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  • 32001ef124 PeerConnection shutdown-time fixes - TCPPort::~TCPPort() was leaking incoming_ sockets; now they are deleted. - PeerConnection::RemoveStream() now removes streams even if the PeerConnection::IsClosed(). Previously such streams would never get removed. - Gave MediaStreamTrackInterface a virtual destructor to ensure deletes on base pointers are dispatched virtually. - VideoTrack.dispose() delegates to super.dispose() (instead of leaking) - PeerConnection.dispose() now removes streams before disposing of them. - MediaStream.dispose() now removes tracks before disposing of them. - VideoCapturer.dispose() only unowned VideoCapturers (mirroring C++ API) - AppRTCDemo.disconnectAndExit() now correctly .dispose()s its VideoSource and PeerConnectionFactory. - CHECK that Release()d objects are deleted when expected to (i.e. no ref-cycles or missing .dispose() calls) in the Java API. - Create & Return webrtc::Traces at factory birth/death to be able to assert that _all_ threads started during the test are collected by the end. - Name threads attached to the JVM more informatively for debugging. - Removed a bunch of unnecessary scoped_refptr instances in peerconnection_jni.cc whose only job was messing with refcounts. fischman@webrtc.org 2013-08-12 23:26:21 +00:00
  • a5506690b4 Update libjingle to 50733053. mallinath@webrtc.org 2013-08-12 21:18:15 +00:00
  • 4ca7d3f9fe Replace MapWrapper with std::map<>. pbos@webrtc.org 2013-08-12 19:51:57 +00:00
  • dd14b2add1 libjingle gyp: signal errors during gyp time to avoid cryptic failures during build time. fischman@webrtc.org 2013-08-12 18:06:29 +00:00
  • 1928d0ef67 Updated WebRTC version to 3.39 elham@webrtc.org 2013-08-12 17:12:44 +00:00
  • 468e19aa93 Signal when shutting down DirectTransport. pbos@webrtc.org 2013-08-12 14:28:00 +00:00
  • 0d94c2f81c Avoid acquiring VCM::_receiveCritSect during decode callback. wuchengli@chromium.org 2013-08-12 14:20:49 +00:00
  • 9668467d87 Run loopback tests with network thread. pbos@webrtc.org 2013-08-12 12:59:04 +00:00
  • ecbe0aa543 Added Opus stereo support minyue@webrtc.org 2013-08-12 06:48:09 +00:00
  • 91053e7c5a Update libjingle to 50654631. wu@webrtc.org 2013-08-10 07:18:04 +00:00
  • bf853f2732 Fix crash in screen capturer on Mac sergeyu@chromium.org 2013-08-10 01:30:23 +00:00
  • 6cd9341801 Hand over loopback packets to a network thread. pbos@webrtc.org 2013-08-09 21:11:57 +00:00
  • 80865fd611 Don't pace out packets or generate padding when the pacer is disabled. stefan@webrtc.org 2013-08-09 11:31:23 +00:00
  • 2ab209ef14 Remove include_dirs from test/test.gyp. pbos@webrtc.org 2013-08-09 08:49:48 +00:00
  • a3b7406219 Remove unused unreferenced code in webrtc/ pbos@webrtc.org 2013-08-09 08:47:51 +00:00
  • f4081ab8d8 Revert "Avoid acquiring VCM::_receiveCritSect during decode callback." wuchengli@chromium.org 2013-08-09 04:42:51 +00:00
  • a717ee9962 Avoid acquiring VCM::_receiveCritSect during decode callback. wuchengli@chromium.org 2013-08-09 04:08:38 +00:00
  • 64799da6c6 Allowing decoding with errors, when disabling nack. mikhal@webrtc.org 2013-08-08 22:45:33 +00:00
  • e270331481 Fix duplicate code niklas.enbom@webrtc.org 2013-08-08 22:23:48 +00:00
  • 5a27e49f35 This CL will add support of passing all turn urls returned by the CEOD to PeerConnection object. mallinath@webrtc.org 2013-08-08 19:52:08 +00:00
  • 58d76cb635 Delete Channels without ChannelManager lock. pbos@webrtc.org 2013-08-08 17:32:21 +00:00
