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

0d03514 Revert "Use libvpx's obj_int_extract and unpack_lib_posix to generate offset header file." by michaelbai@google.com · 10 years ago
d1e7fac Add stats of incoming frame delays for debugging bandwidth estimation. by jiayl@webrtc.org · 10 years ago
0e53607 Use libvpx's obj_int_extract and unpack_lib_posix to generate offset header file. by michaelbai@google.com · 10 years ago
c864f63 Add experiment: SkipEncodingUnusedStreams by sprang@webrtc.org · 10 years ago
1e04cfb Roll chromium_revision 245382:249215 by kjellander@webrtc.org · 10 years ago
9add2bb Fix WindowCapturerWin to unselect bitmap before destroying DC. by sergeyu@chromium.org · 10 years ago
0feb8fa Make VideoReceiveStream::GetStats() const. by sprang@webrtc.org · 10 years ago
c8ab721 Wire up statistics in video receive stream of new API by sprang@webrtc.org · 10 years ago
de93ce0 Plot the capacity of a trace-based delivery filter. by stefan@webrtc.org · 10 years ago
ee94ef9 Use system's cpu_features library by michaelbai@google.com · 10 years ago
df1f99f Add delay and send/receive throughput plots to BWE simulation. by stefan@webrtc.org · 10 years ago
aa42c87 Implementing replacement audio support in neteq_rtpplay by henrik.lundin@webrtc.org · 10 years ago
7993c94 Fixing a bug in DummyRTPpacket by henrik.lundin@webrtc.org · 10 years ago
03e6a88 Update AudioProcessing::Create docs. by andrew@webrtc.org · 10 years ago
4845e6e Fix a cursor capturing issue on Windows. by jiayl@webrtc.org · 10 years ago
abe3b85 Handle the invalid case of setting multiple stream_bitrates if there is only a single send stream registered. by stefan@webrtc.org · 10 years ago
05384c0 Revert "Disable the test: DtmfTest.ManualSuccessfullySendsIn/OutOfBandTelephoneEvents" by pbos@webrtc.org · 10 years ago
6b15b51 Fix locking in LoopBackTransport::StorePacket. by pbos@webrtc.org · 10 years ago
7387e23 Trivial rename of non-compile time consts. by andrew@webrtc.org · 10 years ago
a84ddcd Disable the test: DtmfTest.ManualSuccessfullySendsIn/OutOfBandTelephoneEvents by marpan@webrtc.org · 10 years ago
76dcaae Wire up feedback to VideoSender. by stefan@webrtc.org · 10 years ago
9c06023 Re-enabling audio processing tests by aluebs@webrtc.org · 10 years ago
87c8b86 Moved the new OnData interface to AudioTranport, and expose the AudioTransport pointer via voe_base by xians@webrtc.org · 10 years ago
1d23d5e Implement single monitor capture on Mac. by jiayl@webrtc.org · 10 years ago
f65e883 Fixing test name for NetEqPerformanceTest by henrik.lundin@webrtc.org · 10 years ago
8ef6548 Add configuration for cpu overuse detection to video send stream. by asapersson@webrtc.org · 10 years ago
748acc2 Add gyp_webrtc script to generate projects. by kjellander@webrtc.org · 10 years ago
e1c9902 Add BWE tools for parsing RTP files. by stefan@webrtc.org · 10 years ago
5e9b730 Fix the mouse cursor offset issue on Mac. by jiayl@webrtc.org · 10 years ago
c13a537 Move out typing detection to its own class. by henrikg@webrtc.org · 10 years ago
54ad929 Moves the display reconfiguration callback into a separate class, by jiayl@webrtc.org · 10 years ago
942ba53 Added new capture callback interface to pass the capture callback to a specific voe channel from libjingle webrtcvoiceengine.cc. by xians@webrtc.org · 10 years ago
2bb7ad5 Fix race when deleting video receive streams in Call. by solenberg@webrtc.org · 10 years ago
