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  • 31c8d2eac5 Update with new default boringssl no-aes cipher suites. Re-enable tests. Torbjorn Granlund 2016-01-14 15:17:53 +01:00
  • e5e0e57bdf Revert of Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket. (patchset #3 id:40001 of https://codereview.webrtc.org/1577873003/ ) tommi 2016-01-14 04:56:59 -08:00
  • 688e308a35 Re-land: "Use an explicit identifier in Config" aluebs 2016-01-14 04:32:46 -08:00
  • 7307952a5b Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket. Stefan Holmer 2016-01-14 13:15:45 +01:00
  • 268493a96b Revert of Delete remnants of non-square pixel support from cricket::VideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/1586613002/ ) nisse 2016-01-14 02:35:22 -08:00
  • 35aae2e5ca Remove libfuzzer trybot from default trybot set. kjellander 2016-01-14 02:03:23 -08:00
  • ff2a6351e0 Add ramp-up tests for transport sequence number with and w/o audio. Stefan Holmer 2016-01-14 10:00:21 +01:00
  • 709513d413 Delete remnants of non-square pixel support from cricket::VideoFrame. nisse 2016-01-13 23:43:52 -08:00
  • beed8280d8 Fix IPAddress::ToSensitiveString() to avoid dependency on inet_ntop(). Sergey Ulanov 2016-01-13 18:14:49 -08:00
  • 2d110be77f Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ ) deadbeef 2016-01-13 12:00:26 -08:00
  • 8432e1f4b8 Re-enable tests that failed under Linux_Msan. marpan 2016-01-13 08:35:44 -08:00
  • fca54f41ad Revert of Use an explicit identifier in Config (patchset #4 id:60001 of https://codereview.webrtc.org/1538643004/ ) tommi 2016-01-13 08:12:00 -08:00
  • 09d944f82d Roll chromium_revision 346fea9..099be58 (369082:369139) kjellander 2016-01-13 07:52:38 -08:00
  • 306efadffa Disable WebRtcVideoChannel2BaseTest.SendManyResizeOnce for TSan kjellander 2016-01-13 07:51:28 -08:00
  • 292e192f17 Add build_protobuf variable. kjellander 2016-01-13 05:46:58 -08:00
  • a276e73168 Clean the code for external denoiser. jackychen 2016-01-13 05:36:31 -08:00
  • 2f7dea164d [rtp_rtcp] rtcp::Empty moved into own file and renamed to CompoundPacket on the way danilchap 2016-01-13 02:03:04 -08:00
  • ea8c0f6fcb Fix capture ntp time issue introduced with r11187. Stefan Holmer 2016-01-13 08:58:38 +01:00
  • 365543d0e7 Roll chromium_revision 131167b..346fea9 (368784:369082) kjellander 2016-01-12 21:05:25 -08:00
  • 25249d92d3 Use an explicit identifier in Config aluebs 2016-01-12 18:50:25 -08:00
  • e591f9377f Storing raw audio sink for default audio track. deadbeef 2016-01-12 16:44:59 -08:00
  • 6955870806 Convert channel counts to size_t. Peter Kasting 2016-01-12 16:26:35 -08:00
  • 92e677a1f8 [rtp_rtcp] rtcp::Sli packet moved into own file and got Parse function danilchap 2016-01-12 10:04:52 -08:00
  • 5584bf4c4d Make :rtc_base_approved a public dep of :rtc_base. jbroman 2016-01-12 09:46:55 -08:00
  • e84e96e8be NetEq: Fix a typo in a comment Henrik Lundin 2016-01-12 16:36:13 +01:00
  • 36220ae24f Slap deprecation notices on Pass methods kwiberg 2016-01-12 07:24:19 -08:00
  • d20e651327 Fix test bug introduced in r11101. Stefan Holmer 2016-01-12 15:51:22 +01:00
  • 3e1cfa7edb Delete unused method webrtc::VideoRendererInterface::SetSize. nisse 2016-01-12 06:39:20 -08:00
  • 3235a27e7a Updated chromium/.gclient and sync_chromium.py to not ignore third_party/ffmpeg. Was forgotten in this CL: https://codereview.webrtc.org/1575913003/ Henrik Boström 2016-01-12 15:05:29 +01:00
  • 2845a02339 Remove unused enum RTPDirections. terelius 2016-01-12 05:00:53 -08:00
  • 3842c5c7f7 Wire-up BWE feedback for audio receive streams. Stefan Holmer 2016-01-12 13:55:00 +01:00
  • 6183de6da0 Remove tools/refactoring. Peter Boström 2016-01-12 13:40:58 +01:00
  • 127782bbb1 Add default dummy implementation of cricket::VideoRenderer::SetSize, to easy later removal. nisse 2016-01-12 03:39:14 -08:00