  • bd21fb5f8d Adding call to Opus PLC tina.legrand@webrtc.org 2013-08-08 11:01:07 +00:00
  • d177c10e2d Added logic for kSelectiveErrors to VCMJitterBuffer and corresponding unit tests. agalusza@google.com 2013-08-08 01:12:33 +00:00
  • 676ff1ed89 Ref-counted rewrite of ChannelManager. pbos@webrtc.org 2013-08-07 17:57:36 +00:00
  • 825e9b0a9b talk/objc/README: s/libjingle/webrtc/ in repository path. fischman@webrtc.org 2013-08-07 16:52:03 +00:00
  • a165d9c0a4 Code formatting on files touched in r4447. pbos@webrtc.org 2013-08-07 14:17:05 +00:00
  • 401ef361ac Added configuration of max delay to ACM and NetEq pwestin@webrtc.org 2013-08-06 21:01:36 +00:00
  • c883fdc273 PeerConnection.java: enable setting trace & log levels from Java fischman@webrtc.org 2013-08-06 19:00:53 +00:00
  • c4e1ab515b Added Decoding with errors API to video_coding.h and removed unused DecodeError enum. agalusza@google.com 2013-08-06 18:27:41 +00:00
  • 0fc2558503 Add turaj@webrtc.org to NetEq owners. turaj@webrtc.org 2013-08-06 17:07:18 +00:00
  • 94aca5c7de Disabled flaky HardwareTest.BuiltInWasapiAECWorksForAudioWindowsCoreAudioLayer. phoglund@webrtc.org 2013-08-06 08:20:47 +00:00
  • bd69d1beaf Disabled SsrcPropagatesCorrectly on Linux. phoglund@webrtc.org 2013-08-06 08:03:16 +00:00
  • 7bb5436e5d Better error treatment in NetEqImpl::InsertPacketInternal() minyue@webrtc.org 2013-08-06 05:40:57 +00:00
  • 9721db799c removed NetEq::EnableDtmf() minyue@webrtc.org 2013-08-06 05:36:26 +00:00
  • 6e7c203aee Modified apprtc demo code to detect browser by checking user_agent in apprtc.py. Now we will use Mozilla stun server if FF is detected as the browser. The CL is an improvement to r4388. vikasmarwaha@webrtc.org 2013-08-05 22:05:20 +00:00
  • 9dba525627 * Update libjingle to 50389769. * Together with "Add texture support for i420 video frame." from wuchengli@chromium.org. https://webrtc-codereview.appspot.com/1413004 wu@webrtc.org 2013-08-05 20:36:57 +00:00
  • f696f253b2 Invert dependency between webrtc_utility and media_file targets to reflect reality. fischman@webrtc.org 2013-08-05 18:45:19 +00:00
  • 9b8861c358 Updated WebRTC version number to 3.38 elham@webrtc.org 2013-08-05 17:19:16 +00:00
  • 12dc1a38ca Switch C++-style C headers with their C equivalents. pbos@webrtc.org 2013-08-05 16:22:53 +00:00
  • c3d93c6921 talk/PRESUBMIT: Accept copyright years going back to 2004. fischman@webrtc.org 2013-08-05 15:01:33 +00:00
  • ccdcbae177 Fix implicit int->bool conversion in VideoSendStream::DeliverRtcp. pbos@webrtc.org 2013-08-05 13:25:51 +00:00
  • 4052370e89 Use RtpHeaderParser in VideoCall implementation. pbos@webrtc.org 2013-08-05 12:49:22 +00:00
  • bbb07e69e5 Glue code and tests for NACK in new VideoEngine API. pbos@webrtc.org 2013-08-05 12:01:36 +00:00
  • 7fb9ce0cf5 Fix send times in video_full_stack. pbos@webrtc.org 2013-08-05 09:29:50 +00:00
  • 735a7c8b93 Add back is.FrameProvider() call lost in r4194. pbos@webrtc.org 2013-08-05 09:03:03 +00:00
  • 94349552de Disable P2PTransportChannelTest.* on memcheck and tsan bots due to issue 1972. wu@webrtc.org 2013-08-02 23:30:50 +00:00
  • 2cbb429323 Remove redundant conditions key. andrew@webrtc.org 2013-08-02 20:52:54 +00:00
  • 7df9706a01 Add one API for implementing Initial delay. turaj@webrtc.org 2013-08-02 18:07:13 +00:00
  • 89c674053e Adds all unittests to android NDK-APK framework. henrike@webrtc.org 2013-08-02 16:53:47 +00:00
