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gerrit-public.fairphone.software / platform / external / webrtc / 5256d8bc4b67f4ec777a72d0c6cbeb0f182963c2 / audio / channel_receive.h
  1. 7a9a092 Delete media transport integration. by Bjorn A Mellem · 5 years ago
  2. c3d1f9b Enable injection of a custom NetEqFactory into PeerConnectionFactory. by Ivo Creusen · 5 years ago
  3. fcf79cc Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats. by Åsa Persson · 5 years ago
  4. ac0a4cb Reland "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Niels Möller · 5 years ago
  5. ef0627f Revert "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Mirko Bonadei · 5 years ago
  6. fbde32e Fix GetStats bytesSent/Received, wireup headerBytesSent/Received by Niels Möller · 5 years ago
  7. 65f17ca Move MediaTransportInterface out of the libjingle_peerconnection_api target by Niels Möller · 5 years ago
  8. a837030 Split out RtpSource from libjingle_peerconnection_api by Niels Möller · 5 years ago
  9. 224c69d Delete ext_seqnum member from VoiceSenderInfo and VoiceReceiverInfo by Niels Möller · 5 years ago
  10. 70efdde Set local ssrc at construction of Rtp module by Erik Språng · 5 years ago
  11. ed44f54 In ChannelReceive, use AcmReceiver directly, not AudioCodingModule by Niels Möller · 5 years ago
  12. 054e3bb Reland "Replace the implementation of `GetContributingSources()` on the audio side." by Chen Xing · 5 years ago
  13. a4d8737 Format almost everything. by Jonas Olsson · 5 years ago
  14. 67008df Revert "Replace the implementation of `GetContributingSources()` on the audio side." by Artem Titov · 5 years ago
  15. 8fa7151 Replace the implementation of `GetContributingSources()` on the audio side. by Chen Xing · 5 years ago
  16. 3472b9a Delete RTCInboundRTPStreamStats::fraction_lost by Niels Möller · 5 years ago
  17. 4f08faa Introduce MediaTransportConfig by Anton Sukhanov · 5 years ago
  18. 01738c6 Wire up RTCInboundRtpStreamStats.lastPacketReceivedTimestamp. by Henrik Boström · 6 years ago
  19. 8fb1a6a Delete a few return values from audio streams and video send streams. by Niels Möller · 6 years ago
  20. 977b335 Injecting Clock into audio streams. by Sebastian Jansson · 6 years ago
  21. 3b50f9f Propagate base minimum delay to audio_receiver_stream by Ruslan Burakov · 6 years ago
  22. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  23. 53eae87 Add PeerConnection option to enable RTX handling in the audio jitter buffer. by Jakob Ivarsson · 6 years ago
  24. f693bfa Remove CodecInst pt.2 by Fredrik Solenberg · 6 years ago
  25. 10403ae Add PeerConnection option to configure minimum audio jitter buffer delay. by Jakob Ivarsson · 6 years ago
  26. 2222a80 Delete unneeded includes of common_types.h and gn deps on webrtc_common. by Niels Möller · 6 years ago
  27. 78e88fe Move NetworkStatistics and AudioDecodingCallStats from common_types.h by Fredrik Solenberg · 6 years ago
  28. dced9f6 Delete class ChannelSendProxy by Niels Möller · 6 years ago
  29. 349ade3 Delete class ChannelReceiveProxy. by Niels Möller · 6 years ago
  30. 8fb5746 Delete obsolete interface class RtpData by Niels Möller · 6 years ago
  31. 80c6762 Tweak ChannelReceive interface, to make it closer to ChannelReceiveProxy by Niels Möller · 6 years ago
  32. 7d76a31 Use MediaTransportInterface, for audio streams. by Niels Möller · 6 years ago
  33. 78410ad Fixes use after free error when setting a new FrameEncryptor on ChannelSend. by Benjamin Wright · 6 years ago
  34. bfb444c Adds new CryptoOption crypto_options.frame.require_frame_encryption. by Benjamin Wright · 6 years ago
  35. ae4237e Set ChannelReceive transport at construction time. by Niels Möller · 6 years ago
  36. 84583f6 Enable End-to-End Encrypted Audio Payloads. by Benjamin Wright · 6 years ago
  37. 530ead4 Split voe::Channel into ChannelSend and ChannelReceive by Niels Möller · 6 years ago