- 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
- b669e60 Remove default implementations for SuspendBelowMinBitrate by henrik.lundin@webrtc.org · 11 years ago
- 3bcea52 Fix bug where fraction_lost is always set to 0 when getting received RTCP statistics. by stefan@webrtc.org · 11 years ago
- 8911937 Add callbacks for send channel rtcp statistics by sprang@webrtc.org · 11 years ago
- d90bebb Make RTPSender::SendPadData public. by stefan@webrtc.org · 11 years ago
- 991d58c Remove unused ThreadData struct. by andrew@webrtc.org · 11 years ago
- 5459e0b Remove the long disabled WEBRTC_SVNREVISION define. by andrew@webrtc.org · 11 years ago
- 382cfdd Removing DropDeltaAfterKey functionality which is unused. by andresp@webrtc.org · 11 years ago
- 9435a17 Add send frame rate statistics callback by sprang@webrtc.org · 11 years ago
- f2c136b Added a delay measurement, measures the time between an incoming captured frame until the frame is being processed. Measures the delay per second. by asapersson@webrtc.org · 11 years ago
- 0cb8020 Fixes a crash in fullstack tests introduced with r5209. by stefan@webrtc.org · 11 years ago
- 29a9669 Small fixes to plot_neteq_delay.m by henrik.lundin@webrtc.org · 11 years ago
- da3ae7c Adds support for sending redundant payloads over RTX. by stefan@webrtc.org · 11 years ago
- c101a27 Fix a typo in neteq.gypi by henrik.lundin@webrtc.org · 11 years ago
- c749348 Compile-out functions only used by the bit-exact test. by andrew@webrtc.org · 11 years ago
- b1f4a72 Don't HANDLE_EINTR(close). Use IGNORE_EINTR(close). by fischman@webrtc.org · 11 years ago
- adaa8b5 Add baseline generation/verification to BWE test framework. by solenberg@webrtc.org · 11 years ago
- 4ed6832 Utility class for reading/writing network-byte-ordered integers. by sprang@webrtc.org · 11 years ago
- 397aae0 Change BitrateStats to more generalized RateStatistics by sprang@webrtc.org · 11 years ago
- 309b2c8 Set local SSRC for VideoReceiveStream. by pbos@webrtc.org · 11 years ago
- f91a14f Do not use recursive calling in NetEq test tools by henrik.lundin@webrtc.org · 11 years ago
- 02817f8 Fixing NetEq tests for new Opus version by tina.legrand@webrtc.org · 11 years ago
- 169a27a Disable check for all sent SSRCs being valid. by pbos@webrtc.org · 11 years ago
- 758ef4c This CL adds an API to enable robust validation of delay estimates. by bjornv@webrtc.org · 11 years ago
- c8918cb Fixes a crash in the pacer where it fails to find a normal prio packet if there are no high prio packets, given that the queue has grown too large. by stefan@webrtc.org · 11 years ago
- 0e2571d Lock frame in ViECapturer::IncomingFrameI420. by pbos@webrtc.org · 11 years ago
- 9105cbd Set up SSRCs correctly after switching codec. by pbos@webrtc.org · 11 years ago
- b8b2a23 Recommit CL5184 by bjornv@webrtc.org · 11 years ago
- 151cd25 Refactor Remote Estimators Test into a more reusable form. by solenberg@webrtc.org · 11 years ago
- 66d634f Revert 5184 "Small refactoring change in delay_estimator." by bjornv@webrtc.org · 11 years ago
- 2c75d4e Small refactoring change in delay_estimator. by bjornv@webrtc.org · 11 years ago
- 801822c Ensure that no packet stays in the pacer queue for longer than 2 seconds. by stefan@webrtc.org · 11 years ago
- 8f9da30 Create default implementation to fix issue in libjingle by sprang@webrtc.org · 11 years ago
- 4a9843f Implement and test EncodedImageCallback in new ViE API. by sprang@webrtc.org · 11 years ago
- 2622be1 Added measure of encode time. Added encode time to the ViE CpuOveruseMeasure api. by asapersson@webrtc.org · 11 years ago
- 58b912b Remove const in vie_rtp_rtcp, where there is conflict with by sprang@webrtc.org · 11 years ago
- e7270f5 Faster implementation of BitRateStats. by mikhal@webrtc.org · 11 years ago
- 1a5aa03 Updated WebRTC version to 3.47 TBR=wu@webrtc.org by elham@webrtc.org · 11 years ago
- 5892ce5 Made video quality toolchain more configurable. by phoglund@webrtc.org · 11 years ago
- c86d1c6 Add include stdlib.h to files using abs. by stefan@webrtc.org · 11 years ago
