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webrtc / src / webrtc / ba5e7e291afb87d7eccf524fa26f0ed81f8a34cd
  1. ba5e7e2 Delete unused method PayloadRouter::MaxPayloadLength. by nisse · 8 years ago
  2. dd25b7f Style cleanup in RTCPReceiver by danilchap · 8 years ago
  3. be87638 Revert of Add disabled certificate check support to IceServer PeerConnection API. (patchset #8 id:140001 of https://codereview.webrtc.org/2557803002/ ) by magjed · 8 years ago
  4. e1a9899 Add a unit test for Opus complexity adaptation by henrik.lundin · 8 years ago
  5. d77888a Add an abstract class for IceTransport by zhihuang · 8 years ago
  6. cd20de1 Add GUARDED_BY's in FlexfecReceiver. by brandtr · 8 years ago
  7. 1358b5a Clean up storage of FlexFEC payload type in webrtc::VideoCodecSettings. by brandtr · 8 years ago
  8. 994a137 Revert of Rename RTPVideoHeader.isFirstPacket to .is_first_packet_in_frame. (patchset #1 id:1 of https://codereview.webrtc.org/2574943003/ ) by danilchap · 8 years ago
  9. 2e72d35 Removed undefined method from webrtcsession.h. by hbos · 8 years ago
  10. 75e309e Rename RTPVideoHeader.isFirstPacket to .is_first_packet_in_frame. by johan · 8 years ago
  11. 9858aec Move histogram for number of pause events to per stream: by asapersson · 8 years ago
  12. dc85c51 Reland of Properly report number of quality downscales in stats. (patchset #1 id:1 of https://codereview.webrtc.org/2586783003/ ) by kthelgason · 8 years ago
  13. 2e103c1 Re-enable Opus complexity tests on Android by henrik.lundin · 8 years ago
  14. 75e866a Revert of Properly report number of quality downscales in stats. (patchset #11 id:220001 of https://codereview.webrtc.org/2564373002/ ) by kthelgason · 8 years ago
  15. 1c0ef6d CodecInst operator<< by kwiberg · 8 years ago
  16. 3de4f88 Reland of Disabling NOTREACHED which we're hitting flakily in browser tests. (patchset #1 id:1 of https://codereview.webrtc.org/2585183002/ ) by asapersson · 8 years ago
  17. 81ed475 Add multithreaded fake encoder and corresponding FlexFEC VideoSendStreamTest. by brandtr · 8 years ago
  18. a3eddc5 Fix segfault when PeerConnection is destroyed during stats collection. by hbos · 8 years ago
  19. dacedae Properly report number of quality downscales in stats. by kthelgason · 8 years ago
  20. 19a13ce RTCStatsCollector: Utilize network thread to minimize thread hops. by hbos · 8 years ago
  21. 782e221 Now expect the correct number of streams in EndToEndTest.GetStats. by philipel · 8 years ago
  22. 300049d Add disabled certificate check support to IceServer PeerConnection API. by hnsl · 8 years ago
  23. 1591196 Add QP stats to the statsview in AppRTCMobile for ios. by denicija · 8 years ago
  24. 6d6e845 Revert of Re-enable Opus complexity tests on Android (patchset #1 id:1 of https://codereview.webrtc.org/2589673002/ ) by henrik.lundin · 8 years ago
  25. 1d9fb0b Re-enable Opus complexity tests on Android by henrik.lundin · 8 years ago
  26. 02a6431 Move FlexfecReceiveStream from api/call/ to call/. by brandtr · 8 years ago
  27. 289c7a2 Revert of Disabling NOTREACHED which we're hitting flakily in browser tests. (patchset #1 id:1 of https://codereview.webrtc.org/2477663002/ ) by asapersson · 8 years ago
  28. 993d3d3 Move tools/mb -> tools-webrtc/mb by Henrik Kjellander · 8 years ago
  29. c6eb673 Put iOS H264 High profile under a field trial by magjed · 8 years ago
  30. 1ed5da6 Move tools/valgrind-webrtc -> tools-webrtc/valgrind by kjellander · 8 years ago
  31. 8495a81 Revert of Delete unused code from systeminfo. (patchset #3 id:40001 of https://codereview.webrtc.org/2578323005/ ) by skvlad · 8 years ago
