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  • 5b51ebc179 Removing DropDeltaAfterKey functionality which is unused. andresp@webrtc.org 2013-12-04 15:53:24 +00:00
  • 71f055fb41 Add send frame rate statistics callback sprang@webrtc.org 2013-12-04 15:09:27 +00:00
  • 9e5b0342f6 Added a delay measurement, measures the time between an incoming captured frame until the frame is being processed. Measures the delay per second. asapersson@webrtc.org 2013-12-04 13:47:44 +00:00
  • 79b63206b9 Fixes a crash in fullstack tests introduced with r5209. stefan@webrtc.org 2013-12-04 13:34:28 +00:00
  • b477fa6d21 Small fixes to plot_neteq_delay.m henrik.lundin@webrtc.org 2013-12-04 12:28:47 +00:00
  • 7e9315b42e Adds support for sending redundant payloads over RTX. stefan@webrtc.org 2013-12-04 10:24:26 +00:00
  • 9523b55826 Fix a typo in neteq.gypi henrik.lundin@webrtc.org 2013-12-04 08:24:49 +00:00
  • d7696c4ed1 Compile-out functions only used by the bit-exact test. andrew@webrtc.org 2013-12-03 23:39:16 +00:00
  • d3865e9124 Don't HANDLE_EINTR(close). Use IGNORE_EINTR(close). fischman@webrtc.org 2013-12-03 19:10:20 +00:00
  • 812dd11f8c Add baseline generation/verification to BWE test framework. solenberg@webrtc.org 2013-12-03 15:11:14 +00:00
  • 499631c1e4 Utility class for reading/writing network-byte-ordered integers. sprang@webrtc.org 2013-12-03 13:22:48 +00:00
  • 37968a9be7 Change BitrateStats to more generalized RateStatistics sprang@webrtc.org 2013-12-03 10:31:59 +00:00
  • b613b5ab2b Set local SSRC for VideoReceiveStream. pbos@webrtc.org 2013-12-03 10:13:04 +00:00
  • 5ecdef11cc Do not use recursive calling in NetEq test tools henrik.lundin@webrtc.org 2013-12-03 08:26:49 +00:00
  • e0034557a7 RTCPeerConnection(objc): avoid leaking ICE candidate on addition. fischman@webrtc.org 2013-12-02 18:49:54 +00:00
  • 8418e9696b Fixing NetEq tests for new Opus version tina.legrand@webrtc.org 2013-11-29 09:30:43 +00:00
  • 54e8bfafba Apprtc demo: add DSCP support. braveyao@webrtc.org 2013-11-29 02:38:20 +00:00
  • 03c7a35ac0 Fixing long lines in apprtc.py. phoglund@webrtc.org 2013-11-28 17:45:08 +00:00
  • e1fc3f22ea Disable check for all sent SSRCs being valid. pbos@webrtc.org 2013-11-28 15:40:12 +00:00
  • bd41a84694 This CL adds an API to enable robust validation of delay estimates. bjornv@webrtc.org 2013-11-28 14:58:35 +00:00
  • b627f676b3 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. stefan@webrtc.org 2013-11-28 14:00:09 +00:00
  • 1f7c8d8b6a Lock frame in ViECapturer::IncomingFrameI420. pbos@webrtc.org 2013-11-28 13:26:33 +00:00
  • 13d38a13e3 Set up SSRCs correctly after switching codec. pbos@webrtc.org 2013-11-28 11:59:31 +00:00
  • d1a1c353ac Recommit CL5184 bjornv@webrtc.org 2013-11-28 11:45:05 +00:00
  • c8f76ddc19 Refactor Remote Estimators Test into a more reusable form. solenberg@webrtc.org 2013-11-28 10:11:35 +00:00
  • 82eb3a690e Revert 5184 "Small refactoring change in delay_estimator." bjornv@webrtc.org 2013-11-28 09:44:47 +00:00
  • eea079a376 Small refactoring change in delay_estimator. bjornv@webrtc.org 2013-11-28 07:59:04 +00:00
  • 19a40ff05b Ensure that no packet stays in the pacer queue for longer than 2 seconds. stefan@webrtc.org 2013-11-27 14:16:20 +00:00
  • b3ea3afa60 Create default implementation to fix issue in libjingle sprang@webrtc.org 2013-11-27 10:28:27 +00:00
  • 4070935f4f Implement and test EncodedImageCallback in new ViE API. sprang@webrtc.org 2013-11-26 11:41:59 +00:00
  • c7ff8f990a Added measure of encode time. Added encode time to the ViE CpuOveruseMeasure api. asapersson@webrtc.org 2013-11-26 11:12:33 +00:00
  • bd51d9324e LSan suppressions for libjingle_peerconnection_unittest kjellander@webrtc.org 2013-11-26 09:37:12 +00:00
  • 7f959980f8 Remove const in vie_rtp_rtcp, where there is conflict with mock defines in fakewebrtcvideoengine. sprang@webrtc.org 2013-11-26 09:30:04 +00:00