  • 51b2459d37 Add some virtual and OVERRIDEs in webrtc/common_audio/ pbos@webrtc.org 2013-08-02 11:44:38 +00:00
  • 9162080527 Fix some chromium-style warnings in webrtc/modules/audio_processing/ pbos@webrtc.org 2013-08-02 11:44:11 +00:00
  • 4ebd8efc09 Supress libjingle_unittest fails on TSan. wu@webrtc.org 2013-08-02 00:14:41 +00:00
  • a054569c15 Fix memory leak in datachannel and its test. wu@webrtc.org 2013-08-01 22:08:14 +00:00
  • 0dc0f172a3 sscanf isn't safe with strings that aren't null-terminated. In such case, create a local copy that is null-terminated first. wu@webrtc.org 2013-08-01 21:20:46 +00:00
  • 17758e96c5 Fix crash in DesktopRegion::Intersect(). sergeyu@chromium.org 2013-08-01 19:51:04 +00:00
  • 86d7a198ec ObjC PeerConnection README: note workaround needed for crbug.com/248168 fischman@webrtc.org 2013-08-01 19:27:54 +00:00
  • 1bc1954174 AppRTCDemo: builds using ninja on iOS for simulator and device! fischman@webrtc.org 2013-08-01 18:29:45 +00:00
  • 6abb750993 Delete gtest_exclude for asan which doesn't have effect with how the bots are setup now Add gtest_exclude for tsan to disable some flakey tests. Change tsan suppression since the function name has been changed from DecodeWithErrors to DecodeErrorMode. wu@webrtc.org 2013-08-01 18:00:02 +00:00
  • a2a2718a6c Fix some chromium-style warnings in webrtc/system_wrappers/ pbos@webrtc.org 2013-08-01 17:26:15 +00:00
  • a7e360e89b Removed lines preventing simultaneous kHardNack and decoding with errors. Also made changes recommended by gcl lint (with the exception of changing non-const references to pointers). Propagated orthogonal API for decoding with errors from VideoCodingModule to VCMJitterBuffer. Modified VCMJitterBuffer to allow three error modes: kNoErrors, kSelectiveErrors, kWithErrors. agalusza@google.com 2013-08-01 03:15:08 +00:00
  • d64719d895 Update libjingle to 50191337. wu@webrtc.org 2013-08-01 00:00:07 +00:00
  • d3ae3c7b1f Unbreak clang/android build of webrtc. fischman@webrtc.org 2013-07-31 23:53:07 +00:00
  • 7fdbb1c832 We don't need to link with libssl.so when we already depend on openssl. wu@webrtc.org 2013-07-31 22:41:36 +00:00
  • 27c0408a16 Suppressing tsan errors on libjingle_unittest and libjingle_peerconnection_unittest. wu@webrtc.org 2013-07-31 22:41:15 +00:00
  • caa7024b86 PeerConnectionTest.java: build on android bots as well as linux ones. fischman@webrtc.org 2013-07-31 21:56:30 +00:00
  • a543114004 Removes no longer needed valgrind-libjingle folder. Was workaround for some bots using wrong valgrind script. henrike@webrtc.org 2013-07-31 17:53:39 +00:00
  • d40b4d9685 Fix libjingle memory bots by suppressing some of the errors. wu@webrtc.org 2013-07-31 17:32:36 +00:00
  • d4412feeb0 Adding possibility to use encoding time when trigger underuse for frame based overuse detection. mflodman@webrtc.org 2013-07-31 16:42:21 +00:00
  • 09e8c47ee5 Merge r4374 from stable to trunk. xians@webrtc.org 2013-07-31 16:30:19 +00:00
  • 8fff1f065e Merge r4394 from stable to trunk. xians@webrtc.org 2013-07-31 16:27:42 +00:00
  • 2f84afad30 Merge r4326 from stable to trunk. xians@webrtc.org 2013-07-31 16:23:37 +00:00
  • 7126b38d8f Handel zero correlation if at the same time distortion is also zero. turaj@webrtc.org 2013-07-31 16:05:09 +00:00
  • 2d1a55caed Add some virtual and OVERRIDEs in webrtc/modules/audio_coding/ pbos@webrtc.org 2013-07-31 15:54:00 +00:00
  • e72428442d Fix some chromium-style warnings in webrtc/modules/desktop_capture/ pbos@webrtc.org 2013-07-31 15:32:43 +00:00
