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  • bff9620116 Fix log build error for Chromium builds. henrikg@webrtc.org 2013-11-08 10:37:27 +00:00
  • 4c828e145e Remove update_resources.py as it's no longer used. kjellander@webrtc.org 2013-11-08 09:08:36 +00:00
  • f1a48174d4 Replace disabled logging with a restricted logging mode. andrew@webrtc.org 2013-11-07 23:47:26 +00:00
  • 5adc89747a Updated WebRTC version to 3.46 elham@webrtc.org 2013-11-06 22:27:51 +00:00
  • a7855a88b3 Fix for xgetbv on Visual Studio 2010. BUG=none TEST=local build of webrtc with 2010. python build\gyp_chromium --depth=. -G msvs_version=2010 -fninja all.gyp & ninja -C out\Debug R=niklas.enbom@webrtc.org fbarchard@google.com 2013-11-06 21:18:48 +00:00
  • bde3056567 Fix for video_processor_intergration_tests to run in parallel. BUG=2601. R=pbos@webrtc.org, stefan@webrtc.org marpan@webrtc.org 2013-11-06 20:59:29 +00:00
  • c4225b63bb Update getUserMedia W3C conformance tests. kjellander@webrtc.org 2013-11-06 13:26:34 +00:00
  • 8bad50e845 Sending status fix for module. asapersson@webrtc.org 2013-11-06 10:45:58 +00:00
  • 16d6254e8c Update talk to 56183333. wu@webrtc.org 2013-11-05 23:45:14 +00:00
  • d16d307218 Fix bad Google Storage uploads of resource files. kjellander@webrtc.org 2013-11-05 21:03:04 +00:00
  • 0e03360591 Add OWNERS for resources/ kjellander@webrtc.org 2013-11-05 21:02:49 +00:00
  • 7a36cb408b Add missing dependencies to .isolate files kjellander@webrtc.org 2013-11-05 14:28:57 +00:00
  • 1e8b671d05 Roll chromium_revision 231713:232627 kjellander@webrtc.org 2013-11-05 12:32:05 +00:00
  • da7f6589aa Add svn:ignore to avoid re-download of resources kjellander@webrtc.org 2013-11-05 09:27:51 +00:00
  • b8cb85b348 Fix broken build on x86 Android BUG=2545 R=fischman@webrtc.org, henrike@webrtc.org fischman@webrtc.org 2013-11-04 19:06:08 +00:00
  • 7b273a545d PeerConnection iOS: update README instructions This is needed to account for https://codereview.chromium.org/25535004/ fischman@webrtc.org 2013-11-04 18:48:12 +00:00
  • 07a6fbe83d Update talk to 56092586. wu@webrtc.org 2013-11-04 18:41:34 +00:00
  • 3779c1cb0a Fix invalid .sha1 files for audio_coding kjellander@webrtc.org 2013-11-04 14:54:47 +00:00
  • 80174583bd Replace old resources download script with depot_tools kjellander@webrtc.org 2013-11-04 12:07:57 +00:00
  • a452fc29e6 Remove resources/ svn:ignore to prepare for updated resource handling kjellander@webrtc.org 2013-11-04 12:07:09 +00:00
  • 58bcdeee2c Roll chromium_revision 229708:231713 kjellander@webrtc.org 2013-11-04 09:40:03 +00:00
  • 766154aa1d Removed unused code. asapersson@webrtc.org 2013-11-04 08:35:50 +00:00
  • e2df8b7f01 Make video quality analysis unittests print to log instead of stdout. kjellander@webrtc.org 2013-11-03 18:34:51 +00:00
  • 5dd2ecb32d Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420" sheu@chromium.org 2013-10-31 23:41:04 +00:00
  • 74e6e8458e Remove extra copy in VideoCaptureImpl::IncomingFrameI420 sheu@chromium.org 2013-10-31 21:48:16 +00:00
  • d705649edf Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420" sheu@chromium.org 2013-10-31 21:20:15 +00:00
