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  • fe1eafb71a Call SSL_shutdown in OpenSSLStreamAdapter::Cleanup. jiayl@webrtc.org 2014-09-24 21:13:39 +00:00
  • 30be827e6a Enable render downmixing to mono in AudioProcessing. andrew@webrtc.org 2014-09-24 20:06:23 +00:00
  • e1bba60792 Add missing DesktopConfigurationMonitor Unlock in webrtc::ScreenCapturerMac jiayl@webrtc.org 2014-09-24 17:23:46 +00:00
  • 3987b6de50 Fix a problem in Thread::Send. Previously if thread A->Send is called on thread B, B->ReceiveSends will be called, which enables an arbitrary thread to invoke calls on B while B is wait for A->Send to return. This caused mutliple problems like issue 3559, 3579. The fix is to limit B->ReceiveSends to only process requests from A. Also disallow the worker thread invoking other threads. jiayl@webrtc.org 2014-09-24 17:14:05 +00:00
  • a0ce9fa2a6 Call NS AnalyzeCaptureAudio before AEC aluebs@webrtc.org 2014-09-24 14:18:03 +00:00
  • 70e2d11ea8 Reduce jitter delay for low fps streams. Enabled by finch flag. sprang@webrtc.org 2014-09-24 14:06:56 +00:00
  • 275dac2c1d Moved the filter calculation from analyze to process in ns_core aluebs@webrtc.org 2014-09-24 13:23:49 +00:00
  • 634c926928 audioproc: Now also writes to output file in simulation mode bjornv@webrtc.org 2014-09-24 12:21:51 +00:00
  • 7ee24a7906 WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type encoded stream as uint8_t kwiberg@webrtc.org 2014-09-24 10:31:02 +00:00
  • d60d79a145 Thread annotation of rtc::CriticalSection. pbos@webrtc.org 2014-09-24 07:10:57 +00:00
  • 38344ed280 Move thread_annotations.h to webrtc/base/. pbos@webrtc.org 2014-09-24 06:05:00 +00:00
  • 8166faeff3 Change Android video renderer to maintain video aspect ratio when displaying camera or decoded video frames. glaznev@webrtc.org 2014-09-23 23:58:52 +00:00
  • 90668b1633 Switch HW video decoder to output byte buffers if video renderer EGL context is not provided by app. glaznev@webrtc.org 2014-09-23 21:42:15 +00:00
  • 1b7dcc1647 (Auto)update libjingle 76169599-> 76176062 buildbot@webrtc.org 2014-09-23 17:41:48 +00:00
  • 94ff92ceec Use VPX_IMG_FMT_*/VPX_PLANE_* defines johannkoenig@google.com 2014-09-23 17:31:47 +00:00
  • 2c1bcea1bc Enable ipv6 by default for webrtc under a Finch experiment. guoweis@webrtc.org 2014-09-23 16:23:02 +00:00
  • 3987f10c11 Revert "Remove DTMF status methods from Voice Engine" r7276 henrik.lundin@webrtc.org 2014-09-23 13:15:14 +00:00
  • bf7b9e0081 Remove DTMF status methods from Voice Engine henrik.lundin@webrtc.org 2014-09-23 12:54:04 +00:00
  • e34a2e7475 Revert "Set minimum SDK level to 10.7 for Mac and iOS" (r7175) kjellander@webrtc.org 2014-09-23 12:43:14 +00:00
  • faf2410a32 gn: Hide modules/video_capture:video_capture_internal_impl behind an arg pbos@webrtc.org 2014-09-23 12:37:06 +00:00
  • 0e6e4d2ff2 Reland "Converting five tests to use new AudioCoding interface" (r7258) henrik.lundin@webrtc.org 2014-09-23 12:05:34 +00:00
  • 4f6f22f0c6 Reland (rev 7259) "Convert AcmReceiverTest to new AudioCoding interface" andresp@webrtc.org 2014-09-23 11:37:57 +00:00
  • ea29787df0 audio_processing/agc: Solved building with AGC_DEBUG + few style changes bjornv@webrtc.org 2014-09-23 11:21:39 +00:00
  • 0a2087a711 Skeleton for registering external encoders/decoders. pbos@webrtc.org 2014-09-23 09:40:22 +00:00
