Commit Graph

  • 34c50c1de1 Makes OpensSL default audio implementation/device on Android. henrike@webrtc.org 2013-09-27 22:34:35 +00:00
  • a39b323a7e Add tools/sharding_supervisor to .gitignore kjellander@webrtc.org 2013-09-27 20:20:47 +00:00
  • 8b7ec82cf4 Exclude P2PTransportChannelSameNatTest.TestConesBehindSameCone for TSan Linux kjellander@webrtc.org 2013-09-27 20:03:03 +00:00
  • 4475905613 Disable flaky RapidSpeakerChange test. andrew@webrtc.org 2013-09-27 18:24:40 +00:00
  • 6049787252 Add protection to few more methods of AudioDeviceLinuxALSA. Those methods can be called from a different thread. wu@webrtc.org 2013-09-27 18:19:25 +00:00
  • 137b3793d9 Only use -lm on Linux in ISAC. andrew@webrtc.org 2013-09-27 18:16:28 +00:00
  • 287f07b174 Add sharding_supervisor to DEPS to prepare for swarm/isolated testing. kjellander@webrtc.org 2013-09-27 16:57:10 +00:00
  • 2e246b4e78 Remove test parameters from CallTest. pbos@webrtc.org 2013-09-27 10:54:10 +00:00
  • 3223a3d4ee Roll libvpx 212975:225010 to pick up iOS Release fixes fischman@webrtc.org 2013-09-26 15:59:48 +00:00
  • 663da0a8fc With ACM2 and NetEq4, VoE fuzz test very often fails. minyue@webrtc.org 2013-09-26 15:21:26 +00:00
  • f26e8f6f57 Remove include_dirs from tools. pbos@webrtc.org 2013-09-26 14:04:27 +00:00
  • f8b296628f Remove include_dirs from test. pbos@webrtc.org 2013-09-26 14:03:57 +00:00
  • 544b17c6a9 Implemented AutoMuter in MediaOptimization henrik.lundin@webrtc.org 2013-09-26 12:05:15 +00:00
  • 04b61790d1 Remove include_dirs from pacing. pbos@webrtc.org 2013-09-26 09:23:31 +00:00
  • 97eefb77e0 Remove include_dirs from remote_bitrate_estimator. pbos@webrtc.org 2013-09-26 09:22:54 +00:00
  • 339fe12f67 Remove include_dirs from bitrate_controller. pbos@webrtc.org 2013-09-26 09:22:34 +00:00
  • 054ccd2e35 Remove include_dirs from video_coding. pbos@webrtc.org 2013-09-26 09:22:09 +00:00
  • 73f207611c Remove include_dirs from video_processing. pbos@webrtc.org 2013-09-26 09:17:56 +00:00
  • dc3fa08331 Remove include_dirs from rtp_rtcp. pbos@webrtc.org 2013-09-26 09:17:13 +00:00
  • 7b75ac6756 Sync-packet insertion into NetEq4. This is related to r3883 & r4052 for NetEq 3. turaj@webrtc.org 2013-09-26 00:27:56 +00:00
  • 6b1e21924a Move the Config DelayCorrection struct to audio_processing.h. andrew@webrtc.org 2013-09-25 23:46:20 +00:00
  • 1760a17b8e Add an extended filter mode to AEC. andrew@webrtc.org 2013-09-25 23:17:38 +00:00
  • becbefaee6 Fix WindowCapturerWin to capture window decorations after window size changes. sergeyu@chromium.org 2013-09-25 22:56:59 +00:00
  • 3fdeddb59a Disable a NetEq unittest on Android. The test tries to register iSAC-swb as send codec and fails. turaj@webrtc.org 2013-09-25 22:19:22 +00:00
  • 3e7703640f Remove unused constants, so chrome can enable a warning for that. Patch from thakis@ niklas.enbom@webrtc.org 2013-09-25 22:05:05 +00:00
