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  • cb0ea43e57 audio_processing: Forces extended filter to be used in splitting filter test. bjornv@webrtc.org 2014-06-09 08:21:52 +00:00
  • 9c55f0f957 Rename neteq4 folder to neteq henrik.lundin@webrtc.org 2014-06-09 08:10:28 +00:00
  • 31f967c611 Fix Dr Memory download kjellander@webrtc.org 2014-06-09 07:30:37 +00:00
  • 9221ab420d Re-enable AudioCodingModuleMtTest again henrik.lundin@webrtc.org 2014-06-08 21:43:45 +00:00
  • 9359edaf78 PRESUBMIT: Add Android ARM64 and remove Linux TSan kjellander@webrtc.org 2014-06-08 17:55:51 +00:00
  • e3cdd9959e Revert "Fix the "Failed unprotect audio RTP packet" error when SCTP is bundled with audio." jiayl@webrtc.org 2014-06-06 22:32:57 +00:00
  • 013bdf802a APPRTCDemo(objc): Remove regex parsing in favor of JSON struct. tkchin@webrtc.org 2014-06-06 22:29:10 +00:00
  • 24c1778651 Revert r6358 "AppRTCDemo(Android): only stop the cameraThread's looper after stopping the camera." fischman@webrtc.org 2014-06-06 22:24:40 +00:00
  • c3288c130d Add OpenGL Android video renderer which can display multiple yuv420 images in a single GLSurfaceView. Start using new video renderer in AppRTC demo app. glaznev@webrtc.org 2014-06-06 21:57:46 +00:00
  • b8f582591f Use XErrorTrap in MouseCursorMonitorX11 to catch the error if the shared window has been closed. jiayl@webrtc.org 2014-06-06 21:42:00 +00:00
  • 171d94177b AppRTCDemo(Android): only stop the cameraThread's looper after stopping the camera. fischman@webrtc.org 2014-06-06 21:22:37 +00:00
  • b464618c84 Unbreak NDEBUG compile by RTC_UNUSED()ing an assert()d variable. fischman@webrtc.org 2014-06-06 20:13:49 +00:00
  • 745a39cced Fix the "Failed unprotect audio RTP packet" error when SCTP is bundled with audio. jiayl@webrtc.org 2014-06-06 19:24:02 +00:00
  • b273b60154 ViEAutoTestAndroid: Unbreak compile by casting void* to jobject. fischman@webrtc.org 2014-06-06 18:59:30 +00:00
  • 9512719569 AppRTCDemo(android): support app (UI) & capture rotation. fischman@webrtc.org 2014-06-06 18:40:44 +00:00
  • 42694c5937 VideoCaptureImpl::IncomingFrame(): avoid deadlock by acquiring _apiCs. fischman@webrtc.org 2014-06-06 18:28:28 +00:00
  • 91c910469f (Auto)update libjingle 68701339-> 68703656 buildbot@webrtc.org 2014-06-06 16:29:00 +00:00
  • 910473b31a Fix C++11 -Wnarrowing in channel_unittest.cc. pbos@webrtc.org 2014-06-06 15:44:00 +00:00
  • 7b6cbb3aa0 (Auto)update libjingle 68689052-> 68689059 buildbot@webrtc.org 2014-06-06 10:54:08 +00:00
  • 6ae48c6609 Make VideoSendStream/VideoReceiveStream configs const. pbos@webrtc.org 2014-06-06 10:49:19 +00:00
  • 4b83a471de (Auto)update libjingle 68646004-> 68648993 buildbot@webrtc.org 2014-06-05 21:11:28 +00:00
  • 4e5f65a4c6 Rebase webrtc/base with r6345 version of talk/base: cd webrtc/base svn diff -r 6249:6300 http://webrtc.googlecode.com/svn/trunk/talk/base > 6300.diff patch -p0 -i 6300.diff ls genericslot* | xargs rm cp ../../talk/base/sigslottester* . manual edits of sigslottester* to get rid of talk and talk_base. henrike@webrtc.org 2014-06-05 20:40:11 +00:00
