1. 0ba2419 Merge to M71: AEC3: Simplify render buffering by Gustaf Ullberg · 7 years ago
  2. 745dd01 Merge to M71: AEC3: Turn off the specific suppressor mode for ... by Per Åhgren · 7 years ago
  3. f8051d0 Merge to M71: AEC3: Compensate comfort noise level for loss due to filter bank by Gustaf Ullberg · 7 years ago
  4. e07640a Merge to M71: AEC3: Computation of comfort noise gains from suppression gains by Gustaf Ullberg · 7 years ago
  5. 49169a4 Merge to M71: JitterEstimator: Add trial for upper bound. by Erik Språng · 7 years ago
  6. cfdc368 Merge to M71: AEC3: Decrease latency until the delay has been detected by Per Åhgren · 7 years ago
  7. d2362be Disable probe delay warning in release builds. by philipel · 7 years ago
  8. 4f2fd3d Merge to M71: AEC3: Allow limiting dominant nearend to the non-initial.. by Per Åhgren · 7 years ago
  9. 6a731ba Merge to M71: AEC3: Introduce partial adaptive filter resets at echo... by Per Åhgren · 7 years ago
  10. 3af7bbb Merge to M71: AEC3: Improve dominant nearend detection by Per Åhgren · 7 years ago
  11. b6f3b95 Merge to M71: AEC3: Changes to initial behavior and handling of ... by Per Åhgren · 7 years ago
  12. 939b527 Merge to M71: AEC3: Enabling by default the use of the stationarity... by Per Åhgren · 7 years ago
  13. ea1dd98 Adds support for "-" to a=ssrc msid lines. by Seth Hampson · 7 years ago
  14. b83fb90 Adding NetEq buffer full metric to UMA. by Minyue Li · 7 years ago
  15. cac376c Add field trials for configuring Opus encoder packet loss rate. by Jakob Ivarsson · 7 years ago
  16. f23e5cf Create WebRTC 71 branch for Chrome 71 by Patrik Höglund · 7 years ago
  17. fb226af Remove some old logging in goog_cc for congestion window. by Ying Wang · 7 years ago
  18. a1d9ca4 Revert "Add ability to specify if rate controller of video encoder is trusted." by Oleh Prypin · 7 years ago
  19. cdc959f Compute video freeze metrics on rendered frames instead of on decoded by Ilya Nikolaevskiy · 7 years ago
  20. 3bdbc84 Moves pushback controller to GoogCC by Sebastian Jansson · 7 years ago
  21. f81170b Add error logs to RtpPacketHistory::GetBestFittingPacket when no packet is found. by Per Kjellander · 7 years ago
  22. ade98c9 Adds srte to WATCHLISTS. by Sebastian Jansson · 7 years ago
  23. 2b15626 Revert "Use unique_ptr and ArrayView in SSLFingerprint" by Henrik Grunell · 7 years ago
  24. 703259c Don't CHECK when parsing AEC3 parameters from json by Sam Zackrisson · 7 years ago
  25. 80bf775 Roll chromium_revision 2499289737..f34485ffde (598606:598711) by chromium-webrtc-autoroll · 7 years ago
  26. f7fcaf0 Use zero octets for rtp packet padding by Danil Chapovalov · 7 years ago
  27. 3d25530 Reland "Export symbols needed by the Chromium component build (part 1)." by Mirko Bonadei · 7 years ago
  28. 3e335d1 Add ability to specify if rate controller of video encoder is trusted. by Erik Språng · 7 years ago
  29. 88be972 Delete post_encode_callback by Niels Möller · 7 years ago
  30. 74f6c7e AEC3: Cleanup test code for platforms with clock-drift by Per Åhgren · 7 years ago
  31. d6b0796 AEC3: Ensure that the usage of stationary signal properties is not unset by Per Åhgren · 7 years ago
  32. 23b2a25 Remove unlimited retransmission for screenshare experiment code by Ilya Nikolaevskiy · 7 years ago
  33. cc21e61 Use unique_ptr and ArrayView in SSLFingerprint by Steve Anton · 7 years ago
  34. e8d2b1b Roll chromium_revision 8afdf16764..2499289737 (598496:598606) by chromium-webrtc-autoroll · 7 years ago
