1. ef45669 Adds GetSentPacket to PacketResult. by Sebastian Jansson · 6 years ago
  2. 449c1c0 Adds unit tests for safe reset trial. by Sebastian Jansson · 6 years ago
  3. 7286496 Download aap2 and bundletool as part of required dependencies. by Yves Gerey · 6 years ago
  4. 6fcf6ca Modified PressEnterToContinue() to actualy check if Enter is pressed by Danail Kirov · 6 years ago
  5. 2c16cc6 Replace some usage of EventWrapper with rtc::Event. by Niels Möller · 6 years ago
  6. 88d8d7d Add missing assignment in RTCConfiguration.mm by Piotr (Peter) Slatala · 6 years ago
  7. f3ff14c Properly setup MockPeerConnectionObserver in tests. by Yves Gerey · 6 years ago
  8. 22a8f98 Formatted sslidenty.cc and moved non referenced functions into an by Benjamin Wright · 6 years ago
  9. 428320c Formatting OpenSSLCertificate and doing some minor code cleanup. by Benjamin Wright · 6 years ago
  10. 5d35554 Rename private member functions to use CamelCase. by Benjamin Wright · 6 years ago
  11. 61c5cc8 Makes OpenSSL concrete implementations final. by Benjamin Wright · 6 years ago
  12. 2616594 Refactor: Make SSLCertChain a final class. by Benjamin Wright · 6 years ago
  13. 150a907 FrameEncryption Video End To End Testcase. by Benjamin Wright · 6 years ago
  14. c462a6e Prevent the frame decryptor being set if the channel is stopped. by Benjamin Wright · 6 years ago
  15. 625771d Roll chromium_revision a539a24569..62e33bd2f0 (603045:603177) by chromium-webrtc-autoroll · 6 years ago
  16. 59ebf23 Refactor structs in rtc_event_log_parser_new.h by Elad Alon · 6 years ago
  17. ff43541 Delta compression efficiency improvement for non-existent base by Elad Alon · 6 years ago
  18. 436ebca Fix extra setdscp call introduced by bad merge. by Tim Haloun · 6 years ago
  19. 0f08d22 Add a function for enabling the congestion window and pushback controller in the webrtc::SendSideCongestionController. by erikvarga@webrtc.org · 6 years ago
  20. 99b71df Use function_video_(en|de)coder_factory from api by Danil Chapovalov · 6 years ago
  21. 88c2c50 Use monotonic clock to derive NTP timestamps in RTCP module by Ilya Nikolaevskiy · 6 years ago
  22. fdee701 Add parser and unittests for new RTC event log format. by Bjorn Terelius · 6 years ago
  23. 916ae08 Makes critsect_.Leave() more visible in PacedSender. by Sebastian Jansson · 6 years ago
  24. 6dd7f91 Remove deprecated deregistration functions in RtcpTransceiver by Danil Chapovalov · 6 years ago
  25. 06aa209 Add support to adapt video without preserving aspect ratio by Magnus Jedvert · 6 years ago
  26. 9049037 Simplify api/DEPS presubmit check. by Mirko Bonadei · 6 years ago
  27. 7d76a31 Use MediaTransportInterface, for audio streams. by Niels Möller · 6 years ago
  28. 2769cd5 Roll chromium_revision f54583b6a0..a539a24569 (602763:603045) by chromium-webrtc-autoroll · 6 years ago
  29. 0d24772 Allocate CMBlockBuffers using a memory pool. by Kári Tristan Helgason · 6 years ago
  30. c35096d Reland "Encode RTC event logs in new format." by Bjorn Terelius · 6 years ago
  31. a5d543c Set minSdkVersion to 16 for AppRTCMobile_stubbed_video_io_test_apk. by Mirko Bonadei · 6 years ago
  32. 95dfa52 Clarify the desired semantics of AsyncResolverInterface::GetResolvedAddress. by Zach Stein · 6 years ago
  33. 327b753 Split out a separate target for VP8EncoderSimulcastProxy by Jonathan Yu · 6 years ago
