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  • 615fabb661 Add looping sound button to AppRTCDemo Zeke Chin 2016-02-24 10:58:52 -08:00
  • f6ff9714c0 Fix division by zero in FindTMMBRBoundingSet danilchap 2016-02-24 09:23:37 -08:00
  • 07fb9be37f Move RTCP histograms from vie_channel to video channel stats proxies. sprang 2016-02-24 07:55:00 -08:00
  • f14c47a58c Remove ignored return code from modules. Peter Boström 2016-02-24 16:51:18 +01:00
  • 985177c757 Keep disabled RtpRtcp modules registered. Peter Boström 2016-02-24 16:12:33 +01:00
  • c379fcb248 Break out pacer thread from CongestionController to increase testability. Stefan Holmer 2016-02-24 16:02:55 +01:00
  • 23353ab465 Increase encoder-overuse thresholds for HW. Peter Boström 2016-02-24 15:19:55 +01:00
  • 3e60bf0ff3 Adds low complexity audio mode for single core CPUs henrika 2016-02-24 14:27:09 +01:00
  • c2b785df5d Replace scoped_ptr with unique_ptr in webrtc/common_audio/ kwiberg 2016-02-24 05:22:32 -08:00
  • 837b39e8f4 Fix ubsan warnings in BWE tests. Stefan Holmer 2016-02-24 14:03:01 +01:00
  • f01633e667 Replace scoped_ptr with unique_ptr in webrtc/modules/audio_device/ kwiberg 2016-02-24 05:00:36 -08:00
  • 58e08cbea8 Reset indexer upon initialization in AudioLoop. minyue 2016-02-24 03:49:19 -08:00
  • 2346c5a728 Add tools/mb to .gitignore Henrik Kjellander 2016-02-24 12:44:15 +01:00
  • 0665f0518f Fix OOB read in pacing test. stefan 2016-02-24 03:04:17 -08:00
  • 12f4cda086 Histograms for H264EncoderImpl/H264DecoderImpl initialization and errors. hbos 2016-02-24 03:03:05 -08:00
  • 0ab8e81e12 Move histograms for rtp receive counters to ReceiveStatisticsProxy sprang 2016-02-24 01:35:40 -08:00
  • b7261fd3ae iSAC float: Check for end of input buffer while decoding kwiberg 2016-02-24 01:34:29 -08:00
  • b01c7816a8 Added functional variants of Buffer::SetData and Buffer::AppendData. ossu 2016-02-24 01:05:56 -08:00
  • 17849fcc8b Reland of Add tools/mb to setup_links.py (patchset #1 id:1 of https://codereview.webrtc.org/1691723003/ ) kjellander 2016-02-24 00:04:44 -08:00
  • 3e8043b01e Roll chromium_revision 4d6ba6e..b97dbeb (376909:376966) kjellander 2016-02-23 23:39:09 -08:00
  • 1cfe94096c Added the agc digital_agc.c file to the ubsan blacklist in order to suppress UBSan errors from the AGC. peah 2016-02-23 23:06:50 -08:00
  • f75d008235 Bitrate controller for VideoToolbox encoder. tkchin 2016-02-23 22:49:42 -08:00
  • 0ed85b2ee3 Track pending ICE restarts independently for different media sections. deadbeef 2016-02-23 17:24:52 -08:00
  • 8df5d4f15b Moved the AEC C code to be built using C++. In order for the change to be reviewable, the move was made into two steps consisting of the first two patches in this CL. peah 2016-02-23 14:34:59 -08:00
  • e80f9d0218 Revert of Replace scoped_ptr with unique_ptr in webrtc/common_audio/ (patchset #4 id:60001 of https://codereview.webrtc.org/1712513002/ ) kjellander 2016-02-23 13:33:32 -08:00
  • 9788534c77 Removing some redundant ostringstreams declarations. Taylor Brandstetter 2016-02-23 12:58:15 -08:00
  • 71d9721bdc iOS: Fix JSON for tryserver configurations. kjellander@webrtc.org 2016-02-23 21:16:02 +01:00
  • fffa42b57e Replace scoped_ptr with unique_ptr in webrtc/audio/ kwiberg 2016-02-23 10:46:32 -08:00
  • f4d8441aed Disabled flaky tests philipel 2016-02-23 18:56:36 +01:00
  • 77f3e0da5a Screen was flickering when the picker for desktop medias showed up in Windows platform. Keeping track of window size for each window so that BitBlt() instead of PrintWindow() will be called for windows with unchanged sizes. gyzhou 2016-02-23 08:57:45 -08:00
  • b1eaa8df0d Only average positive quality stats. Peter Boström 2016-02-23 17:30:49 +01:00
  • b68e02fd86 Revert of CQ: Disable linux_baremetal pending installation fix. (patchset #1 id:1 of https://codereview.webrtc.org/1710363002/ ) kjellander 2016-02-23 08:28:06 -08:00
  • 80e12072cf Move congestion controller to a separate module. Stefan Holmer 2016-02-23 13:30:42 +01:00
