Log - 10b8135a78daa30cadacecc8931c272329939ccf - fp2-dev/platform/external/chromium_org/third_party/webrtc

10b8135 Re-enable libjingle_peerconnection_java_unittest since bug 2952 is fixed. by fischman@webrtc.org · 10 years ago
ecee063 Adds APIs for reporting pacer queuing delay. by jiayl@webrtc.org · 10 years ago
873f357 Remove ProcessingComponent's dependence on AudioProcessingImpl. by andrew@webrtc.org · 10 years ago
f66d92c Call PrintWindow for the first time of capturing to capture the window frames correctly. by jiayl@webrtc.org · 10 years ago
40cb3ed Clean up CPU detection defines in SincResampler a little. by andrew@webrtc.org · 10 years ago
44ef0e6 Invalidate the whole screen when the frame size is changed. by jiayl@webrtc.org · 10 years ago
7a38402 Use scoped_ptr<T[]> in SincResampler to avoid .get()[] weirdness. by andrew@webrtc.org · 10 years ago
a51b238 Add SetConfig method to FakeNetworkPipe and to DirectTransport by henrik.lundin@webrtc.org · 10 years ago
44f8467 Add experimental noise suppression flag to audioproc test by aluebs@webrtc.org · 10 years ago
ea59061 Missing include in experiments.h by sprang@webrtc.org · 10 years ago
3dc57d9 Split the implementation of VP8Encoder|Decoder::Create into a seperated file by wu@webrtc.org · 10 years ago
c0d56c0 Fix to get total number of sent and received rtcp packets. by asapersson@webrtc.org · 10 years ago
5f1ea5b AviRecorder is missing a critical section. by braveyao@webrtc.org · 10 years ago
ee03b3b Disable libjingle_peerconnection_java_unittest by kjellander@webrtc.org · 10 years ago
69a0eca Removed unused mock methods in audio_processing by bjornv@webrtc.org · 10 years ago
65bf249 Add RTCP packet class. Adds packet types: sr, rr, bye, fir. by asapersson@webrtc.org · 10 years ago
f59fce9 MIPS optimizations for AEC audio processing module by andrew@webrtc.org · 10 years ago
23c8d6b Updated WebRTC version to 3.50 TBR= wu@webrtc.org by elham@webrtc.org · 10 years ago
f8722d5 Add an AlignedFreeDeleter and remove scoped_ptr_malloc. by andrew@webrtc.org · 10 years ago
27c1536 Minor improvement in RoundToInt16 implementation. by turaj@webrtc.org · 10 years ago
072bab2 Modified overuse detection thresholds. by asapersson@webrtc.org · 10 years ago
04e6137 Removing a variable that was never read by henrik.lundin@webrtc.org · 10 years ago
a8498d9 ifdef the alsa code based on macro USE_X11 by fbarchard@google.com · 10 years ago
cb06a6b Fix the break caused by r5579. by turaj@webrtc.org · 10 years ago
554bd44 Resolves memcheck issue in AudioCodingModuleTest. The issue is coditional jumnp based on uninitialized variable. by turaj@webrtc.org · 10 years ago
4b2ef8b Make WindowCapturerLinux handling window resize events. by jiayl@webrtc.org · 10 years ago
1d68dc1 Added architecture for compiling under chrome NaCl. by andresp@webrtc.org · 10 years ago
0657a86 This CL separate all ACM tests with new and old implementation of ACM and NetEq. The reason is to debug an issue with failure on Android try bots. We need to see if the error only occurs with the new ACM/NetEq, or if it is a flakiness that affects both. by tina.legrand@webrtc.org · 10 years ago
2fa9f7e Add RTCP packet type counter (for getting statistics such as sent/received NACK and FIR). by asapersson@webrtc.org · 10 years ago
fa28e37 Removes VoERTP_RTCP::InsertExtraRTPPacket. by henrika@webrtc.org · 10 years ago
