- 3046b84 Adding new data files for audio classifier unit testing on Android try bots by jan.skoglund@webrtc.org · 10 years ago
- d3d6bce (Auto)update libjingle 62865357-> 62871616 by henrike@webrtc.org · 10 years ago
- d32797f Add a float interface to PushSincResampler. by andrew@webrtc.org · 10 years ago
- bc206ea iOS video_render: omit no-op setNeedsDisplay by fischman@webrtc.org · 10 years ago
- f792d17 AppRTCDemo(iOS): video support; part 1 of 2: webrtc/. by fischman@webrtc.org · 10 years ago
- 0537634 (Auto)update libjingle 62713454-> 62865357 by henrike@webrtc.org · 10 years ago
- 4a47be0 Disable CallTest.ReceivesAndRetransmitsNack for TSan by kjellander@webrtc.org · 10 years ago
- 36b6221 Adding a link to issue by henrik.lundin@webrtc.org · 10 years ago
- 6b0cbcb Roll chromium_revision 249215:255773 by kjellander@webrtc.org · 10 years ago
- 9b5f4d8 Fix build breakage introduce with r5665. by stefan@webrtc.org · 10 years ago
- f9e7c9d Add option to bwe_rtp_to_text to output arrival times only in nanoseconds. by stefan@webrtc.org · 10 years ago
- a01daf0 RTCPeerConnectionTest(objc): deflake by ignoring ICECompleted. by fischman@webrtc.org · 10 years ago
- 13320ea PeerConnectionTest(objc): expect ICE Completed state post 61460797-p10 by fischman@webrtc.org · 10 years ago
- 7811469 Roll libvpx 251850:254609 by marpan@webrtc.org · 10 years ago
- 11aab0e Populate VoiceReceiverInfo::delay_estimate_ms, jitter_buffer_ms, and jitter_buffer_preferred_ms to getStats. by jiayl@webrtc.org · 10 years ago
- 64e0405 Avoid crash in ViEEncoder::DeRegisterExternalEncoder(). by fischman@webrtc.org · 10 years ago
- cc08e3f Moves WEBRTC_POSIX define from header file to gyp-settings. by henrike@webrtc.org · 10 years ago
- 3ecc162 Remove std:: prefixes from C functions in webrtc/. by pbos@webrtc.org · 10 years ago
- 371243d Remove std:: prefixes from C functions in talk/. by pbos@webrtc.org · 10 years ago
- 46509c8 adding FEC support to WebRTC Opus wrapper and tests. by minyue@webrtc.org · 10 years ago
- 0454688 This CL is to add Opus complexity knob and to test it. by minyue@webrtc.org · 10 years ago
- ebdb0e3 Help to land 7969005 on behalf of solenberg. The review and try is done in 7969005. by wu@webrtc.org · 10 years ago
- 79047f9 (Auto)update libjingle 62691533-> 62713454 by henrike@webrtc.org · 10 years ago
- 2d213e4 (Auto)update libjingle 62550414-> 62691533 by henrike@webrtc.org · 10 years ago
- f714e7f Remove abs() use in PseudoTcp::process. by pbos@webrtc.org · 10 years ago
- 4584697 Fixes a bug in the simulation framework where the time offset is accumulating as the packet trace is repeated, causing increasingly large gaps with no packets being transmitted. by stefan@webrtc.org · 10 years ago
- ed865b5 NetEq4: Changing the behavior of playout_timestamp_ update by henrik.lundin@webrtc.org · 10 years ago
- 60ad5fd Potential deadlock in VideoSendStreamTest::ProducesStats by sprang@webrtc.org · 10 years ago
- 998cb8f Use DISABLE_ instead of commenting out tests by henrik.lundin@webrtc.org · 10 years ago
- 845862f Adding a new ramp-up-down-up test by henrik.lundin@webrtc.org · 10 years ago
- a0d11da Remove upper check for number of cores in VCM, I didn't find any good reasons for checking this. by mflodman@webrtc.org · 10 years ago
- cf85f1c Reorganize libjingle path variables. by kjellander@webrtc.org · 10 years ago
- 9f4d212 adding sha1 files for audio classifier test by jan.skoglund@webrtc.org · 10 years ago
