1. 80b7c6b Delete Call dependency on ProcessThread as unused by Danil Chapovalov · 2 years, 7 months ago
  2. 3176ef7 Rename AudioReceiveStream to AudioReceiveStreamInterface by Tommi · 2 years, 8 months ago
  3. f6f4543 Rename VideoReceiveStream to VideoReceiveStreamInterface by Tommi · 2 years, 8 months ago
  4. 641a1b1 Adopt absl::string_view in call/ by Ali Tofigh · 2 years, 8 months ago
  5. 1331c18 Reland: Update local_ssrc without needing to recreate video streams. by Tommi · 2 years, 8 months ago
  6. c92ee5f Revert "Update local_ssrc without needing to recreate video streams." by Tomas Gunnarsson · 2 years, 9 months ago
  7. 16a8b25 Update local_ssrc without needing to recreate video streams. by Tommi · 2 years, 9 months ago
  8. cf4ed15 Add GetRtpExtensionMap to ReceiveStream and remove GetRtpExtensions. by Tommi · 2 years, 9 months ago
  9. e62c2f2 WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf by Jonas Oreland · 2 years, 10 months ago
  10. d5f414c Add class comments to Call and ChannelInterface by Harald Alvestrand · 3 years ago
  11. 6b7b255 Update call/ to not use implicit T* --> scoped_refptr<T> conversion by Niels Möller · 3 years ago
  12. 55107c8 Update the sync_group id without recreating audio receive streams. by Tommi · 3 years, 7 months ago
  13. 08be9ba Don't recreate the audio receive stream when updating the local_ssrc. by Tommi · 3 years, 7 months ago
  14. 504fc19 Add ability to pass factory for RtpTransportControllerSend to PeerConnectionFactoryDependencies. by Vojin Ilic · 3 years, 8 months ago
  15. e984aa2 Add thread accessors to Call. by Tomas Gunnarsson · 3 years, 9 months ago
  16. ceb4495 Reland: Wires up WebrtcKeyValueBasedConfig in media engines. by Erik Språng · 4 years, 4 months ago
  17. 5956a17 Revert "Wires up WebrtcKeyValueBasedConfig in media engines." by Artem Titov · 4 years, 4 months ago
  18. 591b2ab Wires up WebrtcKeyValueBasedConfig in media engines. by Erik Språng · 4 years, 5 months ago
  19. 4c50e70 Ensure CreateTimeControllerBasedCallFactory use simulated time in Call::SharedModuleThread by Per Kjellander · 4 years, 7 months ago
  20. f4a9991 [Adaptation] Adding adaptation resources from Call. by Henrik Boström · 4 years, 8 months ago
  21. 25c77c1 Add SharedModuleThread class to share a module thread across Call instances. by Tommi · 4 years, 8 months ago
  22. 78a7138 Remove MediaTransport from Call. by Tommi · 5 years ago
  23. 0182a03 Reland "Remove the injectable bitrate allocation strategy API." by Jonas Olsson · 6 years ago
  24. e95b57c Revert "Remove the injectable bitrate allocation strategy API." by Mirko Bonadei · 6 years ago
  25. 80cb3f6 Remove the injectable bitrate allocation strategy API. by Jonas Olsson · 6 years ago
  26. 359fe33 Move TaskQueueFactory from Call::Create parameter to CallConfig by Danil Chapovalov · 6 years ago
  27. c42f1a7 Revert "Move TaskQueueFactory from Call::Create parameter to CallConfig" by Danil Chapovalov · 6 years ago
  28. 90705cb Move TaskQueueFactory from Call::Create parameter to CallConfig by Danil Chapovalov · 6 years ago
  29. 7fbfaa4 PeerConnection::SetBitrate now also configures media transport. by Piotr (Peter) Slatala · 6 years ago
  30. 4e5f5ed Allow Clock injection in Call. by Sebastian Jansson · 6 years ago
  31. 896b47c Injecting ProcessThread and TaskQueueFactory in Call. by Sebastian Jansson · 6 years ago
  32. 547a1dc Removes injection of RtpTransportControllerSend from Call::Create. by Sebastian Jansson · 6 years ago
  33. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  34. 2222a80 Delete unneeded includes of common_types.h and gn deps on webrtc_common. by Niels Möller · 6 years ago
  35. cc8e8bb Pass the media transport from JsepTransportController to Call. by Piotr (Peter) Slatala · 6 years ago
  36. 1298541 Removing unnecessary dependencies on socket.h. by Sebastian Jansson · 6 years ago
  37. 64be7fa Move FecController to RtpVideoSender. by Stefan Holmer · 6 years ago
  38. 7008287 Delete struct webrtc::PacketTime. by Niels Möller · 6 years ago
  39. b6b29e0 Convert video quality test from a TEST_F to a TEST fixture. by Patrik Höglund · 7 years ago