4e266ff Fix DesktopAndCursorComposer not to crash by sergeyu@chromium.org · 10 years ago
8513671 Move the volume quantization workaround from VoE to AGC. by andrew@webrtc.org · 10 years ago
c8529ab Remove obsolete voe_unit_test. by solenberg@webrtc.org · 10 years ago
387b944 Don't print a warning if RTPPacketHistory::SetStorePacketStatus is called by mflodman@webrtc.org · 10 years ago
aeb2e9e Remove unnecessary warnings. by turaj@webrtc.org · 10 years ago
ae50521 Remove external encryption API for VoE. by solenberg@webrtc.org · 10 years ago
3f3e951 Incorrect overhead calculation when using FEC + RTP extension headers. by sprang@webrtc.org · 10 years ago
15e3511 Reset estimate if no frame has been seen for a certain time (to avoid large jitter if stop sending). by asapersson@webrtc.org · 10 years ago
e3da97c Misc small nits in NetEq by henrik.lundin@webrtc.org · 10 years ago
6f8b051 AudioProcessing is not a Module. by andrew@webrtc.org · 10 years ago
cd15790 Refactoring common_audio/signal_processing: Removed two macros used by isac only. by bjornv@webrtc.org · 10 years ago
46b22d8 Adding a critical section missing in r5543. by stefan@webrtc.org · 10 years ago
0f5010d Initialize output_will_be_muted_. by andrew@webrtc.org · 10 years ago
8e98655 Increase overuse and normal use thresholds for Mac. by asapersson@webrtc.org · 10 years ago
8cb4c8d Fixes a race when writing to send_padding_. by stefan@webrtc.org · 10 years ago
5d5e87d Small refactoring of NetEq unittest for CNG with clock drift by henrik.lundin@webrtc.org · 10 years ago
f4f1d1a Add a method to inform AudioProcessing that its output will be muted. by andrew@webrtc.org · 10 years ago
96b5dfa Change the type of propagation delta from int64 to int. by jiayl@webrtc.org · 10 years ago
9e3cb7b Initialize key_pressed_. by andrew@webrtc.org · 10 years ago
6ec403d Add a keypress field to the audioproc debug proto. by andrew@webrtc.org · 10 years ago
6cfc58d Set pacing bitrates in SetEncoder. by pbos@webrtc.org · 10 years ago
0fd5775 Remove unused and not working voe_extended_test. by solenberg@webrtc.org · 10 years ago
48a5cdb Reduce mixing threshold in test to avoid flakiness. by andrew@webrtc.org · 10 years ago
247df83 Add an interface for accepting keypress signals to AudioProcessing. by andrew@webrtc.org · 10 years ago
4112a51 Rename merged webrtc lib to libwebrtc_merged.a. by andrew@webrtc.org · 10 years ago
e2d2804 Remove "Too long processing time of Incoming frame" logspam. by fischman@webrtc.org · 10 years ago
ff986f4 Add boundary checking to supress gcc 4.8.3 warning. by turaj@webrtc.org · 10 years ago
ddbd31e Remove ViE external encryption API. by solenberg@webrtc.org · 10 years ago
e08d28e Use libvpx's obj_int_extract and unpack_lib_posix to generate offset header file. by michaelbai@google.com · 10 years ago
dd1d6ce Restore mixing integration tests. by andrew@webrtc.org · 10 years ago
89a0796 Revert "Use libvpx's obj_int_extract and unpack_lib_posix to generate offset header file." by michaelbai@google.com · 10 years ago
a68379b Add stats of incoming frame delays for debugging bandwidth estimation. by jiayl@webrtc.org · 10 years ago
bac08b3 Use libvpx's obj_int_extract and unpack_lib_posix to generate offset header file. by michaelbai@google.com · 10 years ago
85835a0 Add experiment: SkipEncodingUnusedStreams by sprang@webrtc.org · 10 years ago
