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webrtc / src.git / 273f31b85c1c87616ee6b8f2a540f2070b341416 / webrtc
  1. 273f31b Fix for flaky RemoveOverheadFromBandwidth test. by michaelt · 8 years ago
  2. 87d11cd Reland of Avoid calling PostTask in audio callbacks (patchset #1 id:1 of https://codereview.webrtc.org/2684913003/ ) by henrika · 8 years ago
  3. 5d83780 Fix flaky test introduced by r16478 by stefan · 8 years ago
  4. 0e3213a Fix bug in BitrateProber where an old probe added at a high bitrate will stay active indefinitely if the bandwidth estimate becomes too low to probe at that bitrate. by Stefan Holmer · 8 years ago
  5. 488c5dc Add new target direct_transport and remove fake_network and direct_transport from test_common. by perkj · 8 years ago
  6. e525d6a Revert Make the new jitter buffer the default jitter buffer. by stefan · 8 years ago
  7. 498ee8e Remove repeat flag from SendRTCP by danilchap · 8 years ago
  8. fd8f102 Revert of Avoid calling PostTask in audio callbacks (patchset #6 id:100001 of https://codereview.webrtc.org/2663383004/ ) by henrika · 8 years ago
  9. 2192089 Adding full initial version of delay estimation functionality in echo by peah · 8 years ago
  10. d4ed7f5 New tool for printing basic packet information from an RTC event log to stdout. by terelius · 8 years ago
  11. abcef5d Replace std::tr1::tuple by ::testing::tuple. by ehmaldonado · 8 years ago
  12. b10f32f Adding more comments to every header file in api/ subdirectory. by deadbeef · 8 years ago
  13. 54b6e98 Added gn target for rtc_event_log2rtp_dump. by ivoc · 8 years ago
  14. 7798501 Fix the Chrome crash caused by RtcEventLog by zhihuang · 8 years ago
  15. 9dd77ba Clarifying error messages in ParseIceServerUrl for invalid transport parameters. by zstein · 8 years ago
  16. 69fb2cc Revert of Add QP sum stats for received streams. (patchset #10 id:180001 of https://codereview.webrtc.org/2649133005/ ) by skvlad · 8 years ago
  17. ed02c6d Revert of RTCInboundRTPStreamStats.qpSum collected. (patchset #4 id:80001 of https://codereview.webrtc.org/2675943002/ ) by skvlad · 8 years ago
  18. 76bc8e8 Delete VideoReceiveStream::Config::pre_render_callback. by nisse · 8 years ago
  19. cd195be RTCInboundRTPStreamStats.qpSum collected. by hbos · 8 years ago
  20. c16fa5e Replace all use of the VERIFY macro. by nisse · 8 years ago
  21. ff0e72f Add QP sum stats for received streams. by sakal · 8 years ago
  22. 7de8d64 Wire up audio packet loss to BWE. by stefan · 8 years ago
  23. 2bc6864 Reland of Drop frames until specified bitrate is achieved. (patchset #1 id:1 of https://codereview.webrtc.org/2666303002/ ) by kthelgason · 8 years ago
  24. 338f78a RTCIceCandidatePairStats.available[Outgoing/Incoming]Bitrate collected. by hbos · 8 years ago
  25. 3443bb7 RTCRTPStreamStats.ssrc changed type to uint32_t. by hbos · 8 years ago
  26. 87b8e9f Add missing dependency to audio_decoder_unittests. by ehmaldonado · 8 years ago
  27. 585a9b1 Refactor and clean-up relating to RTCCodecStats. by hbos · 8 years ago
  28. b99b596 Add chromium-junit4 tag to instrumentation test AndroidManifests. by sakal · 8 years ago
  29. e0ac5a6 Use std::unique_ptr in VideoProcessorIntegrationTest. by asapersson · 8 years ago
  30. 1b21b9b Replace occurences of string by std::string. by ehmaldonado · 8 years ago
  31. 1634e16 Remove use of selectors matching Apple private API names. by kthelgason · 8 years ago
  32. 4a9a595 Make rtcp packets copyable by danilchap · 8 years ago
  33. 1959b63 Remove Assert lint suppression. by sakal · 8 years ago
  34. 4709e89 Move RemoteBitrateEstimator::RemoveStream calls from receive streams to Call. by nisse · 8 years ago