  • 1a4ed0d70c Remove extra copy in VideoCaptureImpl::IncomingFrameI420 sheu@chromium.org 2013-10-31 20:32:28 +00:00
  • de305014c6 Update talk to 55906045. wu@webrtc.org 2013-10-31 15:40:38 +00:00
  • 58cd31665c Address Clag Analyzer issues. turaj@webrtc.org 2013-10-31 15:15:55 +00:00
  • 7d6bd22019 Propagate estimated RTT from receivers to rtt observer. asapersson@webrtc.org 2013-10-31 12:14:34 +00:00
  • da2c37b759 Video bandwidth not reported correctly sprang@webrtc.org 2013-10-31 09:49:03 +00:00
  • 773e72797f Provide a MouseCursorMonitor::CreateForWindow implementation in *_null.cc sergeyu@chromium.org 2013-10-31 01:51:21 +00:00
  • de748c806c Remove unused make_scoped_ptr which causes an "ambiguous" error with chromium build. wu@webrtc.org 2013-10-30 20:43:27 +00:00
  • dce70ccb0b Add delay limit to ChokeFilter. solenberg@webrtc.org 2013-10-30 19:18:07 +00:00
  • f424cb8e13 Update talk to 55863981. wu@webrtc.org 2013-10-30 17:57:33 +00:00
  • d6e46638ec Logging for BWE test framework. solenberg@webrtc.org 2013-10-30 16:06:26 +00:00
  • cecfd1832d Update talk to 55821645. wu@webrtc.org 2013-10-30 05:18:12 +00:00
  • ec4cccc6b6 Update libyuv to 832. wu@webrtc.org 2013-10-29 21:02:20 +00:00
  • 47ebbaddbb Make video/ only depend on video_engine_core. pbos@webrtc.org 2013-10-29 13:11:56 +00:00
  • def22b455b Stop DirectTransports in VideoSendStreamTests. pbos@webrtc.org 2013-10-29 10:12:10 +00:00
  • 55e1723713 Avoid a leak in AudioCodingModuleTest.TestIsac. The leak was caught by LSAN. turaj@webrtc.org 2013-10-29 04:40:09 +00:00
  • 9ca93a8b8e Explicitly @synthesize ObjC @properties fischman@webrtc.org 2013-10-29 00:14:15 +00:00
  • 0aeb22e32c Adding tl0idx consideration for continuity mikhal@webrtc.org 2013-10-28 22:26:14 +00:00
  • 0803c03f9a Fix build/isolate.gypi path in webrtc_tests.gypi. pbos@webrtc.org 2013-10-28 18:10:29 +00:00
  • b7a171825b Drop ViEDecoderObserver::DecoderTiming impl now that WebRtcDecoderObserver rolled in r5038. fischman@webrtc.org 2013-10-28 17:36:59 +00:00
  • 16e03b7bd8 Separate Call API/build files from video_engine/. pbos@webrtc.org 2013-10-28 16:32:01 +00:00
  • 850bcbe855 Remove frame_callback.h include in webrtcvie.h. pbos@webrtc.org 2013-10-28 15:41:17 +00:00
  • 1a3a6e5340 Removing the threshold from the auto-mute APIs henrik.lundin@webrtc.org 2013-10-28 10:16:14 +00:00
  • fe5d36b6fe Move RtcpStatistics to webrtc/common_types.h, to be used by vie as well. sprang@webrtc.org 2013-10-28 09:21:07 +00:00
  • 97077a3ab2 Update libjingle to 55618622. Update libyuv to r826. wu@webrtc.org 2013-10-25 21:18:33 +00:00
  • 728bc0fa4c Add qiang.lu@intel.com to WATCHLISTS. fischman@webrtc.org 2013-10-25 19:20:02 +00:00
  • c94abd313e Use clang-format -style=chromium to correct the format in webrtc/modules/interface/module_common_types.h xians@webrtc.org 2013-10-25 18:15:09 +00:00
  • e4e5683b41 Clean up tsan suppression file: 1) remove suppressions that are already fixed. 2) merge duplicated suppressions. wu@webrtc.org 2013-10-25 16:29:33 +00:00
