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  • 74bf7a6523 Add tkchin@ to OWNERS. tkchin@webrtc.org 2014-07-01 18:38:28 +00:00
  • 974bbbb352 Fix uninitialized value in DtlsTransport and TransportDescription. jiayl@webrtc.org 2014-07-01 18:33:07 +00:00
  • 08564546cb roll libyuv to r1025 for mips n32 support, arm nacl port, psnr tool jpeg support. BUG=none TESTED=untested R=andrew@webrtc.org fbarchard@google.com 2014-07-01 18:17:21 +00:00
  • 6335645400 (Auto)update libjingle 70329914-> 70330023 buildbot@webrtc.org 2014-07-01 16:46:01 +00:00
  • 37b4e1bbcb webrtc/base: add dependent setting for gtest include directory that was missed when creating base_tests.gyp. Same as https://code.google.com/p/webrtc/source/browse/trunk/talk/libjingle_tests.gyp?r=6484#39 henrike@webrtc.org 2014-07-01 16:39:17 +00:00
  • 0402515d35 Implement command line flags for peerconnection client example on Windows kjellander@webrtc.org 2014-07-01 16:28:13 +00:00
  • 9138eb649b Fix compile error introduced with r6571. stefan@webrtc.org 2014-07-01 12:44:05 +00:00
  • 5779ca478d Fixes a potential BWE clock mismatch bug. stefan@webrtc.org 2014-07-01 12:14:49 +00:00
  • 6d21ddca5f audio_processing/aec: Refactors NonLinearProcessing to prepare for NEON optimizations bjornv@webrtc.org 2014-07-01 10:03:42 +00:00
  • d5a0506e84 Use X509_NAME, not struct X509_name_st. henrike@webrtc.org 2014-06-30 20:38:56 +00:00
  • 59adb1dcd7 Neon version of cftmdl_128() bjornv@webrtc.org 2014-06-30 19:34:33 +00:00
  • 9825afc3bd Add ExperimentalNs support in Config aluebs@webrtc.org 2014-06-30 17:39:53 +00:00
  • 2be53a3b00 Disable CanSwitchToUseAllSsrcs on DrMemory. pbos@webrtc.org 2014-06-30 15:53:07 +00:00
  • be9d2a4549 Reserve RTP/RTCP modules in SetSSRC. pbos@webrtc.org 2014-06-30 13:19:09 +00:00
  • cd9b90ab53 Neon version of cft1st_128() bjornv@webrtc.org 2014-06-30 12:05:18 +00:00
  • e9b9ec5ced Removing W3C conformance tests after move to web-platform-tests. phoglund@webrtc.org 2014-06-30 09:54:27 +00:00
  • ae7cfd7bc8 Make MediaOptimization thread-safe. wuchengli@chromium.org 2014-06-30 08:01:47 +00:00
  • 62711f8227 GN: Fix build by disabling compiler warning in base. kjellander@webrtc.org 2014-06-29 13:37:08 +00:00
  • 7497fa74e4 GN: Refactor base/BUILD.gn and fix dbus-glib error. kjellander@webrtc.org 2014-06-28 18:05:22 +00:00
  • b3c188f27b Use the libvpx rev from Chromium's DEPS, not the Chromium rev. andrew@webrtc.org 2014-06-28 17:49:31 +00:00
  • ee4e466661 Roll libvpx: follow the Chromium revision. marpan@webrtc.org 2014-06-27 21:00:22 +00:00
  • 6f833c332c Rebase webrtc/base with r6555 version of talk/base: cd webrtc/base svn diff -r 6521:6555 http://webrtc.googlecode.com/svn/trunk/talk/base > 6555.diff patch -p0 -i 6555.diff henrike@webrtc.org 2014-06-27 16:21:49 +00:00
  • bfa758a54c (Auto)update libjingle 70004190-> 70103367 buildbot@webrtc.org 2014-06-27 16:04:43 +00:00
  • 680555f923 constructormagic.h macros are duplicated in several repositories. undef them in webrtc to prevent conflict for some build configurations. henrike@webrtc.org 2014-06-27 15:49:02 +00:00
  • f4d6d7c27e Add DrMemory suppression for AsyncWriteTest aluebs@webrtc.org 2014-06-27 13:55:36 +00:00
