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  • 823c9b8e36 Add histograms stats for sent/received fraction loss for a stream: - "WebRTC.Video.SentPacketsLostInPercent" - "WebRTC.Video.ReceivedPacketsLostInPercent" asapersson@webrtc.org 2015-01-08 07:50:56 +00:00
  • d730b288c8 Remove WebRtcSpl_ScaleAndAddVectorsWithRoundNeon andrew@webrtc.org 2015-01-07 21:34:23 +00:00
  • 59062d5aef Rename SendAndReceiveH264SvcQqvga to VP8 instead. pbos@webrtc.org 2015-01-07 19:21:18 +00:00
  • 8af11042cb Avoid reading past end of string in GetLine. decurtis@webrtc.org 2015-01-07 19:15:51 +00:00
  • 3663fb08ff Reenable dlclose() for InternalUnloadDll on TSan. pbos@webrtc.org 2015-01-07 18:02:39 +00:00
  • bab79951ca Convert FileMediaEngineTest to use more expects. pbos@webrtc.org 2015-01-07 18:01:29 +00:00
  • 69472e711c Add a dummy implemenation of SChannelAdapter::SetMode that makes sure that StartSSL fails if the mode is set to DTLS. pthatcher@webrtc.org 2015-01-07 18:01:07 +00:00
  • c10eceab6e Always tag SRTP_PROTECTION_PROFILE and BIO_METHOD as const. henrike@webrtc.org 2015-01-07 17:59:28 +00:00
  • dfef02824c Ignore virtual box interfaces. pthatcher@webrtc.org 2015-01-07 17:20:52 +00:00
  • 25dd754fff Excluding a flaky test from DrMemory tina.legrand@webrtc.org 2015-01-07 14:25:55 +00:00
  • 7fbf278f3f Suppress memcheck error in video_engine_tests kjellander@webrtc.org 2015-01-07 11:04:57 +00:00
  • 1777880f54 Roll gtest-parallel. pbos@webrtc.org 2015-01-07 11:03:19 +00:00
  • 07c83a1385 Disable WebRtcVideoMediaChannelSimulcastTest.SimulcastSend tests on Win (take 2) kjellander@webrtc.org 2015-01-07 10:36:53 +00:00
  • 4e5115ae73 RTCPeerConnectionFactory: Explicitly create new worker and signaling threads. tkchin@webrtc.org 2015-01-07 06:35:18 +00:00
  • f6a9714760 Remove peer connection and signaling calls from UI thread. glaznev@webrtc.org 2015-01-06 22:24:09 +00:00
  • 2ec50f2b0f Memcheck suppression for uninitalized memory in WebRtcIsac_Decode kjellander@webrtc.org 2015-01-06 11:33:08 +00:00
  • d95435c17a Disable WebRtcVideoMediaChannelSimulcastTest.SimulcastSend tests on Win kjellander@webrtc.org 2015-01-06 11:01:35 +00:00
  • cbe7ca8796 Roll chromium_revision 8e72e1d..271c6cc (307131:309333) kjellander@webrtc.org 2015-01-06 07:24:27 +00:00
  • 3a63a3c35d iOS AppRTC: First unit test. tkchin@webrtc.org 2015-01-06 07:21:34 +00:00
  • 4796cb93dc Disable flaky RelayServerTest.TestExpiration on all platforms. andrew@webrtc.org 2015-01-05 23:56:19 +00:00
  • fb7a039e9d Use array geometry in Beamformer aluebs@webrtc.org 2015-01-05 21:58:58 +00:00
  • a37bf2c4fe Hack clock_unittest fix for parallel execution. andrew@webrtc.org 2015-01-05 19:08:58 +00:00
  • c37e72e890 Make setting identical RTP extensions a no-op. pbos@webrtc.org 2015-01-05 18:51:13 +00:00
  • e5a921a82d Use tmp files in file_utils_unittests aluebs@webrtc.org 2015-01-05 18:45:22 +00:00
  • 76bc981b2d Use a temp file in FileLockTest. pbos@webrtc.org 2015-01-05 17:56:33 +00:00
  • 433006a6c2 Fixed style issues from lint and got rid of unused fields. wzh@webrtc.org 2015-01-05 17:39:43 +00:00
  • c4ad157d8d Re-enable test: VideoProcessorIntegrationTest.ProcessNoLossChangeFrameRateFrameDropVP9. marpan@webrtc.org 2015-01-05 17:31:34 +00:00
