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f59fce91dac3b8b6b1e67b340a367e8bdba03f06
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224933c
Add callbacks for receive channel RTP statistics
by sprang@webrtc.org
· 11 years ago
bd389d5
Android, fixes crash on devices with only front cameras.
by henrike@webrtc.org
· 11 years ago
ccee3c3
Output logs to stderr from voe_cmd_test by default.
by andrew@webrtc.org
· 11 years ago
e412bb6
Android example apps: fixes issue where useful failure information was suppressed.
by henrike@webrtc.org
· 11 years ago
7262655
Potential dead lock in receive statistics
by sprang@webrtc.org
· 11 years ago
c85c797
Fix for libtalkmobile build error bug=b/12549061
by elham@webrtc.org
· 11 years ago
e1b9880
Removes script for generating supplement.gypi also adds git ignore for tools/gn.
by henrike@webrtc.org
· 11 years ago
2752bb1
Set up receiver RTX config using a std::map.
by pbos@webrtc.org
· 11 years ago
64339f0
Add configuration and test for extended RTCP reference time reports to new video api.
by asapersson@webrtc.org
· 11 years ago
b6a78e5
Android, OpenSlDemo: moved to webrtc/examples/android/opensl_loopback
by henrike@webrtc.org
· 11 years ago
5426f84
Implement screen enumeration and individual screen capturing for Windows.
by jiayl@webrtc.org
· 11 years ago
ac4c7ea
Android, OpenSlDemo: fixes issue where app would crash as soon as the application is started.
by henrike@webrtc.org
· 11 years ago
6b84a14
Android, WebRTCDemo: fix issue where changing remote IP was not working properly.
by henrike@webrtc.org
· 11 years ago
1240a3d
Add full path to headers
by aluebs@webrtc.org
· 11 years ago
d8163b6
Adds back set_sample_rate_hz() when Init is called in recordings.
by bjornv@webrtc.org
· 11 years ago
f180890
MIPS optimizations for NS audio processing module
by andrew@webrtc.org
· 11 years ago
08ef082
Fix crash in MouseCursor::CopyOf()
by sergeyu@chromium.org
· 11 years ago
0a71d11
Exclude protoc objects from merge_libs.py.
by andrew@webrtc.org
· 11 years ago
df9f867
Update needed to MockScreenCapturer after new methods addition to webrtc::ScreenCapturer.
by mallinath@webrtc.org
· 11 years ago
7f1a443
Extends the ScreenCapturer interface for individual display screen cast.
by jiayl@webrtc.org
· 11 years ago
c1792c5
Roll Chromium 238260 -> 243863
by wjia@webrtc.org
· 11 years ago
ef3b24e
Remove empty VideoCodecGeneric struct.
by pbos@webrtc.org
· 11 years ago
e6882db
Changing to using factory methods for some classes in NetEq
by henrik.lundin@webrtc.org
· 11 years ago
90c6679
Temporarily disabling some more audio processing tests.
by aluebs@webrtc.org
· 11 years ago
282a4d8
Fix MouseCursorMonitorMac to return correct hotspot position.
by sergeyu@chromium.org
· 11 years ago
1c27316
Removes the remaining uses of the list wrapper class and the list wrapper class.
by henrike@webrtc.org
· 11 years ago
8340614
WebRTCDemo: fix out-of-bounds array read.
by fischman@webrtc.org
· 11 years ago
b95f445
Updated Webrtc version to 3.49
by elham@webrtc.org
· 11 years ago
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
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