  • c569a49a3d Unit tests for SSLAdapter tkchin@webrtc.org 2014-09-23 05:56:44 +00:00
  • dc0b37dcb1 modules_unittests: Turned on ApmTest.Process test for Android bjornv@webrtc.org 2014-09-23 05:03:44 +00:00
  • a3c4d4dd2c Revert 7266 "WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type en..." andrew@webrtc.org 2014-09-23 01:32:57 +00:00
  • 8c5740b485 WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type encoded stream as uint8_t kwiberg@webrtc.org 2014-09-22 23:04:14 +00:00
  • 83f95ba9a6 Remove engine-level SetOptions. pbos@webrtc.org 2014-09-22 16:07:18 +00:00
  • 99e404c84a Revert "Converting five tests to use new AudioCoding interface" (rev 7258). andresp@webrtc.org 2014-09-22 15:49:56 +00:00
  • 35850ff71f Adding test file path as argument of the rtcBot run command's arguments. houssainy@google.com 2014-09-22 15:24:56 +00:00
  • 64a2f10f4b Remove Get/SetNetEQPlayoutMode APIs henrik.lundin@webrtc.org 2014-09-22 14:30:10 +00:00
  • 07ca949346 Adding webrtc_video_streaming test This test is streaming video and audio between two bots using webrtc js api. houssainy@google.com 2014-09-22 13:52:39 +00:00
  • c570761288 Revert "Convert AcmReceiverTest to new AudioCoding interface" (rev 7258). andresp@webrtc.org 2014-09-22 13:18:34 +00:00
  • cfe073539c Convert AcmReceiverTest to new AudioCoding interface henrik.lundin@webrtc.org 2014-09-22 12:10:44 +00:00
  • eb1de5cb72 Converting five tests to use new AudioCoding interface henrik.lundin@webrtc.org 2014-09-22 12:07:12 +00:00
  • bdfdc96b22 Clang-format ns_core aluebs@webrtc.org 2014-09-22 10:59:46 +00:00
  • 759982d357 Set number of temporal layers for VideoSendStream. pbos@webrtc.org 2014-09-22 09:32:46 +00:00
  • 612171527e Ensure that NetEq recovers after a large timestamp jump henrik.lundin@webrtc.org 2014-09-22 08:30:07 +00:00
  • 88772874da Disabled several rtc_unittests so the tests can be turned on in the waterfall henrike@webrtc.org 2014-09-22 07:30:48 +00:00
  • 97ed39344a Reapply 23529005 after fixing the build break issue (Chromium:582133002) guoweis@webrtc.org 2014-09-19 21:06:12 +00:00
  • ed5ca1f122 (Auto)update libjingle 75925673-> 75926712 buildbot@webrtc.org 2014-09-19 20:30:44 +00:00
  • c98f217c65 (Auto)update libjingle 75924589-> 75925673 buildbot@webrtc.org 2014-09-19 20:18:10 +00:00
  • 0c9fe72b21 (Auto)update libjingle 75922684-> 75924589 buildbot@webrtc.org 2014-09-19 20:05:02 +00:00
  • ebf2757339 Fix HW video decoder crash on some Android KK devices. glaznev@webrtc.org 2014-09-19 19:36:13 +00:00
  • c1eebfa107 Fix the libjingle_media_unittest failure in Windows build by modifying libjingle_tests.gyp and sctpdataengine_unittests.cc instead of ssladapter.cc. thorcarpenter@google.com 2014-09-19 17:54:00 +00:00
  • e65812427d Fixing compilation failure in peerconnection_jni.cc with WEBRTC_CHROMIUM_BUILD. glaznev@webrtc.org 2014-09-19 16:53:46 +00:00
  • fbf3bfe172 Separate between Analyze and Process in NS aluebs@webrtc.org 2014-09-19 15:18:59 +00:00
  • 95705602bd Additional disabled tests in rtc_unittests. kjellander@webrtc.org 2014-09-19 14:49:37 +00:00
  • 34ac7762e0 Additional disabled tests in rtc_unittests. kjellander@webrtc.org 2014-09-19 13:47:47 +00:00