  • d89b52af80 Faster implementation of BitRateStats. mikhal@webrtc.org 2013-11-25 17:49:28 +00:00
  • 326bcff879 Updated WebRTC version to 3.47 TBR=wu@webrtc.org elham@webrtc.org 2013-11-25 17:19:46 +00:00
  • 4e3161dd55 Style-option file for clang-format. pbos@webrtc.org 2013-11-25 15:30:37 +00:00
  • 3260f109e3 Made video quality toolchain more configurable. phoglund@webrtc.org 2013-11-25 14:10:20 +00:00
  • 47fadba750 Add include stdlib.h to files using abs. stefan@webrtc.org 2013-11-25 12:03:56 +00:00
  • 4ab4fc0044 Add test for automatically disabling padding when no video is being captured. stefan@webrtc.org 2013-11-25 11:54:24 +00:00
  • b5bc098e20 Clear empty video frames in unittest so DrMemory will allow them to be read without an uninitialized read error. BUG=libyuv:263 TESTED=drmemory out\Debug\modules_unittests.exe --gtest_filter=*PreprocessorLogic R=kjellander@webrtc.org, mflodman@webrtc.org fbarchard@google.com 2013-11-25 09:06:33 +00:00
  • aa74b5d690 Add success/error callback to set sdp calls. Add a workaround for crbug/322756 to append a line break to the end of sdp if needed. wu@webrtc.org 2013-11-23 00:37:50 +00:00
  • 5272eb8d83 Don't register iSAC-swb and iSAC-fb in NetEqDecodingTest. turaj@webrtc.org 2013-11-23 00:11:32 +00:00
  • e839da02c1 Fix MouseCursor to MouseCursorShape conversion in ScreenCapturerWin. sergeyu@chromium.org 2013-11-22 20:39:16 +00:00
  • 78b41a09bd Fix issues with sequence number wrap-around in jitter statistics. turaj@webrtc.org 2013-11-22 20:27:07 +00:00
  • 832bd74cfa libyuv r874 for build improvements on ios/android, and improved YUV scale performance. BUG=libyuv:288 TESTED=try bots R=andrew@webrtc.org fbarchard@google.com 2013-11-22 19:30:52 +00:00
  • b43202d839 Disable PeerConnectionEndToEndTest for tsanv2 build. BUG=1205 TEST=try R=kjellander@webrtc.org wu@webrtc.org 2013-11-22 19:14:25 +00:00
  • 1e8c93c953 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. turaj@webrtc.org 2013-11-22 17:04:49 +00:00
  • 2ffb149c2c Replace VideoFrameI420 with I420VideoFrame. pbos@webrtc.org 2013-11-22 13:10:13 +00:00
  • b0ed8f8a08 Don't reset the AEC filter in extended mode. andrew@webrtc.org 2013-11-22 06:39:42 +00:00
  • 9e85c01ec8 Add exception handling when configuring MediaCodc in order to prevent break in the new sdk release. dwkang@webrtc.org 2013-11-22 02:49:17 +00:00
  • 9fe3603dc1 Renaming ViEEncoderObserver::VideoSuspended henrik.lundin@webrtc.org 2013-11-21 23:00:40 +00:00
  • 484ee962b5 Protect reads of ViEEncoder::video_suspended_. pbos@webrtc.org 2013-11-21 18:44:23 +00:00
  • 1977960866 AppRTCDemo(ios): remove codesigning hack now that gyp signs by default. fischman@webrtc.org 2013-11-21 16:48:51 +00:00
  • ef2d55461b Increase size of pacer window to 500 ms as that better matches the encoder. stefan@webrtc.org 2013-11-21 14:37:11 +00:00
  • 331d4402fc Connect pacer/padding to SuspendBelowMinBitrate henrik.lundin@webrtc.org 2013-11-21 14:05:40 +00:00
  • ffe1b17b57 Lock access to ModuleRtpRtcpImpl::simulcast_. pbos@webrtc.org 2013-11-21 13:53:13 +00:00
  • 2c46f8d854 Rename DestroyStream methods to include Video. pbos@webrtc.org 2013-11-21 13:49:43 +00:00
  • 6f6ba6edee Fix issues with sequence number wrap-around in jitter statistics henrik.lundin@webrtc.org 2013-11-21 13:17:29 +00:00
  • b3cc78de28 Add -Wnon-virtual-dtor warning for C++ code. pbos@webrtc.org 2013-11-21 11:42:02 +00:00
  • 72964bd4fb Make interface destructor virtual sprang@webrtc.org 2013-11-21 09:09:54 +00:00
  • 8d02f5dc71 Added API for enabling/disabling RTCP Receiver Reference Time extension. asapersson@webrtc.org 2013-11-21 08:57:04 +00:00
  • 54a05518e2 Increase run-time for full stack test for the rtt to be added reliably to the delay measurement. asapersson@webrtc.org 2013-11-21 07:45:08 +00:00
  • 425e1d0fb9 Typo in vie_autotest_win.cc braveyao@webrtc.org 2013-11-21 02:17:01 +00:00
  • a750044396 Fixes a crash in VoE when unregistering JNI hooks. henrike@webrtc.org 2013-11-20 22:32:12 +00:00
