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  • 992febb997 (Auto)update libjingle 74873066-> 74873164 buildbot@webrtc.org 2014-09-05 16:39:08 +00:00
  • a3344cfda4 Fix webrtcvideoframe tests. thorcarpenter@google.com 2014-09-05 16:34:13 +00:00
  • ddb85ab85b Updated SCTP SDP attributes according to draft-ietf-mmusic-sctp-sdp-07 jiayl@webrtc.org 2014-09-05 16:31:56 +00:00
  • 8f073c5054 Create a new interface for AudioCodingModule henrik.lundin@webrtc.org 2014-09-05 13:16:23 +00:00
  • af5fa95258 (Auto)update libjingle 74857067-> 74860820 buildbot@webrtc.org 2014-09-05 13:03:50 +00:00
  • 7e3bd3d7de (Auto)update libjingle 74851128-> 74857067 buildbot@webrtc.org 2014-09-05 11:45:42 +00:00
  • bc6fa1876e (Auto)update libjingle 74825992-> 74851128 buildbot@webrtc.org 2014-09-05 11:08:01 +00:00
  • 287e9614b3 Disable TestDrain test on memcheck bots. pbos@webrtc.org 2014-09-05 10:11:24 +00:00
  • cdb48dbc23 Enable VideoAdapterTest.BlackOutput on DrMemory. pbos@webrtc.org 2014-09-05 09:46:34 +00:00
  • fed47dc205 Drop buildbot_tests.py script kjellander@webrtc.org 2014-09-05 08:25:38 +00:00
  • a2da031dc0 Remove use_relative_paths from DEPS kjellander@webrtc.org 2014-09-05 08:25:24 +00:00
  • bcf75e32a3 Modifying audio_coding/codecs/OWNERS henrik.lundin@webrtc.org 2014-09-05 07:18:50 +00:00
  • c2c4117477 common_audio: Replaced WEBRTC_SPL_LSHIFT_U32 with << in audio_processing bjornv@webrtc.org 2014-09-05 06:01:53 +00:00
  • 2c03a97d37 Roll chromium_revision f0a439d..94532b1 kjellander@webrtc.org 2014-09-05 05:33:31 +00:00
  • 818b7b3ac9 (Auto)update libjingle 74825084-> 74825992 buildbot@webrtc.org 2014-09-05 00:14:03 +00:00
  • dfbcf8161e Fix an issue in MediaStreamSignaling that a remotely create DataChannel is added to the list twice. jiayl@webrtc.org 2014-09-05 00:01:12 +00:00
  • f1427c6731 Revert 7070 "TurnPort should retry allocation with a new address on error STUN_ERROR_ALLOCATION_MISMATCH." henrike@webrtc.org 2014-09-04 22:21:33 +00:00
  • 4b234044d5 Reduce maximum video resolution for Android. glaznev@webrtc.org 2014-09-04 19:50:07 +00:00
  • 574f2f60fe TurnPort should retry allocation with a new address on error STUN_ERROR_ALLOCATION_MISMATCH. jiayl@webrtc.org 2014-09-04 19:11:34 +00:00
  • 021e76fd39 Add support for WAV output in audioproc aluebs@webrtc.org 2014-09-04 18:12:00 +00:00
  • 52055a276d 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 2014-09-04 17:12:25 +00:00
  • afa77cd803 Add direct_dependent_config to desktop_capture in GN build. brettw@chromium.org 2014-09-04 17:00:55 +00:00
  • ceb956b29d Abort Negotiate() if DoCreateOffer() fails. pbos@webrtc.org 2014-09-04 15:27:49 +00:00
  • d57c95fde4 Change Chromium .gclient to not use Managed mode. kjellander@webrtc.org 2014-09-04 14:58:55 +00:00
  • fa822b940f Fix strange owners files with comments that crashs "git cl presubmit" andresp@webrtc.org 2014-09-04 14:25:07 +00:00