  • 215bbbdcdd Fix for log typo in ViEExternalCodecImpl::RegisterExternalReceiveCodec. mflodman@webrtc.org 2015-01-05 14:56:02 +00:00
  • aeb0dd3079 Disable RelayServerTest.TestExpiration on Mac. kjellander@webrtc.org 2015-01-03 17:47:05 +00:00
  • 8390c2762e Add two unit tests for Android AppRTCDemo. glaznev@webrtc.org 2015-01-02 19:51:12 +00:00
  • 896888b7e4 Remove min bitrate from simulcast streams. pbos@webrtc.org 2015-01-02 15:40:56 +00:00
  • bac0012120 Extend delay estimation window in AEC to 500 ms on all platforms bjornv@webrtc.org 2015-01-02 09:23:49 +00:00
  • 9eacb8cc59 Make P2PTestConductor use VirtualSocketServer. pbos@webrtc.org 2015-01-02 09:03:19 +00:00
  • c62749fb47 Parallelize MediaRecorder unittests. pbos@webrtc.org 2015-01-02 09:01:20 +00:00
  • 3a70625caf audio_processing: Added back ATTRIBUTE_UNUSED lost in r7877 bjornv@webrtc.org 2015-01-01 22:04:12 +00:00
  • 27f5317560 Use the prod GAE server in AppRTCDemo for iOS. jiayl@webrtc.org 2014-12-31 00:26:20 +00:00
  • 5eb71eb4f4 Fix style issues from lint. jiayl@webrtc.org 2014-12-30 22:44:11 +00:00
  • 34ac956706 Do not use openmax_dl for MIPS64 platform. andrew@webrtc.org 2014-12-30 18:19:56 +00:00
  • b2bda67497 Removing old channel code from a few more places. glaznev@webrtc.org 2014-12-30 18:15:43 +00:00
  • a9b1ec0247 Support for DTLS in OpenSSLAdapter pthatcher@webrtc.org 2014-12-29 23:00:14 +00:00
  • c5fd66dcdf Accept incoming pings before remote answer is set to reduce connection latency. jiayl@webrtc.org 2014-12-29 19:23:37 +00:00
  • 84d84471f5 Minor fixes regarding accumulator usage on MIPS platforms. andrew@webrtc.org 2014-12-29 17:08:44 +00:00
  • b024da3122 Add support for audio device selection in AppRTCDemo. henrika@webrtc.org 2014-12-29 10:35:06 +00:00
  • 5ad4178137 Move the Jingle-specific network code into webrtc/libjingle. pthatcher@webrtc.org 2014-12-23 22:14:15 +00:00
  • 46d4d29a75 Add field trial for screenshare bitrates when using temporal layers. sprang@webrtc.org 2014-12-23 15:19:35 +00:00
  • 1be0a78f45 Removing giles@mozilla.com from WebRTC watchlist. mflodman@webrtc.org 2014-12-22 12:49:14 +00:00
  • 53cb74107f Make RelayServerTest use VirtualSocketServer. pbos@webrtc.org 2014-12-22 07:56:42 +00:00
  • 086c8d5a02 Use a temporary buffer to scale a screencast in OnFrameCaptured braveyao@webrtc.org 2014-12-22 05:46:42 +00:00
  • 4c0544ab07 Move Jingle-specific files from talk/session/media to webrtc/libjingle/session/media. This is part of an ongoing effort to remove Jingle-specific files from the WebRTC repository. pthatcher@webrtc.org 2014-12-19 22:29:55 +00:00
  • ed1a48b0cd Fix mac video capture leak. tkchin@webrtc.org 2014-12-19 20:51:02 +00:00
  • 7ce4a584aa Add initWithCoder to RTCEAGLVideoView. tkchin@webrtc.org 2014-12-19 20:47:35 +00:00
  • ae643ce280 Wire up Beamformer in AudioProcessing aluebs@webrtc.org 2014-12-19 19:57:34 +00:00
  • 8817256373 Fix the ramp-up-down-up test which was using ts-offset extension with the abs-send-time estimator. stefan@webrtc.org 2014-12-19 18:00:21 +00:00
  • 50f7db8a77 Remove unneccessary lock causing a potential deadlock. stefan@webrtc.org 2014-12-19 17:55:20 +00:00
  • a6f7ba6848 Add a AppRTCDemo setting to change the GAE server. jiayl@webrtc.org 2014-12-19 17:32:14 +00:00
