Commit Graph

  • 8118f1861f Set pacing bitrates in SetEncoder. pbos@webrtc.org 2014-02-12 14:50:29 +00:00
  • 67e70442b5 Remove unused and not working voe_extended_test. solenberg@webrtc.org 2014-02-12 09:58:49 +00:00
  • 5591046ab1 .gitignore: + /third_party/{clang_format,usrcsctp} pbos@webrtc.org 2014-02-12 09:33:22 +00:00
  • 14d80793a8 PeerConnectionClient needs to initialize SSL. BUG=2911 R=fischman@webrtc.org jiayl@webrtc.org 2014-02-12 00:41:59 +00:00
  • b659e2844d Reduce mixing threshold in test to avoid flakiness. andrew@webrtc.org 2014-02-11 21:52:50 +00:00
  • 75dd2885c5 Add an interface for accepting keypress signals to AudioProcessing. andrew@webrtc.org 2014-02-11 20:52:30 +00:00
  • aa1278de46 Rename merged webrtc lib to libwebrtc_merged.a. andrew@webrtc.org 2014-02-11 18:22:29 +00:00
  • 8685af7ea0 Remove "Too long processing time of Incoming frame" logspam. fischman@webrtc.org 2014-02-11 17:48:11 +00:00
  • a80be4b23c Add boundary checking to supress gcc 4.8.3 warning. turaj@webrtc.org 2014-02-11 16:38:45 +00:00
  • fc320466d1 Remove ViE external encryption API. solenberg@webrtc.org 2014-02-11 15:27:49 +00:00
  • 82ebb463fd Use libvpx's obj_int_extract and unpack_lib_posix to generate offset header file. michaelbai@google.com 2014-02-11 04:48:27 +00:00
  • dd82fa726c Revert 5516 "Thread annotation of talk_base::CriticalSection." wjia@webrtc.org 2014-02-10 23:20:15 +00:00
  • 16c08f03da Restore mixing integration tests. andrew@webrtc.org 2014-02-10 23:04:39 +00:00
  • c68d046bb1 Fix BUILD.gn to load all Chromium GN configurations. kjellander@webrtc.org 2014-02-10 21:28:55 +00:00
  • a65abf9d3a Revert "Use libvpx's obj_int_extract and unpack_lib_posix to generate offset header file." michaelbai@google.com 2014-02-10 19:26:26 +00:00
  • 1f64f06784 Add stats of incoming frame delays for debugging bandwidth estimation. BUG=crbug/338380 R=stefan@webrtc.org jiayl@webrtc.org 2014-02-10 19:12:14 +00:00
  • 82387e4608 Add ability to receive calls for iOS BUG=2701 R=fischman@webrtc.org fischman@webrtc.org 2014-02-10 18:47:11 +00:00
  • 7686f0ddda Use libvpx's obj_int_extract and unpack_lib_posix to generate offset header file. michaelbai@google.com 2014-02-10 17:42:34 +00:00
  • 0a7085ffc2 Thread annotation of talk_base::CriticalSection. pbos@webrtc.org 2014-02-10 13:58:37 +00:00
  • 9cba2e4802 Exclude libjingle_p2p_unittest tests failing TSan Race verifier. kjellander@webrtc.org 2014-02-10 12:43:40 +00:00
  • 6f0ca57fb2 Add experiment: SkipEncodingUnusedStreams sprang@webrtc.org 2014-02-10 09:20:51 +00:00
  • 4723dc88b3 Revert 5511 "Revert 5510 "Disable failing libjingle_p2p_unittest..." kjellander@webrtc.org 2014-02-09 18:48:56 +00:00
  • 607c805b87 Roll chromium_revision 245382:249215 kjellander@webrtc.org 2014-02-09 18:38:31 +00:00
  • ce2b44532e Revert 5510 "Disable failing libjingle_p2p_unittest test on Linux" kjellander@webrtc.org 2014-02-09 18:21:00 +00:00
  • 8d2ddd00f1 Disable failing libjingle_p2p_unittest test on Linux kjellander@webrtc.org 2014-02-08 21:35:20 +00:00
  • 6f8acdc76f Suppressions for libjingle_unittest after roll in r5502 kjellander@webrtc.org 2014-02-08 21:15:59 +00:00
  • cc685acbdf Disable AsyncInvokeTest.CancelInvoker test sergeyu@chromium.org 2014-02-08 19:59:00 +00:00
