- a48c91d Complete rewrite of demo application. by henrike@webrtc.org · 11 years ago
- 4fcb2f5 Remove overloaded CpuOveruseMeasure function. by asapersson@webrtc.org · 11 years ago
- 27f0841 Enables mixing and matching Java and native audio. It is used for getting best of both worlds capabilities (AEC and low latency). by henrike@webrtc.org · 11 years ago
- 6bf67db Fix common_video_unittests in apk_tests.gyp. by pbos@webrtc.org · 11 years ago
- c33d37c Add SwapFrame() to VideoSendStreamInput. by pbos@webrtc.org · 11 years ago
- ffea4ce Revert 5259 "Callback for send bitrate estimates" by sprang@webrtc.org · 11 years ago
- ca63ad9 Roll chromium_revision 232627:238260 by kjellander@webrtc.org · 11 years ago
- 1430bc3 Callback for send bitrate estimates by sprang@webrtc.org · 11 years ago
- b46e68d Removed unnecessary Pulse init from VoE startup. by fischman@webrtc.org · 11 years ago
- aa9e768 Correctly define OVERRIDE when building with g++ 4.7 and C++11 support by andrew@webrtc.org · 11 years ago
- b589c65 Improve VideoSendStreamTest::MaxPacketSize by sprang@webrtc.org · 11 years ago
- 2cafda4 Change uses of the obsolete armv7 setting to arm_version==7. by kjellander@webrtc.org · 11 years ago
- 5424c16 Fix compilation errors on Fedora 20. by fischman@webrtc.org · 11 years ago
- 241103f Ensure WEBRTC_MODULE_UTILITY_VIDEO is undefined for enable_video==0. by andrew@webrtc.org · 11 years ago
- 935c8c7 Add shape in DesktopFrame. by sergeyu@chromium.org · 11 years ago
- 8beba83 Add new method to MockAudioProcessing. by andrew@webrtc.org · 11 years ago
- e388f9e Revert 5229 "Make VideoSendStreamTest::MaxPacketSize test a whol..." by andrew@webrtc.org · 11 years ago
- 7b72264 Allow opening an AEC dump from an existing file handle. by henrikg@webrtc.org · 11 years ago
- 4383539 Stop video capturers in multi-stream test. by pbos@webrtc.org · 11 years ago
- 539670c Fix use of uninitialized memory in RtpSenderTest::StreamDataCountersCallbacks by sprang@webrtc.org · 11 years ago
- cebd1d7 Fraction lost statistics not being reported by sprang@webrtc.org · 11 years ago
- 532b8f7 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
- 0b16527 Use the RTT from RtcpRttStats class if provided when sending/receiving NACK. by asapersson@webrtc.org · 11 years ago
- b113981 Add callbacks for send channel rtp statistics by sprang@webrtc.org · 11 years ago
- 3adf058 Remove CallTest dependency on voice_engine/test/. by pbos@webrtc.org · 11 years ago
- ee234be Add API to query video engine for the send-side delay. by stefan@webrtc.org · 11 years ago
- a4fae33 Fixing the android build by henrik.lundin@webrtc.org · 11 years ago
- d05597a Move implementation files out of the webrtc/ root. by pbos@webrtc.org · 11 years ago
- db04941 Remove default implementations for SuspendBelowMinBitrate by henrik.lundin@webrtc.org · 11 years ago
- d964bf5 Fix bug where fraction_lost is always set to 0 when getting received RTCP statistics. by stefan@webrtc.org · 11 years ago
- 9b30fd3 Add callbacks for send channel rtcp statistics by sprang@webrtc.org · 11 years ago
- f03505e Make RTPSender::SendPadData public. by stefan@webrtc.org · 11 years ago
- 6d1a71b Remove unused ThreadData struct. by andrew@webrtc.org · 11 years ago
- c7d7363 Remove the long disabled WEBRTC_SVNREVISION define. by andrew@webrtc.org · 11 years ago
- 1fc02eb Removing DropDeltaAfterKey functionality which is unused. by andresp@webrtc.org · 11 years ago
- 5fdd10a Add send frame rate statistics callback by sprang@webrtc.org · 11 years ago
- f1630b1 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
- c0d6b2d Fixes a crash in fullstack tests introduced with r5209. by stefan@webrtc.org · 11 years ago
- a5be230 Small fixes to plot_neteq_delay.m by henrik.lundin@webrtc.org · 11 years ago
- 47f0c41 Adds support for sending redundant payloads over RTX. by stefan@webrtc.org · 11 years ago
- 76c6ac4 Fix a typo in neteq.gypi by henrik.lundin@webrtc.org · 11 years ago
- 3f0b77f Compile-out functions only used by the bit-exact test. by andrew@webrtc.org · 11 years ago
- 38aa817 Don't HANDLE_EINTR(close). Use IGNORE_EINTR(close). by fischman@webrtc.org · 11 years ago
- af00735 Add baseline generation/verification to BWE test framework. by solenberg@webrtc.org · 11 years ago
- 09299b0 Utility class for reading/writing network-byte-ordered integers. by sprang@webrtc.org · 11 years ago
- 7374da3 Change BitrateStats to more generalized RateStatistics by sprang@webrtc.org · 11 years ago
- 4b50db1 Set local SSRC for VideoReceiveStream. by pbos@webrtc.org · 11 years ago
- cc19dce Do not use recursive calling in NetEq test tools by henrik.lundin@webrtc.org · 11 years ago
- db085fa Fixing NetEq tests for new Opus version by tina.legrand@webrtc.org · 11 years ago
- 1131f9b Disable check for all sent SSRCs being valid. by pbos@webrtc.org · 11 years ago
