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gerrit-public.fairphone.software / platform / external / webrtc / 5963c7cf0a70e1abdf0e3f313c10cb8a161f6cdb / audio / audio_receive_stream.cc
  1. f13df86 Delete audio methods SignalNetworkState by Niels Möller · 5 years ago
  2. 224c69d Delete ext_seqnum member from VoiceSenderInfo and VoiceReceiverInfo by Niels Möller · 5 years ago
  3. 70efdde Set local ssrc at construction of Rtp module by Erik Språng · 5 years ago
  4. 5b5d97c Reland of "Reporting of decoding_codec_plc events"" by Alex Narest · 5 years ago
  5. 054e3bb Reland "Replace the implementation of `GetContributingSources()` on the audio side." by Chen Xing · 5 years ago
  6. bedb7a8 Revert "Reporting of decoding_codec_plc events" by Mirko Bonadei · 5 years ago
  7. 0a88ea0 Reporting of decoding_codec_plc events by Alex Narest · 5 years ago
  8. 67008df Revert "Replace the implementation of `GetContributingSources()` on the audio side." by Artem Titov · 5 years ago
  9. 8fa7151 Replace the implementation of `GetContributingSources()` on the audio side. by Chen Xing · 5 years ago
  10. 3472b9a Delete RTCInboundRTPStreamStats::fraction_lost by Niels Möller · 6 years ago
  11. 4f08faa Introduce MediaTransportConfig by Anton Sukhanov · 6 years ago
  12. 8d8ffdb Expose new audio stats on the API by Ivo Creusen · 6 years ago
  13. 44125fa Reland "Piping audio interruption metrics to API layer" by Henrik Lundin · 6 years ago
  14. fc02a79 Revert "Piping audio interruption metrics to API layer" by Henrik Andreassson · 6 years ago
  15. 299c4e6 Piping audio interruption metrics to API layer by Henrik Lundin · 6 years ago
  16. 01738c6 Wire up RTCInboundRtpStreamStats.lastPacketReceivedTimestamp. by Henrik Boström · 6 years ago
  17. c01367d Deprecating ThreadChecker specific interface. by Sebastian Jansson · 6 years ago
  18. 232b3fd Expose relative packet arrival delay metric in stats API. by Jakob Ivarsson · 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. 0acffb5 Expose `jitterBufferEmittedCount` in addition to the existing `jitterBufferDelay` for `getStats()`. by Chen Xing · 6 years ago
  23. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  24. 53eae87 Add PeerConnection option to enable RTX handling in the audio jitter buffer. by Jakob Ivarsson · 6 years ago
  25. f693bfa Remove CodecInst pt.2 by Fredrik Solenberg · 6 years ago
  26. 10403ae Add PeerConnection option to configure minimum audio jitter buffer delay. by Jakob Ivarsson · 6 years ago
  27. 352ce5c Expose delayed packet outage as a cumulative metric of samples in the new getStats API. by Jakob Ivarsson · 6 years ago
  28. 8af8896 Expose jitter buffer flushes metric in new getStats api. by Ruslan Burakov · 6 years ago
  29. 2222a80 Delete unneeded includes of common_types.h and gn deps on webrtc_common. by Niels Möller · 6 years ago
  30. 349ade3 Delete class ChannelReceiveProxy. by Niels Möller · 6 years ago
  31. 7d76a31 Use MediaTransportInterface, for audio streams. by Niels Möller · 6 years ago
  32. 988cc08 [Cleanup] Add missing #include. Remove useless ones. by Yves Gerey · 6 years ago
  33. bfb444c Adds new CryptoOption crypto_options.frame.require_frame_encryption. by Benjamin Wright · 6 years ago
  34. ae4237e Set ChannelReceive transport at construction time. by Niels Möller · 6 years ago
  35. 84583f6 Enable End-to-End Encrypted Audio Payloads. by Benjamin Wright · 6 years ago
  36. 530ead4 Split voe::Channel into ChannelSend and ChannelReceive by Niels Möller · 6 years ago
  37. b222f49 Split ChannelProxy into send and receive classes. by Niels Möller · 6 years ago
  38. 30b4839 Refactor voe::Channel to not use RtpReceiver. by Niels Möller · 6 years ago
  39. fa4e185 Delete class voe::RtcEventLogProxy by Niels Möller · 6 years ago
  40. 848d6d3 Change Channel::GetRtpRtcp to return only RtpRtcp, not RtpReceiver. by Niels Möller · 6 years ago
  41. 665174f Reformat the WebRTC code base by Yves Gerey · 7 years ago
  42. b9b146c Replace rtc::Optional with absl::optional in audio, call and video by Danil Chapovalov · 7 years ago
  43. f782492 Delete RtpFeedback. The ssrc for a receive stream should be known at by Niels Möller · 7 years ago
  44. 5f22365 Remove unnecessary proxy+lock code around RtcpRttStats pointer by Tommi · 7 years ago
  45. 0812634 Pass a real audio codec pair ID to decoders that we create by Karl Wiberg · 7 years ago
  46. 881f168 Make SimpleStringBuilder into a non-template by Karl Wiberg · 7 years ago
  47. fef0500 Adding a new string utility class: SimpleStringBuilder. by Tommi · 7 years ago
  48. f120cba Delete AudioMonitor and related code. by Niels Möller · 7 years ago
  49. 24ea822 Remove logging in audio/* from release builds. by Jonas Olsson · 7 years ago
  50. a8b7c7f Move remaining traces of VoiceEngine by Fredrik Solenberg · 7 years ago
  51. 90ea504 Delete Channel::OnRecoveredPacket. by Niels Möller · 7 years ago
