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  • c3e51acb30 To use the channel_transport on the iOS platform, some #if directives are changed. sjlee@webrtc.org 2013-09-17 05:14:32 +00:00
  • 15e979b571 Call AllowCommandLineReparsing in unit tests. andrew@webrtc.org 2013-09-17 00:54:09 +00:00
  • b3af8aea3e Verify local and remote transport description before negotiation. mallinath@webrtc.org 2013-09-17 00:11:05 +00:00
  • f6ae62ffa3 Add Win TSan exclude and Dr Memory suppressions kjellander@webrtc.org 2013-09-16 21:16:48 +00:00
  • eddbfb8cf8 Add more Dr Memory suppressions for common_audio_unittests kjellander@webrtc.org 2013-09-16 20:53:20 +00:00
  • e401c2e391 Split video coding module unit tests into sender and receiver unit tests. andresp@webrtc.org 2013-09-16 20:29:13 +00:00
  • ab800f64bc Disable flaky libjingle tests under tsan and memcheck. stefan@webrtc.org 2013-09-16 15:22:32 +00:00
  • 5860de02aa Implement NACK over RTX for VideoSendStream. pbos@webrtc.org 2013-09-16 13:01:47 +00:00
  • 8fa436bd65 Remove use of vcm->ResetDecoder from modules/utility. andresp@webrtc.org 2013-09-16 11:26:35 +00:00
  • 62b816afcf Fixed pylint warnings. phoglund@webrtc.org 2013-09-16 11:26:12 +00:00
  • 15b8871e4a Allocate float_buffer_ in the initializer list. andrew@webrtc.org 2013-09-14 01:57:55 +00:00
  • 8a1448950c Disable WebRtcSessionTest.TestCreateOfferWithSctpEnabledWithoutStreams sergeyu@chromium.org 2013-09-14 00:30:51 +00:00
  • f7eb75be1a Split VideoCodingModuleImpl into VideoSender and VideoReceiver. andresp@webrtc.org 2013-09-14 00:25:28 +00:00
  • a59696b2a5 Update libjingle to 52300956 sergeyu@chromium.org 2013-09-13 23:48:58 +00:00
  • 48af652ea5 Prepare to compile ACM1 and ACM2. turaj@webrtc.org 2013-09-13 23:06:59 +00:00
  • bc189fb3b9 * Prefer to send ISAC on clank. * Add url option asc and arc to allow setting preferred audio send/receive codec. wu@webrtc.org 2013-09-13 20:11:47 +00:00
  • 6ab45b9dab Implement DesktopRegion subtraction. sergeyu@chromium.org 2013-09-13 19:53:16 +00:00
  • 1f09dbe353 Moving test-only code (stream_generator) out of vcm implemention. andresp@webrtc.org 2013-09-13 19:17:54 +00:00
  • 2553450959 Fix win trybot errors due to r4729. andrew@webrtc.org 2013-09-13 00:02:13 +00:00
  • 6a5cc9d899 Fix crash in the window capturer on windows sergeyu@chromium.org 2013-09-12 19:17:26 +00:00
  • 7959e16cc2 ACM2 integration with NetEq 4. turaj@webrtc.org 2013-09-12 18:30:26 +00:00
  • 82a846f0cb Adding Ami to the video renderer and capturer modules. mallinath@webrtc.org 2013-09-12 17:43:17 +00:00
  • 36cf4d2309 The video render module for iOS. fischman@webrtc.org 2013-09-12 17:39:53 +00:00
  • e509f943ed This issue is related to https://chromereviews.googleplex.com/9908014/ minyue@webrtc.org 2013-09-12 17:03:00 +00:00
  • 8fa03a15ab Make PCM16 available in Chromium builds. andrew@webrtc.org 2013-09-12 01:30:30 +00:00
  • 89df092807 Make the destructor of AudioCodingModule public. andrew@webrtc.org 2013-09-12 01:27:43 +00:00