  • 16979e35b3 Update .gitignore Henrik Kjellander 2016-01-12 08:17:37 +01:00
  • 67e94fb6f2 Add unit test for stand-alone denoiser and fixed some bugs. jackychen 2016-01-11 21:34:07 -08:00
  • b2328d11dc Remove additional channel constraints when Beamforming is enabled in AudioProcessing aluebs 2016-01-11 20:32:29 -08:00
  • e93ad1b129 Roll chromium_revision 8c958e0..131167b (368561:368784) kjellander 2016-01-11 20:04:30 -08:00
  • 2a34688f86 Make Beamforming dynamically settable for Android platform builds aluebs 2016-01-11 18:04:30 -08:00
  • 2bc63a1dd3 clang-format audio_device/mac. andrew 2016-01-11 15:59:17 -08:00
  • a7446d2a50 Change DTLS default from 1.0 to 1.2 for webrtc. Guo-wei Shieh 2016-01-11 15:27:03 -08:00
  • f6c318ebae Update API for Objective-C RTCMediaSource. Jon Hjelle 2016-01-11 14:39:01 -08:00
  • e799badacc Move Objective-C video renderers to webrtc/api/objc. Jon Hjelle 2016-01-11 13:47:11 -08:00
  • 81028796bc Update API for Objective-C RTCMediaStreamTrack. Jon Hjelle 2016-01-11 13:16:13 -08:00
  • a2c353f815 Update API for Objective-C RTCStats. Jon Hjelle 2016-01-11 13:11:38 -08:00
  • 7e8145f05d [rtp_rtcp] rtcp::Tmmbr moved into own file danilchap 2016-01-11 11:49:19 -08:00
  • 27ed3cc28c SCTP: Stopped accepting SSRCs higher than max. Seems to fix asan-related crash. lally 2016-01-11 10:24:33 -08:00
  • a9a1d2acaf H.264: Default flags and pulling in openh264 and ffmpeg. hbos 2016-01-11 10:19:02 -08:00
  • 7823495698 Move RTCI420Frame to webrtc/api/objc/RTCVideoFrame with minor style changes. Jon Hjelle 2016-01-11 09:47:07 -08:00
  • fd99dea4f6 Roll chromium_revision 42ab10e..8c958e0 (368534:368561) kjellander 2016-01-11 04:57:22 -08:00
  • ef3d805f6e [rtp_rtcp] rtcp::Tmmbn moved into own file explicetly unchanged. danilchap 2016-01-11 03:31:08 -08:00
  • d36efeb622 Roll chromium_revision e738b54..42ab10e (368533:368534) kjellander 2016-01-10 19:55:57 -08:00
  • 4de003722d Roll chromium_revision 7d97c94..e738b54 (368514:368533) kjellander 2016-01-10 03:58:36 -08:00
  • 3c05e6c9c3 Disable EndToEndTest.TransportSeqNumOnAudioAndVideo for Dr Memory. kjellander 2016-01-09 21:46:36 -08:00
  • daa87497e8 Revert of Roll chromium_revision 7d97c94..951c006 (368514:368525) (patchset #1 id:1 of https://codereview.webrtc.org/1577573002/ ) guoweis 2016-01-09 19:27:29 -08:00
  • db21f633a2 fix GN build break on native_client Guo-wei Shieh 2016-01-09 13:12:04 -08:00
  • 6109fc13aa Roll chromium_revision 7d97c94..951c006 (368514:368525) kjellander 2016-01-09 11:59:32 -08:00
  • 0697db6f1b Roll chromium_revision 8a15a7f..7d97c94 (368391:368514) kjellander 2016-01-08 20:09:29 -08:00
  • 684e995464 Disable 2 video tests which fail on DrMemoryFull Guo-wei Shieh 2016-01-08 19:03:38 -08:00
  • f475d365a2 Properly handle different transports having different SSL roles. Taylor Brandstetter 2016-01-08 15:35:57 -08:00
  • 25702cb162 Misc. small cleanups. pkasting 2016-01-08 13:50:27 -08:00
  • 5de688ed34 Roll chromium_revision ede5d4f..8a15a7f (368310:368391) kjellander 2016-01-08 13:09:28 -08:00
  • 49c454e748 Cleaning neteq_unittest resource files. minyue 2016-01-08 11:30:14 -08:00
  • f1685c771d Disable RampUpTest.UpDownUp* in webrtc_perf_tests on Mac kjellander 2016-01-08 10:43:41 -08:00
  • e74eef19bd Add CreateSend/ReceiveTransport() methods to CallTest. stefan 2016-01-08 06:47:13 -08:00
  • 37ebcf0ce5 Reland "Add APK targets to build libjingle tests for Android." patchset #10 id:180001 of https://codereview.webrtc.org/1511633002/ phoglund 2016-01-08 05:04:57 -08:00
  • b71b4f0c7a Update attributes to match gclibc's ansidecl.h kjellander 2016-01-08 04:51:38 -08:00
  • 004851caca Roll chromium_revision 32569c6..ede5d4f (368258:368310) kjellander 2016-01-08 03:59:09 -08:00
  • e1ca167217 Add tracing to NetEqImpl::GetAudio henrik.lundin 2016-01-08 03:50:08 -08:00
