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  • ca91e38a3a Update API for Objective-C RTCAudioTrack and RTCVideoTrack. Jon Hjelle 2016-01-21 15:36:47 -08:00
  • 97888bd95a Swap use of CriticalSectionWrapper for rtc::CriticalSection in webrtc/video. While doing this, I made a couple of minor changes: * Removed unused variables (one lock and one video frame variable) * Switched over to a scoped lock in remb.cc and removed an if() in a function where we can just return the expression being checked. Tommi 2016-01-21 23:24:59 +01:00
  • 7ac8babbc6 Move RTCAVFoundationCapturer to webrtc/api/objc. Jon Hjelle 2016-01-21 11:44:55 -08:00
  • 891a446a92 Update/move RTCVideoRendererAdapter to webrtc/api/objc. Jon Hjelle 2016-01-21 11:42:05 -08:00
  • 31fc21f454 Swap use of CriticalSectionWrapper with rtc::CriticalSection in voice_engine/ Also remove mischievous tab character! This is a part of getting rid of CriticalSectionWrapper and makes the code slightly simpler. tommi 2016-01-21 10:37:37 -08:00
  • 8947a01e05 Fixing an uninitialized variable in webrtcsession_unittest. deadbeef 2016-01-21 10:26:38 -08:00
  • fa156696dd Fix probing breakage with send-side BWE introduced by r11322. stefan 2016-01-21 08:55:03 -08:00
  • fea3dd83fc Fix a bug in InputAudioFile::Read henrik.lundin 2016-01-21 08:19:57 -08:00
  • af9e6637c0 Make rtc::CriticalSection lockable from f() const. Peter Boström 2016-01-21 16:56:52 +01:00
  • 3c16978c84 Remove cast to LocalAudioSource from AudioRtpSender. Tommi 2016-01-21 16:12:17 +01:00
  • 32be07bc36 Remove RentACodec::GetEncoderStack kwiberg 2016-01-21 07:10:01 -08:00
  • 693a1147c6 Add stefan@webrtc.org to webrtc/test/OWNERS. Peter Boström 2016-01-21 15:32:50 +01:00
  • 3313ec901f Enable transport seq num extension on receive channel to suppress log warning. stefan 2016-01-21 06:32:43 -08:00
  • d664836efa Added EncodedImage::GetBufferPaddingBytes. hbos 2016-01-21 05:43:11 -08:00
  • 429c345b02 Fixes a bug which incorrectly logs incoming RTCP as outgoing. terelius 2016-01-21 05:42:04 -08:00
  • b304e2686d Roll chromium_revision 1728ddf..4623ce8 (370595:370665) kjellander 2016-01-21 05:09:27 -08:00
  • 1f150b35f8 Add new NetEq resources to modules_unittests.isolate. kjellander 2016-01-21 04:42:21 -08:00
  • 902c03e724 rtc_use_h264 flag (replacing use_third_party_h264 flag) for building OpenH264/FFmpeg, false by default but can be overridden in supplement.gypi and build_overrides/webrtc.gni. hbos 2016-01-21 03:34:40 -08:00
  • 0b98cf72c6 Delete CaptureRenderAdapter::VideoRenderInfo struct, it is unused since the recent deletion of SetSize. nisse 2016-01-21 03:04:30 -08:00
  • 5082c83320 Make type and constructors in EglBase14 public. noahric 2016-01-21 02:12:53 -08:00
  • becf9ee48c Roll chromium_revision 6a04368..1728ddf (370362:370595) kjellander 2016-01-21 01:45:55 -08:00
  • d9f641e341 Reallocate encoded buffer size if needed. Initially set to the input image size. Issue may occur for very small input images (e.g. 4x4) when encoded image length > input image size. asapersson 2016-01-21 01:11:35 -08:00
  • d26fadb454 Delete GetRenderer method, used only by the tests. nisse 2016-01-20 23:26:37 -08:00
  • 057ecf01e4 Making WebRtcSession fire a destroyed signal. deadbeef 2016-01-20 14:30:43 -08:00
  • da99da81c9 Update API for Objective-C RTCPeerConnectionFactory. Jon Hjelle 2016-01-20 13:40:30 -08:00