  40. 665174f Reformat the WebRTC code base by Yves Gerey · 7 years ago
  41. 11b34f4 Remove chromium clang style errors affecting sdk/android/media_jni by Paulina Hensman · 7 years ago
  42. 8f83b42 Moved bitrate config interface from Call class. by Sebastian Jansson · 7 years ago
  43. fc8d26b Reland "Moved BitrateConfig out of Call::Config." by Sebastian Jansson · 7 years ago
  44. e4bf600 Revert "Moved BitrateConfig out of Call::Config." by Lu Liu · 7 years ago
  45. 5897fe2 Moved BitrateConfig out of Call::Config. by Sebastian Jansson · 7 years ago
  46. 0dd1b0a Revert "Revert "Enables PeerConnectionFactory using external fec controller"" by Ying Wang · 7 years ago
  47. 0073301 Revert "Enables PeerConnectionFactory using external fec controller" by Taylor Brandstetter · 7 years ago
  48. 4f07bdb Enables PeerConnectionFactory using external fec controller by Ying Wang · 7 years ago
  49. 8366e17 Rename Call::Config to CallConfig, keep old name as alias. by Niels Möller · 7 years ago
  50. 3b790f3 Make fec controller plug-able. by Ying Wang · 7 years ago
  51. 292a73e Deliver packet to Call as rtc::CopyOnWriteBuffer by Danil Chapovalov · 7 years ago
  52. 78609d5 Reland of BWE allocation strategy by Alex Narest · 7 years ago
  53. dc9ca93 Revert "BWE allocation strategy" by Alex Narest · 7 years ago
  54. a5fbc23 BWE allocation strategy by Alex Narest · 7 years ago
  55. 39260c4 Revert "BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic." by Lu Liu · 7 years ago
  56. 54d1da1 BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic. by Alex Narest · 7 years ago
  57. 7120742 Adding NOLINT for typedefs.h and common_types.h by Mirko Bonadei · 7 years ago
  58. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  59. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/call/call.h]
  60. 440b6d9 Move video send/receive stream headers to webrtc/call. by aleloi · 7 years ago
  61. db2a9fc Wire up RTP keep-alive in ortc api. by sprang · 7 years ago
  62. e5c4a81 Move RTP keep-alive config from VideoSendStream::Config to Call::Config by sprang · 8 years ago
  63. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  64. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  65. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  66. 38ede13 Support building WebRTC without audio and video. by zhihuang · 8 years ago
  67. a5e0df6 Move MinPositive to call.h as discussed here: https://codereview.chromium.org/2888303005/#msg19 by zstein · 8 years ago
  68. 4b97980 Relanding: Adds PeerConnectionInterface::UpdateCallBitrate to give clients more control of the bandwidth estimator. PeerConnection implements this method by passing a BitrateConfigMask to its associated Call, which is combined with the existing BitrateConfig and passed on to the SendSideCongestionController as necessary. The existing BitrateConfig generally comes from the x-google-{min,start,max}-bitrate params in the SDP. by zstein · 8 years ago
  69. 441718e Revert of Add PeerConnectionInterface::UpdateCallBitrate. (patchset #7 id:120001 of https://codereview.webrtc.org/2888303005/ ) by charujain · 8 years ago
  70. 9641c13 Adds PeerConnectionInterface::UpdateCallBitrate to give clients more control of the bandwidth estimator. PeerConnection implements this method by passing a BitrateConfigMask to its associated Call, which is combined with the existing BitrateConfig and passed on to the SendSideCongestionController as necessary. The existing BitrateConfig generally comes from the x-google-{min,start,max}-bitrate params in the SDP. by zstein · 8 years ago
  71. 7cb69d5 This will allow me to test that Call invokes SendSideCongestionController::SetBweBitrates as expected (for https://codereview.chromium.org/2793913008). by zstein · 8 years ago
  72. b8a654c Delete declaration of non-existing function webrtc::Version(). by nisse · 8 years ago
  73. fb45c6c Inform jitter buffer about FlexFEC protection. by brandtr · 8 years ago
  74. 7250b39 Move FlexfecReceiveStream from api/call/ to call/. by brandtr · 8 years ago
  75. 446fcb6 Clean up FlexfecReceiveStream ctor signatures. by brandtr · 8 years ago
  76. f515ab8 Moved call.h and most of api/call/* into call/ by ossu · 8 years ago