  • ec80f03b3c Check the mic volume only periodically on Mac. andrew 2016-01-08 01:16:17 -08:00
  • fbeb97e01f Fix clang warning in peerconnection_jni.cc perkj 2016-01-08 00:43:07 -08:00
  • 59bac1a4c5 Fix for stats updated twice when switching content type (realtime <-> screenshare). Add unittest. asapersson 2016-01-07 23:36:00 -08:00
  • 95ab30ce88 Roll chromium_revision 6dd04c2..32569c6 (368115:368258) kjellander 2016-01-07 23:31:08 -08:00
  • a2b1e03c66 Disable AudioDeviceAPITest.MicrophoneVolumeTests on Linux. kjellander 2016-01-07 23:26:06 -08:00
  • 893505d0fb Adding unit test to ensure TURN server priorities are unique. Taylor Brandstetter 2016-01-07 15:12:48 -08:00
  • e5ba13bc09 Adding a way for a Java RtpSender to set a track without taking ownership. Taylor Brandstetter 2016-01-07 15:11:25 -08:00
  • ced8ec9069 Roll chromium_revision bd5949f..6dd04c2 (368055:368115) kjellander 2016-01-07 13:22:46 -08:00
  • bedc17be5c Fixing integer underflow in FileAudioDevice (webrtc issue 4554) A.Brauckmann 2016-01-07 12:38:31 -08:00
  • 69387930e0 vp9 tests: Adjust some parameters and re-enable the tests. Marco 2016-01-07 12:00:38 -08:00
  • 6f5ca080b8 Update API for Objective-C RTCMediaConstraints. hjon 2016-01-07 09:29:29 -08:00
  • 9fea80f50d Add audio streams to CallTest and a first A/V call test. Stefan Holmer 2016-01-07 17:43:18 +01:00
  • ecd21b481f Add ImplementationName to SimulcastEncoderAdapter. pbos 2016-01-07 08:03:05 -08:00
  • 01f364ef06 Remove always-on options in OveruseFrameDetector. Peter Boström 2016-01-07 16:38:25 +01:00
  • 30166cb1a8 iOS stability improvement for device switching, including BT devices henrika 2016-01-07 07:23:01 -08:00
  • 7776e782d6 Remove unused methods in VideoCodingModule. Peter Boström 2016-01-07 15:42:47 +01:00
  • 3886fc8f57 Use pointer to generated FEC packet. Peter Boström 2016-01-07 15:33:49 +01:00
  • 46ea3ce580 AudioDeviceTest.StartPlayoutOnTwoInstances now verifies two active playing streams henrika 2016-01-07 15:02:53 +01:00
  • a49ad9856b Roll chromium_revision 4662d4f..bd5949f (368042:368055) kjellander 2016-01-07 05:57:46 -08:00
  • cea7c2f783 Replace manual casting to rvalue reference with calls to std::move kwiberg 2016-01-07 05:52:04 -08:00
  • a46a4c92d0 Roll chromium_revision 2a70cb1..4662d4f (367468:368042) kjellander 2016-01-07 02:54:22 -08:00
  • 1fe48a5e1d Add implementation in metrics.h that uses atomic pointer. asapersson 2016-01-07 01:02:42 -08:00
  • 4331fcd517 Remove duplicate code in SocketDispatcher jbauch 2016-01-06 22:20:28 -08:00
  • 44cc795016 Roll chromium_revision 4df108a..2a70cb1 (367307:367468) kjellander 2016-01-06 22:12:27 -08:00
  • 4f3c772015 Update default trybot names for changed Mac bots kjellander 2016-01-06 12:12:09 -08:00
  • 67e83d6539 Update API for Objective-C RTCSessionDescription. Jon Hjelle 2016-01-06 12:05:22 -08:00
  • 29d5e570b5 Update API for Objective-C RTCIceCandidate. Jon Hjelle 2016-01-06 11:49:11 -08:00
  • 1aa6e4e095 Update default trybot names for changed Mac bots kjellander 2016-01-06 09:14:42 -08:00
  • 335ecf59d0 Disable VideoCaptureTest.Capabilities and CreateDelete fails on Mac kjellander 2016-01-06 05:23:09 -08:00
  • b6802749f1 Fix a flaky turnport test failure Honghai Zhang 2016-01-05 13:51:03 -08:00
  • 6b9ab9204b Cease all future TURN requests when a TURN refresh request fails for a given TURN port. This fixes an assert error in Turnport::OnSendStunPacket honghaiz 2016-01-05 09:06:12 -08:00
  • 37389b42b4 Don't delete an ICE connection until it has been pruned or timed out on writing in the case where it hasn't received anything yet. Deleting an ICE connection before it is pruned or timed out when it hasn't received anything yet leads to ICE connections being deleted before they have a chance to send a ping and receive a response. BUG= honghaiz 2016-01-04 21:57:33 -08:00