c0b1926 Roll chromium_revision 245382:249215 by kjellander@webrtc.org · 10 years ago
992076c Fix WindowCapturerWin to unselect bitmap before destroying DC. by sergeyu@chromium.org · 10 years ago
f2c28a0 Make VideoReceiveStream::GetStats() const. by sprang@webrtc.org · 10 years ago
fa7c4c4 Wire up statistics in video receive stream of new API by sprang@webrtc.org · 10 years ago
8a431ef Plot the capacity of a trace-based delivery filter. by stefan@webrtc.org · 10 years ago
74ffc7b Use system's cpu_features library by michaelbai@google.com · 10 years ago
94c5692 Add delay and send/receive throughput plots to BWE simulation. by stefan@webrtc.org · 10 years ago
618154f Implementing replacement audio support in neteq_rtpplay by henrik.lundin@webrtc.org · 10 years ago
395e1b4 Fixing a bug in DummyRTPpacket by henrik.lundin@webrtc.org · 10 years ago
680d3ca Update AudioProcessing::Create docs. by andrew@webrtc.org · 10 years ago
1ca2c1f Fix a cursor capturing issue on Windows. by jiayl@webrtc.org · 10 years ago
55367d5 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
1eba384 Revert "Disable the test: DtmfTest.ManualSuccessfullySendsIn/OutOfBandTelephoneEvents" by pbos@webrtc.org · 10 years ago
4f41016 Fix locking in LoopBackTransport::StorePacket. by pbos@webrtc.org · 10 years ago
3634228 Trivial rename of non-compile time consts. by andrew@webrtc.org · 10 years ago
0a7d406 Disable the test: DtmfTest.ManualSuccessfullySendsIn/OutOfBandTelephoneEvents by marpan@webrtc.org · 10 years ago
cc8de94 Wire up feedback to VideoSender. by stefan@webrtc.org · 10 years ago
54a9a32 Re-enabling audio processing tests by aluebs@webrtc.org · 10 years ago
910910a Moved the new OnData interface to AudioTranport, and expose the AudioTransport pointer via voe_base by xians@webrtc.org · 10 years ago
2b38fc1 Implement single monitor capture on Mac. by jiayl@webrtc.org · 10 years ago
622a139 Fixing test name for NetEqPerformanceTest by henrik.lundin@webrtc.org · 10 years ago
4b1817f Add configuration for cpu overuse detection to video send stream. by asapersson@webrtc.org · 10 years ago
aaac959 Add gyp_webrtc script to generate projects. by kjellander@webrtc.org · 10 years ago
098ffb2 Add BWE tools for parsing RTP files. by stefan@webrtc.org · 10 years ago
28429ea Fix the mouse cursor offset issue on Mac. by jiayl@webrtc.org · 10 years ago
25bec2a Move out typing detection to its own class. by henrikg@webrtc.org · 10 years ago
c4fa5fa Moves the display reconfiguration callback into a separate class, by jiayl@webrtc.org · 10 years ago
4f23307 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
fdb30d1 Fix race when deleting video receive streams in Call. by solenberg@webrtc.org · 10 years ago
50afcf1 Fix deadlock in video_receiver.cc. by stefan@webrtc.org · 11 years ago
49e9e15 Connect webrtc::Config to WrappingBitrateEstimator by henrik.lundin@webrtc.org · 11 years ago
9d5a547 Add Config struct for experimental AGC. by andrew@webrtc.org · 11 years ago
a1e140d Revert 5444 "Revert 5421 "Fix deadlock on register/unregister ob..." by mallinath@webrtc.org · 11 years ago
78fae4b Add clean test to NetEq perf test by henrik.lundin@webrtc.org · 11 years ago
76d028d VideoCaptureAndroid: stop preview in opposite order of starting. by fischman@webrtc.org · 11 years ago
c091c50 Revert 5421 "Fix deadlock on register/unregister observer while ..." by mallinath@webrtc.org · 11 years ago