  • 767d98ebff TSan: Move suppressions to source file. kjellander@webrtc.org 2014-06-27 09:18:51 +00:00
  • 994d0b7229 Refactor Call-based tests. pbos@webrtc.org 2014-06-27 08:47:52 +00:00
  • 35d46fbe1a Roll chromium_revision 277350:280149 kjellander@webrtc.org 2014-06-27 07:54:02 +00:00
  • c8e98187d1 Receiver bit-exactness test for AudioCoding Module henrik.lundin@webrtc.org 2014-06-26 19:07:04 +00:00
  • 7ea71de396 clock.h: Removed GUARDED_BY annotation as it breaks som builds. henrike@webrtc.org 2014-06-26 16:13:58 +00:00
  • 1d1e40f36e Add Chromium's src/buildtools to DEPS. kjellander@webrtc.org 2014-06-26 14:02:16 +00:00
  • 19db3e3164 Don't forward declare RWLockWrapper in clock.h henrik.lundin@webrtc.org 2014-06-26 14:01:18 +00:00
  • aa0e56e8e8 Fixes a bug causing NACKs to be dropped excessively at the send-side. stefan@webrtc.org 2014-06-26 11:44:49 +00:00
  • 269605ce45 Implement SetSendCodecs() unit tests for WebRtcVideoChannel2. pbos@webrtc.org 2014-06-26 08:49:03 +00:00
  • 420ca434b1 (Auto)update libjingle 69860953-> 70002228 buildbot@webrtc.org 2014-06-26 08:08:40 +00:00
  • a2142caa2f Bump version number to 3.55 tnakamura@webrtc.org 2014-06-25 22:04:27 +00:00
  • fe526ff10f fix after r6472 in rtp_sender, comparison between signed and unsigned integer expressions. henrike@webrtc.org 2014-06-25 20:59:51 +00:00
  • 4ddcc40d32 pkg-config-wrapper should not be run when build_nss is disabled (=0). henrike@webrtc.org 2014-06-25 20:14:13 +00:00
  • 3b84b3a58c Add RTCP packet types to packet builder: REMB, TMMBR, TMMBN and extended reports: RRTR, DLRR, VoIP metric. asapersson@webrtc.org 2014-06-25 12:22:17 +00:00
  • 6568e97d10 This is to compare NetEq with various codecs under a shared packet loss pattern. minyue@webrtc.org 2014-06-25 12:17:41 +00:00
  • d5075bdbb5 Neon version of FilterFar() bjornv@webrtc.org 2014-06-25 12:15:46 +00:00
  • 1ed1af9b31 Remove payload duplication in AudioDecoderTest henrik.lundin@webrtc.org 2014-06-25 07:59:40 +00:00
  • ec9f5fb34c Change SdpSerializeCandidate to output candidate line without the "a=" and without the leading \r\n", i.e. candidate-attribute as defined in section 15.1 of [ICE]. wu@webrtc.org 2014-06-24 17:05:10 +00:00
  • 1da152d7f6 talk/base and webrtc/base suppression had the same names for their suppressions which is not allowed. Renamed the talk/base ones as they are going away. henrike@webrtc.org 2014-06-24 14:24:37 +00:00
  • eecf5e6ba7 Removing neteq decode lock and friends henrik.lundin@webrtc.org 2014-06-24 13:11:22 +00:00
  • 05f1464df3 Exclude AsyncWriteTest.TestWrite from Win DrMemory Full bot and suppress the reported errors aluebs@webrtc.org 2014-06-24 11:08:51 +00:00
  • 04fbc38c4a Neon version of ScaleErrorSignal() bjornv@webrtc.org 2014-06-24 10:25:00 +00:00
  • 9a4f651037 Disable PhysicalSocketTest.TestUdpReadyToSendIPv4 for TSAN2 aluebs@webrtc.org 2014-06-24 08:35:39 +00:00
  • 71dffb76dc (Auto)update libjingle 69648312-> 69830415 buildbot@webrtc.org 2014-06-24 07:24:49 +00:00
  • b338ca6557 Annotating the rest of AcmGenericCodec henrik.lundin@webrtc.org 2014-06-24 05:51:34 +00:00