  • 0193158634 Fix some chromium-style warnings in webrtc/modules/pacing/ pbos@webrtc.org 2013-07-31 15:18:19 +00:00
  • f3e4ceee47 Fix some chromium-style warnings in webrtc/modules/rtp_rtcp/ pbos@webrtc.org 2013-07-31 15:17:19 +00:00
  • 8f23df51d4 Fix some chromium-style warnings in webrtc/modules/remote_bitrate_estimator/ pbos@webrtc.org 2013-07-31 15:16:52 +00:00
  • 4fac8a4699 Fix some chromium-style warnings in webrtc/modules/bitrate_controller/ pbos@webrtc.org 2013-07-31 15:16:20 +00:00
  • a96d8771f2 Added libjingle_peerconnection_java_unittest to buildbot_tests.py phoglund@webrtc.org 2013-07-31 10:50:30 +00:00
  • 0a4ca8f0bb Move internal aec_core defines out of header. andrew@webrtc.org 2013-07-31 08:13:08 +00:00
  • 7446870a0f Suppress failing tests on Linux Memcheck bot. wu@webrtc.org 2013-07-30 23:36:42 +00:00
  • 9c9fc767b1 Fixing the memory check bots by suppressing some of the tests. wu@webrtc.org 2013-07-30 22:54:08 +00:00
  • 933946ac55 Suppress libjingle_peerconnection_unittest failures on linux memcheck build bot. wu@webrtc.org 2013-07-30 22:28:29 +00:00
  • 0342e65f8d Disable peerconnection tests that are failing on memcheck. wu@webrtc.org 2013-07-30 22:28:14 +00:00
  • ae7bf1525b Disable p2p tests that are failing on memory test. wu@webrtc.org 2013-07-30 21:40:39 +00:00
  • b59c6dd397 Add svn:ignore properties for all spuriously-removed dirs on Linux64 Release (internal). fischman@webrtc.org 2013-07-30 19:34:07 +00:00
  • 85f07f59ee PeerConnectionTest.java: use java_home gyp var instead of hardcoding /usr. fischman@webrtc.org 2013-07-30 18:11:07 +00:00
  • fd7e3c52d8 Correcting Turaj's email. turaj@webrtc.org 2013-07-30 17:25:07 +00:00
  • 3d496fb046 Roll chromium_revision 205140:214260 to pick up build fixes for ninja iOS device build. fischman@webrtc.org 2013-07-30 17:14:35 +00:00
  • 9638564340 Adds no parent to talk folder. henrike@webrtc.org 2013-07-30 15:51:54 +00:00
  • 7f7162a003 Fix some chromium-style warnings in webrtc/modules/video_coding/ pbos@webrtc.org 2013-07-30 15:18:31 +00:00
  • e6c3966530 Fix some chromium-style warnings in webrtc/test/ pbos@webrtc.org 2013-07-30 13:08:38 +00:00
  • a6f56acc53 Fix some chromium-style warnings in webrtc/tools/ pbos@webrtc.org 2013-07-30 12:50:59 +00:00
  • 096515b070 Fix some chromium-style warnings in webrtc/modules/audio_device/ pbos@webrtc.org 2013-07-30 12:32:59 +00:00
  • 10bbfeff5b Apprtc: add 'event' parameter to onkeydown event handler. braveyao@webrtc.org 2013-07-30 09:27:49 +00:00
  • d818dcb939 Sets up framework for decoding with errors: collects frame sizes (in number of packets) in JB and passes this information to VCMSessionInfo with rtt_ms as FrameData. agalusza@google.com 2013-07-29 21:48:11 +00:00
  • a0b2f1794b Adds files still expected by the libjingle bots. henrike@webrtc.org 2013-07-29 21:34:08 +00:00
  • d6134c7cfd PeerConnectionTest.java: make the test work for the bots' v4l2loopback. - Make the test agnostic to the actual resolution used, since v4l2_file_player is playing a non-640x480 file (go/httfw) - Teach DeviceInfoLinux::FillCapabilityMap() about I420 since that's what v4l2_file_player is feeding. fischman@webrtc.org 2013-07-29 20:43:15 +00:00
  • 147d44a450 AppRTCDemo: replace the use of query-string parameters for pre-JB devices. fischman@webrtc.org 2013-07-29 19:07:33 +00:00
  • 7694562805 Land http://webrtc-codereview.appspot.com/1632005/ niklas.enbom@webrtc.org 2013-07-29 18:37:32 +00:00