  • cecaae2e4c Updated WebRTC version to 3.43 TBR=mallinath@webrtc.org elham@webrtc.org 2013-09-25 21:45:23 +00:00
  • 0c0fae8a5e Re-enable verbose logging in NetEq4. turaj@webrtc.org 2013-09-25 17:42:17 +00:00
  • 69fc315fd9 Convert DeviceInfoImpl::_captureCapabilities from a map to a vector. fischman@webrtc.org 2013-09-25 17:01:42 +00:00
  • ce014d97cd Revert 4837 "Add an extended filter mode to AEC." asapersson@webrtc.org 2013-09-25 12:27:27 +00:00
  • 2f240b43f5 Disable some flaky libjingle base tests. andrew@webrtc.org 2013-09-25 02:33:50 +00:00
  • 26e02f0ee4 Add an extended filter mode to AEC. andrew@webrtc.org 2013-09-25 02:17:47 +00:00
  • d6a7a5f385 Small fixes to run ACM2 tests. turaj@webrtc.org 2013-09-25 01:09:23 +00:00
  • ff43c85ef1 API add to set background noise mode. turaj@webrtc.org 2013-09-25 00:07:27 +00:00
  • 8d757ac0a2 Fix window capturer not to leak HDC. sergeyu@chromium.org 2013-09-24 23:13:51 +00:00
  • 958cdf68f3 Fix window capturer to stop capturing when the target is minimized. sergeyu@chromium.org 2013-09-24 22:10:13 +00:00
  • f832a551cc Disable flaky TestPartialFrameHeader. andrew@webrtc.org 2013-09-24 20:09:30 +00:00
  • c2ac5ed588 Add more TSAN suppressions for libjingle_media_unittest. andrew@webrtc.org 2013-09-24 18:45:19 +00:00
  • 641587f938 Disable some VP8 tests on Android. andrew@webrtc.org 2013-09-24 18:43:28 +00:00
  • f0f92fae12 Disable flaky SendDataMultipleClocks. andrew@webrtc.org 2013-09-24 16:26:41 +00:00
  • 9b6eefcedf Fix for Heap-use-after-free in webrtc::voe::Channel::SendRTCPPacket henrika@webrtc.org 2013-09-24 15:42:49 +00:00
  • ebd3ab0ce1 Add libjingle_peerconnection_objc_test to buildbot_tests.py kjellander@webrtc.org 2013-09-24 13:41:16 +00:00
  • 6e86349273 Disable tests that crash the OS X kernel when run under memcheck. kjellander@webrtc.org 2013-09-24 12:34:51 +00:00
  • b0e6eb50b5 Revert r4823 "Reenable test and remove flaky expects." stefan@webrtc.org 2013-09-24 10:38:57 +00:00
  • 01aad09a01 Reenable test and remove flaky expects. stefan@webrtc.org 2013-09-24 10:16:52 +00:00
  • b426c469b9 MediaOptimization: Converting a few members to scoped_ptrs henrik.lundin@webrtc.org 2013-09-24 07:41:53 +00:00
  • 6ffc74ee0e Disable flaky RunsRtpRtcpTestWithoutErrors. andrew@webrtc.org 2013-09-23 23:25:39 +00:00
  • eb524d997b Remove deprecated AudioCodingModule::Destroy. andrew@webrtc.org 2013-09-23 23:02:24 +00:00
  • 1112c30e1e Update libjingle to 53057474. mallinath@webrtc.org 2013-09-23 20:34:45 +00:00
  • e2af622edf - Reset capture deltas at resolution change. - Applied smoothing of capture jitter. - Adjusted thresholds. asapersson@webrtc.org 2013-09-23 20:05:39 +00:00
  • bec11ef632 Reformatting media_optimization.cc and .h henrik.lundin@webrtc.org 2013-09-23 19:54:25 +00:00
  • b533a82bf9 Disabled flaky tests. BUG=2409 R=andrew@webrtc.org, mallinath@webrtc.org asapersson@webrtc.org 2013-09-23 19:47:49 +00:00
  • 7a7b929882 Updated dc1.html to support SCTP transport. vikasmarwaha@webrtc.org 2013-09-23 18:03:33 +00:00