  • f6d37de466 Fix array declarations in aec_core.c andrew@webrtc.org 2014-06-23 22:40:58 +00:00
  • ceb5a1d724 Annotating the rest of AudioCodingModuleImpl henrik.lundin@webrtc.org 2014-06-23 19:52:27 +00:00
  • 1227ab89a7 GN: Add BUILD.gn files + kjellander to OWNERS kjellander@webrtc.org 2014-06-23 19:21:07 +00:00
  • c00ca627fd Rebase webrtc/base with r6521 version of talk/base: cd webrtc/base svn diff -r 6466:66521 http://webrtc.googlecode.com/svn/trunk/talk/base > 6521.diff patch -p0 -i 6521.diff henrike@webrtc.org 2014-06-23 16:15:27 +00:00
  • 948f768580 Roll libvpx 269083:278497 fgalligan@google.com 2014-06-23 15:07:15 +00:00
  • b6ebe75806 Disables tests that breaks Android bots bjornv@webrtc.org 2014-06-23 09:14:03 +00:00
  • a36a259858 TSan v2 deadlock suppressions. kjellander@webrtc.org 2014-06-22 08:01:42 +00:00
  • a97f6f34b2 Exclude flaky libjingle_peerconnection_unittest test for Memcheck. kjellander@webrtc.org 2014-06-22 07:11:44 +00:00
  • c70b2f9a54 Add third_party/colorama to DEPS kjellander@webrtc.org 2014-06-21 19:54:15 +00:00
  • 27ab19d9b4 Roll chromium_revision 272489:277350 + fix sanitizer options kjellander@webrtc.org 2014-06-21 19:30:29 +00:00
  • 78f440c5e7 GN: BUILD.gn for system_wrappers kjellander@webrtc.org 2014-06-21 14:25:16 +00:00
  • ff1b1bf094 When creating an answer, takes the codec preference from the offer. wu@webrtc.org 2014-06-20 20:57:42 +00:00
  • a24d366e1c - Exit from a camera thread lopper loop() method only after all camera release calls are completed. This fixes camera exceptions observed from time to time when calling camera functions on a terminated looper. - Allocate real texture for camera preview. - Add fps and camera frame duration logging. - Get camera frame timestamp in Java code and pass it to jni code so the frame timestamp is assigned as soon as possible. Jni code will not use these timestamps yet until timestamp ntp correction and zeroing in webrtcvideengine.cc will be addressed. glaznev@webrtc.org 2014-06-20 20:55:54 +00:00
  • 0d15159b04 (Auto)update libjingle 69634309-> 69640360 buildbot@webrtc.org 2014-06-20 19:02:09 +00:00
  • b43c99de29 Limits the send and receive buffer by bytes, not by packets. The new limit is 16MB for each buffer. Also refactors the code to handle send failure more consistently. jiayl@webrtc.org 2014-06-20 17:11:14 +00:00
  • db397e5c6c Re-evalutes the ICE role on ICE restart. Also unifies the logic of ICE restart. jiayl@webrtc.org 2014-06-20 16:32:09 +00:00
  • 0b893b1e05 Do not hold the critical section in VideoCaptureAndroid::SetCaptureRotation since it would case possible deadlock with OS Camear thread. braveyao@webrtc.org 2014-06-20 16:00:30 +00:00
  • bb2d65895b (Auto)update libjingle 69617317-> 69623266 buildbot@webrtc.org 2014-06-20 14:58:56 +00:00
  • 75ce92086c (Auto)update libjingle 69600065-> 69617317 buildbot@webrtc.org 2014-06-20 12:30:24 +00:00
  • f425b55eeb Add tests of texture frames in video_send_stream_test. wuchengli@chromium.org 2014-06-20 12:04:05 +00:00
  • 83785d37d1 Remove unused ALLOCATE_DELAY constant. pbos@webrtc.org 2014-06-20 10:28:39 +00:00
  • 4c25c67146 (Auto)update libjingle 69589535-> 69600065 buildbot@webrtc.org 2014-06-20 04:42:34 +00:00