  • 065aacc249 Move RTCVideoSource to webrtc/api/objc. Jon Hjelle 2016-01-20 13:25:44 -08:00
  • d8dccd57ea uses standard types instead of RTCPUtility type to store data. got member read accessors, got Parse function. danilchap 2016-01-20 12:08:51 -08:00
  • 72c08edced Reenables several NetEq unittests on android. ivoc 2016-01-20 07:26:24 -08:00
  • 32f81542c2 Support REMB in combination with send-side BWE. stefan 2016-01-20 07:13:58 -08:00
  • a5dec16b42 Name SimulcastEncoderApdater on InitEncode. Peter Boström 2016-01-20 15:53:55 +01:00
  • a2b4c40928 Roll chromium_revision 15d94b7..6a04368 (370289:370362) kjellander 2016-01-20 05:38:40 -08:00
  • 9090e0b147 Switch CriticalSectionWrapper->rtc::CriticalSection in modules/audio_coding. This is a part of cleaning up CriticalSectionWrapper in general. Tommi 2016-01-20 13:39:36 +01:00
  • 84df580d52 Switch to rtc::CriticalSection in IncomingVideoStream and remove one lock. tommi 2016-01-20 04:37:54 -08:00
  • e8493326f2 Remove ConditionVariableWrapper. ConditionVariableEventWin remains for now since it's still needed for the rw lock on Windows XP. Tommi 2016-01-20 13:36:31 +01:00
  • 63cb434691 Switch use of CriticalSectionWrapper -> rtc::CriticalSection in call/ This is a first cl of removing use of CriticalSectionWrapper after a series of cleanup CLs that have been landing recently (and still are landing). tommi 2016-01-20 02:32:54 -08:00
  • 1d61a518ac Send key frame if time difference between incoming frames exceeds a certain limit. asapersson 2016-01-20 01:13:46 -08:00
  • 436ff3129e Update exclude files for renamed test kjellander 2016-01-20 00:19:19 -08:00
  • a927dcf9cd Roll chromium_revision 542b77a..15d94b7 (370158:370289) kjellander 2016-01-19 20:09:37 -08:00
  • f0b8a3784f Allow disabling denoiser when it is enabled. jackychen 2016-01-19 18:18:52 -08:00
  • 3a6bf2d68b Enable full screen windows to be shown in window picker for mac. Before this patch a full screen window can be shared if sharing is started before the window is entered into full screen mode, but not if it's already in full screen. niklas.enbom 2016-01-19 17:34:14 -08:00
  • 95c8b401fd Roll chromium_revision f527e86..542b77a (370073:370158) kjellander 2016-01-19 14:15:01 -08:00
  • f01ea4f847 Remove use of ConditionVariableWrapper and CriticalSectionWrapper from UdpSocket2Windows. This helps with untangling CriticalSectionWrapper from ConditionVariableWrapper and looks like we can just delete ConditionVariableWrapper and use rtc::Event instead. Tommi 2016-01-19 22:49:49 +01:00
  • cd255cc07b Remove unused ConditionVariableWrapper on POSIX platforms tommi 2016-01-19 13:13:14 -08:00
  • 7b971e728b Remove extra_options from VideoCodec. Peter Boström 2016-01-19 16:26:16 +01:00
  • ee5a309f12 Make CriticalSectionWrapper non-virtual. There's no need for this class to have a vtable since there exists only a single implementation (per platform). It's also not good for performance. Tommi 2016-01-19 15:42:45 +01:00
  • dd45eb6801 Remove use-after-free when quality tests stall. Peter Boström 2016-01-19 15:22:32 +01:00
  • 8a2c31d208 Make it possible to run peerconnection_unittests on Android. - renamed libjingle_peerconnection_unittest to peerconnection_unittest to circumvent cr issue http://crbug.com/543820 perkj 2016-01-19 06:20:00 -08:00
  • 0edb05b344 Declare that rent_a_codec depends on the audio codecs kwiberg 2016-01-19 05:54:28 -08:00