  • ba3e25e502 Simple RTCP receiver fuzzer. Peter Boström 2016-02-23 11:35:30 +01:00
  • 79d7a499c0 Replace scoped_ptr with unique_ptr in webrtc/common_audio/ kwiberg 2016-02-23 01:26:44 -08:00
  • 0be9df4293 Roll chromium_revision aa04eb9..4d6ba6e (376768:376909) Henrik Kjellander 2016-02-23 07:18:43 +01:00
  • dc0e381eb5 Add more camera resolutions to camera scaling slider. Alex Glaznev 2016-02-22 16:48:29 -08:00
  • 18fcbcf48c Use VAD to get a better speech power estimation in the IntelligibilityEnhancer Alejandro Luebs 2016-02-22 15:57:38 -08:00
  • 67b81f92f4 Tune QP thresholds for HW H.264 encoder. Alex Glaznev 2016-02-22 15:05:15 -08:00
  • 18f9ddd7b3 Roll chromium_revision 14bbbf2..aa04eb9 (376710:376768) kjellander 2016-02-22 12:28:25 -08:00
  • a094fd1550 RTT intermediate calculation use ntp time instead of milliseconds. Compact NTP representation was designed exactly for that purpose: calculate RTT. No need to map to ms before doing arithmetic on this values. Because of this change there is no need to keep mapping between compact ntp presentation and milliseconds in the RTCPSender. Danil Chapovalov 2016-02-22 18:59:36 +01:00
  • 723ead844b Move simple RtpRtcp calls to VideoSendStream. Peter Boström 2016-02-22 15:14:01 +01:00
  • 2e67ae1222 Roll chromium_revision fbc4ecf..14bbbf2 (376680:376710) kjellander 2016-02-22 05:14:38 -08:00
  • eee7d9e8e8 iOS: Promote iOS simulator testing to main waterfall. kjellander@webrtc.org 2016-02-22 12:48:57 +01:00
  • 7ddc9deb4d Reduce the scope of rtc::Event::Wait() locking. Peter Boström 2016-02-22 11:31:49 +01:00
  • d1f718bb1e Changes in the wav_file implementation in order to avoid clang warnings. The changes does not change any of the functionality in the code. peah 2016-02-22 02:13:28 -08:00
  • 253d8fa82c Simplified the function for detecting whether capture data is modified. peah 2016-02-22 02:00:09 -08:00
  • ada8fe5c00 iOS: Don't run modules_unittests on iOS simulator kjellander@webrtc.org 2016-02-22 07:34:39 +01:00
  • 0077272a47 Roll chromium_revision 4101b15..fbc4ecf (376664:376680) kjellander 2016-02-21 20:09:01 -08:00
  • da1d656f9e Roll chromium_revision 789f25d..4101b15 (376663:376664) kjellander 2016-02-21 12:04:06 -08:00
  • b9f943d60d Roll chromium_revision 1120bd3..789f25d (376660:376663) kjellander 2016-02-21 04:04:06 -08:00
  • a18f638ab1 Include "sharedexclusivelock.cc" in Chromium GN build. jbauch 2016-02-21 01:56:16 -08:00
  • fa830dcc28 Roll chromium_revision 9dc1788..1120bd3 (376655:376660) kjellander 2016-02-20 20:12:38 -08:00
  • bf811752ad Roll chromium_revision 5618e25..9dc1788 (376642:376655) kjellander 2016-02-20 12:04:52 -08:00
  • 330d3d82a0 Roll chromium_revision fa5d546..5618e25 (376142:376642) kjellander 2016-02-20 04:12:32 -08:00
  • b9dd7c5b3a Remove GetTransport() from TransportChannelImpl mikescarlett 2016-02-19 20:43:45 -08:00
  • 9bf5cde91a Update build_ios_libs.sh script to build new Objective-C API and gather header files. hjon 2016-02-19 17:15:49 -08:00
  • 91fe304b0f vp9: Adjust parameter for a test in videoprocessor_integrationtest.cc Marco 2016-02-19 15:31:18 -08:00
  • a9d0892946 Add initial bitrate and frame resolution parameters to quality scaler. Alex Glaznev 2016-02-19 15:24:06 -08:00
  • 0013dcc0c1 Simplify SSRC usage inside ViEEncoder. Peter Boström 2016-02-19 20:42:19 +01:00
  • 7254890b28 Nuke SetSenderBufferingMode. Peter Boström 2016-02-19 19:58:34 +01:00
  • da9ae0c23f Revert of CQ: Change Android trybots to not run device tests. (patchset #1 id:1 of https://codereview.webrtc.org/1715643002/ ) kjellander 2016-02-19 10:44:22 -08:00
  • e2d83d6560 Use CallStats for RTT in Call, rather than VideoSendStream::GetRtt() sprang 2016-02-19 09:03:26 -08:00
  • 45c44f0b94 Simplify EncoderStateFeedback. Peter Boström 2016-02-19 17:36:01 +01:00
  • 9674d7cb89 Revert of Prevent data race in MessageQueue. (patchset #3 id:40001 of https://codereview.webrtc.org/1675923002/ ) jbauch 2016-02-19 07:16:16 -08:00