  • 334865e2a1 Re-enable VideoCaptureTest.CreateDelete Previously the test insisted on non-zero delay, but 0 is not a crazy delay value (esp. on a fake camera device!). Instead we now test for delay>=0 being set at all. fischman@webrtc.org 2013-09-23 17:58:45 +00:00
  • 038e8e64ef Updated WebRTC version to 3.42 elham@webrtc.org 2013-09-23 16:45:25 +00:00
  • 98fcd2d4c3 Adding unit tests for default temporal layer strategy. andresp@webrtc.org 2013-09-23 11:12:59 +00:00
  • cdd3d4d139 Revert test change in r4808. stefan@webrtc.org 2013-09-23 09:43:07 +00:00
  • 269dd4264f Reduce flakiness in network down test. stefan@webrtc.org 2013-09-23 08:42:39 +00:00
  • 63fe8e1f38 Enable SetInitialPlayoutDelay on Android. dwkang@webrtc.org 2013-09-23 05:42:22 +00:00
  • 2edb642810 Fix bugs in DesktopRegion::Subtract(). sergeyu@chromium.org 2013-09-20 21:29:18 +00:00
  • cee0dfb57a Made sure that DTLS/SRTP is set to false in apprtc demo when testing loopback. See crbug/294881 for details. vikasmarwaha@webrtc.org 2013-09-20 21:26:07 +00:00
  • 10e6cc7e23 VAD changes ported to ACM2. turaj@webrtc.org 2013-09-20 16:38:26 +00:00
  • 362a55e7b0 Address Windows 64-bits warnings. turaj@webrtc.org 2013-09-20 16:25:28 +00:00
  • 0e63e76781 Enable FEC for VideoSendStream. pbos@webrtc.org 2013-09-20 11:56:26 +00:00
  • 9c74be7bd1 Disable flaky video capture test. stefan@webrtc.org 2013-09-20 08:52:31 +00:00
  • 4f3624d39e Avoid recursively taking critical section. stefan@webrtc.org 2013-09-20 07:43:17 +00:00
  • dd57cd6ed5 Removing the tsan text exclusion since the tests should be passing now. BUG=2299, 2290, 2291 R=wu@webrtc.org jiayl@webrtc.org 2013-09-20 00:51:53 +00:00
  • d29ab4e17c Suppress SSL error strings on mac_asan to unbreak that build Example borkedness: http://chromegw/i/client.webrtc/builders/Mac%20Asan/builds/642/steps/libjingle_p2p_unittest/logs/stdio fischman@webrtc.org 2013-09-19 23:54:56 +00:00
  • 76fe9309b9 Use link_settings instead of all_dependent_settings to pacify xcode gyp generator fischman@webrtc.org 2013-09-19 21:11:08 +00:00
  • ccddd0a941 Roll webrtc's chromium_revision 217707:224141 fischman@webrtc.org 2013-09-19 20:27:32 +00:00
  • 6917e19ad4 Rename EngineTest to CallTest. pbos@webrtc.org 2013-09-19 14:22:12 +00:00
  • a03e34e9ab Heap-use-after-free in WebRtcNetEQ_RecInRTPStruct tina.legrand@webrtc.org 2013-09-19 13:32:14 +00:00
  • ab6549562b Refactor frame generation code so it can be used by multiple modules. andresp@webrtc.org 2013-09-19 12:14:03 +00:00
  • 7a30dfdc69 Disable NACK bandwidth statistics test due to being too flaky. stefan@webrtc.org 2013-09-19 12:08:55 +00:00
  • b5a191bfe7 Fixes a flake in network down tests. stefan@webrtc.org 2013-09-19 11:14:35 +00:00
  • d8a9b86671 Disable tests for TSan v2 kjellander@webrtc.org 2013-09-19 07:15:36 +00:00
  • 967bfff54d Update talk to 52534915. wu@webrtc.org 2013-09-19 05:49:50 +00:00