- 586becf Add test for automatically disabling padding when no video is being captured. by stefan@webrtc.org · 11 years ago
- 5ae14be Clear empty video frames in unittest so DrMemory will allow them to be read without an uninitialized read error. by fbarchard@google.com · 11 years ago
- e8f79c5 Don't register iSAC-swb and iSAC-fb in NetEqDecodingTest. by turaj@webrtc.org · 11 years ago
- 8bdb87f Fix MouseCursor to MouseCursorShape conversion in ScreenCapturerWin. by sergeyu@chromium.org · 11 years ago
- 5fd393f Fix issues with sequence number wrap-around in jitter statistics. by turaj@webrtc.org · 11 years ago
- 6508af1 Distinguish instances of ACM1 from ACM2 by a version string. This is fpr testing purposes and will be removed when the experiment is done and ACM1 is fade out. by turaj@webrtc.org · 11 years ago
- 44b21e7 Replace VideoFrameI420 with I420VideoFrame. by pbos@webrtc.org · 11 years ago
- 51fa6ac Don't reset the AEC filter in extended mode. by andrew@webrtc.org · 11 years ago
- ce4a0b8 Add exception handling when configuring MediaCodc in order to prevent break in the new sdk release. by dwkang@webrtc.org · 11 years ago
- 970c5e5 Renaming ViEEncoderObserver::VideoSuspended by henrik.lundin@webrtc.org · 11 years ago
- a706baf Protect reads of ViEEncoder::video_suspended_. by pbos@webrtc.org · 11 years ago
- 9c15a62 Increase size of pacer window to 500 ms as that better matches the encoder. by stefan@webrtc.org · 11 years ago
- d7d60c8 Connect pacer/padding to SuspendBelowMinBitrate by henrik.lundin@webrtc.org · 11 years ago
- 0e6558b Lock access to ModuleRtpRtcpImpl::simulcast_. by pbos@webrtc.org · 11 years ago
- f8486d0 Rename DestroyStream methods to include Video. by pbos@webrtc.org · 11 years ago
- b87f528 Fix issues with sequence number wrap-around in jitter statistics by henrik.lundin@webrtc.org · 11 years ago
- 3c3a953 Add -Wnon-virtual-dtor warning for C++ code. by pbos@webrtc.org · 11 years ago
- e92aec9 Make interface destructor virtual by sprang@webrtc.org · 11 years ago
- 3fe2e7f Added API for enabling/disabling RTCP Receiver Reference Time extension. by asapersson@webrtc.org · 11 years ago
- 402f34c Increase run-time for full stack test for the rtt to be added reliably to the delay measurement. by asapersson@webrtc.org · 11 years ago
- fa7ac56 Typo in vie_autotest_win.cc by braveyao@webrtc.org · 11 years ago
- 36fb531 Fixes a crash in VoE when unregistering JNI hooks. by henrike@webrtc.org · 11 years ago
- 13a4d31 Interface changes to old api, for use by new api transition. by sprang@webrtc.org · 11 years ago
- b06a926 Added ViE API for getting overuse measure. by asapersson@webrtc.org · 11 years ago
- 04bcc9d Deliver I420VideoFrames from VideoRender module. by pbos@webrtc.org · 11 years ago
- c2162d1 Add possibility to get the last processed RTT from the call stats class (to be used by RTP/RTCP module). by asapersson@webrtc.org · 11 years ago
- f3b4602 Rename newapi::Transport::SendRTP()->SendRtp(). by pbos@webrtc.org · 11 years ago
- 60108c2 Rename RTP-extension constants. by pbos@webrtc.org · 11 years ago
- 48cc9dc Rename video streams' start/stop methods. by pbos@webrtc.org · 11 years ago
- 162021c Rename Call::Create{Receive,Send}Stream(). by pbos@webrtc.org · 11 years ago
- 4bfa866 Add experimental noise suppression dummy API. by aluebs@webrtc.org · 11 years ago
- 2b9794b Fix DesktopAndCursorComposer to restore frames to the original state. by sergeyu@chromium.org · 11 years ago
- dbc2a35 Adding back main() to the test. Now it is possible to choose between ACM1 and ACM2, furthermore, the test can simulate a channel with packet loss and FEC can be activated. Packet loss pattern is based on channel implementation in Channel{.cc,.h}, which currently is a determenistic pattern with 1 every 3rd packet is discarded. by turaj@webrtc.org · 11 years ago
- 8fdf191 Rename AutoMute to SuspendBelowMinBitrate by henrik.lundin@webrtc.org · 11 years ago
- 8d2354a Fix test broken with r5128. by stefan@webrtc.org · 11 years ago
- 26a736f Hook up audio/video sync to Call. by stefan@webrtc.org · 11 years ago
- 5eca0c7 Fix breakage after introducing new test. by stefan@webrtc.org · 11 years ago