  • fc968a283c Fix sequence-number replay race for padding. Peter Boström 2016-02-19 16:14:37 +01:00
  • 88788adcfd Replace scoped_ptr with unique_ptr in webrtc/modules/audio_processing/ kwiberg 2016-02-19 07:04:49 -08:00
  • df88460372 Prevent data race in MessageQueue. jbauch 2016-02-19 07:03:31 -08:00
  • 1e80ce438e webrtc::RtpPacket name freed for better RtpPacket Danil Chapovalov 2016-02-19 16:02:15 +01:00
  • c51d6947e4 CQ: Disable linux_baremetal pending installation fix. kjellander@webrtc.org 2016-02-19 15:26:30 +01:00
  • 728012e49f Changed the semantics of Buffer::Clear to not alter the capacity ossu 2016-02-19 02:38:32 -08:00
  • ecdeb4cb94 CQ: Change Android trybots to not run device tests. kjellander@webrtc.org 2016-02-19 10:14:40 +01:00
  • c4e3ead352 Blacklist "build/c++11" cpplint filter. jbauch 2016-02-19 00:25:55 -08:00
  • 4458d09ee4 Drop support for playing output through aplay in intelligibility_proc Alejandro Luebs 2016-02-18 19:16:08 -08:00
  • b3fb71c101 Add RTCAudioSession proxy class. Zeke Chin 2016-02-18 15:44:07 -08:00
  • 9ac4df1ba6 iOS: Enable modules_unittests and common_audio_unittests kjellander 2016-02-18 13:15:09 -08:00
  • 235aaa7468 Fix Linux 32-bit compilation after sysroot switch. Henrik Kjellander 2016-02-18 21:52:21 +01:00
  • 66a99283be Roll chromium_revision 1d144ca..fa5d546 (375480:376142) kjellander@webrtc.org 2016-02-18 20:30:14 +01:00
  • 0e2e50ca1c Always append the BYE packet type at the end aleungbroadsoft 2016-02-18 08:33:26 -08:00
  • 452df1cb44 Suppress UBSan errors in common_audio henrik.lundin 2016-02-18 07:41:15 -08:00
  • f45381e1e5 VideoCapturerAndroid: Report onFirstFrameAvailable() for textures as well Magnus Jedvert 2016-02-18 15:47:47 +01:00
  • 5199c74d25 AndroidVideoCapturer getSupportedFormats(): Change interface from JSON string to List/vector Magnus Jedvert 2016-02-18 13:09:54 +01:00
  • 347c0bb5b5 Android GLShader: Check return value of glCreateShader() magjed 2016-02-18 03:47:44 -08:00
  • 3ee73a59ad Make RemoteBitrateEstimator::GetStats() virtual. Stefan Holmer 2016-02-18 11:42:30 +01:00
  • fd22e6cf2d Change PeerConnectionFactory.setVideoHwAccelerationOptions to be able to replace Egl context. Per 2016-02-18 11:35:48 +01:00
  • 74db777d64 Revert of Remove GetTransport() from TransportChannelImpl (patchset #3 id:40001 of https://codereview.webrtc.org/1691673002/ ) guidou 2016-02-18 01:57:49 -08:00
  • 59c634b605 Re-add RemoteBitrateEstimator::GetStats. Stefan Holmer 2016-02-18 10:14:43 +01:00
  • 32348192cc Fix and simplify the power estimation in the IntelligibilityEnhancer Alejandro Luebs 2016-02-17 20:04:19 -08:00
  • ee18220ddd Remove GetTransport() from TransportChannelImpl mikescarlett 2016-02-17 15:20:18 -08:00
  • ee75c7a78f Compile rtc_base_objc for Mac. tkchin 2016-02-17 14:44:52 -08:00
  • e3c6c82717 When doing continual gathering, remove the local ports when a corresponding network is dropped. honghaiz 2016-02-17 13:00:28 -08:00
  • a08bb0d163 Disabled the test EndToEndTest RestartingSendStreamPreservesRtpState due to the test being flaky. peah 2016-02-17 10:27:52 -08:00
  • b7f89d6e66 Replace scoped_ptr with unique_ptr in webrtc/voice_engine/ kwiberg 2016-02-17 10:04:18 -08:00
  • dabf07f477 Replace scoped_ptr with unique_ptr in webrtc/modules/audio_processing/vad/ kwiberg 2016-02-17 07:59:48 -08:00
  • a293ef0618 Apply VideoOptions per stream. nisse 2016-02-17 07:24:50 -08:00
  • 789ba92e14 Simplify CongestionController. Stefan Holmer 2016-02-17 15:52:17 +01:00
  • bad7804b99 Remove unused VideoSendStream TransportAdapter. Peter Boström 2016-02-17 15:42:52 +01:00
  • 62eaacf5ee Replace scoped_ptr with unique_ptr in webrtc/modules/audio_processing/test/ kwiberg 2016-02-17 06:39:05 -08:00
  • 28c99bc44a iOS: Include legacy objc API in all.gyp + fix H264 libyuv dependency kjellander 2016-02-17 05:38:29 -08:00
  • 4b4dc86c61 Remove conference_mode flag from AudioOptions and VideoOptions. nisse 2016-02-17 05:25:36 -08:00