  35. f7dd9df Change TurnPort::Create to return a unique_ptr by Steve Anton · 7 years ago
  36. 9cfce17 Roll chromium_revision 0d09089dd5..8afdf16764 (598349:598496) by chromium-webrtc-autoroll · 7 years ago
  37. 0854eb6 Respond to SDP request extmap-allow-mixed. by Johannes Kron · 7 years ago
  38. a8f1e56 Change Port::Create methods to return a unique_ptr by Steve Anton · 7 years ago
  39. 7940da0 Integration of media_transport in JSepTransportController by Anton Sukhanov · 7 years ago
  40. 6cc9cca Don't reset streams for the FrameEncryptor / FrameDecryptor unless they changed. by Benjamin Wright · 7 years ago
  41. da67c16 Roll chromium_revision 8a25f94ac2..0d09089dd5 (598237:598349) by chromium-webrtc-autoroll · 7 years ago
  42. ca27091 Remove rtc_base:rtc_base_approved_generic. by Mirko Bonadei · 7 years ago
  43. ede8796 Print per-frame VMAF score instead of average. by Paulina Hensman · 7 years ago
  44. b3b0179 Fix backwards logic in rtc::Buffer::OnMovedFrom() by Karl Wiberg · 7 years ago
  45. 0213786 Add certificate gen/set functionality to bring Android closer to JS API by Michael Iedema · 7 years ago
  46. dcc0238 Don't increment timestamp on drop/reencode in LibvpxVp8Encoder. by Erik Språng · 7 years ago
  47. 5526e45 vp9: change x-google-profile-id to profile-id by Philipp Hancke · 7 years ago
  48. 028248c Add `rtc_enable_symbol_export` to incrementally create a WebRTC component. by Mirko Bonadei · 7 years ago
  49. b686396 Makes AudioSendStream signal that it's part of allocation. by Sebastian Jansson · 7 years ago
  50. 99a70a2 Remove rtc_base_approved_objc and introduce rtc_base:logging_mac. by Mirko Bonadei · 7 years ago
  51. edc49c1 [Cleanup] Remove unused swap function. by Yves Gerey · 7 years ago
  52. a4c8514 Add JSON parsing and corresponding ToString to EchoCanceller3Config by Sam Zackrisson · 7 years ago
  53. 2558c4e Remove ortc folder. by Mirko Bonadei · 7 years ago
  54. 88b68ac Create field trial for setting a minimum value for Opus encoder packet loss rate by Jakob Ivarsson · 7 years ago
  55. f08dd9d Disable flaky tests on mac perf bot by Ilya Nikolaevskiy · 7 years ago
  56. 1bca65b Makes RtpSender indicate allocation and feedback status on packets. by Sebastian Jansson · 7 years ago
  57. 81125f0 Implement (mostly) standards-compliant RTCIceTransportState. by Jonas Olsson · 7 years ago
  58. 5f35e96 Roll chromium_revision 476ae6d661..8a25f94ac2 (598136:598237) by chromium-webrtc-autoroll · 7 years ago
  59. c87b8c1 Moves GoogCC factory to API. by Sebastian Jansson · 7 years ago
  60. 0d8c100 AEC3: Decrease the suppression during the echo-only case by Per Åhgren · 7 years ago
  61. 463c764 Roll chromium_revision cfe6e706d0..476ae6d661 (598018:598136) by chromium-webrtc-autoroll · 7 years ago
  62. aabf204 Remove container typedefs from RelayServer by Steve Anton · 7 years ago
  63. 11358fe Use unique_ptr in port_unittest by Steve Anton · 7 years ago
  64. 13d392d AEC3: Utilize dominant nearend functionality to increase transparency by Per Åhgren · 7 years ago
  65. 3a3f027 Roll chromium_revision 0cf8926390..cfe6e706d0 (597915:598018) by chromium-webrtc-autoroll · 7 years ago
  66. 0378997 Adds flags indicating presence in allocation and feedback per packet. by Sebastian Jansson · 7 years ago
  67. 30e2d6e Moves locking outside function in RtpSender. by Sebastian Jansson · 7 years ago