  35. 6b34124 Remove unnecessary RTPHeaderParser, following https://codereview.webrtc.org/2659563002/ by solenberg · 8 years ago
  36. f748ca4 Change order of tear down/create of default audio stream, to avoid starting/stopping audio card playout unnecessarily. by solenberg · 8 years ago
  37. bd9a77f Remove most of the remaining calls to VoE APIs from Audio[Send|Receive]Stream. by solenberg · 8 years ago
  38. f9b6e5e Fix KeepsHighBitrateWhenReconfiguringSender to avoid flakiness if probing succeeds in between encoder reconfigurations. by Stefan Holmer · 8 years ago
  39. 1134b7b Reland of Improve and re-enable FEC end-to-end tests. (patchset #1 id:1 of https://codereview.webrtc.org/2672373002/ ) by brandtr · 8 years ago
  40. b77c716 Enable send-side BWE by default for video in call tests. by stefan · 8 years ago
  41. fd8d265 Revert of Improve and re-enable FEC end-to-end tests. (patchset #3 id:40001 of https://codereview.webrtc.org/2675573004/ ) by brandtr · 8 years ago
  42. d40b0f3 Improve and re-enable FEC end-to-end tests. by brandtr · 8 years ago
  43. cb789bb Remove NewApi lint suppression. by sakal · 8 years ago
  44. 93e1e23 Use RateAccCounter for sent bitrate stats. Reports average of periodically computed stats over a call. by asapersson · 8 years ago
  45. 447dba9 Add debuggable=true to AppRTCMobile manifest. by henrika · 8 years ago
  46. b114e9c Camera2Session: Add return statements after reportError where needed. by sakal · 8 years ago
  47. 61202ac Ensure that AEC3 is not run in tandem with AEC2 by peah · 8 years ago
  48. 237e1bb Fix potential use after free in H264 packetizer. by sprang · 8 years ago
  49. 60f7c63 Remove temporary AddRtxInfo member function. by brandtr · 8 years ago
  50. d44ce05 Reland of Always call RemoteBitrateEstimator::IncomingPacket from Call. (patchset #1 id:1 of https://codereview.webrtc.org/2668973003/ ) by nisse · 8 years ago
  51. 656610f Move frame_generator_capture.{cc, h} and video_capturer.h to video_test_common. by ehmaldonado · 8 years ago
  52. a7111eb Fixing an error in ANA FrameLengthController unittest. by minyue · 8 years ago
  53. e702b30 Adding C++ versions of currently spec'd "RtpParameters" structs. by deadbeef · 8 years ago
  54. d1f5fda Allow changing the minimal ICE ping timeout with PeerConnection.SetConfiguration. by skvlad · 8 years ago
  55. 98c4374 Allow passing network config constraint as base64 encoded string to preserve values of serialized protos. The values are a serialized byte stream packed into a std::string. To be represented as a NSString they must be base64 encoded or bytes outside of the ASCII range will be encoded into multi byte UTF8 sequences by default. by haysc · 8 years ago
  56. 390e64d Add VP9 full stack tests: - ConferenceMotionHd2000kbps100msLimitedQueueVP9 by jianj · 8 years ago
  57. 53b6cc3 Reland of Enable audio streams to send padding. (patchset #4 id:60001 of https://codereview.webrtc.org/2652893004/ ) by stefan · 8 years ago
  58. b11fb25 Protect APM in webkit builds. by agouaillard · 8 years ago
  59. 9d58d94 Fix and improve FlexFEC configuration for RTP/RTCP. by brandtr · 8 years ago
  60. 4cb1b75 Extends timer from 10 to 30 seconds for output volume check on Android. by henrika · 8 years ago
  61. 77ce9a5 Avoid calling PostTask in audio callbacks. by henrika · 8 years ago
  62. 5f47126 Added VP8 simulcast tests. Fixed analyzer to correctly infer timestamps. by ilnik · 8 years ago
  63. 4b512d7 Fix Chromium FYI bot by skvlad · 8 years ago
  64. d030912 Pick the DTLS handshake timeout based on the ICE RTT estimate by skvlad · 8 years ago
  65. a24a9e2 Get rid of unqualified std:: types. by deadbeef · 8 years ago
  66. 6741516 Implement new PeerConnection certificate policy API in ObjC API by hnsl · 8 years ago