  52. 8f5787a Move ownership of voe::Channel into Audio[Receive|Send]Stream. by Fredrik Solenberg · 7 years ago
  53. 3b903d0 Reconfigure, not reconstruct, AudioReceiveStreams. by Fredrik Solenberg · 7 years ago
  54. d524751 Replace VoEBase::[Start/Stop]Playout(). by Fredrik Solenberg · 7 years ago
  55. 2707fb2 Optional: Use nullopt and implicit construction in /audio by Oskar Sundbom · 7 years ago
  56. 675513b Stop using LOG macros in favor of RTC_ prefixed macros. by Mirko Bonadei · 7 years ago
  57. c3fa8e1 New method RtpReceiver::GetLatestTimestamps. by Niels Möller · 7 years ago
  58. b0a0207 Added RTCMediaStreamTrackStats.jitterBufferDelay for audio by Gustaf Ullberg · 7 years ago
  59. 1c239d4 Remove voe::Statistics. by solenberg · 7 years ago
  60. 2397b9a Remove voe::OutputMixer and AudioConferenceMixer. by solenberg · 7 years ago
  61. 9a2e906 Added RTCMediaStreamTrackStats.concealmentEvents by Gustaf Ullberg · 7 years ago
  62. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  63. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/audio/audio_receive_stream.cc]
  64. 0e320ec Wiring discard rate of audio FEC/RED packets up to StatsReport. by minyue-webrtc · 7 years ago
  65. 2dbc69f Add stats totalSamplesReceived and concealedSamples by Steve Anton · 7 years ago
  66. abbc430 Make ~webrtc::AudioSendStream public, and s/config()/GetConfig(), as well as make public. by eladalon · 7 years ago
  67. 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
  68. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  69. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  70. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  71. 0f15f92 Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface. by nisse · 8 years ago
  72. 8d609f6 Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ ) by hbos · 8 years ago
  73. fbcc5cb Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ ) by olka · 8 years ago
  74. 292084c Added the GetSources() to the RtpReceiverInterface and implemented by zhihuang · 8 years ago
  75. fdbfdc9 Let PacketRouter separate send and receive modules. by nisse · 8 years ago
  76. 1c07c70 Reland "WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType" by kwiberg · 8 years ago
  77. 670a7f3 Revert of WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType (patchset #13 id:260001 of https://codereview.webrtc.org/2686043006/ ) by kwiberg · 8 years ago
  78. 1724cfb WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType by kwiberg · 8 years ago
  79. 796b8f9 Remove usage of VoEVolumeControl from WVoE and Audio[Send|Receive]Stream. by solenberg · 8 years ago
  80. 922246a Replace NULL with nullptr or null in webrtc/audio/ and common_audio/. by deadbeef · 8 years ago
  81. 657bab2 Replace AudioReceiveStream::DeliverRtp with OnRtpPacket. by nisse · 8 years ago
  82. 08b19df Remove VoEVideoSync interface. by solenberg · 8 years ago
  83. 4709e89 Move RemoteBitrateEstimator::RemoveStream calls from receive streams to Call. by nisse · 8 years ago
  84. 6b34124 Remove unnecessary RTPHeaderParser, following https://codereview.webrtc.org/2659563002/ by solenberg · 8 years ago
  85. bd9a77f Remove most of the remaining calls to VoE APIs from Audio[Send|Receive]Stream. by solenberg · 8 years ago
  86. d44ce05 Reland of Always call RemoteBitrateEstimator::IncomingPacket from Call. (patchset #1 id:1 of https://codereview.webrtc.org/2668973003/ ) by nisse · 8 years ago
  87. 14245cc Revert of Always call RemoteBitrateEstimator::IncomingPacket from Call. (patchset #9 id:160001 of https://codereview.webrtc.org/2659563002/ ) by nisse · 8 years ago
  88. 6d4dd59 Always call RemoteBitrateEstimator::IncomingPacket from Call. by nisse · 8 years ago
  89. 3ebbcb5 Stop using VoEVideoSync in Call/VideoReceiveStream. by solenberg · 8 years ago
  90. d32bf75 Pass SdpAudioFormat through Channel, without converting to CodecInst by kwiberg · 8 years ago
  91. 95aa964 Support external audio mixer in webrtc 2. by gyzhou · 8 years ago
  92. 39ce11f Revert of Support external audio mixer. (patchset #5 id:140001 of https://codereview.webrtc.org/2539213003/ ) by gyzhou · 8 years ago
  93. f6bcac5 Support external audio mixer in webrtc. by gyzhou · 8 years ago
  94. 0245da0 Move ownership of PacketRouter from CongestionController to Call. by nisse · 8 years ago
  95. 04c0722 Replace AudioConferenceMixer with AudioMixer. by aleloi · 8 years ago
  96. 1acfbd2 Expose RtpCodecParameters to VoiceMediaInfo stats. by hbos · 8 years ago
  97. d4adce4 Remove Absolute Send Time from list of supported header extensions for audio streams. by solenberg · 8 years ago
  98. 7602aab Remove usage of VoEBase::AssociateSendChannel() from WVoMC. by solenberg · 8 years ago
  99. 572ae12 Fix crash when registering abs-send-time to AudioSend/ReceiveStream. by stefan · 8 years ago
  100. b521aa7 Clean up abs-send-time for audio. by stefan · 8 years ago