  • 79ee97cf43 [MIPS] Fix gn gen failure for MIPS in webrtc kjellander@webrtc.org 2014-09-04 14:10:49 +00:00
  • 38ef664418 Moving the api.js and bot.js to /rtcbot/bot/ to be shared between /borwser and /android houssainy@google.com 2014-09-04 13:44:47 +00:00
  • 262e676a08 Reland rev 7041 with BUILD.gn files. andresp@webrtc.org 2014-09-04 13:28:48 +00:00
  • 3cbd6c26c8 Fix MSVC warnings about value truncations, webrtc/common_audio/ edition. bjornv@webrtc.org 2014-09-04 13:21:44 +00:00
  • f6ab6f86e7 Rename Audio[Multi]Vector.CopyFrom to .CopyTo henrik.lundin@webrtc.org 2014-09-04 10:58:43 +00:00
  • 3c0aae17f0 Change gflags and gmock includes to be full paths. kjellander@webrtc.org 2014-09-04 09:55:40 +00:00
  • 51bb33cc18 ACMOpus: Remove useless member variable fec_enabled_ kwiberg@webrtc.org 2014-09-04 08:42:44 +00:00
  • 7825b1abf9 Add support for multi-channel DTMF tone generation henrik.lundin@webrtc.org 2014-09-04 07:39:21 +00:00
  • bcb6bcfe6c Remove HybridVideoEngine. pbos@webrtc.org 2014-09-04 07:32:26 +00:00
  • 9d453931c5 Change return value for number of discarded packets to be int. asapersson@webrtc.org 2014-09-04 07:07:44 +00:00
  • 01581da711 Fix audio/video sync when FEC is enabled. stefan@webrtc.org 2014-09-04 06:48:14 +00:00
  • bfd7a8c448 Fix compile errors on webrtc/base. andresp@webrtc.org 2014-09-04 04:59:52 +00:00
  • 0229cbae33 Remove ambiguous call to MakeCheckOpString. andresp@webrtc.org 2014-09-04 04:53:29 +00:00
  • 95c2458766 * Move test data assests required by video frame tests to be in libjingle instead of elsewhere and co-located with other libjingle test data files. thorcarpenter@google.com 2014-09-03 23:17:36 +00:00
  • 9328f39a3e cast return values in uint16_t RTPFile::Read() to uint16_t to avoid compile error BUG=3663 TESTED=ninja local build on windows. R=andrew@webrtc.org, kwiberg@webrtc.org, thorcarpenter@google.com fbarchard@google.com 2014-09-03 23:05:07 +00:00
  • 5b83af49c1 Fix leak of NSAutoreleasePool. tkchin@webrtc.org 2014-09-03 22:53:34 +00:00
  • 609f987488 (Auto)update libjingle 74696326-> 74723281 buildbot@webrtc.org 2014-09-03 21:50:32 +00:00
  • 1b8b4c4959 Revert 7041 " Audio codecs to include webrtc/typedefs.h" henrike@webrtc.org 2014-09-03 19:42:16 +00:00
  • fa4535b270 (Auto)update libjingle 74694022-> 74696326 buildbot@webrtc.org 2014-09-03 16:49:04 +00:00
  • 26c0c41a06 Network up/down signaling in Call. pbos@webrtc.org 2014-09-03 16:17:12 +00:00
  • ebee401230 Remove flake in SendsLowerResolutionOnSmallerFrames. pbos@webrtc.org 2014-09-03 15:52:02 +00:00
  • c4175b9fdf Set resolution based on incoming VideoFrames. pbos@webrtc.org 2014-09-03 15:25:49 +00:00
  • 9730d3aae9 Audio codecs to include webrtc/typedefs.h andresp@webrtc.org 2014-09-03 14:37:18 +00:00
  • 0372b93118 Partial revert of r7014 (Android APK refactor) kjellander@webrtc.org 2014-09-03 14:34:46 +00:00
  • bac072667b Use the sample rate as a temporary solution to unpack aecdumps with wrong sizes aluebs@webrtc.org 2014-09-03 13:39:01 +00:00