  • 73674f8064 Replace hardcoded constant in video capture with macro. kjellander 2016-01-19 05:49:18 -08:00
  • 3c85cad1d4 Roll chromium_revision 7a4fb8d..f527e86 (370025:370073) kjellander 2016-01-19 04:47:17 -08:00
  • 61046eb38d Rename RWLockGeneric to RWLockWinXP to more accurately reflect when it's used. tommi 2016-01-19 02:59:56 -08:00
  • 3860c7f873 Fix parsing of CLANG_REVISON from tools/clang/scripts/update.py kjellander 2016-01-19 02:04:41 -08:00
  • c4c8485662 Deleted renderer-related SetSize methods, and all uses. nisse 2016-01-19 00:52:47 -08:00
  • 81354f5ef6 Added mute logic to VideoTrackRenderers. nisse 2016-01-19 00:23:24 -08:00
  • 8d6fab8fac Remove two dead 'using' instances. Peter Boström 2016-01-19 00:24:27 +01:00
  • 2067826a5e Remove dependency on ConditionVariableWrapper and CriticalSectionWrapper in UdpSocketPosix. This is a part of cleaning up 'friend' parts of ConditionVariableWrapper's implementation where it accesses private variables of CriticalSectionWrapper, which is not good since it makes assumptions about the implementation on all posix platforms. Instead I'm using rtc::Event, another condition variable based implementation we have, and fits the requirements of UdpSocketPosix. Tommi 2016-01-18 20:35:39 +01:00
  • 233bfd2da4 Move keyframe requests outside encoder mutex. Peter Boström 2016-01-18 20:23:40 +01:00
  • 49c740264c Roll chromium_revision ad2f344..7a4fb8d (370010:370025) kjellander 2016-01-18 11:05:28 -08:00
  • aff4b70db0 Simplify the implementation of LoggingTest. This removes dependency on ConditionVariableWrapper and CriticalSectionWrapper which currently have a 'friend' relationship that I'd like to get rid of. tommi 2016-01-18 10:20:17 -08:00
  • f8c2baca4e Add a gyp/gn variable for whether to use iLBC or not kwiberg 2016-01-18 06:38:32 -08:00
  • f5a3a93d26 Add 5-argument wrapper WebRtcVideoFrame::InitToBlack For some reason, inheriting it doesn't work. Niels Möller 2016-01-18 15:24:48 +01:00
  • d1420673e3 Roll chromium_revision 1c9621e..ad2f344 (369979:370010) kjellander 2016-01-18 06:03:25 -08:00
  • 34ed2b95a5 [rtp_rtcp] rtcp::SenderReport moved into own file and got Parse function danilchap 2016-01-18 02:43:32 -08:00
  • 8b1e431231 Delete remnants of non-square pixel support from cricket::VideoFrame. nisse 2016-01-18 01:46:27 -08:00
  • 33c1dca48b Roll chromium_revision 89ca041..1c9621e (369966:369979) kjellander 2016-01-17 20:01:41 -08:00
  • 9d2a3c51d9 Roll chromium_revision 4b805fe..89ca041 (369965:369966) kjellander 2016-01-17 12:00:36 -08:00
  • e110e5c09b Roll chromium_revision 6058a7b..4b805fe (369961:369965) kjellander 2016-01-17 04:03:18 -08:00
  • d7db862622 Roll chromium_revision 9e8fb7a..6058a7b (369957:369961) kjellander 2016-01-16 20:33:37 -08:00
  • c1cf0d3ba7 Roll chromium_revision 0a79aa1..9e8fb7a (369950:369957) kjellander 2016-01-16 13:21:32 -08:00
  • 011df0acd2 Roll chromium_revision 553c2cb..0a79aa1 (369932:369950) kjellander 2016-01-16 04:02:32 -08:00
  • f624a22c86 Roll chromium_revision 46fd746..553c2cb (369797:369932) kjellander 2016-01-15 20:04:04 -08:00
  • cec0a08275 Add a new interface for creating a udp socket in which it binds the socket to a network if the network handle is set. Plus, in stunport, turnport and allocation sequence, create a socket using the new interface. honghaiz 2016-01-15 14:49:09 -08:00
  • 56271ed889 fix bug 5430 guoweis 2016-01-15 14:45:06 -08:00