  34. 5abd541 Stop exporting simulcast_encoder_adapter.h in :rtc_internal_video_codecs by Jonathan Yu · 6 years ago
  35. b19b497 Refactor: Removing IgnoreBadCert from SSLStreamAdapter. Make test methods more explicit. by Benjamin Wright · 6 years ago
  36. dcd40ca Roll chromium_revision d68fb50e14..f54583b6a0 (602627:602763) by chromium-webrtc-autoroll · 6 years ago
  37. 8c27cca Promotoing webrtc::CryptoOptions to RTCConfiguration. by Benjamin Wright · 6 years ago
  38. 78410ad Fixes use after free error when setting a new FrameEncryptor on ChannelSend. by Benjamin Wright · 6 years ago
  39. f26e290 fuchsia: Stub out timing and memory functions by Scott Graham · 6 years ago
  40. 9c8ae4b Disable probe delay warning in release builds. by Jamie Walch · 6 years ago
  41. 6c6c9df Refactor: Renaming ssl_cert_chain to GetSSLCertificateChain() by Benjamin Wright · 6 years ago
  42. 359d60a Adds target rate to audio send stream stats. by Sebastian Jansson · 6 years ago
  43. 57ba7e1 Normalize baseline in network delay plot to RTT/2. by Bjorn Terelius · 6 years ago
  44. 039743e Reland "Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase" by Niels Möller · 6 years ago
  45. e2754c9 Fixes bug in AudioPriorityBitrateAllocationStrategy field trial. by Sebastian Jansson · 6 years ago
  46. c0e4d45 Adds BitrateAllocation struct to OnBitrateUpdated. by Sebastian Jansson · 6 years ago
  47. 4ba6c26 Delete MessageData when a message is posted to a quitting MessageQueue by Niels Möller · 6 years ago
  48. 9516c38 [Fuzzer] Check FieldTrial bitmask size at compile time. by Yves Gerey · 6 years ago
  49. 1803bb2 Fix for clock read race in FakeNetworkPipe. by Christoffer Rodbro · 6 years ago
  50. 3284b61 Fix for packet loss tracking in network emulation. by Christoffer Rodbro · 6 years ago
  51. 2620470 Update fuzzer max input length handling by Sam Zackrisson · 6 years ago
  52. ddc84e9 Publish function_video_(en|de)coder_factory into api by Danil Chapovalov · 6 years ago
  53. 23524ce Add HDR metadata struct by Johannes Kron · 6 years ago
  54. 977b46a Export symbols needed by the Chromium component build (part 7). by Mirko Bonadei · 6 years ago
  55. 3eb1c72 Removes deprecated BitrateAllocation alias. by Sebastian Jansson · 6 years ago
  56. 2506839 Add DCHECK for wrap around in RtpVideoSender::OnBitrateUpdated. by Bjorn Terelius · 6 years ago
  57. 370bae4 APM: Adding more explicit handling of failures in the json config data by Per Åhgren · 6 years ago
  58. 487e694 Use default value if field trial switch is set to an invalid number by Johannes Kron · 6 years ago
  59. 273c851 Remove obsolete android ndk copy from //third_party/android_tools/ndk by Yongje Lee · 6 years ago
  60. 7a95e0f APM: Add ability to turn on/off dumping of internal data by Per Åhgren · 6 years ago
  61. e2fd86a Move encoder metadata into EncoderInfo struct. by Erik Språng · 6 years ago
  62. 2d3a1fb Don't add webrtc-reviews@ to CC, it can be added globally on Gerrit by Oleh Prypin · 6 years ago
  63. 4191a81 Revert "Move relay server code to a test-only target p2p_server_utils." by Oleh Prypin · 6 years ago
  64. 8fb20ce Roll chromium_revision f57bd4785e..d68fb50e14 (602511:602627) by chromium-webrtc-autoroll · 6 years ago
  65. e284c52 Move relay server code to a test-only target p2p_server_utils. by Niels Möller · 6 years ago