  • adee8f9242 Renaming SetOpusMaxBandwidth to SetOpusMaxPlaybackRate minyue@webrtc.org 2014-09-03 12:28:06 +00:00
  • 0a214ffa8a Setting marker bit on DTMF correctly stefan@webrtc.org 2014-09-03 11:46:54 +00:00
  • 74cf916924 Fix issues in audioproc for float aecdumps aluebs@webrtc.org 2014-09-03 11:05:01 +00:00
  • 48f2568d89 audio_processing/nsx: Bug fix that could cause divide by zero bjornv@webrtc.org 2014-09-03 07:58:37 +00:00
  • d944a6887d Suppressing VideoAdapterTest.AdaptResolutionWide and VideoAdapterTest.AdaptResolutionNarrow on DrMemory minyue@webrtc.org 2014-09-03 07:43:32 +00:00
  • 72e448559d (Auto)update libjingle 74628537-> 74648573 buildbot@webrtc.org 2014-09-03 00:43:48 +00:00
  • 90750482fa Remove deprecated RTCVideoRenderer constructor. tkchin@webrtc.org 2014-09-02 20:50:00 +00:00
  • 34a6764981 Remove the checks.h dependence on logging.h in a standalone build. andrew@webrtc.org 2014-09-02 19:00:45 +00:00
  • 8e24d87778 Fix race in Voice Engine's Channel where it accesses RemoteNtpTimeEstimator from both the audio playback thread and the network thread without locking. stefan@webrtc.org 2014-09-02 18:58:24 +00:00
  • 9f341283f6 Remove WebRtcVideoEngine::default_codec_format(). pbos@webrtc.org 2014-09-02 16:33:09 +00:00
  • 03655143db Remove files from talk/PRESUBMIT.py. pbos@webrtc.org 2014-09-02 16:17:36 +00:00
  • d72a7599d4 Create a copy of talk/xmllite under webrtc/xmllite. henrike@webrtc.org 2014-09-02 15:41:12 +00:00
  • 6f729e8a74 Disable video_engine_tests and webrtc_perf_tests on Android. kjellander@webrtc.org 2014-09-02 15:13:55 +00:00
  • ee0fb187a5 Divide-by-zero problem in NetEq's Normal::Process fixed henrik.lundin@webrtc.org 2014-09-02 13:22:11 +00:00
  • 94da2034b0 Remove retired android_apk[_rel] trybots from PRESUBMIT.py kjellander@webrtc.org 2014-09-02 13:05:58 +00:00
  • 324b72dda6 Disable video_capture_tests for Android. kjellander@webrtc.org 2014-09-02 12:37:50 +00:00
  • e281f7fba3 GN: Update webrtc/base to recent GYP changes. kjellander@webrtc.org 2014-09-02 11:22:06 +00:00
  • 468516c959 RTCBot is a framework that allows to write tests where logic runs on a single host that controls multiple endpoints ("bots"). Thus allowing to create more complex scenarios that would otherwise require non-trival signalling between multiple parties. andresp@webrtc.org 2014-09-02 10:52:54 +00:00
  • 561a9eccc5 Update checkedeps.py rules in DEPS. kjellander@webrtc.org 2014-09-02 09:39:35 +00:00
  • 76a42577ad Remove build_with_chromium==1 conditions for Android kjellander@webrtc.org 2014-09-02 08:40:39 +00:00
  • 841f58f64c Unpacking aecdumps generates wav files aluebs@webrtc.org 2014-09-02 07:51:51 +00:00
  • c3f42f37b5 Fix audio_decoder_unittests.isolate kjellander@webrtc.org 2014-09-01 15:06:14 +00:00
  • 8dbeb5b301 Adding more codecs to the AcmSenderBitExactness henrik.lundin@webrtc.org 2014-09-01 14:19:00 +00:00
