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  1. 36690cd Fix inverted RTC_DCHECK in RtpVideoStreamReceiver::RtcpFeedbackBuffer by Elad Alon · 5 years ago
  2. ba96e2f In FrameEncodeMetadataWriter don't clear known bitrate on Reset. by Ilya Nikolaevskiy · 5 years ago
  3. 015ff80 Roll chromium_revision abb1a36732..fd17362e28 (665960:666098) by chromium-webrtc-autoroll · 5 years ago
  4. 06a9926 Roll chromium_revision f8c14c5353..abb1a36732 (665857:665960) by chromium-webrtc-autoroll · 5 years ago
  5. 835baf7 Add amithi@ as pc OWNERS by Steve Anton · 5 years ago
  6. e0f3704 Add cap to video jitter buffer size/latency in experiment branches only. by “Michael · 5 years ago
  7. 479a3c0 Add support for enabling and negotiating raw RTP packetization. by Mirta Dvornicic · 5 years ago
  8. 961407f Delete unused method RtpRtcp::GetRtpPacketLossStats by Niels Möller · 5 years ago
  9. 65853dd Roll chromium_revision 1070231d7d..f8c14c5353 (665750:665857) by chromium-webrtc-autoroll · 5 years ago
  10. 31f18e1 Android SurfaceTextureHelper: Avoid crashing if size hasn't been set by Magnus Jedvert · 5 years ago
  11. f4c7ab1 in test/scenario pass TaskQueueFactory explicitly by Danil Chapovalov · 5 years ago
  12. c8501f7 Fix bug in neteq_quality_test by Pablo Barrera González · 5 years ago
  13. 90bc1e1 Fix comment typo about degradation preference. by Åsa Persson · 5 years ago
  14. bd00271 Roll chromium_revision 584b49b1a7..1070231d7d (665633:665750) by chromium-webrtc-autoroll · 5 years ago
  15. 292ce4e Move datagram transport to JsepTransport by Anton Sukhanov · 5 years ago
  16. 9005e23 Roll chromium_revision a3e71ebfa3..584b49b1a7 (665525:665633) by chromium-webrtc-autoroll · 5 years ago
  17. 1716d39 Let SessionDescription take ownership of MediaDescription by Harald Alvestrand · 5 years ago
  18. 1fe119f Change the gating of surfacing candidates on ICE transport type change by Qingsi Wang · 5 years ago
  19. e86af2c Allowing buffering a LNTF (loss notification) feedback message in RTCPSender by Elad Alon · 5 years ago
  20. 4e34c18 Check input file extension is not wav by Pablo Barrera González · 5 years ago
  21. 102b728 Prevent howling in RunPlayoutAndRecordingInFullDuplex by Gustaf Ullberg · 5 years ago
  22. 15f2200 Roll chromium_revision aaa0f87a5c..a3e71ebfa3 (665423:665525) by chromium-webrtc-autoroll · 5 years ago
  23. d2a6686 Add RtpPacketInfo to hold information about a received RtpPacket. by Chen Xing · 5 years ago
  24. 1df841d Target SDK level 29 in AppRTCMobile. by Sami Kalliomäki · 5 years ago
  25. ef09c5b Buffer RTCP feedback messages in RtpVideoStreamReceiver by Elad Alon · 5 years ago
  26. 4cd1c6a Lockless SwapQueue by Gustaf Ullberg · 5 years ago
  27. 89bbf37 Allow neteq_quality_test to read a complete file by Pablo Barrera González · 5 years ago
  28. 7537838 Add fhernqvist to watchlist. by Fredrik Hernqvist · 5 years ago
  29. 62838fe Expose audio decoder factory in neteq_quality_test by Pablo Barrera González · 5 years ago
  30. 695cf6a Delete deprecated StartRtcEventLog override with PlatformFile by Niels Möller · 5 years ago
  31. f330183 A threading explanation by Harald Alvestrand · 5 years ago
  32. 8db36de Roll chromium_revision 97b44755d9..aaa0f87a5c (665322:665423) by chromium-webrtc-autoroll · 5 years ago
  33. 114e8bb Roll chromium_revision ebd9263281..97b44755d9 (665197:665322) by chromium-webrtc-autoroll · 5 years ago
