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webrtc / src / 62dc65b5375aea6bb846927c2be6ab6893ca89b3 / call / audio_receive_stream.h
  1. b3e5969 stats: use uint64_t for RTCSentRtpStreamStats.packetsSent by Philipp Hancke · 2 years ago
  2. 0c126ed De-flake NonSenderRttStats and make it faster to run on average. by Henrik Boström · 2 years ago
  3. 9ece54f Delete unnecssary AudioReceiveStreamInterface::GetRtpExtensions by Per K · 2 years, 2 months ago
  4. af51228 audio: make packets lost a signed integer by Philipp Hancke · 2 years, 4 months ago
  5. 1a84b56 Implement RTCInboundRTPStreamStats.JitterBufferMinimumDelay by Ivo Creusen · 2 years, 8 months ago
  6. a136ed4 Add SetTransportCc to ReceiveStreamInterface. by Tommi · 2 years, 10 months ago
  7. 5ac19df Remove deprecated alias, AudioReceiveStream by Tommi · 2 years, 10 months ago
  8. 3176ef7 Rename AudioReceiveStream to AudioReceiveStreamInterface by Tommi · 2 years, 10 months ago
  9. 1def899 Remove legacy (unused) config param: jitter_buffer_enable_rtx_handling by Tommi · 2 years, 10 months ago
  10. 6fb674e Rename MediaReceiveStream to MediaReceiveStreamInterface by Tommi · 2 years, 10 months ago
  11. cf4ed15 Add GetRtpExtensionMap to ReceiveStream and remove GetRtpExtensions. by Tommi · 2 years, 10 months ago
  12. 7a15ff3 Add a transport_cc() getter and remove rtp_config(). by Tommi · 2 years, 10 months ago
  13. cb7c736 Separate reading remote_ssrc from using the rtp_config() getter. by Tommi · 2 years, 10 months ago
  14. 1f38a38 Add ability to set rtp header extensions without recreating streams. by Tommi · 3 years, 6 months ago
  15. 2562cf0 Reland "Wire up non-sender RTT for audio, and implement related standardized stats." by Ivo Creusen · 3 years, 6 months ago
  16. 2c41cba Revert "Wire up non-sender RTT for audio, and implement related standardized stats." by Björn Terelius · 3 years, 6 months ago
  17. fb0dca6 Wire up non-sender RTT for audio, and implement related standardized stats. by Ivo Creusen · 3 years, 7 months ago
  18. 28a2c63 Adding packetsDiscarded to RTCReceivedRtpStreamStats. by Minyue Li · 3 years, 8 months ago
  19. e54914a Implement nack_count metric for inbound audio rtp streams. by Jakob Ivarsson · 3 years, 9 months ago
  20. 3008bcd Don't recreate audio receive streams on header extension update. by Tommi · 3 years, 9 months ago
  21. 1c1f540 Factor out common receive stream methods to a common interface. by Tommi · 3 years, 9 months ago
  22. e097282 Avoid recreating the audio stream when a frame decryptor is set. by Tommi · 3 years, 9 months ago
  23. 6eda26c Reland "Remove AudioReceiveStream::Reconfigure() method." by Tommi · 3 years, 9 months ago
  24. 8a18e5b Revert "Remove AudioReceiveStream::Reconfigure() method." by Andrey Logvin · 3 years, 9 months ago
  25. e2561e1 Remove AudioReceiveStream::Reconfigure() method. by Tommi · 3 years, 9 months ago
  26. f7b1b95 Add `RTCRemoteOutboundRtpStreamStats` for audio streams by Alessio Bazzica · 4 years ago
  27. 8467cf2 Reduce redundant flags for audio stream playout state. by Tomas Gunnarsson · 4 years, 2 months ago
  28. 6b4d962 Fix standard GetStats to not modify NetEq state. by Niels Möller · 4 years, 6 months ago
  29. 3e9af7f Insert audio frame transformer between depacketizer and decoder. by Marina Ciocea · 5 years ago
  30. e618cc9 Add jitterBufferTargetDelay as RTCNonStandardStatsMember to new GetStats API by Artem Titov · 5 years ago