  • f4decb5469 Add QP statistics logging to Android HW encoder. glaznev 2016-01-15 13:49:22 -08:00
  • 305ca25a67 Roll chromium_revision ff895e2..46fd746 (369726:369797) kjellander 2016-01-15 12:02:59 -08:00
  • 884f58523a Storing raw audio sink for default audio track. deadbeef 2016-01-15 09:20:04 -08:00
  • 1567d0bd98 [rtp_rtcp] rtcp::Sdes moved into own file Cleaning/Parsing will be done in the https://codereview.webrtc.org/1439553003/ Danil Chapovalov 2016-01-15 17:34:27 +01:00
  • 79a5a83e10 Adapt to boringssl's new defaults. torbjorng 2016-01-15 07:16:51 -08:00
  • 2c13297bf5 [rtp_rtcp] rtcp::Rpsi moved into own file Cleaning/Parsing will be done in the https://codereview.webrtc.org/1550293003/ Danil Chapovalov 2016-01-15 15:21:21 +01:00
  • 256e5b23f8 Cleaning/Parsing will be done in the https://codereview.webrtc.org/1557593002/ Danil Chapovalov 2016-01-15 14:16:24 +01:00
  • a132197f10 Roll chromium_revision 6e188de..ff895e2 (369712:369726) kjellander 2016-01-15 04:38:23 -08:00
  • 5679da1291 [rtp_rtcp] rtcp::Fir moved into own file Cleaning/Parsing will be done in the https://codereview.webrtc.org/1544403002 Danil Chapovalov 2016-01-15 13:19:53 +01:00
  • a5eba6c98b [rtp_rtcp] rtcp::Remb moved into own file Cleaning/Parsing will be done in the https://codereview.webrtc.org/1552773002/ Danil Chapovalov 2016-01-15 12:40:15 +01:00
  • d66b44d565 Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87. ivoc 2016-01-15 03:06:36 -08:00
  • 74e8df81ae Roll chromium_revision 9946592..6e188de (369667:369712) kjellander 2016-01-15 02:34:51 -08:00
  • 0f7d2939e0 Revert changes to default option setting in https://codereview.webrtc.org/1500633002/ solenberg 2016-01-15 01:40:39 -08:00
  • 5602f65659 setup_links.py fix so that FFmpeg compiles on windows. hbos 2016-01-15 01:38:34 -08:00
  • 6a59ad3ab5 Revert of Remove libfuzzer trybot from default trybot set. (patchset #1 id:1 of https://codereview.webrtc.org/1585963002/ ) kjellander 2016-01-15 00:28:42 -08:00
  • 301830f79b Roll chromium_revision 099be58..9946592 (369139:369667) kjellander 2016-01-14 20:18:23 -08:00
  • dc305db059 Add ApplyPacketOptions() Sergey Ulanov 2016-01-14 17:14:54 -08:00
  • 20ac434010 Fix a test bot failure. That was caused by https://codereview.webrtc.org/1581903002/ Honghai Zhang 2016-01-14 15:56:26 -08:00
  • e1f9d837ae Adding AddTrack/RemoveTrack to native PeerConnection API. deadbeef 2016-01-14 15:35:42 -08:00
  • d9e62f5837 Fixed sending Rtp packets with non zero csrcs and certain extensions. Added test that fails because of given issue. danilchap 2016-01-14 14:55:19 -08:00
  • 67b1e1ab0b Put options as the argument of the java PeerConnectionFactory constructor. honghaiz 2016-01-14 14:45:38 -08:00
  • 5d332ac8ff Fix expectation bug in the RTPSender unit test. terelius 2016-01-14 14:37:39 -08:00
  • 04cb763955 Add tests for verifying transport feedback for audio and video. Stefan Holmer 2016-01-14 20:34:30 +01:00
  • fcfc804e43 Eliminate defines in talk/ kjellander 2016-01-14 11:01:09 -08:00
  • 3542013f58 Revert of Update with new default boringssl no-aes cipher suites. Re-enable tests. (patchset #3 id:40001 of https://codereview.webrtc.org/1550773002/ ) sprang 2016-01-14 09:14:02 -08:00
  • 2734d77c95 Remove assert which was incorrectly added to TcpPort::OnSentPacket. Stefan Holmer 2016-01-14 17:03:53 +01:00
  • 55674ffb32 Reland Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket. Stefan Holmer 2016-01-14 15:49:16 +01:00