4a3208e Fix deadlock in video_receiver.cc. by stefan@webrtc.org · 10 years ago
d2f95a8 Connect webrtc::Config to WrappingBitrateEstimator by henrik.lundin@webrtc.org · 10 years ago
4d088c6 Add Config struct for experimental AGC. by andrew@webrtc.org · 10 years ago
7dc7514 Revert 5444 "Revert 5421 "Fix deadlock on register/unregister ob..." by mallinath@webrtc.org · 10 years ago
a6b9a3f Add clean test to NetEq perf test by henrik.lundin@webrtc.org · 10 years ago
cfbde68 VideoCaptureAndroid: stop preview in opposite order of starting. by fischman@webrtc.org · 10 years ago
9afb6cf Revert 5421 "Fix deadlock on register/unregister observer while ..." by mallinath@webrtc.org · 10 years ago
4406e5e Avoid potential dead lock in StreamStatisticianImpl by sprang@webrtc.org · 10 years ago
312a049 Race condition in RTPSender::UpdateRtpStats by sprang@webrtc.org · 10 years ago
0a29815 Drop early packets when not sending in TransportAdapter. by sprang@webrtc.org · 10 years ago
be35c57 Fix bug introduced during replace of list wrapper with std equivalents in r5378. by andresp@webrtc.org · 10 years ago
50d6975 Race in StreamStatisticianImpl::GetStatistics vs. ::IncomingPacket by sprang@webrtc.org · 10 years ago
3021162 Fix "field '_testNo' is uninitialized" warnings. by pbos@webrtc.org · 10 years ago
a370f24 Always initialize Trace in Call TraceDispatcher. by pbos@webrtc.org · 10 years ago
3fbe666 Add a Config parameter to AudioProcessing::Create(). by andrew@webrtc.org · 10 years ago
b2c6a45 Android, WebRTCDemo: fixes crash issue when pressing switch camera button on devices with only one camera. by henrike@webrtc.org · 10 years ago
01d06c8 Remove loopback setup in RtpRtcpImplTest. Changed to use two separate rtp/rtcp modules. by asapersson@webrtc.org · 10 years ago
ed592c7 Add new API (webrtc.gyp:webrtc) to merge_libs.gyp. by pbos@webrtc.org · 10 years ago
ef6a602 Add trace-based delivery filter to BWE test framework. by stefan@webrtc.org · 10 years ago
c71929d Wire up RTX in VideoReceiveStream. by pbos@webrtc.org · 10 years ago
630939f Fix deadlock on register/unregister observer while there is a an going callback. by andresp@webrtc.org · 10 years ago
d242006 Fix array declarations in aec_rdft.h. by andrew@webrtc.org · 10 years ago
e822c84 Set NACKed packet to -1 in TestNackRetransmission. by pbos@webrtc.org · 10 years ago
7d99cd4 Add callbacks for receive channel RTP statistics by sprang@webrtc.org · 10 years ago
888b839 Android, fixes crash on devices with only front cameras. by henrike@webrtc.org · 10 years ago
0584274 Output logs to stderr from voe_cmd_test by default. by andrew@webrtc.org · 10 years ago
83dd51d Android example apps: fixes issue where useful failure information was suppressed. by henrike@webrtc.org · 10 years ago
83a37c8 Potential dead lock in receive statistics by sprang@webrtc.org · 10 years ago
57e8ba9 Fix for libtalkmobile build error bug=b/12549061 by elham@webrtc.org · 10 years ago
7238373 Removes script for generating supplement.gypi also adds git ignore for tools/gn. by henrike@webrtc.org · 10 years ago
5b08052 Set up receiver RTX config using a std::map. by pbos@webrtc.org · 10 years ago
b4263e0 Add configuration and test for extended RTCP reference time reports to new video api. by asapersson@webrtc.org · 10 years ago
4be4e8a Android, OpenSlDemo: moved to webrtc/examples/android/opensl_loopback by henrike@webrtc.org · 10 years ago