  • 0729460acb Added a "interleaved_" flag to webrtc::AudioFrame. xians@webrtc.org 2013-10-25 12:50:46 +00:00
  • 442c5e47cd Update adapter.js to use TURN transport parameters for FF version 27 & above. vikasmarwaha@webrtc.org 2013-10-24 20:31:57 +00:00
  • d674a566d3 Update dc1 demo as it was using invalid data Constraint (Reliable:true) for SCTP. The constraint Reliable is not supported by Standard and ignored in our implementation. See issue 2511. vikasmarwaha@webrtc.org 2013-10-24 19:38:47 +00:00
  • b3731da68f Prefix MOVE_ONLY_TYPE_FOR_CPP_03 with WEBRTC_. andrew@webrtc.org 2013-10-24 15:16:53 +00:00
  • b56d0e383e Change the low-bitrate handling in BitrateControllerImpl henrik.lundin@webrtc.org 2013-10-24 09:24:06 +00:00
  • 37bb4974e7 Expose VideoCodingModule's decoder stats up the stack from VCMTiming to chrome://webrtc-internals. fischman@webrtc.org 2013-10-23 23:59:45 +00:00
  • d371a29227 Fix tsan failures for libjingle_unittest. 1) Change AsyncSocket's SignalReadEvent and SignalWriteEvent's thread mode to multi_threaded_local as they can be accessed from different threads. 2) Protect NATServer::TransEntry::whitelist. 3) Protect PhysicalSocket:error_. wu@webrtc.org 2013-10-23 23:56:09 +00:00
  • d1bcf1180a Check if WARN_UNUSED_RESULT and COMPILE_ASSERT are defined. andrew@webrtc.org 2013-10-23 19:11:32 +00:00
  • 22858d4785 Add an extended filter option to audioproc. andrew@webrtc.org 2013-10-23 14:07:17 +00:00
  • 042e91c2b2 Fix for incorrect RTT estimation. A too low RTT value could be estimated. asapersson@webrtc.org 2013-10-23 13:58:31 +00:00
  • ba975e2078 Porting auto mute to new ViE API henrik.lundin@webrtc.org 2013-10-23 11:04:57 +00:00
  • 886aef09a8 Fixing broken tests in voe_auto_test extended tina.legrand@webrtc.org 2013-10-23 10:39:56 +00:00
  • 8804a29951 Add CriticalSection to fakeaudiocapturemodule to protect the variables which will be accessed from process_thread_ and the main thread. wu@webrtc.org 2013-10-22 23:09:20 +00:00
  • 4d7116be7a Fix tsan failures on filevideocapturer.cc. 1) init start_time_ns_ before the file_read_thread_ is started to avoid data racing as the start_time_ns_ will also be read by the file_read_thread_. 2) add CriticalSection to protect |finished_| that is accessed by FileReadThread and the main thread wu@webrtc.org 2013-10-22 18:41:17 +00:00
  • 90d8719fd7 Radix should be specified when calling ParseInt function in adapter.js. Refer to issue 2490. vikasmarwaha@webrtc.org 2013-10-22 18:02:41 +00:00
  • 8575980e16 Add default trybots for WebRTC try server. kjellander@webrtc.org 2013-10-22 16:47:40 +00:00
  • 31628aae7e Upgrade scoped_ptr to Chromium's latest version. andrew@webrtc.org 2013-10-22 12:50:00 +00:00
  • 06b60c07b7 Roll chromium_revision 228675:229708 kjellander@webrtc.org 2013-10-22 12:09:48 +00:00
  • 621df678c8 WEBRTC_{BIG, LITTLE}_ENDIAN -> WEBRTC_ARCH_{BIG, LITTLE}_ENDIAN. andrew@webrtc.org 2013-10-22 10:27:23 +00:00
  • 943e3b95a6 Add CurrentLayerId() to temporal layers. marpan@webrtc.org 2013-10-22 01:55:07 +00:00
  • 50bc553852 Reenable DTLS renegotiation unittest in libjingle. mallinath@webrtc.org 2013-10-21 17:58:35 +00:00