  66. 01c68b8 Roll chromium_revision b9a687f112..f57bd4785e (602396:602511) by chromium-webrtc-autoroll · 6 years ago
  67. 09a49fa Roll chromium_revision 869181c2dc..b9a687f112 (602275:602396) by chromium-webrtc-autoroll · 6 years ago
  68. 4cb4786 Add expected default values to video configuration tests. by Benjamin Wright · 6 years ago
  69. 44a262a Declares BitrateAllocator methods const. by Sebastian Jansson · 6 years ago
  70. 583d6d9 Add missing directory to api/DEPS and PRESUBMIT.py. by Mirko Bonadei · 6 years ago
  71. 62ae178 Remove deprecated pipe field from VideoQualityTestFixtureInterface::Params by Artem Titov · 6 years ago
  72. 825f83b Revert "Encode RTC event logs in new format." by Mirko Bonadei · 6 years ago
  73. e943d43 Remove deprecated DefaultNetworkSimulationConfig by Artem Titov · 6 years ago
  74. a418e67 Use checkdeps to ensure API headers don't include internal headers. by Mirko Bonadei · 6 years ago
  75. ec9b77b Remove deprecated API: NetwrokSimulationInterface. by Artem Titov · 6 years ago
  76. 257ed43 Add support for optional fields in FixedLengthDeltaEncoder by Elad Alon · 6 years ago
  77. c6ec4b1 Fix w3c URL for RTCIceTransport by Niels Möller · 6 years ago
  78. 5e58bcb Forward audio rtp frequency to Rtcp sender and use it for SR packets by Ilya Nikolaevskiy · 6 years ago
  79. ece3c22 Encode RTC event logs in new format. by Bjorn Terelius · 6 years ago
  80. fb5c1ec AEC3: Included missing parsing of config parameter by Per Åhgren · 6 years ago
  81. 8e6749e Improve fileutils_override implementation internal API. by Artem Titov · 6 years ago
  82. e068ad6 Use a sufficiently large bitmask. by Jonas Olsson · 6 years ago
  83. 511fe0b Roll chromium_revision 5e5003737d..869181c2dc (602066:602275) by chromium-webrtc-autoroll · 6 years ago
  84. d38a2b8 Increase the UDP receive buffer for video by Johannes Kron · 6 years ago
  85. f0c449e APM: Correct includes required for the data dumping functionality by Per Åhgren · 6 years ago
  86. 700b4a4 AEC3: Allow limiting dominant nearend to the non-initial phase by Per Åhgren · 6 years ago
  87. 41ed3e0 Roll chromium_revision b1cb85713b..5e5003737d (601125:602066) by chromium-webrtc-autoroll · 6 years ago
  88. 73f3917 Add support for signed deltas in FixedLengthDeltaEncoder by Elad Alon · 6 years ago
  89. 4b31cf5 Disable CertificateTest.CertificateIsUsedInConfig by Elad Alon · 6 years ago
  90. 087e9be AGC2 Limiter class renamed. by Alessio Bazzica · 6 years ago
  91. 4842c78 Increasing APM fuzzer coverage. by Alex Loiko · 6 years ago
  92. c6de47e Added supported H264 profiles for new iPhones by Yura Yaroshevich · 6 years ago
  93. 8f726be Add ability to override detection of resource location and source root by Artem Titov · 6 years ago
  94. 877dc89 Fix errors in AEC3 JSON parsing by Sam Zackrisson · 6 years ago
  95. 6e8e299 Revert "Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase" by Oleh Prypin · 6 years ago
  96. 7e6b528 Removes FakeBaseEngine. by Sebastian Jansson · 6 years ago
  97. 362cb50 Remove redundant RTC_DCHECK of max/min RTP header extension id by Johannes Kron · 6 years ago
  98. 93922dc Fix flaky unit test in rtc_unittests by Johannes Kron · 6 years ago
  99. 848273a Revert "Increase coverage of AEC3 JSON config unit tests, fix bugs" by Sam Zackrisson · 6 years ago
  100. 80cd25b Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase by Niels Möller · 6 years ago