28bd30a Implement screen enumeration and individual screen capturing for Windows. by jiayl@webrtc.org · 10 years ago
aea6053 Android, OpenSlDemo: fixes issue where app would crash as soon as the application is started. by henrike@webrtc.org · 10 years ago
89b5ae8 Android, WebRTCDemo: fix issue where changing remote IP was not working properly. by henrike@webrtc.org · 10 years ago
0902491 Add full path to headers by aluebs@webrtc.org · 10 years ago
a7c7e2c Adds back set_sample_rate_hz() when Init is called in recordings. by bjornv@webrtc.org · 10 years ago
556423f MIPS optimizations for NS audio processing module by andrew@webrtc.org · 10 years ago
4c20797 Fix crash in MouseCursor::CopyOf() by sergeyu@chromium.org · 10 years ago
96961e5 Exclude protoc objects from merge_libs.py. by andrew@webrtc.org · 10 years ago
390a5ae Update needed to MockScreenCapturer after new methods addition to webrtc::ScreenCapturer. by mallinath@webrtc.org · 10 years ago
a2c2654 Extends the ScreenCapturer interface for individual display screen cast. by jiayl@webrtc.org · 10 years ago
fba4f1c Roll Chromium 238260 -> 243863 by wjia@webrtc.org · 10 years ago
84350a9 Remove empty VideoCodecGeneric struct. by pbos@webrtc.org · 10 years ago
37fb66d Changing to using factory methods for some classes in NetEq by henrik.lundin@webrtc.org · 10 years ago
b3ff385 Temporarily disabling some more audio processing tests. by aluebs@webrtc.org · 10 years ago
59fdf2d Fix MouseCursorMonitorMac to return correct hotspot position. by sergeyu@chromium.org · 10 years ago
4b5d36e Removes the remaining uses of the list wrapper class and the list wrapper class. by henrike@webrtc.org · 10 years ago
8c03c4c WebRTCDemo: fix out-of-bounds array read. by fischman@webrtc.org · 10 years ago
eed1f11 Updated Webrtc version to 3.49 by elham@webrtc.org · 10 years ago
083049f Removes usage of ListWrapper from several files. by henrike@webrtc.org · 10 years ago
ad584b6 Revert "Activate ACM test for Android in modules_tests." (rev5364). by andresp@webrtc.org · 10 years ago
572cc28 Temporarily disabling audio processing tests. by aluebs@webrtc.org · 10 years ago
2904b71 Increasing simulation time for NetEqPerformanceTest by henrik.lundin@webrtc.org · 10 years ago
b7c1e03 Enables robust delay validation in AEC delay logging. by bjornv@webrtc.org · 10 years ago
22470b5 Minor voice engine improvements around AGC. by andrew@webrtc.org · 11 years ago
f3a2ef3 Android: Fixes crash when exiting WebRTCDemo. by henrike@webrtc.org · 11 years ago
3ab10f9 Activate ACM test for Android in modules_tests. by turaj@webrtc.org · 11 years ago
75e7da3 Permitting double start/stopping of streams. by pbos@webrtc.org · 11 years ago
5b1467d Adding NetEq performance test to webrtc_perf_tests by henrik.lundin@webrtc.org · 11 years ago
8d4f9ca Delay Estimator: Adds unittests for robust validation. by bjornv@webrtc.org · 11 years ago
201049c Fixing lint errors in NetEq4 by henrik.lundin@webrtc.org · 11 years ago
8b6867b Make code simpler on VCMEncodedCallback. by andresp@webrtc.org · 11 years ago
88ece35 Isolate register post encode callback in video coding module to simplify code and critical sections. by andresp@webrtc.org · 11 years ago
093b960 Isolate debug recording from video sender into a thread safe small class. by andresp@webrtc.org · 11 years ago
0b9d7ce Add another test case for AST/TOF switching. by solenberg@webrtc.org · 11 years ago