  32. 37d7b3f Add support for content hints to VideoTrack. by pbos · 8 years ago
  33. 09c258f Add full stack tests: by asapersson · 8 years ago
  34. fe71d23 Remove redundant local variable from OveruseDetector::Detect. by terelius · 8 years ago
  35. 14f3811 Remove device HW id -> marketing name mapping table for iOS devices. by kthelgason · 8 years ago
  36. 767c6f5 Delete unused code from systeminfo. by kthelgason · 8 years ago
  37. e4bd6b1 Fix for integer overflow in NetEq. by ivoc · 8 years ago
  38. 6fc1bd1 In RtpPacket do not keep pointer to RtpHeaderExtensionMap by danilchap · 8 years ago
  39. 5ea8737 Fix wrong log message. by kthelgason · 8 years ago
  40. fca6e17 Use NtpTime in RTCPSender::RtcpContext instead of pair of uint32_t by danilchap · 8 years ago
  41. 82d102d Pass arrival time as an int64_t rather than a double to the MedianSlopeEstimator to avoid precision loss. by terelius · 8 years ago
  42. afde8da Move ios_helpers to sdk folder by kthelgason · 8 years ago
  43. 4bd9eb0 Adding Åsa and Erik as video owners. by mflodman · 8 years ago
  44. 75a0edb Don't report packets with id -1 to the transport feedback adapter as they provide no value. by stefan · 8 years ago
  45. b90203c Initialize packetization mode in VideoToolbox by kthelgason · 8 years ago
  46. d7960fd iOS: Add trendline filter to field trials. by tkchin · 8 years ago
  47. 6419534 Fix integer overflow in ProbeController. by sergeyu · 8 years ago
  48. 0ab767a Avoid precision loss in TrendlineEstimator by passing the arrival time as an int64_t instead of a double. by terelius · 8 years ago
  49. e10c50d Add ossu@ to OWNERS of audio/ and modules/audio_coding/ by ossu · 8 years ago
  50. 7fc9834 Fixing loopback video test by reconfiguring the encoder to correct size. by mflodman · 8 years ago
  51. ecee889 Create VideoReceiver with external VCMTiming object. by philipel · 8 years ago
  52. 536d5c9 Improves release of allocated audio resources on Android. by henrika · 8 years ago
  53. 076a2cf Revert of Avoid precision loss in TrendlineEstimator from int64_t -> double conversion (patchset #7 id:120001 of https://codereview.webrtc.org/2577463002/ ) by terelius · 8 years ago
  54. e86008a Revert of Avoid precision loss in MedianSlopeEstimator from int64_t -> double conversion (patchset #3 id:40001 of https://codereview.webrtc.org/2578543002/ ) by terelius · 8 years ago
  55. ecf9e25 Pass arrival time as an int64_t rather than a double to the MedianSlopeEstimator to avoid precision loss. by terelius · 8 years ago
  56. 0e41e0d Reland of Delete VideoFrame default constructor, and IsZeroSize method. (patchset #1 id:1 of https://codereview.webrtc.org/2574123002/ ) by nisse · 8 years ago
  57. c9999ff Wire-up audio BWE with overhead. by michaelt · 8 years ago
  58. 7383d05 Avoid precision loss in TrendlineEstimator by passing the arrival time as an int64_t instead of a double. by terelius · 8 years ago
  59. a75e2b0 Rename RTCOutboundRTPStreamStats *_rtt members to *_round_trip_time. by hbos · 8 years ago
  60. 4d3e5ef Revert of Delete rtc::linked_ptr. Only use, in statstypes.h, replaced bu std::unique_ptr. (patchset #1 id:1 of https://codereview.webrtc.org/2581663002/ ) by nisse · 8 years ago
  61. 36b4bcf RTCStatsIntegrationTest: TestMemberIsIDReference on all defined IDs. by hbos · 8 years ago
  62. 1c0dcc2 Reland of Delete rtc::linked_ptr. Only use, in statstypes.h, replaced bu std::unique_ptr. (patchset #1 id:1 of https://codereview.webrtc.org/2576673002/ ) by nisse · 8 years ago
  63. 6148462 MB: Enable memcheck for the linux_memcheck trybot. by Henrik Kjellander · 8 years ago