  34. 36e3147 Surface the standardized ICE connection state to mobile clients. by Qingsi Wang · 5 years ago
  35. 2dbc627 Check H264 packetization mode when using IsSameCodec by Steve Anton · 5 years ago
  36. 2229cf7 Roll chromium_revision c1296cf1c0..ebd9263281 (665078:665197) by chromium-webrtc-autoroll · 5 years ago
  37. 85b8ce2 In media engine replace forward declaration with proper includes by Danil Chapovalov · 5 years ago
  38. d7e2fb3 mb: Implement 'quiet' flag in mb lookup by Oleh Prypin · 5 years ago
  39. cecf87f Reland "Change default secure SCTP protocol to UDP/DTLS/SCTP" by Guido Urdaneta · 5 years ago
  40. 4436887 Revert "Change default secure SCTP protocol to UDP/DTLS/SCTP" by Guido Urdaneta · 5 years ago
  41. a937c6e Remove Win32 ASan from mb config. by Mirko Bonadei · 5 years ago
  42. ad4a3c8 Roll chromium_revision 81e506385d..c1296cf1c0 (664522:665078) by chromium-webrtc-autoroll · 5 years ago
  43. e93d109 Add "Win asan 64-bit" in order to migrate away from the 2-bit version. by Mirko Bonadei · 5 years ago
  44. eb22227 Add OnDatgramLost and default value for receive_timestamp. by Bjorn A Mellem · 5 years ago
  45. d91969e Explicitly close PeerConnections when using ScopedFieldTrials by Steve Anton · 5 years ago
  46. 220f4be Remove some media/ --> pc/ test dependencies by Steve Anton · 5 years ago
  47. 57dc02a Add receive_timestamp to DatagramAcks. by Bjorn A Mellem · 5 years ago
  48. 0c1c1b4 Move ownership of ICE from DtlsTransport to JsepTransport. by Bjorn A Mellem · 5 years ago
  49. a913c12 Roll chromium_revision 2d1120f0c1..81e506385d (664417:664522) by chromium-webrtc-autoroll · 5 years ago
  50. 74bebc5 Add OnDatagramAcked interface by Anton Sukhanov · 5 years ago
  51. 72055b1 Roll chromium_revision 8891f34d24..2d1120f0c1 (664289:664417) by chromium-webrtc-autoroll · 5 years ago
  52. 740cc35 Roll chromium_revision d4906ebd49..8891f34d24 (664184:664289) by chromium-webrtc-autoroll · 5 years ago
  53. 6806550 Fix build with recent linux kernel. by Emilio Cobos Álvarez · 5 years ago
  54. 85a9d91 Add ability to set min/start/max bitrate on peer's PC in PC quality tests by Artem Titov · 5 years ago
  55. 845c6aa Add support for early loss detection using transport feedback. by Erik Språng · 5 years ago
  56. b3b3e3f Add acked bandwidth estimator config for sample uncertainty in ALR. by Christoffer Rodbro · 5 years ago
  57. 7eb0a5e AudioDecoderOpus: Add support for 16 kHz output sample rate by Karl Wiberg · 5 years ago
  58. ed69d41 Remove deprecated RtcEventLog Create functions by Danil Chapovalov · 5 years ago
  59. 2f5554d Make KeyFrameRequestSender injectable in RtpVideoStreamReceiver by Niels Möller · 5 years ago
  60. e8e7d7b Move Connection into it's own .h/.cc file. by Jonas Oreland · 5 years ago
  61. 28f0eb2 Move H.264 SPS VUI rewriting to FrameEncodeMetadataWriter. by Mirta Dvornicic · 5 years ago
  62. a1d1a1e WebRTC Opus C interface: Add support for non-48 kHz decode sample rate by Karl Wiberg · 5 years ago
  63. 232b6a1 Propagate screenshare info into video track and it's source. by Artem Titov · 5 years ago
  64. 98266a4 Roll chromium_revision 99181c0bec..d4906ebd49 (664078:664184) by chromium-webrtc-autoroll · 5 years ago
  65. 6737841 Add jitterBufferDelay and jitterBufferEmittedCount stats for video by Guido Urdaneta · 5 years ago
  66. e4470cd Roll chromium_revision 9b60f86c15..99181c0bec (663961:664078) by chromium-webrtc-autoroll · 5 years ago
  67. 686be20 Fix ICE connection in datagram_transport. by Anton Sukhanov · 5 years ago
