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gerrit-public.fairphone.software / platform / external / webrtc / 1e257cacbfb0a49d80e62993ee65369c9d08b7b6 / media / base / media_channel.h
  1. 1e257ca Migrate media/ to webrtc::Mutex. by Markus Handell · 4 years ago
  2. 2a70703 Delete MediaTransportInterface and DatagramTransportInterface by Niels Möller · 4 years, 1 month ago
  3. 772b149 MediaChannel: remove lock recursions. by Markus Handell · 4 years, 2 months ago
  4. a0ff50c Reland "Improve outbound-rtp statistics for simulcast" by Henrik Boström · 4 years, 2 months ago
  5. 9a925c9 Revert "Improve outbound-rtp statistics for simulcast" by Henrik Boström · 4 years, 2 months ago
  6. da6cda8 Improve outbound-rtp statistics for simulcast by Eldar Rello · 4 years, 3 months ago
  7. 2c99264 Remove std::vector<BandwidthEstimationInfo> template instantiation. by Mirko Bonadei · 4 years, 3 months ago
  8. 3e98368 Reland "Distinguish between send and receive codecs" by Johannes Kron · 4 years, 4 months ago
  9. e618cc9 Add jitterBufferTargetDelay as RTCNonStandardStatsMember to new GetStats API by Artem Titov · 4 years, 4 months ago
  10. b05ca4b Implement new specification for degradation preference by Florent Castelli · 4 years, 4 months ago
  11. 412a31b Insert frame transformer between Depacketizer and Decoder. by Marina Ciocea · 4 years, 5 months ago
  12. e77912b Insert frame transformer between Encoded and Packetizer. by Marina Ciocea · 4 years, 5 months ago
  13. 8e8b36a Revert "Reland "Reland "Reland "Distinguish between send and receive codecs"""" by Johannes Kron · 4 years, 5 months ago
  14. 184ea66 Reland "Reland "Reland "Distinguish between send and receive codecs""" by Johannes Kron · 4 years, 6 months ago
  15. 749f660 Enable SSRC 0 in MediaChannel methods by Saurav Das · 4 years, 8 months ago
  16. 32565f6 WebRtcVideoEngine: Enable encoded frame sink. by Markus Handell · 4 years, 8 months ago
  17. 934afc6 Deprecate RtpReceiver's SetParameters method by Saurav Das · 4 years, 8 months ago
  18. 7a9a092 Delete media transport integration. by Bjorn A Mellem · 4 years, 8 months ago
  19. 00376e1 Add totalInterFrameDelay to RTCInboundRTPStreamStats by Johannes Kron · 4 years, 8 months ago
  20. 16cec3b Added allow_codec_switching parameter to RTCConfig. by philipel · 4 years, 9 months ago
  21. fcf79cc Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats. by Åsa Persson · 4 years, 9 months ago
  22. 0855e2d Delete unused members of MediaReceiverInfo and MediaSenderInfo by Niels Möller · 4 years, 9 months ago
  23. 9429888 Delete deprecated bytes_sent/bytes_rcvd stat values by Niels Möller · 4 years, 9 months ago
  24. ac0a4cb Reland "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Niels Möller · 4 years, 9 months ago
  25. ef0627f Revert "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Mirko Bonadei · 4 years, 9 months ago
  26. fbde32e Fix GetStats bytesSent/Received, wireup headerBytesSent/Received by Niels Möller · 4 years, 9 months ago
  27. ff27da5 Add/remove receive streams with SSRC 0 from media channels by Saurav Das · 4 years, 10 months ago
  28. 1b83a9e Only handle each RTCP once. by Sebastian Jansson · 4 years, 10 months ago
  29. 65f17ca Move MediaTransportInterface out of the libjingle_peerconnection_api target by Niels Möller · 4 years, 10 months ago
  30. cc62b16 Add qualityLimitationResolutionChanges stat by Evan Shrubsole · 4 years, 10 months ago