  • 94454b71ad Fix the chain that propagates the audio frame's rtp and ntp timestamp including: * In AudioCodingModuleImpl::PlayoutData10Ms, don't reset the timestamp got from GetAudio. * When there're more than one participant, set AudioFrame's RTP timestamp to 0. * Copy ntp_time_ms_ in AudioFrame::CopyFrom method. * In RemixAndResample, pass src frame's timestamp_ and ntp_time_ms_ to the dst frame. * Fix how |elapsed_time_ms| is computed in channel.cc by adding GetPlayoutFrequency. wu@webrtc.org 2014-06-05 20:34:08 +00:00
  • 130fa64d4c AppRTCDemo(android): remove HTML/regex hackery in favor of JSON struct. fischman@webrtc.org 2014-06-05 20:31:41 +00:00
  • 65d61c3924 Opus send rate overflows if over 65 kbps tina.legrand@webrtc.org 2014-06-05 13:42:51 +00:00
  • b51d3ea593 Revert 6341 "Fixes and enables SystemDelayTests." bjornv@webrtc.org 2014-06-05 13:41:33 +00:00
  • 681aaae71b Remove remaining samples (AppRTC) since moved to Github kjellander@webrtc.org 2014-06-05 13:09:59 +00:00
  • 1f971b5788 Fixes and enables SystemDelayTests. bjornv@webrtc.org 2014-06-05 10:58:55 +00:00
  • 2f816bbae7 NetEq: Add thread annotation to const scoped_ptrs henrik.lundin@webrtc.org 2014-06-05 10:37:13 +00:00
  • eae7924836 Adding back platform specific renderer to video loopback test. mflodman@webrtc.org 2014-06-05 09:32:51 +00:00
  • 0d523eea83 Remove static initializer from WebRtcVideoEngine2. pbos@webrtc.org 2014-06-05 09:10:55 +00:00
  • aafd7a88c5 The correct fix of workaround in r6261. bjornv@webrtc.org 2014-06-05 08:53:51 +00:00
  • edbe886a0b common_audio/signal_processing: Removed macro WEBRTC_SPL_MUL_16_16_RSFT_WITH_FIXROUND bjornv@webrtc.org 2014-06-05 08:42:53 +00:00
  • ef92755780 Have RTX be enabled by setting an RTX payload type instead of by setting an RTX SSRC. stefan@webrtc.org 2014-06-05 08:25:29 +00:00
  • c578962006 Disable a test in libjingle_peerconnection_unittest for DrMemory henrik.lundin@webrtc.org 2014-06-05 07:38:31 +00:00
  • f1adbeedb4 (Auto)update libjingle 68562943-> 68571194 buildbot@webrtc.org 2014-06-04 21:57:16 +00:00
  • e6e139159f Android: cleanup gtest_target_type conditions. henrike@webrtc.org 2014-06-04 20:46:50 +00:00
  • 738df8913d Fix retain cycle in RTCEAGLVideoView. tkchin@webrtc.org 2014-06-04 20:19:39 +00:00
  • c6db88b0cf Make it possible to build webrtc for arm64. solenberg@webrtc.org 2014-06-04 17:15:42 +00:00
  • 6f237769b3 (Auto)update libjingle 68507189-> 68543735 buildbot@webrtc.org 2014-06-04 16:23:10 +00:00
  • 40b45fc07a (Auto)update libjingle 68506654-> 68507189 buildbot@webrtc.org 2014-06-04 14:48:33 +00:00
  • d3dcebf6b4 Disable P2PTransportChannelMultihomedTest.TestFailover under Memcheck henrik.lundin@webrtc.org 2014-06-04 13:23:07 +00:00
  • 147f4fe3c0 Disables SystemDelayTest.CorrectDelayDuringDrift on Android bjornv@webrtc.org 2014-06-04 13:17:58 +00:00
  • b616e1211f Disables some modules_unittests on Android. bjornv@webrtc.org 2014-06-04 12:12:58 +00:00
  • 4436b4436a Moved verbose logging in rtcp_receiver.cc to LS_VERBOSE. andresp@webrtc.org 2014-06-04 09:05:30 +00:00
  • 2bdd3994e3 Suppress memcheck error in VideoProcessorIntegrationTest henrik.lundin@webrtc.org 2014-06-04 08:47:29 +00:00
  • 19fc09efba Adding missing break in media_file_utility.cc. mflodman@webrtc.org 2014-06-04 05:21:56 +00:00
  • 0cdcd23a03 (Auto)update libjingle 68501302-> 68506654 buildbot@webrtc.org 2014-06-04 01:31:14 +00:00
  • af81b9bffd (Auto)update libjingle 68499439-> 68501302 buildbot@webrtc.org 2014-06-04 00:08:54 +00:00