  64. 21fd4ab RTCOutboundRTPStreamStats.roundTripTime: Only report non-negative values. by hbos · 8 years ago
  65. 8dca466 Move all codec specific definitions from modules_include by hta · 8 years ago
  66. 4bb94db Add WriteIsolatedOutput() functions by zijiehe · 8 years ago
  67. 0d580e9 Guard against uninitialized packetization modes. by hta · 8 years ago
  68. fa849a2 Skip RTCMediaStreamTrackStats.echoReturnLoss[Enhancement] default value. by hbos · 8 years ago
  69. f427d57 Adds basic Bluetooth support to AppRTCMobile by henrika · 8 years ago
  70. 4a16f60 Run 'git cl format --full' on Base64. by johan · 8 years ago
  71. c5805c3 Remove deprecated RTPSender::SendPadData by danilchap · 8 years ago
  72. 28cdc2d Remove static cast from H264SpropParameterSets. by johan · 8 years ago
  73. 87e0263 Improvements to the reliability of the echo detector perf test. by ivoc · 8 years ago
  74. 3986fab Add vector<uint8_t> to Base64 decoded data types. by johan · 8 years ago
  75. 19fd630 Delete webrtc/transport.h. by aleloi · 8 years ago
  76. 604324f Corrected access of null pointer in audioproc_f: by peah · 8 years ago
  77. 9817ab9 Removes verification of audio parameters on Android by henrika · 8 years ago
  78. d8cd9cc Revert of Delete VideoFrame default constructor, and IsZeroSize method. (patchset #5 id:80001 of https://codereview.webrtc.org/2541863002/ ) by nisse · 8 years ago
  79. 2410df4 Fix for left shift of negative value in NetEq. by ivoc · 8 years ago
  80. ad6c450 Delete method Pathname::url and base/urlencode* by nisse · 8 years ago
  81. 0c0354e Disable flaky test VideoProcessorIntegrationTest.ProcessNoLossChangeFrameRateFrameDropVP9 by skvlad · 8 years ago
  82. d7e5b79 This CL adds the basic framework for AEC3 in the audio processing module. by peah · 8 years ago
  83. 10a6614 Delete unused class rtc::RegKey. by nisse · 8 years ago
  84. e7a1876 Delete stl_util.h. Unused since cl https://codereview.webrtc.org/2447103002 by nisse · 8 years ago
  85. 0170590 Revert of Delete rtc::linked_ptr. Only use, in statstypes.h, replaced bu std::unique_ptr. (patchset #1 id:1 of https://codereview.webrtc.org/2567143003/ ) by nisse · 8 years ago
  86. 0cf81d9 Delete rtc::linked_ptr. Only use, in statstypes.h, replaced with std::unique_ptr. by nisse · 8 years ago
  87. 3dc2796 Disable AudioDeviceTest.StartPlayoutOnTwoInstances on iOS by Henrik Kjellander · 8 years ago
  88. d6b07c2 Fixing possible crash due to RefCountedChannel assignment operator. by deadbeef · 8 years ago
  89. 9a2667f Fixing integer overflow when parsing bandwidth attribute. by deadbeef · 8 years ago
  90. 800a8f1 Support external audio mixer in webrtc 2. by gyzhou · 8 years ago
  91. 7e1f0d5 Removing "crypto_required" from MediaContentDescription. by deadbeef · 8 years ago
  92. e5aa121 ParseCandidate(): Refactor to fix memcheck false positive. by hnsl · 8 years ago
  93. f031dec Update common_audio/smoothing_filter. by minyue · 8 years ago
  94. 6dab241 Delete VideoFrame default constructor, and IsZeroSize method. by nisse · 8 years ago
  95. a0aa420 Disable flaky QualityScaler tests for now. by kthelgason · 8 years ago
  96. 43bf21d Refactor "secure bool" into explicit PROTO_TLS. by hnsl · 8 years ago
  97. 4fefda7 Add a gtk3 port of peerconnection_client on Linux by thomasanderson · 8 years ago
  98. a77cc7c Logging basic bad call detection by palmkvist · 8 years ago
  99. f522e03 Revert of New PeerConnectionInterface::GetStats: No bogus default implementation. (patchset #1 id:1 of https://codereview.webrtc.org/2566143002/ ) by hbos · 8 years ago
  100. f3b4b21 Rename RtpStreamReceiver::SetCodec() to ::AddCodec(). by johan · 8 years ago