  • 9c735c4e25 Updated WebRTC version to 3.45 elham@webrtc.org 2013-10-21 16:34:50 +00:00
  • 8215106371 Framework for testing bandwidth estimation. solenberg@webrtc.org 2013-10-21 14:23:26 +00:00
  • 29dd0de5b3 Changing the bitrate clamping in BitrateControllerImpl henrik.lundin@webrtc.org 2013-10-21 14:00:01 +00:00
  • 0d19ed9a06 AutoMute: Adding channel_id parameter to callback. henrik.lundin@webrtc.org 2013-10-21 12:37:13 +00:00
  • fe1ef935e7 Implement I420FrameCallbacks in Call. pbos@webrtc.org 2013-10-21 10:34:43 +00:00
  • e05362916c Make sure the first frame isn't dropped. pbos@webrtc.org 2013-10-21 09:02:30 +00:00
  • eb61a851d5 Move audio_e2e_harness into include_tests==1 condition. kjellander@webrtc.org 2013-10-21 08:40:56 +00:00
  • 88a310886e Add audio_e2e_test target to tools.gyp kjellander@webrtc.org 2013-10-19 18:10:04 +00:00
  • fb648da2b9 Protect _transportPtr, which can be accessed by different threads in the case of external transport. This change avoid the potential use-after-free, e.g. the case in the reported bug. wu@webrtc.org 2013-10-18 21:10:51 +00:00
  • 3c5d2b43ec Thread::Stop() must be called before any subclass's destructor completes. Update Thread documentation, fix all subclasses that had a problem. wu@webrtc.org 2013-10-18 16:27:26 +00:00
  • 3e00505e9a Have padding decay to zero if no frames are being captured. stefan@webrtc.org 2013-10-18 15:05:29 +00:00
  • 893c07f209 Disable the -Wno-unused-const-variable Clang warning on Mac kjellander@webrtc.org 2013-10-18 14:42:27 +00:00
  • 89b1e688ca Minor comment fix after clang reformat. andrew@webrtc.org 2013-10-18 14:23:29 +00:00
  • 1c82037494 AppRTCDemo(android): remove vestigial mentions of PowerManager fischman@webrtc.org 2013-10-17 20:53:12 +00:00
  • 2df89c0c8b MouseCursorMonitor implementation for OSX and Windows. sergeyu@chromium.org 2013-10-17 19:47:18 +00:00
  • 6b426ba5c5 Final round of LSan suppressions (take 2) kjellander@webrtc.org 2013-10-17 19:26:27 +00:00
  • 6342066974 Fix tsan failures in channel.cc regarding to the volume settings. wu@webrtc.org 2013-10-17 18:28:55 +00:00
  • b22049b295 Final round of LSan suppressions. kjellander@webrtc.org 2013-10-17 18:13:28 +00:00
  • 8a7b89fd41 More libjingle LSan suppressions. kjellander@webrtc.org 2013-10-17 17:15:24 +00:00
  • 675e260ad1 Check the number of playout channels instead of the send channels in StopPlayout() xians@webrtc.org 2013-10-17 16:15:34 +00:00
  • e61da8c391 Suppressions and PRESUBMIT.py for LSan kjellander@webrtc.org 2013-10-17 14:16:17 +00:00
  • c11148b352 Compound/reduced-size RTCP in VideoReceiveStream. pbos@webrtc.org 2013-10-17 14:14:42 +00:00
  • 603ed98bca Suppress race condition warn in CallTest_ReceivesAndRetransmitsNack_Test sprang@webrtc.org 2013-10-17 07:49:26 +00:00
  • 54e729bb73 Remove tsan suppression for the failure that's already fixed. wu@webrtc.org 2013-10-16 23:19:54 +00:00
  • 853dd07890 Add issue links to the tsanv2 suppressions. wu@webrtc.org 2013-10-16 22:50:59 +00:00
  • e7771e26f2 Add /webrtc/modules/audio_device/android/test/{bin,gen,libs} to .gitignore fischman@webrtc.org 2013-10-16 21:08:17 +00:00