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  • 595b23c66f Revert 7184 "Enable ipv6 by default for webrtc under a Finch exp..." kjellander@webrtc.org 2014-09-16 08:58:22 +00:00
  • c75f607042 audio_processing/aec: Ported NEON optimizations of SubbandCoherence() and its sub-functions to SSE2 bjornv@webrtc.org 2014-09-16 05:01:42 +00:00
  • 6ae5a6d7fe Add a target for the approved subset of rtc_base. andrew@webrtc.org 2014-09-16 01:03:29 +00:00
  • b3cbeb31cc Fix memory leak in webrtc::MouseCursorMonitorMac sergeyu@chromium.org 2014-09-15 20:11:23 +00:00
  • ab7073a1e8 Partial implementation of rtc::LogMessage in chromium overrides. glaznev@webrtc.org 2014-09-15 19:16:21 +00:00
  • 996784548d HW video decoding optimization to better support HD resolution: glaznev@webrtc.org 2014-09-15 17:52:42 +00:00
  • cd309e3168 Enable ipv6 by default for webrtc under a Finch experiment. guoweis@webrtc.org 2014-09-15 16:31:13 +00:00
  • 000d86792d Make BW checks > 0 in peerconnection_unittest.cc. pbos@webrtc.org 2014-09-15 14:38:07 +00:00
  • 7bb2586c55 audio_processing: Correct sample rate in aec_debug_dump bjornv@webrtc.org 2014-09-15 13:23:07 +00:00
  • 76ba7caae8 Re-enable neteq_performance_unittest.cc for android. andresp@webrtc.org 2014-09-15 12:29:50 +00:00
  • 541753f96c Re-enable rampup_tests.cc for Android. andresp@webrtc.org 2014-09-15 12:27:35 +00:00
  • 4a6c5b3b01 Re-enable video send stream tests for android. andresp@webrtc.org 2014-09-15 12:24:34 +00:00
  • 18617cfde8 Fix ThreadChecker unittests when DCHECK_ALWAYS_ON is defined henrik.lundin@webrtc.org 2014-09-15 11:19:35 +00:00
  • 7f826350e3 Stop building talk/xmllite since it is no longer used. henrike@webrtc.org 2014-09-15 08:13:36 +00:00
  • 192ab710ce Set minimum SDK level to 10.7 for Mac and iOS. kjellander@webrtc.org 2014-09-15 08:02:43 +00:00
  • a42a3ade54 (Auto)update libjingle 75390072-> 75428737 buildbot@webrtc.org 2014-09-13 01:09:18 +00:00
  • 7e31197cb2 Revert 7170 "Revert 7121 "ValidateFrame, When dumping the first ..." BUG=3789 TESTED=drmemory out\Debug\libjingle_media_unittest.exe --gtest_catch_exceptions=0 --gtest_filter=*Validate* fbarchard@google.com 2014-09-13 00:52:42 +00:00
  • 91ee7468dd Add enable flag for Android device orientation change event. glaznev@webrtc.org 2014-09-12 16:48:12 +00:00
  • 192a54ff2f Temporary revert maximum video codec resolution back to 1080p. glaznev@webrtc.org 2014-09-12 16:40:35 +00:00
  • 3decd9b776 Revert 7121 "ValidateFrame, When dumping the first 4 samples of a frame, first copy it to a temporary buffer that is zero padded, them use that." henrike@webrtc.org 2014-09-12 16:31:29 +00:00
  • 1fb5d1204b Initialize restored_packet in nack_rtx_unittest.cc. pbos@webrtc.org 2014-09-12 16:16:00 +00:00
  • c3c9015bc6 linux: remove stray libcrypto dependency henrike@webrtc.org 2014-09-12 16:11:38 +00:00
  • 78b2d56ac6 Disable MethodNotAllowedOnDifferentThreadInDebug. henrike@webrtc.org 2014-09-12 15:57:08 +00:00
  • d2cf48de1a Fix mac video_render implementation on cocoa. andresp@webrtc.org 2014-09-12 13:57:47 +00:00
  • f7e5f22f98 Fix stack limit exceeded in http client. andresp@webrtc.org 2014-09-12 13:35:05 +00:00
  • a0d7827b16 Add ability to downscale content to improve quality. pbos@webrtc.org 2014-09-12 11:51:47 +00:00
