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gerrit-public.fairphone.software / platform / external / webrtc / 6725b648c869689fd28038f95124e1de239b2ef1 / audio / audio_receive_stream.cc
  1. e618cc9 Add jitterBufferTargetDelay as RTCNonStandardStatsMember to new GetStats API by Artem Titov · 4 years, 8 months ago
  2. 74d2b1d Add periodic logging of sync delays. by Åsa Persson · 4 years, 9 months ago
  3. 7a9a092 Delete media transport integration. by Bjorn A Mellem · 5 years ago
  4. c3d1f9b Enable injection of a custom NetEqFactory into PeerConnectionFactory. by Ivo Creusen · 5 years ago
  5. fcf79cc Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats. by Åsa Persson · 5 years ago
  6. 9429888 Delete deprecated bytes_sent/bytes_rcvd stat values by Niels Möller · 5 years ago
  7. ac0a4cb Reland "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Niels Möller · 5 years ago
  8. ef0627f Revert "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Mirko Bonadei · 5 years ago
  9. fbde32e Fix GetStats bytesSent/Received, wireup headerBytesSent/Received by Niels Möller · 5 years ago
  10. f13df86 Delete audio methods SignalNetworkState by Niels Möller · 5 years ago
  11. 224c69d Delete ext_seqnum member from VoiceSenderInfo and VoiceReceiverInfo by Niels Möller · 5 years ago
  12. 70efdde Set local ssrc at construction of Rtp module by Erik Språng · 5 years ago
  13. 5b5d97c Reland of "Reporting of decoding_codec_plc events"" by Alex Narest · 5 years ago
  14. 054e3bb Reland "Replace the implementation of `GetContributingSources()` on the audio side." by Chen Xing · 5 years ago
  15. bedb7a8 Revert "Reporting of decoding_codec_plc events" by Mirko Bonadei · 5 years ago
  16. 0a88ea0 Reporting of decoding_codec_plc events by Alex Narest · 5 years ago
  17. 67008df Revert "Replace the implementation of `GetContributingSources()` on the audio side." by Artem Titov · 5 years ago
  18. 8fa7151 Replace the implementation of `GetContributingSources()` on the audio side. by Chen Xing · 5 years ago
  19. 3472b9a Delete RTCInboundRTPStreamStats::fraction_lost by Niels Möller · 5 years ago
  20. 4f08faa Introduce MediaTransportConfig by Anton Sukhanov · 5 years ago
  21. 8d8ffdb Expose new audio stats on the API by Ivo Creusen · 6 years ago
  22. 44125fa Reland "Piping audio interruption metrics to API layer" by Henrik Lundin · 6 years ago
  23. fc02a79 Revert "Piping audio interruption metrics to API layer" by Henrik Andreassson · 6 years ago
  24. 299c4e6 Piping audio interruption metrics to API layer by Henrik Lundin · 6 years ago
  25. 01738c6 Wire up RTCInboundRtpStreamStats.lastPacketReceivedTimestamp. by Henrik Boström · 6 years ago
  26. c01367d Deprecating ThreadChecker specific interface. by Sebastian Jansson · 6 years ago
  27. 232b3fd Expose relative packet arrival delay metric in stats API. by Jakob Ivarsson · 6 years ago
  28. 8fb1a6a Delete a few return values from audio streams and video send streams. by Niels Möller · 6 years ago
  29. 977b335 Injecting Clock into audio streams. by Sebastian Jansson · 6 years ago
  30. 3b50f9f Propagate base minimum delay to audio_receiver_stream by Ruslan Burakov · 6 years ago
  31. 0acffb5 Expose `jitterBufferEmittedCount` in addition to the existing `jitterBufferDelay` for `getStats()`. by Chen Xing · 6 years ago
  32. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  33. 53eae87 Add PeerConnection option to enable RTX handling in the audio jitter buffer. by Jakob Ivarsson · 6 years ago
  34. f693bfa Remove CodecInst pt.2 by Fredrik Solenberg · 6 years ago
  35. 10403ae Add PeerConnection option to configure minimum audio jitter buffer delay. by Jakob Ivarsson · 6 years ago
  36. 352ce5c Expose delayed packet outage as a cumulative metric of samples in the new getStats API. by Jakob Ivarsson · 6 years ago
  37. 8af8896 Expose jitter buffer flushes metric in new getStats api. by Ruslan Burakov · 6 years ago
  38. 2222a80 Delete unneeded includes of common_types.h and gn deps on webrtc_common. by Niels Möller · 6 years ago
  39. 349ade3 Delete class ChannelReceiveProxy. by Niels Möller · 6 years ago
  40. 7d76a31 Use MediaTransportInterface, for audio streams. by Niels Möller · 6 years ago
  41. 988cc08 [Cleanup] Add missing #include. Remove useless ones. by Yves Gerey · 6 years ago
  42. bfb444c Adds new CryptoOption crypto_options.frame.require_frame_encryption. by Benjamin Wright · 6 years ago
  43. ae4237e Set ChannelReceive transport at construction time. by Niels Möller · 6 years ago
  44. 84583f6 Enable End-to-End Encrypted Audio Payloads. by Benjamin Wright · 6 years ago
  45. 530ead4 Split voe::Channel into ChannelSend and ChannelReceive by Niels Möller · 6 years ago
  46. b222f49 Split ChannelProxy into send and receive classes. by Niels Möller · 6 years ago
  47. 30b4839 Refactor voe::Channel to not use RtpReceiver. by Niels Möller · 6 years ago
  48. fa4e185 Delete class voe::RtcEventLogProxy by Niels Möller · 6 years ago
  49. 848d6d3 Change Channel::GetRtpRtcp to return only RtpRtcp, not RtpReceiver. by Niels Möller · 6 years ago