  • 364f204d16 Update talk to 56698267. wu@webrtc.org 2013-11-20 21:49:41 +00:00
  • dc50aaeaa8 Interface changes to old api, for use by new api transition. sprang@webrtc.org 2013-11-20 16:47:07 +00:00
  • b24d33565c Added ViE API for getting overuse measure. asapersson@webrtc.org 2013-11-20 13:51:40 +00:00
  • d29d4e9c08 Deliver I420VideoFrames from VideoRender module. pbos@webrtc.org 2013-11-20 13:19:54 +00:00
  • 1ae1d0c471 Add possibility to get the last processed RTT from the call stats class (to be used by RTP/RTCP module). asapersson@webrtc.org 2013-11-20 12:46:11 +00:00
  • 27326b6a42 Rename newapi::Transport::SendRTP()->SendRtp(). pbos@webrtc.org 2013-11-20 12:17:04 +00:00
  • ce90eff345 Rename RTP-extension constants. pbos@webrtc.org 2013-11-20 11:48:56 +00:00
  • 53c8573525 Rename video streams' start/stop methods. pbos@webrtc.org 2013-11-20 11:36:47 +00:00
  • 5a63655ab0 Rename Call::Create{Receive,Send}Stream(). pbos@webrtc.org 2013-11-20 10:40:25 +00:00
  • 0b72f5863b Add experimental noise suppression dummy API. aluebs@webrtc.org 2013-11-19 15:17:51 +00:00
  • 5d85819dd2 Fix DesktopAndCursorComposer to restore frames to the original state. sergeyu@chromium.org 2013-11-19 02:15:47 +00:00
  • 7a05ae5c69 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. turaj@webrtc.org 2013-11-18 18:16:53 +00:00
  • 9c5fb76662 Exclude AV-sync test from Valgrind platforms. pbos@webrtc.org 2013-11-18 16:22:50 +00:00
  • ce8e0936d9 Rename AutoMute to SuspendBelowMinBitrate henrik.lundin@webrtc.org 2013-11-18 12:18:43 +00:00
  • 28bf50f0ec Fix test broken with r5128. stefan@webrtc.org 2013-11-18 11:58:24 +00:00
  • b082ade3db Hook up audio/video sync to Call. stefan@webrtc.org 2013-11-18 11:45:11 +00:00
  • 4cfa6050f6 Fix breakage after introducing new test. stefan@webrtc.org 2013-11-15 13:15:56 +00:00
  • 69969e2e2f Improve Call tests for RTX. stefan@webrtc.org 2013-11-15 12:32:15 +00:00
  • 6e95d7afab Increment RTP timestamps for padding packets henrik.lundin@webrtc.org 2013-11-15 08:59:19 +00:00
  • 6488761f2e Implement VideoSendStream::SetCodec(). pbos@webrtc.org 2013-11-14 08:58:14 +00:00
  • 183c727bca Disable datachannel_unittest.cc sergeyu@chromium.org 2013-11-13 22:59:20 +00:00
  • a23f0ca4ba Update talk to 56619788 sergeyu@chromium.org 2013-11-13 22:48:52 +00:00
  • e8722856f9 Disable all vie_auto_tests on Linux for now (take 2) kjellander@webrtc.org 2013-11-13 15:51:49 +00:00
  • c8489852ec Disable all automated vie_auto_tests on Linux for now kjellander@webrtc.org 2013-11-13 15:44:54 +00:00
  • 9b82f5a6ed Fix for RTX in combination with pacing. stefan@webrtc.org 2013-11-13 15:29:21 +00:00
  • 03f33709f8 Inject config when creating channels to override the existing one. turaj@webrtc.org 2013-11-13 00:02:48 +00:00
  • e8433eb115 Reimplementing NetEq4's AudioVector henrik.lundin@webrtc.org 2013-11-12 13:15:02 +00:00
  • 38599510df Parse next RTCP XR report block after an unsupported block type. asapersson@webrtc.org 2013-11-12 08:08:26 +00:00
  • 3e427263ee Reducing opus_test runtime to pass Android test minyue@webrtc.org 2013-11-11 22:03:52 +00:00
  • e03cafaebc MIPS optimizations for AECM audio processing module andrew@webrtc.org 2013-11-11 20:10:01 +00:00
  • b0730108a2 Move audio_processing dependencies to a variable. andrew@webrtc.org 2013-11-11 17:20:27 +00:00
  • 57eb858698 Remove ".." from include_dirs in build/common. pbos@webrtc.org 2013-11-11 10:20:27 +00:00
  • 6e908b3adf Remove unnecessary include_dirs from audio_processing. andrew@webrtc.org 2013-11-08 19:52:05 +00:00
  • 00ed170795 Roll libvpx 225010:232686. marpan@webrtc.org 2013-11-08 18:37:18 +00:00
  • 5973f3a24a Remove unneeded includes from trace_posix.cc. andrew@webrtc.org 2013-11-08 17:30:07 +00:00
  • 48df38114d Fix for making sure that the packet in order checks are done prior to updating the last received packet state. stefan@webrtc.org 2013-11-08 15:18:52 +00:00