  68. 789f459 Adds fields for unacknowledged data to transport feedback. by Sebastian Jansson · 7 years ago
  69. 20a49f3 Don't try to use CN if voice codec isn't mono by Karl Wiberg · 7 years ago
  70. 5fcc4de Roll chromium_revision f362b3e857..0cf8926390 (597811:597915) by chromium-webrtc-autoroll · 7 years ago
  71. 759f959 Refactor tests with ConfigurableFrameSizeEncoder by Niels Möller · 7 years ago
  72. 040f87f AEC3: Allow a more stable filter during double-talk by Gustaf Ullberg · 7 years ago
  73. 7730193 Remove SetExecutablePath, simplify ResourcePath by Patrik Höglund · 7 years ago
  74. 7004571 AEC3: Decrease the modelling of the reverb by Per Åhgren · 7 years ago
  75. d76a0fc Throttle the RTP decryption error messages in the SrtpSession and SrtpTransport by [email protected] · 7 years ago
  76. b674cd1 Enable multithreading in libvpx VP9 decoder. by Sergey Silkin · 7 years ago
  77. d0bc462 Check if __IPHONE_OS_VERSION_MAX_ALLOWED is defined before reference by Joel Sutherland · 7 years ago
  78. 0414040 Fix race condition for SupportsFlexfecWithMultithreadedH264/0 test. by Yves Gerey · 7 years ago
  79. bf47198 Roll chromium_revision ba2e073e2c..f362b3e857 (597606:597811) by chromium-webrtc-autoroll · 7 years ago
  80. 4ff7214 Using TaskQueue for congestion controller by default. by Sebastian Jansson · 7 years ago
  81. 4b14416 Roll chromium_revision 0cdd2e3eab..ba2e073e2c (597498:597606) by chromium-webrtc-autoroll · 7 years ago
  82. e0c2e97 Pass MediaTransportFactory to PeerConnectionFactory. by Piotr (Peter) Slatala · 7 years ago
  83. 1e05486 Added the new generic descriptor extension to WebRtcVideoEngine::GetCapabilities. by philipel · 7 years ago
  84. ab09039 Add comment that xcode version needs to be updated in two places by Oleh Prypin · 7 years ago
  85. 16fe3f2 Revert "Export symbols needed by the Chromium component build (part 1)." by Mirko Bonadei · 7 years ago
  86. 99eea42 Reland "Reland "Export symbols needed by the Chromium component build (part 1)."" by Mirko Bonadei · 7 years ago
  87. 2e068e8 Adds RTT based backoff trial to SendSideBandwidthEstimation. by Sebastian Jansson · 7 years ago
  88. d2fb1bf Generate module.modulemap file when building Mac Framework by Joel Sutherland · 7 years ago
  89. e6708f3 Notify a rotation about autoroll CLs by Oleh Prypin · 7 years ago
  90. 75e3647 Switch usages of DefaultNetworkSimulationConfig to BuiltInNetworkBehaviorConfig by Artem Titov · 7 years ago
  91. 3a74239 Fix compilation issues on media_transport_interface.h by Niels Möller · 7 years ago
  92. 788c51c Pass HeaderExtensionMap by reference in rtc_event_log2rtp_dump. by Bjorn Terelius · 7 years ago
  93. b6a8942 Fix race condition for GetContributingSources test. by Yves Gerey · 7 years ago
  94. 666fb32 Rename DefaultNetworkSimulationConfig into BuiltInNetworkBehaviorConfig. by Artem Titov · 7 years ago
  95. 6a8327f Roll chromium_revision ccb83d4a55..0cdd2e3eab (597330:597498) by chromium-webrtc-autoroll · 7 years ago
  96. 7c1744d Reland "Reland "Using units in SendSideBandwidthEstimation."" by Sebastian Jansson · 7 years ago
  97. 841c912 Changed FakeVp8Encoder to write dimensions in payload. by Per Kjellander · 7 years ago
  98. a4de9c8 Revert "Reland "Using units in SendSideBandwidthEstimation."" by Sebastian Jansson · 7 years ago
  99. e2cb26c Reland "Using units in SendSideBandwidthEstimation." by Sebastian Jansson · 7 years ago
  100. 917e596 Revert "Using units in SendSideBandwidthEstimation." by Oleh Prypin · 7 years ago