  • 0178810659 Don't use LOG() in callback.h sergeyu@chromium.org 2014-02-08 03:18:03 +00:00
  • bfc26dcc10 gitignore: ignore webrtc android demo apps build artifacts, and sort list fischman@webrtc.org 2014-02-08 01:05:45 +00:00
  • 5a59ccbb6d Switching to NSS random number generator and adding init method to unittests. mallinath@webrtc.org 2014-02-07 23:22:00 +00:00
  • ad3035fc9e Fix WindowCapturerWin to unselect bitmap before destroying DC. sergeyu@chromium.org 2014-02-07 21:24:04 +00:00
  • c5a839c3a9 Updated demos so they work on FF, the createOffer api cannot have null parameters according to spec. Same applies to createAnswer. vikasmarwaha@webrtc.org 2014-02-07 19:08:38 +00:00
  • 9cf037b831 Update libjingle to 61168196 sergeyu@chromium.org 2014-02-07 19:03:26 +00:00
  • 9510e53cc0 Make VideoReceiveStream::GetStats() const. sprang@webrtc.org 2014-02-07 15:32:45 +00:00
  • f7d4993940 Remove Coverity script. kjellander@webrtc.org 2014-02-07 13:50:40 +00:00
  • 09315705b9 Wire up statistics in video receive stream of new API sprang@webrtc.org 2014-02-07 12:06:29 +00:00
  • 90ce73a0d5 Remove svn:ignore for third_party/cygwin kjellander@webrtc.org 2014-02-07 10:04:21 +00:00
  • b307e86076 Updated demos to use the sucess and failure callback in addIceCandidate api. vikasmarwaha@webrtc.org 2014-02-06 22:38:37 +00:00
  • 60de116687 libyuv.gyp fix for ios sim which is intel not neon, fixing a link error. BUG=none TEST=try bots R=andrew@webrtc.org fbarchard@google.com 2014-02-06 21:17:16 +00:00
  • dfef7ba971 Roll libvpx 241571:248011 marpan@webrtc.org 2014-02-06 18:40:35 +00:00
  • 77c917a6ee Plot the capacity of a trace-based delivery filter. stefan@webrtc.org 2014-02-06 16:34:47 +00:00
  • ea1c5ad58f Fix gunit compilation on VS2012. pbos@webrtc.org 2014-02-06 13:17:20 +00:00
  • f928f5c87c Use system's cpu_features library michaelbai@google.com 2014-02-05 18:43:46 +00:00
  • c88d3368d5 Add delay and send/receive throughput plots to BWE simulation. stefan@webrtc.org 2014-02-05 15:57:14 +00:00
  • 75642fcd9a Implementing replacement audio support in neteq_rtpplay henrik.lundin@webrtc.org 2014-02-05 08:49:13 +00:00
  • e6ab21b9ca Fixing a bug in DummyRTPpacket henrik.lundin@webrtc.org 2014-02-05 08:46:46 +00:00
  • 54744918ef Update AudioProcessing::Create docs. andrew@webrtc.org 2014-02-05 06:30:29 +00:00
  • 20a60ea39d Fix a cursor capturing issue on Windows. The input position to WindowFromPoint should be relative to the desktop, not relative to the window; if the result from WindowFromPoint is a child window of the shared top window, it should be captured. jiayl@webrtc.org 2014-02-04 17:49:12 +00:00
  • 0e5a2b5de6 Handle the invalid case of setting multiple stream_bitrates if there is only a single send stream registered. stefan@webrtc.org 2014-02-04 14:38:25 +00:00
  • 3e6c41c48f Revert "Disable the test: DtmfTest.ManualSuccessfullySendsIn/OutOfBandTelephoneEvents" pbos@webrtc.org 2014-02-04 10:45:14 +00:00
  • 064b32acbb Fix locking in LoopBackTransport::StorePacket. pbos@webrtc.org 2014-02-04 09:42:02 +00:00
  • 36291da197 Pull Chromium's clang-format binaries. andrew@webrtc.org 2014-02-04 01:45:10 +00:00
  • f6a638e001 Trivial rename of non-compile time consts. andrew@webrtc.org 2014-02-04 01:31:28 +00:00
  • e88c186dbe Revert r5480 marpan@webrtc.org 2014-02-04 00:02:35 +00:00