- 3e0b60f Switch to correct interpretation of int and float input data in audio_processing_unittest by bjornv@webrtc.org · 10 years ago
- 17e4064 Add a deinterleaved float interface to AudioProcessing. by andrew@webrtc.org · 10 years ago
- b90991d Update libjingle 62472237->62550414 by henrike@webrtc.org · 10 years ago
- 7bd4a27 VideoCaptureAndroid: don't deliver frames after stopCapture(). by fischman@webrtc.org · 10 years ago
- be50ab6 Including algorithm header to avoid VS2013 breakage by henrik.lundin@webrtc.org · 10 years ago
- 52e898d Add .bin and .rx files to svn:ignore in resources by kjellander@webrtc.org · 10 years ago
- 24dae94 Add pthatcher@webrtc.org to talk/OWNERS. by pbos@webrtc.org · 10 years ago
- a25a92e Add third_party dependencies to svn:ignore by kjellander@webrtc.org · 10 years ago
- db41b4d Remove the deprecated GetStats method from PeerConnectionInterface. by jiayl@webrtc.org · 10 years ago
- 80bbf4c Enable test SSLStreamAdapterTestDTLS.TestDTLSConnectWithSmallMtu since it does not fail anymore. by jiayl@webrtc.org · 10 years ago
- 40b3b68 Update libjingle 62364298->62472237 by henrike@webrtc.org · 10 years ago
- 1bbfb57 Rollback of r5629 "(Auto)update libjingle 62364298-> 62368661". by henrike@webrtc.org · 10 years ago
- 0117d1c Fix compilation errors under clang 3.5. by pbos@webrtc.org · 10 years ago
- 31413dc (Auto)update libjingle 62364298-> 62368661 by henrike@webrtc.org · 10 years ago
- 10adbef Exclude /out* instead of just /out from pylint checks. by fischman@webrtc.org · 10 years ago
- 2bd5944 Re-enable libjingle_peerconnection_java_unittest since bug 2952 is fixed. by fischman@webrtc.org · 10 years ago
- d3dc424 Remove posting of ICE messages from WebRTCSession in PeerConnection to signaling thread. by mallinath@webrtc.org · 10 years ago
- bcfc167 AppRTCDemo(android): don't send local SDP until it's set. by fischman@webrtc.org · 10 years ago
- b898ce9 Revert of r5622 "disable unit tests" as it should be fixed in r5623. by henrike@webrtc.org · 10 years ago
- b8395eb (Auto)update libjingle 62293974-> 62364298 by henrike@webrtc.org · 10 years ago
- eec3843 TSAN only disable of two of libjingle's tests for atomic ops as they are failing for TSAN-bot. by henrike@webrtc.org · 10 years ago
- 9fd8d87 Adds APIs for reporting pacer queuing delay. by jiayl@webrtc.org · 10 years ago
- 56e4a05 Remove ProcessingComponent's dependence on AudioProcessingImpl. by andrew@webrtc.org · 10 years ago
- 806768a (Auto)update libjingle 62281784-> 62293974 by henrike@webrtc.org · 10 years ago
- 704bf9e (Auto)update libjingle 62063505-> 62278774 by henrike@webrtc.org · 10 years ago
- f0fc72f Call PrintWindow for the first time of capturing to capture the window frames correctly. by jiayl@webrtc.org · 10 years ago
- 00073aa Clean up CPU detection defines in SincResampler a little. by andrew@webrtc.org · 10 years ago
- 0231e80 Invalidate the whole screen when the frame size is changed. by jiayl@webrtc.org · 10 years ago
- 2038920 Use scoped_ptr<T[]> in SincResampler to avoid .get()[] weirdness. by andrew@webrtc.org · 10 years ago
- c0e9aeb Add SetConfig method to FakeNetworkPipe and to DirectTransport by henrik.lundin@webrtc.org · 10 years ago
- eaadeca iOS, AppRTCDemo: Fixes exception due to JSON for ice using "urls" instead of "url", which is introduced by r5599. by braveyao@webrtc.org · 10 years ago
- 90173e1 Roll libvpx 248011:251850 by marpan@webrtc.org · 10 years ago