  • fded02c164 base: disabled several base tests on Mac so that rtc_unittests can be turned back on henrike@webrtc.org 2014-09-19 13:10:10 +00:00
  • bbe0a8517d Config struct for VideoEncoder. pbos@webrtc.org 2014-09-19 12:30:25 +00:00
  • 02686115cc Re-enable missing android tests disabled due to issue 3770. andresp@webrtc.org 2014-09-19 08:24:19 +00:00
  • 2036a7bb40 Clean directx_sdk_path as it is already defined in base/common.gypi andresp@webrtc.org 2014-09-19 08:14:12 +00:00
  • 5ca6008236 Creating a test helper class TimestampJumpRtpGenerator henrik.lundin@webrtc.org 2014-09-19 07:14:31 +00:00
  • 6e5c78422d (Auto)update libjingle 75875619-> 75878731 buildbot@webrtc.org 2014-09-19 06:46:37 +00:00
  • b5a5c44ef7 (Auto)update libjingle 75865376-> 75875619 buildbot@webrtc.org 2014-09-19 05:36:18 +00:00
  • d7acf11e8d (Auto)update libjingle 75854833-> 75865376 buildbot@webrtc.org 2014-09-19 02:01:09 +00:00
  • ccb3e3f3db (Auto)update libjingle 75854418-> 75854833 buildbot@webrtc.org 2014-09-18 23:31:03 +00:00
  • dcc1f0426b (Auto)update libjingle 75852725-> 75853560 buildbot@webrtc.org 2014-09-18 23:14:12 +00:00
  • 0b435ba597 A few fixes to avoid crash in HW codec on device orientation change. glaznev@webrtc.org 2014-09-18 23:01:03 +00:00
  • 143ffa4bd5 Update iOS video capture to use non-deprecated APIs. tkchin@webrtc.org 2014-09-18 21:44:54 +00:00
  • 83af77bf3c Revert maximum video codec resolution on Android back to 720p again. glaznev@webrtc.org 2014-09-18 21:11:29 +00:00
  • 933d88af58 (Auto)update libjingle 75818332-> 75837294 buildbot@webrtc.org 2014-09-18 20:23:05 +00:00
  • c3091a6c26 Remove the 'webrtc_test_video_render_dependencies' target. pbos@webrtc.org 2014-09-18 17:22:18 +00:00
  • 42731bdded Avoid writing a double/float to a string to avoid a crash. jiayl@webrtc.org 2014-09-18 16:51:51 +00:00
  • ba737cba1a Do not require synchronization access on the thread if called from rtc::Thread::WrapCurrent. The synchronization access is unnecessary for rtc::Thread::WrapCurrent (called from JingleThreadWrapper) since JingleThreadWrapper never calls rtc::Thread::Stop or rtc::Thread::Join. Failing to get the access caused crashes in Chrome since rtc::Thread::Current will be NULL when rtc::Thread::WrapCurrent fails. jiayl@webrtc.org 2014-09-18 16:45:21 +00:00
  • 611606297e Trying to fix Chrome FYI bots. andresp@webrtc.org 2014-09-18 15:50:05 +00:00
  • e94f83a191 Cleanup .gclient_entries to avoid sync problems. kjellander@webrtc.org 2014-09-18 13:47:23 +00:00
  • 205c15a224 Adds asan suppresions for rtc_unittests henrike@webrtc.org 2014-09-18 13:32:43 +00:00
  • 6cd6ba8ae0 Expose VP8/H264 defaults through video_encoder.h. pbos@webrtc.org 2014-09-18 12:42:28 +00:00
  • c7134f8286 Fix proper deps in BUILD.gn files. This should make Chrome GN bots happy. andresp@webrtc.org 2014-09-18 10:06:54 +00:00
  • fda2c2e810 Add Analyze API to NS aluebs@webrtc.org 2014-09-18 09:54:06 +00:00
  • ab071daab8 Split video_render_module implementation into default and internal implementation. Targets must now link with implementation of their choice instead of at "gyp"-time. andresp@webrtc.org 2014-09-18 08:58:15 +00:00
  • 369a637ac8 Implemented Network::GetBestIP() selection logic as following. guoweis@webrtc.org 2014-09-17 22:37:29 +00:00
  • 3b67f8e0ca Enable HW video decoding on Qualcomm devices. glaznev@webrtc.org 2014-09-17 21:25:51 +00:00