  • 5eb997a2fd Fix unsigned/signed comparison error due to r4729. andrew@webrtc.org 2013-09-12 01:01:42 +00:00
  • 8f94013651 Reduce frequency of high audio delay warning logs. andrew@webrtc.org 2013-09-11 22:35:00 +00:00
  • 256b83146c Removes function that is not used anywhere but somehow still causing library load issues on Android Release build. henrike@webrtc.org 2013-09-11 20:43:13 +00:00
  • 5c678eabd9 Implement 'abs-send-time' extension in VideoSendStream. pbos@webrtc.org 2013-09-11 19:00:39 +00:00
  • 6138c5cfa4 OpenSl: fixes crashes externally reported in issue 2361 and 2362. henrike@webrtc.org 2013-09-11 18:50:06 +00:00
  • 036b7436df Adding APIs. These APIs are not implemented yet, they are to help developement of ACM. turaj@webrtc.org 2013-09-11 18:45:02 +00:00
  • a80ee74f69 AppRTC: using a footer element instead of div#footer in CSS. braveyao@webrtc.org 2013-09-11 16:24:07 +00:00
  • d4d59ac871 Remove FrameForStorage:Follow up on r4688 mikhal@webrtc.org 2013-09-11 15:18:15 +00:00
  • 2902328cce Implement 'toffset' extension in VideoSendStream. pbos@webrtc.org 2013-09-11 10:14:56 +00:00
  • 554d158ce6 Reset jitter buffer and timing if frames are getting too much delay. stefan@webrtc.org 2013-09-11 08:45:26 +00:00
  • 835ef67d14 Remove repeated conditions key. andrew@webrtc.org 2013-09-11 00:16:00 +00:00
  • 82f014aa0b OpenSL (not default): Enables low latency audio on Android. henrike@webrtc.org 2013-09-10 18:24:07 +00:00
  • 641340944b Show the signaling state and ice connection state in AppRTC by hooking up the peerconnections .onsignalingstatechange and .oniceconnectionstatechange events. braveyao@webrtc.org 2013-09-10 17:37:16 +00:00
  • 319c98d663 Fix format string in video_quality_analysis.cc. pbos@webrtc.org 2013-09-10 15:23:50 +00:00
  • 182d025d94 Remove include_dirs from voice_engine.gyp. pbos@webrtc.org 2013-09-10 15:01:26 +00:00
  • df531a2eee Test that VideoSendStream responds to NACK. pbos@webrtc.org 2013-09-10 14:56:33 +00:00
  • f880f863dd Convert printing in video quality tests to Chromium's perf format. kjellander@webrtc.org 2013-09-10 12:10:01 +00:00
  • e07049f19f Lock RTPSender statistics. pbos@webrtc.org 2013-09-10 11:29:17 +00:00
  • 744fbc7fe4 Split up EngineTests and RampupTests. pbos@webrtc.org 2013-09-10 09:26:25 +00:00
  • eda189be14 Remove redundant STR_CASE_CMP macro definitions. andrew@webrtc.org 2013-09-09 17:50:10 +00:00
  • a19c9f4173 Updated WebRTC version to 3.41 elham@webrtc.org 2013-09-09 17:23:44 +00:00
  • 021c42bfa8 Lock use of _packetRequestCallback in VCM. pbos@webrtc.org 2013-09-09 17:18:31 +00:00
  • 7ebf0e7f44 Remove include_dirs from video_engine_core.gypi. pbos@webrtc.org 2013-09-09 16:56:31 +00:00
  • 59f20bb735 Break out RTCPSender dependency on ModuleRtpRtcpImpl. pbos@webrtc.org 2013-09-09 16:02:19 +00:00
  • 26b0d77baf Suppress RTPSender race regardless of codec. pbos@webrtc.org 2013-09-09 15:34:36 +00:00
  • 841c8a44bb Rename VideoCall to Call. pbos@webrtc.org 2013-09-09 15:04:25 +00:00