  31. a837030 Split out RtpSource from libjingle_peerconnection_api by Niels Möller · 4 years, 11 months ago
  32. 0c141c5 Fix frames dropped statistics by Johannes Kron · 5 years ago
  33. 224c69d Delete ext_seqnum member from VoiceSenderInfo and VoiceReceiverInfo by Niels Möller · 5 years ago
  34. 6b43086 Reland "[GetStats] Expose video codec implementation in standardized metrics." by Henrik Boström · 5 years ago
  35. df625f4 Revert "[GetStats] Expose video codec implementation in standardized metrics." by Henrik Andreassson · 5 years ago
  36. 2b9fa09 [GetStats] Expose video codec implementation in standardized metrics. by Henrik Boström · 5 years ago
  37. bbeb109 Reporting audio device underrun counter by Alex Narest · 5 years ago
  38. 5b5d97c Reland of "Reporting of decoding_codec_plc events"" by Alex Narest · 5 years ago
  39. bedb7a8 Revert "Reporting of decoding_codec_plc events" by Mirko Bonadei · 5 years ago
  40. 0a88ea0 Reporting of decoding_codec_plc events by Alex Narest · 5 years ago
  41. e1795f4 Adds remote estimate RTCP packet. by Sebastian Jansson · 5 years ago
  42. d2c336f [getStats] Implement "media-source" audio levels, fixing Chrome bug. by Henrik Boström · 5 years ago
  43. bfd343b Add totalDecodeTime to RTCInboundRTPStreamStats by Johannes Kron · 5 years ago
  44. 2efae77 Add RTCStats for keyFramesEncoded, keyFramesDecoded. by Rasmus Brandt · 5 years ago
  45. 3472b9a Delete RTCInboundRTPStreamStats::fraction_lost by Niels Möller · 5 years ago
  46. 6737841 Add jitterBufferDelay and jitterBufferEmittedCount stats for video by Guido Urdaneta · 5 years ago
  47. ce33b6a Implement QualityLimitationReasonTracker and expose "reason". by Henrik Boström · 5 years ago
  48. 87e3f9d [video] Plumbing of ReportBlockData from RTCPReceiver to MediaSenderInfo by Henrik Boström · 5 years ago
  49. 23aff9b Implement RTCOutboundRtpStreamStats.totalEncodedBytesTarget. by Henrik Boström · 5 years ago
  50. 4f08faa Introduce MediaTransportConfig by Anton Sukhanov · 5 years ago
  51. 9fe1834 Implement RTCOutboundRtpStreamStats.totalPacketSendDelay for video. by Henrik Boström · 5 years ago
  52. 8d8ffdb Expose new audio stats on the API by Ivo Creusen · 5 years ago
  53. 44125fa Reland "Piping audio interruption metrics to API layer" by Henrik Lundin · 5 years ago
  54. fc02a79 Revert "Piping audio interruption metrics to API layer" by Henrik Andreassson · 5 years ago
  55. 299c4e6 Piping audio interruption metrics to API layer by Henrik Lundin · 5 years ago
  56. cf96e0f Implement RTCOutboundRtpStreamStats.retransmitted[Bytes/Packets]Sent. by Henrik Boström · 5 years ago
  57. 01738c6 Wire up RTCInboundRtpStreamStats.lastPacketReceivedTimestamp. by Henrik Boström · 5 years ago
  58. f71362f Wire up RTCOutboundRtpStreamStats.totalEncodeTime. by Henrik Boström · 5 years ago
  59. e7a5f7b Modifying MediaChannel to accept CopyOnWriteBuffer by value. by Amit Hilbuch · 5 years ago
  60. e25f595 Guard preferred_dscp with the network interface lock by Steve Anton · 5 years ago
  61. 232b3fd Expose relative packet arrival delay metric in stats API. by Jakob Ivarsson · 5 years ago
  62. 493a650 Propagate base minimum delay from video jitter buffer to webrtc/api. by Ruslan Burakov · 5 years ago
  63. ea7ef2a Refactoring RtpSenderInternal to share implementation for Audio & Video. by Amit Hilbuch · 5 years ago
  64. 7ea4605 Add latency to remote source api. by Ruslan Burakov · 5 years ago
  65. 0237106 Expose video freeze metrics in GetStats. by Sergey Silkin · 5 years ago