  68. 44bd71c Create a composite implementation of RtpTransportInternal. by Bjorn A Mellem · 5 years ago
  69. 64e97cf Roll chromium_revision 09fae7ef1b..9b60f86c15 (663849:663961) by chromium-webrtc-autoroll · 5 years ago
  70. f94e3d9 Roll chromium_revision 9809faf8ca..09fae7ef1b (663719:663849) by chromium-webrtc-autoroll · 5 years ago
  71. ce33b6a Implement QualityLimitationReasonTracker and expose "reason". by Henrik Boström · 5 years ago
  72. 07fc398 Roll chromium_revision 13f6824c51..9809faf8ca (663612:663719) by chromium-webrtc-autoroll · 5 years ago
  73. 787f4b2 Fix text logging of ALR detector experiment settings. by Bjorn Terelius · 5 years ago
  74. 0b97e17 Cleanup of CongestionWindowDownlinkDelay trial. by Sebastian Jansson · 5 years ago
  75. 9ab520e Reland "Avoid encrypting empty audio packet." by Minyue Li · 5 years ago
  76. 9a57350 Use ';' to escape '/' characters in path to dumped received video stream by Ilya Nikolaevskiy · 5 years ago
  77. 4ffed7c Add field trial for selecting potentially useful packets as padding. by Erik Språng · 5 years ago
  78. a33a860 Deprecate functions returning cricket::DataContentDescription. by Harald Alvestrand · 5 years ago
  79. f2e9cab Fix BWE simulation graph in event log visualization by Bjorn Terelius · 5 years ago
  80. ca2c430 Allow both LNTF to coexist with NACKs and key frame requests by Elad Alon · 5 years ago
  81. 3a072de Roll chromium_revision 60cc82f9b7..13f6824c51 (663509:663612) by chromium-webrtc-autoroll · 5 years ago
  82. 8b27910 Include downlink delay into congestion window size. by Ying Wang · 5 years ago
  83. 2370242 Enable flex fec support in PC quality test framework by Artem Titov · 5 years ago
  84. 36bc4f8 Add thread guards to cricket::P2PTransportChannel. by Harald Alvestrand · 5 years ago
  85. 2e8d78c Allow overriding subsets of probing field trials by Jonas Olsson · 5 years ago
  86. 6019d43 Removes using imports from flexfec_receiver. by Sebastian Jansson · 5 years ago
  87. 126f2b3 AudioEncoderOpus: Add support for 16 kHz input sample rate by Karl Wiberg · 5 years ago
  88. 883eefc Implement RTCRemoteInboundRtpStreamStats for both audio and video. by Henrik Boström · 5 years ago
  89. 6e436d1 [audio] Plumbing of ReportBlockData from RTCPReceiver to MediaSenderInfo by Henrik Boström · 5 years ago
  90. 87e3f9d [video] Plumbing of ReportBlockData from RTCPReceiver to MediaSenderInfo by Henrik Boström · 5 years ago
  91. e0eb325 AudioEncoderOpusImpl: Remove unused static methods by Karl Wiberg · 5 years ago
  92. 87da109 Make ReceiveStatisticsImpl::SetMaxReorderingThreshold apply per ssrc by Niels Möller · 5 years ago
  93. ad44b75 Roll chromium_revision e1ec78e27e..60cc82f9b7 (663034:663509) by chromium-webrtc-autoroll · 5 years ago
  94. 15baf5e Remove last mention of ortc from the codebase. by Mirko Bonadei · 5 years ago
  95. 3a1b927 Remove rtp_ and rtcp_packet_transport() from the RtpTransport interface. by Bjorn A Mellem · 5 years ago
  96. 8b096a0 LogToSderr by default in WebRTC tests by Anton Sukhanov · 5 years ago
  97. 34cd485 Delete the remaining ORTC interfaces. by Bjorn A Mellem · 5 years ago
  98. 039a714 VP9 screenshare: drop base layer separately by Ilya Nikolaevskiy · 5 years ago
  99. d9b4f33 Cleanup of AudioAllocationSettings flags. by Sebastian Jansson · 5 years ago
  100. 4c29546 Add test to cover bug in vp9 wrapper, triggered by field trial by Erik Språng · 5 years ago