  • 86136a0e8f Re-enable tests for Remote Bitrate Estimator solenberg@webrtc.org 2013-09-09 13:06:52 +00:00
  • 0181b5f8dd ExternalVideoDecoder for new VideoEngine API. pbos@webrtc.org 2013-09-09 08:26:30 +00:00
  • 30e055c4dd Handle empty RTP video packets agnostic to codec. pbos@webrtc.org 2013-09-08 11:15:00 +00:00
  • 1b476d9a56 Disabling channelmanager unittest. This test is causing TSAN error. The problem could be in thread Invoke method. mallinath@webrtc.org 2013-09-07 18:59:12 +00:00
  • ab5a0912a3 Fixing the build error on Windows. Problem is in coversion from size_t to int. mallinath@webrtc.org 2013-09-07 00:12:57 +00:00
  • 1b15f4226f Update talk to 51960985. mallinath@webrtc.org 2013-09-06 22:56:28 +00:00
  • b159c2e3dd Reduce cost of PushSincResampler::Resample(). andrew@webrtc.org 2013-09-06 21:15:55 +00:00
  • c7f708679d Clamp camera id to legal values. fischman@webrtc.org 2013-09-06 18:17:45 +00:00
  • b2c8a952a7 Improving padding rules and breaking out bw allocation to ViEEncoder. stefan@webrtc.org 2013-09-06 13:58:01 +00:00
  • 7bb8f02274 Adds support for combining RTX and FEC/RED. stefan@webrtc.org 2013-09-06 13:40:11 +00:00
  • 5500d93fe5 Add temporal layer factory. andresp@webrtc.org 2013-09-06 11:26:15 +00:00
  • 016eec0983 Unbreak build by adding new mandatory ICE username param. fischman@webrtc.org 2013-09-05 23:11:55 +00:00
  • f1e807c0e5 Removing FrameForStorage mikhal@webrtc.org 2013-09-05 22:34:41 +00:00
  • c31d4d0324 AppRTCDemo(iOS): prefer ISAC as audio codec This makes audio flow well bidirectionally to an iPod Touch (5th gen). Also: - Update to new turnserver JSON style: - separate username field - multiple URLs for the same server (e.g. both UDP & TCP) - Added more explicit logging for ICE Connected since it's useful for debugging - Give focus to the input field on app launch since that's the only useful thing to have focus on, anyway. - Fix minor typos - Cleaned up trailing whitespace and hard tabs fischman@webrtc.org 2013-09-05 21:49:58 +00:00
  • aa3d1c8169 Make unittest log printouts opt-in with a --logs flag. andrew@webrtc.org 2013-09-05 21:16:29 +00:00
  • bebf3995ce Pre-multiply images for MouseCursorShape. alexeypa@chromium.org 2013-09-05 19:32:46 +00:00
  • 31b4a5ac82 Recognize armv7 target_arch for ios support in webrtc common.gyp fischman@webrtc.org 2013-09-05 16:46:36 +00:00
  • be588f9a58 Apprtc Demo: calling createOffer/Answer without failureCallback is deprecated in FF braveyao@webrtc.org 2013-09-05 16:44:55 +00:00
  • 9080518a39 Restore severity precondition to logging.h. andrew@webrtc.org 2013-09-05 16:40:43 +00:00
  • 95e51f509c Remove send and receive streams when destroyed. pbos@webrtc.org 2013-09-05 12:38:54 +00:00
  • 164c4f71ba Add clockdrift to RtpGenerator henrik.lundin@webrtc.org 2013-09-05 12:16:38 +00:00
  • 7e1bf318bf Allow unknown flags in test_main.cc. pbos@webrtc.org 2013-09-05 10:27:46 +00:00
  • 36439bf906 NetEq4: Small change to reduce allocs in AudioMultiVector henrik.lundin@webrtc.org 2013-09-05 06:02:56 +00:00
