demo application. by henrike@webrtc.org · 11 years ago
0ceb51f Remove overloaded CpuOveruseMeasure function. by asapersson@webrtc.org · 11 years ago
2f70422 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
f49d16c Fix common_video_unittests in apk_tests.gyp. by pbos@webrtc.org · 11 years ago
7123a80 Add SwapFrame() to VideoSendStreamInput. by pbos@webrtc.org · 11 years ago
66e84b0 Revert 5259 "Callback for send bitrate estimates" by sprang@webrtc.org · 11 years ago
894dab9 Roll chromium_revision 232627:238260 by kjellander@webrtc.org · 11 years ago
f1d22d4 Callback for send bitrate estimates by sprang@webrtc.org · 11 years ago
72b0d40 Removed unnecessary Pulse init from VoE startup. by fischman@webrtc.org · 11 years ago
e8ca064 Correctly define OVERRIDE when building with g++ 4.7 and C++11 support by andrew@webrtc.org · 11 years ago
090f37f Improve VideoSendStreamTest::MaxPacketSize by sprang@webrtc.org · 11 years ago
ba8b32c Change uses of the obsolete armv7 setting to arm_version==7. by kjellander@webrtc.org · 11 years ago
934be30 Fix compilation errors on Fedora 20. by fischman@webrtc.org · 11 years ago
4adc7ad Ensure WEBRTC_MODULE_UTILITY_VIDEO is undefined for enable_video==0. by andrew@webrtc.org · 11 years ago
e681a01 Add shape in DesktopFrame. by sergeyu@chromium.org · 11 years ago
e8dd108 Add new method to MockAudioProcessing. by andrew@webrtc.org · 11 years ago
e4d591a Revert 5229 "Make VideoSendStreamTest::MaxPacketSize test a whol..." by andrew@webrtc.org · 11 years ago
c8bd975 Allow opening an AEC dump from an existing file handle. by henrikg@webrtc.org · 11 years ago
9e40eba Stop video capturers in multi-stream test. by pbos@webrtc.org · 11 years ago
5b23ce6 Fix use of uninitialized memory in RtpSenderTest::StreamDataCountersCallbacks by sprang@webrtc.org · 11 years ago
ed8c496 Fraction lost statistics not being reported by sprang@webrtc.org · 11 years ago
6dccf13 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
2de68d6 Use the RTT from RtcpRttStats class if provided when sending/receiving NACK. by asapersson@webrtc.org · 11 years ago
cf5c552 Add callbacks for send channel rtp statistics by sprang@webrtc.org · 11 years ago
f0d9b20 Remove CallTest dependency on voice_engine/test/. by pbos@webrtc.org · 11 years ago
8db148e Add API to query video engine for the send-side delay. by stefan@webrtc.org · 11 years ago
adc238a Fixing the android build by henrik.lundin@webrtc.org · 11 years ago
3d70641 Move implementation files out of the webrtc/ root. by pbos@webrtc.org · 11 years ago