  • e35ecb476b Roll libvpx 241571:248011 marpan@webrtc.org 2014-02-03 22:53:10 +00:00
  • f6b8f496ee Disable the test: DtmfTest.ManualSuccessfullySendsIn/OutOfBandTelephoneEvents marpan@webrtc.org 2014-02-03 21:34:35 +00:00
  • 6e08228525 PeerConnectionTest(java): remove the obsolete magical names of streams & tracks. fischman@webrtc.org 2014-02-03 19:15:44 +00:00
  • a06ebab1e1 PeerConnectionTest(java): test SCTP DataChannels. fischman@webrtc.org 2014-02-03 19:11:29 +00:00
  • ecd622eec3 Updating libjingle.gyp after addition new files yuvframescapturer.cc. mallinath@webrtc.org 2014-02-03 17:17:05 +00:00
  • 67ee6b9a62 Update talk to 60923971 mallinath@webrtc.org 2014-02-03 16:57:16 +00:00
  • 422fdbf502 Wire up feedback to VideoSender. stefan@webrtc.org 2014-02-03 16:33:50 +00:00
  • c9ee412070 Re-enabling audio processing tests aluebs@webrtc.org 2014-02-03 14:41:57 +00:00
  • c1e28038ba Moved the new OnData interface to AudioTranport, and expose the AudioTransport pointer via voe_base xians@webrtc.org 2014-02-02 15:30:20 +00:00
  • 1af5ea0538 Implement single monitor capture on Mac. jiayl@webrtc.org 2014-02-01 02:03:24 +00:00
  • 83aee8f450 Fixing test name for NetEqPerformanceTest henrik.lundin@webrtc.org 2014-01-31 11:46:34 +00:00
  • bdc5ed2e7d Add configuration for cpu overuse detection to video send stream. asapersson@webrtc.org 2014-01-31 10:05:07 +00:00
  • 7d7f08957c Add gyp_webrtc script to generate projects. kjellander@webrtc.org 2014-01-31 09:34:51 +00:00
  • 1dd9b4d98e Add BWE tools for parsing RTP files. stefan@webrtc.org 2014-01-31 09:15:48 +00:00
  • 668a23b402 Fix MIME type on new demo pages. juberti@webrtc.org 2014-01-30 23:42:01 +00:00
  • 5db9a3f32a Added new create-offer and ice-servers demos to test the exact output of createOffer and .onicecandidate. Updated a few demos to work on Firefox. juberti@webrtc.org 2014-01-30 23:38:44 +00:00
  • bda5fa77af Fix the mouse cursor offset issue on Mac. The problem is that MouseCursorMonitor returns coordinates in DIPs, while DisplayAndMouseComposer assumes that they are in physical pixels. The fix is to convert the position to physical pixels in MouseCursorMonitorMac. jiayl@webrtc.org 2014-01-30 23:27:35 +00:00
  • c693704cc2 Move out typing detection to its own class. henrikg@webrtc.org 2014-01-30 09:50:46 +00:00
  • cf1b51b6fb Moves the display reconfiguration callback into a separate class, so that it can be shared with the cursor monitor when single monitor capturing is added (https://webrtc-codereview.appspot.com/4679005/). This Cl should have no functionality change. jiayl@webrtc.org 2014-01-29 21:59:12 +00:00
  • 808b99b111 Disable a test assert which fails due to usrsctp not cleaned up in SctpDataEngine.cc BUG=2749 R=fischman@webrtc.org jiayl@webrtc.org 2014-01-29 19:44:40 +00:00
  • a576faf82a Enable SCTP and use OPENSSL on Anroid and NSS on other platforms. It includes unit test fixes to properly initialize SSL if DTLS or SSL random number generator is used in the tests. The private key and certificate constant strings used in some tests are updated to be compatible with NSS. A few potentially overflow type conversions caused compiling warning on Windows and they are fixed by importing and using Chromium's checked_cast, which aborts the program if overflow occurs. It also fixes a leak in nssstreamadapter.cc by releasing the PRFileDesc* in StreamClose. jiayl@webrtc.org 2014-01-29 17:45:53 +00:00