  50. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  51. b9b146c Replace rtc::Optional with absl::optional in audio, call and video by Danil Chapovalov · 6 years ago
  52. f782492 Delete RtpFeedback. The ssrc for a receive stream should be known at by Niels Möller · 6 years ago
  53. 5f22365 Remove unnecessary proxy+lock code around RtcpRttStats pointer by Tommi · 7 years ago
  54. 0812634 Pass a real audio codec pair ID to decoders that we create by Karl Wiberg · 7 years ago
  55. 881f168 Make SimpleStringBuilder into a non-template by Karl Wiberg · 7 years ago
  56. fef0500 Adding a new string utility class: SimpleStringBuilder. by Tommi · 7 years ago
  57. f120cba Delete AudioMonitor and related code. by Niels Möller · 7 years ago
  58. 24ea822 Remove logging in audio/* from release builds. by Jonas Olsson · 7 years ago
  59. a8b7c7f Move remaining traces of VoiceEngine by Fredrik Solenberg · 7 years ago
  60. 90ea504 Delete Channel::OnRecoveredPacket. by Niels Möller · 7 years ago
  61. 8f5787a Move ownership of voe::Channel into Audio[Receive|Send]Stream. by Fredrik Solenberg · 7 years ago
  62. 3b903d0 Reconfigure, not reconstruct, AudioReceiveStreams. by Fredrik Solenberg · 7 years ago
  63. d524751 Replace VoEBase::[Start/Stop]Playout(). by Fredrik Solenberg · 7 years ago
  64. 2707fb2 Optional: Use nullopt and implicit construction in /audio by Oskar Sundbom · 7 years ago
  65. 675513b Stop using LOG macros in favor of RTC_ prefixed macros. by Mirko Bonadei · 7 years ago
  66. c3fa8e1 New method RtpReceiver::GetLatestTimestamps. by Niels Möller · 7 years ago
  67. b0a0207 Added RTCMediaStreamTrackStats.jitterBufferDelay for audio by Gustaf Ullberg · 7 years ago
  68. 1c239d4 Remove voe::Statistics. by solenberg · 7 years ago
  69. 2397b9a Remove voe::OutputMixer and AudioConferenceMixer. by solenberg · 7 years ago
  70. 9a2e906 Added RTCMediaStreamTrackStats.concealmentEvents by Gustaf Ullberg · 7 years ago
  71. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  72. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/audio/audio_receive_stream.cc]
  73. 0e320ec Wiring discard rate of audio FEC/RED packets up to StatsReport. by minyue-webrtc · 7 years ago
  74. 2dbc69f Add stats totalSamplesReceived and concealedSamples by Steve Anton · 7 years ago
  75. abbc430 Make ~webrtc::AudioSendStream public, and s/config()/GetConfig(), as well as make public. by eladalon · 7 years ago
  76. e76bd3a Adding stats that can be used to compute output audio levels as described here https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy. by zstein · 7 years ago
  77. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  78. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  79. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  80. 0f15f92 Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface. by nisse · 7 years ago
  81. 8d609f6 Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ ) by hbos · 8 years ago
  82. fbcc5cb Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ ) by olka · 8 years ago
  83. 292084c Added the GetSources() to the RtpReceiverInterface and implemented by zhihuang · 8 years ago
  84. fdbfdc9 Let PacketRouter separate send and receive modules. by nisse · 8 years ago
  85. 1c07c70 Reland "WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType" by kwiberg · 8 years ago
  86. 670a7f3 Revert of WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType (patchset #13 id:260001 of https://codereview.webrtc.org/2686043006/ ) by kwiberg · 8 years ago
  87. 1724cfb WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType by kwiberg · 8 years ago
  88. 796b8f9 Remove usage of VoEVolumeControl from WVoE and Audio[Send|Receive]Stream. by solenberg · 8 years ago
  89. 922246a Replace NULL with nullptr or null in webrtc/audio/ and common_audio/. by deadbeef · 8 years ago
  90. 657bab2 Replace AudioReceiveStream::DeliverRtp with OnRtpPacket. by nisse · 8 years ago
  91. 08b19df Remove VoEVideoSync interface. by solenberg · 8 years ago
  92. 4709e89 Move RemoteBitrateEstimator::RemoveStream calls from receive streams to Call. by nisse · 8 years ago
  93. 6b34124 Remove unnecessary RTPHeaderParser, following https://codereview.webrtc.org/2659563002/ by solenberg · 8 years ago
  94. bd9a77f Remove most of the remaining calls to VoE APIs from Audio[Send|Receive]Stream. by solenberg · 8 years ago
  95. d44ce05 Reland of Always call RemoteBitrateEstimator::IncomingPacket from Call. (patchset #1 id:1 of https://codereview.webrtc.org/2668973003/ ) by nisse · 8 years ago
  96. 14245cc Revert of Always call RemoteBitrateEstimator::IncomingPacket from Call. (patchset #9 id:160001 of https://codereview.webrtc.org/2659563002/ ) by nisse · 8 years ago
  97. 6d4dd59 Always call RemoteBitrateEstimator::IncomingPacket from Call. by nisse · 8 years ago
  98. 3ebbcb5 Stop using VoEVideoSync in Call/VideoReceiveStream. by solenberg · 8 years ago
  99. d32bf75 Pass SdpAudioFormat through Channel, without converting to CodecInst by kwiberg · 8 years ago
  100. 95aa964 Support external audio mixer in webrtc 2. by gyzhou · 8 years ago