  • 251fdf64cb (Auto)update libjingle 68495561-> 68499439 buildbot@webrtc.org 2014-06-03 23:43:48 +00:00
  • 09a71cd9ce talk/ios: Fixes source after corrupt sync in r6305 (which corrupted r6291). henrike@webrtc.org 2014-06-03 22:46:23 +00:00
  • 53217848b2 (Auto)update libjingle 68465410-> 68487517 buildbot@webrtc.org 2014-06-03 21:09:11 +00:00
  • 4ef254f781 Enable videoprocessor_integrationtest tests on android. marpan@webrtc.org 2014-06-03 16:42:03 +00:00
  • 83eb7dff5c PeerConnection(java): disable wait for flaky ICEConnection.COMPLETED. fischman@webrtc.org 2014-06-03 16:38:08 +00:00
  • ddc6bc9347 Revert 6312 "Re-enable AudioCodingModuleMtTest" turaj@webrtc.org 2014-06-03 15:25:34 +00:00
  • 289a35c56d Add empty webrtcmediaengine.cc. pbos@webrtc.org 2014-06-03 14:51:34 +00:00
  • 8d13cd1956 Re-enable AudioCodingModuleMtTest henrik.lundin@webrtc.org 2014-06-03 12:53:21 +00:00
  • 8e4401b5a0 Reformat integer accessors to look like their float counterparts kwiberg@webrtc.org 2014-06-03 10:04:13 +00:00
  • f2e4a99a39 Add kwiberg@webrtc.org to watchlist for audio_coding and audio_processing kwiberg@webrtc.org 2014-06-03 10:01:39 +00:00
  • b525a9d790 (Auto)update libjingle 68379861-> 68445177 buildbot@webrtc.org 2014-06-03 09:42:15 +00:00
  • 044bdacfef Remove kMaxWaitForStatsMs from tsanv2 compilation. pbos@webrtc.org 2014-06-03 09:40:01 +00:00
  • c0035a67a1 Remove an optimization that's no longer worth the extra complexity it causes kwiberg@webrtc.org 2014-06-03 09:10:06 +00:00
  • 34a08b4fb8 (Auto)update libjingle 68275107-> 68379861 buildbot@webrtc.org 2014-06-02 15:48:10 +00:00
  • a28c697d93 - Get rid of 'using' from .h - Add parenthesis to make order of evaluation clearer. solenberg@webrtc.org 2014-06-02 15:22:33 +00:00
  • 2f7c7ce020 Remove old perf_expectations no longer used. kjellander@webrtc.org 2014-06-02 15:03:21 +00:00
  • 2bd032e11c Disable MouseCursorMonitorTest henrik.lundin@webrtc.org 2014-06-02 14:52:34 +00:00
  • 4ecae6e753 Disable MouseCursorMonitorTest.FromScreen henrik.lundin@webrtc.org 2014-06-02 14:17:06 +00:00
  • fe41a8f68d Adding thread annotations to parts of Audio Coding Module henrik.lundin@webrtc.org 2014-06-02 11:45:26 +00:00
  • 2812b59acd Re-enables CommonFormats test for Android. bjornv@webrtc.org 2014-06-02 11:27:29 +00:00
  • 174a67439b Enable -Wall, -Wextra and -Wunused-variable for talk/ on clang. pbos@webrtc.org 2014-06-02 07:58:30 +00:00
  • 8a09af3f67 Fix the build error from OpenSSLStreamAdapter::SSLVerifyCallback jiayl@webrtc.org 2014-05-30 23:24:08 +00:00
  • 360507b12b VideoCaptureAndroid: don't synchronized on camera thread. fischman@webrtc.org 2014-05-30 23:17:38 +00:00
  • 0163674f99 Make OpenSSLStreamAdapter verify the leaf certificate digest for chained certificates. jiayl@webrtc.org 2014-05-30 23:14:08 +00:00
  • 222d8d3b1f Add a TSAN suppression for a benign TRACE_EVENT race. andrew@webrtc.org 2014-05-30 23:14:00 +00:00
  • 56d114627b Fix AppRTC target configuration in libjingle_examples.gyp. tkchin@webrtc.org 2014-05-30 23:04:39 +00:00
  • acca675bcf Implement mac version of AppRTCDemo. tkchin@webrtc.org 2014-05-30 22:26:06 +00:00
