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gerrit-public.fairphone.software / platform / external / webrtc / e7a5f7bfae7e619d2c9711a0b13a17d02e407563 / call / call_unittest.cc
  1. 3d2ed19 Remove Transport implementation from ChannelSend by Fredrik Solenberg · 6 years ago
  2. 179a392 Implement TargetBitrate, NetworkRoute and overhead features of media transport interface. by Piotr (Peter) Slatala · 6 years ago
  3. cc8e8bb Pass the media transport from JsepTransportController to Call. by Piotr (Peter) Slatala · 6 years ago
  4. 7d76a31 Use MediaTransportInterface, for audio streams. by Niels Möller · 6 years ago
  5. ae4237e Set ChannelReceive transport at construction time. by Niels Möller · 6 years ago
  6. 918f50c Use absl::make_unique and absl::WrapUnique directly by Karl Wiberg · 6 years ago
  7. 665174f Reformat the WebRTC code base by Yves Gerey · 7 years ago
  8. e5447fb Removed fake rtp transport controller send. by Sebastian Jansson · 7 years ago
  9. fc8d26b Reland "Moved BitrateConfig out of Call::Config." by Sebastian Jansson · 7 years ago
  10. e4bf600 Revert "Moved BitrateConfig out of Call::Config." by Lu Liu · 7 years ago
  11. 5897fe2 Moved BitrateConfig out of Call::Config. by Sebastian Jansson · 7 years ago
  12. 8f5787a Move ownership of voe::Channel into Audio[Receive|Send]Stream. by Fredrik Solenberg · 7 years ago
  13. fedc00c Optional: Use nullopt and implicit construction in /call by Oskar Sundbom · 7 years ago
  14. 2a87797 Remove voe::TransmitMixer by Fredrik Solenberg · 7 years ago
  15. d319534 Move ADM initialization into WebRtcVoiceEngine by Fredrik Solenberg · 7 years ago
  16. f3850f6 Voice Engine: Require caller to supply an AudioDecoderFactory by Karl Wiberg · 7 years ago
  17. a32dd01 Reland "Remove AudioDeviceObserver and make ADM not inherit from the Module interface." by Fredrik Solenberg · 7 years ago
  18. d4404c2 Revert "Remove AudioDeviceObserver and make ADM not inherit from the Module interface." by Fredrik Solenberg · 7 years ago
  19. 34cdd2d Remove AudioDeviceObserver and make ADM not inherit from the Module interface. by Fredrik Solenberg · 7 years ago
  20. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  21. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/call/call_unittest.cc]
  22. 5c8942a Move PacedSender ownership to RtpTransportControllerSend. by Stefan Holmer · 7 years ago
  23. db2a9fc Wire up RTP keep-alive in ortc api. by sprang · 7 years ago
  24. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  25. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  26. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  27. a9cc40b Allow an external audio processing module to be used in WebRTC by peah · 8 years ago
  28. 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
  29. 441718e Revert of Add PeerConnectionInterface::UpdateCallBitrate. (patchset #7 id:120001 of https://codereview.webrtc.org/2888303005/ ) by charujain · 8 years ago
  30. 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
  31. c3d4b48 Store/restore RTP state for audio streams with same SSRC within a call by ossu · 8 years ago
  32. 8c96a14 Simple tests for Call::SetBitrateConfig. by zstein · 8 years ago
  33. 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
  34. 37e99fd Move AudioDecoder and AudioDecoderFactory mocks to webrtc/test/ by kwiberg · 8 years ago
  35. 1c07c70 Reland "WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType" by kwiberg · 8 years ago
  36. 670a7f3 Revert of WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType (patchset #13 id:260001 of https://codereview.webrtc.org/2686043006/ ) by kwiberg · 8 years ago
  37. 1724cfb WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType by kwiberg · 8 years ago
  38. 8313a6f Make |rtcp_send_transport| mandatory in FlexfecReceiveStream::Config. by brandtr · 8 years ago
  39. 1cfbd60 Generalize FlexfecReceiveStream::Config. by brandtr · 8 years ago
  40. f515ab8 Moved call.h and most of api/call/* into call/ by ossu · 8 years ago
  41. 10111bc Passed AudioMixer to AudioState::Config. by aleloi · 8 years ago
  42. dd31071 Added an empty AudioTransportProxy to AudioState. by aleloi · 8 years ago
  43. 7602aab Remove usage of VoEBase::AssociateSendChannel() from WVoMC. by solenberg · 8 years ago
  44. 25445d3 Integrate FlexfecReceiveStream with Call. by brandtr · 8 years ago
  45. 11a9cbf Refactoring: move ownership of RtcEventLog from Call to PeerConnection by skvlad · 8 years ago
  46. ac9f876 Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/ by kwiberg · 8 years ago
  47. 77eab70 Enable the -Wundef warning for clang by kwiberg · 8 years ago
  48. a69d973 Move webrtc/audio_*.h to webrtc/api/call by kjellander · 8 years ago
  49. 29b1a8d Moved creation of AudioDecoderFactory to inside PeerConnectionFactory. by ossu · 9 years ago
  50. b25345e Replace scoped_ptr with unique_ptr in webrtc/call/ by kwiberg · 9 years ago
  51. 3a94154 Move some send stream configuration into webrtc::AudioSendStream. by solenberg · 9 years ago
  52. 566ef24 Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats(). by solenberg · 9 years ago
  53. 0ccae13 Changed FakeVoiceEngine into a MockVoiceEngine. by Fredrik Solenberg · 9 years ago
  54. 4f4ec0a Re-Land: Implement AudioReceiveStream::GetStats(). by Fredrik Solenberg · 9 years ago
  55. 43e83d4 Revert of Implement AudioReceiveStream::GetStats(). (patchset #19 id:360001 of https://codereview.webrtc.org/1390753002/ ) by solenberg · 9 years ago
  56. a457752 Implement AudioReceiveStream::GetStats(). by Fredrik Solenberg · 9 years ago
  57. c7a8b08 Add webrtc::AudioSendStream and methods on webrtc::Call to create and delete AudioSendStreams. by solenberg · 9 years ago