  • 58e7c8660c (Auto)update libjingle 69588980-> 69589535 buildbot@webrtc.org 2014-06-20 00:26:50 +00:00
  • 0970dd8767 (Auto)update libjingle 69588608-> 69588980 buildbot@webrtc.org 2014-06-20 00:18:36 +00:00
  • 8563ef448a (Auto)update libjingle 69587333-> 69588608 buildbot@webrtc.org 2014-06-20 00:13:01 +00:00
  • 1ef789d455 (Auto)update libjingle 69568113-> 69587333 buildbot@webrtc.org 2014-06-19 23:54:12 +00:00
  • 594aefa807 Do not call CaptureCursor in ScreenCapturerWinGdi if no MouseShapeObserver. It's wasted work and affects frame rate adaptation in Chrome. jiayl@webrtc.org 2014-06-19 22:04:41 +00:00
  • df9bbbee56 (Auto)update libjingle 69567902-> 69568113 buildbot@webrtc.org 2014-06-19 19:54:33 +00:00
  • fbd13286dc (Auto)update libjingle 69555283-> 69567902 buildbot@webrtc.org 2014-06-19 19:50:55 +00:00
  • 21794f9862 (Auto)update libjingle 69543894-> 69555283 buildbot@webrtc.org 2014-06-19 17:14:19 +00:00
  • 304ca76be1 Revert 6481 and 6482 fgalligan@google.com 2014-06-19 17:08:46 +00:00
  • 8de8c9155e Maintain constantness of the input to iSAC-fix decoder, and prevent heap-buffer overflow. turaj@webrtc.org 2014-06-19 15:47:09 +00:00
  • 9158df2aa4 Adding an empty constructor implementation to the AudioSink class henrik.lundin@webrtc.org 2014-06-19 12:34:31 +00:00
  • 84f8ec1f9c Changes to tests and tools in audio_processing. bjornv@webrtc.org 2014-06-19 12:14:33 +00:00
  • 077593b805 Ensure that the start bitrate can be set multiple times. stefan@webrtc.org 2014-06-19 12:13:00 +00:00
  • 496a98463b Adding test::AudioSink interface and derived classes henrik.lundin@webrtc.org 2014-06-19 10:02:11 +00:00
  • 5c3f4e3b0f Fixes and re-enables tests disabled on Android bjornv@webrtc.org 2014-06-19 09:51:29 +00:00
  • d27d9ae644 (Auto)update libjingle 69506154-> 69515138 buildbot@webrtc.org 2014-06-19 01:56:46 +00:00
  • 6ce1d58613 Exclude flaky test PeerConnectionEndToEndTest.CreateDataChannelAfterNegotiate on memcheck. jiayl@webrtc.org 2014-06-19 00:06:36 +00:00
  • acede34aea Fix a memory leak in SctpDataMediaChannelTest. jiayl@webrtc.org 2014-06-18 23:36:16 +00:00
  • 85b19a1a12 Exclude SctpDataMediaChannelTest on Win DrMemory for third_party/usrsctp issues. jiayl@webrtc.org 2014-06-18 23:34:18 +00:00
  • f8063d34de Properly shut down the SCTP stack. jiayl@webrtc.org 2014-06-18 21:30:40 +00:00
  • a19b930b5b Update webrtc to fix unpack_lib expansion. fgalligan@google.com 2014-06-18 19:20:45 +00:00
  • 8f06a8aeb0 Update generated asm offsets scripts. fgalligan@google.com 2014-06-18 17:38:08 +00:00
  • b947d954a5 Neon version of FilterAdaptation() bjornv@webrtc.org 2014-06-18 14:55:49 +00:00
  • 12396aba42 Update PacketSource and RtpFileSource henrik.lundin@webrtc.org 2014-06-18 12:20:31 +00:00
  • d8de0669c9 Revert "Restore ptypes.txt file" henrik.lundin@webrtc.org 2014-06-18 11:09:53 +00:00
  • ec869bf781 Revert 6473 "Update generated asm offsets scripts." turaj@webrtc.org 2014-06-17 19:07:56 +00:00
  • e398954658 Update usrsctp to r8875 jiayl@webrtc.org 2014-06-17 18:16:08 +00:00
  • 32196decd6 Update generated asm offsets scripts. fgalligan@google.com 2014-06-17 17:55:23 +00:00