  • b5e6bfc76a Make RTPSender/RTPReceiver generic. pbos@webrtc.org 2014-09-12 11:05:55 +00:00
  • 6071b0636d Mark all virtual overrides in the hierarchy of RtpData and RtpReceiver as such. stefan@webrtc.org 2014-09-12 07:42:33 +00:00
  • cc774a69cb Mark all virtual overrides in the hierarchies of RtpDump and VCMPacketizationCallback as such. henrike@webrtc.org 2014-09-11 22:45:54 +00:00
  • ea77334c30 (Auto)update libjingle 75302540-> 75327856 buildbot@webrtc.org 2014-09-11 21:52:48 +00:00
  • 31c285b333 Update AUTHORS file. henrike@webrtc.org 2014-09-11 21:12:59 +00:00
  • 89959966a9 Fix window capturing on Windows when the window is minimized. jiayl@webrtc.org 2014-09-11 19:33:58 +00:00
  • f520ea5eed Skip dlclose() on AddressSanitizer. pbos@webrtc.org 2014-09-11 17:29:11 +00:00
  • 1d8f780779 Stop building talk/sound since it is no longer used. henrike@webrtc.org 2014-09-11 17:16:56 +00:00
  • 1d53f64b0f Disabling initializeAndroidGlobals when built with WEBRTC_CHROMIUM_BUILD. glaznev@webrtc.org 2014-09-11 16:58:25 +00:00
  • b9906743da Split suppressons of thread.cc and messagequeue.cc. pbos@webrtc.org 2014-09-11 14:59:06 +00:00
  • 4b049fcabe Remove developing code in ns_core aluebs@webrtc.org 2014-09-11 11:19:56 +00:00
  • f5bdd54ac3 Add myself to common_audio and audio_processing watchlists aluebs@webrtc.org 2014-09-11 10:11:43 +00:00
  • 307d3dbdee Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h." henrikg@webrtc.org 2014-09-11 09:48:30 +00:00
  • 665d861115 Restore webrtc_base target until r7140 is rolled into Chromium. kjellander@webrtc.org 2014-09-11 09:22:13 +00:00
  • 8dd60cc855 audio_processing_unittests: Enabled ApmTest.Process for all platforms but Android bjornv@webrtc.org 2014-09-11 08:36:35 +00:00
  • c665dcb205 Revert 7145 "Stop building talk/sound since it is no longer used." sprang@webrtc.org 2014-09-11 08:29:53 +00:00
  • 2b58a4433f Calculating round-trip-time in send-only channel in VoE. minyue@webrtc.org 2014-09-11 07:51:53 +00:00
  • 1972ff8a6e Mark all virtual overrides in the hierarchy of Module as virtual and OVERRIDE. henrik.lundin@webrtc.org 2014-09-11 06:20:28 +00:00
  • 4c876453c8 Stop building talk/sound since it is no longer used. henrike@webrtc.org 2014-09-10 22:18:04 +00:00
  • 47658f1269 Mark all virtual overrides in the hierarchy of AudioPacketizationCallback, RTPStream, and NetEq as such. Also mark all other virtual overrides in the same files. henrike@webrtc.org 2014-09-10 22:14:59 +00:00
  • 1711104b8a Fix MSVC warnings about value truncations, webrtc/base/ edition. henrike@webrtc.org 2014-09-10 22:10:24 +00:00
  • 3472dcd7b0 Fix frame rate selection for Android camera. glaznev@webrtc.org 2014-09-10 19:24:57 +00:00
  • 67eabc0938 Add schannel webrtc_base build using a new use_schannel gyp variable. tpsiaki@google.com 2014-09-10 18:06:47 +00:00
  • b2efb6771c Put base tests in webrtc_tests.gyp henrike@webrtc.org 2014-09-10 17:28:19 +00:00
  • a8d2ee7f3b Roll chromium_revision ea769fd..6455c69 (re-land) kjellander@webrtc.org 2014-09-10 16:51:37 +00:00
  • b6d69282f5 Enable shared socket for TurnPort. In AllocationSequence::OnReadPacket, we now hand the packet to both the TurnPort and StunPort if the remote address matches the server address. jiayl@webrtc.org 2014-09-10 16:31:34 +00:00