  • 7e86049d21 Roll chromium_revision 681cc8e..f0a439d (r292217:r292861) kjellander@webrtc.org 2014-09-01 11:41:56 +00:00
  • 3bd4156d75 Android APK tests built from a normal WebRTC checkout. kjellander@webrtc.org 2014-09-01 11:06:37 +00:00
  • c4870bb221 GN: Audio device module kjellander@webrtc.org 2014-09-01 04:24:11 +00:00
  • 524b8f7304 GN: Implement voice engine, common audio, audio coding and audio processing kjellander@webrtc.org 2014-08-31 20:32:53 +00:00
  • 1b9a188ba5 GN: Fix webrtc/video/BUILD.gn for Chromium build. kjellander@webrtc.org 2014-08-29 21:39:35 +00:00
  • a22485eaf0 MIPS optimizations for AEC audio processing module andrew@webrtc.org 2014-08-29 17:51:28 +00:00
  • af7fdfcde8 Add LTO support for Android Chromium. andrew@webrtc.org 2014-08-29 17:41:13 +00:00
  • f554d75288 Allow same src and dst in InputAudioFile::DuplicateInterleaved henrik.lundin@webrtc.org 2014-08-29 07:26:40 +00:00
  • 44010f3e52 win: Replace custom assert() macro with regular assert.h thakis@chromium.org 2014-08-29 03:00:15 +00:00
  • bc3f333905 Add jiayl to talk OWNERS. jiayl@webrtc.org 2014-08-28 23:24:36 +00:00
  • e21cc9ae2a When the peerconnection creates the offer with a constraint to disable the audio offering, stats will not get properly updated. jiayl@webrtc.org 2014-08-28 22:21:34 +00:00
  • b0dc3d7204 Precompile out our standalone CHECK macros in a Chromium build. andrew@webrtc.org 2014-08-28 19:00:15 +00:00
  • a5b7869f3d Add CHECK and friends from Chromium. andrew@webrtc.org 2014-08-28 16:28:26 +00:00
  • 11c6bde474 Specify an ECDH group for ECDHE. jiayl@webrtc.org 2014-08-28 16:14:38 +00:00
  • 55e9da1772 Add talk owners to migrated talk folders henrike@webrtc.org 2014-08-28 16:03:58 +00:00
  • 4431fd6ad5 Add 60 fps video support niklas.enbom@webrtc.org 2014-08-28 14:57:46 +00:00
  • 788f0581c7 GN: Implement video_engine, video_capture and video_render. kjellander@webrtc.org 2014-08-28 13:51:08 +00:00
  • df9fef6638 common_audio: Removed macro WEBRTC_SPL_DIV bjornv@webrtc.org 2014-08-28 12:57:32 +00:00
  • 1f8a23757a (Auto)update libjingle 74235596-> 74297316 buildbot@webrtc.org 2014-08-28 10:52:44 +00:00
  • 59a1b1b928 Fix the different samples per channel in aecdump aluebs@webrtc.org 2014-08-28 10:43:09 +00:00
  • deaece6ac0 Disable VideoAdapterTest.BlackOutput on DrMemory. pbos@webrtc.org 2014-08-28 09:55:34 +00:00
  • f8723d666a Add unit tests to rtcp_receiver_test. asapersson@webrtc.org 2014-08-28 07:35:06 +00:00
  • 2dbb47abb4 Roll chromium_revision b1748b:681cc8 marpan@webrtc.org 2014-08-28 02:32:45 +00:00
  • 956f281d2f Re-enable all VideoAdapterTests on DrMemory. pbos@webrtc.org 2014-08-27 18:41:58 +00:00
  • 75c3ec1763 Fix data races during VideoAdapterTest tear-down. pbos@webrtc.org 2014-08-27 18:16:13 +00:00
  • 573a1eef3d (Auto)update libjingle 74202294-> 74230205 buildbot@webrtc.org 2014-08-27 17:21:19 +00:00
  • 18584fcde4 Move end of namespace inside #ifdef henrik.lundin@webrtc.org 2014-08-27 10:17:22 +00:00