- cd8c2b1 This CL adds an API to enable robust validation of delay estimates. by bjornv@webrtc.org · 11 years ago
- 1003b7d 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
- 5a669d5 Lock frame in ViECapturer::IncomingFrameI420. by pbos@webrtc.org · 11 years ago
- 66f4394 Set up SSRCs correctly after switching codec. by pbos@webrtc.org · 11 years ago
- c3cce18 Recommit CL5184 by bjornv@webrtc.org · 11 years ago
- cda3cf3 Refactor Remote Estimators Test into a more reusable form. by solenberg@webrtc.org · 11 years ago
- 85ef326 Revert 5184 "Small refactoring change in delay_estimator." by bjornv@webrtc.org · 11 years ago
- cd96113 Small refactoring change in delay_estimator. by bjornv@webrtc.org · 11 years ago
- 04d6593 Ensure that no packet stays in the pacer queue for longer than 2 seconds. by stefan@webrtc.org · 11 years ago
- 163393e Create default implementation to fix issue in libjingle by sprang@webrtc.org · 11 years ago
- 2e98d45 Implement and test EncodedImageCallback in new ViE API. by sprang@webrtc.org · 11 years ago
- dd4f866 Added measure of encode time. Added encode time to the ViE CpuOveruseMeasure api. by asapersson@webrtc.org · 11 years ago
- 4ee440a Remove const in vie_rtp_rtcp, where there is conflict with by sprang@webrtc.org · 11 years ago
- 66fba2b Faster implementation of BitRateStats. by mikhal@webrtc.org · 11 years ago
- b0d97ed Updated WebRTC version to 3.47 TBR=wu@webrtc.org by elham@webrtc.org · 11 years ago
- d95137e Made video quality toolchain more configurable. by phoglund@webrtc.org · 11 years ago
- b1d7931 Add include stdlib.h to files using abs. by stefan@webrtc.org · 11 years ago
- 88d3a17 Add test for automatically disabling padding when no video is being captured. by stefan@webrtc.org · 11 years ago
- 98d217d 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
- 01a09ac Don't register iSAC-swb and iSAC-fb in NetEqDecodingTest. by turaj@webrtc.org · 11 years ago
- c5a2831 Fix MouseCursor to MouseCursorShape conversion in ScreenCapturerWin. by sergeyu@chromium.org · 11 years ago
- 936fd5f Fix issues with sequence number wrap-around in jitter statistics. by turaj@webrtc.org · 11 years ago
- f1fccf7 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
- 50293f5 Replace VideoFrameI420 with I420VideoFrame. by pbos@webrtc.org · 11 years ago
- 84d69d4 Don't reset the AEC filter in extended mode. by andrew@webrtc.org · 11 years ago
- db43763 Add exception handling when configuring MediaCodc in order to prevent break in the new sdk release. by dwkang@webrtc.org · 11 years ago
- 82b883c Renaming ViEEncoderObserver::VideoSuspended by henrik.lundin@webrtc.org · 11 years ago
- 7673871 Protect reads of ViEEncoder::video_suspended_. by pbos@webrtc.org · 11 years ago
- c41f11b Increase size of pacer window to 500 ms as that better matches the encoder. by stefan@webrtc.org · 11 years ago
- b9f1eb8 Connect pacer/padding to SuspendBelowMinBitrate by henrik.lundin@webrtc.org · 11 years ago
- c3f827c Lock access to ModuleRtpRtcpImpl::simulcast_. by pbos@webrtc.org · 11 years ago
- 12a93e0 Rename DestroyStream methods to include Video. by pbos@webrtc.org · 11 years ago
- 31bd97d Fix issues with sequence number wrap-around in jitter statistics by henrik.lundin@webrtc.org · 11 years ago
- e1e050e Add -Wnon-virtual-dtor warning for C++ code. by pbos@webrtc.org · 11 years ago
- 21dc10d Make interface destructor virtual by sprang@webrtc.org · 11 years ago
- 3dc7ff3 Added API for enabling/disabling RTCP Receiver Reference Time extension. by asapersson@webrtc.org · 11 years ago
- 3b7da1e 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
- 63e3810 Typo in vie_autotest_win.cc by braveyao@webrtc.org · 11 years ago
- 7950b98 Fixes a crash in VoE when unregistering JNI hooks. by henrike@webrtc.org · 11 years ago
- 4673674 Interface changes to old api, for use by new api transition. by sprang@webrtc.org · 11 years ago
- 4747585 Added ViE API for getting overuse measure. by asapersson@webrtc.org · 11 years ago
- 3051951 Deliver I420VideoFrames from VideoRender module. by pbos@webrtc.org · 11 years ago
- c4af4cf 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
- 3009c81 Rename newapi::Transport::SendRTP()->SendRtp(). by pbos@webrtc.org · 11 years ago
- 346dbe7 Rename RTP-extension constants. by pbos@webrtc.org · 11 years ago
- 7f9f840 Rename video streams' start/stop methods. by pbos@webrtc.org · 11 years ago
- 964d78e Rename Call::Create{Receive,Send}Stream(). by pbos@webrtc.org · 11 years ago
- 09b40ec Add experimental noise suppression dummy API. by aluebs@webrtc.org · 11 years ago
- 8b0791c Fix DesktopAndCursorComposer to restore frames to the original state. by sergeyu@chromium.org · 11 years ago
- eb45a20 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