  • 0867f69cc6 Convert GN visibility to be lists. brettw@chromium.org 2014-09-10 16:24:11 +00:00
  • 5c20bb27a0 Remove suppressions for VideoFrame::Validate. pbos@webrtc.org 2014-09-10 14:59:09 +00:00
  • 33aa095896 Simplify gyp rules on video_render_module. andresp@webrtc.org 2014-09-10 14:48:48 +00:00
  • e0761d06b0 Fix printing of error stack in rtcbot when a test fails via test.fail(). houssainy@google.com 2014-09-10 14:35:35 +00:00
  • 49fa212bcd Fix compile error on JDK 1.7. kjellander@webrtc.org 2014-09-10 12:35:59 +00:00
  • 0fa04755af Roll gtest-parallel. pbos@webrtc.org 2014-09-10 09:29:12 +00:00
  • 23a5e3c3b0 Remove DestructEncoderInst and its codec-specific implementations. henrik.lundin@webrtc.org 2014-09-10 08:52:26 +00:00
  • a2e6a52563 Revert 7128 "Roll chromium_revision ea769fd..6455c69" kjellander@webrtc.org 2014-09-10 08:38:27 +00:00
  • 5d639b3ef3 (Auto)update libjingle 75141932-> 75179475 buildbot@webrtc.org 2014-09-10 07:57:12 +00:00
  • fdba9ee64a Roll chromium_revision ea769fd..6455c69 kjellander@webrtc.org 2014-09-10 07:42:55 +00:00
  • 4ca66d691e include cstdlib for free() and abort() andrew@webrtc.org 2014-09-10 03:24:36 +00:00
  • fa603981f2 Add a new class InterfaceAddress inherited from IPAddress to keep track of IPv6 Address flags. guoweis@webrtc.org 2014-09-09 23:42:40 +00:00
  • 87ff9c8efa Fix up configs applying to GN build. brettw@chromium.org 2014-09-09 23:34:56 +00:00
  • 7d4891d3f1 Fixes two issues in how we handle OfferToReceiveX for CreateOffer: 1. the options set in the first CreateOffer call should not affect the result of a second CreateOffer call, if SetLocalDescription is not called after the first CreateOffer. So the member var options_ of MediaStreamSignaling is removed to make each CreateOffer independent. Instead, MediaSession is responsible to make sure that an m-line in the current local description is never removed from the newly created offer. jiayl@webrtc.org 2014-09-09 21:43:15 +00:00
  • a941970d4a Change explicit static cast from int to uint16_t to implicit cast of 0u. BUG=3663 TESTED=local windows build with VS2013. R=harryjin@google.com, tina.legrand@webrtc.org fbarchard@google.com 2014-09-09 21:37:27 +00:00
  • 9fe11010f7 Fix the RTC+Chromium GN build. brettw@chromium.org 2014-09-09 19:15:33 +00:00
  • 54cf1505e2 ValidateFrame, When dumping the first 4 samples of a frame, first copy it to a temporary buffer that is zero padded, them use that. BUG=3789 TESTED=drmemory out\Debug\libjingle_media_unittest.exe --gtest_catch_exceptions=0 --gtest_filter=*Validate* R=tommi@webrtc.org fbarchard@google.com 2014-09-09 18:34:53 +00:00
  • 22406fcc9b TurnPort should retry allocation with a new address on error STUN_ERROR_ALLOCATION_MISMATCH. jiayl@webrtc.org 2014-09-09 15:44:05 +00:00
  • 04b853b56a Bot Browser files moved to /bot/browser/ houssainy@google.com 2014-09-09 14:50:09 +00:00
  • 3d81b1b22a Relanding https://code.google.com/p/webrtc/source/detail?r=7093, after it got reverted due to some internal compile failures. mallinath@webrtc.org 2014-09-09 14:38:10 +00:00
  • 4bbd3c83a8 fix a bug in the logic when new Networks are merged. This happens when we have 2 networks with the same key guoweis@webrtc.org 2014-09-09 13:54:45 +00:00