  67. a5d94ff Objective-C API to set the ICE check rate through RTCConfiguration. by skvlad · 8 years ago
  68. b55bd5f Don't capture variables explicitly in lambda expression. by ehmaldonado · 8 years ago
  69. 5107246 Allow applications to limit the ICE check rate through RTCConfiguration by skvlad · 8 years ago
  70. e5bd702 Reland of Make the new jitter buffer the default jitter buffer. (patchset #2 id:260001 of https://codereview.chromium.org/2656983002/ ) by philipel · 8 years ago
  71. 8c61924 video_coding::PacketBuffer now group all H264 packets with the same timestamp into the same frame. by philipel · 8 years ago
  72. 1dffc62 Remove all occurrences of "using std::string". by ehmaldonado · 8 years ago
  73. e372d3c Add event log visualization of rtp timestamps over time. by stefan · 8 years ago
  74. a55f021 Add 120ms frame ability to ANA by michaelt · 8 years ago
  75. ed01647 Remove bad DCHECK added as part of https://codereview.webrtc.org/2452163004/ by solenberg · 8 years ago
  76. b33eed2 Fix perf issue when timinig out receiver infos in RTCP. by stefan · 8 years ago
  77. cc99bc2 Change StunMessage::AddAttribute return type from bool to void. by nisse · 8 years ago
  78. f7826d6 Remove InlinedApi lint ignore. by sakal · 8 years ago
  79. a29d5ec Make 'webrtc' target a complete static library on Linux, Android and Windows by kjellander · 8 years ago
  80. 24af663 Adding Java wrapper for DtmfSender. by deadbeef · 8 years ago
  81. 20cb0c1 Move DTMF sender to RtpSender (as opposed to WebRtcSession). by deadbeef · 8 years ago
  82. 2e03c66 Adding build switch for Opus that supports 120ms ptime. by minyue · 8 years ago
  83. d3d3ba5 Revert of Enable audio streams to send padding. (patchset #4 id:60001 of https://codereview.webrtc.org/2652893004/ ) by deadbeef · 8 years ago
  84. e35f89a Enable audio streams to send padding. by stefan · 8 years ago
  85. b1ca073 Rename adaptation api methods, extended vie_encoder unit test. by sprang · 8 years ago
  86. d83b967 Replace consecutive-losses count by a calculation of first-order-FEC recoverability. by elad.alon · 8 years ago
  87. 14245cc Revert of Always call RemoteBitrateEstimator::IncomingPacket from Call. (patchset #9 id:160001 of https://codereview.webrtc.org/2659563002/ ) by nisse · 8 years ago
  88. 77f0580 Add new step graph type to event log visualization tool. Currently used for bitrate estimate and accumulated packet count, but could in general be used for any metric that is piecewise constant. by terelius · 8 years ago
  89. 099110c Don't send audio packets if the network is down. by stefan · 8 years ago
  90. 4637b6a Consistent 30% improvement in audio mixer running time. by aleloi · 8 years ago
  91. 35fc2aa Revert of Drop frames until specified bitrate is achieved. (patchset #12 id:240001 of https://codereview.webrtc.org/2630333002/ ) by minyue · 8 years ago
  92. 6d4dd59 Always call RemoteBitrateEstimator::IncomingPacket from Call. by nisse · 8 years ago
  93. 803dc29 Enable cpplint and fix cpplint errors in webrtc/api by oprypin · 8 years ago
  94. 83399ca Drop frames until specified bitrate is achieved. by kthelgason · 8 years ago
  95. dc20e26 Use correct calling convention for CreateThread callback on Windows. by deadbeef · 8 years ago
  96. ac61b74 Refactor FakeAudioDevice to have separate methods for starting recording and playout. by perkj · 8 years ago
  97. 1c05625 Fix race condition in FrameBuffer2 by philipel · 8 years ago
  98. 54340d8 Change opus min bitrate. by michaelt · 8 years ago
  99. 7f08e82 Fix per regression in probing. by stefan · 8 years ago
  100. 6fb4f56 Reland of move usage of deprecated g_type_init API (patchset #1 id:1 of https://codereview.webrtc.org/2666103002/ ) by oprypin · 8 years ago