  • 5570769210 Remove the last getters from VideoReceiveStream stats. pbos@webrtc.org 2014-12-19 15:45:03 +00:00
  • 742386a136 Enable payload-based padding by default and remove the API. stefan@webrtc.org 2014-12-19 15:33:17 +00:00
  • aa21f2765b Unify the two copies of move.h kwiberg@webrtc.org 2014-12-19 14:35:57 +00:00
  • d16e839c6d Rtp-Rtcp sender cleanup. pbos@webrtc.org 2014-12-19 13:49:55 +00:00
  • 556caffb36 GN: Fix build for Mac kjellander@webrtc.org 2014-12-19 13:28:37 +00:00
  • 11d8176cb3 Move updating nack bitrate inside UpdateNACKBitRate. stefan@webrtc.org 2014-12-19 09:52:24 +00:00
  • 5647877b2d Breakup Transports and TransportParsers and move TransportParsers into webrtc/libjingle. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository. pthatcher@webrtc.org 2014-12-19 03:32:59 +00:00
  • 0c39e91cc8 Merge beamformer aluebs@webrtc.org 2014-12-18 22:22:04 +00:00
  • 1090a6eccf Remove obsolete target_arch == armv7. andrew@webrtc.org 2014-12-18 21:36:18 +00:00
  • aacc23465b Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository. pthatcher@webrtc.org 2014-12-18 20:31:29 +00:00
  • 16a05dddb8 Clean up the Channel code in AppRTCDemo and use GAE prod server for new signaling mode. jiayl@webrtc.org 2014-12-18 20:12:03 +00:00
  • f5847d7746 Move session/tunnel to webrtc/libjingle. This is part of the ongoing effort to move Jingle-specific things out of WebRTC and into its own repository. I won't submit this until all other projects have moved off of compiling this as well. pthatcher@webrtc.org 2014-12-18 17:09:11 +00:00
  • cb79141eab Store the received report blocks map (mapped per remote ssrc) in a map per source ssrc. When using rtx, receiver reports with two report blocks are received. The report blocks have the same remote ssrc and therefore the first report block was overwritten by the second report block when stored in the ReportBlockInfoMap. asapersson@webrtc.org 2014-12-18 14:30:32 +00:00
  • ce4e9a3562 Refactor some receive-side stats. pbos@webrtc.org 2014-12-18 13:50:16 +00:00
  • 98c04b38a8 Get avg_delay_ms from DecoderTiming callback. pbos@webrtc.org 2014-12-18 13:12:52 +00:00
  • 9b79197c80 Suppress REMB in bitrate ctrl if it seems lika a short network glitch. sprang@webrtc.org 2014-12-18 11:53:59 +00:00
  • f832a6d090 Remove _t from function pointer typedefs. pbos@webrtc.org 2014-12-18 09:56:09 +00:00
  • eed7a22bbf Make an AudioEncoder subclass for iSAC redundant encoding henrik.lundin@webrtc.org 2014-12-18 09:52:36 +00:00
  • dd8f6f3d48 Rename rtpDumpPktHdr_t to RtpDumpPacketHeader. pbos@webrtc.org 2014-12-18 09:18:42 +00:00
  • a9cf079248 Rename external_hmac_ctx_t to ExternalHmacContext. pbos@webrtc.org 2014-12-18 09:12:21 +00:00
  • e468bc9e60 Rename _t struct types in audio_processing. pbos@webrtc.org 2014-12-18 09:11:33 +00:00
  • cab1291745 Fixing the memory leak in AudioEncoderCopyRedDeathTest.NullSpeechEncoder henrik.lundin@webrtc.org 2014-12-18 06:58:42 +00:00
  • 4fba293c87 Workaround for issue 3927 to allow localhost IP even if it doesn't match the local turn port guoweis@webrtc.org 2014-12-18 04:45:05 +00:00
  • 4cb3856a4d Revert "Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository." pthatcher@webrtc.org 2014-12-18 02:28:25 +00:00