  • 1b088ee316 More suppressions, uninitialized read in cricket::VideoFrame::Validate sprang@webrtc.org 2014-09-09 11:50:19 +00:00
  • 4d19e05ab2 Peerconnection_jni to use webrtc/base/checks.h instead of implementing its own. andresp@webrtc.org 2014-09-09 11:45:44 +00:00
  • b420191743 Expose VideoEncoders with webrtc/video_encoder.h. pbos@webrtc.org 2014-09-09 10:40:56 +00:00
  • 641bda6f9c Initialize ChannelBuffer's memory to avoid uninitialized reads. andrew@webrtc.org 2014-09-08 23:11:44 +00:00
  • 8b0b21161a Revert 7093: "Implementing ICE Transports type handling in libjingle transport." henrike@webrtc.org 2014-09-08 22:46:28 +00:00
  • 519c9e207d Convert GN visibility to be a list. brettw@chromium.org 2014-09-08 22:45:18 +00:00
  • 7118e61669 Finish work queue in SctpDataMediaChannelTest. pbos@webrtc.org 2014-09-08 21:44:07 +00:00
  • 0e52772aa9 Fix a bot-breaking memory leak from early returning in ParseMediaDescription. jiayl@webrtc.org 2014-09-08 21:43:43 +00:00
  • c172320bd2 Revert "Fixes two issues in how we handle OfferToReceiveX for CreateOffer:" because it broke content_browsertests on Android. jiayl@webrtc.org 2014-09-08 20:44:36 +00:00
  • 17454f79dc Add ctors to ChannelBuffer to enable copying on construction. andrew@webrtc.org 2014-09-08 20:27:04 +00:00
  • fd42f9dd6f (Auto)update libjingle 74955991-> 75042522 buildbot@webrtc.org 2014-09-08 19:45:36 +00:00
  • 1272ee59b3 Suppress uninitialized read warning in cricket::VideoFrame::Validate sprang@webrtc.org 2014-09-08 14:00:38 +00:00
  • c64246f42c Set a default speech type in iSAC wrapper henrik.lundin@webrtc.org 2014-09-08 13:40:58 +00:00
  • ed8bcd3ac5 Starting to implement the new ACM API henrik.lundin@webrtc.org 2014-09-08 13:13:19 +00:00
  • 9600519147 Adding the ability to test on Chrome for Android. use "android-chrome" as type in rtcbot running command. Example: node test.js android-chrome houssainy@google.com 2014-09-08 13:01:40 +00:00
  • 37c39f3784 audio_processing: Removed use of macro WEBRTC_SPL_UMUL_16_16 bjornv@webrtc.org 2014-09-08 11:21:56 +00:00
  • 0d394f3609 video_processing: Removed usage of WEBRTC_SPL_UMUL_16_16 bjornv@webrtc.org 2014-09-08 11:19:39 +00:00
  • c77e4d6aef - Adding AndroidDeviceManager to botManager.js to help in selecting devices, in case running test on Android devices. houssainy@google.com 2014-09-08 10:36:11 +00:00
  • 142bb9d870 Roll chromium_revision 94532b1..ea769fd kjellander@webrtc.org 2014-09-08 10:06:37 +00:00
  • fe16167507 Fix RTT calculations for send-only channels. stefan@webrtc.org 2014-09-08 08:45:25 +00:00
  • c30e9e2300 Ignore FEC packet in stats, if it is first packet on ssrc. sprang@webrtc.org 2014-09-08 08:20:18 +00:00
  • 6d08ca6379 GN: Prefix WebRTC specific variables with "rtc_" kjellander@webrtc.org 2014-09-07 17:36:10 +00:00
  • f68cf93e1b Add video_capture_tests_apk_target kjellander@webrtc.org 2014-09-07 17:35:51 +00:00
  • 7256d31d28 Implementing ICE Transports type handling in libjingle transport. mallinath@webrtc.org 2014-09-07 04:08:44 +00:00
  • a781f68712 Fix rm command for class cleanup in r7091 kjellander@webrtc.org 2014-09-06 22:11:28 +00:00
  • 9510022e1f Cleanup temporary class files for OpenSlDemo kjellander@webrtc.org 2014-09-06 18:03:45 +00:00
  • cc060563f3 Remove unnecessary include from testutils.cc. thorcarpenter@google.com 2014-09-05 21:19:00 +00:00