- bc1d224 Add experimental noise suppression flag to audioproc test by aluebs@webrtc.org · 10 years ago
- 050892a Missing include in experiments.h by sprang@webrtc.org · 10 years ago
- 7f52a6e Split the implementation of VP8Encoder|Decoder::Create into a seperated file by wu@webrtc.org · 10 years ago
- 79a1cff Android, AppRTCDemo: Fixes java exception due to JSON for ice using "urls" instead of "url". by henrike@webrtc.org · 10 years ago
- bf88ecc Added turn-prober.sh: a super-simple prober for TURN servers & candidates. by fischman@webrtc.org · 10 years ago
- 78ea3d5 Check pcConfig (which can be null) before use. by wu@webrtc.org · 10 years ago
- 91cbaa4 (Auto)update libjingle 61966318-> 62063505 by henrike@webrtc.org · 10 years ago
- 23caa2d Fix to get total number of sent and received rtcp packets. by asapersson@webrtc.org · 10 years ago
- 4f0801b AviRecorder is missing a critical section. by braveyao@webrtc.org · 10 years ago
- bc0470f AppRTC Sample: Switch AppRTC to use RTCIceServer.urls. by braveyao@webrtc.org · 10 years ago
- 55fcd71 Disable libjingle_peerconnection_java_unittest by kjellander@webrtc.org · 10 years ago
- 33af96c Removed unused mock methods in audio_processing by bjornv@webrtc.org · 10 years ago
- d43aa9d Update libjingle 61901702->61966318 by henrike@webrtc.org · 10 years ago
- a7b9818 Unrevert 5590 "description"(=(Auto)update libjingle 61834300->61901702). by henrike@webrtc.org · 10 years ago
- 125a66a Memory and Tsan tests: Turn off the new-ACM tests by tina.legrand@webrtc.org · 10 years ago
- ef22151 Revert 5590 "description" by xians@webrtc.org · 10 years ago
- 0f2809a Add RTCP packet class. Adds packet types: sr, rr, bye, fir. by asapersson@webrtc.org · 10 years ago
- c0907ef MIPS optimizations for AEC audio processing module by andrew@webrtc.org · 10 years ago
- 2643805 description by henrike@webrtc.org · 10 years ago
- 3f170dd Updated WebRTC version to 3.50 TBR= wu@webrtc.org by elham@webrtc.org · 10 years ago
- d617a44 Add an AlignedFreeDeleter and remove scoped_ptr_malloc. by andrew@webrtc.org · 10 years ago
- d4d5be8 Minor improvement in RoundToInt16 implementation. by turaj@webrtc.org · 10 years ago
- a0a6df3 Modified overuse detection thresholds. by asapersson@webrtc.org · 10 years ago
- 04a691a Removing a variable that was never read by henrik.lundin@webrtc.org · 10 years ago
- 6606199 ifdef the alsa code based on macro USE_X11 by fbarchard@google.com · 10 years ago
- 056176b Presubmit script that prohibits cls to both trunk/webrtc and trunk/talk. by henrike@webrtc.org · 10 years ago
- 78f0db4 Fix the break caused by r5579. by turaj@webrtc.org · 10 years ago
- 571df2d Update libjingle 61759961->61834300 by henrike@webrtc.org · 10 years ago
- c2d69d3 Resolves memcheck issue in AudioCodingModuleTest. The issue is coditional jumnp based on uninitialized variable. by turaj@webrtc.org · 10 years ago
- 97e7a64 Make WindowCapturerLinux handling window resize events. by jiayl@webrtc.org · 10 years ago
- 2421025 Added architecture for compiling under chrome NaCl. by andresp@webrtc.org · 10 years ago
- 056287e 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
- 8098e07 Add RTCP packet type counter (for getting statistics such as sent/received NACK and FIR). by asapersson@webrtc.org · 10 years ago
- b7a91fa Removes VoERTP_RTCP::InsertExtraRTPPacket. by henrika@webrtc.org · 10 years ago
- e384104 Fix DesktopAndCursorComposer not to crash by sergeyu@chromium.org · 10 years ago