  31. 7a9a092 Delete media transport integration. by Bjorn A Mellem · 5 years ago
  32. fcf79cc Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats. by Åsa Persson · 5 years ago
  33. 9429888 Delete deprecated bytes_sent/bytes_rcvd stat values by Niels Möller · 5 years ago
  34. ac0a4cb Reland "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Niels Möller · 5 years ago
  35. ef0627f Revert "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Mirko Bonadei · 5 years ago
  36. fbde32e Fix GetStats bytesSent/Received, wireup headerBytesSent/Received by Niels Möller · 5 years ago
  37. 65f17ca Move MediaTransportInterface out of the libjingle_peerconnection_api target by Niels Möller · 6 years ago
  38. a837030 Split out RtpSource from libjingle_peerconnection_api by Niels Möller · 6 years ago
  39. 224c69d Delete ext_seqnum member from VoiceSenderInfo and VoiceReceiverInfo by Niels Möller · 6 years ago
  40. 5b5d97c Reland of "Reporting of decoding_codec_plc events"" by Alex Narest · 6 years ago
  41. bedb7a8 Revert "Reporting of decoding_codec_plc events" by Mirko Bonadei · 6 years ago
  42. 0a88ea0 Reporting of decoding_codec_plc events by Alex Narest · 6 years ago
  43. 3472b9a Delete RTCInboundRTPStreamStats::fraction_lost by Niels Möller · 6 years ago
  44. 4f08faa Introduce MediaTransportConfig by Anton Sukhanov · 6 years ago
  45. 8d8ffdb Expose new audio stats on the API by Ivo Creusen · 6 years ago
  46. 44125fa Reland "Piping audio interruption metrics to API layer" by Henrik Lundin · 6 years ago
  47. fc02a79 Revert "Piping audio interruption metrics to API layer" by Henrik Andreassson · 6 years ago
  48. 299c4e6 Piping audio interruption metrics to API layer by Henrik Lundin · 6 years ago
  49. 01738c6 Wire up RTCInboundRtpStreamStats.lastPacketReceivedTimestamp. by Henrik Boström · 6 years ago
  50. 647d5e6 Increase the default maximum jitter buffer size to 200 packets. by Jakob Ivarsson · 6 years ago
  51. 232b3fd Expose relative packet arrival delay metric in stats API. by Jakob Ivarsson · 6 years ago
  52. 3b50f9f Propagate base minimum delay to audio_receiver_stream by Ruslan Burakov · 6 years ago
  53. d970807 Remove rtc_base/scoped_ref_ptr.h. by Mirko Bonadei · 6 years ago
  54. 0acffb5 Expose `jitterBufferEmittedCount` in addition to the existing `jitterBufferDelay` for `getStats()`. by Chen Xing · 6 years ago
  55. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  56. 53eae87 Add PeerConnection option to enable RTX handling in the audio jitter buffer. by Jakob Ivarsson · 6 years ago
  57. 10403ae Add PeerConnection option to configure minimum audio jitter buffer delay. by Jakob Ivarsson · 6 years ago
  58. 352ce5c Expose delayed packet outage as a cumulative metric of samples in the new getStats API. by Jakob Ivarsson · 6 years ago
  59. 8af8896 Expose jitter buffer flushes metric in new getStats api. by Ruslan Burakov · 6 years ago
  60. 2222a80 Delete unneeded includes of common_types.h and gn deps on webrtc_common. by Niels Möller · 6 years ago
  61. 7d76a31 Use MediaTransportInterface, for audio streams. by Niels Möller · 6 years ago
  62. bfb444c Adds new CryptoOption crypto_options.frame.require_frame_encryption. by Benjamin Wright · 6 years ago
  63. 84583f6 Enable End-to-End Encrypted Audio Payloads. by Benjamin Wright · 6 years ago
  64. a12c42a Delete root header file typedef.h. by Niels Möller · 7 years ago
  65. b9b146c Replace rtc::Optional with absl::optional in audio, call and video by Danil Chapovalov · 7 years ago