  • 532f3dc548 Compile ACM2 and ACM1. turaj@webrtc.org 2013-09-19 00:12:23 +00:00
  • f3930e941c Small refactoring of AudioProcessing use in channel.cc. andrew@webrtc.org 2013-09-18 22:37:32 +00:00
  • 0d5da25e6c NetEq4: Making a few more members scoped_ptrs henrik.lundin@webrtc.org 2013-09-18 21:12:38 +00:00
  • 5a43370cdb Dedicated speed test for NetEq3 henrik.lundin@webrtc.org 2013-09-18 20:58:33 +00:00
  • 7a968a8f07 Add more TSan and Dr Memory suppressions for modules_unittests kjellander@webrtc.org 2013-09-18 19:18:20 +00:00
  • 8d1e4d6149 Increase the dtmfsender test toleration to 100ms to avoid flaky. wu@webrtc.org 2013-09-18 18:01:07 +00:00
  • 8bf755d5c5 MIPS optimizations for the functions WebRtcSpl_SqrtFloor, WebRtcSpl_CrossCorrelation, WebRtcSpl_ScaleAndAddVectorsWithRound and the inline functions from signal_processing spl_inl.h file. andrew@webrtc.org 2013-09-18 17:40:46 +00:00
  • 5f1051631a Fix disabling of tests. stefan@webrtc.org 2013-09-18 15:28:10 +00:00
  • 1c77dfd521 Revert r4772 "Compile ACM1 and ACM2." stefan@webrtc.org 2013-09-18 12:34:05 +00:00
  • 40d3fc65f5 NetEq4: Make some DSP operation classes member variables henrik.lundin@webrtc.org 2013-09-18 12:19:50 +00:00
  • 8db81c5112 Fix races in vcm::Process(). stefan@webrtc.org 2013-09-18 11:57:34 +00:00
  • e75a1bf45f Break out glue for old->new Transport. pbos@webrtc.org 2013-09-18 11:52:42 +00:00
  • fe84fda488 Changing 'frame' method to 'bounds' method. sjlee@webrtc.org 2013-09-18 01:37:31 +00:00
  • 367baa6eb3 Compile ACM1 and ACM2. turaj@webrtc.org 2013-09-18 00:36:11 +00:00
  • c8dea6a00f Use the native sample rate for OpenSL recording. henrike@webrtc.org 2013-09-17 18:44:51 +00:00
  • bf00740c92 Adds a new voice engine warning for the typing noise off state. The old VE_TYPING_NOISE_WARNING is unchanged and fired whenever typing noise is detected. The new VE_TYPING_NOISE_OFF_WARNING is fired when typing noise was detected and is gone now. This is necessary for converting the typing state to a PeerConnection stats. jiayl@webrtc.org 2013-09-17 18:09:20 +00:00
  • da79008ab4 Disabling crashing or flaky tests in peerconnection_unittest. stefan@webrtc.org 2013-09-17 13:11:38 +00:00
  • 32d640e03d Fix typo in r4765. pbos@webrtc.org 2013-09-17 10:36:30 +00:00
  • da2c4cede0 Fix dangling pointer _encoder in video_sender.cc. pbos@webrtc.org 2013-09-17 09:38:41 +00:00
  • be63fd644f Initialize CodecInst structs in test_api_audio.cc. pbos@webrtc.org 2013-09-17 09:33:17 +00:00
  • d1fc5d4e17 Dedicated speed test for NetEq4 henrik.lundin@webrtc.org 2013-09-17 08:38:02 +00:00
  • 28a331eede Add support for multiple report blocks. stefan@webrtc.org 2013-09-17 07:49:56 +00:00
  • fc10c5c97f This is related to https://code.google.com/p/webrtc/issues/detail?id=1341 sjlee@webrtc.org 2013-09-17 05:15:37 +00:00
  • e6ac163145 This is related to https://code.google.com/p/webrtc/issues/detail?id=846 sjlee@webrtc.org 2013-09-17 05:15:19 +00:00