  • e2d4da6586 Enable EngineTest.ReceivesPliAndRecoversWithNack and fix memcheck suppression filter. mflodman@webrtc.org 2013-09-04 14:21:57 +00:00
  • be23b32727 Adding tsan suppression for BUG 2349. mflodman@webrtc.org 2013-09-04 13:36:52 +00:00
  • 77bf5c28c8 Clean capture timestamp code. andresp@webrtc.org 2013-09-04 11:35:43 +00:00
  • 06f1f74331 Disable EngineTest.ReceivesPliAndRecoversWithNack. mflodman@webrtc.org 2013-09-04 11:00:07 +00:00
  • b21e528c60 Protecting Bitrate to avoid data race found by tsan. mflodman@webrtc.org 2013-09-04 08:42:44 +00:00
  • 65abb6b1ed Revert 4671 "Enable SetInitialPlayoutDelay on Android." mflodman@webrtc.org 2013-09-04 07:47:39 +00:00
  • 310ac91d2a Enable SetInitialPlayoutDelay on Android. dwkang@webrtc.org 2013-09-04 01:19:12 +00:00
  • 3abb82d8df Suppress video engine test mikhal@webrtc.org 2013-09-03 22:19:31 +00:00
  • 3c5a9242fe Don't force cont' when enabling kWithErrors mikhal@webrtc.org 2013-09-03 20:45:36 +00:00
  • 635b2b88e4 Removing some TODO's from libyuv mikhal@webrtc.org 2013-09-03 20:06:18 +00:00
  • 2b810bf77b Removing non decodable count from session info: This value isn't used, and therfore can be removed. This is a step towards the refactor of the session info to use maps. mikhal@webrtc.org 2013-09-03 19:09:49 +00:00
  • ccf8b56670 AppRTCDemo(android): prefer ISAC for audio codec. fischman@webrtc.org 2013-09-03 19:02:11 +00:00
  • 8788167b9b PeerConnection Java: explicitly cast DataChannel* to jlong for Java. Otherwise on 32-bit ARM Android the nativeDataChannel param the Java ctor sees is a 64-bit value whose low 32 bits are the pointer, and whose high 32-bits are garbage. fischman@webrtc.org 2013-09-03 18:58:12 +00:00
  • c8c32638be Remove JpegEncoder suppression as jpeg is now removed. kjellander@webrtc.org 2013-09-03 18:38:52 +00:00
  • f5f5da0df1 Adding TSAN suppression for test posix udp transport. mflodman@webrtc.org 2013-09-03 14:15:21 +00:00
  • 3a6ff41e8f Document the source of test scenarios for Dummynet wrapper script. kjellander@webrtc.org 2013-09-03 13:01:31 +00:00
  • cac7325b84 Adding critsect for child_modules_ in ModuleRtpRtcpImpl::Process() to avoid race with ModuleRtpRtcp::RegisterChildModule. mflodman@webrtc.org 2013-09-03 12:11:12 +00:00
  • cb5118c14c Add FakeEncoder to VideoSendStream tests. pbos@webrtc.org 2013-09-03 09:10:37 +00:00
  • 8fb89533af Correcting two nits in InputAudioFile henrik.lundin@webrtc.org 2013-09-03 08:43:28 +00:00
  • 8d32066073 Changed method name. mflodman@webrtc.org 2013-09-02 12:45:30 +00:00
  • 814d5e9133 Renamed method. mflodman@webrtc.org 2013-09-02 12:45:18 +00:00
  • d51bcffc1e Function name change. mflodman@webrtc.org 2013-09-02 12:45:09 +00:00
  • dfbf52baac Fixing capture frame race in ViECapturer. mflodman@webrtc.org 2013-09-02 12:44:57 +00:00
  • 5aedb295d5 Add TSan and Dr Memory suppressions for Windows kjellander@webrtc.org 2013-09-02 11:50:46 +00:00
  • b3e905cd91 Disable all LS_VERBOSE logging in NetEq4 henrik.lundin@webrtc.org 2013-09-02 09:41:06 +00:00