  66. 0acffb5 Expose `jitterBufferEmittedCount` in addition to the existing `jitterBufferDelay` for `getStats()`. by Chen Xing · 6 years ago
  67. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  68. 1c05765 (3) Rename files to snake_case: move the files by Steve Anton · 6 years ago[Renamed from media/base/mediachannel.h]
  69. 514f084 New statistic added to VideoReceiveStream to determine latency to first decode. by Benjamin Wright · 6 years ago
  70. 352ce5c Expose delayed packet outage as a cumulative metric of samples in the new getStats API. by Jakob Ivarsson · 6 years ago
  71. 8af8896 Expose jitter buffer flushes metric in new getStats api. by Ruslan Burakov · 6 years ago
  72. 5f2ffee Clean up deprecated APM stats by Sam Zackrisson · 6 years ago
  73. dd9390c Prevent channels being set on stopped transceiver. by Amit Hilbuch · 6 years ago
  74. e693381 Delete struct rtc::PacketTime. by Niels Möller · 6 years ago
  75. 9190b82 Propagate SDP negotiation of extmap-allow-mixed to RtpHeaderExtensionMap by Johannes Kron · 6 years ago
  76. 648d28a Media engine and channel support for per-channel dscp values, specified by RtpParameter by Tim Haloun · 6 years ago
  77. 98a462c Reland "Reland "Propagate media transport to media channel."" by Anton Sukhanov · 6 years ago
  78. 9accc9f Revert "Reland "Propagate media transport to media channel."" by Oleh Prypin · 6 years ago
  79. da65ed2 Reland "Propagate media transport to media channel." by Anton Sukhanov · 6 years ago
  80. 37cf245 Revert "Propagate media transport to media channel." by Oleh Prypin · 6 years ago
  81. 8c16f74 Propagate media transport to media channel. by Anton Sukhanov · 6 years ago
  82. 84583f6 Enable End-to-End Encrypted Audio Payloads. by Benjamin Wright · 6 years ago
  83. 6ca9836 Prepare for per-media DSCP values. Push dscp for stun packets to the port layer where they are created. by Tim Haloun · 6 years ago
  84. 49ac595 Add GetSources to VideoRtpReceiver by Jonas Oreland · 6 years ago
  85. 84df1c7 Make fewer copies when using StringBuilder. by Jonas Olsson · 6 years ago
  86. bfd412e Adds integration of the FrameEncryptor/FrameDecryptor into the MediaChannel. by Benjamin Wright · 6 years ago
  87. 366a50c Remove simple stringstream usages. by Jonas Olsson · 6 years ago
  88. f5f5373 Delete unused member MediaSenderInfo::packets_cached. by Niels Möller · 6 years ago
  89. e41c433 Move sigslot to proper third_party directory by Artem Titov · 6 years ago
  90. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  91. 00c7183 Replace rtc::Optional with absl::optional in media, ortc, p2p by Danil Chapovalov · 6 years ago
  92. 97e0488 Delete unused stats for preferred_bitrate. by Niels Möller · 6 years ago
  93. 0327c2d Move VideoStreamEncoderInterface to api/. by Niels Möller · 6 years ago
  94. 65ec0fc Delete unneeded includes of basictypes.h. by Niels Möller · 6 years ago
  95. c948fe6 Delete unneeded includes of call/video_config.h. by Niels Möller · 6 years ago
  96. c6ce9c5 New file api/video/BUILD.gn by Niels Möller · 6 years ago
  97. 11b34f4 Remove chromium clang style errors affecting sdk/android/media_jni by Paulina Hensman · 6 years ago
  98. ff40b14 Delete obsolete enable argument to SetVideoSend. by Niels Möller · 6 years ago
  99. 5897a6e Adds support for signaling a=msid lines without a=ssrc lines. by Seth Hampson · 6 years ago
  100. 003930a Fix MID not always getting set with audio by Steve Anton · 6 years ago