  • 9f8164c060 Fix two bugs in DataChannel state transition. 1. OnStateChange should not be fired if state is not changed. 2. RemotePeerRequestClose should be a no-op if it's already closed. jiayl@webrtc.org 2014-05-30 21:53:17 +00:00
  • 1fddd6185d Add a Reset() method to AudioFrame. andrew@webrtc.org 2014-05-30 17:28:50 +00:00
  • af48aaadf4 Disable AudioCodingModuleMtTest due to memcheck and tsan failures. andrew@webrtc.org 2014-05-30 17:11:15 +00:00
  • 1678db9df6 (Auto)update libjingle 68230113-> 68244456 buildbot@webrtc.org 2014-05-30 14:02:09 +00:00
  • 288bd15db8 Multi-threaded test for Audio Coding Module henrik.lundin@webrtc.org 2014-05-30 13:00:35 +00:00
  • b4e3c254ee Add native_test dependency to webrtc_perf_tests. pbos@webrtc.org 2014-05-30 12:42:10 +00:00
  • 420b2567f3 Fix bug where RTP headers in the packet history were replaced with the RTX wrapped headers. stefan@webrtc.org 2014-05-30 12:17:15 +00:00
  • a816180f93 Fixing a bug regarding VOE packet loss rate feedback to ACM minyue@webrtc.org 2014-05-30 09:28:07 +00:00
  • 6e732c6765 Revert 6272 "Update generated asm offsets scripts." sprang@webrtc.org 2014-05-30 09:19:03 +00:00
  • 540a2251aa (Auto)update libjingle 68230011-> 68230113 buildbot@webrtc.org 2014-05-30 07:40:35 +00:00
  • 35efb839ed Implement new-API test RecvStreamWithoutRtx. pbos@webrtc.org 2014-05-30 07:40:04 +00:00
  • c34bb3a886 Log default receive stream creation. pbos@webrtc.org 2014-05-30 07:38:43 +00:00
  • 198647473b Implement and fix new-API NackIsEnabled test. pbos@webrtc.org 2014-05-30 07:35:47 +00:00
  • 1d66be22c8 (Auto)update libjingle 68203780-> 68206793 buildbot@webrtc.org 2014-05-29 22:54:24 +00:00
  • 8dcd43c4f7 Make MediaSessionDescriptionFactory accept offers with UDP/TLS/RTP/SAVPF. This is the first step toward switching completely to UDP/TLS/RTP/SAVPF. jiayl@webrtc.org 2014-05-29 22:07:59 +00:00
  • abe01dd634 AppRTCDemo(android): run in full-screen & immersive mode. fischman@webrtc.org 2014-05-29 21:46:52 +00:00
  • 21a5d449b7 Increase VPMVideoDecimator's initial max_frame_rate_ to 60, which allow us potentially do 60fps. wu@webrtc.org 2014-05-29 19:43:26 +00:00
  • 7a9a3b70b3 * Revert clock.cc changes made in 6178, but keep the changes to the test. * Use the new appoach proposed by jib in https://review.webrtc.org/10439004/ to fix the windows clock issue. wu@webrtc.org 2014-05-29 19:40:28 +00:00
  • 2a8efa8971 Update generated asm offsets scripts. fgalligan@google.com 2014-05-29 17:08:34 +00:00
  • caa01b172e Rebase webrtc/base with r6250: cd webrtc/base svn diff -r 6249:6250 http://webrtc.googlecode.com/svn/trunk/talk/base > 6250.diff patch -p0 -i 6250.diff henrike@webrtc.org 2014-05-29 15:53:39 +00:00
  • 5dc51fbe50 Closes the DataChannel when the send buffer is full or on transport errors. As stated in the spec. jiayl@webrtc.org 2014-05-29 15:33:54 +00:00
  • 001fd2d503 Fire OnRenegotiationNeeded only for the first SCTP DataChannel. Subsequent DataChannels do not need renegotiation since SCTP data streams are not negotiated through SDP. jiayl@webrtc.org 2014-05-29 15:31:11 +00:00
  • 9aa7d8df95 Increase the threshold for CallPerfTest.CaptureNtpTimeWithNetworkDelay to avoid flaky. wu@webrtc.org 2014-05-29 05:03:52 +00:00
  • d6a0efdc86 VideoCaptureAndroid: quit & join the camera thread on stopCapture. fischman@webrtc.org 2014-05-28 18:37:07 +00:00