  • 07e5196414 Added new capture callback interface to pass the capture callback to a specific voe channel from libjingle webrtcvoiceengine.cc. xians@webrtc.org 2014-01-29 13:54:02 +00:00
  • 094ac39b5a Fix race when deleting video receive streams in Call. solenberg@webrtc.org 2014-01-29 11:21:58 +00:00
  • f7c6e743b3 Fix deadlock in video_receiver.cc. stefan@webrtc.org 2014-01-29 10:27:51 +00:00
  • 41907748cb Connect webrtc::Config to WrappingBitrateEstimator henrik.lundin@webrtc.org 2014-01-29 08:47:15 +00:00
  • c7c7a531f3 Add Config struct for experimental AGC. andrew@webrtc.org 2014-01-29 04:57:25 +00:00
  • 7433a088d2 Revert 5444 "Revert 5421 "Fix deadlock on register/unregister ob..." mallinath@webrtc.org 2014-01-29 00:56:02 +00:00
  • 84eb0e952e Add clean test to NetEq perf test henrik.lundin@webrtc.org 2014-01-28 21:50:35 +00:00
  • 45a60c7fdc Add tools/gn and tools/swarming_client to svn:ignore kjellander@webrtc.org 2014-01-28 19:30:26 +00:00
  • 83dd95432e rolling Opus 1.1 minyue@webrtc.org 2014-01-28 08:46:58 +00:00
  • 0dac5378e5 Revert 5447 "Update talk to 60420316." mallinath@webrtc.org 2014-01-28 06:58:42 +00:00
  • 752a017809 Update talk to 60420316. mallinath@webrtc.org 2014-01-28 06:45:52 +00:00
  • 69ff90e832 libyuv r976 for MJPGToI420 return code. BUG=2847 TESTED=libyuv MJPGToI420 unittest added which passes invalid MJPG and expects a failure. R=andrew@webrtc.org, braveyao@webrtc.org fbarchard@google.com 2014-01-28 03:58:46 +00:00
  • 932b0193e7 VideoCaptureAndroid: stop preview in opposite order of starting. fischman@webrtc.org 2014-01-27 22:32:05 +00:00
  • 18586d38bc Revert 5421 "Fix deadlock on register/unregister observer while ..." mallinath@webrtc.org 2014-01-27 22:00:57 +00:00
  • ecc96af15b Expose errors in apprtc demo to div. Currently the errors only show in the console, the CL attempts to expose critical errors on to the div element. BUG=2786 R=fischman@webrtc.org vikasmarwaha@webrtc.org 2014-01-27 21:13:54 +00:00
  • 776d8df25f Fix hooks in DEPS to allow read-only checkout to succeed. wjia@webrtc.org 2014-01-27 19:55:16 +00:00
  • a45cac0fb7 Avoid potential dead lock in StreamStatisticianImpl sprang@webrtc.org 2014-01-27 16:22:08 +00:00
  • 2a260d9fab Enable Android APK trybots by default. kjellander@webrtc.org 2014-01-27 16:08:43 +00:00
  • 5314e85926 Race condition in RTPSender::UpdateRtpStats sprang@webrtc.org 2014-01-27 13:20:36 +00:00
  • d9b9560ee5 Drop early packets when not sending in TransportAdapter. sprang@webrtc.org 2014-01-27 13:03:02 +00:00
  • 2397a17c6b Fix bug introduced during replace of list wrapper with std equivalents in r5378. andresp@webrtc.org 2014-01-27 12:33:30 +00:00
  • c8b99a49d1 This is to roll a more recent Chromium, which contains latest Clang, so as to be able to roll Opus 1.1, which will the next step. minyue@webrtc.org 2014-01-27 12:21:42 +00:00
  • c00adbed73 Race in StreamStatisticianImpl::GetStatistics vs. ::IncomingPacket sprang@webrtc.org 2014-01-27 10:42:48 +00:00
  • 99eab02fb1 Fix "field '_testNo' is uninitialized" warnings. pbos@webrtc.org 2014-01-27 09:30:35 +00:00
  • c98882dcd3 Always initialize Trace in Call TraceDispatcher. pbos@webrtc.org 2014-01-27 09:11:10 +00:00