Google Git
Sign in
webrtc / src / / 2eacbbc03a4a41ea658661225eb1c8fc07884c33 / audio / channel_receive.h
  1. 6a7bf10 Replace "rcvd" with "received" for readability by Philipp Hancke · 1 year, 11 months ago
  2. b3e5969 stats: use uint64_t for RTCSentRtpStreamStats.packetsSent by Philipp Hancke · 2 years ago
  3. af51228 audio: make packets lost a signed integer by Philipp Hancke · 2 years, 5 months ago
  4. c92338a Remove `CallReceiveStatistics::rttMs` by Alessio Bazzica · 2 years, 7 months ago
  5. 3176ef7 Rename AudioReceiveStream to AudioReceiveStreamInterface by Tommi · 2 years, 10 months ago
  6. 1def899 Remove legacy (unused) config param: jitter_buffer_enable_rtx_handling by Tommi · 2 years, 10 months ago
  7. 2562cf0 Reland "Wire up non-sender RTT for audio, and implement related standardized stats." by Ivo Creusen · 3 years, 7 months ago
  8. 2c41cba Revert "Wire up non-sender RTT for audio, and implement related standardized stats." by Björn Terelius · 3 years, 7 months ago
  9. fb0dca6 Wire up non-sender RTT for audio, and implement related standardized stats. by Ivo Creusen · 3 years, 7 months ago
  10. e54914a Implement nack_count metric for inbound audio rtp streams. by Jakob Ivarsson · 3 years, 9 months ago
  11. eb61b7f ModuleRtcRtcpImpl2: remove Module inheritance. by Markus Handell · 3 years, 9 months ago
  12. 08be9ba Don't recreate the audio receive stream when updating the local_ssrc. by Tommi · 3 years, 9 months ago
  13. e097282 Avoid recreating the audio stream when a frame decryptor is set. by Tommi · 3 years, 10 months ago
  14. f0adf38 Fix timestamps for the remote outbound audio stream stats by Alessio Bazzica · 4 years ago
  15. bc1c93d Add remote-outbound stats for audio streams by Alessio Bazzica · 4 years, 1 month ago
  16. dea374a Deliver packet info to source listeners when audio playout is disabled. by Ranveer Aggarwal · 4 years, 1 month ago
  17. 6b4d962 Fix standard GetStats to not modify NetEq state. by Niels Möller · 4 years, 6 months ago
  18. bef7b05 Make AV sync robust to failures to set a desired minimum delay by Ivo Creusen · 4 years, 7 months ago
  19. 3e9af7f Insert audio frame transformer between depacketizer and decoder. by Marina Ciocea · 5 years ago
  20. 7a9a092 Delete media transport integration. by Bjorn A Mellem · 5 years ago
  21. c3d1f9b Enable injection of a custom NetEqFactory into PeerConnectionFactory. by Ivo Creusen · 5 years ago
  22. fcf79cc Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats. by Åsa Persson · 5 years ago
  23. ac0a4cb Reland "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Niels Möller · 5 years ago
  24. ef0627f Revert "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Mirko Bonadei · 5 years ago
  25. fbde32e Fix GetStats bytesSent/Received, wireup headerBytesSent/Received by Niels Möller · 5 years ago
  26. 65f17ca Move MediaTransportInterface out of the libjingle_peerconnection_api target by Niels Möller · 6 years ago
  27. a837030 Split out RtpSource from libjingle_peerconnection_api by Niels Möller · 6 years ago
  28. 224c69d Delete ext_seqnum member from VoiceSenderInfo and VoiceReceiverInfo by Niels Möller · 6 years ago
  29. 70efdde Set local ssrc at construction of Rtp module by Erik Språng · 6 years ago
  30. ed44f54 In ChannelReceive, use AcmReceiver directly, not AudioCodingModule by Niels Möller · 6 years ago
  31. 054e3bb Reland "Replace the implementation of `GetContributingSources()` on the audio side." by Chen Xing · 6 years ago
  32. a4d8737 Format almost everything. by Jonas Olsson · 6 years ago
  33. 67008df Revert "Replace the implementation of `GetContributingSources()` on the audio side." by Artem Titov · 6 years ago
  34. 8fa7151 Replace the implementation of `GetContributingSources()` on the audio side. by Chen Xing · 6 years ago
  35. 3472b9a Delete RTCInboundRTPStreamStats::fraction_lost by Niels Möller · 6 years ago
  36. 4f08faa Introduce MediaTransportConfig by Anton Sukhanov · 6 years ago
  37. 01738c6 Wire up RTCInboundRtpStreamStats.lastPacketReceivedTimestamp. by Henrik Boström · 6 years ago
  38. 8fb1a6a Delete a few return values from audio streams and video send streams. by Niels Möller · 6 years ago
  39. 977b335 Injecting Clock into audio streams. by Sebastian Jansson · 6 years ago
  40. 3b50f9f Propagate base minimum delay to audio_receiver_stream by Ruslan Burakov · 6 years ago
  41. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  42. 53eae87 Add PeerConnection option to enable RTX handling in the audio jitter buffer. by Jakob Ivarsson · 6 years ago
  43. f693bfa Remove CodecInst pt.2 by Fredrik Solenberg · 6 years ago
  44. 10403ae Add PeerConnection option to configure minimum audio jitter buffer delay. by Jakob Ivarsson · 6 years ago
  45. 2222a80 Delete unneeded includes of common_types.h and gn deps on webrtc_common. by Niels Möller · 6 years ago
  46. 78e88fe Move NetworkStatistics and AudioDecodingCallStats from common_types.h by Fredrik Solenberg · 6 years ago
  47. dced9f6 Delete class ChannelSendProxy by Niels Möller · 6 years ago
  48. 349ade3 Delete class ChannelReceiveProxy. by Niels Möller · 6 years ago
  49. 8fb5746 Delete obsolete interface class RtpData by Niels Möller · 6 years ago
  50. 80c6762 Tweak ChannelReceive interface, to make it closer to ChannelReceiveProxy by Niels Möller · 6 years ago
  51. 7d76a31 Use MediaTransportInterface, for audio streams. by Niels Möller · 6 years ago
  52. 78410ad Fixes use after free error when setting a new FrameEncryptor on ChannelSend. by Benjamin Wright · 6 years ago
  53. bfb444c Adds new CryptoOption crypto_options.frame.require_frame_encryption. by Benjamin Wright · 6 years ago
  54. ae4237e Set ChannelReceive transport at construction time. by Niels Möller · 6 years ago
  55. 84583f6 Enable End-to-End Encrypted Audio Payloads. by Benjamin Wright · 6 years ago
  56. 530ead4 Split voe::Channel into ChannelSend and ChannelReceive by Niels Möller · 6 years ago