  • 536f999e58 Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository. pthatcher@webrtc.org 2014-12-18 01:22:02 +00:00
  • c51fb9348d Fix an assert failure caused by race condition guoweis@webrtc.org 2014-12-18 00:30:55 +00:00
  • 0ab42bc3f6 Make safe_conversions suitable for rtc_base_approved. andrew@webrtc.org 2014-12-17 22:56:09 +00:00
  • bc03192560 Move jingle examples from talk/ into webrtc/libjingle. This is part of the effor to move Jingle out of WebRTC and into its own repository. pthatcher@webrtc.org 2014-12-17 22:15:11 +00:00
  • 0eb6eec5cb Move VirtualSocket into the .h file to allow unit tests more control over behavior. guoweis@webrtc.org 2014-12-17 22:03:33 +00:00
  • 6f10ae25ea Support block_size greater than chunk_size in Blocker aluebs@webrtc.org 2014-12-17 17:28:31 +00:00
  • eb544460e4 Rename _t struct types in audio_coding. pbos@webrtc.org 2014-12-17 15:23:29 +00:00
  • 209df9bf77 Change MockStatsObserver to grab values inside of OnComplete. This is done since StatsReportCopyable is going away and the list of supported properties of the mock class is known. StatsReports holds a list of pointers to objects that cannot be cached, so this is a simple way to grab the values when they're available. tommi@webrtc.org 2014-12-17 14:09:05 +00:00
  • e728ee03ba Remove or rename typedefs with _t prefixes. pbos@webrtc.org 2014-12-17 13:43:55 +00:00
  • 5263c58923 Add a little utility to capture cpu graphs. tommi@webrtc.org 2014-12-17 12:35:29 +00:00
  • 70f74f3f7b Add overshoot of target bitrate for screenshare with temporal layers. sprang@webrtc.org 2014-12-17 10:57:10 +00:00
  • 45a272ab22 Change aggregated fraction loss to be calculated from the cumulative loss and extended sequence number diff between the current and the last report block of two get stats calls. asapersson@webrtc.org 2014-12-17 10:27:57 +00:00
  • e102e8147b Enable the iSACfix AudioDecoder test (and make it work again) kwiberg@webrtc.org 2014-12-17 07:30:23 +00:00
  • 38881be912 If one of the bundled content is missing in SDP, return false to MaybeEnalbeMuxingSupport(). Verified in chromium. Now the existing content still could work. braveyao@webrtc.org 2014-12-17 05:59:41 +00:00
  • 950c518251 Add adapter_type into Candidate object. guoweis@webrtc.org 2014-12-16 23:01:31 +00:00
  • 971bf557e2 Fix path to mock_agc.h andrew@webrtc.org 2014-12-16 22:28:20 +00:00
  • f050791ba0 Revert "Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository." pthatcher@webrtc.org 2014-12-16 22:28:03 +00:00
  • 4afb59903c Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository. pthatcher@webrtc.org 2014-12-16 21:37:37 +00:00
  • e2b7585bc2 Move ViewRequest and MediaStreams to streamparams.h, and remove dependency on mediasessionclient.h and mediamessages.h. This is part of the effort to remove Jingle-specific code from WebRTC and into its own repository. pthatcher@webrtc.org 2014-12-16 21:09:08 +00:00
  • a32487f97b Disable AudioEncoderCopyRedDeathTest.NullSpeechEncoder henrik.lundin@webrtc.org 2014-12-16 21:04:55 +00:00
  • 02c21dbef1 Make one OWNERS files for all of webrtc/libjingle so we don't need approval from webrtc/OWNERS every time we want to add a directory. pthatcher@webrtc.org 2014-12-16 21:04:41 +00:00