  66. 11b34f4 Remove chromium clang style errors affecting sdk/android/media_jni by Paulina Hensman · 7 years ago
  67. 0812634 Pass a real audio codec pair ID to decoders that we create by Karl Wiberg · 7 years ago
  68. f120cba Delete AudioMonitor and related code. by Niels Möller · 7 years ago
  69. a8b7c7f Move remaining traces of VoiceEngine by Fredrik Solenberg · 7 years ago
  70. 8f5787a Move ownership of voe::Channel into Audio[Receive|Send]Stream. by Fredrik Solenberg · 7 years ago
  71. 3b903d0 Reconfigure, not reconstruct, AudioReceiveStreams. by Fredrik Solenberg · 7 years ago
  72. b0a0207 Added RTCMediaStreamTrackStats.jitterBufferDelay for audio by Gustaf Ullberg · 7 years ago
  73. 9a2e906 Added RTCMediaStreamTrackStats.concealmentEvents by Gustaf Ullberg · 7 years ago
  74. 7120742 Adding NOLINT for typedefs.h and common_types.h by Mirko Bonadei · 7 years ago
  75. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  76. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/call/audio_receive_stream.h]
  77. 84f6a3f Move optional.h to webrtc/api/ by kwiberg · 8 years ago
  78. 1acbd68 Move RtpExtension to api/ directory and config.h/.cc to call/. by Stefan Holmer · 8 years ago
  79. 0e320ec Wiring discard rate of audio FEC/RED packets up to StatsReport. by minyue-webrtc · 8 years ago
  80. 2dbc69f Add stats totalSamplesReceived and concealedSamples by Steve Anton · 8 years ago
  81. 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 · 8 years ago
  82. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  83. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  84. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  85. 8d609f6 Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ ) by hbos · 8 years ago
  86. fbcc5cb Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ ) by olka · 8 years ago
  87. 292084c Added the GetSources() to the RtpReceiverInterface and implemented by zhihuang · 8 years ago
  88. 796b8f9 Remove usage of VoEVolumeControl from WVoE and Audio[Send|Receive]Stream. by solenberg · 8 years ago
  89. 087bd34 Move AudioDecoder and related stuff to the api/ directory by kwiberg · 8 years ago
  90. d32bf75 Pass SdpAudioFormat through Channel, without converting to CodecInst by kwiberg · 8 years ago
  91. f515ab8 Moved call.h and most of api/call/* into call/ by ossu · 8 years ago[Renamed (96%) from webrtc/api/call/audio_receive_stream.h]
  92. a8eb756 Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies. by aleloi · 8 years ago
  93. 1acfbd2 Expose RtpCodecParameters to VoiceMediaInfo stats. by hbos · 8 years ago
  94. 6348978 Add new decoding statistics for muted output by henrik.lundin · 8 years ago
  95. a69d973 Move webrtc/audio_*.h to webrtc/api/call by kjellander · 9 years ago[Renamed (96%) from webrtc/audio_receive_stream.h]
  96. 217fb66 Add AudioReceiveStream::SetGain() method and use that in WVoMC::SetOutputVolume(). by solenberg · 9 years ago
  97. 8189b02 Configure VoE NACK through AudioReceiveStream::Config, for receive streams. Also minor refactoring of WVoE unit test. by solenberg · 9 years ago
  98. 29b1a8d Moved creation of AudioDecoderFactory to inside PeerConnectionFactory. by ossu · 9 years ago
  99. 1ba8d39 Remove webrtc/stream.h and unutilized inheritance. by pbos · 9 years ago
  100. 3d7db26 Switch voice transport to use Call and Stream instead of VoENetwork. by mflodman · 9 years ago