  • d91608dd2d The 2x2 black frame on windows when the shared window is minimized caused an assert from vp8 and may lead to memroy corruption. It's changed to 1x1 to match the behavior on Mac. The Chromium code will detect the size and convert it to a black frame in the original size. jiayl@webrtc.org 2014-09-17 16:12:49 +00:00
  • 5422e724d3 Modifying NetEqExternalDecoderTest henrik.lundin@webrtc.org 2014-09-17 15:09:08 +00:00
  • 4a5061fbff talk/p2p/base: removed unused variable "port_" henrike@webrtc.org 2014-09-17 12:33:07 +00:00
  • 5a098c51ea Refactor VP8 de-packetizer. stefan@webrtc.org 2014-09-17 11:58:20 +00:00
  • 3bd5603b18 Revert "Disable video_capture_tests for Android." (revision 7023). andresp@webrtc.org 2014-09-17 11:56:25 +00:00
  • a74eda1b6f Split video_capture_module specific implementation (external vs internal capture) into its own targets. Dependencies must link directly with the desired one. andresp@webrtc.org 2014-09-17 11:50:19 +00:00
  • 85ef770d92 Split video engine android initialization into each internal module initialization. andresp@webrtc.org 2014-09-17 11:44:51 +00:00
  • ab990ae43a Revert 7151 "Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h."" pbos@webrtc.org 2014-09-17 09:02:25 +00:00
  • 6a9b155798 (Auto)update libjingle 75683337-> 75695882 buildbot@webrtc.org 2014-09-17 08:08:38 +00:00
  • e387cc0d37 webrtc/overrides: add OWNERS-file. henrike@webrtc.org 2014-09-17 08:04:28 +00:00
  • dc8dcb4b8c Narrower include for constructormagic.h in Chromium. pbos@webrtc.org 2014-09-17 07:44:33 +00:00
  • eb43264f26 Remove linux_memcheck from default trybots. kjellander@webrtc.org 2014-09-17 06:46:38 +00:00
  • a59c501c99 Java VideoRenderer class may be backed by two different native classes depending on type of rendering. Fix crash in AppRtcDemo by calling correct destructor on exit. glaznev@webrtc.org 2014-09-17 03:26:59 +00:00
  • 40c2aa36f2 Implemented Network::GetBestIP() selection logic as following. guoweis@webrtc.org 2014-09-16 20:29:41 +00:00
  • f8bff762d1 Implemented Network::GetBestIP() selection logic as following. guoweis@webrtc.org 2014-09-16 20:17:22 +00:00
  • 7351d4d698 Add a gyp target for producing a voice engine merged library. andrew@webrtc.org 2014-09-16 18:48:53 +00:00
  • a6cefcaceb gn: Fix cflags usage pbos@webrtc.org 2014-09-16 17:57:02 +00:00
  • cddd17c0f8 Recreate VideoStreams when setting resolution. pbos@webrtc.org 2014-09-16 16:33:13 +00:00
  • 88e85ad64d Add pbos@webrtc.org (myself) to talk/media/webrtc/. pbos@webrtc.org 2014-09-16 16:14:51 +00:00
  • dae612ebf8 Mark all virtual overrides in the hierarchies of UdpTransportData and UdpSocketWrapper as such. henrikg@webrtc.org 2014-09-16 15:29:02 +00:00
  • 80132e4d70 (Auto)update libjingle 75610402-> 75610402 buildbot@webrtc.org 2014-09-16 15:24:15 +00:00
  • 699c46ac57 rtc_unittest: prevent execution of broken tests. henrike@webrtc.org 2014-09-16 11:19:32 +00:00
  • 44360200e3 Fix GN for rtc_base_approved target. kjellander@webrtc.org 2014-09-16 11:16:12 +00:00
  • 178015d8f9 memcheck: suppressions didn't map over directly when moving base from talk to webrtc (part of the suppression that is not related to the signature differed). Fixed suppressions accordingly. henrike@webrtc.org 2014-09-16 09:41:21 +00:00